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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

1 0

ffmpeg-devel.log.20130222
by burek 23 Feb '13

23 Feb '13

[00:08] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:29436c1ab2c5: libavcodec/mips/aacsbr_mips: fix ;; [03:01] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:b16bf913d8ed: dxva2: move _WIN32_WINNT handling to public header [04:43] <tonsofpcs> just ran ffprobe -show_packets on a .ts and it is showing a dts = pts on all packets that don't have a dts. Why is this? (it's an ATSC TS if that matters) [05:42] <llogan> i'm wondering if "qualification tasks" are an impediment to getting shit done for gsoc. [06:59] <av500> by experience, a lot of "shit" has been done [08:33] <highgod> Hi,Stefano Sabatini,I have re-submit the opencl patch [11:01] <cone-583> ffmpeg.git 03Matt Wolenetz 07master:311443f6c7eb: x86: h264: Don't use redzone in AVX h264_deblock on Win64 [11:01] <cone-583> ffmpeg.git 03Michael Niedermayer 07master:db05f7ab1626: Merge remote-tracking branch 'qatar/master' [13:18] Action: durandal_1707 looking at drawutils, and what to change for >8 bit support ; result = headache [13:40] <Compn> (Remove the sws project, its mentor thinks it is pointless and a complete rewrite is needed instead.) [13:40] <Compn> snort, rewrite of swscale :D [13:44] <durandal_1707> hmm how to know what number of threads are appear when calling execute? [13:48] <durandal_1707> i could allocate 16 * height bytes in worst case [13:49] <durandal_1707> because each image scan line could be decoded in separate thread [13:50] <durandal_1707> but having 8000 thread jobs may be too much [13:57] <durandal_1707> ught there is memory leak in lavf/mxg [14:42] <cone-449> ffmpeg.git 03Paul B Mahol 07master:a72b41035cc4: mxg: do not leak mxg->buffer [14:52] <cone-449> ffmpeg.git 03Michael Niedermayer 07master:21f68528e39d: aacdec: dont pop random configurations [14:55] <cone-449> ffmpeg.git 03Paul B Mahol 07master:c46400ddecab: cafenc: do not leak caf->pkt_sizes [15:38] <tclarke> hi, I'm trying to write a custom muxer which write raw frames (using the rawvideo codec) to a circular framebuffer in a memmapped file...the problem I have is that ffmpeg is opening the file I specify and performing flush's, etc. so the second time my write_packet function is called the mmap section seems to be overritten with 0's [15:39] <tclarke> how can I tell ffmpeg that my AVOutputFormat will handle it's own I/O and an avio context should not be opened? [15:41] <Compn> sounds complicated [15:43] <tclarke> you have a better solution? [15:43] <ubitux> tclarke: maybe look how it's done in the segment muxer [15:44] <ubitux> or lavf/hlsenc [15:44] <tclarke> ok, I'll look there, thanks for the pointers [16:31] <Compn> j-b : could just do what winamp does for interfaces , put it in html , make user use his web browser to see bittorrent status :) [16:31] <Compn> for bittorrent playback thing [17:27] <creep> hi [17:27] <creep> what kind of music do you consume ? [17:27] <creep> Flyleaf - i'm so sick, Fully Alive http://www.youtube.com/watch?v=iWIADZKU9dw hackers used to listen stuff like this on films [17:32] <gnafu> Cool story, bro. [17:44] <Compn> hey hes back [17:45] <mystica555> [libvorbis @ 0x119c8e0] Que input is backward in time [17:45] <mystica555> ...que hora es? >.> [17:46] <Compn> is that a libvorbis error or our error ? [17:47] <mystica555> im not entirely sure [17:48] <mystica555> it visibly appears similar (with a teal background) to [17:48] <mystica555> [libx264 @ 0x11b0600] frame I:46 Avg QP:25.17 size: 26041 [17:48] <mystica555> que... was brown [17:48] <mystica555> libvorbis teal [17:50] <Compn> i dont see that error message in the code. [17:50] <mystica555> then its probably in libvorbis. [17:50] <Compn> no [17:50] <Compn> its in libfaac to [17:50] <Compn> somehow [17:50] <Compn> lets see where it comes from... [17:51] <mystica555> hmm [17:51] <creep> nah [17:52] <creep> spam? :P [17:52] <mystica555> /usr/lib64/libavcodec.so.54.59.100: Que input is backward in time [17:52] <mystica555> hm [17:52] <mystica555> says strings [17:52] <mystica555> so definitely there, somewhere. [17:53] <Compn> audio_frame_queue.c: av_log(afq->avctx, AV_LOG_WARNING, "Queue input is backward in time [17:53] <Compn> is it somehow eating "ue input" ? [17:53] <mystica555> likely a newer version patched it [17:53] <mystica555> ive maybe not updated in a few months [17:53] <Compn> maybe its just a typo [17:53] <Compn> que > queue [17:54] <Compn> english isnt the first language of many devs in the project [17:54] <Compn> and there are multiple ways to spell the same english words :\ [17:54] <mystica555> well yes im sure it is a typo [17:54] <mystica555> que just isnt one of the right ways [17:54] <mystica555> either you "cue" something - to advance it to a certain poinbt in time in preperation for playback for example; or you put something into a queue [17:55] <mystica555> que is spanish however [17:55] <mystica555> ;) [17:56] <Compn> que is slang for queue then :P [17:56] <Compn> in this case [17:56] <mystica555> que paso [17:56] <mystica555> ;) [17:56] <Compn> vato loco! [17:58] <Compn> but yeah it appears its been fixed in the meantime [18:20] <cone-345> ffmpeg.git 03Michael Niedermayer 07master:6f77122bf571: aac: reconfigure output on pop [18:28] <saste> ubitux: I'm asking because an av_free() would have fail. [18:28] <saste> what do you mean? [18:40] <BBB-work> "Description: Not many VP7 samples are in the wild, but no open-source decoder exists although Google has released a specification. Write a decoder that reuses as much as possible of existing FFmpeg code, it is likely that functions of the existing decoders for On2-based formats will be useful." ? [18:40] <BBB-work> I didn't know google released a vp7 spec? [18:43] <nevcairiel> all i know of is this: http://multimedia.cx/mirror/VP7_Data_Format_and_Decoder_Overview.pdf [18:44] <nevcairiel> it says to be used in conjunction with the reference decoder [18:45] <BBB-work> that is not released by google, that is dropped off a truck [18:45] <BBB-work> that's not the same thing :) [18:45] <nevcairiel> since when do we care about that specifically? :d [18:46] <BBB-work> you're writing a wiki page to give it to a company named google for inclusion in gsoc, right? [18:47] <BBB-work> that very same company is claimed to have done something it didn't do on that very wiki page [18:51] <nevcairiel> for all i know, they have released it, whoever added that shall be questioned [19:21] <ubitux> saste: mmh sorry, it would have fail in some configuration only [19:21] <saste> ubitux, i still can't understand [19:22] <saste> how can av_free fail? [19:22] <ubitux> in case of memalign hack for instance [19:22] <ubitux> if the allocated buffer is not from av_malloc or similar [19:22] <ubitux> it would fail [19:23] <saste> but we are using av_malloc [19:23] <saste> strdup -> av_strdup should be obviously fixed [19:23] <ubitux> yes that's the point [19:23] <ubitux> on a memalign config, the buffer allocated with strdup() would have fail if the pointer was freed with a av_free() [19:24] <ubitux> memalign-hack* configu [19:24] <saste> so if i fix strdup -> av_strdup() it will be safe, right? [19:25] <ubitux> sure [19:25] <saste> good i'll fix and push then [19:28] <ubitux> btw, about the [HH:] thing the guy actually sent an updated patch [19:29] <saste> ubitux, for some reason I got my mails stucked on the server [19:29] <saste> so they are released just after many hours [19:30] <saste> the more time i don't fetch mails, the more time they tend to stay stucked [19:30] <saste> so in my mailbox i don't always have all the mails [19:31] <saste> it sucks big time, but nothing i can do much about without spending a few hours debugging or nailing tech support [19:31] <ubitux> :( [19:34] <ubitux> ffcat [19:34] <ubitux> fast forward cat [19:42] <tonsofpcs> I ran ffprobe -show_packets on a .ts and it shows dts = pts on all packets (video and audio) that don't have a dts. Why is this? (it's an ATSC TS if that matters) [19:46] <saste> ffcat -> fast frenzy kittie [19:55] <creep> http://images.puszta.com/hazi_kifli/Hazi%20kifli%201000.jpg ! [19:55] <Tjoppen> oh mama [20:01] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:f30979e0954e: lavfi/boxblur: add support to named options [20:01] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:8902f5b89c16: doc/filters: apply minor spacing fixes to boxblur docs [20:18] <durandal_1707> michaelni: have anything to add to exr patches? (i have improved version with slice threading) [20:21] <cone-345> ffmpeg.git 03Paul B Mahol 07master:345689129102: exr: check version too [20:29] <michaelni> durandal_1707, the patches should be fine [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:88d7259fd246: exr: merge multiple if () blocks [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:1178868683d2: exr: fix overflow check in lineoffset processing [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:cc0bd7841389: exr: support more line orders [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:9c75c24de40d: exr: check if line offset table is complete only once per frame [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:fefd4b0924cc: exr: cosmetics: reindent [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:7a930f310ad2: exr: remove superfluous check [20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:89c77b6b155c: exr: shorter channel buffer names [20:58] <cone-345> ffmpeg.git 03Michael Niedermayer 07master:2c21d34ea44d: avutil/mem: do a small set of checks for memalign hack before freeing. [21:30] <code-guru> hi all [21:31] <code-guru> I can't set mpegts metadata when use hls muxer. [21:31] <code-guru> for example when I add "-metadata service_provider="Some provider" it doesn't change service_metadata when I use hls m3u8 muxer [21:31] <code-guru> can anybody help me with that? [21:33] <llogan> code-guru: sorry. wrong channel. #ffmpeg is for user help. [21:34] <code-guru> llogan, it seems more of a bug to me ? isn't it ? [21:53] <ubitux> btw, about GSoC [21:53] <ubitux> i can suggest some subtitles stuff [21:54] <ubitux> and i'd be very happy to see improvements in the gif muxer/encoder [21:54] <ubitux> like non-intra frames, and better colors [21:56] <llogan> would you be interested in mentoring? [21:56] <ubitux> subtitles yes, gif maybe [21:58] <llogan> can you add this to the ideas page? [21:58] <ubitux> maybe in an hour or two [21:58] <llogan> or email me the info and i can do it if that will be easier for you. [21:59] <ubitux> i think i need an account too :p [22:00] <llogan> but i can't do it until later either...more video editing to do [22:09] <cone-345> ffmpeg.git 03Malcolm Bechard 07master:239b88c284af: Improve QTRLE encoding performance, no change to output file size/content. [22:50] <code-guru> https://ffmpeg.org/trac/ffmpeg/ticket/2296 [22:53] <cone-345> ffmpeg.git 03Martin Storsjö 07release/1.1:65bf4c9c4578: mpegvideo: Do REBASE_PICTURE with byte pointers [22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:0b2b8ab97962: qdm2: check array index before use, fix out of array accesses [22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:52b18c1fde65: roqvideodec: check dimensions validity [22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:ab434bf0d051: vqavideo: check chunk sizes before reading chunks [22:53] <cone-345> ffmpeg.git 03Vicente Jimenez Aguilar 07release/1.1:6626a7df5345: doc: Fix some obsolete references to av* tools as ff* tools [22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:80ddf7889eec: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [23:11] <ubitux> fun fact: there are around 12k additionnal commits in ffmpeg against l. (without counting merges) [23:12] <nevcairiel> now do that excluding commits that they ported over [23:12] <nevcairiel> :D [23:12] <ubitux> 12k-3 ? [23:15] <ubitux> ok so anyway [23:15] <ubitux> i need to mail mike for a multimedia.cx account, right? [23:15] <wm4> simply subtracting the number of commits shows that ffmpeg has slightly more commits, I think? [23:15] <wm4> though you'd have to subtract merge commits too, hm [23:15] <ubitux> i didn't count them [23:16] <ubitux> otherwise it's 14k or something iirc [23:16] <ubitux> but TBH i should have count them somehow :) [23:16] <ubitux> llogan: ping? [23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:a642be972d48: h264: ensure that get_format() is called when changing format but not otherwise. [23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:469cb6119386: avcodec_decode_audio4: check got_frame_ptr before handling initial skip [23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:02d1efdd5b61: h264: check that luma and chroma depth match [23:21] <cone-345> ffmpeg.git 03Andrea3000 07release/1.1:ef688e742557: matroska: fix missing , [23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:705e89d75fde: update for 1.1.3 [23:41] <michaelni> BBB-work, how should the VP7 text on the SOC page be worded ? (i dont know at all where which spec came from) [23:42] Action: michaelni just found his un/pw for wiki-multimedia again ;) [23:42] <ubitux> can you find mine if it exists? [23:44] <michaelni> ubitux, if you had an account try to login theres a button to get a pw by email [23:44] <michaelni> if you didnt then well i cant help [23:44] <ubitux> yeah i think i never had one :) [23:45] <michaelni> you need to talk to mike then [23:45] <michaelni> thats the disadvantage of it over our wiki [23:45] <michaelni> delay to get access [23:47] <ubitux> yep right, just sent an email [23:49] <michaelni> hey beastd, do you have a wiki multimedia account ? if so you could help improve our SOC page [23:50] <cone-345> ffmpeg.git 03Senthilnathan M 07master:12a269a5229d: lavu/parseutils: allow MM:SS format for duration in av_parse_time() [23:51] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:ce84c14ab173: lavu/parseutils: remove spurious "]]" in date format specification [23:51] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:160def32da43: doc/muxers: fix reference to @anchor defined in another manual [23:51] <beastd> hi michaelni, i have. will try to have a look at it this weekend. [23:51] <michaelni> ok thx [00:00] --- Sat Feb 23 2013

1 0

ffmpeg.log.20130222
by burek 23 Feb '13

23 Feb '13

[00:17] <tonsofpcs> just ran ffprobe -show_packets on a .ts and it is showing a dts = pts on all packets that don't have a dts. Why is this? [00:18] <tonsofpcs> (the .ts is an ATSC transport stream, if it helps) [02:22] <rcombs> not directly ffmpeg related, but... [02:22] <rcombs> I've got some files here with the .WAV extension, but which start with 14 28 08 32. Does anyone recognize that header? [02:37] <jagginess> hi [02:37] <jagginess> "ERROR: libx264 not found", been trying to get this fixed.. been using flags, no use :( [02:40] <klaxa> did you try installing x264? [02:40] <jagginess> yup that's installed.. this is a prefix of ~/local [02:40] Action: jagginess is following https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#x264 [02:40] <klaxa> what is your compilation configuration for ffmpeg? [02:41] <jagginess> latest git, [02:41] <jagginess> ./configure --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame \ [02:41] <jagginess> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora \ [02:41] <jagginess> --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 [02:41] <jagginess> and --prefix=/home/username/local [02:41] <jagginess> i also have 3 export variables of flags [02:41] <klaxa> i take you compiled x264? [02:42] <klaxa> and have a libx264.a in the x264 directory? [02:42] <jagginess> ( export CPPFLAGS=-I/home/user/local/include; export CXXFLAGS=-I/home/user/local/include; export LDFLAGS=-L/home/user/local/lib ) [02:43] <klaxa> mhh... did you install the x264 libraries to /home/user/local? [02:43] <jagginess> ./usr/local/include/x264.h [02:43] <jagginess> ./usr/local/include/x264_config.h [02:43] <jagginess> ./usr/local/lib/pkgconfig/x264.pc [02:43] <jagginess> ./usr/local/lib/libx264.a [02:43] <jagginess> ./usr/local/bin/x264 [02:43] <jagginess> ./bin/x264 [02:43] <jagginess> (in ~/local) [02:43] <klaxa> that's.... well it certainly *should* work i think [02:43] <jagginess> hmm [02:43] <jagginess> i think i need to use ~/local/usr/local/lib too [02:49] <jagginess> klaxa, how can i add multiple paths for flags? (export CPPFLAGS='-I/path/abc -I/path/def') [02:49] <jagginess> :/ [02:49] <klaxa> yes that should work like that [02:50] <jagginess> i no expert, i usually dont' compile at all.. [02:50] <jagginess> you sure ? :} [02:50] Action: jagginess replaces comma with space [02:50] <klaxa> it works whenever i compile stuff [02:52] <jagginess> can i use --extra-cflags twice with ./configure ? [02:53] <jagginess> hmm [02:53] <klaxa> i'm not sure, but you should be able to add all the flags with using it only one time [02:54] <jagginess> klaxa, how would i separate paths, i use a comma? [02:54] <klaxa> in what context? commas are genereally not used i think [02:54] <jagginess> --extra-cflags=/path/abc,/path/def ? [02:55] <klaxa> i doubt that that will do anything even, since it lacks parameters like -L or -I [03:15] <jagginess> klaxa, mind if i ask you can try it? :} lol [03:16] <klaxa> mmmh my system is probably quite different from yours and i compile ffmpeg without problems every now and then [03:16] <relaxed> jagginess: --extra-cflags="-I/path/to/prefix/include" --extra-ldflags="-L/path/to/prefix/lib" [03:17] <relaxed> if you need multiple, --extra-cflags="-I/path/to/prefix/include -I/some/other/place/include" [03:17] <jagginess> yeah tried both :/ [03:17] <relaxed> make distclean and try again [03:18] <jagginess> hmm did that too [03:18] <jagginess> ./configure passes without x264 enabled. [03:18] <jagginess> always. [03:18] <relaxed> pastebin or it didn't happen [03:19] <jagginess> i'll live without x264 for now.. thanks kla [03:19] <jagginess> klaxa [06:04] <magn3ts> Is there ever any hope for libav and ffmpeg remerging? [06:06] <klaxa> magn3ts: doubtful i think [06:07] <Guest82135> hi [06:09] <llogan> magn3ts: it's hard to say, but i'd leave open the possibility at least. [06:26] <Guest82135> /msg nickserv ghost code-guru [06:26] Last message repeated 1 time(s). [06:26] <Guest82135> /msg nickserv regain code-guru [06:26] <Guest82135> /msg nickserv help [06:27] <Guest82135> /msg nickserv ghost code-guru [06:27] <Guest82135> /msg nickserv identify code-guru 1qaz1qaz [07:02] <llogan> i hope he changes his password [07:37] <hendry> hi, I'm looking for test videos that print their elapsed time. Working on a syncronized video player. Any tips? [08:34] <klaxa> hendry: if you succeed i'd be interested in a synchronized video player (you are talking about synchronized over a network right?) [08:55] <SpmP> cross purpose... can ffmpeg be used to convert over under to side by side? [09:02] <relaxed> SpmP: yes [09:04] <SpmP> relaxed: great. fair call... <cough> how 8) cheers. [09:07] <relaxed> SpmP: Use the exapnd filter to create a "canvas" and then the movie/overlay filter to place it. [09:07] <relaxed> expand* [09:09] <hendry> klaxa: yes [09:09] <relaxed> SpmP: sorry, s/expand/pad/ [09:11] <SpmP> yaR your way too ahead of me.. still trying to find expand in your docs 8)... [09:11] <relaxed> SpmP: I meant the pad filter [09:12] <SpmP> cool... [09:15] <relaxed> SpmP: http://pastie.org/6315146 [09:19] <relaxed> That's an old example, you may have to change it a little and use -filter_complex. [09:23] <SpmP> yarrr... still getting my head around the simple syntax, oh dear, ffmpeg is amazingly complete. I guess this is imagemagik for video 8)... dammit, ille get there [10:11] <knobo> Hello, I get "moov atom not found" when I play I video I have. [10:12] <knobo> I read somewhere that AtomicParsley could help me fix the problem. But How do I use it? [10:12] <knobo> Or is there any other way, using ffmpeg? [10:12] <Mavrik> um [10:13] <Mavrik> if you don't have a moov atom in video [10:13] <Mavrik> your video is badly broken [10:14] <code-guru> try this. "ffmpeg -i broken.mp4 -acodec copy -vcodec copy -y new.mp4" [10:15] <klaxa> say code-guru did you change your nickserv password after pasting it in here? [10:15] <code-guru> no :-o [10:15] <klaxa> better do that :X [10:16] <knobo> An instruction video I got from china. Maybe the Great Wall changed it... [10:16] <code-guru> thanks [10:16] <JEEB> knobo, basically the moov atom is the "index" of the file [10:16] <JEEB> with all the offsets and so forth [10:16] <JEEB> if that is borked [10:17] <JEEB> you could as well throw the file to the fjords [10:17] <code-guru> what else I pasted here ? :-D [10:17] <JEEB> in other words, try to use a way of transfer that tries to keep the data in tact :) [10:19] <code-guru> that's a funny FAIL ! [10:19] <klaxa> huh? why? [10:19] <knobo> How do I transfere videos back and forth from China? [10:19] <code-guru> sending out your password to the channel [10:19] <knobo> off topic, maybe.. [10:20] <JEEB> no idea, personally I'd use rsync via ssh? [10:20] <JEEB> that tries to keep the data intact [10:21] <code-guru> if it's a live stream, you can use ffserver [10:22] <JEEB> and what would ffserver do with a live stream? It's a streaming component itself. [10:22] <knobo> But I don't have a server in china.. Well this is off topic. And the sender in china probably don't know how to use ffmpeg, linux, command line or facebook. [10:23] <klaxa> upload on a filesharing site? [10:23] <JEEB> or just keep downloading that video until the file size and md5 hash match, if you even have an md5 hash of the file available :D [10:25] <code-guru> I mean if it's video coming from a closed circuit camera, ffserver can broadcast it to the net based on a protocol like rtp [10:25] <knobo> That's what they did. http://mail.qq.com/cgi-bin/ftnExs_download?k=7c376165859204c1fbc71f7e166102… [10:28] <relaxed> curl has support for resuming a download [10:39] <SpmP> relaxed: Thanks for the intro [13:01] <spinx60> Hello im trying to stream live to twitch.tv with ffmpeg but when i try and start my script i get this http://pastebin.com/QgPM9PRE [13:02] <ubitux> you didn't paste the command you used [13:02] <ubitux> and since this is a syntax error, the error lies into the command you used [13:02] <spinx60> anyone who can [13:02] <spinx60> ohh sorry 1 sec [13:03] <spinx60> streaming live_XXXXXXXXXXXXXX where X is my stream key [13:03] <spinx60> i have the script in my .bashrc file [13:04] <spinx60> its wierd couse i got my it to start my stream last night using that command. and i got sound but no video on my stream' [13:10] <spinx60> Anyone got any idés ? [13:11] <ubitux> since you still didn't paste the command we can't help you [13:15] <spinx60> i execute the following command [spinx60@GreenPeace ~]$ streaming live_XXXXXXXXXXXXXXXXXXXXX where XXX is my own stream key [13:15] <spinx60> streaming live_xxxxxxxxxxxxxx is the command [13:17] <spinx60> i put X in becouse i dont want to show my stream key ofc [13:19] <ubitux> how are we supposed to guess what the "streaming" command is? [13:21] <spinx60> how you mean.. guess the command.. i pasted my script in my -bashrc file and that is the command kind of couse i start it by streaming live_XXXXXX Key etc ? dont you get it = [13:21] <spinx60> look at this aswell [13:22] <ubitux> there is a syntax error in the command defined in your .bashrc or wherever you put it [13:22] <ubitux> unless you show it we can't guess the problem [13:22] <spinx60> http://pastebin.com/XrvpMYQL [13:22] <ubitux> ah, finally. [13:22] <spinx60> there is it [13:22] <spinx60> sorry my bad there totally [13:24] <ubitux> with this exactly, you get the exact error pasted above ? [13:24] <spinx60> yes [13:25] <ubitux> sh -x streaming xxxx [13:25] <spinx60> and the thing is my script looks the same and is the same as i used last night when i got it to stream with sound but whitout video [13:25] <ubitux> does it show a sane exec? [13:26] <spinx60> gives me sh: streaming: No such file or directory [13:26] <ubitux> erm. [13:26] <ubitux> well [13:26] <ubitux> try to exec it yourself then [13:26] <spinx60> its something in my script that makes me able to run the command streaming live_XXX [13:26] <ubitux> i guess yes. [13:27] <spinx60> yeah that works.. now il paste the outcome but now im back to square one where i dont get any picture/capture on my stream its only a black screen with sound [13:29] <ubitux> then pastebin ffmpeg output again [13:29] <spinx60> http://pastebin.com/hcAwEj42 [13:30] <spinx60> i need to go now but i will be back later on check that out and send any message and i llook at it later.. thanks for the help so far [13:31] <shevy> "Encoder (codec id 86017) not found for output stream #0.0" [13:31] <shevy> anyone has a listing for what is codec ix 86017? [13:32] <shevy> command was: ffmpeg -i foo.mp3 -ss 00:00:11 -t 00:00:22 test.mp3 [13:33] <sacarasc> shevy: Try adding -c:a copy [13:53] <jarno> Hi all, what happens when encoding a video and the PTS (presentation time stamp) overflows? PTS is supposed to be monotonically increasing function (i.e. value), but if encoding either with high frame rate or a long video, it might overflow, right? [15:54] <spinx60> Im back again and still having issues with my Twich.tv streaming. i use the following command http://pastebin.com/pFVJa5TF [15:55] <spinx60> and the outcome is http://pastebin.com/ee67TY1n [15:56] <spinx60> im having my stream live with sound but whitout any video/capture at all. [16:02] <spinx60> Anyone who can shed some light on this ? must be something rather simple.. i mean i cant be the only one having/had this problem [16:38] <Wihl-w> Hi, I have 20 audio files (audio1.wav etc) and 20 images (image1.png etc) that I'd like to combine so that image1.png is swn for the duration of audio1.wav and so on [16:39] <Wihl-w> Is that something that would be possible to do with ffmpeg in an automated fashion? [16:40] <Wihl-w> *shown for the duration [16:40] <relaxed> spinx60: remove -s "1280x720" and add -filter:v scale=1280:720,format=yuv420p [16:42] <relaxed> yuv444p was most likely the problem. [16:55] <spinx60> testing it now [17:04] <relaxed> Wihl-w: bash for loop; -loop 1 -i blah.png -shortest [17:09] <addisonj> hey, looking for a bit of help to transcode a video, copy the audio, and write to stdout, here is my command ffmpeg -i video.mpg -c:v libx264 -preset superfast -f:a copy -f:v h264 - [17:09] <addisonj> problem right now, video is working, but not getting any audio streams [17:09] <addisonj> doing an ffprobe confirms its being stripped [17:10] <relaxed> you're using the wrong format [17:10] <relaxed> and it's '-c:a copy' [17:10] <addisonj> I had that earlier as welll... [17:10] <Mavrik> hmm [17:11] <relaxed> what kind of audio? [17:11] <Mavrik> any idea how can I clear the "orientation" setting from transform matrix in mp4? [17:11] <relaxed> what are you piping to? [17:12] <addisonj> relaxed: writing to stdout, which is picked up by a little node.js script and then send down in response to http request :P [17:12] <addisonj> the video works just fine (when I buffer in mplayer) but no audio [17:12] <relaxed> -f flv [17:14] <masch> hi. I need to change the frame rate of a clip from 29.97 to 23.976 without dropping frame. i tried decoding the clip to rawvideo and pipe that to another ffmpeg process that reencodes the video but it seems to lose the audio by that :-/ [17:14] <masch> can i tell ffmpeg to keep all frames but just change the framerate? [17:14] <addisonj> flv? isn't that flash video? I am trying to do this with html5 vid player [17:14] <relaxed> ok, try -f webm [17:15] <addisonj> I was going to try webm next, is there something wrong about outputting h264? still don't understand why the video would drop out... [17:16] <relaxed> masch: Yes. Pipe rawvideo back into ffmpeg like ffmpeg ... - 2>/dev/null|ffmpeg -r 23.976 -s WxH .... [17:16] <relaxed> addisonj: h264 is not a container. [17:17] <addisonj> ah... [17:18] <relaxed> masch: oyu'll have to demux the audio and remux them back together. [17:18] <relaxed> you'll* [17:18] <addisonj> relaxed: also, going to webm, I am going to have to transcode audio to vorbis right? [17:18] <masch> relaxed: oh okay, any idea how to change the audio tempo ? [17:19] <masch> so it fits to the new duration? [17:19] <relaxed> masch: sox can do it [17:19] <relaxed> maybe there's a ffmpeg filter for it, I'm not sure [17:20] <relaxed> addisonj: be sure to use -c:a libvorbis [17:27] <relaxed> masch: you know, with mplayer you can simply play back at a different framerate. [17:29] <relaxed> spinx60: \o/ [17:36] <spinx60> relaxed thanks alot [18:38] <undercash> hello [18:38] <undercash> is there a way to hardcode the title in the output? [18:38] <undercash> i watched the filter options but i didnt see anything related [18:39] <saste> undercash, drawtext? [18:39] <undercash> https://github.com/FFmpeg/FFmpeg/blob/master/doc/ffmpeg.texi [18:39] <undercash> this is yours i assume [18:39] <undercash> i tried the -metadata option but won't work [18:40] <undercash> donno about drawtext, let me look [18:40] <saste> "hardcode the title" it's ambiguous [18:41] <undercash> well to display the filename or metadata title in the output [18:42] <saste> undercash, script it and use drawtext [18:42] <undercash> it can retrieve the filename automatically or you need to specify it everytime? [18:47] <Fjorgynn> Kör fram! [18:50] <undercash> ok thx saste, i ll check this , but i need to recompile with libfree [18:50] <undercash> at least i know what to look at [18:51] <saste> undercash, you're passing the filename, you know it [18:51] <saste> no the filter does not know nothing about the input filenames [18:52] <undercash> $file as text to draw maybe [18:52] <undercash> need to try :) [18:52] <spinx60> hello im finally got ffmpeg to stream sucessfully to Twitch.tv with http://pastebin.com/jnS1ra2H command [18:54] <spinx60> my question now is is there any way for me to make a script to this command where i can easier set more options like quality/resolution/framerate etc can anyone point me in the right direction to a guide or something mabee a complete script that i can edit etc ? [19:04] <spinx60> Anyone at all who can point me to a good guide of some sort ? [19:08] <klaxa> spinx60: maybe look at general x264 options ( http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide ) in general crf, preset, computation power and your upload speed limit video quality [19:09] <spinx60> klaxa Thanks for your answer im not so good at this thats why i asked. but im learning [19:41] <hughmanwho> I'm trying to pull a H264 stream via RTSP and decode it into YUV frames [19:42] <hughmanwho> However I've allocated memory via avformat_alloc_context() then called 'avformat_open_input' [19:43] <hughmanwho> Afterward I am getting a value of 0 for the packet_size within curStream->context->packet_size [19:45] <hughmanwho> Any thoughts? Due to a conversation I had yesterday I was told that changing the value of packet_size can change how fast avformat_open_input takes to execute. [19:46] <hughmanwho> But I'm thinking packet_size shouldn't be equal to zero. [20:40] <hughmanwho> Having some problems connecting to a RTSP stream, my connection code is here: http://pastebin.com/3v5wNMf4 [20:41] <hughmanwho> Then when I call av_read_frame(streamStream->context,&(streamStream->packet)), I always get back an error [20:41] <hughmanwho> Anyone have any thoughts? [21:26] <code-guru> hi anyone can help me with hls muxer please? [21:27] <code-guru> I can't set mpegts metadata when use hls muxer [21:28] <code-guru> for example when I add "-metadata service_provider="Some provider" it doesn't change service_metadata when I use hls m3u8 [21:34] <durandal_1707> code-guru: if it does not but should it is bug, and should be reported at usual location [21:37] <code-guru> I'm going to implement the commands functionality in my code finally, if there it's possible to fix this using the api, it's also ok [21:37] <code-guru> any help please ? [21:41] <PigDude> hi, i get "Uknown format" for a .flac. [21:44] <PigDude> sure llogan [21:44] <PigDude> here is my command and verbose output: http://dpaste.com/970491/ [21:44] <PigDude> with -v, i see error "Unable for find a suitable output format for 'some.flac'" [21:45] <PigDude> ffmpeg -formats shows i have flac, aac, and alac formats available, I think [21:46] <PigDude> here is a paste of a grepped -format ouput, showing those: http://dpaste.com/970505/ [21:47] <PigDude> I don't know what the flags D, E, or A mean in this context [21:47] <llogan> 2004. Why are you using something from 8 years ago? [21:47] <PigDude> i'm using ffmpegx distribution because it takes a very long time to build ffmpeg on my machine and i wanted to do this quickly [21:47] <llogan> that isn't supported here. you're wasting your time using it. [21:48] <PigDude> very well [21:48] <llogan> there are static builds available [21:48] <PigDude> where can I find these? [21:48] <PigDude> these are decent? http://www.evermeet.cx/ffmpeg/ [21:48] <llogan> yes. use most recent one from "snapshots" [21:48] <PigDude> thanks, installing now [21:49] <llogan> ubitux: he is OSX user, AFAIK. [21:49] <ubitux> oh mac os [21:49] <ubitux> yeah right ok sorry :) [21:50] <llogan> don't worry. i almost did the same. [21:50] <ubitux> llogan: btw, it seems libmodplug was added :)) [21:51] <ubitux> that's nice, i can ffplay the keygen musics with the packaged ffmpeg now \o/ [21:51] <llogan> ubitux: as repetence several users are asking how to get a better looking gif output: ffmpeg -i input -pix_fmt rgb24 out.gif any suggestions? [21:51] <llogan> ah, nice. i'll have to try it out [21:51] <ubitux> i used to add two scale filters [21:52] <ubitux> 05:59 <@ubitux> this is generated with: ./ffmpeg -i ~/samples/big_buck_bunny_1080p_h264.mov -vf 'select=gt(scene\,.4),scale=-1:120,format=rgb8,format=rgb24' -vsync vfr -frames:v 10 -r 1 -y bbb-scenes.gif [21:52] <ubitux> i have this in my irc log [21:52] <ubitux> try to adapt [21:52] <llogan> ah, thanks. [21:53] <PigDude> llogan, thanks for showing me that, this new version of ffmpeg has a much nicer command line interface [21:53] <ubitux> llogan: it's far from perfect, but that's nice anyway [21:53] <llogan> PigDude: good. did it work as expected? [21:54] <PigDude> i now see a more detailed error message, and the conversion makes some progress. my output file is written and the operation itself takes perhaps 15 seconds to error out. [21:54] <PigDude> here is the command and output: http://dpaste.com/970549/ [21:55] <PigDude> i thought on a whim that "operation not permitted" might be related to unix permissions, but `sudo` does not remedy this (I would not think this an issue anyway, as this is all occurring in my home directory) [21:56] <PigDude> note that i am using the "libvo_aacenc" format and not "aac", as recommended by ffmpeg when I first specified "aac" [21:56] <llogan> libvo_aacenc isn't any better than aac [21:57] <divVerent> yes, libfdk_aac is the good one [21:58] <PigDude> oh, ok, i will attempt with that format [21:58] <PigDude> it appears i did not build with that [21:59] <PigDude> in any case, i see this error with mp3 output also ... so my input is the source of this issue [22:00] <PigDude> the audio tracks play in VLC well, so it's not like i'm throwing crap at ffmpeg [22:00] <PigDude> though VLC certainly copes well with crap [22:01] <llogan> PigDude: you most likely won't find one with libfdk_aac support due to it being non-free and non-redistributable, IIRC. [22:02] <llogan> does it only occur with that input? [22:03] <PigDude> it occurs with other flac files too. i have not tested each file, but none of my tests have a different result [22:03] <PigDude> what does "Error while decoding stream #0:0: Operation not permitted" mean? [22:07] <PigDude> huh, i'm not even sure what sort of an error this is. the generated audio file works fine [23:27] <hughmanwho> what is the difference between avcodec_open & avcodec_find_decoder? Do you need both of them? [23:33] <exutux> hi [23:41] <exutux> http://pastebin.ubuntu.com/5556480/ can someone can help me to convert this type of file into mp3 audio? [23:42] <exutux> when I try to convert it I have a rustle audio on .mp3 [23:42] <exutux> :( [23:47] <exutux> http://pastebin.ubuntu.com/5556480/ can someone can help me to convert this type of file into mp3 audio? when I try to convert it I have a rustle audio on .mp3 [23:49] <exutux> using ffmpeg -i test.wav -acodec libmp3lame -ac 1 -ab 92k test.mp3 [00:00] --- Sat Feb 23 2013

1 0

ffmpeg-devel.log.20130221
by burek 22 Feb '13

22 Feb '13

[00:12] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:47e92153b61b: mjpegdec: check for packed rgb mjpeg. [00:28] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:551d8b58ebc0: brender_pix: use a grayscale palette as fallback instead of a uninitialized one. [01:01] <michaelni> someone wants to test(&review) "v4l2: bug #1570 and possible solution" ? [01:03] <ubitux> saste: [01:03] <ubitux> libavfilter/vf_overlay.c:201:19: warning: overlay_formats may be used uninitialized in this function [-Wmaybe-uninitialized] [01:03] <ubitux> libavfilter/vf_overlay.c:202:19: warning: main_formats may be used uninitialized in this function [-Wmaybe-uninitialized] [01:04] <ubitux> michaelni: isn't it specific to some devices? [01:04] <ubitux> iirc originally i wasn't able to reproduce [01:04] <ubitux> but the issue might have evolved [01:05] <michaelni> could be specific yes but some testing is still better than none [01:06] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20130220225403&slot=x86_64-archlinux… erm it seems we have some more in vf idet too [01:06] <ubitux> (which looks unrelated to the others) [01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:580e2285577c: lavf/swfdec: remove 8-bit audio FIXME. [01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:7346354b5597: lavf/microdvd: fix muxing. [01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:9ed6af4be2c6: fate/subtitles: add MicroDVD remux test. [01:59] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:dc9edb06b011: vf_idet: Fix macro arguments sideeffect [01:59] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:cdb7a1ac6345: vf_idet: adapt default scores due to bugfix [02:01] <ubitux> thx :) [02:30] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.8:2b6f3be08250: avcodec_align_dimensions2: Ensure cinepak has large enough buffers. [02:30] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.8:dfeef3a209ec: cook: check js_subband_start for validity [02:36] <pajamaboy> quick question, does ffmpeg support authentication against an rtmp target, it appears to in a recent patch, but doesn't seam to be working for me [03:26] <cone-561> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.8.14': unknown revision or path not in the working tree. [03:26] <cone-561> Use '--' to separate paths from revisions [03:26] <cone-561> refs/tags/n0.8.14:HEAD: vf_idet: adapt default scores due to bugfix [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:1fd86f9a2136: vqavideo: fix return type [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:8aedb7515674: tiff: check bppcount [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:5af2fd317d84: wma: check byte_offset_bits [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:17966ae5bb43: qdm2: increase noise_table size [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:0a57df38f468: Update for 0.8.14 [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:2b6f3be08250: avcodec_align_dimensions2: Ensure cinepak has large enough buffers. [03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:dfeef3a209ec: cook: check js_subband_start for validity [03:55] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:a92cdd896345: Merge branch 'release/0.8' into release/0.7 [03:55] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:3be6a8e1eb97: update for 0.7.15 [04:04] <Compn> ubitux : i have an interesting question. do any containers have subtitle codepage metadata ? [04:05] <Compn> like does mkv have stream 3 srt rus cp1251 ? [04:06] <Compn> or utf8 maybe [04:18] <cone-561> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.7.15': unknown revision or path not in the working tree. [04:18] <cone-561> Use '--' to separate paths from revisions [04:18] <cone-561> refs/tags/n0.7.15:HEAD: vf_idet: adapt default scores due to bugfix [08:10] <Belinda> hi [08:11] <Belinda> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas? [09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07master:d714576037e3: Use SMPTE standard notation when showing mxf Universal Labels. [09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:4c2b6ce26cf9: Fix bits_per_coded_sample when encoding png with frame-level multithreading. [09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:4be63111d1a2: Fix bits_per_coded_sample when encoding png with frame-level multithreading. [10:10] <highgod> Hi, I want to ask a question. I want to modify the configure file to make it detect cl/cl.h and libOpenCL.a and make the ffmpeg link libOpenCL.a,the code I add is "enabled opencl && require opencl CL/cl.h OPENCL_init -lOpenCL", but the output is "ERROR: opencl not found", I am not familiar with modifing configure file. I used --extra-cflags=-I/opt/include --extra-ldflags=-L/opt/lib/x86 [10:10] <highgod> --extra-libs="-lOpenCL" to compile the opencl code [10:31] <Belinda> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas? [10:31] <durandal_1707> why? [10:32] <durandal_1707> aacdec is floating point decoder [10:34] <nevcairiel> decoders intentionally only output their native format, if you need another format, use swresample to convert it [10:49] <Belinda> actually i am using it on android and android only recognizes s16, so i wanted to add support for "requested_name". [10:55] <durandal_1707> the aac decoder is float one, but there are fixed decoders on web [10:55] <av500> one can also convert float to fixed [10:56] <av500> clamp( lrintf( *(float*)src * (1 << 15))) [10:57] <durandal_1707> does that give good results? [10:58] <av500> define good [10:59] <divVerent> it sure is good enough :P [10:59] <divVerent> actually, shouldn't one use ((1 << 15) - 1)? nitpicking :P [10:59] <Belinda> i tried float_to_int16_interlave_c but it didn't work [10:59] <av500> why not? [10:59] <nevcairiel> like i said, use swresample to convert float to int16 [11:00] <nevcairiel> thats what its there for [11:00] <nevcairiel> among other things [11:00] <durandal_1707> av500: because floats from aacdec are between -1 and 1 [11:00] <nevcairiel> a patch to allow s16 output would not really be accepted [11:01] <Belinda> :/ [11:01] <nevcairiel> it used to be possible, but that was removed from all sorts of decoders in favor of native output [11:02] <nevcairiel> so instead of converting inside the decoder, just convert after the decoder [11:02] <av500> yes, but one can easily find the old code [11:02] <nevcairiel> same code required, jus [11:02] <nevcairiel> just in your app [11:02] <av500> yes [11:02] <nevcairiel> or just use the library which has functions for this =p [11:03] <Belinda> actually i found the old code but there are many structural changes, it is easy to ruin [11:04] <Belinda> since i am doing this on the fly, i try to reduce cpu and ram usage [11:04] <av500> its only audio [11:04] <av500> doing that conversion wont kill you [11:04] <av500> and you need to do it anyway [11:05] <nevcairiel> the decoder works in float anyway, so it needs conversion at some point, the cpu required is the same no matter where its done [11:06] <av500> yep [11:07] <durandal_1707> so on slow arch with float it is slow already [11:07] <av500> aac takes like 30mhz on an ARM [11:08] <av500> find me an android device that you can even clock that low [11:08] <av500> its a non issue [11:10] <durandal_1707> av500: on arm? maybe [11:10] <av500> yes [11:10] <av500> on an armv7 [11:11] <av500> basically you have more cpu power due to leakage than due to that bit of audio decoding [11:12] <durandal_1707> so fixed point decoder would not be faster? [11:13] <av500> probably [11:13] <av500> well [11:14] <av500> since armv7 with neon etc has float I am not even sure fixed is faster [11:27] <Belinda> AV_SAMPLE_FMT_S16 is not supported by swr_convert? [11:28] <durandal_1707> it is supported [11:28] <durandal_1707> read documentation [11:28] <Belinda> ok sorry, i've made a mistake [11:37] <ubitux> Compn: some formats contain the info yes [11:38] <ubitux> (microdvd for instance) [11:54] <durandal_1707> michaelni: this ^ spammer is really retarded [11:55] <durandal_1707> delete him and png (that links to another site) [11:56] <durandal_1707> also can permission for WikiStart be more restrictive? [11:56] <durandal_1707> is there better bug tracker which have better way of handling fine grained permissions? [11:57] <durandal_1707> if yes, switch to it [12:38] <cone-212> ffmpeg.git 03Nicolas George 07master:01649c79c87c: lavfi/buffersrc: set channel layout if it is known. [12:43] <cone-212> ffmpeg.git 03Nicolas George 07master:9fa5a47197c6: ffmpeg: set resample_pix_fmt for sub2video. [12:51] <highgod> Hi,I add check opencl code to configure file, and set the check code to enabled opencl && require opencl CL/cl.h clEnqueueNDRangeKernel -lOpenCL,clEnqueueNDRangeKernel is a function of libopencl.a, but it failed,Then I review the libopencl.a and found that the function is clEnqueueNDRangeKernel@36, but if I set the code to enabled opencl && require opencl CL/cl.h clEnqueueNDRangeKernel@36 [12:51] <highgod> -lOpenCL, it si still wrong. does anyone konws how to fix it? Thanks [13:01] <highgod> require_cpp? [13:03] <cone-212> ffmpeg.git 03Vicente Jimenez Aguilar 07master:b5ad422bf4e6: doc: filters: Correct BNF FILTER description [13:03] <cone-212> ffmpeg.git 03Diego Biurrun 07master:3e85b46ecf3c: arm: vp8: Add missing #includes for header to compile standalone [13:03] <cone-212> ffmpeg.git 03Diego Biurrun 07master:a0a1ac0f091d: dxva2: Add missing #define to make header compile standalone [13:03] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:6b8f21190d5a: Merge remote-tracking branch 'qatar/master' [13:05] <cone-212> ffmpeg.git 03Paul B Mahol 07master:a3d890527ed3: exr: reset compression in decode_frame() [13:05] <cone-212> ffmpeg.git 03Paul B Mahol 07master:59015c1118ac: exr: constify s in rle_uncompress() [13:05] <highgod> sill don't work [13:16] <highgod> require2 works [13:21] <Compn> durandal_1707 / michaelni : also, it would help to give devels more access on trac to ban spammers. [13:27] <michaelni> theres a bayesian spam filter only problem is we have too few spamers so it hasnt enabled itself yet [13:29] <michaelni> also trac can use various external spam detectors like akismet but IIUC we need a api key for each of these [13:34] <michaelni> and about banning, that slavon888 guy used a different IP each time IIRC [13:35] <iive> might be good idea to block known exit nodes of tor network and vpn/proxy tunnels. [13:35] <iive> probably some spam lists have them all. [13:37] <michaelni> trac already uses multipe ip blacklists, i dunno whats on them exactly though [13:46] <michaelni> Compn, if you want to ban IPs or content see for example trac.edgewall.org/wiki/BadContent and trac.openpsa2.org/wiki/BadIP [13:50] <michaelni> also in case i worded it unclearly, some volunteer needs to get us api keys for akismet and the others listed at http://trac.edgewall.org/wiki/SpamFilter with exception of where they key is just for training the external service) [13:50] <michaelni> also i dont know if these keys are free [13:51] <michaelni> that is if someone wants trac to use these external spam detectors ... [14:04] <durandal_1707> i gonna push rest of exr patches [14:04] <durandal_1707> than gonna try slice threading [15:04] <blackmage> Does anyone know if pushing to an rtmp destination that requires authentication is supported [15:04] <blackmage> it looks like it was added in a patch, but i'm not certain, based upon some of the docs i've seen [15:06] <durandal_1707> what docs? [15:06] <durandal_1707> didn't you said in another channel that your buid is missing ssl support? [15:06] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:bdeb61ccc679: h264: check that luma and chroma depth match [15:09] <blackmage> i'm seeing mentions of it, but here is the one that says you need ssl [15:09] <blackmage> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=723#p2581 [15:10] <durandal_1707> ssl is needed [15:12] <durandal_1707> or not [15:32] <durandal_1707> could fate survive 8 mb of new samples? [15:52] <durandal_1707> it looks like it needs user[:pass]@hostname [15:52] <durandal_1707> (reading source code) [15:53] <blackmage> okay, i'll try that, thanks durandal [16:29] <wm4> so it seems some libavformat demuxer actually expect a special AVIOContext and use pb->opaque without checking [16:30] <wm4> e.g. hls.c accesses this, and if the caller passes a custom AVIOContext, it will crash [16:45] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:669d50b44596: ec: check that reference pictures have matching parameters before using them. [17:06] <Compn> wm4 : btw , write down any info you are learning, so the next person who wants to do what you do, doesnt have to guess and figure out ... :) [17:06] <Compn> aka please update the docs or make source code comments, or wiki entry , anything [17:08] <wm4> uh what, like what [17:52] <Compn> wm4 : i thought you were adding a new format, not investigating a bug, nm [18:07] <durandal_1707> finally! ffmpeg is recognised by google+: https://plus.google.com/+ffmpeg [18:08] <Compn> finally, they can track ffmpeg people on google servers, yay... [18:17] <durandal_1707> https://docs.google.com/presentation/d/1awg1CHM1w128iOBp_JOxE2DgHfywBeyjDe2… [18:20] <ubitux> :) [18:23] <ubitux> i like slide 6 too [18:31] <durandal_1707> you can even chat with other readers [18:52] <durandal_1707> http://revolution.byu.edu/programmingconcepts/potatoPseudocode.png [19:31] <durandal_1707> anyone want to review my patches? [20:04] <ubitux> durandal_1707: can i help? [20:05] <saste> ubitux, durandal_1707: a second look at blend may help, especially with regards to interface/syntax [20:08] <ubitux> blend looked good to me when i had a look at it [20:08] <saste> ubitux, also boxblur and overlay patch [20:08] <ubitux> named options? [20:09] <ubitux> saste: btw, did you see the warnings in overlay? [20:09] <saste> yes, and we're still missing a few filters if you want to help with low hanging fruits [20:09] <saste> ubitux, nope [20:25] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:8036ba299d33: h264: move next/last picture init to decode_slice_header [20:27] <durandal_1707> sase left [20:46] <ubitux> erm max_samples isn't honored. [20:58] <ubitux> looks like a mistake in 82541d83 [20:58] Action: ubitux mailing Nicolas [20:59] <durandal_1707> "Failed to inject frame into filter network" <--- what that means [21:01] Action: ubitux reviewing boxblur [21:18] <cone-48> ffmpeg.git 03Paul B Mahol 07master:47619e41c4a1: blend filter [21:19] <ubitux> yay :) [21:19] <durandal_1707> now fix bugs in it [21:19] <ubitux> :)) [21:20] <ubitux> durandal_1707: how i imagine you currently: http://z0r.de/5024 [21:21] <durandal_1707> ubitux: whatever it is i do not have flash/java or some other plugin [21:22] <ubitux> i think it works with ffplay [21:22] <ubitux> but it won't loop [21:22] <ubitux> too bad :) [21:22] <ubitux> indeed it works [21:22] <ubitux> you need to imagine the audio and video loop now [21:23] <ubitux> (they are not in sync) [21:23] <durandal_1707> ubitux: ??? [21:23] <ubitux> durandal_1707: wget 'http://z0r.de/L/z0r-de_5024.swf' [21:24] <ubitux> the basic idea of the loop (at least most of them) is to have a/v not really in sync [21:24] <ubitux> for instances 5 frames looping, and a longer background audio, looping too [21:24] <durandal_1707> wget: Command not found. [21:24] <ubitux> (the a/v sync doesn't really matter) [21:24] <ubitux> mmh [21:24] <ubitux> fetch? [21:25] <ubitux> curl ... > out.swf? [21:25] <durandal_1707> hmm why clip is in rawvideo/audio, feared of patents? [21:27] <ubitux> no idea [21:27] <ubitux> happens often [21:27] <ubitux> (in swf) [21:48] <durandal_1707> saste: gonna review [PATCH] Handling special characters in a URL ? it take really to loong [21:49] <saste> durandal_1707, too long? he posted a reply tomorrow [21:49] <durandal_1707> ping reply [22:00] <ubitux> actually the min/max samples works really well [22:00] <ubitux> awesome Nicolas is awesome [22:00] <ubitux> saste: you should use that feature too ;) [22:01] <ubitux> i wonder if asetnsamples couldn't use it too [22:02] <wm4> why isn't there libswblend :( [22:02] <saste> ubitux, you should thank anton to be fair [22:02] <saste> wm4, you know the answer [22:02] <wm4> "patches welcome"? [22:02] <saste> yes [22:02] <wm4> "no thank you" [22:03] <ubitux> saste: no [22:03] <ubitux> it's a feature from Nicolas [22:03] <ubitux> not present in libav [22:03] <saste> is it? [22:03] <ubitux> yes :) [22:03] <ubitux> see 0689d5e17ac0a3269c604b4df2c01140be328647 [22:04] <durandal_1707> wm4: why libswblend? [22:04] <ubitux> durandal_1707: to blend subtitles \o/ [22:04] <ubitux> (fast!) [22:04] <wm4> durandal_1707: it would gather useless functionality into a library, instead of shattering it across libavfilter where it's needed for various things [22:04] <durandal_1707> there are actually many libs that do blend but they all sucks afaik [22:05] <llogan> saste: worthless fact for the day: there are more than 1500 -user subscribers. [22:05] <ubitux> how many -devel? [22:05] <llogan> also i heard back from mike. he'll be able to deal with account stuff. [22:05] <durandal_1707> day before were 1800 (so it is 0.3 % drop) [22:06] <llogan> ubitux: 1327 [22:06] <ubitux> nice [22:06] <ubitux> 10 left [22:06] <durandal_1707> that all trackers (gmane,google, etc) [22:06] <llogan> digest members are included but digests are disabled and they do not recieve mail [22:08] <llogan> saste: so i'll try to get the gsoc ideas page up later today (unless someone gets to it first), but i've been in a timestuck editing videos for the last few days... [22:08] <saste> llogan, no hurry [22:09] <saste> i'll be able to work actively on gsoc not before the first week of march [22:09] <durandal_1707> llogan: so is number of users/devels stagnating or they are all spammers? [22:10] <llogan> i don't actively watch the subscriber numbers and there are no spammers with accounts AFAIK (not including hijacked [yahoo mostly] accounts) [22:14] <durandal_1707> there is no log or something like graph? [22:17] <llogan> durandal_1707: not in the admin web panel, AFAIK. [22:20] <llogan> but a graph would be interesting [22:20] <wm4> durandal_1707: have you found a sample for vf_stereo3d yet? [22:21] <saste> anyone wants to review the partial overlay patch? [22:21] <saste> it's the missing one in my overlay patchset [22:21] <durandal_1707> wm4: not yet there, why? [22:21] <wm4> just wondering [22:22] <durandal_1707> it is not that mandatory for porting [22:22] <wm4> its output looks like modern art for me [22:22] <durandal_1707> anything with bunch of different colors can be used [22:23] <durandal_1707> wm4: get glasses [22:23] <wm4> and I wonder what stuff like "anaglyph_green_magenta_half_color" actually means [22:24] <durandal_1707> nothing relevant for anime scene [22:29] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:8a6449167a6d: avcodec_decode_audio4: check got_frame_ptr before handling initial skip [22:29] <cone-48> ffmpeg.git 03Andrea3000 07master:8d8c59480e1b: matroska: fix missing , [22:31] <durandal_1707> wm4: looks like that filter is only way to experience 3d at home for free [22:43] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:04220b473e9d: h264: ensure that get_format() is called when changing format but not otherwise. [22:43] <cone-48> ffmpeg.git 03Matt Wolenetz 07master:82a4a4e7caa9: Fix Win64 AVX h264_deblock by not using redzone on Win64 [23:06] <cone-48> ffmpeg.git 03James Almer 07master:67b3fcf65578: configure: Use libiconv on systems that lack built-in iconv [23:06] <cone-48> ffmpeg.git 03Mirjana Vulin 07master:8d2eb5fe58ae: mips: optimization for float aac decoder (sbr module) [23:06] <cone-48> ffmpeg.git 03Mirjana Vulin 07master:46d52a0b098f: mips: optimization for float aac decoder (ps module) [23:10] <ubitux> it almost makes me wanting a mips device [23:11] <michaelni> ubitux, ask the @mips.com people, maybe they would give you one for free [23:16] <ubitux> not sure if i'll really make use of it, it's just a whim :) [23:16] <ubitux> but it's nice to see all these optims [23:18] <wm4> durandal_1707: it appears matroska even has hints for this 3D crap... [23:20] <JEEB> H.264 can have hints as well, but yeah -- matroska had some crap too [23:22] <durandal_1707> there is unsupported H.264 extension IIRC [23:23] <JEEB> MVC you mean? that's separate, and pure H.264 decoders just don't have to care as it will get one of the views that way just fine [23:23] <JEEB> I mostly meant the SEI (or VUI was it?) stuff [23:23] <JEEB> where you can note what kind of multi-view stream it is [23:23] <JEEB> x264 has support for those [23:24] <Skyler_> SEI [23:41] <ubitux> huh [23:41] <ubitux> seems i have the audio normalization working [00:00] --- Fri Feb 22 2013

1 0

ffmpeg.log.20130221
by burek 22 Feb '13

22 Feb '13

[00:08] <xangadix> I have two questions [00:08] <xangadix> 1) so I can code to quicktime animation codec using -f mov -vcodec qtrle -ac 2 [00:08] <xangadix> but how to set the quality of the image sequence? [00:08] <xangadix> in most editing programs, I can set a quality from 0 to 100%, but I don't know how to tell ffmpeg that [00:08] <xangadix> 2) [00:09] <xangadix> is it possible to have ffmpeg make dxv files ? [00:09] <xangadix> http://resolume.com/software/dxv.php [00:10] <xangadix> and how would i add that :) [00:10] <xangadix> these are my humble questions [00:10] <xangadix> ow great and mighty people in #ffmpeg [00:15] <ubitux> specs anyway? [00:16] <ubitux> or it will need some reverse? [00:16] <xangadix> specs for dxv ? [00:16] <ubitux> yes [00:16] <xangadix> I'm not sure where to find those either [00:16] <xangadix> google stumbles on suggestions for divx, and dxva of which I'm not sure is the same [00:17] <ubitux> for your first question, the quality is controlled by -qscale:v and relatex (qmin, qmax, etc) [00:17] <xangadix> much obliged [00:18] <xangadix> saw that for making jpgs, I was hoping it was kind of the same :) [00:19] <xangadix> if the codec is not open source, can it still be used in ffmpeg? I ran through this conversation, http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2010-October/000227.html but couldnt make it out completely [00:21] <ubitux> it will need some reverse [00:21] <xangadix> by that you mean reverse engeneering ? [00:21] <ubitux> yes [00:21] <xangadix> darn [00:21] <ubitux> you can a open a feature request on the trac [00:21] <ubitux> and wait for a decade [00:21] <ubitux> ;) [00:21] <xangadix> :) [00:21] <xangadix> I will [00:21] <ubitux> unless more ppl are interested [00:22] <ubitux> but it seems no one care atm [00:22] <xangadix> in the meantime I will have to render to photojpg first, and then batch it with sony vegas or so [00:22] <xangadix> how come that most video editing software will render it then? [00:22] <xangadix> they are using the same dlls I'm assuming ? [00:50] <sabton> i call avformat_open_input() with an AVFormatContext that I have pre-allocated, and have specified custom AVIOContext on, but it is opening a new http connection... so i end up with 2 connections. how can i get avformat_open_input() to use my custom AVIOContext? [01:03] <meelu> i don't get it, i do "/usr/bin/ffmpeg /var/www/videos/filename.flv -vol 256 -y -acodec libmp3lame -ab 256k /var/www/output/filename.mp3 2> /var/www/logs/filename.txt" and it outputs "File '/var/www/videos/filename.flv' already exists. Overwrite ? [y/N]" [01:10] <klaxa> meelu: you're missing "-i" [01:10] <meelu> oh thanks [01:10] <meelu> :_) [01:13] <meelu> klaxa [01:13] <klaxa> meelu [01:13] <meelu> nothing [01:13] <meelu> sorry bout that [01:13] <klaxa> kk [01:27] <votz> How can I instruct ffmpeg to avoid writing any id3 tags (id3v1 and id3v2) for an mp3? I just want the mp3 output, starting with the encoder information, LAME3.87.4UUU... [01:49] <spinx60> Hello are there any friendly and late night ppl like me out here ? [01:59] <xangadix> spinx60 not sure about the friendly part :p [02:32] <pajamaboy> is it possible to perform rtmp authentication when trying to push a stream to an rtmp target [02:32] <pajamaboy> such as akamai [02:33] <xangadix> does rtmp support something like rtmp://user:pass@address.com ? [02:33] <xangadix> like ftp [02:33] <pajamaboy> don't think so, i tried that actually [02:34] <xangadix> hm :-/ [02:35] <pajamaboy> i also tried pubUser and pubPasswd [02:35] <pajamaboy> maybe i need to ask the devs [02:35] <pajamaboy> :( [02:36] <xangadix> dunno, I have only done it through wowza, and that uses a local autherisation file (on the server in that case) [02:37] <pajamaboy> yea, thats how i'm doing it now too [02:37] <pajamaboy> i'm trying to move to using nginx-rtmp [02:39] <xangadix> can't help you there :( [02:41] Action: xangadix zZz [08:20] <Garryco> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas? [08:21] <Garryco> I am trying to get s16 pcm format out from decoded aac, but it currently only gives FLTP [09:25] <firewalker> hello everyone! [09:26] <Garryco> yo [09:26] <firewalker> I am not sure if a problem I face with FFmpeg 1.1.2 is an ffmpeg bug or not [09:26] <firewalker> the issue is this one https://bugs.archlinux.org/task/33901 [09:27] <firewalker> should I open a bug report for ffmpeg, or not? [10:32] <divVerent> double WTF: [10:32] <divVerent> WTF1: ffmpeg play MIDI files?!? [10:32] <divVerent> WTF2: it totally fails at them [10:32] <divVerent> as if notes never end [10:32] <durandal_1707> how are they detected? [10:32] <divVerent> cause is apparently libmodplug "supporting" MIDI files [10:32] <divVerent> but failing totally at them [10:33] <divVerent> Input #0, libmodplug, from 'VEGGIESG.MID': [10:33] <durandal_1707> what happens? [10:33] <divVerent> instrument : Timidity GM patches [10:33] <divVerent> it sounds TOTALLY wrong, many stuck notes [10:33] <divVerent> can't even recognize the song [10:34] <divVerent> apparently, from listening closely... apparently the last note on each channel continues playing until a next note comes [10:34] <divVerent> also, it doesn't sound AT ALL like the timidity GM patches I am using [10:34] <divVerent> haha, I see in strace... it did ATTEMPT to load timidity patches, but didn't succeed at any [10:35] <divVerent> [pid 17742] open("/usr/local/share/timidity/instruments/.pat", O_RDONLY) = -1 ENOENT (No such file or directory) [10:36] <divVerent> I actually remember noticing this like 3 years ago when libmodplug ADDED midi support... and was really disappointed at the bad output... so much, I made darkplaces whitelist file extensions to pass to modplug [10:37] <malinens> ffmpeg version git-2012-07-02: ffmpeg -i "a.mp4" -an -ss 3 -t 00:00:01 -r 1 -threads 1 -y "a.%d.jpg" and I get this error on one video: Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) [10:38] <malinens> when started from -ss 10 then screenshots are created correctly. is there fool-proof way to generate screenshot with ffmpeg? [10:41] <malinens> http://pastebin.com/ZPmkSWCL [11:02] <juanfont> hi. is there any way to force the use of mpegtsraw demuxer from the stream (or the url, or any place except the code)? thanks :) [11:02] <juanfont> *from the stream -> from the transport stream [11:24] <n1n0> the movie I need to extract images from is in the home folder. when using this command ffmpeg -ss 00:07:00 -t 00:02:00 -i i.MOV -r 0.1 P1000072/image%3d.jpg [11:25] <n1n0> I get this output ffmpeg -ss 00:07:00 -t 00:02:00 -i i.MOV -r 0.1 P1000072/image%3d.jpg [11:25] <n1n0> ffmpeg version 0.8.5-4:0.8.5-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers [11:25] <n1n0> built on Jan 24 2013 18:03:14 with gcc 4.6.3 [11:25] <n1n0> *** THIS PROGRAM IS DEPRECATED *** [11:25] <n1n0> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [11:25] <n1n0> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'i.MOV': [11:25] <n1n0> Metadata: [11:25] <n1n0> major_brand : qt [11:25] <n1n0> minor_version : 512 [11:25] <n1n0> compatible_brands: qt [11:25] <n1n0> creation_time : 1970-01-01 00:00:00 [11:25] <n1n0> encoder : Lavf52.93.0 [11:25] <n1n0> Duration: 00:03:10.00, start: 0.000000, bitrate: 713 kb/s [11:25] <n1n0> Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 634x360, 613 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc [11:25] <n1n0> Metadata: [11:25] <n1n0> creation_time : 1970-01-01 00:00:00 [11:25] <n1n0> Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 95 kb/s [11:25] <n1n0> Metadata: [11:25] <n1n0> creation_time : 1970-01-01 00:00:00 [11:25] <n1n0> Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [11:25] <klaxa> don't... [11:25] <n1n0> [buffer @ 0x87b2780] w:634 h:360 pixfmt:yuv420p [11:25] <n1n0> [avsink @ 0x87b2920] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [11:25] <n1n0> [scale @ 0x87b2da0] w:634 h:360 fmt:yuv420p -> w:634 h:360 fmt:yuvj420p flags:0x4 [11:25] <klaxa> post all that shit in the channel [11:25] <n1n0> Output #0, image2, to 'P1000072/image%3d.jpg': [11:25] <n1n0> Metadata: [11:25] <n1n0> major_brand : qt [11:25] <n1n0> minor_version : 512 [11:25] <n1n0> compatible_brands: qt [11:25] <n1n0> creation_time : 1970-01-01 00:00:00 [11:25] <n1n0> encoder : Lavf53.21.1 [11:25] <klaxa> never do that [11:25] <klaxa> ever [11:25] <n1n0> Stream #0.0(eng): Video: mjpeg, yuvj420p, 634x360, q=2-31, 200 kb/s, 90k tbn, 0.10 tbc [11:25] <n1n0> Metadata: [11:25] <n1n0> creation_time : 1970-01-01 00:00:00 [11:25] <n1n0> Stream mapping: [11:25] <n1n0> Stream #0.0 -> #0.0 [11:25] <n1n0> Press ctrl-c to stop encoding [11:25] <n1n0> frame= 0 fps= 0 q=0.0 Lsize= -0kB time=10000000000.00 bitrate= -0.0kbits/s dup=0 drop=10 [11:25] <n1n0> video:0kB audio:0kB global headers:0kB muxing overhead -inf% [11:26] <n1n0> ok, can you check it this time only [11:26] <klaxa> you can go to #libav because you are not using ffmpeg [11:27] <klaxa> and please... use pastebin or something to paste logs and post the link [11:27] <klaxa> spare them from this spam [11:29] <n1n0> ok, cause on another software support channels (within the freenode) was ok to post terminal lines [11:29] <n1n0> thanx anyway [11:29] <klaxa> you do not post more than 3 lines in any irc channel that is sane [11:30] <n1n0> ok, thanks [11:38] <durandal_1707> malinens: what you are actually trying to do? [11:39] <malinens> durandal_1707: create screenshot from video [11:41] <durandal_1707> malinens: single screenshot at exact time? [11:41] <malinens> Basically at the start of the video, single [12:47] <malinens> is there chat log for #ffmpeg channel somewhere on interwebs? [12:49] <durandal_1707> http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/ [12:50] <ubitux> only the devel channel is logged afaik [12:51] <durandal_1707> not really [12:51] <durandal_1707> user too [12:51] <malinens> durandal_1707: I just wanted to check if You wrote someting to me when I was offline about ffmpeg screenshots [12:52] <durandal_1707> malinens: i think using select filter may work better [12:52] <malinens> ok, I'll check it [12:58] <ubitux> durandal_1707: oh, right, my bad [14:06] <ubitux> heh it seems the ffmpeg package in archlinux is now linked against a bunch ffmpeg-specific libraries [14:06] <ubitux> nice. [14:07] <ubitux> fontconfig, libass, libbluray, libmodplug, libv4l2, postproc, ... [14:08] <ubitux> that's gonna get tricky if libav wants to replace the package at some point :) [14:08] <durandal_1707> for example? [14:09] <ubitux> for example? [14:09] <ubitux> all the libs i quoted are not supported in libav afaik [14:13] <durandal_1707> but libmp3lame is, so i still do not get what you mean [14:15] <ubitux> what i mean is that if libav wants to replace the ffmpeg package here, they will need to drop a lot of configure switches, which might be very tricky for them to do [14:15] <ubitux> (because feature-loss) [14:19] <durandal_1707> why would they want to drop ffmpeg package? [14:19] <sacarasc> Because many distros are doing it these days. [14:20] <durandal_1707> many? [14:20] <sacarasc> Ubuntu and all the Ubuntu likes, Debian... [14:22] <durandal_1707> really? [14:23] <durandal_1707> perhaps if every libav developer becomes package maintainer for separate distro that may be true [14:35] <meelu> any idea why this cuts off without completing? i mean it doesn't run in the background untill completion like i want it to $exec_string = '/usr/bin/ffmpeg -i /var/www/videos/filename.flv -vol 256 -y -acodec libmp3lame -ab 256k /var/www/output/filename.mp3 2> /var/www/logs/filename.txt >/dev/null 2>&1 &'; exec($exec_string, $output, $return); if (!$return){ echo 'yey'; } [14:43] <ubitux> sacarasc: "many" is a bit excessive; it's present in two distro, gentoo (where ffmpeg is also available), and debian (because there is a libav maintainer here) [14:43] <ubitux> of course being in debian make it spread in all the variants [14:46] <blackmage> do you guys know how to do rtmp authentication with ffmpeg? [14:46] <blackmage> i'm getting an error 403 saying it needs authmod=adobe [14:46] <blackmage> but i can't find any documentation [14:48] <ubitux> add authmod=adobe&user=... in the url? [14:53] <blackmage> i tried that already, doesn't seam to work [14:53] <blackmage> :( [14:59] <blackmage> apparently i need to compile with ssl support [14:59] <blackmage> and add pubUser= and pubPasswd= [15:17] <meelu> how do i run an ffmpeg command with php in the background [15:17] <meelu> anyone knows? [15:20] <blackmage> http://pastie.org/6291660 [15:21] <durandal_1707> you are using old and wrong version [15:21] <durandal_1707> which does not have native rtmp support either [15:23] <blackmage> okay, i'll compile it, and try instead of using the ubuntu package [15:23] <blackmage> i was able to record some streams from an rtmp source, so i thought this one had rtmp support [15:27] <durandal_1707> adobe and limelight published authentication is available from 1.1 release [15:28] <durandal_1707> previouse version (guessing) used whatever librtmp needed [15:29] <blackmage> i have a compiled version of the latest from github, i just didn't use it in that command i ran on my local box (it has the same error) [15:30] <blackmage> so i'm probably just missing some options, i'm thinking [15:30] <durandal_1707> post pastie of that one [15:36] <blackmage> in the v1.1 [15:36] <blackmage> is it passed in by doing username=blah password=blah [15:37] <sabton> Hello, can someone please have a look at this? trying to convert a flac file, this used to work but now seems like FFmpeg behavior changed when dealing with multiple streams: http://pastebin.com/T15cJZA5 [15:37] <durandal_1707> blackmage: dunno, i don't use it [15:38] <blackmage> http://pastie.org/6292723 [15:43] <relaxed> sabton: it doesn't know how to handle the cover art, add -vn [15:43] <Mavrik> *grumble* Compiling ffmpeg for ARM is PITA [15:45] <durandal_1707> blackmage: looks like args you give are wrong [15:46] <sabton> relaxed: thanks. it appears -vn disables video extraction. this i fine, any other gotchas like this come to mind? [15:47] <Mavrik> well, yeah, passing "disable video" probably does that ;) [15:47] <relaxed> sabton: No. I'm not sure if ogg supports album art, but you could try -vcodec copy to see if it works. [15:48] <sabton> relaxed: what was the old behavior? did ffmpeg know to ignore the video stream since i don't provide a matching output video stream? [15:50] <relaxed> I think it's considered metadata and ffmpeg ignored it. [15:50] <relaxed> s/it's/it was/ [15:50] <relaxed> both work :) [15:53] <sabton> gotcha, thanks :) [16:53] <JodaZ> anyone got an idea how i could make webm thumbnails of pictures ? like a few seconds from a few points of a video cut together [17:14] <relaxed> JodaZ: webm is a container [17:48] <JodaZ> relaxed, the sky is blue [18:16] <MattCz> Any way to speed up 'avformat_find_stream_info'? It's taking about 75 seconds to execute right now but seems like it should be a pretty fast one... [18:21] <Mavrik> MattCz, are you reading a network stream? [18:57] <spinx60> Hello. im new to ffmpeg and im trying to stream to Twitch.tv / Justin.tv and my script seems to work very well and my stream goes live. But theres no capture at all stream is black screen and even if i try and record desktop to .mkv file i cant even open it in like vlc player etc. i have no clue what to do been searching since last night and now im out of ideés can anyone shed some light on my problem... this is m [18:57] <spinx60> y scrip http://pastebin.com/b1wsdPH7 [18:58] <JodaZ> spinx60, i usually use rtmpdump to stream twitch, in fact, i am doing that right now [18:59] <spinx60> then Jodaz can you help me out plz :) [18:59] <El_Angelo> hi [18:59] <spinx60> im starting my stream with this command streaming live_xxxxxxxxxxxx for the key ofc [19:00] <JodaZ> ooh, wait, stream to twitch not from, never mind [19:00] <El_Angelo> i used ffmpeg to concatenate 2 ts files, but now the timestamps seem to be missing from the final ts files [19:00] <Mavrik> spinx60, pastebin also the output [19:00] <El_Angelo> i can't ff in the file... [19:01] <spinx60> ehhh ? what so thats what im missing ? :) how schould a output look like ? im read guides and watched youtube clips etc and this is all they talked about.. [19:02] <Mavrik> spinx60, the text output of ffmpeg [19:02] <Mavrik> as it runs. [19:02] <spinx60> ohh yeah 1 sec [19:03] <spinx60> here it is http://pastebin.com/AYMwEyqX [19:04] <spinx60> thats what happends when i execute the command streaming live_xxxxxxxxxxxx the x is my key [19:04] <Mavrik> hmm [19:04] <Mavrik> and you say you get black frame only? with or without audio? [19:05] <Mavrik> spinx60, um [19:05] <spinx60> whitout audio [19:05] <Mavrik> spinx60, pass "-pix_fmt yuv420p" to ffmpeg after libx264 [19:05] <spinx60> and yes only a bland screen on my twitch channel [19:06] <spinx60> schould i paste it in to my scripe after libx264 you mean ? [19:07] <Mavrik> spinx60, add it as parameter to ffmpeg command after you set vcodec [19:10] <spinx60> yeah i did so [19:10] <spinx60> no change [19:10] <spinx60> my stream is http://www.twitch.tv/undate btw [19:12] <spinx60> think i did what you wanted me to il paste the new script file with the changes for you too look at to see that i did right [19:13] <spinx60> stream is live now btw so you can se for yourself [19:15] <spinx60> http://pastebin.com/Q4uRiW1J [19:16] <spinx60> hmm not seen it before but in the stream window i got a message now saying. seems to be a problem downloading the stream. not seen it before [19:20] <JodaZ> spinx60, no "" around the -pix_fmt [19:21] <spinx60> yeah tried that to. no change in that either [19:23] <spinx60> what about these lines in the output http://pastebin.com/7V1Adtce [19:23] <spinx60> can it have something to do with it ? [19:24] <spinx60> i really cant wait to begin streaming. would love it soo much [19:24] <JodaZ> dunno, but definitely remove the "" arround the -pix_fmt, thats definitely wrong [19:25] <spinx60> yeah i did and i tried it whiout those aswell and no differense at all [19:26] <JodaZ> paste output without them [19:27] <spinx60> okey i can do that just started the stream again so. [19:32] <spinx60> here ya go http://pastebin.com/7gNcpscn [19:32] <spinx60> im using arch linux if it relevant [19:35] <MattCz> Sorry, wrote up my question then walked away for a second 'avformat_find_stream_info' is currently taking me 75 seconds to execute. [19:35] <MattCz> Which seems long, mavrik asked me if I'm reading a networks stream. [19:36] <MattCz> I'm reading from a 'rtsp://' URL coming from a camera [19:38] <spinx60> you guys finding anything or does it look dark for me to get it to work ? [19:39] <spinx60> JodaZ: Thanks for helping me anyway i will be here if you find any answers [19:52] <frogprince_mac> hi is this also a support channel for ffmpeg-php? [19:58] <durandal_1707> frogprince_mac: nope [20:03] <spinx60> this is frustrating as hell.. going crazy here :P [21:05] <dceo> what is the difference between -vcodec h264 and -vcodec libx264 -- I have an ffmpeg compiled on May 16 2006 using the former, and when I play back an mp4 file encoded from it using other media players like vlc it says 64bit atoms are not supported [21:07] <sabton> when calling av_read_frame(), where is the stream index that it is reading from assigned? [21:10] <dceo> i'm looking now, it shows av_read_frame( p_sys->ic, &pkt ); I will keep looking around a bit through those structures [21:10] <JodaZ> wow, 2006 *_* they had 264 back then [21:11] <dceo> seems old but 2006 was leading up to 2009 so its connected to an era ;) [21:13] <sabton> it looks like AVFormatContext->cur_st holds the selected stream to decode... [21:13] <sabton> i guess my question is now, when is cur_st initially set? I am doing custom IO, i think i need to set the stream index myself' [21:14] <dceo> i guess I can try and print out the number of streams, nb_streams as a first debugging step [21:15] <Mavrik> dceo, "-vcodec h264" will try to use the internal ffmpeg (experimental) encoder [21:15] <Mavrik> while libx264 uses external "libx264" encoder for h.264 [21:15] <Mavrik> using the internal one isn't recommended :) [21:16] <JEEB> uhh [21:16] <Mavrik> MattCz, try lowering probe_size on AVFormatContext if your stream is very low on bitrate [21:16] <JEEB> I don't think the libavcodec H.264 encoder never got in [21:16] <JEEB> it's the decoder [21:16] <JEEB> s/never/ever/ [21:17] <Mavrik> JEEB, it's in for awhile no AFAIK, just needs the "-experimental 2" switch afaik [21:17] <Mavrik> even though, if I remember correctly, ffmpeg prefers libx264 when you type "h264" [21:18] <dceo> I would imagine they have test streams that they run against, maybe support has somehow dropped for these older encoders then [21:18] <JEEB> no, there is no h264enc [21:18] <JEEB> there is only libx264 as a H.264 encoder in ffmpeg [21:19] <JEEB> the very low-feature H.264 encoder (baseline only) got to the mailing list a few years ago [21:19] <JEEB> but never, never got into the repository as far as I know [21:22] <JEEB> yeah, it was on the mailing list in '06 [21:22] <Mavrik> hrm, of course, [21:22] <Mavrik> I was thinking of AAC [21:22] <Mavrik> I'm gonna go to sleep now, stop talking crap. Sorry. [21:22] <JEEB> http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2006-June/012884.html [21:22] <JEEB> I don't think this ever got into libavcodec, mainly because it was completely useless for real usage :) [21:22] <JEEB> even back then [22:32] <timofonic> Hello [22:33] <timofonic> What about Daala from Xiph? I hope it surpasses H265, H264 and WebM :) [22:35] <Zeeflo> is .sup supported by nightly builds (burek) ? [22:36] <JEEB> timofonic, yes -- a format that is not even specified yet... [22:37] <JEEB> the plan for them is to make something better than HEVC, H.264 or VP8/9 [22:37] <JEEB> but currently there is code to test ideas and discussion on what kind of design the format should take [22:37] <timofonic> JEEB: I hope Google merges efforts instead being lone gunmen... [22:38] <timofonic> JEEB: And use a matroska-based container, at least. As for audio... Opus? [22:38] <JEEB> uhh [22:38] <JEEB> > opus > matroska [22:38] <JEEB> > implying they will get that finished [22:38] <JEEB> also [22:38] <JEEB> xiph is making up their own new container format as far as I can see [22:38] <timofonic> What about Dirac too? [22:39] <timofonic> Another one container? :P [22:39] <timofonic> Opus < matroska? Uhm? [22:39] <JEEB> no [22:40] <JEEB> > is used in many places to mark quoted content [22:40] <JEEB> also given how early in design and development daala is, I would pretty much say to you that you should keep it "behind your ear" and check up on it every now and then [22:40] <JEEB> but [22:40] <JEEB> it's at the very least a few years from being a reality [22:45] <Zeeflo> why the f all these different containers.. [22:45] <Zeeflo> Its madness [22:45] <Zeeflo> and makes no sense [22:45] <Zeeflo> a container is a container.. [22:45] <Zeeflo> doesnt matter if its blue or green [22:45] <Zeeflo> ffs [22:45] <JEEB> well, various containers have various problems [22:46] <JEEB> generally when a sane community makes a new one, it fixes some of the problems related to the previous ones [22:46] <Zeeflo> its idiotic [22:46] <JEEB> not really [22:46] <JEEB> matroska is already starting to show its limitations as opus is showing [22:47] <JEEB> hacking stuff into "mp4" really isn't worth it because nothing outside of some OSS stuff will support it [22:47] <JEEB> the new xiph container's ideas finally seemed more sane than their stuff with the original ogg "container" >_> [22:48] <Zeeflo> the future is streaming containers.. [22:48] <JEEB> and? the new xiph one should be better for that, too [22:48] <JEEB> than matroska [22:48] <JEEB> for one [22:48] <Zeeflo> and no players would support it.. [22:48] <Zeeflo> (flash) [22:49] <Zeeflo> like weve seen with mkv [22:49] <Zeeflo> its a great container, but you cant use it for streaming.. [22:49] <Zeeflo> So who cares about it [22:49] <JEEB> you have a very limited view on what is "streaming" then [22:49] <Zeeflo> no [22:49] <Zeeflo> mkv's cant rtmp stream [22:49] <Zeeflo> not through flash [22:49] <JEEB> it's an adobe protocol, are you fucking surprised? [22:49] <Zeeflo> not through html5 [22:50] <Zeeflo> no [22:50] <Zeeflo> im not surprised. [22:50] <JEEB> and html5 doesn't have rtmp in it to begin with as far as I know [22:50] <Zeeflo> But when 90% of the worlds devices use flash to receive audio/video then whos stupid? [22:50] <Zeeflo> the people who create containers that flash will support or those who dont? [22:50] <JEEB> ... [22:50] <JEEB> so you say EVERYONE should use FLV for EVERYTHING? [22:50] <Zeeflo> you cant really debate that [22:50] <Zeeflo> no [22:51] <JEEB> and you do understand how flash doesn't even support shit that is defined in "mp4" official, for example? [22:51] <JEEB> you do, right? [22:51] <Zeeflo> the only container that will go with rtmp streaming is mp4 [22:51] <JEEB> ... [22:51] <Zeeflo> and webm [22:51] <JEEB> rtmp does NOT use mp4 [22:51] <JEEB> and it does NOT use webm [22:51] <Zeeflo> but webm sucks even harder [22:51] <JEEB> rtmp is FLV [22:51] <Zeeflo> in its truest form yes [22:52] <Zeeflo> but its a shitty container [22:52] <JEEB> there's no other form [22:52] <JEEB> you can use various containers as sources for rtmp streams, yes [22:52] <Zeeflo> you can rtmp stream mp4's [22:52] <JEEB> yes, it gets remuxed on fly [22:52] <Zeeflo> and ogg [22:52] <Zeeflo> etc [22:52] <JEEB> jesus christ have you ever actually looked at rtmp streams' content!? [22:52] <JEEB> instead of talking out of your ass [22:52] <Zeeflo> yes [22:53] <Zeeflo> i look at it every day [22:53] <JEEB> then why the fuck are you talking out of your ass [22:53] <Zeeflo> cause flv sucks [22:53] <Zeeflo> for high definition i would never use flv [22:53] <JEEB> I agree, flv sucks -- but still, look at your fucking rtmp streams [22:53] <JEEB> and no, SD/HD does not matter [22:53] <JEEB> flv is a CONTAINER [22:53] <Zeeflo> not when you have other containers that can be streamed with rtmp [22:53] <JEEB> not a format [22:54] <JEEB> oh for fuck's sake [22:54] <timofonic> JEEB: Opus audio or opus container? [22:54] <JEEB> GO LOOK AT WHAT DATA YOU GET FROM THE RTMP STREAM [22:54] <JEEB> I BEG OF YOU FAGGOT, TAKE A FUCKING LOOK [22:54] <Zeeflo> im not a faggot, and you need to calm down bro [22:54] <JEEB> I just don't know where to start making you understand [22:55] <Zeeflo> i do understand [22:55] <JEEB> what you push to an rtmp(e) service and what is coming out of the rtmp(e) stream do not equal [22:55] <Zeeflo> I just dont give a crap? [22:55] <JEEB> then why the hell are you talking as if you know your stuff [22:55] <Zeeflo> what comes out of the rtmp stream is flv [22:55] <timofonic> JEEB: So with not make a modern, network robust, low latency, suitable for bad connections, streaming/voip/webcam protocol? [22:56] <timofonic> JEEB: So say just one instead a dozen depending on the kind of media stream [22:56] <JEEB> Zeeflo, then it doesn't matter what the hell the source file is as long as the streaming server can process it into what is needed [22:56] <JEEB> your whole point goes trembling down [22:56] <Zeeflo> no [22:56] <Zeeflo> like i said, theres only a few containers that supports rtmp [22:56] <JEEB> I mean, if it's H.264/AAC it can be whatever you want container-wise as long as your thing that is serving an rtmp(e) stream can remux it into flv [22:57] <JEEB> .. [22:57] <JEEB> man, you are talking shit. Please stop that. [22:57] <Zeeflo> ok [22:57] <Zeeflo> bro [22:57] <mauricio> hi, I need some tips about how to stream rtp/rtsp at lowest latency [22:57] <JEEB> your rtmp(e) streaming provider is limited to some formats, which then get converted/remuxed/used as-is when streamed over rtmp(e) [22:57] <Zeeflo> you go make a video stream and audiostream in a MKV, put it on CloudFront and try serve it to e.g. JW Player [22:57] <Zeeflo> let me know what you find out [22:57] <JEEB> Zeeflo, read my last line [22:58] <Zeeflo> thats what Ive been saying the whole time! [22:58] <JEEB> no it hasn't been [22:58] <Zeeflo> Only a few containers are suited for RTMP [22:58] <JEEB> ... [22:58] <JEEB> that is one incorrect way of putting that [22:58] <Zeeflo> i dont know any CDN's that lets you stream AVI, MKV etc [22:58] <Zeeflo> do you? [22:59] <Zeeflo> please tell me [22:59] <Zeeflo> OH no! They gotta support the shittiest containers of them all! [22:59] <Zeeflo> webm.. [22:59] <Zeeflo> mp4 [22:59] <timofonic> JEEB: So why not do a proper streaming protocol instead? One that it's very robust and can be even suitable for sucky internet connections [22:59] <JEEB> everything that can use f.ex. ffmpeg or libav for the input can handle pretty much everything [22:59] <JEEB> I have no fucking idea about CDNs [22:59] <JEEB> and that's a whole fucking separate discussion [22:59] <Zeeflo> its like the whole world is run by freaking Apple! [22:59] <Zeeflo> fuck em! [22:59] <JEEB> blame your CDN for providing crappy service [23:00] <Zeeflo> I do! Every day [23:00] <JEEB> don't go out saying that the containers are the problem [23:00] <Zeeflo> well FLV is still a shit poor stream to output! [23:00] <Zeeflo> The problems arent really the containers.. Its the webbrowsers [23:00] <timofonic> JEEB: Well, I have tons of problems when playing streams from a bad connection. Tons of cache problems and such... [23:00] <JEEB> that's a separate discussion, although to be honest FLV is an "OK" simple container for H.264/AAC [23:01] <JEEB> it's a very simple container, and does what it's supposed to do [23:01] <timofonic> JEEB: But well, using mplayer/mplayer2/mpv solves partially the problem... [23:01] <JEEB> I don't like it, but its crappiness comes from something else [23:01] <JEEB> and yes, video playback in browsers suck dongs [23:01] <timofonic> JEEB: Adobe? [23:01] <JEEB> but that's a whole separate discussion [23:01] <Zeeflo> That actually reminds me [23:01] <timofonic> JEEB: But the streaming protocol might be something that needs to get improved, too [23:02] <JEEB> timofonic, I don't give a fuck, I was correcting Zeeflo's incorrect statements [23:02] <Zeeflo> Can anyone tell me a better way to stream to pc/mac/consoles other then web browsers? [23:02] <timofonic> JEEB: This HTML5 trend is bullshit, still we don't have proper streaming/voip [23:02] <timofonic> JEEB: ok... [23:02] <Zeeflo> would it be to have a player/software created that would link to our CDN? [23:02] <mauricio> rtsp isnt good enouh? [23:02] <timofonic> JEEB: You just want to have grumpy discussions :) [23:03] <Zeeflo> and have our customers download it to their pc's? [23:03] <Zeeflo> shut up timofonic [23:03] <Zeeflo> JEEB, [23:03] <timofonic> Zeeflo: No, thanks. Stop acting rude and Asperger-like [23:03] <timofonic> Zeeflo: You and JEEB should be more polite [23:03] <Zeeflo> what do you think would be the ideal way to stream RTMP to a clients desktop? [23:04] <timofonic> Seriously, IRC is full of morons [23:04] <Zeeflo> or a clients console (xbox, ps3 etc) [23:04] <JEEB> timofonic, sorry -- Zeeflo's general attitude of "I know even if I do not know or am saying it incorrectly" basically made me go xkcd 386 [23:05] <timofonic> JEEB: I agree [23:05] <JEEB> but your comments from the side that were unrelated kind of pissed me off [23:05] <Zeeflo> FFS shut up with the crying about rude irc dude! you pussies [23:05] <Zeeflo> im debating [23:05] <timofonic> JEEB: ok, I don't have enough level and got confused. Sorry [23:06] <timofonic> Zeeflo: No, you aren't. You are argumenting and requesting in a rude way, so you don't deserve a reply at all [23:06] <timofonic> Zeeflo: I don't cry, I ignore morons. Sorry, acting like Linus Torvalds isn't cool anymore. Specially is you are older than 15yo [23:06] <timofonic> is=if [23:06] <Zeeflo> jesus christ you are a faggot timofonic [23:06] <Zeeflo> go cry in some corner [23:06] <timofonic> Zeeflo: No, you are a moron ;) [23:07] <Zeeflo> "waah the bad internet dude yelled at me" [23:07] <Zeeflo> ... [23:07] <timofonic> Zeeflo: YOu are the reason geeky people are known as childish and stupid [23:07] <Zeeflo> yea sure [23:07] <timofonic> Zeeflo: No, I just point at you [23:07] <Zeeflo> youre right, now fuck off [23:07] <Zeeflo> dont loose your finger while youre at it.. [23:07] <timofonic> Zeeflo: Get a life, get emotional skills and stop acting like a teenage boy [23:08] <Zeeflo> oh my.. Dad is yelling at me! [23:08] <timofonic> Zeeflo: You aren't capable of cause danger at all, just annoying [23:08] Action: Zeeflo is so heartbroken now [23:08] <timofonic> Like a child [23:08] <timofonic> Like an antisocial nerd, that's all [23:08] <Zeeflo> yea, ill be 9 this month, and you still suck.. [23:08] <timofonic> Get a life [23:08] <Zeeflo> sure, Ill take a life, any life but yours. [23:09] <Zeeflo> cause it must suck really bad! [23:09] <Zeeflo> youre still arguing with me, what the f? [23:09] <Zeeflo> suck it up big boy! [23:09] <timofonic> Zeeflo: At least I know how to act properly and recognize my errors or stuff I'm ignorant about [23:09] <Zeeflo> get pas it [23:09] <Zeeflo> ... [23:09] <timofonic> Zeeflo: You can be temporarily funny, because you act in a very pathetic way ;) [23:10] <timofonic> Zeeflo: If I had OP, I would ban you [23:10] <Zeeflo> let me know when I should cry and beg for remorse ok? [23:10] <Zeeflo> of course you would. [23:10] <Zeeflo> people like you end discussions in such ways. [23:10] <timofonic> Zeeflo: You aren't capable of it, you must develop any kind of social ability first [23:10] <Zeeflo> i got plenty, just not enough for the internet. And especially whiners like you on it. [23:10] <timofonic> LOL [23:11] <Zeeflo> you whine like my 5 year old girl who cant get candy from the store,, [23:11] <Zeeflo> no.. [23:11] <Zeeflo> You whine worse.. [23:11] <timofonic> Zeeflo: I'm laughing at your childish attitude, but sad about it too :) [23:11] <Zeeflo> are you done any time soon? [23:11] <timofonic> Zeeflo: Poor children, I hope you mature very soon [23:11] <Zeeflo> gues not! [23:12] <Zeeflo> at first you pissed me off.. Now you just bore me.. [23:12] <timofonic> Zeeflo: At first you were funny, now you just bore me ;) [23:13] <Zeeflo> Is Bureks build supporting .sup? E.g. subtitles="blabla.sup" [23:17] <ubitux> .sup are not supported, if it's what i'm thinking about [23:18] <ubitux> also, since it's similar to vobsub, that wouldn't be supported by the filter, even if there was a demuxer (but it could be hardcoded differently) [23:18] <Zeeflo> whats up ubitux :) [23:18] <Zeeflo> thats what I figured.. [23:18] <Zeeflo> .sup is some kind of blueray "vob" format or something [23:18] <Zeeflo> bluray* [23:23] <Zeeflo> ubitux, i have some ideas in my head, wanna give your thoughts to em? [23:23] <Zeeflo> we stream alot of media, but to web browsers only. [23:24] <Zeeflo> Theres services out there like Netflix, Hulu etc etc. [23:24] <Zeeflo> They have desktop applications.. Would it be a better way to serve streaming media? [23:24] <Zeeflo> Have some software/player developed? [23:25] <ubitux> i don't understand [23:25] <Zeeflo> or would the result be the same as if a user just used his webbrowser [23:26] <Zeeflo> Like, if you got netflix. You can view their media on your webbrowser, what ever it might be. But they also got a desktop player / software of some sort.. [23:27] <Zeeflo> Would it be a better video experience for the end user to have some kind of desktop player instead of using a web browser? [23:27] <Zeeflo> Using RTMP [23:27] <Zeeflo> a stream is a stream.. And a player is a player, isnt it? [23:28] <mauricio> I want to know more about rtp/rtsp [23:30] <Zeeflo> http://en.wikipedia.org/wiki/Real-time_Transport_Protocol [23:32] <mauricio> ok, testing using ffmpeg outputing -f rtp rtp://.....:9696 [23:32] <mauricio> I cant see anything using ffplay [23:32] <Zeeflo> uhm [23:33] <Zeeflo> you need to define the transport layer, dont you? [23:33] <Zeeflo> tcp, udp etc [23:33] <Zeeflo> im not sure though. Im not to much into RTP [23:33] <mauricio> I believe its use tcp by default [23:34] <Zeeflo> what about port forwarding then? [23:34] <Zeeflo> RTP is IP to IP [23:34] <Zeeflo> so if youre behind NAT, you would probably have to redirect the port to a WAN IP [23:34] <Zeeflo> LAN IP [23:34] <Zeeflo> * [23:34] <mauricio> I'm testing inthe same computer [23:34] <Zeeflo> got no clue then [23:35] <Zeeflo> sorry [23:35] <mauricio> to stream rtsp do I need a server? [23:35] <mauricio> because rtsp fails [23:35] <mauricio> to stream [23:36] <Zeeflo> hmm [23:36] <Zeeflo> i dont know if you are able to stream RTP to and from the same server/pc/whatever [23:37] <mauricio> btw I'm trying to measure wich config will give me the lowest latency [00:00] --- Fri Feb 22 2013

1 0

ffmpeg-devel.log.20130220
by burek 21 Feb '13

21 Feb '13

[00:17] <wm4> is there a bug report yet for VDA failing with recent ffmpeg? [00:18] <wm4> I think I heard something about this, but I haven't found a ticket [00:28] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:c69315a5deb0: cook: check js_subband_start for validity [01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:1fd86f9a2136: vqavideo: fix return type [01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:8aedb7515674: tiff: check bppcount [01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:5af2fd317d84: wma: check byte_offset_bits [01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:17966ae5bb43: qdm2: increase noise_table size [01:28] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:0a57df38f468: Update for 0.8.14 [04:15] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:2b10d414648b: h264: add videodsp dependency. [04:15] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:f5c00b347dc7: avcodec_align_dimensions2: Ensure cinepak has large enough buffers. [04:15] <cone-841> ffmpeg.git 03u-bo1b(a)0w.se 07master:4e635e10e29f: cinepak: Use correct colorspace. [04:15] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a3adbedf9b63: cinepak: simplify [04:15] <cone-841> ffmpeg.git 03u-bo1b(a)0w.se 07master:8d1dd5bd5040: cinepak: More correct Cinepak decoder. [07:48] <highgod> Hi, I want to ask a question, I want to move the opencl init function to filter module. Can I call this function at avfilter_init_filter? because I this function will compile all opencl kernel functions and create a bin file, when ran ffmpeg next time, opencl will read the bin file and do not compile the kernel again [07:49] <highgod> it can save time [10:22] <durandal_1707> how i pad both inputs with -filter_complex ? [10:26] <durandal_1707> got it: they are named: [0:v], [1:v], ... [10:46] <durandal_1707> michaelni: calling setdar/setsar with no arguments causes segv [11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:f859678f3f27: vp56: Remove clear_blocks call, and clear alpha plane U/V DC only [11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:0c0828ecc565: x86: Use simple nop codes for <= sse (rather than <= mmx) [11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:e5ffffe48d20: h264chroma: Remove duplicate 9/10 bit functions [11:06] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:536d1ed7934d: Merge commit 'e5ffffe48d20642acc079166f0fa7d93a6a9f594' [11:21] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:2ed008204d54: h264: Add add_pixels4/8() to h264dsp, and remove add_pixels4 from dsputil [11:21] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:088f20a33251: Merge commit '2ed008204d5467be03a0a3af1e293b2f7038d0a0' [11:31] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:d709a18ac227: lavfi/hue: simplify parsing by making use of option shorthands [11:31] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:dd3732634528: lavfi/hue: reindent after previous commit [11:49] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:9918f57dcfc2: get_bits: Return pointer to buffer that is the result of the alignment [11:49] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:7ebfb466aec2: h264: Don't store intra pcm samples in h->mb [11:49] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:98f63e1f34b3: Merge commit '7ebfb466aec2c4628fcd42a72b29034efcaba4bc' [11:51] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:fae6fd5b8739: h264/svq3: Stop using draw_edges [11:51] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:8c6d9ca6387f: Merge commit 'fae6fd5b873911cfcd7f7b3d72de6598af5e003a' [11:58] <cone-768> ffmpeg.git 03Martin Storsjö 07master:26ee5ca03d76: h264: Get rid of unnecessary casts [11:58] <cone-768> ffmpeg.git 03Diego Biurrun 07master:8a11ce43d083: build: Ensure that output directories for header objects are created [11:58] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:39b03932c2ed: Merge commit '8a11ce43d08352f7a290355ebb5b29c495ad9609' [12:03] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:271f4112670f: lavf/matroskaenc: avoid assert failure in case of cuepoints with duplicated PTS [12:05] <cone-768> ffmpeg.git 03Diego Biurrun 07master:c9ad5f59aa84: intreadwrite: tomi: Add missing attributes.h #include for av_always_inline [12:05] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:f7b46d86dee8: Merge remote-tracking branch 'qatar/master' [12:36] <ubitux> any hungarian here? [12:39] <ubitux> mmh hungarian won't do actually [12:41] <ubitux> Compn: would you happen to have some russian subtitles? [12:41] <ubitux> or with a weird codepage/charset [12:41] <ubitux> (to add a fate test) [12:45] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:951d39fecd1f: avfilter: fix segfault with setsar/dar [12:55] <ubitux> btw, is there a "is codec id pcm" function helper somewhere? [12:56] <durandal_1707> nope [12:56] <ubitux> ok thx [13:08] <Compn> ubitux : hmmmmmmm, try opensubtitles.org for any big hollywood movie, then look for different languages and hope they arent uft ? [13:08] <Compn> but i'll look around [13:10] <Compn> utf* [13:11] <Compn> ubitux : http://samples.ffmpeg.org/sub/bg.sub [13:12] <Compn> thats probably hungarian or bulgarian [13:12] Action: Compn has no idea :) [13:12] <Compn> strange we didnt keep samples for different languages [13:14] <Compn> diff subcp that is [13:16] <ubitux> Compn: nice [13:16] <ubitux> any idea what's the charset? [13:16] <Compn> no [13:16] <Compn> subcp=cp1251 [13:16] <Compn> try that [13:17] <ubitux> i tried already [13:17] <ubitux> koi8-r and various other cp [13:17] <Compn> ah [13:19] <Compn> -subcp ISO-8859-1 ? [13:19] <Compn> i'm not actually sure what its supposed to look like :) [13:20] <Compn> oh, -subcp windows-1251 [13:20] <Compn> that looks much better [13:20] <ubitux> doesn't seem so, unless i'm doing something wrong [13:21] <Compn> you try in mplayer ? [13:21] <ubitux> no :) [13:21] <Compn> :) [13:21] <Compn> mr testing subtitles in a untested project [13:21] <JEEB> subs in various text encodings remind me of a guy telling me that his English subs wouldn't show up correctly and instead showed random symbols [13:21] <JEEB> I get a sample [13:22] <JEEB> and I get EUC-KR Korean subtitles :D [13:22] <Compn> lol [13:22] <ubitux> :D [13:22] <Compn> people think whatever subs that come with the movie are for their language [13:22] <Compn> but most new releases are only the language that the movie is in... [13:22] <Compn> korean movie, korean subs [13:36] <ubitux> mmh actually it worked [13:37] <ubitux> thx Compn [13:50] <Compn> ubitux : so subtitle filter converts it to ass ? [13:51] <ubitux> subtitles filter is for hardcoding [13:51] <ubitux> not sure to understand your question [13:52] <Compn> uhhh [13:52] <Compn> i guess i dont know what my question was indending to ask [13:52] <Compn> ehe [13:52] <ubitux> @_@ [13:52] <Compn> intending* [13:52] Action: Compn kinda distracted [13:54] <Compn> but thats cool. subtitle encoding support [13:54] <Compn> whats next on your list ? [13:54] <ubitux> world domination [13:55] <ubitux> there are some important changes to do to subtitles [13:56] <ubitux> notably fix the ass packets (which is far from being obvious) [13:56] <ubitux> the StyledSubtitles API (to avoid the internal representation in ASS) [13:56] <ubitux> subtitles in lavfi... [13:56] <ubitux> teletext [13:56] <ubitux> and other related crazy shit :) [14:01] <Compn> are vobsub in mkv supported ? [14:01] <Compn> i remember something like that wasnt [14:01] <nevcairiel> should work fine [14:01] <Compn> ok :) [14:12] <ubitux> <@Compn> are vobsub in mkv supported ? // you can mux and demux vobsub yes [14:13] <ubitux> (into mkv) [14:18] <ubitux> hey, it seems that in haskell "f $ g x $ h y $ a + b" is equivalent to "f (g x (h y (a + b)))" [14:18] <ubitux> who is up to add this awesome syntax to ffeval ? [14:20] <saste> ubitux, tools/ffeval -h | head -n [14:20] <saste> ubitux, tools/ffeval -h | head -n1 [14:21] <ubitux> saste: i mean, in eval() api [14:21] <ubitux> of course [14:21] <ubitux> so everyone can enjoy it [14:27] <Compn> ubitux : i thought i gave you a "windows-1251" subcp but i see "cp1251" in your fate test... [14:27] <Compn> arent they different? [14:27] <av500> 1251 is 1251 [14:27] <ubitux> Compn: i believe it's the same [14:28] <av500> 850 was the non-windows one [14:28] <Compn> oh , i guess so. [14:29] Action: Compn should probably take a break from computer now [15:50] <durandal_1707> saste: eval is slow [15:51] <saste> durandal_1707, one evaluation per frame? [15:51] <saste> it is slow in geq, when you evaluate an expression for each pixel [15:51] <durandal_1707> that usecase is fine [15:52] <saste> but one/few evaluations per frame are hardly measurable [15:52] <saste> (unless you do really fancy stuff in the expression of course) [16:04] <ubitux> saste: what do you think of making filter request window sizes? [16:04] <ubitux> (to auto-insert asetnbsamples filters) [16:06] <saste> ubitux, implies auto-reconfiguration [16:06] <saste> which is still missing [16:06] <ubitux> auto reconfig? how is that different that auto inserting scale filter etc? [16:07] <saste> ubitux, which use case do you have in mind? [16:07] <saste> i don't usually do things just because it is possible to do them [16:08] <ubitux> i need a fixed granularity of samples if ebur128 [16:08] <ubitux> in* [16:08] <ubitux> so i want ebur128 to receive fixed frames of 100ms [16:09] <ubitux> otherwise i would need to introduce a fifo in ebur128, add a request_frame, etc and basically reproduce the asetnbsamples code [16:09] <ubitux> which is a pain [16:11] <cone-137> ffmpeg.git 03Michael Niedermayer 07master:2ccaab9709aa: mov: fix self referencing timecode tracks [16:17] <ubitux> saste: so anyway, i was thinking of having such thing in the format negociation (maybe) [16:17] <ubitux> which might be semantically incorrect [16:18] <ubitux> the other solution might be to let the current filter responsible from auto inserting the filter itself [16:18] <ubitux> i don't think we do that already, that might be a bit tricky [16:18] <ubitux> and another one, call the filter internally [16:19] <saste> ubitux, what if the filter is not compiled? [16:19] <ubitux> error out at configuration? [16:19] <saste> or if the filter changes its syntax? [16:19] <ubitux> it's not a hard dep [16:19] <saste> another option would be to factorize the code at the C-api level [16:19] <ubitux> it's just needed/used in a particular context [16:19] <saste> but otoh should not be so hard to use audio queue internally [16:21] <ubitux> asetnbsamples filter code is not that obvious, duplicating its code might be pretty ugly [16:23] <saste> you could have a filter hook which auto-inserts a filter at the configuration stage [16:24] <ubitux> how would you do that? [16:26] <ubitux> <@saste> another option would be to factorize the code at the C-api level // basically export all the asetnbsamples symbols? :P [16:27] <saste> ubitux, no, i mean create an internal API which provides the correct level of abstraction [16:27] <saste> in case this is doable [16:28] <ubitux> btw, this may be very useful with show{spectrum,waves} [16:28] <ubitux> to avoid flickering [16:30] <saste> ubitux, showwaves should not need it, as it auto-adjusts itself already [16:30] <saste> but from what i see it, the solutions are two: C-level internal API, or per-filter hooking [16:31] <ubitux> can you detail the per-filter hooking? [16:31] <saste> the latter could address other use cases as well, even if I can't say at the moment [16:31] <saste> you define a callback in your filter, which is executed at some point during the configuration stage [16:31] <saste> this may do something such as inserting a filter or such [16:32] <ubitux> mmh sounds good, will check [16:32] <saste> for example, earwax may auto-insert an aresample [16:32] <saste> the problem in that case is that you don't have control over the many params of aresample [16:33] <saste> unless you pass them to earwax, which may be ugly (and imply the usual escaping problems) [16:33] <ubitux> what kind of aresample feature non handled by the format negociation earwax needs? [16:33] <saste> an audiophile (the kind of guy who may want to use earwax) may be picky about the resampling algo [16:33] <mateo`> i'm looking for some interlaced j2k/mxf samples, is there a place where i can find this kind of files ? [16:34] <durandal_1707> saste: you picked broken filter for your explanation [16:34] <saste> durandal_1707, then fix it! :-) [16:34] <ubitux> mateo`: maybe ask kierank or TimNich :p [16:35] <durandal_1707> saste: thats why there is ERROR for reson if sample rate is wrong [16:35] <ubitux> saste: mmh well yeah ok why not [16:36] Action: durandal_1707 gonna write another showaudio filter [16:36] <ubitux> NIH! [16:37] <durandal_1707> ubitux: are you saying if wrong frame size is given to ebu you get useless output? [16:38] <ubitux> durandal_1707: nope, but the metadata will if i add the feature [16:38] <ubitux> since you need to get information on very small slice of audio [16:39] <ubitux> OTOH, i can also just put the normalization inside ebur128 filter... [16:40] <ubitux> but loosing volume optimizations would be too bad [16:41] <durandal_1707> ubitux: play ape with insane profile with ebur128 [16:41] <durandal_1707> each frame have bunch of samples [16:42] <ubitux> we could also have a asetnbsamples=random :) [16:42] <ubitux> that would be nice to do some testing [16:44] <ubitux> an eval in the n field should do the trick [16:45] <durandal_1707> i added asetnsamples (there is no asetnbsamples filter, want to write one :-/ ) and it does not help [16:46] <ubitux> yes sorry asetnsamples [16:46] <ubitux> what do you mean it does not help? [16:46] <durandal_1707> i added asetnsamples=n=1024,ebur.... [16:47] <ubitux> wait, you actually have issues with ebur128? [16:48] <durandal_1707> yes [16:48] <saste> ubitux, also i never loved that name "asetnsamples", what about asetframesize? [16:48] <ubitux> saste: oh please, we had this discussion thousand times :D [16:48] <ubitux> durandal_1707: how to reproduce? [16:49] <saste> ubitux, i always forget [16:49] <saste> because to me it is a natural association: asetnsamples -> no good [16:49] <saste> will send a patch then soon or later [16:51] <Compn> that ebur128 sound sreally complicated [16:53] <durandal_1707> ubitux: http://samples.ffmpeg.org/A-codecs/lossless/luckynight.ape [16:56] <ubitux> durandal_1707: ffplay -f lavfi amovie=luckynight.ape looks broken [16:56] <ubitux> sounds like lavfi has some trouble with it [16:57] <durandal_1707> lol [16:57] <ubitux> it doesn't even play the same thing [16:57] <ubitux> wtf [16:57] <durandal_1707> http://en.wikipedia.org/wiki/Goniometer_%28audio%29 [16:59] <ubitux> anyway, doesn't look like a problem in ebu [17:00] <saste> anyone knows what's the problem with that file? [17:00] <durandal_1707> how does agoniometer sounds? [17:01] <ubitux> saste: the format [17:01] <saste> i suppose may depend on the fact that it contains *huge* packets [17:02] <saste> so may be a problem in movie or in the lavfi device [17:02] <saste> but i bet on movie [17:41] <durandal_1707> how this one actually works? By just ploting dots in 2dimensions when L is x-axis and R is y-axis? [18:24] <durandal_1707> ahh i found one that use jack audio system [18:33] <durandal_1707> you don't like my filters? [18:50] <wm4> so uh how do I specify the video decoder with ffmpeg [18:50] <wm4> it seems -codec:v sets the encoder? [18:51] <durandal_1707> you set it before -i ? [18:52] <wm4> no [18:58] <Compn> ubitux : that subtitle seeking -ss bug is kinda important :P [18:58] <ubitux> it's not a bug [18:58] <ubitux> it's a feature request [18:58] <ubitux> and not easy to solve [18:58] <Compn> i dont think its a tough fix, dont just pass the current timepos to the sub code and it goes from there ? [19:00] <ubitux> you can't pass it easily [19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:49c8fe304f69: doc/filters: apply misc fixes to drawbox documentation [19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:a2a1e20fa2c8: lavfi/decimate: use named options [19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:7f07c61c2f6d: doc/filters: rework introductory description for decimate [19:21] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:3d77a275487b: lavfi/overlay: implement shortest option [19:21] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:2db0056f9680: lavfi/overlay: apply minor consistency fixes [19:37] <saste> ubitux: libavfilter/vf_subtitles.c:313:14: warning: assignment discards const qualifier from pointer target type [enabled by default] [19:42] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:22b06103f019: lavfi/overlay: add format option [19:46] <ubitux> huh strange i didn't spot it [19:48] <cone-561> ffmpeg.git 03Clément BSsch 07master:9b9d996b34ae: lavfi/subtitles: fix recently introduced warning about discarded const qualifier. [19:49] <ubitux> saste: thx [20:05] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:039f55c80333: h264: call alloc_scratch_buffers() only once linesize is known [20:05] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:da7672b20a74: h264: Fix handling of changing reference counts between slices [21:04] <durandal_1707> ubitux: why you do not use table in swf? [21:05] <ubitux> i could [21:05] <ubitux> but the 16-bits bit flag is not part of the passed format [21:06] <ubitux> i could construct one and extrand the tab [21:06] <ubitux> not sure that's a better idea [21:06] <ubitux> afaict the s16 in the tab are supposed to be "pcm" by the specs [21:07] <ubitux> and the bit flag i'm using somehow describe it [21:08] <durandal_1707> ubitux: you could use ff_get_pcm_codec_id() [21:09] <ubitux> mmh interesting [21:09] <ubitux> maybe a bit overkill [21:11] <durandal_1707> ubitux: yea, not really important... [21:11] <durandal_1707> how you found flag? [21:12] Action: durandal_1707 wonders why this was not already done [21:12] <ubitux> web & specs [21:12] <ubitux> i'll re-send a patch soon though [21:16] <ubitux> durandal_1707: i wonder if it doesn't affect the adpcm [21:16] <ubitux> but i don't have any adpcm 8bit in swf samples [21:16] <ubitux> only pcm 8bit, and very few [21:19] <durandal_1707> ubitux: you cant encode one? [21:20] <ubitux> no idea how i'm supposed too [21:20] <ubitux> to* [21:21] <ubitux> sounds like i likely need some windows-specific black magic tools [21:21] <ubitux> i really don't like the dup of code in swfdec [21:22] <ubitux> not sure to understand it very well too.. [21:22] <durandal_1707> ubitux: adpcm not mentioned in spec? [21:24] <ubitux> the specs says (about the 16-bit flag): "This parameter only pertains to uncompressed formats" [21:25] <ubitux> do we consider adpcm uncompressed? [21:32] <durandal_1707> no [21:39] <ubitux> ok [21:39] <ubitux> i'll send a new patch set soon [22:06] <cone-561> ffmpeg.git 03Clément BSsch 07master:9a0076f50cfc: lavf/swfdec: factorize the creation of a new stream. [22:06] <cone-561> ffmpeg.git 03Clément BSsch 07master:624fb5f96544: lavf/swfdec: support 8-bits PCM audio. [22:44] <jojva> Hi, I'm a student of a french university, and I'd be happy to know about how far the need to implement the MVC decoder has been brought to. Is there any decoder ? A partly functional, or even a beginning of implementation of MVC ? Cheers. [22:47] <ubitux> D.VIL. mvc1 Silicon Graphics Motion Video Compressor 1 [22:47] <ubitux> D.VIL. mvc2 Silicon Graphics Motion Video Compressor 2 [22:47] <ubitux> ? [22:48] <ubitux> (Decode, Video, Intra-only, Lossy) [22:48] <jojva> Oh, sorry I should have been more precise. It's about H.264 / MVC (Multiview Video Codec), not Motion Video Compressor. [22:50] <jojva> I can see on this thread for example, that I'm not the only one hoping for this : http://ffmpeg.org/pipermail/ffmpeg-devel/2012-June/126139.html [22:56] <cone-561> ffmpeg.git 03Carl Eugen Hoyos 07master:c4dc6c4c86a0: Fix bits_per_coded_sample when encoding png with frame-level multithreading. [22:56] <burek> im not sure if this is a bug, so i dont know should i put it on the bug tracker or what.. [22:56] <burek> when using -acodec aac (without any other audio encoding options), and when the audio input is like this: Stream #0:1: Audio: pcm_u8 ([1][0][0][0] / 0x0001), 8000 Hz, mono, u8, 64 kb/s [22:57] <burek> ffmpeg i guess adds default -ab 128k [22:57] <burek> and final stream ends up like this: Stream #0:1: Audio: aac, 8000 Hz, mono, fltp, 128 kb/s [22:57] <burek> which causes: [aac @ 0360f720] Too many bits per frame requested [23:00] <burek> for more reference: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=825&p=1627 [23:09] <ubitux> is there a way, from a muxer, to get the packet pts exactly as it was stored originally in the demuxer? [23:10] <ubitux> it seems i'm able to get it back with something like av_rescale(ts, stream->time_base, (AVRational){1,25}) [23:10] <ubitux> but the 1/25 is a lucky guess [23:11] <nevcairiel> shouldnt the stream time base match the timebase used for storage in the source format? [23:12] <nevcairiel> depends on the demuxer, i guess [23:12] <ubitux> where is the "timebase used for storage" stored? [23:13] <ubitux> (did i misunderstood what you said?) [23:13] <nevcairiel> for some files, its always the same, say mpegts is always 1/90000, mkv has it in its headers somewhere, and in both cases its put in the avstream timebase and the file timestamp just read and not changed at all [23:14] <nevcairiel> i'm sure there are exceptions that do change the timestamps [23:37] <burek> geez, forum really got big.. [23:37] <burek> a lot of people is using it now [23:38] <burek> i wasnt there for a couple of days and now i need to spend couple of hours to reply to all the questions :/ [23:38] <ubitux> (ok get it, i needed the tb from the stream->codec) [23:39] <ubitux> burek: it will only get worse! [23:39] <burek> that's the (murphy's) spirit :) [23:41] <wm4> so, you don't have a machine where you'd be able to run a VDA FATE test or so? [23:41] <wm4> same for other hwaccel stuff [23:41] <wm4> though these would be harder [23:41] <wm4> as the vda pseudo decoder works like a normal decoder [23:52] <burek> what is ffmbc? [23:53] <ubitux> a gpl fork of ffmpeg for broadcasting [23:53] <burek> something like embedded ffserver or worse? :) [23:53] <ubitux> just a ffmpeg specialized for broadcasters needs [23:54] <burek> ok, thx :) [00:00] --- Thu Feb 21 2013

1 0

ffmpeg.log.20130220
by burek 21 Feb '13

21 Feb '13

[00:15] <ezekiel> saste: v4l2 capture with out-of-sync audio to MP4: http://ezekiel.endpoint.com/vid/sync-test-video1.txt and resulting video: http://ezekiel.endpoint.com/vid/sync-test-video1.mp4 [00:16] <ezekiel> I understand if you don't personally have time to troll through my logs [00:17] <ezekiel> seems like I _may_ be a pretty good test case, though. Willing to rebuild and test. [00:17] <saste> ezekiel, check that audio timestamps start with 0 as well [00:17] <saste> video timestamps at least should be good [00:17] <saste> -af ashowinfo [00:19] <ezekiel> is it fairly significant that a 30fps capture would show: n:30 pts:1000068 pts_time:1.00007 [00:24] <ezekiel> wow - now that I'm looking at the ashowinfo output, I can see that the above is not nearly as significant as I might've imagined [00:29] <saste> ezekiel, audio ts start from 0? [00:30] <saste> also ffmpeg has a host of tools for dealing with timestamps [00:30] <saste> check for example resample, setpts, asetpts filters just to name a few [00:31] <saste> and the v4l2 ts option [00:36] <stephanedev> i wrote some code to stream a video encoded with vp8 over rtp using ffmpeg C API and i play it using ffplay. when packets are dropped, many messages related to missed RTP packets are written to the console [00:36] <stephanedev> i noticed those 2 in particular: [00:36] <stephanedev> [vp8 @ 00a6e960] RTP: missed 1 packets [00:36] <stephanedev> [NULL @ 00a6e960] RTP: missed 1 packets [00:37] <stephanedev> i thought the first part (e.g. vp8) was extracted from AVClass and the part after '@' was an address [00:37] <stephanedev> am i correct? [00:46] <stephanedev> if my understanding is correct, what could explain the fact that the information extracted from the AVClass ("vp8") turn into "NULL"? [00:47] <stephanedev> turns* [00:47] <stephanedev> as obviously the address is the same in the 2 log messages above [01:49] <defaultro> saste [01:53] <ezekiel> still investigating, comparing the first 12 ashowinfo of out-of-sync git master: http://pastebin.com/ibCGHt76 with in-sync v1.0.4: http://pastebin.com/ymxh8jxK [02:05] <ezekiel> recording audio and video using separate ffmpeg processes, then combining the videos, still totally out of sync [02:05] <ezekiel> maybe an alsa issue, not a v4l2 issue [02:08] <ezekiel> ffprobe seems to pretty regularly show "start: 0.032000" - which I know is mere tenths of a second... [02:24] <stephanedev> is pthreads support required to stream over rtp? [02:24] <stephanedev> i am asking because of this message when running ffmpeg on windows: [02:24] <stephanedev> [udp @ 00645b80] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required) [02:25] <stephanedev> using the zeranoe build [06:51] <jagginess> hi [06:51] <jagginess> there good guidelines for merging timing information from a video file to an audio one? [13:24] <kvz> if i set bitrate, is that per channel, or do channels share that bitrate? [13:24] <durandal_1707> share [13:24] <kvz> durandal_1707: thank you! [13:49] <robertbuhren> Hi there, i'm using a product which uses fmmpeg to transcode video files to a lower resolution/codec/container. I'm having the problem that audio and video is out-of sync, could this be anyhow related to running ffmpeg inside a virtual machine (qemu-kvm) I know the information is very vague, but maybe someone here has a hint for me [13:50] <robertbuhren> the reason for asking is that the fact that it's run from a virtualized evinvironment is the only thing which differs from other setups that work well [14:18] <creep> how do you like this? :) http://images.puszta.com/hazi_kifli/Hazi%20kifli%201000.jpg [16:03] <Ziarkaen> I am running an ffmpeg screen capture command in a Makefile, for conveinience. I can run the command separately with no problem, but have to use Ctrl-C to stop recording. When this is put in the Makefile, I believe make captures the ctrl-C and exits with error. Is there a way to pass Ctrl-C directly into FFMPEG, or another way to stop recording? [16:22] <Ziarkaen> I have solved the problem by spawning a graphical terminal running the command, although this isn;t too nice. [16:26] <sacarasc> What do you need the capture for? [16:50] <teratorn> Ziarkaen: Make does not intercept SIGINT [16:50] <teratorn> Ziarkaen: the process exits with non-zero on SIGINT, so make fails at that point [16:51] <teratorn> Ziarkaen: you can add "| true" at the end of the command in your Makefile, probably [16:51] <teratorn> err, || true [17:03] <n1n0> which is the command line, first to set a certain video to be used , then to set ffmpeg to extract frames at the original frame rate? [17:36] <pedro_sland> should it make a difference if i switch the output order when i'm doing output to 2 files? eg ffmpeg -i file -filter:v ... out1.mp4 -filter:v ... out2.webm [17:39] <relaxed> pedro_sland: using -map sure wouldn't hurt [17:41] <relaxed> also -filter:v:$stream_id [17:42] <pedro_sland> I appologise for not being more explicit. here is what i'm using http://pastebin.com/zFRtCeqE [17:43] Action: pedro_sland try again [17:44] <pedro_sland> i'll go to the beginning :) i am trying to convert one video into 2 suitable for the web. i'm happy with just webm and mp4. [17:45] <pedro_sland> sure, i was going to give the story first? [17:45] <relaxed> durandal_1707 loves stories [17:46] <pedro_sland> i thought some context might help. @relaxed guessing that was sarcasm [17:49] <pedro_sland> i checked my command http://pastebin.com/ThZ5WBk2 [17:50] <relaxed> not really, I just like messing with him. [17:50] <relaxed> (or her?) [17:51] <pedro_sland> k. my problem is that the webm freezes for a second at the beginning (at about 1s in). [17:51] <pedro_sland> i started cutting down the parameters [17:52] <pedro_sland> eventually i tried switching round the output files (and their associated commands) and the problem seems to be fixed [17:53] <pedro_sland> The difference in output is this "[Parsed_overlay_1 @ 0x2e99260] Buffer queue overflow, dropping." line only shows when it jumps (when webm is last) [17:53] <relaxed> pedro_sland: try, ffmpeg -i input -map 0 (first output options) $first_output -map 0 (second output options) $second_output [17:55] <relaxed> oh yeah, remove 99% of those options and use "-t 5" for both outputs to get a 5 second sample. [17:56] <relaxed> until you iron this out [17:56] <pedro_sland> yep, that's what i was doing hence the first paste. [17:56] <pedro_sland> it was fine without the filter but the filter broke it [18:02] <durandal_1707> does it happen if you write only webm file? [18:03] <pedro_sland> i think i tested that but i'll test again just now [18:03] <pedro_sland> http://pastebin.com/GQYwvpBA [18:03] <relaxed> pedro_sland: did you check out ffmpeg from an old repo? Your version is much older than mine: ffmpeg version N-49946-ga08ebf0 [18:03] <pedro_sland> oh [18:03] <pedro_sland> hmm, i'll check [18:03] <relaxed> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg [18:04] <relaxed> they switched at some point [18:04] <relaxed> once you update, look at -filter_complex in the man page. [18:06] <pedro_sland> i've moved that repo and downloaded another. compiling now zzzz [18:08] Action: relaxed sticks a beer under burek's nose [18:09] <durandal_1707> relaxed: how you found out that pedro_sland is using old version? [18:10] <relaxed> his pastebin, http://pastebin.com/ThZ5WBk2 , shows "ffmpeg version N-38907-g39b0393" [18:10] <pedro_sland> i downloaded a new one anyway but looking at it i don't think it is old. last commit i have is 9d9420d2 at Wed Feb 20 11:54:06 2013 +0100 [18:11] <relaxed> 11,000 commits behind [18:11] <pedro_sland> :O [18:12] <pedro_sland> something fishy here. .git/config says [remote "origin"] url = git://source.ffmpeg.org/ffmpeg [18:12] <pedro_sland> nvm almost done compiling [18:14] <durandal_1707> relaxed: how you get that number? [18:14] <pedro_sland> latest version should say "ffmpeg version git-2013-02-20-2ccaab9" right? [18:15] <durandal_1707> yes [18:15] <pedro_sland> :) [18:15] <durandal_1707> and previous build you used was just few commits older [18:17] <pedro_sland> that's what i thought i was using - a build from earlier today but nvm [18:17] <relaxed> durandal_1707: hmm, I thought the number in N-38907-g39b0393 was the number of commits. Am I mistaken? [18:18] <durandal_1707> relaxed: yes you are [18:20] <pedro_sland> i ran the broken command again and same behaviour http://pastebin.com/UqKXRBgv. checking with just webm [18:21] <pedro_sland> that worked fine http://pastebin.com/ya30Zgvk [18:22] <relaxed> durandal_1707: then what is it? [18:25] <durandal_1707> relaxed: last commit hash [18:25] <durandal_1707> (partial one) [18:25] <relaxed> isn't this the commit hash --> g39b0393 [18:29] <durandal_1707> relaxed: without g [18:37] <pedro_sland> relaxed: thanks for the -filter_complex hint. i imagine that this way only loads the overlay once and "movie=" loads it 2x? other than looking nicer on the command line, are there more benefits of this way? [18:41] <relaxed> I thought it was required with 2 or more inputs, but I guess the old way still works for now. [18:42] <pedro_sland> well, i just tested, this way works properly unlike the other way :) [18:42] <pedro_sland> ty much for your help. i really appreciate it. [18:43] <relaxed> unless you read the man page from time to time they'll change things right under your nose. [18:43] <pedro_sland> :) [18:43] <relaxed> you're welcome [18:45] <pedro_sland> if they're going to remove that other way, does that mean that we won't be able to have different filters on different streams? not that i want that but just curious. [18:56] <relaxed> pedro_sland: no, -filter_complex is just a better implementation for use with multiple inputs. [18:57] <durandal_1707> s/better/red [18:58] <relaxed> red? [18:59] <pedro_sland> i don't get it either, unless he/she's playing with you [19:00] <relaxed> I think she meant "rad" [19:00] <relaxed> which it totally is [19:00] <pedro_sland> totally. i love it already [19:03] <pedro_sland> thanks again and bye all [19:45] <Emmanuel_Chanel> hi [19:45] <Emmanuel_Chanel> Someone is alive? [19:46] <Emmanuel_Chanel> I want to encode Columbo: Friend in Deed aired on BSP. But the audio is one track with ual monaural. [19:47] <Emmanuel_Chanel> It has only 1 Stereo sound track that has Japanese "left" and English "right". How can I split the track into two, Japanese and English? [19:55] <relaxed> Emmanuel_Chanel: look at -map_channel in the man page [19:57] <relaxed> Emmanuel_Chanel: there are example commands [20:06] <someperson> anyone happen to know anything about video editing ? particularly automated skin softening/retouching ? just wondering what tools i need to look at [20:07] <Emmanuel_Chanel> Yeah, I did. But I couldn't get a good result. Anyway, ok, I try. Thanks. [20:09] <durandal_1707> Emmanuel_Chanel: use filters [20:12] <durandal_1707> Emmanuel_Chanel: see documentation for channelsplit [20:19] <Aaronds> Hi, I'm trying to convert a video to .mp4 using ffmpeg CLI. However, at the moment my video plays normally in Firefox but in Chrome I only get audio, no video. I'm using -pix_fmt yuv420p as I've seen on the net that this is needed for chrome, but otherwise I have no idea what is causing the problem. Can anyone help? [20:19] <Aaronds> I'll paste command/output shortly [20:23] <durandal_1707> Aaronds: pasted anything? [20:23] <Aaronds> Command & output for the aforementioned question: http://pastebin.com/fE51jgVX [20:24] <Aaronds> IRC bugged out on me briefly [20:25] <durandal_1707> does chrome plays aac? [20:25] <Aaronds> durandal_1707, sorry aac? [20:25] <JEEB> yeah [20:25] <JEEB> H.264 + AAC [20:25] <JEEB> in mp4 [20:26] <Aaronds> I don't know... AFAIK others have done the same conversion and had everything worked, the only problem I picked up on was the yuv240p thing. [20:26] <Aaronds> obviously H.264 is fine [20:29] <durandal_1707> maybe it support only certain samplerates? [20:30] <durandal_1707> it is little strange because AFAIK chrome use libavcodec [20:30] <JEEB> but he is getting the audio :D [20:30] <Aaronds> durandal_1707, are you able to provide a set of parameters which you'd expect to work? [20:31] <JEEB> that should in all theory work [20:31] <Aaronds> >_< [20:31] <JEEB> are you testing on chromium or chrome_ [20:31] <JEEB> ? [20:31] <Aaronds> chrome [20:31] <JEEB> weird [20:33] <Emmanuel_Chanel> durandal_1707: How shall I fix this command for that? ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4 [20:36] <Aaronds> Oh, the issue is with chrome... Going to try and have to use a different file format for now then -.- [20:39] <durandal_1707> Emmanuel_Chanel: try to add "-af channelsplit" before -vf [20:40] <Emmanuel_Chanel> ok. [20:42] <Emmanuel_Chanel> I got "Cannot map stream #0:2 - unsupported type." with "ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -af channelsplit -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4" [20:44] <durandal_1707> Emmanuel_Chanel: replace "-af" with "-filter_complex" [20:48] <Emmanuel_Chanel> ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -filter_complex channelsplit -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4 [20:48] <Emmanuel_Chanel> says: Cannot map stream #0:2 - unsupported type. [20:48] <Emmanuel_Chanel> ... [20:51] <Emmanuel_Chanel> ok. [21:10] <Emmanuel_Chanel> durandal_1707: thx. It seems that I can encode TS with Dual Monaural now. [21:12] <durandal_1707> Emmanuel_Chanel: how? what happened? [21:12] <Emmanuel_Chanel> It seems that -filter_complex channelsplit helps. [21:15] <Emmanuel_Chanel> durandal_1707: Where on the document can I see channel_split , the option? [21:20] <durandal_1707> Emmanuel_Chanel: http://ffmpeg.org/ffmpeg-filters.html [21:24] <Emmanuel_Chanel> thx. [22:53] <burek> beer! :D \o/ [22:53] Action: burek bows to relaxed :) [23:00] <burek> wieshka, trouble in paradise? :) [23:02] <burek> does anyone use ffplay extensively enough to answer this q: http://ffmpeg.gusari.org/viewtopic.php?f=26&t=826 [23:02] <ubitux> burek: hey long time no see. [23:02] <burek> it's related to controlling ffplay programatically [23:02] <burek> ubitux hello :) [23:02] <burek> it was a busy holiday week :))) [23:02] <Mavrik> holiday week? O.o [23:03] <burek> couple of holidays merged together in a whole week :D [23:03] <ubitux> burek: ffplay has no "slave" mode like MPlayer has [23:03] <ubitux> burek: redirect him to MPlayer/mplayer2/mpv [23:03] <burek> ok thanks [23:04] <burek> is there any interesting news around here :) [23:05] <beastd> ubitux: didn't mpv drop slave-mode? or has it something different now? [23:05] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=Changelog;hb=HEAD ? [23:05] <ubitux> beastd: mmh good question [23:06] <ubitux> beastd: yeah right seems they dropped it [23:06] <ubitux> so MPlayer/mplayer2 then [23:30] <burek> halp :) again :) [23:30] <burek> "How can i change the "scan type" into interlaced? How can i change the "scan order" (Bottom Field First - Top Field First) fot interlaced encoding?" [00:00] --- Thu Feb 21 2013

1 0

ffmpeg-devel.log.20130219
by burek 20 Feb '13

20 Feb '13

[00:29] <wm4> there are no FATE tests for seeking, right? [00:43] <ubitux> wm4: fate/seek.mak? [00:48] <wm4> ok [00:51] <wm4> that doesn't seem to test decoding [00:51] <wm4> i.e. whether the packets are valid? [00:52] <wm4> or maybe the packet size is always good enough for this purpose? [01:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:c6f59b95c529: h264: avoid calling get_format() multiple times [01:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:56b6909b395e: movenc: hotfix, dont store fiel for h264 / mpeg4-asp / dnxhd [01:14] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:7acfa7758c7f: configure: Make warnings from -Wreturn-type fatal errors [01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:1ca25bc38775: libopencore-amrnb: cosmetics: Group all encoder-related code together [01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:e492818d8925: libopencore-amr: Conditionally compile decoder and encoder bits [01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:a23d6ea1e443: libopencore-amrwb: Make AMR-WB ifdeffery more precise [01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:6c62098827d3: build: The libopencore-amrnb encoder depends on audio_frame_queue [01:25] <cone-967> ffmpeg.git 03Luca Barbato 07release/1.1:b9a287f23700: build: make audio_frame_queue a stand-alone component [01:25] <cone-967> ffmpeg.git 03Matti Hamalainen 07release/1.1:d61c6ebccfde: svq3: unbreak decoding [01:25] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:0f5a0a415510: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [01:35] <cone-967> ffmpeg.git 03Hendrik Leppkes 07release/1.1:e14564b9262d: lavfi/kerndeint: use av_pix_fmt_desc_get instead of directly accessing the table [01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:f5955d9f6f9f: targa: Fix y check in advance_line [01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:1ea5bbc5940d: sanm: add forgotten check for decoded_size in old_codec37() [01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:1400f1a1e46d: sanm: Use the correct height variable in the decoded_size checks [01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:f6687bbb6464: pngdec/filter: dont access out of array elements at the end [01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:75211f2b8cfb: tiff: Check buffer allocation and pointer increment more carefully in shorts2str() and double2str() [01:36] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:811a504c6bc2: shorten: dont leave invalid channel counts in the context. [01:36] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:7c40a0449b47: swr: check channel layouts before using them. [01:50] <BBB> michaelni: ok, fixed for that one crash at least [01:50] <BBB> michaelni: it's hard for me to run complete fate, but if you find more crashers, let me know and I'll fix them too [01:50] <BBB> michaelni: seems to work for me now [03:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:a3a97129e79d: mpegvideo: make ff_print_debug_info independant of mpegvideo [03:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:d41efc1f267c: h264: put visualization and debug support back [03:41] <BBB> michaelni: where do I find svq3_watermark.mov? [03:51] <Compn> should be listed in allsamples.txt [03:51] <Compn> http://samples.ffmpeg.org/mov/watermark/svq3_watermark.mov [03:51] <Compn> BBB : ^^ [03:53] <BBB> ty [03:53] <BBB> I don't know how allsamples.txt works [03:56] <michaelni> http://samples.ffmpeg.org/allsamples.txt should list all samples so one can easily grep in it [03:59] <Compn> ehe [03:59] <Compn> BBB : you dont have a copy of the samples repo on your hd ? [03:59] <Compn> michaelni : how many tests do you run on the samples? i mean, how did you find out svq3_watermark.mov had a problem? :) [04:00] <BBB> Compn: how much HD space do you think I have? [04:00] <BBB> I have a little laptop that is easily portable at home, that's all [04:00] <Compn> BBB : last i checked it was only ~60gb [04:00] <Compn> 64G unified/ [04:01] <BBB> my hd generally is full of pictures, music and stuff [04:01] <BBB> you know, I have kids and stuff, I have ... baggage, as they'd say [04:01] <Compn> no problem :) [04:01] <Compn> ssd arent there yet with the size [04:01] <BBB> I'll just download them as needed [04:05] <BBB> hm right the 8x8 only clears one half of the block data [04:05] <BBB> let me try to fix that [04:07] <BBB> ok, that's fixed now, now idct8_add4 seems broken also [04:07] <BBB> blegh [04:11] <BBB> ok h264 fixed [04:11] <BBB> will test svq3 in a bit, may well be fixed by the same thing [04:12] <BBB> yeah looks mostly the same to me now [04:12] <BBB> michaelni: care to test again? [04:22] <BBB> oh nm the crcs are different [04:22] <BBB> lemmecheck [05:45] <highgod> Hi, I want to ask a question. When will the next ffmpeg release version be published? where can I get the time schedule about ffmpeg release version be published? thanks [05:50] <Compn> no idea :) [05:51] <Compn> maybe michaelni knows release schedule [05:51] <Compn> usually after a bunch of bugs are fixed, and new bugs are merged in [05:59] <highgod> @Compn:thanks, does ffmpeg will publish a version on march? I have read the comments, and I have some quesions, and I have sent the email. [06:04] <BBB> michaelni: nice bug you detected there in svq3_watermark.mov [06:04] <BBB> congrats on that [06:27] <highgod> michaelni:do you know when the next ffmpeg release version be published? [06:27] <highgod> Thanks [10:10] <ubitux> highgod: please fix your mailer and reply below each comment [10:10] <ubitux> currently you're duplicating the whole mail, without quoting, so the previous mail is mixed with your answer [10:10] <ubitux> it's unreadable [10:52] <durandal_1707> ubitux: should i sent new blend patch or just commit? [10:53] <ubitux> saste is more familiar than me with the multiple input thing [10:53] <ubitux> you may want to ask him before [10:53] <ubitux> or maybe Nicolas [12:05] <michaelni> highgod, a new major release every 3 month (more or less) minor releases randomly [12:16] <cone-841> ffmpeg.git 03Paul B Mahol 07master:f2d200d460ab: lavfi/overlay: yuv444p & yuva444p support [12:16] <cone-841> ffmpeg.git 03Paul B Mahol 07master:03b711d95e28: tmv: initialize unused pallete entries with 0 [12:21] <highgod> Thanks [12:21] <highgod> @ubitux:I got it. thanks [12:24] <cone-841> ffmpeg.git 03Vicente Jimenez Aguilar 07master:202b5f6deb65: doc: Fix some obsolete references to av* tools as ff* tools [12:24] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:4e469842b525: Merge commit '202b5f6deb65e405b07b9b5c20f97c8cb925cf49' [12:36] <cone-841> ffmpeg.git 03Daniel Kang 07master:7a03145ed7cb: x86: dsputil: int --> ptrdiff_t for ff_put_pixels16_mmxext line_size param [12:36] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:cf10616cc0db: Merge commit '7a03145ed7cb4f1ce794b5126559dd6f38029243' [12:41] <highgod> @ubitux:should I just write the replies under the comments of the orignal mail? [12:54] <mateo`> hello, i'm trying to fix the frame duplication issue i have with interlaced jpeg2000 patch. Since the decoder output a single frame for 2 video packets, is there a proper way to correct the pts/dts of the decoded frame so that ffmpeg won't duplicate frames ? (-vsync 0 does the trick) [12:55] <michaelni> mateo`, sounds like something sets wrong timestamps [12:55] <michaelni> did you look at the demuxer timestamps ? and the framerate choosen ? [12:56] <mateo`> yes pkt_pts and pkt_dts seems ok within the decoder [12:58] <mateo`> i tried to devide by 2 the pkt_pts and pkt_dts within the decoder when it outputs a full frame but it does nothing [12:58] <mateo`> (maybe i'm not supossed to do that in the decoder) [12:59] <cone-841> ffmpeg.git 03Daniel Kang 07master:9acd23d655b5: x86: dsputil: Fix h263 loop filter link error in some configurations [12:59] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:fa09ad5c9e68: Merge remote-tracking branch 'qatar/master' [12:59] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:b9237aa7b090: x86/h263_loopfilter: Fix author attribution after code has been moved/splited around [15:12] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:b97d61f924ee: avfilter/ff_merge_formats: only merge if doing so does not loose chroma or alpha [15:16] <Compn> this bug looks invalid, AWS problem not ours ... https://ffmpeg.org/trac/ffmpeg/ticket/1486 [15:26] <Compn> yay i closed a bug :) [15:31] <durandal_1707> nooooooooooooooooooooooooooooooooooooo [16:02] <ubitux> durandal_1707 ? [16:21] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:a1f1ca96b426: svq3: fix decoding residual blocks of b-frames. [16:43] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:1acd7d594c15: h264: integrate clear_blocks calls with IDCT. [16:50] <Compn> ubitux : i think he was making a joke about me closing a bug [16:51] <ubitux> oh. my bad. [17:02] <wm4> why does linking to libav* add -SDLmain? [17:03] <durandal_1707> because of sdl in lavd? [17:03] <wm4> but why -SDLmain? [17:03] <wm4> this is completely retarded [17:04] <wm4> I wonder why you'd ever need -lSDLmain at all [17:04] <wm4> it's a crap hack [17:04] <wm4> I almost feel like adding a horrible hack to configure to filter it out [17:04] <durandal_1707> i don't see it anywhere [17:05] <durandal_1707> perhaps that is what pkgconfig returns [17:05] <wm4> yes, it is [17:05] <wm4> hm maybe it's recursively added from the SDL .pc [17:05] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:633f9974790e: bmp: check available space when reading palette [17:06] <wm4> indeed it is [17:06] <wm4> got to report this as SDL bug [17:06] <wm4> because... seriously. [17:08] <durandal_1707> i don't have that here with old SDL (one from FreeBSD ports though) [17:11] <wm4> I get it with 1.2.15 [17:11] <wm4> but well, it's a windows-only thing [17:11] <wm4> so it won't happen with a unix native libsdl [17:11] <wm4> just mingw packages [17:12] <durandal_1707> same version here [17:12] <durandal_1707> how would you describe lavfi in few words? [17:13] <wm4> "new ways of hiding semi-useful functionality under awkward APIs" [17:15] <durandal_1707> that is too much words, i need 5 max [17:21] <Compn> lol [17:22] <Compn> durandal_1707 : powerful filters with confusing syntax [17:22] <Compn> :D [17:22] <Compn> sorry, i am too honest :) [17:22] <Compn> i love the filters :) [17:22] <Compn> i dont want to spend 20 minutes to figure how to write the right command line tho [17:23] <Compn> the examples are good, i want more of them. [17:23] <wm4> even Compn (mplayer and mencoder veteran) thinks the syntax is confusing [17:24] <Compn> by confusing i mean, most people only know 8th grade math (and probably lower than that) so creating equations is difficult [17:25] <Compn> the filters are complex, so they need to have the ability to provide details like that. but most users wont be able to use them as-is [17:25] <Compn> without some kind of frontend, script, examples, etc [17:26] <Compn> i never understood custom matrices in mpeg encoding either... [17:26] Action: Compn not the smartest [17:28] <wm4> it just means it's not userfriendly [17:31] <Compn> j-b : is there someone working on the g2m decoder? [17:31] <Compn> j-b : kostya updated his blog http://codecs.multimedia.cx [17:45] <durandal_1707> what about those cinepak patches? [17:51] <divVerent> Compn: what annoys me most about the filters is how the tags work [17:51] <divVerent> as long as your filters are a single chain, you can use comma to avoid most tag use [17:52] <divVerent> but it is quite annoying that to insert a filter into the middle of e.g. [in] foo [mid]; [mid] [in2] bar [out] [17:52] <divVerent> you suddenly have to RENAME a tag [17:53] <divVerent> and e.g. turn it into [in] foo [new]; [new] added [mid]; [mid] [in2] bar [out] [17:53] <divVerent> this is especially bad if you want to programmatically edit filter chains [17:53] <divVerent> of course, in this case, comma syntax helps too ([in foo, added [mid]; [mid] [in2] bar [out] [17:53] <divVerent> ) [17:53] <divVerent> but that can't be used in the general case [17:54] <divVerent> so to add a filter before [tag], you need to rename existing OUTPUT uses of [tag] to [newtag], then add [newtag] addedfilter [tag]... the "rename" part here is annoying [17:56] <divVerent> however, having said that this syntax sucks... I have no constructive idea about doing it better :( [17:56] <ubitux> it looks like you are complaining in a situation where you have a variable foo and when wanting to add a similar variable you need to rename it to foo0 and add foo1 [17:56] <ubitux> :)) [17:56] <durandal_1707> using single line for just this is wrong [17:56] <divVerent> ubitux: in C this issue is simple to solve :P [17:57] <divVerent> just add functioncall(...) around the expression [17:57] <divVerent> but C syntax doesn't help us for filters, as a function can only have one return value there [17:57] <divVerent> I wonder if a Haskell-style "let" would actually help :P [17:58] <durandal_1707> script reader [17:59] <wm4> and then there's this idea to use Lua to build the filtergraph [17:59] <divVerent> https://gist.github.com/4987734 [17:59] <divVerent> actually, the concept of Haskell's "let" would help [17:59] <divVerent> wm4: does Lua fix the issue, is the question [18:00] <divVerent> basically, the idea of this "let" use is to have a way to insert filters at a point by JUST inserting text [18:00] <durandal_1707> wm4: using single line to describe filtergraph is silly [18:00] <wm4> divVerent: it seems vsynth is doing fine with using Python for building graphs [18:00] <divVerent> durandal_1707: that is mostly the issue [18:00] <divVerent> wm4: wonder if it has the same issue, for dynamically generated graphs [18:00] <divVerent> or mkanually edited graphs [18:01] <durandal_1707> where are devs when you need them [18:02] <divVerent> haha, C/Perl-like syntax WOULD help a great deal already [18:02] <divVerent> https://gist.github.com/4987753 [18:02] <BBB-work> michaelni, opinions on https://github.com/rbultje/ffmpeg/commit/be3d3b499bd241b7d6f7d9b4b6d91cb2dc… ? [18:02] <divVerent> i.e. clear distinction between input and output args [18:02] <divVerent> in C done by &, in perl done by creating the variables using "my", ... [18:02] <divVerent> this way, the filter insertion can be done by pure text operations [18:02] <divVerent> which is basically my goal [18:02] <wm4> well, Lua and python can return tuples [18:03] <divVerent> right, but we still need a way to easily programmatically build a filter graph [18:03] <divVerent> e.g. to add the subtitle renderer only if subtitles are present [18:03] <Compn> durandal_1707 : well the patch author just said the code for rejecting strips could be moved to detecting incomplete blocks... i think someone said we shouldnt make a regression by not playing some files due to his patch [18:03] <Compn> durandal_1707 : so it would be nice if his patch doesnt affect the playing of (possibly broken) files [18:03] <saste> divVerent, lua scripting will be possibly a gsoc task i'd like to mentor [18:04] <Compn> will the patch author do that or will someone else have to is the question [18:04] <saste> otherwise i'll tackle it myself as soon as i'll have some time (not before a few months) [18:04] <Compn> saste : should i start writing gsoc 2013 page ? [18:04] <saste> Compn, wait for llogan [18:04] <divVerent> wm4: BTW, actually found now that my "ffstuff" can't handle my test case :P [18:04] <saste> he asked mike if we should use multimediawiki.cx [18:04] <divVerent> or... it can? why did it work at all then [18:04] <BBB-work> actually let's send that to the ml [18:05] <Compn> saste : thats what we've always used .... [18:05] <divVerent> AH, because one tag is defined later :P [18:05] <saste> if we use trac we should make sure that can't be hacked by a random user [18:05] <divVerent> that's the sole reason why it even works+ [18:05] <Compn> trying to avoid conflict or something ? [18:05] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a34418c28e0a: tiff: check bppcount [18:05] <BBB-work> if you want to see h264 w/o mpegvideo and dsputil, that three patches finish it [18:05] <divVerent> because in ffmpeg filter graphs, it's hard to detect by text checks whether a tag is an input or an output [18:05] <saste> Compn, i don't know if mike is still actively maintaining the wiki [18:06] <saste> if he's not, he could be non responsive for granting access to new contributors [18:06] <Compn> oh [18:06] <saste> so i'd say let's wait a day or two [18:06] <Compn> who needs new wiki contribs ? :P [18:06] <Compn> students claim on the ml, devels change it on wiki [18:06] <Compn> but ok [18:06] <merbanan> BBB-work: did you really untie the Gordian knot ? [18:07] <saste> Compn, i don't know who has access to the (multimedia.cx) wiki [18:07] <Compn> merbanan : he also fixed svq3 breaking too :) [18:07] <saste> also we don't want to depend on it if mike is not maintaining it [18:07] <Compn> saste : http://wiki.multimedia.cx/index.php?title=Special:Listusers [18:07] <Compn> hehe [18:07] <Compn> yeah i got it [18:08] <saste> e.g. in case of attacks or spamming [18:08] <Compn> somewhat trolling [18:08] <BBB-work> merbanan, it doesn't compile yet if you actually make dsputil files selectable, but it's close [18:08] <saste> that's why we should wait his reply [18:08] <BBB-work> merbanan, what's left is the constants in dsputil_mmx (pw_N) [18:08] <Compn> BBB-work : did you run benchmarks on it ? [18:09] <Compn> oh i guess that doesnt change too much, nm [18:10] <michaelni> BBB-work, lots of ifdefs in be3d3b499bd241b7d6f7d9b4b6d91cb2dce5f5fe [18:11] <michaelni> not really pretty [18:11] <michaelni> does it have some users / usecase ? [18:12] <ubitux> i guess the reason is to have a config without the dspcontext need [18:16] <BBB-work> michaelni, chrome would use it [18:16] <BBB-work> michaelni, note how it removes dspcontext from h264 [18:16] <BBB-work> michaelni, that's a major size reduction [18:17] <BBB-work> merbanan, and secondly, the zigzag scan table permutators; thirdly, hpeldsp is still in there (used by vp3); fourthly, all the rectangular fullpel mc functions are currently in dsputil, but shared between chromamc (bilin), h264/mpeg4qpel, vp8mc, hpeldsp; all this needs to be made more generic for it to be truly finished [18:17] <BBB-work> merbanan, but it's almost there [18:18] <BBB-work> michaelni, might be worth doing the same for mpegvideo-derived codecs also; a lot of cases, you don't want er, you just want to error out [18:19] <nevcairiel> why would you want to error out instead of trying to mask an error? too risky that there are other bad errors in there? [18:21] <BBB-work> showing a corrupt frame only makes sense if the transport mechanism somehow allows that to have any meaning [18:21] <BBB-work> e.g. multi-slice frame, one slice corrupt: all is fine [18:21] <BBB-work> single-slice frame, one slice corrupt: yeahright [18:21] <BBB-work> vp8 same with partitions [18:21] <nevcairiel> doent er also cover single macroblock corruptions? [18:21] <BBB-work> over tcp, if the file is corrupt, it means the source was corrupt, so you want to error out if you're chrome [18:22] <BBB-work> over udp, the stream commonly becomes corrupt because of natural transport issues [18:22] <BBB-work> so it all depends on the use case [18:22] <BBB-work> for chrome, we don't want er; error means error out [18:22] <nevcairiel> not that ffmpegs ER is the best in the world anyway, my GPUs hardware decoder does a better job at covering things up in some clips =p [18:23] <BBB-work> plus er is slow, introduces extra security risks (more code to cover), is not as well tested, doesn't support >8bpp, I'm not even sure it supports non-420 [18:23] <BBB-work> isn't integrated with ffvp8 [18:23] <BBB-work> there's tons of issues with it [18:23] <BBB-work> we'd prefer to not use it [18:23] <michaelni> is there a big disadvantage over making ER optional while leaving most things in h264.c so there are few ifdefs? [18:24] <BBB-work> no, that should be fine [18:24] <BBB-work> I just did the ifdef route so I know I fixed all users [18:24] <BBB-work> but I can just add if (CONFIG_ER) around the relevant calling blocks and leave the assignments that don't require er symbols in place [18:24] <BBB-work> I don't care about 100bytes of unused ERContext memory in H264Context [18:25] <michaelni> if(CONFIG_ER) sounds alot better than ifdefs [18:25] <BBB-work> sure, can adapt [18:25] <BBB-work> we might need to make ER explicitely settable in configure than [18:25] <BBB-work> since currently, --disable-everything --enable-decoder=h264 will disable er [18:25] <BBB-work> with no way to enable it other than --enable-decoder=vc1 or so [18:26] <BBB-work> which isn't quite ideal [18:26] <nevcairiel> cant you make it enabled by default if a decoder is selected which uses it, unless overriden at command line? [18:26] <BBB-work> I don't know enough about configure to know how to do that [18:27] <ubitux> with the _select vars maybe [18:27] <michaelni> about lack of >8bpp support, it also lacks support for interlaced concealment, i suspect these are why the GPU is better, our error concealmentz was actually quite good in the cases that it supports when i last compared to other decoders but thats long ago [18:28] <michaelni> interlace, >8bpp support should of course be added ... [18:28] <nevcairiel> my clip was 420 8bpp [18:28] <nevcairiel> looks like a slice was failing [18:29] <nevcairiel> ffmpeg overwrote the previous reference frame with a flat color more or less, while the gpu decoder just seemed to try to interpolate some motion vectors on top of it, which looked OKish because there wasnt much movement [18:29] <ubitux> (btw, it would be nice to fix all the uninitialized read issues recently introduced...) [18:29] <ubitux> libav might have not realized it because their valgrind+undef is broken [18:30] <michaelni> nevcairiel, probably makes sense to open a ticket with sample file and screenshoot [18:30] <nevcairiel> i'll put it on my list of things todo for a rainy day [18:31] <nevcairiel> at least i still know which sample it was [18:33] <michaelni> nevcairiel, also try with threads 1, could be a thread issue [18:34] <michaelni> on that rainy day thart is [18:34] <nevcairiel> i suppose [18:34] <nevcairiel> i'll try to remember this [18:34] <nevcairiel> lately i'm happy to have time to sleep [18:37] <BBB-work> it's certainly true er is broken wiht threading [18:38] <BBB-work> mpeg-er has ff_thread_await_progress() indicators [18:38] <BBB-work> h264-er doesn't, they're explicitely marked as FIXME [18:38] <BBB-work> lack of a fate test means I couldn't be sure it worked or not so I just left it out [18:38] <BBB-work> (i.e. yes we need a h264-er test) [18:39] <nevcairiel> just out of curiosity, does the spec outline how missing slices for example should be handled? [18:41] <BBB-work> no [18:41] <BBB-work> hm does implement something for it, I believe [18:41] <BBB-work> but you're not required to follow that, from what I understand [18:41] <BBB-work> jm, not hm [18:41] <BBB-work> lol [18:43] <durandal_1707> stupid spammer [18:43] <BBB-work> I also wonder if I should make ercontext embed dspcontext, so the caller doesn't have to set it anymore [18:45] <durandal_1707> how to permanently remove files uploaded to trac? [18:46] <durandal_1707> saste: gonna comment on blend filter? [18:47] <saste> durandal_1707, yes in a few hours, please wait [18:59] <michaelni> slavon888 Smartmil8 smartmil888(a)gmail.com [18:59] <michaelni> IP address: 93.84.19.159 [19:05] <michaelni> BBB-work, for frame threading with EC not only does the EC code need to wait for its references to be available subsequent frames also need to wait as soon as a slice is missing or damaged, i think they dont currently [19:06] <saste> i'm going to implement an annoying feature in overlay [19:06] <michaelni> slice threading should work with EC though [19:08] <durandal_1707> saste: what that means? [19:11] <saste> durandal_1707, that people will suffer because of that [19:11] <durandal_1707> saste: what feature? [19:11] <saste> durandal_1707, surprise, but you can easily guess what [19:12] <durandal_1707> i lost my pil [19:12] <saste> no more teasers [19:13] <BBB-work> michaelni, no they do (at least for mpeg1/2) [19:13] <BBB-work> michaelni, for h264 they probably don't, you may be right [19:20] <cone-841> ffmpeg.git 03Paul B Mahol 07master:76415a942051: configure: fix libavfilter pkgconfig description [19:21] <michaelni> mpeg1/2 do only slice threads [19:23] <michaelni> also theres the problem that a error in MB x,y can produce the first detectable error a row or more later [19:23] <michaelni> (not in mpeg2 which is a special case here) [19:25] <BBB-work> maybe it's mpeg4 [19:25] <BBB-work> it's the one covered by the -error fate test [19:26] <BBB-work> I don't know which mpeg it is, they all look so much alike (and are - honestly - all equally useless) [19:31] <michaelni> yes, seems mpeg4 has a simple check [20:37] <saste> ubitux, durandal_1707: how hard would be to make showspectrum output a frame only when it is completely filled? [20:38] <saste> this should allow to compose a global spectrum image [20:54] <durandal_1707> saste: aren't you lavfi expert? [20:55] <durandal_1707> saste: fix your connection [20:55] <durandal_1707> by calling ff_filter_frame only once [20:56] <durandal_1707> lol [20:57] <saste> damn [20:59] <durandal_1707> saste: got any of my replies? [21:01] <saste> i got: by calling ff_filter_frame only once [21:02] <saste> i tried to look at the code, and changing the logic for outputting just one filled frame doesn't look trivial [21:02] <saste> that's why i asked [21:03] <nevcairiel> michaelni: i took two screenshots of the EC behaviour on this clip http://screenshotcomparison.com/comparison/8966 .. while you can see that there is something wrong in the GPU decoded image, its by far not as obvious (and even less so when the image is moving) .. and the second one is even worse: http://screenshotcomparison.com/comparison/8968 [21:03] <nevcairiel> its single threaded, so not related to threading in this case [21:04] <durandal_1707> saste: why it is not trivial? [21:06] <nevcairiel> a user actually asked me once for this particular clip if avcodec was broken, because he couldnt see the corruption with the gpu decoder, and thought it might be a software issue =p [21:06] <nevcairiel> i will upload the clip and open a ticket, i guess [21:07] <saste> durandal_1707, why? because of the caching logic, which is not immediate to grasp [21:14] <ubitux> between two timeouts, feel free to tell him "no" to answer his question [21:15] <ubitux> damn it's not answering the question [21:15] <ubitux> then "hard at 5%" [21:16] <durandal_1707> ^ what that last one means? [21:16] <ubitux> it means it's not hard [21:17] <durandal_1707> than do it before saste comes ;) [22:04] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:ce153eef8ffd: vc1dec: dont decode slices when the slice header failed to decode. [22:22] <durandal_1707> saste: fixed connection? [22:23] <saste> durandal_1707, don't know, it doesn't depend on me [22:23] <saste> let's see if it lasts [22:24] <durandal_1707> saste: what caching logic you see in showspectrum? [22:25] <saste> durandal_1707, in showwaves i send the frame only when I fill the frame [22:25] <durandal_1707> so? [22:25] <saste> with showspectrum it is different, since the frame is updated at each audio frame, right? [22:25] <saste> so it is different [22:26] <durandal_1707> i dont think updating frame is good idea (even you could store frame data in another buffer) [22:26] <durandal_1707> generally see what sox does [22:27] <saste> durandal_1707, sox is not meant for audio visualization [22:27] <saste> i want that feature mostly to be able to generate a single image containing the audio spectrum [22:29] <durandal_1707> saste: sox does just that [22:30] <durandal_1707> for some duration - can be whole file [22:30] <saste> durandal_1707, will you implement that feature of should I look at it? [22:30] <durandal_1707> but lavfi do not know what is duration of file yet, but this does not stop you to implement it [22:31] <durandal_1707> saste: i plan to work on exr and filters (blend, stereo3d, noise, etc ...) [22:32] <saste> durandal_1707, i don't mind, as long as I can script it and use tile and scale if necessary [22:32] <saste> my problem is that don't want overlapping frames [22:32] <durandal_1707> you return frame at EOF [22:35] <durandal_1707> saste: those doxygen issues you mentioned are present in geq/overlay too [22:35] <durandal_1707> i just copy-pasted it [22:36] <saste> durandal_1707, i know [22:36] <durandal_1707> saste: can av_image_copy() copy only one plane? [22:37] <saste> durandal_1707, av_image_copy_plane [22:37] <durandal_1707> nice [22:39] <saste> durandal_1707, i was sleeping when reviewing your overlay patch [22:40] <saste> working on a fix right now [22:40] <durandal_1707> saste: i had conversation on issue with michaelni [22:40] <durandal_1707> there are two solution: elegant and less elegant one [22:40] <saste> an ugly fix [22:41] <saste> yes I'm working on the less elegant/simpler [22:41] <durandal_1707> elegant is very simple [22:41] <durandal_1707> and makes filter simpler [22:42] <saste> durandal_1707, how? [22:42] <saste> does it involve scaling? [22:42] <durandal_1707> what you mean with more explicit check for inputs dimensions? [22:42] <durandal_1707> saste: no, using single pix fmt list [22:42] <saste> i meant a more explicit *error message* [22:42] <saste> like the one in concat [22:43] <saste> durandal_1707, i don't understand about overlay [22:43] <saste> what I'm doing now is to select the "format", either yuv420, yuv444, rgb [22:44] <saste> it is a generalization of the rgb option [22:44] <durandal_1707> removing slow rgba code handling different rgbs in inputs [22:45] <durandal_1707> i see no point to allow inputs to be of different pix fmts [22:45] <saste> durandal_1707, the overlay is usually alpha [22:45] <saste> why do you need to add alpha to the main input? [22:47] <durandal_1707> than API have limitations [22:48] <durandal_1707> there should be way to force inputs to have same subsampling and etc [22:48] <durandal_1707> but anyway rgba handling is funny [22:48] <saste> how? [22:49] <durandal_1707> converting both inputs to same one would be faster than conversion code in filter [22:49] <durandal_1707> because you would do just memcpy in filter [22:50] <durandal_1707> saste: also why if i give single images as inputs to blend filter i get nothing in output? [22:50] <durandal_1707> looks like there is of by 1 error somewhere [22:51] <durandal_1707> eg, using 2 images for inputs give 1 output frame [22:51] <saste> durandal_1707, two images for *both* inputs? [22:51] <saste> or two images in total? [22:52] <saste> also I see no synchronization mechanism in your code [22:52] <durandal_1707> two images in total = 0 [22:52] <saste> that's what makes overlay more complex [22:52] <durandal_1707> saste: why would i need to add one (same code in overlay)? [22:52] <durandal_1707> i used alphamerge, and it does not have this issue [22:53] <saste> alphamerge is supposed to be used on synched inputs [22:53] <saste> but i don't mind, that can be done on top of your patch [22:53] <saste> ideally the synching code should be shared [22:53] <durandal_1707> so i just add setpts? [22:54] <saste> durandal_1707, i suppose you have this situation [22:54] <saste> 1, 1, EOF, 2 [22:54] <saste> 1, EOF, 2 [22:54] <saste> or something like that [22:55] <saste> when the first EOF is found, it exists [22:55] <saste> *exits [22:55] <saste> so you have no chance to blend the two frames [22:55] <durandal_1707> could be [22:56] <durandal_1707> so if EOF is found on first input but not on second i just return EAGAIN? [22:56] <saste> setpts can't help in your case since it depends on how frames are cached on the filtergraph [22:56] <saste> durandal_1707, the complete logic is in overlay, so that should be ideally the model to follow [22:57] <saste> see if nicolas wants to help, since he proposed to write common utilities for input synching [22:59] Action: llogan should have picked a less generic name [23:16] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a5153b1d1692: shorten: Fix signedness of comparission [23:39] <wm4> how do you force a video codec with ffplay? [23:41] <saste> wm4: -vcodec? [23:41] <wm4> Unrecognized option 'vcodec' [23:41] <wm4> Failed to set value 'h264' for option 'vcodec' [23:41] <wm4> I tried that first [23:46] <michaelni> wm4 -codec:v h264 [23:53] <wm4> runtime CPU detection is enabled by default, right? [23:54] <BBB-work> not for libpostproc I think [23:55] <BBB-work> then again, libpostproc... [23:55] <wm4> yeah, nobody cares about libpostproc [23:55] <wm4> and isn't it full of 2001 code anyway? [23:58] <saste> wm4, how old is your ffplay? [23:58] <wm4> saste: oh, that's the distro version, probably a bit older [23:59] <saste> -vcodec should work, i fixed it some months ago [23:59] <wm4> yeah, it works with the git version [00:00] --- Wed Feb 20 2013

1 0

ffmpeg.log.20130219
by burek 20 Feb '13

20 Feb '13

[01:00] <teratorn> is command like this possible? ffplay -f rawvideo -pixel_format yuv420p -s 352x288 '/tmp/%04d.yuv420p.raw' [01:24] <dionoea> /win 0 [03:45] <relaxed> teratorn: I think only the image2 format can use printf's %d naming as input. [03:46] <relaxed> but, you can cat raw yuv in as input. [03:49] <defaultro> hey guys, how can we mimic a warm filter? [03:50] <defaultro> it's different from hue filter [03:51] <defaultro> or maybe another solution is to superimpose a png image with 50% or less for alpha [03:51] <defaultro> is that possible? [08:53] <mista_D> anyway to run libass on same file using two srt files (eng and spa)? without retranscoding twice? [09:14] <juanfont> hi. Is there any way to reduce max_analyze_duration at runtime? or discard those 5M of data? I'm trying to reduce the lag between server and client as much as I can [09:14] <juanfont> thanks :) [09:29] <eli___> hello [09:33] <eeli_> hello [09:33] <eeli_> would you please help me `md? [09:34] <eeli_> what can i do for can not chatting freenode channel? [10:52] <jjido> Hello, I learnt that .VOB files in a VIDEO_TS folder are MPEG2. Do you know if it is possible to replace the video and audio streams in it to occupy less space? I want to do what DVDShrink does to fit a 2H DVD on a 4.7GB disk [11:51] <invariant> I have a wav file which is the result of a cdrip. I want to turn that in a sequence of flac files. How can that be done? [11:53] <relaxed> you need to rip the cd into a sequence of wav files [11:55] <invariant> relaxed, alternatively, how can I make the wav file available as a rippable device? [11:56] <invariant> I don't get why all the rippers work on a device file myself. [11:56] <relaxed> Does it have a cue sheet? [11:59] <invariant> relaxed, yes [11:59] <invariant> relaxed, I have a cue file. [12:00] <relaxed> Then you need software to extract the tracks using that file. ffmpeg can't do it. [12:01] <durandal_1707> ffmpeg can do it but not in friendly way like some tools designed to do just that [12:02] <relaxed> invariant: mplayer2 has cue://$file support, use that with -dumpstream to extract a wav for each track. [12:04] <durandal_1707> http://www.etree.org/shnutils/shntool/ [12:04] <durandal_1707> it have cue mode [12:05] <relaxed> real men use mplayer [12:05] Action: relaxed flexes [12:15] <durandal_1707> this one can create cue files [12:41] <Aktau> Hey guys. I've been using ffmpeg to decode videos very well, but I'd like to know if it does (maybe through libavcodec) support HW accelerated decode through VA-API on linux? [12:41] <Aktau> Does anyone know about that? [14:00] <JEEB> WmA, sending me private messages for nothing that needs to be private or is specifically related to me will only lead to /ignore :P [15:35] <otters> being a total noob, what option do I pass to map_metadata to convert a FLAC to Apple Lossless and keep per-track metadata intact (like album name etc.) [15:37] <durandal_1707> they are usually completly copied and encoder is added [15:38] <otters> so adding map_metadata is redundant? [15:38] <durandal_1707> dunno, did you tried it ? does it work as expected? [15:38] <otters> yeah, I tried it, and the tracks don't have *any* metadata when I open them in itunes [15:39] <otters> oh, hey, yeah, omitting the option works. [15:40] <otters> I must be some sort of moron [15:46] <otters> thanks everyone [15:53] <e2b> Is there any plan to support the Raspberry Pi GNU-acceleration for en-/decoding? The CPU itself is pretty slow, but thanks to the GPU it can already play HD videos witch optimized players. [15:55] <durandal_1707> no plans, people usually come with patch that implements such funcionality - that is how FOSS works [15:58] <divVerent> if "nm -a binary" shows an entry: [15:58] <divVerent> 000000010087ca70 T _avcodec_default_execute [15:58] <divVerent> does that mean I have the right to the source code of the binary? ;) [15:59] <divVerent> I found this in "iTap Mobile RDP.app" for Mac OS X [15:59] <divVerent> *hint* want to bring that Hall of Shame back? [16:00] <durandal_1707> only if you have enough money to sue them [16:00] <divVerent> putting them on the HoS costs no money :P [16:00] <durandal_1707> and pay lawyers and etc [16:00] <divVerent> also, they violated none of MY copyright [16:01] <divVerent> so I wouldn't have a case there [16:01] <divVerent> but I mainly am wondering, did I read the "nm" output right and this is not an external reference? [16:01] <divVerent> I THINK I did [16:02] <divVerent> also, can I somehoe figure out if they used ffmpeg or libav? [16:02] <divVerent> ah, it's PROBABLY libav [16:02] <divVerent> I see an URL with upload.libav.org [16:38] <MIG-> e2b: raspbian includes source code for x264 encoding using the gpu [16:42] <jeje34> Hi to all ;-) [16:43] <MIG-> hello [16:44] <jeje34> I'm want to use FFMPEG library in a windows application to decode H264 video streams from camera. I have a question about the memalign-hack for the configure compilation. I'm not sure if it needed on Windows OS [16:46] <JEEB> if it is enabled by default in the configuration for your way of building for windows on the current ffmpeg, then it is either needed or not [16:46] <JEEB> you shouldn't have to set it manually [16:46] <jeje34> yes but in the configure file there: --enable-memalign-hack emulate memalign, interferes with memory debuggers [16:47] <jeje34> So if I use Visual Studio VC++ debuggers, it can make me memory problems? [16:47] <JEEB> I repeat [16:47] <JEEB> you shouldn't have to enable it manually [16:48] <JEEB> configure should look it up by the way you are compiling it and for which arch [16:48] <jeje34> yes but you tell me it is enables by default [16:48] <JEEB> DEPENDS ON THE CONFIGURATION AND WHAT YOU ARE BUILDING WITH [16:48] <JEEB> AS I SAID [16:48] <jeje34> I use msys and mingw to compile FFMPEG on my windows [16:48] <JEEB> look at the gawd damn configure script's output [16:48] <JEEB> seriously [16:49] <JEEB> and it shouldn't make the thing any worse debug-wise [16:49] <JEEB> if it is needed, it is needed [16:49] <JEEB> if it isn't, it isn't [16:49] <JEEB> that's it [16:49] <JEEB> if you are building with mingw MSVS's debugger won't be able to load the damn debug symbols anyways, if you are building with MSVC then you are getting the debug symbols if you enable debug [16:50] <jeje34> I use Visual Studio to make my application, but I use the library of FFMPEG I built and don't try to debug "inside" FFMPEG [16:51] <jeje34> just use debugger to debug my application [16:52] <JEEB> go over what I said, the answers to your question(s) are there [17:14] <ozone89> hello :) [17:21] <jeje34> I'm sorry but there is something I can't explain in my code. I have an AVPacket m_lpPacket and an AVPicture m_lpPict. In order, I call m_lpPict.data[0] and after decoding I use avpicture_alloc( &m_lpPict, AV_PIX_FMT_RGB32, dwWidth, dwHeight); to use swscale to convert my YUV420 to RGB32. But at the next frame, when I call av_init_packet(&m_lpPacket), m_lpPict.data[0] become NULL. [17:41] <mpfundstein> jeje34: didnt know people still use hungarian notation [17:44] <jeje34> I find it more clear when writing classes [17:58] <fenduru> How can I avoid artifacts between segments when segmenting an audio file? [17:59] <fenduru> I need to be able to play them back sequentially without gaps/clicks between them [18:36] <sabton> hey guys, just got caught up and built for Android at head release/1.1, after making necessary adjustments for deprecated API everything is running stable... except certain codecs (ogg/mp3) are decoding to garbled audio; however FLAC for example is decoding perfect [18:36] <sabton> does this immediately indicate anything to anyone? [18:48] <JEEB> sabton, sounds like you're expecting the decoder to output interleaved int while it is giving you planar float? [18:48] <JEEB> although if flac works then it's the float vs int difference [18:48] <JEEB> most lossy decoders were switched to output float by default, as that's how it basically gets encoded [18:49] <JEEB> lossless stuff like flac naturally should output exactly what is put in [18:49] <durandal_1707> flac did not changed sample format [18:49] <JEEB> yes [18:49] <durandal_1707> so it is GrEaT sample format switch [18:49] <JEEB> I already noted it [18:49] <JEEB> if you didn't see my second line :P [18:50] <JEEB> and then the third+fourth noting the fact that the sample format was changed for most lossy formats [18:50] <JEEB> sabton, ffmpeg does provide you a library to then convert the float audio to integer audio that you probably want [18:51] <durandal_1707> and from planar to interleaved [18:51] <JEEB> yup, if that is needed [18:53] <sabton> ooh [18:55] <sabton> that sure does explain things [18:55] <sabton> the sound of my garbled audio is consistent with that explanation [18:56] <sabton> so in addition to converting the samples, i guess the codecs will tell me what sample format is used? [18:56] <sabton> i would need to find that API as well [18:56] <JEEB> yes, I would think the avframe tells you what the output format is [18:57] <sabton> ah, thank you [18:57] <sabton> man, im playing with compiler settings and such.. =\ [18:57] <sabton> thanks :) [19:13] <Dmole> is there a repo rpm with a recent ffmpeg with all libs enabled or is everyone compiling for some reason? [19:17] <sophie_> hi [19:17] <Dmole> what's p? [19:17] <sophie_> do you know it's possible to run ffmpeg with ubuntu 11.04 ? [19:18] <Dmole> ye it is. [19:18] <Dmole> *yes [19:18] <sophie_> i have a problem with yasm [19:18] <sophie_> i have difficulties to upgrade it [19:19] <JEEB> there shouldn't really be anything hard in upgrading it [19:19] <JEEB> build it and put it somewhere in your PATH [19:20] <Dmole> might need to purge from apt before did you try compiling yasm? [19:20] <sophie_> yes [19:20] <Dmole> so what exactly is the issue? [19:20] <sophie_> terminal tell me : [19:20] <sophie_> Found yasm 1.1.0.2352 [19:20] <sophie_> Minimum version is yasm-1.2.0 [19:20] <sophie_> If you really want to compile without asm, configure with --disable-asm. [19:21] <JEEB> that just means that you have 1.1.0 installed [19:21] <sophie_> but i've already done apt-get upgrade yasm [19:21] <JEEB> yes, because the repositories don't have 1.2.0 [19:21] <JEEB> you have to grab the tarball [19:21] <JEEB> extract it [19:21] <sophie_> so, i can also unabled i ? [19:21] <sophie_> it [19:21] <JEEB> what [19:21] <Dmole> disable [19:22] <JEEB> no, don't disable asm [19:22] <JEEB> that will be SLOW [19:22] <sophie_> ok [19:22] <sophie_> the tarball ? [19:22] <JEEB> http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz [19:22] <JEEB> wget that [19:22] <JEEB> tar xvf yasm[tab] [19:22] <JEEB> cd yasm-[tab] [19:23] <JEEB> do you want it to be available to everyone in the system automatically? [19:23] <JEEB> or are you OK with a local install? [19:23] <sophie_> local install it's ok [19:24] <JEEB> ./configure --prefix=${HOME}/ownapps [19:24] <JEEB> make [19:24] <JEEB> make install (no sudo needed) [19:24] <sophie_> ok, i try [19:24] <JEEB> export PATH=${HOME}/ownapps/bin:${PATH} [19:24] <JEEB> then try what yasm --version gives [19:25] <JEEB> if that works, then you can add that thing into your dot-bashrc file [19:25] <JEEB> so that directory gets added to the PATH every time [19:35] <sophie_> thank you JEEB [19:35] <sophie_> it has worked! [19:41] <sophie_> JEEB what do you mean by "add that thing into your dot-bashrc file" [19:41] <sophie_> ? [19:41] <sophie_> you mean these phrase : [19:41] <sophie_> export PATH=${HOME}/ownapps/bin:${PATH} [19:42] <JEEB> yes, that thing. dot-bashrc is a file that gets loaded every time bash gets loaded :P [19:42] <JEEB> basically so you don't have to run it every time you start a terminal [19:43] <JEEB> manually [19:43] <sophie_> and where it is ? [19:44] <JEEB> /home/your_username/ is where .bashrc should reside [19:44] <JEEB> (that file name is dot-bashrc) [19:45] <JEEB> although you generally should have one I guess [19:45] <sophie_> when i want to ./configure ffmpeg [19:45] <sophie_> it tells me : [19:45] <sophie_> :~/ffmpeg-source/x264$ ./configure [19:45] <sophie_> ./configure: ligne 241: rm : commande introuvable [19:45] <sophie_> ./configure: ligne 243: dirname : commande introuvable [19:45] <sophie_> ./configure: ligne 245: grep : commande introuvable [19:45] <JEEB> uh-oh [19:45] <JEEB> you did something wrong :P [19:45] <sophie_> Unknown system , edit the configure [19:45] <sophie_> yes [19:45] <sophie_> i know [19:45] <JEEB> your original ${PATH} got derped [19:46] <sophie_> i just pick some command on the net [19:46] <sophie_> when it wasn't good [19:46] <sophie_> what can i do ? [19:46] <sophie_> reinstall my all system ? [19:47] <JEEB> no... [19:47] <JEEB> you can just close the terminal and open a new one :P [19:47] <sophie_> :) [19:49] <sophie_> i was on this site : [19:49] <sophie_> http://buildall.wordpress.com/2011/05/09/how-to-compile-and-install-ffmpeg-… [19:50] <sophie_> perhap's it will be more easy by installing new version of ubuntu ? [19:50] <JEEB> no [19:50] <JEEB> also that "guide" is a few years old :P [19:51] <JEEB> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide use this if you want to use some damn guide :P [19:51] <JEEB> also this installs yasm on the whole system [19:52] <sophie_> yes, it's ok too [19:53] <sophie_> This guide supports Ubuntu 12.10 (Quantal Quetzal), Ubuntu 12.04 (Precise Pangolin), and Ubuntu 11.10 (Oneiric Ocelot). [19:54] <sophie_> but i have version 11.04 [19:54] <sophie_> natty [19:54] <JEEB> shouldn't matter too much in most cases [19:54] <sophie_> it will work either ? [19:54] <sophie_> ok [19:54] <JEEB> there are some things you actually will have to think about, yes [19:54] <JEEB> now you're not afraid of thinking, right? [19:55] <JEEB> just try to understand what the fuck is going on instead of mindless copy-pasting shit and you will be effing fine [19:56] <sophie_> yeah! [19:56] <sophie_> you know how to speak my godness! [20:00] <JEEB> Do pardon me on my french, but I was kind of trying to think how to explain some very basic *nix things to you, but ended up just not thinking about it because it made my head hurt, and I'm not here to teach basics of *nix [20:12] <Dmole> or if you don't want to think stick to apt [20:29] <reactormonk> how do I drop the video stream? -vcodec null doesn't work [20:30] <JEEB> if you want no video at all in the output, -vn [20:30] <reactormonk> thanks [20:45] <WmA> can anyone tell me what I am doing wrong please? [20:45] <WmA> http://pastebin.com/zZpSwrEq [20:46] <WmA> I would like to pipe a http ts stream into libx264 [20:46] <WmA> to record it straight to mp4 [20:49] <relaxed> WmA: why wouldn't you use the stream as ffmpeg's input? [20:58] <stephanedev> hi everybody. i wrote a program using ffmpeg C API which streams a video encoded with vp8 over rtp. when streaming the video with ffmpeg and playing it with ffplay and simulating a 5% datagram loss, the video still plays ok and most frames are not too degraded. [20:59] <stephanedev> but when using my program, it works fine if no datagram is lost, but with a 5% loss, ffplay shows a much more degraded video with most frames showing big pixel blocks or sometimes unrecognizable images [21:00] <stephanedev> of course i am not surprised that ffmpeg is doing a much better job than my program, but i would like to understand what it does to achieve that [21:01] <stephanedev> i tried to play with gop size, bitrate and other settings, but to no avail [21:04] <stephanedev> does anyone have any hint on what i could try to improve my streaming code? [21:10] <stephanedev> is there any parameter in AVCodecContext or AVFormatContext which would help? [22:15] <vulkanr> hello. need some help with libavfilter. anyone in the house? [22:15] <saste> vulkanr, don't ask to ask, just ask [22:16] <vulkanr> trying to transcode video and pass the raw frames though libavfilter. i am getting really bad quality output. i think i am doing something wrong with the frame pts. [22:17] <vulkanr> looking at the output file - the frames seem to be ordered by the pts which is weird and not the case when not using libavfilter [22:19] <vulkanr> i followed the example code and am converting the pts between timebases as showed [22:34] <ezekiel> in recent versions of ffmpeg, has the "lowpass"(http://ffmpeg.org/ffmpeg-filters.html#lowpass) filter been removed? Or perhaps replaced with something better? [22:35] <saste> ezekiel, it has been added one week ago or so [22:35] <ezekiel> okay, so only master ? [22:35] <saste> yes [22:35] <ezekiel> saste: much appreciated! [22:36] <ezekiel> I've been building from git, but I've had to step back a few versions to achieve good audio+video sync from v4l2 [22:36] <saste> ezekiel, how old? we recently fixed a few v4l2 issues [22:36] <ezekiel> 1.0.4 works great - 1.1.X is out of sync :( [22:37] <saste> ezekiel, try master [22:38] <ezekiel> I will try again and hopefully come back with some better info [23:19] <ezekiel> eck, bad news for me - build from master still doesn't capture synchronized audio [23:33] <ezekiel> to AVI: http://pastebin.com/HHddvm8W [23:36] <rsumi> hello, i'm trying to build mediainfo on my box but i'm getting the following errors - http://pastebin.com/ZEJ9sQNA - how can i fix this? [23:38] <llogan> rsumi: this is #ffmpeg. mediainfo is not from the FFmepg project. [23:38] <llogan> ezekiel: N-36755-gce153ee does not appear to be current git head [23:39] <rsumi> llogan: yes i know but i thought it's likely that someone from ffmpeg has ran into this issue. thanks anyways [23:39] <llogan> we usually just use ffprobe instead [23:39] <ezekiel> hmm, git log tells me - commit ce153eef8ffdd10ead74ad4646eb784be1c5b107 [23:40] <saste> ezekiel, how can be sure that the timestamps are synched? [23:40] <llogan> current is N-50103-ga5153b1 [23:41] <saste> use setpts/showinfo to make sure [23:41] <llogan> rsumi: did you run ./autogen before configure? [23:41] <ezekiel> llogan: yeah, that commit was pushed while I was building - so now my build is one commit behind [23:42] <llogan> close enough, but your N- number is odd [23:42] <llogan> i mean weird [23:43] <rsumi> llogan: no. i just did now and i get these errors [23:43] <rsumi> http://pastebin.com/gXNcxc94 [23:44] <llogan> i don't know. make sure you have libtool, automake, autoconf as build deps and libmediainfo as a depends. [23:45] <llogan> other than that i don't know, especially for PPC. [23:47] <rsumi> llogan: yeah i have all that stuff except for libmediainfo. i suspect libmediainfo is what i get after building [00:00] --- Wed Feb 20 2013

1 0

ffmpeg-devel.log.20130218
by burek 19 Feb '13

19 Feb '13

[00:05] <BBB> michaelni: have you tested h264 w/ and w/o edges (i.e. w/ and without draw_edges) lately (since I optimized ff_emulated_edge_mc)? [00:06] <BBB> michaelni: it's the last dsp function (not counting er) [00:07] <BBB> michaelni: I'd very much fancy killing it [00:09] <michaelni> Any tests i did where in the distant past i think [00:10] <BBB> hm ok I'll do some testing [00:10] <BBB> on vp8, not using draw_edge was never slower and sometimes faster [00:10] <BBB> (only tested on x86 ofc) [00:10] <BBB> I think the same would hold for arm also though [00:11] <ubitux> yay blend filter. [00:11] <ubitux> Paul is on steroids recently [00:12] <nevcairiel> i think he is on a mission to kill mpcodecs [00:12] <ubitux> yes, that's good, but afaict blend is not related to it :) [00:12] <ubitux> same for all the awesome audio filters [00:21] <Compn> gasp! [00:21] <Compn> hes gone! [00:21] <Compn> :p [00:31] <cone-485> ffmpeg.git 03Paul B Mahol 07fatal: ambiguous argument 'refs/tags/n0.8.13': unknown revision or path not in the working tree. [00:31] <cone-485> Use '--' to separate paths from revisions [00:31] <cone-485> refs/tags/n0.8.13:HEAD: lavfi/noise: switch to AVLFG noise generator [00:34] <BBB> yeah h264 is definitely faster with -flags +emu_edge on x86 [00:34] <BBB> (on cathedral) [00:34] <BBB> so same as vp8 [00:34] <BBB> I'll kill draw_edges - fun fun fun [00:53] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:54b2bddd22fb: v4l2: try to fix build on BSD [01:09] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:b102d5d97dae: h264: allow cropping to AVCodecContext.width/height [01:09] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:0054d70f23ed: mov: set AVCodecContext.width/height for h264 [01:09] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:c497d71a026a: Clarify that -passlogfile has a different syntax when used with -vcodec libx264. [01:10] <cone-485> ffmpeg.git 03Ronald S. Bultje 07release/0.7:9a5e81235e62: dxva2: include dxva.h if found [01:10] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:8582e6e9a3ce: Fix muxing mjpeg in swf. (cherry picked from commit 7680d99b4302e476076cc1b8f2567f47c2aaef4d) [01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:a60eb6ef12df: ffmpeg: fix -force_key_frames [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:0173a7966b33: vc1dec: add flush function for WMV9 and VC-1 decoders [01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:f31170d4e7f9: nuv: check RTjpeg header for validity [01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:8812b5f16410: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:899d95efe12f: mpegvideo: Don't use ff_mspel_motion() for vc1 [01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:77d43bf42d76: lavf: don't segfault when a NULL filename is passed to avformat_open_input() [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:b6ba39f931a8: alsdec: check opt_order. [01:10] <cone-485> ffmpeg.git 03Mina Nagy Zaki 07release/0.7:b6c5848a1f8f: lavfi: avfilter_merge_formats: handle case where inputs are same [01:10] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:61ece4137298: vorbisenc: check all allocations for failure [01:10] <cone-485> ffmpeg.git 03Alex Converse 07release/0.7:d6e250abfc36: vorbis: Validate that the floor 1 X values contain no duplicates. [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e28814e0e16c: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 [01:10] <cone-485> ffmpeg.git 03Diego Biurrun 07release/0.7:7b91e52eb9fd: x86: Require an assembler able to cope with AVX instructions [01:10] <cone-485> ffmpeg.git 03Clément BSsch 07release/0.7:80b8dc30dc96: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [01:10] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:a2ae183a382f: h264: allow cropping to AVCodecContext.width/height [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:e39fc137aeac: vc1dec: add flush function for WMV9 and VC-1 decoders [01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:d3e2f35f7add: bmpdec: only initialize palette for pal8. [01:10] <cone-485> ffmpeg.git 03Max Lazarov 07release/0.7:0892a6340f86: eval: fix swapping of lt() and lte() [01:10] <cone-485> ffmpeg.git 03Martin Storsjö 07release/0.7:a81c1ea2eb66: h263: Add ff_ prefix to nonstatic symbols [01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:99008ba3667a: rv34: use AVERROR return values in ff_rv34_decode_frame() [01:10] <cone-485> ffmpeg.git 03Mina Nagy Zaki 07release/0.7:10c244cc89e8: lavfi: avfilter_merge_formats: handle case where inputs are same [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:7a0ff7566b66: avsdec: Set dimensions instead of relying on the demuxer. [01:10] <cone-485> ffmpeg.git 03Aneesh Dogra 07release/0.7:42c3a3719b05: bytestream: add a new set of bytestream functions with overread checking [01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:4c849c69910c: dfa: check that the caller set width/height properly. [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:2d63f9b4effc: dfa: add some checks to ensure that decoder won't write past frame end [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:c0df6a24ce9e: indeo: check custom Huffman tables for errors [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:601fa565823b: indeo: clear allocated band buffers [01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:3c0f84402b2b: indeo: check for invalid motion vectors [01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:8148833193c6: indeo5: prevent null pointer dereference on broken files [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:aa097b4d5fd4: indeo5: check tile size in decode_mb_info(). [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:44da556815fc: lagarith: check count before writing zeros. [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:4a636a5e4368: wmaprodec: check num_vec_coeffs for validity [01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:05f5a2eb62db: avidec: use actually read size instead of requested size [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:6996a2f79689: cavsdec: check for changing w/h. [01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e3e369f6962c: alsdec: Check that quantized parcor coeffs are within range. [01:10] <cone-485> ffmpeg.git 03Thilo Borgmann 07release/0.7:1b48a426a96f: alsdec: Fix out of ltp_gain_values read. [01:10] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:7e070cf2025f: alsdec: remove dead assignments [01:10] <cone-485> ffmpeg.git 03Thilo Borgmann 07release/0.7:9474c9302844: alsdec: fix number of decoded samples in first sub-block in BGMC mode. [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:aa45b90804ab: alsdec: Check k used for rice decoder. [01:11] <cone-485> ffmpeg.git 03Reinhard Tartler 07release/0.7:642d758a2d82: Update RELEASE file for 0.7.7 [01:11] <cone-485> ffmpeg.git 03Piotr Bandurski 07release/0.7:fe9cbf582b09: tiffdec: Use the correct height field. [01:11] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:9f8071245491: ivi_common: check that scan pattern is set before using it. [01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:700fb8c8dd06: vp56: make parse_header return standard error codes [01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:7fd7950174f9: vp56: release frames on error [01:11] <cone-485> ffmpeg.git 03Sami Pietila 07release/0.7:bfbff1c7483c: vp8: reset loopfilter delta values at keyframes. [01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:f3f22f183fa2: yuv4mpeg: reject unsupported codecs [01:11] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:3d0c9c9af687: flacenc: ensure the order is within the min/max range in LPC order search [01:11] <cone-485> ffmpeg.git 03Reinhard Tartler 07release/0.7:ce8910d861c1: h264: Fix parameters to ff_er_add_slice() call [01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:4ede95e69cf9: vp6: properly fail on unsupported feature [01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:10ff052c6013: lavf: avoid integer overflow in ff_compute_frame_duration() [01:11] <cone-485> ffmpeg.git 03Alex Converse 07release/0.7:b143844ea0f6: aacdec: Fix an off-by-one overwrite when switching to LTP profile from MAIN. [01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:5fa739e685bc: h264: enable low delay only if no delayed frames were seen [01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:08d9fd611eac: ppc: always use pic for shared libraries [01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:4457e6137d83: h264: check sps.log2_max_frame_num for validity [01:11] <cone-485> ffmpeg.git 03Victor Lopez 07release/0.7:884a9b0d298a: h264: fix sps parsing for SVC and CAVLC 4:4:4 Intra profiles [01:11] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:a39c6bf1b878: alacdec: do not be too strict about the extradata size [01:11] <cone-485> ffmpeg.git 03Martin Storsjö 07release/0.7:808187965570: rtsp: Recheck the reordering queue if getting a new packet [01:11] <cone-485> ffmpeg.git 03Dale Curtis 07release/0.7:55065315caf1: Fix uninitialized reads on malformed ogg files. [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:391e0fc6c90c: roqvideodec: check dimensions validity [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:af343f5cddc5: eamad: fix out of array accesses [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:2cac35086c9e: vqavideo: check chunk sizes before reading chunks [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e6ac11e41734: aacdec: check channel count [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:41eda870483f: pngdec/filter: dont access out of array elements at the end [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:377fabc9e687: Update for 0.8.13 [01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:7378101d4102: Merge branch 'release/0.8' into release/0.7 [01:14] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:f86da592c339: update for 0.7.14 [01:23] <llogan> saste: are you interested in administering an application to GSoC this year? [01:24] <saste> llogan, if there are no other volunteers [01:25] <saste> possibly, i'd prefer to just act as co-administrator [01:26] <ubitux> @_@ [01:31] <llogan> i can help format/cleanup/prettify the ideas list at least... [01:34] <llogan> should we continue to use multimedia.cx, or use the trac wiki this time? .cx is more of an actual wiki, but it might look nice to use our own domain [01:35] <saste> llogan, you can already start to do it [01:35] <saste> i'm biased towared multimedia.cx, because of stricter access policy [01:35] <llogan> shit. responsibilities... [01:35] <saste> we don't want to let random j. hacker to edit the page just before review [01:36] <llogan> multimedia.cx is fine with me as long as whoever needs to edit it can get access. does it still depend on mike? [01:36] <saste> llogan, i think so [01:36] <saste> we should probably ask him before [01:37] <llogan> good idea. i can do that. [01:37] <saste> then we could simply copy the 2012 page to a new one, and let candidate mentors edit the proposed tasks [01:37] <llogan> ok [01:38] <llogan> i'll try to do that later today or tomorrow if i hear back from mike [01:39] <saste> llogan, thanks [01:46] <cone-485> ffmpeg.git 03Ronald S. Bultje 07master:c63f9fb37a7b: h264: don't store intra pcm samples in h->mb. [02:08] <ubitux> what is cathedral? [02:09] <BBB> a sample clip often used for profiling in this community [02:10] <ubitux> where can i find it? [02:10] <Compn> oh [02:10] <Compn> that clip [02:10] <Compn> raw yuv [02:10] <BBB> google for cathedral-beta2-400extra-crop-avc.mp4 [02:10] <ubitux> thx [02:11] <BBB> it's somewhere on mplayersamples or whatever that site is called [02:11] <BBB> but I'm too lazy to find it [02:12] <ubitux> http://samples.ffmpeg.org/V-codecs/h264/cathedral-beta2-400extra-crop-avc.m… [02:12] <ubitux> thx :) [02:12] <BBB> tadaah [02:13] <Compn> what? [02:13] <Compn> dont use the compressed version :P [02:14] <ubitux> oh, never saw this diablo-elephant-dream animation [02:14] Action: ubitux looking for a higher res now [02:16] <Compn> http://samples.ffmpeg.org/raw-video/ [02:16] <Compn> well foremans in there [02:18] <ubitux> looks like there are some other interesting stuff on platige.com [02:55] <ubitux> oh, seems libav realized they broke another codec with cosmetics [02:56] <ubitux> mmh, already fixed in ffmpeg. [02:59] <ubitux> actually, michaelni merged it properly [03:02] <Compn> we're not supposed to talk/troll/complain about libav, apparently [03:02] <Compn> :P [03:02] <ubitux> who told this? [03:02] <Compn> that guy daemon404 who complains about ffmpeg development [03:02] <ubitux> yes but he ragequited so that's ok [03:03] <Compn> lol [03:03] <ubitux> anyway, it's just that i find very funny when Diego is saying Michael is merging blindly, while he is actually fixing the crap he introduce with his cosmetics madness [03:05] <Compn> i'm curious how they arent breaking fate ? [03:09] <BBB> you know, svq3 is a very elaborate format [03:09] <BBB> it looks simple, but isn't [03:10] <BBB> it uses hpel, tpel, for example [03:10] <BBB> and has many coding modes [03:10] <BBB> code coverage of the sample in fate is likely in the 50% range [03:10] <BBB> in terms of functional coverage, not lines of code [03:11] <ubitux> in term of LoC it's about 70% [03:11] <ubitux> http://lucy.pkh.me/ffmpeg-coverage/ffmpeg/libavcodec/svq3.c.gcov.html [03:12] <ubitux> and it seems the broken code was exactly in the non-covered area [03:13] <ubitux> (~L564) [03:13] <BBB> there you go [03:13] <BBB> so one could say that we need more fate coverage [03:13] <BBB> (not that I care) [03:15] <Compn> svq3 is tied into h264 heavily [03:15] <Compn> or was [03:16] <Compn> BBB : i like your .2percent speedups :) [03:20] <BBB> it still is [03:20] <BBB> this is more cleanup than speedup, strictly speaking [03:20] <BBB> but thanks anyway [03:56] <BBB> isn't draw_horiz_band utterly broken for h264? [03:56] <BBB> (unless SLICE_FLAG_CODED_ORDER is set) [03:56] <BBB> michaelni: ^^ [04:04] <michaelni> BBB, it does look broken, i wonder why noone reported an issue [04:38] <BBB> michaelni: I can only imagine nobody is using it [04:38] <BBB> can we unshare that between mpeg and h264? [04:38] <BBB> it makes no sense to share after my last patch, there's virtually no code shared at all [05:50] <highgod> Hi,all,I want to ask a question. I use the hwaccel and software both decode the same h264 video to yuv. I want to ask that are the both yuv file are the same? totally? [05:56] <Compn> you want to compare each file ? [05:57] <Compn> use ffmpeg with the framemd5 , i think [05:57] <Compn> if they are exact, the md5 should match [05:57] <Compn> ffmpeg -i file.yuv -f framemd5 [06:02] <Compn> ffmpeg -i file.yuv -f framemd5 out.txt that is [06:02] <Compn> highgod : it work for you ? [06:02] <Compn> michaelni : are you going to review the dxva2 patch? i dont think anyone else is [06:02] <highgod> No,I use dxva2 and ffmpeg software decode the same h.264 video to yuv,but the two files is mismatch in some bits [06:03] <Compn> highgod : if you are using ffmpeg to decode using dxva2, cant you directly use -f framemd5 ? just to make sure the yuv part isnt adding additional bug [06:04] <Compn> are some frames the same ? is it one type of frame that causes problems? like b-frame or p-frame or i-frame ? [06:04] <Compn> and are you sure you are using single thread h264 software decoder ? i think multithreaded h264 decoder will cause slight differences when decoding [06:05] <BBB> no it won't [06:05] <Compn> oh ok, was that fixed ? [06:05] <BBB> multithreaded h264 is entirely bitexact for all test sequences [06:05] <Compn> good :) [06:05] <BBB> I fixed that before it was ever merged in either tree [06:05] <Compn> its hard to remember all of these bugs [06:06] <BBB> you can test it locally [06:06] <BBB> make THREADS=2 fate-h264 [06:06] <Compn> yeah well i could do that [06:06] <BBB> if that passes, all sequences are bitexact [06:06] <Compn> or i could go to sleep [06:06] Action: Compn sleeps [06:06] <BBB> fate.ffmpeg.org does it every day [06:06] <BBB> http://fate.ffmpeg.org/report.cgi?time=20130218001421&slot=x86_64-archlinux… [06:12] <highgod> I just use the merge tool to compare the two yuv [06:31] <BBB> you'd want to use jm to decode the h264 file and check which is correct [06:32] <highgod> sorry,I tested use ffmpeg -s 1920x1080 -i ./testfile/output.yuv -f framemd5 out.txt, and ffmpeg -s 1920x1080 -i ./testfile/output_dxva2.yuv -f framemd5 out_dxva2.txt.all the same [06:32] <highgod> 10frames [06:32] <highgod> I will test more frames [06:38] <highgod> I test all the 912 for my video,all the md5 code is the same [06:39] <BBB> then they are identical [06:44] <highgod> @BBB: the orignal code I implement modified the pthread code, is it can cause the difference? the difference happened on my orignal code. the patch version I tested just now is OK [06:44] <BBB> I don't know, I didn't see the patch [06:44] <BBB> messing with pthread is a bad idea in general, unless you know what you're doing [06:44] <BBB> (I know what I'm doing, so I know how much a bad idea it is) [06:46] <highgod> hehe,yes.Thanks [11:11] <ubitux> durandal_1707: about the ffmpeg -help full, i was wondering if it didn't go in infinite loop or something, because of the cross references [11:11] <ubitux> like it does for instance when you have several filters with the same class [11:16] <durandal_1707> ubitux: doesn't happen [11:17] <ubitux> okay, good [11:17] <durandal_1707> i dont see cross reference [11:17] <ubitux> seems valgrind is not happy since a little while [11:17] <durandal_1707> ? [11:17] <ubitux> durandal_1707: i was refering to multiple options refering to the same flags [11:17] <ubitux> (my valgrind remark is unrelated) [11:17] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20130218023828&slot=x86_64-archlinux… [11:17] <ubitux> http://fate.ffmpeg.org/history.cgi?slot=x86_64-archlinux-gcc-valgrindundef [11:18] <ubitux> i guess d2a25c4032ce6ceabb0f51b5c1e6ca865395a793 is the cause [11:19] <ubitux> i wonder if libav finally added a real valgrind instance to spot these [11:20] <ubitux> ah yeah, there is one, in the yellow as well. [11:20] <michaelni> Compn, did you ask fenrir ? he is the author of libavcodec/dxva2* ... [11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:39bec05ed42e: qdm2: check array index before use, fix out of array accesses [11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:f7d18deb73d1: vqavideo: check chunk sizes before reading chunks [11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:488f87be8735: roqvideodec: check dimensions validity [11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:96008082dbd3: Merge commit '488f87be873506abb01d67708a67c10a4dd29283' [11:36] <durandal_1707> ubitux: i gonna write lua filter if none will, using eval is slow and unfriendly [11:39] <durandal_1707> ubitux: guess what this does: ffmpeg -f lavfi -i color=c=white:size=720x480 -i INPUT -filter_complex "blend=all_expr=B*(if(gte(T\,10)\,1\,mod(T\,10)/10))+A*(1-(if(gte(T\,10)\,1\,mod(T\,10)/10)))" [11:59] <ubitux> fade? :) [11:59] Action: ubitux trying [12:05] <ubitux> yeah e [12:06] <cone-967> ffmpeg.git 03Luca Barbato 07master:0b70fb1d518c: dsputil: convert remaining op_pixels_func [12:06] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:f88f7b86a962: Merge commit '0b70fb1d518cbd796545fd6eef02772cd0d892c7' [12:14] <cone-967> ffmpeg.git 03Diego Biurrun 07master:56632fef65c0: libopencore-amrnb: cosmetics: Group all encoder-related code together [12:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:4ff0e63a015a: Merge commit '56632fef65c0cb6946ed3648ded3d7b82e5c5c17' [12:19] <cone-967> ffmpeg.git 03Diego Biurrun 07master:e6bda9a9fd86: libopencore-amr: Conditionally compile decoder and encoder bits [12:19] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:ab213b5360e8: Merge commit 'e6bda9a9fd86505927a2e095e495eae104860701' [12:26] <cone-967> ffmpeg.git 03Diego Biurrun 07master:8837f4396a1a: libopencore-amrwb: Make AMR-WB ifdeffery more precise [12:26] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:2220f13d8227: Merge commit '8837f4396a1a458a0efb07fe7daba7b847755a7a' [12:34] <cone-967> ffmpeg.git 03Diego Biurrun 07master:870a0c669e53: build: The libopencore-amrnb encoder depends on audio_frame_queue [12:34] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:4b7c81c3e059: Merge commit '870a0c669e536d56c6325d84f65e34c53792398e' [13:00] <cone-967> ffmpeg.git 03Luca Barbato 07master:aa11cb79318b: build: make audio_frame_queue a stand-alone component [13:00] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:82e82fa2e5bb: Merge commit 'aa11cb79318baa3415d553424ba378f6c62e1f9b' [13:11] <cone-967> ffmpeg.git 03Matti Hamalainen 07master:7f3624dc81a1: svq3: unbreak decoding [13:11] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:0789374ed39a: Merge remote-tracking branch 'qatar/master' [13:19] <divVerent> haha, I have the very oppposite bug... so maybe this regression fixes my issue :P [13:19] <divVerent> I just couldn't figure out why -vn doesn't prevent album art from being copied, and also could not find a -map_metadata option to do this [13:22] <divVerent> BTW, I have four features I cannot "really" enable in ffmpeg... is this normal? [13:22] <divVerent> http://paste.debian.net/235197/ [13:22] <divVerent> frei0r and libopencv only really can be used with ffmpeg if they are built without ffmpeg support, or we get a cycle :P [13:22] <divVerent> (yes, I know, cycles can actually be worked around, but not when packaging software) [13:23] <divVerent> libflite and libilbc apparently don't work, any idea about these? as in, which version of these is ffmpeg known to work with? [13:24] <divVerent> I noticed these when making an arch linux PKGBUILD that tries to enable virtually everything :P [13:25] <divVerent> the cycle can be resolved when dynamically linking by first building ffmpeg without the dependency, building the dependency with ffmpeg support, then building ffmpeg with it [13:25] <divVerent> just can't apply this to packaging [13:25] <divVerent> just find it funny that ffmpeg supports using libraries that in turn depend on ffmpeg in the usual configurations :P [13:27] <ubitux> libflite is very recent afaik [13:27] <ubitux> ask saste when he's back [13:28] <ubitux> divVerent: btw, don't want your name in the showspectrum copyright? [13:28] <ubitux> not that is has much legal value in comparison to the git history, but well [13:30] <divVerent> ubitux: don't care much for that, my changes aren't that big [13:30] <divVerent> the infrastructure in there is not by me, I only did the drawing part [13:31] <divVerent> so for my taste, the git history's info alone is ok [13:31] <divVerent> if you want to add me to the file, go ahead, though [13:35] <ubitux> ok :) [13:36] <cone-967> ffmpeg.git 03Clément BSsch 07master:f894246304b3: lavf/showspectrum: add divVerent in the (c) for his recent work on the filter. [13:36] <divVerent> yes, in retrospect, I should have added that... it's better for ffmpeg to have clean copyright notices, even on minor changes... one too many is better than one too little [14:09] <durandal_1707> ubitux: so you have no opinion on lua filter? [14:11] <ubitux> i would be happy to have a lua filter, but actually i think what is lacking is not a lua filter, but a lua filtergraph architecture [14:11] <ubitux> typically if you want to script an editing scenario [14:12] <ubitux> basically, a "lua binding" for the lavfi api [14:12] <durandal_1707> and how it should be implemented? [14:12] <ubitux> i have no idea [14:15] <durandal_1707> ubitux: not as filter that does filtergraph? [14:18] <ubitux> might be interesting [14:18] <durandal_1707> or ffmpeg script [14:19] <durandal_1707> ? [14:20] <ubitux> i'm not sure [14:20] <ubitux> you should look for common workflow in the anime ripping world [14:20] <ubitux> we talked about a while ago about that stuff [14:21] <ubitux> it was things like detect the interlacing (so calling filters and analyze results), then do some processing, maybe do some ad cutting in the middle, or change the de-interlacing method for the op/ed part [14:22] <ubitux> whatever the solution, imo, this kind of operation should be solvable easily [14:22] <ubitux> you can imagine processing some filters on part of the video too [14:23] <ubitux> anyway, that's what the lua scripting should be used for [14:23] <ubitux> that might be different from what you originally had planned though (aka a geq-like in lua) [14:23] <durandal_1707> what? no [14:23] <ubitux> ah i misunderstood then, sorry [14:24] <durandal_1707> lua filter would just create filtergraph using own stuff [14:24] <durandal_1707> because using eval for if/else is slow and ugly [14:25] <durandal_1707> and idea is that "lua filter" would take input as file script [14:25] <durandal_1707> (i do not use ed for programming ....) [14:26] <ubitux> a script.lua? [14:26] <ubitux> that reminds me there was some security concerns about this [14:27] <ubitux> (about accessing any FFmpeg C functions from a user PoV) [14:27] <durandal_1707> isn't that implementation problem [14:27] <ubitux> maybe [14:28] <durandal_1707> doesn't avisynth/vapoursynth do just that? [15:14] <ubitux> no idea, i never used that [15:26] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:51513b98d62d: h264/svq3: stop using draw_edges. [15:26] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:71ae8d50b283: x86/dsputil: fix compilation when h263 decoder/encoder are disabled. [15:31] <durandal_1707> so in the end ffmpeg logo on web page is strange [15:48] <Compn> durandal_1707 : you should see mplayer's scriptable 'run' command if you want security trouble ... :) [15:50] <durandal_1707> that just execute shell? [15:59] <Compn> yea run sudo rm -rf / [16:02] <Compn> michaelni : thanks for looking out for mplayer compilation : [16:02] <Compn> :) [16:14] <durandal_1707> michaelni: ffmpeg -ss 3 -i /tmp/1.nut gives assertion if nut is ffv1 video with single frame [17:26] <cone-967> ffmpeg.git 03Paul B Mahol 07master:1f4ab61b746b: lavc/cdxl: clear palette before reading it [17:31] <BBB> does anyone have qemu-system-arm set up on his system and want to test a series of patches for me? [17:37] <av500> or a real ARM system [17:37] <av500> they exist.... [17:37] <BBB> that would be nice also [17:37] <BBB> anyone? [17:42] <michaelni> I still have that old beagle board that neither thilo nor me could get fully working with fate, its not fun to use but should do for a single fate test (compared to all fate tests which it just doesnt seem to pass without failing somehow) [17:43] <michaelni> i dont recommand that though if theres any other solution [17:44] <cone-967> ffmpeg.git 03Paul B Mahol 07master:85921499c747: cdgraphics: set palette to zero too [17:44] <michaelni> but if you or another ffmpeg devel wants the board iam happy to put it in a box and send it ... [17:47] <BBB> I have enough arm devices if I wanted to run fate myself [17:47] <BBB> I dont' :) [17:47] <BBB> they're all a pain [17:49] <BBB> e.g. current version (http://pastebin.com/x4Rcd4RD) may fix arm, but I'm not sure [17:49] <BBB> it would take me many hours to hook up any arm device and run a test; so if anyone has arm set up, that would be a lot quicker [18:07] <cone-967> ffmpeg.git 03Paul B Mahol 07fatal: ambiguous argument 'refs/tags/n0.7.14': unknown revision or path not in the working tree. [18:07] <cone-967> Use '--' to separate paths from revisions [18:07] <cone-967> refs/tags/n0.7.14:HEAD: cdgraphics: set palette to zero too [18:09] <durandal_1707> ^ can something be done with this failure? [18:10] <michaelni> durandal_1707, talk with thresh [18:11] <BBB> yeah and qemu setup doesn't work either on mac [18:11] <BBB> bah [18:12] <BBB> slightly better version: http://pastebin.com/EDT4BWV8 [18:12] <BBB> still untested, but maybe it works [18:12] <av500> one will never know... [18:12] <BBB> I s'pose [18:13] <BBB> I believe one of my computers has an android sdk installed somewhere, and I was able to connect to my phone at least at some point for some testing [18:15] <durandal_1707> michaelni: anyway, I did not touched 0.7.14 and i dot plan to, also i dont see it in web interface [18:18] <michaelni> durandal_1707, yes, cone has a bug with tags it posts nosnense, thresh and j-b are cones masters, ask them what is wrong with code / if it can be fixed [18:19] <michaelni> s/code/cone/ [18:20] <michaelni> the repo is always fine / unaffected its just what is printed to IRC thats a bit off [19:04] <durandal_1707> are there files/samples where stereo3d filter is useful [19:04] <durandal_1707> i plan to port this filter [19:08] <durandal_1707> michaelni: gonna fix that assert in nut or should i fill bug? [20:14] <durandal_1707> -filter_complex "showwaves=mode=line:size=976x240[out3],split[one][two],[one]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[out1],[two]pad=976:720[out2],[out2][out1]overlay=720:0[x],[x][out3]overlay=0:480" [20:19] <ubitux> does it work? [20:19] <ubitux> (doesn't seem so here for unknown reason) [20:20] <durandal_1707> what error you get? [20:20] <ubitux> "Open inputs in the filtergraph are not acceptable" [20:21] <ubitux> but i'm not using filter_complex here [20:21] <durandal_1707> you should [20:21] <durandal_1707> thought i dunno what that error means [20:21] <durandal_1707> i display it with ffmpeg sld muxer [20:21] <durandal_1707> *sdl [20:22] <ubitux> oh good idea the ffmpeg sdl [20:22] <durandal_1707> ffplay is in bad shape [20:22] <ubitux> will try, after i figure out the problem [20:27] <ubitux> it's strange.. [20:27] <durandal_1707> blend=all_expr=if(eq(mod(X\,2)\,mod(Y\,2))\,A\,B) [20:33] <ubitux> ok get it. [20:45] <ubitux> Open inputs in the filtergraph are not acceptable. Orphan inputs: "in" "(null)" [20:45] <ubitux> erm. [20:46] <ubitux> actually, just "(null)" [20:50] <ubitux> durandal_1707: ./ffplay -f lavfi "amovie=niea7-op.mp3,showwaves=mode=line:size=976x240,split[waves1][waves2]; [waves1]split[a][b]; [a]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[hist]; [b]pad=976:720[padded]; [padded][hist]overlay=720:0[over]; [over][waves2]overlay=0:480" [20:50] <ubitux> is that the same? [20:50] <ubitux> maybe a split=3 would make more sense [20:52] <ubitux> so ./ffplay -f lavfi "amovie=niea7-op.mp3,showwaves=mode=line:size=976x240,split=3[a][b][c]; [a]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[hist]; [b]pad=976:720[padded]; [padded][hist]overlay=720:0[over]; [over][c]overlay=0:480" [21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:695a766bff4c: ff_read_timestamp: check stream_index before using it as array index [21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:6cd650dbd2be: ff_gen_search: Fix finding the maximum timestamp in a really small file [21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:e39821a65e42: nutenc: only write an index if there are syncpoints [21:04] <durandal_1707> michaelni: what? seek does not return -1 any more for lavf-ogg? [21:05] <durandal_1707> :) [21:11] <michaelni> yes it matches asf/mpg in that now [21:11] <michaelni> or nut [21:18] <cone-967> ffmpeg.git 03Paul B Mahol 07master:480ddf2bc9c0: lavfi/histogram: overlay display mode for levels histogram mode [21:50] <Compn> durandal_1707 : youtube outputs some 3d stuff ,like side-by-side samples [21:50] <Compn> stereo3d can convert between side by side, top/bottom, and red/blue (amongst a bunch of other colors for analgylph 3d) [21:50] <Compn> also frame interleaved or something [21:50] <Compn> probably [21:53] <durandal_1707> and there is sample, with which filter is used? [21:55] <durandal_1707> i can use any video, but would like to have one where output will make more sense [21:57] Action: durandal_1707 should probably google it [22:05] <Compn> durandal_1707 : oh, well you take a side by side and use stereo3d to convert it to red/blue i guess [22:05] <Compn> i think i wrote examples on the wiki [22:05] <Compn> one sec [22:06] <Compn> oh no [22:06] <Compn> i wrote this one [22:06] <Compn> http://wiki.multimedia.cx/index.php?title=How_to_make_a_3d_movie_with_ffmpeg [22:06] <Compn> this page has mencoder options > http://www.mtbs3d.com/phpbb/viewtopic.php?f=27&t=4521 [22:07] <Compn> er page 3 of that does > http://www.mtbs3d.com/phpbb/viewtopic.php?f=132&t=4521&start=30 [22:31] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:e2789d3e3399: split out ff_hwaccel_pixfmt_list_420[] over individual codecs. [23:05] <wm4> what is the state of a demuxer after a failed av_seek_frame()? [23:05] <wm4> in particular, what does av_read_frame() return? [23:10] <wm4> I think it_'s reproducible with ffplay and a .flac file [23:11] <wm4> hit the cursor down and cursor left keys at the same time a few times at the beginning of the file [23:11] <wm4> and you get errors like [flac @ 0x9106220] invalid sync code [23:11] <wm4> and it only happens after "error while seeking" is printed [23:12] <saste> llogan, any news from mike? [23:14] <llogan> saste: no reply yet. [23:14] <llogan> although i would like to get started on it soon [23:15] <durandal_1707> on what? [23:17] <wm4> hm I guess a ticket will generate more attention [23:17] <llogan> i asked him about using his wiki again for the GSoC idea list and if he didn't mind keeping up with any new requests for edit permissions [23:18] <michaelni> wm4, the state after failed seek is undefined (that is how it is currently, not neccesarily what we want it to be) [23:18] <durandal_1707> llogan: you need permisison to edit gsoc page? [23:18] <wm4> michaelni: well, it appears for some formats it works, and for some (flac) it doesn't [23:19] <wm4> michaelni: and by "works" I mean unconditionally resetting the decoder, and then calling av_read_frame() and feeding that to the decoder [23:19] <durandal_1707> flac use generic seek [23:19] <llogan> durandal_1707: you need an account to edit http://wiki.multimedia.cx ...or maybe an account with editing permissions, either way mike has to manually do something [23:19] <michaelni> wm4, reset is easy the hard part is to be in the same state as before the failed seek was attempted [23:20] <wm4> michaelni: so I would suggest: seeking before start of the file -> clamp seek position to 0 (and always succeed seeking), and seeking past the end of the file => return EOF (and let av_read_frame() return EOF too) [23:20] <michaelni> llogan, you can just use trac wiki [23:21] <llogan> i know, and i like the trac wiki for the most part, but saste mentioned that a more strict permission might be good in this case [23:21] <llogan> last minute troll vandalism, etc [23:22] <durandal_1707> trac wiki can't have finer permissions? [23:22] <michaelni> can be done somehow AFAIK but i dunno how [23:23] <llogan> probably, but michael has better things to deal with [23:23] <llogan> if i don't hear from him by tomorrow i'll just use our wiki. [23:25] <durandal_1707> i could do it... [23:32] <saste> http://trac.edgewall.org/wiki/TracFineGrainedPermissions [23:32] <saste> so yes it is possible [23:32] <wm4> how can I seek with "ffmpeg" again? [23:32] <durandal_1707> wm4: only once -ss [23:33] <saste> i have no objection into using our wiki if we can implement dev-only access on the gsoc page [23:33] <wm4> ok... that's not enough [23:34] <durandal_1707> saste: it appears doable, but from friendly [23:34] <durandal_1707> *far [23:36] <saste> durandal_1707, what do you mean, unfriendly towards whom? [23:36] <durandal_1707> the trac [23:37] <saste> gsoc page should be edited only by candidate mentors or developers [23:37] <saste> there is no much point into allowing a random user to edit it [23:38] <saste> we never did like that in the past years [23:41] <durandal_1707> i mean cofiguring finer permissions for trac are unfriendly. [23:42] <durandal_1707> llogan: why you plan to put for gsoc? [23:43] <durandal_1707> *what [23:43] Action: durandal_1707 needs to sleep [23:44] <llogan> i was going to copy most of last year's and let interested devs/mentors edit it. then i was going to make sure it is formatted well and in a good layout before google views it [23:44] <llogan> scrolling marquees, page counter, some flash, gifs, etc. [23:45] <durandal_1707> flash? [23:45] <llogan> bad joke [23:45] <llogan> durandal_1707 needs to sleep more [00:00] --- Tue Feb 19 2013

1 0

ffmpeg.log.20130218
by burek 19 Feb '13

19 Feb '13

[02:36] <llogan> relaxed: were you able to work on your compile guide? [03:09] <t4nk422> hi all, I'm trying to use ffmpeg to convert an FLV file to MP4. I managed to do this successfully for some FLV files, but other seem to be failing. The encoding on the ones that fail are saying "flv1, yuv420p". I'm running on Ubuntu and followed the following link to compile ffmpeg: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide. Am I missing a flag or a codec that I need to add, any assistance or pointers would be very [03:09] <t4nk422> My command line output is: http://pastebin.com/pJVyfirD [03:16] <llogan> t4nk422: never seen that before. can you supply a sample file? [03:17] <llogan> also please include the command and output for one that does work [03:19] <t4nk422> Yeah sure, let me put that together [03:26] <t4nk422> It looks like the one that was working is encoded differently (see pastebin). The working file plays in VLC & Movie Player on Ubuntu. The notworking file ONLY plays in VLC, would it be a codec I'm missing on my system that might be causing this? http://pastebin.com/Gv0eLS7N [03:27] <t4nk422> The links listed in the pastebin, should make those random sample files playable [03:30] <Tvilling> speex and flv1 vs. vp6 and mp3 [03:32] <llogan> t4nk422: looks like the audio stream is the source of the issue, AFAIK. probably worthy of a bug report. [03:33] <llogan> ah, i see. i think you need --enable-libspeex. [03:34] <t4nk422> ah thanks, I'll give that a try [03:34] <llogan> i assumed ffmpeg can decode speex natively, but i was wrong and that error isn't very informative [03:34] <llogan> https://ffmpeg.org/trac/ffmpeg/ticket/1668 [03:35] <llogan> so install libspeex-dev and recompile ffmpeg with --enable-libspeex [03:35] <Tvilling> Try running it more verbosely next time. [03:36] <llogan> the verbose output was not very useful in this case. [03:36] <t4nk422> ah ok, thanks very much for the help to both of you, installing that now [03:38] <llogan> t4nk422: refer to the "Upgrading FFmpeg" section of the guide [03:38] <Tvilling> llogan: The debug level weren't helpful as well? [03:39] <llogan> not for the general user. [03:39] <Tvilling> Eh. [03:43] <llogan> t4nk422: i updated the guide so it will now install libspeex. [08:04] <defaultro> llogan, the slowmovideo result was awesome :) [14:30] <jarno> ...using libav* and libx264 to encode and mux a video stream received from a camera and I get "nvalid DTS: PTS is less than DTS"...I'm setting PTS in my code and it indeed is monotonically increasing.... [14:30] <jarno> ...so what's the cure? [14:32] <jarno> ...anyone? [14:44] <Mavrik> jarno: obviously you need to set DTS on output packets as well. [14:47] <jarno> ...I took a quick peak related to this issue and I (like others) thought that the DTS was something generated by the libx264 [14:47] <jarno> but, then again, I must have been wrong [14:47] <Mavrik> it is [14:47] <Mavrik> PTS and DTS is generated by libx264 [14:48] <Mavrik> so if you're getting that error [14:48] <Mavrik> you're doing something wrong with them after encode [14:48] <jarno> in my code I'm setting PTS for the frame but not the DTS.... [14:48] <jarno> ...since I thought it was not necessary to set the DTS.... [14:49] <Mavrik> I see, [14:49] <Mavrik> well PTS and DTS of output packets are set by libx264 encoder [14:50] <Mavrik> you need to make sure your frames going to the encoder have proper PTS [14:50] <Mavrik> are you by any chance trying to set PTS on encoded frames? [14:50] <jarno> I'm setting PTS on the frame that I give to the encoder [14:51] <jarno> ...i.e. the frame that I want to be encoded... [14:52] <Mavrik> well, step through the code with a debugger and find out where do your PTS change [14:53] <jarno> ...okis...will do...however, I'm pretty much following the decoding_encoding example.... [14:54] <jarno> ...in any case, thanx for the hint... [14:54] <JEEB> also, the reason for the error is simple -- PTS never can be smaller than DTS [14:54] <JEEB> DTS = when is this frame decoded [14:54] <JEEB> PTS = when is this frame shown [14:54] <JEEB> you cannot show it before it's decoded :P [14:55] <JEEB> but yes, if you give libx264 valid PTS values, it should give you valid PTS and DTS [14:55] <Mavrik> yeah [14:55] <Mavrik> jarno: basically, libx264 should never give you invalid PTS/DTS pairs, so you're breaking them somewhere in your code before passing them to the muxer [14:56] <Mavrik> unless you have broken/buggy versions of libav*/x264 [14:56] <Mavrik> but that's extremely unlikely [14:58] <jarno> ...ok...I'm pretty sure that I'm the cause of the problem since I'm new to this... [14:58] <Mavrik> so I suggest you first just output PTS/DTS pairs right after encoding to see if the order there is ok (as JEEB said, DTS is ALWAYS <= PTS) and then follow your code through for changes [14:59] <JEEB> remember to output both nominator and the denominator (timescale and the pts value) for all cases [14:59] <jarno> ...ok...thanx guys!!!!!!! [14:59] <JEEB> printf debugging om nom nom [15:00] <Mavrik> ^^ [15:00] <jarno> ...okis...I'll output the whole stuff.... :) with a bit of luck I can get the initial version of this thing running today....!!!!! [15:06] <brx_> im using an ffmpeg compiled for arm [15:06] <brx_> on one device: ffmpeg -i a.jpg returns ifo on the file [15:07] <brx_> on another device the same command returns "No such file or directory" [15:07] <brx_> it does the same for png files, why is this? its the same ffmpeg binary on both devices [15:07] <JEEB> sounds like a file rights issue [15:07] <brx_> ahh i see [15:10] <brx_> JEEB, i did chmod 777 on a.png, same error [15:10] <brx_> im running the process as root btw [15:10] <JEEB> is the ffmpeg binary run at all btw? [15:11] <brx_> yes, it runs fine ffmpeg outputs the "file not found" error [15:11] <JEEB> ok [15:11] <brx_> also, when i try to overay a png using -VF it compains about the file [15:11] <brx_> the same command works on another device [15:11] <JEEB> I do remember someone on ARM/android having fun with rights and group ownership and other stuff [15:12] <brx_> this is a major setback for me, the app im writing depends on overlaying an image onto a video [15:13] <JEEB> see the code for what exactly is being checked for and how far the whole thing gets? [15:13] <JEEB> and why it errors out [15:13] <JEEB> that's all I can think of [15:14] <brx_> ok thanks for the suggestions JEEB [15:14] <Mavrik> usually strace helps with those problems [15:16] <brx_> Mavrik, ok i will copy the console output now [15:28] <brx_> Mavrik, JEEB http://pastebin.com/SA8ea0rF [15:28] <brx_> both those commands work on nexus 7 but not my htc desire. both devices are rooted and im in the terminal as root [15:30] <JEEB> first try more verbose output from ffmpeg, second of all, start tracing to th error through the source code :P [15:30] <JEEB> also do note that there have been at least two major releases of ffmpeg since that version [15:30] <JEEB> or three [15:30] <brx_> ! ahh ok [15:30] <JEEB> no, four [15:30] <brx_> shit [15:31] <JEEB> 0.10, 0.11, 1.0, 1.1 [15:31] <Mavrik> also, make sure your sdcard is actually mounted. [15:31] <Mavrik> and not plugged into the computer [15:31] <Mavrik> (which would cause difference in behaviour from Nexus 7 to Desire) [15:31] <brx_> it is mounted, i can do cat /sdcard/image.png and it outputs the file [15:31] <brx_> cable in or out, same errors [15:32] <JEEB> I'm pretty sure that this is similar to that one other person's problems with rights on android devices before, but that possibly was already sorted over at the newer releases [15:32] <JEEB> also without this person actually going there and checking where and why that error occurs we can't say anything [15:32] <brx_> it was a bitch getting ffmpeg binary in the first place, myb thats why i went with an older version :s [15:33] <brx_> thanks for the help ayway guys, im going to try and build the latest version of ffmpeg [15:39] <Mavrik> well as I said [15:39] <Mavrik> check strace and see what goes wrong [15:46] <brx_> ok im on it now thanks [15:58] <shevy> I have a .flv file here, which has Audio codec aac [15:59] <shevy> I would like to snatch the audio stream and dump it somehow to a file [15:59] <shevy> it however tells me: Encoder (codec id 86017) not found for output stream #0.0 [15:59] <shevy> is there perhaps a simplified way how to just split up audio and video data into separate streams? [15:59] <shevy> I don't even know what aac exactly is, whether there could be a .aac file or not [16:15] <shevy> relaxed hmm ok [16:16] <shevy> I first tried to convert a .flv into a .mp3 actually [16:16] <shevy> Here is the raw pastie: http://pastie.org/pastes/6213332/text [16:19] <relaxed> Your ffmpeg version is very old. Try, ffmpeg -i test.flv -vn -acodec copy output.m4a [16:22] <shevy> aha interesting, that created a file called output.m4a [16:23] <shevy> now I seem to have an audio stream, thanks relaxed [17:23] <terr1> Are there any x264 best practices for calculating which profile (baseline/main/high) and level (1-5.1) to use from a given input video source file? [17:26] <klaxa> mhh i think profile depends more on what level of compression you want to have in your output file [17:27] <klaxa> i don't know about the level parameter that much myself, i think this should give a general idea though? http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Levels [17:28] <klaxa> preset and crf are important factors for the resulting output too, in general have a look at this guide http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [17:29] <terr1> klaxa, ok thanks I will read thoes [21:32] <Sander^home> DOes it exist a standar for recording video in a browser yet? [21:32] <Fjorgynn> recording video in a browser? [21:33] <Sander^home> Fjorgynn, Yes.. To do it without flash. [21:34] <Fjorgynn> no [21:34] <Fjorgynn> different for each browser [21:34] <Fjorgynn> webm is one of them [21:36] <Sander^home> Fjorgynn, Do you have any link on how to do it in html5? [21:38] <Sander^home> Fjorgynn, in webkit eg? [22:41] <teratorn> anyone can recommend an image viewer application that can handle raw formats? [22:43] <ubitux> ffplay [22:43] <teratorn> ubitux: I had a sneaking suspicion... [22:44] <durandal_1707> raw formats? [22:45] <durandal_1707> ffplay is bad because of sdl only output ...., and image viewer should try hard to display correct color [22:45] <teratorn> durandal_1707: YUV420P, specifically... assuming adjacent plane alignment [22:46] <teratorn> perhaps I can use image magick [22:46] <durandal_1707> ffplay can display yuv420p just fine (though i think it messes up color range sometimes, i dunno what is workaround, -vf colormatrix?) [22:47] <durandal_1707> image magick is so memory hungry, last time i used it ... [22:47] <durandal_1707> generaly there is rawvideo demuxer [22:48] <durandal_1707> and with it you can play raw frames with mplayer1/2/mpv [22:49] <teratorn> i really wish there was some kind of text-based standard for YUV images [22:49] <teratorn> like there are for RGB [22:49] <durandal_1707> there is, it just some codecs use different colors, so you use colormatrix then .... [22:50] <llogan> i wonder if graphicsmagick is a memory hog too [22:50] <durandal_1707> so you will get correct colors with mplayer2/mpv [22:50] <teratorn> durandal_1707: there is? what? [22:51] <durandal_1707> teratorn: different color spaces [22:51] <durandal_1707> BT.709, BT.601, SMPTE-240M, FCC, ... [22:52] <durandal_1707> it really depends on your source [22:54] <durandal_1707> and i dunno if imagemagick viewer displays correct color [22:55] <durandal_1707> it certainly does not have opengl output IIRC [22:55] <durandal_1707> _if_ opengl is relevant at all in your scenarion... [22:57] <teratorn> nah, i'm just debugging a thing that makes test frames [22:57] <teratorn> and need to dump/view them [22:57] <durandal_1707> but there is bunch of application that use lavc/lavf/swscale to display images ... [00:00] --- Tue Feb 19 2013

1 0

ffmpeg-devel.log.20130217
by burek 18 Feb '13

18 Feb '13

[00:20] <nevcairiel> michaelni: avpriv_color_frame asserts when using a hwaccel in h264, pixel format is not planar [00:25] <nevcairiel> michaelni: i sent a small patch about that [00:35] <cone-323> ffmpeg.git 03James Almer 07master:a56fd9edab57: lavu: Fix checkheaders for x86/emms.h [00:35] <cone-323> ffmpeg.git 03Hendrik Leppkes 07master:1d6f6ff4d9fa: h264: don't initialize missing pictures when using a hwaccel [00:44] <cone-323> ffmpeg.git 03James Almer 07master:2e988fd68964: lavc/tta: Use a safer check for encrypted streams [00:44] <cone-323> ffmpeg.git 03James Almer 07master:b8bb661dab8e: Changelog: Add entry for encrypted TTA stream decoding [01:55] <Compn> j-b : how goes the bittorrent demuxer? :P [01:55] <Compn> j-b : bonus points if you have it implemented into ffmpeg. that way more players can utilize ffmpeg code :) [01:56] <j-b> Compn: you _really_ think the issue is the demuxer or protocol paet? [01:56] <j-b> part? [01:56] <j-b> Of course the issue is the UI, the management and the downloading services [01:56] <Compn> hmm? i mean the overall project [01:57] <Compn> i dont know which part is the difficult . probably cache, demuxer quirks (mov/avi index at end of file) [01:57] <Compn> implementing an entire bt client ... [01:58] <Compn> j-b : but is my idea of putting it in ffmpeg good or bad ? [01:59] <Compn> or its already started in vlc ? [01:59] Action: Compn is slow with the ideas today [02:00] <Compn> j-b : its not important, just another dumb idea from me :) [02:02] <j-b> Compn: so far, we've done nothing, and we're sponsoring nothing [02:02] <j-b> so, I got no idea [02:04] <Compn> can vlc use ffmpeg's network stuff ? [02:04] <j-b> sure. [02:04] <Compn> ok [02:04] <Compn> so at least it would be possible to implement it in ffmpeg [02:04] <Compn> and vlc could use it from there [02:05] <j-b> sure [02:05] <j-b> but, once again, the hard part is NOT the networking part [02:06] <Compn> it sounds kinda similar to apple / adobe fragmented thing [02:06] <Compn> unless i'm completely missing something [02:06] <Compn> thing = http [02:09] Action: Compn keeps focusing on networking part for some reason [02:24] <iive> j-b: what is the problem with the UI? [02:24] <saste> iive, interface are usually a mess to do (correctly) [02:24] <iive> but why does it need ui at all? [02:25] <j-b> iive: well, you have media player UI, and you need to rape it, in order to have a torrent-management-system in it [02:25] <Compn> iive : limit bandwidth, specify download dir [02:25] <j-b> of course not [02:25] <Compn> j-b : cant just put it all in settings ? [02:25] <j-b> all those are trivial [02:25] <Compn> whats the hard part ? [02:25] <Compn> i mean [02:25] <Compn> what ui do you need ? [02:25] <j-b> look at Utorrent [02:25] <Compn> like utorrent status or something [02:25] <Compn> ah [02:26] <j-b> You need a playlist-like-thing to list your torrent [02:26] <j-b> because if you play just one, it will buffer for a long time [02:26] <iive> see it is network problem :P [02:27] <Compn> lolol [02:27] <j-b> moreover, the UI needs to talk to this module in a very specific way (it ain't your average demuxer) [02:28] <iive> instead of utorrent, take a look of the original bittorrent client, or even transmision... not much to click there. [02:28] <Compn> just text output? lol [02:28] <Compn> use ffmpegs ascii decoder :D [02:29] <Compn> wont be pretty [02:29] <j-b> iive: even for that, you need to control your demuxer in different way than RTSP/RTMP [02:29] <iive> the problem is elsewhere, e.g. do you want disk caching... because then you create permanent copy on your system. [02:30] <Compn> you need disk caching , for indexes at end of files :P [02:30] <j-b> So, you need a special API, and even with one-torrent-at-a-time, you need status that is a bit more than "buffering" [02:30] <iive> the biggest problem of course it bandwidth, you need to both download blocks in realtime playback and download blocks in random order to keep the seeding balanced. [02:31] <iive> ...in order for realtime playback... [02:31] <Compn> i dont think the implementation needs seeding balance [02:32] <Compn> its the same thing anyways, in order or random [02:32] <Compn> still the same amount of data [02:33] <iive> no it is not. torrent always uses random. [02:33] <j-b> you can [02:33] <Compn> torrent can be configured to set priority to rares pieces, starts/ends of files, or download in order [02:34] <iive> it is same amount of data, but if everybody have the first 10% of the movie and only 1 seed have the rest, you will get a lot worse speed when trying to get something from the 90% [02:34] <Compn> its the same speed! [02:34] <iive> no. for the 10% you have 10 seeders, so you get 10x speed. [02:35] <Compn> until everyone catches up, then its the same speed [02:35] <iive> and if you use it in video content, you are most likely to start from the start :P [02:35] <j-b> in-order is possible [02:35] <j-b> but it is bad [02:36] <iive> yes, it is bad for the swarm. [02:36] <Compn> doesnt miro already do this btw ? [02:36] <Compn> and isnt it based on vlc... [02:37] <Compn> http://www.getmiro.com/download/features/ [02:38] <Compn> or is it just a mix of the two. i dunno never used it [02:54] <highgod> Hi, all I want to ask a question. Can ffmpeg be compiled using mingw64 and create 64bits libs and dlls? thanks [03:09] <Compn> highgod : yes, it can [03:10] <Compn> oh, maybe not [03:10] <Compn> we dont have a fate box for it [03:11] <Compn> er [03:11] <Compn> yeah [03:11] <Compn> highgod : http://ffmpeg.zeranoe.com/builds/ [03:11] <Compn> zeranoe has 64bit mingw ffmpeg builds [03:11] <Compn> and instructions for doing it ^^ [03:43] <cone-323> ffmpeg.git 03Reinhard Tartler 07release/0.5:deb650c69231: Release notes and changelog for 0.5.10 [03:43] <cone-323> ffmpeg.git 03Michael Niedermayer 07release/0.5:0360dbefad8a: Merge remote-tracking branch 'qatar/release/0.5' into release/0.5 [04:24] <highgod> OK,thanks Compn. [04:29] <Zeranoe> Could anyone explain to me the license details of fdk-aac? A lot of users are asking for fdk-aac, and reading the license for this lib I find "Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions:" Which then lists a few conditions. Is there any way to load the library dynamically and still compl [04:30] <cone-323> ffmpeg.git 03Michael Niedermayer 07release/0.5:a23a3dba2544: qdm2: check array index before use, fix out of array accesses [04:30] <cone-323> ffmpeg.git 03Michael Niedermayer 07release/0.5:fee26d352a52: roqvideodec: check dimensions validity [04:30] <cone-323> ffmpeg.git 03Michael Niedermayer 07release/0.5:13093f9767b9: vqavideo: check chunk sizes before reading chunks [04:30] <cone-323> ffmpeg.git 03Michael Niedermayer 07release/0.5:d34cfb33afb8: update for 0.5.11 [04:48] <Compn> Zeranoe : got a link? [04:48] <Zeranoe> Compn: A link for the license? [04:48] <Compn> ya [04:49] <Zeranoe> Compn: https://github.com/mstorsjo/fdk-aac/blob/master/NOTICE#L29 [04:49] <Compn> Zeranoe : thats wbs's repo, maybe he can explains it [04:49] <Zeranoe> The license looks very similar to the GPL and I'm surprised FFmpeg wont allow it in a GPL build. [04:50] <Zeranoe> Compn: Explain why FFmpeg wont allow it under the GPL? [04:50] <Compn> its very similar to gpl build [04:50] <Compn> possibly You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec [04:50] <Compn> software or your modifications thereto. [04:50] <Compn> goes against gpl [04:50] <Compn> since you can charge for gpl [04:51] <Compn> not sure what a 'copyright license fee' would even be ... [04:51] <Zeranoe> Compn: If the lib cannot be included in a GPL build, would it comply with the license if FFmpeg was built to accept a fdk-aac lib file, like a .dll and use it externally when it needs it? [04:53] <Zeranoe> Compn: Since both libs allow for distribution of a binary, only thing would be the license for the fdk-aac code that FFmpeg uses to call the lib. [04:54] <Compn> you dont just want to put it in 'non-free' ? [04:54] <Compn> but i dunno, i'm no license master [04:54] <Compn> your idea sounds good [04:54] <Zeranoe> libfdk-aacenc.c is under the LGPL so it should be ok [04:55] <Zeranoe> Compn: Non-free means I cant redistribute the binary, which is my goal. [04:55] <Zeranoe> I want to provide my uses with a FFmpeg that can use the best AAC encoder lib. [04:56] <Zeranoe> users** [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:ef4c71e8f83a: lavfi/unsharp: add check on matrix x/y size values oddity [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:d2cadea3f085: lavfi/unsharp: directly access in-loop variables in apply_unsharp() [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:64e592eef26b: lavfi/unsharp: merge definition and declaration in init_filter_param() [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:89505f2c3f8e: lavfi/unsharp: add missing NULL check [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:8c85a9f046fb: lavfi/mp: drop mp=unsharp filter [11:53] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:7ca2f8b1130b: lavfi/mp: drop mp=kerndeint filter [11:56] <ubitux> can anyone tell me if the avcodec.h change in the subtitle character encoding patch is fine? [11:56] <ubitux> Nicolas seems to have some concern about it [11:56] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/139117.html [11:56] <ubitux> "Wait for [11:56] <ubitux> confirmation about the avcodec.h part though. [11:56] <ubitux> " [11:57] <ubitux> (basically +#define AV_CODEC_PROP_BITMAP_SUB (1 << 16)) [12:02] <ubitux> oh maybe he was refering to the rest though [12:05] <ubitux> well i'll resend a patch [12:48] <ubitux> wm4: subtitles character encoding, soon. [12:48] <ubitux> (finally) [12:48] <wm4> ok [12:48] <wm4> what are you going to use? [12:48] <ubitux> it's done, i'm just waiting for a last comment [12:49] <ubitux> no utf-16 yet though, Nicolas seems to be interested to working on it [12:50] <ubitux> https://github.com/ubitux/FFmpeg/compare/master...sub-recode-nofilter [12:56] <ubitux> should i increment micro or minor when adding a field to AVCodecContext? [12:56] <nevcairiel> wasnt minor for such things [12:57] <ubitux> ok [13:01] <ubitux> saste: [13:01] <ubitux> -unsharp 7d72d2ab7b7f60159c822a097e01068b [13:01] <ubitux> +unsharp d41d8cd98f00b204e9800998ecf8427e [13:01] <ubitux> :( [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:8732271e4025: lavc: mark bitmap based subtitles codecs as such. [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:f796399344a1: lavc: support subtitles character encoding conversion. [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:3a0a959dcbf8: lavfi/subtitles: split options between ass and subtitles. [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:08d149d6889b: lavfi/subtitles: support charenc option. [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:2a14b23484be: lavfi: rename vf_ass.c to vf_subtitles.c [13:04] <cone-485> ffmpeg.git 03Clément BSsch 07master:90fb3e9bee60: lavfi/subtitles: set a different description between ass and subtitles. [13:04] <ubitux> wm4: have fun ^ [13:07] <wm4> hm, vf_subtitles doesn't actually use this new flag? [13:08] <ubitux> yes it does [13:08] <ubitux> -vf subtitles=foo.srt:charenc=cp1251 [13:09] <cone-485> ffmpeg.git 03Clément BSsch 07master:b1e6b144ed92: lavc/utils: reindent in avcodec_decode_subtitle2() after f7963993. [13:09] <wm4> I mean AV_CODEC_PROP_BITMAP_SUB [13:10] <ubitux> mmh good point, maybe it should now [13:10] <wm4> and maybe codecs which decode to ass should be marked as well (in fact, AV_CODEC_PROP_BITMAP_SUB is entirely useless for now, because of backwards compatibility) [13:11] <ubitux> it's mainly meant to be used internally [13:11] <ubitux> notably to set sub->format (0 or 1) [13:11] <ubitux> or enable/disable charset re-encoding [13:14] <wm4> I still don't quite see how vf_subtitles prevents opening an image subtitle decoder as ass decoder [13:14] <ubitux> it doesn't [13:15] <ubitux> it will basically do nothing [13:15] <ubitux> first bitmap sub expected break the loop, do nothing [13:15] <ubitux> i'll add an error soon now that we have the flag [13:15] <ubitux> for backward compat, you can rely on sub->format [13:15] <ubitux> for the first decoded sub [13:23] <wm4> so I have to try to decode the first subtitle just to detect a property statically known before any decoding? [13:23] <ubitux> or use AV_CODEC_PROP_BITMAP_SUB now [13:24] <ubitux> i can't do much more for the old versions [13:24] <ubitux> any subtitles format not with the AV_CODEC_PROP_BITMAP_SUB prop is supposed to decode to ass [13:30] <cone-485> ffmpeg.git 03Stefano Sabatini 07master:894ed8fbb71d: tests: fix values for the unsharp test [13:39] <cone-485> ffmpeg.git 03Clément BSsch 07master:fe150a48a72e: lavc: fix codec_descriptor and pkt_timebase doxy. [13:39] <cone-485> ffmpeg.git 03Clément BSsch 07master:f3147917bf00: lavf/subtitles: error out in case of bitmap subtitles. [13:40] <wm4> ubitux: any news on fixing the ass packet format? [13:40] <ubitux> i'm going back to it slowly [13:40] <ubitux> i wanted to get done with the charset thing first [14:33] <cone-485> ffmpeg.git 03Anton Khirnov 07master:1f8f43a5b5e9: error_resilience: add required headers. [14:33] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:59e46ef63ab4: Merge remote-tracking branch 'qatar/master' [16:03] <cone-485> ffmpeg.git 03Nicolas George 07master:dcc73aaaa995: doc/examples: do not allocate AVFrame directly. [16:58] <cone-485> ffmpeg.git 03Clément BSsch 07master:1e860f16688b: doc/codecs: simple sub_charenc option documentation. [17:35] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07master:cf36180143b3: Only set accelerated arm fft functions if fft is enabled. [17:37] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:09ece9fa6c45: eval: print() support [17:37] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:29c8619a4977: fate: add print() to the tests of eval [17:41] <durandal_1707> anyone working on lua filter? [18:06] <cehoyos> nevcairiel: If you look at the mkvalidator output in ticket 2263, is it sufficient for you to see why libavformat does not know where to find keyframes? [18:06] <cehoyos> And if you know why: How can such files be produced: I have the same version of mkvmerge installed here and if I remux the file, FFmpeg reads the resulting file fine (seeking does not take ages) [18:07] <nevcairiel> most software will probably only read the first seekhead, and without cues seeking in mkv is butt-slow [18:08] <nevcairiel> especially because the second seekhead as at the end of the file, it appears [18:09] <cehoyos> How can I check if the seekhead is really at the end of the file? [18:09] <cehoyos> And since our sample is 11G: How can I produce such a sample? [18:09] <cehoyos> ... such a shorter sample;-) [18:10] <nevcairiel> well mkvalidator says the second seekhead is at 11568150341 [18:10] <nevcairiel> that sounds like 11g =p [18:10] <cehoyos> ok, sorry.. [18:11] <nevcairiel> no idea how to produce something like this, maybe it didnt have the ability to seek in the output and then decided to write it at the end is better then not at all [18:11] <cehoyos> Something like mkvmerge -o - ? [18:12] <nevcairiel> never tried anything like this [18:15] <cehoyos> mkvmerge -o - produces a file ./- [18:17] <cehoyos> nevcairiel: Does this link describe why mkvmerge put a second seek head at the end of the file? https://trac.bunkus.org/ticket/290 [18:19] <nevcairiel> but in this case it references the second seek head from the first [18:19] <nevcairiel> so you know where it is and can parse it directly [18:20] <nevcairiel> so i can only guess that it simply failed to reference the secondary one in the primary one [18:22] <cehoyos> So basically I have to keep the file;-( [18:26] <cehoyos> Is my comment correct? https://ffmpeg.org/trac/ffmpeg/ticket/2263#comment:8 [18:27] <nevcairiel> from what i know without having seen the file, yes [18:28] <cehoyos> Thank you! [18:41] <ubitux> the print thing in eval will be very useful [18:42] <ubitux> thx :) [19:47] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07master:fdec49cbe83f: Add gray16 as a supported v4l2 input format. [19:47] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07master:fdbe7628a957: Add yvu410 as a supported v4l2 input format. [20:21] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07master:8d0757e1079f: Revert "swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size" [20:23] <Daemon404> http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=8d0757e1079f588ed69f62… <-- protip: bug ids are not a replacement for an actual descriptio [20:23] <Daemon404> n [20:24] <Daemon404> http://ffmpeg.org/developer.html#Development-Policy [20:24] <Daemon404> #3 [20:25] <Daemon404> in this case, the bug ID still doesnt have a dexription of why the patch reverts it [20:27] <durandal_1707> cehoyos: that bug do not have reproduced and analyzed set [20:27] <Compn> Daemon404 : its reverted because it fixes the bug. i think your real question is "why is that previous change causing this swf bug" [20:27] <durandal_1707> Compn: please, stop [20:28] <Daemon404> Compn, let us name all our commits "fix bug" [20:28] <Compn> hehe [20:28] <Compn> it should be fix regression :P [20:34] <durandal_1707> swf can have mp3 which is not covered by av_get_audio_frame_duration [20:35] <cehoyos> durandal_1707: I cannot reproduce it because I don't know how to feed FlashPlayer [20:35] <durandal_1707> and how you found out reverting fixes it? [20:37] <cehoyos> This is difficult to answer because I am not sure what the deeper meaning of your question is. [20:38] <cehoyos> (But the answer would be: Because I used git bisect for some time and afaict, the original test was not just unreviewed but untested.) [20:38] <cehoyos> s/test/patch [20:38] <durandal_1707> wasn't it reproduced with ffmpeg too? eg. durationts from demuxer are set to 0? [20:39] <cehoyos> Daemon404: My patch description contains an exact description why the patch was needed [20:39] <Daemon404> where [20:39] <Daemon404> its not in the commit message. [20:39] <Daemon404> how did you test this patch [20:39] <Daemon404> why does it fix it [20:39] <Daemon404> etc [20:40] <durandal_1707> also coverage for muxing lot of codecs in swf or any other container is small [20:42] <durandal_1707> cehoyos: but bug report also have also near to 0 useful info [20:42] <durandal_1707> so user might not use mp3 in swf at all [20:43] <durandal_1707> bug report is just incomplete, and fix/revert is (even if correct) pure guesswork [20:46] <cehoyos> durandal_1707: No, the duration is ok with ffmpeg -i file.swf [20:46] <durandal_1707> lol the ticket have enough info [20:47] <durandal_1707> it is just hidden in stupid log file [20:47] <durandal_1707> cehoyos: that can be from parser ... [20:48] <cehoyos> Daemon404: I have a suggetion; Could you find out which sample / bug report / hypothetical case was fixed by 9fb7e14? [20:49] <Daemon404> if it switches to a new config and fails, instead of just giving up, it will revert to the old one [20:49] <Daemon404> its pretty obvious [20:49] <Daemon404> also piitning out random Libav commits does not magically make it OK. [20:49] <kierank> oh nice, someone just *had* to try putting latm in mkv [20:49] <cehoyos> durandal_1707: I completely agree that posting logs is significantly more useful than attaching, but I tried to help the user because it was his first report afair. [20:50] <Daemon404> the amount of english comprehension in this channel is painful. [20:51] <cehoyos> Daemon404: (Do you honestly think it is a random commit? Because if yes, this whole discussion makes even less sense.) You misunderstand: I don't need an explanation what the patch does, I probably don't understand it anyway, and if I would be interested, looking at it would probably be sufficient, the point is *why* [20:52] <Daemon404> you need an explanaion of why it was reverted. "fixes bug X" with ZERO technical details is not valid [20:52] <durandal_1707> cehoyos: you should reopen bug and wait for poster to reply that issue have really been fixed (you cant or did not tried to reproduce it after all) [20:52] <cehoyos> I disagree, but that is probably not the point... [20:52] <cehoyos> durandal_1707: I disagree because OP's often never react and as you know we have too many open reports. [20:53] <cehoyos> (In this specific case, no testing is necessary, but that is beside the point.) [20:53] <cehoyos> Daemon404: Wow, I just wanted to leave because of you being quite offensive;-) [21:04] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/0.11:381e3e7e44f9: Revert "swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size" [21:04] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:98d06b046dfe: Revert "swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size" [21:04] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:6407800521d2: Revert "swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size" [21:11] <Compn> kierank : no not more latm stuff :V [21:12] <kierank> :) [21:13] <durandal_1707> there is some other latm open bug? [21:42] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07fatal: ambiguous argument 'refs/tags/n0.5.11': unknown revision or path not in the working tree. [21:42] <cone-485> Use '--' to separate paths from revisions [21:42] <cone-485> refs/tags/n0.5.11:HEAD: Revert "swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size" [22:14] <durandal_1707> michaelni: i sent patch for lavfi/noise to stop using rand/srand - i looked at output and quality of noise looks almost same [22:18] <peper03> Is a patch to add passing PRIVATE_STREAM_2 MPEG2 packets to the user likely to be accepted? I would only add a new codec ID and pass the packet as-is. There doesn't seem to be much point with a decoder as it's a pure data packet and from what I can see, used for different purposes (DVD, Dreamcast - Sofdec). [22:19] <peper03> I couldn't see any other data-only codec IDs so I can't base my patch on other implementations. [22:21] <durandal_1707> peper03: there is patch on ml? [22:23] <peper03> durandal_1707: Not yet. I've basically got everything prepared (I think) but I just wanted to check a couple of basic points first as I'm not subscribed to the ml yet etc. etc. [22:24] <peper03> Did I understand the guidelines correctly, indentation due to an 'if' should be submitted separately? [22:24] <durandal_1707> i'm not mpegts expert so can't really help [22:24] <durandal_1707> peper03: yes, easier to review [22:25] <peper03> durandal_1707: Sure. No problem. Again, just wanted to ensure I hadn't misunderstood. [22:26] <peper03> durandal_1707: Are you aware of any other data-only packets that are currently handled? [22:26] <peper03> Handled = passed back to the caller? [22:28] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:dece584a639c: h264: avoid calling get_format() multiple times [22:29] <durandal_1707> peper03: i'm not qualified enough to answer such question [22:30] <peper03> durandal_1707: Ok, np. Thanks. [22:31] <durandal_1707> michaelni: so what I should use instead of RAND_MAX? [22:32] <michaelni> UINT_MAX IIRC [22:34] <durandal_1707> michaelni: is it relevant that rand returns int and lfg unsigned? [22:37] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07master:259603b91767: h264: don't initialize missing pictures when using VDPAU. [22:38] <michaelni> it could make a difference depending on how its used [22:42] <durandal_1707> i replaced RAND_MAX with UINT_MAX and nothing chaged, still noise on screen [22:43] <durandal_1707> rand(3) here returns positive numbers [22:44] <durandal_1707> and RAND_MAX is INT_MAX [22:48] <cone-485> ffmpeg.git 03Paul B Mahol 07master:f4e29c408cea: lavfi/noise: switch to AVLFG noise generator [22:48] Action: llogan wonders if saste will administer GSoC this year. [22:57] <durandal_1707> so g2m is comming [23:01] <Compn> someone is working on it durandal_1707 ? [23:02] <durandal_1707> Compn: codecs.multimedia.cx [23:03] <Compn> hah, someone must have trolled kostya :) [23:03] <Compn> wonder where he found the smallest go2 binary codec [23:03] <Compn> i spent a while in archive.org looking for it [23:09] <Compn> oh yes [23:09] <Compn> http://codecs.multimedia.cx/?p=431 [23:09] <Compn> that was the one i found :) [23:10] Action: Compn was the troll :D [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:760929117df0: alsdec: check block length [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:caeca53a09fd: qdm2: check array index before use, fix out of array accesses [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:391e0fc6c90c: roqvideodec: check dimensions validity [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:af343f5cddc5: eamad: fix out of array accesses [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:2cac35086c9e: vqavideo: check chunk sizes before reading chunks [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:e6ac11e41734: aacdec: check channel count [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:41eda870483f: pngdec/filter: dont access out of array elements at the end [23:45] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.8:377fabc9e687: Update for 0.8.13 [23:46] <Compn> j-b : is there any g2m1 samples hiding in vlc trac or ftp? :) [23:54] Action: Compn looking for webinars [23:54] <Compn> wmv3 , wmv1 , mss2 [00:00] --- Mon Feb 18 2013

1 0

ffmpeg.log.20130217
by burek 18 Feb '13

18 Feb '13

[00:06] <Xgates> klaxa: YEAH 1.1 works now :) [00:06] <klaxa> :) [00:06] <klaxa> i've messed a bit with mp3 in the last weeks, and it is very easily truncated [00:06] <Xgates> now to see if the presets in WinFF work with this version [00:07] <Xgates> you mean just to shorten mp3? [00:07] <Xgates> or cut... [00:08] <klaxa> shorten or lengthen, anything goes [00:08] <Xgates> ok [00:09] <klaxa> like i said i messed with that a bit in the last weeks, mp3 is arranged in frames, which are a fixed amount of samples (depending on the layer version) you can add and/or take out any frame of an mp3 and it's still spec compliant [00:10] <Kristian_> I need a freelancer that is working with ffmpeg and ffserver, it's about live audio streaming from usb soundcard. Anyone interested? [00:15] <Xgates> ok thanks [00:16] <Xgates> crap I forgot to compile in xvid support errr [00:16] Action: Xgates has to recompile hehe [00:16] <defaultro> saste, looks like I found something fishy. When I use the -vf lutrgb with ffplay, it works. However, when I use -vf lutrgb with ffmpeg, it creates a grey video. Most likely, the video I made last night didn't have the lutrgb option [00:44] <defaultro> saste, will do [00:45] <defaultro> saste, http://pastie.org/6199394 [00:45] <defaultro> saste, that command generated a blank grey video [00:46] <defaultro> however, ffplay shows it properly. I can see the channel gets changed [00:48] <saste> try this one: ffmpeg -f lavfi -i testsrc=d=3 -vf 'lutrgb=r=10+val:g=val-10:b=0+val' out.mp4 [00:48] <saste> also keep in mind that it may be a player problem [00:48] <saste> it is converting to yuv444p, which is not properly supported by many players [00:49] <saste> you may want to force the output to yuv420p [00:49] <defaultro> ok [00:49] <defaultro> it created out.mp4 [00:50] <defaultro> do you want the output in console? [00:50] <saste> no try to play it [00:50] <defaultro> k [00:50] <defaultro> I played it with mplayer and it's color gray [00:50] <defaultro> mplayer -fs out.mp4 [00:50] <saste> try with ffplay [00:50] <defaultro> k [00:51] <defaultro> i see it [00:51] <defaultro> very nice :) [00:51] <saste> then try this: ffmpeg -f lavfi -i testsrc=d=3 -vf 'lutrgb=r=10+val:g=val-10:b=0+val' -pix_fmt yuv420p out.mp4 [00:51] <defaultro> k [00:52] <defaultro> done [00:52] <defaultro> mplayer is able to play it [00:52] <saste> if you have an old mplayer it may not properly support yuv444p H.264 [00:52] <defaultro> got it [00:52] <saste> that's why it is better to avoid that [00:52] <defaultro> avoid mplayer? [00:53] <saste> many players/devices still need to catchup [00:53] <defaultro> got it [00:53] <saste> ipad just to name one [00:53] <defaultro> so from now on, I should use ffplay for playing [00:53] <saste> no you should avoid H.264 + yuv444p [00:53] <defaultro> ok [00:53] <defaultro> i wasn't aware [00:54] <defaultro> i don't even know what yuv444 is :) [00:54] <defaultro> so when encoding, should I use the params you used on the last command? [00:56] <saste> force the pix_fmt to yuv420p, and you got no surprises [00:56] <defaultro> k [00:56] <defaultro> i just tried the long command that was failing, it works now :) [00:58] <defaultro> Thank you so much!!! :) [00:59] <defaultro> that was one complex troubleshooting [00:59] <defaultro> :D [01:01] <defaultro> was it because the recording format was in yuv444 format? [01:02] <defaultro> recording from my video camera [01:21] <Diamonte> If I have a video that was recorded upside down. How could I rotate it in one shot with ffmpeg? [01:22] <Diamonte> from a command line [01:22] <JEEB> see the video filters listing in ffmpeg.org's documentation [01:23] <JEEB> there probably is a filter that has a name that rings a bell [01:23] <juanmabc> -vf "vflip" ? [01:25] <Diamonte> I just saw that too. [01:25] <Diamonte> I'll try it out. [01:56] <Kristian_> Anybody know a freelancer that is working with ffmpeg? Would be grateful for any kind of information. Thank You. [02:23] <saste> Kristian_, check the consulting page on ffmpeg.org [02:24] <saste> what do you need a consultant for? [06:13] <defaultro> wow, the mp4 I uploaded in youtube was 250mb. I downloaded it from youtube, it's downto 33mb. LOL [06:14] <defaultro> i ran strings then grep crf against it, it didn't have any [10:09] <alesan> hi! I have an H.264 byte stream file. This is the output of a hardware H.264 encoder. [10:09] <Kristian_> I need a freelancer that is working with ffmpeg and ffserver, it's about live audio streaming from usb soundcard. Anyone interested? [10:09] <alesan> Any idea how to treat this with ffmpeg to have programs like Vegas Video read it? [10:30] <Fjorgynn> alesan: it doesn't work now? [10:31] <Fjorgynn> alesan: $ ffmpeg -i file.mp4 -c:a copy -c:v copy file.mp4 [10:31] <Fjorgynn> or reencode it with $ ffmpeg -i file.mp4 -c:a copy -c:v libx264 file.mp4 [10:53] <t3ch> hello, why ffmpeg converting so slow command: ffmpeg -i test.mpg -f ogg test.ogg [10:53] <t3ch> what i need to do to convert little faster [10:53] <t3ch> :) [10:55] <sacarasc> Try using more threads, getting a better CPU... [10:56] <t3ch> aha ok :) [10:56] <t3ch> how to set more threads ? [10:56] <sacarasc> -threads X as an output option... Not sure how well theora does multithreading, though... [10:57] <t3ch> ok [10:57] <t3ch> will try [12:09] <pythonsnake> hi [12:09] <pythonsnake> is there a way to ffmpeg -i x.mp4 -o x.mp4? [12:15] <ubitux> only remux? [12:15] <ubitux> oh you mean same filename? [12:16] <ubitux> doesn't sound safe, especially with mp4 [12:35] <pythonsnake> ubitux: why not [12:36] <ubitux> because of seeking [12:36] <pythonsnake> ubitux: the problem i'm trying to fix is [12:37] <pythonsnake> gstreamer's discoverer can't discover a video [12:37] <pythonsnake> and im trying to use ffmpeg (remux to the same file) so it fills the missing fields [12:38] <ubitux> remux to another file, replace the old one [12:38] <pythonsnake> yeah, that's what i thought of [12:38] <pythonsnake> thanks [12:39] <ubitux> it will also reduce race risks on your side [12:39] <pythonsnake> ubitux: what happens if i dont specify extension but vcodecs and acodecs [12:39] <pythonsnake> nothing right? [12:39] <pythonsnake> as in works [12:40] <ubitux> you need to -f mp4 too [12:40] <pythonsnake> ffmpeg -i i.mpg -vcodec copy -acodec copy -scodec copy -y o [12:40] <pythonsnake> o is gonna be mpg? [12:41] <ubitux> not sure it will work [12:41] <ubitux> ffmpeg -i in.mpg -c copy -f mpeg out [12:50] <pythonsnake> thanks ubitux [13:33] <defaultro> I'm reading this thread. I don't know much about yuv. Why would someone convert yuv422 to yuv420, or viceversa? The videos that we see in youtube, what formats are they? Are they yuv420? [13:33] <defaultro> http://forum.videohelp.com/threads/316712-Where-can-i-get-YUV444-YUV422-raw… [13:34] <JEEB> pretty much all video around us is 4:2:0 YCbCr (colloquially called 'yuv') [13:34] <JEEB> and most decoders of video formats can't even take anything else but that [13:38] <defaultro> k [13:39] <defaultro> btw, the 250mb file I uploaded to youtube last night became 33mb, lol. That's a lot of image quality loss with their compression [13:40] <defaultro> i don't even know how they re-encoded it. I didn't see any 264 string from their mp4 file [13:41] <JEEB> they've not written that into their SEI for a long time :P [13:42] <defaultro> was wondering if it was possible to get the picture quality as close as our original [13:49] <defaultro> am I right that my source which is at 60fps will always be downconverted by youtube maybe to 30 fps? [13:50] <JEEB> yes, that's how it used to do it at least [13:52] <defaultro> k [13:57] <WmA> How can one feed a http ts stream into libx264? Do I need to encode from pipe? [14:04] <defaultro> JEEB, this is the video I've been playing last night. I'm trying to fix the greencast(bondi blue). I think I got it to a point that it pleases me - http://www.youtube.com/watch?v=ElZAwRgPbvQ [14:04] <defaultro> glad to hear some comments [14:45] <hid> Hi I'd like to compare videos on youtube cause I suspect somebody to have the same video as mine (a plagiat). How can I do this with ffmpeg ? [14:49] <Mavrik> hmm [14:49] <Mavrik> that's a wierd question [14:49] <Mavrik> what do you want to compare exactly? content? [14:50] <hid> yeah metadatas [14:51] <Mavrik> you see all metadata when doing ffprobe <file> [14:51] <Mavrik> youtube probably strips all of it [14:56] <hid> I checked with mediainfo and gstd was the same [14:57] <hid> evrything was the same [15:10] <Mavrik> hid: that means nothing since Youtube encodes everything to same quality [15:10] <Mavrik> hid: you should look at actual video content not metadata [15:15] <hid> Mavrik : Quality, sound, duration everthing is the same. [15:16] <hid> How can I do to erase all video/audio metadatas ? [15:16] <hid> I've never found any such tool. [15:26] <defaultro> Mavrik, I disagree with what you said that youtube encodes everything to the same quality. I've uploaded videos and pic quality has changed [15:27] <Mavrik> and where did I say that youtube encodes video to the same quality as you gave them? [15:27] <Mavrik> their outputs have same parameters and quality. [15:28] <JEEB> they don't encode for quality, they just have a set bit rate as far as I know [15:29] <JEEB> thus, they are encoding for a set average bit rate [15:29] <JEEB> not same quality [15:29] <Mavrik> *sigh* [15:30] <JEEB> as far as I know, 'tube doesn't use crf [15:30] <JEEB> some video thingies use crf with first pass log, and if it becomes too big, they use a second pass with a set bit rate [17:33] <terr1> I'm playing with ffmpeg (lastest git) and libx264 (lastest git). I'm trying to find some good generel transcoding settings for a html5 website player (progessive download - pseudo streaming). So I was playing with the different x264 presets from ultrafast to veryslow, made a benchmarking shell script to test each setting. I thought this preset setting only effected filesize (compression) not the bitrate? It seems the slower setting the lower bitrat [17:33] <terr1> e? (I'm using cfr 21 to test) [17:34] <Fjorgynn> placebo? [17:52] <xm3> Hi, could anyone help me figure out how to transcode a folder of avis & merge them into one file? [18:12] <grepper> xm3: have you looked at http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join,%20merg… ? [18:40] <xm3> grepper: thanks looks helpful :) [18:56] <Emmanuel_Chanel> Helllo! [18:57] <Emmanuel_Chanel> Do you know how to encode a TS file with dual monaural sounds into a mp4 with two sound tracks? [19:21] <Fjorgynn> thanks to ffmpeg I can listen to this awesome song [19:43] <creep> how much power you use your headphones on? i listen to music on my 2x40 ohm headphone at 2x1mW 0.2V/5mA rms and i don't feel like i want more [20:26] <ioni> hi guys [20:26] <Mavrik> evenin [20:26] <ioni> does anyone in here happens to use latest mplayer? [20:27] <ioni> the configure script is pulling latest ffmpeg [20:27] <ioni> yesterday mplayer started to crash when using vdpau and i'm not sure where the bug is [20:27] <ioni> in mplayer or ffmpeg [20:27] <ioni> http://paste.xinu.at/h9YC/ [20:28] <ioni> Assertion desc->flags & 16 failed at libavcodec/utils.c:511 [20:29] <WmA> blame Charly Sheen! [20:29] <WmA> ;) [20:29] <ioni> =) [20:59] <terr1> Using ffmpeg and x264, how important is customizing keyint, when doing offline transcoding to be played in html5 web player afterwards? does it matter that much? or mostly when doing live streaming ? [21:01] <Mavrik> mosty when doing live streaming [21:01] <Mavrik> if you choose low keyint you'll make compression (and by extension - quality) worse [21:07] <terr1> Ok thanks.. Ill roll with default 250 for now then... If this setting would create problems how would it manifest itself? Mostly when doing manually seeking ? [21:08] <Mavrik> terr1: by taking a long time to start after seeking [21:09] <Mavrik> also by long periods of time when blocky errors repair themselves [21:10] <terr1> Mavrik: Blocky errors.. thoes artifact squares that sometimes appear when seeking, that takes x amount of time / frames to be corrected? [21:10] <Mavrik> terr1: those only appear if there's something wrong with transmission [21:10] <Mavrik> but yeah [21:10] <Mavrik> usually they don't get repaired until next keyframe [21:11] <terr1> ah ok thanks for the information :) [22:08] <defaultro> hey guys, is it possible to add a film look color via ffmpeg? [22:10] <durandal_1707> defaultro: what film look color means? [22:11] <defaultro> let me find some videos [22:13] <Youka> What's the most simple video format to de-/encode with selfwritten code? [22:13] <defaultro> jump to 1:50 http://www.youtube.com/watch?v=Jis97Eup9_E [22:15] <defaultro> durandal_1707, this one is nice - http://www.youtube.com/watch?v=JhtHvynfcTQ [22:15] <Mavrik> Youka: probably motion jpeg or huffyuv [22:16] <Youka> thx [22:20] <durandal_1707> defaultro: i still not sure what you mean, there are some hue/equalization filters [22:23] <defaultro> ok. looks like I will have to experiment to get the desired color [22:24] <Youka> Isn't there a video format similar simple like wav? Saving raw RGB images with some header data? [22:24] <Mavrik> not really, since it's very inefficient [22:24] <Mavrik> huffyuv is probably the closest [22:24] <Mavrik> or any of other losless codecs [22:24] <Mavrik> plus, most video isn't recorded in RGB [22:26] <Youka> Strange :S thx for these informations [22:34] <defaultro> you guys familiar with twixtor? Can ffmpeg achieve that kind of effect? [22:35] <durandal_1707> twixtor is some kind of interpolated slow motion? [22:36] <defaultro> yup [22:36] <llogan> defaultro: you could look into slowmovideo. it has ffmpeg as a dependency, IIRC. [22:36] <defaultro> really nice very slow motion [22:36] <defaultro> k [22:37] <durandal_1707> there is no such filter in libavfilter, but wrapper or native filter is welcome [22:37] <llogan> and it costs $0 compared to $595. [22:37] <durandal_1707> llogan: where is link with price? [22:38] <llogan> durandal_1707: http://www.revisionfx.com/products/twixtor/pricing/#ae_compatible_4 [22:38] <llogan> defaultro: http://slowmovideo.granjow.net/ [22:39] <defaultro> cool [22:40] <llogan> i've never tried it because i hate cmake. [22:47] <llogan> ioni: see commit 259603b [22:54] <ioni> llogan, yes, i was talking with Carl [00:00] --- Mon Feb 18 2013

1 0

ffmpeg-devel.log.20130216
by burek 17 Feb '13

17 Feb '13

[00:12] <ubitux> saste: mmh i don't remember [00:13] <ubitux> i think i didn't submit because i had some output issues [00:13] <ubitux> i'll have a look in a moment [00:14] <ubitux> we have almost the same code [00:15] <ubitux> i guess it's ok [00:15] <ubitux> huh that was my code maybe [00:16] <ubitux> let me re-bench it locally [00:18] <cone-941> ffmpeg.git 03Stefano Sabatini 07master:fbcc584d3abb: lavfi/unsharp: use named options, and add missing checks on matrix size values [00:18] <cone-941> ffmpeg.git 03Stefano Sabatini 07master:37b42bcb7db9: doc/filters: itemize unsharp examples [00:18] <cone-941> ffmpeg.git 03Stefano Sabatini 07master:2042cd37699f: lavfi/unsharp: extend range for amount values, to reflect the behavior of mp=unsharp [00:18] <cone-941> ffmpeg.git 03Clément BSsch 07master:51ba843f195c: lavfi/kerndeint: use aligned linesizes for the temporary buffer. [00:18] <cone-941> ffmpeg.git 03Stefano Sabatini 07master:f84ef19f3f8b: lavfi/kerndeint: move is_rgb_packed definition to the internal context [00:18] <saste> ubitux, uh I already pushed [00:19] <ubitux> haha it simplifies my test then [00:19] <ubitux> don't remove mp=kerndeint now [00:19] <ubitux> give me a few minutes :P [00:19] <saste> anyway, i think that the benchmarking differences may be due to the fact that i don't enable optims [00:19] <saste> no rush, take your time [00:19] <ubitux> what do you mean with the optim? [00:19] <saste> you can use my benchmarking patch on top of git master [00:20] <saste> --disable-optimizations [00:20] <ubitux> yes i have the same patch locally [00:20] <ubitux> yes but why would that affect the benchmarks diff? [00:21] <saste> yes, indeed it shouldn't [00:26] <ubitux> 68908034 decicycles in kerndeint, 2048 runs, 0 skips [00:26] <ubitux> 68417747 decicycles in mp=kerndeint, 2048 runs, 0 skips [00:27] <ubitux> 68329763 decicycles in mp=kerndeint, 2048 runs, 0 skips [00:28] <ubitux> 69982854 decicycles in kerndeint, 2048 runs, 0 skips [00:28] <ubitux> i'm doing ./ffmpeg -f lavfi -i testsrc=s=720x480 -vf tinterlace,kerndeint -f null - [00:28] <ubitux> (btw, surprise with s=720x479, with both filters) [00:29] <ubitux> saste ^ [00:30] <ubitux> anyway, i don't think the performance matters, but i'm actually very curious about the reason of that difference [00:31] <ubitux> i wasn't able to get it last time [00:34] <cone-941> ffmpeg.git 03James Darnley 07master:724061888be1: lavfi/yadif: remove unnecessary cast [00:34] <cone-941> ffmpeg.git 03James Darnley 07master:87d359ff8f55: lavfi/yadif: support 9, 12, and 14 bits per pixel [00:40] <saste> ubitux: you discovered a bug in testsrc [00:42] <ubitux> oh it's in testsrc? okay [00:42] <ubitux> i didn't dig [00:44] <saste> ubitux, I'm testing with 720x479, and still mp=kerndeint is slower [00:49] <ubitux> so with the command i gave, you get faster results? [00:49] <ubitux> maybe some magic somewhere [00:50] <ubitux> let's not bother too much about it then [00:51] <saste> 166831123 decicycles in mp=kerndeint, 1024 runs, 0 skips [00:51] <saste> 166170803 decicycles in kerndeint, 1024 runs, 0 skips [00:51] <saste> the difference is minimal, but yet it is consistently faster [00:52] <saste> i'll drop mp=kerndeint tomorrow, unless we have more benchmarks which prove the contrary [01:02] <saste> ping on "lavfi/unsharp: add check on matrix x/y size values oddity" [01:02] <saste> so i can finally drop mp=unsharp [03:13] <cone-941> ffmpeg.git 03Marton Balint 07master:b853cfe7eaf1: ffplay: remove refresh thread to achieve more precise frame delay [03:13] <cone-941> ffmpeg.git 03Marton Balint 07master:5534d8f75e27: ffplay: remove remains of video_clock [03:13] <cone-941> ffmpeg.git 03Marton Balint 07master:77ab702a848a: ffplay: only display frames if there are no pending events [03:13] <cone-941> ffmpeg.git 03Michael Niedermayer 07master:b9c544891b05: Merge remote-tracking branch 'cus/stable' [08:55] <cone-323> ffmpeg.git 03Anton Khirnov 07master:ccd70d9c1689: vf_yadif: factorize initializing the filtering callbacks [08:55] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:fab8a89abeae: Merge commit 'ccd70d9c1689990e5aef2de383199bbc7cf60d13' [09:14] <cone-323> ffmpeg.git 03Anton Khirnov 07master:64ed397635ef: vf_yadif: fix out-of line reads [09:14] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:446f7c62a2d1: Merge commit '64ed397635ef2666b0ca0c8d8c60a8bc44581d82' [11:09] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:1e78679768b3: mpegvideo_enc: draw edges on input [12:47] <durandal_1707> michaelni: yadif have unused absrefs variable, probably included with merge [12:49] <durandal_1707> s/included with/left from/ [12:56] <cone-323> ffmpeg.git 03Paul B Mahol 07master:af882e1819ca: lavfi: port MP noise filter [13:05] <durandal_1707> michaelni: why ff_* is global in swr? [13:15] <michaelni> mplayer and our libmpcodecs used some of these i think [13:21] <cone-323> ffmpeg.git 03Anton Khirnov 07master:d2a25c4032ce: get_buffer(): do not initialize the data. [13:21] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:a53b144ec041: avcodec: add avpriv_color_frame() [13:21] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:1c5b8b916d3d: h264: color frames gray [13:21] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:bb29ee62e9ef: Merge commit 'd2a25c4032ce6ceabb0f51b5c1e6ca865395a793' [13:21] <michaelni> durandal_1707, rand()/srand() is not good in a lib it can randomly affect applications that expect a repeatable sequence from rand() [13:24] <michaelni> in vf_noise that is [13:31] <durandal_1707> michaelni: so you wait until i commit this to comment? [13:32] <durandal_1707> also it was already in mp [13:33] <durandal_1707> it can be replaced by some fibonaci stuff saste mentioned [13:41] <michaelni> yes i wait until iam aware of a problem before mentioning, ill try to mention it next time before iam aware of it but iam not sure how to do that yet [13:41] <michaelni> mp is no real lib so it can do things like that [13:42] <durandal_1707> michaelni: i don't follow [13:43] <wm4> uh doesn't lavu have RNGs [13:43] <michaelni> it does [13:43] <michaelni> durandal_1707, i wasnt aware of the problem before noise was commited .... [13:45] <durandal_1707> michaelni: i see srand is used in lavu [13:46] <michaelni> its under #ifdef TEST [13:46] <durandal_1707> michaelni: so what should be used? av_lfg? [13:49] <durandal_1707> michaelni: ok to remove mp=noise it is slower that native one(for me, the SIMD stuff is not enabled at compilation, but this can be added later to native one) [13:50] <michaelni> av_lfg is fine, other generators would be fine too [13:50] <Compn> 26 cycles faster. :) [13:52] <Compn> BBB-work : dxva2 is windows hwaccel , chrome could take advantage of gpu speed up ... [13:55] <durandal_1707> really? [14:11] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:b4bb996d6f41: yadif: remove unused variable [14:11] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:0a707da37b41: cinepack: print an error if cinepak_decode() failed [14:20] <cone-323> ffmpeg.git 03Anton Khirnov 07master:68f930d2188a: h264: simplify calls to ff_er_add_slice(). [14:20] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:131b9c924d6d: Merge commit '68f930d2188aba5b32624887dcbf688c23482834' [14:34] <cone-323> ffmpeg.git 03Anton Khirnov 07master:2491f9ee29ae: h264: remove a pointless if() [14:34] <cone-323> ffmpeg.git 03Anton Khirnov 07master:d9ebb00dcbaa: svq3: remove a pointless if() [14:34] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:8b7568cc3482: Merge commit 'd9ebb00dcbaac3812b8b1fbc3d6e027506c11cbc' [15:58] <durandal_1707> what is max allowed value for color components in YUV 16 bit? [16:08] <cone-323> ffmpeg.git 03Stefano Sabatini 07master:2bac1535db97: Changelog: add missing empty line after the version line [16:24] <saste_> mp=unsharp is faster than unsharp [16:42] <durandal_1707> saste_: really? how much? [16:42] <saste_> durandal_1707, it's due to indirect variable access [16:43] <saste_> fp->x vs. x [16:43] <saste_> since this is in a loop it seems it is measurable [16:44] <saste> the other difference is due to a call to av_clip() [16:44] <saste> which is only relevant with no optims [16:44] <durandal_1707> look if it is different by 100 cycle I do not care [16:44] <saste> so all in all i'm going to push these changes and drop mp=unsharp [16:45] <durandal_1707> what compiler you are using? [16:45] <saste> 68524757 decicycles in unsharp, 4096 runs, 0 skips [16:45] <saste> 68856107 decicycles in mp=unsharp, 4096 runs, 0 skips [16:45] <saste> so with the "optims" unsharp is a bit faster now [16:46] <saste> durandal_1707, gcc [16:46] <durandal_1707> what version? [16:47] <saste> gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3 [16:49] <durandal_1707> saste: it is expected for mp to be slow as it does some extra stuff... [16:49] <durandal_1707> also i think it is still using slices.... [16:50] <saste> durandal_1707, no, i'm just benchmarking the processing code [16:53] <durandal_1707> saste: also lavfi have problems when using mpegts as input in VV->V filter(nut works fine) [16:54] <durandal_1707> i guess it is related to probing [16:54] <saste> which kind of problems? [16:55] <saste> related to framerate or something? [16:55] <saste> there is a lavfi open ticket for that [16:56] <durandal_1707> [buffer @ 0x2940c3a0] Unable to parse option value "0x0" as image size [16:56] <durandal_1707> interestingly it works if input is 2nd [16:57] <durandal_1707> it have 2 video streams, but only one is actually ther [16:57] <durandal_1707> some mpegts bug [17:10] <mateo`> what is the consequences of setting AVFrame.interlaced_frame to 1 for a picture ? does it mean that the picture only contain odd or even fields ? [17:11] <durandal_1707> it means frame is interlaced and not progressive, nothing more - nothing less [17:12] <nevcairiel> an avframe typically always contains both fields weaved together, so yeah, it just indicates that this is the fact [17:13] <mateo`> thx [17:15] <durandal_1707> why would frame contain only odd/even fields? [17:16] <mateo`> it was a dumb question, i just wanted to be sure [17:17] <mateo`> by definition a frame contains both fields [17:18] <Daemon404> nevcairiel, from other channel: i feel like im taking crazy pills [17:22] <Shiz> Daemon404: sounds like guys are having fun [18:59] <cone-323> ffmpeg.git 03Anton Khirnov 07master:54974c62982a: error_resilience: decouple ER from MpegEncContext [18:59] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:1fad547cefb4: Merge commit '54974c62982ae827becdbdb9b620b7ba75d079a0' [19:10] <cone-323> ffmpeg.git 03Anton Khirnov 07master:1d0feb5d1ac0: mpegvideo: split ff_draw_horiz_band(). [19:10] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:ddbbfd95bee4: Merge commit '1d0feb5d1ac04d187b335f0e8d411c9f40b3a885' [22:36] <ubitux> mega accepts bitcoins @_@ [23:33] <cone-323> ffmpeg.git 03Anton Khirnov 07master:2c541554076c: h264: deMpegEncContextize [23:33] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:b7fe35c9e50e: Merge remote-tracking branch 'qatar/master' [00:00] --- Sun Feb 17 2013

1 0

ffmpeg.log.20130216
by burek 17 Feb '13

17 Feb '13

[00:23] <undercash> hello [00:23] <undercash> having a weird issue, i wanted to burn subtitle .srt, so i compiled with libass,. so far so good [00:24] <undercash> but it only display the first sentence then nothing [00:24] <undercash> it s kinda weird.. [00:27] <undercash> http://gyazo.com/db3303aa9e8253459d9e57eba157b106 [00:34] <Kapsel> hi, I'm trying to use ffmpeg to extract video frames from an ARRI Alexa video camera, as PNG's, and then use oiiotool (opencolorio) to apply a color conversion LUT to them, and write it back to an H264. But I see some color shifting, when writing the initial PNG frames, I guess its because ffmpeg reads the data as one color space, and then converts it. any ideas on how to solve rthis? [00:35] <Kapsel> https://sites.google.com/site/openimageio/ this is the tool that I am using [00:36] <Kapsel> works extremely well [00:36] <Kapsel> and converts an absolutely perfect color correct frame, but again, the colors drifts slightly when exporting to png from ffmpeg [00:37] <klaxa> Kapsel: i think you are right, default colorspace for h264 is yuv and not rgb [00:38] <Kapsel> yep, but this happens even when I do my source -> rgb [00:38] <alesan> can h264 support RGB at all? [00:38] <Kapsel> if I compare the PNG's with PNG's taht I export from another program, they differ slightly in colors [00:38] <Kapsel> err, source -> png [00:38] <klaxa> alesan: there is libx264rgb at least... [00:39] <klaxa> >H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (decoders: h264 h264_vdpau ) (encoders: libx264 libx264rgb ) [00:39] <alesan> hm [00:41] <klaxa> Kapsel: i have not completely understood your work-chain [00:41] <klaxa> so you have a video-file [00:41] <klaxa> then you extract one rgb frame from that [00:41] <klaxa> apply a color filter [00:41] <klaxa> and add it to an h264 stream with ffmpeg [00:42] <klaxa> the resulting stream does not look like the png you exported from your source video? [00:42] <Kapsel> klaxa: sorry. My video file is a ProRes 4444 file, that I extract to PNG's. Then I run them over oiiotool, a program that applies a color LUT, write them to a new file, and then I encode an H264 with ffmpeg. when I do the first source -> png, the colors shifts compared to if I do the same frame with my other software, DaVinci Resolve [00:43] <Kapsel> the DaVinci Resolve extracted PNG is totally accurate, but my ffmpeg PNG is different. [00:43] <Kapsel> I came across this thread, when searching: http://blenderartists.org/forum/showthread.php?209240-Lut-Support-in-Blende… [00:43] <Kapsel> Someone mentions this: [00:43] <Kapsel> ffmpeg -i input -sws_flags lanczos+full_chroma_int+full_chroma_inp+accurate_r nd -f image2 output%05d.png [00:43] <Kapsel> this command extracts to 550d footage to using full range levels 1-254. [00:43] <Kapsel> but I am not sure if those parameters matters in my case. [00:46] <klaxa> so if you are using DaVinci Resolve to extract the frames from the ProRes 4444 file, it's perfect, but if you use ffmpeg, the colors are off? [00:46] <Kapsel> yeah [00:47] <undercash> well sorry, works perfectly.. was just the wrong sub file..... [00:50] <klaxa> Kapsel: can't find a lot in the codecs section of my ffmpeg binary other than 3 encoders and 2 decoders [00:50] <klaxa> i don't know whether the 4444 profile is supported though [00:50] <klaxa> however, http://en.wikipedia.org/wiki/ProRes_422#Open_Source_Projects [00:51] <klaxa> maybe that fork is of interest for you [00:53] <Kapsel> I know of ffmbc, we use it a couple of places. [00:54] <Kapsel> I do think that Prores 4444 decoding is fully supported by ffmpeg, though. [00:55] <klaxa> mmhh... have a look at the output log and see if you can find something like swscaler or something that indicates a colorspace change [00:57] <Kapsel> I have both ffmbc and ffmpeg installed on my laptop, downloading a Prores 4444 file from the office right now, going to do some quick tests. [00:57] <Kapsel> good point, will try that [01:02] <bcoudurier> Kapsel, is your source hd ? [01:02] <Kapsel> bcoudurier: its 2K [01:02] <bcoudurier> is it using bt 709 as colorspace ? [01:03] <Kapsel> no, thats what I'm trying to convert it to. its some logC colorspace right now. [01:03] <Kapsel> afaik. [01:03] <Kapsel> and hey, thank you for your work on ffmpeg and ffmbc for us in broadcast and post [01:04] <Kapsel> http://www.arri.com/camera/digital_cameras/learn/log_c_and_rec_709_video.ht… [01:05] <Kapsel> we shoot in Log C [01:06] <Kapsel> I can send you a .mov if you'd like to see how it looks like. [01:09] <Kapsel> http://dpaste.com/933645/ info on file here [01:14] <Kapsel> http://dpaste.com/933662/ and this is what I get when I export a frame (ignore the errors) [01:16] <Kapsel> interestingly enough, ffmpeg produces a 12.8MB PNG file, ffmbc creates a 3.8MB PNG [01:16] <Kapsel> using the same command, but different bins. [01:18] <Kapsel> full output here: http://dpaste.com/933679/ [01:19] <Kapsel> colors varies slightly, maybe ffmbc does it better for some reason(?) - will try to compare with some material exported from Resolve [01:20] <Kapsel> Stream #0:0(eng): Video: png, rgb48be, 2048x1152 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc [01:20] <Kapsel> Stream #0.0(und): Video: png, rgb24, 2048x1152p [PAR 1:1 DAR 16:9], vbr, lossless, 25.00 fps [01:20] <Kapsel> rgb24 vs rgb28be, maybe thats the answer [01:21] <Kapsel> rgb48be [02:10] <defaultro> other than -vf hue, what other color correction filters we can use in ffmpeg? [02:36] <tds5016> hey. Is it possible to use ffserver to receive an rtp stream, and rebroadcast it as ffmpeg? [02:37] <tds5016> rebroadcast as hls* [07:15] <bigmac> how can i convert a video to a smaller size for streaming over http [07:15] <bigmac> vcodec examples? [08:02] <defaultro> hey folks, I have an existing -vf "mp ......", how do I add hue=h? [08:03] <defaultro> this failed, ffplay 00013.MTS -vf "mp=eq2=1.3:1.1:-.1,hue=h=30:0" [08:11] <defaultro> found it, using comma :) [12:12] <WmA> anyone sees the error in my syntax? [12:12] <WmA> ffmpeg -i "http://192.168.1.2:7522/upnp/channelstream/0.ts" -c:v libx264 -preset slow -crf 19 -x264opts keyint=123:min-keyint=20 -c:a copy output.mp4 [12:12] <WmA> At least one output file must be specified [13:22] <octe_> Trying to follow this guide: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20capture%20a%20lightning%20(th… , but it's not working to play back the output from samplicate [13:22] <octe_> http://pastebin.com/LMCgjnmD [13:22] <octe_> that is the output i get [15:33] <Aktau> Hey guys :). Does anyone know what some good settings are to make libvpx encode with variable bitrate. Leaving me to just choose some kind of quality setting? [15:33] <Aktau> I've tried crf but it appears to be ignored [15:33] <Aktau> The ffpresets are cool but they set a bitrate of 2M. Which is for some videos too much and for some perhaps too little [15:34] <Aktau> I'd like to suffer as little quality loss from the source H.264 file as possible [15:34] <Aktau> I'm using the newest ffmpeg freshly compiled [15:34] <Mavrik> sec [15:36] <JEEB> Aktau, -crf is libx264-only [15:37] <Aktau> JEEB: I've heard many conflicting things. This is from the WebM project itself: http://wiki.webmproject.org/ffmpeg [15:37] <Mavrik> Aktau: -crf should work [15:37] <Aktau> JEEB: it actually lists a crf [15:37] <Mavrik> Aktau: if you have new enough ffmpeg [15:37] <JEEB> o_O [15:37] <Mavrik> I see that in older versions only bitrate CBR mode was available [15:37] <JEEB> uhh [15:38] <Aktau> Mavrik: ffmpeg 1.1.2 [15:38] <Aktau> I compiled it yesterday via homebrew for mac, works like a charm [15:38] <Aktau> Just looking for the right flags [15:38] <JEEB> I'd be surprised if there was a -crf option for libvpx :s [15:38] <JEEB> ask on #vp8 [15:38] <JEEB> as far as I know, crf has only been in libx264 [15:39] <Mavrik> JEEB: I'm looking at the source. [15:39] <Aktau> Yea that's what many people on the web say as well [15:39] <Aktau> Been looking for almost 2 days how to do this [15:39] <Mavrik> JEEB: "-crf" parameter maps to libvpx's "constant quality" option [15:39] <JEEB> lol [15:39] <Mavrik> http://ffmpeg.org/doxygen/1.0/libvpxenc_8c-source.html [15:39] <Mavrik> line 271. [15:39] <JEEB> that is not crf >_< [15:39] <JEEB> IIRC the constant quality in libvpx is something quite different [15:39] <JEEB> at least looking at their documentation [15:39] <Aktau> ffmpeg -i ironman3-biggame_h1080p.mov -codec:v libvpx -threads 8 -vpre 1080p-aktauvar -crf 11 -codec:a libvorbis ironman3-biggame_h1080p-var.webm [15:39] <Aktau> Is my current commandline [15:40] <Aktau> This creates a really small file, source is 64MB, output is about 4 [15:40] <Aktau> Quality is really bad so I assume it does 200kb/s as it shows after entering the command [15:40] <Mavrik> JEEB: terminology aside, "-crf" just switches libvpx to "CQ" mode and passes the number as CQ factor :) [15:40] <JEEB> yes [15:41] <Mavrik> which is what Aktau wants I guess [15:41] <JEEB> I understand that [15:41] <JEEB> no [15:41] <JEEB> he wants x264's crf [15:41] <JEEB> something like that [15:41] <JEEB> while libvpx's "constant quality" as far as I know [15:41] <JEEB> is something else [15:41] <JEEB> unless they've recently implemented something better [15:41] <JEEB> http://blog.webmproject.org/2011/03/vp8-constrained-quality-cq-encoding.html [15:41] <Aktau> I guess I want something to regulate quality without fixing the bitrate [15:42] <Mavrik> JEEB: ffs, their documentation calls "VPX_CQ" the "constant quality" parameter [15:42] <Mavrik> and in other docs they call it "constrained quality" [15:42] <Mavrik> -_- [15:42] <JEEB> yes, their documentation sucks on it and is not definitive [15:43] <JEEB> and since I don't really care for vp8, I've never really derped at the #vp8 folk about whether or not this is something similar to x264's crf [15:43] <JEEB> putting it under the -crf option in ffmpeg is another piece of craziness as well [15:43] <Aktau> That, it sure is [15:43] <Aktau> I'll inquire there as well [15:44] <Mavrik> slides from senior engineer: "This is not a constant quality mode; but can be made to [15:44] <Mavrik> behave like one (almost). [15:44] <Mavrik> " [15:44] <Mavrik> they need better docs -_- [15:45] <JEEB> I hope they'll make their docs and the encoder better for when they start finishing up VP9 :P [15:46] <Mavrik> considering how they rushed it to Chromium... I doubt it :P [16:30] <defaultro> is it possible to reduce green color? [16:30] <saste_> defaultro, lutrgb [16:31] <defaultro> awesome, I'll search for it [16:37] <defaultro> what's default value of rgb, is it 1? [16:37] <defaultro> i got it working btw :) just need to know the default values [16:42] <saste_> there is no default value for a pixel color [16:42] <saste_> defaultro, what you mean? [16:49] <defaultro> just got back [16:49] <defaultro> the default value that will not change rgb [16:49] <defaultro> i'll try 0 [16:50] <defaultro> 0 made video all black :) [16:50] <durandal_1707> defaultro: there is expression which is evaluated [16:51] <durandal_1707> using constant values is not much of use [16:51] <octe> did anyone see my question? [16:51] <defaultro> k [16:51] <octe> Trying to follow this guide: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20capture%20a%20lightning%20(th… , but it's not working to play back the output from samplicate [16:51] <octe> http://pastebin.com/LMCgjnmD that's the output i get [16:51] <octe> maybe i need to specify more parameters explicitly? [16:53] <durandal_1707> octe: you sure it is mpegts stream? [16:53] <octe> that is the format specified in the streamer [16:54] <durandal_1707> does it work if you do not force it? [16:55] <octe> udp://127.0.0.1:5678: Invalid data found when processing input [16:57] <durandal_1707> increse probe/analyize duration [16:57] <octe> durandal_1707, trying it without samplicate in the middle doesnt seem to work either [16:58] <octe> ah [16:58] <durandal_1707> octe: if it fails to auto detect it usually forcing it will not help at all [17:00] <octe> durandal_1707, just following the tutorial from the wiki.. [17:00] <octe> what's a good value to try for probesize / analyzeduration? [17:00] <durandal_1707> big one [17:21] <rindolf> Hi all. [17:23] <rindolf> How do I record the screen and a mono sound track from my microphone at the same time on Linux using ffmpeg-1.1.2-1.mga3.tainted on Mageia Linux 3/Cauldron? [17:23] <rindolf> Something like a screencast. [17:52] <octe> rindolf, something like this works for me: ffmpeg -f x11grab -s 1920x1200 -r 15 -i :0.0 -f pulse -ac 2 -i default -vcodec libx264 -tune zerolatency -vb 5000k -pix_fmt yuv420p -s 1920x1200 -acodec libmp3lame -ab 96k -ar 22050 -threads 0 -f flv out.flv [17:52] <rindolf> octe: thanks. [17:53] <Fjorgynn> why 420p? [17:53] <Fjorgynn> why not 720P? [17:53] <Fjorgynn> ah, I read wrong [17:53] <Mavrik> evenin' [17:54] <rindolf> octe: http://paste.debian.net/234689/ - I'm getting this. [17:55] <rindolf> Mavrik: evening. [17:56] <Fjorgynn> rindolf: http://www.youtube.com/watch?v=eFM9ZI4BY0w [17:56] <Fjorgynn> maybe? [17:56] <Fjorgynn> and why flv, use x264 imo [17:56] <rindolf> Fjorgynn: OK. [17:57] <rindolf> Fjorgynn: windows? [17:57] <rindolf> Fjorgynn: I'm on Linux. [17:57] <rindolf> Fjorgynn: Give me freedom or give me death. [17:57] <Mavrik> ... [17:59] <rindolf> Fjorgynn: someone I know become much more cynical/etc. after he switched from using Debian Linux to Windows (for Photoshop and for video stuff). [17:59] <Mavrik> dude [17:59] <rindolf> Mavrik: what? [17:59] <Mavrik> please leave OS evangelism out of this channel. [17:59] <rindolf> Mavrik: OK. [18:00] <rindolf> Mavrik: I want to record a screencast on Linux using ffmpeg. [18:00] <octe> Fjorgynn, flv is just a container isnt it? [18:00] <Mavrik> anyway, what seems to be the problem? [18:00] <octe> it's still x264 [18:00] <rindolf> Not on Windows, not on whatever. [18:00] <Mavrik> mhm [18:00] <octe> rindolf, did you change the screen size parameter? [18:00] <Mavrik> rindolf: and I see you're using x11grab [18:00] <octe> to match [18:00] <rindolf> Mavrik: http://paste.debian.net/234692/ [18:00] <rindolf> octe: no, I did not. [18:00] <octe> rindolf, is your screen smaller than 1920x1200? [18:00] <octe> then that is probably the issue. [18:00] <Mavrik> rindolf: can you pastebin the actual command as well? [18:01] <Mavrik> I see only a bash script [18:01] <rindolf> Ah, it's x1080 [18:01] <Mavrik> oh, it's in the log [18:01] <rindolf> Mavrik: it's what octe gave. [18:01] <Mavrik> yup, your resolution is probably wrong [18:02] <Fjorgynn> rindolf: I know that hak5 have a tutorial [18:03] <rindolf> OK, now I'm getting no sound. [18:04] <rindolf> Fjorgynn: will you forget about hak5? http://www.shlomifish.org/philosophy/computers/web/use-qmail-instead/ ? [18:04] <rindolf> Mavrik: OK, resolution is fine now and the video was recorded but no sound. [18:04] <rindolf> Let me paste the new script. [18:04] <rindolf> Mavrik, octe : thanks. [18:04] <Mavrik> rindolf: hmm, do you have pulseaudio server on your machine? [18:04] <Mavrik> IIRC Debian doesn't install it by deafult [18:04] <Mavrik> *default [18:05] <octe> yeah, that command uses pulseaudio [18:05] <octe> change the -f pulse to -f alsa to use alsa i think? [18:05] <rindolf> Mavrik: http://paste.debian.net/234694/ - see this. [18:05] <Mavrik> -f alsa -i hw0,0 I think [18:05] <Mavrik> or is it hw0:0 [18:05] <Mavrik> not sure what alsa device aliases are [18:05] <octe> if you're using pulse check what input you get in the pulseaudio volume control [18:06] <rindolf> Mavrik: I'm using ALSA. [18:06] <rindolf> Mavrik: there's "default". [18:07] <octe> do you have audio recording working in general with alsa? [18:07] <rindolf> octe: yes, works fine in Audacity. [18:17] <rindolf> http://paste.debian.net/234696/ - this is what I get - script + output - in the console. [21:57] <defaultro> anyone alive :) what's the best compression setting so that when we upload to youtube, it won't affect the image quality too much [22:00] <klaxa> defaultro: youtube will fuck up your quality anyways, don't bother too much [22:00] <defaultro> k [22:01] <rindolf> defaultro: maybe try using an .mp4 file. [22:01] <defaultro> i am on mp4 [22:05] <Xgates> hi guys [22:05] <divVerent> defaultro: just use high quality settings [22:05] <Xgates> if I use this to trim a video the audio is messed up; [22:05] <Xgates> ffmpeg -ss 00:00:00 -i 1.avi -c:v copy -c:a copy -t 00:03:13 output.avi [22:05] <divVerent> so the quality loss will be mostly on youtube's side [22:06] <divVerent> then your encoding process just won't matter, quality loss wise [22:06] <divVerent> that's as good as you can get then [22:06] <Xgates> then I found this for mencoder and it works; [22:06] <Xgates> mencoder -ss 00:00:00 -endpos 00:03:13 -oac copy -ovc copy 1.avi -o trimmed.avi [22:06] <Xgates> so not sure if mencoder works better for simple video trimming... [22:08] <defaultro> i read somewhere that if our encoding is really high, the loss will be little [22:08] <Xgates> it's mpeg4 & mp3 and if I leave out the -c:a copy it works fine, not sure why it won't do the copy and work... [22:09] <Xgates> so is there one command that will work on any codec for trimming? [22:10] <Xgates> without transcoding, just copying... [22:12] <divVerent> defaultro: "little", no [22:12] <divVerent> but the higher your quality settings are [22:12] <defaultro> k [22:12] <divVerent> the less your own quality loss affects the result [22:12] <divVerent> so it then is only youtube's quality loss, which you can't control [22:12] <defaultro> yup [22:13] <divVerent> I don't know which crf to use, but if you want to find out... [22:13] <defaultro> what would be the highest quality setting? q=15? [22:13] <defaultro> i mean crt [22:13] <divVerent> upload a HQ video to youtube, with a very low CRF [22:13] <divVerent> then download it from youtube [22:13] <divVerent> and see what crf the file has [22:13] <defaultro> ah [22:13] <divVerent> but, yes, crf 15 sounds safe to me [22:14] <defaultro> can i find out what crf was use in my encoding? [22:14] <divVerent> yes [22:14] <divVerent> run "strings" on the video file [22:14] <defaultro> ok [22:14] <divVerent> then you get a long string from x264 [22:14] <divVerent> which contains all x264 settings [22:14] <defaultro> k [22:14] <divVerent> not sure if youtube strips that, though [22:14] <divVerent> if they DO strip it, go by file size [22:15] <divVerent> make your upload twice the size download from youtube will yield, then you also are pretty safe [22:15] <divVerent> or just use paranoidly high quality settings, like crf 15 :P [22:15] <defaultro> ah, this one used 23 for crf. That's why isn't that good [22:15] <divVerent> and PCM or REALLY high bitrate MP3 or AAC for audio [22:15] <defaultro> i'm fine with audio [22:15] <divVerent> crf 23 is default [22:15] <defaultro> got it [22:23] <defaultro> this is failing [22:23] <defaultro> ffmpeg -y -i $2 -crf $1 -vcodec libx264 -s 1920x1080 -r 30000/1001 -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.5, mp=eq2=1:1.5:0' -acodec libfaac -ac 2 -vpre slow -threads 0 00013.mp4 [22:23] <defaultro> /usr/local/share/ffmpeg/libx264-slow.ffpreset: Invalid option or argument: 'directpred=3 [22:23] <JEEB> don't use preset files [22:23] <defaultro> what should I change? [22:23] <JEEB> use -preset and the actual internal x264 presets [22:24] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [22:24] <defaultro> did you mean -vpre slow? [22:24] <JEEB> ^ list of presets [22:24] <JEEB> yes [22:24] <JEEB> also I am not fully sure why you are setting the frame rate [22:24] <JEEB> esp. after -i [22:25] <defaultro> it's because video is at 60p [22:25] <defaultro> oh, where should I place it? [22:25] <JEEB> ok, so you are converting frame rate [22:25] <JEEB> then it's ok [22:25] <defaultro> k [22:25] <JEEB> well, -r after -i converts frame rate [22:25] <JEEB> -r before -i sets a frame rate (in certain cases) [22:25] <JEEB> and usually people don't have to set -r [22:26] <JEEB> because the input time stamps get used [22:26] <defaultro> so mine is fine? [22:26] <JEEB> yes [22:26] <defaultro> k [22:26] <defaultro> ffmpeg is running now :) I removed -vpre and changed it to -preset slow [22:26] <defaultro> Thanks a lot [22:36] <defaultro> hmmm. video became blank gray [22:36] <defaultro> i'm rerunning it. Command is ffmpeg -y -i 00013.MTS -crf 15 -vcodec libx264 -s 1920x1080 -r 30000/1001 -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.5, mp=eq2=1:1.5:0' -acodec libfaac -ac 2 -preset slow -threads 0 00013.mp4 [22:42] <defaultro> it's still color gray :( what could be the culprit parameter? [22:46] <Mavrik> um [22:46] <Mavrik> maybe the video filter where you change colors? [22:47] <defaultro> it works i'm sure because i used it last night but without the other parameters [22:48] <defaultro> i took out -r and -s [22:48] <defaultro> let's set if it's going to work this time [22:50] <defaultro> still blank gray :( ffmpeg -y -i 00013.MTS -crf 23 -vcodec libx264 -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.5, mp=eq2=1:1.5:0' -acodec libfaac -ac 2 -preset slow -threads 0 00013.mp4 [22:50] <defaultro> this is what I used and it works, ffmpeg -i 00013.MTS -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.3' 00013.mp4 [22:51] <defaultro> but crf on that is 23 which is default [22:51] <defaultro> maybe -preset slow is affecting it [22:52] <JEEB> no, no it shouldn't [22:52] <defaultro> not sure why when using the long command, output becomes gray [22:52] <JEEB> you are most probably doing something wrong, as simple as that [22:52] <JEEB> check your filter chain and so forth [22:52] <JEEB> have fun [22:52] <defaultro> it works if using the short command I showed above [22:58] <defaultro> i took out -vf and video is now being displayed [22:58] <defaultro> where should -vf be placed? [22:58] <JEEB> the placement of -vf itself is OK [22:58] <JEEB> I'm pretty sure you have problems with the filter chain [22:59] <defaultro> k [22:59] <defaultro> i took out two from the filter chain and left lutrgb [23:00] <defaultro> video is back to blank gray again with filter lutrgb [23:01] <defaultro> now trying filter hue [23:01] <defaultro> it works [23:02] <defaultro> trying with both hue and mp [23:02] <defaultro> works again. so lutrgb is the culprit [23:13] <defaultro> played with lutrgb values and it wouldn't cooperate :( [23:13] <defaultro> maybe lutrgb is not compatible with libx264 [23:13] <JEEB> try converting to rgb before it and back to yuv420p after it [23:14] <defaultro> how do I do that? [23:14] <defaultro> the reason I am using lutrgb is to just reduce the green channel [23:15] <JEEB> look at the listing of video filters, there should be one to change colorspaces around [23:15] <defaultro> k [23:16] <JEEB> because either you are failing at life regarding the usage of that filter, or you were and are using a diff. colorspace between those two command lines, and in the latter case you could just manually convert before and after the filter [23:17] <defaultro> k [23:23] <saste> defaultro, looks like you didn't read lutrgb docs [23:23] <defaultro> why? [23:28] <defaultro> saste, can you please tell me where the lutrgb doc is? [23:28] <defaultro> is this it, http://multivac.fatburen.org/localdoc/ffmpeg/ffmpeg.html#lut_002c-lutrgb_00… [23:30] <Xgates> I edited a video from my digital camera with ffmpeg and now I want to trim off the end with this command but the audio doesn't work; [23:30] <Xgates> ffmpeg -i 6.avi -vcodec copy -mp3 copy -ss 00:00:00 -t 00:03:13 trimmed.avi [23:30] <Xgates> is there a one size fits all command for when audio is remuxed how to do this, no matter what audio format? [23:31] <Xgates> errr my bad no -mp3 copy but -acodec copy doesn't work [23:31] <Xgates> this is the output; http://pastebin.ca/2314537 [23:31] <klaxa> that is weird because -c:a copy should work especially with mp3 [23:31] <Xgates> yeah it's not [23:31] <Xgates> hmm [23:32] <klaxa> you might also want to update your ffmpeg [23:32] <Xgates> I compiled ffmpeg-0.11.2 yeah thinking maybe to get the latest 1x the only reason I didn't is because I mainly use WinFF to convert videos and working good with 0.11x [23:33] Action: Xgates checks [23:34] <saste> defaultro, lutrgb=r=1.1*val:g=.9*val:b=1*val [23:34] <saste> this looks mostly right, but 1.1*val is going to generate out-of-range values [23:34] <defaultro> that's what I used [23:34] <saste> which are clipped to 255 IIRC [23:35] <defaultro> it works when I don't add the other parameters [23:35] <defaultro> so float is not allowed? [23:35] <defaultro> but i've been trying non-float, it still generates gray video [23:36] <saste> defaultro, complete command [23:36] <defaultro> one sec, it's running [23:37] <defaultro> for my recent test, I took out lutrgb and used lutyuv, it showed the video [23:37] <defaultro> this one generated grey video, ffmpeg -y -i 00013.MTS -crf 23 -vcodec libx264 -vf lutrgb="r=1*val" -acodec libfaac -ac 2 -preset ultrafast -threads 0 00013.mp4 [23:38] <Xgates> ok compile time for 1.1x [23:39] <saste> defaultro, what about the original source? [23:39] <defaultro> what about the original source? [23:39] <saste> also "lutrgb=r=1*val" is not changing the video [23:40] <defaultro> based from my earlier test, it will increase red channel [23:40] <defaultro> what is your question about original source? [23:41] <saste> defaultro, no, r=1*val is just setting red to the original value, while the other components are kept to the same value [23:41] <saste> that is it is not changing the input at all [23:41] <defaultro> got it [23:41] <defaultro> now I remember why I did 1.1 [23:41] <saste> also if you have gray input, you get gray output [23:41] <defaultro> my original source is full color [23:42] <defaultro> but it has some slight green cast [23:42] <saste> but then i don't know what you're trying to achieve [23:42] <defaultro> saste, check this out, http://www.youtube.com/watch?v=twwWOsWUSIk&list=UUrcz0Vxiv4ZOKFEDScLw5UQ&in… [23:43] <defaultro> that's the one I created last night using ffmpeg. However, the command was very simple. crf was set to default of 23 [23:43] <defaultro> now, I am retrying with more ffmpeg parameters but when I added the same -vf filters, it turned color gray [23:43] <saste> defaultro, what about lutrgb=g=0.5*val [23:44] <saste> but in that case you are pixel intensity [23:44] <defaultro> ok [23:44] <saste> you are *decreasing* [23:44] <defaultro> fyi, this is what I used last night to create that youtube video - ffmpeg -i 00013.MTS -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.3, mp=eq2=1:1.5:0' 00013.mp4 [23:45] <defaultro> now, when I added that same filter to my longer command line, it generated grey video [23:46] <defaultro> this one generated grey video [23:46] <defaultro> ffmpeg -y -i 00013.MTS -crf 15 -vcodec libx264 -s 1920x1080 -r 30000/1001 -vf 'lutrgb=r=1.1*val:g=.9*val:b=1*val, hue=0:1.5, mp=eq2=1:1.5:0' -acodec libfaac -ac 2 -preset slow -threads 0 00013.mp4 [23:46] <saste> not here [23:47] <defaultro> what do you mean "not here"? [23:47] <relaxed> defaultro: you may want to build ffmpeg with libfdk (--enable-libfdk-aac), as it's supposed to be a much better aac encoder. [23:47] <saste> defaultro, that I tried that filterchain and I get no grey video [23:48] <defaultro> ok saste. I don't get grey video if I use the shorter command line [23:48] <defaultro> i get grey video once I use the longer command [23:48] <defaultro> relaxed, I'll retry it later [23:48] <defaultro> i would like to make the filter work [23:49] <defaultro> do you think i have a bad ffmpeg version? [23:50] <defaultro> ffmpeg version N-48389-g341e40f [23:50] <Xgates> ok done, I gues 10 mins. on an i3 not bad :) [23:56] <relaxed> defaultro: latest from git is usually recommended [23:56] <defaultro> k [23:57] <defaultro> when specifying multiple filters, how do you combine them? Do we use comma, space or semicolon? [23:58] <defaultro> i tried two -vf, the latter is only one that gets executed [00:00] --- Sun Feb 17 2013

1 0

ffmpeg-devel.log.20130215
by burek 16 Feb '13

16 Feb '13

[00:16] <cone-893> ffmpeg.git 03Vignesh Venkatasubramanian 07master:003be0a9c33e: Removing network.h from matroskadec.c [00:16] <cone-893> ffmpeg.git 03rogerdpack 07master:12c71f648cbc: dshow: Fix MSVC support, remove av_export, which was apparently unneeded anyway. [00:27] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:8102a097a5ce: doc/APIchanges: List merge commit hashes and version numbers [00:29] <cone-893> ffmpeg.git 03Stefano Sabatini 07master:772b949d8eea: examples/scaling_video: fix typo [01:06] <Paranoialmaniac> michaelni: fiel atom is only in mov. at least, fiel atom is not defined in 14496-12(iso base media)/14 (mp4)/15 (avc), even if fiel atom is present in a non-mov file, reader shall ignore it [01:09] <cone-893> ffmpeg.git 03Stefano Sabatini 07master:7ac3ccc5f238: lavfi/unsharp: use the same macros used in the original MP filter [01:59] <michaelni> Paranoialmaniac, thanks! [02:07] <Paranoialmaniac> undefined atoms under a certain brand shall be ignored by reader. they are just junk under that brand [04:03] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:4a6fa7963bd2: lavf: dont try to find fps for attached pictures [08:40] <cone-466> ffmpeg.git 03Cyrille Faucheux 07master:75758f84def2: build: fixes a "can't cd to..." issue when installing shared libraries. [09:07] Action: ubitux still trying to figure out what Nicolas really has in mind about the sub_charenc mode thing [12:16] <cone-241> ffmpeg.git 03Diego Biurrun 07master:ab441e20ffdf: avutil: Move emms code to x86-specific header [12:16] <cone-241> ffmpeg.git 03Diego Biurrun 07master:759a3a217783: configure: Move MinGW CPPFLAGS setting to libc section, where it belongs [12:16] <cone-241> ffmpeg.git 03Diego Biurrun 07master:4db96649ca70: avutil: Ensure that emms_c is always defined, even on non-x86 [12:16] <cone-241> ffmpeg.git 03Michael Niedermayer 07master:61fbb4cd57d7: Merge commit '4db96649ca700db563d9da4ebe70bf9fc4c7a6ba' [12:27] <cone-241> ffmpeg.git 03Diego Biurrun 07master:49fe280753e0: h264idct: Replace duplicate scan8 table by appropriate #include [12:27] <cone-241> ffmpeg.git 03Michael Niedermayer 07master:048ecbd3f833: Merge commit '49fe280753e0f167ac3d9f227f0c0f7744501fc1' [12:34] <cone-241> ffmpeg.git 03Diego Biurrun 07master:3594554a064d: sparc: dsputil: Simplify high_bit_depth checks [12:34] <cone-241> ffmpeg.git 03Michael Niedermayer 07master:f98598942f7f: Merge remote-tracking branch 'qatar/master' [13:24] <Shiz> >archlinux [13:24] <Shiz> no wonder it failed [13:31] <kode54> hah [13:53] <ubitux> Shiz: libav moved some stuff again, no wonder it failed [16:02] <Shiz> ubitux: you're not libav though, are you :-) [16:15] <ubitux> kierank: give me an hour [16:15] <kierank> no problem [16:16] <ubitux> sorry for the delay, i'm trying to catch up with all i missed this month :) [16:16] <ubitux> kierank: btw, you're using that filter? [16:16] <Compn> Shiz : ffmpeg merges from libav, including their bugs :D [16:16] <kierank> i am investigating using it [16:16] <kierank> ubitux: the biggest issue of course is getting the amount to normalise by to volume [16:17] <kierank> but i think that'll be doable in the latest libav merge [16:17] <ubitux> can't you just use af volume as a second pass? [16:17] <ubitux> doing a simple substraction [16:17] <kierank> i am filtering in realtime [16:17] <Shiz> Compn: I'm aware, it's just kind of hypocitic to bitch about libav if you merge their changes without apparently checking if it works for your own repo :) [16:18] <ubitux> Shiz: this fate instance has for very particular purpose to test such thing [16:19] <Shiz> ic [16:19] <kierank> ubitux: do you know where it says what length you are meant to use per "time slice" of audio in the documents [16:19] <Compn> libav has their own fate system too, http://fate.libav.org [16:19] <ubitux> Shiz: bitching about the distro wasn't innocent either, it's a bit hypocrit to critic such behaviour [16:19] <Compn> oh burn [16:19] <ubitux> kierank: the 100ms and 3s? [16:20] <ubitux> erm 400ms [16:20] <kierank> are those segments the amounts of time that you then pass to the volume filter? [16:20] <kierank> and normalise separately [16:20] <ubitux> i don't know about the real time normalization [16:21] <ubitux> i've really spent only 2 weeks on this :D [16:21] <kierank> that's the part i don't quite understand about r128 [16:21] <kierank> r128 or atsc a/85 is going to be mandatory in some countries for realtime encoding [16:21] <ubitux> i'd say adjust volume in 400ms windows [16:22] <ubitux> based on the overall loudness [16:22] <kierank> yeah [16:22] <ubitux> (integrated) [16:23] <ubitux> or maybe using the other loudness [16:23] <ubitux> to get something smoother [16:26] <mateo`> hi guys ! [16:27] <kierank> hello [16:30] <mateo`> is there a way to force the number of threads of a decoder from a demuxer ? [16:30] <mateo`> i'm trying to solve this ticket https://ffmpeg.org/trac/ffmpeg/ticket/1102 [16:32] <mateo`> the problem i'm facing is that i need to reconstruct a full frame from 2 jpeg2000 frame [16:32] <mateo`> i have a current wip here https://github.com/mbouron/FFmpeg/commits/mxfdec_j2ki but it removes threading support from the libopenjpeg wrapper [16:33] <mateo`> (and i'm not familiar at all with the threading api) [16:38] <mateo`> any hints would be appreciated ;) [16:40] <cone-462> ffmpeg.git 03David A. Sedacca 07master:de21e6736e36: lavfi/ebur128: fix channel weights [16:40] <cone-462> ffmpeg.git 03sedacca(a)comcast.net 07master:b64de24fd7a5: lavfi/ebur128: advance pointer to samples [16:41] <ubitux> erm, i should have checked the authorship [16:41] <TimNich> :P [16:42] <kierank> thanks [16:48] <ubitux> 16:16:59 <+kierank> ubitux: the biggest issue of course is getting the amount to normalise by to volume [16:48] <ubitux> 16:17:12 <+kierank> but i think that'll be doable in the latest libav merge [16:48] <ubitux> what are you refering to? [16:48] <ubitux> lavfi already has metadata injection [16:48] <kierank> elenril said it may be possible with per frame metadata [16:48] <ubitux> we actually can already do that given relatively little effort [16:48] <ubitux> we already have them in buffer ref in lavfi [16:48] <ubitux> so it's the sasme [16:48] <ubitux> same* [16:49] <kierank> well yes but that's going to become AVBuffer or something [16:49] <kierank> to be unified [16:49] <ubitux> currently it's copied at the AVFrame at the end [16:49] <ubitux> the feature you need is already available [16:49] <kierank> ok [16:50] <ubitux> it's already used in two filters iirc [16:50] <ubitux> silencedetect and scenedetect [19:06] <cone-462> ffmpeg.git 03Stefano Sabatini 07master:0018221c03c2: doc: fix reference to ffmpeg-bitstream-filters.html page [19:17] <cone-462> ffmpeg.git 03Hendrik Leppkes 07master:5ad43af9a62c: lavfi/kerndeint: use av_pix_fmt_desc_get instead of directly accessing the table [19:53] <saste> ubitux: i'm benchmarking kerndeint vs. mp=kerndeint and noticing no serious difference (even without the linesize "optimizations") [19:53] <saste> how was you testing it? [19:54] <saste> also, do you have numbers to share? [21:39] <BBB-work> who is "Owner fate" on fate.ffmpeg.org? [21:40] <BBB-work> (I'm looking for the owner of "http://fate.ffmpeg.org/report.cgi?time=20130215202302&slot=x86_64-msvc10-wi…") [21:41] <llogan> damnit roger [21:43] <BBB-work> nevcairiel, Daemon404: is that one of you? [21:44] <nevcairiel> thats mine [21:44] <nevcairiel> why? [21:44] <BBB-work> \o/ [21:44] <BBB-work> does that machine have avx? [21:44] <nevcairiel> yes [21:44] <BBB-work> uhm... [21:44] <BBB-work> ok [21:44] <BBB-work> so why does it pass? [21:45] <BBB-work> I've been finding 2 avx-specific win64 bugs already [21:51] <nevcairiel> i havent seen any issues recently [21:54] <BBB-work> so one issue (patch coming sometime soon) is that h264 loopfilter uses redzone [21:54] <BBB-work> win64 doesn't support that [22:01] <Compn> BBB-work : any chance you can review the dxva2 patch on the list ? [22:01] <Compn> everyone seems to ignore it :\ [22:03] <nevcairiel> because we hate it [23:02] <BBB-work> Compn, dxva2? I don't even know what that is [23:14] <BBB-work> (if you're asking me because it's windows; I have absolutely no love for windows, and don't ever use it; I keep msvs working for chrome :) ) [23:39] <cone-941> ffmpeg.git 03Michael Niedermayer 07master:5e947aeb5945: sws/x86: improve rounding for yuv2yuvX [23:49] <saste> ubitux: ping kerndeint [00:00] --- Sat Feb 16 2013

1 0

ffmpeg.log.20130215
by burek 16 Feb '13

16 Feb '13

[00:10] <tds5016> so if I disable-all, what all would I need to enable for ffmpeg -i fs3 -c:v copy -f mpegts stream3.ts to work? [00:10] <tds5016> ./configure --disable-yasm --disable-all seems not to work :-( [00:12] <tds5016> correction: it works, but it doesn't output an ffmpeg binary anywhere :-( [00:19] <ubitux> --disable-all is what it means& [00:24] <tds5016> I'm now getitng osmething produced, but I'm missing something to make that command work... [00:24] <tds5016> I keep getting invalid input. [00:25] <ubitux> --enable-demuxer=mpegts --enable-protocol=file and stuff like that i guess [00:26] <ubitux> (assuming you disabled everything) [00:26] <ubitux> you'll likely need a few ones like this [00:39] <stephanedev> hi everybody. i would like to stream a live video over rtp without any noticeable latency on a LAN. right now i have a latency ~1s. [00:40] <stephanedev> i am already using some codec options to reduce latency [00:41] <stephanedev> namely (("quality", "realtime"), ("lag-in-frames", "0")) for vp8 and ("tune", "zerolatency") for h264 [00:42] <stephanedev> this reduced the latency from a couple of seconds to around one second [00:42] <stephanedev> even when using the loopback interface this latency happens [00:44] <stephanedev> when streaming directly the output of the vp8 encoder without using rtp (and using my own code instead of letting ffmpeg lib handling the network bits), there is no noticeable latency [00:45] <stephanedev> do you know of any other option in the ffmpeg C API to reduce this latency? [00:46] <stephanedev> is it due to some internal buffering in the muxer or the demuxer? [03:47] <lobotomius> hi ffmpeg! [03:47] <lobotomius> how do i use this unsharpen filter [03:47] <lobotomius> what are the params to sharpen the clip [03:47] <lobotomius> ? tia [06:33] <bigmac> i have a question about a movie collection and is there a way to convert them all to a smaller size streamable on my droid phone... [06:33] <bigmac> maybe one command will work for all different formats, sizes and sizes? [06:34] <bigmac> [700mb.mkv, 3gig.avi, 5gig.flv] [06:34] <bigmac> those would be example source files [06:36] <bigmac> src= mp4 avi mkv [06:36] <bigmac> output= 300mb.flv [06:36] <bigmac> or something [06:38] <bigmac> or 300mb.swf [06:46] <sacarasc> bigmac: Input matters relatively little. [06:46] <sacarasc> Except for length, if you're wanting to have a specific size. [06:49] <bigmac> i can batch script a directory but ultimatly i want to use a single ffmpeg command to get decent quility video i want to embed but download speed is my main concern [06:50] <bigmac> maybe a example command if a embeddable output video? [06:52] <bigmac> can i specify output file size? with width and hight? [06:57] <bigmac> ffmpeg -i INPUT.mp4 -vcodec copy -acodec copy OUTPUT.flv [06:58] <bigmac> thats same quality, im not sure how to loose file size [07:01] <bigmac> -fs 3000 [07:20] <bigmac> i dont know what im doing lol [07:21] <bigmac> how to reduce the output video size for better streaming [10:20] <llogan> bigmac: see https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [10:20] <llogan> two-pass if you're targeting specific output file size [11:06] <barnseenio> relaxed: about? [11:09] <relaxed> barnseenio: yes [11:10] <barnseenio> relaxed: pm? [16:17] <captainloc> Hi guys. Im broadcasting a live stream from one place to another and then i'm capturing this stream, cutting it into chunk. It's a mpeg-ts stream with H264 and AAC. I'm saving this stream locally, cutting it based upon a guide. It can appens that my live stream stop and restart but i keep recording and appending in the same file. Then i transcode this file to MP4. But randomly i get problems with Video : audio seems correct and plays but video stop and i [16:22] <haole> hey there... I'm working in a digital TV application in which I need to send a private binary file over a MPEG stream... I need to develop a software to "break" my binary as MPEG sections with a possibly private type/descriptor/whatever... can ffmpeg help me achieving this? [18:13] <alesan> hi [18:13] <alesan> I am recording from three video sources at the same time and audio. So far I have captured the output of each camera, encoded in a different video file, and the audio in a .ogg [18:14] <alesan> but it's a nightmare to synchronize everything [18:14] <alesan> is there a way to record and sync right away multiple video sources? and maybe one or more audio sources too [18:16] <klaxa> maybe synchronize recording devices clocks and add timestamps? :X [18:17] <klaxa> *devices' [18:21] <alesan> klaxa, the device is one and I am reading out three USB webcams [18:22] <klaxa> are you using one ffmpeg instance or are you spawning multiple ones? [18:22] <alesan> adding timestamps... but shouldn't the three videos be in the same file at that point? otherwise how can the timestamps relsato correctly to each other? [18:22] <alesan> right now it's three different processes, each one records on a separate file completely indipendently [18:23] <klaxa> you can put multiple video streams into one file and extract them for later processing [18:23] <alesan> if they are in the same file, I do not want to extract them later :) [18:23] <alesan> how do I put more than one video stream in the same file? [18:23] <klaxa> they would be in the same file [18:24] <klaxa> the matroska container for example supports multiple video streams [18:24] <klaxa> you could extract them for post-editing [18:24] <alesan> I know mkv supports more tracks, that would be awesome but... how to do that? I have searched for the past two days [18:25] <alesan> I see no example at all, only joining several video files that already exist with mkvmerge [18:26] <klaxa> ffmpeg -i <input 0> -i <input 1> -i <input 2> -map 0 -map 1 -map 2 everything_in_one_file.mkv [18:26] <klaxa> i *think* that should work [18:26] <klaxa> i never tried it myself [18:27] <alesan> -map ? let me see what it does [18:28] <klaxa> https://www.ffmpeg.org/ffmpeg.html#Main-options [19:32] <Aktau> Hey guys [19:33] <Aktau> I'm trying to convert an apple h.264 file (1080p) to a decent WebM file (libvpx and libvorbis) [19:33] <Aktau> I can do that just fine with the libvpx-1080p.ffpreset [19:33] <Aktau> But that sets the nitrate fixed [19:33] <Aktau> In this case to 2M [19:34] <Aktau> It looks nice and all but I would like to equal the nitrate of the original, which was a bit more than 16M [19:34] <Aktau> Is there a nice way to tell ffmpeg to take the same nitrate? [19:34] <Aktau> s/nitrate/bitrate/ [19:46] <Aktau> I guess I would like to specify quality and let the bitrate be determined by libvpx [19:46] <Aktau> 't would be nice if anyone with experience with vp8 could enlighten me :) (using 1.1.2 on mac btw, freshly compiled) [20:23] <john_____> hi, I'm using ffmpeg 0.11.2 with android 2.3.3, When I play h264 videos, there's this hiccup effect. With 0.10.8, everything was fine... Any idea ? [20:24] <bahar> john_____ what do you mean by hiccup event? [20:25] <john_____> bahar: the video plays normally, then suddenly, goes back few frames. Then proceeds forward, but jumps back few frame. and so on [20:35] <llogan> john_____: does it occur with ffmpeg from git head? [20:38] <john_____> llogan: tried to compile and run 1.0 but didn't succeed. We sticked to 0.x But to fix the deadlock in libestagefright, we need to use 0.11.1 or 0.11.2 [20:40] <llogan> development is active, and testing git head is always the first step in case the issue has already been resolved [20:44] <john_____> llogan: ok, will try. Is it working with android 2.3.3 ? Does somebody has a recipe to build libstagefright with 1.x ? [20:48] <peper03> Is there any way to get av_read_frame to return MPEG2 packets with 0x1bf (PRIVATE_STREAM_2) as start code? I need them in sync with the normal packets to make improvements to DVD playback but mpegps_read_pes_header() actively filters them out. [20:55] <rcombs> hey, all [20:56] <rcombs> I'm trying to use FileSalvage and dd to recover a file [20:56] <rcombs> it's an MOV [20:56] <rcombs> I've been alternating between "invalid data" and "no MOOV atom found" [20:57] <rcombs> in ffplay, which of those would come first? [20:57] <rcombs> e.g. if I get "no MOOV atom", could it indicate that I'm not extracting enough data? [21:00] <llogan> probably, especially if the moov atom is near the end of the file [21:00] <rcombs> if the file was written by a DSLR, I'd expect the MOOV to be at the end, yeah? [21:03] <llogan> i'm not sure. it's not something i'm really that familiar with [21:04] <llogan> but an uneducated guess would be "yes". [21:11] <rcombs> OK, which would indicate I've got the right start to the file: invalid data, or no MOOV? [21:22] <qubitnik> hello! [21:22] <Fjorgynn> cake? [21:23] <qubitnik> thx! [21:27] <qubitnik> I have an ffmpeg question (suprise, suprise), its about reusing outputs: I want one Single Input to be transcoded in multiple bitrates+resoultions (e.g. 720,480,240) but also pushed to other servers (e.g. rtmp), one quality should be saved to file, and the 3 qualities should be segmented for hls [21:29] <qubitnik> I know how to do all the single Steps, but my Question is: What is the preferred way to provide each ffmpeg process with the already encoded streams, avoiding reencoding, and beeing able to use the streams any time not just at initial start [21:34] <Mavrik_> qubitnik: ffmpeg really doesn't support this use case well [21:34] <Mavrik_> (or at all) at the moment [21:34] <qubitnik> hmm i made some experiments with a rtmp server in the middle [21:34] <Mavrik_> qubitnik: if you have different resultions and bitrates you WILL have to reencode for each output [21:34] <qubitnik> but i am not happy with that [21:37] <qubitnik> yes i know, that is not the problem, but is there a way other ffmpeg processes can grab these outputs and use it again just copying audio+video and only doing remux for pushing,recording,segmenting? [21:37] <llogan> maybe the tee muxer will be useful to you but i don't know if it will work with segment [21:38] <Mavrik_> hmm [21:38] <qubitnik> @llogan: yes, but it is a bit too static for me, using an rtmp server in the middle works, but adds another component+latency [21:39] <Mavrik_> well another rather inefficient option is to just send UDP packets over localloop [21:39] <Mavrik_> have one ffmpeg encode to MPEG2-TS and send to udp://127.0.0.1:<port> [21:39] <Mavrik_> and other ffmpegs read from it and remux [21:39] <Mavrik_> it'll add latency of course [21:40] <Mavrik_> but for anything better you'll just have to pay for a "real" solution like Wowza :) [21:40] <qubitnik> i also experimented with udp://127.0.0.1:5000 Streams and 1 Output is using them, and a second output is a copy to another port [21:41] <qubitnik> i did so because this udp is point-2-point only one ffmpeg instance could use it, multicast is not an option in my scenario [21:41] <Mavrik_> well as I said [21:41] <Mavrik_> Wowza licenses aren't so expensive for pro usage [21:42] <qubitnik> well, if you need one wowza license for each client, it is... [21:43] <qubitnik> what i want to do tooks too much performance, its 1, maximum 2 liveshows per server [21:43] <qubitnik> and they have a per server license [21:43] <Mavrik> I had to develop my own solution based on libav* libs to do just what you want to [21:44] <Mavrik-> everything else is just sub-par :\ [21:45] <qubitnik> yes using the libav* apis is the right way to go, but i am not good in c [21:45] <qubitnik> more into java [21:45] <Mavrik-> mhm, it's a problem then [21:45] <qubitnik> yes [21:45] <qubitnik> and libav* is not so well documented :) [21:45] <Mavrik-> true ^^ [21:47] <qubitnik> what did you wrote with libav? [22:01] <peper03> Is there a better place to ask about the PRIVATE_STREAM_2 issue? I tried on the libav-user mailing list several days ago and got no response there either. Just don't want to waste my (or anyone else's) time asking in the wrong place. [22:02] <Mavrik_> well, you can ask [22:02] <Mavrik_> but I doubt there's much people here that would know more than libav-user list [22:07] <peper03> Ok. Doesn't look hopeful, then. I'm guessing there must be a reason for explicitly filtering those packets out but I'm blown if I can understand why. Thanks anyway. [22:47] <neXyon> hi [22:48] <neXyon> what are the standard codecs (video, audio) for avi files? I know the container supports many codecs, but which ones are used and best supported from devices? [22:49] <neXyon> mpeg4 + mp3? that's what ffmpeg uses by default [22:51] <Mavrik_> something like that [22:51] <Mavrik_> neXyon: what device are you targeting? [22:51] <Mavrik_> usually choosing anything else than avi is a wise idea :) [22:52] <neXyon> Mavrik_: yeah, the problem is, I don't know, that's why I'm gonna take my video in different formats and resolutions with me xD [22:52] <Mavrik_> hmm [22:52] <Mavrik_> mp4 with H.264/AAC is much more widely supported than anything in .avi really [22:54] <neXyon> Mavrik_: yeah, that's the original format anyway [23:15] <hi117> how do i extract a text stream from a video file? [23:19] <hi117> er actually its the skeleton stream from a ogg [23:23] <neXyon> ffmpeg -i short.mp4 -sameq -s 1280x720 short_hd.mp4 => gives an error, is that size not supported? o_O [23:24] <Mavrik_> 1.) never use sameq [23:24] <Mavrik_> 2.) what error? [23:24] <neXyon> Mavrik_: what else? xD [23:24] <neXyon> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [23:25] <Mavrik_> look higher [23:29] <neXyon> http://pastie.org/private/tuzgssgnsb8gc34qrypmg [23:29] <neXyon> no clue why it writes none @ audio, and why not use sameq? [23:30] <JEEB> because sameq is "copy quants from source", and hasn't really worked even as such for years [23:30] <sacarasc> -sameq uses the same quantizer as the source, not the same quality. [23:30] <JEEB> yup [23:30] <JEEB> also the reason why your line didn't work is written on line 31 [23:30] <neXyon> ah ok xD [23:32] <neXyon> and how can I tell it to use the same quality? [23:32] <JEEB> long story short, you can't [23:32] <JEEB> you don't have thousands of chinese kids working in your computer [23:32] <JEEB> the closest one has to such is libx264's -crf option [23:32] <JEEB> which is the closest to "constant quality" [23:32] <JEEB> default is 23 [23:33] <JEEB> higher is more compression, possibly worse quality, lower is less compression and possibly better quality [23:33] <JEEB> the idea is to find the highest crf value that still looks good [23:33] <neXyon> ah ok :) thanks [23:33] <JEEB> which is the maximum amount of compression you can get while the video still looks good [23:34] <JEEB> -c:v libx264 -crf 23 -preset medium <- these would be libx264's defaults [23:34] <JEEB> preset is speed vs compression, while crf controls quality level [23:34] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset <- all the presets listed [23:34] <JEEB> so yeah, find the highest crf value that still looks good [23:35] <JEEB> and then pick the slowest preset that is still fast enough for you [23:35] <JEEB> and that's it :) [23:35] <neXyon> k thanks :D [23:35] <JEEB> you can use -ss and -t to cut 1-2 minute parts from the middle of a film or something [23:36] <JEEB> for testing :) [23:36] <JEEB> so you don't have to encode the whole thing [23:40] <neXyon> JEEB: thanks, but I don't have enough time to do testing xD [23:40] <neXyon> JEEB: have a very soon deadline :D [00:00] --- Sat Feb 16 2013

1 0

ffmpeg-devel.log.20130214
by burek 15 Feb '13

15 Feb '13

[00:02] <someone-noone> Hey! If I want to reorder packets by pts (decoder is not ffmpeg), how can I know when to "flush" those packets? When new key-frame arrives or when dts==pts? Or may be some other algorithm? [00:04] <nevcairiel> for mpeg2 and vc1 its pretty easy, you basically delay the last I/P frame until the next one arrives, for H.264 thats quite a bit more complex [00:06] <someone-noone> nevcairiel, the first codec I'm working on is exactly h264 [00:06] <nevcairiel> well good luck with that [00:06] <nevcairiel> i wouldn't know how to do it :D [00:06] <someone-noone> :D [00:06] <nevcairiel> you could just store the packets and then simply sort by time [00:07] <nevcairiel> since pts is out of order, you can just re-sort them to give you a linear time [00:07] <nevcairiel> but its meh :) [00:07] <nevcairiel> anyway, sleeping time [00:09] <someone-noone> nevcairiel, pts is in milliseconds time. So& it's not so obvious. However, now I do following: while (pts!=dts || list.size() < 10) wait() [00:09] <someone-noone> But it's not very accurate :) [00:10] <someone-noone> okay, what is a "good time" for this channel? I mean, it's night here too, but I don't know when is it better time to ask [00:11] <someone-noone> not || but && [00:14] <michaelni> someone-noone, if you want to know how h264 pixture reordering works, the authorative source is the itu h264 spec [00:14] <michaelni> its not a nice easy to read doc though [00:15] <someone-noone> anyway thanks :_ [00:43] <cone-580> ffmpeg.git 03Ronald S. Bultje 07master:faf8eca08dcb: h264: remove clear_blocks call in threading init. [00:43] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:701e9b82547c: h264: Use mb itself as memcpy anchor and assert the other anchors position [01:00] <cone-580> ffmpeg.git 03Vignesh Venkatasubramanian 07master:30c5c45b123c: Adding support for parsing BlockAdditional [01:01] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:a08ebf04b1b0: matroskadec: simplify additional_id writing code [03:35] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:9f16cb9e50a5: doc/APIchanges: fix odd .01 versions [06:58] <wm4> so, how am I supposed to use libavformat in presence of sparse video packets (lots of sound packets, but video packets come only after a number of seconds, like mjpeg "slide shows" in mp4 with audio) - I think this is an API weakness, or am I overlooking something? [06:58] <wm4> trying to find the next video packets ends in being flooded with audio packets [06:58] <wm4> ffplay has the same problem [06:59] <wm4> and in presence of the cover art hack, you'd end up demuxing the whole audio file just to find out that there are no more video packets [10:47] <wm4> ffmpeg still fails to inject cover art images when trying to seek before start of the file [10:47] <wm4> or, at least my code just Works on libav [10:48] <wm4> haven't managed to reproduce it with 'ffmpeg' [10:49] <durandal_1707> how you recognise cover art on libav? [10:49] <wm4> I don't think I'm doing anything special here [10:52] <durandal_1707> so what not special are you doing? [10:53] <durandal_1707> do you know why it happens? [10:53] <wm4> not really [10:53] <durandal_1707> also what container? [10:53] <wm4> mp3 [10:54] <wm4> also converting mp3s with cover art to mkv produces strange files [10:54] <wm4> maybe ffmpeg should repeat the picture in the video stream [10:55] <durandal_1707> what "strange files" means? [10:55] <durandal_1707> doesn't ffmpeg disposition for covers and libav doesn't? [10:56] <durandal_1707> lets try again: [10:56] <durandal_1707> ffmpeg have disposition for covers and libav does not? [10:57] <wm4> they both have it set [10:57] <wm4> the AV_DISPOSITION_ATTACHED_PIC flag on the stream [10:58] <wm4> how can I seek to negative timestamps with ffplay? [10:58] <durandal_1707> ok, can we get back with strange mkv files? [10:58] <durandal_1707> wm4: with left/right cursor/arrow keys? [10:59] <wm4> the "strange" thing about the produced mkv file is that it 1. starts at 10 seconds timestamp (?), and 2. nothing can seek in it (ffplay just jumps to the start) [11:00] <durandal_1707> and resulted video stream (covert arts) is not marked as cover art? [11:00] <nevcairiel> many things try to seek to a video frame [11:00] <nevcairiel> idealy, such an attached pic should not result in a video stream in the mkv [11:00] <nevcairiel> but also an attached pic [11:00] <wm4> durandal_1707: no, the produced mkv picture is a "real" video stream [11:00] <durandal_1707> that is bug, please report it [11:01] <durandal_1707> nevcairiel: you are talking to the void [11:01] <wm4> wut [11:06] <wm4> durandal_1707: can I make ffprobe to seek? [11:12] <durandal_1707> wm4: seek in what sense? [11:12] <wm4> durandal_1707: call av[format]_seek_ [11:13] <wm4> before dumping packets with -show_packets [11:15] <durandal_1707> wm4: don't think so, feature request should be reported in usual place [11:16] <durandal_1707> wm4: why you did not tell in bug report [11:17] <durandal_1707> #2270 that disposition flag is not set? [11:17] <wm4> isn't that obvious from the output [11:18] <wm4> isn't that why you ask everyone to post the full output [11:18] <durandal_1707> not for cehoyos, instead you mentioned metadata and attachments which confused him even more [11:19] <nevcairiel> attachments are just metadata [11:19] <nevcairiel> its all in id3 tags for example in mp3 [11:19] <nevcairiel> the disposition flag probably is set, ffmpeg output just doesnt show it, btw [11:19] <wm4> oops [11:20] <durandal_1707> why would muxer create dispostion stream in h264? [11:20] <wm4> it really doesn't [11:20] <nevcairiel> the muxer just creates a video stream [11:20] <nevcairiel> it ignores the disposition, apparently [11:22] <durandal_1707> this doesn't happen with avconv? [11:22] <cone-893> ffmpeg.git 03Diego Biurrun 07master:c6507946d428: dsputil: Move STRIDE_ALIGN macro to the only place it is used [11:22] <cone-893> ffmpeg.git 03Derek Buitenhuis 07master:130cefc9dced: doc/platform: Fix 10l typo [11:22] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:8bbb487e443a: Merge remote-tracking branch 'qatar/master' [11:22] <wm4> durandal_1707: didn't check [11:23] <wm4> durandal_1707: yes avconv does the same [11:24] Action: durandal_1707 is not motiviated to fix their mess rigth now [11:27] <durandal_1707> wm4: about seek, does ffplay show same issue? [11:31] <wm4> durandal_1707: no, ffplay doesn't mind if there's no video packet after a seek, I think [11:33] <durandal_1707> whatever export attachment as stream does not work when seeking [11:40] <durandal_1707> michaelni: michaelni what is supposed output when qp filter is applied before pp one? [11:43] <michaelni> pp filter applies postprocessing depending on qp values, qp filter can change qp, higher qp stronger postprocessing [11:48] <durandal_1707> michaelni: it doesn't work because qcale<->qp_table are not copied? [11:49] <michaelni> yes, dunno if thats the only reason [11:52] <saste> durandal_1707, can you send an updated noise? [11:53] <saste> if you see no review from me within today, then feel free to push it [12:00] <wm4> durandal_1707: I made a craptacular test program: http://pastebin.com/QCFPXdUU [12:01] <wm4> durandal_1707: when I use this on a mp3 with cover art, I never see a packet from stream 1 [12:01] <wm4> durandal_1707: but when changing the negative seek target pts to something positive, it shows up [12:04] <durandal_1707> saste: i want to remove mp=softskip and mp=dsize (and mp=qp, non functional but will port it) [12:05] <durandal_1707> and maybe also ilpack [12:07] <durandal_1707> michaelni: libavcodec/h264.c:1257:30: warning: comparison of distinct pointer types ('CABACContext *' [12:07] <durandal_1707> (aka 'struct CABACContext *') and 'int16_t (*)[512]') [12:07] <durandal_1707> av_assert0(&h->cabac == &h->mb_padding + 1); [12:07] <durandal_1707> ~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~ [12:07] <durandal_1707> ./libavutil/avassert.h:38:11: note: expanded from macro 'av_assert0' [12:07] <durandal_1707> if (!(cond)) { \ [12:07] <durandal_1707> ^ [12:09] <nevcairiel> its meant to compare those pointers [12:18] <cone-893> ffmpeg.git 03Diego Biurrun 07release/1.1:077beee4653f: x86: ac3: Fix HAVE_MMXEXT condition to only refer to external assembly [12:18] <cone-893> ffmpeg.git 03Tim Walker 07release/1.1:5393a5600ddb: mlpdec: set the channel layout. [12:18] <cone-893> ffmpeg.git 03Tim Walker 07release/1.1:59f22ef91a1e: mlpdec: TrueHD: use Libav channel order. [12:18] <cone-893> ffmpeg.git 03Tim Walker 07release/1.1:5af78cc98d80: mlp: store the channel layout for each substream. [12:18] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:e1a86b1433f1: mlpdec: dont leave a invalid huff_lsb in the context. [12:18] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:e67491a2a49a: Merge commit '99ccd2ba10eac2b282c272ad9e75f082123c765a' [12:18] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:1d20d975aa89: Merge commit '3ffcccb4fbaae4d5ad775506f1f2761f2029affa' [12:18] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:9e3e11a348e0: Merge commit '1fd2deedcc6400e08b31566a547a5fac3b38cefb' [12:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:6baaaa0174bf: Merge commit '5af78cc98d807f3b43510410dad46e1840c5c99f' into release/1.1 [12:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:358e4081edb8: mlp: fix channel order. [12:24] <saste> durandal_1707, i'm not against removing mp filters if useless, but i prefer if it is michael to give the ok, since that's mostly his code [12:24] <saste> ubitux, still willing to optimize kerndeint? [12:25] <nevcairiel> try to figure out why it fails in msvc-dll builds while you're in there :D [12:28] <durandal_1707> michaelni: your opinion on mp=dsize and mp=softskip removal? [12:29] <durandal_1707> saste: michaelni did not write those two filters [12:29] <michaelni> i leave it to saste todecide [12:29] <durandal_1707> maybe thay where useful with mplayer/mencoder but thay are not with ffmpeg [12:30] <durandal_1707> i tried my best to prove they do anything and i failed [12:30] <durandal_1707> unlike mp=qp, which is useless but port could be useful [12:33] <durandal_1707> also my goal is to write more generic filters - they are way more useful [12:35] <cone-893> ffmpeg.git 03Michael Karcher 07release/1.1:901682ff78da: atrac3: use correct loop variable in add_tonal_components() [12:35] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:e0e425042193: dnxhdenc: fix invalid reads in dnxhd_mb_var_thread(). [12:35] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:e835ce83e2ac: vf_delogo: fix an uninitialized read. [12:35] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:5bee21d724dc: vf_delogo: fix copying the input frame. [12:36] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:5a3c8f95d5bf: Merge commit '5bee21d724dc47d115faae3f5065a6db74e1594a' into release/1.1 [12:41] <wm4> hm, so under some condition, libav actually fails to inject a video packet too... actually the seek can sometimes fail [12:43] <durandal_1707> saste: sent new noise [12:46] <saste> durandal_1707, thanks for the noise :) [12:46] <durandal_1707> what is different between unsharp and mp=unsharp? [12:46] <saste> durandal_1707, interface [12:46] <saste> also there is a possible out-of-buffer read in unsharp [12:47] <saste> give me some time and i'll send you a link to the thread [12:47] <durandal_1707> saste: so no reason to keep mp=unsharp? [12:47] <saste> i'd prefer to keep it until unsharp is fixed [12:47] <durandal_1707> what needs fixing? [12:47] <saste> for theoretical security reasons people may prefer to use mp=unsharp [12:48] <durandal_1707> nonsense, it is nowhere documented [12:48] <saste> durandal_1707, i forgot the details, but it is all in the thread [12:50] <durandal_1707> no, i just found thread that you found that output differs [12:51] <durandal_1707> which is WTF, filter should not be commited if it have obvious bug [12:51] <durandal_1707> like earwax [13:00] <durandal_1707> unsharp have strange macro UPSHIFT [13:00] <durandal_1707> *SHIFTUP [13:01] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:00bf66785f7d: fraps: fix off-by one bug for version 1. [13:01] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:7e35c50b81d9: yop: check that extradata is large enough. [13:01] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:108ca6fad1e0: yop: check for input overreads. [13:01] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:81bcf9454e19: Merge commit '108ca6fad1e0e9af8d6337f908bfd23807b7fbd6' into release/1.1 [13:03] <durandal_1707> ok i removing softskip and dsize [13:11] <durandal_1707> what is happening with mplayerhq.hu? I cant read its manual [13:17] <michaelni> elaborate what where doesnt work ? [13:17] <michaelni> manual i mean ? [13:17] <durandal_1707> native unsharp indeed gives different output, maybe because its syntax is broken [13:18] <durandal_1707> michaelni: not acessible, stalled [13:22] <michaelni> i dont know what you mean by "not acessible, stalled" it doesnt timeout it doesnt give http errors [13:23] <michaelni> also ive no time for guessing games [13:23] <michaelni> which URL does what ? [13:25] <durandal_1707> itsn't ffmpeg so why would it be so imporant? [13:25] <durandal_1707> no it works [13:26] <durandal_1707> *now [13:26] <michaelni> if theres a problem with the server it may be important [13:26] <michaelni> also mplayerhq.hu uses a mirror network [13:27] <michaelni> www1.mplayerhq.hu www2, ... [13:30] <iive> meaning you may have been routed to some of the mirrors that is having problems. [13:30] <durandal_1707> lol, unsharp can do some unintentional effect [13:31] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:1f8bf163e4b4: aasc: fix output for msrle compression. [13:31] <cone-893> ffmpeg.git 03Kostya Shishkov 07release/1.1:d0249f1c2e55: qtrle: fix the topmost line for 1bit [13:31] <cone-893> ffmpeg.git 03Anton Khirnov 07release/1.1:b7765d00f911: msrledec: check bounds before constructing a possibly invalid pointer, [13:31] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:7f8846405e8a: Merge commit 'b7765d00f911fe0f8fcda21b93a540f27d2ba2f5' into release/1.1 [13:32] <durandal_1707> saste: actually unsharp gives same output as mp=unsharp [13:32] <saste> durandal_1707, yes [13:33] <saste> durandal_1707, but it accepts param values for which it will read oob values [13:33] <durandal_1707> so what should be done? change syntax? [13:33] <saste> the port was ill-done [13:33] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.11:470ee0c660ed: h264_refs: Print default in case we are missing a reference. [13:34] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.11:ebe645f02b3f: h264: Reset last_pocs in case of reference or frame number inconsistencies [13:34] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.0:9c659b3a678f: h264_refs: Print default in case we are missing a reference. [13:34] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.0:169d84934423: h264: Reset last_pocs in case of reference or frame number inconsistencies [13:34] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:2ac6b573a408: h264: Reset last_pocs in case of reference or frame number inconsistencies [13:34] <saste> since the unsharp syntax is different from mp=unsharp [13:34] <saste> and it accepts different ranges [13:35] <durandal_1707> it can also crash [13:35] <durandal_1707> i want to fix obvious nonsense/useless nonfeatures/misfeatures/bugs/and atc [13:35] <durandal_1707> so what solution you prefer? [13:36] <durandal_1707> keep syntax and add missing checks? [13:45] <durandal_1707> config_props can be called several times in row without uninit? [13:55] <cone-893> ffmpeg.git 03Kostya Shishkov 07release/1.1:5479e08cc44a: xxan: properly handle odd heights. [13:55] <cone-893> ffmpeg.git 03Derek Buitenhuis 07release/1.1:4eede1fca24a: doc/platform: Fix 10l typo [13:55] <cone-893> ffmpeg.git 03Martin Storsjö 07release/1.1:5310da7e83ec: arm: Fall back to runtime cpu feature detection via /proc/cpuinfo [13:55] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:7d3e21762317: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [14:17] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/1.1:71fee2ab1e77: sws: dont write out of array on bigendian [14:19] <cone-893> ffmpeg.git 03Xi Wang 07release/0.8:e163d884ef6c: rtmp: fix multiple broken overflow checks [14:19] <cone-893> ffmpeg.git 03Xi Wang 07release/0.8:b59ee5dcf119: rtmp: fix buffer overflows in ff_amf_tag_contents() [14:19] <cone-893> ffmpeg.git 03Janne Grunau 07release/0.8:801eff785aa1: rv34: error out on size changes with frame threading [14:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.8:03ddc260668b: indeo5dec: Make sure we have had a valid gop header. [14:19] <cone-893> ffmpeg.git 03Anton Khirnov 07release/0.8:604d72aa0d05: dfa: improve boundary checks in decode_dds1() [14:19] <cone-893> ffmpeg.git 03Anton Khirnov 07release/0.8:440e98574bde: indeo4/5: check empty tile size in decode_mb_info(). [14:19] <cone-893> ffmpeg.git 03Anton Khirnov 07release/0.8:301761792a69: mpeg12: do not decode extradata more than once. [14:19] <cone-893> ffmpeg.git 03Reinhard Tartler 07release/0.8:db5b454c3d20: Update changelog for 0.7.7 release [14:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.8:e4831bb9a678: huffyuvdec: Check init_vlc() return codes. [14:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.8:4f91c4564493: huffyuvdec: Skip len==0 cases [14:19] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.8:acada70ffbd7: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 [14:23] <durandal_1707> last time to say no to dsize & softskip removal [14:26] <durandal_1707> cehoyos, bug is not wish [14:46] <durandal_1707> saste: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-February/120640.html <- why this was never applie? [14:49] <cone-893> ffmpeg.git 03Paul B Mahol 07master:968f8acec882: lavfi: remove dsize libmpcodecs wrapper [14:49] <cone-893> ffmpeg.git 03Paul B Mahol 07master:41ae43cade37: lavfi: remove softskip libmpcodecs wrapper [14:56] <saste> durandal_1707, i was not anymore sure it was a good idea [14:56] <durandal_1707> why? [14:56] <durandal_1707> please explain [14:57] <saste> maybe we should keep separate options [14:57] <saste> and use shorthands to keep syntax compatibility [14:58] <saste> having an ad-hoc specification language for the params is a bit inconsistent [14:59] <durandal_1707> saste: so the only reason why you prefer to keep mp=unsharp is because native one crash? [14:59] <saste> especially now that we have shorthands and options [14:59] <saste> does it crash? [14:59] <durandal_1707> yes [15:00] <saste> durandal_1707, as i explained, i was not confortable with the other patch which was limiting the ranges [15:00] <saste> because i couldn't tell why those ranges were choosen [15:00] <saste> and i'm sorry that i've being busy but i'd still would like to update my unsharp patchset [15:02] <durandal_1707> sorry but too much time passed [15:03] <durandal_1707> mp=unsharp clip them somehow [15:04] <ubitux> saste: not soon [15:04] <ubitux> got some other priorities somehow now [15:12] <durandal_1707> saste: mp=unsharp set blocks size to odd values between [3, 63] [15:12] <saste> durandal_1707, missing micro bumps [15:13] <saste> durandal_1707, give me some time to work on unsharp, now i gonna leave [15:14] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:7d0e3b197c81: Write the fiel atom to mov files independently of the used video coded. [15:14] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/0.10:7a21b089c275: sws: dont write out of array on bigendian [15:14] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/0.11:2f98537ea074: sws: dont write out of array on bigendian [15:14] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:140ca8f6c035: sws: dont write out of array on bigendian [15:19] <durandal_1707> this is unacceptable [15:30] <cone-893> ffmpeg.git 03Kostya Shishkov 07release/0.10:a94f789c334c: indeo3: initialise pixel planes on allocation [15:30] <cone-893> ffmpeg.git 03Luca Barbato 07release/0.10:1076ea8115ad: mp3: exit on parsing error in mp_decode_frame [15:30] <cone-893> ffmpeg.git 03Anton Khirnov 07release/0.10:a4c9260e6914: pthread: set the frame properties from the thread context, not user. [15:30] <cone-893> ffmpeg.git 03Xi Wang 07release/0.10:165f783235a0: rtpenc: fix overflow checking in avc_mp4_find_startcode() [15:30] <cone-893> ffmpeg.git 03Xi Wang 07release/0.10:69b3fedc09d9: rtmp: fix multiple broken overflow checks [15:30] <cone-893> ffmpeg.git 03Xi Wang 07release/0.10:ef953f760ef1: rtmp: fix buffer overflows in ff_amf_tag_contents() [15:30] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.10:ba4b57e8024a: huffyuvdec: Check init_vlc() return codes. [15:30] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.10:b07c79125270: huffyuvdec: Skip len==0 cases [15:30] <cone-893> ffmpeg.git 03Michael Niedermayer 07release/0.10:5b7f7f3809a8: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10 [15:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/0.10:83446017128b: Write the fiel atom to mov files independently of the used video coded. [15:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/0.11:ec18baadfa17: Write the fiel atom to mov files independently of the used video coded. [15:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:397e769f74dd: Write the fiel atom to mov files independently of the used video coded. [15:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:057051b84879: Write the fiel atom to mov files independently of the used video coded. [16:19] <michaelni> Paranoialmaniac, is there a fiel atom in mp4 like in mov ? [16:25] <TimNich> I don'y have the mp4 spec to hand but Quicktime Pro writes them to .mp4 files [16:25] <JEEB> not like QuickTime really cares about the specs [16:26] <michaelni> well iam asking because carls commit makes ffmpeg write them into mp4 too now [16:26] <nevcairiel> i probably have the iso spec somewhere, but i hate reading it :D [16:26] <JEEB> the ISO base media format spec is freely available [16:26] <JEEB> the ones that are not are the short MP4 file format spec [16:26] <JEEB> and the -15 AVC-in-"mp4" spec [16:27] <JEEB> (AVC file format) [16:27] <TimNich> Yes, but I tend to have qtff to hand at all times, but not the part 10 specs [16:27] <nevcairiel> -12 should be the isom [16:28] <JEEB> oh right, 12 [16:29] <nevcairiel> where in the hierarchy is fiel? [16:29] <JEEB> http://www.iso.org/iso/home/store/catalogue_tc/catalogue_detail.htm?csnumbe… <- click the ITTF link [16:29] <JEEB> for a free copy of -12 [16:39] <cone-893> ffmpeg.git 03Nicolas George 07master:bb3303b94c1c: lavfi/vf_ass: ignore subtitles decoding errors. [16:39] <cone-893> ffmpeg.git 03Nicolas George 07master:c557a5b08efe: lavfi/vf_ass: reindent after last commit. [16:44] <michaelni> TimNich, QT pro writes fiel also into h264 in mp4 files ? [16:47] <TimNich> not for H264, but for some other codecs..\ [16:50] <JEEB> also stuff becomes even more fun because "mp4" is not a single entity [16:50] <JEEB> given the written brand [16:51] <TimNich> ISO/IEC 14496-12:2012(E) doesn't specifically mention the 'fiel' atom, but that doesn't necessarily mean its disallowed.. [16:51] <JEEB> various brands have various feature sets [16:51] <JEEB> f.ex. QT does have some brands of its own [16:51] <JEEB> which then follow qtff [16:51] <JEEB> just like DTS and Dolby have their own, too [16:51] <JEEB> "oh ze joy" [16:52] Action: michaelni fails to find a single mp4 with fiel except what ffmpeg generates now [16:54] <TimNich> If I put proves in mp4 with QTPro I get a fiel atom [16:55] <JEEB> check the brands written [16:55] <TimNich> s/proves/prores/ [16:57] <TimNich> QTPro brand QT compatible QT&.. [16:57] <nevcairiel> why would something like h264 get a fiel atom anyway? [16:58] <TimNich> quite, why should it need it? but then why should Prores need it. they all have the infer in the stream parameters anyway... [16:58] <nevcairiel> i have no idea about prores, just know that h264 has it :D [16:59] <michaelni> so write fiel if mov || prores ? [16:59] <TimNich> nevcairiel: btw the file is in trak->mdia->stbl->stsd [16:59] <nevcairiel> there are a lot more formats that will probably get it [17:00] <TimNich> I think the matrix is more complicated than that [17:01] <michaelni> what about write if !h264 && !mpeg4 [17:02] <michaelni> mpeg4-asp video that is [17:04] <michaelni> id like to avoid it for at least things like ipod and other hw player specific files [17:05] <michaelni> i doubt it will do any good to them ... [17:06] <TimNich> The problem I see is that for .mov it is only mandated for raw video *but* included in other formats that shouldn't need it (like Prores) while similar formats, like DNX don't use it. The big question is does its presence do any harm? It should be silently ignored by applications that do not want or expect it, but can we garuantee that? [17:07] <nevcairiel> sadly plenty applications arent smart enough to simply ignore unknown data, but tend to explode [17:09] <michaelni> also theres the case that what fiel says differs from what the codec stream says where it can store it like h264 [17:10] <TimNich> I think thats important and afaik untested, which is why I wanted to do extensive testing on DX before being completely happy. [17:12] <TimNich> If the section of mov.c with all those 'if else' was turned into a set of 'case' sections it would be easier to fine tune, and only add in codecs as needed. [17:22] <durandal_1707> nobody working on bitexact dirac? [17:24] <kierank> dirac is largely dead [17:24] <durandal_1707> as codec? [17:25] <kierank> yes [17:25] <kierank> jpeg2k is all the rage [17:26] <TimNich> this week at least.... [17:28] <durandal_1707> someone from libav is working on fully working native decoder, i think i had cotact with repo but i forgot where it was [17:28] <michaelni> TimNich, dnxhd has fiel or no fiel ? [17:30] Action: Compn tried to tell people about reinventing dirac a) before it was finished and b) before there were even samples / codec was used [17:30] <Compn> no one listens to me :P [17:30] <TimNich> dnxhd has no fiel in my "genuine" samples [17:30] <michaelni> ok, thx [17:31] <TimNich> got to go now back tomorrow [17:33] <durandal_1707> vf_transpose have interesting way to detect number of data[planes] [17:50] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:405cc0d90520: movenc: hotfix, dont store fiel for h264 / mpeg4-asp / dnxhd [17:50] <cone-893> ffmpeg.git 03James Almer 07master:8c95d177dfa9: fate: Add encrypted tta stream test [19:11] <durandal_1707> i cant use ffplay with -filter_complex? [19:25] <llogan> durandal_1707: "Set noise seed for pixel component or all of them" <- this phrase doesn't make sense to me. what is "them"? [19:27] <durandal_1707> all pixel components [19:32] <llogan> "Set noise seed for specific pixel component of all pixel components" or something would be more clear to lusers like me [19:32] <llogan> s/of/or [19:32] <iive> how about "set noise seed for one or all pixel components" ? [19:34] <llogan> "set noise seed for all-one, all-one, all-one: all-one or none" (can you guess the reference?) [19:38] <iive> nope [19:46] <durandal_1707> i think i will add float and double pixel format [19:52] <llogan> iive: some weird-ass hippy soap in the US. Dr. Bronners. [19:54] <iive> oh, i guess they haven't ported it to EU market yet. [19:56] <ubitux> durandal_1707: i've been away for a while and saw a lot of activity from you, are you waiting for some particular review? [20:11] <durandal_1707> ubitux: yes, unsharp, kerndeint and tinterlace removal from mp [20:17] <durandal_1707> ubitux: there is also trivial overlay patch that adds yuv444p but it somehow is broken with ffplay, because it inject strange conversion in middle of chain, breaking chroma values [20:18] <durandal_1707> dunno if that is lavfi bug [20:22] <ubitux> michaelni: do you expect someone to check all the hash or you're waiting for more general reviews? [20:22] <ubitux> durandal_1707: about unsharp and tinterlace, dunno [20:22] <ubitux> about kerndeint i had a relative performance concern, which i wasn't really able to understand [20:23] <ubitux> i have no idea when i'll have time to look again at it [20:23] <ubitux> i'm not sure to be the best person for the yuv444p thing; i think i've actually already expected a similar problem with ffplay [20:28] <michaelni> ubitux, dunno, i had no specific expectattions [20:28] <ubitux> sorry i'm not motivated to check every single hash :)) [20:29] <michaelni> :) [20:29] <michaelni> they should be ok [20:29] <michaelni> version numbers are trickier to lookup so higher chance that theres a wrong one in it [20:30] <michaelni> i mean version numbers are not unique before we added .100 [20:30] <michaelni> also there are typos in the file [20:36] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:ad6802f975a9: apichanges: fix date [20:36] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:33d6330652c0: apichanges: Use , instead of / to seperate multiple hashes [20:55] <durandal_1707> ubitux: also you may review noise [21:42] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:2f3bc5122822: apichanges: fix 2 wrong hashes [21:43] <cone-893> ffmpeg.git 03Vignesh Venkatasubramanian 07master:ce6a8e5947ca: Adding AlphaMode element to Matroska Parser [21:49] <ubitux> durandal_1707: ok, i'm having a look [22:16] <ubitux> michaelni: i won't make any more review for APIChanges, so you can consider it a LGTM from me [22:51] <durandal_1707> ubitux: filters that use filter_complex need PRESERVE? [22:53] <ubitux> filters that use filter_complex? [22:53] <ubitux> what does that mean? [22:54] <ubitux> i'm not familiar with the preserve thing btw [22:54] <durandal_1707> filter that operates on > 1 buffers [22:54] <durandal_1707> one need to queue such buffs [22:55] <durandal_1707> i encountered problem with my filter if i use 2 images as inputs i will get nothing as output [22:56] <durandal_1707> may not be related to it at all [22:56] <ubitux> that's what i was going to say [22:58] <ubitux> * Filters that intend to keep a reference after the filtering process [22:58] <ubitux> is finished (after filter_frame returns) must have the PRESERVE [22:58] <ubitux> permission on it and remove the WRITE permission if they create a new [22:58] <ubitux> reference to give it away. [22:58] <ubitux> next point might be related too [23:02] <ubitux> divVerent: ping [23:11] <cone-893> ffmpeg.git 03Clément BSsch 07master:cf8dec7d6421: lavfi/showspectrum: pretty-align constants. [23:11] <cone-893> ffmpeg.git 03Clément BSsch 07master:022437518086: lavfi/showspectrum: simplify intensity_color_table declaration. [23:47] <cone-893> ffmpeg.git 03Clément BSsch 07master:35a995f4518a: doc/resampler: fix two typo. [23:47] <cone-893> ffmpeg.git 03Clément BSsch 07master:d5ce725cb32b: Fix a few "its" vs "it's" typo. [23:59] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:c3fb20bab4f0: ffmpeg: Check for parameter changes at the output of the audio filter graph [00:00] --- Fri Feb 15 2013

1 0

ffmpeg.log.20130214
by burek 15 Feb '13

15 Feb '13

[00:32] <meelu> is there a list of codecs ffmpeg supports, where is allows libmp3 conversion [00:32] <meelu> from video [00:32] <meelu> i mean is there a list of video codecs it supports where it can convert to mp3 [00:33] <llogan> i don't quite understand your question, but did you see "ffmpeg -codecs"? [00:33] <meelu> tried ffmpeg -formats, is this it [00:34] <meelu> i want to know if i can use "-y -acodec libmp3lame -ab " on all llogan [00:34] <llogan> you want to know what formats support mp3 audio? [00:35] <llogan> i meant "containers", not "formats" [00:35] <meelu> i want to know which type of videos i can run this command on [00:35] <meelu> %ffmpeg% -i %tempFile% -vol %volume% -y -acodec libmp3lame -ab %quality%k %newFile% [00:35] <llogan> you can run that on any input file [00:35] <meelu> ah great [00:35] <meelu> so i can do that with anything [00:36] <llogan> although consider using -q:a instead of -ab [00:36] <meelu> im going to make something using php to verify a file is supported when a user uploads it [00:36] <meelu> why? [00:36] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/Encoding%20VBR%20%28Variable%20Bit%20Ra… [00:42] <tseth> hi. can someone help me with this error? [00:42] <tseth> h264_mp4toannexb failed for stream 0, codec copy: Invalid argume [00:42] <tseth> argument* [00:42] <tseth> ffmpeg -i fs0 -vcodec copy -acodec copy -vbsf h264_mp4toannexb -f mpegts ts_stream0.ts [00:42] <tseth> that is my command... fs0 is a rawdump of an h264 stream nals [00:50] <tseth> better question... does anoyne know how to convert raw h.264 over to mpeg2-ts [00:50] <tseth> ? [00:52] <cps> Hi, I'm working on the open source animation program Pencil2D. I'm trying to get file export working using ffmpeg. [00:53] <cps> As a test, I'm trying to encode a vector of RGB image data (derived from a png). [00:54] <cps> I have a couple of questions about the decoding_encoding.c example. [00:54] <cps> Firstly, is this the right place to ask? [00:55] <klaxa> cps: if you are using ffmpeg and not the fork (avconv) then yeah probably [00:56] <cps> Great! [00:58] <cps> The example uses "AV_PIX_FMT_YUV420P". What I'm not clear about is how to convert and then write the image data to the AVFrame [01:00] <cps> Am I right in thinking that I need to convert the RGB data to YCbCr for each pixel, and then average the YCbCr data for each quad? [01:06] <tseth> so I basically realized what's happening. I'm trying to put an mpeg2-ts container around an h264 payload... when I run this command: ffmpeg -i fs0 -acodec copy -vbsf h264_mp4toannexb -f mpegts stream0.ts the 10 second stream pops out a 00:00:0.4 seconds long. [01:06] <tseth> any idea what I'm doing wrong? [01:16] <tseth> can someone please help me out? I'm getting incredibly frustrated? [01:18] <tseth> llogan: http://pastie.org/6158686 [01:18] <tseth> the video clip is 10 seconds long. [01:19] <tseth> it's a raw h264 dump, all I'm trying to do, is to convert it to an mpeg2-ts format that safari will play nice with. [01:19] <tseth> if anyone has any idea how to help, it would really mean the world to me. [01:21] <llogan> ask in #libav. you're not using ffmpeg from FFmpeg. [01:21] <llogan> we do not support third party tools here. [01:22] <tseth> llogan: am I able to do this with ffmpeg? [01:23] <llogan> you're not using ffmpeg. [01:23] <tseth> damn... I just used the apt-get for ffmpeg.... [01:23] <tseth> how can I get ffmpeg? [01:23] <llogan> compile or use a static build [01:23] <llogan> the "ffmpeg" is a result of others misrepresenting the name ffmpeg. [01:24] <llogan> *the "ffmpeg" in the ubuntu repo [01:24] <tseth> does ubuntu not have an ffmpeg. [01:24] <tseth> and is libav essentially the same thing? (honest question... I have no idea.) [01:24] <llogan> no. it's a fork. [01:24] <llogan> see http://stackoverflow.com/a/9477756/1109017 [01:24] <llogan> and http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html [01:25] <llogan> ubuntu no longer provides ffmpeg from FFmpeg, but a fake "bizarro" version from libav. [01:25] <tseth> why would they do that? [01:26] <tseth> Would the real ffmpeg solve the issue I'm bumping into here? [01:27] <llogan> because the debian/ubuntu maintainer is a member of the fork and they want users to switch to libav (while allowing users to bleieve they are using ffmpeg from FFmpeg). [01:27] <llogan> i'm not sure if ffmpeg will help, but we simply dont support fake ffmpeg here. [01:28] <tseth> If this were a real version of ffmpeg, what would you suggest to do this? [01:29] <tseth> libav looks dead, and I'm frustrated, and you're the only one who seemed to be willing to help :-( [01:29] <tseth> #libav* [01:29] <llogan> i suggest trying it with a static build from the link i provided. it will be simple to try. [01:29] <llogan> i've never been in that channel. i don't know if they often try to help users. [01:30] <tseth> will downloading that ffmpeg break the current one? [01:31] <klaxa> tseth: no, you can put it in any directory you want and execute it from there [01:31] <llogan> no. you simply download the archive, extract the archive, and run the binary [01:31] <klaxa> no installation needed whatsoever [01:34] <llogan> do you mean to be re-encoding the video? also your input has no audio, so -acodec copy is superfluous. [01:35] <tseth> oh wow. looks like this version worked. [01:41] <tseth> llogan: you are correct. [01:42] <tseth> any idea why what I'm doing wouldn't work with http live streaming? it half worked :-) [01:42] <klaxa> that is often the case :) [01:42] <tseth> it starts the video, plays through my section, then craps out. [01:42] <tseth> :-) [01:42] <llogan> it's what I call YALB...yet another libav bug [01:43] <llogan> "it" as in safari? [01:48] <tseth> ah, gotchya. [01:48] <tseth> hahaha. [01:48] <tseth> yeah, I think I tracked the issue down :-) [01:48] <tseth> one last question... what is the licensing on ffmpeg? [01:48] <tseth> (hoping for mit or berkeley, expecting gpl v2) [01:53] <klaxa> i think most is lgpl [01:53] <klaxa> http://ffmpeg.org/legal.html [02:06] <skered-> I have an mp3 file that only has right audio, can ffmpeg make an audio file that has the right channel for both left/right (or a mono file using the right side)? [02:08] <tseth> lgpl = more restrictive, no? [02:21] <base698> Hi All! I have this command: ffmpeg -i pipe:0 -acodec libvorbis test.ogg and it works great. Just wanted to know if you can do something like this: ffmpeg -i pipe:0 -acodec libvorbis pipe:1 > somefile.ogg [02:21] <base698> I had to rebuild ffmpeg for ogg support and in doing that it changed the man page, which did have a huge section about pipes [02:25] <klaxa> base698: you probably have to specify a container format with -f [02:25] <klaxa> for example -f ogg [02:25] <llogan> base698: man pages have been segmented. see the end of "man ffmpeg". [02:25] <llogan> such as: man ffmpeg-protocols [02:25] <base698> holy shit success! [02:26] <base698> and thanks even more for the man info [02:26] <base698> that's the man page I remember [02:47] <iive> tseth: lgpl is less strict than gpl. [03:10] <defaultro> so happy, -vf hue=30 fixed the greencast from my videos :) [03:16] <tseth> new question... [03:16] <tseth> is it possible to use ffmpeg to resize a video? [03:17] <klaxa> yes [03:17] <klaxa> see -s [03:17] <tseth> what would the options be for doing such said black magic? [03:17] <klaxa> i think it should also be possible to use -vf scale [03:17] <tseth> so if I wanted to reencode to 720p, how would I do that? [03:18] <klaxa> ffmpeg -i <input> <input options> -s 1280x720 output.container [03:18] <klaxa> i think that should work [03:18] <klaxa> i think you can put -s 1280x720 even before <input options> but not before -i <input> [03:19] <tseth> hmm, looks like this should already be done :-/. [03:19] <tseth> for some reason the ts I cvreate doesn't work on the ipad. [03:19] <klaxa> it's also rather output options really [03:19] <klaxa> not input options, silly me [03:19] <tseth> it does work on the safari desktop thoguh. [03:19] <tseth> for live streaming. [03:19] <tseth> I know this isn't your area, but.... [03:20] <tseth> ffmpeg -i fs0 -vbsf h264_mp4toannexb stream0.ts am I missing some options to make this a valid http live stream? [03:20] <tseth> for the ipad* [03:20] <klaxa> heh, no idea [03:20] <klaxa> i have no ipad and everything i stream via http is done with matroska [03:21] <llogan> -vf scale is more flexible than -s and has the advantage of working easier with other filters in a filterchain [03:21] <llogan> such as -vf scale=1280:-1 if you want it to automatically figure out the height to preserve aspect [03:22] <tseth> gotchya. [03:22] <tseth> hmm, that's really weird. [03:22] <tseth> haha. [03:24] <llogan> i'm assuming your command is re-encoding H.264 to mpeg2 video, so -vbsf h264_mp4toannexb is not needed if that is what you want [03:25] <tseth> llogan: I don't think I want to be reencoding either. I think what I want is just to reencapsulate? [03:26] <tseth> unless I'm completely misunderstanding. [03:26] <llogan> add "-codec copy" as an output option with "-bsf:v h264_mp4toannexb" [03:30] <tseth> this may sound like a really strange question... [03:30] <tseth> but does ffmpeg use cron at all? [03:35] <llogan> you can invoke just about anything with cron [03:38] <tseth> you're right. ffmpeg is the only thing I was running. I ended up with about 100 copies of it. [03:38] <tseth> haha. [03:38] <tseth> there's something screwed with that server.... [03:39] <tseth> Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [03:39] <tseth> llogan: I'm getting that :-) [03:40] <tseth> http://pastie.org/6159610 [03:41] <tseth> llogan: that's the output :-) [03:51] <llogan> tds5016: i don't know why it's doing that and I can't duplicate it [03:51] <tds5016> any ideas llogan ? [03:51] <tds5016> no worries. Sorry. :-/. I always get lucky with this. [03:52] <tds5016> This problem is so stupidly obnoxious... haha. [03:53] <llogan> ask on ffmpeg-user mailing list if you don't figure it out. make sure to include your complete ffmpeg command and the complete console output (you can omit the repeated "failed" messages but leave at least one) [03:54] <tds5016> could I be missing timecode info? [03:57] <tds5016> Stream format for the stream segmenter must be MPEG elementary audio and video streams, wrapped in an MPEG-2 transport stream, and using the following encoding. [03:57] <tds5016> hmmmm..... [03:57] <tds5016> how do I check if I've already got an elementary video stream? [04:00] <tds5016> sorry if that's a stupid question... I'm just triyng ot figure all of this out :-/. [04:14] <tds5016> is there a differennce between baseline and elementary? [04:18] <relaxed> elementary stream means a raw h264 file. (not in a container) [04:18] <tds5016> I thinkt hat's what I have :-/. how could I tell? [04:19] <relaxed> run ffmpeg -i file [04:19] <relaxed> look at input format [04:20] <relaxed> For example: Input #0, h264, from 'fs0': [04:20] <tds5016> http://pastie.org/6159849 [04:20] <tds5016> also, btw...... pastie.com and pastie.org... NOT the same thing ;-) [04:20] <relaxed> so, yes, your input is a h264 elementary stream [04:21] <tds5016> then I have no idea why in the hell mobile safari is choking on it :- [04:21] <tds5016> :-( [04:21] <tds5016> http://pastie.org/6159854 [04:22] <tds5016> is that the right output for what should be an mpeg2-ts with a payload of an elementary h264 stream? [04:22] <relaxed> your video is now mpeg2, is that what you want? [04:22] <relaxed> no [04:22] <tds5016> I want h264 wrapped in mpeg2-ts [04:22] <tds5016> relaxed: can you help me figure out how I'd do that? [04:23] <relaxed> "ffmpeg -i fs0 -c:v copy -f mpegts output.ts" [04:23] <tds5016> http://pastie.org/6159861 [04:24] <relaxed> NO, run what I just gave you. [04:24] <relaxed> -vbsf h264_mp4toannexb is only needed when demuxing h264 from the mp4 container. [04:25] <relaxed> or stream copying [04:25] <tds5016> hmm, that looked like it broke :-/. [04:25] <tds5016> it only plays for a portion of a second, then breaks. [04:26] <relaxed> pastie.org or it didn't happen [04:26] <tds5016> Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [04:26] <tds5016> I keep getting that :-( [04:26] <tds5016> sad face. [04:26] <tds5016> haha. [04:26] <tds5016> when I add that section back in. [04:27] <relaxed> I need pastebins and not a personal recount from you. [04:28] <tds5016> http://pastie.org/6159883 [04:28] <tds5016> that's without the mp4toannexb [04:29] <tds5016> for some reason when I get hte file, it only plays a portion of a second of it, then it stops playing. [04:30] <tds5016> oh, it looks like it's a problem with the video tag auto playing.... [04:30] <tds5016> it looks like it gets what it can from http, then just says "screw it, I'm at the end" then dies if it doesn't have the whole file. [04:31] <defaultro> does ffmpeg support denoise? [04:32] <tds5016> relaxed: do you have any idea how I'd be able to fix that? or that's a web problem? [04:35] <tds5016> hmmm.... magically started working. [04:35] <tds5016> you guys rock! [04:35] <tds5016> thank you so much! [04:35] <tds5016> well, partially worked... [04:36] <tds5016> still breaks on ipad in the video tag [04:53] <tds5016> thanks for all of the help guys. you are phenomenal people, all of you! [05:31] <relaxed> defaultro: ffmpeg -filters | grep -i denoise [05:38] <defaultro> thanks relaxed! [05:39] <defaultro> hqdn3d V->V Apply a High Quality 3D Denoiser. [08:13] <willbradley> hey is ffmpeg good for extracting h264 data from inside a large binary? [08:13] <willbradley> google is failing me [08:49] <creep> At the end of the month, Jaimie Alexander was injured on the London film set, after she slipped while walking in the rain. [08:49] <creep> breaking news haha [08:50] <creep> willbradley<< ffmpeg plays it, you do "extracting" with dd for example [08:52] <robotics> Hi everyone! Sorry for interrupting you, but, does anybody know, why while converting avi to mpeg2, one second of the end of video is lost? [08:52] <robotics> *why when... [08:53] <creep> robotics<< hi, no problem, i'll put the chips down while writing then [08:54] <robotics> creep: i'm sorry, i don't know english very well. Could you explain in detail [08:56] <creep> robotics<< you know, when you eat you hold food in your hand you would want to eat and take it to your mouth usually [08:57] <robotics> ok, don't waste your time. [11:10] <jeje34> hello to all :-) [11:21] <jeje34> I have a question about av_init_packet and av_new_packet. In my code, when I have a H264 frame to decode with FFMPEG 1.1.1, I use av_init_packet and av_free_packet. But, I see av_new_packet seems be better because it can realloc if necessary with FF_INPUT_BUFFER_PADDING_SIZE and set the av_destruct pointer. With av_init_packet, I call av_init_packet, I make a temporary buffer sizing it to my [11:21] <jeje34> (H264 frame size+FF_INPUT_BUFFER_PADDING_SIZE), memset it to 0, copy my H264 frame in this one and so affect the AVPacket.data to this buffer and the size to the H264 frmae size. When finish decoding, I use av_free_packet to free the memory. I try to use av_new_packet because I don't have to create my temporary buffer, but I have an error on the av_free_packet (in reality in av-destruct-packet) [11:23] <jeje34> between the av_new_packet and av_free_packet I call the avcodec_decode_video2 on this packet, so the size of this packet after decoding is set to 0 [11:55] <jeje34> another question I have is how to use the av_lock_mgr because in liv AVCodec 5.2 (I know it's old) I had a function called av_lockmgr_addref and I use it doing if (av_lockmgr_addref()==1)av_lockmgr_register(&ff_lockmgr); But now, it seems there's only the function av_lockmgr_register. Do I need to call it at each thread using ffmpeg? [12:50] <Bor0> how can I correctly increase the audio volume of a video that I'm displaying with SDL? I tried something like for (int i=0;i<len1;i++) {... *p *= (is->volume / 100.0); } but increasing amplitude like this will generate noise at a point (and above it) [12:50] <Bor0> are there any native functions for this? [12:54] <Mavrik> there's an audio filter for that [12:54] <Mavrik> btw, volume is always adjusted on logarithmic scale [12:55] <Bor0> can you point me to some examples? can libavfilter do this? I'd rather not do it manually [12:55] <durandal_1707> see volume filter [12:58] <Bor0> ok, and regarding the logarithmic scale thing, if I were to do it manually (out of curiousity), then I'd multiply by log(x), instead of x? 0 < x < 100 [13:00] <Mavrik> http://dsp.stackexchange.com/questions/2990/how-to-change-volume-of-a-pcm-1… [13:00] <Mavrik> this should help a little ;) [13:19] <Bor0> excellent, thanks. [13:27] <funyun_> hi. if i want to encode 2 videos into 1, how would i go about doing that? ffmpeg -i vid1 -i vid2 ? [13:37] <durandal_1707> burek: the nick arg works for !pb only. why? [13:46] <funyun_> saste: here's what i understood from that but it's not working. any idea why? i did this, ffmpeg -i vid1.wmv -i vid2.wmv -filter_complex '[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a]' -map '[v]' -map '[a]' -vcodec libx264 -preset veryfast -tune film -x264opts level=41:deblock=-2,-2 -crf 18 test.mp4 [13:46] <funyun_> the error i get is Stream specifier ':0' in filtergraph description [0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a] matches no streams. [13:51] <funyun_> anyone know what i'm doing wrong? [14:14] <saste> funyun_, you're missing to link the graph outputs to the output file [14:14] <saste> -map [a] -map [v] -codec:v libx264 ... [14:15] <saste> uhm not really [14:17] <funyun_> saste: http://pastie.org/6163183 [14:21] <saste> funyun_, can't see nothing wrong in your command, and it works here [14:22] <saste> you might try to update your ffmpeg [14:22] <funyun_> alright, i'll try [14:52] <funyun_> saste: now i get "Unrecognized option 'preset'." [16:30] <diroots> hi there [16:31] <jeje34> Hi, is it better to in order:1- avcodec_all_context3(NULL) 2- avcodec_find_decoder(CODEC_IH_H264) to finally 3- avcodec_open2( CodecCtx, Codec, NULL); or is better to directly do 1- Codec = avcodec_find_decoder(CODEC_ID_H264) , 2- avcodec_alloc_context3(Codec) and so 3- avcodec_open2( CodecCtx, Codec, NULL); because the 2 methods works but which is the best??? [16:31] <diroots> I have a source (H.264) video with a Stream #0.2 : Data: unknown (tmcd), do you know wtf this is? [16:32] <diroots> and when I re-encode it to x264, I loose this stream #0.2 is there a way to keep it within the output video ? [18:10] <taylanub> Is $CC being ignored by ffmpeg's build ? I see a line in config.log, but then it doesn't seem to use it. [18:45] <ubitux> taylanub: yes iirc it's ignored [18:45] <ubitux> i sent a patch a long while ago for this [18:45] <ubitux> but it was somehow rejected for a reason like& [18:46] <ubitux> "ppl tend to put crap in it" [18:47] <taylanub> ... [18:48] <ubitux> actually it was at the time of the fork [18:48] <ubitux> http://lists.libav.org/pipermail/libav-devel/2011-April/001454.html the thread died after this [18:49] <relaxed> taylanub: can't you use --cc= [18:52] <taylanub> relaxed: I can, but of course I'd prefer it to just use my global $CC. [19:17] <relaxed> taylanub: -cc=$CC done [19:20] <taylanub> Still not as good as: <nothing> done :P [19:46] <ubitux> relaxed: that's especially troublesome when you are using an automated build of ffmpeg, like let's say, in ffmpeg [19:47] <ubitux> in mplayer* [21:24] <john_____> hi, does anyone has some knowledge about libstagefright.cpp ? [21:50] <durandal_1707> john_____: what is issue? [22:09] <funyun_> hi. i'm trying to encode 2 videos into 1 video. but i get this error http://pastebin.com/1r2tumWx , can anyone help me? [22:15] <funyun_> anyone have any idea? [22:23] <funyun_> JEEB: are you familiar with concat in ffmpeg? [22:25] <durandal_1707> funyun_: concat what? [22:25] <durandal_1707> protocol?filter?demuxer? [22:25] <ubitux> seems to be the filter here [22:31] <funyun_> durandal_1707: yes, filter [22:31] <funyun_> durandal_1707: here's my problem http://pastebin.com/1r2tumWx [22:51] <llogan> ubitux: looks like ffmpeg in Extra now has your config requests. [22:52] <llogan> s/extra/testing [22:53] <ubitux> llogan: oh? all of them? [22:55] <ubitux> funyun: try to put the audio streams after the video ones maybe [23:15] <tds5016> hi. I was wondering if anyone could send a list of dependencies for ffmpeg? [23:16] <llogan> ubitux: i think so [23:17] <ubitux> nice ok [23:17] <llogan> tds5016: depends on what you want [23:18] <tds5016> ffmpeg -i fs3 -c:v copy -f mpegts stream3.ts [23:18] <tds5016> enough to do that :-P [23:20] <tds5016> how would I track down the dependencies on that? [23:20] <tds5016> thanks for the help man. llogan you are awesome. [23:26] <ubitux> tds5016: gcc [23:26] <ubitux> i guess that's all [23:26] <tds5016> really? no deps aside from that for what I'm doing there? [23:26] <ubitux> for remuxer i don't think so [23:26] <ubitux> remuxing* [23:27] <ubitux> yasm is generally recommended for a ffmpeg build though [23:27] <tds5016> yasm? [23:27] <ubitux> assembler [23:27] <ubitux> for the optims [23:27] <ubitux> (mostly in codecs) [23:27] <tds5016> I'll be crosscompiling using gcc. [23:28] <tds5016> I don't think I can use yasm for this. [23:28] <tds5016> what would the configure options be to allow what I have there to work? [23:28] <ubitux> none [23:29] <ubitux> you can disable a lot of things though [23:29] <ubitux> like --disable-everything and just enable the few flags you need [23:29] <ubitux> s/flags/things/ [23:31] <tds5016> awesome. so just configure and it'll install? I don't need to disable/enable anything and it'll all work? [23:31] <tds5016> correction: SHOULD all work? [23:35] <ubitux> yes [23:36] <ubitux> note: you don't need to make install to test, make is enough, you can ./ffmpeg [23:37] <tds5016> thanks! [23:37] <tds5016> you rock :-) [00:00] --- Fri Feb 15 2013

1 0

ffmpeg-devel.log.20130213
by burek 14 Feb '13

14 Feb '13

[00:31] <j-b> Compn: indeed [01:16] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:4f1279154ee9: shorten: dont leave invalid channel counts in the context. [04:36] <cone-884> ffmpeg.git 03Daniel Verkamp 07master:2acd5fb55325: doc/demuxers: quote command line arguments with @code [04:52] <Compn> nevcairiel : highgod is going to post his 3rd patch ,adding opencl support to scale filter [04:53] <Compn> one of these days i'm just going to commit his stuff :P [06:45] <Compn> https://github.com/nblock/pkgbuild/blob/master/otrdecoder/PKGBUILD [06:45] <Compn> what this otrkey thing is [06:46] <Compn> will have to check it later, maybe some germans can shine a light [10:01] <nevcairiel> Compn: why would i care about opencl [10:01] <nevcairiel> there is also general uglyness in the patches, anyone can review them :p [11:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:a3de4010c2d0: 8svx: decode directly to the user-provided AVFrame [11:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:ffd2123095bd: aac: decode directly to the user-provided AVFrame [11:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:55d2e12aefa2: ac3: decode directly to the user-provided AVFrame [11:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:e57daa876bf0: adpcm: decode directly to the user-provided AVFrame [11:32] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:4789955ec426: Merge commit 'e57daa876bf0cf50782550e366e589441cd8c2bd' [11:44] <cone-580> ffmpeg.git 03Justin Ruggles 07master:0588935e64fe: adx: decode directly to the user-provided AVFrame [11:44] <cone-580> ffmpeg.git 03Justin Ruggles 07master:bae4f47938bb: adxenc: alloc/free coded_frame instead of keeping it in the ADXContext [11:44] <cone-580> ffmpeg.git 03Justin Ruggles 07master:5cd597f22ff5: alac: decode directly to the user-provided AVFrame [11:44] <cone-580> ffmpeg.git 03Justin Ruggles 07master:9b0b355e9731: als: decode directly to the user-provided AVFrame [11:44] <cone-580> ffmpeg.git 03Justin Ruggles 07master:e3db34291f44: amrnb: decode directly to the user-provided AVFrame [11:44] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:d13f434dbb55: Merge commit 'e3db34291f4401a16f6ac92721617a9f33cd4c31' [11:51] <cone-580> ffmpeg.git 03Justin Ruggles 07master:da28bb3f4a7c: amrwb: decode directly to the user-provided AVFrame [11:51] <cone-580> ffmpeg.git 03Justin Ruggles 07master:5932e2d7d2c0: ape: decode directly to the user-provided AVFrame [11:51] <cone-580> ffmpeg.git 03Justin Ruggles 07master:0ded61b57a3d: atrac1: decode directly to the user-provided AVFrame [11:51] <cone-580> ffmpeg.git 03Justin Ruggles 07master:9a75ace2b2cb: atrac3: decode directly to the user-provided AVFrame [11:51] <cone-580> ffmpeg.git 03Justin Ruggles 07master:5cc0bd2cb47c: binkaudio: decode directly to the user-provided AVFrame [11:51] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:f03cdbd0454f: Merge commit '5cc0bd2cb47cbb1040f2bb0ded8d72a442c79b20' [11:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:a6bb39add273: pcm: decode directly to the user-provided AVFrame [11:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:e42e5a89d623: bmvaudio: decode directly to the user-provided AVFrame [11:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:cddf8998f1af: comfortnoise: decode directly to the user-provided AVFrame [11:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:7b78321597a9: cook: decode directly to the user-provided AVFrame [11:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:182821cff43f: dca: decode directly to the user-provided AVFrame [11:58] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:0a5138695aa4: Merge commit '182821cff43f5f977004d105b86c47ceb20d00d6' [12:03] <cone-580> ffmpeg.git 03Justin Ruggles 07master:97c7bdc6d417: cinaudio: decode directly to the user-provided AVFrame [12:03] <cone-580> ffmpeg.git 03Justin Ruggles 07master:b8e9c99ef19c: flac: decode directly to the user-provided AVFrame [12:03] <cone-580> ffmpeg.git 03Justin Ruggles 07master:036e9b045f14: g722: decode directly to the user-provided AVFrame [12:03] <cone-580> ffmpeg.git 03Justin Ruggles 07master:7e52fd6bca83: g723.1: decode directly to the user-provided AVFrame [12:03] <cone-580> ffmpeg.git 03Justin Ruggles 07master:cb7b47a61dba: g726: decode directly to the user-provided AVFrame [12:03] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:d88e674a15dc: Merge commit 'cb7b47a61dba0b9329ecede5dd3211dc0662dc05' [12:12] <cone-580> ffmpeg.git 03Justin Ruggles 07master:6fdfdb23d36f: gsm: decode directly to the user-provided AVFrame [12:12] <cone-580> ffmpeg.git 03Justin Ruggles 07master:9b28e583572c: imc/iac: decode directly to the user-provided AVFrame [12:12] <cone-580> ffmpeg.git 03Justin Ruggles 07master:b8788671288e: dpcm: decode directly to the user-provided AVFrame [12:12] <cone-580> ffmpeg.git 03Justin Ruggles 07master:a8ea936a0a00: libilbc: decode directly to the user-provided AVFrame [12:12] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:5459cf411412: Merge commit 'a8ea936a0a00570f61a16a588821b52f6a3115c2' [12:18] <cone-580> ffmpeg.git 03Justin Ruggles 07master:4f69612d3e61: libgsm: decode directly to the user-provided AVFrame [12:18] <cone-580> ffmpeg.git 03Justin Ruggles 07master:0cd08367dd6a: libopencore-amr: decode directly to the user-provided AVFrame [12:18] <cone-580> ffmpeg.git 03Justin Ruggles 07master:19b2cb268f5e: libopus: decode directly to the user-provided AVFrame [12:18] <cone-580> ffmpeg.git 03Justin Ruggles 07master:bed957bb1116: libspeex: decode directly to the user-provided AVFrame [12:18] <cone-580> ffmpeg.git 03Justin Ruggles 07master:86bfcfcf2364: mace: decode directly to the user-provided AVFrame [12:18] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:afe30fe06008: Merge commit '86bfcfcf2364bc837b7bb582c66a8a15a332414f' [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:dc33fbbfd0f4: mlp/truehd: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:2c785e250a1f: mpegaudio: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:3a23752c5a06: mpc7/8: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:0fe4056f953a: nellymoser: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:0905c96390b0: pcm-bluray: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Justin Ruggles 07master:1b9b6d6e5ea5: qcelp: decode directly to the user-provided AVFrame [12:27] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:65da7007043c: Merge commit '1b9b6d6e5ea556b6d307f9d473f54f6406fdc3c8' [12:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:e01e1a4673a5: qdm2: decode directly to the user-provided AVFrame [12:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:cbeb3ed38d9d: ralf: decode directly to the user-provided AVFrame [12:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:f7e8c87c02dc: ra144: decode directly to the user-provided AVFrame [12:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:79fb2a1f1725: ra288: decode directly to the user-provided AVFrame [12:32] <cone-580> ffmpeg.git 03Justin Ruggles 07master:5d5c248c3df3: s302m: decode directly to the user-provided AVFrame [12:32] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:08059f6150fc: Merge commit '5d5c248c3df30fa91a8dde639618c985b9a11c53' [12:40] <cone-580> ffmpeg.git 03Justin Ruggles 07master:fed74c0ae497: shorten: decode directly to the user-provided AVFrame [12:40] <cone-580> ffmpeg.git 03Justin Ruggles 07master:09d6831f4901: sipr: decode directly to the user-provided AVFrame [12:40] <cone-580> ffmpeg.git 03Justin Ruggles 07master:903b62cc0b88: smackaud: decode directly to the user-provided AVFrame [12:40] <cone-580> ffmpeg.git 03Justin Ruggles 07master:4a2b26fc1b1a: tak: decode directly to the user-provided AVFrame [12:40] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:2becf21d9fd8: Merge commit '4a2b26fc1b1ad123eba473a20e270f2b0ba92bca' [12:47] <cone-580> ffmpeg.git 03Justin Ruggles 07master:3997fef95262: truespeech: decode directly to the user-provided AVFrame [12:47] <cone-580> ffmpeg.git 03Justin Ruggles 07master:ad2104ba449a: tta: decode directly to the user-provided AVFrame [12:47] <cone-580> ffmpeg.git 03Justin Ruggles 07master:3b7d43383f5e: twinvq: decode directly to the user-provided AVFrame [12:47] <cone-580> ffmpeg.git 03Justin Ruggles 07master:f80f8dd4c25a: vmdaudio: decode directly to the user-provided AVFrame [12:47] <cone-580> ffmpeg.git 03Justin Ruggles 07master:ee6ca11b6575: vorbis: decode directly to the user-provided AVFrame [12:48] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:dca6fb08a71b: Merge commit 'ee6ca11b657515ad736ec0d2b8635e098d0a2680' [12:53] <cone-580> ffmpeg.git 03Justin Ruggles 07master:9873d71f316c: wavpack: decode directly to the user-provided AVFrame [12:53] <cone-580> ffmpeg.git 03Justin Ruggles 07master:f4a283eec49b: wmapro: decode directly to the user-provided AVFrame [12:53] <cone-580> ffmpeg.git 03Justin Ruggles 07master:c815ca36412e: wma: decode directly to the user-provided AVFrame [12:53] <cone-580> ffmpeg.git 03Justin Ruggles 07master:205a95f7b517: wmaenc: alloc/free coded_frame instead of keeping it in the WMACodecContext [12:53] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:0dff771f31cf: Merge commit '205a95f7b5178362874bc1e65eae9866723491c1' [12:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:5a7288822f24: wmavoice: decode directly to the user-provided AVFrame [12:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:8ae50d87e771: ws-snd1: decode directly to the user-provided AVFrame [12:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:646831e697a5: lavr: print out the mix matrix in ff_audio_mix_set_matrix() [12:58] <cone-580> ffmpeg.git 03Justin Ruggles 07master:1647da89dd8a: lavr: make sure that the mix function is reset even if no mixing will be done [12:58] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:91043de8256f: Merge commit '1647da89dd8ac09a55c111589f7a30d7e6b87d90' [13:05] <cone-580> ffmpeg.git 03Justin Ruggles 07master:a3735bb92a6b: lavr: cosmetics: reindent [13:05] <cone-580> ffmpeg.git 03Justin Ruggles 07master:600b4c973fb9: lavr: fix matrix reduction for upmixing in certain cases [13:05] <cone-580> ffmpeg.git 03Justin Ruggles 07master:157542ebc15d: lavr: fix mixing matrix reduction when normalization is disabled [13:05] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:845fa2f5c964: Merge remote-tracking branch 'qatar/master' [13:36] <cone-580> ffmpeg.git 03Paul B Mahol 07master:9145818ecd51: vima: decode directly to the user-provided AVFrame [13:36] <cone-580> ffmpeg.git 03Paul B Mahol 07master:2cced2a85442: paf: decode directly to the user-provided AVFrame [13:36] <cone-580> ffmpeg.git 03Paul B Mahol 07master:8f7c7ff2d605: evrcdec: decode directly to the user-provided AVFrame [13:38] <cone-580> ffmpeg.git 03Anton Khirnov 07release/0.5:0b25c3b67cef: mpeg12: do not decode extradata more than once. [13:38] <cone-580> ffmpeg.git 03Michael Niedermayer 07release/0.5:deaaacbc3f8d: Merge remote-tracking branch 'qatar/release/0.5' into release/0.5 [14:26] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:b2e57eb5a3cb: mjpegdec: pass nb_components into ljpeg_decode_yuv_scan [15:04] <cone-580> ffmpeg.git 03Nicolas George 07master:6d680d20ad8b: lavf/concatdec: add safe option. [15:04] <cone-580> ffmpeg.git 03Nicolas George 07master:db9dc5154d1a: lavf/concatdec: add signature directive. [15:04] <cone-580> ffmpeg.git 03Nicolas George 07master:b68dd8a1614b: lavf/concatdec: allow probing. [15:13] <cone-580> ffmpeg.git 03Nicolas George 07master:a17ececcc792: libcelt: decode directly to the user-provided AVFrame. [15:32] <durandal_1707> michaelni: libvorbis decoder is broken [15:34] <kierank> is ubitux back [15:43] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:fe84577273c9: libvorbisdec: set sample type [16:07] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:324d96644b57: libvorbisdec: Remove AVFrame from context [16:22] Action: ubitux is back kierank [16:24] <j-b> ubitux: we went and met your former Bosses this morning [16:25] <ubitux> "we"? [16:25] <j-b> we! [16:25] <ubitux> wepee~ [16:48] <durandal_1707> there are not deps for filters in fate? [16:51] <durandal_1707> can something be done about FATE_AVCONV nonsense in makefiles? [17:00] <michaelni> it could be renamed trivially but it would add conflicts on merges [17:07] Action: durandal_1707 waiting for s/FATE/LATE [17:09] <ubitux> Libav Automated Mess Environment [17:09] <ubitux> you can also have this variant [17:11] <av500> +1 [17:17] <durandal_1707> now i have big pix fmts table, and it is big and ugly [17:37] <durandal_1707> why all image formats and even raw cant be displayed when only part of data is present? [17:40] <cone-580> ffmpeg.git 03Paul B Mahol 07master:cf73e0bff3a2: lavu/intreadwrite: add AV_[RW][BLN]48 [17:40] <cone-580> ffmpeg.git 03Paul B Mahol 07master:34126cd090c3: lavfi/transpose: support more pix fmts [18:14] <durandal_1707> michaelni: what QP changer filter does? [18:16] <durandal_1707> lol, it changes some quantization tables.. [18:26] <michaelni> changes Quantization Parameter [18:26] <durandal_1707> -> does nothing in lavfi [18:28] <durandal_1707> why there are 3 blur filters - they could be merged into one [18:35] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:21cd905cd44a: swr: check channel layouts before using them. [18:47] <cone-580> ffmpeg.git 03Daniel Verkamp 07master:5dd78ef34e20: WAV muxer: reindent [18:49] <durandal_1707> burek: you still want to write playlist(m3u) demuxer? [20:19] <cone-580> ffmpeg.git 03Clément BSsch 07master:15e7533ca1a1: doc/dev: clarify public symbols prefixes. [21:32] <ubitux> don't worry about this one [21:32] <ubitux> most likely because of the upgrade in the middle of the tests [21:43] <kierank> ubitux: were there any more problems with those r128 patches? [21:43] <ubitux> i'll test them and apply, but i wanted a comment from the author about the back speakers masks [22:15] <tg2> when was version syntax changed from git-2012-11-08-aeb4d27 to N-49695-g4e2c636 ? [22:17] <llogan> tg2: i don't know if it was changed. [22:17] <tg2> what does the N-49695-g4e2c636 signify [22:18] <tg2> build # - commit? [22:18] <tg2> or something [22:18] <llogan> git-2012-11-08-aeb4d27 probably means you're using a shallow clone [22:18] <llogan> "git clone --depth 1 ..." [22:18] <ubitux> 49695th commit, hash=4e2c636? [22:19] <tg2> yeah trying ot figure out what tha tsecond hash is [22:19] <llogan> i'm guessing you're using ffmpeg-git from arch or followed the ubuntu compile guide [22:20] <tg2> used git clone [22:20] <tg2> then compiled [22:21] <tg2> anyway, have a weird bug i'm seeing betweeen two versions, both compiled within a day of eachother, one does it one doesnt with teh same file same command [22:21] <llogan> ah, yes, in version.sh: Shallow Git clones (--depth) do not have the N tag [22:21] <tg2> [mpeg4 @ 0x6be18a0] Invalid and inefficient vfw-avi packed B frames detected [22:21] <tg2> on one [22:21] <tg2> ah ok llogan [22:21] <tg2> that makes sense [22:23] <tg2> http://pastie.org/6157210 [22:24] <tg2> with cmd: http://pastie.org/6157216 [22:55] <BBB-work> someone broke msvc dll builds by marking ff_codec_bmp_tags[] with av_export [22:55] <BBB-work> can whoever did that fix it? [22:55] <BBB-work> https://github.com/libav/c99-to-c89/issues/5 [23:09] <beastd> BBB-work: ^^^^ i think it could have been him. need to check. [23:11] <michaelni> aa715cdc <-- [23:21] <beastd> BBB-work: A generic question? Do you have a strategy handle shared data symbols between DLLs? Or you completely avoid it? [23:25] <llogan> It's GSoC time again. [23:34] <cone-580> ffmpeg.git 03Michael Niedermayer 07master:c230af9bccc3: h264: Reset last_pocs in case of reference or frame number inconsistencies [23:48] <nevcairiel> beastd: av_export can in general share data symbols, however its a bit of a pain. You cant use them as static initializers anymore (which causes the current breakage), and stuff like that [23:48] <nevcairiel> additionally, a ff_* symbol should not be exported. :p [23:50] <nevcairiel> it may be easier in many cases to add a getter functionf or such tables [23:50] <BBB-work> that doesn't work for static users like this [23:50] <BBB-work> generally, avoid it [23:51] <BBB-work> if at all possible [23:51] <beastd> nevcairiel: well, i said generally (not this instance) [23:51] <beastd> BBB-work, nevcairiel: but thanks for your answers [23:52] <nevcairiel> its generally a painful thing to do, so don't :) [23:52] Action: beastd also tends to think that avoiding it is the probably the best strategy [23:53] <beastd> so if it was roger (dshow guy) who broke this someone should send a mail (i do not think he is here on irc) [23:59] <beastd> Need to get some sleep now. [23:59] <beastd> bye... [00:00] --- Thu Feb 14 2013

1 0

ffmpeg.log.20130213
by burek 14 Feb '13

14 Feb '13

[00:21] <fatpony> nice tip llogan thanks [00:21] <tds5016> hi. [00:21] <tds5016> Is it possible to use ffmpeg to convert from h264 stream over to an mpegts for http live streaming? [00:26] <tds5016> I've basically got an h264 file, and I'm trying to convert it over to an http live streaming format to share out. [01:47] <jankarlitos> Hello. Is anyone capable of get the streams out of this shoutcast URL?: http://johnny.serverroom.us:6336 [01:49] <klaxa> um... [01:49] <klaxa> that's pretty much the stream already [01:50] <klaxa> wget http://johnny.serverroom.us:6336/ -O stream_dump.mp3 [01:51] <jankarlitos> klaxa, yes that's the only way i could get it. But i need to transcode it with ffmpeg [01:51] <klaxa> ffmpeg -i http://johnny.serverroom.us:6336/ -c:a libvorbis stream_dump_encoded.ogg [01:51] <klaxa> or whatever [01:52] <jankarlitos> It doesn't work [01:53] <klaxa> mmhh.. it appears to hang that's weird [01:54] <klaxa> this shouldn't be a problem, for the time being you could create a pipe with wget [01:54] <klaxa> wget http://johnny.serverroom.us:6336/ | ffmpeg -i - -c:a libvorbis stream_dump.ogg [01:54] <klaxa> argh [01:55] <klaxa> wget http://johnny.serverroom.us:6336/ -O - | ffmpeg -i - -c:a libvorbis stream_dump.ogg [01:55] <klaxa> like that [01:56] <jankarlitos> Already try that. The result: "Cannot write to - (Broken pipe)" [01:56] <klaxa> what is your command line? [01:58] <jankarlitos> The same as yours: wget http://johnny.serverroom.us:6336/ -O - | ffmpeg -i - -acodec libvorbis out.ogg [01:59] <klaxa> mmhh that sounds like a restriction set by your system [02:01] <klaxa> hmm maybe you can use named pipes but that would get ugly [02:14] <jShaf> how do I skip the beginning to the very first keyframe for video INPUT? [02:14] <jShaf> like -ss <time of the first keyframe> [05:19] <happarappa> ffmpeg.exe -i test.mp4 -sameq test.avi <-- This results in an .avi of poored quality than the input .mp4. What the hell? Just when I thought I had ffmpeg figured out, it goes and does this to me. [05:20] <klaxa> sameq just means same quantizer not same quality [05:20] <klaxa> i think it should be avoided if possible [05:23] <happarappa> Same quantizer? [05:24] <happarappa> Never even heard of any such thing as "quantizer". [05:24] <Plorkyeran> then you definitely don't want to be using sameq [05:25] <fenduru> Anyone know a way to force an encoder to NOT add padding? [05:26] <fenduru> ffprobe reports a .wav as 0.97 seconds long, but after encoding to aac it is 10.0, and after encoding to mp3 it is 10.03. These are audio segments so playing them in sequence is causing gaps [05:27] <happarappa> F... F... S. :| [05:34] <relaxed> it was removed from ffmpeg because it most cases it didn't even work. [05:37] <happarappa> What was? [05:43] <relaxed> -sameq [05:45] <happarappa> relaxed: Huh? It's not removed? [05:50] <relaxed> You must be using an older version. [05:52] <relaxed> happarappa: use '-q:v 3' instead for good quality [06:05] <happarappa> Not using an old version... [06:05] <happarappa> I want the same exact quality. [06:14] <relaxed> then use -c:v copy, otherwise it's impossible [06:17] <Plorkyeran> or a lossless codec [06:50] <happarappa> :| [07:52] <asher^> hi guys. does ffmpeg have an option similar to -ss where you can seek to half way through a video rather than a specific time? [07:54] <klaxa> yes [07:54] <klaxa> asher^: see: http://ffmpeg.org/trac/ffmpeg/wiki/Seeking%20with%20FFmpeg [07:54] <asher^> thanks [07:56] <asher^> klaxa, i cant see it there. i should be more specific, id like to know if theres an option where i dont need to know the duration in advance [07:57] <klaxa> um... [07:57] <klaxa> i don't think so? [07:57] <klaxa> for what usecase would you need that? [07:57] <asher^> im just taking screen grabs of videos programatically where i dont know the length in advance. i thought id just grab them from the middle [07:58] <asher^> some are 5 mins, some an hour [07:58] <klaxa> so you want to specify like... -ss 50% ? [07:59] <asher^> yeah something like that [08:00] <klaxa> hmm... well you could certainly script it with the help of ffprobe [08:00] <asher^> yeah i guess ill have to do something like that [08:00] <asher^> thanks [08:01] <asher^> hmm, looks like something called ffmpegthumbnailer might do it [08:01] <asher^> although that looks like linux too [08:38] <spaam> klaxa: nice page. [08:39] <klaxa> i know, i didn't write it though [08:39] <klaxa> but it's rather useful, i didn't know ffmpeg could do fast-seeking with keyframes [08:39] <spaam> it was burek and some other guy D: [09:20] <happarappa> Why is ffmpeg the most confusing nonsense ever made? [09:21] <happarappa> I just want to convert from one file format to another with the same amount of quality/untouched. [09:21] <klaxa> happarappa: because it's the leading FOSS in video encoding [09:21] <klaxa> that's not possible unless you use lossless codecs [09:21] <klaxa> or copy the codec [09:21] <klaxa> this is a general truth and has nothing to do with ffmpeg [09:22] <klaxa> oh ffmpeg also does video filtering [09:22] <klaxa> same stuff with audio :V [09:23] <klaxa> i mean come on, sure it's not easy to use, but it's still one of the greatest open source projects [09:25] <happarappa> :( [09:25] <happarappa> Yes, it is amazing... if it were just nicer to use. [09:25] <happarappa> Supports an insane amount of formats. [09:26] <klaxa> actually i find it rather easy to use [09:26] <klaxa> everything is well documented [09:26] <happarappa> I get a crappy video result from my conversion. [09:26] <happarappa> I want it to look just like the original. [09:26] <klaxa> <klaxa> that's not possible unless you use lossless codecs [09:26] <klaxa> <klaxa> or copy the codec [09:26] <happarappa> ... What? [09:27] <happarappa> Of course it's possible. [09:27] <happarappa> It's damaging the video. [09:27] <klaxa> you cannot achieve the exact s-- [09:27] <JEEB> a result close to the original is possible [09:27] <JEEB> not the exact same [09:27] <happarappa> It's MUCH worse. [09:27] <happarappa> ffmpeg.exe -i test.mp4 -sameq test.avi [09:27] <JEEB> pastebin your command line and terminal output [09:27] <JEEB> ugh [09:27] <JEEB> sameq is not what you think [09:27] <JEEB> and was actually broken in most cases [09:27] <happarappa> Whatever it means, the result is not what I want... [09:28] <JEEB> it is not same quality but "same quantizer", which doesn't even make sense between two different formats [09:28] <JEEB> also I think it was removen later [09:28] <JEEB> as an option [09:28] <JEEB> because people kept using it even if it was broken [09:28] <JEEB> so what do you want your output to be? [09:29] <JEEB> for what are you encoding? [09:29] <happarappa> Visually impossible to tell apart. [09:29] <JEEB> no, i mean [09:29] <happarappa> For what am I encoding? [09:29] <JEEB> avi is a container [09:29] <JEEB> what do you want into it? [09:29] <JEEB> just like mp4, avi is a "box" [09:29] <JEEB> you can put stuff into the "box" [09:29] <happarappa> I don't care. When files are jungled through the black box that is "ffmpeg", they work in various software such as video editors. [09:30] <JEEB> :| [09:30] <happarappa> Prior to being put into the black box, they are not accepted. [09:30] <happarappa> (Renaming the files does not work.) [09:30] <JEEB> well, naturally [09:30] <JEEB> if the editor doesn't support the container format [09:30] <JEEB> or the video inside it, then naturally [09:30] <JEEB> for video editors I recommend utvideo [09:30] <JEEB> because you can get decoder components for every OS for it [09:31] <JEEB> Windows, OS X, Linux [09:31] <JEEB> ffmpeg -i input -c:v utvideo -c:a pcm_s16le out.avi [09:32] <klaxa> heh promoting utvideo again :) [09:32] <aji> i wonder what percentage of this channel's messages are ffmpeg command lines [09:32] <klaxa> (btw, is it multi-threaded yet?) [09:32] <JEEB> the decoder is, the encoder is harder to MT [09:32] <JEEB> at least with slices [09:32] <JEEB> frame-based MT would be easier [09:33] <JEEB> happarappa, also it kind of would help me if you would tell which OS and editor you're going to use [09:33] <JEEB> and I asked for the terminal output because I kind of could have seen the version of ffmpeg you have :V [09:34] <JEEB> pretty much everything I ask for is for a reason [09:35] <JEEB> klaxa, also why would I not promote it? It has components for every OS after all [09:35] <JEEB> I really don't see a reason not to use it if you're doing video editing :) [09:35] <klaxa> i wasn't complaining, just noting :) [09:36] <jeje34> Hi to all ;-) [09:36] <JEEB> Daemon404 actually multithreaded the prediction per-slice but that ended up being bottlenecked by the huffman afterwards [09:36] <JEEB> so it seems like the easiest way to make the libavcodec utvideo encoder faster is to just frame thread it :< [09:37] <klaxa> hmm okay i don't understand this well enough yet to comment on it :P [09:41] <jeje34> I already have integrated FFMPEGs Library to decode H264 video streaming from IP Cameras and display them in windows of a Windows OS. But now, I'm trying to decrease the CPU usage when I have several decompressions at the same time. So I compile FFMPEG (version release 1.1.1) with --enable-dxva2 --enable-decoder=h264_dxva2 --enable-hwaccel=h264_dxva2 [09:43] <jeje34> After do this, I try to modify my code to use AVHWAccel. but it's really hard to find documentation or sample code source to use it...hte only thing I found is I have to use av_hwaccel_next to find the good one (hwaccel->id == CODEC_ID_H264) and register it by av_register_hwaccel but It seems it isn't enougth to use the hardware decompression [09:43] <JEEB> yes [09:43] <jeje34> if someone can help me, thanks [09:44] <JEEB> hwaccel has you do work too, because libavcodec can have no idea on how you want to use DXVA [09:44] <JEEB> https://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-May/006600.html [09:44] <JEEB> take a look at this post [09:45] Action: JEEB goes back to doing laundry [09:48] <jeje34> thanks fot this reply! for the point 3- Create a dxva_context structure and initialize it's fields with the above objects, is it a context like an AVCodecContext? And how initialize the fields? [09:50] <jeje34> if someone has already use AVHWAccel in his code, it will be very helpfull [10:09] <JEEB> jeje34, there is an app mentioned in the post, so if you want you can look at that as an example [10:13] <happarappa> JEEB: I don't see what my OS or my video editor has to do with this. [10:15] <JEEB> happarappa, because if you are using a packaged ffmpeg it can be old, if you are using linux video editors they can be built with old libavcodec [10:15] <JEEB> those naturally limit the alternatives [10:15] <jeje34> JEEB> you telling about "a useful tool for diagnosing DXVA2 is DXVAChecker" [10:15] <JEEB> jeje34, no [10:15] <JEEB> far before that [10:15] <jeje34> ok the msdn link [10:16] <JEEB> well, that is related yes, but you know -- the reference to VLC [10:16] <JEEB> because DXVA is a MS thing, you will find all the docs related to it from MSDN :P [10:17] <jeje34> but do you think it really decrease my CPU time when I decode several streams at the same time (because it need me to make a lot of changes in my implementation...) [10:17] <happarappa> JEEB: None of that made any sense to me. Sorry. [10:17] <happarappa> "a packaged ffmpeg"? [10:17] <JEEB> package management [10:18] <JEEB> on linux most distributions have a package management system [10:18] <JEEB> where most people get their software [10:18] <JEEB> on OS X you have macports and homebrew for similar usage [10:18] <happarappa> Windows here. [10:18] <JEEB> ok [10:18] <JEEB> then the case B) of you having an editor that can't use the VFW component is unrelated [10:19] <JEEB> then Ut Video should work just fine and be lossless as long as you install the windows codec for it (DirectShow/VFW) [10:19] <JEEB> http://umezawa.dyndns.info/wordpress/?p=3655 [10:19] <JEEB> of course your windows build of ffmpeg has to be new enough to support ut video [10:19] <JEEB> but yes, that line I gave you up there should work then [10:20] <JEEB> <JEEB> ffmpeg -i input -c:v utvideo -c:a pcm_s16le out.avi [10:20] <JEEB> it will be big, but it will also be lossless as well as intra so it will be quick to seek through [10:20] <JEEB> when editing [10:21] <JEEB> (lossless does not mean uncompressed, it is compressed) [10:21] <JEEB> also if you have too old of an ffmpeg, then you can grab a newer one from zeranoe http://ffmpeg.zeranoe.com/builds/ [10:29] <jeje34> I also have another question about my FFMpeg use. I compile it with --enable-w32threads (because I'm a window user). But when I try to set a thread_count in my AVCodecContext (like 3), I always have a log error The maximum value for lowres supported by the decoder is 0. when calling avcodec_open2 [10:29] <jeje34> So I can't use FFMPEG Multithreading to decode H264? [10:30] <JEEB> no idea if you're just using libavcodec wrong [10:30] <JEEB> because there has been H.264 multithreaded frame decoding since summer 2011 [10:31] <JEEB> sliced decoding is a more derpy thing, and only works with slices [10:31] <JEEB> sliced threading [10:34] <jeje34> ok JEEEB, I set my ffmpeg code part here: http://pastebin.com/sNpcpHjf [10:36] <jeje34> but I want to decompress H264 frames coming from a camera. So in my code I recompose one entire frame before passing it to FFMPEG so I think it will be more FF_THREAD_FRAME I need to use [10:36] <JEEB> I have no idea, I have only coded within libavcodec, not used it myself :P Unfortunately you'll have to have someone else look at that code, or look at other examples using libavcodec's threading (various things use it) [10:36] <jeje34> ok thanks [10:43] <jeje34> another one question: do you think in my case, using a CodecCtx->thread_count could decrease my CPU usage to decode my H264 stream? [10:44] <jeje34> when I have several streams to decode at the same time [10:46] <JEEB> depending on how many things you have going at the same time, you can make them derp less with each other by limiting the overall amount of threads, the notion of "could decrease my CPU usage" is kind of derp -- if you are bound by, say, a single thread it will use less CPU but be slower, while more threads mean more power can be used for one stream. In case of you doing multiple parallel jobs of course, you might want to limit the [10:46] <JEEB> amount of threads depending on the amount of cores in the system and the amount of streams you're doing, but that really won't "decrease your CPU usage", it will just make those separate processes not poke each other as much [11:00] <jeje34> ok JEEB, thanks for all your answers. If I don't set a thread_count but have compiling FFMPEG with --enable-w32threads, what happen with FFMPEG? because when calling avcodec_alloc_context3, it initialze the AVCodecContext with avcodec_get_context_defaults3, so setting the threadcount to 0 [11:02] <JEEB> that should be the default now > --enable-w32threads [11:02] <JEEB> on windows [11:02] <JEEB> and it just enables threading [11:02] <JEEB> I think it depends on the default in the decoder? No idea, as I've said I've never used the API [11:02] <JEEB> so I don't know the defaults and how it is going to function [11:07] <jeje34> ok, I'm going to seek in the FFMPEG source code to understand a little bit more what happen [11:14] <fatpony> i can't seem to crop one pixel, is that normal? can i only crop to even values? [11:15] <JEEB> if your source has chroma subsampling (4:2:2 or 4:2:0 YCbCr [colloquially called "YUV"]), then yes [11:16] <JEEB> because with 4:2:0 you have one value for a 2x2 area, and 4:2:2 has one value for a 2x1 area [11:17] <fatpony> ah so i guess that's what yuv420p stands for [11:17] <JEEB> yup [11:18] <fatpony> it's strange because i can clearly see that a black border beginning at an odd height [11:18] <fatpony> s/that// [11:19] <JEEB> luma (grayscale image) is full resolution, but the chroma planes (color information) are subsampled [11:19] <JEEB> so yes, I guess having the black area start at an odd height is possible I guess [11:21] <fatpony> ah i see, thanks! [13:49] <jShaf> how do i trim the beginning up to first keyframe on the video input? [13:59] <xroberx> hi [13:59] <xroberx> If I only want to do pixel format conversion, which is the right functions ? sws_scale() ? [14:00] <durandal_1707> yes [14:00] <xroberx> I do not want to perform scaling though [14:01] <xroberx> durandal_1707: so... I guess I'll have to set the source width/height equal to the destination width/height, right ? [14:01] <durandal_1707> same function can do most of unscaled conversions too [14:01] <durandal_1707> depends on input->output formats [14:01] <durandal_1707> and obviously same dimensions [14:02] <weecka> Hello guys, maybe you can help me. I am receiving lots of these warnings: max resync size reached, could not find sync byte.78 bitrate=N/A. Whole output is here http://pastebin.com/Ke9wse1a. What does this means? [14:03] <durandal_1707> weecka: how you create file? what can play it? and soo on .. [14:05] <weecka> Honestly to say, I'm trying to convert flv to mpegts, which then could be played using apple streaming protocol. [14:05] <xroberx> durandal_1707: do you know if there is any ARM optimized version ? By looking in the libswscale folder I can see there is only optimizations for x86, ppc, sparc and bfin. [14:06] <weecka> but I'm not sure, what I'm doing wrong. [14:06] <durandal_1707> xroberx: as you noticed, there is noone [14:07] <xroberx> ok [14:08] <durandal_1707> weecka: the mpegts muxer may be just buggy, so what is your command that do flv->mpegts? [14:09] <weecka> I've it did like this: ffmpeg -y -i mindmovie-5173.flv -f mpegts -acodec libmp3lame -ar 48000 -ab 64k -s 320x480 -vcodec libx264 -b 512k -maxrate 512k -bufsize 512k -qcomp 0.6 -qmin 30 -qmax 51 -qdiff 4 -level 30 -aspect 320:480 -g 30 -async 2 output-iphone-big.ts [14:14] <weecka> Also, I've tried to do flv->m4v and then to segment it, but the segment durations somehow are then way of target. [14:41] <xroberx> hi [14:43] <weecka> durandal_1707: any thoughts? [14:43] <xroberx> durandal_1707: I've found this ARM optimized yuv2rgb conversion library : http://wss.co.uk/pinknoise/yuv2rgb/ The license is BSD. Now the question is: would it be possible to integrate it in ffmpeg's libswscale ? [14:45] <durandal_1707> xroberx: what kind of integration? [14:46] <durandal_1707> writing patches that add arm optimization to swscale is certainly possible and likely to be accepted and applied [14:47] <xroberx> durandal_1707: ok, good to know, I'll contact the author then, but considering it's BSD licensed I guess it can be done [14:47] <durandal_1707> if you can't code you find someonce who can [14:47] <creep> h [14:47] <durandal_1707> xroberx: why would you contact author? [14:48] <xroberx> durandal_1707: because of this sentence in his webpage: "If you do use use this code as part of a piece of software (or hardware), please let me know, purely for my own interest." [14:48] <durandal_1707> it's BSD license which means you can do almost anything with it (except claiming you write that code ...) [14:48] <durandal_1707> xroberx: good, but first code must be used ... [14:48] <durandal_1707> currently is not [14:49] <xroberx> yes [14:52] <xroberx> durandal_1707: do you know happen to know who is the person to contact if I wanted to send some patches to integrate the library ? [14:53] <JEEB> ffmpeg-devel mailing list for patches [14:53] <jarno> Hi Jeeb [14:53] <jarno> ...and everyone else... [14:54] <xroberx> JEEB: ok, thanks [14:55] <jarno> ...I have a question related to encoding and mux(ing)...if I have understood it correctly, the steps should be 1) find encoder 2) allocate context and 3) create output stream, right? [14:56] <jarno> ...so the avcodec_alloc_context3() sets default values as per the chosen encoder, right? [14:57] <jarno> ...and then these can be changed as needed? [15:34] <Macey> hi all whats the output format for just streaming a TS file to UDP? [15:36] <Mavrik> mpegts? [15:36] <Macey> if i do -f mpegts it just recreates the container [15:38] <Mavrik> mhm [15:38] <Mavrik> what do you want to do exactly? [15:41] <Macey> i have a fully formed TS file (NIT/SDT/PMTs e.t.c) that i want to push over UDP [15:41] <Macey> no transcoding nothing, just a straight tx [15:41] <Mavrik> ffmpeg isn't the right tool for the job then [15:41] <Macey> ah ok [15:41] <Macey> vlc? [15:41] <Mavrik> why don't you use something made to just dump data to network like netcat? [15:42] <Macey> needs to read the PCR and tx it at a rate [15:42] <Mavrik> both vlc and ffmpeg are meant to actually change streams, not be dumb network pipes [15:42] <Mavrik> VLC might know that, but I'm not really sure if it can do what you want withou modifying the stream [15:42] <Macey> ok thanks [15:50] <jeje34> Hello, just to know ... are functions av_lockmgr_addref and av_lockmgr_register are deprecated? [15:55] <jeje34> I can find av_lockmgr_register but not av_lockmgr_addref in avcodec.h (using last FFMPEG release 1.1.1) [16:36] <catalinb> Hi, I'm trying to build ffmpeg on windows using ./chromium/scripts/build_ffmpeg.sh [16:36] <catalinb> I'm calling the script from a mingw shell [16:37] <catalinb> I encounter the error: 'c99wrap is unable to create an executable file.' [16:37] <catalinb> I have a mingw installation and gcc is in path [16:38] <catalinb> How do I specify the right compiler? [17:48] <jeje34> hi to all [17:49] <jeje34> I have some question about using FFMPEG to decode h264 videos streaming from IPCamera. My code works (I can decode) but I have some questions about the initialization of the AVCodecContext fields [17:50] <jeje34> there are lot of them set - decoding: Set by user [17:51] <jeje34> I can really understand what are the best values for flags and flags2 of the AVCodecContext structure [17:54] <Mavrik> jeje34: usually most fields don't have to be set [17:54] <Mavrik> unless you have a very-nonstandard use case [17:55] <Mavrik> jeje34: those "set by user" don't always mean MUST be set by user ;) [17:55] <jeje34> I have a standard use case (I hope all manufacturer of IP camera have a good implementation of H264 video stream) [17:55] <jeje34> ;-) [17:56] <Mavrik> ^^ [17:56] <jeje34> but when I see some flags like CODEC_FLAG2_FAST [17:57] <jeje34> I'm not very sure if it's better to set [17:58] <Mavrik> jeje34: usually, leaving default settings is prefered [17:58] <Mavrik> seems like CODEC_FLAG2_FAST gains some speed at cost of decoded image quality [18:56] <Guest17651> hi, I'm trying to export VOB videos to ogg (theora/vorbis) for the web, using kdenlive. The videos are about 45 minutes each. Do you have good settings to recommend? [19:55] <someone-noone> if dts==pts does it mean that frame is a key-frame? [19:56] <someone-noone> or there is some cases when it's not true [19:56] <someone-noone> ? [19:58] <JEEB> no, it just means that it's not a b-frame [19:59] <someone-noone> Okay, thanks. That was a silly question. But I have another one, probalby less silly :) [20:00] <someone-noone> If I want to reorder packets in pts way and I want to flush packets when they are ordered. If pts==dts is a good event for that? [20:00] <someone-noone> If I want to reorder packets in pts order* [20:01] <someone-noone> Or better to look-up for key-frames for this event? [20:06] <Paranoialmaniac> JEEB: {k? it is true if pts always equals dts. but, if not, a frame with pts==dts is an un-referenced b-frame [22:15] <mordonez1> Hu guys, how can I put an image in a video in a specified time and position? [22:25] <llogan> the "specified time" is the tricky part. i don't think overlay filter has a temporal option [22:26] <llogan> of course you could section the video into parts, add the image overlay (you want an overlay, right), and then concat the sections together. [22:26] <mordonez1> yeah, I want to put an image inside a square on the video [22:26] <mordonez1> the square only appear on second 10 to 12 [22:27] <llogan> you can do that with the drawbox and overlay filters [22:28] <llogan> but i don't think either have "make filter work from 10 to 12 seconds" type of option [22:28] <mordonez1> you have any example or something that I can use to start with it? [22:38] <llogan> mordonez1: ffmpeg -i video.mp4 -i image.png -filter_complex drawbox,overlay output [22:38] <llogan> http://ffmpeg.org/ffmpeg-filters.html#drawbox [22:38] <llogan> http://ffmpeg.org/ffmpeg-filters.html#overlay-1 [23:40] <mordonez1> Thaks llogan [23:56] <someone-noone> Hey! If I want to reorder packets by pts (decoder is not ffmpeg), how can I know when to "flush" those packets? When new key-frame arrives or when dts==pts? Or may be some other algorithm? [00:00] --- Thu Feb 14 2013

1 0

ffmpeg-devel.log.20130212
by burek 13 Feb '13

13 Feb '13

[00:30] <saste_> wow av_url_split() sucks [00:35] <ubitux> unlimited free wifi at airport \o/ [00:35] <ubitux> i don't want to leave :( [00:42] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:3b1f037df56b: ffmpeg: fix streamcopy of lsf mp3. [00:43] <michaelni> BBB-work, stuttering should be fixed [00:43] <saste> ubitux, too bad for you [00:43] <saste> better for us ;-) [00:44] <ubitux> maybe i forgot how to write code [00:44] <saste> ubitux, now you write code in kanji [00:44] <ubitux> :) [00:45] <wm4> ubitux: btw., what's the status of the subtitle stuff? [00:45] <ubitux> untouched for one month? [00:46] <wm4> maybe I'll feel the need to kill of mplayer's subreader.c on a whim or so in the near future [00:46] <ubitux> wm4: i'll be back to it as soon as i'm home [00:46] <wm4> there was something about codepages still [00:47] <ubitux> but i need to catch up with one mo nth of semi hibernation [00:47] <ubitux> wm4: yeah there is a pending patch for this [00:47] <ubitux> codepage, char encoding etc [00:47] <wm4> I found libass doesn't even read utf-16 natively :( [00:47] <ubitux> it seems to be in need of some more thinking for the utf-16 shit [00:47] <ubitux> wm4: yeah well of course [00:48] <ubitux> SSA/ASS is utf-8 only afaik [00:48] <wm4> well I've seen utf-16 files in the wild [00:48] <wm4> http://code.google.com/p/libass/issues/detail?id=89 [00:48] <ubitux> lol [00:48] <wm4> chinese stuff [00:48] <wm4> fortunately not multibyte something :) [00:57] <saste_> i wonder if there is some way to avoid all the MP3 application-level hacks [01:02] <saste> michaelni: i summon you on "v4l2: bug #1570 and possible solution" [01:03] <saste> after some thinking i can't find a better solution of the one proposed [01:03] <saste> even if it is somehow hackish [01:12] <J_Darnley> What the... Who thought 99162f8d46db7ec02facae035c4ff573d2d8d612 was a good idea? [01:22] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:365270aec5c2: sanm: add forgotten check for decoded_size in old_codec37() [01:24] <saste_> J_Darnley, I don't like it very much, but what's the problem with it? [01:28] <J_Darnley> No real problem. I just think it is ugly. [01:28] <J_Darnley> Plus it missed one line. [01:29] <J_Darnley> All that to silence one warning? A warning that I thought an existing void pointer cast was there to hide. [01:42] <BBB-work> michaelni, are timestamps assigned to all non-b-frames now? can you check that? [01:44] <BBB-work> michaelni, we had issues in chrome where non-b-frames had AV_NOPTS_VALUE assigned to them for that video [02:04] <BBB-work> michaelni, and new patch sent that should apply cleanly on head for that other patch [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:f0d9ccac8937: sanm: reset frame sizes on buffer destruction [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:4ee88db840b3: sanm: init sizes with the full frame size [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:5260edee7e5b: sanm: Use the correct height variable in the decoded_size checks [02:23] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:7ff1a4b10f2d: Add add_pixels4/8() to h264dsp, and remove add_pixels4 from dsputil. [02:34] <michaelni> BBB-work, you need "-fflags genpts" or equivalent for pts from avi with b frames [02:34] <michaelni> this will add latency/delay though [02:47] <BBB-work> michaelni, in decoder? [02:49] <michaelni> latency in demuxer to delay packets till the next non B frame whichs dts is the previous frames pts [02:57] <creep> »! [03:41] <Plorkyeran> @ubitux> SSA/ASS is utf-8 only afaik <-- according to the spec it's ANSI, except for when you use \fe to override the encoding for a specific line [03:41] <Plorkyeran> everyone ignores that these days because it's retarded [04:32] <mfg> Hi, I have an mp4 file that wont play (its by Neulion) and I think they screw up the header on purpose to protected their content. Is there someone that can have a look at it? [04:39] <llogan> mfg: wrong channel. try #ffmpeg. [12:26] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=faac85005cadb5065d218b… [12:26] <ubitux> http://git.libav.org/?p=libav.git;a=commitdiff;h=fd1abf42693308ff99b9473c23… [12:26] <ubitux> rofl [12:27] <durandal_1707> you mean E1 thing? [12:27] <ubitux> yeah [12:27] <ubitux> E1 is obviously better [12:28] <av500> it saves storage space :) [12:28] Action: durandal_1707 blames nevcairiel [12:29] <nevcairiel> clearly i should've suggested E1 instead of EXPAND [12:48] <cone-102> ffmpeg.git 03Martin Storsjö 07master:ab8f1a698990: arm: Fall back to runtime cpu feature detection via /proc/cpuinfo [12:48] <cone-102> ffmpeg.git 03Derek Buitenhuis 07master:375ef6528c9d: libfdk-aacenc: Actually check for upper bounds of cutoff [12:48] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:3ccda2b02ba6: Merge commit '375ef6528c9dd2db7f9881e232cb0ec3aa16970d' [13:07] <iive> nevcairiel: actually E7 is better, E is short of Exp (like in the float format), 7 is the same key as & that is user for AND, so e7 reads as EXPAND. [13:08] <av500> iive: on your keyboard [13:09] <iive> av500: on most keyboards :P [13:10] <av500> iive: so maybe git post-checkout hook that modifies it for the locale :) [13:11] <cone-102> ffmpeg.git 03Diego Biurrun 07master:6fb296e97edc: avcodec/rectangle: Remove nonsense assert [13:11] <cone-102> ffmpeg.git 03Diego Biurrun 07master:62a43d30cc3b: crc: Move static data table from header to .c file [13:11] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:e2b703f53612: Merge commit '62a43d30cc3b9b2460f2c926beb522df32493625' [13:12] <durandal_1707> wasn't there patch on ml that fix VC1 interlaced? [13:15] <burek> just to mention this, we could integrate something like this into our wiki: https://github.com/mpenkov/ffmpeg-tutorial [13:15] <burek> A set of tutorials that demonstrates how to write a video player based on FFmpeg [13:16] <burek> :D [13:17] <burek> did you manage to get the list of cmds? :) [13:21] <durandal_1707> burek: i have private correspodence with ffloger [13:21] <durandal_1707> now if only it could write code .... [13:22] <burek> :) [13:25] <cone-102> ffmpeg.git 03Martin Storsjö 07master:d192ac3e03d1: swscale: Disallow conversion to GBRP16 [13:25] <cone-102> ffmpeg.git 03Martin Storsjö 07master:4c51fe48ba6d: h264: Copy h264chroma dsp context to slice thread copies [13:25] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:fe54f6cb453c: Merge commit '4c51fe48ba6dde059360b7451db7f6bbf2f11db3' [13:34] <cone-102> ffmpeg.git 03Daniel Kang 07master:b3f2a3fe3fe4: x86: mpeg4qpel: Make movsxifnidn do the right thing [13:34] <cone-102> ffmpeg.git 03Diego Biurrun 07master:304b806cb524: build: Make library minor version visible in the Makefile [13:34] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:7491356111e7: Merge commit '304b806cb524fb040f8e09a241040f1af2cb820b' [14:05] <cone-102> ffmpeg.git 03Diego Biurrun 07master:69dca3a4f45e: openbsd: Add minor number to shared library install name [14:05] <cone-102> ffmpeg.git 03Diego Biurrun 07master:56dea32f09e7: doc: developer: Fix wording in "naming conventions" section [14:05] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:08db519e1275: Merge commit '56dea32f09e71460add14a1b6d0fbe6eb16f02a4' [14:21] <cone-102> ffmpeg.git 03Diego Biurrun 07master:3658d2dfd48a: doc: developer: Add a note about reserved system name space [14:21] <cone-102> ffmpeg.git 03Luca Barbato 07master:fd1abf426933: lavu: avoid clashing definition of E [14:21] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:a63dc846b502: Merge remote-tracking branch 'qatar/master' [14:42] <durandal_1707> why transpose filter does not work for h != v chroma subsampling? [15:17] <divVerent> 14:42:28 @durandal_1707 | why transpose filter does not work for h != v chroma subsampling? [15:17] <divVerent> because then transposing would be lossy [15:17] <divVerent> and that's simply not implemented (one can always convert to 4:4:4 first though) [15:34] <durandal_1707> ok -> no reply -> applying patches at will [15:39] <durandal_1707> divVerent: ahh, because of spam i did not see you reply, sorry [15:43] <cone-102> ffmpeg.git 03James Almer 07master:0957041221fa: lavc/tta: Add support for encrypted streams [15:43] <durandal_1707> yes, that makes sense [16:02] <cone-102> ffmpeg.git 03Daniel Verkamp 07master:1ecce266fa5b: WAV muxer: add RF64 support [16:21] <ubitux> michaelni: "visible" [16:22] <ubitux> (commit message) [16:22] <av500> better than invisible commit message [16:23] <durandal_1707> ubitux: what/where? [16:24] <ubitux> "[FFmpeg-devel] [PATCH] developer.texi: replace "externally vissible" by public" [16:25] <ubitux> just a typo [16:25] <ubitux> 9h in the transit is a nightmare [16:25] <ubitux> in about ~15 hours i'm back for new adventures [16:26] <av500> going round the world? [16:26] <durandal_1707> you need 24h of sleep to sync [16:26] <ubitux> no, i'll be in sync directly [16:28] <ubitux> i just need to stay away for about 8 extra hours, which is easy with all the movie stuff available [16:29] <ubitux> i wonder what they use for the movies btw [16:29] Action: durandal_1707 removed base64 from its system [16:29] <ubitux> it's interlaced as hell [16:29] <ubitux> also, they display multiple subtitles at once [16:29] <ubitux> which is kinda interesting [16:29] <ubitux> the system is a linux but that's common knowledge [16:29] <ubitux> i wonder if someone has dumped a copy or two of these [16:30] <ubitux> av500: no, just back from a month in a country at the other side of the planet [16:31] <ubitux> and i'm waiting at the transit right now :P [16:31] <av500> australia? [16:31] <ubitux> transit is HK [16:35] <ubitux> (fr->china->jpn and th other way around) [16:36] <durandal_1707> hmm, why china? [16:43] <durandal_1707> nut doesnt support qtrle by default [17:00] <aballier> maybe i've missed something, but doesnt the AVFrame.channel change from int64 to int implies sizeof(AVFrame) decreases ? [17:00] <aballier> and in this case, accessing AVFrame.pkt_size will break [17:03] <aballier> emailed, better ask there directly [17:04] <nevcairiel> these fields are not part of the public API, you are supposed to use the getter functions [17:05] <aballier> ah ok, makes sense, i was missing that part [18:30] <durandal_1707> saste: i may be stupid but what is way to pad silence at start of stream? [18:30] <durandal_1707> sox have delay efffect which delays each channel in audio to some value [18:49] <saste> durandal_1707, aresample=min_comp=0.001:min_hard_comp=0.1 is one way [18:49] <saste> then there is a swr option which can be used to achieve the same effect [18:49] <saste> and yes adding an example may be useful [18:49] <saste> I don't know if apad can be used as well [18:54] <durandal_1707> what that numbers means and what ammount of silence it adds? [18:56] <saste> durandal_1707, wait, you want to *ADD* silence at the beginning of the stream? [18:58] <durandal_1707> yes [18:58] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/43315/focus=43318 [18:59] <durandal_1707> that resets pts to some 0 value [18:59] <saste> durandal_1707, plus, you can add an initial offset with asetpts=PTS+TIME_OFFSET/TB [19:00] <saste> I didn't try the first_pts option yet [19:00] <durandal_1707> well i would certainly need working command [19:02] <saste> durandal_1707, e.g. if you want to add 100 seconds of silence, you do: asetpts=PTS+100/TB,aresample=min_comp=0.001:min_hard_comp=0.1 [19:02] <saste> I suspect first_pts may be a more robust solution [19:03] <saste> apad could be extended to achieve the same effect [19:03] <saste> also now the audio filters order is messed up [19:04] <saste> what about factorizing documentation for all biquad filters? [19:05] <saste> you could have section and a subsection for all variants [19:05] <durandal_1707> saste: i'm more interested in writing new filters [19:05] <saste> you could have *a main section* and a subsection for all variants [19:05] <saste> i can do the work, just wanted to know your opinion [19:05] <durandal_1707> also I really do not know doxygen [19:05] <saste> texinfo i guess [19:05] <saste> not very much to know [19:09] <durandal_1707> saste: feel free to do whatever you feel is needed, my only opinion about documentation is that it should be present & correct and not cryptic [19:11] <saste> durandal_1707, about mp and optims, yes they are disabled [19:11] <saste> don't ask me why [19:12] <durandal_1707> saste: i'm trying to write VV->V filter where both inputs are readonly and i create new output buffer, i get buffer queue overflow errors [19:12] <durandal_1707> and nothing in output [19:12] <durandal_1707> ffmpeg -f lavfi -i testsrc -f lavfi -i mptestsrc -filter_complex blend /tmp/out.png [19:46] <Compn> http://torrentfreak.com/open-source-giant-vlc-mulls-bittorrent-support-1302… [19:51] <durandal_1707> seed!!! [19:57] <Compn> j-b talking to torrentfreak :D [19:58] <durandal_1707> saste: or should i have another queue for filtered buffers? [20:01] <durandal_1707> i really dunno what should I do and is that even possible [20:07] <michaelni> it should work to dereference the input and create new output [20:08] <durandal_1707> yes, but i'm clearly missing some code [20:11] <durandal_1707> https://github.com/richardpl/ffmpeg/tree/blend [20:28] <michaelni> I think you are missing a request_frame() implementation but i dont know if this is related to the problem [20:36] <durandal_1707> michaelni: and i would request frame for each input? [20:37] <durandal_1707> s/for/from [20:38] <durandal_1707> i just need template and i did 99% of filter already :D [20:47] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:1ac0fa50eff3: pngdec/filter: dont access out of array elements at the end [20:57] <durandal_1707> yay [23:52] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:6f9ae391deea: faxcompr: Dont read ref when the end has been reached in pass mode [23:52] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:e1219cdaf9fb: tiff: Check buffer allocation and pointer increment more carefully in shorts2str() and double2str() [00:00] --- Wed Feb 13 2013

1 0

ffmpeg.log.20130212
by burek 13 Feb '13

13 Feb '13

[00:50] <sam___> any suggestions .ffmpeg [libx264 @ 0x670520e0] Unable to parse option value "flags2 +bpyramid+wpred+mixed_refs-dct8x8" [00:51] <sam___> o.ffmpeg [Eval @ 0x7f8f4fffd4c0] Undefined constant or missing '(' in 'flags2' o.ffmpeg [libx264 @ 0x522f8140] Unable to parse option value "+bpyramid+wpred+mixed_refs-dct8x8" [00:53] <sam___> whats the problem with ffmpeg am i missing something? [00:53] <sam___> Undefined constant or missing '(' in 'flags2'. [00:54] <saste> sam___, use libx264 presets [02:44] <mishehu> greetings folks. I was wondering if openjpeg 2.0.x is supported in ffmpeg currently, or only the older 1.5.x branch. [03:07] <kode54> fuck, more distributions switching to avconv? [03:10] <llogan> kode54: which one(s) are you referring to? [03:11] <kode54> friend on another network caught my whining [03:12] <kode54> I spotted a capture and broadcast to twitch.tv tutorial [03:12] <kode54> which I first complained about because it was a freaking youtube video instead of a document [03:12] <kode54> then I complained because he was displaying a bash script in the video for viewers to transcribe, without actually linking to it in plain text form [03:12] <kode54> then I noticed he was using avconv instead of ffmpeg [03:13] <kode54> friend said he doesn't want to get into the whole ego war between the projects [03:13] <sacarasc> Which distro is your friend using? [03:13] <kode54> debian [03:13] <kode54> but he says gentoo is "switching" to avconv as the primary of the two as well [03:14] <kode54> his complaint is that ffmpeg gets more important features faster than avconv does [03:14] <kode54> but he's stuck with people making decisions that favor using the other [03:16] <demonjester> hey everyone, I was wanting to know if someone can help me with audio recording on my openwrt router [03:19] <llogan> kode54: at least the ffmpeg package in gentoo is actually FFmpeg upstream unlike debuntu [03:19] <kode54> that's cool, at least [03:20] <llogan> debian has dropped the "ffmpeg" package from experimental apparently [03:21] <llogan> so a real ffmpeg package may be one step closer to a reality...assuming the DDs don't simply disregard it [03:28] <demonjester> can anyone help me? i am using these comands and it wont save the file after I ctrl + c "ffmpeg -i /dev/audio test.mp3" [03:35] <llogan> relaxed: interested in changing the CentOS guide to static build instructions? [03:36] <defaultro> good evening folks. Is it possible to fix the color of my video? Looks like there is a a little cyan cast. The blue sky is like not blue [03:38] <llogan> defaultro: in all players? [03:38] <defaultro> yes [03:38] <defaultro> i remember reading an issue with this video camcorder [03:39] <defaultro> but Panasonic never made a fix for it :( [03:40] <llogan> there are probably several filters that can do this [03:40] <llogan> http://ffmpeg.org/ffmpeg-filters.html#hue [03:40] <defaultro> cool. Checking it out now [03:40] <demonjester> here is one with using oss http://pastebin.com/pGg9CnYA [03:41] <demonjester> default its not for you [03:41] <demonjester> sorry [03:41] <demonjester> here is the one without using oss http://pastebin.com/LfXsZ7rp [03:41] <defaultro> k [03:41] <llogan> demonjester: you have no mp3 encoder [03:42] <llogan> missing --enable-libmp3lame [03:42] <demonjester> so in my command i add --enable-libmp3lame? [03:42] <llogan> no. in your ffmpeg ./configure [03:42] <llogan> did you compile it? [03:42] <demonjester> kk [03:43] <demonjester> no i used opkg install in openwrt it did the configuring [03:43] <llogan> if you have the lame binary you could possibly pipe to lame [03:44] <llogan> ffmpeg -i input -f wav - | lame - output.mp3 [03:44] <demonjester> i downloaded a package where i thought it was lame but said lame is not found and idk what the package is in openwrt for lame [03:46] <llogan> you'll need to install lame headers first before you compile ffmpeg with --enable-lame [03:46] <llogan> *--enable-libmp3lame [03:48] <relaxed> llogan: Hey. What kernel does it ship with? [03:48] <demonjester> i cant fine headers for lame so it seems ffmpeg is out of the question. [03:48] <demonjester> thanks anyways [03:49] <llogan> https://dev.openwrt.org/browser/packages/sound/lame/Makefile [03:49] <defaultro> is it possible to send output to mplayer? This way I don't have to create a temporary file [03:50] <llogan> why do you want to do that? so you can see the effects of the filter? [03:50] <defaultro> yup [03:51] <llogan> ffplay -i input -vf hue [03:51] <defaultro> ah [03:54] <defaultro> llogan, can ffmpeg manipulate each RGB channel? [03:56] <defaultro> here is what I did. I took a snap of the problematic video and opened it in Photoshop. I modified hue settings and I was able to correct the sky. :) FFMPEG's hue filter should be able to do it too :) [03:59] <relaxed> llogan: I've been working on a distro agnostic guide for compiling ffmpeg to a custom locations, and avoiding package managers entirely. I could add a blurb about building it statically too. [04:00] <llogan> relaxed: sounds good to me. are you going to add it to the FFmpeg wiki? [04:01] <llogan> defaultro: i'm not sure. [04:01] <relaxed> I may, at first it will be a simple txt file in dropbox. [04:02] <relaxed> If work is slow tonight I'll finish it up and let you look at it. [04:05] <llogan> relaxed: thanks. [04:11] <defaultro> llogan, I can't get this to work with vf, mp=eq2=1.0:2:0.5 [04:11] <defaultro> it's contrast brightness [04:13] <ivor> I'm looking at http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide and notice many errors making it obvious nobody uses it [04:13] <ivor> Is there a better page for this? [04:14] <defaultro> llogan, i got it to work [04:17] <ivor> What does one do when one wants to compile the latest version of ffmpeg on linux mint? [04:19] <llogan> ivor: what are the many errors? [04:20] <llogan> defaultro: what did you do? [04:20] <ivor> Well, take a look at the page at http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide and notice how it deletes yasm and then pulls down a new version but never sets the path [04:21] <defaultro> i wrapped it with double quotes [04:21] <llogan> yasm in the repo is too old [04:21] <defaultro> however, i can't combine hue and contrast/brightness [04:21] <ivor> llogan: exactly, so the script attempts to download a new version but then never uses it [04:21] <llogan> the path defaults to /usr/local [04:22] <ivor> llogan: well the version that is pulled down only gets installed into the ~/yasm-1.... directory [04:23] <ivor> llogan: so if the script is used then it complains that there is no yasm on the system [04:23] <ivor> llogan: that error can be fixed if one updates their path to include the location the script installs yasm but that's just the first error with the script/directions [04:25] <defaultro> llogan, I can't get this to work. ffplay -i 00107.MTS -vf "mp=eq2=0.0:1:-.1:hue=h=26" [04:25] <defaultro> the hue is not working [04:26] <defaultro> ah. I got it to work using comma :) [04:27] <llogan> ivor: works for me. sounds like you skipped the checkinstall command. [04:27] <llogan> that guide will be changed soon (relatively) so there is no system installation [04:27] <llogan> and no screwing around with checkinstall/package manager. [04:28] <llogan> defaultro: which Panasonic model? [04:28] <defaultro> TM700 [04:28] <defaultro> Quality is amazing but the cyan/green cast is so annoying [04:29] <defaultro> I'm not sure if I should sell it and buy another video camera [04:29] <llogan> it occurs with default settings? are you sure it's not set to tungsten white balance when shooting in daylight? [04:29] <defaultro> yup, it's AWB [04:29] <llogan> do you have a small sample? [04:29] <defaultro> so many people have the same issue [04:29] <llogan> lame. [04:29] <defaultro> i can extract few frames [04:30] <defaultro> there are so many blogs about it [04:30] <llogan> i have a HMC150 laying around here that seems fine [04:30] <ivor> llogan: I'm not familiar with what checkinstall does but I did copy it in as shown... [04:31] <ivor> llogan: I'm reading the man pages on checkinstall now [04:32] <llogan> you can see if it installed with: dpkg -L checkinstall [04:33] <llogan> or: dpkg -s checkinstall | grep Status [04:33] <ivor> llogan: What would I be looking for in there, yasm, ffmpeg, or something else? [04:33] <llogan> duh... i meant yasm, not checkinstall [04:34] <llogan> dpkg -s yasm | grep Status [04:34] <llogan> should show "Status: install ok installed" [04:35] <ivor> llogan: no, it did not get installed [04:35] <ivor> llogan: Do I need to run the script with sudo? [04:35] <llogan> what script are you referring to? [04:36] <ivor> llogan: oh, sorry, I put the directions into a script so I could be sure I got everything right [04:36] <defaultro> llogan, where can I upload the 5 sec extracted clip? [04:37] <llogan> you don't have your own server? [04:37] <defaultro> not anymore, I shut it down llast week [04:37] <llogan> let me guess...you got rid of the neckbeard too? [04:37] <defaultro> :D [04:37] <llogan> get. out. [04:37] <defaultro> let me turn on httpd here [04:37] <llogan> datafilehost or mediafire are fine [04:38] <defaultro> oh ok [04:38] <llogan> i'd like to shut mine down, but i still have some legacy clients. im bored of server admining [04:39] <defaultro> :) [04:39] <defaultro> i didn't want to pay $120/year [04:40] <defaultro> upload should be done soon [04:40] <llogan> we're paying more than that per month [04:40] <llogan> not for long though [04:42] <llogan> ivor: as i mentioned earlier, it seems as if checkinstall was skipped or did not run successfully [04:43] <defaultro> :) [04:44] <defaultro> i'm only uploading 4.5mb but it's taking a long time [04:44] <defaultro> http://www.datafilehost.com/download-48faa237.html [04:45] <defaultro> maybe the reason why I'm having a hard time correcting the sky is because I used a circular polarizer [04:45] <llogan> where is that? [04:45] <ivor> llogan: I guess I have to go back and look at my script [04:46] <defaultro> Playa del Carmen, Mexico [04:47] <llogan> it looked familar but i couldn't think of the name [04:47] <defaultro> Blue something. Saw nude women there [04:48] <llogan> were they cyan? [04:48] <defaultro> yes [04:48] <defaultro> beautiful water [04:48] <defaultro> or turquoise [04:49] <defaultro> but look at the sky, it's not blue but i remember it was a really nice blue sky [04:50] <defaultro> i'll check some new videos. That mexico was from 2010 [04:54] <defaultro> some of my videos have green/cyan cast [04:54] <defaultro> on some videos, I remember using custom white balance. The skies looked fine [05:10] <fenduru> Anyone know how to avoid any padding at the end of each segment when using -f segment [06:27] <bigmac> i have a question about hosting lots of large images on my local webserver... current format is jpg, can i convert them to load faster and not loose much quality? [06:44] <klaxa> you could create thumbnails and have the user click on those before downloading the real thing [06:46] <fenduru> is there a way to do a 2 step conversion in one command? i.e. mp3 > wav > ogg [06:49] <klaxa> you want to have wav and ogg? [06:49] <klaxa> in one command line, yeah with many pipes though [06:50] <fenduru> well, my problem is that using -f segment [06:50] <fenduru> playing the outputted segments in sequence has gaps [06:51] <fenduru> but it seems to be an encoder or container issue, as straight PCM isn't giving me this issue [06:51] <klaxa> ffmpeg -i mp3_file.mp3 -c:a pcm_s16le -f wav pipe: | tee somefile.wav | ffmpeg -i - -c:a libvorbis somefile.ogg [06:52] <fenduru> so I'm trying to see if decoding to PCM.wav and then splitting that guy will alleviate my problems. [06:52] <fenduru> klaxa, thanks. I had considered that I was just trying to avoid multiple ffmpeg processes [06:52] <fenduru> but I'll give that a shot, ty [06:52] <klaxa> i didn't test it, but it should work [06:57] <fenduru> well, the pipe worked. still getting gaps though [10:56] <Bor0> could anyone help me with http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe… ? I am experiencing the same exact problem [11:59] <wahaha__> There is a file named input.txt file '/home/1.flv' file '/home/2.flv' [12:00] <wahaha__> I can use the command to Concatenate 1.flv and 2.flv into a mp4 file. [12:00] <wahaha__> ffmpeg -f concat -i input.txt -c copy output.mp4 [12:00] <durandal_1707> and? [12:00] <wahaha__> When i use the following command [12:01] <wahaha__> ffmpeg -i "concat:1.flv|2.flv" -c copy output.mp4 [12:01] <wahaha__> can run ,but wrong result ,only one file 1.flv in the output.mp4 [12:01] <wahaha__> ffmpeg -f concat -i input.txt -c copy output.mp4 is ok [12:02] <wahaha__> ffmpeg -i "concat:1.flv|2.flv" -c copy output.mp4 ,problem [12:02] <durandal_1707> perhaps it needs full path [12:04] <wahaha__> wait a minute ,let me try [12:09] <wahaha__> 'full path' can not solve it [12:10] <durandal_1707> with -v debug [12:11] <wahaha__> wait more moment [12:12] <wahaha__> i am in china , www.pastie.org was blocked by chinese gfw [12:15] <wahaha__> i have to prepare a ladder to break the blocked gfw [12:15] <durandal_1707> wahaha__: couldn't you just find paste site that is not blocked? [12:25] <wahaha__> please wait a moment [12:28] <relaxed> wahaha__: mkvmerge -o temp.mkv 1.flv +2.flv; ffmpeg -i temp.mkv -c copy done.mp4 [12:29] <wahaha__> let me try [12:30] <relaxed> ffmpeg's concat only works well with mpeg containers from what I've seen. [12:31] <durandal_1707> relaxed: but op issue is that one syntax works and another one do not [12:32] <wahaha__> mkvmerge -o /tmp/temp.mkv /tmp/1.flv +/tmp/2.flv bash: mkvmerge: command not found [12:32] <wahaha__> apt-get install mkvmerge Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package mkvmerge [12:34] <wahaha__> now i can get pastie.org,but i can't use it,it is difficult to use [12:35] <wahaha__> do you have email?i can send you my screen-grap picture [12:36] <relaxed> durandal_1707: oh :) [12:38] <durandal_1707> wahaha__: pastebin is blocked too? [12:39] <wahaha__> my screen-grap picture is on my desktop in png file,how can i paste it into the pastie.org [12:39] <durandal_1707> paste text [12:39] <durandal_1707> copy -paste text [12:40] <wahaha__> i have break it with some tools for several minutes,i can link pastie now [12:41] <wahaha__> had never used pastie.org [12:41] <durandal_1707> pastie* sites is for copy-paste of text [12:41] <durandal_1707> for uploading images there is bunch of other hosts [12:43] <wahaha__> here it is [12:43] <wahaha__> http://pastie.org/6124324 [12:46] <wahaha__> http://pastie.org/6124377 [12:46] <wahaha__> please see http://pastie.org/6124377 [12:50] <wahaha__> are you still there durandal_1707? [12:54] <leosaeba> hi to all. I'm trying to compile a self contained (not system installed) opencv (with ffmpeg video capabilities enabled) on raspberrypi to be used by an application I'm writing. [12:54] <leosaeba> I have problems compiling it [12:55] <leosaeba> I managed to compile and install x264, ffmpeg and opencv on x86_64 [12:55] <leosaeba> then I tried to do the same on the raspberry [12:55] <leosaeba> but opencv fails with: lib//libavcodec.so.54: undefined reference to `ff_flacdsp_init_arm' [12:56] <wahaha__> durandal_1707,what is the matter? [12:57] <durandal_1707> wahaha__: open bugreport [12:58] <burek> leosaeba, are you sure you really want to use libx264 on RPi? [12:59] <leosaeba> I'm pasting the whole compilation process on http://pastebin.com/G2GQx1y7 [12:59] <burek> no need to look at that [12:59] <leosaeba> burek: for now I'm just trying to get things up [12:59] <burek> most probably you'll get unusable system [12:59] <wahaha__> think you [12:59] <burek> and probably shouldnt waste your time with it [13:00] <burek> RPi is just to weak for any kind of video encoding [13:00] <leosaeba> I just need opencv to append jpeg frames on a video (after some computations) [13:00] <burek> find some better hardware for such purpose [13:00] <leosaeba> (my main constrain is money....) [13:00] <burek> well, ok, good luck with it, what can i say.. [13:01] <burek> i assume you'll use some usb webcam [13:01] <burek> so most probably you'll have serious usb issues too [13:01] <burek> im just trying to save your time=money here [13:01] <leosaeba> burek: [13:02] <leosaeba> no, actually I have remote nodes (with usb camera) that mount raspi filesystem [13:02] <burek> your RPi will generate video frames? [13:02] <leosaeba> they write frames there and raspi do some little assembly. [13:03] <burek> what do you mean by "remote nodes" ? [13:03] <leosaeba> rpi will append received frames to a video [13:03] <leosaeba> tcp/ip [13:03] <burek> i see [13:03] <burek> that might maybe work [13:03] <burek> but then you dont need libx264 on RPi [13:03] <leosaeba> I think so [13:03] <burek> you can remove it [13:04] <leosaeba> and ffmpeg will allow opencv to use VideoWriter? [13:04] <burek> just so you know, on RPi, ethernet controller is usb-based, so you won't have 100 mbps if you expect that too [13:04] <burek> the most you can hope for is < 5-6 mbps [13:05] <leosaeba> the problem I got on raspi (not on pc) is about flac-arm something [13:05] <burek> im not familiar with opencv, so you just need to try and test it [13:05] <burek> flac is an audio codec [13:05] <leosaeba> I've done some math, and for the bandwidth required it should suffice.. [13:05] <burek> avoid any kind of encoding on RPi.. [13:07] <burek> things might seem to be working at first, but you'll notice that at some point your RPi got frozen [13:07] <burek> rebooted itself and stuff [13:07] <leosaeba> I actually could work with uncompressed video, but my problem now is just to set opencv up. [13:07] <burek> so don't consider is as a cheap version of less-powerfull x86 machine [13:07] <burek> it* [13:07] <burek> what is the image size of the video [13:08] <burek> it shouldn't be a problem to set opencv as long as it doesnt require ffmpeg's encoding abilities [13:08] <burek> if you only need muxing, that might work [13:08] <leosaeba> burek: I'm not happy about broadcom gpu too....but I accepted the compromise on openness/cost [13:09] <burek> low price comes at the cost of everything else [13:09] <burek> you'll experience it sooner or later [13:09] <burek> i just wouldnt like that you spend a lot of time setting all up just to realize what i said above [13:09] <leosaeba> imagesize is not a constrain...I will start with HD (720 I believe) and gracefully downgrade until the system works. [13:10] <burek> forget hd.. [13:10] <burek> i had a camera with internal h264 encoder [13:10] <burek> and all RPi was supposed to do was to mux the stream and re-stream it [13:10] <burek> and it failed.. [13:10] <burek> so.. i think the math is pretty obvious [13:11] <leosaeba> I see... [13:11] <burek> cpu could manage it all, but the data bus couldn't [13:11] <burek> everything was usb-based and all the data travel over that usb bus [13:11] <leosaeba> My remote node feeds is jpeg frames [13:11] <burek> so conflicts are numerous at least [13:11] <burek> that's why i noted the ethernet usb controller too [13:11] <leosaeba> i understand [13:12] <leosaeba> well..anyway I will continue to try (at leas until I will reach the bottle-neck you are referring to) [13:12] <burek> ok [13:12] <leosaeba> ..I've got no alternatives right now [13:13] <Bor0> could anyone help me with http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe… ? I am experiencing the same exact problem [13:13] <burek> you have an rpi compilation guide on our wiki though [13:13] <burek> it might help [13:13] <leosaeba> actually I have set up slackwarearm, and everything is ok. [13:14] <burek> this "lib//libavcodec.so.54: undefined reference to `ff_flacdsp_init_arm'" means that you need a specific version of ffmpeg [13:14] <leosaeba> compilation is slow, but yesterday I compiled ffmpeg (stripped of a lot of things) in a few hours. [13:14] <burek> which was in mind of opencv programmers at the time they decided to use ffmpeg [13:14] <leosaeba> I compiled ffmpeg-1.1.2 [13:14] <burek> and you probably got latest [13:14] <burek> so it failed [13:14] <leosaeba> ah! [13:15] <leosaeba> I didn't think of that. [13:15] <leosaeba> on opencv documentation they refer to minimum version of ffmpeg, so I thought to use the lates version. [13:16] <burek> well see what they recommend [13:16] <burek> if it is 0.x then use 0.x [13:16] <burek> 1.x had a lot of changes [13:16] <burek> so it might be no wonder why it doesnt work [13:16] <leosaeba> I see [13:17] <leosaeba> I got tricked by the fact that everything worked on x86_64....but the error is arm related. [13:17] <burek> hmh.. most probably [13:18] <burek> on your x64, you have both ffmpeg on your system [13:18] <burek> one installed through package management [13:18] <burek> and another (unneeded) your own compiled [13:18] <leosaeba> Now I will try with the latest 0.x version (in the meantime I investigate which version of ffmpeg opencv 2.4.3 used) [13:19] <leosaeba> ...to avoid this chance I unistalled system ffmpeg ( but I think something could still be present with vlc) [13:19] <burek> well, dpkg -l | grep libav [13:20] <leosaeba> dpkg is debian/ubuntu? [13:20] <burek> yes [13:20] <leosaeba> I will check slackware way, just a sec [13:22] <leosaeba> I've looked in /usr/lib64 for any libav* (none present) [13:22] <leosaeba> then I checked if vlc used some libav* somewhere else [13:23] <leosaeba> and I found this: /usr/lib64/vlc/plugins/codec/libavcodec_plugin.la and /usr/lib64/vlc/plugins/demux/libavformat_plugin.la [13:24] <leosaeba> but I don't think it is it. [13:24] <burek> Bor0, it might be easier to ask the author of those tutorials, rather than us here, because i dont believe anyone here has got time to read those coding samples just in order to tell you what the issue was [13:25] <burek> libavcodec_plugin.la is vlc's wrapper for libav* libraries [13:25] <Bor0> I see. is there any official tutorial for this? [13:25] <burek> installed libav* libraries [13:25] <burek> Bor0, also see docs/examples in your ffmpeg dir [13:26] <Bor0> alright, will check those, thanks. [13:26] <burek> :beer: :) [13:28] <Bor0> yoghurt with burek ;) [13:28] <burek> +1 :) [13:37] <leosaeba> burek: Ok, now I'm trying to check if using ffmpeg-0.11.2 (without x264) works on my pc, then I will try the same on raspi. In the meanwhile thank you for helping. [15:49] <jarno> Hi all, I have a question about ffmpeg/libx264...am I in the right place to make such a question? [15:50] <durandal_1707> yes [15:52] <jarno> ...great...I'm using ffmpeg - libx264 to write video (actually it's OpenCV that's calling ffmpeg) and I get the infamous "broken ffmpeg default settings detected" [15:53] <jarno> I'm using Kubuntu 12.10 and OpenCV, ffmpeg, x264 etc. is all git:ted today (should be stable) and I was wondering from where (which source code file) the default settings get written from [15:55] <jarno> ...so that I could experiment with different settings...I wonder if it is the static const AVCodecDefault x264_defaults[] in libx264.c [15:56] <jarno> ....I would really appreciate if someone could set me on the right track..... [15:56] <durandal_1707> you should see what OpenCV use/calls and what expects [15:57] <durandal_1707> "because broken ffmpeg default settings detected" in nonsense [15:57] <jarno> I was checking out the OpenCV part and as far as I can see, it would seem that so called "ffmpeg" defaults are being used [15:57] <durandal_1707> than call ffmpeg with other options [15:58] <jarno> ...hmmm, what do you mean that "broken ffmpeg default settings detected" is nonsense? [16:00] <durandal_1707> broken in what sense? what is broken and why it is broken? .... [16:00] <jarno> ...I mean that I get the following error message from libx264: [libx264 @ 0x12c7440] broken ffmpeg default settings detected [16:04] <durandal_1707> what ffmpeg version are you using? [16:06] <jarno> ffmpeg version N-38355-ge2b703f Copyright (c) 2000-2013 the FFmpeg developers [16:06] <jarno> ...I just git:ted it today... [16:08] <Mavrik> huh [16:08] <Mavrik> that "broken ffmpeg settings" is a message from x264 [16:08] <Mavrik> very old [16:08] <Mavrik> jarno: default settings aren't read from files [16:08] <Mavrik> you set them via context options [16:08] <jarno> ...yes, I know, but as far as I understand it, the default settings are written by libffmpeg, but I might be mistaken... [16:08] <Mavrik> however if you get that error the OpenCV bindings must be ancient [16:09] <jarno> ...ok...that might be the case [16:10] <jarno> ...since I am not an expert (very far from it), from where I should search for examples etc. in order to get a better idea of how to use libffmpeg? [16:11] <Mavrik> hmm, no idea [16:12] <Mavrik> since libffmpeg seems like some 3rd party product :\ [16:12] <jarno> really? [16:12] <Mavrik> yep, ffmpeg libraries are named libav* (libavcodec, libavformat, libavutil) [16:12] <Mavrik> and have such prefixes when calling them (avformat_, avcodec_) [16:13] <Mavrik> libffmpeg seems like some wrapper around them to ease programming [16:13] <jarno> ...ok... [16:13] <Mavrik> sadly, i gotta run now, hopefully you find a solution :) [16:13] <jarno> thanks!!!! [16:13] <jarno> ...hmmm...perhaps I should stick to gstreamer then...it seems to be a more straight forward solution... [16:22] <JEEB> jarno, depends on what you're doing [16:23] <jarno> ...I'm reading images from uEye cameras... [16:23] <jarno> ...and then I just need to write them to a video without loosing too much information... [16:24] <jarno> ...I'm not really an expert when it comes to gstreamer nor ffmpeg, but I have done some simple stuff using gstreamer and it seems to be fairly well documented [16:24] <jarno> ...and "easy" to use... [16:24] <JEEB> well, I've just seen way too many people derp at it over at #x264 , but I have no idea what's the best API to do the camera reading for you to be honest :s [16:25] <jarno> ....actually, I'm getting the images using the API from uEye.... [16:25] <jarno> but writing videos does not function in Linux with the same API [16:25] <JEEB> Basically I'd use whatever is the best API to grab the pictures, depending on the colorspace push it through swscale or something to convert it if your output needs to be something != input, then encode with libx264, and then mux into a container with libavformat or something [16:27] <jarno> I'll check out what you are proposing [16:27] <jarno> ...and see what is the "easiest" / "fastest" way of doing it for the time being [16:29] <jarno> but I quess that what Mavrik was saying about the OpenCV bindings is right [16:29] <jarno> ...they are just damn old...but I'll check that out tomorrow [16:30] <JEEB> I have no idea of your use case, but I am kind of reading through the lines that you might get raw video frames from the camera, in some colorspace -> if the colorspace is something you don't want to encode as-is (f.ex. RGB with libx264 is possible, but you won't really be capable of playing it with anything but FOSS, no flash or hardware decoder will like that) you feed the raw frames to swscale, or another converter of your choice [16:30] <JEEB> that converts the picture to the colorspace you want -> libx264 itself to encode the pictures into raw H.264 (Annex B) -> take the data that libx264 outputs and put it through libavformat to put the raw H.264 streams into a container [16:30] <JEEB> and that would be it [16:30] <JEEB> libswscale and libavformat are libraries that ffmpeg provides, and libx264 is well, WhatItSaysOnTheTin [16:33] <jarno> ...thanx a lot for the description man!!! [16:34] <jarno> I really appreciate it!!! [16:43] <jarno> ...since I'm pretty noob when it comes to this, from where I should/could start learning about using libav:s?... [16:44] <JEEB> depends on what exactly you want to do? [16:44] <JEEB> I mean, the trunk doxygen is as good as it gets [16:45] <JEEB> (you will rip your hair off at some point, but on the other hand that unfortunately generally is better than the alternatives) [16:45] <jarno> ...just encode and write into a container... [16:45] <jarno> ...so there are no good "tutorials" going around as normally? [16:46] <JEEB> I would probably recommend just using libx264 itself for the encoding, thus x264.h is the thing to read through [16:46] <JEEB> and then libavformat, well that is in the libavformat doxygen [16:46] <jarno> ok [16:46] <JEEB> and yes, there are samples but if they are out-of-date you get the fun time of screaming at the developers [16:46] <JEEB> http://libav.org/doxygen/master/index.html [16:47] <jarno> ...hahahahah :D [16:47] <JEEB> the doxygen is auto-generated so you at least have it be generally up-to-date [16:48] <JEEB> also do make sure what the picture format of what you're getting from the camera and what you want to feed to x264 are the same thing, otherwise you'll have to stick libswscale in the middle [16:48] <JEEB> or something else [16:48] <JEEB> which pain to take is up to you [16:49] <jarno> ...ok...happily I know what the image format is and can actually change it using the API supplied by the vendor... [16:49] <JEEB> okies, as long as that is something you a) can and b) want to stick to x264 :) [16:50] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=tree;f=doc/examples;h=36f7e2da0aa85… <- and the samples are in doc/examples btw [16:50] <JEEB> I would guess scaling_video is swscale stuff and muxing is well, that [16:51] <jarno> ...thanx...I'll check it up tomorrow...gotta go now...and once more, thanx for your help... [16:51] <jarno> cheers!!!!! [19:00] <Sam_> yesterday i had asked one question about high CPU usage in ffmpeg, one fellow member said update ffmpeg to latest version. I did migrate to latest version but still i see during flv transcoding CPU reaches about 120% [19:01] <Sam_> any suggestions what could be the reason behind it? [19:03] <Sam_> why ffmpeg is eating up so much CPU during transcoding? [19:03] <sacarasc> Because it's trying to go as fast as possible. [19:03] <sacarasc> You could make it go real time, which would presumably use less CPU... [19:03] <stqn> you could use https://github.com/opsengine/cpulimit [19:03] <Sam_> <sacarasc> but this will reduce throughput right? [19:05] <sacarasc> If you want it as fast as possible, it's going to use as much CPU as possible. [19:06] <Mavrik> Sam_: well it's supposed to use ALL CPU [19:09] <Mavrik> your question is... strange. [19:09] <Sam_> i am limiting it to 1 thread [19:09] <Sam_> <sacarasc> Correct but while doing 20 parallel transcoding it reduce the throughput of the system by significant amount [19:09] <Sam_> <Mavrik> can you please put some light why question is strange [19:26] <framer99> I have a working ffmpeg cmdline that has jitter when run as part of initscripts on centos 6.3 [19:27] <framer99> I've tried many many things to figure it out, no luck. I can restart the process after bootup and it runs clean with no jitter. [19:27] <framer99> full details here : http://pastebin.com/L2wwX3iP [19:28] <framer99> I had it setup as a Upstart process/task and I could just ssh in and restart that Upstart task and it will run fine. I just can't figure out what is different about running during boot [19:29] <framer99> thansk in advance for any advice [19:29] <durandal_1707> that is so old, more than 2 years [19:30] <framer99> yeah i was goign to compile myself eventualy just never got to it.. let me do that and report back [19:55] <framer99> rrrr.. remaking with x11grab support [20:22] <charleszivko> in a program like Audacity I can see the waveforms of the each file and manually match the peaks of them. I'm probably using the wrong term , sorry. [20:22] <charleszivko> I know programs like final cut pro auto match two audio files [20:23] <charleszivko> trying to figure out how I can implement such a feature myself [20:24] <durandal_1707> you mean same data in several bytes? [20:25] <charleszivko> similar, sound files are recorded on different devices. A mic. B video from a iPhone [20:25] <charleszivko> so it's not exactly the same data [20:26] <durandal_1707> not currently, but nobody stops you to fill feature request on bug tracker for such audio filter [20:28] <charleszivko> cool. any pointers on a alternative library that may do this? [20:28] <durandal_1707> if i know one, i would already told you, i do not think sox have anything like that [20:29] <charleszivko> oh thanks, yeah I have been researching and already found sox. doesn't seem to handle what I am looking for either. [20:29] <charleszivko> appreciate your input [20:32] <durandal_1707> there is showwaves filter so you could just watch it ;) [20:33] <charleszivko> haha, yeah I found that in the docs. Was hoping I missed something that I might be able to leverage. [20:34] <durandal_1707> but you are basically looking for something like diff tool [20:35] <charleszivko> yeah. I know audio A and audio B have a match within a 1 - 2 second timestamp. So I need to analyze those to get the exact timestamp from each where they first match so I can use it as a offset to sync them up. [20:36] <charleszivko> B would replace A (video audio) but needs to match the video [21:00] <framer99> durandal_1707: build from latest git seems to be running smoother from initscripts now.... thanks for the pointer that I was way out of date, should have paid more attention [22:09] <FFmpeg> Hello friends :) [22:09] <fatpony> can somebody explain me why cropdetect constantly yields values that very clearly overcrop the source? [22:13] <FFmpeg> I am having some trouble with ffserver. I am trying to live stream via asf. Everything seems to be working, but when I try to open my stream in windows media player I get an error that it can't play the file. [22:48] <llogan> fatpony: adjust the limit threshold? [22:49] <fatpony> llogan: i tried to do that but i don't get realistic values [22:49] <fatpony> low threshold = less aggressive crop, right? [22:49] <llogan> yes, AFAIK [22:54] <llogan> you could use drawbox to get an idea using the values from cropdetect [22:54] <llogan> ffplay -f lavfi color=s=1024x720:c=blue -vf "pad=iw:ih+80:0:40,drawbox=0:40:1024:720:red@0.5" [22:54] <llogan> or similar [22:54] <llogan> of course your input will be a video and not a color source from lavfi [22:55] <llogan> ...and you won't need pad obviously [23:23] <fenduru> I have a .wav (pcm_s24le encoded) that ffprobe reports as 9.97s long, however when I reencode that to aac (.mp4), ffprobe reports 10.00 seconds [23:24] <fenduru> How can I avoid this? I need the .mp4 to be precisely the same as the wav in terms of duration [23:27] <fenduru> encoding to mp3 gives it a duratnoi of 10.03s [23:27] <fenduru> So I'm assuming it is a container issue (i.e. frame length is fixed, so the end is padded), but I need to force it to be precise [00:00] --- Wed Feb 13 2013

1 0

ffmpeg-devel.log.20130211
by burek 12 Feb '13

12 Feb '13

[01:11] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:8491ac3ad2e9: mlpdec: check ch_assign more completely [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:55e35c98d5f3: h264 intra pred: add missing includes. [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:c7e3e55429cb: Move ff_emulated_edge_mc prototypes to videodsp. [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:5e74a2b68563: bit_depth_template: don't depend on dsputil.h. [01:12] <Darkarnium> Hey Compn, sorry, I didn't see your message last night until late [01:18] <Darkarnium> Pretty damned nasty, and only being used in a development environment as a PoC at the moment: http://pastebin.com/3LNcery6 [01:19] <Darkarnium> Worth noting, I'm using the snprintf and av_dict_set and 'headers' as 'cookie' does not seem to apply for my test. Whether I specify the options as per RFC6265 or as just an attribute value pair. [01:30] <wm4> someone is reporting broken decoding with h264_vda in recent ffmpeg git (might be my fault, but I don't know and it used to work fine) [02:36] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:fd6a021d8e3d: get_bits: return pointer to buffer that is the result of the alignment. [11:31] <Tjoppen> someone's improving my cinepak encoder toy. this makes me happy :) [11:50] <durandal_1707> i want that filter! - http://slowmovideo.granjow.net/videos.html [11:53] <Tjoppen> ooh [11:59] <durandal_1707> michaelni: when gbrp16 output? [12:09] <cone-615> ffmpeg.git 03Paul B Mahol 07master:5d7009dac28b: evrcdec: fix wrong condition [12:10] <michaelni> ill look into adding gbrp16 support if noone else does ... [12:17] <someone-noone> Hello! Question related to vda_h264 decoding. Does VDADecoderDecode in sync mode returns frames in display order? Docs are saying that frames are always in decode order, however ffmpeg reorders it only in async mode and do not reorder in sync mode. I'm confused. Is there smth I've missed in sources? [12:23] <michaelni> maybe ask dilaroga or xidorn quan [12:27] <someone-noone> thanks. How often are they online on this channel? [12:27] <someone-noone> or better try mail? [12:28] <cone-615> ffmpeg.git 03Diego Biurrun 07master:d46d87d1d3de: sparc: dsputil_vis: Fix silly variable name search and replace typo [12:28] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:c3a62d3507b2: Merge remote-tracking branch 'qatar/master' [12:31] <michaelni> probably better mail, ive seen dilaroga here but not common enough for waiting to be a good idea IIRC [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:272e7f6443b7: huffyuvdec: Check init_vlc() return codes. [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:ac476bfa9f90: huffyuvdec: Skip len==0 cases [12:41] <cone-615> ffmpeg.git 03Reinhard Tartler 07release/0.5:5c9d2d837783: Bump version number for the 0.5.10 release [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:0f6d4da8def2: bmpdec: only initialize palette for pal8. [12:41] <cone-615> ffmpeg.git 03Mina Nagy Zaki 07release/0.5:4475a7d88b89: lavfi: avfilter_merge_formats: handle case where inputs are same [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:6b97e76dfca8: avsdec: Set dimensions instead of relying on the demuxer. [12:41] <cone-615> ffmpeg.git 03Aneesh Dogra 07release/0.5:c28c631d29a8: bytestream: add a new set of bytestream functions with overread checking [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:d1729c3715af: avidec: use actually read size instead of requested size [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:4fac60d56863: cavsdec: check for changing w/h. [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:2ae6bdbb9b17: avidec: return 0, not packet size from read_packet(). [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:d4e4234147c9: yuv4mpeg: return proper error codes. [12:41] <cone-615> ffmpeg.git 03JindYich Makovi ka 07release/0.5:6731776795a8: h264: avoid stuck buffer pointer in decode_nal_units [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:5235db68c062: mpegaudiodec: fix short_start calculation [12:41] <cone-615> ffmpeg.git 03Alex Converse 07release/0.5:80f89a9b40cc: tiffenc: Check av_malloc() results. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:1f1b2f18067b: yuv4mpeg: reject unsupported codecs [12:41] <cone-615> ffmpeg.git 03Janne Grunau 07release/0.5:2e1474fd9988: lavf: avoid integer overflow in ff_compute_frame_duration() [12:41] <cone-615> ffmpeg.git 03Dale Curtis 07release/0.5:c3761b661874: Fix uninitialized reads on malformed ogg files. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:fe4409a396d7: oggdec: check memory allocation [12:41] <cone-615> ffmpeg.git 03Janne Grunau 07release/0.5:a49599b1255e: h264: check context state before decoding slice data partitions [12:41] <cone-615> ffmpeg.git 03Justin Ruggles 07release/0.5:7aeb281aa507: shorten: check for realloc failure [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:9def5c466648: shorten: Use separate pointers for the allocated memory for decoded samples. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:4f8f4458a5a8: vp56: release frames on error [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:b9500bf864e9: vp6: properly fail on unsupported feature [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:10ec2308b0d4: Merge remote-tracking branch 'qatar/release/0.5' into release/0.5 [12:57] <durandal_1707> michaelni: what was idea of noise filter? and how addgrain (film grain) relates to it? [13:12] <cone-615> ffmpeg.git 03Carl Eugen Hoyos 07master:182e342fdbc7: Print mxf Universal Label if the user requested verbose logging. [13:19] <durandal_1707> shouldn't colormatrix filter actually use libswscale? [13:21] <michaelni> the idea of noise was to add different kinds of noise and grain ... [13:24] <durandal_1707> i don't think there is way in lavfi to overlay frames with differen opacity [13:52] <durandal_1707> how should i name blend filter? mix/merge/blend [13:56] <iive> mix sounds too audio related. [13:58] <durandal_1707> iive: its called amix [14:00] <iive> the filter may be called amix, but I do associate the mix word with audio. [14:12] <durandal_1707> ubitux: want to add different edge detectors (not just canny) to edgedetect? [15:23] <ubitux> durandal_1707: i called the filter "edgedetect" and not "canny" for that reason, so feel free to add some enhancement [15:26] Action: ubitux is going back tomorrow :( [15:44] <durandal_1707> ubitux: i already have bunch of other work, and you wrote edgedetect and are most qualified person to extend it [15:46] <cone-615> ffmpeg.git 03Paul B Mahol 07master:8160864b0027: lavfi/amerge: make description consistent with documentation [16:02] <ubitux> durandal_1707: i have no plans for it atm [16:21] <cone-615> ffmpeg.git 03Derek Buitenhuis 07master:c177f2ec4a21: libfdk-aacenc: Actually check for upper bounds of cutoff [17:14] <durandal_1707> ubitux: your opinion on blend filter? [17:14] <durandal_1707> overlay already have some blend code - very limited one though [17:16] <durandal_1707> also i dont understand this complicated logic with overlay/alphamerge request_frame thing [17:57] <durandal_1707> michaelni: i added yuv444 support to overlay and it is broken, works if i add format=yuv444p to begin and end any pix fmt of filter chain [17:58] <durandal_1707> this is strange [18:01] <durandal_1707> filter expect that overlay and main are of same format but it does not happen to actually be in config_input [18:10] <durandal_1707> negate/lut debug filter output is boring spam [18:11] <durandal_1707> it inserts scaler 444 -> 420 in middle [18:11] <durandal_1707> perhaps ffplay nonsense [18:15] <durandal_1707> indeed, it happens with ffplay only [18:25] <durandal_1707> so i want basically VV->V filter without pts nonsese stuff invading my code [18:26] <durandal_1707> is it doable? [18:30] <durandal_1707> another thing is that filters write to first frame, I dont find this much useful for blend and want to have two input read only frames and output writeonly [18:34] <michaelni> you can set permission on the inputs [18:34] <michaelni> or a filter can output read only buffers so they wont be written into [18:34] <durandal_1707> yes, I know that, but what about this PRESERVE and pts nonsense [18:35] <durandal_1707> i just read pixels from 2 inputs and write pixel to output [18:36] <durandal_1707> and both 2 inputs should have same format [18:37] <michaelni> for same format just make them use the same list and you should get the same format [18:41] <durandal_1707> michaelni: how? see bug with ffplay and overlay when i added yuv444p [18:44] <michaelni> i think vf_overlay uses seperate lists for its 2 inputs not sure thats the issue though [18:46] <durandal_1707> formats are set with query_formats and its global for all inputs/outputs [18:59] <durandal_1707> so filter_frame() can take two inputs at same time? [19:03] <durandal_1707> looks like i need to do some kind of buffering [19:04] <durandal_1707> this is nonsense, why would any filter would need to do this? [19:07] <durandal_1707> i will just copy-paste code [19:13] <durandal_1707> saste: about il: feel free to simplify options if you think that per plane options do not make sense [19:14] <durandal_1707> saste: what would be template of filter that takes 2 readonly inputs and 1 write only output? [19:14] Action: durandal_1707 thinks that having filter templates could help novice devs [19:20] <saste_> durandal_1707, overlay? [19:21] <durandal_1707> overlay writes to first input [19:21] <durandal_1707> that is useless for blend [19:23] <durandal_1707> well not always useless, but i imagine scenario where it would be useful to have original input [19:24] <durandal_1707> blend is just geq but instead on single frame it operates on VV->V [19:29] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:0aa1d848ece6: ffv1enc: better heuristic to calculate initial states [20:14] Action: durandal_1707 now that is priority ^ [22:04] <llogan> why is rbg24 not automatically chosen for gif output instead of the current AV_LOG_ERROR message? [22:05] <llogan> it is confusing to users [22:05] <llogan> *rgb24 [22:18] <llogan> Compn: i'm not against your request, FYI. [22:31] <Compn> llogan : i know, i'm just pointing out -ss 30 is seconds, so -ss 30:30 is minutes+seconds [22:35] <llogan> gif appears to support pal8, (if you use -f image2) but that message does make that apparent. [23:00] <llogan> s/does/does not/ [23:07] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:972771dcf27d: h264chroma: remove duplicate 9/10 bit functions. [23:39] <BBB-work> michaelni, one more for you, a file encoded with "./ffmpeg -i bear_mpeg4_mp3.avi -acodec copy -vcodec mpeg4 -flags mv4+qpel+mv0+cbp+aic -trellis 1 -dia_size 1040 -g 250 -bf 8 -b_strategy 2 -cmp sad -subcmp rd -mbd 2 -precmp sad -last_pred 4 -subq 8 -bidir_refine 4 -qns 3 bear_mpeg4asp_mp3.avi" apparently fails to playback properly in ffplay (stutters a lot) because of a bug in compute_pkt_fields() for files with b-frames that are not [23:39] <BBB-work> h264 [23:39] <BBB-work> [let's continue that here] [23:40] <michaelni> ok [23:42] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:b582af1ed778: Use simple nop codes for <= sse (rather than <= mmx). [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:4c69cafceea5: doc/filters: fix position of il filter docs [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:af60e451509a: doc/filters: apply minor fixes to il filter docs [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:64353be81135: doc/filters: rework idet filter description [23:56] <michaelni> BBB-work, i see the stuttering too without encoding with -bf 8 [00:00] --- Tue Feb 12 2013

1 0

ffmpeg.log.20130211
by burek 12 Feb '13

12 Feb '13

[00:34] <rbrauer> I'm piping raw video (rgba) to h264 in mp4 via Java which produces a valid video if I use -t to set duration. Does anyone know how I would signal ffmpeg to close the file without an explicit duration? I tried sending 0x1a, but that didn't work. I'm doing this on Windows 7. [01:00] <fenduru> rbrauer: I think 0x1A is only for text files. Are you closing the stream properly? [01:01] <rbrauer> I tired closing both input and output streams on the ffmpeg process, but it still didn't complete the file correctly [01:03] <rbrauer> after closing the video pipe it just sits there with no messages [01:03] <fenduru> are you closing the stream without sending an EOF character? [01:04] <FergusL> anybody with a mp3->ogg ffmpeg command at hand ?! I can't seem to find one working... [01:06] <rbrauer> fenduru: what byte am I supposed to send for EOF? [01:08] <fenduru> rbrauer: Honestly I'm not sure. But I would try just closing the stream without one. [01:08] <rbrauer> I tried closing it from the Java side (ffOut.close()), wasn't working [01:09] <rbrauer> I mean I get no repsonse from ffmpeg and the file isn't correct [01:09] <fenduru> What type of stream are you dealing with on the Java side? (so I can lookup javadoc) [01:10] <fenduru> FergusL: try using -acodec libvorbis and have your output file be NAME.ogg [01:10] <FergusL> fenduru: I have vorbis but not libvorbis, weird [01:11] <FergusL> oh wait [01:11] <FergusL> it's in avconv now... [01:11] <FergusL> Ubuntu here [01:12] <rbrauer> fenduru: its a plain outputStream coming out of a Process wrapped with a bufferedOutputStream [01:12] <rbrauer> I'm calling ffOut.flush() then ffOut.close() [01:22] <fenduru> I'm out of ideas. I'm not familiar with piping/stdin on windows. [01:22] <fenduru> Looking around it seems that there is no standard EOF in the spec [01:22] <fenduru> Others suggest using -1 [01:23] <fenduru> I'm piping data from node.js into ffmpeg's stdin [01:24] <fenduru> and using stream.end() to send the eof, which works great on Linux [01:24] <fenduru> Perhaps try that, and if it works take a look at node.js' source for what it sends for EOF? (or if not node.js something else that has a similar .end() function) [01:26] <fenduru> FergusL: did you figure out your problem? [01:26] <FergusL> fenduru: yes I used avconv and there I could use -acodec libvorbis [01:27] <FergusL> that's just crazy, avconv is a wrapper [01:27] <FergusL> could just be a symlink [01:27] <FergusL> to ffmpeg, but well... enough arguing over this I think [01:33] <rbrauer> I figured out what was happening I think. Apparently when I closed the BufferedOutputStream it didn't close the actual one. I did ffProc.getOutputStream.close() after closing and flushing the buffered stream and it seemed to work. [01:35] <rbrauer> well I thought that worked...maybe not [01:37] <fatpony> /join #avcon [01:37] <fatpony> duh [01:41] <rbrauer> yeah that worked my code is just really messy now and I keep putting stuff in the wrong place, thanks for the help fenduru [01:43] <fenduru> Sure thing rbrauer. [02:14] <fenduru> Any reason that ffmpeg would return successfully if it isn't segmenting the output as specified? [02:43] <fenduru> Oh, the ffmpeg in the Ubuntu repo is very out of date [02:44] <klaxa> the ffmpeg in the ubuntu repo isn't ffmpeg [02:46] <fenduru> yeah, but even the one listed on the ffmpeg site is old [02:50] <klaxa> huh? [02:50] <klaxa> fenduru: elaborate [02:52] <fenduru> klaxa: https://launchpad.net/~jon-severinsson/+archive/ffmpeg is listed as the repo to use for ffmpeg [02:52] <fenduru> but is at version 0.10 instead of 1.x [02:53] <klaxa> oh true that [02:54] Action: fenduru is compiling from source now [02:54] <klaxa> the only sane thing to do really :) [02:55] <fenduru> yep :) I've been spoiled with arch's rolling release / package system [02:56] <fenduru> although I should really just use one of the static builds that are online... this amazon EC2 server is quite slow [03:09] <asher^> is there a way to have ffmpeg add metadata to an existing file? [03:17] <klaxa> asher^: https://www.ffmpeg.org/ffmpeg.html#Main-options see: -metadata [03:17] <klaxa> to not touch the rest of the streams, do: -map 0 -c copy [03:18] <asher^> great, thanks, ill try that out [03:18] <fenduru> klaxa, any idea how to avoid choppiness when segmenting your output? [03:18] <klaxa> what do you mean with choppyness? [03:19] <fenduru> if i recombine sequential output segments, sometimes there are clicks between the segments [03:19] <fenduru> I assume there is some sort of padding going on here [03:19] <klaxa> i.e. audio clicks? [03:19] <fenduru> yep [03:19] <asher^> klaxa, Unrecognized option 'c'. Any idea why this would happen? [03:19] <klaxa> asher^: ffmpeg too old [03:20] <klaxa> fenduru: that's probably because the last sample of the prepending segment, or the first sample of the following segment are not at 0 amplitude [03:20] <klaxa> at least that's what i would think [03:21] <klaxa> those things produce audio-clicks [03:21] <klaxa> open the audio stream in a graphical audio editor and you should be able to see that [03:21] <klaxa> audacity is pretty well suited for that [03:22] <fenduru> I'll take a look. I'm segmenting for streaming on the web so I'd like to be able to reconstruct the segments losslessly [03:22] <klaxa> asher^: install a newer version from your distributions repository, if they don't offer a newer one, compile from source or grab a static build [03:22] <asher^> yeah ive got the latest from my distro, looks like ill have to compile. thanks [03:25] <asher^> also, anyone know an alternative to ffmpeg-php? it causes 502 errors in some circumstances on my machine [13:19] <veleno> hello everyone. is it possible to use ffmpeg/ffprobe to statically analyze a file and detect lost packets/lost frames ? [13:25] <durandal_1707> veleno: i guess it depends on container/codecs [13:25] <durandal_1707> which means more info is needed [13:26] <durandal_1707> some containers simply are not designed that way [13:26] <durandal_1707> also lost packets/lost frames is not same as corrupted packets/frames that fail to decode [13:27] <veleno> durandal_1707: what would be the ideal container/codecs to detect those info? i have the liberty to chhoose, a this point. [13:28] <veleno> lost packets/frame do have a mapping to corrupeted packets/frame, do they ? [13:28] <durandal_1707> corrupted/incomplete frames are reported by lavc when decoding fails [13:29] <durandal_1707> this depends on decoder, because some decoders may try to recreate data [13:29] <veleno> (when I refer to lost/corrupted packets, it imply an input file received by a network streaming, not rtp-based) [13:30] <durandal_1707> .. but than usually it is displayed in log what is happening .. [13:30] <veleno> i see. [13:31] <durandal_1707> veleno: than you need to look at that stream protocol/format [13:32] <veleno> durandal_1707: that i know that is udp. the stream is sent over the network with somehting like: ffmpeg -re -i video.mp4 -f mpegts rtp://127.0.0.1:60000 -loglevel debug -report -vstat [13:32] <veleno> with udp instead of rtp in the streaming url [13:35] <durandal_1707> then you need to find out if mpegts support some kind of packet drop detection [13:36] <mf_> I've read it somewhere, but has forgotten what it stated, but what does max-rate really make for a difference when you run 2-pass encoding? Did it distribute the bitrate more evenly across all frames, or did it "move" the unused bits from a non/barely-moving sequence to the one that has more movement? How can I optimize this further? (i.e. it it possible to look ahead a lot longer for it to calculate more accurately?). I have a s [13:36] <JEEB> in mpeg-ts there are continuity bits IIRC [13:36] <JEEB> 0-15 was it? [13:36] <JEEB> and if it doesn't go +1 then you've lost a packet [13:37] <JEEB> (an mpeg-ts packet) [13:37] <veleno> JEEB: that's useful indeed. i'm checking http://en.wikipedia.org/wiki/MPEG_transport_stream#Packet [13:37] <JEEB> then you've got the 192 and 204 byte mpeg-ts packets that contain checksums [13:37] <JEEB> I have no idea what libavformat supports and how [13:41] <veleno> and i guess there's a way to see which are the stream transport protocols available [13:41] <durandal_1707> there is some log under debug: "Continuity check failed for pid %d expected %d got %d\n" [13:42] <veleno> durandal_1707: under the logs of ffmpeg or ffprobe ? [13:42] <durandal_1707> veleno: it is in lavf, so anything that use lavf and care for its logs will get it [13:43] <JEEB> what about the checksums for >188 byte mpeg-ts? [13:47] <durandal_1707> i see some check_crc but it its result is not loged, i'm not mpegts expert and mpegts in lavf can be improved for sure [13:50] <JEEB> yeah, lavf lacks support for PID switching, too [13:53] <veleno> if a transport other than mpegts is preferrable for my goals, i'd rather switch to it. open for suggestions. [13:57] <Mavrik> JEEB: PID switching? [14:03] <JEEB> Mavrik, a PMT says that one or all of the streams will switch the PID they will be transferred over [14:03] <Mavrik> ah, yes, that's not supported :) [14:03] <JEEB> oh I know it's not supported [14:04] <JEEB> which is why I use tvtest for Tokyo MX recordings :) [14:04] <JEEB> (Tokyo MX switches between PIDs when they switch between SD and HD) [14:04] <JEEB> NHK on the other hand keeps the video PID the same afaik [14:06] <veleno> JEEP: in the mpegts packets i don't see CRC fields . where did you see the infos about checksum transported by byte 192 and 204 ? [14:06] <veleno> JEEB: ^ [14:06] <Mavrik> veleno: there's an option of adding CRC fields to the 188 byte packets [14:07] <Mavrik> if your MPEG-TS packets are 188B then you don't have those :) [14:07] <JEEB> http://wiki.multimedia.cx/index.php?title=MPEG-2_Transport_Stream#Packet [14:08] <JEEB> "Packets are most often 188 bytes in length, but some transport streams consist of 204-byte packets which end in 16 bytes of Reed-Solomon error correction data." [15:32] <iive> isn't that part of the FEC? [16:15] <skarm> Hi to all. Is it possible to stream /dev/video0 to local network, without using ffserver? [16:24] <veleno> I guess there are CRC infos in the TS packets of the stream being analyzed http://codepad.org/GrvX6094 .. [16:24] <veleno> i wonder if when using ffmpeg to stream the video it's possible to add those [16:26] <durandal_1707> veleno: yes, modify mpegts muxer and send patch [16:40] <veleno> durandal_1707: ok [17:53] <espr3ss0> hilo [17:54] <espr3ss0> need some help, is anyone here ? [17:55] <Fjorgynn> duno [17:55] <stqn> Cant help, but Im here. Hi. [17:55] <Fjorgynn> where's the cake? [17:56] <espr3ss0> lol [17:57] <Fjorgynn> espr3ss0: shoot? [17:57] <espr3ss0> anyone know how to set the timescale ? [17:57] <Fjorgynn> what? [17:57] <espr3ss0> mp4 works with QuickTime but to iPad complains [17:57] <espr3ss0> iTunes even plays fine :-/ [17:59] <klaxa> espr3ss0: a bit more information about the problem would be great [17:59] <klaxa> i.e. information about the produced file and ffmpeg output logs [17:59] <espr3ss0> " [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180e800] stream 0, timescale not set " [17:59] <klaxa> paste everything on pastebin [18:00] <espr3ss0> not wanting to do that soz ;-p [18:01] <espr3ss0> does ffmpeg have a timescale parameter ? [18:21] <durandal_1707> espr3ss0: yes [18:54] <iamtakin1iteasy> hi, which codec (and/or settings) do you recommend for recording desktop? [18:55] <iamtakin1iteasy> from x11grab for instance [18:55] <iamtakin1iteasy> so text won't get blurred [18:55] <JEEBsv> ffvhuff is a fast and simple lossless encoder [18:55] <iamtakin1iteasy> after a couple of seconds [18:55] <iamtakin1iteasy> thank you [18:56] <Guest94> in FLV transcoding using ffmpeg+libx264 i could see CPU usage is around 115% [18:56] <Guest94> any suggestions? [18:56] <Guest94> i am using single thread option [18:57] <durandal_1707> use faster preset/more threads/faster hardware/ .... [18:57] <Guest94> here is the commandline http://pastebin.com/nJpGpt26 [18:58] <Guest94> <durandal_1707> do you think using faster presets CPU usage will go down? [18:59] <durandal_1707> try it [19:00] <Guest94> let me give a shot [19:02] <Guest94> i tried superfast/ultrafast and veryfast but nothing seems to take the cpu usage below 100% [19:02] <Guest94> its always going >108% [19:05] <durandal_1707> those preset and without extra options? [19:06] <Guest94> here is the command i am using now ffmpeg -y -i /root/fGFNmEOntFA.flv -vcodec libx264 -preset superfast -threads 1 /tmp/test.3g1_super.flv [19:09] <klaxa> there is a superfast preset? [19:09] <klaxa> wtf? [19:11] <iamtakin1iteasy> also, can ffmpeg stream over the network? [19:11] <iamtakin1iteasy> without medium of local file [19:11] <klaxa> yes [19:11] <iamtakin1iteasy> how it looks like? [19:12] <Guest94> <klaxa> Read the article http://dev.gentoo.org/~beandog/x264_preset_reference.html [19:12] <klaxa> write to udp or rtp, if you have an rtmp server, you can stream to that too [19:12] <klaxa> oh cool didn't know that it had that many presets [19:13] <Guest94> now if you can suggest something which is causing high CPU usage would be great [19:13] <iamtakin1iteasy> hmm [19:14] <klaxa> well too much computation needed for compression, also ffmpeg will use as much cpu as possible to decrease encoding time [19:14] <Guest94> right but this will reduce the throughput of the system when we do multiple transcode [19:15] <klaxa> you can increase the nice-value for ffmpeg [19:15] <klaxa> however, it will still use as much cpu time as it can get [19:16] <durandal_1707> and need [19:17] <Guest94> any other better thought? [19:17] <klaxa> no [19:18] <klaxa> if you have multiple encodes at the same time obviously all encodes will be slower than if they are run sequentially [19:18] <klaxa> if you increase nice values for selected ffmpeg instances, those should work slower than those instances with unchanged nice values [19:18] <Guest94> all the encode in parallel, just like independent thread [19:19] <klaxa> yes, that doesn't increase productivity of a system [19:19] <iamtakin1iteasy> i am trying to stream x11grab over network, but apparently doing something wrong [19:19] <iamtakin1iteasy> here is my commandline output: http://paste.eientei.org/show/39/ [19:19] <iamtakin1iteasy> any ideas where i got wrong? [19:19] <iamtakin1iteasy> might it be wrong USE-flags i built ffmpeg with? [19:20] <klaxa> iamtakin1iteasy: you have to specify a container-format [19:20] <klaxa> for example: -f matroska [19:20] <klaxa> i *think* that should work, i'm not sure though [19:20] <klaxa> also, note that you will have rather high latency [19:20] <klaxa> and make sure you have high network throughput [19:21] <iamtakin1iteasy> i have added -f matroska to options list and now i am getting udp://192.168.1.4:6000: No such file or directory [19:21] <iamtakin1iteasy> 192.168.1.4 is the ip of current machine [19:21] <iamtakin1iteasy> should it be some remote listening server? [19:21] <klaxa> --disable-network [19:21] <klaxa> heh [19:22] <iamtakin1iteasy> ah [19:22] <iamtakin1iteasy> i see [19:22] <klaxa> that will be it [19:22] <iamtakin1iteasy> thank you [19:22] <klaxa> you could get it working with pipes and netcat though :V [19:22] <iamtakin1iteasy> nah, i'll just re-emerge the ffmpeg with proper use-flag [19:23] <iamtakin1iteasy> thanks for noticing [19:23] <klaxa> i can't say how reliable udp streaming is, i don't think i ever got it to work really [19:23] <iamtakin1iteasy> what about rtp servers? [19:23] <iamtakin1iteasy> is there any opensourced of them? [19:24] <iamtakin1iteasy> so i can setup it on my server with higher bandwitch [19:24] <iamtakin1iteasy> and stream to other users [19:24] <klaxa> dunno i haven't really found many good streaming solutions that are open-source [19:24] <klaxa> i think live555 is rather popular though [19:24] <Mavrik> red5? [19:25] <klaxa> oh yeah that sounds like what we are looking for [19:25] <klaxa> reading that it's implemented in java makes me think it must be bloated :V [19:26] <iamtakin1iteasy> i am java coder. i am getting paid for work with bloatware >:3 [19:26] <iamtakin1iteasy> (shouldn't be so proud of it) [19:27] <iamtakin1iteasy> but meh, there is no way i'll be scared by any kind of java monstrocity [19:27] <iamtakin1iteasy> thank you [19:57] <espr3ss0> ok ... had a look for a command to set the timescale but no luck :-( [19:58] <espr3ss0> and everybody has gone :-/ [19:59] <durandal_1707> timescale what? [19:59] <espr3ss0> an mp4 ? o_o [20:00] <durandal_1707> can you explain in finer detail [20:00] <espr3ss0> warning, " timescale not set " [20:00] <espr3ss0> i assume it's why i can't copy it to mi' iPad [20:01] Action: espr3ss0 shrugs [20:01] <espr3ss0> why need the whole info ? [20:01] <espr3ss0> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 2027 kb/s, 23.98 fps, 23.98 tbr, 96k tbn, 47.95 tbc [20:02] <durandal_1707> i cant help if i cant reproduce - is this so hard to understand? [20:02] <espr3ss0> oh, no :-) [20:02] <espr3ss0> how can u re-produce when u don't have the file ? [20:03] <durandal_1707> so instead of providing requested info you ask questions? [20:03] <klaxa> espr3ss0: simple: you can't, but you can try to encode a file with the same output parameters [20:04] <klaxa> that should produce similar encodes [20:04] <espr3ss0> ahh, i c [20:04] <klaxa> also, what durandal_1707 said, if you don't want to provide information, we can't help [20:04] <klaxa> "I cannot get my car to start, I think the battery might be empty, also it's read, how do i fix it?" [20:04] <espr3ss0> fair enuff ;-p [20:04] <klaxa> *red [20:05] <espr3ss0> i will post some info [20:05] <durandal_1707> klaxa: call 911 [20:07] <espr3ss0> http://pastebin.com/tdJ5tquf [20:11] <klaxa> looks like it's something with the input file rather than with ffmpeg [20:13] <espr3ss0> what's wrong with the input file ? [20:14] <klaxa> timescale is not set :V [20:14] <klaxa> no idea how to fix it myself [20:15] <klaxa> except maybe re-encode but you want to avoid that if possible [20:15] <durandal_1707> time scale is set from demuxer [20:15] <durandal_1707> or muxer [20:15] <durandal_1707> so perhaps png stream messed up something [20:16] <espr3ss0> tried from the original file, extract (copy) to an MKV then to MP4, which usually solves just about everything so far [20:16] <espr3ss0> oh, that's without completely re-encoding [20:16] <durandal_1707> perhaps it copy png "stream" too? [20:17] <durandal_1707> because timescale is set for video [20:17] <durandal_1707> and audio [20:17] <durandal_1707> it is just not set for cover png [20:19] <espr3ss0> stream 0.1 ? [20:19] <espr3ss0> unless that's the "Artwork" (meta), then it shouldn't be there :-/ [20:20] <klaxa> heh now that i see it... [20:20] <espr3ss0> listed in Subler is only Vid & Aud [20:20] <klaxa> 1080p video but 64 kbps 22050hz aac [20:20] <espr3ss0> ? o_O [20:20] <klaxa> the audio track is extremly low quality [20:21] <durandal_1707> espr3ss0: stream 0.2 is png [20:21] <klaxa> it's funny that one shouldn't get copied even [20:22] <espr3ss0> why not ? [20:22] <klaxa> but try: -map 0:0 instead of -an [20:23] <durandal_1707> anyway if you could make smaller sample (eg it happens when partially copied into mkv) that you could report it on trac and upload sample [20:24] <Pitfall> Hi folks [20:24] <Pitfall> I have a question about licensing [20:25] <Pitfall> I want to use FFMPEG in my project which is closed source [20:25] <Pitfall> Is that allowed? [20:25] <JEEBsv> see the LGPL license [20:26] <Pitfall> Is there a way to distinguish code between GPL and LGPL? [20:26] <Pitfall> I heard that it had some GPL too in it [20:26] <JEEBsv> yes, but if you do not do --enable-gpl ffmpeg will be LGPL [20:26] <JEEBsv> it will only compile code that is LGPL [20:26] <JEEBsv> (or LGPL-compatible) [20:26] <Pitfall> OK [20:27] <JEEBsv> you can then separately get a license from x264 LLC if you want to use x264 in a closed source app [20:27] <JEEBsv> because it otherwise is GPL [20:27] <Pitfall> We already have a x264 license [20:29] <JEEBsv> nice [20:29] <JEEBsv> anyways, just keep your ffmpeg configuration as LGPL, and you should be able to use it under the LGPL license :) [20:29] <Pitfall> OK thanks [20:29] <Pitfall> I do not want to be blamed for my actions :) [20:31] <JEEBsv> just keep away from enabling GPL or nonfree [20:31] <Pitfall> What if a library actually is GPL? [20:31] <JEEBsv> then you can't use it [20:31] <Pitfall> You cannot purchase a license for that? [20:32] <JEEBsv> no unless the developers give such out [20:32] <JEEBsv> you can always ask the developers [20:32] <JEEBsv> also, a quick go-over for how to be "OK" with LGPL [20:32] <durandal_1707> hmm, wasn't you can use gpl but using dynamic lib and not modifying code? [20:33] <JEEBsv> that's /very/ iffy [20:33] <JEEBsv> I'd rather keep away from that [20:34] <JEEBsv> LGPL: if you do dynamic linking, just the exact source code of the LGPL component(s) used in the binary [20:34] <JEEBsv> if you do static linking, the exact source code of the LGPL component(s) as well as the object files for the proprietary part [20:34] <JEEBsv> so that the user can switch around the librarie(s) if he/she wants [20:34] <JEEBsv> that is LGPL [20:34] <JEEBsv> more or less [20:35] <durandal_1707> Pitfall: read https://ffmpeg.org/legal.html [20:36] <Pitfall> I did read that, but it is pretty vague [20:36] <durandal_1707> than ask lawyer [20:37] <espr3ss0> just checked with a different file ... muxing (mp4) only a video (x264) track and audio (aac) shows as fine and i didn't realise adding meta artwork adds an extra track. but then it shows the timescale problem [20:38] <durandal_1707> it is definitely bug so report it anyway [20:38] <Pitfall> I guess I will first check if I can use FFMPEG in my project compiled as LGPL [21:08] <iamtakin1iteasy> hi, i hava a low-bandwitch netbook which screen i wish to stream over http protocol and i have a higher-bandwitch server which i want to do actual streaming to end-users. how do i best transfer the stream from netbook to server and restream for each user then? [21:08] <iamtakin1iteasy> i have tried various rtsp servers, but they don't seem to be capabel of http streaming [21:09] <iamtakin1iteasy> so you can't simply use mplayer on http stream they are [not] providing [21:09] <iamtakin1iteasy> maybe i missed something? [22:02] <Pinhole> How can I determine which extra libraries are actually being used (not compiled for)? I have two boxes with the same stream and the same ffmpeg command line. The resulting output files are very different in size. I'm trying to figure out why. [22:03] <Pinhole> Both are Fedora 17 with the same version of ffmpeg installed. One is 32 bit the other is 64 bit. The 64 bit seems right, the 32 bit is very bad. [22:03] <JEEBsv> what is being used should be visible from the terminal output, pastebin the commands and terminal output onto a pastebin provider of your choice, and then link them here [22:08] <Pinhole> Ok, I'll paste these and be back in about 40 minutes (they are in progress) [23:06] <fenduru> is there a way to use -f segment to segment audio and pipe the segments out to stdout? [23:08] <klaxa> use pipe: as an output file? [23:08] <klaxa> that will write to stdout [23:43] <fenduru> klaxa, I think I figured it out. I was overlooking the fact that aac needs a seekable output [23:43] <klaxa> kk [23:48] <humans63> !lista [23:48] <humans63> ciao [00:00] --- Tue Feb 12 2013

1 0

ffmpeg-devel.log.20130210
by burek 11 Feb '13

11 Feb '13

[00:44] <Zeranoe> Slightly off topic here but anyone know why patches are created with -up or "--git" instead of just running a standard diff command with no options? Don't the patches work the same? Is it just for readability purposes? [00:46] <beastd> Zeranoe: --git is what you will get from git naturally [00:48] <Zeranoe> beastd: Do the patches include the context around the modified area just for readability sake? [00:49] <beastd> Zeranoe: not only [00:50] <beastd> Zeranoe: it is also for finding the right spot to apply the patch (probably with "fuzz") [01:02] <J_Darnley> Surely patches are made that way because people ask git to make them rather than running diff on the files themselves... [01:50] <Compn> zera [01:50] <Compn> sfakgj [01:50] <Compn> J_Darnley : patches are made with git send email because you can apply these patches from your email client in one click / keystroke [01:50] <Compn> one or two anyways [02:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:13ae02d03fbb: imgconvert: add get_pix_fmt_score() [02:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:cf8d9b74f306: imgconvert: use get_pix_fmt_score in avcodec_find_best_pix_fmt_of_2() [05:34] <Darkarnium> Afternoon [07:05] <Darkarnium> Bit of a strange query regarding AVOptions. I patched one of the demuxers and added a few AVOptions that can be passed as CLI arguments. This works without issue under ffplay, all options are passed and the relevant calls therein are performed with the options specified via the flags. However, if I instead use the ffmpeg binary, the flags seem to be ignored and the demuxer never uses the CLI arguments. [07:05] <Darkarnium> Is there any major difference between the handling of options between these two applications? [07:06] <Darkarnium> I'm sure it must just be something I've missed. However, I've also taken the newly compiled binary and executed it directly - in case I was having issues with ffplay and ffmpeg being different versions, etc. [07:08] <Darkarnium> Any suggestions, documentation or mailing list threads would be appreciated - if I'm missing something here, which is most probable. [07:18] <Skyler_> what's your ffmpeg commandline? [07:57] <Compn> Darkarnium : wheres your patch ? [07:58] <Compn> Darkarnium : did you specify the options in the right order? it matters in ffmpeg for some reason [07:58] <Compn> eg ffmpeg -demuxeroptions filename is different than ffmpeg filename -options [08:02] <Compn> anyone rememeber what i was supposed to add to ffmpeg tracker ? [08:04] <Compn> Darkarnium : saste found some strange bugs with -user-agent and other options in ffmpeg. maybe there is bug... [08:04] <Darkarnium> I'll have a look at the order then, see how far I get :) [08:04] <Darkarnium> The patch isn;t something I'd be submitting to be honest, as it's for an internal application :) [08:05] <Darkarnium> I can't see there being another user who is doing something this broken to be honest :P [08:06] <Compn> someone remind me if bayer patch didnt get applied ... [08:06] <Compn> oh i think it still crashes [08:07] <Compn> Darkarnium : i meant we can check what you did [08:07] <Compn> to see if you did it wrong [08:07] <Compn> but you might want to ask saste [08:08] <Compn> a lot of option code got moved around, apparently [08:10] Action: Compn gone, night [08:11] Action: Compn pings pross about bayer decoder [08:11] <Compn> now night [10:53] <cone-23> ffmpeg.git 03Carl Eugen Hoyos 07master:f02f231e05e0: Do not write invalid matroska files. [11:12] <saste> may we remove the rfps spam from the log? [12:02] <durandal_1707> why ffplay outputs only yuv420p ? [12:02] <nevcairiel> because of SDL, more or less [12:06] <durandal_1707> and it still cant output 444 or rgb? [12:08] <durandal_1707> sdl 2 version i mean [12:18] <cone-23> ffmpeg.git 03Diego Biurrun 07master:8658e1c68253: mss4, ra288: Remove unused DSPContext local codec context members [12:18] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:36fab50e90d1: asfdec: silence a warning [12:18] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:0af4506e15a6: Merge commit '36fab50e90d15352e403e4cc210890810f2fb4e2' [12:18] <saste> durandal_1707, should we add a test for il? [12:19] <saste> durandal_1707, in fact SDL1.2 supports a few more formats, check lavd/sdl [12:21] <durandal_1707> saste: that are just packed variants [12:21] <durandal_1707> saste: feel free to add test for il,lut,negate, ... [12:38] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:61f9ad2dfcb3: asfdec: read the full Metadata Object, not just aspect ratio information [12:38] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:3600049dae9e: Merge commit '61f9ad2dfcb3f98b7ac5777d19d0e7b61d0be01e' [12:45] <saste> durandal_1707, currently il segfaults on rgb48 [12:45] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:84b721db366c: asfdec: also read Metadata Library Object [12:45] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:30085bd43e35: Merge commit '84b721db366c0734fdfd23c8daaa7da7da21f761' [12:45] <saste> so yes, we should add a test for it [12:49] <durandal_1707> saste: not when i tested rgb48 [12:49] <durandal_1707> saste: don't you have patch for it? [12:50] <saste> durandal_1707, it doesn't segfault with my patch [12:52] <durandal_1707> saste: it does not segfault here [12:52] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:db0a943266be: avplay: apply the stream sample_aspect_ratio to decoded video frames [12:52] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d80820655362: Merge commit 'db0a943266be29ff0596872ebb418dfed75d00de' [12:53] <saste> durandal_1707, -vf format=bgr48be,il=l=d:c=d,showinfo,format=bgr48be [12:53] <saste> anyway i don't mind, I'm going to visually test with my patch on [13:17] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:f5fac6f77752: asfdec: support reading ID3v2 tags in ASF files [13:17] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:9a714d871ac5: Merge commit 'f5fac6f77752931347ab302563802dcaa49c2419' [13:21] <saste> durandal_1707, can you explain this: ffplay -f lavfi color -vf format=monow,il=l=d:c=d,format=monow [13:21] <saste> check the first frame [13:21] <durandal_1707> saste: about your noise source, how you expected to use it with other videos? [13:22] <saste> with my patch it looks ok [13:22] <cone-23> ffmpeg.git 03Anton Khirnov 07master:1ef0e8a6bf89: asfdec: do not assume every AVStream has a corresponding ASFStream [13:22] <cone-23> ffmpeg.git 03Anton Khirnov 07master:81726a4f0b8a: FATE: add tests for additional flavors of asf cover art [13:22] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:e88f392733fe: Merge commit '81726a4f0b8a43e19898e2a36fdde80583bafff0' [13:22] <durandal_1707> saste: if it does not make sense, filter out bitstream formats in query_.. [13:22] <durandal_1707> saste: i said your patch is ok [13:23] <saste> durandal_1707, yes but i can't understand what's wrong [13:23] <durandal_1707> i think i just made it bug to bug compatible with mp=il [13:25] <saste> durandal_1707, noise source? which one? [13:25] <saste> the old and slow patch? [13:26] <durandal_1707> there was only one noise source afaik [13:28] <durandal_1707> saste: ffplay have strange buffer handling [13:28] <durandal_1707> use ffmpeg, ffplay hide several bugs i had with il [13:53] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d6d5ef5534d5: sws: GBRP9, GBRP10, and GBRP16 output support [13:53] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:42431a557e2d: Merge commit 'd6d5ef5534d582f9dbaf18ac2605cf5bb72cd821' [13:55] <saste> durandal_1707, i don't think it is a good idea to have separate modes for il [13:55] <saste> it was a (bad) idea ported from mplayer [13:55] <saste> for example what's about alpha mode and packed? [13:55] <saste> why should I want to set a different interleaving mode for the various components? [13:56] <saste> why do i need to set it up to make it work with a given format (and i have to account for the pixel format planar/packed layout)? [13:59] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:e58013dd8f00: sws: use planarRgbToRgbWrapper only for 8bit per component [13:59] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d1d9b93e9a21: Merge commit 'e58013dd8f00e17dd98ba64e41c72f1d152f6608' [14:03] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:63a99622876f: lavfi/il: simplify/generalize linesize computation [14:03] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:0b1896543446: lavfi/il: move query_formats() definition up [14:16] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:7b41c24c5fa2: sws: dont enable chrSrcHSubSample for planar RGB [14:16] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f2b89f21cc3d: Merge commit '7b41c24c5fa221b55a9302efaf1a9eeb38b12551' [14:27] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:813784a8bf90: sws: disable yuv2rgb warning for planar rgb. [14:27] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:02b3ac87f81a: Merge remote-tracking branch 'qatar/master' [14:34] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:2c5e4ae29180: Revert "Merge commit 'db0a943266be29ff0596872ebb418dfed75d00de'" [14:45] <cone-23> ffmpeg.git 03Carl Eugen Hoyos 07master:034a125c8c51: Support more atrac3-in-mkv samples. [15:18] <durandal_1707> so what is status of vf_psnr? [16:05] <durandal_1707> one cant generate pink/brown/.. noise with aevalsrc? [16:41] <durandal_1707> burek: why ubuntu packages are that old? [16:50] <creep> h [16:55] <wm4> durandal_1707: maybe because ffmpeg APIs change so often, and they can't get all packages to work with newer ffmpeg... that's generally the reason why ffmpeg packages are old [16:55] <burek> durandal_1707 ? [16:56] <nevcairiel> ubuntu doesnt have ffmpeg packages, it only has "fake" ones that actually are libav [16:56] <wm4> it probably has old libav packages then :) [16:58] <nevcairiel> someone should come up with a popular linux tool that only works with ffmpeg and not libav, and see what distributions do 0p [16:58] <durandal_1707> burek: i talk about one in link provided from ffloger [16:59] <durandal_1707> burek: also there could be static build for linux [17:04] <burek> nevcairiel, ffmpeg is already a popular tool ^^ [17:04] <burek> durandal_1707, can you provide more references? [17:04] <burek> so i could know what are you talking about [17:06] <burek> btw, searching for enhancements on trac and trying to implement the most easy/quick ones would make ffmpeg have more popular functionality than libav (which it already has though), but it would allow users to choose ffmpeg, because "it can do something that libav cant" or something [17:08] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:6462268e74fa: pcx: fix rounding in bytes_per_scanline check [17:13] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:e005697af685: tests: add fate-lavfi-il test [17:13] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:60950adc18fe: lavd/v4l2: implement consistent error handling [17:13] <durandal_1707> burek: read #ffmpeg [17:15] <durandal_1707> saste, michaelni, ubitux what is status of lavfi scritpting and dynamic/conditional filtergraph? [17:15] <burek> oh you mean Jon Severinsson's FFmpeg PPA [17:15] <burek> well, ask Jon Severinsson... [17:15] <burek> I just linked his ppa [17:15] <burek> because we don't have anything "ours" [17:15] <saste> durandal_1707, still [17:15] <burek> I mean, probably the guy is also here, so just ask him, i guess [17:16] <saste> durandal_1707, what features do you exactly need? [17:17] <saste> a lua filter is just part of it, in general we may need to add more events to the already implemented filters [17:17] <saste> for example to change position to an overlay (or disable it) [17:18] <durandal_1707> saste: crossfading for example, one apply fade out and mix/blend it with fade in of another input [17:18] <durandal_1707> no caching of any kind is needed [17:20] <saste> durandal_1707, concat+fade? [17:20] <saste> now the problem is that you don't know when a filter stream is going to end [17:21] <durandal_1707> concat is useless here .... [17:21] <saste> yes *cross*-fading i see [17:22] <durandal_1707> it already can be done if several steps - which is PITA [17:22] <durandal_1707> *in [17:33] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f41329c8aec0: pcx: use init_get_bits8() [17:33] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d24de4596c3f: pcx: Add missing padding to scanline buffer [17:35] <michaelni> saste, libavdevice/v4l2.c:688:20: warning: ret may be used uninitialized in this function [-Wuninitialized] [17:39] <saste> Assertion bytes >= needed_bytes failed at libavformat/matroskaenc.c:166 [17:39] <saste> now i wonder why this happens... [17:50] <saste> ^^ anyone has ideas? [17:51] <durandal_1707> saste: how it happens [17:54] <saste> durandal_1707, ffmpeg -i IN.flv -an -codec:v copy -y OUT.mkv [17:54] <saste> this is happening at the end, when closing the file [17:56] <saste> apparently in end_ebml_master() [17:59] <durandal_1707> i ported noise filter [18:00] <saste> durandal_1707, do you plan to port all mp filters? [18:01] <durandal_1707> saste: not just mp useful filters but all useful filters [18:01] <durandal_1707> ilpack for examples is hardly useful [18:01] <wm4> but, 2002! [18:02] <nevcairiel> 2002 called, they want their filters back [18:02] <wm4> I think vf_pullup and vf_unsharp are relatively useful (though still a league below modern filters) [18:03] <wm4> vf_dsize doesn't make sense, you can delete it [18:03] <durandal_1707> so every new year I need to rewrite each decoder/encoder/demuxer/muxer? [18:04] <nevcairiel> every 10 years is enough [18:05] <durandal_1707> the idea is to have addgrain like filter [18:06] <saste> unsharp is already in, but it has some differences with the original unsharp [18:07] <durandal_1707> like what? [18:07] <wm4> uh, so you're keeping the mp one? [18:08] <saste> durandal_1707, like I wrote a patch long time ago, i sent it and it was approved, but then I was not satisfied with the patch and is still there [18:08] <saste> i wanted to read the papers and understand more about the algo, but never found the time/motivation to do that [18:10] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:151067bbc220: dpx: Fix rounding in 10bit total_size calculation [18:10] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:cb85779d459c: dpx: include offset in the total_size calculation [18:14] <durandal_1707> hmm, why i dont get rgba in filter_frame if original video is rgba? [18:23] <durandal_1707> another double ones are kerndeint and tinterlace [18:23] <durandal_1707> mp=tinterlace should be removed [18:23] <saste> durandal_1707, if you can convince michaelni to drop it [18:24] <saste> kerndeint can be optimized, ubitux was working on that [18:24] <saste> it should be easy, but i leave it to someone else [18:24] <saste> tinterlace is slower than mp=tinterlace [18:25] <saste> although the absolute difference should be meaningless [18:25] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:ea1d8465e6ec: Fix division by 0 due to audio frame size [18:25] Action: wm4 wonders why you would optimize these filters [18:27] <wm4> porting vivtc instead would be _actually_ useful: http://code.google.com/p/vapoursynth/source/browse/#svn/trunk/src/filters/v… [18:27] <saste> wm4, patches are... you know what [18:28] <wm4> now I don't want to be too offensive, "patches welcome" has to be the most stupid excuse ever [18:30] <saste> wm4, why? [18:30] <saste> it's not an excuse, it the truth [18:30] <saste> i won't work on that, but i won't prevent you to do so [18:31] <saste> that's how floss works [18:32] <wm4> I was just suggesting that porting vivtc would be a better use of your time rather than _optimizing_ barely useful filters (which probably not even mencoder users ever use) [18:33] <J_Darnley> Don't port it! Write a filter that passes video through Vapoursynth. [18:33] <J_Darnley> This is just the kind of stuff that thread on the ML was talking about [18:33] <wm4> J_Darnley: that certainly would be the best, though connecting these two APIs could be cumbersome [18:35] <saste> J_Darnley, getting rid of mp is more fun [18:35] <wm4> mp shouldn't have been added in the first place [18:36] <saste> about usefulness, it's not like i *personally* need il or any other interlacing filters [18:36] <saste> wm4, tell it to the users which use it [18:36] <saste> also it was useful for testing/porting stuff [18:36] <saste> that was the main reason for inclusion imo [18:45] <saste> that said, you should add ticket to trac/upvote, so we developers know what people really want [19:04] <Compn> durandal_1707 : going to port libvo from mplayer to ffplay ? [19:04] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:478fc7f57bac: pictordec: fix cga palette index limit [19:07] <Compn> wm4 / J_Darnley : why not rally vapoursynth and avisynth together to tie into ffmpeg ? [19:07] <Compn> surely theres plenty of *synth devs to make a measley wrapper [19:09] <Compn> michael did mp wrapper in less than a day ... [19:10] <Compn> of course [19:10] <Compn> you could also bribe michael into doing it :) [19:12] <durandal_1707> wrappers are bad [19:12] <Compn> i dont mean wrapper [19:12] <Compn> maybe i do [19:12] <Compn> external wrapper [19:12] <durandal_1707> same [19:12] <Compn> whats better? native api ? [19:12] <Compn> or porting everything to ffmpeg ? [19:13] <Compn> (or are you talking about libvo ? ) [19:13] <durandal_1707> Compn: there is libavdevice which is actually libavformat [19:16] <Compn> what [19:17] <Compn> oh libavdevice handles hardware and such ? [19:17] <Compn> so what are you saying ? [19:17] <Compn> sorry, i'm kind of dumb sometimes :) [19:31] <durandal_1707> Compn: answer to question about porting libvo [19:31] <durandal_1707> it is bad idea to port it to ffplay [19:34] <saste> who can help with matroskaenc? [19:43] <michaelni> saste, what exactly ? [19:43] <michaelni> about what [19:49] <durandal_1707> saste: having hwaccel pix fmts in supported inputs/outputs of filter is nonsense [19:58] <durandal_1707> saste: why has_alpha have double !! in vf_il [19:58] <durandal_1707> it is not needed at all [19:58] <saste> durandal_1707, flags & ALFA is a mask [19:58] <saste> I need to convert to 1/0 [20:00] <saste> michaelni, what are this cuepoints in matroska? [20:01] <saste> i have a (broken) file duplicated timestamps [20:01] <saste> i suspect this is the problem [20:01] <saste> it is trying to put too many entries into the same cuepoint, and asserting [20:14] <nevcairiel> cuepoints are a seeking index [20:14] <nevcairiel> more or less [20:22] <saste> in mkv_write_cues() is is setting an expectedsize in start_ebml_master [20:22] <saste> which is not honored, and which leads to an assert failure [20:29] <saste> michaelni: in other words, the MAX_CUEPOINT_SIZE is not assuming duplicated timestamps in the same stream [20:30] <saste> next question is, how to deal with that? [20:31] <saste> I could assume unexpected size, and increase filesize [20:32] <saste> now i'm not even sure how matroska copes with duplicated timestamps [20:36] <michaelni> saste, assuming duplicate timestamps arent disallowed how would they be usefull in the cue list ? [20:36] <michaelni> if you have 2 timestamps of 123 and you want to seek to 123 ... [20:37] <saste> michaelni, so what do you suggest to do? [20:37] <saste> the best thing would be to avoid to write duplicated cue points (for the same stream) [20:37] <saste> so we don't hit the bytesize limitation [20:38] <saste> do you agree? [20:41] <saste> michaelni, ^^ [20:45] <saste> durandal_1707, i can exclude the hwaccell from the il list, even if this is not going to do any difference [20:45] <michaelni> saste, yes, i agree dont wrote duplicate cue points [20:47] <durandal_1707> saste: doing it at higher level would be even better, even if lavfi will never use such formats [20:48] <durandal_1707> the noise filter makes it possible to select noise generation parameters per plane [20:48] <saste> durandal_1707, allright, direct commit or do you want to see the patch? [20:48] <durandal_1707> adding opts for all planes is pita [20:48] <saste> durandal_1707, in case of noise, i think it is useful for testing purposes [20:49] <saste> otoh the user should be able to specify with a single option the value for all the planes [20:49] <saste> and btw same for il [20:49] <durandal_1707> saste: it is usefull not only for testing but is there better way to handle this? [20:50] <durandal_1707> i start to hate opts tables. they clearly needs some improvement, like shortname or something.... [20:50] <durandal_1707> instead of duplicating each and every line [20:51] <durandal_1707> so it is 6*2*4 = 48 entries [20:51] <saste> durandal_1707, you can use macros [20:52] <durandal_1707> macros just hide issue [20:52] <saste> it's not like you need duplicated-similar options in most components [20:52] <saste> use them with caution [20:52] <durandal_1707> there should be type of list item [20:53] <durandal_1707> you set number of entries in list and size of each entry [20:54] <saste> for the constants? [20:54] <durandal_1707> no, for 0/1 and int [20:55] <durandal_1707> i could use AV_OPT_TYPE_BINARY [20:56] <durandal_1707> or actually not, but something similar [20:57] <durandal_1707> actually 0/1 items are just flags [20:58] <durandal_1707> and avopt don't care for this [20:58] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f28043d0a34a: tiff: check for failure in search_keyval() [20:58] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:0e3dacb11eac: tiff: dont leave geotag_count in an invalid state on errors. [20:58] <durandal_1707> or actually does ..... [21:00] <durandal_1707> oh so i get now 4*(1+1) = 8, i can handle this [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:f0703b6cba65: lavd/v4l2: fix error logic when enumerating standards [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:6b171f046c6f: lavf/matroskaenc: provide more feedback in case of unsupported codec [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:3acaea26e5b6: lavfi/il: filter out hwaccels formats from the list of supported formats [21:41] <michaelni> an3k / Guest9410, why do you constantly change nick back and forth ? [21:42] <saste> he can't decide [22:16] <Compn> michaelni : hes got nick protect on, and since hes not here to identify, chanserv changes his nickl [23:19] <durandal_1707> Compn: buy how do you explain periodic changes? [23:27] <Compn> durandal_1707 : irc client script auto changes nick or maybe bnc script etc [23:28] <Compn> didnt mean to do that, oops [23:28] <Compn> ehe [23:40] <durandal_1707> michaelni: fine to relicense noise to lgpl? [00:00] --- Mon Feb 11 2013

1 0

ffmpeg.log.20130210
by burek 11 Feb '13

11 Feb '13

[00:10] <norbert_> hi, latest 32 bit ffmpeg binary, how can I specify the location on the screen from where I want the width+height to be recorded? [00:10] <norbert_> used to be -i :0.0+x,y [00:10] <norbert_> that now says "Protocol not found" [00:12] <norbert_> I'm willing to bet that 99% of all ffmpeg lines that Google has indexed no longer work with the latest version [00:12] <norbert_> even the ffmpeg wiki itself is outdated [00:13] <sacarasc> Which site did you get the binary from? Which OS? [00:13] <norbert_> Mint GNU/Linux and http://ffmpeg.gusari.org/static/32bit/ [00:15] <fenduru> I'm trying to decode and reencode an audio file into segments. Right now I am running ffmpeg -i pipe: -vn -b:a 192k -acodec libvorbis -map 0 -f segment -segment_time 10 output%d.ogg [00:16] <norbert_> and it's also funny how 99% of all users run into libx264 being unknown [00:16] <norbert_> gotta love that [00:16] <norbert_> plus the location in the string of input and output parameters [00:16] <fenduru> however, when I decode these files and play them sequentially I get occasional pops/clicks between segments. Is there a way to force ffmpeg to split on the nearest frame boundary? (I've tried -segment_time_delta but this hasn' helped) [00:17] <Samus_Aran> will this command re-encode the audio: ffmpeg -i input.aac -acodec copy output.mp4 [00:18] <Samus_Aran> I wouldn't think so, but it says it is encoding the audio [00:18] <Samus_Aran> I am just trying to put my aac files into a container that will provide the duration of the audio (so MPlayer doesn't incorrectly guess the duration) [00:19] <JEEB> setting encoding code to copy should just mux [00:19] <JEEB> so as long as the log says that it only copies, it doesn't [00:20] <Samus_Aran> it says it is using an AAC encoder [00:20] <Samus_Aran> there is no mention of copying [00:20] <JEEB> pastebin your full command and log, and link here [00:22] <norbert_> JEEB: http://pastebin.com/MEuMpkkn [00:22] <norbert_> :) [00:23] <JEEB> unfortunately I have no idea about screen capture [00:25] <beastd> norbert_: does -f x11grab before -i give a different response? [00:25] <norbert_> well, I tried that, but it gives me: Unknown input format: 'x11grab' [00:25] <norbert_> so I decided to just remove that x11grab thing and hoped that it would magically fix the problem [00:25] <norbert_> now I'm reading http://www.jedi.be/blog/2010/08/30/capturing-the-screen-of-your-virtual-mac… [00:26] <sacarasc> That sounds like they didn't compile with x11grab support. :o [00:26] <norbert_> which is complicated and goes on about what I could do and stuff [00:26] <beastd> norbert_: ffmpeg -formats | grep x11 [00:26] <norbert_> yeah, the guy didn't add --enable-x11grab [00:27] <norbert_> so back to the ancient ffmpeg on my system, I guess, no recent binary this time [00:27] <jure> so, to speed up the tempo of audio, without changing pitch, you would do something like "ffmpeg -i input.mkv -filter:a 'atempo=2.0' -vn output.mkv". how do you change the speed of the audio and get the change in pitch as well? [00:27] <Samus_Aran> JEEB: here's my command and the output: http://pastie.org/pastes/6106599/text [00:27] <norbert_> and then I run into: Unknown encoder 'libx264' [00:27] <norbert_> myeah... sigh [00:28] <beastd> norbert_: this is not ffmpeg's fault. seems you binaries miss a few popular features :( [00:28] <Samus_Aran> it shows libvo_aacenc and Lavf53.21.0 [00:28] <beastd> norbert_: which system are you on? [00:28] <norbert_> beastd: well, the binary doesn't have --enable-x11grab and my system (Mint GNU/Linux) has an ffmpeg without libx264 in its ffmpeg [00:29] <norbert_> which means I need to pick another codec or recompile the thing myself [00:29] <Samus_Aran> norbert_: Ubuntu for instance has repositories with more up-to-date ffmpeg releases [00:29] <JEEB> Samus_Aran, you seem to be using a ffmpeg binary from the libav project, but that is still weird... try with the 'avconv' binary that is more up-to-date in that project [00:29] <Samus_Aran> also more feature-enabled [00:30] <JEEB> avconv -i input -c:a copy out.mp4 [00:30] <norbert_> Samus_Aran: do you know what would be a good codec to pick for desktop recording? out of these: http://pastebin.com/kZFzxP84 [00:31] <norbert_> something that gives me good quality I mean, without grinding my system to a halt [00:31] <norbert_> I can reencode it later if necessary [00:31] <JEEB> ffvhuff is a nice quick video encoder [00:31] <JEEB> it's big, but it's lossless and supports various colorspaces [00:32] <JEEB> also quick to encode [00:32] <Samus_Aran> JEEB: with that command it becomes: Stream #0:0 -> #0:0 (copy) [00:32] <JEEB> ok [00:32] Action: juanmabc guesses game recording [00:32] <JEEB> ubuntu by default uses libav, so #libav would be more correct for that [00:33] <JEEB> basically they later removed the ffmpeg binary from their side and avconv is their up-to-date binary [00:33] <JEEB> but it seems to work now, so that's great :) [00:33] <norbert_> [matroska @ 0x8aa5020] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1 >= 1 [00:33] <norbert_> av_interleaved_write_frame(): Invalid argument [00:33] <norbert_> is what it says when I try: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 25 -s 1280x720 -i :0.0+1920,26 -acodec pcm_s16le -vcodec ffvhuff -crf 0 output.mkv [00:33] <Samus_Aran> norbert_: I would suggest using a dedicated screen capture app, then converting the video it produces. is this OpenGL or simple 2D apps? [00:33] <JEEB> norbert_, might as well use avi for those codecs, try it out :) [00:34] <norbert_> there is no avi or libavi [00:34] <Samus_Aran> JEEB: so you can confirm that the ffmpeg command and output I pasted is in fact re-encoding my audio files? [00:34] <norbert_> it seems [00:35] <JEEB> norbert_, no -- I mean set output file name to dot-avi [00:35] <JEEB> that should switch the container to avi [00:35] <JEEB> Samus_Aran, that's how it reads like :P [00:35] <norbert_> JEEB: hey thanks, now it records [00:37] <Samus_Aran> JEEB: thanks. need to copy over my audio again, then, as I erased several files after I thought I losslessly copied them. oops. [00:38] <Samus_Aran> blah @ re-encoding then re-encoding then re-encoding audio... [00:41] <norbert_> hey thanks again, gotta go [00:45] <Samus_Aran> JEEB: thanks again for your help. [01:52] <suddendad> hello [01:53] <suddendad> is it possible to stream a huffyuv encoded videostream over network? [02:02] <videoman> Can't fade in a png without it appearing as a black square, seemed to work in earlier versions http://pastebin.com/g9PUKXfp [02:20] <videoman> alpha is =1 as in fade=out:80:30:alpha=1 [11:05] <Ertes> Hi guys, I have a mkv with 2 different audio track, both AC3, 48,0 KHz, 16Bit, 6 Channel, but different Kbps...the first is 384, the second is 448 [11:06] <Ertes> My TV can read the second track, but not the first ( Audio not supported ) so i think the problem is the Kbps, ho to change the Kbps from 384 to 448? Thanks [11:10] <Ertes> I have try with this commnad --- ffmpeg -i input.mkv -vcodec copy -acodec copy -ab 448k output.mkv --- but the audio is always 384 after the complete process... [11:25] <Ertes> i have a warning for the channel layout, how to set up for the 6 channel? [13:19] <t4nk461> Hi, when running the ffmpeg configure script with --enable-libfdk-aac, the check of it fails --> /usr/bin/ld: error: cannot find -lfdk-aac ... undefined reference to 'aacEncOpen' ... But libfdk-aac.a is installed under /usr/local/lib [13:19] <Fjorgynn> and you have compiled libfdk-aac? [13:19] <t4nk461> yes [13:19] <Mavrik> t4nk461, did you run ldconfig after installing fdk_aac? [13:20] <JEEB> that shouldn't be needed for a static lib (.a) [13:20] <JEEB> only for .so (shared) [13:20] <Mavrik> ah yes [13:20] <t4nk461> as described under: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#FFmpeg [13:20] <JEEB> t4nk461, does your distribution by default look into /usr/local for headers/libs? [13:21] <t4nk461> how can I check this? [13:23] <JEEB> -print-search-paths for libraries it seems? And just -v when compiling a random c file for header search paths [13:34] <t4nk461> when compiling with -v LIBRARY_PATH does not contain /usr/local. How to set it up? I added LD_LIBRARY_PATH=/usr/local/lib to my .bashrc? [13:35] <JEEB> --extra-ldflags="-L/usr/local/lib" [13:35] <JEEB> when configuring [13:35] <JEEB> LD_LIBRARY_PATH is used for shared libraries when loading them at runtime, so that is unrelated [13:36] <t4nk461> But where do I set it permanently? [13:37] <JEEB> you really can't [13:37] <JEEB> unless you want to recompile your toolchain :P [13:37] <JEEB> you'll just have to get used to the fact that you will have to add -L/usr/local/lib to the linking stages of things [13:38] <t4nk461> ok, thanks [16:23] <Seablade> Real quick question, getting unknown input format concat event hough it is listed int he available filters with the command... ffmpeg -f concat -i temp.txt msg.mp4 [16:23] <Seablade> Any ideas? [16:26] <Seablade> The entire output is what I posted except the ffmpeg version number, which I have tried a HEAD compiled from a few months back and 0.8.5, built Feb5 [16:27] <Seablade> If there is nothing obvious from my command setup that is all I need to know, only have five minutes or so to get this done sadly [16:27] <durandal_1707> cant help yuo until i see full uncut output [16:28] <Seablade> durandal_1707, I repeat what I jsut meantioned but since you insist...http://pastebin.com/D4gdLHCc [16:29] <durandal_1707> Seablade: you are using wrong and old version [16:29] <Seablade> durandal_1707, Thus why I mentioned I have tried with a HEAD fversion from a few months back as well, this is on two different computers [16:30] <Seablade> durandal_1707, What I need to know, is is there anything wrong with the command, or should it work assuming my version is up to date [16:30] <durandal_1707> Seablade: what HEAD version? [16:30] <Seablade> durandal_1707, I compiled SVN head from a few months back, that computer is in a different room right now I can't get you the exact revision [16:30] <durandal_1707> no you are using ffmpeg from Libav [16:31] <durandal_1707> SVN head is old and not maitained/updated at all [16:31] <durandal_1707> you should use git head [16:31] <Seablade> durandal_1707, Ok good to know thanks, my question does still remain, is there anything wrong with the basic command structure? [16:31] <durandal_1707> command should work with latest ffmpeg from FFmpeg project [16:31] <Seablade> durandal_1707, Thank you [16:32] <Seablade> durandal_1707, That is what I needed to know [16:32] <durandal_1707> Seablade: you get list of supported formats with: ffmpeg -formats [16:33] <Seablade> durandal_1707, Ahh thank you that is what I was messing up [16:34] <Seablade> Yea this Ubuntu package deperecation is ebing annoying on this machine, my OS X machine apparently was from the deprecated version, trying to download and build hnow [16:35] <Seablade> durandal_1707, Yep installing that ppa right now in fact;) [16:37] <Seablade> durandal_1707, And that ppa doesn't seem to have an up to date version with concat format, yea;) [16:37] <Seablade> durandal_1707, At least not for precise [16:39] <durandal_1707> Seablade: that is 0.10.6 version? [16:39] <durandal_1707> concat is in 1.1 [16:39] <Seablade> durandal_1707, Yep [16:39] <Seablade> durandal_1707, And yep:) [16:39] <Seablade> durandal_1707, Gotta love it [16:40] <Seablade> durandal_1707, Ok well my time limit passed, was trying to fix something for a live event on the fly, thanks very much for the help [16:40] <Seablade> durandal_1707, Ill fight through setting up the build environment for something up to date later [16:49] <creep> h [17:37] <creep> somebody know where to get gtk+-2.0-dev? it should be here i think... ftp://ftp.gtk.org/pub/ [17:52] <creep> somebody know of a good qr code reader/generator for linux (opensource) ? [19:24] <karlox> hi all [19:25] <karlox> wich is the status of rtmp support in ffmpeg? [19:32] <durandal_1707> karlox: there is native and wrapper one [21:13] <njhbnjhbn> hi all! need help [21:14] <njhbnjhbn> i want to join a still image and an audio file to a video file [21:15] <creep> how do i make 1 image from webcam with ffmpeg and exit ? [21:15] <njhbnjhbn> i used the following command: ffmpeg -loop 1 -f image2 -r 1 -i dragon2.jpg -shortest -i a.mp3 -acodec copy -vcodec libx264 out.mp4 [21:15] <njhbnjhbn> but the -shortest option is ignored, it loops forever [21:15] <njhbnjhbn> the video keeps growing and growing until i cancel [21:27] <njhbnjhbn> i fixed it by using an older version from 2012 [21:42] <saste> creep: ffmpeg -f v4l2 -t 3 -i /dev/video0 -updatefirst out.png or something like that [21:42] <saste> you should keep the device open for a few seconds, so that it can adjust brightness/contrast [00:00] --- Mon Feb 11 2013

1 0

ffmpeg-devel.log.20130209
by burek 10 Feb '13

10 Feb '13

[00:31] Action: Compn wonders why someone is encoding to qtrle in 720p [00:36] <kierank> why not [00:39] <Compn> whats qtrle good for ? seems like an old codec with poor compression ? [00:39] <Compn> maybe i dont know how good it is [00:40] <Compn> but most codecs from the year 2004 arent that great for hd. [00:40] <Compn> probably earlier, but going by copyright year :P [00:42] <kierank> it has alpha channel [00:42] <kierank> that's why its useful [01:02] <llogan> saste: any changes to "sort demuxers" other than moving the text? [01:19] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:a92816c4eb9d: get_se_golomb: fix decoding of very large values [04:06] <cone-778> ffmpeg.git 03James Almer 07master:53b2083f4433: sparc: Revert some parameter name changes in dsputil_vis.c [04:13] <cone-778> ffmpeg.git 03rogerdpack 07master:aa715cdcf5c6: dshow: attempt to build shared as well as static [04:13] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:a35f5c22339c: Merge remote-tracking branch 'rdp/dshow_shared' [05:31] <llogan> haven't seen a gcc seg fault in a while... [05:31] <llogan> i mean until now [06:17] <Compn> llogan : not an ICE then ? [11:26] <durandal_1707> michaelni: the il filter have option to (de)interleave chroma and luma separately [11:26] <durandal_1707> and alpha is completely forgotten [11:26] <durandal_1707> so i will fix this by making it more generic [11:27] <durandal_1707> keepeing old syntax though does not make sense [11:49] <cone-273> ffmpeg.git 03Diego Biurrun 07master:0b016eb99d38: dsputil: Move ff_block_permute to mpegvideo_enc [11:49] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:55151aa1013b: Merge commit '0b016eb99d38738e2c53e36549a4732a0f863b2e' [12:10] <cone-273> ffmpeg.git 03Diego Biurrun 07master:b5b7b75e91e2: dsputil: Move get_penalty_factor() to the only place it is used. [12:10] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:bcb15e66f7c4: Merge commit 'b5b7b75e91e291c810ebd01868a42b53f267a87c' [12:42] <cone-273> ffmpeg.git 03Mans Rullgard 07master:ed62e6e3c3ca: mjpegdec: use put_pixels instead of copy_block8 [12:42] <cone-273> ffmpeg.git 03Mans Rullgard 07master:b9ee5f2cab3f: indeo3: replace use of copy_block4 with put_pixels [12:42] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:e5212354eadd: Merge commit 'b9ee5f2cab3ffe1c962e542346b1ed61394864ec' [12:45] <durandal_1707> michaelni: do not push hidden stuff into merge, like this snowenc change [12:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:29a92c011480: histogram filter [13:13] <michaelni> durandal_1707, the code doesnt compile without that change [13:14] <durandal_1707> michaelni: yes, but hidden commits are worse [13:15] <michaelni> my commit messages suck i know, i should have explained it in the merge commit [13:15] <michaelni> its on my todo to see if they should be merged or [13:15] <michaelni> if we can gain something from seperate functions [13:17] <durandal_1707> michaelni: just commit it before merge [13:17] <durandal_1707> quatar removed snow so it should not matter [13:20] <durandal_1707> michaelni: why #1280 picks gray? [13:22] <michaelni> dunno [13:22] <michaelni> i could look into it later [13:22] <michaelni> libavfilter/vf_histogram.c:208:13: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] [13:24] <durandal_1707> when we will finally upgrade from ISO C90? [13:24] <burek> ..A.L. ilbc iLBC (Internet Low Bitrate Codec) [13:24] <burek> it doesnt say if it is encoder/decoder [13:25] <JEEB> not better than opus [13:25] <durandal_1707> burek: because you do not have either [13:25] <burek> if i do not have, why is it listed? [13:26] <burek> it* [13:26] <durandal_1707> others are listed tooo [13:26] <burek> even worse [13:26] <burek> why [13:26] <durandal_1707> to notify users of it existence [13:26] <burek> i see [13:26] <durandal_1707> you should still be able to demux files [13:27] <durandal_1707> and you would know what codec is it about [13:27] <burek> well, then maybe they should be sorted/groupped somehow at the end/beginning, to make it clear they are not available [13:27] <durandal_1707> burek: there is no native decoder, instead you enable wrapper... [13:27] <burek> yes, i understand that [13:27] <burek> using external library [13:28] <michaelni> durandal_1707, C90, when all compilers we support support that [13:28] <durandal_1707> michaelni: translation: never [13:30] <cone-273> ffmpeg.git 03Paul B Mahol 07master:e5670cb8a1c9: lavfi/histogram: make it C90 compliant [13:30] <iive> imho, declarations in any location in the code is misfeature. it solves a problem caused by another misfeature [13:31] <durandal_1707> iive: not really.... [13:37] <iive> i'm waiting for your explanation. [13:42] <someone-noone> Hello! I'm trying to pass "magic cookie"(extradata) data to mac os Core Audio framework for decoding AAC format. File is in m4a format. For this I'm using extradata from my AVCodecContext. But extradata_size is 7, however if I read "magic cookie" using OS X API, it's size is 44 for same file. Looks like magic cookie != extradata in ffmpeg (but should be). Any help how can I extract this data from ffmpeg? [13:44] <cone-273> ffmpeg.git 03Diego Biurrun 07master:6a4430407404: dsputil: Move ff_h264_idct function declarations to a separate header [13:44] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:0af07495eccc: Merge commit '6a4430407404bfe55581e66a86a496c9adf6fb44' [13:51] <cone-273> ffmpeg.git 03Diego Biurrun 07master:75d5156ac152: dsputil: Move ff_svq3 function declarations to a separate header [13:51] <cone-273> ffmpeg.git 03Diego Biurrun 07master:35685a3c2a1e: dsputil: Move ff_shrink* function declarations to separate header [13:51] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:ffbe19fc84fe: Merge commit '35685a3c2a1ec09f3c62dcfc4368fe9e92bcddf6' [13:52] <burek> does anyone have any idea what to use (in ffmpeg) to produce a low latency speech audio (for use in p2p audio chat)? [13:53] <burek> whatever i try, it doesnt work, except aac/mpegts (over udp) which has a significant delay :( [13:54] <someone-noone> burek, first of all you need low latency codec. The most talkable codec is OPUS for now. [13:56] <burek> well i tried opus, like: ... -acodec opus -f mpegts udp://localhost:5555 [13:56] <burek> but ffplay wont play it back [13:57] <nevcairiel> i dont think there is a defined way to store opus in mpegts [13:57] <someone-noone> mpegts doesn't support opus [13:57] <JEEB> did the "ogg container" have it? [13:57] <burek> no [13:57] <JEEB> then there's that matroska mapping that differs from the one that mkvmerge implemented [13:57] <JEEB> which might or might not have been removed from ffmpeg [13:57] <burek> well is there a standard way to stream voice/speech with ffpmeg (preferably over udp) [13:58] <nevcairiel> ogg is about the only thing that properly supports opus today [13:58] <JEEB> yeah [13:58] <nevcairiel> short of raw opus files, which may just be ogg, which i never checked =p [13:58] <JEEB> :D [13:58] <JEEB> I tried have him find 'opus' from the output of -formats [13:58] <JEEB> and there was none there [13:59] <wm4> is proper opus in mkv defined yet? [13:59] <JEEB> nope [13:59] <nevcairiel> no [13:59] <wm4> why? [13:59] <JEEB> so both mkvmerge and ffmpeg IIRC now have conflicting experimental ways of putting it there [13:59] <wm4> lol [13:59] <burek> well i might use speex i guess, i just need the hint for the output format used with udp [13:59] <nevcairiel> opus is problematic because it requires pre-roll data to start decoding [13:59] <JEEB> ^ [13:59] <cone-273> ffmpeg.git 03Diego Biurrun 07master:b9c2408b19dc: dsputil: Remove commented-out, unused function declarations [13:59] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:a8864615e6cc: Merge commit 'b9c2408b19dc9408e9dc0ce4a099af529bbfba3b' [14:00] <nevcairiel> and how to handle this has caused the difficulty defining the proper standard in mkv [14:00] <JEEB> and matroska, unlike mov/mp4, doesn't have that concept in the container until now [14:00] <JEEB> and they want to "define it properly" [14:00] <JEEB> somehow I feel like I'll be liking xiph next-gen container if matroska just doesn't start evolving >_> [14:01] <burek> is it possible there is no "standard" or preferred way in ffmpeg (so far) to have it send low latency audio to another peer? [14:01] <burek> i mean, it looks so advanced with everything [14:01] <nevcairiel> ffmpeg is mostly a transcoding application, streaming is not its main goal [14:01] <burek> i see [14:02] <burek> well, ok, thanks.. at least i know where not to look then i guess.. [14:10] <someone-noone> Okay, I found difference between magic cookie and extradata for AAC. OS X meaning of magic cookie is esds ATOM in mp4 file. However ffmpeg stores only part (in the middle, only 7 bytes) of it in extradata and it looks like enough for ffmpeg to decode content in AAC. Anyway how can, probably ffmpeg stores somewhere(while decoding) this atom in AVCodecContext structure ? [14:13] <Paranoialmaniac> someone-noone: libavcodec doesn't store atom [14:13] <Paranoialmaniac> atom is mp4/mov thing [14:14] <Paranoialmaniac> aac specific info is stored in extradata as the same format between mp4 and mov [14:15] <someone-noone> Paranoialmaniac, I see. But is there a way I can mux this data into my container that is a part of libavformat, that I could read it in another application that doesn't use libav*? [14:15] <nevcairiel> all the important data is stored in extradata, you just have to put all the atom data back around it [14:15] <nevcairiel> if you need it to be an atom [14:17] <Paranoialmaniac> someone-noone: i have no idea about libavformat muxing :P sorry [14:17] <someone-noone> I have an idead. Look at m4a format& It's somehow playing by OS X :) [14:21] <someone-noone> hm, looks like I can't find m4a format in libavformat folder. Doesn't it name should be mpegaudio or m4a ? [14:21] <nevcairiel> m4a is just a normal mp4 file with a different extension [14:21] <nevcairiel> apple just uses it for audio files, but its the same format [14:21] <Paranoialmaniac> mov.c [14:21] <someone-noone> didn't know that& thanks [14:22] <Paranoialmaniac> mp4 != m4a, m4v. pure mp4 requires object descriptor and bifs for presentation :P [14:24] <nevcairiel> its the same damn format, may just have different atoms somewhere =p [14:24] <Paranoialmaniac> well, yes. mp4, m4a and m4v is derived from iso base media file format [14:25] <someone-noone> well, I found mov_write_esds_tag that is doing some really crazy things :( [14:25] <Paranoialmaniac> esds atom includes not only mpeg-4 audio specific info but also bitrate info [14:26] <Paranoialmaniac> esds atom consists of mpeg-4 system structure [14:26] <someone-noone> I need smth like uint8_t *esds_atom = get_atom(); OSXOpenCodec(esds_atom+4) [14:26] <someone-noone> Really, there is no explanation in Apple's docs :( [14:27] <cone-273> ffmpeg.git 03Diego Biurrun 07master:bf6b3ec924b4: dsputil: Move rnd_avg inline functions to a separate header [14:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:81a11452f8e0: Merge commit 'bf6b3ec924b4fb64d14df33077f4d4541d525dbf' [14:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:5fd6d85d17d5: rnd_avg: fix author attribution [14:32] <cone-273> ffmpeg.git 03Diego Biurrun 07master:059866eb17cd: dsputil: Move WRAPPER8_16_SQ macro to the only place it is used [14:32] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:276c9c0577e9: Merge commit '059866eb17cd3c66422a61d3e9c3305f55830a50' [14:40] <cone-273> ffmpeg.git 03Diego Biurrun 07master:218aefce4472: dsputil: Move LOCAL_ALIGNED macros to libavutil [14:40] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:6b2e65078c02: Merge commit '218aefce4472dc02ee3f12830a9a894bf7916da9' [14:44] <durandal_1707> iive: for (int i ....) [14:45] <durandal_1707> michaelni: i get strange internal.h macro E redefintions warnings [14:46] <durandal_1707> #define E(x) x ???? [14:47] <nevcairiel> its for macro expansion [14:48] <nevcairiel> some compilers need an extra expansion to properly expand all parameters [14:48] <durandal_1707> but there is already E defined in bunch of other places for opt stuff [14:49] <durandal_1707> In file included from libavcodec/options.c:35: [14:49] <durandal_1707> libavcodec/options_table.h:39:9: warning: 'E' macro redefined [14:49] <durandal_1707> #define E AV_OPT_FLAG_ENCODING_PARAM [14:49] <durandal_1707> ^ [14:49] <durandal_1707> ./libavutil/internal.h:63:9: note: previous definition is here [14:49] <durandal_1707> #define E(x) x [14:49] <durandal_1707> ^ [14:49] <durandal_1707> libavcodec/options.c:111:5: warning: 'av_opt_set_defaults2' is deprecated [-Wdeprecated-declarations] [14:49] <durandal_1707> av_opt_set_defaults2(s, flags, flags); [14:49] <durandal_1707> ^ [14:49] <durandal_1707> 2 warnings generated. [14:49] <nevcairiel> apparently libav doesnt have those other E defines =p [14:50] <durandal_1707> nevcairiel: i doubt it [14:50] <nevcairiel> or they dont include that header there [14:50] <durandal_1707> this is the new thing recently happening [14:52] <durandal_1707> nevcairiel: and this is introduced when moving local aligned stuff [14:52] <durandal_1707> fairly recently [14:52] <nevcairiel> i saw the commit that caused it [14:53] <durandal_1707> good. i will just undefine macro [14:53] <nevcairiel> that macro is needed [14:54] <cone-273> ffmpeg.git 03Diego Biurrun 07master:5d3d39c72e57: dsputil: Move fdct function declarations to dct.h [14:54] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:5cfc0ae825e6: Merge remote-tracking branch 'qatar/master' [14:54] <durandal_1707> but if you did not yet notice this breaks when using own E in opt stuff before using LOCAL.... [14:55] <nevcairiel> maybe just rename E to EXPAND or something [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:111697170e14: doc/demuxers: sort demuxers by name [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:9b9b5a475b99: doc/demuxers: fix copy&paste error in "@c man end" line [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:765dbea9fe1f: lavf/rawvideodec: fix/extend option descriptions [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:bad82243cd83: doc/demuxers: add docs for rawvideo demuxer [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:b46a8f055ee5: doc/filters: end table in histogram docs [15:00] <durandal_1707> beastd: so native resamples is fast as sox one? [15:01] <beastd> hi [15:01] <durandal_1707> welcome on board [15:02] <beastd> durandal_1707: i did not measure the speed of native vs sox resampling. i just found that new thread on HA somewhat misleading. should check if something was added [15:03] <cone-273> ffmpeg.git 03Carl Eugen Hoyos 07master:7a259a5fcdc3: Fix libpostproc compilation on x86_32 with disable-optimizations. [15:04] <durandal_1707> beastd: what thread, there is trollolo old few days [15:06] <beastd> i understood from michaelni that you talked about ffmpeg resampling on HA. Or did you mean the old thread? I found this one http://www.hydrogenaudio.org/forums/index.php?showtopic=99286 [15:06] <durandal_1707> i mean whatever thread that was recently active [15:07] <durandal_1707> yes that thread ..... [15:10] <beastd> durandal_1707: did you compare speed of native vs libsoxr resampling? [15:10] <durandal_1707> beastd: no, i'm busy with removing mp [15:12] <durandal_1707> and negate filter is really broken on big endian? [15:12] <durandal_1707> why is there no fate coverage? [15:13] <durandal_1707> saste: ping [15:15] <saste> durandal_1707, what? [15:16] <durandal_1707> saste: i ported il filter, did you mind if old syntax is not keept? [15:16] <durandal_1707> it does not make sense to keep it [15:16] <saste> durandal_1707, how hard would be to keep the old syntax? [15:16] <saste> lemme check [15:17] <durandal_1707> il filter does not copy alpha plane at all [15:17] <saste> about lut/negate, I cannot test [15:29] <cone-273> ffmpeg.git 03Paul B Mahol 07master:da726a8763d3: paf, sanm: remove unused header after b05830169d [15:30] <saste_> michaelni, do you want to review the il filter? [15:30] <saste_> also tell if you're ok with GPL -> LGPL [15:31] <durandal_1707> saste: read logs, he already approved change [15:31] <durandal_1707> it is few memcpy calls after all [15:31] <saste> durandal_1707, ok [15:31] <michaelni> saste, review for() memcpy ? no ;) [15:31] <durandal_1707> :)) [15:32] <saste> that's not only that [15:32] <saste> i'm going to review it, leave me some time [15:32] <durandal_1707> i said, i need to fix 4 and 2 plane situation [15:33] <durandal_1707> this means keeping old syntax is not possible at all [15:33] <saste> durandal_1707, i don't mind about that [15:35] <saste> durandal_1707, is mp=il broken or what? [15:36] <durandal_1707> saste: not it is working fine [15:36] <durandal_1707> saste: mp=il=i,mp=il=d [15:36] <saste> ffplay IN -vf mp=il=d [15:37] <saste> i got chroma artefacts [15:37] <durandal_1707> saste: mp=il=d:d [15:37] <saste> otoh i confess i don't know what the filter really does [15:37] <saste> yeah [15:42] <saste> durandal_1707, what's the plan with regards to mp=fil? [15:42] <durandal_1707> remove [15:43] <durandal_1707> it is just il that works correctly only when width match linesize [15:43] <saste> i mean it is possible to port such a filter (which according to docs, "does not always work" in unspecified circumstances)) [15:43] <durandal_1707> and fast when works [15:43] <wm4> haha [15:43] <wm4> saste: yeah that freaked me out when I saw it [15:44] <wm4> then I removed it from my mplayer fork [15:44] <durandal_1707> wm4: because you do not know what you are removing .... [15:45] <wm4> well, have fun rewriting it [15:45] <durandal_1707> saste: i could add option to il to use fil way when it will work ... [15:45] <saste> durandal_1707, seems a good idea [15:53] <durandal_1707> wm4: now one cant filter interlaced video with mpv and display it on tv [15:54] <wm4> durandal_1707: how would one achieve this with "fil"? [15:55] <durandal_1707> calling fil=d, fitlers, fil=i (note code is from 2002) [15:57] <wm4> filtering interlaced material and then playing it on a interlaced display...? [15:58] <durandal_1707> yes [15:59] <wm4> well, I guess there are quite some 2002 use cases lost by removing some of these filters, so if you care about these... [16:00] <wm4> I think extended filtering should be left to tools which are suitable for these tasks, such as avisynth (and hopefully vsynth) [16:06] <ramiro> michaelni: I don't remember having commit rights to the git repo. I don't think I ever sent my public key (even if I did, I've changed it a few times in the past two years) [16:22] <michaelni> ramiro, theres a key for Ramiro Polla on the server since Jan 19 2011 [17:03] <durandal_1707> saste: the nv21/nv12 works with il [17:25] <cone-273> ffmpeg.git 03Paul B Mahol 07master:4fa9defc7fde: lavfi/histogram: make waveform mode more useful [17:29] <cone-273> ffmpeg.git 03Paul B Mahol 07master:faac85005cad: lavu/internal: rename macro E(x) to EXPAND(x) [18:02] <ubitux> hey beastd [18:02] <ubitux> how is the subtitles thing going on? [18:02] <beastd> hi ubitux [18:02] <ubitux> you have 3 days left [18:02] <ubitux> ;) [18:03] <beastd> yeah, unfortunately it is not going so well. but that is not a reason to not enjoy your remaining 3 days on another planet ;) [18:03] <ubitux> alright, then i'm going to sleep [18:03] <ubitux> 'night all :) [18:03] <beastd> n8, see you [18:16] <cone-273> ffmpeg.git 03Carl Eugen Hoyos 07master:e149127bb217: Call the common 7.1 wide channel layout "7.1(wide)". [18:55] <durandal_1707> michaelni: what happened to dificulty tag for trac/ [19:03] <michaelni> durandal_1707, could be added but iam not sure its that usefull, open tickets with smaller numbers are harder [19:05] <durandal_1707> michaelni: not always, but yes ... [19:12] <durandal_1707> can lavfi open file and write to it and in another run another filter read file? [19:21] <durandal_1707> did performance dropped with dsputil split? [19:22] <nevcairiel> the functions just get moved to different files and different pointers, why would it get slower? [19:29] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:4401958fdc9a: motionpixels: Propagate errors in vlc table init [19:29] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:b5fc95e77f9e: motionpixels: Check that the vlc table has been fully inited [20:05] <durandal_1707> michaelni: what kind of data rv10 encodes? major/minor/micro/ version... [20:17] <michaelni> durandal_1707, dunno you would have to compare it with the decoder if theres no comment [20:18] <durandal_1707> michaelni: thing is rv10 encoders does not create extradata [20:18] <durandal_1707> and i stops here with invalid slice_count [20:18] <durandal_1707> and nothing sets avctx->slice_count [20:20] <durandal_1707> matroska expects extradata, and decoder too. the last one needs only last 8 bytes [20:22] <durandal_1707> actually i do same thing as rm muxer [20:24] <durandal_1707> which i think is hack because ignores extradata in case of -c copy i think [20:24] <durandal_1707> ^the rm muxer [20:35] <cone-273> ffmpeg.git 03Paul B Mahol 07master:350128b28fe1: rv10: remove check that is always true [20:39] <durandal_1707> the rm muxers adds some extra stuff to each video packet [20:49] <durandal_1707> i think that rv10/rv20 encoder & rm muxer are just wrong [20:50] <durandal_1707> eg muxer does encoder job [21:18] <cone-273> ffmpeg.git 03Nicolas George 07master:f43d09cd60a1: lavf: add tee pseudo-muxer. [21:20] <durandal_1707> michaelni: why ./ffmpeg -f lavfi -i testsrc -vf format=gray16 /tmp/out.bmp picks monob? [21:22] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:796012af6c78: targa: Fix y check in advance_line [21:53] <durandal_1707> saste: i fixed il to work with >8 bpp pix fmts [21:53] <durandal_1707> also mp one crash with pal8 [21:56] <durandal_1707> so i will apply it soon and remove mp once [21:57] <durandal_1707> *one [22:48] <durandal_1707> michaelni: fine to remove ilpack? [22:51] <michaelni> i could only state my oppinon, you still should probably ask on the ML [22:52] <durandal_1707> michaelni: i asked for you opinion? [22:52] <michaelni> i suspect its ok but iam not sure [22:53] <durandal_1707> it is some hybrid scaler and deinterlacer [22:53] <durandal_1707> *reinterlacer [22:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:6e9abce5e2d2: lavfi: port MP il filter [22:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:c0a8918f2af7: lavfi: remove mp=il filter [23:00] <cone-273> ffmpeg.git 03Paul B Mahol 07master:969039eb6e10: lavfi/il: fix typo in Copyright [00:00] --- Sun Feb 10 2013

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ffmpeg.log.20130209
by burek 10 Feb '13

10 Feb '13

[00:34] <stephanedev> hi everybody. i have issues when streaming a video file using rtp and playing it with ffplay. it works fine when the 2 hosts are on a LAN, but the video is very jerky and most frames are dropped when doing it over the internet [00:35] <stephanedev> the file contains a vp8 video stream 640*360, 25 fps, 500 kb/s [00:36] <stephanedev> ffplay is passed in the sdp file made of the ffmpeg output in the terminal [00:37] <stephanedev> command is: ffmpeg -re -i big_buck_bunny.webm -vcodec copy -an -f rtp rtp://<target_ip>:50004 [00:39] <stephanedev> in the ffplay output, there are a lot of messages like: [00:39] <stephanedev> [vp8 @ 026bc380] RTP: missed 3 packets [00:39] <stephanedev> [sdp @ 026a3e00] Missed unknown data, sequence broken [00:41] <stephanedev> any idea as to what may be wrong? am i missing some command-line option to ffplay? [00:42] <stephanedev> is it expected to receive the sdp error about a broken sequence? [00:49] <Mavrik_> stephanedev: well, obviously it's telling you that you're missing packets [00:49] <Mavrik_> either your bandwidth isn't large enough [00:49] <Mavrik_> or you're having ordering problems [00:59] <stephanedev> Mavrik_: thanks for the answer [01:00] <stephanedev> bandwidth shouldn't be an issue. downloading and video streaming using skype works just fine between the 2 endpoints [03:58] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [04:52] <st000> hello guys is anyone of you familiar with the problem that a from ffmpeg created (avi) file has the right aspect ratio in ffplay, but not in vlc or windows media player? [05:24] <Sam__> Guys i am doing simple flv transcoding using libx264+ffmpeg using single thread, when i see CPU usage using top i could see it cross 112% [05:24] <Sam__> and can see 4 instances of ffmpeg [05:25] <Sam__> 214358 root 20 0 300m 23m 4788 R 90.3 0.5 0:07.17 ffmpeg 214361 root 20 0 300m 23m 4788 S 5.0 0.5 0:00.35 ffmpeg 214359 root 20 0 300m 23m 4788 S 4.4 0.5 0:00.34 ffmpeg 214360 root 20 0 300m 23m 4788 S 4.4 0.5 0:00.35 ffmpeg [05:26] <Sam__> http://pastebin.com/TX5GqjWz [05:27] <st000> llogan: what do you mean with !pb ? [05:29] <llogan> st000: that is a shortcut to tell the #ffmpeg channel bot, fflogger, to tell you to, "please use a pastebin site..." [05:30] <Sam__> any suggestions please [05:30] <llogan> you could always... [05:31] <st000> llogan okay^^ [05:32] <st000> llogan: ehm here my pastbin http://pastebin.com/yHve1nzZ [05:35] <llogan> st000: not that it really matters, but your input is already 704 wide, so why add scale? [05:36] <Sam__> http://pastebin.com/fhqMPuRz [05:36] <Sam__> @ fflogger [05:36] <Sam__> @fflogger http://pastebin.com/fhqMPuRz [05:36] <elkng> anyone around ? [05:37] <llogan> fflogger is a bot, not a person [05:37] <st000> llogan: it was a try [05:37] <Sam__> and CPU usage in Pasebin http://webchat.freenode.net/ [05:37] <Sam__> http://pastebin.com/TX5GqjWz [05:38] <st000> even if i let it go it will not change anything [05:39] <llogan> what does vlc or WMP think the aspect ratio is? [05:39] <st000> 4/3 [05:39] <st000> or 1/1 [05:40] <st000> i think it is the last thing [05:40] <llogan> do you have a sample? [05:41] <st000> yes [05:41] <Sam__> @<fflogger> any suggestions? [05:42] <st000> it is crud it anything between 5:4 and 4:3 [05:45] <klaxa> Sam__: suggestion 1) update your ffmpeg 0.10 is pretty damn old [05:45] <klaxa> suggestion 2) add: -threads 1 [05:45] <Sam__> -threads 1 is already add in commandline [05:46] <klaxa> probably add it before -x264ops [05:46] <klaxa> *x264opts [05:46] <klaxa> also i'm not sure that's supported anymore [05:46] <klaxa> i could be wrong though [05:47] <Sam__> no effect adding it before x264opts as well [05:49] <klaxa> mmh dunno [05:49] <Sam__> @ <klaxa> I disagree with you here about options as these all are latest options for both [05:50] <Sam__> ffmpeg and libx264 [05:50] <klaxa> your ffmpeg version is 0.10 [05:50] <klaxa> that's, if i may say so, old as fuck [05:52] <Sam__> ok [05:54] <klaxa> grab the latest ffmpeg and take a look at this http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [06:02] <gudy> Hi, is it valid for a GOP to not start with an I frame, but with B frames which reference a future I frame. The I frame is first in the decode sequence, but not first in terms of presentation time ? [06:03] <gudy> (in h264 high profile) [06:12] <st000> llogan: do you want to have the sample ? [08:28] <jac1d> Hello all. I have a number of PCM streams coming from FM tuners to a single x86 CPU and I want to encode them to AC3 stereo in real time. What kind of CPU power would I need to do this? [09:42] <int> hello guys, I want to compress video file a bit without a big quality loss, but I receive the followint error: http://paste2.org/p/2841453 [09:42] <int> could somebody help me? [09:42] <int> maybe I should change codec? [09:45] <relaxed> int: ffmpeg -i avn1.mkv -c:a libmp3lame -b:a 128k -c:v libx264 -preset veryslow -crf 18 output.mkv [09:45] <sacarasc> Why are you using h263? [09:45] <int> sacarasc: I thought it is the best for compression [09:46] <int> relaxed: thank you very much [09:46] <sacarasc> H264 is what you wanted, using libx264. [09:46] <int> sacarasc: hm [09:46] <relaxed> The source is h264, so I doubt you'll gain much more compression. [09:46] <int> ok, I will try [09:47] <int> does redusing resoultion help me? [09:48] <sacarasc> Yes, it would. [09:53] <int> -s 1024x570? [10:17] <sacarasc> -vf scale=1024:570 I think is the proper way these days. [10:20] <int> ok, thank you [11:08] <creep> h [11:08] <creep> another nice nuclear meltdown in result of stupidity from japan, have a good appetite for fish from sea http://en.wikipedia.org/wiki/Lists_of_nuclear_disasters_and_radioactive_inc… [11:10] <elkng> creep: when was the latest ? in fukushima ? [11:10] <elkng> or there is new incident ? [11:12] <creep> elkng<< about a year ago, you know, it taskes some time to share this unimportatnt information :) [12:02] <sceriffo> hi , where can I find a code for generation extradata for AAC decoder ? [12:03] <creep> elkng<< soon, chinese will start selling radioactive flashlights, then they will be able to get rid of their radioactive waste legally, no recharge, 100 years of light, cheap, made in china... [12:13] <elkng> creep: "radioactive flashlights", its creepy [12:29] <iive> creep: funny. but not really accurate. [12:29] <iive> all flashlights are radioactive, because light is radioactive radiation :) [12:35] <Mavrik> hmm, does anyone know what "afterburner" parameter on Fraunhofer AAC encoder actually does? [12:36] <JEEB> seemingly optimizes more on quality than speed [12:36] <JEEB> what it *actually* does tho is something you'll have to ask wbs [12:38] <Mavrik> hmm [12:38] <Fjorgynn> seems like I [12:38] <Mavrik> basically I have a Motorola STB here which chokes as soon as I enable afterburner [12:39] <Mavrik> JEEB, wbs? [12:41] <JEEB> IIRC that's the IRC nickname for Martin Storsjö, who made fdk-aac from an Android thing to a library :P [12:41] <JEEB> and also who pushes fixes towards back to Android [12:41] <Mavrik> ah, ok :) [13:05] <DeLoreann> Hi! I have a problem with ffmpeg. While recording rtmp stream from ustream, quality decreases during recording. Anyone? (sorry for my english ;) [13:05] <DeLoreann> Here is a sample: http://bocik.pl/ss/player.php?v=njIdVI.flv [13:18] <burek> ive just set up ffmpeg as a p2p voice tool :) [13:18] <burek> btw, which audio codec is the most suitable for p2p voice chat [13:18] <burek> (the least delay, speech codec) [13:18] <JEEB> the one mumble uses, opus [13:21] <burek> hmh, it doesnt produce any errors [13:21] <burek> but sending it over udp with -f mpegts doesnt work not even in vlc [13:21] <burek> i mean, i cant hear anything [13:22] <JEEB> are you sure vlc is equipped with libopus decoding? are you sure its muxing into mpeg-ts is somehow supported by Various Things? [13:22] <burek> oh [13:22] <burek> how should i send it then? [13:22] <burek> i can use ffplay [13:23] <JEEB> to be honest I have no idea whatsoever, I think mumble just sends out a raw opus stream or whatever. [13:23] <JEEB> not sure if ogg has a set up mapping for opus, matroska's one is still in the works [13:25] <burek> ill try ogg too, thanks :) [13:25] <JEEB> as the container, that is [13:25] <JEEB> or just raw opus [13:25] Action: JEEB shrugs [13:26] <burek> how to specify -f raw [13:28] <DeLoreann> Mavrik: http://pastebin.com/SSjM5Gx9 [13:28] <JEEB> burek, see -formats for opus [13:28] <burek> thanks [13:29] <Mavrik> DeLoreann, your quality drops because you're re-encoding video to 200kb/s [13:29] <Mavrik> DeLoreann, use "-codec copy" as parameter [13:29] <Mavrik> to force only copy of stream, not reencode [13:30] <DeLoreann> "Unrecognized option 'codec'" [13:30] <Mavrik> DeLoreann, ah, you have an older version, it's "-vcodec copy" [13:31] <DeLoreann> [flv @ 0x808040020] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 100 >= 0 [13:31] <DeLoreann> av_interleaved_write_frame(): Invalid data found when processing input [13:32] <DeLoreann> output file: http://bocik.pl/ss/player.php?v=abfwcZ.flv [13:32] <DeLoreann> 51kB, (one frame?) [13:33] <Mavrik> hmm [13:33] <JEEB> DeLoreann, update your ffmpeg [13:33] <Mavrik> that shouldn't happen really... but you do have a rather old fmfpeg [13:33] <Mavrik> *ffmpeg [13:33] <JEEB> yes, it's /very/ old [13:34] <DeLoreann> im doing it on shell account, this will take time :) [13:34] <burek> JEEB, sorry for bugging you, but what exactly should i be looking in the -formats ? [13:34] <DeLoreann> anyway, thanks Mavrik! [13:34] <JEEB> it should have a raw format for opus, if there isn't... then there isn't :D [13:34] <burek> there isn't :) [13:34] <Mavrik> DeLoreann, grab static builds :) [13:35] <JEEB> Mavrik, he's on freebsd [13:35] <JEEB> lol [13:35] <Mavrik> oh. [13:35] <Mavrik> ah. [13:35] <Mavrik> that could be problematic then :D [13:35] <JEEB> burek, well then try -f ogg or something, can't really come up with anything better :D [13:36] <burek> ok, i'll try various test and see which one works then, thank you :) [13:42] <someone-noone> Hello! I'm trying to pass "magic cookie"(extradata) data to mac os Core Audio framework for decoding AAC format. File is in m4a format. For this I'm using extradata from my AVCodecContext. But extradata_size is 7, however if I read "magic cookie" using OS X API, it's size is 44 for same file. Looks like magic cookie != extradata in ffmpeg (but should be). Any help how can I extract this data from ffmpeg? [16:24] <elkng> anyone around [16:24] <elkng> ? [16:26] <elkng> that is the file http://sprunge.us/QQeM I'm trying to convert, that is the command http://sprunge.us/MAgd I use to convert it, that is the output http://sprunge.us/FHRa, it's length is 23:19 but it converts only first 4:57 and then stoped convert [16:26] <elkng> with errors, what is wrong ? [16:27] <elkng> and similar problems happened for last couple days about 4 times, it converts first few minutes and then stopped with errors [16:31] <durandal_1707> elkng: does number of channels really change to that big number? [16:31] <elkng> what number ? [16:31] <durandal_1707> in log, ich [16:32] <elkng> ffprobe says: "Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 192 kb/s" [16:32] <durandal_1707> [SWR @ 0x8a20640] Value 24.000000 for parameter 'ich' out of range [16:32] <durandal_1707> Unsupported number of input channels [16:32] <elkng> something wrong with video ? [16:33] <durandal_1707> maybe something get corrupted, does it play ok with other software? [16:35] <elkng> durandal_1707: no, seems like I need to redownload video, because mplayer crashes playing original mp4 file after 4 minutes [16:36] <durandal_1707> mplayer uses old lavf/lavc too [16:37] <durandal_1707> you need qt time or wmp that does not use lavf/lavc [21:12] <llogan> DeLoreann: did you figure it out. there is at least one other *BSD user here that may be able to provide assistance. [21:14] <DeLoreann> i managed to solve the problem by "-acodec copy -vcodec copy", here is sample: http://bocik.pl/ss/player.php?v=AiBeEC.flv [00:00] --- Sun Feb 10 2013

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ffmpeg-devel.log.20130208
by burek 09 Feb '13

09 Feb '13

[00:15] <saste> anybody going to review the opencl deshake patch? [00:21] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:0d194ee51ed4: vc1dec: Fix destination pointers in context for fields [00:25] <michaelni> saste, iam lazy, you can review it [00:25] <saste> michaelni, i'm busy [00:25] <saste> i'll try to review it anyway, in the weekend [00:26] <saste> also i know no s*#t about opencl [00:27] <michaelni> anyone has any odd platforms/hw that are missing from fate.ffmpeg.org ? ... and wiling to setup fate clients ? [00:28] <michaelni> we now also have a SH4 client since a few days (qemu based though) [00:29] <saste> michaelni, i have a ZX spectrum, might that be helpful? [00:30] <michaelni> can it run ffmpeg / fate ? [00:42] <michaelni> saste, you could optimize and port ffmpeg to ZX spectrum ;) [00:43] <michaelni> maybe a striped down version that does support just one container and codec or so [00:43] <saste> uhm in 128KiB it's going to be hard [00:44] <michaelni> uhm why ? [00:44] <michaelni> there was a 3d first person shooter in less than 64k IIRC [00:44] <michaelni> not on ZX spectrum though id guess ;) [00:51] <gnafu> If anyone could do it, it would be Fabrice Bellard. And so I would think a project he started could be transformed to do it. [00:51] <gnafu> ;D [01:13] <saste> well at least we could have a player on zx [01:13] <saste> after all we have an encoder for c64 [01:48] <ubitux> hey [01:49] <ubitux> michaelni: a00c4b4db seriously? oO [01:49] <michaelni> yes seriously [01:49] <michaelni> no joke [01:50] <ubitux> why does this happen? [01:50] <michaelni> no idea, gcc i guess [01:51] <michaelni> but + is nicer for gcc [01:51] <michaelni> it allows o2*=a2; o2+= ... [01:51] <michaelni> with - that style would not be possible for gcc to assemble to [01:51] <michaelni> but i didnt check if gcc actually does this [01:52] <ubitux> fun [01:53] <ubitux> kyowa, tokyo desu!~ [01:53] Action: ubitux & [02:12] <Compn> michaelni : wei was asking in the dxva2 mail "Also I want to ask a question that is the chroma_format always stored in the ext sequence header?" [02:12] <Compn> he has some other questions in that mail too [02:27] <michaelni> Compn, what mail (subj/date) ? [02:54] <michaelni> Compn, found the mail & replied [03:00] <Compn> oop, ok good :) [03:00] <Compn> thanks for looking at it [11:45] <cone-778> ffmpeg.git 03Paul B Mahol 07master:810cd0dd9c03: riff: chunks must be at 2 byte boundary [12:14] <durandal_1707> michaelni: the fil filters is buggy, it interleaves incorrectly [12:51] <michaelni> durandal_1707, libmpcodecs vf_fil ? [12:51] <michaelni> does the isssue happen just with ffmpeg or also mplayer ? [12:51] <michaelni> its a mplayer filter and not maintained in ffmpeg [12:52] <durandal_1707> michaelni: i dont have mplayer installed [12:52] <durandal_1707> i just did fil=i,fil and expect to see same output [12:55] <michaelni> fil was designed to be fast (0 copy) IIRC it can be wrong sometimes [12:55] <durandal_1707> i want to port it [12:56] <durandal_1707> so i want to know what is expected result [13:00] <michaelni> i think these artifacts are expected for stride != width [13:06] <durandal_1707> michaelni: so its not useful at all? [13:07] <michaelni> might be usefull if you have a slow box and need it for realtime playback ... [13:08] <michaelni> but the copy would not be that slow so its probably not that usefull [13:08] <durandal_1707> michaelni: so what about making port that would do copy and be morae usefull [13:09] <michaelni> yes if it cant do it without copy & without artofacts it should copy [13:10] <michaelni> maybe with a user override switch that allows forcing non copy [13:10] <michaelni> but that would have very few users [13:10] <michaelni> also see vf_il [13:24] <durandal_1707> there is no filter that blends to images? [13:24] <durandal_1707> *two [13:36] <Compn> durandal_1707 : blend how ? [13:37] <Compn> durandal_1707 : the stereo3d puts images side by side [13:37] <Compn> or pad or that other filter [13:40] <iive> or interlace, or one above the other, or codes them red/blue .. ;) [14:16] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:47335126c049: ff_end_tag: assert that the start is aligned. [14:16] <cone-778> ffmpeg.git 03James Almer 07master:89946c840375: dxva2: Prevent redefinition of _WIN32_WINNT [16:14] <durandal_1707> ubitux: when you will become live again? [16:14] <ubitux> around wednesday [16:15] <durandal_1707> ok, don't drink too much blood [16:20] <durandal_1707> michaelni: so fil is just "fast" variant of il? [16:28] <michaelni> yes [16:28] <durandal_1707> ok, why they exist at all? [16:30] <michaelni> il/fil allow for example frame based filters to be easily applied to fields [16:30] <michaelni> probably they have other uses too [16:31] <durandal_1707> hmm, so i will just port il - could i pick less retarded name? [16:32] <durandal_1707> and fil will be just special case of il [16:43] <durandal_1707> michaelni: hmm mp=il=d gives obviously broken output here [16:55] <durandal_1707> nvm, i deinterleave only luma plane ... [16:55] <durandal_1707> michaelni: are you ok to relicense this trivial code to LGPL? [18:14] <michaelni> durandal_1707, the vf_il.c (4289byte) code (the parts i wrote) can be used under LGPL, its just some memcpies anyway [23:52] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:b8820e7aa338: dsp: make put_no_rnd_pixels_tab 4x4 big [00:00] --- Sat Feb 9 2013

1 0

ffmpeg.log.20130208
by burek 09 Feb '13

09 Feb '13

[00:56] <stephanedev> saste: i just want to thank you again for your help. i am now setting "lag-in-frames" to zero for the vp8 encoder and latency is much lower. [00:56] <saste> stephanedev, of course you can't achieve the same performance with all encoders [00:57] <saste> libx264 -tune zerolatency is fu***ng brilliant [00:57] <saste> most other encoders may not be suited for that kind of use [00:57] <stephanedev> yeah i saw that option for x264 [00:58] <stephanedev> for now we settled on vp8 because we are not sure about x264 license [00:58] <JEEBsv> the software license or licensing of the H.264 format? [00:59] <JEEBsv> x264 at least is available for payware licensing if you don't want to touch GPL [00:59] <stephanedev> the format [00:59] <JEEBsv> yeah, that depends on how you're using it, don't remember where the MPEG-LA pdf about it was [00:59] <stephanedev> for ffmpeg we use lgpl [01:00] <JEEBsv> also IIRC MPEG-LA is rather open for talks, unlike VIA licensing f.ex. [01:00] <JEEBsv> so they should be rather easily contactable [01:01] <stephanedev> ok, thanks for the advice. we may contact them if vp8 does not do the job or customers want x264 [01:02] <stephanedev> by the way, i assume i need to build my own ffmpeg DLL with the proper options to remove everything that is not lgpl if this is the license i use, even if i don't use the glp bits? [01:02] <stephanedev> gpl* [01:03] <JEEBsv> yes [01:04] <JEEBsv> people have a tendency of checking the license you built/used your ffmpeg with [01:04] <stephanedev> thank you [01:04] <JEEBsv> and trolling you if you linked against a GPL ffmpeg [01:04] <JEEBsv> so yeah, building an LGPL library is a *good* idea [01:05] <stephanedev> it makes sense. we want to respect every third-party license, so i will build a lgpl version before we distribute our software [01:06] <JEEBsv> heh, I wish every developer was like you :P [01:07] <stephanedev> that's the least we can do, given how helpful some of these libraries are [01:08] <JEEBsv> people copying their ffmpeg confiuration from random GPL project X, and then subsequently using it with their non-open source app... [01:08] <JEEBsv> I wish that didn't happen as much as it does [01:10] <stephanedev> some developers don't bother to check the license text and just assume they can do anything with source code they find online [01:11] <JEEBsv> yup [01:42] <Keshl> stephanedev: Sorry for randomly going afk. Mind slipped. I swear I have early onset alzheimers. <É< http://pastebin.com/uN9RnWng [01:44] <stephanedev> hmm i usually get the fps on the video stream line, i.e. the one starting with Stream#0:0: Video [01:45] <stephanedev> let me try with a flv file on my system [01:49] <stephanedev> Keshl: i don't have fps either when probing a flv file. but it seems you can rely on the tbr: http://stackoverflow.com/questions/3199489/meaning-of-ffmpeg-output-tbc-tbn… [01:49] <Keshl> oÉo. [01:50] <Keshl> 25 tbs == 25 fps, or "about 25, maybe 24"? [01:53] <stephanedev> Keshl: i guess so, but check the last paragraph of the second message about tbr on: http://ffmpeg-users.933282.n4.nabble.com/What-does-the-output-of-ffmpeg-mea… [01:53] <Keshl> "I hope that helps a little.", oÉo? [01:53] <stephanedev> the one just before that :p [01:53] <Keshl> Ohhh oÉo [01:54] <Keshl> ... What kind of video is 25 or 12.5 fps? [01:54] <Keshl> o.O' [01:54] <stephanedev> i don't know what the field rate is, sorry [01:54] <Keshl> How does vlc know what rate to play it at? o_o' [01:54] <stephanedev> i am just a ffmpeg API user still struggling with some of the concepts involved :p [01:55] <stephanedev> it knows with the container rate and the codec rate [01:55] <Keshl> So how does I calculate that? [01:55] <stephanedev> the time bases in AVStream and AVCodecContext [01:55] <Keshl> o_o [01:56] <stephanedev> each packet is timestamped [01:57] <Keshl> <.< Can I get a dump of the first 60 timestamps [01:57] <Keshl> *? [01:57] <Keshl> =D [01:57] <stephanedev> the unit for the packet timestamp read from the file is in stream time base units (hope i am correct) [01:58] <stephanedev> not sure how to do it with ffmpeg command-line [01:58] Action: Keshl dies. [01:58] <Keshl> x.x [01:58] <stephanedev> but if you can program, the ffmpeg C API allows you to do that [01:59] <Keshl> o_o Anything beyond the standard library makes me wet myself. [01:59] <stephanedev> the demuxing.c example is a good starting point if you want to try [02:01] <stephanedev> you can easily modify demuxing.c (http://ffmpeg.org/doxygen/trunk/doc_2examples_2demuxing_8c-example.html) by adding some traces around the line calling avcodec_decode_video2 [02:02] <stephanedev> some printf's to show some of the members of pkt and frame (like pkt.pts, ...) [02:08] <stephanedev> Keshl: for ffmpeg command-line, "-debug_ts" may be useful [10:19] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [10:19] <portal`> would love for someone to chime in on why it might be. [10:19] <portal`> ffmpeg version N-49527-g985e93a [10:26] <portal`> running the same command usually will stop at around 8 seconds, but i just had it go to 48 seconds and stop there [14:07] <jeje34> hi to all [14:08] <jeje34> I have a problem using the last ffmpeg release (1.1.1) to decode h264 stream [14:10] <jeje34> when I want to enable multithreading to decompress, I always got an error when I open the codec (The maximum value for lowres supported by the decoder is 0) [14:10] <jeje34> But in the code of utils.c, this message appear when: if (avctx->codec->max_lowres < avctx->lowres || avctx->lowres < 0) [14:11] <jeje34> if I trace my code I have avctx->lowres=0 and avctx->codec->max_lowres=0 [14:11] <jeje34> I can't understand why I have this error [14:12] <jeje34> my codecctx have thread_count=3 (I have a dualcore) and I set thread_type = FF_THREAD_FRAME|FF_THREAD_SLICE [14:12] <jeje34> Thanks for your responses [14:15] <jeje34> for informations, I compile FFMPEG with Mingw with --enable-w32threads because I use ffmpeg libs in a windows application (using Visual Studio vc++ 2003) [14:29] <jeje34> the error code resulting from avcodec_open2 is -22 [14:52] <iive> jeje34: i don't think you are supposed to change max_lowres. [14:53] <iive> imho, if you don't use it, don't touch lowres too. [14:53] <iive> you may want to print the context after you get the error and see the value of lowres. if you think this is what caused the error. [14:54] <jeje34> I don't modify the value of max_lowres and lowres. When I look at the value, the two values are set to 0 (I think the default values) [14:54] <iive> aha. good... [14:55] <jeje34> I just call avcodec_alloc_context3 to get my context and so avcodec_get_context_defaults to set the default values [14:55] <iive> another possibility is that something got wrong with some of the other threads. [14:55] <iive> they have their own contexts, that are cloned from the main one... [14:55] <iive> most likely some tricky memory corruption. [14:57] <iive> just something for you to check. win32 doesn't have memalign, so ffmpeg have special memory allocation, that allocates more memory and then uses aligned memory into that allocation. [14:57] <jeje34> but what I don"t understand is why before calling avcodec_open2(codecCtx, codec, NULL) I have lowres=0 of my codecCtx and max_lowres=0 in my codec but I have the error message [14:57] <iive> be sure that memory allocated by ffmpeg function is free'd by ffmpeg functions and vice versa. [14:58] <iive> jeje34: put some debug printf's or breakpoints into the code that prints the error, and check the values there. [14:58] <jeje34> I use the ffmpeg calling to allocate and free memory [14:59] <iive> good. [15:00] <jeje34> I try to activate multithreading because I have a dualcore. When I decompress one stream, my CPU usage is near 1-2%. When I decompress 2 streams, I have 3-4%. But when I decompress 3 streams, i tkae me 25% of CPU usage. [15:00] <jeje34> Do you think activate Multithreading could decrease my cpu usage? [15:01] <iive> i didn't quite get that. [15:03] <jeje34> I'm going to rebuild FFMPEG with more traces I think [15:10] <iive> i would have recommended running it under valgrind, but I'm not sure if it is available under windows. [15:12] <JEEBsv> nope [15:12] <JEEBsv> there's a crap port, but that doesn't really work :S [15:14] <jeje34> I already have a look trying using GDB tools to debug FMMPEG under VisualStudio but I don't find naything working [15:14] <jeje34> opps anything sorry [15:15] <jeje34> Is someone have an idea how to configure the AVContext to use the lowest CPU usage when decoding H264 video streaming? [15:16] <jeje34> or perhaps in all the configurationof compilinf FFMPEG too [15:16] <jeje34> there's too much options... [15:22] <jeje34> my configure option are: ./configure --prefix=/mingw/i686-pc-mingw32 --enable-shared --disable-static [15:22] <jeje34> --enable-w32threads --enable-runtime-cpudetect --enable-memalign-hack --enable- [15:22] <jeje34> hwaccels --cpu=i686 --enable-debug=3 (I had enable-debug to try to see more reason) [15:33] <jeje34> for the CodecContext, I use error_concealment= FF_EC_GUESS_MVS | FF_EC_DEBLOCK, err_recognition = AV_EF_CAREFUL, debug = FF_DEBUG_THREADS [15:55] <flexnsniff> So i'm trying to stream RTMP to AMS5& streaming a movie to it. It works when i use the live/ app, but then is a green screen when i use the livepkgr/ app. Should i set ffmpeg differently or change how AMS works? [15:57] <jeje34> I add some trace in Utils.c of libavcodec, and so on, I have avctx->codec->max_lowres=0, avctx->lowres=3 [15:57] <jeje34> very strange [15:57] <jeje34> but it explain why I have this message when using avcodec_open2 [16:08] <Damik> Hi, I've problem with compiling ffmpeg under msvc toolchain. I've got - c99wrap cl is unable to create an executable file. [17:16] <mpfundstein> is here someone who has experience with sar/dar stuff? [17:16] <JEEBsv> in what way? [17:17] <mpfundstein> i have a video which is 720x480 (par 32/27, dar 16/9). all my calculations lead me to an actual width of 853x480 but quicktime reports 872x480 , which also works well in my formulas. i cant find any formula which leads to an actual width of 872 [17:17] <mpfundstein> i tried so many formulas, all of them say actual width = 853, but i need 872 :-) [17:18] <JEEBsv> are you sure there is no other metadata related? [17:18] <JEEBsv> also what video format is that in? [17:18] <mpfundstein> :( i dont know [17:18] <mpfundstein> video format? [17:18] <mpfundstein> mpeg2 [17:18] <mpfundstein> one sec [17:18] <mpfundstein> ill give you a ffprobe output [17:18] <JEEBsv> k [17:19] <mpfundstein> https://gist.github.com/MarkusXite/e7f3b032fe698041c985 [17:19] <mpfundstein> 720*((16/9)/1.5) = 853 [17:20] <mpfundstein> 1.5 = 720/480 [17:21] <JEEBsv> 875 or so is the DVD "16:9" aspect ratio (the whole size of the active+non-active picture areas), but unfortunately that really doesn't give us enough info on what the actual values either in the MPEG-2 video track or in the MOV container are [17:22] <mpfundstein> how can quicktime calculate it right [17:22] <mpfundstein> when i open the video in quicktime it says 720x480 (actual: 872x480) [17:22] <JEEBsv> depends on several things, none of which I can say anythin about definitely [17:22] <JEEBsv> because as I said, not enough info [17:22] <mpfundstein> ok [17:22] <mpfundstein> what do you think is missing? [17:22] <mpfundstein> :-) [17:23] <JEEBsv> I have no fucking idea, a full dump of the MPEG-2 Video headers and the full dump of data in the container would be needed for that [17:23] <JEEBsv> for dump of the MOV file, you can try http://code.google.com/p/l-smash/ 's boxdumper [17:23] <JEEBsv> (output boxdumper's output into a text file, then go through it) [17:24] <mpfundstein> ok [17:24] <mpfundstein> so its probably encoded with non visible data on the left and right [17:24] <JEEBsv> as for the MPEG-2 stream, you'd need something like elecard streameye to check the headers [17:24] <JEEBsv> mpfundstein: no [17:24] <JEEBsv> the 720x480 picture in DVDs is by the specs encoded with a SAR of 40:33 [17:24] <JEEBsv> which ends up being 875x480 or so [17:24] <mpfundstein> AH [17:25] <JEEBsv> (the active picture is 704x480) [17:25] <mpfundstein> that is really f**** up [17:25] <JEEBsv> (and 704x480 with a 40:33 AR is 853x480 or so) [17:25] <mpfundstein> mh [17:25] <JEEBsv> not really [17:26] <JEEBsv> so in the simplest case, QT just takes MPEG-2 Video in with these aspect ratios, but somehow I /really/ wouldn't like to think that's all it bases its guess upon [17:26] <JEEBsv> thus [17:26] <JEEBsv> go look at 1) the boxdumper's dump of the mov's metadata 2) the MPEG-2 Video headers [17:26] <mpfundstein> im screwed -.- [17:26] <Mavrik> hmm, how did I manage to miss the existence of AAC Main profile [17:27] <mpfundstein> ok thx JEEBsv, ill do more research on that [17:32] <casimiro_> Helloa all, is there anybody here who have combined ffmpeg with libev ? [17:35] <casimiro_> Or better, is there anyway that I can give a buffer with mpeg ts received from an tcp socket to libavformat so that I can extract the mpeg ts packets ? [18:27] <iive> casimiro_: ffmpeg merges most of libav, but it also adds its own stuff and fixes. [19:23] <casimiro_> thank you for your response iive. Do you know of anyway I pass a buffer containing an mpeg ts stream to some ffmpeg function and I get the mpeg ts packets ? [19:24] <iive> casimiro_: i'm not familiar with libavformat. but imho, mpeg-ts packets are the easiest to parse. they are always of same size and start with same sync byte. [19:28] <casimiro_> iive: In fact! But I have written an little app that receives mpeg ts from an TCP socket and send the stream through an UDP socket. In this case I need to splice the stream in order to send the udp packets correctly. I am able to find the sync bytes, but it seems that after some few packets I need to synchronize (find the sync bytes again..) [19:29] <casimiro_> iive: It seems that it's not only a matter of finding a sync byte and split every 188 bytes ahead [19:29] <iive> casimiro_: do you send exact multiples of the ts packet size? [19:30] <iive> udp could easily be lost or delivered out-of-order. [19:30] <Mavrik> casimiro_, so basically you want to read TCP stream [19:31] <Mavrik> casimiro_, and then forward it over UDP? [19:31] <Mavrik> without remuxing? [19:31] <casimiro_> Mavrik: yes [19:31] <Mavrik> hmm [19:32] <Mavrik> MPEG2-TS packetization is handled via MPEG-TS muxer in ffmpeg, so I'm not sure if there's an elegant way of doing that with ffmpeg libs [19:32] <Mavrik> (without remuxing that is) [19:32] <casimiro_> I see [19:33] <casimiro_> Should not it be just a matter of splicing the incoming stream on every 188 bytes ? [19:33] <Mavrik> casimiro_, yes, but you need to splice them on the packet boundaries [19:33] <Mavrik> casimiro_, basically you're looking for 0x47 sync byte at the start of each TS packet [19:35] <casimiro_> Exactly. And everything works fine at the beginning. But after a few packets, there is not anoter sync byte after 188 bytes from the current sync byte [19:35] <casimiro_> I understand that I will need to synchronize things after some time, but I am having to synchronize to often... [19:36] <casimiro_> I'm sorry if it's too much noob problem. =) [19:36] <Mavrik> not a noob problem really :) [19:36] <casimiro_> Thx! [19:39] <Mavrik> casimiro_, btw, are you checking continuity counters if your packets are in order? [19:41] <casimiro_> No...All I do for now is identify a sync byte send one UDP message for every 188 bytes ahead [19:42] <Mavrik> casimiro_, check those so you see if your packets are ordered properly on the input [19:42] <Mavrik> it's a 4-bit field in packet head [19:44] <casimiro_> But, an not ordered packet would implicate on bytes between packets ou packets longer or shorter ? [19:44] <casimiro_> Also, I am receiving the data from an HTTP request... [20:09] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [21:06] <shevy> quick question - can I easily combine .mp4 files via ffmpeg? [21:31] <shevy> ah well, the mencoder variant worked fine. cya. [21:32] <walker> Hi folks - I'm using ffmpeg to transcode a live video stream, from RTP to RTMP. [21:33] <walker> It works fine in good conditions, but I have an instance where the RTP sender starts with video muted. [21:33] <walker> So it doesn't start sending RTP until a few seconds in. During that time, ffmpeg starts up, can't analyze the video stream, and just does audio. [21:34] <walker> And won't do video after that (packets received from video stream are marked as NULL stream in ffmpeg -v debug) [21:34] <walker> How can I get around this? [21:34] <cheeseboy> with -f image2 -usefirst can i limit # of writes per second or something? [21:35] <beastd> cheeseboy: what do you want to do? [21:36] <cheeseboy> beastd, i want to load a preview frame ron my website but if its writting it 20 times a second its lil hard to load [21:36] <beastd> cheeseboy: you could use -re and probably probably set a different framerate [21:37] <cheeseboy> beastd, i want it to still encode at right rate thoo [21:37] <Fjorgynn> can I do -ss and copy and burn in the subs? [21:39] <beastd> cheeseboy: hmm. probably i still do not understand what you want. [21:39] <beastd> Fjorgynn: probably not [21:41] <beastd> cheeseboy: if you encode faster you will write out more frames per second and therefore you will be overriding the outputfile more often. [21:42] <beastd> Fjorgynn: I meant, you cannot use copy for the video stream you want to burn the subs onto. [21:52] <Fjorgynn> seems like I can't copy an ass file to that [22:15] <Youka> @shevy: have a look at <http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join,%20merg…> [00:00] --- Sat Feb 9 2013

1 0

ffmpeg-devel.log.20130207
by burek 08 Feb '13

08 Feb '13

[01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:8eda88868399: sanm: check image dimensions before using them [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:bc58fe0309a0: rtpenc: fix overflow checking in avc_mp4_find_startcode() [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:7df878ac1507: rtmp: fix multiple broken overflow checks [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:c3744b415a97: rtmp: fix buffer overflows in ff_amf_tag_contents() [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:b666debffec1: huffyuvdec: Check init_vlc() return codes. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:db0f7f7394e1: huffyuvdec: Skip len==0 cases [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:85a14dbd5dca: adpcm: fix off by 1 error and out of array access in DK4 [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:6df0d3e2916c: alsdec: check block length [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:2502914c5f8e: atrac3dec: Check coding mode against channels. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:3e196e4def03: error_concealment: Check that the picture is not in a half setup state. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:4cd1dad91ae9: avcodec_decode_audio: do not trust the channel layout, use the channel count. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:89e16e675d3c: sanm: Check decoded_size. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:08e2c7a45f82: vqavideo: check chunk sizes before reading chunks [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:c459c7b23eff: aacdec: check channel count [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:3f2715804019: h264: Only apply error concealment if theres a frame [01:26] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:1690c1fa8860: update for 1.0.4 [02:13] <cone-533> ffmpeg.git 03Stefano Sabatini 07fatal: ambiguous argument 'refs/tags/n1.0.4': unknown revision or path not in the working tree. [02:13] <cone-533> Use '--' to separate paths from revisions [02:13] <cone-533> refs/tags/n1.0.4:HEAD: doc: replace '×' symbol with 'x' [03:55] <Compn> ubitux : did you see the opencl filter patch ? [03:55] <Compn> michaelni : any idea if that opencl patch is an acceptable api ? [09:43] <ubitux> <@Compn> ubitux : did you see the opencl filter patch ? // i did, but i'm still not available [10:39] <cone-93> ffmpeg.git 03Tim Walker 07master:08797c599df5: flac: don't check the number of channels before setting the channel layout [12:01] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:199683c898c3: mov: remove unused function [12:44] <cone-93> ffmpeg.git 03Nicolas George 07master:3dea8b375ed5: lavfi: remove disabled code. [12:44] <cone-93> ffmpeg.git 03Nicolas George 07master:b5884db437ee: lavfi: fix missing space in error message. [13:06] <burek> michaelni, i think CVE ids are missing for 1.0.2 at http://ffmpeg.org/security.html [13:29] <cone-93> ffmpeg.git 03Diego Biurrun 07master:79dad2a93253: dsputil: Separate h264chroma [13:29] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:c4e394e46008: Merge commit '79dad2a932534d1155079f937649e099f9e5cc27' [13:41] <cone-93> ffmpeg.git 03Diego Biurrun 07master:82bd04b170de: rv34: Drop now unnecessary dsputil dependencies [13:41] <cone-93> ffmpeg.git 03Martin Storsjö 07master:a846dccb29d2: h264chroma: x86: Fix building with yasm disabled [13:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:60a0bc46cdba: Merge commit 'a846dccb29d2bb0798af1d47d06100eda9ca87cc' [13:44] <burek> Forum stats :) Total posts 992 Total topics 245 Total members 215 [13:44] <burek> we are approaching 1000th post :) [13:47] <cone-93> ffmpeg.git 03Tim Walker 07master:41244e13d545: flac: don't check the number of channels before setting the channel layout. [13:47] <cone-93> ffmpeg.git 03Tim Walker 07master:9d083d64172e: flac: add channel layout masks for streams with 7 or 8 channels. [13:47] <cone-93> ffmpeg.git 03Anton Khirnov 07master:ca1fe6c0e608: h263: remove an unused parameter from ff_h263_decode_init_vlc [13:47] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:d5518545d1b6: Merge commit 'ca1fe6c0e60808da45d4dfd8728f45e843b9f9b0' [13:52] <durandal_1707> burek: where is list of stuff fflogger know about? [13:54] <burek> what do you mean [13:55] <burek> list of commands? [13:55] <durandal_1707> exactly [13:55] <burek> well !help should do [13:55] Action: durandal_1707 feels dumb [13:55] <burek> you probably blocked noticies [13:55] <durandal_1707> no, i ignored them [13:56] <burek> thats what i meant :) [13:56] <durandal_1707> i expect it to spam here [13:56] <burek> why [13:56] <burek> you need the help, not everyone else ^^ [13:57] <someone-noone> hello! I'm developing some format for ffmpeg. How should I specify fps value while demuxing? [14:00] <cone-93> ffmpeg.git 03Anton Khirnov 07master:9782c778a205: h264: remove silly macros [14:00] <cone-93> ffmpeg.git 03Anton Khirnov 07master:c5fcdb440237: error_resilience: remove a useless if() and FIXME [14:00] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:7f23b2a9d0da: Merge commit 'c5fcdb440237f06f6c954185ab60970cabf786a2' [14:00] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:b8d6108cfcb8: error_resilience: assert that unsupported B frame interpolation isnt used with h264 [14:00] <michaelni> someone-noone, if fps matches timebase setting the timebase alone should work fine see avpriv_set_pts_info [14:01] <Compn> someone-noone : what format are you working on? :) [14:02] <durandal_1707> funny but ffplay eats 1-3 last lines of image if height is not multiply of 4 [14:03] <Compn> is it sdl bug ? [14:04] <someone-noone> michaelni, actually I don't understand how avpriv_set_pts_info works(is it just settings time_base?). If I set directly time_base it doesn't work (values are 1 and 24000). If I put those values to avpriv_set_pts_info fps got some extraordinary values too. [14:04] <someone-noone> Compn, smth similar to mpegts but with some differences. [14:05] <Compn> michaelni : wow theres a ton of history to look for to find original author copyright in that h264 code :\ [14:05] <Compn> it all gets moved around like crazy [14:05] <Compn> http://git.videolan.org/?p=ffmpeg.git;a=commit;f=libavcodec/x86/dsputil_mmx… [14:13] <someone-noone> I do following: avpriv_set_pts_info(context->video_stream,32,1,24000); [14:13] <someone-noone> but ffmpeg -i testfile shows 3.17 fps [14:16] <michaelni> your timestamps or timebase maybe arent correct, also 24000 is probably not your framerate [14:17] <someone-noone> michaelni, I got those values(1 and 24000) while muxing. Pseudo code: put_be32(time_base.den); put_be32(time_base.num) [14:18] <someone-noone> and then passed this values to avpriv_set_pts_info, while demuxing [14:18] <someone-noone> the original file is in .mp4 format [14:18] <someone-noone> what am I doing wrong? [14:20] <someone-noone> also, original .mp4 file doesn't have r_frame_rate value. So,AFAIK, I should calculate it directly from time_base [14:20] <someone-noone> r_frame_rate = 1/time_base [14:24] <someone-noone> as well as: ffmpeg -i test.mp4 shows 23.94 fps [14:24] <cone-93> ffmpeg.git 03Anton Khirnov 07master:605b047bcc48: rv10: improve buffer size check. [14:24] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:1c65393906ab: Merge commit '605b047bcc48482dc8a356f56629da259fbddbf1' [14:24] <someone-noone> sorry, not 23.94, but 23.98 [14:26] <someone-noone> Another question is, what is tbr, tbn and tbc values? If I remember right, those values are stored in codec->flags and codec->flags2 [14:27] <someone-noone> I thought that providing codec->extradata is enough for correct decoding any codec in ffmpeg [14:33] <cone-93> ffmpeg.git 03Anton Khirnov 07master:12b54a1f39fe: mpegvideo: remove an unused function parameter [14:33] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:c7002e3d3dad: Merge commit '12b54a1f39fee22fa0399825ae47a43e60bad4c5' [14:34] <michaelni> someone-noone, the tbr, tbc, tbn values printed by the command line tool are unrelated to flag fields [14:36] <someone-noone> michaelni, thanks. Then what are they and should I specify them? [14:36] <someone-noone> And any advice about fps? What am I doing wrong? [14:36] <michaelni> someone-noone, i said it already, your values are wrong [14:37] <michaelni> either timebase or timestamps [14:37] <someone-noone> michaelni, timestamps you mean pts? [14:37] <michaelni> pts and dts [14:38] <michaelni> it could be other issues too but these are most likely [14:38] <someone-noone> hm, but I put time_base, pts and dts while muxing directly.... [14:48] <cone-93> ffmpeg.git 03Daniel Kang 07master:659d4ba5af5d: dsputil: x86: Convert h263 loop filter to yasm [14:48] <cone-93> ffmpeg.git 03Daniel Kang 07master:a1d36730342e: dsputil: x86: Fix compile error [14:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:54d8322355c0: Merge remote-tracking branch 'qatar/master' [16:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:48870853b2b7: x86/dsputil: Fix author attribution after code has been moved/splited around [16:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:4e2c63685e03: sws: dont write out of array on bigendian [18:47] <Compn> Fix the value of TWO [18:47] <Compn> haha [19:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:9f956611e33f: af_biquad: unroll loop, remove variable copies [19:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:a00c4b4db044: af_biquad: avoid subtraction in loop [20:32] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:1afbb4c22620: doc/filters: remove another instance of '×' [21:31] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:872855e2a5ea: lavfi/select: avoid double -> int lossy conversion in debug log [21:31] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:c0a4871c1687: lavfi/select: correctly set prev_t and prev_pts [21:55] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:7c00d853a658: matroskadec: support seek to non keyframes [21:55] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:a8e748eee506: lavf: add seek2any to allow forcing seeking to non keyframes via AVOptions [00:00] --- Fri Feb 8 2013

1 0

ffmpeg.log.20130207
by burek 08 Feb '13

08 Feb '13

[00:30] Action: klaxa is listening to Louise (CV: Kugimiya Rie) - Gomen ne [07:08] <stephanedev> hi everyone. i wrote some code using ffmpeg C API to stream a video generated in real time over RTP, but on the decoding side, there are ~2s of latency. [07:09] <stephanedev> does anyone know how to get rid of that latency? [07:10] <stephanedev> is it due to my use of avformat_open_input which may scan the received stream to guess the format? [07:10] <stephanedev> avformat_open_input is called with the sdp file generated by the encoding side [07:11] <stephanedev> codec is vp8 [07:12] <stephanedev> i have found several articles related to the ffmpeg command line parameters but i am having a hard time finding anything about how to get rid of this latency when using the ffmpeg C API [07:18] <stephanedev> when streaming the output of the libvpx encoder over tcp and then decoding it, there is no latency [07:20] <stephanedev> from what i have read, i would think the latency occurs on the decoding side which buffers some input to guess the stream format. is it correct? [07:34] <gudy> Hi, I'm running ffprobe on a ts segment, and I'm not seeing the same number of video frames compared to video packets. In particular, the first packet of type video has the "K" flag, but the first video frame is not a key frame (and their pts differ). It seems that the first few video frames are not being considered. What could explain that ? [10:28] <Bor0> hello, I'm experiencing the same exact issue as http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe…. could anyone give a hint about it? [10:51] <mxmx> who is alive? [10:52] <durandal_1707> noone [10:53] <mxmx> heh [10:59] Action: relaxed kills durandal_1707 [11:02] <mxmx> so i changed my mind, i rather add a png image to the video instead a subtitle [11:03] <mxmx> "movie=logo.png" <-- can logo.png be a path to the image? [11:03] <durandal_1707> yes [11:03] <mxmx> "movie=/path/to/image.png"? [11:04] <durandal_1707> yes [11:05] <mxmx> ok [11:32] Last message repeated 1 time(s). [11:32] <mxmx> what is wrong with this command [11:32] <mxmx> ffmpeg -i video.mp4 -vf "movie=/home/www-data/mysite.com/public_html/include/settings/logo.png [logo]; [in][logo] overlay=main_w:main_h [out]" -acodec libmp3lame -ar 22050 -ab 32 -f flv video.flv"; [11:46] <relaxed> hmm, the bot is gone. [11:46] <relaxed> mxmx: paste your command and all output to pastie.org [11:51] <mxmx> ok sec [11:56] <mxmx> http://pastie.org/6086277 [11:58] <relaxed> that's impossible to read, try again [11:58] <mxmx> i am using putty and thats how it displays it [11:58] <mxmx> sec [11:59] <mxmx> yeah its stupid putty [12:00] <mxmx> these are teh error messges tough [12:00] <relaxed> Then `command` 2>ffmpeg.log (and upload to dropbox or something. [12:00] <mxmx> http://pastie.org/6086299 [13:12] <theos> so i finally compiled x264 and everything else :D [13:13] <theos> now i have to learn how to use it properly for max compression [13:15] <JEEB> 1) encode 2500-5000 frame clips with crf values starting from 23, and going up if it still looks good enough for you, and down if it looks bad for you - until you hit the magical "highest value that still looks good for you" crf value [13:15] <JEEB> 2) pick the slowest -preset you can still manage time-wise [13:15] <JEEB> 3) ??? [13:15] <JEEB> 4) profit [13:15] <JEEB> yes, that's it [13:17] <theos> hi JEEB :) [13:18] <JEEB> and yes, the outlook of the same crf value will be somewhat different between settings (presets), but the tweaking you might or might not need should be relatively small [13:20] <JEEB> and for the ~1-2 minutes of input you'll probably want to learn about -ss and -t , the first one lets you seek and then -t limits how long will be encoded [13:20] <theos> is msmpegv2 better? [13:21] <JEEB> what? [13:21] <theos> i assume h264 is vcodec? [13:21] <JEEB> H.264 is a format, called h264 in the libavcodec/ffmpeg limbo. libx264 is an encoder for it [13:21] <theos> or should i rtfm? [13:21] <theos> oh [13:22] <JEEB> and no, there are no better encoders than libx264 for lossy, and at times nowadays even for lossless encoding [13:22] <theos> wont ffmpeg use x264 by default [13:23] <JEEB> to be honest, no idea -- they have now some defaults depending on the type of container you start outputting to, but I've never trusted ffmpeg's automatic selection of "what you probably want" [13:23] <theos> says Stream #0:0 -> #0:0 (mpeg4 -> mpeg4) Stream #0:1 -> #0:1 (mp3 -> libmp3lame) [13:24] <JEEB> so the mpeg4 (MPEG-4 Visual, what xvid and divx "conformed" to) encoder got used and audio got re-encoded with lame into mp3 [13:25] <JEEB> sounds like you tried to output into the avi container [13:25] <JEEB> at least defaults-wise [13:25] <theos> yes i tried "ffmpeg -i input.avi output.avi" :D [13:25] <JEEB> yes, don't do that, in the worst case you will get mpeg-1 at 200kbps rubbing its balls against your face [13:26] <theos> so what should i use? [13:26] <JEEB> I would prefer you to move to some container that can handle b-frames properly, and then just set video and audio formats to whatever you need [13:27] <JEEB> ffmpeg -i input.derp -c:v libx264 -crf 23 -preset medium -c:a copy out.mkv [13:27] <JEEB> now take a look at the first four lines I listed [13:28] <theos> increase crf and decrease 1 when its not looking good? [13:28] <theos> ah its called libx264 :/ i was using h264 and it was giving error [13:28] <JEEB> also use -ss to seek to some timestamp that has content in it, and -t to then limit it to a couple of minutes so you can actually do some testing [13:29] <JEEB> h264 is the name for the internal decoder methinks, libx264 is the library that handles H.264 encoding, which you can build ffmpeg with [13:31] <theos> can the output be avi too? [13:31] <JEEB> in theory yes, but I will bludgeon you to hell for it. AVI does not take b-frames in well, and that was found out with MPEG-4 Visual already. But at the time we had no alternatives, really -- so people pretty much got used to hacking b-frames into AVI [13:31] <JEEB> let's not continue that shit in the 21st century, thank you [13:32] <theos> haha ok. so mkv is a good option? [13:32] <JEEB> use mkv, mp4 or flv -- whatever floats yer boat. As long as it's not AVI [13:32] <theos> thanks a lot :) [13:32] <JEEB> that said, both mp4 and flv tend to limit audio formats and don't support most commonly used subtitle track formats, so mkv is the current "supermarket of containers" [13:33] <theos> so mkv it is! [13:35] <DEATH> 9 out of 10 pirates use MKV [13:35] <DEATH> Even hardware makers acknowledged it [13:35] Action: theos hides [13:35] <DEATH> Only Apple and Microsoft continue to ignore it with extreme prejudice [13:36] <theos> i dont use either of those ^^ [13:36] <DEATH> Samsung likes MKV [13:36] <burek> can ffmpeg use something like this: ffmpeg -f image2 -i image%03d.tiff -c copy output.WHAT ? [13:36] <theos> i have a samsung android phone if that counts :) [13:37] <burek> meaning is there any video format that stores tiff images in the video, just like jpegs can be saved in mjpeg [13:37] <JEEB> burek, I would guess that's possible, but I have no idea what container takes in tiff [13:37] <JEEB> try mkv? [13:37] <JEEB> lol [13:37] <burek> :) [13:37] <burek> ok, ill just re-encode :) its safer :) [13:37] <JEEB> yeah [13:37] <theos> does h264 have an audio codec? [13:37] <JEEB> it's a video format [13:37] <theos> :( [13:38] <theos> should i use mp3 for audio? [13:38] <JEEB> MPEG-4 family does have an audio format, but to be honest if your input is mp3 you might as well keep it that way [13:38] <JEEB> aka -c:a copy [13:38] <JEEB> aka "just copy the audio from the input" [13:38] <theos> yes i understand copy. thanks :) [13:39] <theos> its 160kbps though. 128kbps is enough for me. but i dont think it will have much difference on the output [13:39] <JEEB> yes, that kind of difference is miniscule compared to the video bit rate [13:39] <JEEB> and it will not make an already highly compressed audio clip any better [13:40] <burek> try aac+ to get as low as 32 kbps ^^ [13:40] <theos> is it lossless? [13:40] <burek> right.. [13:40] <burek> :D [13:40] <theos> :) [13:42] <JEEB> in general I tend to just copy audio tracks over if they're not grossly oversized compared to what they are -- as in, lossy tracks in pretty much all cases will just get copied over, if it's something lossless (PCM from DVDs or BDs, or TrueHD etc. from the BDs) I encode it with whatever I deem good enough for that job [aotuv's vorbis, fdk-aac's AAC, some lossless format possibly?] [13:42] <JEEB> also, depends highly on wtf the stuff will get played on [13:43] <DEATH> Good luck getting "hardware players" to recognize Vorbis or FLAC, heh [13:43] <JEEB> surprisingly both are supported in some things [13:43] <JEEB> but do note that I don't even know what the person is encoding for [13:43] <JEEB> and neither do you [13:46] <DEATH> Yeah [13:46] <DEATH> PC playback with MPC-HC = encode any format you like [13:46] <DEATH> MPC-HC or other good software player [13:46] <DEATH> I'm having great fun with Samsung TV vs "acquired" encodes [13:47] <DEATH> Some H.264 configs just won't decode [13:50] <someone-noone> hello! What is syntax for <rational> values? [13:51] <theos> i use my pc for every multimedia purpose [13:55] <theos> why is the output file longer than the input file? :) conversion isnt complete. maybe it will get fixed [16:55] <Diogo> hi i having problems using ffmpeg and threads.. [16:55] <Diogo> ffmpeg -threads 1 (guarantees that only one thread is running)? [16:56] <Diogo> i'm having one more than one thread [17:05] <Diogo> i'm doing something wrong? [17:33] <kittonian_> morning all [17:35] <kittonian> i am trying to figure out the command line to turn an MKV into an M4V for AppleTV3 while converting DTS surround sound to AC3 surround sound at a bitrate 640 [17:35] <kittonian> i've been using Handbrake and most of the time it works just fine, but there are some files that Handbrake will not work with for some reason [17:46] <kittonian> will this accomplish what I'm searching for? ffpmeg -i input.mkv -c:v libx264 -crf 18 -acodec ac3 -ac 6 -ab 640k output.m4v [17:48] <Mavrik> if you're trying to re-encode video and audio to H.264/ac3 then yes [17:48] <Mavrik> :) [17:48] <Mavrik> why are you reencoding video though? [17:49] <kittonian> trying to convert an MKV with DTS surround sound into an M4V for Apple TV 3 [17:49] <Mavrik> ah [17:49] <Mavrik> there's probably no need to reencode video [17:49] <Mavrik> (afaik AppleTV can handle pretty much any H.264) [17:49] <kittonian> i've been using Handbrake to do it but on some MKV files it doesn't output correctly [17:50] <Mavrik> so replace "-c:v libx264 -crf 18" with "-codec:v copy" [17:51] <kittonian> will that make sure the video quality isn't touched? [17:51] <kittonian> that's my goal [17:51] <kittonian> i'm trying to have the highest quality video and audio possible in an ATV3 compatible file [18:16] <frnk1> does anyone here knows a codec to make ffmpeg _demodulate_ a wav file containing the recorded sound of e.g. a modem connection/fax/datasette [18:17] <frnk1> i'm not that familiar in whistling modem sounds like the terminatrix in T3, but there might only be some parameters, and then some bytestream, txt-file, should result [18:18] <kittonian> Mavrik: hmm, got an error using -codec:v copy but my string seems to be working [18:19] <Mavrik> "an error" [18:19] <Mavrik> ? [18:19] <kittonian> Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted [18:20] <Mavrik> the actual error is probably written a line or few higher [18:20] <kittonian> Stream #0:0: Video: vc1 (WVC1 / 0x31435657), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 23.98 fps, 90k tbn, 48003.07 tbc (default) [18:20] <kittonian> Stream #0:1(eng): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1, fltp, 640 kb/s (default) [18:20] <kittonian> Stream mapping: [18:20] <kittonian> Stream #0:0 -> #0:0 (copy) [18:20] <kittonian> Stream #0:1 -> #0:1 (dca -> ac3) [18:28] <gudy> Hi, I'm running ffprobe on a ts segment, and I'm not seeing the same number of video frames compared to video packets. In particular, the first packet of type video has the "K" flag, but the first video frame is not a key frame (and their pts differ). It seems that the first few video frames are not being considered. What could explain that ? [18:39] <Mavrik> gudy: it's quite possible the decoder drops input frames until it gets an I-frame [18:40] <Mavrik> are you looking at frames (After decode) or packets (before decode)? [18:52] <casimiro> Hi all, I am trying to broadcast through UDP an mpeg-ts received from TCP. For what I have tested, I need to parse the incoming stream in order to deliver the mpeg packets correctely. I have tried to implement the parse by my self, but it's getting too complicated. I would like to know if anybody could give me some lead in how to use libav to do such parse. [18:53] <Mavrik> casimiro: do you have to do anything else besides pass-through of mpeg-ts? [18:54] <casimiro> No [18:54] <Mavrik> well, you could probably just do [18:54] <Mavrik> ffmpeg -i <your tcp input> -codec copy -f mpegts udp://<output-ip>:<output-port> [18:56] <casimiro> Thank you Mavrik! But actually I am developing a program to do that. Do you know if I could have something similar with some lib ? [18:56] <Mavrik> yeah, you can use libavformat calls to do that [18:56] <Mavrik> casimiro: grab ffmpeg source and check doc/examples/demuxing_muxing.c [18:57] <Mavrik> that's essentialy what you have to do [18:57] <casimiro> Thank you so much for your help! [19:02] <gudy> Mavrik: both [19:03] <gudy> Mavrik: so the first video packet has the "K" flag, but will not show up as a video frame [19:05] <gudy> here are the frames : http://pastebin.com/bBvi7q3X [19:05] <gudy> and the packets : http://pastebin.com/pt1BBAVa [19:24] <kittonian> is there a difference between -codec:v copy and -vcodec copy? [19:27] <JEEB> kittonian, -c:v and -codec:v are a newer way of setting it [19:41] <kittonian> JEEB: cool thanks [19:41] <kittonian> i'm trying to figure out why when i set -crf 18 it still shows q=23 during encoding [19:41] <JEEB> probably unrelated [19:42] <JEEB> as long as libx264 outputs crf=18 , all's fine [19:42] <kittonian> i read through the docs and it said to use -qp instead of -crf [19:42] <JEEB> -qp should never be used by end users [19:42] <kittonian> oh ok [19:42] <JEEB> or well, it can be used with other encoders than libx264 [19:42] <JEEB> it's constant quant [19:42] <JEEB> which is what that q probably shows [19:42] <JEEB> libx264 on the other hand [19:42] <JEEB> has something better [19:42] <JEEB> crf [19:42] <kittonian> gotcha [19:43] <JEEB> before you basically either had 2pass abr or 1pass constant quant [19:43] <JEEB> now you have 1pass crf which takes the better sides of both [19:43] <JEEB> one pass, but still distributes bits better than just setting a constant quantizer \o/ [19:44] <JEEB> thus being the closest to "constant quality" we've ever been [19:44] <kittonian> makes sense [19:44] <kittonian> i'm trying to convert an MKV with DTS surround to an M4V for Apple TV3 but retain the highest possible video and audio quality. [19:44] <kittonian> right now i'm using | ffmpeg -i input.mkv -c:v libx264 -crf 18 -acodec ac3 -ac 6 -ab 640k output.m4v [19:45] <JEEB> if the input is something that the apple tv3 can play back as-is without re-encoding, then that'd be the best way [19:45] <JEEB> but that would depend on what exactly is in there :) [19:46] <JEEB> anyways, I'd recommend going over with -c:v copy once [19:46] <JEEB> (and audio being re-encoded) [19:46] <kittonian> i tried -codec:v copy and it returned an error and wouldn't work [19:46] <JEEB> k [19:46] <kittonian> i would love to not have to transcode the video. that was my goal [19:47] <kittonian> and it's already in h.264 format [19:47] <JEEB> yeah, the problem becoming in what profile/level the apple tv3 likes its stuff [19:47] <kittonian> ffmpeg threw the error - Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted [19:47] <JEEB> oh [19:47] <JEEB> would probably want to pastebin the whole log there [19:50] <kittonian> http://pastebin.com/wmvZiAd0 [19:51] <JEEB> [ipod @ 0x7feac2843e00] track 0: could not find tag, codec not currently supported in container [19:51] <JEEB> Stream #0:0: Video: vc1 (Advanced) (WVC1 / 0x31435657) [19:51] <JEEB> so yeah, that's VC-1 [19:52] <kittonian> i'm not familiar with VC-1 [19:52] <JEEB> which actually /does/ have mappings for "mp4"-style containers but I'd still be surprised if your Apple thingy could play it :) [19:52] <kittonian> i assume that's a different codec than H.264 [19:52] <JEEB> it's a format from Microsoft that tried to be royalty-less for a moment and got onto both HD-DVD and Blu-ray specs [19:52] <kittonian> ahh [19:52] <JEEB> ...and then promptly got itself dragged into a patent pool of MPEG-LA's [19:53] <JEEB> Warner Bros IIRC have a long history regarding using VC-1 with their discs [19:53] <JEEB> basically VC-1 is Windows Media Video 3 on steroids [19:54] <kittonian> hmm, interesting. alright, well, do you see anything I should be doing to improve video and/or audio quality (as well as speed of transcoding) in my command line? [19:54] <kittonian> oh, and i want to ensure that it doesn't screw with the picture at all (i.e. exact same pixel dimensions, no cropping, etc.) [19:54] <kittonian> in Handbrake i was doing it strict [19:55] <JEEB> ffmpeg doesn't do that by default, unlike many other things it might poke if you don't set them (such as codecs f.ex.) [19:55] <JEEB> crf keeps the quality more or less constant with those settings so quality wise as long as you're around the highest crf value that still looks good to you, you should be fine [19:55] <JEEB> 23 is the default and if it looks good you go higher, and if it looks bad you go lower [19:56] <kittonian> from what i've read 18 is visually lossless [19:56] <JEEB> depends on who is looking at it and what the resolution is etc. [19:56] <kittonian> this is all 1080p movie stuff [19:56] <JEEB> then i'd say you're on the very low end of the line [19:57] <JEEB> as in, you could raise it gratuitiously by a few before you would get anything else but a free minification [19:57] <JEEB> also, regarding compression/speed settings x264 has you covered: http://mewiki.project357.com/wiki/X264_Settings#preset [19:57] <kittonian> ok, so i'm more than safe [19:57] <JEEB> you can set these with -preset [19:57] <JEEB> yes, you're safe but you might be wasting a nice amount of bits [19:58] <kittonian> and do i no longer need the -f mp4 directive? [19:58] <kittonian> i've seen that in a couple of examples but i'm not using it [19:58] <JEEB> if your output file name includes mp4 or m4v, nope [19:59] <JEEB> ffmpeg looks it automagically [19:59] <kittonian> great. that's handy [19:59] <kittonian> the only other thing had to do with the -profile option [19:59] <JEEB> yeah, profile/level (and making sure not too many refs get used) [19:59] <kittonian> i looked at atv3's specs and it says high or main with high quality audio is supported [20:00] <kittonian> but apple's specs don't even mention that it supports AC3 surround at 640k [20:00] <kittonian> and it does just fine [20:00] <JEEB> looking at every apple device since the 3GS (not looking at the apple tvs), they've all actually supported high profile, level 4.1 content [20:00] <kittonian> and is that the default? [20:00] <JEEB> just that itunes usually barfed at it [20:01] <JEEB> libx264's default is high profile, yes -- and the level is set by your preset's settings [20:01] <kittonian> since i didn't use a preset, should i be setting the level directly? [20:01] <JEEB> basically profile = feature set, and level = how much memory the decoder needs to decode, reference frame count [20:01] <JEEB> uhh [20:01] <JEEB> default is preset=medium [20:01] <JEEB> then you can set it to be either faster or slower [20:01] <kittonian> what's the better way to do it? [20:01] <JEEB> ? [20:02] <kittonian> for better looking/sounding output [20:02] <JEEB> huh [20:02] <kittonian> or does that not have anything to do with it [20:02] <kittonian> sorry, trying to wrap my head around all of this [20:03] <kittonian> the reason i asked is that i read that slower yields better results [20:03] <kittonian> but if this literally just has to do with computing power than i guess it doesn't matter [20:03] <JEEB> 1) find your highest, still looking fine crf value 2) pick the slowest preset that is still fast enough for you 3) look at the limitations of your playback device/application and set profile/level accordingly 4) since libx264 doesn't limit the amount of reference frames used to the level's, set it if needed [20:05] <kittonian> i guess the reference frames is what's throwing me off here. not sure what that should be [20:05] <JEEB> 1) sets the visual look 2) sets the compression ratio and encoding speed 3) limits 2's feature set 4) makes you limit the amount of reference frames that can be used to the levels [20:05] <JEEB> reference frames are the amount of frames a frame can reference for predictive encoding, more means better compression [20:05] <JEEB> also more means more memory usage [20:06] <JEEB> for whatever reason libx264 itself doesn't limit itself if you set a level [20:06] <JEEB> (the command line x264 encoder actually does) [20:06] <JEEB> so you have to set the maximum amount of reference frames if you want to actually adhere to the level [20:07] <kittonian> so medium is a good compromise between everything? [20:07] <JEEB> it's a default, offers good enough stuff in general while not being overtly slow [20:07] <JEEB> then the user can easily set it to something either faster or slower while getting more or losing some compression [20:08] <kittonian> isn't less compression always the better quality output? [20:08] <JEEB> no? [20:08] <JEEB> better compression means less size for pretty much the same output [20:09] <kittonian> ahh [20:09] <kittonian> that's cool [20:09] <kittonian> i'm from the audio world and you always want as little compression as possible to retain the highest quality [20:09] <kittonian> just getting into the video side of things [20:09] <JEEB> no, you don't want as little compression as possible, you want the least loss of information [20:10] <JEEB> and that's the same with video, basically. Least loss of information while still compressing things [20:10] <kittonian> ok, so what would your perfect world scenario command line be for Blu-Ray quality outputs? [20:11] <kittonian> without a concern for disk space as I have huge drives [20:11] <JEEB> well since crf sets more or less the quality level, if you have no concern towards the quality level, you might just as well set some faster preset then [20:12] <JEEB> (the quality given by a certain crf value differs between settings because the algorithms that lead to the certain rate factor start differing) [20:12] <JEEB> (but of course the difference usually isn't too big) [20:13] <kittonian> hmm, ok [20:13] <JEEB> the idea really isn't harder than 1) find your highest crf value that still looks good 2) find the preset of your choice regarding the compression/speed ratio 3) check what limitations whatever you're encoding for has 4) set profile/level/x264opts ref=NUMBER as needed per the limitations [20:14] <kittonian> i noticed one example i found referenced -threads 0 as the way to use all the computing power possible and greatly speed up encoding times [20:14] <JEEB> that is the default now [20:14] <kittonian> oh, great [20:14] <JEEB> and it just sets threads to 'auto' in x264 [20:14] <kittonian> though from my pastebin it seems -threads=6 is what is set [20:14] <JEEB> which is 1.5 the amount of virtual cores [20:15] <JEEB> anyways, select crf value and preset as you like (you can encode 2500-5000 frame sequences of actual content to test) [20:16] <JEEB> -ss seeks and -t sets the amount of time to encode [20:16] <JEEB> encode a minute or two for testing a few times and you should find out [20:16] <kittonian> ok [20:16] <kittonian> right now i'm only getting 6FPS on a 4-core mac pro [20:16] <JEEB> then if your thing can handle high profile/level 4.1 you don't really have to set profile as high profile is the default, and level 4.1 is -level 41 [20:17] <kittonian> i think i read that ATV3 can only handle level 4 [20:17] <kittonian> 4.0 [20:17] <JEEB> then -level 4 for that [20:17] <JEEB> and you'll have to limit reference frames to that and the resolution you're using... lemme go look up the numbers [20:19] <JEEB> ok, so level 4 has a MaxDpbMbs of 32768 [20:19] <JEEB> macroblock width and height of 1920x1080 is 120x68 [20:19] <JEEB> thus the amount of macroblocks in a single frame is 8160 [20:20] <JEEB> 32768 / 8160 = 4.something, thus 4 [20:20] <JEEB> so -x264opts ref=4 [20:20] <JEEB> would limit the encoder to four reference frames at a time [20:21] <JEEB> I've been thinking of implementing the automatic limitation of reference frames to the level you select into the libavcodec's libx264 wrapper [20:21] <JEEB> so it does the same as x264's cli encoder [20:23] <kittonian> this is interesting [20:24] <kittonian> just read that even though apple doesn't officially support high profile 4.1 people are playing back without issue [20:24] <kittonian> on both atv2 and atv3 [20:24] <kittonian> but not atv1 [20:25] <JEEB> yes, the first one was atrocious [20:25] <JEEB> it only played baseline profile [20:25] <JEEB> anyways, 4.1 has the same MaxDpbMbs limit [20:28] <kittonian> ok, well i guess i'll see whether or not it'll work when it's finished [20:28] <kittonian> if it doesn't and i need to do it again with different settings for high profile 4, is this the command? ffmpeg -i input.mkv -c:v libx264 -crf 18 -level 4 -x264opts ref=4 -acodec ac3 -ac 6 -ab 640k output.m4v [20:28] <JEEB> yes [20:29] <JEEB> also do note that itunes has its own limitations so if you have to go through it, that's a whole separate can of worms [20:29] <kittonian> i have to import the movie to the itunes library but itunes doesn't play back any of the movies properly. only atv3 plays them fine [20:30] <JEEB> depending on how nazi itunes is about that kind of stuff nowadays (it used to outright block >main profile stuff with the 3GS f.ex., while 3GS and newer phones could play high profile level 4.1 stuff JustFine) [20:30] <kittonian> i'm really hoping the jailbreak comes soon so i can just play the MKV files directly [20:30] <JEEB> you might be fine or might have to use something else to transfer the files over [20:31] <kittonian> you can't use anything else on a non-jailbroken atv. you have to use itunes on a computer and then connect that computer to the atv over ethernet sharing (wi-fi is very choppy) [20:32] <kittonian> i've been considering building a standalone HTPC with the latest XBMC installed to replace atv3 but i can't get netflix unless i run windows and use some crappy add-on [20:33] <kittonian> i really appreciate all the help. thank you [20:40] <gudy> when I run ffprobe on .ts segment with h264 content in it, -show_frames and -show_packets do not see the same number of video frames / packets. What could explain that ? [20:47] <saste> gudy: one packet can contain several audio frames [20:48] <saste> for video it should be the same number [20:48] <saste> unless you have problems decoding them [20:48] <saste> in that case you have frames < packets [20:49] <Hammerhead2011-S> Hi All, I have a rtsp stream coming into VLC and transcoded to rtp(rtsp) out on port "xxxx"/blah.mov [20:49] <gudy> saste: thanks, how can I find out what is wrong with a video frame ? [20:49] <Hammerhead2011-S> totem in Linux plays the stream no problem. Windows(Quicktime player) plays it for about 1 minute then stops. no error just stops. [20:50] <saste> gudy, check the log [20:50] <Hammerhead2011-S> I'm trying to keep the stream size smaller...encoding to FLV triples the size of the stream... [20:50] <gudy> saste: things like this : [h264 @ 0x100bc5280]no picture ? [20:50] <gudy> (that's from mplayer) [20:50] <saste> gudy, yes [20:51] <gudy> is there a way to get more info on what went wrong ? [20:51] <Hammerhead2011-S> transcoded using h264 BTW [20:51] <saste> gudy, -loglevel [20:51] <gudy> with mplayer, of ffprobe ? [20:51] <saste> ffprobe [20:51] <Hammerhead2011-S> The feed is from a video camera....My question....is there an easier way to do this? [20:54] <gudy> saste: "./ffprobe -loglevel 9 -analyzeduration 10000000 -i input.ts" doesn't output anything ... [20:56] <saste> gudy, asking on irc is not a substitute for the fine manual [20:56] <saste> you're not decoding, unless you select -show_frames [20:57] <gudy> ok, thanks [21:06] <gudy> saste: thanks, that helped. I'm now down to "[h264 @ 0x7fd6fb000000] mmco: unref short failure" which I'm trying to find out what it could be [21:06] <saste> gudy, bingo [21:07] <saste> corrupted stream [21:08] <gudy> mmmm [21:09] <gudy> this is an HLS stream [21:09] <gudy> the first problem is that there is a massive audio/video sync problem (segments are 10 seconds long, and audio/video frames are 3.5 seconds appart) [21:10] <gudy> and then, most segments will lack a key frame at the beginning, however, the key frame seems to be there, but simply invalid ... [21:13] <saste> gudy, how are you creating the segments? [21:16] <gudy> I am not creating the segments [21:16] <gudy> I am getting them from a partnet [21:16] <gudy> and my player doesn't like them too much [21:16] <saste> gudy, which player? [21:16] <gudy> the default android hls player [21:17] <saste> did you try with an I* device? [21:17] <gudy> on android 4.0 [21:17] <saste> also i'm curious, which android player? [21:17] <gudy> well, the player will play [21:18] <saste> also you should ask your partner to validate the HLS stream using the apple (proprietary) tools [21:18] <saste> most segments will lack a key frame at the beginning, however, the key frame seems to be there, but simply invalid ... < what do you exactly mean? [21:18] <gudy> here's the output with both packets and frames enabled [21:18] <gudy> http://pastebin.com/D7mgwe1j [21:19] <gudy> I mean that there is a video packet which is supposed to contain a video key frame [21:20] <gudy> if you look at the log I've pasted, first "FRAME" is after a bunch of video packets [21:21] <saste> gudy, a decoding problem [21:26] <gudy> how likely is it a bug in the stream vs a bug in ffmpeg's h264 decoder ? [21:29] <saste> gudy, H.264 decoder is very well tested [21:30] <saste> seems more likely a bug in the stream/encoder [22:07] <stephanedev> hi everybody. do you have any hint on how to reduce latency when using ffmpeg C API? [22:08] <stephanedev> i am talking about latency when streaming using rtp [22:09] <stephanedev> is it due to the encoder, the decoder, or both? [22:10] <stephanedev> from what I have read, i think there may be some option to pass to the encoder so that it does not buffer frames [22:26] <boot> hello, is there another option to record the desktop and games besides x11grab? [22:29] <stephanedev> boot: why? is there anything wrong with x11grab? [22:31] <boot> I'm using KDE and have an ATI with proprietary divers, they don't play well together. So by default KDE disable direct rendering mode. If I have a game in fullscreen ffmpeg only records a black screen... [22:34] <stephanedev> boot: on linux, x11grab is the only option i know of to grab the desktop. but i have very little experience with ffmpeg command-line tool so my answer is not authoritative at all [22:38] <saste> boot, dshow, vfw on windows [22:38] <saste> ah kde [22:38] <saste> no what's wrong with x11grab? [22:38] <boot> explained above [22:39] <saste> i see [22:39] <boot> i'm reading it might be a codec issue, i'll try a different one [22:39] <saste> i think it's related to x11 [22:39] <saste> so nothing related to codec [22:49] <stephanedev> when using avcodec_open2, is there some generic option (i.e. not codec dependent) i can pass in the dictionary that will prevent the encoder from buffering frames? [22:50] <stephanedev> something along ("tune", "zerolatency")? [22:50] <saste> stephanedev, don't think so [22:50] <saste> you need to find a specific solution for each encoder [22:51] <stephanedev> saste: thank you for your answer. does it mean it is impossible to create a rtp video streamer without latency which would be codec-agnostic? [22:52] <saste> stephanedev, in principle, no [22:52] <saste> in practice, yes [22:52] <saste> since you're not going to support all the possible codecs but just a subset [22:52] <saste> so you need to find meaningful options for each different codec [22:53] <stephanedev> yes that's how i understood your "in practice answer" [22:55] <stephanedev> is there any document (other than the source code) describing codec options that can be passed in the dictionary? [22:57] <saste> stephanedev, ffmpeg-codecs(1) [22:57] <saste> and you can do ffmpeg -h fullhelp to get a short description of all the private options [22:59] <stephanedev> saste: thank you again for your help. is ffmpeg-codecs a man page? [22:59] <saste> stephanedev, check also the docs on the website [23:02] <stephanedev> i have been looking at the ffmpeg doxygen doc and manual but maybe i missed it [23:02] <stephanedev> i'll check the output of ffmpeg -h full [23:12] <stephanedev> saste: thank you very much. i found the document providing all the available options so i will try to play with them to hopefully get rid of this latency [23:12] <saste> http://ffmpeg.org/ffmpeg-codecs.html [23:12] <saste> mostly incomplete, but better than nothing [23:12] <saste> (and patches are welcome) [23:12] <stephanedev> yes that's the one i am looking at :) [23:52] <Keshl> Blah. How do I find a video's framerate with ffmpeg? x.x [23:52] <Keshl> I feel like I'm mis-reading the output of "ffmpeg -i video.flv" -- Is "25 tbr" what I'm looking for? [23:55] <stephanedev> Keshl: what about ffprobe <your_video_file> and then looking at the fps value for each stream? [23:58] <Keshl> stephanedev: ... <.< So now I know ffprobe exists. [23:58] <Keshl> stephanedev: It's not saying it. [23:59] <Keshl> Same output as ffmpeg, still no useful information. xwx [23:59] <Keshl> I'm expecting either 15, 24 or 30 and I'm not seeing those numbers anywhere. [23:59] <stephanedev> can you paste the command output to pastebin.com? [00:00] --- Fri Feb 8 2013

1 0

ffmpeg-devel.log.20130206
by burek 07 Feb '13

07 Feb '13

[00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.0:0c2a350762c3: movtextenc: fix pointer messup and out of array accesses [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.0:21185028b4fb: dvenc: dont fail hard if the timecode is invalid [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:a55c274f519f: movtextenc: fix pointer messup and out of array accesses [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:f6b50924a5fd: dvenc: dont fail hard if the timecode is invalid [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:5a97a5291a83: riff: fix infinite loop [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:d2c1a8dc2de8: ljpegenc: allocate needed scratch-buffer [00:24] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:58c4b0f99082: ffplay: extend feedback in case of video changes [00:24] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:212bb6490fa6: ffplay: report video frame changes only in debug mode [00:37] <ubitux> llogan: awesome :) [00:40] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:a12787960e47: lavf/segment: extend PTS debugging log with -reset_timestamps [01:31] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:ccf9dd00da05: ffmpeg: dont allow -flags to override -pass [01:47] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:68dfe530e0fb: vdpau: Rename union, avoid name conflict with vdpau [02:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:0e1bb99f26c2: update for 1.1.2 [02:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:5f3fa5f93087: ffmpeg: dont allow -flags to override -pass [03:14] <llogan> michaelni: ping on rmlist [03:15] <michaelni> llogan, iam busy with 1.1.2 release, is it ok i do it tomorrow or later ? [03:16] <michaelni> i dont want to quickly enter a random command as root ;) [03:17] <llogan> fine with me. i also personally archived the archives incase you do blow it up by accident [03:23] <michaelni> llogan, done [03:28] <llogan> michaelni: thanks [04:15] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:127ff88639e7: ffmpeg: do not call exit from exit_program() [04:22] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:b48cf5412bb4: ffmpeg: do not call exit from exit_program() [04:46] <cone-290> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.1.2': unknown revision or path not in the working tree. [04:46] <cone-290> Use '--' to separate paths from revisions [04:46] <cone-290> refs/tags/n1.1.2:HEAD: ffmpeg: do not call exit from exit_program() [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.10:4d9bde86d024: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.11:3cecef5b0389: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.9:9f0bf48b5ce4: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:86f072ceb3e9: ffmpeg: dont allow -flags to override -pass [11:08] <durandal_1707> michaelni: there is some talk on HA how ffmpeg native resample is slow and ugly quality [11:18] <LoopHoldYoaBrown> Hi [11:19] <LoopHoldYoaBrown> How can i make sure libavcodec can't call get_format over and over again during hw decoding? [11:30] <durandal_1707> is there way to change audio pitch? changing sample rate without changing audio? [13:22] <cone-533> ffmpeg.git 03Diego Biurrun 07master:25841dfe806a: Use ptrdiff_t instead of int for {avg, put}_pixels line_size parameter. [13:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:ede45c4e1da8: Merge commit '25841dfe806a13de526ae09c11149ab1f83555a8' [13:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:0ddca7d416cc: dsputil: fixup half a dozen bugs with ptrdiff vs int linesize [13:29] <iive> these commits sound interesting. [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:c9f933b5b62d: Add av_cold attributes to arch-specific init functions [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:70b348ec5b6a: configure: Group all hwaccels together in a separate variable [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:620289a20e02: sh4: Fix silly type vs. variable name search and replace typo [13:33] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:6c3888487686: Merge commit '620289a20e022b9c16c10d546ef86cc0bb77cc84' [13:33] <iive> I actually have 2 questions. Is there any improvement for using ptrdiff over int? is ptrdiff_t signed? [13:34] <durandal_1707> in reality it is just cosmetic change with minor if any gain [13:35] <DEATH> ptrdiff_t is signed size_t pretty much? [13:38] <cone-533> ffmpeg.git 03James Almer 07master:8ab2173ed141: lavc/bink: Chech for malloc failure [13:44] <ubitux> hey there [13:44] <Compn> hey you [13:44] <michaelni> hey [13:44] <ubitux> seems there are some laught in the cvs-log [13:44] <Compn> ours ? [13:45] <ubitux> 25841dfe 0ddca7d4 [13:45] <ubitux> iive: isn't the ptrdiff thing useful for the asm? [13:45] <ubitux> avoid overread etc [13:46] <iive> i don't see how. [13:53] <Compn> devs using unfamiliar calls tend to forget signedness and sizes with the new types. well , afaik anyway [13:58] <iive> i do see now how. some 64bit asm function did sign extend, so it was removed. [13:59] <iive> or rather moved in the C code. So in reality it may even affect execution speed. [14:00] <durandal_1707> at what magnitude? [14:01] <iive> let's hope it makes it faster. [14:01] <cone-533> ffmpeg.git 03Tim Walker 07master:7af876a93fa8: flac: add channel layout masks for streams with 7 or 8 channels [14:09] <iive> but imho it just puts 64bit variable in the stack instead of 32bit. the sign extend should be done somewhere. [14:51] <cone-533> ffmpeg.git 03Luca Barbato 07master:b2a722a87e3b: sh4: drop unused functions [14:51] <cone-533> ffmpeg.git 03Vladimir Pantelic 07master:77bcb89600e7: vc1dec: use codec_id instead of codec_tag for VC1IMAGE [14:51] <cone-533> ffmpeg.git 03Diego Biurrun 07master:48a4ffa722c0: asf: K&R formatting cosmetics [14:51] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:fadf845973ae: Merge commit '48a4ffa722c0874b251de9d201babed52cef0bcb' [14:59] <cone-533> ffmpeg.git 03Anton Khirnov 07master:99162f8d46db: vf_yadif: silence a warning. [15:00] <cone-533> ffmpeg.git 03Anton Khirnov 07master:6ed9fc44badb: svq1: replace struct svq1_frame_size with an array. [15:00] <cone-533> ffmpeg.git 03Anton Khirnov 07master:238614de679a: cdgraphics: do not rely on get_buffer() initializing the frame. [15:00] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:6e9f3f3b6522: Merge commit '238614de679a71970c20d7c3fee08a322967ec40' [15:13] <michaelni> durandal_1707, about ha & ffmpeg resampler, beastd and his friends did some double blind tests to adjust the resampler parameters, so best you ask him [15:14] <michaelni> ill also ping him about it when i see him [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:6a399854517f: cmdutils: remove ansi from the list of broken codecs. [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:9bd6375d5f16: msrledec: check bounds before constructing a possibly invalid pointer, [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:da7baaaae79b: aasc: fix output for msrle compression. [15:17] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:ddc3ca2caa4a: Merge commit 'da7baaaae79b4d7d715d35ea6bcfbdd149edc177' [15:26] <cone-533> ffmpeg.git 03Anton Khirnov 07master:2cd4068071b9: fraps: fix off-by one bug for version 1. [15:26] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:a845ac78c942: Merge commit '2cd4068071b9a8908823a3107f97e938211045ce' [15:31] <cone-533> ffmpeg.git 03Kostya Shishkov 07master:685e6f2e3939: xxan: properly handle odd heights. [15:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:68c0144d028b: Merge commit '685e6f2e3939f124b41c7801cc541dad8252af3d' [16:03] <cone-533> ffmpeg.git 03Kostya Shishkov 07master:89f11f498b9c: qtrle: fix the topmost line for 1bit [16:03] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:769eba75eeaa: Merge commit '89f11f498b9c15bc71494a11a7ec560f4adf630d' [16:09] <cone-533> ffmpeg.git 03Anton Khirnov 07master:06cf597c3525: yop: check that extradata is large enough. [16:09] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:3138b158dd4b: Merge commit '06cf597c352519d2b70f293518c61b8c312f5d4f' [16:28] <cone-533> ffmpeg.git 03Anton Khirnov 07master:8136f2344458: yop: check for input overreads. [16:29] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:73562f1f834e: Merge commit '8136f234445862c94d1c081606b2d1e3d44fccf3' [16:40] <aballier> michaelni: you should probably change the comment after the qtrle 1bit merge; row_ptr became a pointer to the current row after the merge [16:41] <michaelni> aballier, will do, thanks [16:42] <michaelni> aballier, otherwise the code is ok after the merge ? I tested it with the files from the ticket both ways produce the same output [16:43] <aballier> wasnt sure if the commit was correct, but it seems so [16:43] <aballier> lemme check again [16:45] <aballier> for what i remember, all the 1st lines have the skip & 0x80 bit set, I have never been sure if that's the spec or just luck [16:45] <cone-533> ffmpeg.git 03Anton Khirnov 07master:39a9fdd00fea: yop: initialize palette to 0 [16:45] <cone-533> ffmpeg.git 03Anton Khirnov 07master:dff6197dfb9b: nuv: do not rely on get_buffer() initializing the frame. [16:46] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:0b74fd113412: Merge commit 'dff6197dfb9b2d145729885c63f865c5fdd184aa' [16:47] <aballier> meaning doing "pixel_ptr = row_ptr;" before the loop is useless [16:50] <aballier> and i'm not sure how to interpret 1st line not having skip & 0x80 bit set; it may mean we should process lines_to_change lines and not lines_to_change+1, but i'm lacking a sample like that [16:50] <michaelni> yes was thinking the same [16:51] <michaelni> either its correct/unused or we have to wait for a bugreport to provide a sample to know which way it should be interpreted [16:52] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:d71f8d74db54: qtrle: fix 1bit row_ptr comment [16:53] <aballier> yep, anyway, if i'm not mistaken, if skip & 0x80 has to be set for 1st lines, then both versions are correct, if it doesnt have to, then its likely both versions are incorrect for that case [17:08] <cone-533> ffmpeg.git 03Anton Khirnov 07master:293065bdb56e: mpegvideo: initialize dummy reference frames. [17:08] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:78dc4cb26d5a: Merge remote-tracking branch 'qatar/master' [18:20] <Compn> seriously? [18:21] <Compn> i cant write -t 36:30 but have to write -t 00:36:30 ? [18:21] <michaelni> 36:30 is 36minutes 30 seconds or 36 hours 30 minutes ? [18:21] <Compn> 36 minutes 30 seconds [18:22] <Compn> if i write 1:30 , 1 minute, 30 seconds [18:22] <Compn> if i write -t 30 , it accepts it as seconds ... how does it know its not hours ? :P [18:22] <michaelni> is this AA:BB thing a regression ? [18:23] <Compn> i'm just used to it working in mencoder :) [18:23] <Compn> the docs mention hh:mm:ss so , its feature request, not regression [18:23] <Compn> -ss behaves the same way [18:23] <Compn> as -t , requiring hh:mm:ss [18:24] <michaelni> probably best to open a ticket so its not forgotten [18:24] <Compn> i guess :) [18:24] <Compn> new question, how to specify container aspect ratio ? [18:25] <cone-533> ffmpeg.git 03Vignesh Venkatasubramanian 07master:579c4add6334: Fate test to print side_data crc if present [18:25] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:b2c443194484: framecrcenc: Also print the size of side data elements [18:25] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:864fdfa0627e: framecrcenc: print all side data elements, not just the first [18:25] <Compn> maybe i should ask in #ffmpeg :) [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:6d33026668c4: dshow: fix log message [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:3b9166fc7efb: dshow: better error reporting [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:c916389bd286: dshow: allow for more codec ID's [20:20] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:474519433634: Merge remote-tracking branch 'rdp/combined2' [20:33] <Zeranoe> Why does FFmpeg have "double versions", such as "libavcodec 54. 91.100 / 54. 91.100" both sides of the / are the same. [20:33] <wm4> isn't that header version vs. library version [20:34] <wm4> they can be different if you use shared libraries and are criminally insane enough to use newer library binaries with programs compiled against older versions [20:59] <michaelni> anyone wants to review the 32bit fixed point FFT patch ? [20:59] <michaelni> Nedeljko Babic ( 46K) [FFmpeg-devel] [PATCH 1/2] libavcodec: Implementation of 32 bit fixed point FFT [21:00] <Compn> thats the mips version ? [21:02] <michaelni> theres generic C code for it and MIPS code [21:03] <michaelni> ive been wrong, rechecking, its just C code no asm/mips code [21:09] <wm4> <Rodeo> FFmpeg applied http://patches.libav.org/patch/33823/ without http://patches.libav.org/patch/33822/ [21:09] <wm4> <Rodeo> I wonder if I should tell them? [21:09] <wm4> <Rodeo> at the very least <= 6 should be <= 8 [21:09] Action: wm4 wonders if he's a spy now [21:12] <llogan> I think the question is: why not tell us? [21:13] <wm4> because dicks? [21:48] <michaelni> beastd, "<durandal_1707> michaelni: there is some talk on HA how ffmpeg native resample is slow and ugly quality" [21:49] <beastd> michaelni: hydrogenaudio? [21:49] <michaelni> yes [21:55] <wm4> is libswsresample better? [22:05] <Compn> michaelni : unless HA did a double blind study like they did for vorbis, opinions dont mean squat :) [22:08] <beastd> the infinitywave graph is done with 0.10.4 from 2012-06 FWIW [22:11] <llogan> anyone try libsoxr? [22:48] <cone-533> ffmpeg.git 03Matthieu Bouron 07master:59d40fc7e66e: lavf/mov: skip version and flags attributes in mov_read_chan function [22:57] <cone-533> ffmpeg.git 03Stefano Sabatini 07master:b3d2c6f8b9d0: lavf/segment: fix DTS inconsistencies with -reset_timestamps [22:57] <cone-533> ffmpeg.git 03Stefano Sabatini 07master:d11ac64ecf03: doc: replace '×' symbol with 'x' [23:37] <cone-533> ffmpeg.git 03Matthieu Bouron 07release/1.0:f3144b0cc694: lavf/mov: skip version and flags attributes in mov_read_chan function [23:37] <cone-533> ffmpeg.git 03Matthieu Bouron 07release/1.1:02d3ad860957: lavf/mov: skip version and flags attributes in mov_read_chan function [00:00] --- Thu Feb 7 2013

1 0

ffmpeg.log.20130206
by burek 07 Feb '13

07 Feb '13

[01:04] <pigoz> what's the channel name for mono in -af pan? trying to map mono to stereo [01:09] <saste> pigoz, does ffmpeg -layouts help? [01:11] <pigoz> saste: dumb me I mispelled it to --layouts. I got what I needed from the source. [01:34] <llogan> cheeseboy: why not just use libx264 with ffmpeg [01:35] <cheeseboy> llogan, too lazy to learn arguments plus x264's progress is prettier [01:35] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [01:37] <cheeseboy> llogan, lazy [03:30] <FergusL> Are you around llogan? [03:32] <llogan> FergusL: for a little longer. [03:32] <FergusL> llogan: you helped me yesterday about compilation [03:33] <FergusL> I still have issues, unrelated to ffmpeg itself, rather to how I now do it. I didn't "check install" for all the packages [03:33] <FergusL> could that create problems regarding libraries linking ? [03:34] <llogan> ffmpeg will tell you if it couldn't find things [03:37] <FergusL> hm... [03:38] <FergusL> okay so I _have_ to run checkinstall ? [03:39] <FergusL> I had hoped I could do what I want without touching a single bit of the current packages etc... [03:39] <FergusL> as in, I want the ffmpeg version I compiled to be only used by the program I cmpile, all other programs using ffmpeg should find it in the usual place [03:41] <relaxed> FergusL: I have current static binaries for 64bit linux or you can use a custom prefix and avoid using your package manager. [03:42] <FergusL> I did that, too, relaxed [03:42] <relaxed> and? why would you need checkinstall? [03:43] <FergusL> but the guide I followed ( https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide) asks to runs checkinstall instead of make install [03:43] <FergusL> btw doesn't "make install" only copy files to the desired location ? [03:43] <relaxed> yes, to your prefix [03:44] <FergusL> exactly, ffmpeg is there, now [03:44] <FergusL> well I'm asking there but my issue now is unrelated to ffmpeg, really [03:45] <relaxed> I don't agree with these guides instructing people to replace software using their package manager. [03:46] <FergusL> well that's the suggestion I was about to do ! [03:47] <FergusL> moreover, someone noted the same thing as you yesterday [03:48] <relaxed> It's trivial to install software using --prefix=/home/blah/ffmpeg-custom, then running "LD_LIBRARY_PATH=/home/blah/ffmpeg-custom/lib /home/blah/ffmpeg-custom/bin/ffmpeg -i ..." [03:49] <FergusL> wait wait wait [03:49] <FergusL> oh okay [03:49] <relaxed> You can have infinite, multiple installs and no conflicts. [03:49] <FergusL> that's like an export $LD_blabla before [03:50] <relaxed> yes, but maybe you don't want it exported. [03:50] <relaxed> anyway, you get the idea. [03:53] <relaxed> install all recent external libs to --prefix=/home/blah/ffmpeg-custom, the configure ffmpeg with --extra-cflags="-I/home/blah/ffmpeg-custom/include" --extra-ldflags="-L/home/blah/ffmpeg-custom/lib" [03:53] <relaxed> There- a one minute guide that's not retarded. [03:54] <relaxed> If yasm is old, install it using the same prefix and then modify your path so the new binary is first. done. [03:58] <FergusL> relaxed: what do you mean by "then configure" ffmpeg ? [03:58] <FergusL> configure the compilation flags of what I'm trying to compile ? [03:59] <FergusL> (yes I'm trying to compile some small program using ffmpeg) [04:00] <relaxed> Install libx264 from git using --prefix=/home/blah/ffmpeg-custom; then when you go to run ffmpeg's ./configure, add --extra-cflags="-I/home/blah/ffmpeg-custom/include" --extra-ldflags="-L/home/blah/ffmpeg-custom/lib" so that the configure script will find the libs/headers of the recent version you just installed to the custom prefix (and not an old system version) [04:01] <FergusL> yes, i've done than... I... I think I've done that [04:01] <FergusL> but I'm pretty sure the issue I have now is not related to that [04:02] <relaxed> I have zero idea what issue you're having. [04:03] <FergusL> I know I know, I didn't mention it ! [04:04] <FergusL> the initial question was just to know if the issue I had could be related to previous omissions [11:17] <LoopHoldYoaBrown> Hello howto make sure libavcodec won't call h264 deocder get_format constantly during HW decoding? [11:49] <sceriffo> hi , I have to build a function that decode an array of AAC data into PCM data. My input is only this array. I've not the header frame. What kind of informations I need to perform this? (bitrate, etc..). Best regards, sceriffo [11:54] <sledges> how can I multiple-trim mp4 without intermediate files (and hell of concatenating them)? something like: ffmpeg -ss 00:00:05 -t 00:00:04 -ss 0:00:20 -t 00:00:04 -i in.mp4 acodec copy -vcodec copy -async 1 out.mp4 [11:55] <sledges> (hello all btw ;)) [11:55] <sledges> *-acodec [12:06] <aleray> hi, I have a set of VOB files and the sound quality is pretty bad and the audio level is low (amateur DV recording of dance performances). Is there any way to improve the sound quality, The files will then be converted to Ogg for being shown on the web. [12:35] <sceriffo> hi , I have to build a function that decode an array of AAC data into PCM data. My input is only this array. I've not the header frame. What kind of informations I need to perform this? (bitrate, etc..). Best regards, sceriffo [14:15] <rcombs> 'ehlo, wondered if anyone could help me track down a possibly-ffmpeg-related bug involving green artifacts around subtitles being burned in [14:17] <rcombs> the issue is present in Plex Media Server, which uses a modified version of ffmpeg (so it might have nothing to do with ffmpeg itself at all) [14:17] <rcombs> the current version, which has the issue, is based on git-2013-01-22-0c46ebe [14:19] <rcombs> issue is demonstrated in screenshots shown here: https://plexapp.lighthouseapp.com/projects/14382/tickets/1414-transcoder-ba… [14:20] <rcombs> once again, not sure if this is an issues with Plex's transcoder modifications, ffmpeg itself, libass, or any number of other things [14:20] <rcombs> but I figured I'd try here to see if anyone had ever heard of a similar issue [14:20] <rcombs> also present here: https://twitter.com/11rcombs/status/297898849245216768/photo/1 [14:22] <relaxed> it's called hardsubbing [14:23] <rcombs> ? [14:23] <relaxed> overlaying subs into the video stream [14:23] <rcombs> yes, I'm aware of the term [14:24] <relaxed> Do you have a command and samples to reproduce the problem? [14:25] <relaxed> If not, that's where you need to start. [14:25] <rcombs> sure, sec [14:25] <rcombs> well, I'll need to cut a sample file [14:26] <relaxed> I don't want to see the result of the problem, just to be clear. [14:26] <rcombs> and the commands (from my logfiles) are for the Plex Transcoder, which has a few differences in input syntax from stock ffmpeg [14:26] <relaxed> ok [14:27] <relaxed> Provide a link to the sample of the input and pastebin the command used. [14:27] <rcombs> OK [14:33] <rcombs> Feb 06, 2013 07:32:54 [0xa85ffb70] DEBUG - Job running: /usr/local/plexmediaserver/Resources/Plex New Transcoder -i http://127.0.0.1:32400/library/parts/4571/file.mkv -threads 0 -y -f mp4 -loglevel quiet -progressurl http://127.0.0.1:32400/video/:/transcode/session/20444012b84d521c6a3737829b… -vcodec libx264 -crf 23 -vsi 640x568 -x264opts ref=3:bframes=3:cabac=1:8x8dct=1:weightp=1:b-pyramid=normal:b-adapt=2:direct=spatial:me=hex:subq= [14:33] <rcombs> 7:analyse=most:trellis=1 -map 0:0 -burnsidx 3 -burnsscale 1 -acodec copy -map 0:1 /mnt/user/Plex-Media/Library//Application Support/Plex Media Server/Cache/Transcode/Sync/35941/103588/4a58b8b31592a6d99f9c.mp4.temp [14:33] Action: rcombs cuts a sample [14:39] <rcombs> will be up on Dropbox momentarily: https://dl.dropbox.com/u/46348978/Baka%20and%20Test%20001%20%281%29.mkv [14:40] <rcombs> (sample is the series' opener) [14:40] <rcombs> up [14:56] <rcombs> relaxed: ping ^ that was for you [15:14] <relaxed> rcombs: Sorry, ffmpeg doesn't have "-burnsidx 3 -burnsscale 1" so I can't reproduce. [15:18] <veleno> hello. is it possible with ffmpeg to receive an UDP stream and, while receiving it, to produce statistics about the quality it ? [15:21] <rcombs> relaxed: yeah, and I'm not sure what options that might correspond to [15:22] <rcombs> relaxed: (-vsi corresponds to -s, for instance, but that's no help with -burnsidx) [16:21] <jeje34> Hi to all ;-) [16:23] <jeje34> I have a problem with the last FFMPEG (1.1.1). I use the libavcodec to decompress H264 stream [16:26] <jeje34> But I'd like to have less CPU usage during decompressing my H264 stream. I want to try Multithreading (m_lpCodecCtx->thread_type = FF_THREAD_FRAME | FF_THREAD_SLICE; m_lpCodecCtx->thread_count = 3;) [16:26] <jeje34> but I have an error The maximum value for lowres supported by the decoder is 0 [16:27] <jeje34> I doesn't reallyunderstand how I can fix my lowres and max_lowres [16:27] <jeje34> if someone can explain me, thks a lot [17:01] <veleno> where do i find some doc about the format of the vstats files ? [18:13] <mm_> Hi there. I'm using ffmpeg as a client for an udp stream ( $ffmpeg -report -vstats -i udp://localhost:60001 output_drops.mpg -loglevel verbose ) and I need to know how can I extract information regarding potential errors from the logs. For instance when packets are dropped I observe errors like "end mismatch left=4677 147B52" and "concealing 520 DC, 520 AC, 520 MV errors in P frame". If you could provide some pointers regarding the semantics of those errors, i [18:13] <mm_> what they actually mean. The reason for this is that I'm developing a network protocol using ffmpeg as a sink and I need to adjust some parameters according to the video quality at the sink, so I need to quantify that quality, even if in a naive way. Thanks for your time. [18:23] <jeje34> no one can give me some reason about the lowres message error I have when I want set multithreading to decode h264 [18:23] <jeje34> ? [18:25] <Compn> anyway to set par/dar in mkv with ffmpeg while -vcodec copy ? [18:28] <Compn> i see -vf setsar but doesnt look like it works for copy.. [18:29] <Mavrik> videofilters get applied only when actually transcoding the steram [18:29] <Mavrik> *stream [18:29] <Mavrik> not while copying [18:29] <Compn> yeah obviously [18:29] <Compn> no way to set container aspect ? [18:29] <Compn> mkvmerge can do this i think... [18:29] <Mavrik> hmm [18:29] <Mavrik> I'm quite sure SAR is set inside H.264 stream, not on the container [18:30] <Compn> pretty sure h264 itself has codec aspect too... [18:30] <Compn> can ffmpeg modify either of them using -vcodec copy ? [18:30] <Mavrik> no because ffmpeg doesn't touch the stream when doing "copy" [18:30] <Compn> so mkvmerge it is , grumble [23:52] <Pinhole> I'm trying to transcode a ts stream from mpeg4 to mpeg2. I'm having trouble with mapping. http://pastebin.com/QkfdB9U4 [00:00] --- Thu Feb 7 2013

1 0

ffmpeg-devel.log.20130205
by burek 06 Feb '13

06 Feb '13

[00:02] <cone-831> ffmpeg.git 03Stefano Sabatini 07master:2d22e97f6d73: doc/filters: fix @item syntax in showspectrum docs [00:13] <cone-831> ffmpeg.git 03Stefano Sabatini 07master:0e2b0033f161: doc/filters: apply various rendering changes to the showspectrum docs [00:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:b77d94dc6cc6: ffv1enc: check for malloc failure [00:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:5a8311513091: ffv1: check for malloc failure [00:41] <llogan> saste: any comments (since you were the only one with input on ML)? http://pastebin.com/raw.php?i=6B1ege28 [00:44] <saste> llogan, i find annoying when I'm subscribed to a new ML without my explicit approval [00:45] <llogan> one reason i'm not going to auto-subscribe [00:45] <saste> so I'd suggest to advertise the current ffserver-user subscribers of the move [00:45] <saste> you could keep the ML active, but tell all the posters to move to ffmpeg-user [00:46] <saste> yes an option would be to forward ffserver-user mails to ffmpeg-user [00:46] <saste> i don't have a strong opinion on that [00:46] <Compn> i wasnt suggesting subscribing them w/ email subscriptions [00:46] <llogan> i already made two announcements on ffserver-user. [00:46] <Compn> only putting their emails in so they could post, automatically [00:46] <saste> you could leave ffserver-user for a month or so [00:47] <saste> i don't think there is much traffic anyway [00:47] <saste> and send a mail every week advertising that the list will be closed in X days [00:47] <saste> and providing instructions/link to subscribe to ffmpeg-user [00:48] <llogan> i already said today is it's last day (on jan 27 and again today). [00:51] <saste> llogan, i think it's safe to remove the list anyway [00:52] <saste> it would be a bit annoying for the few subscriber to move to an high traffic list [00:52] <saste> but should be allow a better support, since there will be more feedback [00:52] <llogan> and they can always disable delivery on their choice. [00:54] <saste> i'm fine with it [01:06] <llogan> michaelni: please rmlist ffserver-user at your convenience [01:45] <michaelni> llogan, ill wait 24h to give people a last chance to raise objections and then delete it [02:03] <llogan> ok [03:09] <cone-831> ffmpeg.git 03multiple authors 07master:708ed15d8ccd: libmpcodecs/vf_stereo3d: update to latest version from mplayer [03:32] <Compn> ehehe [03:43] <cone-831> ffmpeg.git 03multiple authors 07master:103a2c2b1757: libmpcodecs: update img_format.* & mp_image.* to latest from mplayer [10:36] <j-b> morning [11:54] <michaelni> morning [11:55] <TimNich> moening [11:58] <durandal_1707> its 12 [12:02] <cone-290> ffmpeg.git 03Diego Biurrun 07master:e66240f22e24: avfilter: x86: consistent filenames for filter optimizations [12:02] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:0b6f34cc9f28: Merge remote-tracking branch 'qatar/master' [12:16] <TimNich> not in my timezone [12:20] <kierank> it is always morning on irc [12:23] <michaelni> that proofs the earth cant be flat [12:36] <TimNich> sometimes feels like a moebius loop! [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:a54737f7ead0: libmpcodecs/cpudetect: update to latest from mplayer [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:745cb39d0558: libmpcodecs/help_mp: update to latest from mplayer [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:99a0813a7e0b: libhmpcodec/mp_msg: update to latest from mplayer [13:09] <michaelni> TimNich, a moebius loop is unlikely because if it was that way that moebius loop could be seen by simply looking up from some places [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:5d0115118df4: libmpcodecs/vf_divtc: update to latest version in mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:e02c713fb149: libmpcodecs/vf_filmdint: update to latest version from mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:2b9eb6167b32: libmpcodecs/av_helpers: update to latest version from mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:0b36a05c1752: libmpcodecs/vf_fspp: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:011702f5a934: libmpcodecs/vf_ivtc: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:25ed6e99d789: libmpcodecs/vf_ilpack: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:b7e6622971eb: libmpcodecs/vf_mcdeint: update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:fdd574ed3f60: libmpcodecs/vf_qp: update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:3ca46d0c10aa: libmpcodecs/vf_pp7:update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:8f7e3e678b58: libmpcodecs/vf_noise: update to latest version from mplayer [14:41] <juanmabc> mmm, mplayer pulseaudio got excellent, not even 1% [14:41] <juanmabc> cpu [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:fe8bc6ddfcd2: libmpcodecs/pullup: update to latest version from mplayer [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:14132599a372: libmpcodecs/vf_sab: update to latest version from mplayer [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:7a4885600ff0: libmpcodecs/vf_spp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:3ce3d4cc3c0a: libmpcodecs/vf_unsharp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:13afcdd00eec: libmpcodecs/vf_uspp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:81e357db6259: libmpcodecs/vf_pullup: update to latest version from mplayer [16:11] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:2b20397e1fbe: libmpcodecs/vd_ffmpeg.h: remove, its unused and removed upstream [16:25] <wm4> lol old broken mplayer sources lingering in ffmpeg [16:27] <cone-290> ffmpeg.git 03Xi Wang 07release/1.1:8d0631c8fa89: mpegvideo: fix loop condition in draw_line() [16:27] <cone-290> ffmpeg.git 03Luca Barbato 07release/1.1:01050448cfbc: get_bits: change the failure condition in init_get_bits [16:27] <cone-290> ffmpeg.git 03Janne Grunau 07release/1.1:1a28948eb3e5: videodsp_armv5te: remove #if HAVE_ARMV5TE_EXTERNAL [16:27] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:62de693a17f9: rtp: Make sure priv_data is set before reading it [16:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:85e94a30eeb9: Merge commit '62de693a17f9b107be7867d822d5accacd4be544' into release/1.1 [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:348cd84fc82e: rtp: Make sure the output format pointer is set [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:a856623e87d3: rtpenc_chain: Use the original AVFormatContext for getting payload type [16:33] <cone-290> ffmpeg.git 03Justin Ruggles 07release/1.1:d7e7e12abc7e: lavr: call mix_function_init() in ff_audio_mix_set_matrix() [16:33] <cone-290> ffmpeg.git 03Reinhard Tartler 07release/1.1:8d55c2441c67: Prepare for 9.2 Release [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:3d67f52f9dcc: arm: vp8: Fix the plain-armv6 version of vp8_luma_dc_wht [16:33] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:8a4464514fac: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [17:01] <cone-290> ffmpeg.git 03Piotr Bandurski 07release/1.1:22561bc0e963: aasc: fix 16bpp on big-endian [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:6baa54924980: huffyuvdec: Check init_vlc() return codes. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:75e88db33013: huffyuvdec: Skip len==0 cases [17:01] <cone-290> ffmpeg.git 03Diego Biurrun 07release/1.1:0e68b6ddce29: Use proper "" quotes for local header #includes (cherry picked from commit 6c1a7d07eb41b4cf36e48ebb560fecd9504fd968) [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:62c9beda0c18: sanm: Check decoded_size. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:9547034f9120: gifdec: gif_copy_img_rect: Fix end pointer [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:f3d16706060a: vqavideo: check chunk sizes before reading chunks [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:47e462eecc0a: aacdec: check channel count [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:3ef1538121fa: h264: check the pixel format directly and force a reinit on mismatches. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:d3bec2473972: h264: Only apply error concealment if theres a frame [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:eaa9d2cd6b8c: h264: skip error concealment when SPS and slices are mismatching [18:26] <wm4> how do I use avcodec_copy_context() when I want to use a codec different from the source context? maybe call avcodec_get_context_defaults3() with the wanted AVCodec after that? [18:36] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:a8343bfb6a3f: riff: fix infinite loop [19:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:c2dd5a18b27b: ljpegenc: allocate needed scratch-buffer [21:05] <Zeranoe> Could anyone tell me the current status of GPU encoding/decoding? I see NVIDIA provides a H.264 encoder/decoder that works off of CUDA. I'm interested in starting a LGPL GPU encoder [21:08] <nevcairiel> in general? only fixed presets, quality below software encoders without a significant speed advantage in many cases [21:08] <nevcairiel> for encoding, that is [21:09] <nevcairiel> decoders are usually working quite well, provide bit-exact output, speed is usually below any modern CPU, only real use-case is on embedded/mobile [21:10] <nevcairiel> and limited to mainstream profiles of course, for ex. h264 only 8-bit 4:2:0 baseline/main/high, both for enc/dec [21:12] <Zeranoe> nevcairiel: Is the speed below CPU encoding because of limitations of the GPU? Is there a possibility to make an encoder/decoder that would out preform CPU encoders/decoders? [21:12] <nevcairiel> it always depends what quality you're looking for [21:13] <nevcairiel> and what kind of gpu you have =p [21:13] <nevcairiel> latest nvidia models for example have a fixed function encoder, earlier models implemented it in cuda [21:13] <nevcairiel> the tests i remembered, running x264 on settings that resulted in similar quality and filesize, the speed was pretty similar [21:13] <wm4> what's the use case for GPU encoding? cameras? [21:14] <nevcairiel> video conferencing, i guess [21:14] <Zeranoe> nevcairiel: [21:14] <nevcairiel> or if you go into the phone market, camera recordings as well, i suppose [21:15] <Zeranoe> nevcairiel: Does FFmpeg support any GPU features? [21:15] <nevcairiel> ffmpeg supports gpu decoding [21:15] <nevcairiel> technically, avcodec does, you need some user code to make it work [21:15] <Zeranoe> nevcairiel: Is it automatically used, or is there special way to test it? [21:15] <nevcairiel> which the ffmpeg application does not implement [21:16] <nevcairiel> you could run vlc, it uses it [21:16] <Zeranoe> nevcairiel: Is there a way to test it on the command line? [21:16] <nevcairiel> is mplayer considered command line? =P [21:17] <Zeranoe> nevcairiel: is it only available in other applications? [21:17] <nevcairiel> gpu decoding is not useful for transcoding because the gpu decoders are so slow in comparison [21:17] <nevcairiel> the ffmpeg applications dont support it [21:18] <Zeranoe> nevcairiel: Interesting, so it only really helps with playing a steam? [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:a50049a197a5: dshow: use standardized raw pixel format lookup [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:47e88486b4b3: hdyc col [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:c540312ac3b5: Merge remote-tracking branch 'origin/master' into combined [21:18] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:f7b7f021e528: Merge remote-tracking branch 'rdp/combined' [21:19] <llogan> "hdyc col". not very descriptive. [21:20] <Zeranoe> nevcairiel: How many encoders support CPU threading in FFmpeg? [21:20] <nevcairiel> no idea, i dont encode much [21:22] <J_Darnley> Wasn't a flag added to codecs to state whether they supported frame and/or sliced threads? Does ffmpeg -codecs tell you? [21:28] <J_Darnley> Answer: My (slightly) old ffmpeg doesn't but a search shows 59 matches in 39 files. [21:35] <llogan> ubitux: how is the adventure? [22:02] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:b0635e2fcf80: movtextenc: fix pointer messup and out of array accesses [23:04] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:f0eacbc760c8: dvenc: dont fail hard if the timecode is invalid [00:00] --- Wed Feb 6 2013

1 0

ffmpeg.log.20130205
by burek 06 Feb '13

06 Feb '13

[00:26] <llogan> FergusL: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [00:27] <llogan> blitz: https://ffmpeg.org/trac/ffmpeg/wiki/Encoding%20VBR%20%28Variable%20Bit%20Ra… [00:35] <FergusL> thanks llogan [00:36] <FergusL> but ... erh... haven't tried it yet but I suspect the first command will ask for the removal of like tens of packages [00:39] <llogan> you could compile and not install the new ffmpeg but still use it [00:40] <FergusL> okay, that's what i hought [00:43] <FergusL> llogan: I just do make and then I should have everything in the output dir ? [00:45] <llogan> FergusL: the examples here may help http://ffmpeg.org/pipermail/ffmpeg-user/2012-August/008552.html [00:46] <FergusL> perfect llogan I would have been looking for someting like this indeed :) [00:47] <FergusL> what about /include llogan ? will follow the same --prefix ? [00:47] <FergusL> well I'll just look at ./configure anyway first [00:47] <llogan> you will have to refer to ./configure --help for each thing you're compiling [00:49] <FergusL> yes, a bit used to this, thanks again for your help llogan [00:49] <llogan> a "local install" version of the guide would be good [01:12] <saste> llogan, what do you mean by "local install"? [01:13] <saste> i have some notes about general configuration/compilation/installation in *NIX, I plan to put them on the wiki soon or later [01:16] <llogan> saste: using --prefix=$HOME for dependencies and then pointing ffmpeg to those with cflags, ldflags, etc. [01:17] <llogan> would be useful for users with no superuser/root access, and could be good to keep system "ffmpeg" from repo for other repo shit and a compiled ffmpeg for command-line usage [01:21] <llogan> repo package conflict has always been the biggest issue with the ubuntu guide. or a simple copy binary to ~/bin or whatever in $PATH might suffice. [01:21] <saste> llogan, I see [01:21] <saste> i'll try to write it (hardly before the weekend) [01:22] <llogan> i actually haven't done much of that myself though since i use a more flexible (IMO) distro [01:26] <FergusL> llogan which is... ? :) [01:37] <llogan> FergusL: arch linux [01:38] <FergusL> I knew it would be that or Gentoo :) [02:00] <ldionmarcil> a mp4 is like an container right? it has a sound track + Video? then how can I see what type the video is? like h.264 etc [02:03] <ldionmarcil> ffmpeg -i on file, got my annswer, thanks guys [02:04] <llogan> FergusL: gentoo? i'm not that neckbeard. [02:04] <FergusL> Haha [02:35] <quup> can I make ffprobe just print dimensions '%dx%d' for first video stream? [02:38] <sacarasc> With grep/sed/awk I am sure you can! [02:40] <quup> sacarasc· no doubt, I just don't have a shell :p [02:40] <quup> oh well, I'll just bite the sour apple and regex it [03:11] <FergusL> trying to compile, compiled yasm to /opt/yasm, now x264 doesn't find it, how can I specify yasm location ? [03:12] <FergusL> although it correclty isntalled with checkinstall... [03:12] <FergusL> aptitude shows the right version and location [03:44] <FergusL> compiling... still expecting it to break at some point... haha [06:07] <relaxed> FergusL: PATH=/path/to/prefix/bin:$PATH; ./configure ... [09:59] <gotwig> Hey [09:59] <gotwig> please help me, why does -ss option to cut some seconds from the beginning make the audio so bad? the audio offset is awful [10:00] <gotwig> in combo with a video [10:10] <relaxed> gotwig: -async 1 [10:10] <gotwig> relaxed, thank you reall much ^_^ [10:10] <gotwig> ^really :/ [10:17] <gotwig> can I add audio pitches with ffmpeg? [10:19] <leo2013> How to add filters for ffmpeg? [10:20] <leo2013> add one new filter or another functions based on old filters. [10:22] <gotwig> Is there anything I can do with ffmpeg, so the audio music files dont get recoginzed by youtube, with ffmpeg? [10:22] <gotwig> like filters (^) ? [10:23] <gotwig> because in Germany there are big problems with that... [10:36] <gotwig> somehow -t does not work for me... [10:36] <gotwig> what is 2:35:53 - 2:29:27 :/? [10:36] <gotwig> 6: [10:36] <gotwig> oups... [10:45] <gotwig> can I cut the endning of the video with the help of -t ? [10:45] <gotwig> *ending [10:45] <gotwig> like -t 350s ? or -t 350 ? [11:00] <gotwig> Any ide how to add audio pitch? [11:11] <durandal_1707> gotwig: isn't that just changing sample rate? [11:13] <gotwig> yeah I guess so [11:13] <gotwig> I just have to find a trick to get my video widly available on youtube... [11:16] <gotwig> and somehow I am not able to use -t in combination with -ss [11:16] <durandal_1707> sox have pitch effect that just change sample rate of audio, and audio is apparently unchanged [11:16] <gotwig> why does -t not work in combo with -ss? [11:17] <durandal_1707> -ss just seek [11:17] <gotwig> -ss cuts the specified seconds before the video, -t before the end [11:17] <gotwig> oh damn... [11:17] <gotwig> do you use avconv? [11:18] <gotwig> copy does not work for me... [11:18] <durandal_1707> avconv is another project, man [11:18] <gotwig> yes I know... [11:18] <gotwig> but its somehow the same application... [11:18] <durandal_1707> not really .... [11:18] <gotwig> from the syntax :X [11:18] <durandal_1707> not really .... [11:19] <gotwig> I always lose quality, I cant find a way to keep it... [11:19] <gotwig> durandal_1707, can I use -ss in combo with -t? [11:19] <durandal_1707> what you want to archieve? [11:19] <gotwig> or reach my goal in another way [11:19] <gotwig> cut a bit from the beginning [11:19] <gotwig> and a bit from the end [11:20] <gotwig> with one run [11:20] <durandal_1707> with -c copy? [11:20] <gotwig> no, I talk about cutting here... [11:20] <gotwig> with -ss and -t [11:20] <gotwig> but you told me -ss is just seaking.. [11:21] <durandal_1707> you want to cut file without transcoding? [11:22] <gotwig> yes [11:22] <gotwig> without [11:22] <gotwig> just cut the thing [11:23] <gotwig> can I do that? [11:25] <durandal_1707> thing is you can't precisely cut with -c copy, it is simply not possible [11:25] <gotwig> :( [11:25] <gotwig> can I use -sameq? [11:26] <gotwig> the thing is, that my video file shrinked down to 300 mb, from 900 mb... and I only cut a few minutes [11:26] <durandal_1707> because compressed bitstream packets of audio can have several samples... [11:26] <gotwig> can I try with -c copy and -sameq? [11:26] <gotwig> btw, is avconv faster or ffmpeg..? [11:27] <durandal_1707> video patckets should be fine except you cant cut in middle of non-keyframe because you will get nothing for few frames.... [11:27] <durandal_1707> gotwig: avconv is slower and buggy [11:27] <gotwig> ubuntu ships it... [11:27] <durandal_1707> gotwig: try for yourself [11:27] <durandal_1707> gotwig: so what? [11:28] <gotwig> durandal_1707, so I am going to do two runs? [11:28] <durandal_1707> -sameq is removed, and it worked only for transcoding.... [11:28] <durandal_1707> gotwig: i can help you only if you use ffmpeg, you are on #ffmpeg after all [11:28] <gotwig> yeah [11:28] <gotwig> I use ffmpeg now xD [11:28] <gotwig> but -c copy does not work for me... [11:29] <gotwig> http://paste.ubuntu.com/1611935/ [11:30] <durandal_1707> gotwig: do you understand what word complete means? [11:30] <durandal_1707> i'm just wasting my time here [11:32] <gotwig> durandal_1707, what? [11:33] <DEATH> gotwig: Complete console output? [11:33] <gotwig> :) [11:34] <gotwig> http://paste.ubuntu.com/1611946/ [11:39] <durandal_1707> gotwig: you are using ffmpeg from project that is not FFmpeg [11:39] <gotwig> it seems... [11:40] <durandal_1707> gotwig: put -c copy after -i ... [11:41] <gotwig> Failed to set value 'copy' for option 'c' [11:43] <gotwig> it seems my version is too old? [11:43] <durandal_1707> you are using wrong version [11:43] <durandal_1707> use ffmpeg [11:43] <gotwig> -sameq works for me [11:43] <gotwig> can I use that? even if its an older version [11:45] <durandal_1707> -sameq is not supposed to be used at all with -c copy [11:45] <durandal_1707> it does nothing [11:45] <durandal_1707> -sameq is not same quality [11:45] <durandal_1707> -sameq is same quantizier [11:45] <durandal_1707> it works when transcoding by pure luck [11:46] <durandal_1707> it sets quality to very high, so get that feeling you did something good when if fact you did nothing [11:46] <gotwig> :| [11:47] <gotwig> http://paste.ubuntu.com/1611965/ [11:48] <durandal_1707> gotwig: this is not avconv/Libav support channel [11:48] <durandal_1707> either leave or install ffmpeg [11:51] <gotwig> that is ffmpeg, isnt it? :( [11:51] <durandal_1707> no it is not, it is ffmpeg from Libav [11:51] <gotwig> I installed ffmpeg with apt-get install.. from my software resources [11:51] <durandal_1707> that is not ffmpeg but libav [11:51] <durandal_1707> ubuntu is bad [11:52] <gotwig> :( ! [11:52] <JEEB> basically it's an old ffmpeg from the libav project, the up-to-date binary in libav is called 'avconv' and works like a more up-to-date ffmpeg, also libav channel is #libav [11:52] <DEATH> Libav situation is annoying, too confusing for new users [11:52] <DEATH> Shot myself in the foot as well trying libav first not realizing what it really was [11:52] <DEATH> Well lesson learned [11:52] <gotwig> when I install just avconv, I dont am able to use ffmpeg [11:52] <JEEB> I think the worst part is how they kept the ffmpeg binary there without updating it, throughout at least 0.8 [11:52] <gotwig> when I install ffmpeg, I can use ffmpeg.. [11:53] <gotwig> I better should ask in #ubuntu... [12:01] <durandal_1707> gotwig: afaik ubuntu do not yet provide ffmpeg package [12:02] <gotwig> I grab the binary... [12:02] <durandal_1707> gotwig: you can download ffmpeg from www.ffmpeg.org ... [12:02] <gotwig> :) https://launchpad.net/~jon-severinsson/+archive/ffmpeg [12:04] <dentex> hi everyone. [12:05] <dentex> Just a little question on compiling flags... [12:05] <gotwig> actually ffmpeg in the newer version does give me the same results... [12:05] <dentex> what would be a minimum flag set to have just the ability to extract audio from mp4, webm, flv, 3gpp containers? [12:06] <dentex> --enable-decoders and --enable-demuxers ? [12:06] <durandal_1707> gotwig: i dont see full uncut output [12:06] <durandal_1707> and -c copy just works [12:07] <gotwig> http://paste.ubuntu.com/1612197/ [12:09] <durandal_1707> gotwig: 10.6 is very old version [12:09] <durandal_1707> dentex: and store it where? [12:10] <durandal_1707> you need muxer [12:10] <gotwig> its just 16 weeks old [12:10] <gotwig> I mean 12 weeks [12:10] <JEEB> yes, but 0.10.* as a version is already unfortunately old :P [12:11] <durandal_1707> gotwig: build date? [12:11] <dentex> you're right. I want to strip audio only. so "-vn -acodec copy $AUDIO_CODEC" [12:11] <gotwig> I get the static builds.. [12:11] <durandal_1707> dentex: in what container? [12:13] <gotwig> mummy, it works... [12:13] <gotwig> is that the right command? ./ffmpeg -ss 152 -i OUTPUT.mkv -c copy output3.mkv [12:14] <JEEB> yes, if you want to leave stream selection to ffmpeg (one video, one audio and not sure about subtitles) [12:14] <gotwig> lol, finally I see a nice output from ffmpeg... [12:14] <JEEB> anyways, that just means that it was fixed later than either libav 0.8 or ffmpeg 0.10 :3 [12:14] <gotwig> avconv does not offer that, just without any time formats [12:17] <gotwig> and ffmpeg is really very fast :( [12:17] <JEEB> remuxing generally is :) [12:46] <gotwig> And now I no longer have problems with audio offset [12:46] <gotwig> even when I dont use async 1 [12:47] <JEEB> yes, generally async 1 shouldn't be needed [12:49] <gotwig> I guess because of -c copy [12:49] <JEEB> I would rather guess at the much newer code that you are now testing :D [12:50] <FergusL> relaxed: thanks for your tip about PATH. I think I used a symlink instead [13:15] <_raven_> hi [13:15] <_raven_> how to merge some mp3-files together? [13:16] <gotwig> ffmpeg -i MP3_FILE -i MP3_FILE OUTPUT ? [13:17] <gotwig> no, sry, I am a noob... [13:17] <_raven_> i need any option with * [13:17] <gotwig> oh ok [13:17] <gotwig> so you need a loop? [13:17] <_raven_> not sure perhaps ffmpeg is able to handle* [13:17] <gotwig> I dont think so [13:19] <gotwig> something like that? http://ubuntuforums.org/showthread.php?t=2005963 [13:20] <gotwig> but that is for converting each one as individual... [13:20] <JEEB> that just re-encodes all files separately, yeah [13:20] <gotwig> how can I cut the end of the video? [13:20] <gotwig> like remove the last 20 seconds? [13:20] <JEEB> have fun calculating the right timestamp for -t [13:21] <JEEB> I don't think you can set it to minus-something to just cut stuff off the end [13:21] <JEEB> (mostly because in general ffmpeg doesn't know or care about the "length" until it hits the end of stream) [13:21] <_raven_> cat *.mp3 | ffmpeg ? [13:22] <JEEB> I would not recommend that, at least use mkvmerge to append them or something and then demux the stream :P [13:22] <JEEB> although depending on your luck that might or might not work [13:23] <DEATH> While on the subject: it would be handy to have an alternate -t syntax that specifies end offset instead of duration (same as -t if no -ss specified, subtracts -ss time from the specified value if -ss is specified). Less math to do in my brain. [13:24] <dentex> Hi. I wanted to re-issue my previous question, if I may. [13:47] <WillieNL> ola i have something weird with ffserver version 0.7.1 theire is no data send on the output stream untill i stop the input stream [13:47] <WillieNL> is this a known problem ? [13:48] <WillieNL> next to what is best advice for streaming audio only (aac) mp4 seems not always working without video [14:55] <WillieNL> wb burek [15:44] <makara> racking my brains with this math. I have 1920x1080 and want to author DVD (PAL 720x576). What steps to take? [15:44] <Mavrik> makara: first of all, your aspect ratio is problematic :) [15:45] <Mavrik> makara: widescreen DVDs are usually anamorphic [15:46] <WillieNL> or place the 4:3 content in a 16:9 frame with subtitle bar [15:47] <Mavrik> yeah, you can pad it aswell. Not as optimal of a situation :) [15:47] <WillieNL> well indeed even when my description is way wrong [15:47] <WillieNL> here i still do 16:9 -> 4:3 [15:47] <WillieNL> but visa versa is also possible [15:48] <makara> which does the DVD expand to reach 16:9 aspect? Height or width? [15:49] <Mavrik> makara: width [15:50] <Mavrik> makara: you just "squish" the 16:9 video to 720x576 [15:50] <Mavrik> then set SAR so it tells the decoder the video is 16:9 [15:50] <Mavrik> and it'll stretch it when playing back [15:50] <WillieNL> makara: an example http://forum.videohelp.com/images/guides/p1538554/aspectratio.jpg [15:52] <WillieNL> personal keep the aspect ratio as is else you endup with really tall or fat people onscreen :) [15:52] <Mavrik> WillieNL: that's why you set the SAR. [15:53] <WillieNL> well i'm no expect on ffmpeg since i can't make rtmp/rtsp restreaming functional for audio only [15:54] <makara> WillieNL, that easy? [15:54] <WillieNL> makara: well WowzaMediaServer aac audio fetch works but restream part stalls untill i stop the feeding then vlc starts playing the buffered content :S [15:54] <WillieNL> i render clueless at that stage [15:54] <makara> just resize to 720x576; create mpeg-ps; author with widescreen on (16:9) [15:55] <Mavrik> makara: just make sure your SAR is correct when resizing [15:55] <makara> Mavrik, what is SAR [15:55] <Mavrik> sample aspect ratio [15:55] <Mavrik> basically, it tells how to stretch each pixel [15:57] <Mavrik> makara: SAR has to be set so that "720 * SAR = 1024" [15:57] <Mavrik> (because your video should be 1024x768) [15:58] <Mavrik> er [15:58] <Mavrik> 1024x576, sorry :) [16:01] <JEEBsv> something munch, I am expecting that your HD source is BT.709 and with a 1:1 aspect ratio: -vf colormatrix=bt709:bt601,scale=width=702:height=576,pad=width=720:height=576,setsar='16:11' [16:01] <JEEBsv> *something to munch at [16:05] <JEEBsv> (this contains some of the 'dirty secrets' of certain things that are not exactly how they look like) [16:06] <Mavrik> hmm, so colormatrix conversion is necessary? [16:07] <JEEBsv> well, yes -- naturally most people just wouldn't care about most of this stuff 8) After all, ffmpeg does have a PAL DVD -target option, which would take care of most of it all 8) [16:10] <Mavrik> JEEBsv: btw, why the black padding with anamorphic video? [16:11] <JEEBsv> http://www.x264bluray.com/home/576p-pal (DVD uses the same aspect ratios) [16:11] <JEEBsv> also a calculator that uses the "exact values" instead of their simplifications http://ps-auxw.de/cgi-bin/ar-calc.pl [16:12] <JEEBsv> basically the 702x576 part of the picture is going to be widened to 1024x576, and the rest goes further [16:13] <brontosaurusrex> JEEB; continued from yesterday: the player (tv) does play "bluray" vc1 mkvs, only it cant read the "bluray" subs ... [16:13] <JEEBsv> so if you want to hifist, you can do it like this (resize to 702x576, and then pad), if you just want to get off easily, you can just resize to 720x576 :) [16:13] <brontosaurusrex> and the bluray subs are always bitmaps? [16:13] <JEEBsv> that is, if your source content is 16:9 [16:13] <JEEBsv> brontosaurusrex: not always, but pretty much nothing handles the few cases of text-based subtitles [16:14] <JEEBsv> I think only some French BDs actually used the text-based subtitles [16:14] <brontosaurusrex> so is there a way to hardburn those on a non-windows machine, using ffmpeg only perhaps? [16:16] <JEEBsv> as I said, pretty much nothing except for payware players support that part of blu-ray subtitles thanks to the fact that they are barely used [16:16] <brontosaurusrex> no, i mean bitmap based [16:17] <JEEBsv> not that I know [16:17] <JEEBsv> ffmpeg can render them, but I don't think it could hardcode them [16:20] <brontosaurusrex> i mean transcoding from vc1+subs > h.264 [18:13] <serard> hello [18:13] <serard> I'm looking to replace mp4split use using ffmpeg, any howto or link ? :) [18:27] <Fjorgynn> serard: for what? [18:29] <serard> to get some examples of command line arguments [18:31] <Fjorgynn> for what? [18:31] <Fjorgynn> I don't know what mp4split does or what you're using it for [18:31] <Fjorgynn> My car doesn't work. It is red. [18:35] <juanmabc> split mp4s? [18:36] Action: juanmabc guesses hard [18:44] <WillieNL> Fjorgynn: well more examples on how to stream only audio in a couple formats will be welcome and solve the 1 hr trial and error [18:44] <WillieNL> here ffmpeg feeding to ffserver still is not sending any data untill stop the feed [18:44] <WillieNL> or only to the first client [18:45] <WillieNL> highly frustrating if unsure how to do it right :) [18:46] <WillieNL> dunno if these bugs are fixed afther version 0.7.1 but dinner time here [18:46] <Fjorgynn> aha [19:20] <pskosinski> ffmpeg -f x11grab -r 25 -i :0.0 file.mpg [19:20] <pskosinski> Will it give me "raw" video? [19:21] <pskosinski> Or it will be compressed using a codec. How to get as raw video as I can? [19:22] <pskosinski> Because I want to "modify" it a bit, and if I compress it to webm and then edit result is awful [19:22] <klaxa> pskosinski: -c:v rawvideo [19:23] <klaxa> i'm not sure the mpeg container supports it though, i think avi can handle it [19:23] <pskosinski> ok, thanks, I will try :) [19:24] <klaxa> you will need a reasonably fast disk though (ram, ssd?) otherwise you can use a lossless videocodec for compression, needs more cpu though [19:25] <sweb> http://superuser.com/questions/547027/ffmpeg-multiple-video-filter [19:25] <pskosinski> Unrecognized option 'c:v' [19:26] <klaxa> pskosinski: update your ffmpeg :P (or use -vcodec) [20:01] <pskosinski> So, I would like to cut beginning and end of video. I used: ffmpeg -i in.mkv -ss 8 -t 541 -vcodec copy -an oot.mkv and it is cutting but there is one problem: when I play video in mplayer it shows that video starts from 8th second. Can I do something with that? [20:02] <pskosinski> Without re-encoding. [20:03] <pskosinski> Well, looks like for those 8 second video is staying in one place& [20:03] <pskosinski> seconds * [20:15] <llogan> pskosinski: you mean the image is still for 8 seconds and then plays? [20:15] <pskosinski> Yes [20:15] <pskosinski> But I just reencoded and I don't see quality lose, so kinda no problem now. :p But would be nice to know if it's possible anyway [20:17] <pskosinski> Ah& It seems that I have a bit old ffmpeg, I have Debian Wheezy& Maybe first I [20:17] <pskosinski> I'll try to update * stupid enter [20:31] <llogan> pskosinski: this could help https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuideHardy [20:31] <llogan> or a static build is an option [20:32] <pskosinski> Yeah, thanks. :) I see that Debian switched to libav& [20:38] <llogan> that has caused much confusion for users [21:16] <creep> hey http://www.androidinabox.com/mini-joystick-capacitive-touch-screen-joystick… what do you think about this stuff? [22:26] <taylanub> The ffplay, ffserver etc. executables aren't supposed to be very big, right ? 140M here, I'm guessing dynamic linking was disabled because I'm on OpenBSD. [22:28] <cheeseboy> [image2 @ 0xbe2d00] Could not get frame filename number 2 from pattern '/var/log/encodes/file.png' [22:29] <cheeseboy> i want to write to same file every time. how I accomplish this? [22:32] <cheeseboy> http://pastie.org/6074719 [22:40] <beastd> cheeseboy: not sure i understood you correctly. but please look at this http://ffmpeg.org/ffmpeg-formats.html#image2-1 and check updatefirst option [22:40] <saste> cheeseboy, -updatefirst [23:06] <cheeseboy> ffmpeg -f yuv4mpegpipe - -i - -updatefirst 1 -f image2 "/srv/http/file.png" | x264 - why isnt x264 recivng pipe? [23:09] <klaxa> because ffmpeg isn't writing to stdout [23:10] <klaxa> the first '-' is somewhat misplaced and incorrect completely i think [23:10] <klaxa> what are you trying to do even? [23:12] <sacarasc> cheeseboy: ffmpeg [input options] -i input [output options] output [23:14] <cheeseboy> ah [00:00] --- Wed Feb 6 2013

1 0

ffmpeg-devel.log.20130204
by burek 05 Feb '13

05 Feb '13

[00:02] <burek> michaelni, these powerpcs also need a different timeout value? [00:04] <michaelni> burek, probably [00:10] <burek> judging by this history log, it seems powerpcs are also stuck or something: http://fate.ffmpeg.org/history.cgi?slot=powerpc-linux-gnu-gcc-4.3.5 [00:11] <burek> they used to provied results every couple of hours [00:11] <burek> provide* [01:30] <michaelni> burek, maybe, i have no access to them so i cant tell ... [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:91debec976d2: swr/resample: fix filter rounding and cliping for s32 [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:b8c55590d54d: swr/resample: fix integer overflow, add missing cast [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:24244cec2fdd: swr-test: Fix clip to 32bit [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:be989f923678: swr: reorder init code to make rematrix status available earlier [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:b5f544a0f998: swr: Dont use floats for S32->S32 when possible [11:11] <elk> I'm trying to get ffmpeg to read from a file which is growing (ongoing recording) but to not exit when it reaches the files EOF, I have tried retuning AVERROR(EAGAIN) instead of a AVERROR_EOF when it reaches the EOF.. [11:12] <elk> Because I'm reading from a mov-file i looked at mov.c in libavformat but I can find where the actual file is read [11:12] <elk> I was thinking of peeking into stream and wait if the next data is a eof [11:58] <ubitux> JEEB: arg @captioning :) [12:02] <mateo`> ubitux: \o/ [12:02] <ubitux> :) [12:16] <JEEB> ubitux, ARIB captions are quite fabulous, are they not 8) [13:04] <cone-831> ffmpeg.git 03Anton Khirnov 07master:d9df93efbf59: mpegvideo: simplify REBASE_PICTURE [13:04] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:a2bff49e26a9: Merge remote-tracking branch 'qatar/master' [13:31] <cone-831> ffmpeg.git 03Michaël Cinquin 07master:60f06f986f81: libopenjpegenc: make dci compliant j2c [13:31] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:c67ddb56da55: Merge remote-tracking branch 'mjbshaw/master' [13:32] <ubitux> JEEB: well, it's not surprising to see such thing [13:38] <JEEB> it's been around for quite a while but I finally found a kind of open source renderer for it [13:39] <JEEB> caption2ass has been around for a while, but the conversion is shoddy at best [14:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:63c61f0be036: vc1dec: Fix 2 "warning: may be used uninitialized in this function" [14:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:353bd158f5d4: sws: Fix "warning: comparison of distinct pointer types lacks a cast" [14:41] <cone-831> ffmpeg.git 03Rudolf Polzer 07master:81f2549ec9ab: lavfi/showspectrum: display multiple channels in separate row [15:01] <durandal_1707> michaelni: what patch is supposed to fix #2227 [15:05] <ubitux> divVerent: you should add yourself in the showspectrum filter copyright [15:26] <ubitux> showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt i e this [15:26] <ubitux> thanks divVerent :) [15:27] <ubitux> we should find a friendly way of making the playback fluid& [15:49] <michaelni> durandal_1707, [FFmpeg-devel] [PATCH 2/2] ffmpeg: Avoid matching attachment pics to video on wildcard maps. [15:57] <durandal_1707> what is policy about pathes that quatar folks just forget to apply? [15:57] <durandal_1707> *patches [17:07] <michaelni> durandal_1707, i dont think theres a policy about that, but which patch ? [17:43] <burek> can anyone please help me to properly answer this question about security vulnerabilities in ffmpeg 1.0.2: http://ffmpeg.gusari.org/viewtopic.php?f=16&t=818 [17:47] <burek> also, if someone knows what is a NEON simd, I would appreciate any help on that too: http://ffmpeg.gusari.org/viewtopic.php?f=8&t=819 [17:48] <JEEB> NEON simd is just simd for armv7 cpus [17:48] <iive> burek: simd - single instruction multiple data. NEON is used on arm cpu's [17:48] <JEEB> he is expecting some kind of hw H.264 encoding/decoding or whatever, and the reality hits him :P [17:48] <JEEB> yay armv7 [17:48] <burek> i see, how can someone check if ffmpeg is using simd or main cpu for encoding [17:49] <burek> that's assentially what the guy is asking [17:49] <JEEB> SIMD is main cpu [17:49] <burek> oh [17:49] <JEEB> it's just instructions that deal with "multiple data" at the same time (like SSE/MMX and their derivatives) [17:50] <burek> yes, i understand, i guess the guy thought it was some kind of additional gpu or something [17:50] <JEEB> yeah [17:50] <iive> MMX and SSE are SIMD [17:50] <JEEB> yup [17:50] <JEEB> for IA32 and x86_64 [17:50] <burek> ok, at least now i know what to answer there :) thank you both :) [17:52] <wm4> burek: didn't we have this question here on this channel recently [17:52] <JEEB> yeah [17:52] <JEEB> I remember a similar guy coming around here [17:53] <burek> probably, i didnt follow the channel in detail, but this one was from the forum, so it might be the same guy perhaps [18:04] <wm4> which mp3 decoder is actually preferred, "mp3" or "mp3float"? [18:08] <wm4> I would have thought it's mp3float, but both ffplay and ffmpeg seem to select "mp3" by default [18:09] <wm4> avcodec_find_decoder(AV_CODEC_ID_MP3) returns "mp3" too [18:12] <burek> if anyone knows this thing about security vuln. in ffmpeg 1.0.2 ( http://ffmpeg.gusari.org/viewtopic.php?f=16&t=818 ), just ping me and I'll see the message later :) and thanks in advance :) [18:23] <burek> one more thing :) is there any relation between "FFserver version SVN-r25767" and some git version currently hosted on ffmpeg.org ? [18:24] <burek> i.e. is there a way to find out what is the git commit hash of that svn version? [18:26] <wm4> burek: it was 47d2ddca802f4c1bc4b454c5ac40f06f79b740a0 [18:27] <wm4> the svn commits imported in git always have this in their log message: "Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk" [18:27] <wm4> so you can just do git log and search by text [18:27] <burek> oh great [18:27] <wm4> and that version is from 2010 (lol) [18:27] <burek> thanks [20:31] <michaelni> burek, ffmpeg 1.0.2 security fixes are listed at ffmpeg.org/security.html [21:13] <llogan> michaelni: i'd like to remove ffserver-user today: rmlist ffserver-user [21:13] <llogan> that should keep the archives [21:13] <llogan> actually i'll send out one more notice beforehand... [21:15] <michaelni> llogan, assuming theres consensus on this, is there anything i will need to do ? [21:16] <Compn> move all ffserver-user subscribers to ffmpeg-user , but turn off messages, so they are allowed to post, but dont recieve ffmpeg-user mails [21:16] <Compn> ? [21:17] <llogan> michaelni: i asked on -devel if there were no objections. only saste replied saying he supported this. i made an [announce] on ffserver-user describing the situation on jan 27. [21:18] <llogan> Compn: i am not migrating users due to the volume of messages in ffmpeg-user. they will have to do that themselves as mentioned in the announce. [21:19] <michaelni> so i dont have to do anything ? :) [21:19] <llogan> michaelni: you will have to do the actual list removal with "rmlist". [21:19] <michaelni> work :( [21:19] <llogan> i know...i would do it if i could. [21:20] <llogan> you should check first with "rmlist --help" [21:20] <llogan> in case the defaults/syntax differs from what i refered to [21:21] <llogan> i also already updated contact [21:22] <michaelni> isnt it better to leave the list and forward mails to ffmpeg-user ? [21:23] <michaelni> that is if someone from somewhere uses the ML address to post a mail [21:23] <michaelni> or reply to an old mail [21:23] <llogan> if you prefer that method you can do it that way [21:23] <michaelni> you can do that yourself i think :) [21:23] <llogan> i didn't see an option to set that up in the admin panel, but maybe i missed it [21:24] <michaelni> remove all subscribers, put the ffmpeg-user list as only subscriber to ffserver-user then it would get mails sent to it [21:27] <llogan> there did not seem to be a trivial method to disable subscribers. [21:28] <llogan> and considering the low-volume of ffserver i thought rmlist was the best tradeoff of work vs potential user convienence [21:28] <llogan> *disable subscribing [21:29] <michaelni> well i can rmlist it if thats what the ffserver-user people want [21:35] <llogan> alternatively, you can setup an alias in postfix to forward ffserver-user mail to ffmpeg-user. [21:36] <llogan> but truthfully i don't see much "reply to old mail" happening. [23:05] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:96a08d862730: wmavoice: silence may be used uninitialized warnings [23:05] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:fc0d069feb69: flacdec: silence several "warning: X may be used uninitialized in this function" [00:00] --- Tue Feb 5 2013

1 0

ffmpeg.log.20130204
by burek 05 Feb '13

05 Feb '13

[00:01] <taylanub> Is Diego Biurrun in this channel ? [00:02] <taylanub> I'll mail 'em I guess. [01:05] <pskosinski> Hi. Can I increase/decrease video speed while converting it? [01:07] <pskosinski> Oh, I was goind to ask what I should use as PTS& Now I see that it's variable, no need to replace. >.> Ok, nvm, thanks. [01:08] <pskosinski> Oh, but result for webm video is very bad& [01:42] <Jiyuu|C2> [00:40:15] <@saste> burek, can't be hader than subtitles=in.mkv [01:42] <Jiyuu|C2> tried that [01:42] <Jiyuu|C2> doesnt work [01:42] <Jiyuu|C2> in the documentation i dont see any mention of something like that [02:09] <glisignoli> Hello, I'm trying to use avconv to convert a mp3 to a 64k ogg. I'm doing "avconv -i somefile.mp3 -b 64k output.ogg" except the file output ends up as a flac, and much larger [02:10] <sacarasc> #libav is the avconv channel. [02:10] <glisignoli> ah sorry [02:10] <sacarasc> Hint: -c:a libvorbis [02:12] <glisignoli> Nice! [02:28] <Animawish> so I just transcoded an mkv file to mp4, and the video was fine, but the audio went to shit. I used the command "ffmpeg -i file.mkv newfile.mp4". What should I do to make the sound survive the conversion? [02:28] <sacarasc> -c:a copy [02:29] <sacarasc> If you have an MP4 compatible audio stream, anyway. [02:29] <Animawish> how do I know if I have an mp4 compatible audio stream? [02:30] <sacarasc> ffprobe -i file.mkv [02:30] <sacarasc> Then paste to a pastebin type site and someone will tell you. [02:31] <Animawish> http://pastie.org/6035603 [02:32] <kcm1700> I want to know the required lifetime of the 'extradata' field of AVCodecContext [02:32] <kcm1700> Is it okay to throw away after avcodec_open2? [02:33] <sacarasc> Animawish: I don't think AC3 can go in MP4... [02:33] <Animawish> hm [02:33] <Animawish> well it doesn't have to be mp4 [02:33] <Animawish> idek if i'm doing the right thing [02:33] <Animawish> i'm just trying to get these video files into a burnable format [02:34] <sacarasc> Why is it not burnable now? Or do you mean playable in some kind of device? [02:34] <Animawish> yeah. i want to burn them to DVDs to put in a DVD player to play on an awful TV that doesn't have HDMI in :P [02:35] <Animawish> hate that TV but whatareyagonnado [02:35] <sacarasc> What does the DVD player support? [02:36] <Animawish> idk. I was thinking I'd just use like, iDVD or equivalent to burn it [02:36] <kcm1700> I'm talking about the situation of decoding. [02:38] <sacarasc> Animawish: What is iDVD? :D [02:39] <Animawish> a first party DVD burning program in mac os [02:39] <sacarasc> Does it just burn, or does it convert? [02:41] <Animawish> i'm not sure [02:41] <Animawish> like [02:41] <Animawish> i can't use mkvs with it for sure [04:04] <Soho_> hi all [04:05] <Soho_> I am new in ffmpeg field [04:05] <Soho_> I have question about :how to get video frame from usb web cam then straming to remote client [04:05] <Soho_> the video format is h.264, [04:06] <Soho_> is there any example about this kind of application [04:07] <Soho_> it is not necessary to encode for video frame, because the content has been encoded by webcam [04:07] <Soho_> Does anyone have idea ? [04:15] <Soho_> Can I send video frame from usb webcam to network without transcode ? [04:18] <Soho_> And [04:18] <Soho_> if the video is H.264 and audio is PCM from USB web CAm, [04:19] <Soho_> which container is suitable for streaming by ffserver? [04:19] <Soho_> Does anyone have idea about my question? [04:20] <Soho_> I would like to send video stream to remote client by ffmpeg, is there any exmaple for reference? [08:08] <elkng> I use command ffmpeg -i "1.mp4" -vcodec mpeg4 -vb 1500k -s 650x276 -ab 128k -ar 44100 -ac 2 -acodec libmp3lame -y -f avi "1.avi" to convert video from mp4 to avi and thats the error output: http://sprunge.us/ZdbW what is wrong ? [08:35] <relaxed> elkng: replace "-s 650x276" with "-filter:v scale=650:-1" [08:36] <relaxed> you may need to specify an output frame rate too. [08:50] <SubJunk> How can I load videos with commas in the name? I've tried using -i file:'filename.mpg' with no luck [09:02] <relaxed> SubJunk: -i ./input,name.ext [09:02] <elkng> or './input,name.ext' ? [09:12] <SubJunk> Thanks, got it working by escaping the comma with a backslash :) [09:12] <SubJunk> Is it normal for subtitles to ignore seeking? Like if I use -ss 100 then the video and audio start 100 seconds in but the subtitles start at the beginning [09:31] <XATRIX> Guys, can you advice me what do i do wrong ? I'm tryting to glue/concat 3 mp3 files with voice [09:31] <XATRIX> Into a one, result.mp3 [09:32] <XATRIX> Encoding does good, but the result mp3 is unstable on a few browsers + it has broken pitch rate [09:32] <XATRIX> The command i issue to endoe it is : ffmpeg -i "concat:en1.mp3|ru1.mp3|ru2.mp3" -acodec copy /mnt/www/rf.ua/result.mp3 [09:32] <XATRIX> You can check the result mp3 by http://rf.ua/result.mp3 [09:33] <XATRIX> Also you can check the initial (good ones) mp3s by http://rf.ua/en1.mp3 http://rf.ua/ru1.mp3 http://ru2.mp3 [09:33] <XATRIX> English voice is ok, but the next 2 mp3s pitch rate failed [09:34] <XATRIX> What can be wrong ? [09:34] <XATRIX> http://fpaste.org/gxcT/ - the log of encoding process [09:52] <Soho> hi [09:52] <Soho> I got the trouble, when I build ffmpeg 1.1.1 [09:52] <Soho> ffmpeg is not build [09:53] <Soho> but fferver and fforibe are built [09:53] <Soho> the config.h shows #define CONFIG_FFMPEG 0 [09:53] <Soho> does anyone have idea? [09:53] <Soho> that I can use ffmpeg [09:53] <Soho> that I canNOT use ffmpeg [09:55] <Soho> Does someone help me? [10:51] <XATRIX> Sorry guys, were network connection problems... [10:52] <XATRIX> So any ideas on my question of ffmpeg and concat a few files ? [11:14] <elk> I'm trying to get ffmpeg to read from a file which is growing (ongoing recording) but to not exit when it reaches the files EOF, I have tried retuning AVERROR(EAGAIN) instead of a AVERROR_EOF when it reaches the EOF.. [11:14] <elk> Because I'm reading from a mov-file i looked at mov.c in libavformat but I can find where the actual file is read [11:14] <elk> I was thinking of peeking into stream and wait if the next data is a eof [11:25] <XATRIX> I have a trouble with ffmpeg encoding of a few files, can you look at ? http://fpaste.org/O8cf/ [12:20] <XATRIX> Guys, please help me with my trouble, i'm really stucked here ... :( http://fpaste.org/O8cf/ [13:08] <XATRIX> As long as i see my troubles is that en1.mp3 has 16000Hz freq, and ru1.mp3&ru2.mp3 has 22050 [13:09] <XATRIX> So when it does concat of en1.mp3+ru1.mp3+ru2.mp3 it prolong 16000 freq along the whole resulting mp3 [13:09] <XATRIX> Can't change it :( [13:09] <XATRIX> Any ideas ? [14:07] <sacarasc> XATRIX: You'd have to resample before concatenating... [14:14] <XATRIX> sacarasc: Any possibility to resample while concat ? [14:15] <sacarasc> Only if you want to re-encode all of them, I think. [14:16] <XATRIX> Yes, i tried to reencode all of them to -ab 32kb -ar 22050 [14:16] <XATRIX> But the resulting mp3 has the same shit [14:16] <p4plus2> If I had a set of 6 videos and wanted to arrange them in a 3x2 grid how would one go about this in ffmpeg? [14:17] <XATRIX> I can only fix it out with recoding the 'broken' mp3s from 16000 -> 22050 and then apply concat [14:17] <p4plus2> (optionally there can be a 7th video as a background generated black if needed -- not a problem) [14:17] <XATRIX> But isn't there any possibility to resample/recode it on fly with only one command ? [14:58] <Zeeflo> hi guys [14:58] <Zeeflo> the -sn option leaves out subtitles in mkv's [14:58] <Zeeflo> what the option if I dont want chapters as well? [15:00] <JEEB> see the documentation on ffmpeg.org about metadata, I think chapters are included in that part of it [15:00] <JEEB> since it's not really a "codec" I don't think you have a simple option like an/vn/sn for it [15:00] <Zeeflo> hmm [15:00] <JEEB> as those are what "codecs" handle [15:00] <JEEB> there was something to null all metadata or not copy it around [15:00] <JEEB> that might be what you want [15:01] <Zeeflo> that would be useful as well [15:03] <JEEB> oh [15:03] <JEEB> there's something separate [15:04] <JEEB> -map_chapters [15:04] <JEEB> http://ffmpeg.org/ffmpeg.html [15:04] <JEEB> -map_chapters input_file_index (output) [15:04] <JEEB> "Copy chapters from input file with index input_file_index to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying." [15:04] <JEEB> see the last sentence [15:17] <Zeeflo> JEEB, use a negative file index? [15:17] <Zeeflo> what would that look like? [15:18] <JEEB> -map_chapters -1 ? [15:19] <happarappa> Hello. I am trying to convert some .wavs to .flacs to save storage space with ZERO loss of quality. Is ffmpeg.exe -i test.wav -sameq test.flac safe for doing this? [15:20] <sacarasc> No need for -sameq. [15:20] <Zeeflo> ahh like that [15:20] <happarappa> I cannot HEAR a difference to the resulting file, but Winamp says "14H" in the case of the WAV and 925 kbps for the FLAC. [15:20] <sacarasc> Flac is lossless. [15:20] <D0minat0r> ffmpeg -vcodec copy -acodec copy -ss 0 -t 00:01:00 -i VIDEO_SOURCE.mp4 VIDEO_PART.mp4 That command will be the easiest way for me if i want to 1min split video_source.mp4 to video_part.mp4, right? [15:21] <happarappa> sacarasc: Well, it does say it ignores the "-sameq". [15:21] <happarappa> But if FLAC is lossless, why doesn't Winamp say "14H" in the FLAC file? [15:21] <sacarasc> happarappa: If you have a text file and you put it into a zip file, does it not get smaller without losing anything? FLAC is the same thing, but for music. [15:21] <Zeeflo> because winamp sucks [15:21] <happarappa> sacarasc: I know what lossless means... [15:21] <D0minat0r> And i will be able to split everything that ffmped supports? [15:21] <sacarasc> It reduced the bitrate. [15:21] <D0minat0r> I will get audio and everything? [15:22] <sacarasc> So it won't show the same one. [15:22] <happarappa> So basically, you are saying that it reduced the bitrate to the smallest possible WITHOUT losing a single bit of info? [15:22] <sacarasc> Yes. [15:22] <happarappa> Ah. Makes sense. [15:22] <happarappa> So CD quality is in other words 925 kbps. [15:22] <sacarasc> Well, maybe not the smallest, as you didn't tell it which compression setting to use. [15:22] <happarappa> I who thought 320 kbps sounded great... [15:23] <sacarasc> And it varies between tracks. [15:23] <happarappa> That last thing you said confused me entirely. [15:23] <happarappa> "as you didn't tell it which compression setting to use." [15:23] <happarappa> Why would it not use the maximum? [15:23] <sacarasc> It can compress lightly or heavily. [15:23] <sacarasc> Because ffmpeg's defaults aren't always what you want. [15:23] <happarappa> What possible reason could there be to not do it as efficiently as possible? [15:24] <sacarasc> It uses SLIGHTLY more CPU time to encode/decode higher compressed audio. [15:24] <happarappa> I suppose... in 1994. These days, the computers barely need to grasp for air to do it. [15:24] <happarappa> What option would that be, BTW? I would like to make them as small as possible. [15:25] <sacarasc> -q 9, I think, but I am unsure. [15:25] <happarappa> :( [15:25] <happarappa> Will the difference be noticable? [15:25] <happarappa> I mean in resulting file size. [15:25] <durandal_1707> happarappa: what is your issue? [15:25] <happarappa> Or is this an exercise in futility? [15:26] <sacarasc> happarappa: You could get it down another few hundred kbps... [15:26] <sacarasc> Depending on track. [15:26] <happarappa> durandal_1707: Um... I guess I'm trying to figure out how to convert a .wav to .flac with the best possible compression without losing any data. [15:26] <sacarasc> It's variable bitrate, so could be anywhere. [15:26] <durandal_1707> -q is for lossy not for lossless [15:26] <durandal_1707> happarappa: -h full and look for flac [15:27] <durandal_1707> there are options that control compression [15:27] <happarappa> But... [15:28] <durandal_1707> what? [15:29] <durandal_1707> use -prediction_order_method search [15:29] <durandal_1707> other options like -q do nothing for lossless codecs [15:29] <happarappa> I am confused about what you mean. [15:29] <happarappa> "use -prediction_order_method search" is not English to me. [15:30] <sacarasc> ffmpeg -i foo.wav -prediction_order_method search out.flac [15:32] <happarappa> Can't find a mention of prediction_order_method in http://ffmpeg.org/ffmpeg.html [15:32] <sacarasc> ffmpeg -h full [15:32] <sacarasc> That would have told you about it. [15:32] <sacarasc> ffmpeg -h full > blah.txt #so you can read it after. [15:33] <happarappa> It's just that it's not PATHed, so it's very annoying to call. [15:33] <happarappa> But why is the HTML manual not complete? [15:34] <happarappa> I have it now. Reading. [15:36] <sacarasc> The HTML docs aren;'t complete because just a text output of the full help is over 90kB long. :D [15:37] <happarappa> Hehe. 8 kB larger was the file that was not -prediction_order_method searched. [15:38] <happarappa> At least for the test track. [15:38] <happarappa> I would never in a million years have guessed that I would need to use that parameter/argument. [15:38] <happarappa> Why can't the defaults be optimized? [15:40] <durandal_1707> because defaults are to be fast [15:41] <durandal_1707> and flac can be documented in encoders, but it just is not.... [15:41] <durandal_1707> could file bug report [15:42] <happarappa> "flac can be documented in encoders"? [15:46] <durandal_1707> happarappa: encoders is file that document encoders present in ffmpeg [15:47] <durandal_1707> see http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders [15:54] <Youka> How i can extract attachments and subtitles as raw data with the API? It seems that AVPacket doesn't contain attachments and subtitles just as event lines (which seems to be buggy because in an example the first 15 lines are missing) [15:57] <happarappa> durandal_1707: That is confusing. [15:58] <durandal_1707> what is confusing? [16:01] <happarappa> "encoders is file that document encoders present in ffmpeg" [16:01] <durandal_1707> ok full name is doc/encoders.texi [16:02] <happarappa> Sigh... [16:02] <happarappa> Are people getting more vague and confusing? [16:02] <happarappa> It feels like it to me. [16:05] <durandal_1707> if you are not interested in helping you should not care at all for "confusing lines" [16:06] <durandal_1707> actually the easiest way for you to help is filling bug report for anything you feel should be improved/documented/etc... [16:07] <happarappa> I don't understand what you are saying. [16:07] <happarappa> That's why I ask. [16:07] <retard> if you have a general feeling of other people getting more vague and confusing the root cause might not be the other people [16:07] <happarappa> Or it might actually be just that. [16:08] <retard> or it might not [16:08] <retard> like i said [16:08] <happarappa> But it is. [16:08] <retard> not necessarily [16:08] <happarappa> Not necessarily, but it is. [16:10] <durandal_1707> funny [16:13] <retard> it is... but is it really? [16:50] <D0minat0r> Got so many errors when i try to use ffmpeg ;< [16:50] <D0minat0r> isnt it just to apt-get install ffmpeg [16:58] <D0minat0r> configuration: --extra-version='4:0.7.6-0ubuntu0.11.10.3' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static [16:59] <D0minat0r> did i miss'd something or wtf? ;< [17:05] <D0minat0r> cmon.... someone? [17:33] <D0minat0r> still no one here? [17:49] <Youka> ask in stackoverflow, ffmpeg forum or google for it when nobody here can help you [17:50] <burek> D0minat0r probably because [17:51] <burek> you didnt read the topic [17:51] <burek> and also didnt provide log [17:51] <burek> -ffmpeg command +configure command [18:07] <WillieNL> ola i wonder what happend with libva ? [18:07] <WillieNL> original project host seems to be offline [18:07] <burek> which host [18:08] <WillieNL> burek: http://www.splitted-desktop.com/~gbeauchesne/libva/ [18:09] <burek> did you try google [18:09] <burek> for example http://cgit.freedesktop.org/libva/ [18:09] <WillieNL> yes found via other sources but unsure if the project is still maintained [18:09] <burek> btw, isn't there some kind of irc channel for that or ML or something [18:09] <burek> i mean, is libva related to ffmpeg in some way or what [18:10] <WillieNL> dunno ffmpeg costh me half my day here due a old bug in old release... [18:10] <WillieNL> ffserver from 0.6.x -> Aspect ratio mismatch between encoder and muxer layer [18:10] <WillieNL> known issue [18:10] <burek> oh man... 0.6... [18:11] <WillieNL> well before we argue i only need to test something on this old rig [18:11] <WillieNL> somehow ffserver is having a issue wich made update ffmpeg more usefull then reinstall a complete machine [18:11] <burek> well i can understand you need that exact version, i just feel sorry, knowing what all does it bring with itself :S [18:11] <WillieNL> well it sucks if need to debug a remote rtmp host like this yes [18:11] <WillieNL> :S [18:12] <WillieNL> well it do work but no meta data [18:18] <WillieNL> burek: do ffmpeg hold old releases on a ftp or some like that ? [18:18] <D0minat0r> I did try everything that the wiki said, still nothing so [18:19] <D0minat0r> Got alot of errors thar i havent enable everything [18:21] <burek> WillieNL, what do you mean by "old releases" [18:21] <burek> you do use git, right? [18:21] <burek> D0minat0r, right above what I already answered you earlier [18:21] <WillieNL> -> ffmpeg-0.6_p28715.tar.bz2 [18:21] <burek> read that [18:22] <burek> WillieNL, when you type ffmpeg in your shell [18:22] <burek> what is the version string that you get in return [18:22] <WillieNL> FFserver version SVN-r25767, [18:22] <burek> oh, i see.. well, it's just too old [18:23] <burek> i was hoping it was a version after the transition to git [18:23] <burek> so you could simply checkout that version [18:23] <burek> wait, let me ask other people [18:23] <WillieNL> burek: this is my problem -> http://roundup.libav.org/issue2467 [18:24] <D0minat0r> burek http://pastie.org/6044359 [18:24] <D0minat0r> that what i got [18:24] <burek> D0minat0r, you are not using ffmpeg [18:24] <burek> you are using a fork of ffmpeg [18:24] <burek> and it's a very old one [18:25] <burek> either ask your question in a support channel for that fork project [18:25] <burek> or use real ffmpeg [18:25] <D0minat0r> Oh, shit, sry [18:25] <D0minat0r> apt-get install ffmpeg was the wrong one then [18:26] <burek> yes it was [18:27] <D0minat0r> Sry, [18:27] <burek> there was an issue with ubuntu/debian community which they didnt properly resolve [18:27] <burek> causing all that mess you are experiencing right now [18:27] <D0minat0r> Oh i see, so i have to build it from the source? [18:27] <beastd> D0minat0r: Debian does not have FFmpeg anymore because the maintainer of FFmpeg Debian packages is one of the forkers. [18:28] <D0minat0r> But i use ubuntu server [18:28] <burek> ubuntu is based on top of debian [18:28] <beastd> it is related. but look here [18:28] <beastd> D0minat0r: http://ffmpeg.org/download.html [18:28] <burek> WillieNL, good new [18:28] <burek> s [18:28] <burek> you can find a corresponding git version based on your svn version [18:28] <D0minat0r> Hm i see [18:29] <D0minat0r> But do you guys know any other program like ffmpeg? [18:29] <D0minat0r> I just need to make samples [18:29] <burek> if you check out the official ffmpeg's git log, you'll see something like this in one of the commits: "Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk" [18:29] <burek> which happens to be this one [18:29] <burek> 47d2ddca802f4c1bc4b454c5ac40f06f79b740a0 [18:30] <WillieNL> burek: ahh but if can upgrade to 0.7.1 then also ensured not bumping into the same bug i hope while unsure if can play/stream/transcode mp4 without h264 [18:30] <D0minat0r> Ah okey [18:31] <burek> WillieNL, using git you can always move forward/backward with different ffmpeg versions and test your issues [18:31] <burek> with tar.bz2 file, you just have one version [18:32] <WillieNL> burek: jup but gentoo is not always using git in theire ebuilds [18:32] <burek> D0minat0r, it's actually very easy to compile your own ffmpeg [18:32] <burek> did you check out our wiki for example [18:33] <WillieNL> burek: manual plugin versions in a outdated gentoo setup can be tricky :) [18:33] <D0minat0r> Yepp, tried this one: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [18:33] <D0minat0r> Didnt work [18:33] <WillieNL> but thanks do have a running build gona get a smoke and look if can make the ffserver functional [18:34] <burek> D0minat0r, what exactly didnt work [18:34] <WillieNL> ubuntu and compilation sounds wrong :S [18:34] <burek> WillieNL ok :) good luck with all that [18:35] <WillieNL> ffserver version 0.7.1 ffserver.conf:22: Unknown VideoCodec: libx264 [18:35] <WillieNL> well looks much better :) [18:35] <WillieNL> Mon Feb 4 18:35:40 2013 FFserver started. [18:35] <WillieNL> ;-) [18:54] <JEEBsv> ls [18:54] <JEEBsv> ugh, did the usual [19:27] <quup> hi, can I export all frames as PNG back-to-back to the same (fifo) file? [19:29] <quup> or just to stdout [20:34] <DimStar> Good evening... do you know of issues with ffmpeg when playing .wma / .wmv files, as described in http://trac.videolan.org/vlc/ticket/1817 (the 'solution' seems to be to revert to ffmpeg 1.0) [20:51] <juanmabc> http://www.youtube.com/watch?v=BHPzjpojlz0 [20:52] <juanmabc> mistery intended [20:52] <shevy> hmm can .lpd files be converted via ffmpeg? [20:52] <shevy> ik.lpd: Invalid data found when processing input [21:11] <llogan> DimStar: can you duplicate the issue in current ffmpeg? one of the samples in that report sounds fine to me with ffplay. [21:15] <DimStar> llogan: I tried with vlc 2.0.5 linked against ffmpeg 1.1.1 (which is the latest release) [21:15] <WillieNL> burek: well quick testing looks like even working :) but no audio due my own misconfig somewheire so i have something todo :p [21:46] <llogan> DimStar: if it worked for me in ffplay then i would guess it is a VLC issue (unless the file was created by ffmpeg and it also does not work in WMP). [21:46] <DimStar> llogan: seems to be the interaction VLC / ffmpeg 1.1... whereas it was fine with 1.0 [22:09] <veleno> hello. for a research project, i'm trying to convert an input video file into so that i have several versions of it at various bitrates. i need the output files to have a constant bit rate. [22:11] <veleno> now, this is the coomand-line that i use http://pastebin.com/k9pUH1cR and then i play with VLC to see what the media information statistics report [22:27] <relaxed> veleno: for cbr you need to set bitrate, minrate and maxrate to the desired bitrate [22:28] <veleno> relaxed: are those encoder-specific options ? [22:28] <veleno> relaxed: i was using already " -x264opts bitrate=256:vbv-maxrate=256:vbv-bufsize=166 " [22:29] <relaxed> -b:v 256k -minrate 256k -maxrate 256k [22:29] <relaxed> I guess you can do it with -x264opts too [22:35] <veleno> so the muxrate is something not related [22:35] <relaxed> correct [22:42] <blitz> if I wanted to encode something to vbr mp3 can I do that through ffmpeg? I'm trying to use subsonic and transcode anything that isn't mp3 over to v0, right now it uses ffmpeg -i %s -ab 320k -v 0 -f mp3 - where %s is the filename [22:44] <blitz> or do I need to use lame [22:53] <veleno> mmm does ffmpeg also provide some tool to check the bitrate (average, instantaneous) of a given video file ? [23:40] <brontosaurusrex> would vc1 in mkv be something common/supported, or ....? [23:46] <JEEB> brontosaurusrex, not sure if VC-1 has an official mapping in matroska [23:47] <brontosaurusrex> hmm, mkvmerge just removes a ts to mkv with vc1 with no hickups, lets see if my tv can play those ... [23:47] <brontosaurusrex> removes = remuxed [23:48] <JEEB> don't find it in http://www.matroska.org/technical/specs/codecid/index.html at least [23:48] <JEEB> wouldn't stop it from muxing it some other way of course [23:49] <FergusL> Hi, I'm on Ubuntu, trying to compile something coded on Debian, I'm running into an error with libavutil-dev... which is not being updated on Ubuntu in favor of Libav. What could I do to compile ? [23:51] <veleno> is it possible to tell ffprobe to use the same size unit for all the frames? sometimes it uses byte, others Kibyte [00:00] --- Tue Feb 5 2013

1 0

ffmpeg-devel.log.20130203
by burek 04 Feb '13

04 Feb '13

[00:43] <Compn> whats this image_to_j2k thing in the libopenjpeg patch ? [00:43] <Compn> is it another project? [00:43] <Compn> is it lgpl ? [00:45] <Compn> oh i guess its not much code [03:12] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:0cecaa9e2edf: ffmpeg: free attachments, fix memleak [09:43] <burek> are these two image-based? dvb_teletext, dvb_subtitle [09:51] <nevcairiel> dvbsubs are bitmaps, teletext is text based [09:52] <burek> ok, thanks [09:53] <burek> just one more thing, is it possible to copy dvb_subtitle from a random input file into another output file (with completely different video) ? [09:53] <burek> assuming that it makes sense to do such thing [09:53] <nevcairiel> i suppose thats possible [09:54] <burek> cool :) [09:54] <burek> did i mention lately that ffmpeg is great? :) [10:42] <wm4> what's with fate [11:20] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:c91f1f3f926d: ffplay: drop redundant NULL sws_freeContext() check in stream_close() [11:20] <cone-9> ffmpeg.git 03Carl Eugen Hoyos 07master:90020d7f4bf2: lavf/segment: use correct spelling and value for EXT-X-ALLOW-CACHE tag [11:50] <DEATH> Is --disable-mmx meant to disable all SSE etc optimizations too? Because --enable-runtime-cpudetect --disable-mmx apparently means that no CPU specific optimizations are in use [11:51] <nevcairiel> its not necessarily meant to do this, but in practive it sometimes does [11:51] <nevcairiel> practice* [11:51] <DEATH> Also my build only needs audio decoders, build issues that I've been having affect video specific code, configure option to disable all video code could be nice (smaller output). [11:52] <Skyler_> I think --disable-mmx disables all mmx-and-above [11:52] <beastd> DEATH: did you look at current configure options? you should be able to disable most video stuff [11:52] <DEATH> Yeah, I do --disable-everything then enable specific decoders I need [11:53] <DEATH> I got this yesterday: [11:53] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: error: 'gmc_mmx' undeclared (first use inthis function) [11:54] <DEATH> As far as I can tell it's for motion compensation, so while I don't really have clue about FFmpeg internals, it does not concern my usage case as I'm using FFmpeg to decode AAC+MP3+Vorbis audio only. [11:55] <DEATH> Tried to remedy by --disable-mmx but that disabled more stuff than I'd like to disable. [11:55] <beastd> DEATH: ok, i see. [11:57] <DEATH> With all instruction set extensions disabled, FFmpeg is only as fast as most of the competing decoders :) [11:57] <DEATH> (On MP3/AAC/Vorbis audio) [11:57] <beastd> DEATH: your problem looks like a bug to me. so it probably should be fixed. [11:58] <DEATH> OK, thanks for acknowledging [11:58] <beastd> maybe the assignment at dsputil_mmx.c:2309 needs just more #if protection ? [12:00] <beastd> DEATH: AFAICT the sse init has "&& CONFIG_VIDEODSP" . would that fix your compile problem? and hopefully run too ;) [12:02] <beastd> i am just poking around though. i do not know this code well at all. so please do not take my suggestions too serious. [12:05] <saste> beastd, are you going to work on kerndeint? [12:05] <saste> ubitux mentioned that you wanted optimize it, so we can drop mp=kerndeint [12:05] <beastd> saste: i think this is a misunderstanding. i have shown no interest in kerndeint [12:06] <saste> beastd, ok [12:06] <beastd> saste: an lavfi version was posted? it was slower thant mpcodecs wrapper? [12:06] <saste> i think i'll wait for ubitux then, it's not anything like high priority to get rid of mp=kerndeint [12:06] <beastd> agree [12:06] <saste> beastd, check the archive [12:06] <saste> kerndeint is already in [12:07] <beastd> saste: oh ok. sorry, for my ignorance. [12:10] <DEATH> Hmm yes looks like it's missing CONFIG_VIDEODSP check in the #if line, compiles fine after adding it: [12:10] <DEATH> #if (ARCH_X86_32 || !HAVE_YASM) && CONFIG_VIDEODSP [12:10] <DEATH> c->gmc = gmc_mmx; [12:10] <DEATH> #endif [12:10] <DEATH> Thanks for the input. [12:11] <beastd> DEATH: ok, i will sent on ffmpeg-devel list for discussion/approval [12:12] <beastd> thanks for reporting and testing [12:15] <michaelni> burek, can you do something about the FATE-IDLE logger thing ? [12:15] <michaelni> its 99% nonsense ATM [12:15] <michaelni> we have only 1 client (haiku) that actually is stuck [12:16] <michaelni> well, was stuck, i hope it works now again [12:17] <michaelni> the duron box needs >2days for a cycle, blame valgrind & amd if you like but it is that slow it was not stuck [12:19] <nevcairiel> imho it should also say the names of boxes which are newly added to the stuck list, the number is not really useful [13:14] <cone-9> ffmpeg.git 03James Almer 07master:d1511c02f49e: proresdsp: Fix checkheaders [13:14] <cone-9> ffmpeg.git 03James Almer 07master:13eb9fcf5686: build: Remove superfluous MAKE variable for the build suffix [13:55] <beastd> DEATH: Just found out that the problem you described was fixed in exactly that way on Fri Jan 25 by Carl Eugen. So no problem with current git master. [13:55] <beastd> I was actually looking at an older version because I did not notice I was on a topic branch... [16:11] <saste> michaelni, how do we handle mid-stream frame parameter changes before feeding the filtergraph in ffmpeg? [16:11] <saste> or don't we handle them at all? [16:14] <wm4> saste: would it be possible to let the lavfi glue code reconfigure the immediate following part of the filter chain if the frame fed to a filter differs from the config of the previous frame? [16:14] <wm4> or would that be too complex, or impossible? [16:17] <saste> wm4: http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/153146 [16:20] <wm4> saste: I'd say always cal config, and filters which can handle the change don't reset themselves - does this make sense? [16:20] <wm4> *call [16:21] <michaelni> saste, also see -reinit_filter [16:21] <saste> wm4, please comment on the thread [16:21] <wm4> marking filters with bit flags seems fragile and complex [16:21] <saste> wm4, that's why it is an RFC [16:36] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:cb573f7fbca1: avcodec/x86: Add daniels copyright to the recent gcc->yasm convertions he did. [16:46] <wm4> how do you know if new metadata has been added during streaming? [16:52] <JEEB> ubitux, would you have interest in looking at the source code of a renderer for the ARIB B-24 captioning system? :D https://dl.dropbox.com/u/175558/screenshots/tvtest_caption_lol_positioning.… [16:58] <wm4> how's the source? is it full of japanese comments? [17:02] <JEEB> haven't really looked at it yet, but it consists of a captions parsing library (that's used by caption2ass as well f.ex.), and then the actual renderer [17:02] <JEEB> I would guess Japanese Shift-JIS comments tho :) [17:02] <wm4> lol [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:6e32b377bbb5: ffplay: remove pts_ptr argument from audio_decode_frame() [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:02af4e9a97cc: ffplay: rename audio_decode_frame() variable "pts" to "audio_clock0" [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:c44281906a87: ffplay: remove misleading comment from audio_decode_frame() [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:1897109c00a0: lavfi/pad: add support to named options [17:09] <cone-9> ffmpeg.git 03Dave Rice 07master:3bba91d32d86: doc/syntax: fix frame rate documentation [17:09] <cone-9> ffmpeg.git 03Dave Rice 07master:fe63f3d301cc: doc/syntax: add missing frame sizes [17:34] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:4d37d2bfc53c: put_vp_no_rnd_pixels8_l2_mmx: fix type [17:57] <durandal_1707> divVerent: mutt messed up patch somehow, you should use git send-mail instead [17:59] <wm4> so I heard mp4 file always have a perfect index that contains the timestamp for each frame (I don't know mp4,s o this might be wrong) [18:00] <wm4> doesn't that mean libavformat could set the duration for each packet exactly? [18:02] <nevcairiel> there is still complications with frame reordering [18:02] <wm4> oh, right [18:02] <nevcairiel> but it does provide some duration, not sure where it comes from in mp4 [18:03] <wm4> ah well, I'm mostly interested in an API that doesn't force the player to go haywire when video frames are very sparse [18:03] <wm4> like it happens with some "slide" videos containing audio and mjpeg frames [18:03] <wm4> the duration field seems to be mostly unreliable, whatever it is [18:04] <nevcairiel> most containers just have timestamps, not duration info [18:08] <cone-9> ffmpeg.git 03Rudolf Polzer 07master:773fc6e0c712: lavfi/showspectrum: set default height to 512 [19:42] <divVerent> 17:57:01 @durandal_1707 | divVerent: mutt messed up patch somehow, you should use git send-mail instead [19:43] <divVerent> I cannot add my own text when using git-send-email [19:43] <divVerent> also, it'd use the wrong from address [19:43] <divVerent> next time I'll just attach the patch as a file then [19:48] <divVerent> durandal_1707: wtf... [19:48] <divVerent> [rpolzer@grawp ffmpeg (git)-[master]-]$ git apply ../0001-lavfi-showspectrum-display-multiple-channels-in-sepa.patch [19:48] <divVerent> error: patch failed: Changelog:10 [19:48] <divVerent> git-am also fails it [19:50] <durandal_1707> changelog changed [19:50] <durandal_1707> the @ somehow become at in your patch .... [19:56] <durandal_1707> saste: i sent new patch, anxiously expecting your fine review [20:00] <saste> durandal_1707, thanks [20:00] <divVerent> 19:50:19 @durandal_1707 | the @ somehow become at in your patch .... [20:00] <divVerent> durandal_1707: very sure mutt didn't do that [20:00] <divVerent> maybe the mail archive did? [20:01] <divVerent> durandal_1707: ah, thanks... this solved the missing link [20:01] <divVerent> didn't know how to make a named choice [20:02] <divVerent> and yes, I can still fully sign off on the new patch, great work [20:08] <saste> bah ffplay -af is growing complicate... [20:11] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:ccfd8cffe867: qdm2: Fix data type used in multiplication. [20:11] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:d106679fb526: qdm2: disable superblocktype_2_3==0 code [20:14] <michaelni> burek, i suggest you disable the FATE-IDLE thing until its fixed, its of no use like this [20:15] <michaelni> it just prints a random meaningless number every hour [20:23] <Daemon404> \o/ someone broke ffmpeg's configure [20:23] <Daemon404> in such a suble, hidden way [20:23] Action: Daemon404 guess he does a git bisect [20:30] <durandal_1707> what got broken in configure? [20:33] <Daemon404> every single math check function [20:33] <Daemon404> e.g. lrint [20:33] <Daemon404> instea of: foo(floata,floatb) [20:33] <Daemon404> int main() { return (int) foo(...): } [20:33] <Daemon404> it has return 0 [20:33] <Daemon404> which does fuck all [20:34] <Daemon404> if the compiler has DCE [20:34] <Daemon404> and i have a feeing the breakage is hidden in a merge commit... [20:34] <Daemon404> behind wallas of shit [20:35] <wm4> lol [20:36] <wm4> maybe ffmpeg devs should send patches to libav instead to make merging easier [20:36] Action: wm4 runs [20:41] <divVerent> 20:08:33 @saste | bah ffplay -af is growing complicate... [20:41] <divVerent> saste: yes [20:42] <divVerent> ffplay -i lavfi 'amovie=filename.mp3, showspectrum' [20:42] <Daemon404> what if i have a file named lavfi? ;) [20:42] <divVerent> we also need a better way for this eventually... preferably by (later) getting rid of ffplay's own copy [20:42] <divVerent> sorry, -f lavfi ;) [20:42] <Daemon404> :P [20:42] <saste> divVerent, not that convenient [20:43] <divVerent> I agree [20:43] <divVerent> but that's lavfi [20:43] <divVerent> that's something to be fixed separately [20:43] <saste> ffplay IN -af volume=+2dB [20:43] <divVerent> yes, but an -af cannot create a video stream... or can it? [20:43] <saste> divVerent, no [20:43] <divVerent> then maybe that's what is missing ;) [20:45] <divVerent> ffplay filename.mp3 -filter_complex "[0:0] asplit [a] [va]; [va] ashowspectrum [v]" -map "[a]" -map "[v]" [20:45] <divVerent> would be great ;) [20:45] <divVerent> in other words: "consistency with ffmpeg" [20:46] <divVerent> at least that can be hidden in a shell alias then [20:51] <saste> divVerent, that's correct in principle [20:52] <saste> but it's hard to implement in practive, because of the high inconsistencies between the tools [20:53] <saste> ffmpeg/avconv has lots of specific hacks which have never been ported to the other tools [20:54] <teratorn> wm4: i think its pretty clear whos making whos job harder, as an outside observer of sorts :)) [20:54] <wm4> teratorn: the interesting question would be whether libav makes merging intentionally harder [20:55] <teratorn> my answer is to care less i think :/ [20:55] <teratorn> and also to maybe put pressure on debian to repackage ffmpeg [20:55] <teratorn> since that is just silly [20:56] <divVerent> saste: I know [20:56] <divVerent> that this is the issue [20:56] <divVerent> sort of like mplayer/mencoder, just other way round [20:57] <divVerent> ffplay has become the neglected stepchild [20:57] <divVerent> to a point where I wonder if it'd be better to make ffmpeg learn what ffplay can do - i.e. seek [20:57] <divVerent> anything else already can be done by lavfi :P [20:58] <wm4> wut, so video playback with ffmpeg+lavd? [20:58] <divVerent> i.e. if the console, and the video out window of the "sdl" output device would take keyboard input [20:58] <divVerent> and allow seeking [20:58] <divVerent> then basically ffplay could be killed [20:59] <divVerent> oh, possibly AV sync too, possibly in communication between video and audio driver... that sounds ugly ;) [20:59] <divVerent> but MAYBE both video and audio out syncing to pts would work well enough (for audio this is tricky, though :P) [20:59] <wm4> how would you do user input and seeking? [21:00] <divVerent> that's the thing I don't know [21:00] <divVerent> currently both ffmpeg and lavfi movie source lack seeking support [21:00] <divVerent> it's just wishful thinking [21:00] <saste> wm4, input and seeking: you program events (e.g. in lua) which are sent to the filtegraph as events [21:00] <divVerent> yes [21:00] <divVerent> THAT half is the easy half ;) [21:00] <wm4> divVerent: avf_videoplayer? xD [21:00] <divVerent> we already have this event handling [21:00] <divVerent> as in, sending commands to the filter graph [21:01] <saste> divVerent, in the meanwhile i'll add -af to ffplay ;-) [21:01] <wm4> I'd be careful with this commands and filters thing [21:01] <divVerent> sure :P [21:01] <divVerent> not a bad idea either way [21:01] <wm4> that'll become a mess in no time [21:01] <divVerent> just, in the long run, PROBABLY ffplay should be killed [21:01] <divVerent> or maybe replaced by a hardlink to ffmpeg that detects how it should output by binary name ;) [21:02] <divVerent> but talking the REALLY long run here [21:02] <saste> and i'm still awaiting for glplay (or in general an opengl output device) [21:02] <wm4> divVerent: actually I'm glad if "they" can't change their own APIs so easily (due to the code duplication in ffmpeg/ffplay) [21:02] <divVerent> saste: you SORT of could have it by changing the API from SDL 1.2 o SDL 2 ;) [21:03] <divVerent> but... don't [21:03] <wm4> saste: that'd be a good thing, since SDL sucks for video [21:03] <divVerent> I know how to use SDL 2 so it actually works [21:03] <divVerent> but it was tricky [21:03] <divVerent> well, actually, SDL 1.2 also has video surface support [21:03] <divVerent> but by Xvideo [21:03] <wm4> saste: you could also display all sorts of pixfmts natively (think 10 bit 444...) [21:03] <saste> divVerent, what's the status of libsdl2? [21:03] <divVerent> while SDL 2.0 throws that away and introduces a simplified common 2D interface for OpenGL and D3D [21:03] <divVerent> can't recommend it yet [21:03] <saste> what would be the gain for switching from 1.2 -> 2.0 [21:04] <divVerent> I implemented sdl2 output for mpv, and it works [21:04] <divVerent> but it was tricky [21:04] <divVerent> OpenGL output, mainly [21:04] <divVerent> basically, if you DO use sdl2, you WILL hit bugs [21:04] <saste> the last time I tried libsdl dev tree it was buggy as hell [21:04] <divVerent> exactly [21:04] <wm4> they're still changing APIs [21:04] <divVerent> yes, but not much any more [21:04] <divVerent> I am reading the ML [21:04] <divVerent> but, yes, I can't recomemdn it yet [21:05] <divVerent> I will recommend SDL2 as only output device for ffmpeg the day SDL2 enters Debian stable [21:05] <divVerent> and that's still like 2, 3 years in the future [21:05] <wm4> lol [21:05] <divVerent> I assume SDL2 will be stable by end of the year [21:05] <wm4> so long [21:05] <divVerent> given it's now being pushed by app developers for iOS, Android [21:05] <wm4> they sure are taking their time [21:05] <divVerent> indeed [21:06] <divVerent> if the iOS/Android guys wouldn't be already using it for apps they sell [21:06] <divVerent> we could assume SDL2 will NEVER finish [21:06] <divVerent> you know, SOMEONE has to push them ;) [21:06] <divVerent> saste: also, as for SDL2 bugs [21:06] <divVerent> the HW accelerated outputs are mostly bugfree [21:06] <wm4> hm does that mean they can use native code easily on android? [21:06] <divVerent> i.e. opengl, direct3d [21:06] <divVerent> but the SW output... will cause nightmares ;) [21:06] <divVerent> if you REALLY want to try tzhat [21:06] <divVerent> look in mpv source for the APIs I used, and use ONLY these [21:07] <divVerent> wm4: "sort of" ;) [21:07] <divVerent> not sure if it's for rooted devices only [21:07] <wm4> I remember point scaling... [21:07] <divVerent> just know people ARE making money with it [21:07] <divVerent> wm4: that's the missing features part [21:07] <divVerent> but the bugs when you tell SDL a texture is "static!" [21:07] <divVerent> and it then converts the alpha channel to 1bit RLE... [21:07] <divVerent> that's awful [21:08] <divVerent> that's why all textures in mpv are marked and used as dynamic ones [21:08] <divVerent> even the OSD ones which COULD be seen as kinda static [21:08] <divVerent> given they tend to update only once per sec [21:08] <divVerent> (IMHO that's already static enough :P) [21:09] <divVerent> note: these bugs ONLY happen with the SW renderer. Nevertheless awful. [21:09] <divVerent> these bugs ALSO quite likely affect the old SDL_Surface API too [21:09] <divVerent> wonder if SDL 1.2 has these bugs too ;) [21:09] <wm4> divVerent: the worst for video output, though, is libva [21:09] <divVerent> given the SW renderer is just a wrapper for SDL_Surface [21:09] <wm4> well, at least for OSD [21:10] <Daemon404> oh fuckin boy.... [21:10] <Daemon404> the configure / return 0 problem only happens on msys [21:10] <Daemon404> FML [21:11] <Daemon404> and only in ffmpeg [21:16] <durandal_1707> so when new fate look will finally come in? [21:18] <michaelni> Daemon404, theres no return 0 in ffmpegs check_mathfunc [21:19] <Daemon404> yet someone it ends up there [21:19] <Daemon404> it's mind boggling [21:19] <Daemon404> somehow* [21:22] <durandal_1707> so any new filter comming in? [21:23] <saste> durandal_1707, yours ;-) [21:24] <saste> you're getting a filter addicted [21:24] <saste> who's your favourite (git-)pusher? [21:26] <durandal_1707> i dont have favorite (git-)punisher [21:32] <durandal_1707> to download vapoursynth source code i need install svn - no way [21:33] <wm4> use git-svn? [21:33] <durandal_1707> that needs svn too [21:34] <wm4> but you don't need to use svn directly [21:35] <Daemon404> you see, durandal_1707 uses an antiquated OS called BSD [21:35] <Daemon404> which uses an antiuated system called ports [21:35] <Daemon404> and this to install svn he needs to invest time [21:35] <Daemon404> compiling... [21:35] <Daemon404> compiling.... [21:35] <Daemon404> compiling..... [21:35] Last message repeated 1 time(s). [21:35] <Daemon404> etc [21:35] <durandal_1707> ^troll [21:36] <durandal_1707> there are nice and smart people with clones on github [21:36] <Daemon404> none of which are up to date i suspect [21:36] <durandal_1707> what, why this depens on lav* stuff? [21:37] <Daemon404> what? [21:37] <wm4> lav* actually contains a useful video decoder [21:37] <durandal_1707> i read INSTALL [21:37] <Daemon404> vapoursynth uses swscale [21:37] <durandal_1707> ignores lavfi [21:37] <Daemon404> i think i just died of laughter. [21:37] <durandal_1707> should be perished into vapour [21:39] <durandal_1707> now i need to install waf.... [21:39] <wm4> ouch [21:39] <durandal_1707> this is just too complicated for me.... [21:48] <durandal_1707> michaelni: is it really true that S32 when going to/from planar is converted to float? [21:48] <durandal_1707> that does not make any sense... [22:03] <michaelni> durandal_1707, no silly convertions should be done, in what case exactly does this happen? [22:04] <durandal_1707> dunno, that i read from nicolas reply about reordeing in lavd.... [22:23] <durandal_1707> saste: actually i need to do this from my histogram filter: http://avisynth.org/oldwiki/index.php?page=histogram [22:30] <saste> durandal_1707, vectorscope? [22:33] <durandal_1707> all [23:02] <cone-9> ffmpeg.git 03Dave Rice 07master:71956371a7bb: lavu/parseutils: add digital cinema frame sizes [23:17] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:cf7b71b0b812: doc/filters: reformat various filter tables items [23:26] <burek> michaelni, what should i fix about the fate-idle? [23:26] <burek> do you want to have the names of machines always displayed or something? [23:27] <nevcairiel> personally i think it would be great if it just names the machine which was just added to the list once, and doesnt list all machines all the time [23:31] <burek> hmh, i could make a list i guess, but i think it would be easier to have idle time defined for each machine [23:31] <burek> for example, the default of 24h would be applied to each machines except those configured with a different idle value [23:32] <burek> that way it wouldnt spam too much [23:32] <burek> and it would always provide a list of all stuck machines, so you don't have to scroll back and calculate which one is new in the list and if previous ones in the list got unstuck or not, etc [23:40] <burek> also, which one of these is that "duron box" ? I see a lot of durons in the list, so I don't know for which one to set the idle timeout to > 48h [23:41] <michaelni> burek, all that say duron [23:41] <michaelni> they all run on the same box [23:41] <michaelni> thats part of the reason why it takes lots of time to finish a cycle [23:42] <burek> i see [23:42] <burek> ok, how many hours should i set for timeout then? 72 hrs perhaps? [23:43] <michaelni> yep [23:44] <michaelni> or maybe you can automate it [23:44] <michaelni> like using 24h+average past update time or so [23:46] <burek> i'll try first 72hrs, just in order not to complicate things if i dont need to [23:51] <saste> nevcairiel, why RGB -> YUVJ in the showspectrum patch? [00:00] --- Mon Feb 4 2013

1 0

ffmpeg.log.20130203
by burek 04 Feb '13

04 Feb '13

[00:28] <Youka> could it be that ASS subtitle decoding is buggy? i tried it, but it always skip first 15 lines [00:29] <Youka> https://github.com/Youka/NyuFX/blob/master/src/lva.cpp#L493 [01:46] <taylanub> Today I compiled from git and ffplay (and mplayer) give corrupted output. ffplay error output: http://sprunge.us/gOBS [01:46] <taylanub> Weird thing is, libav has the same issue (tested with mplayer, since libav has no ffplay). [03:09] <pietro10> is ther ea way I can capture the audio from either my speakers only (not the mic) or from a specific program? -f alsa -i pulse records from my mic. Thanks. [03:09] <klaxa> pietro10: open pulseaudio volume control and move the record stream to the monitor of your soundcard [03:11] <pietro10> I could do that but I mean with ffmpeg, so I don't have to do that each time if I actually want to use my mic [03:14] <klaxa> you could script it with pacmd or something, i think there is a way to choose that with ffmpeg itself though i don't remember it [03:25] <pietro10> ok thanks [03:59] <blue_misfit> anyone have experience with Smooth Streaming? [06:05] <makara> I want to check that my m4a's aren't ALAC. Why doesn't this work: ffprobe *.m4a | grep alac [06:20] <drv> makara: ffprobe output is on stderr, you could use e.g. ffprobe *.m4a 2>&1 | grep alac [06:23] <makara> drv, could you please explain "2>&1" [06:23] <drv> it redirects stderr to stdout [06:33] <makara> drv, it doesn't work [06:33] <makara> it will output on screen, but can't pipe to grep or a file [06:42] <makara> drv, it works, thanks [14:18] <paste> hi, is there an ffmpeg player? reason is when using yadif with mplayer clearly field 1 is incorrect, but when encoding with ffmpeg I get jutters regardless if I choose top (0) field or bottom (1) field. Or I'm DIW? [14:19] <saste> paste: ffplay [14:19] <paste> cool thanks. [14:20] <paste> hmm. [14:20] <paste> looks like its not available. [14:21] <paste> ugh [14:21] <paste> nvm [14:21] <paste> bed time. [14:21] <paste> thanks. [14:30] <jastr> hi.. can anyone help me with a raw video format issue? [14:30] <jastr> when i run this ./ffmpeg -i MyStream_0.f4v -vb 1200k -f rawvideo -t 00:05:00 -pix_fmt yuv420p -s 832x468 - | ./ffmpeg -f rawvideo -r 30 -pix_fmt yuv420p -s 832x468 -crf 0 -i - out.mp4 the output keeps being in 200kb/s i've tried -qscale and -b but either helps [14:31] <jastr> now it detects the input to be 1200kb after i set the -vb on the raw input but the output it writes is always approx 260kb/s no matter what option i give the output [14:32] <jastr> here is the pastebin http://pastebin.com/hz27cXEj [14:33] <spyvingen> anyone know how to set swe as first choice in winff and if swe dont exist then eng should be choosen? [14:34] <spyvingen> audio that is [14:35] <Mavrik> jastr: why are you trying to set bitrate on input? [14:35] <Mavrik> jastr: input has a bitrate of what it has, forcing anything won't change things [14:35] <Mavrik> plus... why two ffmpeg processes? [14:36] <Mavrik> what are you trying to achieve? [14:36] <jastr> well when i didn't use the -vb it detected the input at 116812 [14:37] <Mavrik> it doesn't matter [14:37] <Mavrik> you can't change what you already have :P [14:37] <jastr> Mavrik i have a movie that looks fine when played back in quicktime but it's a movie recorded in 25fps on a 30fps canvas so it looks a bit stuttering when played back in flash. So what i wanted was ffmpeg to kind of playback the movie and reencode the video stream. If i reencode from file to file in ffmpeg either at 25fps or 30fps it's the same. [14:38] <jastr> i know but the input is 1200kb [14:39] <Mavrik> again, last time: detection doesn't friggin matter [14:39] <Mavrik> the decoder reads stream as it is [14:40] <Mavrik> plus, you're probably looking at totally wrong output anyway [14:40] <Mavrik> now. [14:40] <Mavrik> WHY are you using 2 ffmpeg processes that essentially do the same a simple one-process command line would do? [14:40] <Mavrik> also, crf 0? losless? [14:40] <jastr> thought i had to. that's how i did it in the old days. [14:41] <Mavrik> hmm :) [14:41] <Mavrik> jastr: so if I understand this correctly, you're just trying to convert 25fps video to 30 fps video? [14:41] <jastr> the reason for the two processes was that i just wanted ffmpeg to playback the movie and recapture it hoping to get a little more smooth result [14:42] <jastr> no i'm trying to smooth it. It is 25fps and should be 25fps BUT the program that encoded the file was working with 30fps so it's not smooth when it's played back in flash but quicktime handles the file fine [14:44] <Mavrik> I see. [14:44] <Mavrik> well, your command line won't have the result you seek [14:44] <jastr> i see.. :D any other commandlines that could have? [14:45] <Mavrik> it's equivalent of just reencoding video with parameters with added overhead of piping :) [14:45] <spyvingen> is it possible to choose what audio languege to choose [14:45] <spyvingen> *to rip [14:45] <Mavrik> jastr: well, you can try just dropping frames [14:45] <Mavrik> down to 25fps [14:45] <jastr> also when the file is 25fps? [14:46] <Mavrik> oh doh [14:46] <Mavrik> hmm [14:46] <Mavrik> now that's quite a problem [14:46] <jastr> hehe exactly.. that's what the program did and now i just want to smoothen it as much as possible [14:48] <Mavrik> ffmpeg -r 30 -i yourfile.flv -codec:v libx264 -crf 20 -codec:a copy -r 25 output.mp4 [14:48] <Mavrik> try this :) [14:55] <jastr> that dropped a lot of fps but didn't really get more smooth. more like the other way around. [14:55] <jastr> maybe -f 25 and exporting a 30fps? [14:56] <Mavrik> probably, but I'm afraid you already lost frame info :\ [14:59] <jastr> yeah i know.. that's why people who don't know about video encoding shouldn't do it.. arhh [15:34] <Guest17665> Hi I'm using this options to convert avi to mp4: [15:35] <Guest17665> "-i $INPUT -vcodec libx264 -threads 0 -r 25 -g 25 -b 500k -bt 500k -acodec libvo_aacenc -ar 44100 -ab 64k -y $OUTPUT" [15:35] <Guest17665> But I get: [15:35] <Guest17665> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [15:36] <Mavrik> Guest17665: there has to be a message actually explaining what's wrong. [15:37] <Guest17665> Mavrik: you mean, to pastebin it? [15:37] <Mavrik> well [15:38] <Mavrik> you can read it as well :P [15:40] <Guest17665> ok [15:41] <TheLink> if I set hqdn3d=:::0, does this leave the first 3 parameters at default and the fourth at 0? [15:41] <Guest17665> http://pastebin.com/k8t58KGi [15:42] <Guest17665> What are advantages of avconv vs ffmpeg [15:42] <Mavrik> *shrug* [15:43] <Mavrik> TheLink: hmm, I'm not sure that's even valid syntax [15:43] <Mavrik> TheLink: newer ffmpegs prefer hqdn3d=chroma_tmp=0 type syntax [15:43] <Mavrik> Guest17665: [libvo_aacenc @ 0x1ba5b60] Unable to set encoding parameters [15:43] <Mavrik> is seems your audio encoder doesn't initialize properly [15:44] <Mavrik> ah [15:44] <Mavrik> Guest17665: you have 5.1 audio [15:44] <TheLink> well, ffmpeg didn't throw an error but I guess your syntax is much cleaner [15:44] <TheLink> thanks [15:44] <Mavrik> Guest17665: libvo_aacenc doesn't support 5.1 audio [15:48] <Guest17665> so i should pick some other acodec? [15:48] <Mavrik> yep, you'll need another aac encoder if you want to keep 5.1 audio [16:14] <Guest17665> well what's 5.1 audio? [16:16] <TheLink> multi channel audio [16:16] <TheLink> in your case 6 channels [16:21] <Guest17665> why it says: Unknown encoder 'libfaac' [16:25] <Guest17665> I get now: [16:25] <Guest17665> [ac3 @ 0x2684e60] frame sync error [16:25] <Guest17665> Error while decoding stream #0:1 [16:25] <Guest17665> Application provided invalid, non monotonically increasing dts to muxer in stream 1: 22906 >= 22871 [16:25] <Guest17665> av_interleaved_write_frame(): Invalid argument [16:25] <Guest17665> I change acodec to libfaac [16:34] <LithosLaptop> ewww :) [16:34] <Guest17665> and what is that: [16:34] <Guest17665> encoder 'aac' is experimental and might produce bad results. [16:34] <Guest17665> Add '-strict experimental' if you want to use it. [16:34] <Guest17665> Or use the non experimental encoder 'libfaac'. [16:35] <Guest17665> literally -strict experimental ? [16:36] <LithosLaptop> the quality of the experimental aac encoder is worse than libfaac [16:36] <LithosLaptop> and libfaac aint that great [16:36] <Guest17665> I've built it with [16:36] <Guest17665> ./configure --enable-libx264 --enable-libvo-aacenc --enable-libfaac --enable-gpl --enable-nonf [16:36] <Guest17665> ree --enable-version3 [16:36] <Guest17665> why say not libfaac? [16:37] <LithosLaptop> not really transparent at 128Kbit/s for 2 channel audio [16:37] <LithosLaptop> thats why I don't use it [16:37] <LithosLaptop> even libmp3lame will give you better quality than libffac [16:38] <LithosLaptop> try to use fdkaac in ffmpeg [16:39] <LithosLaptop> libvo-aacenc is the worst btw [16:39] <Guest17665> LithosLaptop: well, I need whatever gets the job done [16:40] <LithosLaptop> --enable-libfdk_aacenc [16:40] <LithosLaptop> oh [16:41] <LithosLaptop> well I haven't really tried more than 2 channels before [16:42] <Guest17665> Unknown option "--enable-libfdk_aacenc". [16:43] <Guest17665> oh it's libfdk_aac [16:44] <Guest17665> oops not found [16:54] <taylanub> So is ffmpeg from git broken for anyone else ? Doesn't play H.264, gives corrupted video. [16:58] <Guest17665> taylanub: it isn't suppoused to play, it's for converts [16:58] <taylanub> Eh ? Doesn't mplayer use libav to decode H.264 for playing ? What does ffplay do ? [17:02] <taylanub> I'll be compiling old commits and see what commit broke it. Assuming the problem isn't elsewhere. [17:03] <Guest17665> LithosLaptop: how do I use libmp3lame? [17:03] <Guest17665> instead of aac [17:03] <LithosLaptop> -acodec libmp3lame [17:04] <Guest17665> Unknown encoder 'libmp3lame' [17:05] <LithosLaptop> strange [17:05] <LithosLaptop> maybe it is not built with it [17:07] <Guest17665> oh [17:20] <Guest17665> I cloned git://github.com/mstorsjo/fdk-aac.git how to build it? [17:22] <Guest17665> after I install libmp3lame do I rebuild avconv? [17:24] <saste> Guest17665, avconv != ffmpeg [17:25] <saste> and we can't support avconv here [17:25] <Guest17665> saste: sorry [18:18] <Zeranoe> Is qt-faststart now part of FFmpeg? If so, how do I use it? If not, are they any other ways to move the moov atom? [18:19] <JEEB> there's a movflag for that IIRC [18:19] <beastd> Zeranoe: movflags IIRC [18:20] <JEEB> movflags +faststart was it? [18:20] <beastd> "-movflags faststart", according to the fine manual [18:21] <JEEB> :) [18:24] <taylanub> OK, I'm back to some December commit, and ffmpeg works again. :\ [18:24] <taylanub> This is OpenBSD 5.1 by the way. [18:25] <beastd> taylanub: i hope you did ffbisect. might have been lots of work otherwise [18:26] <taylanub> What's that ? I use the magit Emacs library to visualize the log and reset from there. [18:28] <beastd> taylanub: ffbisect is a script to ease biscecting ffmpeg code (because of merges). bisecting is the process of choosing an old (working commit) can be way back in time. and than doing a binary search on the interval [18:29] <beastd> anyway could you pinpoint exact commit that is causing you trouble? [18:36] <taylanub> I have yet to find that out .. Indeed, traversing the history seems hard. [18:36] <taylanub> I'm on some December 22 commit now, but the log also shows commits from Jan 23. :\ [18:36] <taylanub> (Not very good at git yet.) [18:39] <beastd> taylanub: maybe you should try ot bisect from now on [18:40] <beastd> i assume you have git clone of ffmpeg and found a commit in the past that worked your you (e.g one in the beginning of december) [18:44] <beastd> taylanub: are you testing for the regression with ffplay? [18:47] <taylanub> Yes. [18:51] <taylanub> Is there a way to not re-compile everything when changing commits ? A full compilation takes 5-10 minutes. [18:51] <beastd> taylanub: not sure that is a goode idea [18:51] <taylanub> OK then, will just take my time. [18:51] <beastd> taylanub: try it this way: [18:52] <beastd> taylanub: remember the bad revision or just checkout the bad revision [18:53] <beastd> then execute tools/bisect-create [18:53] <beastd> tools/ffbisect start [18:54] <beastd> tools/ffbisect need ffplay [18:54] <beastd> tools/ffbisect bad # marks current revision as bad [18:54] <beastd> oop wrong [18:54] <beastd> tools/ffbisect good # assuming you checked out a revision where it was working for you [18:55] <beastd> tools/ffbisect bad master [18:55] <beastd> then it should chose a commit for you to test [18:56] <beastd> make disclean; ./configure; make # use -j4 or whatever brings speed up on your systems, i assume you have multiple cores [18:56] <taylanub> -j4 tends to randomly break; or is that just due to bad Makefiles ? [18:57] <beastd> taylanub: too bad it shouldn't :( [18:57] <taylanub> I mean in-general .. dunno about ffmpeg. [18:57] <beastd> test if it works. depending if it works/or doesn't work you shoult then mark the revision as good or bad with: tools/ffbisect good/bad [18:58] <beastd> continuing that cycle you should arrive at the offending commit in i guess about 10 steps [18:58] <taylanub> OK [18:59] <beastd> taylanub: i don't know. GNU make with -j<n> usually works for me. i remember that on windows i had some warnigns from make that some assertions weren't met. on linux it always worked for me so far. and i use it on ffmpeg all the time. [19:00] <taylanub> I'll just use it then. [19:01] <beastd> taylanub: good luck! if you found the offending commit please read http://ffmpeg.org/bugreports.html and report it as regression [19:02] <taylanub> OK [20:49] <taylanub> On another note, the configure script bails out with "too many arguments" for me. Bash works. I guess I should just report a bug. [20:55] <taylanub> The first bad commit could be any of: [20:55] <taylanub> 528878ee7b377e23a194d7c801571d97793047e0 [20:55] <taylanub> 9c555bca68cf3f168c2e4ffcb5b4dacbda8366bf [21:00] <taylanub> [528878ee7b377e23a194d7c801571d97793047e0] openbsd: configure: Stop enabling PIC by default [21:00] <taylanub> Since it mentions OpenBSD, I'm betting on that. [21:00] <taylanub> Oh, in fact the other one just merges that. [21:05] <durandal_1707> taylanub: you reverted that commit and confirm it fixes your problem? [21:05] <durandal_1707> or you are just using old release with broken shell? [21:06] <taylanub> durandal_1707: Excuse me ? The configure problem is something unrelated; I've always had it and just change the script to use bash. [21:06] <taylanub> The PIC commit is likely what introduced corrupt H.264 video playback. [21:07] <taylanub> (And re. configure, OpenBSD's ksh is a POSIX-compliant sh, so it would be configure script that's somehow broken.) [21:07] <taylanub> (Assuming that this ksh is not simply buggy itself.) [21:08] <durandal_1707> taylanub: not really, there is ksh bug that got fixed in later version [21:08] <taylanub> Oh, when exactly ? I'm using OpenBSD 5.1. [21:08] <durandal_1707> how old is that? [21:08] <taylanub> The previous release .. they release approximately every 6 months, so it would be maximally about a year old. [21:10] <taylanub> May 1 2012 apparently. [21:26] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source? [21:27] <Jiyuu|C2> i.e. avoiding the need to use mkvextract to get a subtitles file out of the mkv file [21:27] <Jiyuu|C2> also [21:27] <Jiyuu|C2> is is possible to use attached font files or do i have to extract and install them? [21:34] <trosh> i am on linux, and when i use ffmpeg to convert a part of a movie to still frames, using -ss for starting time, i have to wait for a hell of a long time until it reaches the desired time and then convert normally. is this normal behaviour? thanks. [21:34] <Fjorgynn> nah [21:35] <nronksr> trosh: try placing the -ss before the input file for a faster seek. add -ss for exact accuracy after the input file for a slower seek once you are in the general timeframe vicinity. [21:35] <trosh> oh thanks [21:35] <trosh> is -ss relative ? [21:36] <Fjorgynn> yeah -ss hh:mm:ss -i file.avi -t hh:mm:ss [21:37] <bkmz> hi guys, what is the exactly computation complexity of AAC-LC ?? where i can find complexity function for it [21:37] <nronksr> Ideas on extracting 6 still images every second from a 59.96fps clip? [21:42] <durandal_1707> nronksr: -vf select? [21:56] <nronksr> durandal_1707: thanks, I thought that too; I was experiencing a typo that boggled me. XD [22:02] <trosh> nronksr: just to be sure, -r 6 works in your case ? [22:05] <nronksr> trosh: for 6 frames extracted every second? I used this: -vf 'select=not(mod(n\,6))' [22:05] <vlt> Hello. Does anyone know what pkg I need to install additionally to ffmpeg to be able to use -vcodec libx264 on Ubuntu 12.04? [22:05] <JEEB> vlt, libavcodec-extra [22:06] <nronksr> vlt: ubuntu uses avconv by default, but yeah, libavcodec-extra and if I recall, something with 53 in it? [22:06] <JEEB> and yeah, if you want to use the packaged libav (which is not ffmpeg, long story), you should use the avconv command instead of the ffmpeg one :) [22:06] <JEEB> just a newer binary in that project [22:07] <trosh> nronksr: how come? [22:08] <vlt> JEEB, nronksr: Thank you! [22:11] <nronksr> trosh: I'm not an expert by any stretch of the imagination and that was the way I've done it and it works? [22:17] <vlt> :-( Installing libavcodecs-extra-53 breaks my whole pkg system on ubuntu o_O [22:17] <vlt> I should try a chroot env & [22:18] <JEEB> more like it wants to install packages that use it instead of the standard libavcodec package [22:18] <JEEB> it's OK [22:18] <JEEB> unless you have something weird in there [22:18] <JEEB> if you look at the list of installed and removed packages you should see it :P [22:19] <vlt> JEEB: (In the chroot env) apt is removing like 280 pkgs now. [22:20] <JEEB> and it should be installing something instead of them, no? [22:21] <vlt> JEEB: I dont know (and care) exactly now. I only know I cant do this on a production machine. [22:21] <vlt> Hence the chroot. [22:21] <JEEB> uhh [22:21] <JEEB> it should just remove the packages that are configured and linked against the standard (without x264 etc.) libavcodec packages [22:22] <JEEB> and then it should at the same time replace them with packages that are configured with x264 etc. enabled in the libavcodec [22:22] <JEEB> you should look at what it says exactly [22:22] <JEEB> because that is what it should do [22:22] <JEEB> if it does something else then something's wrong [22:22] <JEEB> but I do know that in the cases I've done it, it has worked rather nicely [22:23] <vlt> JEEB: Yes, Ill have a look at the apt log files later and show them here in case youre interested. [22:23] <JEEB> no, I'm not interested. I just want to tell you why it looks like it's going to "break" something [22:23] Action: JEEB sighs [22:27] <JEEB> vlt, although it might be apt failing to see the resolution, too [22:28] <JEEB> http://askubuntu.com/questions/132164/ubuntu-restricted-extras-wants-to-rem… [22:28] <JEEB> looking at this [22:28] <JEEB> if this is what happens then take a look at the answer [22:30] <relaxed> there are static ffmpeg binaries available with libx264 support [23:28] <Jiyuu|C2> no help for [22:26:47] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source? [23:28] <Jiyuu|C2> ? [23:29] <burek> Jiyuu|C2, you could use -vf subtitles [23:30] <Jiyuu|C2> would that work with a .mkv file as the source? [23:31] <Jiyuu|C2> or do i still have to extract the subtitles from it? [23:33] <burek> hmh, did you read the docs about that filter? [23:33] <burek> it's kinda new, so i didnt have time to read it thoroughly [23:38] <saste> burek: should be possible, since subtitles uses libavformat to read the sub stream from the file [23:39] <burek> saste, do you have any example/sample cmd line to showcase how to tell it to ffmpeg? [23:40] <saste> burek, can't be hader than subtitles=in.mkv [23:41] <burek> oh right [23:41] <burek> thanks [00:00] --- Mon Feb 4 2013

1 0

ffmpeg-devel.log.20130202
by burek 03 Feb '13

03 Feb '13

[00:03] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07master:985e93a86518: Do not fail for mixed interlaced / non-interlaced YUV4MPEG streams. [00:35] <teratorn> if im generating synthetic audio/video frames for testing, what timestamp-related invariants/constraints should I adhere to in order to keep the muxer happy interleaving the packets? [00:36] <teratorn> oh, err, I meant to ask on #ffmpeg :/ [00:58] <tg2> anybody know where to start looking to override the -RE [01:00] <tg2> to set it more than 1:1? [01:09] <tg2> rate_emu ffmpeg.c line 1642 ? [01:11] <saste> tg2: what's your final goal? [01:12] <tg2> the end goal is to render an mp4 being written to disk to flv so it can be streamed while it is encoding [01:12] <saste> tg2: what's wrong with -re? [01:12] <tg2> can it be made variable? [01:12] <saste> no [01:13] <tg2> if (rate_emu) { [01:13] <tg2> 01643 int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE); [01:13] <tg2> 01644 int64_t now = av_gettime() - ist->start; [01:13] <tg2> 01645 if (pts > now) [01:13] <tg2> 01646 usleep(pts - now); [01:13] <tg2> 01647 } [01:13] <tg2> seems quite easy to override [01:13] <saste> sure it could be extended, but we need at least a meaningful use case [01:13] <tg2> say you wanted to control the speed of encoding [01:13] <tg2> for resource reasons [01:13] <saste> usually you can control that at the OS level [01:14] <tg2> hm [01:14] <saste> e.g. setting the niceness level [01:14] <tg2> yeah you can do that [01:14] <tg2> the goal actually isn't to save rousrces [01:14] <tg2> since the main encoding process is going as fast as it can [01:14] <saste> which is a more general approach (can be applied to all processes) [01:14] <tg2> this is only doing a streamcopy of the mp4 [01:14] <tg2> into flv [01:14] <tg2> basically rewrapping it [01:14] <saste> -re is usually used for streaming [01:14] <tg2> yea [01:14] <tg2> to simulate a live stream [01:14] <tg2> but if it was variable [01:15] <tg2> you could make -re 1.0 by default [01:15] <tg2> and assigning any other value [01:15] <tg2> would give you realtime * that value [01:15] <tg2> in input speed [01:15] <tg2> this works very well for throttling [01:15] <tg2> for example, the main encoding goes about 10x realtime [01:15] <tg2> so 230~fps [01:15] <saste> tg2: patches are welcome, or open a ticket and discuss [01:16] <tg2> ok great [01:16] <tg2> i'll write it up and test it first [01:16] <tg2> it should be nondestructive in the sense that anybody using -re now will see no difference [01:16] <saste> nice [01:16] <tg2> the logic is quite simple [01:16] <saste> tg2: yes [01:16] <tg2> change the argiment parse [01:16] <saste> if you have a meaningful use case it should be fine [01:16] <tg2> and change the rate_emu to be non-boolean [01:16] <tg2> as far as I can see [01:16] <tg2> this is the only place it is used [01:17] <saste> tg2: but you need a separate option [01:17] <tg2> -respd ? [01:17] <tg2> or -refactor ? [01:17] <tg2> -re -refactor 1.2 ? [01:17] <saste> otherwise you break syntax and get many angry users [01:17] <tg2> shouldn't you can just set it to 1.0 if omitted [01:17] <tg2> that way it acts identical [01:17] <saste> yes [01:17] <tg2> ok so you mean make a second case for arg + argv [01:18] <tg2> with the same switch [01:18] <tg2> hmm [01:18] <saste> uhm no i don't think that would be possible [01:18] <saste> you need a separate option [01:18] <saste> then -re code will wrap around the new option (will set a specific value) [01:18] <tg2> ok [01:19] <saste> we will guide you on ML once you have a working patch [01:19] <tg2> great [01:22] <tg2> another quick question, if you will [01:22] <tg2> actually nvm I'll do some more research first ;) [01:22] <tg2> thanks [02:21] <ulatekh> Hello! I submitted a bug-fixing patch to the mailing-list 6 days ago, but no one commented on it. Is there anything else you need me to do? See http://article.gmane.org/gmane.comp.video.ffmpeg.devel/158184 [02:22] <Compn> ulatekh : dunno who checks makefile stuff [02:22] <Compn> cehoyos does sometimes [02:23] <ulatekh> I just didn't want my bug fix to get lost. And cehoyos seems to have his hands full :-) [02:24] <Compn> hes a busy guy [02:25] <ulatekh> Should I just leave my fix to its fate, then? [02:26] <Compn> 6 days old? [02:26] <Compn> ffmpeg docs say wait 2 weeks for reply, then ping [02:26] <Compn> :) [02:26] <ulatekh> OK...there was a lot to read, I must have missed that...I'll ping later...thanks [02:26] <Compn> we are all on volunteer time here :) [02:26] <Compn> no problem [02:26] <Compn> sorry i cant review, i have no clue ... [02:27] <Compn> barely have my head screwed on [02:27] <ulatekh> Still, it's a shame...I'm like a rash on the open-source world. I track down bugs in myriad projects when I find them, and try to submit fixes...and the vast majority of the time, all I get is apathy. [02:28] <Compn> well [02:28] <Compn> you can ask one guy who works on it [02:28] <Compn> ah hes not online atm [02:29] <Compn> ulatekh : try mailing diego biurrun [02:29] <ulatekh> I'll mail him in 8 days, how's that. [02:29] <Compn> or michaelni is the project admin around here too. he can review patches when hes active [02:29] <Compn> lol ok [02:34] <iive> actually I like the idea of creating libav<module>.<major_version> [02:34] <iive> it would allow different api-breaking versions to be installed for the programs that need them. this includes forks too. [02:34] <Compn> ulatekh : see, iive cares :) [02:35] <iive> i'm cheap at talk :P [02:35] <ulatekh> Right now, the --build-suffix option to configure gives you libav<module><buildsuffix>, and --progs-suffix does similarly for the executables. They work fine, but the pkgconfig files needed a fix. [02:35] <ulatekh> Nice to see someone cares...the apathy I get when trying to fix bugs in open-source code really does get me down sometimes. [02:36] <ulatekh> iive: I've got git-master ffmpeg installed on my machine along with the ffmpeg from my dist's yum repos...they coexist just fine. [02:37] <iive> does buildsuffix separate the include locations too? [02:37] <ulatekh> Yes. [02:37] <ulatekh> The only issue left unfixed (if my patch is considered) is the name of man pages. [02:38] <iive> aha, so your patch is just fixing the pkgconfig files [02:39] <ulatekh> Yeah, I couldn't figure out how to fix the man pages...the bash scripting was way too tense for me ;-) [02:40] <ulatekh> I'm fixing the man pages in my rpm .spec file, post buildroot-install. [02:56] <michaelni> ulatekh, your patch is missing a commit message [02:57] <ulatekh> "Fix pkgconfig files when ffmpeg built with --build-suffix"? [02:57] <ulatekh> I dunno, the commit message kinda writes itself... :-) [02:57] <michaelni> it should start with something like build-sys: or configure: [02:58] <michaelni> also it should list [02:58] <michaelni> what you tested this against [02:58] <michaelni> it seems working fine on ubuntu [02:59] <ulatekh> I tested it on Fedora Core 14, 16, and 17. [03:00] <ulatekh> Sorry, I didn't know all the bureaucratic steps it takes to submit a bug fix. It seems like most open-source projects treat "outsiders" like this...then the insiders check in something that breaks the code 8 ways from Sunday but that's OK. (I finally gave up on mono-devel after a few such outrages.) [03:01] <burek> ulatekh reminds me of me :) [03:01] <burek> but it all boils down to the thing that people dont have enough of free time [03:02] <ulatekh> Still, the fish jumped right in the boat, and filleted itself...all you have to do is eat it :-) [03:02] <michaelni> burek, yep, time is much much more an issue than any bueacrat shit [03:02] <burek> well, i now understand both sides so i dont get upset that much as before ^^ [03:02] <burek> still, some kind of improvement in that way would be welcomed :) [03:03] <ulatekh> Mostly, it bugs me when I get the runaround & "insiders' can break code horribly and then give me the runaround on my attempts to fix the bug. (That hasn't happened to me here, so don't take offense. :-) [03:04] <michaelni> ulatekh, do you want to join become an insider and take over configure maintaince ? [03:04] <burek> ulatekh, dont worry that much, if your fix helps anything, it will be applied sooner or later [03:05] <ulatekh> I'd have a HUGE learning curve to take over something like that. [03:06] <ulatekh> My frustration with ffmpeg is that the code is so persistently undocumented. I've tried to trace bugs before, and have encountered nothing but pages and pages of undocumented god-knows-what. [03:06] <burek> we lack the time to document it while coding stuff [03:06] <burek> and people to document it after the thing has been coded [03:06] <burek> consider constant radical changes here and there :) and you get the idea why is it so :) [03:07] <michaelni> if everyone who didnt understand something would document it once he does figure it out then docs should improve alot [03:07] <ulatekh> I must say, I completely reject the idea that "there's no time to document". Documentation SAVES time by reminding you what the heck you were thinking when you wrote it...and telling others what you were thinking when they try to join in. [03:08] <burek> I agree with you, but how to make the developers agree with you too? :) [03:08] <burek> their todo lists are scary [03:09] <ulatekh> I don't know. My 20-year career has been a nightmare, and lack of programmers commenting their code is the biggest reason. [03:10] <Compn> ulatekh : so document one file [03:11] <Compn> :P [03:11] <Compn> no ones stopping you from adding comments to code [03:11] <Compn> what i'm tired of is people who stop by our community for one day (or a week tops) and complain about the code, then never return [03:11] <Compn> its like, put up or shut up [03:11] <burek> Compn, i believe the idea he is standing behind is for developers to document their own code :) [03:11] <Compn> and the developers say no. [03:11] <ulatekh> Well, knowing my patches might actually get applied could convince me it's worth my time...and yes, burek, I'm talking about developers documenting their own code. [03:12] <ulatekh> @compn: and thus my career is a nightmare. [03:12] <ulatekh> I'll never understand the "no to documentation" mentality if I live to be 1000. [03:13] <ulatekh> And as far as me adding comments to ffmpeg's code...I'd have to understand it first. And the lack of any documentation is an insurmountable barrier. Plus, I have to wonder if anyone on the other side will even care if I do something like that. Like I said earlier, my usual experience with submitting fixes in the open-source world, is apathy. [03:14] <Compn> uh oh [03:14] <Compn> no offense [03:14] <Compn> but i am detecting open source troll [03:14] <Compn> my warning flags have gone up [03:15] <ulatekh> "open source troll"? [03:15] <Compn> yes [03:15] <ulatekh> Define? [03:15] <Compn> comparing open source world to commercial software world comes next [03:15] <ulatekh> I hate commercial software. I run it as little as possible. [03:16] <Compn> oh good, then i was wrong :) [03:16] <burek> wait Compn, the guy has got a point [03:16] <burek> no need to flame him [03:16] <Skyler_> the documentation could be improved in many places, patches are probably welcome [03:16] <Compn> patches are always welcome :) [03:16] <Compn> i got a better one [03:16] <ulatekh> Are you suggesting I'm somehow anti-open-source? Dude, Google my username...I'm like a rash all over the open-source world. Mostly I get ignored, but I'm there in the devel mailing lists. [03:16] <Compn> ulatekh : what code confuses you ? [03:16] <michaelni> ulatekh, I will apply the pkgconfig suffix patch, i just want to test it a bit more [03:17] <Skyler_> however video/audio coding has an extra catch that generally people are expected to understand the format before reading the code (unless it's a format where there's no documentation at all, like some proprietary thing), so often it won't document things that are 'obvious' to someone who knows how the format works [03:17] <Compn> ulatekh : i can almost guarentee someone will comment code that you say is hard to understand :) [03:17] <ulatekh> @michaelni: Awesome. All I really wanted was feedback. [03:17] <Skyler_> e.g. an h264 decoder won't explain the h264 spec [03:17] <Compn> h264 spec wont explain h264 :P [03:17] <Compn> ehe [03:17] <Skyler_> Pfffff [03:18] <ulatekh> Well, it'd help if e.g. the h264 decoder would explain how parts of the code relate to the spec...when I'm trying to track down a specific bug, I'd like some idea of what the code is doing, so that I don't necessarily have to be an expert in the codec to track the issue I'm dealing with. [03:18] <Skyler_> yeah, that would help [03:18] <Compn> Skyler_: i dont see many copyrights in h264* [03:18] <Compn> hmm i thought someone just added specs to one of the decoders [03:18] <Compn> but i dont know which one [03:19] <Compn> well , not 'just' but maybe 2 yrs ago [03:19] <burek> :) [03:19] <burek> see what i mean about free time :D [03:19] <burek> people loose the sense of time coding ffmpeg :))) [03:20] <ulatekh> Recently I submitted ticket #2190, while I was trying to get raw video packaged so that kdenlive would be able to import it. Bless cehoyos for knowing what to look for...I would have never been able to find what he did. [03:20] <Compn> i barely know what date is today :\ [03:20] <Compn> ulatekh : its a lot of code to memorize :) [03:21] <ulatekh> @compn: for an example of code that confuses me...MPEG 1/2 encode. One thing I've wanted in ffmpeg for like forever is support for soft telecine, i.e. giving it a 24000/1001 fps video and getting it soft-telecined to 30000/1001. And being able to transcode soft-telecined video w/o it getting mucked up. But digging through the source code, I have no idea what is doing what or why, so I get nowhere. [03:23] <Compn> ulatekh : ehe, ffmpeg -vf mp=telecine or =softskip [03:23] <ulatekh> mpeg2enc from the mjpegtools project does a GREAT job of converting analog video to mpeg1/2, but it doesn't do 2-pass encoding, so I'd like ffmpeg to transcode my soft-telecined video so that I can achieve my desired bitrate (i.e. first encode video with mpeg2enc at the maximum bitrate). [03:25] <ulatekh> If it finally got implemented, that's great...the last time I scoured the Net for how to do that with ffmpeg, what little info I could find seemed to suggest that most ffmpeg developers were non-U.S. and thus didn't care to spend time on soft-telecine. And last week or so, I was on the #ffmpeg IRC channel, discussing soft-telecine with michaelni, and he didn't seem to know it had been implemented... [03:26] <ulatekh> In any case, soft-telecine was an example of something I would have been willing to try to implement, if I could understand the source code at all. [03:27] <ulatekh> And hey, maybe this is a case of "coming home from a bad day at work and kicking the dog"...the programmers ruining my life right now are the ones I have to deal with through my job ;-) [03:28] <cone-874> ffmpeg.git 03Steven Boswell II 07master:6289a8296a89: build-sys: Fix pkgconfig files when ffmpeg is built with --build-suffix [03:28] <ulatekh> Hey, my faith is restored partially...nice :-) [03:29] <ulatekh> Tell you what...let me go get a beer & let's see if I stop whining so much LOL [03:29] <Compn> michaelni was the one who added mplayer filter support [03:29] <Compn> but maybe soft-telecine and telecine are different things that i'm confusing ... [03:30] <ulatekh> Based on what I'm reading in https://ffmpeg.org/trac/ffmpeg/ticket/1782 it seems ffmpeg _does_ support it now. So...hell yeah! [03:32] <Compn> yeah, michaelni works hard for ffmpeg :) [03:33] <ulatekh> My big forays into open-source programming tend to happen when I have time and energy, i.e. when I'm unemployed. While employed, all I have time/energy for is supplying bug fixes for the stuff I run into while trying to do what I want to do. [03:36] <ulatekh> y4mdenoise (i.e. http://mjpeg.cvs.sourceforge.net/viewvc/mjpeg/mjpeg_play/y4mdenoise/ ) happened during a nasty period of unemployment. [06:08] <burek> oh im in gmt+1 so it counts more [06:08] <burek> :) [06:16] <burek> ok it works now :) [06:16] <burek> michaelni, please let me know if something doesn't work as expected, so i can fix it [09:37] <creep> hi [09:37] <creep> i have an mkv file taht ffplay hangs in the middle [09:38] <creep> if a specific part is skipped over, then it will play ok again [10:00] <wm4> did ffmpeg stop extracting cover art images from .aac? [10:01] <wm4> (using ID3v2) [10:01] <wm4> it works with an older version, but not recent git [10:36] <wm4> cehoyos: what the fuck? [10:38] <wm4> I guess I'll post a useless paste of ffplay output on all tickets if it makes you happy [10:39] <wm4> even though it's completely useless, and any developer attempting to reproduce this problem wouldn't get anything out of it (other than wasting his time reading useless crap) [10:39] <cehoyos> wm4: Only paste ffplay output if the problems are not reproducible with ffmpeg, see http://ffmpeg.org/bugreports.html [10:39] <cehoyos> And please do not forget your newer tickets [10:39] <wm4> what's the point? [10:40] <cehoyos> I don't understand. [10:40] <wm4> I've provided all information needed to reproduce the problem [10:42] <cehoyos> I disagree: Either provide the failing ffmpeg command line together with complete, uncut console output (ffplay command line if the problem only affects or is only reproducible with ffplay) or provide minimal source code that allows to reproduce the problem if it is neither reproducible with ffmpeg nor ffplay [10:44] <wm4> it's an illusion to think that simple ffmpeg or ffplay usage with no interaction could show some of these problems [10:44] <wm4> or that people would actually bother writing complicated test programs [10:45] <cehoyos> If this is the case (this is very unusual over the total number of tickets - and issues - ever opened) then please take the time to explain why the problem is not rerproducible with ffmpeg (the application): Very often, this explanation was already sufficient to solve the problem for the reporter [10:52] <wm4> ffmpeg fails to inject an attached picture in the packet stream when seeking before the start of the file - in libav it works [10:53] <wm4> cehoyos: how would you report this issue? [10:54] <cehoyos> Apart from the fact that I don't understand it (and I suspect you don't mean ffmpeg but FFmpeg): What about source code that allows to reproduce this? (I don't think this was ever reported.) [10:54] <cehoyos> Does this correspond to one of the tickets you opened? Which one? [10:54] <wm4> the only source to reproduce this is my mplayer fork [10:55] <wm4> not yet [11:02] <nevcairiel> wouldnt it be better to announce the machines that have newly joined the group of failed/idle machines? [11:10] <divVerent> wm4: one idea to make a test case: [11:11] <divVerent> would it be viable to edit ffmpeg.c to make the issue visible? e.g. a debug print when getting the picture? [11:11] <divVerent> ffplay.c, actually [11:36] <cehoyos> bcoudurier, TimNich: I sent a patch that writes the fiel atom independently of the codec_id. Is there anyhting I miss? Ie, should the atom not be written for some codecs? [11:38] <wm4> cehoyos: is #2010 ok now [11:38] <cehoyos> still downloading... [12:04] <michaelni> burek, i think the duron box is too slow for a dumb 24h threshold [12:29] <ubitux> hey [12:30] <ubitux> please tell Paul i don't mind any change to showspectrum; i'll review history when i come back, if something is wrong i'll do what's necessary [12:32] <cone-454> ffmpeg.git 03Carl Eugen Hoyos 07master:b45a3e167f49: Map the interlaced flag of yuv4mpeg streams to AVCodecContext->field_order. [12:35] <wm4> so how can you set arbitrary avopts with the ffmpeg frontend? [12:35] <cehoyos> ubitux: If you have time, there are still two patches in ticket #2144 [12:36] <wm4> or rather, how do you know where they end up [12:45] <ubitux> cehoyos: i thought i "reviewed" them [12:47] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137682.html http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137684.html http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137686.html [12:47] <cehoyos> ubitux: Did you review this one? http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/157781/focus=157860 [12:48] <ubitux> oh last version, will look [12:49] <cehoyos> I did not apply 3/3 because I thought it might be better to apply 2/3 first [12:50] <ubitux> yep sure [13:19] <wm4> cehoyos: do you still have any tickets reported by me on your close-soon list? [13:20] <ubitux> http://superuser.com/questions/545329/automate-running-a-program-when-input… would be nice to script this with ebur128 filter [13:22] <wm4> oh right, ticket 2002 [13:22] <wm4> but how the hell do you demonstrate a bandwidth issue [13:22] <wm4> maybe there's a program that can measure network bandwidth usage of a single program by playing ld_preload tricks? [13:23] <cone-454> ffmpeg.git 03Diego Biurrun 07master:76d90125cdc0: vf_hqdn3d: x86: Add proper arch optimization initialization [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:0d13a7b786b9: Merge remote-tracking branch 'qatar/master' [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:d593f2b2416f: avfilter/x86/vf_hqdn3d_init: fix author attribution & project name [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:6726fb653320: avfilter/vf_hqdn3d.h: restore author attribution, history and project name [14:04] <DEATH> Hi, I'm having some issues building FFmpeg 1.1.1 on MinGW [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c: In function 'dsputil_init_mmx': [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: error: 'gmc_mmx' undeclared (first use inthis function) [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: note: each undeclared identifier is reported only once for each function it appears in [14:06] <DEATH> Configure line: [14:06] <DEATH> ./configure --disable-everything --enable-decoder=aac --enable-shared --disable-swscale --disable-network --disable-avdevice --disable-postproc --disable-avfilter --disable-avformat --disable-debug --enable-runtime-cpudetect --disable-swresample --disable-ffprobe --enable-decoder=mp3float --enable-decoder=vorbis [14:06] <DEATH> Goes away with --disable-mmx but that appears to also kill all CPU specific optimizations, configure shows SSE and such being all disabled after appending --disable-mmx [14:11] <michaelni> DEATH, cant reproduce this (linux) [14:23] <cehoyos> wm4: Regarding 2002: You don't have to explain why it makes sense not to request all streams. Just add the console output and explain how ffmpeg should know which stream you want to receive. [14:57] <cone-454> ffmpeg.git 03Nicolas George 07master:55910e18948d: lavd/alsa: simplify reordering functions definition. [15:30] <saste> wm4? [15:30] <wm4> saste: ? [15:30] <saste> wm4, ping on my lavf/http cookie patch [15:31] <saste> that should address the first issue [15:31] <wm4> saste: oh ok... I don't follow the ML [15:33] <wm4> saste: I think that's not really ideal... IMO it should always return with an error if an error happened, instead of silently ignoring it [15:33] <wm4> at least on OOM [15:33] <saste> wm4, that's unrelated [15:33] <wm4> but then I don't know what ffmpeg's policy is on OOM issues [15:34] <saste> yes and i don't like how errors are handled there [15:34] <saste> but at least it avoids the (null) [15:34] <wm4> hm looks like I git reset --hard'ed the patch I wrote for this [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:6032a1c977cc: ffplay: extend doxy for audio_decode_frame() [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:fd6b6efbd212: lavfi/crop: fix help message for the keep_aspect option [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:6d9c21dc0e82: doc/indevs: add missing final dot in v4l2 option value description [15:35] <wm4> saste: IMO get_cookies should return whether there is an error (and if there is, return from http_connect too), and the check whether cookies is NULL should be a separate if block entirely [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:571ef42dd4eb: ffplay: dynamically allocate audio buffer [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:4fd6e5af1e33: ffplay: fix order of setting show_mode [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:5de3f724f152: ffplay: remember last window dimensions [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:c5eab4bb70cc: ffplay: move up pause functions [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:4ea7fbb2ec1a: ffplay: step to next frame if paused when seeking [16:19] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:4be0b9109421: Merge remote-tracking branch 'cus/stable' [16:27] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:cdc48860a8cb: h264: silence warning about array index being out of bounds [16:38] <saste> wow we're fast ;-) [18:51] <wm4> with ffmpeg I can't play concatenated ogg files, while I can with libav [18:54] <Daemon404> i've been finding an increasing number of ffmpeg-only bugs recently... [18:55] <wm4> the end is near [18:57] <cehoyos> wm4: Please provide a sample [18:57] <wm4> cehoyos: ffmpeg -i something.mp3 out1.ogg && ffmpeg -i something.mp3 out2.ogg && cat out*ogg > evil.ogg [18:58] <wm4> cehoyos: I'll test with ffmpeg/avconv later and create a proper ticket just to make sure it's not my fault [19:04] <cehoyos> wm4: Why do you think concatenating ogg files produces valid ogg files? (I don't know if it is valid.) [19:04] <michaelni> wm4 works here [19:05] <michaelni> even with changinhg samplerate & channels between the files [19:06] <wm4> cehoyos: apparently it's similar to how ogg streaming can work [19:06] <cehoyos> Daemon404: I find that impressive: Originally (at the time of the 0.7 and 0.8 release) there were around 50 regressions in the fork, perhaps 70. Later, there were a few hundred regressions, now, I can typically not even test new tickets with avconv because it often completely fails to process the given sample [19:06] <cehoyos> And concerning ogg: The reason I asked for a sample is that FFmpeg supports a number of ogg samples (and features) that simply do not work with avconv [19:07] <wm4> cehoyos: I tried it because I saw this: https://bugzilla.mozilla.org/show_bug.cgi?id=455165#c29 [19:09] <cehoyos> Yes, you are right: Chained ogg support is missing in avconv [19:09] <Daemon404> impressive as it may be [19:09] <Daemon404> it's true [19:10] Action: Daemon404 goes to see if the known bugs have been fixed and queue some to report later [19:10] <wm4> cehoyos: who's right? I said it works with libav but not ffmpeg [19:10] <wm4> cehoyos: though I still haven't tested with ffmpeg/avconv directly [19:11] <Daemon404> please provide full uncut output... [19:11] <Daemon404> /troll [19:12] <cehoyos> Yes, that is exactly what I mean! [19:12] <wm4> anyway, I still don't know if libavformat handles ogg streams correctly now... does it still add new tracks on track changes with some ogg streams? [19:21] <michaelni> wm4 it should only create new streams if theres no existing stream with matching codec [19:21] <cone-454> ffmpeg.git 03Nicolas George 07master:23686c72e5ae: doc: update filter_design.txt to API changes. [19:21] <michaelni> i tried vorbis in ogg and flac in ogg both work with chained oggs [19:22] <wm4> michaelni: and how does the application know it has to switch to the new stream, and that the old stream won't have new packets? [19:22] <michaelni> you have chained oggs that mix different codecs ? [19:22] <wm4> probably not [19:23] <wm4> well, the question is whether ogg internet radio streams do this [19:23] <wm4> but they probably don't [19:23] <wm4> what about vorbis extradata? [19:23] <michaelni> will be sent in stream on the point where old swicthes to new [19:24] <wm4> as part of the packet? [19:24] <michaelni> yep [19:25] <wm4> nice [20:02] <wm4> hm I guess I accidentally tested seeking [20:02] <wm4> which appears to behave slightly differently between ffmpeg and libav [20:03] <wm4> ffmpeg still prints a weird message, though: "[NULL @ 0x8ade2c0]Invalid packet" [20:03] <wm4> and reports a bogus length [20:06] <wm4> oh wait, the bogus file length is only with libav [20:08] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:0e9b9a6748aa: flacdec: skip in stream header packets [21:28] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:695af8eed642: h264: skip error concealment when SPS and slices are mismatching [22:37] <cone-454> ffmpeg.git 03Carl Eugen Hoyos 07master:a60530e3ee1d: Require at least three frames to autodetect loas. [00:00] --- Sun Feb 3 2013

1 0

ffmpeg.log.20130202
by burek 03 Feb '13

03 Feb '13

[00:24] <llogan> nilsge: using jack1 or jack2? [00:25] <nilsge> llogan: jack1 [00:26] <nilsge> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -ac 2 -c:v libx264 -preset fast -pix_fmt yuv420p -s 1280x960 -c:a libmp3lame -ab 128k -ar 44100 -threads 0 -f flv "rtmp://live.twitch.tv/app/live_**********************" [00:26] <nilsge> it makes no difference what a/v values I'm using. I went down to 320x240 and -ab 48k -ar 22050 [00:36] <teratorn> if im generating synthetic audio/video frames for testing, what timestamp-related invariants/constraints should I adhere to in order to keep the muxer happy interleaving the packets? [00:48] <nilsge> llogan: I tested it with jack2, it is the same. [00:56] <llogan> you can cut out the repeated messages [00:58] <llogan> are you using realtime scheduling? [00:58] <nilsge> yes [00:59] <nilsge> I am running a pro-audio system with recording chain and super soundcard etc. it is not my system itself [00:59] <nilsge> http://pastebin.com/xuZp8HMZ [01:00] <nilsge> also those xruns do not show up in jack itself (e.g. qjackctl display) [01:00] <nilsge> in this example there was no sound at all. just silence [01:00] <nilsge> the jack ports weren't even connected [01:00] <nilsge> system was idle, except ffmpeg [01:01] <nilsge> The web is full of cry for helps with xruns but no solution. [01:01] <nilsge> xruns in ffmpeg I mean [01:06] <llogan> is the output still ok? [01:07] <nilsge> llogan: absolutely not. You can't listen to it. [01:07] <nilsge> llogan: with the audio hiccpus there also comes a video speed issue. [01:07] <nilsge> I have read in the web that using audio alone with jack seems to work, and video alone also. It is the combination. [01:07] <nilsge> so it might be a sync issue, but I don't know much about these things [01:12] <llogan> i haven't used jack in years, but maybe i'll try to duplicate the issue since i'm going to be using hydrogen for a drumtrack over the weekend [01:14] <llogan> nilsge: maybe you can get an answer at ffmpeg-user mailing list [01:14] <llogan> or linux-audio-user [01:15] <nilsge> lau does know nothing. I am well connected there. It is a known issue with the answer "avoid doing it, don't do live screencasts and use recordmydesktop for the rest" [01:16] <nilsge> (as a sidenote: libav behaves exactly the same) [01:16] <llogan> i guess that's their way of saying "i don't know" [01:16] <llogan> if you think it is a bug you can always search/submit a bug report. [01:17] <nilsge> I don't know if it is a bug. It sounds like not knowing JACK. [01:44] <burek> nilsge [01:44] <nilsge> burek: yes [01:45] <burek> try this: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f null - [01:45] <burek> does it still throw xruns? [01:45] <burek> octe, did you solve your issue [01:46] <nilsge> burek: no xruns [01:46] <nilsge> and my system is under heavy load right now, I am compiling [01:46] <burek> that's good [01:46] <nilsge> plus I am actually sending audio to the default input [01:46] <burek> that means your card can capture at that rate without problems [01:46] <nilsge> so, what does it mean? [01:47] <nilsge> yes, I thought so. [01:47] <nilsge> RME Hammerfall [01:47] <burek> now, we need to see if your bus/cpu is also good enough to sustain the load when both capturing and encoding [01:47] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default output.mpg [01:47] <burek> or [01:48] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c:v libx264 -crf 0 -c:a pcm_s16le output.flv [01:48] <nilsge> that generates xruns on mass. I'll abort the compilation first. [01:48] <burek> ok [01:48] <nilsge> still xruns [01:49] <nilsge> too many. [01:49] <burek> yes i see [01:49] <burek> the problem is (i guess) [01:49] <burek> that your audio card is sending the data at such rate that your machine can't quite respect [01:49] <nilsge> the second command does not work since it does not set a sample rate. I am on 48k which is not in flv [01:49] <burek> since it wastes time on encoding, it can't capture audio properly [01:50] <burek> what kind of cpu do you have [01:50] <burek> (try mp4 or mkv) [01:50] <JEEB> <nilsge> the second command does not work since it does not set a sample rate. I am on 48k which is not in flv <- actually it is [01:50] <JEEB> at least for AAC [01:50] <JEEB> page 71 of the current version of the FLV specification [01:50] <nilsge> yes, I have not read this [01:50] <burek> i wanted to avoid any kind of compression [01:50] <nilsge> [flv @ 0x27bd9c0] FLV does not support sample rate 48000, choose from (44100, 22050, 11025) [01:51] <nilsge> anyway [01:51] <JEEB> nilsge, you should file that as a bug [01:51] <JEEB> as it is one [01:51] <JEEB> if you used AAC [01:51] <burek> nilsge, which version of ffmpeg are you using [01:51] <JEEB> anyways, yeah -- no compression for testing at first is a good idea [01:51] <nilsge> burek: git from 15 minutes ago [01:51] <JEEB> possibly even using a separate wav file as output, otherwise mkv should be fine [01:52] <nilsge> burek: the latest stable in the archlinux repositories was build without jack [01:52] <nilsge> burek: Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz [01:52] <burek> oh ok [01:52] <burek> sorry [01:52] <nilsge> (00:59) ( nilsge) http://pastebin.com/xuZp8HMZ [01:52] <JEEB> I think libav's flvenc.c didn't have that limitation as I poked them about it lately, so I wonder if and why ffmpeg would still have the 44kHz limitation with flv + aac (if it still has such) [01:52] <burek> ok :) [01:53] <JEEB> since ffmpeg merges libav's stuff [01:53] <JEEB> (At least most of it) [01:53] Action: JEEB goes check ffmpeg's flvenc.c [01:53] <burek> oh, you probably have hit an alsa issue [01:54] <burek> try: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f alsa -i hw:0 output.mpg [01:54] <burek> it should still show those xruns [01:54] <JEEB> hmm... [01:54] <JEEB> ffmpeg's flvenc.c seems similar [01:54] <nilsge> that does not work since jack is active. well.. I could deactivate it for now.. one moment [01:54] <JEEB> (unsurprisingly) [01:54] <JEEB> so I guess he wasn't using aac [01:55] <JEEB> because it shouldn't go into that one switch statement in that case [01:56] <nilsge> burek: I can't start this. my hw:0 returns Input/output error. cannot set access type (Invalid argument) [01:56] <nilsge> hw:0,0 the same [01:57] <nilsge> which is indeed my main device [01:58] <burek> did you check with arecord -l [01:58] <nilsge> yes [01:58] <nilsge> and cat /proc/asound/cards [01:59] <nilsge> but I have a loopback device, that works [01:59] <tlp> now that ffmpeg supports encoding 24-bit flac, is there any way to tell it to go back to producing 16-bit? [01:59] <nilsge> produces xruns. not so many [02:00] <nilsge> burek: that looks stable enough so far [02:01] <JEEB> tlp, if this was raw audio I'd say "look at -formats | grep "pcm" and pick a similar 16 one instead of the 24 one, but since your output format will be flac (probably in some container, too) you'd have to call libavresample out in some other way :) [02:01] <JEEB> s/libavresmaple/libswresample/ [02:01] <nilsge> burek: after ctrl+c there were a lot buffer underflow i=0 bufi=235019 size=265650 packet too large, ignoring buffer limits to mux it [02:02] <tlp> hmm [02:02] Action: JEEB wonders if it's -precision [02:02] <JEEB> oh, for soxr only [02:03] <JEEB> oh found [02:03] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#aconvert [02:03] <JEEB> this I'd guess? [02:03] <burek> hmh [02:04] <nilsge> burek: as additional information: all the web search showed me that people who used pulseaudio instead, or sometimes alsa directly, got away with it. [02:04] <nilsge> but pulse is not suitable for a realtime audio system so I would rather not use that. [02:04] <burek> nilsge i wonder if something like this would work: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f SOME_FORMAT_I_DONT_QUITE_KNOW - | ffmpeg -f THAT_SAME_FORMAT -i - -c:v libx264 ... output.flv [02:05] <burek> now, that format could be something like mpegts or nut [02:05] <burek> or im not sure which one is the most suitable for that [02:05] <burek> but the idea is to separate capturing from encoding [02:06] <burek> to see if the encoder spends "too much" time in encoding process, making alsa have to wait with its data, which fails [02:06] <burek> also, as a quick hope, try putting audio input first [02:06] <burek> it worked for me sometimes [02:07] <burek> ffmpeg -f jack -i default -f x11grab -s 1920x1080 -r 60 -i :0.0 output.mpg [02:07] <nilsge> what format shall I use for the first command? [02:09] <nilsge> burek: the second one produces the usual xruns [02:13] <burek> nilsge, try ffmpeg -formats [02:13] <burek> try with -f nut [02:13] <nilsge> the 3 dots...? [02:14] <burek> -c:v libx264 -crf 0 -c:a pcm_s16le [02:15] <nilsge> I have to add -ar 44100 [02:15] <nilsge> xruns [02:15] <nilsge> usual amount [02:17] <nilsge> burek: what kind of computer is needed to encode and record at the same time? [02:18] <nilsge> now that it is mentioned, I remember with recordmydesktop I can't use on the fly encoding. I get sync problems, the video runs too fast, audio hiccups etc. [02:19] <burek> well it could be that alsa issue then [02:19] <burek> the point is (as i understood it that way) that alsa doesnt buffer enough data [02:20] <burek> and when encoder doesn't pick it up quickly enough, the issue happens [02:20] <burek> usually the data is being dropped, encoder then asks for the dropped data and alsa says please wait while i acquire more data for you [02:20] <burek> something like that [02:21] <burek> anyway, you could see if there is any chance to increase that buffer size [02:21] <burek> in alsa directly (or it was in ffmpeg's source code, not sure now) [02:21] <burek> or you might use your own kind of buffering [02:21] <nilsge> some posts said to increase it in ffmpegs source code [02:21] <nilsge> but with no difference, apparently [02:21] <burek> well, one thing that comes to my mind [02:22] <burek> since our very first test didn't produce any errors [02:22] <burek> we could send that output to a localhost/udp port [02:22] <burek> and with another ffmpeg pick it up [02:22] <nilsge> sounds interesting [02:23] <burek> that way udp proto handler in ffmpeg would buffer the data [02:23] <burek> and you might solve your issue [02:23] <nilsge> ok, I want to try that [02:23] <burek> of course, its a dummy workaround, but worth trying if you are in a rush to finish something [02:24] <nilsge> it has become more of a technical quest by now [02:24] <burek> ok, then try for the 1st ffmpeg: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f SOME_FORMAT_I_DONT_QUITE_KNOW udp://localhost:1234 [02:24] <burek> and for the 2nd: ffmpeg -f THAT_SAME_FORMAT -i udp://@:1234 -c:v libx264 ... output.flv [02:25] <burek> usually mpegts format should work, but i cant guarantee [02:25] <nilsge> nut again? [02:25] <nilsge> ah [02:25] <nilsge> mpegts [02:26] <burek> try both [02:26] <burek> mpegts had some issues too recently (at least in my test cases) :) [02:26] <nilsge> the first runs. produces xruns already without the second even started [02:26] <burek> so if nut works better, then great [02:26] <burek> oh sweet [02:26] <nilsge> they stopped...? [02:26] <burek> wait [02:26] <burek> try this [02:27] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy output.mkv [02:27] <nilsge> looks stable after 3 initial xruns. The command before that went stable after a few seconds as well, it seems [02:27] <nilsge> but this is better [02:28] <nilsge> now the second... [02:28] <burek> hmh [02:28] <burek> so it appears to not be even fast enough for just capturing [02:28] <burek> how much ram memory do you have [02:28] <nilsge> (02:17) ( nilsge) burek: what kind of computer is needed to encode and record at the same time? [02:28] <nilsge> 4 GB [02:28] <nilsge> 64bit system [02:28] <burek> that obviously shouldn't be an issue [02:29] <nilsge> nothing of the ram is in use. minimal linux with i3 window manager, no bling [02:29] <burek> it depends on your encoding/capturing parameters [02:29] <burek> try for a test, to lower the frame size to 640x360 [02:29] <burek> (in the last command) [02:30] <nilsge> one moment, now that the xruns have stopped I want to try the current one with the 2nd command first. I had not started it yet [02:31] <burek> well, there's no point in it [02:31] <burek> since i've changed udp output to file output [02:31] <nilsge> that is strange. If I do nothing at all after starting the first (with nut format) it runs ok. Only when I start typing on my keyboard the xruns start as well [02:31] <nilsge> no I meant the udp version [02:31] <burek> yes, that's the same issue i had [02:32] <burek> somehow ffmpeg process doesn't get enough of either cpu time, bus access time or ram memory [02:32] <nilsge> I don't understand it. I started the first command alone again, just ctrl+c, up-arrow, enter [02:32] <nilsge> now zero xruns [02:32] <burek> and the worst thing is, the issue could never be successfuly reproduced at developers' machines.. :( [02:33] <burek> :) [02:33] <burek> you have to admit that it's not boring at least :D [02:33] <nilsge> Such things should not be possible with computers [02:33] <burek> it always throws something new at you :) [02:33] <nilsge> I have done it again to be sure, still everything ok [02:34] <nilsge> now, finally the 2nd command [02:34] <burek> what linux/kernel are you using? [02:34] <nilsge> archlinux 64bit, 3.7.5 [02:35] <burek> you just dont give me any reason to hold on to :) [02:35] <nilsge> there was a series of runs right now again but they stopped. so strange [02:35] <burek> probably those xruns happen when you move the mouse over your desktop or something cpu "intensive" :D [02:35] <nilsge> burek: I am no newbie with all that stuff. I know most details of my system, background processes etc. [02:36] <nilsge> burek: I tried that before I reported "zero xruns". Tried to generate some cpu noise [02:36] <nilsge> I also have an audio player running in the background, the whole time. seems to do nothing [02:37] <nilsge> they start/stop not at random. something I do with the programs/UI triggers it, but I don't know what yet [02:37] <burek> hmh.. did you try asking around in alsa community for some clues maybe? [02:37] <burek> just to make sure it's not (so far) a known issue? [02:37] <nilsge> I am afraid there is no alsa community [02:38] <burek> #alsa ? [02:38] <nilsge> from time to time some user comes to "us" in the #lad (linux audio developer) channel and asks for alsa help. We tell him #alsa and they usually report they send them to #lad in the first place [02:38] <burek> :D [02:38] <burek> that sounds great :) [02:39] <nilsge> Circular buffer overrun. To avoid, increase fifo_size URL option. To survive in such case, use overrun_nonfatal option [02:39] <nilsge> that is the second command [02:39] <burek> yes, that's the same issue i had [02:39] <burek> but that's for udp [02:39] <burek> it's not quite related to alsa issue [02:39] <burek> or it might be [02:39] <nilsge> oh wait, I made a typo [02:40] <nilsge> no, that was nothing. same result [02:41] <burek> :/ [02:42] <nilsge> it is strange [02:42] <burek> take a look at this issue: https://ffmpeg.org/trac/ffmpeg/ticket/615 [02:42] <burek> it's using usb webcam input instead of x11 [02:42] <burek> and alsa [02:43] <burek> and try to increment that buffer size [02:43] <burek> i hope it will work for you [02:43] <nilsge> ah yes, I have read that report earlier today [02:44] <nilsge> ok, lets see... [02:47] <nilsge> why again is this alsa related when using jack? [02:49] <nilsge> If I were using ffado (firewire soundcards)... [02:49] <burek> well, iiuc, jack is the wrapper around alsa lib, right? [02:50] <nilsge> not exactly. You don't need alsa. It runs with ffado, a null driver, over network without any soundcard information at all [02:50] <nilsge> (windows, bsd, osx) [02:51] <nilsge> but I'll try it with alsa buffer size first. If all goes wrong I can attempt to use alsa directly [02:56] <nilsge> burek: ok, compiled and installed [02:57] <nilsge> burek: 1st command is running, one xrun at the start, but that is quite normal for all programs here. [02:58] <nilsge> second does not run. I'm trying my original one. [02:59] <burek> if you compiled that increased buffer size thing, then try your own commands [02:59] <nilsge> doing it right now. I get xruns. too many I fear [02:59] <burek> :( [02:59] <nilsge> but that was with jack, not alsa directly [02:59] <burek> and with alsa? [03:00] <burek> btw, maybe there is the same thing for jack in ffmpeg [03:00] <burek> so if it works for alsa [03:00] <burek> you could try to fix the buffer size for jack too, i guess [03:01] <nilsge> burek: I had a quick look in the jack audio source file but I think it is not there. [03:01] <nilsge> which is quite correct. Jack buffer size is regulated by the daemon, and I have set it quite high to rule that out [03:02] <nilsge> well, maybe I could set it to a ridicoulous value... [03:03] <burek> :D [03:04] <nilsge> set it to 4096 which generates 186ms latency on 44100 Hz ( I set this to 44.1 on my hardware site as well so I can drop the resampling in ffmpeg) [03:05] <nilsge> nope [03:05] <nilsge> still bad [03:09] <burek> :S [03:09] <nilsge> burek: with alsa only... better still a few, but so far apart that you can listen to it [03:10] <nilsge> now I have to find out how to stream my sound to that alsa device :| [03:10] <burek> well, you just have to make sure if that really helped or the errors just turned out to become less frequent out of the blue [03:10] <burek> it's really difficult to debug such a thing [03:10] <burek> when you don't have a constant (expected) errors each time you run your tests [03:13] <nilsge> burek: so far they are less frequent. much less [03:14] <burek> sounds good :) [03:14] <burek> try double-increasing it again :) [03:14] <nilsge> I still have not found out how to use that device [03:24] <nilsge> burek: the video is working fine and no xruns [03:24] <nilsge> but no sound also :) but that should be only a settings problem [03:28] <burek> really? [03:28] <burek> what did you do to make it work? [03:28] <nilsge> the buffer size I think [03:29] <nilsge> the only problem is that I don't know how and which alsa device I have to specify to actually hear my sound [03:30] <nilsge> there is not much choice, though. I have 4 devices, but 3 of them can be ruled out (Webcam, Hdmi, Onboard Intel chip) [03:30] <nilsge> hw0 is my actual card, which is blocked by jack, and hw1 is my loopback device that I use to forward alsa-only sound to jack. [03:31] <nilsge> AND I thought the loopback works backwards as well so I can send jack back to an virtual alsa soundcard, e.g. to use with ffmpeg. But there is nothing coming out when I do that [03:31] <burek> oh i see [03:31] <burek> arecord -l [03:31] <burek> should help [03:31] <burek> also [03:31] <burek> arecord -L [03:32] <burek> probably ffmpeg -f alsa -i hw:X would be enough [03:32] <nilsge> -L is new to me [03:32] <nilsge> that is what I'm trying. hw:1 or hw:1,0 or hw:1,1 [03:32] <burek> but, sometimes you need to use -i hw:X,Y to specify a device Y on the card X [03:32] <nilsge> nothing [03:32] <burek> also, alsamixer might help [03:33] <nilsge> like I said, this sound stuff with alsa and jack is not new too me [03:33] <burek> just run it, press F6 and choose the card [03:33] <burek> then press <tab> to see only recording devices [03:33] <burek> ok [03:33] <burek> :) [03:33] <burek> im just saying it might be something trivial [03:33] <nilsge> I still hope that as well :) [03:34] <nilsge> alsamixer is of no use here. hw0 is RME, it has its own mixer. and hw1 is a virtual device, has no faders at all [03:36] <nilsge> maybe my loop_in device is wrong [03:37] <burek> well, im not sure [03:37] <burek> usually alsa is just enough to me [03:37] <burek> so i didnt experiment too much with other tools [03:38] <nilsge> the situation is a bit more complicated because JACK is running. But that should only be a slight disturbance [03:38] <nilsge> I think the main issue is solved [03:38] <nilsge> so thank you very much for that, burek! [03:38] <burek> well, im glad if we did solve something :) [03:38] <burek> :beer: :) [03:39] <nilsge> I have to check with a few basic tools if I'm getting any signal I think [03:49] <pietro10> Is it possible to have ffmpeg produce the same data on two output sources? I want to stream something over the internet at the same time as I save the stream to a file on my computer. Thanks. [04:31] <nilsge> burek: no luck. I think I give up for now [04:33] <burek> pietro10 check our wiki [04:34] <burek> nilsge :( [04:34] <nilsge> burek: wait! just now I tried arecord --v -fdat foo.wav for the first time and it shows me level [04:34] <burek> i told you, fun never ends with alsa :))) [04:35] <pietro10> ok thanks [04:36] <nilsge> aaargh, I don't understand it. The alsa device clearly receives data. why don't I hear it in ffmpeg [04:36] <pietro10> so just to check [04:36] <pietro10> ffmpeg <output1flags> <output1name> <output2flags> <output2name> ? [04:37] <nilsge> burek: http://de.twitch.tv/nilsge do you hear something? [04:37] <nilsge> or anybody [04:37] <pietro10> ok yes that is it thanks [04:38] <burek> im on a headless server atm without audio card :) [04:39] <burek> try using vlc [04:39] <burek> see if there are any spectrum changes [04:39] <burek> in visualizer [04:45] <nilsge> this is the final mystery: if I just do arecord -vv -fdat foo.wav I can see the VU Meter is going up, I connected my webcam to it, works. Total control. THere is sound coming to my alsa device [04:46] <nilsge> but if I use -D to specify the device is stays 0%. I don't know which device it captures if I don't give one. [04:55] <nilsge> according to this (arecord -vv -fdat foo.wav) http://pastebin.com/qW9NF1f1 it should be hw:Loopback,1 (or hw:1,1) but it is not. The device is silent when used with -D [04:55] <nilsge> burek: I joined #alsa to try it out myself. Maybe I never even get an answer :) [04:59] <burek> :) [04:59] <nilsge> can't I do the same in ffmpeg? Just omitting the device? [05:00] <burek> wait [05:00] <nilsge> somewhere it seems to exist, but not in the hw:x,y terminology [05:00] <burek> oh i see [05:00] <burek> try -f alsa -i dsnoop [05:01] <nilsge> that reads promising [05:01] <nilsge> cannot open audio device dnsoop (No such file or directory) [05:02] <burek> can you pastebin your arecord -L [05:03] <nilsge> http://pastebin.com/pLhc4i11 [05:03] <nilsge> U0x46d0x81d is my webcam [05:06] <burek> and you need to get the input from which device [05:06] <nilsge> http://pastebin.com/Ncbpt9ta [05:06] <nilsge> this is aplay -l and cat /proc/asound/cards [05:06] <nilsge> I am not sure, that is the point [05:06] <nilsge> I would guess it is 1,1 [05:06] <burek> your mic? [05:06] <burek> your cam? [05:06] <burek> what [05:07] <nilsge> it should be Loopback,1 [05:07] <burek> i mean logically from which device you would want to get your input :) [05:07] <nilsge> yes, Loopback. I am sending audio over jack to this [05:07] <burek> wait, what do you use to send your audio to jack [05:08] <nilsge> jack apps. music players etc. [05:08] <burek> what is the source of your audio :) [05:08] <nilsge> the point is, with arecord --vv -fdat foo.wav I actually see the signal when I connect anything to the jack end of the loopback device [05:08] <burek> your voice? your line in? your camera mic? pre-recorded audio file? what? [05:08] <nilsge> moc, the music player [05:08] <burek> oh [05:09] <burek> so "what I hear" [05:09] <burek> right? [05:09] <nilsge> kind of, yes. [05:09] <burek> well yes, you need dsnoop [05:09] <burek> so, try this: ffmpeg -f alsa -i dsnoop output.wav [05:09] <nilsge> cannot open audio device dsnoop (Device or resource busy) [05:09] <nilsge> dsnoop: Input/output error [05:09] <burek> wait let me check [05:11] <burek> i see.. can you try just for test to shut down jack [05:11] <burek> and see if the device gets ready [05:11] <nilsge> which device? [05:11] <burek> dsnoop [05:12] <nilsge> I can shut down jack, but I have no way to test it/send audio in [05:14] <burek> no need to [05:14] <burek> i just want to know if jack is blocking that device [05:15] <nilsge> no, it is not [05:15] <nilsge> arecord says when a device is blocked [05:15] <nilsge> hw:0 is completely blocked. that is my main card and used by Jack [05:15] <burek> well, the point is [05:15] <nilsge> hw:1 is the free loopback. The rest should be uninteresting [05:15] <burek> if the jack has blocked dsnoop for its own purpose [05:15] <burek> you then must use -f jack -i .. [05:16] <burek> cant workaround it [05:16] <nilsge> maybe, but I don't think that is the case. How does arecord receive the audio? It can't use blocked devices [05:16] <burek> good point [05:17] <nilsge> the key is that arecord without -D receives the audio, exactly as I want including the manual jack connection only to that loopback device, and when I use -D with anything it does not work anymore [05:17] <nilsge> and I need -D(evice) to find out what to give to ffmpeg [05:17] <nilsge> so most likely I missed something or there is more syntax than "hw:x(,y)" [05:20] <burek> maybe [05:20] <burek> -i hw:1,0,1 [05:21] <nilsge> I tried that before with an error message in arecord [05:21] <nilsge> ah, that was only a format error, not a device. [05:21] <nilsge> lets see ffmpeg... [05:23] <nilsge> all right! I got it in arecord at least. ffmpeg does not like the format as well [05:24] <burek> what format [05:24] <nilsge> float 32 bit is supported [05:24] <burek> pastebin :) [05:24] <nilsge> FLOAT_LE [05:25] <nilsge> just one line: cannot set sample format 0x10000 2 (Invalid argument) [05:29] <nilsge> burek: PCM 32-bit floating-point little-endian is needed. f32le [05:29] <burek> cant say without pastebin [05:29] <nilsge> pastebin of ffmpeg? [05:30] <burek> yes, show me the log to see what exactly ffmpeg is complaining about [05:30] <nilsge> http://pastebin.com/tG1UT9A7 [05:33] <burek> try using only ffmpeg -f alsa -i ... -f null - [05:33] <burek> remove all the video stuff [05:33] <nilsge> k [05:33] <burek> because when you set -r [05:33] <burek> it may apply to audio too [05:33] <burek> if you don't set -r:v and -r:a [05:34] <burek> maybe -r is not mutual option for a/v but you get the point [05:34] <burek> also, -ac 2 is after -i [05:34] <burek> which means that you are not requesting a 2 channel audio from your card [05:34] <nilsge> http://pastebin.com/Bc0AWjnD [05:35] <burek> but you actually are resampling it after you got the audio (in who knows what format) from your audio card [05:35] <burek> ok, now try [05:37] <burek> ffmpeg -f alsa -ac 2 -i hw:1,0,1 -f null - [05:37] <nilsge> the same [05:39] <nilsge> got it [05:39] <nilsge> burek: ffmpeg -f alsa -acodec pcm_f32le -ac 2 -i hw:1,0,1 -f null - [05:39] <burek> yes [05:39] <burek> i was looking to see what is the proper string :) [05:39] <nilsge> http://ffmpeg.org/trac/ffmpeg/wiki/audio%20types here [05:40] <burek> yes, the same you can get with ffmpeg -sample_fmts i think [05:40] <burek> or -formats [05:40] <burek> anyway, try joining all that now [05:41] <burek> see if it works, and if there are xruns again any more [05:42] <nilsge> http://de.twitch.tv/nilsge [05:42] <nilsge> burek: ! [05:43] <burek> it works ? :) [05:43] <nilsge> ah yes, you have no sound [05:43] <nilsge> it works [05:44] <nilsge> it seems the sound is 10s or so behind [05:45] <burek> buffering i guess [05:45] <burek> but is there any xrun? [05:45] <nilsge> but at least something is coming through [05:45] <nilsge> no, not one [05:45] <burek> oh great [05:45] <burek> it would be lovely if you could just check with the reverted version [05:45] <burek> if that buffer size was really an issue [05:45] <burek> so we can close it once and for good [05:45] <nilsge> the alsa buffer size should be a runtime value [05:45] <burek> i agree [05:46] <nilsge> ok, I'm installing the release version of ffmpeg again [05:46] <burek> thank you [05:46] <burek> im asking that because it is not very easy to reproduce the issue... [05:46] <nilsge> I can do that now since I don't need jack for that anymore [05:46] <nilsge> I found it rather easy. In 999 of 1 cases it did not work :| [05:48] <burek> yes, but you know Murphy... [05:48] <burek> if there is a chance it wont be present at developers' machines, it won't be present :) [05:49] <nilsge> there were a few xruns and the video/audio was not continous anymore [05:49] <nilsge> but not as bad as with jack [05:50] <nilsge> but the audio was still 15 seconds behind my host (and the video 4-5 seconds) [05:50] <nilsge> so +10s for audio [05:51] <burek> so the buffer might not be the only cause for that issue.. :s [05:52] <nilsge> I have to play around with the values a bit to see if that has any influence [05:52] <nilsge> at least I know that my jack bridge only adds ~20ms to that latency [05:53] <nilsge> for example the audio stream was uncompressed right now [05:55] <nilsge> burek: ah! with mp3 and 128k it suddenly gets down to ~5 seconds as well [05:55] <burek> geez [05:55] <nilsge> I have to see how stable it is because now the xruns become audible again [05:55] <nilsge> maybe revert to the buffer version of ffmpeg again [05:55] <burek> :S [05:56] <nilsge> since the delay was the same anyway [05:56] <nilsge> maybe I get no xruns AND proper sync [05:56] <burek> well play around and see what you get.. i lost a lot of time trying to resolve that issue [05:56] <nilsge> yes, thank you very much [05:57] <nilsge> I didn't expect it to be this long [05:57] <burek> :beer: :) [05:57] <burek> you see, it's never boring here :) [05:59] <nilsge> burek: and at last I can confirm that the sync is still as good as before (I don't know how much that is if I stream games with real action sfx sync) and no xruns through the buffersize [05:59] <burek> that's why games should dup their display to some udp port or something [05:59] <burek> so that we don't have to grab the screen [06:00] <nilsge> you still need audiosync then [06:00] <nilsge> the graphiccard and soundcard should internaly be able to generate a video stream [06:00] <nilsge> well, actually that is what a hardware recorder does :) [06:00] <burek> :) [06:01] <nilsge> that must be the most reliable solution. Just dvi through and analog audio through [06:02] <nilsge> if you know the youtube channel Angry Video Nerd this is what he does, playing retro games. Never emulator and screen captures, he just records the output of the consoles directly to a DVD recorder. May sound a bit inconvenient and ancient but it is rock solid [06:03] <nilsge> burek: good night. I should have been in bed 7 hours ago :| [06:03] <nilsge> but a good solution is worth 7 hours of sleep [06:04] <burek> :) [06:04] <burek> good night and take care :) [09:15] <Crosma> Yo. I'm just doing -vcodec copy -acodec copy on an mov to mp4 and it says, "multiple edit list entries, a/v desync might occur, patch welcome" when I try to play the mp4. The mkv only has one audio and one video stream. [10:54] <kode54> http://pastie.org/6028534 [10:54] <kode54> that's the readme for the source package that is available for building ffmpeg 1.0.1 for foobar2000 [10:55] <kode54> is there something about that process which would make it inherently incompatible with any other build of same version numbered av*.dll [10:55] <kode54> avcodec-54.dll etc [10:55] <kode54> same numbers [10:55] <kode54> it's not so much a problem with users building ffmpeg to replace it, which nobody has reported attempting [10:56] <kode54> so much as some random piece of crap Windows software appearing which forcibly loads itself into every process in the system, and auto loads its own copies of the same named but ABI incompatible ffmpeg [10:56] <kode54> which I know is not standard ffmpeg practice [10:56] <kode54> but apparently it's not compatible with some other stock builds of av*.dll either [11:55] <DEATH> Hi [11:56] <DEATH> I'm trying to build FFmpeg DLLs with different file names (as specified in http://www.ffmpeg.org/legal.html - add own suffix to the file names to prevent a clash with certain other software using FFmpeg). [11:56] <DEATH> Is there a ./configure option to do that? I cannot just rename the DLLs because that will break references between avcodec/avutil and so on [11:57] <DEATH> Must have other DLLs reference each DLL by the new name [11:57] <DEATH> In fact avcodec and avutil are all I care about [12:17] <dentex> HI everyone! [12:18] <dentex> I have a question about crosscompiling FFmpeg for Android. [12:21] <dentex> it works with a standard set of flags. But when I add '--enable-libmp3lame' don't [12:23] <dentex> I have also compiled liblame for android, but I don't know where to put the libs for the ffmpeg compiler to find them; or what to change to point there. [12:24] <dentex> I have the error 'libmp3lame not found' [12:34] <klaxa> dentex: did you add -lmp3lame -L/path/to/libmp3lame to ./configure? [12:35] <dentex> no... let me try [12:35] <dentex> thanks [12:36] <dentex> just one thing: [12:36] <dentex> from another project to build lame for android, I have now the structure [12:36] <dentex> ./libs/armeabi/liblame.so [12:37] <dentex> do you know if this is the missing library? [12:45] <nilsge> hello. what does the parameter -bufsize do? [12:53] <dentex> -lmp3lame is unknown... [12:54] <Nano> hi, i get the following error trying to use --enable-libmp3lame [12:54] <Nano> ERROR: libmp3lame >= 3.98.3 not found [12:54] <Nano> this is my configure call: [12:54] <Nano> ./configure --arch=x86 --target-os=mingw32 --disable-asm --disable-doc --disable-ffplay --disable-ffserver --disable-ffprobe --disable-everything --enable-libmp3lame --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib [12:55] <Nano> lame/lame.h exist in /usr/local/include. why does the configure script fail ? [12:56] <klaxa> dentex: ah probably is just -llame [12:57] <klaxa> Nano: are you sure the version is >= 3.98.3? [12:57] <dentex> klaxa: ok. [12:57] <dentex> trying [12:59] <Nano> klaxa: yes it's 3.99.5 [13:02] <klaxa> it also exists as liblame.so in /usr/local/lib then i assume... hmm that should actually work though i think... i'm not too sure though [13:02] <Nano> where is the configure script command check_lib defined ? i'd like to check where it looks for the sources... [13:03] <Nano> or is there a kind of verbose mode ? [13:03] <klaxa> you can check config.log i guess [13:03] <klaxa> not sure whether or not that will help a lot though :X [13:04] <dentex> klaxa: with "did you add -lmp3lame -L/path/to/libmp3lame to ./configure?" you mean an flag when calling ./configure or somewhere inside the file? [13:04] <dentex> *as [13:04] <klaxa> flag when calling ./configure [13:04] <dentex> ok both unknown [13:05] <klaxa> :S [13:05] <dentex> [OT]are those from Mplayer?[/OT] [13:05] <klaxa> i'm not too familiar with compiling ffmpeg although i compile it myself every now and then, but cross-compiling is something different again [13:05] <Nano> the config.log says that it found the lame.h file but there seems to be a reference to the original lame configure location, which doesn't exist any longer.... [13:07] <dentex> klaxa: I know, thanks. And I'm not an expert. [13:07] <Nano> dentex: what you're tryin todo ? [13:07] <dentex> HI: same as you. But crosscompiling for android [13:07] <Nano> :-) [13:08] <Nano> and what's youre problem ? [13:08] <dentex> I cannot make the build script find the lame libs [13:08] <dentex> 5sec [13:08] <dentex> ... [13:08] <Nano> did you configure/make/make install lame ? [13:08] <dentex> it works with a standard set of flags. But when I add '--enable-libmp3lame' don't [13:08] <dentex> <dentex> I have also compiled liblame for android, but I don't know where to put the libs for the ffmpeg compiler to find them; or what to change to point there. [13:09] <dentex> it installs for the host system, but useless for the android build. Anyway I tried [13:09] <dentex> libs not found [13:09] <dentex> same error as for you [13:10] <dentex> do you have any hint? [13:10] <Nano> the make install copiesthe libs to /usr/local/lib [13:10] <dentex> right [13:10] <Nano> see if you have libmp3lame.a in there [13:11] <Nano> for me copying the lame/lame.h failed [13:11] <dentex> 1) no [13:11] <dentex> 2) same [13:11] <Nano> so i created a folder lame in /usr/local/include and copied the lame.h in it. [13:13] <Nano> and then i added the --extra-cflags=-I/usr/local/include to the "./configure --enable-libmp3lame" call [13:13] <Nano> cause the configure script is looking for lame/lame.h file [13:13] <dentex> Nano: did you try adding '--extra-ldflags=-L/usr/local' ? [13:14] <Nano> i've added --extra-ldflags=-L/usr/local/lib [13:16] <dentex> yes, with /lib [13:18] <Nano> ok, the config.log seems to be helpful now. [13:18] <Nano> there i get: error: undefined reference to '_fpgetmask' [13:18] <Nano> /usr/local/lib/libmp3lame.a: error: undefined reference to '_fpsetmask' [13:18] <Nano> collect2: ld returned 1 exit status [13:18] <Nano> ERROR: libmp3lame >= 3.98.3 not found [13:19] <Nano> so both files (header and lib) has been found. but theres a build error in lame. [13:20] <Nano> dentex: check your config.log [13:21] <Nano> if you need help put the output in pastebin [13:21] <dentex> Nano: ok thanks [13:35] <blight> hi [13:36] <blight> can someone tell me what kind of subtitle support ffmpeg has and how to use the API? [13:37] <nilsge> yesterday I searched and tried for hours how to get proper sound for live streaming (together with burek). It is now stable and the quality is good, but I still have a 4 second delay. Video is first, audio is behind. I tried reducing the quality of the audio, but to no audible effect. I searched the web now for an hour again, without luck. So: [13:37] <nilsge> do you know any parameters or tricks to make sure a+v are in sync? [13:38] <nilsge> my current ffmpeg commandline is: ffmpeg -f x11grab -s 640x480 -r 30 -i :1.0 -f alsa -ac 2 -acodec pcm_f32le -i hw:Loopback,0,1 -c:v libx264 -preset fast -pix_fmt yuv420p -s 640x480 -c:a libfaac -ab 128k -threads 0 -f flv "rtmp://live.twitch.tv/app/*********************" & [13:40] <dentex> Nano: http://pastebin.com/K0hfymDF [13:47] <nilsge> additional info to the sync problem: if I record instead of streaming everything looks fine. So I guess it isn't really my fault? [13:48] <rascal999> i have nice --20 ffmpeg -y -f alsa -ac 2 -ar 48000 -i hw:0,0 -f x11grab -r 30 -s 960x540 -i :0.0 -acodec pcm_s16le output.wav -an -vcodec libx264 -preset ultrafast -crf 0 output.mp4 -- which generates and audio and video output, is it possible to create one file with audio and video in it? [13:53] <XATRIX> Hi guys, is it possible to say ffmpeg to encode my mp3 with fraunhofer instead of lame ? [13:57] <JEEB> XATRIX, I don't think fraunhofer's mp3 encoder is available as open source [13:57] <JEEB> unlike their AAC encoder [13:57] <XATRIX> Currently i'm using such a string to catenate multiple files into the one big mp3 file [13:58] <XATRIX> /usr/bin/ffmpeg -i "concat:temp790-1.mp3|temp790-2.mp3|...." -acodec copy output.mp3 [13:58] <XATRIX> But i have a strange issue [13:58] <Nano> dentex: this is the problem i've mentioned [13:58] <XATRIX> Sometimes the output mp3 is not readable by the JS players via web [13:59] <XATRIX> Or it doesn't play the whole mp3 ifles [13:59] <XATRIX> Or some mp3 files missing inside the big one [13:59] <Nano> dentex: call './configure --enable-static --disable-shared --disable-rpath --disable-frontend --disable-gtktest' in lame-3.xx.x folder [14:00] <Nano> dentex: then 'make', then 'make install' [14:00] <Nano> dentex: then copy the lame-3.xx.x/include/lame.h to /usr/local/include/lame [14:01] <JEEB> hmm, shouldn't it copy the header automagically/ [14:01] <JEEB> ? [14:01] <JEEB> during 'make install' [14:11] <XATRIX> how can i list available codecs that i can use to encode my audio ? [14:12] <dentex> Nano: Ok I'm trying [14:12] <dentex> thanks [14:13] <JEEB> XATRIX, I'm not sure if there's a way to only output the audio ones, but ffmpeg -codecs lists all codecs in that ffmpeg [14:13] <XATRIX> yea, thanks [14:13] <dentex> Nano:I'm just not sure that using a toolchain ./configure will look there [14:13] <JEEB> I guess you could use grep or something after -codecs [14:14] <JEEB> because third symbol would be 'A' [14:14] <JEEB> for audio [14:17] <dentex> NAno [14:18] <dentex> Nano: I can't 'make' and then 'make install' in the lame-3.xx.x folder because I already have the libs built for android [14:18] <dentex> Nano: I don't need the libs built for the host system in /usr... [14:18] <dentex> sorry [14:19] <dentex> I just don't know how to make the build script know where they are. [14:20] <makara> any idea why this fails: avconv -i Track\ 1.m4a -acodec copy -f aac track1.aac [14:21] <dentex> makara: did you tried 'avconv -i Track\ 1.m4a -vn -acodec copy track1.aac' ? [14:21] <JEEB> does it tell you that you need a bit stream filter? [14:21] <JEEB> to ADTS [14:21] <dentex> makara: should work [14:22] <makara> dentex, does, but I need to build a script to go through multiple directories, so I want to leave out the output filename and force format type [14:22] <makara> says: Requested output format 'aac' is not a suitable output format [14:23] <makara> but 'ffmpeg -formats' includes aac [14:25] <dentex> makara: is Track\ 1.m4a a video file? [14:26] <makara> no, it works fine with the line you gave me [14:26] <makara> but why it says aac not acceptable format? [14:26] <dentex> :S [14:27] <dentex> but you can leave out '-vn' [14:28] <dentex> makara so you always need .aac output format? from different input types? [14:28] <dentex> all audio input? [14:32] <makara> i'm trying to convert all my m4a's into straight aac's so that I can distinguish between the alac's I'm getting, which also - gag me with a spoon - come as m4a's. [14:34] <relaxed> why don't you ffprobe to see instead of demuxing everything? [14:34] <relaxed> You're probably losing the metadata with that method. [14:37] <dentex> then why don't you simply do it with '-i track.m4a track1.aac' [14:38] <dentex> it will force .aac from the extension [14:48] <makara> no re-encoding, and as relaxed pointed out I'd like to keep metadata [14:51] <makara> relaxed, metadata is preserved conversion [14:58] <XATRIX> guys, can you support me a bit ? I'm totally in a trouble. Can you tell me was the ecoding OK ? Or there was problem ecnountered ? http://fpaste.org/f1Bv/ [14:58] <makara> ok, I see now. can demux but not mux aac [15:01] <makara> raw aac [15:06] <XATRIX> ? [15:07] <Fjorgynn> ? [15:08] <XATRIX> http://fpaste.org/f1Bv - is it ok ? Oh some troubles detected ? [15:09] <XATRIX> s/Oh/Or [15:30] <XATRIX> Guys, any idea how to control "pitch" while encoding ? [15:30] <XATRIX> The trouble is i have a few files to concat [15:30] <XATRIX> It has mixed, mp3 en/ru voice [15:31] <XATRIX> And, after i finally get result mp3, it has en voice ok, but the ru voice is changed, it become like a robo-voice [15:31] <XATRIX> a lower pitch rate [15:33] <XATRIX> Just for example this is the initial ru voice, http://devel.rf.ua/1359732328_118_1.mp3 . This is the file catenated with en.mp3 + ru.mp3 + en.mp3 http://devel.rf.ua/output1.mp3 [15:33] <XATRIX> I do encoding with #ffmpeg -i "concat:ru1359813352_0.mp3|1359732328_118_1.mp3|1359732328_118_2.mp3|ru1359813352_0.mp3" -acodec libmp3lame /mnt/www/rf.ua/output1.mp3 [15:34] <XATRIX> I'm totally screwed up with it :( [15:34] <XATRIX> I have no idea what's the trouble it has while encoding [15:35] <klaxa> shouldn't you use -f for concat? [15:36] <Keshl> Different bitrates, oÉo? Dunno, not an expert here. [15:41] <relaxed> XATRIX: -acodec copy [15:47] <XATRIX> relaxed: the same crap it does [15:47] <XATRIX> Keshl: the bitrates are the same [18:35] <pietro10> I have ffmpeg from https://launchpad.net/~jon-severinsson/+archive/ffmpeg with all the -extra packages and it's complaining that there is no vcodec libx264 when I have libx264 installed on mys ystem. What am I doing wrong? Thanks. [18:38] <pietro10> er unknown decoder libx264 [18:39] <pietro10> http://pastie.org/6030257 here is what I am trying to do [18:39] <pietro10> following the -c copy instructions http://ffmpeg.org/trac/ffmpeg/wiki/Creating multiple outputs [18:46] <pietro10> if I take out the decoder from the second part of the pipeline (just have -f flv), ffmpeg complains with [18:46] <pietro10> pipe:: could not find codec parameters [18:55] <sgfgdf> hello, guys! i have a video with the following audio properties: "Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s (default)". now i want to convert it to stereo AAC audio. i use this arguments -- "-c:a libfaac -ar 48000 -ab 384k -ac 2" and the output has the following audio properties: "Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 165 kb/s". so the bitrate is not the same. why is this happening? [18:56] <pietro10> am I going to hae to do a parallel encoding? [19:38] <pietro10> removing -f flv and just leaving -i - and -c copy gives me "pipe:: Invalid data found when processing input [19:38] <pietro10> " [19:45] <pietro10> am I going to have to repeat the options? [20:21] <Youka> What's the common way to get the frame number from decoded AVFrame? [20:28] <Mavrik> Youka: what do you mean by frame number? [20:29] <Youka> The frame index. [20:30] <Youka> I want it from AVFrame, not counting manually be decoding. [20:30] <Youka> *by [20:32] <Mavrik> Youka: you have frame_number field on your decoder context [20:32] <Mavrik> there are also picture_number fields on AVFrame itself, but I don't see them being set on decoding [20:37] <Youka> frame_number of codec is what's already decoded, so av_seek_frame isn't usable. [20:39] <Mavrik> av_seek_frame takes timestamp not frame number. [20:44] <Youka> thx, i will use coded_picture_number, this seems to solve my problem. frames are not always aligned, so i have to look forward sometimes, but when i have the number of frames i wanted, i can stop [20:45] <Mavrik> um [20:45] <Mavrik> you did hear me that av_seek_frame doesn't take the frame count, but timestamp right? [20:45] <Youka> right [20:58] <mantas322> Hi guys [20:59] <Mavrik> hey [20:59] <mantas322> I'm trying to convert from a .MOV video type to a avi or mpg [20:59] <mantas322> however when i do I lose colors [20:59] <mantas322> twindows says the "Type" is QuickTime Movie [21:00] <Mavrik> mantas322: can you please paste your command line and full output of then command to pastebin? [21:00] <Mavrik> alot easier to help you that way? :) [21:00] <Mavrik> also, what is your goal? where are you trying to play the output file? [21:00] <mantas322> i've tried all sorts of different combinations [21:00] <mantas322> my goal is I want tthe same quality [21:00] <mantas322> and dont want to lose any colors [21:01] <mantas322> ffmpeg -i C:\ffmpeg\bin\1.mov C:\ffmpeg\bin\OUT1.avi [21:01] <mantas322> -i C:\ffmpeg\bin\1.mov -f avi -vcodec mpeg4 -qscale 4 -g 300 -vtag DIVX -acodec libmp3lame -ab 192k -ar 44100 -ac 2 -y output.avi [21:02] <mantas322> I would assume that a simple "ffmpeg -i C:\ffmpeg\bin\1.mov -qscale 0 C:\ffmpeg\bin\OUT1.avi" should get me the same type of video [21:03] <mantas322> anyways I think im being too picky [21:03] <mantas322> its really not that big of a deal. [21:04] <LithosLaptop> mantas322: does this work? ffmpeg -i C:\ffmpeg\bin\1.mov -f avi -vcodec copy -acodec copy -y C:\ffmpeg\bin\OUT1.avi [21:04] <mantas322> lemme try [21:04] <Mavrik> mantas322: .avi can contain a lot of different video and audio formats [21:05] <Mavrik> if you don't pass them ffmpeg will choose a default [21:05] <Mavrik> if you do a copy you'll get another one [21:05] <mantas322> LithosLaptop, no I lose video [21:05] <mantas322> sound only [21:05] <Mavrik> that's why I asked you "What are you trying to achieve?" [21:05] <LithosLaptop> oh ok [21:05] <Mavrik> so we know which format you're trying to create [21:05] <Mavrik> also you still didn't paste full command output so we still don't know what your input file is. [21:05] <mantas322> it doesnt have to be avi [21:05] <mantas322> it could be mpg mpeg [21:05] <Mavrik> "avi" is a format. [21:06] <Mavrik> not video codec, not audio codec [21:06] <Mavrik> what codecs do you need on the output? [21:06] <mantas322> 1.MOV QuickTime Movie from a s100 cannon camera [21:06] <Mavrik> what device are you trying to play the output on? [21:06] <mantas322> whatever codecs achieve the original video quality [21:06] <mantas322> windows movie maker [21:07] <mantas322> I'm trying to make a cute kitten video for youtube [21:07] <Mavrik> ah I see [21:07] <LithosLaptop> oh do you want to be able to use your video in windows movie maker? [21:07] <mantas322> yes. [21:07] <mantas322> but it doesnt except .mov [21:08] <Mavrik> mhm [21:08] <Mavrik> mantas322: can you please do a "ffprobe" on your .mov file and paste output to a pastebin? [21:09] <mantas322> okay [21:09] <Mavrik> it's easier to tell you what to do when we know what resolutoin, quality and pixel format your video is in ;) [21:09] <mantas322> http://imgur.com/Q2qaTLq [21:10] <LithosLaptop> h.264 [21:10] <LithosLaptop> hmm [21:11] <Mavrik> yeah, not supported in movie maker [21:11] <mantas322> is there a free alternative to windows movie maker? [21:11] <Mavrik> mantas322: ffmpeg -i "1.MOV" -vcodec mpeg2video -vb 20000k -acodec mp2 -ab 128k output.mp2 [21:12] <Mavrik> should work [21:12] <Mavrik> without significant loss of quality [21:12] <Mavrik> if I see correctly movie maker will take mp2 files [21:13] <mantas322> 200mb to 700kb [21:13] <mantas322> sound only [21:13] <mantas322> didnt work, but thanks anyways. [21:13] <LithosLaptop> or maybe uncompressed [21:14] <LithosLaptop> what -vcodec is uncompressed? [21:14] <Mavrik> LithosLaptop: none recognised by movie maker :\ [21:14] <Mavrik> plain yuv output [21:14] <Mavrik> mantas322: what? [21:14] <mantas322> your command didnt work, it makes a 700kb mp2 file [21:15] <mantas322> form a 200mb mov file [21:15] <mantas322> "ffmpeg -i "1.MOV" -vcodec mpeg2video -vb 20000k -acodec mp2 -ab 128k output.mp2" [21:16] <LithosLaptop> somebody had the same problem: http://forum.doom9.org/archive/index.php/t-152345.html [21:16] <LithosLaptop> looks like windows movie maker is a bit picky [21:17] <mantas322> oh well [21:17] <Mavrik> mantas322: a 20Mbit/s video can't be 700kb -_- [21:18] <mantas322> no i was discribing what happened when i ran your command [21:18] <mantas322> input file was 200mb and it created an mp2 output file with sound only 700kb [21:57] <sgfgdf> guys, is it possible to use QuickTime AAC codec to encode audio with ffmpeg? [22:00] <pietro10> is it possible to have alsa record the speaker output or the sound from a specific program, not my mic? [22:00] <pietro10> right now I have -f alsa -ac 2 -i pulse \ [22:08] <LithosLaptop> sgfgdf: don't think so. the best AAC output you would be able to get from ffmpeg is with fdkaac [22:15] <sgfgdf> LithosLaptop, i don't know if my message got delivered? i got disconnected. [22:15] <LithosLaptop> didn't receive any messages [22:16] <JEEB> fdk-aac with -afterburner 1 is the best you get with ffmpeg [22:16] <JEEB> which should actually be pretty good [22:16] <JEEB> regarding AAC encoder [22:16] <JEEB> *encoding [22:16] <sgfgdf> LithosLaptop, is the fdkaac and libfaac the same thing? [22:17] <LithosLaptop> nope [22:17] <JEEB> faac is quite worse [22:17] <LithosLaptop> libfaac is old and inferior quality encoder [22:18] <JEEB> it's one of the reference code based encoders, not to mention it was improperly licensed (the reference code wasn't GPL-compatible although the creator tried to pass faac as GPL IIRC) [22:18] <JEEB> the other reference code encoder, vo-aacenc is even worse than the internal libavcodec aac encoder, which is quite hard to achieve to be honest :D [22:19] <sgfgdf> so if i install fdk-aac how should i tell ffmpeg to use it as audio ancoder? -c:a libfdk-aac? [22:19] <JEEB> libfdk_aacenc [22:19] <JEEB> you have to enable it naturally in ffmpeg as well [22:19] <JEEB> when compiling it [22:20] <JEEB> compile fdk-aac -> compile ffmpeg with --enable-libfdk_aacenc (I think it was like this?) -> see it with ffmpeg -codecs |grep "aac" [22:20] <sgfgdf> JEEB, where can i check the references like fdk-aac -> libfdk_aacenc and other libraries matching? [22:22] <JEEB> ./configure --help [22:22] <JEEB> see the External library support: aprt [22:22] <JEEB> *part [22:28] <sgfgdf> JEEB, does people encode their video using ffmpeg and then use external audio encoder such as QuickTime to encode the audio if they want the QuickTime AAC audio? is this popular practice? [22:29] <Mavrik> nah, too much work [22:29] <JEEB> with ffmpeg usage not really, but I've seen that kind of usage with people who use avisynth + x264 and so forth more manually for encoding. Might have been more popular before, but now with fdk-aac I think most people are OK with that quality. [22:29] <JEEB> To be honest, most people were OK with faac, which wasn't that great to begin with, so :D [22:29] <JEEB> fdk-aac is fraunhofer's [22:30] <JEEB> and with -afterburner 1 it should be pretty damn good :3 [22:30] <sgfgdf> JEEB, what does this option do? [22:30] <JEEB> enables fdk-aac's mode with the same name [22:31] <JEEB> seems to enable slower algorithms for better quality [22:34] <sgfgdf> JEEB, where can we put the Nero AAC and the default aac which comes with ffmpeg (which i don't know how to name it)? so far -- 1. QuickTime AAC; 2. fdk-aac; 3. faac; 4. vo-aacenc; if i understood you correctly. [22:35] <JEEB> I wouldn't necessarily put QT's AAC on top [22:35] <JEEB> I remember fraunhofer's other versions doing rather well on tests [22:35] <JEEB> @ hydrogenaudio [22:37] <LithosLaptop> Nero AAC is alot better than FAAC [22:37] <sgfgdf> JEEB, is Nero AAC somewhere near QT AAC and fdk-aac? [22:38] <LithosLaptop> Nero AAC/QT AAC/fdk-aac are very close to each other. hard to judge [22:38] <sgfgdf> JEEB, i mean the three above to be almost equal? [22:38] <sgfgdf> ah okay [22:38] <JEEB> QT | nero | fdk-aac > faac > ffaac (internal) > vo-aacenc [22:38] <sgfgdf> and about ffmpeg's aac (how should i call it?)? [22:40] <sgfgdf> are all of the codecs free for personal use? [22:41] <sgfgdf> and which of the codecs will charge you if you use them for commercial purposes? [22:41] <JEEB> I have no idea about the licensing of nero or QT's encoder for any commercial purpose [22:43] <JEEB> all the others have a license that you can read (Except for faac, that one tries to be GPL, but it isn't because it uses non-GPL reference encoder code) [22:43] <JEEB> and then there's the whole thing about taking an AAC license from VIA licensing [22:44] <JEEB> if you are going to do business it's better to ask a lawyer if you don't want to license AAC [22:45] <sgfgdf> JEEB, no, i was just curious. [22:45] <JEEB> I think I had someone try license nero for their encoding chain [22:45] <JEEB> they didn't get an answer [22:45] <JEEB> not sure if you can license QT either *shrug* [22:47] <JEEB> vo-aacenc is Apache2, ffaac is LGPL, faac is an invalid mish-mash of supposedly GPL code and one of the reference encoders, fdk-aac has a custom license from fraunhofer that is not (L)GPL compatible [22:47] <JEEB> most of these software licenses only apply towards distributing binaries [22:47] <JEEB> and the rules you have to take in case you distribute builds of them, or use the libraries somewhere [22:48] <JEEB> I think fdk-aac's license told you you had to license AAC (via VIA licensing methinks) to distribute it [22:48] <JEEB> (binaries) [22:48] <sgfgdf> btw i forgot to ask about aacplus, but we talked with a guy here about it. it appears to achieve a good quality in very low bitrate. [22:49] <JEEB> that's HE-AACv2 specific methinks, fdk-aac can do it too [22:49] <JEEB> HE-AAC(v2) is a version of AAC that cuts more information out [22:49] <JEEB> it's specifically meant for use cases of 64kbps or less [22:49] <JEEB> aka "it probably sounds close enough for the bit rate" [22:50] <sgfgdf> JEEB, is it normal to use it for encoding a movie audio? [22:50] <JEEB> if your primary need is to lower the bit rate [22:51] <JEEB> if you want keep quality, higher bit rate and LC-AAC cuts less information out [22:54] <sgfgdf> JEEB, if i lower the bit rate i will get smaller file, right? so i think i can't distinguish a difference while using 64kb/s aacplus encoded audio and 192kb/s AAC audio (which i don't know the encoder of). [22:54] <LithosLaptop> you won't expect to see HE-AAC on normal/high bitrate 720p or 1080p videos. LC-AAC more suitable [22:55] <JEEB> sgfgdf, yes -- so if you're encoding for very low bit rate scenarios, HE-AACv2 is very useful [22:55] <JEEB> as you can push the audio down to 48kbps or so while still kind of feeling similar [22:55] <sgfgdf> JEEB, what can be a such scenario? [22:56] <sgfgdf> i mean if i had to choose i will choose smaller filesize. [22:56] <sgfgdf> but you said that it is very unlikely to see it in HD videos. [22:57] <LithosLaptop> the reason is HE-AAC can't acheive transparency [22:57] <JEEB> yes, because in such scenarios generally the video bit rate is already so high that 48kbps becoming 192kbps for better audio quality doesn't really cost you much at all [22:57] <JEEB> and yes, HE-AAC(v2) in general cuts more information out [22:59] <sgfgdf> LithosLaptop, what should i understand under transparency? [22:59] <LithosLaptop> sound just like the original source [23:01] <sgfgdf> LithosLaptop, but isn't it just like the original if i can't distinguish a difference between a video and then the same video re-encoded with HE-AAC(v2)? [23:01] <LithosLaptop> 64Kbps HE-AAC sounds pleasant(not annoying like 64Kbps LC-AAC would), but not transparent. in other words it won't sound like the original [23:02] <sgfgdf> probably it's just me and other people will notice it. i don't know. [23:02] <LithosLaptop> If you can't distinguish then it is transparent to you, but I find it hard to beleive. Are you using headphones? [23:09] <sgfgdf> LithosLaptop, yes, but i'm pretty bad at such things, so probably it is just me. everytime i try to do such a comparison i find myself really struggling. [23:11] <sgfgdf> JEEB, LithosLaptop, thank you a lot for the help guys! [23:11] <LithosLaptop> no prob [00:00] --- Sun Feb 3 2013

1 0

ffmpeg-devel.log.20130201
by burek 02 Feb '13

02 Feb '13

[00:14] <Compn> pross has a bayer swscale eh [00:15] <Compn> now i forgot what codec it was that needed that [00:17] <pigoz> is swig supposed to run ok on libavformat headers? seems like it doesn't like attribute_deprecated at all. Maybe I'm doing something terribly wrong. swig -cffi path/to/avformat.h [00:18] <pigoz> oh well I'll figure out tomorrow :) [00:56] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:131fac1c12ed: vc1dec: fix block_off [00:56] <cone-469> ffmpeg.git 03Sebastian Sandberg 07master:c6779c513117: vc1dec: fieldtx is only valid for interlaced frames [00:59] <michaelni> nevcairiel, do you want to review the remaining patches from "Sebastian Sandb ( *K) [FFmpeg-devel] [PATCH */3] vc1dec: *" ? [01:10] Action: Compn pings bayer rgb patch [01:11] <Compn> michaelni : should i ask kostya about those patches ? [01:14] <Compn> vc1 patches that is [01:15] <michaelni> Compn, nevcairiel worked on interlaced vc1 [01:16] <Compn> ah [01:16] <michaelni> i dunno if kostya did, but he is of course welcome to review/approve/appy the patches [01:16] <michaelni> apply [01:16] <Compn> kostya did not iirc [02:09] <cone-469> ffmpeg.git 03Allan Kristensen 07master:4789c25d83c4: matroska: fixed missing S_DVBSUB subtitles [03:20] <cone-469> ffmpeg.git 03Michael Bradshaw 07master:c58c67398d03: libopenjpegenc: add support for pix fmt gbrp (8-16 bit) [03:20] <cone-469> ffmpeg.git 03Michael Bradshaw 07master:28f36dce3e5c: libopenjpegenc: simplify switch statement [03:20] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:ffd02d0c1974: Merge remote-tracking branch 'mjbshaw/master' [03:39] <highgod> Hi,I use git to get the latest version, but I can't comile correctly.There seems some errors in file libm.h,./libavutil/libm.h:52:32: error: static declaration of 'cbrt' follows non-static declaration [03:53] <michaelni> highgod, you did try make distclean ? [03:58] <Compn> highgod : did you rm the old checkout and pull a new one ? [04:02] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:086566a557ee: dict: fix memleak [04:50] <highgod> Hi,I create a new folder and get the code [05:00] <Compn> does it work highgod ? [05:05] <highgod> I try to reget again,Thanks Compn. [05:05] <highgod> I want to add our openCL filter code in the latest code.we will add a OpenCL wrapper.Is it OK? [05:15] <Compn> highgod : you should ask michaelni [05:15] <Compn> someone on the list said that opencl was only good for newest intel cpu [05:16] <Compn> i dont know enough about opencl to make an opinion [05:30] <highgod> OK, I think add a option to make user can choose whether to use CPU or GPU [05:31] <highgod> I reget the code, but the same error [05:38] <Compn> hmmm [05:39] <Compn> highgod : what ./configure options are you using ? [05:42] <highgod> yes,just ./configure [05:42] <highgod> and my git set autocrlf = false [05:44] <Compn> are you using ccache ? [05:44] <Compn> what are you compiling on ? [05:44] <Compn> what arch [05:45] <Compn> highgod : we ask, because our automated ffmpeg system says everythings ok http://fate.ffmpeg.org/ [05:46] <highgod> no,I just compile on my computer using mingw [05:47] <highgod> I'm confused that why alldevice.o error. [05:48] <Compn> paste make output to pastebin or similar site [05:49] <wm4> Compn: tired of mplayer support? [05:50] <Compn> wm4 : why, people asking for me to return / [05:51] <highgod> sorry,Compn,what is pastebin? [06:00] <Compn> highgod : pastebin.com or pastebin.ca [06:01] <Compn> highgod : just a website you can paste text at for public share [06:01] <Compn> it may be banned in certain countries.... [06:01] <highgod> Thanks [06:01] <Compn> or just put the make output somewhere we can see it [06:02] <Compn> but not in the channel , too much text [06:03] <highgod> http://pastebin.ca/2309543 [06:05] <Compn> highgod : did you try rm libavutil/libm.h && git pull [06:06] <highgod> Compn : No,but I create a new folder and get the code.Should I need to do this operation? [06:07] <Compn> cant hurt to try :) [06:07] <Compn> but i dont know whats wrong [06:07] <highgod> OK,Thanks [06:07] <highgod> I will try [06:08] <Compn> your copy of libm.h looks broken , does it have the windows line endings again ? [06:08] <Compn> dos2unix app might fix it [06:15] <Daemon404> disable autocrlf in git. [06:20] <highgod> Daemon404 : I have set the autocrcf = false [06:20] <Daemon404> you need to re-clone after ofc [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:cb30bf7f03ac: doc/developer: Add web links for all suggested licenses [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:8b0fda180e99: doc/developer: Add ISC license to list of acceptable licenses [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:8787847dc6b3: doc/developer: Drop obsolete MPlayer reference [14:24] <cone-686> ffmpeg.git 03Anton Khirnov 07master:f2960097e42d: bink: fix a check for the first frame. [14:24] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:69a6eddd607a: Merge commit 'f2960097e42ddf9a356bab6547f87906f6999e0a' [14:29] <cone-686> ffmpeg.git 03Anton Khirnov 07master:1730ca2eca42: bink demuxer: check malloc return value [14:29] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:1bc51a7a74fc: Merge commit '1730ca2eca42974af0c29f38a0770997cba2f0da' [14:39] <cone-686> ffmpeg.git 03Anton Khirnov 07master:9ec8971060a0: bink demuxer: set framerate. [14:39] <cone-686> ffmpeg.git 03Diego Biurrun 07master:4eef2ed70744: ppc: fmtconvert: Drop two unused variables. [14:39] <cone-686> ffmpeg.git 03Diego Biurrun 07master:6c1a7d07eb41: Use proper "" quotes for local header #includes [14:39] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:911e27068840: Merge remote-tracking branch 'qatar/master' [14:43] <divVerent> how are patches to be submitted BTW? have a question not answered on the ffmpeg.org/developer.html page [14:43] <divVerent> just git-send-email? but how would I then put in extra explanation (screenshot links, etc.) [14:44] <divVerent> or are attachments allowed on the list [14:46] <iive> attachments are ok, afaik [14:52] <Compn> divVerent : its best to put attachments and screenshots on the bug trac [14:52] <Compn> i think [14:53] <Compn> otherwise you'll be sending a screenshot to 1000+ subscribers (however big ffmpeg-devel is) [14:55] <Compn> oh [14:55] <Compn> i misunderstood [14:55] <Compn> :D [14:56] <Compn> divVerent : isnt there a intro message to get-send-email where you can add links and description and stuff ? [14:56] <Compn> i've seen people send them, a 0/* mail [14:56] <JEEB> yeah, cover thingy [14:56] <JEEB> cover page or whatever it was [14:56] <Compn> yeah that [15:00] <divVerent> Compn: I'd have sent the screenshot as link [15:00] <divVerent> 14:55:54 @Compn | divVerent : isnt there a intro message to get-send-email where you can add links and description and stuff ? [15:00] <divVerent> not that I know of [15:00] <divVerent> but well, can't use git-send-email directly anyway [15:01] <divVerent> or rather, not from the email address I would subscribe [17:48] <Marco-123> hi all. I have a board with an ARM processor (and a NEON hardware accelerator), which is running Ubuntu Linux v10.04. I am building ffmpeg and my goal is to make sure it is taking full advantage of the hardware on the board. It's a freescale IMX53 btw. After much googling, I set a few flags in the configure script, and the script tells me "NEON enabled: yes" but the encoding performance is quite slow (same as when it says " [17:48] <Marco-123> can someone provide advice on how to build this project properly? [17:48] <Marco-123> or if there is a conclusive way to tell if the hardware accelerators are being used? another point is that the configure also tells me "enabled hwaccels: " and nothing is listed there. [17:51] <durandal_1707> the options in table for lut(component options), afade(curve types) are displayed wrong, who is doxygen expert? [17:53] <divVerent> durandal_1707: speaking of the power-of-two division issue BTW which patcheck shows on my patch... just saying because I saw some other post about it [17:53] <divVerent> I intentionally ignored it, as I want to highlight that 32768 is the max sample value of S16 [17:53] <divVerent> so >> 15 is not really an option :P [17:54] <divVerent> oh, and I evaded the patcheck message this way :P [17:54] <divVerent> const double w = 1. / (sqrt(nb_freq) * 32768.); [17:54] <durandal_1707> what patcheck reported? [17:54] <divVerent> and it now says "sqrt(nb_freq) * 32768. is the max value of a fft bucket" [17:54] <divVerent> const double w = 1. / sqrt(nb_freq) / 32768.; [17:54] <divVerent> I had this in a previous version [17:54] <divVerent> then patcheck wanted me to replace the / 32768 by >> 15 [17:55] <divVerent> which doesn't even work for double :P [17:55] <durandal_1707> ahh, patcheck have own bugs... [17:55] <divVerent> just mentioning it, as some other mailing list post mentioned the same patcheck message and that it's useless now :P [17:55] <divVerent> especially given that we can expect compilers to handle this too [17:56] <divVerent> or rather, "mostly handle it" [17:56] <divVerent> i.e. for unsigned types [17:56] <durandal_1707> what other mailing list? [17:56] <divVerent> ffmpeg-devel [17:56] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07master:99eedfc40086: doc/muxers.texi: Fix mp3 picture attachment documentation. [17:56] <divVerent> there was some other unrelated post [17:56] <divVerent> right after I subscribed [17:56] <divVerent> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/138323.html [17:56] <divVerent> this one, last two lines [17:57] <divVerent> oh, oops... haha [17:57] <divVerent> I thought this referred to patcheck, but it does not :P [17:57] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:5c1a1b47f378: doc/muxers.texi: Fix mp3 picture attachment documentation. (cherry picked from commit 99eedfc40086972987aa27df8b1259c8bf15b20c) [17:57] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:9bcb84810f53: doc/muxers.texi: Fix mp3 picture attachment documentation. (cherry picked from commit 99eedfc40086972987aa27df8b1259c8bf15b20c) [18:00] <divVerent> durandal_1707: but anyway, I did notice more false positives with patcheck :P [18:00] <divVerent> it really seems just a crude tool [18:01] <divVerent> possibly never written:intensity_colors [18:01] <divVerent> possibly constant :intensity_colors [18:01] <divVerent> e.g. it says that for the options [18:02] <durandal_1707> patcheck is mostly used for common typos you could not spot and style issues [18:02] <divVerent> yes [18:02] <divVerent> it did find accidental tabs in my code that I then fixed [18:05] <michaelni> Marco-123, you can add some av_log/printf into a neon function to be really sure if its used [18:08] <Compn> grumble [18:08] <Compn> where is mov_text in the documentation ? [18:08] <Compn> hey cehoyos :) [18:09] <Marco-123> michaelni: at the moment i am using a library that works with ffmpeg so I don't issue ffmpeg commands directly. However, for this purpose I guess I should! Would you elaborate more on what I can look for when issuing that command to tell if neon is being used? Im guessing that because its a neon function it wont even work if neon isnt being used. [18:09] <cehoyos> Hi [18:11] <cehoyos> Compn: [15:12] <Compn> durandal_1707 : yes, but is it fixed or is it a mis-duplicate tag ? :P <-- I don't understand this sentence... [18:15] <Marco-123> michaelni: thanks for your help, too. [18:18] <Compn> cehoyos : i just wanted someone to verify the bug was fixed [18:19] <Compn> since it was marked duplicate, that sentence is asking if it was a valid duplicate bug [18:19] <Compn> or a new bug [18:20] <Compn> interesting [18:21] <Compn> i couldnt get quicktime to open our ttext sample [18:22] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:b868275ddbec: doc/muxers: fix level for the segment options subsection [18:22] <Compn> saste : where in the docs is the "3gpp timed text" subtitle option ? [18:23] <saste> Compn, why do you ask me? [18:23] <Compn> because you are active ? :) [18:23] <Compn> and i see you just commit some docs :) [18:23] <Compn> i'd like to fix it, if i can figure out if its missing. or it maybe that i am blind [18:23] <saste> Compn, i don't even know what "3gpp timed text" is [18:24] <Compn> mov text subs format [18:24] <saste> ubitxu? [18:24] <Compn> hes in japan [18:24] <Compn> soaking up all that radiation [18:24] <saste> i don't think it is documented at all [18:25] <saste> yes japan is cheap since then [18:25] <durandal_1707> saste: the lut rgb options and afade curve type options in doxy are displayed wrong what is correct syntax? [18:25] <saste> don't miss the opportunity [18:25] <Compn> so theres no list of subtitle formats and their options ? [18:25] <Compn> it would be cool to visit japan again [18:25] <saste> Compn, ffmpeg -h full may help [18:26] <Compn> saste : it doesnt list either [18:29] <saste> durandal_1707, item foo NL description [18:29] <saste> i'm going to fix lut [18:31] <Compn> http://ffmpeg.org/pipermail/ffmpeg-cvslog/2012-August/053256.html [18:31] <Compn> theres the commit [18:32] <cehoyos> Compn: Do you have any reason to assume it was not a duplicate? [18:34] <cehoyos> Compn: Muxing mov_text in mov is not QT-compatible, see ticket #1845 [18:34] <Compn> cehoyos : no reason. just curiosity [18:36] <Compn> ok thanks for bug 1845 [18:36] <Compn> we have mov_text samples [18:36] <Compn> that work in quicktime [18:36] <Compn> cehoyos : http://samples.ffmpeg.org/mov/subtitles-embedded/ [18:36] <Compn> at least http://samples.ffmpeg.org/mov/subtitles-embedded/BCDisassembly1RightSideTxt… works in quicktime, did not test the rest [18:38] <cehoyos> Compn: Which QT version did you test? [18:38] <Compn> 7.7.3 [18:39] <cehoyos> I ask because this file is around 10years older than the QT version mentioned in ticket 1845 [18:40] <cehoyos> Could you try ffmpeg -i BCDisassembly1RightSideTxt2.mov -codec copy -map 0 out.mov and test if out.mov shows the subtitles in QT? [18:40] <Compn> sure [18:41] <Compn> yes that works [18:41] <Compn> in quicktime [18:41] <cehoyos> Also with -vcodec mpeg4 -qscale 2 -strict -2 -acodec aac ? [18:42] <Compn> theres no -scodec listed [18:42] <Compn> do you want -scodec copy ? [18:42] <cehoyos> Could you test both -scodec copy and scodec mov_text ? [18:42] <Compn> sure [18:43] <Compn> -vcodec mpeg4 -qscale 2 -strict -2 -acodec aac -scodec copy works in quicktime [18:44] <Compn> -scodec mov_text shows weird things in qt [18:44] <Compn> like the text is 100 pt font stuck in a 20pt box, and color dark red maybe ? [18:44] <Compn> yes, if i fullscreen the video, i can see text [18:45] <Compn> kind of hard to explain :) [18:45] <cehoyos> You could try to enlarge the video radically: -s 1280x960 [18:47] <Compn> cehoyos : http://i.imgur.com/n7AeD4l.jpg [18:47] <Compn> thats with -scodec mov_text [18:48] <cehoyos> Thank you for the tests! [18:50] <Compn> in my screenshot, you can see that the subs are at the top with ffmpeg's reencode [18:50] <Compn> but in the original file, the subs are at the bottom... [18:51] <Compn> probably the alignment tag does nt get copied [18:51] <Compn> for some reason [18:51] <Compn> cehoyos : thank you for getting bug reports in order :) [18:52] <cehoyos> Regarding ticket #1840: Your suggestion for -c:s mov_text will not work in 50% of all cases... [18:52] <cehoyos> Especially not for the sample from ticket #756 [18:54] <Compn> if (textsubs) then [18:54] <Compn> i think dvdsub (vobsub?) in mp4/mov is rare [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:902fa77fd716: doc/filters: fix syntax and description for the lut* options [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:986540b65f6e: doc/filters: create a dedicated examples section for lut*, and itemize [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:423856c4ae85: doc/filters: fix syntax and typo for the afade curve option [18:56] <Compn> oh that cpcdemo.mov is Stream #0:2(eng): Subtitle: eia_608 (c608 / 0x38303663) [18:57] <Compn> thats different than mov_text :\ [18:57] <cehoyos> (Apart from the fact that dvdsub, dvbsub, pgssub and maybe even xsub are not rare:) How is the rareness related to giving bad advice? [18:58] <Compn> cehoyos : you are right, how about "please use -scodec (supported mov subtitle list)" then ? [18:58] <cehoyos> Looks like ticket #846 [18:58] <cehoyos> I think the current error message is the best possible [19:01] <Compn> ok, then feel free to close 1840 bug as duplicate of 1845 [19:02] <cehoyos> But 1840 was never a duplicate of 1845 (you insisted yesterday in making it one) [19:02] <Compn> comment #3 pretty much makes it his problem [19:02] <Compn> makes it a duplicate, i mean [19:03] <Compn> ok mark it invalid then :) [19:03] <Compn> no worries [19:03] <Compn> cehoyos : new problem, where is mov_text listed ? [19:03] <Compn> i think its missing in documentation [19:03] <Compn> i cant find supported subtitle list either [19:03] <cehoyos> No, comment three changed the original report to something (more important but) unrelated [19:04] <cehoyos> I sometimes commit documentation fixes (saste: I hope you are ok with my changes an hour ago), but I don [19:04] <cehoyos> ... I don't feel responsible for documenting features I don't know much about. [19:05] <Compn> i dont mind documenting it myself [19:05] <Compn> i mean [19:05] <Compn> i want to fix it [19:05] <Compn> i just dont know where it should go :) [19:05] <cehoyos> Then send a patch [19:06] <Compn> curious why this mov_text does not work, no video stream ? http://samples.ffmpeg.org/mov/subtitles-embedded/nova_eu_3012c01_caps.mov-c… [19:07] <Compn> it works in quicktime i mean [19:07] <Compn> but not ffplay [19:12] <cehoyos> compn: I believe it would work with ffplay but ffplay does not support mov_text (or does it and I just don't know how?) [19:13] <Compn> cehoyos : oh, i forgot ffplay doesnt support mov_text [19:13] <Compn> then its my bad [19:18] <Compn> cehoyos : are there any bugs i should look at ? [19:18] <cehoyos> Compn: mov_text is documented afaict: http://ffmpeg.org/general.html#Subtitle-Formats [19:19] <Compn> i'm not good with programming .. [19:20] <Compn> cehoyos : yes, i found it , mov_text is listed with ffmpeg -codecs [19:21] <Compn> i was trying ffmpeg -formats ... i'm dumb :( [19:21] <Compn> is a subtitle a format or codec, who knows :D [19:25] <durandal_1707> no more comments on showspectrum? [19:26] <saste> durandal_1707, wait a day or so, so ubitux can comment [19:26] <durandal_1707> ubitux: seriously? [19:28] <saste> durandal_1707, he's the author of the filter [19:28] <saste> i can't have a look at it now, i'll try to do it tonight [19:32] <durandal_1707> saste: not listed in MAINTAINERS for that file ........................ [19:34] <nevcairiel> he is listed as author at the top of the file =p [19:39] <durandal_1707> nevcairiel: so i would need to wait for every single persona persona at the top of the file? [19:40] <nevcairiel> its only one person, and if that person is an active commiter to ffmpeg, yes, you should [19:41] <nevcairiel> its just polite to ask the author to review a patch to his code [19:41] <durandal_1707> well i will just add another persona at the top of the file which will simplify things a lot [19:41] <nevcairiel> that doesnt change who the author of the file is [19:45] <durandal_1707> even if code was so badly writen that it needs rewrite or adds more code that was already present .... [19:57] <saste> durandal_1707, i'm asking you to wait a day, not till the end of the world [19:58] <durandal_1707> saste: but end of world is near!!!! [19:58] <saste> durandal_1707, no it's already passed [19:59] <saste> can't say when it will be the next one [20:03] Action: durandal_1707 will ask ubitux for special authorisation code [20:04] <Compn> lol [20:04] <Compn> durandal_1707 : work on something else in the meantime :) [20:04] <Compn> i remember ffmpeg used to have much longer patch review times [20:04] <Compn> 37 patch resubmissions :D [20:05] <Compn> wonder what the highest one was [20:05] <saste> durandal_1707, the-matrix-effect filter [20:05] <Compn> try 42 ? [20:06] <saste> durandal_1707, to say the truth, i have an unfinished port of it, based on jwz code [20:06] <durandal_1707> of what? [20:07] <durandal_1707> matrix effect? it still not in? [20:07] Action: durandal_1707 sad [20:08] <burek> is "Subtitle: dvb_teletext" the same as "Subtitle: dvb_subtitle" ? [20:09] <Compn> no [20:09] <Compn> well [20:09] <Compn> i dont know :D [20:09] <Compn> i dont think so [20:09] <nevcairiel> no its not the same [20:09] <nevcairiel> dvb has two kind of subtitles [20:10] <burek> it seems that ffmpeg converts dvb_teletext to dvb_subtitle when -c copy is used [20:10] <nevcairiel> sounds like a bug [20:10] <burek> yeah, i guess i should report it [20:10] <burek> generally, should i wait and ask for people to report a bug [20:11] <burek> or if i notice in on the forum, should i just report it myself [20:11] <burek> i mean, do i somehow take away their credits or something? [20:11] <burek> because i'd not want to do that [20:12] <burek> notice it* [20:12] <nevcairiel> credit for reporting a bug? [20:12] <burek> well i dunno, i just ask.. :) [20:12] <nevcairiel> you could always mention the person and link to the forum [20:12] <nevcairiel> if you care [20:12] <burek> yeah, that could be ok i guess [20:13] <burek> ok, thanks :) back to reporting it then :) [20:39] <Marco-123> hi all. (my sys: ubuntu 10.04, arm proc, neon media processor). i am using a software library that uses FFMPEG. so i dont work with ffmpeg directly. however, i suspect that my hardware accelerators are not being used. so now I am to get into things. id like to write a minimal app, that will tell me (yes/no, conclusively) whether or not the hardware accelerators are being used by the ffmpeg libraries. could someone help me w [20:42] <Marco-123> i was thinking of having a sample videos, maybe encoding/decoding them with various codecs (h263plus or h264), audio and video, and trying to see if the hwaccel is being used along the way (??). or maybe there is a command that will tell me info directly. [20:42] <burek> does ffmpeg support hwaccel right now? [20:44] <cehoyos> Marco-123: Are you talking about hwaccel or asm / neon support? [20:52] <michaelni> nevcairiel, maybe you want to comment/review on the 2 alterantive [PATCH 1/3] vc1dec: the second field is written wrong to the picture / vc1dec: Fix destination pointers in context for fields [20:53] <michaelni> its better to have a second oppinon on which is better than me just pushing one in a few days [20:53] <nevcairiel> all i can do is check the bitstream parsing against the spec, because thats really all i know about how vc1 works :d [20:55] <michaelni> :( [20:56] <michaelni> sadly no bitstream parsing changes in the ones i pointed to, but thanks anyway [20:57] <nevcairiel> thats also the reason my patch was never finished, i wrote the bitstream parsing, and never had time or motivation to figure out how it fits into everything else [20:59] <Marco-123> cehoyos: hmm. maybe my understanding is off. i have a neon chip (media processor?), and i thought that this was the hardware accelerator. maybe my terminology and understanding of the thing is wrong. [20:59] <michaelni> nevcairiel, [PATCH 3/3] vc1dec: interlaced B-frames <----- contains some bitstream parsing in the added code [21:00] <nevcairiel> neon is not what we usually call a "hardware accelerator", neon is just "SIMD", what in our desktop CPUs is MMX/SSE [21:09] <Marco-123> nevcairiel: ah. ill read about hardware accelerators now. in the mean time, can an SIMD help my encoding/decoding performance? [21:09] <nevcairiel> yes, significantly [21:09] <nevcairiel> without SIMD, de/encoding is much slower [21:10] <Marco-123> if my board has a neon SIMD, might it also have a GPU? or is there an overlap of these two functionalities and only one should be found? [21:10] <nevcairiel> neon is part of the CPU, it has nothing to do with your GPU, so sure, it can have a GPU as well [21:10] <nevcairiel> although in a SoC stuff gets mixed :p [21:13] <Marco-123> alright i will look into my board's datasheet again for GPU details (it probably doesnt have one). so i guess my original question is: what is a good approach for conclusively knowing whether or not my ffmpeg encoding/decoding is acutally making use of the neon simd? [21:15] <wm4> are there fate tests for dvdsub? [21:16] <wm4> Marco-123: my guess is that if there's ARM specific asm in the source code, it uses neon [21:17] <nevcairiel> his question seems to be how to know if that asm is actually used =p [21:17] <Skyler_> see if neon: yes in configure? [21:17] <nevcairiel> isnt there a runtime check now too [21:17] <wm4> Marco-123: compile with asm disabled, and compare speed to normal configuration [21:18] <wm4> or rather, neon disabled, if there's a configure option [21:18] <Marco-123> nevcairiel: after much googling, im able to build ffmpeg so that the configure script reads "NEON enabled: yes" [21:18] <Marco-123> --enable-neon haha [21:18] <Marco-123> still took much googling! [21:18] <nevcairiel> i would've thought configure enables such things on its own by probing [21:18] <wm4> well then try with --disable-neon and compare speed, then you should get some conclusive results [21:19] <nevcairiel> but what do i know about arm [21:19] <Marco-123> wm4: yes this is a good plan. forgive my ignorance but I can't find on google what ASM in this context stands for [21:19] <Marco-123> assembler? [21:20] <wm4> yes [21:20] <nevcairiel> yes, assembly code, hand-optimized code that uses neon [21:20] <Marco-123> thank you [21:21] Last message repeated 1 time(s). [21:21] <Marco-123> so in your opinions: if the configure script reads "NEON enabled: yes", should I stop worrying about ffmpeg using it? [21:21] <Marco-123> is it a good indicator? [21:22] <wm4> sure, why not [21:24] <Marco-123> ok. to answer why not: im still seeing not so great encoding performance. granted my set up is embedded and thus not as fast as a desktop computer, so this might be the cause [21:26] <Marco-123> guys thanks for your help. you've cleared up a lot of fuzziness in my brain about this stuff [21:26] <Marco-123> :D [21:41] <Skyler_> If you want to be absolutely sure, disable NEON and see how much slower it is ;) [23:05] <cone-774> ffmpeg.git 03Matthieu Bouron 07master:5a67e30b1c71: ffmpeg: fix broken channel_layout option [23:10] <cone-774> ffmpeg.git 03Matthieu Bouron 07release/1.1:c8dace272829: ffmpeg: fix broken channel_layout option [23:29] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07master:6a50e8a190bc: Do not change codec in flv streams if the user has forced a codec. [23:42] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:299dc3baeaaf: Do not change codec in flv streams if the user has forced a codec. [23:42] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:3445bec6fc1d: Do not change codec in flv streams if the user has forced a codec. [23:58] <cone-774> ffmpeg.git 03Ronald S. Bultje 07master:8af915c21b03: vp56: remove clear_blocks call, and clear alpha plane U/V DC only. [00:00] --- Sat Feb 2 2013

1 0

ffmpeg.log.20130201
by burek 02 Feb '13

02 Feb '13

[02:51] <ulatekh> I accidentally encoded a .flv video with the wrong audio, and I'm trying to replace it, but "ffmpeg -i old.flv -an -vcodec copy -i new.mp3 -acodec copy -f flv new.flv" exits with "Unknown decoder: 'copy'". Does anyone know how do I do what I want? [02:51] <ulatekh> This is the latest git-master ffmpeg from about a week ago. [03:00] <ulatekh> Anyone awake? [03:05] <ulatekh> OK, one sec. [03:06] <ulatekh> http://pastebin.com/j8q7HW6s [03:07] <llogan> also paste the output of: ffmpeg -i old.flv -i new.mp3 [03:08] <ulatekh> OK, one more sec. [03:09] <ulatekh> http://pastebin.com/ip2AHAGS [03:09] <ulatekh> Thanks for looking into this for me, llogan. [03:09] <llogan> ffmpeg -i old.flv -i new.mp3 -map 0:0 -map 1:0 -c copy -shortest out.flv [03:10] <ulatekh> OK, I'll try that, one sec. [03:10] <llogan> note that your new audio stream is ~6 seconds shorter duration [03:10] <ulatekh> The old audio stream, being the wrong one, was too long. [03:11] <ulatekh> No errors...let me look at the result [03:12] <ulatekh> Looks like it worked! Awesome! I'll have to look up the meaning of "-c copy -shortest". [03:12] <ulatekh> Still...shouldn't "-vcodec copy" have worked? [03:13] <llogan> yes, but you were applying it to the decoder of new.mp3 [03:13] <llogan> in other words, you were using -vcodec copy as an input option [03:14] <ulatekh> Oh...I thought I had to put those options after the input file they related to. [03:15] <ulatekh> What would have been the correct command line with "-acodec copy" and "-vcodec copy"? [03:15] <llogan> -c copy does the same as -acodec copy and -vcodec copy (and more) [03:16] <llogan> An empty stream specifier matches all streams, for example -codec copy or -codec: copy would copy all the streams without reencoding. [03:16] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [03:16] <llogan> and -c is an alias for -codec [03:17] Action: llogan is going to be afk for a while [03:17] <ulatekh> Hmmm, looks like I have to take my knowledge of the ffmpeg command line up to the next level. Thank you very much for your help! [03:55] <sunjun_> http://www.beyond3d.com/content/articles/8/ [04:35] <temhawk> wow, why is the ffmpeg syntax so hard to read?? [04:36] <temhawk> take this for example: http://pastie.org/6003419 wtf!? [04:36] <temhawk> that's just for combining two videos ¬¬ [08:27] <ledil> Hi, Im getting this error: http://pastebin.com/64JFAwjM, Ive used ffmpeg 0.5 before and I have now switched to ffmpeg 1.1 ... [08:40] <Sunil_> for MPEG-ts file , wanted to change service name, provider . tried -metadata service_name="Some Channel" , it seems ti does not change service name [10:59] <buhman> wow, that's some dark magic there: http://pastie.org/6003419 [10:59] <buhman> I would have put them in mpg containers and simply concatenate [11:11] <samuele> Hi everyone, I have a little question on the LICENSE [11:14] <samuele> I found an android port of FFmpeg compiled with --enable-gpl free just for personal use. I can't bundle it in any developed new app. Is this a violation in terms of license, right? [11:16] <khali> samuele: definitely [11:16] <samuele> Thanks, it was just for confirmation. I'm about to say this over there. [11:17] <khali> actually I'm a bit too enthusiastic with "definitely" [11:17] <samuele> over thei website I mean [11:17] <khali> it probably depends on the other build flags [11:17] <Mavrik> wait [11:17] <samuele> I can paste them if you want. [11:17] <Mavrik> samuele: can't really parse your sentence. does it have "--enable-gpl" or not? [11:17] <samuele> yess sir [11:18] <Mavrik> does it have "--enable-nonfree"? [11:18] <khali> samuele: if includes other --enable flags which cause non-GPL code to be included then you must check the terms of these other licenses [11:18] <samuele> nope [11:19] <Mavrik> so, if you only have "--enable-gpl" without "--enable-nonfree" your ffmpeg binary is GPL licensed [11:19] <Mavrik> which means you can distribute it with your app, you can call it and everything [11:19] <samuele> OK: checked. "--enable-nonfree" not included. [11:19] <khali> thanks Mavrik for being more accurate that I was [11:20] <Mavrik> you may not link against ffmpeg libraries without making your code available under GPL license though [11:20] <Mavrik> khali: just being more specific, you were right the entire time ;) [11:20] <samuele> Mavrik: this is wat I meant with my question. [11:20] <samuele> so should I point this out to them? [11:20] <Mavrik> samuele: to who? [11:21] <Mavrik> and as I said... you can distirbute "ffmpeg.exe" and call it from other app to do encoding without violating GPL [11:21] <samuele> sorry: to who built this FFmpeg port to android and released NOT for free if I want to bundle it in an App of mine [11:22] <Mavrik> samuele: are you going to call ffmpeg binary (so use "runtime.exec("ffmpeg -i blahblah")) or are you going to call avcodec_decode... things manually? [11:23] <samuele> I don't really know, because the project is meant to be used as a library. I'm not an expert here, but I can go through further search [11:24] <Mavrik> ok [11:24] <Mavrik> in first case you may use it, in second your code has to be GPL as well [11:24] <Mavrik> that's the distinction [11:24] <samuele> do you want the links to the source code? [11:24] <Mavrik> I don't have time to really check it right now [11:25] <samuele> ah ok. anyway should be free to use. And my app is also GPLv3... [11:25] <samuele> ok. [12:08] <khali> is there a way to ask ffmpeg for the help/format of a specific filter? [12:08] <khali> -filters gives the list of filters but that is very succinct [12:12] <MessedUpHare> Hi all, I'm trying to figure out a way of programatically recording the output of -vf cropdetect - but i'm struggling to do so as the output is all on a single line [12:12] <MessedUpHare> is there something I'm missing here [12:13] <MessedUpHare> In an ideal world, I would like to end up with a text file which states times (or frame number) along with a detected change of "-vf crop" values [12:27] <MessedUpHare> not to worry, awk has saved the day [12:36] <someone-noone> Hello! Where can I read about how can I do Annex B->AVCC and AVCC->Annex B converting in h264 for dummies? :) I found lot information in internet but I there is smth I don't understand (I can't properly parse my h264 annex-b stream) [13:02] <relaxed> someone-noone: To demux from mp4 use ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 [13:02] <someone-noone> relaxed, dude thanks, but I want to make my own demuxer and can't find sutiable specs for that [13:03] <someone-noone> relaxed, I've read ffmpeg code but I'd rather look at specs [13:04] <relaxed> I don't believe the h264 spec is free but that doesn't mean you can't find it for free. [13:05] <JEEB> it is [13:05] <relaxed> it is free? [13:05] <JEEB> they release it for free after six or so months after it's released [13:05] <JEEB> and the last updated spec was released in 12/01 [13:05] <relaxed> ok, maybe you can point him in the right direction. [13:05] <JEEB> I already did [13:05] <JEEB> I linked him the H.264 spec before [13:05] <someone-noone> I've already read 'bout avcc format [13:05] <someone-noone> it's clear for me [13:06] <JEEB> relaxed, if you don't believe it's free here you go http://www.itu.int/rec/T-REC-H.264-201201-I :) [13:06] <someone-noone> but I can't found any information about Annex B(actually, I can, but it doesn't seem to be true) [13:06] <relaxed> I have no reason to doubt you. [13:06] Action: relaxed sometimes makes wrong assumptions [13:06] <JEEB> ISO generally has its specifications locked [13:07] <JEEB> ITU-T seems to be more lenient with some specs [13:07] <JEEB> H.262 / MPEG-2 Video seems to be locked down, though [13:07] <JEEB> I wonder what they'll do with H.265, for adoption a similar protocol as they do with H.264 would be nice [13:14] <JEEB> also, no... seems like ISO actually has this specification available for free as well [13:14] <JEEB> http://standards.iso.org/ittf/PubliclyAvailableStandards/index.html [13:14] <JEEB> I always miss the "you can also get this specification for free from the ITTF" on the ISO spec page [13:27] <someone-noone> how can I produce avcc stream from ffmpeg(using libx264)? I tried to do this from .mp4: ffmpeg -i test.mp4 -c:v copy -an -f h264 test.h264 [13:28] <someone-noone> But it seems that output file has own mp4 format (first byte is not equal to 1) [13:28] <someone-noone> and also result is not demuxable with ffmpeg [13:29] <someone-noone> however, If I do ffmpeg -i test.mp4 -c:v libx264 -an -f h264 test.h264 [13:29] <someone-noone> the result is always in annex b [13:34] <brx_> my ffmpeg question http://pastebin.com/QHHtHL0K [13:44] <khali> brx_: I think your best chance is to add an empty audio channel to one.mp4 [13:45] <brx_> khali, do i do that durig the command where I create the video from jpg? [13:46] <brx_> i couldnt find how to do this when searching the docs [13:46] <brx_> ffmpeg -y -loop 1 -i blah.jpg -t 3 -vcodec libx264 one.mp4 [13:46] <khali> yes I'd do that in the first step [13:46] <brx_> what must i add? [13:46] <khali> I don't know, I never had to do that [13:47] <brx_> ok [13:48] <khali> brx_: I think you can pass -i twice [13:49] <khali> brx_: prepare a silent .wav file and pass it that way [13:49] <brx_> khali, ok i will try that [13:49] <brx_> thanks [13:50] <khali> you may have to pass -loop twice, but I'm not sure if it supports anything else than still pictures [13:50] <khali> brx_: so you may have to make your .wav file the right size it the first place [13:51] <brx_> ok [14:58] <digited> hi [14:59] <digited> are there any examples of encoding video with adding fancy effects from showwaves filter? [14:59] <digited> (also, ffplay example with showwaves/showspectrum doesn't work) [17:39] <Marco-123> hi all. I have a board with an ARM processor (and a NEON hardware accelerator), which is running Ubuntu Linux v10.04. I am building ffmpeg and my goal is to make sure it is taking full advantage of the hardware on the board. It's a freescale IMX53 btw. After much googling, I set a few flags in the configure script, and the script tells me "NEON enabled: yes" but the encoding performance is quite slow (same as when it says "NEON e [17:39] <Marco-123> can someone provide advice on how to build this project properly? [17:40] <Marco-123> or if there is a conclusive way to tell if the hardware accelerators are being used? another point is that the configure also tells me "enabled hwaccels: " and nothing is listed there. [17:53] <samuele> Hi. I have a question about compiling [17:54] <samuele> if I add "--enable-libmp3lame" I get the error "libmp3lame >= 3.98.3 not found" [17:54] <JEEB> see config.log [17:54] <JEEB> have you built lame yourself or trying to utilize a package in your OS? [17:55] <samuele> I downloaded the libmp3lame source, compiled and I have libs in /usr/local/libs [17:56] <samuele> I need mp3 in an ffmpeg for android. The build w/o libmp3lame works OK [17:57] <samuele> config.log : error: lame/lame.h: No such file or directory [17:57] <khali> samuele: odds are that /usr/local isn't looked at [17:58] <samuele> I also tried adding in the build script "export LD_LIBRARY_PATH=/usr/local/lib" [17:59] <khali> samuele: LD_LIBRARY_PATH is for runtime IIRC [17:59] <khali> won't help configure find includes [17:59] <samuele> :8 [17:59] <samuele> ok [17:59] <khali> samuele: did you try --extra-cflags=-I /usr/local/include ? [18:00] <khali> hmm, without space [18:00] <khali> --extra-cflags=-I/usr/local/include [18:00] <samuele> trying now. thanks. [18:00] <JEEB> if your compiler doesn't look there by default, you need both that [18:00] <JEEB> as well as --extra-ldflags=-L/usr/loca/lib [18:00] <khali> and --extra-ldflags=-L/usr/loca/lib [18:00] <samuele> ok [18:00] <JEEB> *local [18:00] <khali> ah, JEEB was faster ;) [18:01] <samuele> ok. call you back. [18:01] <JEEB> also PKG_CONFIG_PATH=/usr/local/lib/pkgconfig is useful [18:01] <JEEB> for some libs [18:21] <samuele> hei khali and JEEB [18:22] <samuele> giving "./configure --enable-libmp3lame" works [18:22] <samuele> but putting --enable-libmp3lame in the build.sh script provided with this project don't [18:22] <samuele> so I think it's the script that has to be checked [18:33] <samuele> http://pastebin.com/D3nqNhpd [18:35] <khali> samuele: ah, you're cross-compiling? [18:35] <samuele> yes, for android. [18:35] <samuele> I think this is bigger than what I can handle [18:36] <samuele> and I think this sounds realy bad in English... [18:37] <samuele> LoL [18:38] <samuele> "--sysroot=$SYSROOT" can be this? [18:40] <samuele> SYSROOT=$NDK/platforms/android-3/arch-arm [18:40] <samuele> NDK=${HOME}/android-ndk [18:41] <khali> samuele: I've never done that, sorry [18:42] <khali> samuele: first question is, how did you build libmp3lame? it must be cross-built too, otherwise this can't work [18:42] <samuele> not cross-built... [18:43] <khali> samuele: so start with this [18:44] <khali> samuele: and then you shouldn't install it to /usr/local, that would be for the host architecture [18:44] <khali> samuele: you must setup a tree for your android libs [18:46] <samuele> sorry. I missed the lines between "so start with this" and "and then you shouldn't..." [18:46] <samuele> I was reading above [18:48] <nilsge> hi. In Archlinux ffmpeg is built withou support for jack audio connection kit (-f jack or -f alsa -i jack both say no jack and ffmpeg -formats does not list it). [18:48] <nilsge> What is the configure switch to include it? --enable-jack or --enable-libjack are unknown [18:53] <nilsge> I built from git anyway and it lists "jack" under indev now, at configure stage, at leas [18:54] <nilsge> t [19:03] <creep> hi [19:03] <creep> why do people share TS and cam movies over net? ;/ [19:03] <creep> it is so much worse than going in a movie and watch it [19:38] <troy_s> Anyone familiar in here with calculating the exact bit width of a raw Y' / Cb' / Cr' data stream in the avframe? [19:38] <troy_s> YUVJ420P reports 12bpp, but the data is in fact only 8 bits wide. [19:42] <JEEBsv> the 12bit comes from the fact that since you have Cb and Cr 8+8 for 2x2 area, you then get 4 bits per one (luma resolution) sample [19:42] <troy_s> fflogger / durandal_1707 I'm using the API. [19:42] <JEEBsv> so yeh, that value does not really tell you the byte size of a single value [19:42] <troy_s> JEEBsv: Right. So is there a method to calculate the actual amount of data stored in a single unit of Cb / Cr? [19:43] <troy_s> JEEBsv: Case in point, if I am attempting to yank out the raw 10 or 12 bit data from a DNxHD or ProRes stream, is there a method to figure out the width of the unit? [19:43] <durandal_1707> isn't that stored in container/bistream [19:44] <troy_s> durandal_1707: Bit depth reports 12 as expected for YUVJ420P for example. I'm willing to admit I am missing some other variable to check. [19:44] <durandal_1707> only if you found out that all lower bits are zero [19:44] <troy_s> durandal_1707: I can't see that as being too effective given the nature that the data might be zero. [19:45] <JEEBsv> troy_s: if it's decoded already, it's probably in a format where a 16bit variable is used if it's >8bit , it depends on the PIX_FMT [19:45] <JEEBsv> YUV420P is 8bit [19:45] <troy_s> I'm thinking there is an obvious method to determine the width of a given unit of Cb / Cr (in the case of YUVJ420P, it's 1 byte per sample.) [19:45] <troy_s> JEEBsv: Yes. I'm just hoping that there is a variable to read. I'd had to make guesses based on PixFmt as to the width of the chroma. [19:46] <troy_s> JEEBsv: Similar to the bitwise shift for chroma widths and height for example. [19:48] <troy_s> Looks like there might be a pixel_shift value. Urgh. [19:49] <durandal_1707> so you actually want bits per pixel? [19:55] <troy_s> durandal_1707: I'm dumping the raw YCbCr [19:55] <troy_s> durandal_1707: So yes... I'm looking for the exact width of a Cb / Cr / Y value in the data, as opposed to me guessing. [19:59] <durandal_1707> troy_s: each pix fmt have its descriptor in libavutil/pixdesc.c [20:03] <troy_s> durandal_1707: Yes. I just found what MN replaced for the avg_bits_per_pixel - av_get_padded_bits_per_pixel. [20:03] <llogan> burek: i should stop opening an reply box and replying half an hour later...i didn't see your post until after. [20:03] <burek> it's ok :) [20:03] <burek> also, i didnt thank you for helping me there [20:03] <burek> so, thank you :) [20:04] <llogan> nice IMPORTANT box [20:04] <burek> i appreciate it :) [20:04] <durandal_1707> troy_s: that is how it is stored in memory i guess ... [20:04] <burek> well, i had to make it ridiculously big and red [20:04] <burek> because people just didnt read it at all.. :S [20:04] <llogan> now lets see how many people still miss it [20:04] <troy_s> durandal_1707: Yes... looks like those are the two tools. I was already referencing the log2_chroma_h and w for shifting, but I need to get the actual value. [20:04] <troy_s> durandal_1707: Thanks for your help by th wya. [20:08] <divVerent> stupid question, but what is the "proper" way to use the showspectrum filter with ffplay? [20:08] <divVerent> as in, is there anything simpler than ffplay -f lavfi 'amovie=filename.mp3, showspectrum' [20:08] <divVerent> because, THIS one can't seek [20:13] <saste> divVerent, no simple solution ATM [20:13] <saste> i'm working right now on -af in ffplay [20:13] <saste> but that doesn't fix the problem [20:13] <saste> another solution would be to port filter_complex to ffplay [20:13] <divVerent> saste: for the specific case BTW [20:13] <saste> or to add seek support in movie [20:14] <divVerent> in the long run, don't we want to get rid of ffplay's own visualization [20:14] <divVerent> and just use showspectrum? [20:14] <saste> divVerent, yes [20:15] <saste> but code doesn't write it itself (yet) [20:15] <divVerent> sure [20:15] <saste> which is a shame [20:15] <divVerent> and it doesn't seem trivial [20:15] <divVerent> also due to lack of -filter_complex :P [20:16] <divVerent> the alternative BTW would be making amovie able to seek [20:16] <divVerent> how likely is that? [20:16] <divVerent> (and movie too, of course) [20:19] <saste> divVerent, i can only say that is on my todo list, but that's a long list [20:20] <divVerent> hehe [20:20] <divVerent> it also doesn't seem trivial [20:20] <divVerent> just wishful thinking... but if it WERE done, it could become ffplay's ONLY input [20:20] <divVerent> (and other options remapped to using the movie source) [20:21] <divVerent> actually, ffmpeg's too... for ffmpeg it seems even more viable [20:21] <divVerent> now if only it wouldn't need weird filename escaping... ;) [20:47] <patbam> hi, how can i convert a NIST Sphere file to wav? I've tried & failed to find it in the docs [20:50] <burek> ffmpeg -i SOMETHING out.wav [20:50] <burek> where SOMETHING is your file [20:51] <patbam> i got this output: http://pastie.org/6019879 does it mean my file is borked? [20:51] <patbam> (& thanks) [20:52] <Gergov> guys how to create an flv file from an mp3 and a .jpg with decent quality ? [20:54] <Gergov> is there some standard terminal command that dont involve 1000 options [21:07] <llogan> Gergov: to what output format(s)? [21:09] <llogan> patbam: can you provide a sample file? [21:10] <Gergov> llogan, .flv [21:10] <Gergov> i want to upload a song to youtube [21:10] <patbam> llogan: i'm afraid i cant because the files are under copyright to a university :( [21:10] <sacarasc> Gergov: You don't need FLV to upload to youtube, you can do pretty much anything. [21:11] <Gergov> sacarasc, well then i dont care about the output format :) if any will do the work [21:11] <llogan> patbam: i found one and ffmpeg decodes it with no problem. [21:11] <patbam> llogan: hmm, did oyu just use ffmpeg -i foo.sph foo.wav? [21:11] <patbam> maybe my files are borked somehow [21:11] <llogan> no, i just did: ffplay speech.sph [21:12] <sacarasc> Gergov: ffmpeg -loop_input -i foo.jpg -i blah.mp3 -shortest -c:v libx264 -preset slow -crf 18 -c:a copy output.mkv [21:12] <sacarasc> I think that would work, but I am a little rusty. [21:12] <llogan> Gergov: https://ffmpeg.org/trac/ffmpeg/wiki/EncodeforYouTube [21:13] <llogan> s/-loop_input/-loop 1 [21:13] <Gergov> sacarasc, i get Unrecognized option 'c:v' Failed to set value 'libx264' for option 'c:v' [21:13] <sacarasc> Or just upgrade to a newer version. [21:14] <llogan> patbam: i guess your file is damaged, or your ffmpeg version is too old, or maybe there is a sph variant that ffmpeg doesn't understand [21:14] <patbam> llogan: this is the file header: http://pastie.org/6020822 i'm wondering if the "sample_coding -s27 ulaw,embedded-shorten-v2.00" bit is suspicious. i think it's just a variant of ulaw. is there a way i can force ffmpeg to try using ulaw to decode (demux?)? [21:14] <Gergov> sacarasc, ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1 is this old? [21:14] <sacarasc> That's probably not even ffmpeg. [21:14] <JEEBsv> that's libav, if you want to use libav, switch to the 'avconv' command [21:15] <Gergov> sacarasc, i typed aptitude install ffmpeg :) it should be it. or not ? [21:15] <JEEBsv> nah, debian/ubuntu use libav [21:15] <JEEBsv> (and gentoo now by default) [21:15] <Gergov> i see [21:16] <llogan> do they laso have a fake ffmpeg package? [21:16] <llogan> *also [21:16] <JEEBsv> the ffmpeg binary in libav is outdated (and was later removed), so 'avconv' is the command you'd want to use when using libav's stuff [21:16] <Gergov> thanks :) ill then install ffmpeg and use the command sacarasc gave me [21:17] <JEEBsv> llogan: gentoo? they have the nice thing of being able to recompile everything that depends on libav* so you can have both ffmpeg and libav there fully [21:20] <llogan> Gergov: you can follow the compile guide or one of the recent static builds if you want ffmpeg from FFmpeg [21:20] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [21:21] <llogan> patbam: yes, you can apply options to the decoder, but maybe your file contains shorten and not pcm. [21:22] <patbam> llogan: ah that sounds likely. do you happen to know know what shorten is? [21:23] <llogan> yet another (dead?) lossless audio format [21:23] <llogan> *losslessly compressed [21:28] <llogan> patbam: just guessing here. maybe try: ffmpeg -ar 8000 -ac 2 -f shn -i input [21:29] <patbam> llogan: hmm cool, it failed in a different way, that's progress :) http://pastie.org/6021755 [21:30] <llogan> i have no more ideas without a sample to monkey with [21:32] <patbam> ok llogan, thanks!! [21:33] <octe> i'm trying to follow this: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20capture%20a%20lightning%20(th… to get a delayed stream [21:33] <octe> but the second instance just says udp://127.0.0.1:5678: could not find codec parameters [21:46] <llogan> burek made that guide, so maybe he'll see your output too [23:30] <nilsge> anyone used ffmpeg with jack input and got rid of the thousands of xruns? [00:00] --- Sat Feb 2 2013

1 0

ffmpeg-devel.log.20130131
by burek 01 Feb '13

01 Feb '13

[00:56] <cone-913> ffmpeg.git 03Piotr Bandurski 07master:9c50e69385e8: loco: fix rgba on big-endian [00:57] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:9df9420dea0f: interplayvideo: Free previous frames on resolution changes. [01:30] <shahriman> guys can anyone download the original source from here, i want to report a bug but to verify you need the source https://vimeo.com/58025358 [02:29] <Compn> shahriman : should be downloadable with youtube-dl or [02:43] <cone-913> ffmpeg.git 03Mirjana Vulin 07master:2b6a8187a6ae: mips: optimization for float aac decoder (core module) [03:53] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:fe6767f849d9: asfdec: fix integer overflow in packet_replic_size check [04:36] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:cdf0877bc341: h264/cabac: check loop index [04:39] <Compn> nevcairiel : sorry if i missed your answer, are you planning to review dxva patch ? [04:49] <highgod> Hi, all, I want to ask a question. Does ffmpeg has any filter to remove mosaic? Thanks [04:51] <Compn> uncensor filter? no , it does not [04:52] <Compn> that would be neat :) [04:52] <Compn> nevcairiel : highgod would like it if someone would review his dxva patch :) [06:06] <highgod> Thanks [07:22] <nevcairiel> Compn: i volunteer you then! [11:25] <durandal_1707> divVerent: you did anything with showspectrum? [11:28] <divVerent> not yet, can I see the code you currently have? [11:28] <divVerent> i.e. where is i [11:29] <divVerent> t [11:29] <durandal_1707> divVerent: you are not subscribed to ffmpeg-devel? [11:29] <divVerent> no, I look in the archives [11:29] <divVerent> ok then [11:30] <durandal_1707> well it should be in archives [11:30] <durandal_1707> search for showspectrum [11:30] <divVerent> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/138172.html this is it probably [11:30] <durandal_1707> there is only one patch that adds new funcionality .... [11:31] <divVerent> right, this one keeps the old code there for compat though [11:33] <durandal_1707> well old code use same row for all channels [11:33] <durandal_1707> compat could be renamed to something better ... [11:33] <durandal_1707> if I know nice way to map 8 ch values into 24bit number [11:35] <divVerent> 24bit number? as in RGB? [11:37] <durandal_1707> yes, rgb24 [11:40] <divVerent> the way I'd prefer is defining channels by hue+ [11:40] <divVerent> it seems to be the best one can do, albeit far from perfect [11:42] <divVerent> durandal_1707: are there already HSV or HSL functions in ffmpeg? [11:43] <divVerent> hm... there is a "hue" filter [11:44] <divVerent> ah, this filter uses unit vectors in U,V space, scaled by Y... so I will try the same and see how it looks [11:48] <durandal_1707> divVerent: it uses rgb24 for output, but that can be changed to yuv444p [11:48] <divVerent> but then it's planar... I wanted to keep the code as close as possible to the original :P [11:48] <divVerent> too bad there is no packed yuv444 format yet [11:49] <divVerent> so I will do RGB for now [11:49] <durandal_1707> what original? [11:49] <durandal_1707> output format can be changed [11:49] <divVerent> the "compat" code :P [11:49] <durandal_1707> compat mode if legacy just for same row [11:49] <divVerent> basically, what I am doing, is adding a "combined" mode to your patch that is SORT of like compat [11:49] <divVerent> but supports multiple channels [11:50] <durandal_1707> there is no point to keep if for some reason like ffplay useless code [11:50] <durandal_1707> showspectrum is just port of ffplay one night hack session [11:51] <durandal_1707> there is no reason to keep such pointless legacy crude whatever code [11:51] <durandal_1707> it is just insane [11:51] <divVerent> sure [11:51] <durandal_1707> also ffplay when outputs rgb24 it fucks up color, everything is too bright [11:51] <divVerent> compiling now [11:52] <divVerent> hehe [11:52] <divVerent> well, trying out the concept on rgb24 now, if it works out, I'll change it to yuv444p [11:52] <divVerent> and then multiple channels will get equal brightness again [11:53] <durandal_1707> i have bunch of ideas to add to this filter to make it actuall useful [11:53] <divVerent> durandal_1707: another thing to notice BTW... the current (and old) code is quite slow... due to working by rows... I wonder if working by columns, and using existing SSE-optimized rotate code after it (if it exists) would be faster [11:54] <durandal_1707> i do not think rotation is needed ... [11:54] <divVerent> but OTOH [11:54] <divVerent> performance is probably ok anyway [11:54] <durandal_1707> code is slow perhaps it do not use something of x^2 +1 [11:54] <durandal_1707> or it is just writtend without performance in mind [11:55] <divVerent> the latter [11:55] <divVerent> probably not needed, it's just processing audio after all [11:55] <durandal_1707> though ffplay output if faster, so port is maybe culprit [11:56] <divVerent> hehe, the old code may indeed work faster, depending on how clever SDL_FillRect and SDL_UpdateRect are [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:d99bc8475232: doc/eval: fix/extend documentation for root() function [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:41e5e28daf0a: doc/eval: fix/extend documentation for taylor() function [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:dcfbe1e0b67b: doc/eval: sort functions by name [12:15] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/b86c263c5a2942382417adf786567173.png http://rm.sudo.rm-f.org/img/uploaded/b07aa627f01447270acc377d9af60b02.png [12:15] <divVerent> is basically my idea [12:15] <divVerent> the views are the same, the combined one is just the colors of the two added [12:16] <divVerent> the colors themselves don't matter yet though, but they are chosen to be so they always all add up to grey [12:17] <divVerent> https://gist.github.com/4682205 - is my code at the moment [12:17] <divVerent> I suppose the new "combined" mode is close enough to the old "compat" so "compat" can be ignored and colorspace can be changed? [12:19] <durandal_1707> divVerent: fine. i just want to see >2 in combined mode [12:20] <divVerent> have no input file with more than 2 channels here, can you point me to one? [12:21] <divVerent> could always generate one though, but it looks more useful with real data [12:21] <durandal_1707> samples.ffmpeg.org/ lossless audio and wavpack ... [12:22] <divVerent> ah, thanks [12:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/0030d8c1a047a74683c5342464beac4f.png [12:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/5adb3378906a2b8b4c08bd3174e82ff6.png [12:25] <divVerent> 6 channel ID in the two modes (still RGB) [12:26] <durandal_1707> hmm you render one after another? [12:27] <divVerent> this audio file does that [12:27] <durandal_1707> 2nd non-combined one looks buggy [12:27] <divVerent> http://samples.ffmpeg.org/multichannel/6_Channel_ID.flac [12:28] <divVerent> durandal_1707: no, but the 6th channel is used weird [12:28] <durandal_1707> ahh then pick less retarded sample [12:28] <divVerent> yes [12:28] <divVerent> but first porting to YUV [12:28] <durandal_1707> eva something [12:28] <divVerent> so the color brightnesses will match [12:28] <divVerent> is there a file list of this whole server? [12:29] <divVerent> oh, stupid question, found it [12:29] <durandal_1707> yes each separate color for channel is good idea for separate option .... [12:29] <divVerent> ./A-codecs/lossless/wavpack/test_suite/num_channels/eva_2.22_6.1_16bit.wv [12:29] <divVerent> AH, should have looked in A-codecs/ [12:29] <durandal_1707> yes i have that sample for testing .... [12:32] <divVerent> haha, got integer overflows ;) [12:33] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ca583f6309bf0323fef2e786752371bc.png separate [12:34] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e42bb921dc915f39c369929c1e7bd63e.png combined (and divided by number of channels) [12:34] <durandal_1707> some kind of projecting X dimenstion to 2.3 is needed [12:34] <divVerent> 2.3? [12:34] <divVerent> you mean the different resolution due to different FFT sizes? [12:35] <durandal_1707> i mean best way to pick color from 8 chans [12:36] <divVerent> what I currently do, is YUV = 1, sin(angle), cos(angle) [12:36] <divVerent> oops, no, first cos, then sin [12:36] <divVerent> where angle is 2pi * channel / num_channels [12:36] <divVerent> that kind of alignment ensures that if all channels output the very same, you get grey in the sum [12:37] <durandal_1707> nice, that could be another option, where other one would be old boring monotonic for all channels.... [12:37] <divVerent> the combined one IS the old monotonic one :P [12:37] <divVerent> but the separate-channel one can easily be colored or uncolored [12:38] <durandal_1707> i mean in separate mode [12:38] <divVerent> now I am wondering... you are good in maths? [12:38] <divVerent> I know there is a mathematical theorem about the max of weighted sums of unit roots [12:38] <durandal_1707> and for separate mode i want something like sox does [12:38] <divVerent> yes, separate mode should IMHO have the choice between intensity-colors (sox) and channel-colors [12:39] <divVerent> intensity-colors being default [12:40] <cone-469> ffmpeg.git 03Giorgio Vazzana 07master:ff23b76899b2: lavd/v4l2: add list_standards option [12:40] <divVerent> basically, I need max real sum(i=0..n-1) omega^i * a_i, where 0 <= a_i <= 1 and omega = a complex unit root of order n [12:40] <divVerent> I THINK there is an easy expression for this [12:42] <divVerent> it SEEMS to be in the order of n/3 [13:05] <divVerent> ARGH, I HATE YOU, TV LEVELS, I HATE YOU [13:06] <divVerent> what is the proper YUV black level again [13:06] <wm4> can't you just output full range? [13:07] <divVerent> I currently do [13:08] <divVerent> and that means lower frequencies become all black [13:08] <durandal_1707> debends what color range you set or application you use think you are using.... [13:08] <divVerent> because converting to RGB doesn't assume full range [13:08] <divVerent> or can the filter specify it outputs full range? [13:08] <wm4> I'd expect that it can set it in the picture that is output [13:08] <durandal_1707> color_range is part of avctx [13:09] <durandal_1707> i dont think lavfi is aware of it at all [13:09] <durandal_1707> try YUVJ444P pix fmt [13:10] <durandal_1707> it is 2013 and there is still no proper way to handle it in code :( [13:11] <divVerent> yes [13:11] <divVerent> it's deprecated [13:11] <divVerent> but this works [13:11] <durandal_1707> swscale have some code that converts between diferent matrix crap [13:12] <divVerent> now wondering how to set the colors so it looks good [13:12] <divVerent> or rather, maybe should match the old code [13:12] <wm4> durandal_1707: swscale + levels works, for YUV [13:12] <divVerent> it used FF0080 and 00FF80 [13:12] <wm4> at least in some cases... [13:12] <divVerent> now what are these colors in YUV [13:13] <divVerent> if my CAS is right, these colors are roughly +U and -U [13:14] <durandal_1707> divVerent: you can find several nice distributions and make them configurable [13:14] <wm4> lol that ticket [13:14] <durandal_1707> Frodo lost his ring [13:21] <divVerent> durandal_1707: there is not many to pick from [13:21] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/c45edab2126015b59ff9aa6665d55427.png http://rm.sudo.rm-f.org/img/uploaded/ce6512b22561426517b5bb5728624d00.png [13:21] <divVerent> my first attempt to match the default colors of ffplay [13:22] <divVerent> could it be that the two filters use different windowing functions? [13:23] <divVerent> yes, indeed [13:23] <divVerent> your code with compat=1 also has less grey stuff around :P [13:23] <durandal_1707> why would they use different function? [13:23] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e929fce848e6f0d09f841bad9183da10.png [13:23] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/c45edab2126015b59ff9aa6665d55427.png [13:23] <divVerent> first is compat=1 [13:23] <divVerent> second is my code [13:23] <divVerent> and the one before was ffplay [13:24] <divVerent> ffplay basically does SOME part of the RDFT differently [13:24] <divVerent> most likely the window function [13:25] <divVerent> ffplay uses a 1-x^2 window [13:25] <ubitux> yes i used a different windowing function [13:25] <durandal_1707> does it make sense? [13:25] <ubitux> (hello!) [13:25] <saste> ehi ubitux-san! [13:25] <divVerent> the avfilter function is better [13:25] <durandal_1707> x^2 + 1 should be used [13:25] <ubitux> the windowing function wasn't sexy so i used a hann one iirc [13:25] <divVerent> durandal_1707: sure not + 1 :P [13:25] <divVerent> yes [13:25] <divVerent> the issue you see on my screenshots :P [13:25] <divVerent> I tested with aevalsrc [13:26] <divVerent> and it's pure sine waves [13:26] <divVerent> ffplay shows LOTS of grey stuff around it [13:26] <divVerent> due to bad window function [13:26] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ce6512b22561426517b5bb5728624d00.png current ffplay [13:26] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e929fce848e6f0d09f841bad9183da10.png current showspectrum [13:26] <durandal_1707> ok, i prefer correct code [13:26] <divVerent> none is correct, but the better window function is less incorrect ;) [13:27] <durandal_1707> i'm against legacy [13:27] <divVerent> there is no way to do this correctly [13:27] <divVerent> but it's pretty good now [13:27] <durandal_1707> whatever, i just cant use current showspectrum as it is useless to me [13:27] <durandal_1707> so either I gonna improve it [13:27] <durandal_1707> or write new filter [13:29] <durandal_1707> so i will apply modified patch of mine for showspectrum, as it got no attention at all [13:29] <durandal_1707> and then will apply divVerent stuff [13:29] <divVerent> you can have my patch now BTW [13:29] <divVerent> what did you modify BTW, as I probably replaced all changed code by now [13:30] <divVerent> https://gist.github.com/4682536 [13:30] <divVerent> is my current changes vs yours [13:30] <divVerent> now all it needs is the intensity-coloring mode [13:30] <divVerent> (the "sox mode") [13:31] <durandal_1707> and then "rotated" mode, where slide goes up<->down [13:32] <durandal_1707> dont stop there, lets your imagination work out..... [13:32] <divVerent> I doubt it'd make sense to have both these modes... it'd duplicate too much code then probably [13:32] <divVerent> maybe better to decide on one mode, and use rotate filter for the other [13:32] <divVerent> but, rotated vs current display would likely be faster [13:33] <divVerent> e.g. that memmove mess can be all done as a single block then [13:33] <durandal_1707> duplicate? cant you just use functions? [13:34] <durandal_1707> yes, making bad code looks good is boring and daunting task.... [13:35] <divVerent> actually... I think I will solve that inefficiency now, which would also help with rotating [13:35] <divVerent> I will use an intermediate float buffer for each line [13:36] <divVerent> that way, you can in the regular display also bump saturation to like 100x [13:36] <divVerent> and then see channel differences better [13:36] <divVerent> in the current code, the clamping to 0..255 at each step would screw that up [13:39] <ubitux> saste: hey :) [13:39] <ubitux> do you miss me? [13:39] <cone-469> ffmpeg.git 03Paul B Mahol 07master:b70ea49ca32c: Port biquads filters from SoX [13:40] <ubitux> durandal_1707: bitexact + fate tests now? ;) [13:40] <durandal_1707> bitexact what? [13:40] <saste> ubitux: i'm feeling a bit annoyed since i have no new subtitles to try since a while [13:40] <ubitux> durandal_1707: integer arithmetic [13:41] <ubitux> saste: no new subtitles patches? [13:41] <durandal_1707> i still wait for that kerndeint improvements... [13:41] <saste> ah yes and that... [13:41] <durandal_1707> ubitux: i still do not get it what it means [13:41] <wm4> hey guys I could need a stable and useful API for lavfi [13:41] <ubitux> saste: nag beastd, he should be doing this [13:41] <durandal_1707> wm4: ask michaelni [13:41] <saste> beastd: I nag you! [13:42] <saste> wm4, you ask too much [13:42] <durandal_1707> ubitux: filters are bixexact with sox if you use aformat=samplefmts=dblp,bass...... [13:42] <saste> what about, unstable and useless? [13:42] <ubitux> durandal_1707: forget this, i though it wasn't supporting s16 etc [13:43] <wm4> e.g. recently I saw a commit that fixes seeking with lavfi and ff/avplay [13:43] <wm4> the solution was recreating the whole filter chain on seek [13:43] Action: ubitux doesn't want to leave this country [13:43] <ubitux> any ffmpeg work in japan for me? :( [13:43] <wm4> is that the proper lavfi solution? [13:49] <divVerent> durandal_1707: ok... https://gist.github.com/4682630 [13:50] <divVerent> now there's not much more of your patch left, so I diffed to master instead [13:50] <divVerent> local copy has the missing freep added [13:50] <divVerent> durandal_1707: the combined mode is now cool with like saturation=5 [13:53] <divVerent> so next part would be research as for how sox mode would work [13:53] <durandal_1707> do you have branch with several commits, it it makes sense to split them in smaller funcionality steps.. [13:54] <divVerent> no, don't have that [13:54] <divVerent> given I don't see many separate steps there [13:54] <durandal_1707> divVerent: you mean colors or output single frame for whole duration? [13:54] <divVerent> colors [13:54] <divVerent> single frame for whole duration can already be done, if calculating the width "right" ;) [13:55] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/257d3a6e6917db49fcf736ae9467da60.png with high saturation [13:55] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ec4ffa6febfc860cbc9aaca340237a01.png - combined mode with high saturation [13:56] <divVerent> I can visually make out 4 channels there [13:58] <durandal_1707> using ch to loop over data planes confused me big time [13:58] <divVerent> I saw it as color channels [13:58] <divVerent> but the main reason was that p for plane was taken [13:58] <divVerent> by p for pointer ;) [13:58] <durandal_1707> :) [13:58] <divVerent> but sure, adding a plane var [14:00] <divVerent> do we use C99 BTW? [14:01] <durandal_1707> silly, but no [14:01] <wm4> ffmpeg requires C99 though [14:02] <divVerent> wm4: it does? [14:02] <divVerent> was mainly asking if I can replace this sqrt(re*re+im*im) by hypot(re,im) [14:02] <divVerent> ;) [14:02] <divVerent> hypot() is C99 though, and MS renamed it to _hypot [14:02] <wm4> divVerent: yes, why else do you think they wrote a c99-to-c89 converter just for libav? [14:02] <divVerent> hypot() is not C89, but existed before C99 on many systems [14:03] <divVerent> e.g. virtually any BSD [14:03] <divVerent> and glibc [14:03] <durandal_1707> i was refering to for (int plane ...) { ... [14:03] <divVerent> durandal_1707: hehe, yes... [14:03] <divVerent> I mainly am asking about hypot() [14:03] <divVerent> as I am now cleaning up the ugly macros [14:04] <durandal_1707> just send patch(es) to ffmpeg-devel when ready [14:04] <divVerent> based on master, I suppose? [14:05] <durandal_1707> yup [14:05] <divVerent> well, I mainly want to know before I do that [14:05] <divVerent> +#define CPLX_MAGNITUDE(re, im) sqrt((re)*(re) + (im)*(im)) /* is actually hypot(re, im) */ [14:05] <divVerent> do we NEED this macro at all then ;) [14:05] <divVerent> hypot() is not on every pre-C99 libc, but on most [14:05] <divVerent> MS is the exception, MS deprecated hypot() in favor of _hypot() in 2005 [14:05] <divVerent> so it'll warn in MSVC [14:05] <divVerent> but C99 added hypot() [14:05] <durandal_1707> do it in separate commit and other who carse will define hypot to something useful [14:06] <durandal_1707> for arch where it is not present... [14:06] <durandal_1707> *care [14:06] <divVerent> it probably exists on all [14:06] <divVerent> just warns on MSVC [14:06] <divVerent> just like fopen() ;) [14:07] <durandal_1707> dont worry, MSVC guys fill provide fix as soon something break.... [14:07] <divVerent> also, c99wrap SURE already fixes this [14:07] <divVerent> given this one can even be fixed by -Dhypot=_hypot [14:08] <divVerent> so hardest part now is cleaning up the history :P [14:08] <durandal_1707> is there someone on #ffmpeg that claims native aac encoder is good? [14:10] <saste> wm4: the "clean" solution, i guess, would be to implement a seek command in the various filters [14:10] <saste> then you simply send a seek to the relevant filters [14:11] <wm4> saste: I did that in mpv [14:11] <durandal_1707> and what seek would do in each filter? [14:11] <saste> we could even script the key-binding, so each user can choose her favored keystrokes [14:11] <wm4> durandal_1707: reset state [14:11] <saste> durandal_1707, it depends [14:11] <saste> some filters don't need to be reset (e.g. hflip, vflip, etc.) [14:11] <saste> movie could actually seek in the input [14:11] <wm4> though I actually wonder if there any filters which really benefit from a seek command, instead of recreating the graph [14:12] <wm4> my problem is more that recreating the graph is a complicated operation [14:12] <saste> wm4, ffplay_input could be changed to support a seek operation [14:13] <saste> again some scripting seems the right solution, so you can change/program the binding at the application level [14:13] <wm4> also, I've said it before, but string commands suck [14:13] <wm4> for both caller and filter implementations [14:16] <saste> wm4, willing to help in that area? [14:17] <wm4> saste: possibly... [14:18] <saste> wm4, first step would be to make all the filters parse the argument string with av_opt_ [14:18] <saste> at that point we can switch and factorize initialization [14:19] <saste> i don't think there are many filters left [14:19] <saste> then we could provide a secondary init() interface, which accepts options as an AVDictionary or whatever [14:20] <cone-469> ffmpeg.git 03Paul B Mahol 07master:402ea625ab7b: lavfi/biquads: remove pointless casts [14:20] <saste> this could simplify scripting, since you would need no more escaping hell since each option can be specified in a separate argument [14:21] <saste> (or you could just set options on the filter) [14:27] <saste> is ffmpeg going to compete with sox? [14:28] <saste> otoh sox doesn't seem to be very active [14:29] <wm4> can't ffmpeg just use sox instead? [14:29] <saste> wm4, yes [14:29] <saste> and ladspa as well [14:32] <durandal_1707> ffmpeg biquads are faster than sox ones (planar<->interleaved) sox is missing optimizations [14:45] <Compn> someone (carl !! you there? :) want to retest if https://ffmpeg.org/trac/ffmpeg/ticket/1957 is fixed now? [14:56] <durandal_1707> Compn: it is duplicate of ticked which is fixed [15:05] <durandal_1707> divVerent: i like your patch [15:06] <divVerent> will soon post it [15:07] <divVerent> I divided it back up into a few commits, but the big change of how your method works can't really be separated out [15:07] <divVerent> also, did I mention that I hate that YUV444P is not actually YUV444P but YVU444P? ;) [15:12] <Compn> durandal_1707 : yes, but is it fixed or is it a mis-duplicate tag ? :P [15:15] Action: Compn going to upset carl by reopening random bugs and turning them into feature requests :D [15:32] <saste> Compn, we should tag tickets with a difficulty level [15:32] <saste> so newbee contributors can pick the easy ones [15:34] <cone-469> ffmpeg.git 03Giorgio Vazzana 07master:514216d8a928: lavd/v4l2: read the correct time per frame from devices that support a standard [15:34] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:a842dc632eca: doc/indevs: apply misc fixes to the v4l2 documentation [15:35] <durandal_1707> saste: can that be added to trac? [15:35] <saste> durandal_1707, sure [15:35] <saste> vlc trac has it at least [15:35] <saste> a tag cloud would be fun as well [15:36] <durandal_1707> than lets add it [15:36] <saste> who's the designated maintainer for trac? [15:41] <Compn> no idea [15:41] <Compn> saste : good idea re difficulty level [15:45] <Compn> i dont think i'm an admin on trac. [15:45] <Compn> probably michaelni [16:00] <durandal_1707> hmm amovie cant use http:// .... ? [16:01] <wm4> durandal_1707: AFAIK that should work? [16:02] <durandal_1707> amovie cant use http:// .... ? [16:03] <ubitux> backquote the ':' [16:03] <ubitux> it's an option separator [16:03] <wm4> d'oh [16:11] <Compn> that info in the docs? [16:11] <Compn> example with http url be nice :) [16:11] Action: Compn gone [16:12] <durandal_1707> what i should put before : ? [16:14] <durandal_1707> i tried / and ` [16:14] <ubitux> \ obviously [16:15] <durandal_1707> it does not work [16:15] <ubitux> the \ must be present, is your shell interpreting it? [16:15] <durandal_1707> \\ works [16:16] <ubitux> yes because otherwise the shell "escape" the :, basically removes the \ [16:16] <durandal_1707> was waste of time, lavfi cant handle mp3 stream [16:18] <saste> durandal_1707, no? [16:19] <durandal_1707> i get spam: [16:19] <durandal_1707> [mp3 @ 0x29878400] Header missing [16:19] <durandal_1707> [Parsed_amovie_0 @ 0x29809140] Decode error: Invalid data found when processing input [16:20] <durandal_1707> even if i download such file [16:42] <saste> Maximum number of registered filters 128 reached, [16:42] <saste> uh... [16:44] <durandal_1707> lol, really? [16:46] <durandal_1707> indeed [16:46] <durandal_1707> saste: how you trigered it? [16:46] <saste> durandal_1707, i just updated my ffmpeg [16:47] <saste> it should be that i enabled more filters than you [16:47] <durandal_1707> exactly [16:47] <saste> patch to raise the max number incoming [16:47] <durandal_1707> isn't it part of ABI/API , we need to bump major [16:48] <durandal_1707> lavf does not have such limit? [16:48] <saste> durandal_1707, no, it's internal [16:49] <durandal_1707> it should raise on its own, so no need to increase number each time limit is reached [16:51] <durandal_1707> i want to do this with ffmpeg: sox -n o.sox synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 [16:53] <saste> yes it's weird, i don't remember why we did filter registration that way [16:56] <saste> durandal_1707, can't reproduce the mp3 issue with amovie [16:57] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:e47114d5459a: lavfi: increment max number of registered filters from 128 to 256 [17:01] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/9db2af3387cdae336f6d831702da5f6e.png [17:01] <divVerent> this is with stolen code from sox though ;) [17:03] <durandal_1707> this is palette or? [17:03] <divVerent> nope [17:03] <divVerent> https://gist.github.com/4683934 [17:03] <divVerent> this is the code [17:03] <divVerent> stolen from sox's palette generation [17:04] <durandal_1707> good, it would be nice to make it generic so other shemes can be added... [17:04] <divVerent> I don't like this though [17:04] <divVerent> it's slow, and nasty (YUV conversion...) [17:04] <divVerent> ideally I'd like to replicate the same curves but in YUV space [17:05] <divVerent> or rather, very similar ones [17:05] <divVerent> I tried to do it, but yellow became pink [17:05] <divVerent> rest works [17:05] <divVerent> wonder if I should just make it an array of (value, y, u, v) items [17:05] <divVerent> and it'd linearily interpolate [17:06] <divVerent> and I'd predefine it for sox's colors at all "critical" points [17:06] <divVerent> i.e. all intensity values mentioned there, and 0.30 (middle of the sin() curve for b) [17:06] <durandal_1707> well having sox as some kind of palette is extra feature [17:07] <divVerent> also, as you see, i also made logarithmic scale for this one [17:07] <durandal_1707> it uses 256 colors only, which i find too low [17:08] <durandal_1707> but it can be used in sox mode, if one wants to have same output as sox does.... [17:09] <divVerent> well, I rather would prefer it to be a table with linear interpolation [17:10] <divVerent> as that CAN be more accurate in the end [17:12] <durandal_1707> sure [17:17] <divVerent> yes, works [17:23] <cone-469> ffmpeg.git 03Paul B Mahol 07master:22fad4e2b8c7: doc/filters: fix afade cuve types listing [17:24] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/180081251f2e54275685c44b505b66bb.png sox colors with square root scale (old ffplay) [17:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/d7fd80be0fd8e92c8f7941f12a87dafb.png sox colors with logarithmic scale (sox) [17:24] <divVerent> both look useful, actually [17:28] <durandal_1707> divVerent: yes [17:31] <durandal_1707> shit, typo [17:31] <durandal_1707> i hate typos [17:39] <divVerent> https://gist.github.com/4684199 for you to test, will post it on the ML later [17:40] <divVerent> I recommend the combinations: [17:40] <divVerent> - showspectrum (separate, otherwise ffplay-like) [17:40] <divVerent> - showspectrum=combined=1 (ffplay replication) [17:40] <divVerent> - showspectrum=intensity=1:logscale=1 (sox) [17:40] <divVerent> - showspectrum=intensity=1 (soxlike but ffplay's scale) [17:41] <divVerent> - showspectrum=saturation=0 (what your original code did) [17:42] <divVerent> saturation=-1 yields an alternate color scheme ;) [17:42] <durandal_1707> my original code was just to not ignore other channels .... [17:42] <divVerent> yes [17:42] <divVerent> I mainly like using saturation in combined mode [17:42] <divVerent> to exaggerate channel differences [17:43] <durandal_1707> does this one still depends on old patch of mine? [17:43] <divVerent> the series includes it [17:43] <divVerent> but I could squash the first two, but didn't to keep your credit [17:43] <durandal_1707> it is splitted? [17:44] <divVerent> my patch has been split, but the first change that basically rewrote yours is not split [17:44] <durandal_1707> divVerent: i dont care for my credits [17:44] <divVerent> then I'd squash the first two [17:47] <divVerent> https://gist.github.com/4684260 would be the same with the first two changes squashed and explained as a single commit [17:48] <divVerent> but for finally committing it, it probably would be best to have all changes in a single commit in the end [17:48] <divVerent> for reviewing it may be better this way though [17:50] <durandal_1707> i'm fine with whatever combination [17:51] <divVerent> so am I... will play around with it a bit more, but probably won't change it much any more [17:51] <divVerent> as for the one-image-output feature, not sure... I also want to look into the Y scale though [17:52] <divVerent> as in, where it comes from [17:52] <divVerent> whether top is nyquist, or something else [17:53] <divVerent> it SHOULD be nyquist ;) but I don't trust the code regarding that yet [18:04] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:7e0fc1a24fa6: CREDITS: redirect to Git log, remove current outdated content [18:09] <divVerent> yes indeed... nyquist is ONLY at the top when the height is a power of two (if not combined, multiplied by channel count) [18:09] <divVerent> does FFT size HAVE to be a power of two? if yes, it's difficult to fix [18:10] <divVerent> compare output with s=512x256 to s=512x257 ;) [18:10] <durandal_1707> divVerent: first commit/ my code have some flaws, see av_size_mult() - this is supposed to check for overflows [18:10] <durandal_1707> but i did not updated it ... [18:12] <durandal_1707> also you should use av_realloc_f or otherwise leaks happens when memory alloc fails [18:15] <divVerent> ok, committed fix for these two locally [18:16] <divVerent> what do you suggest about the nyquist issue? [18:16] <divVerent> the thing is, APPARENTLY only power-of-two FFT sizes are supported [18:17] <divVerent> so if you have (evil case) 7 channels [18:17] <divVerent> you only see the full frequency range if the output size is 7*poweroftwo [18:17] <divVerent> issue existed before too, but then only applied to the full size [18:17] <divVerent> default height of 480 was bad for the same reason :P [18:18] <divVerent> ignore it, like before - but set the output height to 512 so by default, you do see the full spectrum for 1ch and 2ch at least? [18:19] <divVerent> the only alternative would be using a scaling filter to trim the size down [18:19] <divVerent> I'd rather not [19:17] <cone-469> ffmpeg.git 03Ronald S. Bultje 07master:a5ba798c16d0: dsputil: remove unused functions copy_block{2, 4, 8, 16}. [19:17] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:ed341549bbc2: Merge commit 'a5ba798c16d0614d982a76755fdd72b37d437170' [19:27] <cone-469> ffmpeg.git 03Martin Storsjö 07master:a65f965c04bf: mpegvideo: Do REBASE_PICTURE with byte pointers [19:27] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:324498f82470: Merge commit 'a65f965c04bfa27adedc0409c14cc05903f483d0' [19:29] <saste> what happened to the ffmpeg -i option? [19:30] <michaelni> saste, ? [19:31] <saste> michaelni, I'm trying to set -user_agent, and failing [19:31] <saste> also I can't see the "i" entry in the options array (ffmpeg_opt.h) no more [19:31] <saste> ffmpeg_opt.c [19:31] <michaelni> the parsing code was redesigned a few times [19:32] <michaelni> its more convoluted now [19:33] <saste> so now is more "evil" [19:34] <saste> but, it seems I can't set avio options [19:34] <saste> is that a regression or what? [19:34] <michaelni> did it work before ? [19:35] <saste> dunno, but I'm trying an example: ffplay -user-agent foobar URL and it seems to fail with unknown option [19:35] <saste> the example was taken from the docs, so I thought about a regression [19:37] <saste> but maybe it was never possible [19:37] <saste> wm4, how can you reproduce the http cookies issue? [19:39] <saste> in other words, how can I pass protocol options to avformat_open_input()? [19:39] <saste> is that possible? [19:42] <saste> wow, why all these MP3 hacks in avformat_open_input()? [19:42] <michaelni> btw did you try -user-agent or user_agent ? [19:43] <michaelni> ffmpeg -h full has it with - not _ [19:49] <saste> michaelni, i'm debugging the issue [19:52] <saste> it is setting the option, but then for some "evil" reason it fails later in ffmpeg_opt.c code [20:02] <cone-469> ffmpeg.git 03Diego Biurrun 07master:0f5b0b4178b8: avisynth: Change demuxer name to avoid conflicts with AVS [20:02] <cone-469> ffmpeg.git 03Diego Biurrun 07master:29f1fa74225e: configure: Move newlib libc check before mingw libc check [20:02] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:01d012ba3db9: Merge commit '29f1fa74225e9fdbab6d590857edbecb51cf74da' [20:08] <cone-469> ffmpeg.git 03Diego Biurrun 07master:197252f1c58b: configure: Add a comment indicating why uclibc is checked before glibc [20:08] <cone-469> ffmpeg.git 03Diego Biurrun 07master:52acd79165e7: x86: hpel: Move {avg,put}_pixels16_sse2 to hpeldsp [20:08] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:dd87d4a318b3: Merge remote-tracking branch 'qatar/master' [20:20] <saste> michaelni, ffplay -user-agent foobar http://ffmpeg.org/ffmpeg-logo.png [20:24] <saste> oh wait it's a bit of a mess [20:25] <saste> lavf see that the url is an image file, so it doesn't open avio [20:25] <saste> so it doesn't consume the option -> error [20:26] <saste> so i can't see no way to pass the url options here [20:26] <saste> also avformat_open_input() has no way to discriminate avio options from demuxer options [21:00] <Daemon404> does lavfi have a way to generate blank input easily? [21:00] <Daemon404> i.e. i need somthing good to test encoding threading [21:11] <llogan> Daemon404: what do you mean by blank input? [21:11] <Daemon404> llogan, i used /dev/zero. nevermind. [21:12] <saste_> Daemon404, color=c=bikeshed [21:12] <llogan> and "aevalsrc=0" [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:6560625f0aec: lavfi/bufferqueue: add ff_bufqueue_is_full(). [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:ae14887ee7f2: lavfi/af_amerge: check for buffer queue overflows. [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:dcce43340f45: lavfi/af_amerge: set outbuf->audio->channels. [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:d2ccab5f8a5f: lavfi/buffersrc: forward filter_frame errors. [21:23] Action: Compn wonders why saste has a hardcoded max filter number [21:23] <Compn> its like hardcoding a max codec number. a foolish thing! :P [21:30] <durandal_1707> divVerent: just send patches, and size could be set for channel height or rounded from window size so some part of window is black.... [21:31] <durandal_1707> also sox have alternative pallete, if you want to do it too.... [22:14] <durandal_1707> can somewhere help this guy who keep asking user questions in community on ffmpeg? [22:16] <saste_> durandal_1707, what guy? [22:16] <durandal_1707> g+ guy [22:17] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:349e7f423f19: doc/indevs: itemize examples for the v4l2 device [22:17] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:b999774f0b11: doc/indevs: document v4l2 options [22:22] <durandal_1707> so to pad start of audio you use aresample, there really should be example in doc... [22:27] <llogan> durandal_1707: link? [22:36] <durandal_1707> llogan: what link? [22:39] <llogan> nevermind. i see what you're referring to. [22:40] <durandal_1707> why are there two cone- bots? [22:41] <llogan> you already told him to get help at the mailing list. not much else to say. [23:10] <kierank> shahriman: did you see the vc-1 interlaced patch [23:10] <shahriman> no i did not. where is it? [23:10] <shahriman> kierank: ^ [23:10] <kierank> on the ml [23:11] <kierank> j-b: ^ [23:11] <shahriman> kierank: can you please give me the link? [23:11] <kierank> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/138302.html [23:15] <shahriman> kierank: i forgot almost everything i knew about that code, but the patch looks ok to me [00:00] --- Fri Feb 1 2013

1 0

ffmpeg.log.20130131
by burek 01 Feb '13

01 Feb '13

[00:18] <ska> oops. sorry. [00:21] <sacarasc> You should be sorry! [00:21] <sacarasc> What for? [00:37] <K-Rich> Hi all [00:37] <K-Rich> Cananyone tell me why this doesn't work? ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 -f yuv4mpegpipe | tee test.avi | mplayer -cache 1024 -nosound - [00:38] <K-Rich> sorry ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 -f yuv4mpegpipe - | tee test.avi | mplayer -cache 1024 -nosound - [00:45] <K-Rich> anyone? [00:45] <K-Rich> sorry ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 - | tee test.avi | mplayer -cache 1024 -nosound - [00:45] <K-Rich> gives errors [00:47] <K-Rich> http://pastie.org/5976041 [00:57] <K-Rich> and still noone [00:57] <K-Rich> heh [01:10] <klaxa> K-Rich: two things [01:11] <klaxa> actually three [01:11] <K-Rich> ok [01:11] <klaxa> a) you are using avconv, this is the wrong channel unfortunately [01:11] <klaxa> b) apparently you are not allowed to access /dev/video0 [01:11] <klaxa> c) why do you bother to capture audio if you don't play it back with mplayer? [01:11] <K-Rich> (mind you i "just" tried to change to real ffmpeg, with this result http://pastie.org/5976323 )) [01:12] <klaxa> ah didn't see the tee pipe [01:12] <K-Rich> i want the audio captured in the file but want to see the video, if the displayed video had sound it would cause a feedback loop with the recording [01:13] <klaxa> i'm not sure, but i think when playing back raw audio (pcm_s16le) and raw video you will somehow have to tell mplayer that it is raw data [01:13] <K-Rich> also i am in the group video and 'crw-rw----+ 1 root video 81, 0 Jan 30 15:18 /dev/video0' [01:13] <klaxa> also the order of your parameters is a bit weird [01:15] <K-Rich> i'm just learning this, trying to make an applet for screen/webcam recording [01:15] <klaxa> i thnik general practice is ffmpeg -f <in_format> [in_options e.g. codec, filters] -i <in_stream> [-f container format] <out_file> [01:15] <klaxa> *think [01:17] <K-Rich> if put a filename (capture.avi) rather than '-| tee test.avi | mplayer -' it works, just doesn't display [01:18] <klaxa> ah wait... [01:18] <klaxa> right... [01:18] <klaxa> ffmpeg <all your stuff> -f avi pipe: | tee somefile.avi | mplayer - [01:19] <K-Rich> /dev/video0: Operation not permitted STILL ugh, even when i do it with sudo [01:20] <klaxa> move the -i /dev/video0 before -vcodec even [01:22] <K-Rich> http://pastie.org/5976457 [01:22] <K-Rich> same friggin access denied crap [01:22] <K-Rich> err operation not permitted [01:23] <K-Rich> hmmmmmmmm [01:24] <K-Rich> one sec BRB [01:24] <klaxa> try also adding something like -f video4linux2 before -i /dev/video0 [01:24] <K-Rich> i wonder if all the trying somehow left it open [01:24] <K-Rich> ok [01:35] <K-Rich> getting closer [01:35] <K-Rich> i can see it [01:36] <K-Rich> but the vid plays fast and the sound is empty (static only) [01:36] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -f video4linux2 -i /dev/video0 -vcodec rawvideo -f avi pipe: | tee test.avi | mplayer -cache 1024 -nosound - [01:38] <K-Rich> klaxa: you still here? [01:38] <klaxa> yeah [01:38] <klaxa> um... about the audio thing, have you selected the correct recording device in pulse? [01:39] <klaxa> with the video thing, i'm not sure rawvideo has data about framerates [01:40] <K-Rich> okay using mjpeg rather than rawvideo works sorta, way delayed and still no sound [01:41] <klaxa> open pulseaudio volume control and check you have the correct device set for the recording stream [01:45] <K-Rich> okay lol, got sound...... [01:45] <K-Rich> how do i know what encoders i have, i need one with yuv [01:45] <K-Rich> still have the delay though [01:56] <klaxa> K-Rich: ffmpeg -codecs [02:04] <K-Rich> klaxa: woo hoo (sorta) [02:04] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -f video4linux2 -i /dev/video0 -vcodec huffyuv -f avi - | tee test.avi | mplayer -nosound -cache 1024 - [02:05] <K-Rich> works great, except there is a delay on the video... any idea on that? [02:05] <klaxa> mmhh no idea actually, maybe huffyuv is straining your cpu? although it shouldn't really [02:06] <K-Rich> the only one without a delay was rawvideo, but it played at like double speed, though the audio was normal speen (the image pauses half way through) [02:15] <K-Rich> grrrrrr [02:15] <K-Rich> updatung the ffmpeg made my deskrecord script look like ass (pixilated) [02:15] <K-Rich> it was clear before [02:18] <K-Rich> fow is it a command that worked with avconv doesn't work with ffmpeg ? [02:18] <K-Rich> it's become all blocky (pixilated) [02:19] <klaxa> sounds like a quality setting issue [02:21] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 1280x800 -qscale 1 -r 30 -vcodec rawvideo -f x11grab -i $DISPLAY -s 640x356 -vf pad=640:360:0:2:000000 ~/Desktop/Video_$(date +%F_%T).avi [02:23] <klaxa> hmmm that should work though [02:25] <K-Rich> it looks like it's an animated jpeg with low quality, all blocky [02:25] <K-Rich> it worked great with ubuntus avconv ffmpeg whatever [02:26] <K-Rich> Stream mapping: [02:26] <K-Rich> Stream #1:0 -> #0:0 (rawvideo -> mpeg4) [02:26] <K-Rich> Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame) [02:26] <K-Rich> mpeg4 wtf? [02:27] <K-Rich> ffmpeg version 0.10.6-6:0.10.6-0ubuntu0jon1~precise1 [02:29] <K-Rich> Mwahahahaa got it [02:29] <K-Rich> added -sameq [02:29] <K-Rich> works great now :) [02:33] <K-Rich> heh... made the other command better as well [02:33] <K-Rich> the output at least [02:33] <K-Rich> though there is still delay :/ [02:42] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -sameq -r 30 -f video4linux2 -i /dev/video0 -vcodec huffyuv -f avi - | tee test.avi | ffplay -an -window_title Webcam\ Recording - [02:42] <K-Rich> hmmmmmmm still a delay :/ [05:35] <Tronic> Strange problem with my own software... Decoding of MP3 and Ogg (didn't try other formats yet) produces highly distorted sound (lots of white noise etc), but the actual audio can be heard thru. [05:36] <Tronic> Same source code has worked with libav and older ffmpeg versions. Also, ffplay, with current version, decodes audio just fine. Video decoding in my application is also working properly. [05:37] <Tronic> Any suggestions other than trying to go thru my own code and ffplay line by line, seeking for diffferences? [05:40] <Tronic> I was also afraid that different library versions might be mixing up but uninstalling everything but the latest didn't change anything. I am running current git version at the moment. [05:54] <Tronic> With WAV format audio there is no distortion. [09:34] <praveenmarkandu> hi. if a source file is already h.264 and aac and when i transcode it again and use -c:v libx264 -c:a libfaac [09:34] <praveenmarkandu> will it be slower than just using -c:v copy [09:37] <praveenmarkandu> what i actually want to do is reduce the bitrate and resolution of the video [09:37] <praveenmarkandu> will copy make a speed difference? [10:11] <cbsrobot> praveenmarkandu: copy is faster [10:11] <praveenmarkandu> cbsrobot, i just tried copy with my full command and the -vf scale didnt seem to work [10:12] <cbsrobot> yeah stream copy disables the filters [10:13] <praveenmarkandu> oh okay [10:13] <praveenmarkandu> and -s:v is the same as -vf scale [10:13] <praveenmarkandu> so it wont work either right? [10:15] <praveenmarkandu> and it seems it also disables the video bitrate setting [10:15] <praveenmarkandu> hah. okay. should have expected [11:43] <lemonjelly> Hello. With ffmpeg, is it possible to offset all timestamps in the *output* streams by a certain amount? [11:45] <lemonjelly> I've been experimenting with the -itsoffset and -copyts options but I can't get ffmpeg to behave as I need [11:47] <lemonjelly> Basically I'm trying to output mov fragments of an ism stream, but I'm using the -ss <offset> option to start partway through the input stream, and I want the first fragment I output to have a frag_start value matching <offset>. But it is always 0. [13:14] <Zeeflo> burek, are you around? You got a bug of some sort in your build [13:14] <Zeeflo> (which has just set me back 2 days!! :/) [13:35] <someone-noone> Hello! I'm getting crash(from time to time) in avcodec_decode_audio4()->aac_decode_frame()->av_packet_get_side_data() with next code: http://ideone.com/uxeYwz [13:35] <someone-noone> I can't figure out what can be a problem. Sometimes problem occur and sometimes no [13:35] <someone-noone> Any help? [13:39] <Zeeflo> Any ideas? fontconfig: Selected font is not the requested one: 'DejaVu Sans' != 'Arial'its/s dup=1 drop=0 [13:41] <Mavrik> someone-noone: it seems something goes wrong when you're poping the packet [13:41] <Mavrik> are you copying all fields? [13:42] <Mavrik> is your buffer padded with at least FF_BUFFER_PADDING (or what's it) bytes? [13:43] <someone-noone> Mavrik, one momen [13:44] <someone-noone> Mavrik, http://ideone.com/VBliwr this is how I'm reading packets. I'm porting my code from old version of libav. [13:45] <someone-noone> Mavrik, and buffer& you're talking about AVFrame's one? [13:53] <ledil> Hello, Ive installed ffmpeg from jon-severinsson ubuntu ppa repository, but now Im getting "Unknown encoder 'libfaac'". libfaac-dev and libfaac-0 are installed ... any hints ? [13:54] <ledil> this is my version: 0.10.6-6:0.10.6-0ubuntu0jon1~lucid2 [13:54] <ledil> when executing ffmpeg, i cant see a --enable-faac ... is faac deprecated ? [13:54] <someone-noone> ledil, ffmpeg codecs [13:54] <someone-noone> do you see libfaac? [13:55] <Zeeflo> ledil, try the daily build for ubuntu [13:55] <Zeeflo> ah you are [13:55] <Zeeflo> hmm [13:56] <ledil> http://pastebin.com/8xCYWaTc [13:56] <ledil> here is the ouput [13:56] <ledil> of ffmpeg -codecs and faac is missing ... [13:57] <ledil> dont know what to do ? because when Im using the mediaxxx I cant install libavcodec-extras-52 because it conflicts with ffmpeg ... when using jon-severinsson package, im getting "libfaac" unknown ... [13:57] <ledil> I cant find a good ffmpeg package, at the moment :( [13:57] <JEEB> because faac is nonfree and thus impossible to enable if you're making a binary for others [13:58] <ledil> how to solve this ? [13:58] <JEEB> I think even debian stopped linking to it after that was found [13:58] <Zeeflo> nope [13:58] <Zeeflo> works fine here [13:58] <Zeeflo> wheezy 3.2 kernel [13:58] <ledil> my server is lucid [13:59] <JEEB> not sure but in any case I think very recent builds no longer have faac enabled as it's nonfree, methinks [13:59] <JEEB> ledil, 1) switch to fdk-aac, as that is of better quality for AAC 2) compile yourself [13:59] <JEEB> that's the only way [13:59] <Zeeflo> im using Bureks builds.. faac is working here [13:59] <JEEB> both faac and fdk-aac are nonfree [13:59] <JEEB> o_O [13:59] <JEEB> burek has faac enabled? [13:59] <JEEB> Zeeflo, paste ./ffmpeg output [13:59] <Zeeflo> im using the aac libs in my encodings, so yea, id guess so! [13:59] <JEEB> pastebin [13:59] <Zeeflo> sure [14:00] <JEEB> it should have the configure line too [14:00] <someone-noone> ledil, I'd just install ffmpeg from sources [14:00] <Zeeflo> http://pastebin.com/KnJHXz6L [14:01] <JEEB> with fdk-aac instead of faac [14:01] <JEEB> because fdk-aac is fraunhofer's encoder [14:01] <Zeeflo> and this is my input: -c:a aac -strict experimental -ac 2 -b:a 128k [14:01] <JEEB> oh [14:01] <JEEB> that uses the internal one [14:01] <JEEB> not faac [14:01] <Zeeflo> ah [14:01] <JEEB> that'd be called ffaac (ff-something is the internal thing) [14:01] <Zeeflo> but aac is aac right ;) [14:01] <JEEB> yes, but that is experimental and barely as good as vo-aacenc [14:01] <JEEB> (or somewhat better) [14:02] <Zeeflo> it works perfectly well [14:02] <JEEB> but still worse per the same rate than faac or fdk-aac [14:02] <JEEB> and faac isn't exactly great either [14:02] <ledil> Zeeflo: you are using bureks builds ? what ppa repository is that ? i cant switch to fdk-aac at the moment, I need to find a repository with a ffmpeg that has faac support [14:02] <JEEB> there are none [14:02] <JEEB> because faac is nonfree [14:02] <Zeeflo> ledil, you need just the aac [14:02] <JEEB> as I already said, Zeeflo is not using faac [14:02] <ledil> ah,ok [14:02] <JEEB> well, yes -- the "experimental" status mostly comes from the fact that it's not optimal, not that it doesn't work Zeeflo [14:03] <Zeeflo> those two does the same thing [14:03] <Zeeflo> JEEB, I am aware of that, but as long as it works, it works? [14:03] <JEEB> -_- [14:03] <Zeeflo> for my stereo 128k it works like I need it to [14:03] <JEEB> as soon as you hit something that goes below the comfort level of the libavcodec encoder it will be derp [14:03] <JEEB> at least it no longer adds random noise [14:03] <JEEB> it used to do that like 1.5 years ago [14:04] <Zeeflo> dunno about that :) [14:04] <Zeeflo> It works for me :) [14:04] <JEEB> well, if you don't care about audio quality or don't hear the difference then the internal one is OK [14:04] <Zeeflo> you cant hear any difference at stereo level anyways [14:04] <JEEB> you can [14:04] <Zeeflo> u got super ears then [14:04] <JEEB> no [14:04] <JEEB> I've got bad ears [14:05] <Zeeflo> well, im not trying to create THX 7.1 lossless sound here.. [14:05] <JEEB> I have good eyesight for video compression artifacts and bad ears, and I can still easily hear that when you happen to drop below the comfort level of libavcodec's aac encoder, it's worse than fdk-aac here [14:05] <JEEB> and the comfort level is probably somewhere around 128k [14:06] <JEEB> although when I had to use the libavcodec encoder I kept to 192k just in case [14:06] <Zeeflo> you couldnt hear the difference in 128k to 160 or 192k [14:06] <Zeeflo> no one can [14:06] <JEEB> oh yes you can with the libavcodec encoder [14:06] <Zeeflo> dude come on.. [14:06] <JEEB> it's better than the libavcodec vorbis encoder, that's for sure [14:07] <JEEB> probably better than the libavcodec wma encoder, too [14:07] <JEEB> I know that there are encoders where at those rates you really shouldn't be able to distinguish stuff, but the libavcodec's encoder is not one of those [14:08] <Zeeflo> then you probablay got Monster cables with 24k gold all the way through, with oxydized this and that priced at $1000 pr inch on a $200.000 home stereo [14:08] <JEEB> ... [14:08] <Zeeflo> the human ear cannot hear the difference [14:08] <ledil> JEEB, fdk-aac, do I need an additional package to install ? [14:08] <JEEB> yes, you would have to compile it as well [14:08] <Zeeflo> you gotta compile it [14:08] <ledil> or is it only "-acodec fdk-aac" ? [14:09] <JEEB> it is nonfree as well because of the crappy license fraunhofer put on the source [14:09] <ledil> JEEB, is there binary package ? [14:09] <JEEB> so it cannot be distributed either [14:09] <JEEB> in binary [14:09] <JEEB> (linked with (L)GPL at least) [14:09] <Zeeflo> Whats your suggestion instead of bureks aac then? [14:09] <Zeeflo> ill make a sound test [14:09] <Zeeflo> a clinic sound test [14:10] <JEEB> fdk-aac with afterburner on, start around 96k and go up from there with various samples [14:10] <Zeeflo> @ constant bitrate [14:10] <Zeeflo> @128k [14:10] <Zeeflo> vs aac [14:10] <Zeeflo> ill do that [14:11] <JEEB> that's a bit on the high'ish side for fdk-aac probably, but with many sources the libavcodec aac encoder should start showing its skin there [14:11] <Zeeflo> and ill show you the results [14:12] <Zeeflo> crap its cold here [14:12] <JEEB> also please do not compare me with the so-called "audiophiles", I'm all for valid tests and such, and I am just saying that the libavcodec encoder just sucks at its job since no-one finished it >_> [14:12] <Zeeflo> im freezing [14:12] <JEEB> if someone would have the time to put into the libavcodec encoder it'd be great [14:12] <JEEB> unfortunately, it's simpler in most cases for people to just compile fdk-aac and use it locally [14:13] <Zeeflo> people probably just think like me: It works well enough, leave it at that. [14:13] <Zeeflo> also [14:13] <Zeeflo> its the downside of GPL [14:13] <Zeeflo> no-one can force any-one to finish :D [14:13] <JEEB> more like open source [14:13] <JEEB> not specific to GPL in any way [14:14] <JEEB> or all software to be honest [14:14] <Zeeflo> i stand corrected, but you know what I mean [14:14] <JEEB> even with payware there are cases where a developer has stopped developing the app and no matter how much the clients derp he won't finish it [14:14] <JEEB> ("finish" as in fix the bugs etc.) [14:14] <JEEB> also, if you really want a laugh [14:15] <JEEB> try the "ogg" encoder [14:15] <JEEB> not libvorbis, make sure it uses the internal one [14:15] <JEEB> I guarantee you will have a laugh :) [14:15] <JEEB> so compared to that, the aac encoder is much better [14:18] <Zeeflo> h264 wont work with ogg would it? [14:19] <Zeeflo> in a mp4 container [14:19] <JEEB> yes, because it's a container that doesn't specify a way to mux vorbis in it. It has nothing to do with H.264 :) [14:19] <Zeeflo> i use libx264 with aac for mp4 containers, for streaming (rtmp) [14:19] <Zeeflo> flash and html5 fallback [14:20] <Zeeflo> i dont think ogg would work with that? [14:20] <JEEB> I'm not saying you should use it [14:20] <Zeeflo> would flash eat that? [14:20] <JEEB> I just said that if you were going to try out aac and fdk-aac, you might as well quickly encode the same thing with the vorbis encoder [14:20] <Zeeflo> well. that would depend on the player.. [14:20] <JEEB> just for laughs :P [14:20] <JEEB> also rtmp(e) is flv [14:20] <JEEB> not mp4 [14:21] <Zeeflo> nah, im using mp4 [14:21] <JEEB> of course your rtmp(e) service can take in already encoded mp4 [14:21] <JEEB> and remux that on the fly [14:21] <JEEB> into flv [14:21] <JEEB> just that live streaming doesn't fly with that [14:21] <JEEB> unless your mp4 muxer suddenly supports movie fragments [14:21] <Zeeflo> it does [14:21] <Zeeflo> -movflags [14:22] <Zeeflo> +faststart [14:22] <JEEB> uhhh [14:22] <JEEB> that's not it [14:22] <JEEB> that's just putting the index in the front /after/ encoding [14:22] <Zeeflo> do you mean somthing like qtindexing then? [14:23] <Zeeflo> moov atoms? [14:23] <JEEB> no, I mean like something like L-SMASH movie fragments feature. the libavformat demuxer supports it, but muxer AFAIK doesn't [14:23] <JEEB> moov atom moving is moving the index [14:23] <JEEB> can only be done /after/ encoding [14:23] <Zeeflo> i dont know about movie fragtments.. [14:23] <Zeeflo> My setup works.. [14:23] <JEEB> yes, because your thing most probably just reads the mp4 and remuxes it to flv on the fly when feeding it to rtmp(e) [14:23] <Zeeflo> MP4's on amazons CDN (cloudfront) played with JW player 6.1 [14:24] <JEEB> I'm just telling you that rtmp(e) by itself is flv [14:24] <muken> JEEB: ? libavformat muxer supports movie fragments [14:24] <JEEB> muken, oh? [14:24] <JEEB> I've just never seen them used [14:24] <JEEB> lol [14:32] <JEEB> Zeeflo, btw if you really want to see what the CDN is doing to your content, you can point ffmpeg to the rtmp(e) URL the player reads (ffmpeg -i rtmp://url:port ) and see what format etc. it sees coming towards you [14:32] <JEEB> I would guess it just remuxes your mp4 files into flv and transports them like that [14:33] <Zeeflo> i would have to make the file public in the bucket then wouldnt i? [14:33] <JEEB> nope [14:33] <JEEB> at least I have no idea what you're talking about [14:33] <Zeeflo> heh [14:33] <Zeeflo> i know I would have to [14:33] <Zeeflo> :) [14:33] <JEEB> if you are talking about pastebinning, then no -- you can clear out the rtmp(e) url or whatever from the command line and so forth [14:34] <JEEB> and the rtmpe handshake [14:34] <Zeeflo> nono, I am using signed and timed urls for my content [14:35] <JEEB> anyways, just trying to tell you that it pretty surely isn't mp4 when it gets pushed through rtmp(e) [14:35] <Zeeflo> ill make a file public and see :) [14:36] <JEEB> sure, ffmpeg should be able to read what's coming up from the tube with an rtmp:// url as input [14:36] <Zeeflo> BTW [14:37] <Zeeflo> bureks build has a bug [14:37] <Zeeflo> how do I submit it? [14:37] <JEEB> depends on if it's a bug in the build or bug in ffmpeg itself [14:37] <Zeeflo> i dont know.. [14:37] <JEEB> if you still get the bug with building the same revision yourself f.ex. then you poke bug burek [14:38] <JEEB> otherwise your poke the ffmpeg trac issue tracker [14:38] <Zeeflo> when you select subtitles=blabla.srt, if theres no fonts on the server, it doesnt warn you [14:38] <Zeeflo> server=system [14:39] <Zeeflo> it did that on my old server as well [14:39] <Zeeflo> I had to get the ms core fonts [14:39] <Zeeflo> and install fontconfig [14:39] <Zeeflo> and then I had subs in my stream [14:39] <JEEB> so if the system lacks any fonts of any description it just fails to render anything instead of failing with an error? [14:39] <Zeeflo> before I did that, it acted like it was creating a new stream with subs in it, but it wasnt [14:39] <JEEB> uhh [14:40] <Zeeflo> an no warning/error messages [14:41] <Zeeflo> with everything else, ffmpeg is pretty good at warning you or even halting [14:41] <Zeeflo> just not with this! [14:41] <Zeeflo> atleast for the debian build [14:43] <JEEB> that sounds like a ffmpeg bug, but you might want to check how it handles itself in case you build a build with current ffmpeg with libass etc. yourself, and then try it out like that. Although I feel in most cases like that the build would then depend on shared libraries and would actually derp at you for trying to start it up without f.ex. fontconfig to begin with (Before ffmpeg's code actually runs) [14:44] <JEEB> burek's build should have everything in it static methinks, so even if you don't have the stuff around, it would still kind of go forward [14:44] <JEEB> naturally lacking any font cache or anything you wouldn't really get anything [14:45] <JEEB> so I guess that some checks might be looked into, but depending on how "dumb" the checks would have to be done, or where they would have to be done (ffmpeg? or fontconfig? or something else?) it could probably take time to see it get completely "fixed" [14:46] <Zeeflo> yea [14:46] <JEEB> for example, if it's just a case of lax checking in fontconfig [14:46] <JEEB> even if you fix it right now [14:46] <JEEB> it won't fix machines already out there [14:46] <Zeeflo> one just gotta remember to install fonts on his/her new server if they want subs in their streams! Otherwise they might gotta start all over! [14:46] <Zeeflo> <--- like he had to! [14:47] <Zeeflo> :D [14:47] <JEEB> I generally build my stuff myself so I generally grab all the related libraries for that [14:47] <Zeeflo> i tried that as well [14:48] <Zeeflo> but I couldnt get it to support SRT subs [14:48] <Zeeflo> I had to convert everything to ass [14:48] <Zeeflo> untill I got Bureks build [14:48] <JEEB> sure you were building everything new enough? [14:48] <Zeeflo> Besides, I dont think building it myself would make it any bit better! [14:48] <Zeeflo> a wise man said, if it works dont fuck with it [14:49] <Zeeflo> and his build works [14:49] <Zeeflo> atleast for me [14:50] <JEEB> well, it would at least make you not forget fontconfig, fonts you'd still have to grab somewhere but I'd guess in most cases those might be grabbed as a dependency? [14:50] <JEEB> also I'm just used to building ffmpeg and friends by now :V [14:51] <JEEB> I can see why burek's builds are quite useful for many people tho [14:51] <JEEB> they offer an easy way out for those who don't know their ways to compile the stuff they need [14:51] <Zeeflo> i just install it directly [14:52] <Zeeflo> fontconfig isnt a dependency of msttcorefonts [14:52] <Zeeflo> but its needed for ffmpeg [14:52] <JEEB> I know it isn't :| [14:53] <JEEB> also I was talking of it the other way, generally grabbing fontconfig/freetype might end you up with some fonts too in many cases -- I might be completely incorrect tho naturally [14:54] <Zeeflo> im still missing fonts though [14:54] <Zeeflo> ffmpeg is still complaining, but atleast it can fallback to arial [14:54] <Zeeflo> it want Deva Vu [14:55] <Zeeflo> im perfectly fine with arial [14:59] <Zeeflo> JEEB, is there also a command to level the volume fixed? [14:59] <Zeeflo> ive noticed some video files has high volume [14:59] <Zeeflo> some has moderate [14:59] <Zeeflo> some almost has no volume :D [15:04] <durandal_1707> volume , volumedetect filter [15:05] <durandal_1707> they are all listed in filter documentation available on web [15:07] <Zeeflo> i must have missed something i wrote earlier [15:07] <Zeeflo> ;) [15:11] <ubitux> ebur128 is likely more effective than volumedetect [15:12] <ubitux> (but slower) [15:28] <_kud> Hello [15:29] <_kud> I'm trying to do a video with some png; i've got this script= https://gist.github.com/1e0fdf400961e7e5d779 [15:29] <_kud> but it seems not to loop it 20 times [15:29] <_kud> do you have any idea how to make a loop properly? [15:51] <undercash> hello [15:52] <undercash> i m trying to compile ffmpeg on ubuntu precise 64 and i get a ERROR: libx264 not found [15:52] <undercash> which didnt happen for a very long time :) [15:52] <undercash> oops 32bit sorry, testing on laptop [15:53] <JEEB> libx264-dev package is not installed? [15:53] <JEEB> if you want to use the package that is [15:53] <JEEB> if you built libx264 yourself then make sure you had either --enable-static or --enable-shared set [15:53] <undercash> yea it is installed [15:54] <JEEB> check config.log then? [15:54] <undercash> ok let's put it [15:54] <hackeron> hey, I'm trying to record from a granstream video server, I'm doing: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -codec:v copy -codec:a libfaac -flags +qscale -global_quality 1 -afterburner 1 test.mkv --- that gives me broken metalic sound. If I switch libfaac to libmp3lame then I get good audio but it is not in sync with the video. Any ideas? [15:54] <JEEB> lol [15:54] <JEEB> so which is it? [15:54] <undercash> the --enable-static i mean [15:54] <undercash> but note that it wasn't necessary like 2 weeks ago [15:55] <JEEB> undercash, so which is it -- are you building with a pre-built binary package of libx264 or are you building it yourself? [15:55] <undercash> myself [15:55] <JEEB> if you aren't building it yourself you don't add that [15:55] <JEEB> oh [15:55] <undercash> following the well known topic, i want to add libass [15:55] <JEEB> so you had neither --enable-static or --enable-shared on your x264 configure line? [15:55] <undercash> seems interesting [15:55] <undercash> neither yes, but before it wasn't a problem [15:55] <undercash> if i remember well [15:55] <JEEB> that was quite a long time ago then [15:55] <JEEB> not weeks [15:56] <undercash> 2 weeks ago [15:56] <JEEB> no, that wasn't the up-to-date x264 then [15:56] <JEEB> I know for at least a year if not more x264 has needed one of those [15:56] <JEEB> or it would not install the libraries [15:56] <undercash> i guess it was since i had to update yasm too ;) [15:56] <JEEB> no [15:56] <JEEB> yasm update to 1.2 was new [15:56] <JEEB> that was added for the possible addition of haswell asm [15:56] <undercash> anyway thx you i got to recompile x264 [15:57] <undercash> not really useful to debate if it s a recent update or not.. [15:57] <JEEB> you might as well do --disable-cli too [15:57] <XATRIX> Guys, i need some advices... I'm trying to catenate a few mp3 files and, result file is not readable on some players... I'm interested in JS players, and chrome player. Can someone help me to debug ? [15:57] <JEEB> because you only need the library for ffmpeg [15:57] <JEEB> not the command line encoder [15:58] <undercash> i came here coz i thought maybe something was broken in the ffmpeg git [15:59] <undercash> ok JEEB thx, i m recompiling [15:59] <undercash> yes i never use x264 as standalone [16:00] <hackeron> My input audio is pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s from RTSP - any ideas why libfaac would generate broken, metalic sounding audio while libmp3lame sounds fine? - Anything I can do in ffmpeg like maybe force a higher sample rate for the output audio? [16:02] <Mavrik> hackeron: it's quite possible libfaac doesn't support that sample rate [16:02] <Mavrik> hackeron: you can try using libfdk_aac, which is pretty much currently the best ffmpeg supported AAC encoder [16:02] <Mavrik> hackeron: or try resampling the audio to 22050 or some such [16:04] <JEEB> undercash, http://git.videolan.org/?p=x264.git;a=commit;h=c1e60b9032196d204db8dce77051… [16:04] <JEEB> it was may 2011 [16:04] <JEEB> where it changed [16:04] <JEEB> everything from this commit onwards needed --enable-static or --enable-shared [16:06] <hackeron> Mavrik: I tried: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -ar 44100 -codec:v copy -codec:a libfdk_aac -async 1 -flags +qscale -global_quality 1 -afterburner 1 test.mkv -- now it sounds OK but it is out of sync with the video [16:06] <hackeron> Mavrik: it stays in sync if I use libmp3lame though :/ [16:07] <Mavrik> *grumbl* :) [16:07] <Mavrik> hackeron: try without resampling when u sing fdk_aac [16:07] <hackeron> Mavrik: then it dies saying: [libfdk_aac @ 0x7f8ce88b2c00] Unable to encode frame: Encoding error [16:07] <hackeron> Audio encoding failed (avcodec_encode_audio2) [16:08] <undercash> did the job, thx jeeb [16:11] <JEEB> np [16:14] <antonello_> Can I obtain an decoder without reading the file header? [16:14] <hackeron> Mavrik: any ideas? [16:14] <Mavrik> not really, shouldn't go out of sync :\ [16:15] <Mavrik> does your input get probed properly (your analyzeduration is awfully short?) [16:15] <Mavrik> antonello_: yep [16:16] <hackeron> Mavrik: hmm, not sure - but changing libfdk_aac to libmp3lame in the above command I get audio/video sync [16:16] <hackeron> and changing the libfaac I get broken metalic sound [16:17] <Mavrik> antonello_: just call avcodec_find_encoder with right encoder ID [16:18] <antonello_> i do this ... [16:19] <Mavrik> good. [16:20] <antonello_> I obtain a AVCodecContext ... and after i setting the codec parameters [16:21] <antonello_> channels sample_rate bit_rate sample_fmt channel_layoyt [16:21] <antonello_> my error is [16:21] <antonello_> [aac @ 0x10bd800] Not evaluating a further program_config_element as this construct is dubious at best. [16:21] <antonello_> [aac @ 0x10bd800] channel element 1.7 is not allocated [16:22] <antonello_> I don't understand where i miss . [16:23] <hackeron> Mavrik: same story with libvo_aacenc - it seems to be about 1.5 seconds out of sync just like libfdk_aac - but perfect sync with libmp3lame. Does ffmpeg calculate duration different with libmp3lame? [16:23] <Mavrik> hackeron: hmm... wouldn't know, but duration has little to do with it :\ [16:24] <antonello_> this is my code http://pastebin.com/iUGmCq6J line 184 -190 [16:25] <Mavrik> antonello_: um [16:26] <hackeron> Mavrik: any ideas why libmp3lame would be in sync and libfdk_aac and libvo_aacenc audio is around 1.5 seconds behind the video? [16:26] <Mavrik> antonello_: AVFormatContext won't have its fields filled out if you don't read the file header [16:26] <Mavrik> antonello_: which also means your ids and other data will be wrong [16:26] <Mavrik> hackeron: no idea really... that shouldn't happen :\ [16:26] <Mavrik> hackeron: did you try using -async 22050 or something like that [16:28] <antonello_> then.... what should I use? [16:29] <Mavrik> either probe the file to fill out those fields [16:30] <Mavrik> or don't read them at all [16:30] <Macey> Hi All, who looks after the API? [16:31] <Mavrik> O.o [16:31] <hackeron> Mavrik: just tried: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -ar 44100 -async 44100 -codec:v copy -codec:a libfdk_aac -flags +qscale -global_quality 1 -afterburner 1 -y test.mkv -- no change, audio is the same amount behind video - libmp3lame still in sync :/ [16:34] <Diogo> hi i need to capture sound from a blackmagic card using ffmpeg, i don't have sound input...anyone with the same problem? [16:46] <antonello_> Can I obtain an header to add an head of my streeam ? [16:49] <hackeron> I'm reading audio/video from an RTSP source. I have -ar 44100 -async 1 and -flags +qscale - when I use -codec:a libmp3lame - audio and video are in sync, however when I switch to libfdk_aac or libvo_aacenc the video is about a second ahead of the audio. Any ideas? [16:53] <hackeron> ah wait, I tried to create an mp4 file and now the libmp3lame is out of sync by the same amount, hmm [17:06] <hackeron> ah, I had to move -async 1 to be after -codec:a libmp3lame - then I get audio/video sync with libmp3lame - but not with libfdk_aac or libvo_aacenc -- is -async 1 not supported with those? [17:30] <Youka> I'm a bit confused about avcodec_decode_video2 and the auto allocated bitmap inside AVFrame. In the examples, AVFrame bitmap is never freed after decoding, so is it a memory leak (which is ignored because it's small and the program exits after it)? [17:30] <teratorn> Youka: run some tests under valgrind [17:31] <jeje34> hi to all ;-) [17:33] <jeje34> I'm using the last release version of FFMPEG (1.1.1) and it seems the function av_lockmgr_addref and av_lockmgr_release doesn't exist anymore. Before, I was using AVCodec 5.2, and these functions were existing [17:35] <jeje34> sorry I was disconnected [17:35] <jeje34> I use ffmpeg to decode h264 video in a multithread appication [17:40] <jeje34> and in time, I've the error : Insufficient thread locking around avcodec_open/close() whith a crash of my application [17:43] <jeje34> when I was using avcodec52.dll, i call if (av_lockmgr_addref() == 1) [17:43] <jeje34> { av_lockmgr_register(&ff_lockmgr); [17:43] <jeje34> } [17:44] <jeje34> with static int ff_lockmgr(void **mutex, enum AVLockOp op) and casting mutex to CRITICAL_SECTION (I'm on windows envirronment) [17:45] <jeje34> but with the last version, the av_lockmgr_addref doesn't exist [18:03] <jeje34> what happen if several thread call av_lockmgr_register(&ff_lockmgr) [18:03] <jeje34> ? [19:19] <JoeyJoeJo> I have an mkv file that won't play and I'm not sure why. Can ffmpeg repair the file or at least let me know what the problem is? [19:20] <klaxa> probably, try running it through ffprobe [19:20] <klaxa> also some more information on the exact way you can't play it would be great [19:20] <klaxa> i.e. what player and what error message [19:21] <JoeyJoeJo> ffprobe seemed to work. It said "Invalid data found when processing input" [19:21] <klaxa> if you torrented the file, verify your local data, maybe the header is corrupted, i have similar problems from time to time [19:22] <klaxa> mplayer2/libavformat interprets the file as DV Video and starts playing back garbage [19:22] <JoeyJoeJo> I'm trying to play it with VLC and I don't see any picture or hear any sound. The seek bar doesn't even count. And I can't find any errors either [19:23] <JoeyJoeJo> I did torrent it. so I try verifying my local data [21:10] <llogan> burek: perhaps the IMPORTANT note on gusari forum should be even bigger. [21:26] <bparker> is there a way to embed exact pixel colors at certain positions within a compressed video? For example I have a compressed video and I want the first pixel of the video to be 100% blue. It doesn't matter if the video has to be re-encoded, that's fine. Any idea if that's possible? [21:27] <bparker> I'm worried about the compression messing with the color values themselves [21:28] <klaxa> depends on the codec i guess, but yeah compression will probably mess with that stuff [21:29] <klaxa> lossly compression that is [21:35] <bparker> klaxa: right, that's what I'm trying to work with unfortunately [21:35] <bparker> I need some markers placed on each corner of a compressed video [21:35] <bparker> with specific pixel values because I'm essentially encoding auxillary data there [21:36] <klaxa> can't you work with approximate values? [21:36] <bparker> sortof like steganography, but really small bits of data [21:36] <bparker> it's possible, but I'm worried about false positives [21:36] <bparker> if the video happens to contain a color in the range I'm looking for, that's bad [21:37] <klaxa> mh yeah [21:37] <bparker> and I don't want to visibly alter the video enough that it's distracting to users [21:37] <bparker> becaue the displays can see 100% of the pixels [21:37] <bparker> what I'm basically doing is [21:38] <bparker> taking a video signal in from a device, capturing it and doing image analysis on it [21:38] <bparker> then based on that analysis, doing different kinds of post-processing on the image and outputting it to a TV [21:38] <bparker> analysis being some bits I want set that tell me what to do with the video [21:39] <bparker> and the only way I have to communicate that is with the actual video data (pixels) itself [21:40] <klaxa> i don't think you'll be able to achieve that unless you use lossless codecs or settle with approximation [21:40] <klaxa> or establish a second independent data channel of some sort [21:40] <bparker> that's what I thought [21:41] <bparker> unless I can come up with some kind of scheme/algorithm that is resistant to the compression, that's really my only option [21:42] <klaxa> and you have to use the videostream? why not try to code it in binary at the top or bottom and analyze that within some tolerance range? [21:42] <klaxa> like black and white pixels [21:42] <klaxa> you should be able to differenciate black and white even when compressed [21:43] <klaxa> although it may put strain on the codec because hard edges are hard to compress [00:00] --- Fri Feb 1 2013

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