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January 2013

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ffmpeg-devel.log.20130104
by burek 05 Jan '13

05 Jan '13

[00:09] <llogan> saste: thanks for pushing my old-ass patch. i forgot about that one (until that bug report). [01:21] <llogan> so many people keep the "Patches should be submitted..." message in their bug reports. [02:45] <highgod> Hi, when I run fate,the output print Test eval failed." Look at tests/data/fate/eval.err for details."And the teh eval.err file shows that [Eval @ 0028fe58] Undefined constant or missing '(' in '' [02:45] <highgod> [Eval @ 0028fe58] Invalid chars 'gi' at the end of expression '1gi' [02:45] <highgod> [Eval @ 0028fe58] Invalid chars 'Foo' at the end of expression '1GiFoo' [02:45] <highgod> [Eval @ 0028fe58] Invalid chars 'oo' at the end of expression '1Gi*3foo' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in 'foo' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in '' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in ')' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in 'foo)' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in 'sin' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in '' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in ')' [02:45] <highgod> [Eval @ 0028fe58] Undefined constant or missing '(' in 'sin)' [02:45] <highgod> How can I correct it ?thanks [03:04] <weixuan> hi,wei [03:38] <Compn> highgod : try rm the file and get fresh copy from git [03:38] <Compn> i'm guessing your file got windows line endings again, maybe you saved it with notepad program! :P [03:42] <weixuan> hi, we are tring to do the fate for our patch. when i ran the command "make fate", it reported this error:Test eval failed. Look at tests/data/fate/eval.err for details. [03:42] <weixuan> make: *** [fate-eval] Error 126 [03:42] <weixuan> ./tests/fate-run.sh: line 70: /home/wwx/ffmpeg_patch8/ffmpeg_correct_merge1/libavutil/eval-test: Bad file number [03:43] <weixuan> who can tell me why i got this error? [03:44] <weixuan> thank you very much [03:46] <highgod> Thanks Compn,you are right,I get the fate from mingw32 command window,I will use git to reget it, thanks [03:55] <ubitux> weixuan: then you broke something [03:55] <ubitux> make fate-eval V=1 [03:55] <ubitux> and you'll see the command used [03:55] <ubitux> so you can reproduce [03:58] <ubitux> highgod: there error are normal, the eval test is doing some error testing in the eval api [03:58] <ubitux> the test should not fail though [03:58] <ubitux> also, please use pastebin when you have more than 3 lines to paste [04:04] <highgod> OK,I will try it thanks ubitux [04:04] <ubitux> don't try, do [04:04] <ubitux> ;) [04:06] <weixuan> thank you, ubitux [04:06] <highgod> Yes,I will do it.I mean that,my English is not good,hehe [04:07] <ubitux> :) [04:59] <highgod> Hi,ubitux,I run the command line "make fate-eval V=1",still the same error.And I want to ask that if I just add a test case for h264_dxva2,can I just run fate-h264? Thanks [05:02] <ubitux> V=1 will show you the test run by fate [05:02] <ubitux> so you can see the command line [05:02] <ubitux> then you can reproduce that particular test and debug what's going on [05:03] <ubitux> adding a test for your stuff should not break the rest [05:03] <ubitux> if it does, then something is wrong with your patch [05:03] <ubitux> and you need to debug [05:04] <highgod> OK,Thanks [05:09] <cone-2> ffmpeg.git 03Michael Niedermayer 07master:7b5fdd04de66: locodec: flip RGBA [06:21] <cone-2> ffmpeg.git 03Michael Niedermayer 07master:4e0738cec9df: mpegaudiodec/mp3on4: fix buffer size. [06:21] <cone-2> ffmpeg.git 03Michael Niedermayer 07master:b888cea9cb2a: ac3dec: split out pointer update loop for saftey [06:32] <ubitux> anyone has a real world sample of utf16 subtitle? [06:32] <ubitux> (one or more actually) [06:33] <highgod> Hi,ubitux,we didn't add our fate test yet,we just run the fate suit download from ffmpeg.org use "make fate SAMPLES=fate-suite/",and the eval failed.Should I have to debug the eval fate? [06:34] <ubitux> i believe so [06:35] <ubitux> with a fresh checkout of ffmpeg, a make fate SAMPLES=fate-suite should work [06:35] <ubitux> if you add modifications, it should still work [06:54] <highgod> ubitux,the eval output error is "Undefined constant or missing '(' in 'foo' ",I review the code,it is the function "parse_primary" error,but I didn't change code yet,and we didn't add our fate test.Is it the error of eval? [07:20] <ubitux> as i already said [07:20] <ubitux> these errors are fine [07:21] <ubitux> ./libavutil/eval-test > /dev/null [07:21] <ubitux> you'll see various eval error [07:21] <ubitux> they are used to test eval [07:21] <ubitux> OTOH, if the test fails [07:21] <ubitux> then something is wrong [10:12] <cone-513> ffmpeg.git 03Benjamin Kerensa 07master:95016fd1c8e1: Fix id3v1 tag spelling. [10:34] <cone-513> ffmpeg.git 03Carl Eugen Hoyos 07master:2284448775f0: Revert "Fix id3v1 tag spelling." [10:34] <cone-513> ffmpeg.git 03Carl Eugen Hoyos 07master:155cdc1d05e9: Add a comment about an intentional misspelling to the id3v1 tags. [11:45] <saste> do we have a test for yadif? [11:45] <saste> i can't find it... [11:46] <ubitux> in tests/fate/filter.mak [11:47] <saste> ubitux, why avfilter.mak and filter.mak? [11:48] <ubitux> no idea [11:48] <ubitux> avfilter may be related to the lavfi-regressions [11:48] <ubitux> while filter may be independant [11:52] <cone-513> ffmpeg.git 03Stefano Sabatini 07master:b52c1d0c993a: lavfi/yadif: remove unused poll_frame callback [11:52] <cone-513> ffmpeg.git 03Stefano Sabatini 07master:4ea7c179325f: lavfi/yadif: fail during the configuration stage in case of invalid video size [11:52] <cone-513> ffmpeg.git 03Stefano Sabatini 07master:8674597fe531: lavfi/yadif: remove redundant NULL checks in uninit [11:52] <cone-513> ffmpeg.git 03Stefano Sabatini 07master:f7dc6aa6b194: lavfi/yadif: add support to named options and options introspection [11:52] <cone-513> ffmpeg.git 03Stefano Sabatini 07master:cb8d3965fd7e: lavfi/yadif: add support to named constants [12:04] <mateo`> hi there ! how/where are the id3 metadata transmitted from a demuxer to a muxer (typically mp3 -> mp3) ? thanks in advance ! [12:10] <saste> ubitux, yesterday i tried to add an offset option to subtitles, and it wasn't working [12:10] <saste> libass was not recognizing events if i added an offset to time_ms [12:12] <saste> so the question is, is it possible to specify an offset to libass? [12:28] <J_Darnley> mateo`: metadata is read by the demuxer and put into a generic structure for any muxer to then write [12:30] <J_Darnley> For ID3 I would suggest looking at libavformat/id3*.c [12:31] <mateo`> J_Darnley: ok, i'm looking for the code which allocate/affect the MP3Context->id3 which is used in mp3_write_header. [12:33] <J_Darnley> Since I don't know I can't give you an answer now. But I'm sure I can the same searching as you to find out where that comes from. [12:43] <J_Darnley> ID3v2EncContext (the id3 element of that stuct) is defined in id3v2.h. [12:43] <J_Darnley> It will be allocated whenever the MP3Context is allocated. [12:44] <J_Darnley> It will be used by the id3 functions in the write header function. [12:45] <J_Darnley> It gets the metadata through the AVFormatContext pointer [12:45] <J_Darnley> With this last it I really mean the id3 functions [12:47] <ubitux> saste: sorry [12:47] <ubitux> i was away [12:47] <ubitux> saste: what is this offset option you are talking about? [12:49] <ubitux> you want to offset seek? [12:49] <saste> ubitux, no suppose there is a delay in the subtitles with respect to the video [12:50] <saste> you want to fix the subtitles delay by adding an offset to the timestamps [12:50] <ubitux> oh, adding, opposite to a -ss? [12:50] <saste> i tried to change the way time_ms is computed in subtitles, but that doesn't work [12:51] <ubitux> it should work, except when remuxing ass :) [12:51] <saste> i want to shift back/forward in time subtitles overlay [12:51] <saste> ubitux, but no, it doesn't [12:51] <saste> something libass related [12:53] <ubitux> i still don't understand what exactly you did :p [12:55] <saste> ubitux, will send a patch/rfc [12:55] <ubitux> okay :) [12:55] <ubitux> note that ffmpeg -ss 12 -i in.srt -c copy out.srt should do a timing shift [12:56] <ubitux> btw, speaking of this [12:56] <ubitux> it could be nice to have a setpts filter not requiring a re-encode :p [12:59] <saste> muxer-level packet filter? [12:59] <saste> ubitux, i guess so, remuxing the subtitles file may be a suboptimal solution [13:00] <ubitux> yes, muxer level filter [13:01] <saste> ubitux: but that command *should not* do a time shifting [13:01] <saste> it should just cut the first 12 seconds [13:02] <ubitux> mmh [13:02] <ubitux> arg that's something to fix then :$ [13:04] <mateo`> J_Darnley: thx [13:04] <saste> so basically there is no way to add an offset to subtitles, unless we can filter them [13:04] <saste> ubitux, do you have some specific plan with regards to subtitles filtering? [13:05] <ubitux> saste: i'd like to have it [13:05] <ubitux> i don't plan to work on it personally [13:06] <saste> but *you* are the Lord of The Subs! [13:06] <ubitux> yes but i hate them [13:06] <ubitux> i'm not a good lord [13:07] <mateo`> saste: hahaha :) [13:08] <ubitux> :( [13:11] <cone-513> ffmpeg.git 03Clément BSsch 07master:3048fae63c99: build: Avoid detecting bogus components named 'x' [13:11] <cone-513> ffmpeg.git 03Martin Storsjö 07master:a0b7e289075d: aviobuf: Partial support for reading in read/write contexts [13:11] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:d0b450457b3a: matroskadec: fix ffio_init_context() usage [13:11] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:e1cf1a9c89db: Merge commit 'a0b7e289075dccf223b7f407790d8a86fc5d77e8' [13:22] <cone-513> ffmpeg.git 03Martin Storsjö 07master:3f95f0dda55f: rtpdec: Move the URLContext used for RTCP RR out from the context, to a parameter [13:22] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:ea96feddb7bb: Merge commit '3f95f0dda55fca74b646937095a02a8fa9776622' [13:28] <cone-513> ffmpeg.git 03Martin Storsjö 07master:e96406eda4f1: rtsp: Add support for depacketizing RTP data via custom IO [13:28] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:8d0b2aae71f0: Merge commit 'e96406eda4f143f101bd44372f7b2d542183000a' [13:30] <saste> ubitux: more comments on kerndeint? [13:31] <ubitux> not anymor [13:31] <ubitux> +e [13:33] <cone-513> ffmpeg.git 03Martin Storsjö 07master:53c25ee07364: aviobuf: Discard old buffered, previously read data in ffio_read_partial [13:33] <cone-513> ffmpeg.git 03Peter Meerwald 07master:be6cde3ce86e: lavr: fix missing " in header documentation [13:33] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:a08194b4c51e: Merge remote-tracking branch 'qatar/master' [14:44] <kierank> nevcairiel: whywhywhy does carl want to change h264 sar like that... [14:48] <nevcairiel> i knew it was wrong when i saw it, but i double checked the spec anyway just in case... :p [14:52] <nevcairiel> whats with those crazy avc intra files anyway, they come with no SPS of their own? What crazy person would develop such a system [14:53] <JEEBsv> crazy people who want to make sure only their shit can decode those files [14:53] <JEEBsv> same people who make a "spec" for "Pro AVC Intra" yet everyone breaks it [14:54] <JEEBsv> like we found out in the end, practically what the MainConcept/whatever encoder/decoder combo was doing was the "spec" in the end methinks [15:24] <kierank> 13:53:40 <JEEBsv> crazy people who want to make sure only their shit can decode those files [15:24] <kierank> this [15:24] <kierank> a proprietary open standard [15:33] <saste> what's going on on multimedia.cx? [15:33] <saste> i can't find track of "vandalized" pages [15:53] <ubitux> saste: histeq ready for review? [15:54] <saste> ubitux, yes [16:01] <ubitux> oh i just noticed the "RFC" [16:01] <ubitux> erf i'm tired [16:08] <wm4> ubitux: I love that ffmpeg is going to get teletext decoding [16:09] <wm4> ubitux: I haven't fully looked at the patch that was sent, but it looks like this will be exported as text? [16:09] <ubitux> what patch are you talking about? [16:09] <ubitux> libzvbi patch from Wolfram? [16:09] <wm4> yes [16:09] <ubitux> i didn't really look [16:15] <saste> ubitux: the subtitles patch was an RFC [16:15] <saste> the problem is related to libass, as i told you [16:16] <ubitux> yeah i remember reading it [16:16] <ubitux> sorry i'm a bit slow today [16:16] <saste> maybe it expects times starting from 0 or something like that [16:16] <saste> so i can't say what's wrong, if not looking at the libass code [16:18] <ubitux> fun [16:18] <ubitux> though, it's strange [16:18] <wm4> uh what is the problem? [16:18] <ubitux> are you sure you're not simply wrong with the timing? [16:19] <ubitux> (like passing pts in minutes instead of seconds dunno) [16:19] <ubitux> afaik players are able to make it work when seeking from the start [16:22] <saste> ubitux, that's why i increased the log level [16:23] <wm4> link to the issue? [16:24] <ubitux> wm4: http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/136898.html [16:27] <wm4> is ass->offset in milliseconds or microseconds? [16:28] <wm4> you really should have a time option type... [16:29] <wm4> since vf_ass, in my understanding, always demuxes the subtitles completely on start, there should be no issue [16:46] <ubitux> 16:27:53 < wm4> is ass->offset in milliseconds or microseconds? [16:46] <ubitux> 16:28:18 < wm4> you really should have a time option type... [16:46] <ubitux> two of my comments ;) [16:47] <wm4> yeah, I read that [16:47] <wm4> sorry for the redundancy [16:47] <ubitux> vf_ass doesn't demux anything [16:47] <ubitux> it just paste the .ass file into libass [16:47] <ubitux> vf_subtitles does a demux to reconstruct a fake .ass basically [16:47] <ubitux> but no ass filter [16:47] <ubitux> since it's already the good markup :p [16:48] <cone-513> ffmpeg.git 03Alexander Strasser 07master:ac25b31ede03: lswr: Improve default resampler's default parameters [16:48] <wm4> does vf_subtitles handle image subs too? [16:48] <ubitux> nope [16:48] <ubitux> we need a public blending api :'( [16:48] <ubitux> so we could put the ffplay bitmap blending code into it [16:49] <ubitux> factorize the overlay, drawtext, ass and subtitles blending [16:49] <ubitux> and support muxed subtitles blending in ffplay :( [16:49] <wm4> libswblend? [16:50] <ubitux> would be nice [16:50] <ubitux> i have a patch for ffplay to support the muxed subtitles [16:50] <ubitux> but it uses the private api of lavfi [16:50] <ubitux> so it can't be compiled :( [16:50] <ubitux> (in shared) [16:50] <ubitux> it's too bad, because i could finally watch my animes with ffplay! [16:58] <ubitux> wm4: you going to write it? [16:58] <wm4> what [16:58] <ubitux> libswblend [16:58] <ubitux> want to write it? [16:59] <wm4> probably not [16:59] <ubitux> :( [16:59] <wm4> I'm not one to squish out large amounts of asm, which is probably what'd be expected here [16:59] <ubitux> a first C version would be nice [17:00] <ubitux> i would be happy to do it, but hell i have thousands of stuff to do before that :( [17:01] <wm4> in mpv, we have some basic blending, that relies on swscale to bring everything into a common format [17:05] <michaelni> wm4, first bringing things into a common format is probably the correct approach [17:05] <michaelni> especially when adding the same subtitle onto several frames [17:06] <michaelni> only 1 convert [17:10] <wm4> michaelni: I mean what I'm doing is converting the regions covered by subs to 444 (essentially upsampling), blend, and convert back [17:13] <nevcairiel> personally i subsample the overlay in such a case [17:16] <wm4> nevcairiel: this was done with libass in mind, where multiple bitmaps are blended over the target image [17:16] <wm4> so subsampling would lead to quality reduction [17:17] <wm4> and converting the subs to RGBA first would probably be slower (or not, didn't really check) [17:17] <wm4> err, not RGBA, but YUVA [17:17] <wm4> in the common case, at least [17:18] <durandal_1707> wm4: what is this about? [17:18] <wm4> durandal_1707: subtitle rendering [17:19] <wm4> nevcairiel: though, how do you deal with alpha? is it in luma and chroma resolution? [17:19] <ubitux> durandal_1707: http://pastie.org/private/vlntkqvfbchhpk6eo0xj2a [17:19] <nevcairiel> wm4: luma, and for chroma blending i average it [17:20] <nevcairiel> a lot like vf_overlay works in fact [17:22] <wm4> I guess that has less operations per pixel than the other way around [17:31] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:c98d3056cf02: msrle: fix small palette handling [17:38] <durandal_1707> is there >8 channel sample? [17:47] <Compn> like 16 channel sample ? [17:47] <durandal_1707> there is one? [17:47] <Compn> humm [17:48] <Compn> in what format ? [17:48] <Compn> music producers may have such things but otherwise ... [17:48] <durandal_1707> irellevant [17:48] <durandal_1707> wav pcm [17:49] <saste> i dislike when people create a new thread for each patch revision :-( [17:50] <Compn> saste : did you write it in developer rules to keep one thread going? :P [17:51] <saste> Compn, there are so many rules that it is impossible to honor them all [17:51] <durandal_1707> ffmpeg cant decode wav with 512 channes [17:51] <Compn> :D [17:51] <Compn> wow [17:51] <saste> durandal_1707, how did you create the sample? [17:52] <durandal_1707> saste: people are lazy [17:52] <durandal_1707> saste: i found it on web [17:52] <saste> a distinct channel for each voice/instrument in a symphonic orchestra? [17:53] <durandal_1707> i get error: [graph 0 input from stream 0:0 @ 0x29c29200] Invalid sample format '(null)' [17:54] <saste> uh-oh [17:54] <wm4> audiophiles will kill you [17:54] <saste> sure reporting sucks hard [17:54] <saste> there is a missing check somewhere [17:55] <wm4> (lol 512 channels) [17:55] <durandal_1707> this is synthetic one [17:55] <saste> it would be useful to support that [17:57] <wm4> useful? [17:57] <saste> yes in the italian parliament for example [17:57] <saste> everyone speaks and no one listen the others [17:58] <wm4> for that'd you'd use different audio streams [17:58] <wm4> there is no speaker setup called "italian parliament" [17:59] <wm4> anyway, enjoy your API that usually breaks with >8 channels ;D [17:59] <saste> wm4: the <= 8 limitation is supposed to be overcome [17:59] <wm4> by adding a new special case [18:04] <durandal_1707> the only limitation is for planar sample formats [18:04] <durandal_1707> i see no reason why interleaved could not be supported [18:08] <saste> durandal_1707, why is planar limited? [18:08] <saste> you have no limitation on the number of planes you can alloc [18:08] <saste> of course code may be bugged/rely on the old assumption [18:09] <saste> but no design limitation in libavcodec/libavfilter [18:10] <durandal_1707> for >8 one need to use extended_data [18:10] <JEEB> I thought the assumption was that everyone would use extended_data [18:10] <JEEB> for audio [18:10] <wm4> and if it doesn't, it just blows up randomly? [18:10] <durandal_1707> wm4: nope, because nothing supports >8 [18:10] <wm4> or is there something that makes sure it doesn't [18:11] <wm4> ffun [18:11] <durandal_1707> by nothing i mean ffmpeg/lavfi [18:12] <wm4> you could just always force code to use extended_data by not setting data [18:12] <durandal_1707> looks like extended_data was added for fun [18:13] <durandal_1707> l releasing 96 [18:13] <JEEB> more like it was added because data had a limit in how many things could be put there :P Not that I fully agree with how it was handled, but this way things basing upon data wouldn't break until they tried using >8 channels [18:14] <durandal_1707> michaelni: fine to push l patches that are stuck in limbo for eternity? [18:25] <saste> durandal_1707, alsdec: fix typo <- push it directly [18:28] <durandal_1707> saste: no I wait Thilo explicit approval he is als maintainer and prefer to have full control, i cant say him f* off [18:28] <saste> durandal_1707, ok [18:29] <durandal_1707> otherwise i get complaints in cvs log ml that i'm not subscribed and people get angry when i ignore them [18:34] <michaelni> durandal11707, which l patches ? [18:44] Action: Compn wonders what wiki changes [18:53] <cone-513> ffmpeg.git 03Maximilian Seesslen 07master:467c033858a8: fixed granularity of video quality when encoding with theora codec [20:09] <durandal_1707> ubitux: unkown is typo in patcheck typos? [20:29] <cone-513> ffmpeg.git 03Paul B Mahol 07master:d885cc41e526: Fix "knwon" typo and add a check in tools/patcheck [20:56] <cone-513> ffmpeg.git 03Justin Ruggles 07master:f2214c622458: au: use ff_raw_write_packet() [21:04] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:84aba8eed97d: mpegpsenc: restructure SCR handling [21:04] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:2a23f6035ec4: mpegpsenc: Fix SCR handling for DVD [21:04] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:9fd0cf8a3b9e: mpegpsenc: move preload recalculation to where its needed [21:04] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:cf369d444985: mpegpsenc: show first SCR/DTS at debug level [21:04] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:fa11f368764f: mpegpsenc: avoid shifting dts/pts [21:04] <durandal11707> michaelni: if lavf/mxfenc use tables from lavc they should be renamed from ff_ to av_ [21:06] <michaelni> yes or avpriv [21:06] <iive> are these tables to be used by applications? [21:08] <michaelni> durandal11707, which table is it that has a wrong prefix ? [21:08] <durandal11707> ff_dnxd [21:08] <durandal11707> ff_dnxhd [21:09] <durandal11707> 2 tables, mentioned in .v file [21:15] <cone-513> ffmpeg.git 03Paul B Mahol 07master:ddeb299234cc: lavc: remove img_get_alpha_info as it not available any more [21:20] <michaelni> durandal11707, Tjoppen patch to fix ff_dnxhd in mxfenc posted [21:25] <durandal11707> add commit that removes those from .v on top of that set [21:33] <michaelni> removing them from v is a ABI break [21:34] <durandal11707> was img_get_alpha_info removed with ABI break? [21:37] <michaelni> that symbol seems not there anymore so nothing should break [21:37] <michaelni> that wasnt already broken [21:38] <durandal11707> yes, i ask did it got broken .... [21:38] <nevcairiel> if such a function is not part of the public ABI, is it really a break? [21:39] <nevcairiel> i mean, noone should be accessing ff_* symbols [21:39] <durandal11707> it is named img_ [21:39] <nevcairiel> even worse :p [22:13] <durandal11707> anyone run ffmpeg git tree under gource ? [22:58] <cone-513> ffmpeg.git 03Michael Niedermayer 07master:498e1c6bb963: lavu: check av_clip*() limits [00:00] --- Sat Jan 5 2013

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ffmpeg.log.20130104
by burek 05 Jan '13

05 Jan '13

[00:03] <seablade_mbp> Ok so I gave up for the time being on skipping the file, I am recording to the file and trying to come up with a good command line for ffmpeg to stream in realtime a MPEG2 TS h.264/aac file so that I can preview it in VLC for now(Getting it to my server will come later) [00:03] <seablade_mbp> What I have right now is `ffmpeg -i atem.ts -re -acodec libfaac -ab 192k -ar 44100 -ac 2 -strict experimental -vcodec libx264 -vb 2000k -f flv udp://127.0.0.1:58440` [00:03] <seablade_mbp> While it seems to be working, I can't seem to open it in VLC to confirm it is working [00:04] <seablade_mbp> ideally I would like to just copy that video and audio encoding rather than reencode, but for some reason just copying didn't seem to work [00:05] <seablade_mbp> More correctly running with -vcodec copy and -acodec copy seems to try to do it as fast as it can, even if -re is given [00:05] <seablade_mbp> Obviously that doesn't work so well when I need it delivered in realtime [00:06] <seablade_mbp> Moving the -re to be the first thing on the command line causes ffmpeg to lock up here as well, so I don't think that will work to well:) [00:10] <BtbN> ffmpeg -re -i test.mkv test.avi works fine here. [00:10] <BtbN> with -re it limits to the real framerate, without it uses as much CPU power as possible [00:12] <seablade_mbp> BtbN, Sadly it doesn't seem to be working here. Either ffmpeg locks up (-re located before the input file) or it moves as fast as possible(Located after the input file) git version from September [00:13] <BtbN> it has to be before the input to work. [00:13] <seablade_mbp> BtbN, I should note that I am trying to encode to a stream, not to a file [00:13] <BtbN> that's the same [00:13] <seablade_mbp> BtbN, I thought I remembered that yes, but see my comment about it being before:) [00:13] <BtbN> you're doing something else wrong then if it locks up with -re [00:14] <seablade_mbp> BtbN, Doesn't seem to be anything different, lemme grab the command I am using... [00:14] <seablade_mbp> BtbN, ffmpeg -re -i atem.ts -acodec copy -vcodec copy -f flv udp://127.0.0.1:58440 [00:15] <BtbN> you're sure flv supports the codecs you want to copy? [00:15] <seablade_mbp> BtbN, h.264 and aac, yes [00:16] <seablade_mbp> BtbN, Standard codecs I have used for other purposes, wouldn't have any problem switching off fly but haven't yet only because I know it works [07:31] Action: Boon aww [08:15] <macmichael01> Whenever I try muxing a video that contains Dolby TrueHD to the mp4 container I seem to get an error. Is it possible to mux Dolby TrueHD to mp4 container? The spec claims to support it: http://mp4ra.org/filetype.html. [08:50] <thalsharif> hello there, I have question about ffmpeg, so can someone help on that ? [08:52] <JEEB> macmichael01, basically stuff listed by the mp4ra means that someone wrote a specification for it, and registered the name for the thing... Too bad Dolby be Dolby [08:52] <JEEB> I guess the specification isn't exactly easily available [08:53] <JEEB> lemme see if L-SMASH has implemented it [08:55] <JEEB> yeah... nope [08:55] <thalsharif> My question is: I receive from local port video stream which is UDP packet that contains only RGB data, no header no specific protocol used, just only RGB 24bit over UDP packet. My goal is to receive and play my stream on PC using ffmpeg with vlc or ffplay, anybody can guide me how to do that. Thanks [08:56] <JEEB> there was a DTS specification that was somehow gotten by certain people, so that's why L-SMASH could implement DTS-in-"MP4" [08:56] <JEEB> macmichael01, find us the specification and you'll get your support, basically \o/ [09:02] <thalsharif> anybody can help me on my question. [09:02] <thalsharif> hello [09:02] <macmichael01> JEEB: I saw a google summer of code project which looked like they were developing something related to this. http://bit.ly/Us7BEb (MLP/TrueHD encoder) To bad the links are dead :( [09:08] <macmichael01> JEEB: I know MKV supports TrueHD. How hard would it be to translate their specification to work within MP4? [12:10] <thalsharif> hello everyone, I have question and I relley need some help. [12:10] <thalsharif> My Problem is: I receive from local port video stream which is UDP packet that contains only RGB data, no header no specific protocol used, just only RGB 24bit over UDP packet. My goal is to receive and play my stream on PC using ffmpeg with vlc or ffplay, anybody can guide me how to do that. [12:11] <saste> thalsharif, nothing special, it should be like reading from input, but you need to specify -pix_fmt rgb24 and size since there is no header information [12:13] <thalsharif> i am working on windows, can you give command line to do that. [12:13] <saste> thalsharif, what command are you using? [12:15] <thalsharif> i am begginer with ffmpeg but i try to construct the command. ffmpeg -i udp://ip:port, then i don't know how to complete this command [12:15] <thalsharif> upd://ip : port [12:16] <saste> thalsharif, ffmpeg -pix_fmt rgb24 -s SIZE -r RATE -i udp://ip:port OUTPUT [12:16] <saste> check the wiki/documentation for more info [12:18] <thalsharif> ok i will check that. but as you said it's possible to do that with ffmpeg. [12:19] <thalsharif> i just want to make sure ffmpeg has capabilty to do my work. [12:20] <saste> thalsharif, it should, but checking is better [12:21] <thalsharif> saste, thanks for your help :) [12:22] <saste> thalsharif, you should also specify the input codec, ffmpeg -codec:v rawvideo -pix_fmt rgb24 -s SIZE -r RATE -i udp://ip:port OUTPUT [12:24] <thalsharif> good, and what about playing the stream?? [12:24] <saste> ffplay -codec:v rawvideo -pix_fmt rgb24 -s SIZE -r RATE -i udp://ip:port [12:27] <thalsharif> nice, i can use ffplay without ffmpeg to receive and play stream?? [12:28] <saste> thalsharif, why not? [12:31] <Fjorgynn> 688x288 isn't 16:9 [12:31] <iive> it is recommended that you use some container. udp can loose packets and the raw demuxer won't notice it, leading to image shifting [14:35] <An_Ony_Moose> what's a good intermediate format for videos? I have a recording here of which the resolution is a bit too high and intend to scale it down before further processing - what should I output it as, for little loss of quality while keeping a manageable filesize? [14:39] <lnb> trying to convert a .mov to .flv with winff, the preset is: -vcodec flv -f flv -r 29.97 -s 860x484 -aspect 16:9 -b 300k -g 160 -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 -ar 22050 -ab 56k [14:40] <lnb> when it runs, errors are: requested bitrate is too low. Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height [14:40] <lnb> no flv is made [15:02] <klaxa> An_Ony_Moose: either use a lossless codec like ffv1, huffyuv (might get big) or lossless settings for libx264, or use a high quality libx264 settings [15:04] <lnb> it keeps exiting with bitrate too low [15:04] <klaxa> increase bitrate? [15:04] <lnb> -vcodec flv -f flv -preset veryslow -s 860x484 -aspect 16:9 [15:04] <lnb> thats what i have in preset [15:04] <lnb> the file is 2.2gb [15:05] <klaxa> increase the value after -b mabye? [15:05] <klaxa> *maybe [15:08] <lnb> trying again with -vb 400k [15:09] <lnb> trying to convert .mov to .flv [15:09] <sacarasc> Old version of ffmpeg? [15:09] <klaxa> that might be too [15:10] <lnb> ffmpeg version 0.8.4-4:0.8.4-0ubuntu0.12.04.1 [15:10] <lnb> is that old? [15:10] <klaxa> that probably isn't even ffmpeg :P [15:10] <klaxa> yeah ffmpeg is at version 1.0+ [15:11] <lnb> i typed in shell ffmpeg -version [15:14] <limpc> hi guys. is gps info available in video? [15:15] <klaxa> pretty sure there is [15:15] <klaxa> would be weird if photos had it and videos hadn't [15:17] <lnb> when converting it shows Duration: 00:15:38.02, start: 0.000000, bitrate: 19763 kb/s [15:18] <lnb> does that mean i have to set bitrate to 19763? [15:19] <limpc> well the reason i ask is because unalike photos, video is 'on the move'. so its not like you can just add 1 gps coordinate to the entire video [15:19] <saste> limpc, in general: no [15:20] <limpc> as in no, theres no gps info? [15:20] <saste> in theory you could add metadata per frame, but i'm not aware of any format which supports per-frame metadata [15:20] <saste> geotiff does, but is only for images [15:21] <saste> limpc, as in, in theory possible, but i don't know if some formats make use of it [15:21] <limpc> hmm thats a bummer [15:21] <saste> and for sure we only support per-frame metadata only in the Tiff decoder [15:22] <limpc> we were hoping to make use of gps attached to video files for a project. I came across an interesting idea by one person who encoded gps info into the audio stream [15:22] <limpc> so it had me wondering [15:22] <saste> "in the audio stream"? [15:23] <saste> limpc, how was that? [15:23] <limpc> yeah. let me find the page hd a min [15:23] <saste> limpc: BUT: you can store such info in a DATA stream [15:24] <limpc> http://www.sivann.gr/software/gps2dtmf/ [15:25] <lnb> anyone have a preset line for .mov to .flv so it can be used full screen good quality? [15:26] <saste> limpc, yes in theory you can encode anything in audio data, but is that a good idea? [15:26] <lnb> [flv @ 0x1bd8700] requested bitrate is too low [15:26] <limpc> saste, probably not, it'll probably add an odd noise, but I imagine sometimes options are limited [15:27] <limpc> obviously thats not a usable solution by us. we need something thats already there on mobile phones and modern dslr cameras/videocams. [15:28] <limpc> so i guess I'm SOL :) [15:32] <espr3ss0> hi, me again ;-p [15:32] <espr3ss0> Q: does "no symbols" or, "nothing to do" cause any problems ? [15:33] <espr3ss0> installing "gettext" & "glib" @ the moment for ffmpeg [15:34] <luc4> Hi! I'm using avformat_find_stream_info(formatContext, NULL) to get info on the streams. I tried with an file containing an aac stream, but when I get the duration I get a wrong value. Is this supposed to happen? [15:34] <Diogo> hi this is possible use variable frame rate ?? [15:35] <Diogo> (http://pastebin.com/48qcjqJZ) [15:35] <Diogo> media info said always Constant [15:36] <Mavrik> luc4, that depends on the container [15:36] <Mavrik> luc4, but for some the duration is not reliable :\ [15:37] <luc4> Mavrik: no container, it is a aac stream I extracted from a mov file. [15:37] <Mavrik> yeah, that would explain why probing the start can't give you reliable duration :) [15:37] <Mavrik> duration is usually noted in container metadata [15:38] <luc4> Mavrik: so there is no way to get a duration or maybe that value is modified while decoding? [15:39] <Mavrik> well, you can read the file and calculate duration that way [15:39] <luc4> Mavrik: thanks! [15:39] <Mavrik> IIRC that value doesn't change while decoding [15:55] <espr3ss0> Q: how do u compile "Glib" with flags ? o_O [15:55] <espr3ss0> is there a tutorial ? [15:56] <lnb> is -b = bitrate? [15:56] <lnb> as in -vcodec flv -f flv -s 640x360 -aspect 16:9 -qmin 6 -qmax 11 -b 800k [15:57] <lnb> [scale @ 0x17e2720] w:1920 h:1080 fmt:yuvj420p -> w:640 h:360 fmt:yuv420p flags:0x4 [15:57] <lnb> [flv @ 0x17e1700] [lavc rc] Error: bitrate too low for this video with these parameters. [15:57] <espr3ss0> lnb: "http://forum.videohelp.com/threads/277807-Useful-FFmpeg-Syntax-Examples" [15:59] <lnb> it says -b int [15:59] <lnb> do you add a 'k' after the number? [15:59] <lnb> or just -b 800 [16:01] <sacarasc> You add a k. [16:01] <lnb> ok [16:05] <espr3ss0> Q: is compiling "./configure | make | install" ? [16:05] <An_Ony_Moose> klaxa: so what filename extension should I use? [16:05] <sacarasc> espr3ss0: More or less. [16:05] <An_Ony_Moose> why use pipe? [16:06] <An_Ony_Moose> && is preferred to my knowledge [16:06] <An_Ony_Moose> so ./configure && make [16:06] <An_Ony_Moose> make install is installation, no longer compilation [16:07] <klaxa> An_Ony_Moose: i guess .mkv for matroska should be fine for either of those codecs, i'm not sure though [16:07] <klaxa> if you are using ffv1 or huffyuv, i'm not sure it'll work with matroska [16:07] <espr3ss0> sacarasc: i need to 'compile' including ... "LIBFFI_CFLAGS=-I/usr/include/ffi LIBFFI_LIBS=-lffi" [16:07] <An_Ony_Moose> klaxa: so just .mkv and add a -b (somethingsuitable) option? [16:08] <luc4> Hi! I'm trying to decode audio streams and I'm reading the documentation for avcodec_decode_audio4. But can I ask a specific format for the decoded samples? Like the number of channels, but per sample etc... Who is deciding that? [16:08] <luc4> but per sample = bits per sample :-) [16:08] <klaxa> using bitrate based quality measures on any of the codecs i named is not wise [16:09] <klaxa> if you are going lossly (libx264) do some test encodes with -crf [16:09] <klaxa> lower crf values mean higher quality [16:09] <klaxa> start with something like 20 or so [16:09] <klaxa> if it looks like shit reduce the value [16:09] <klaxa> don't encode the whole video just a part of it [16:09] <klaxa> to check quality [16:12] <Fjorgynn> crf seems more used than 2 pass encoding for H.264. Why? [16:13] <JEEB> because most people don't have a set file size limit to abide to [16:13] <An_Ony_Moose> klaxa: ok, thanks [16:13] <lnb> all this does is fail [16:13] <lnb> (#*&@& [16:13] <Fjorgynn> JEEB: but it seems like the "scene standard" also is crf today [16:13] <JEEB> yes, motherfucking finally! [16:14] <JEEB> it took them how many years? [16:14] <Fjorgynn> but then I don't have to do 2 pass encoding, right? [16:14] <JEEB> yes, you don't [16:14] <Fjorgynn> http://scenerules.irc.gs/t.html?id=2012_SDTVx264u1.nfo [16:14] <JEEB> and because the encoder isn't trying to hit a certain average rate, rate control isn't limited either [16:14] <JEEB> (unless you set vbv limits) [16:15] <JEEB> in other words, you are not losing quality/compression at all [16:15] <JEEB> while you are gaining by the fact that you are only doing one pass [16:15] <JEEB> granted, that one pass is not a "fast first pass" like it is with -pass 1, but I think that just makes sense :P [16:15] <Fjorgynn> can I do crf even if I rescale the video? [16:15] <Fjorgynn> I must [16:15] <JEEB> naturally, rescaling etc. doesn't affect the encoder [16:16] <JEEB> just that you might want to tweak your crf value of choice depending on the resolution [16:16] <JEEB> because lower-resolution stuff generally tends to get upscaled more during playback [16:16] <JEEB> and thus it is naturally easier for you to see possible artifacts [16:16] <Fjorgynn> I find stuff somewhere which are 4:3 but they release it as 1280x720... [16:17] <An_Ony_Moose> klaxa: changing the crf doesn't seem to have any effect at all [16:17] <JEEB> anyways, as I have already stated o9k times, you should under no circuimstances look at the "scene rules" as any kind of hint at "what should I do" [16:17] <klaxa> you may just not see the difference [16:17] <Fjorgynn> ;) [16:17] <JEEB> An_Ony_Moose, do you use bit rate at the same time or are your values just so low that you can't see the difference? [16:17] <Fjorgynn> JEEB: so if I am only rescaling I don't have to do -crf at all? [16:18] <Fjorgynn> only -c:v libx264 so it rescales and -c:a copy, right? [16:18] <JEEB> Fjorgynn, why are putting together pre-encoding stuff (deint/resize/crop etc.) and encoding!? [16:18] <An_Ony_Moose> JEEB: only crf [16:18] <JEEB> An_Ony_Moose, what encoder are you using? [16:18] <Fjorgynn> JEEB: I thought you must encode if you rescale? [16:18] <An_Ony_Moose> JEEB: whatever's the default for file extension mkv [16:18] <JEEB> An_Ony_Moose, depends on your build [16:18] <JEEB> there is no single default, see the log [16:19] <JEEB> note: you should never. NEVER imply that ffmpeg is gonna always pick the right one [16:19] <JEEB> (that you want) [16:19] <Fjorgynn> lol default [16:19] <JEEB> Fjorgynn, yes if you resize it implies that you have to re-encode [16:19] <JEEB> but still you were putting apples and oranges together just now [16:20] <An_Ony_Moose> JEEB: ah... it's using mpeg4. I assume I have to change it to libx264 then? [16:20] <JEEB> An_Ony_Moose, yup [16:20] <JEEB> crf is only in libx264 [16:20] <Fjorgynn> dunno [16:20] <JEEB> Fjorgynn, look at it this way: 1) Resizing implies that you have to re-encode 2) Re-encoding does not imply that you have resize [16:21] <JEEB> *have to [16:21] <An_Ony_Moose> bah. Is my problem that I'm using "ffmpeg version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers"? Should I be using a version that's not from libav? [16:21] <Fjorgynn> I must resize a file that's 1280x720 and 1GB > because I want it to be 4:3 and 300 MB [16:21] <JEEB> An_Ony_Moose, you should be using the 'avconv' binary in that case, and you want to install the libavcodec-extra package to get libx264 [16:23] <JEEB> Fjorgynn, anyways I hope you understand more how you took apples and oranges as the same with <Fjorgynn> JEEB: so if I am only rescaling I don't have to do -crf at all? <Fjorgynn> only -c:v libx264 so it rescales and -c:a copy, right? [16:23] <lnb> http://pastebin.com/KEND5PW0 [16:23] <An_Ony_Moose> what's actually the deal with avconv/ffmpeg? Why are they split? [16:23] <JEEB> An_Ony_Moose, in early 2011 there was this split, both projects are in active development and ffmpeg merges most if not all of libav's change [16:23] <JEEB> *changes [16:24] <JEEB> it's better not to go into the possible politics back then because both just seem to want to put that behind themselves [16:24] <An_Ony_Moose> ok [16:24] <Fjorgynn> ffmpeg -i input -c:v libx264 -vf crop=960:720:160:0,size=640:352 -crf 21 -c:a copy output.mp4 [16:24] <An_Ony_Moose> well I think I'll get ffmpeg because I have no idea how the options for avconv work (or are they the same?) [16:24] <JEEB> well, the new commands rewrite mostly came from libav [16:25] <JEEB> so using avconv should work mostly with the avconv binary [16:25] <JEEB> there are some differences [16:25] <An_Ony_Moose> I'll get ffmpeg then [16:25] <JEEB> http://x264.fushizen.eu/random/psp_encoding.txt <- see how libx264 specific settings (you don't generally have to use these at all) get used f.ex. [16:25] <JEEB> well, suit yourself :) That works too [16:25] <espr3ss0> Q: glib needs pkg-config & pkg-config needs glib ??? [16:26] <An_Ony_Moose> 16:25 < JEEB> http://x264.fushizen.eu/random/psp_encoding.txt <- see how libx264 specific settings (you don't generally have to use these at all) get used f.ex. [16:27] <An_Ony_Moose> oops [16:27] <Fjorgynn> lol [16:29] <espr3ss0> used "./configure --with-internal-glib" and ok, so why need glib ? [16:30] <espr3ss0> pkg-config i'm referring to [16:31] <An_Ony_Moose> ooh it wants yasm... What one doesn't do for speed :D [16:33] <An_Ony_Moose> why do glib and glibc have to have such similar names? [16:38] <JEEB> An_Ony_Moose, yes -- yasm is needed for compilation because x264/ffmpeg have a LOT of asm in there [16:38] <mpfundstein> its not needed [16:38] <An_Ony_Moose> I can imagine [16:38] <mpfundstein> but its advised :-) [16:38] <JEEB> mpfundstein, well yes -- but it's not really worth it to make a build without them [16:38] <An_Ony_Moose> http://dpaste.org/dsCCD/ damn# [16:39] <JEEB> also neither will let you configure without an extra option [16:39] <lnb> anyone have a preset to convert .mov to .flv so when watching video in full screen its good quality [16:39] <JEEB> without an assembler [16:39] <mpfundstein> JEEB: yes --disable-yasm [16:39] <mpfundstein> JEEB: but one should not do it [16:39] <JEEB> ...I HAVE SAID THAT QUITE A FEW TIMES [16:39] <JEEB> like, just now [16:39] <An_Ony_Moose> http://dpaste.org/dsCCD/ not very related but how do I fix that? [16:40] <An_Ony_Moose> export PATH=$PATH [16:40] <An_Ony_Moose> fixed it x) [16:44] <corda> hey there [16:45] <Ben_> I'm getting a duration change afte reencoding with ffmpeg [16:46] <Ben_> so my input is Duration: 00:00:02.85, start: 0.013333, bitrate: 1039 kb/s [16:46] <espr3ss0> so installing Glib, is for an update ? [16:46] <Ben_> and after ffmpeg -i myinput.mp4 -acodec copy -vcodec mpeg4 -b:v 500k output.mp4 [16:46] <Ben_> I'm getting [16:46] <Ben_> Duration: 00:00:02.92, start: 0.000000, bitrate: 485 kb/s [16:47] <Ben_> any idea where the pb could be? [16:50] <Fjorgynn> mpeg4 :o [16:50] <espr3ss0> Q: what's FAM ? o_O [16:50] <Fjorgynn> Ben_: no I have no idea [16:51] <Ben_> Fjorgynn: do you think it's a bug? [16:51] <Ben_> I believe it hads an empty frame at the end [16:51] <Fjorgynn> dunno, stay in here and wait for someone to answer [16:51] <Ben_> *adds [16:52] <Ben_> for more details: https://ffmpeg.org/trac/ffmpeg/ticket/2108 [16:52] <Ben_> I'm gonna fetch the git and try to recompile it [16:53] <espr3ss0> Glib "make", "make[4]: *** [gconvert.lo] Error 1" ? [16:53] <Ben_> I used brew to install it so not sure if it uses the HEAD of the git repo [16:57] <seece> hello, any tips how to stream raw video over LAN? [16:58] <seece> i'm on windows, using directshow to capture input [17:13] <espr3ss0> Q: Glib still won't configure properly, even used "./configure --with-libiconv=native" (after updating libiconv) [17:15] <espr3ss0> "#error GNU libiconv not in use but included iconv.h is from libiconv" [17:17] <Ben_> hey the new ffmpeg ask for libavutil.52 [17:18] <Ben_> anybody know the diff between this one and 51? [17:22] <Diogo> any tool to compare quality of 2 video and generate a graphic? command line tool please.. [17:24] <espr3ss0> Q: do i need to specify "--prefix=/usr/local" for libiconv ? [17:29] <Diogo> any tool to compare quality of 2 videos and generate a graphic? a command line tool please.. [17:30] <espr3ss0> what's the latest version of "libconv" ?, i have 1.9.2 [17:36] <An_Ony_Moose> anyone know a good tool for deshaking videos? The video filter included with ffmpeg gives me terrible results. [17:36] <Mavrik> hqdn3d? [17:37] <ubitux> hqdn3d for deshake? wtf [17:37] <ubitux> An_Ony_Moose: https://ffmpeg.org/trac/ffmpeg/ticket/1430 :( [17:37] <espr3ss0> specified "--prefix:/usr/local" still no change [17:38] <espr3ss0> :-9 [17:38] <An_Ony_Moose> http://public.hronopik.de/vid.stab/ seems quite effective but transcode doesn't like me. Or the ubuntu package maintainer for it doesn't. [17:39] <An_Ony_Moose> ah well I guess I'll build transcode from source then [17:40] <An_Ony_Moose> am I in the future? I have transcode 1.1.7 but the website says the latest is 1.1.0 [17:47] <saste> An_Ony_Moose, you are of course in the future, w.r.t. to that website [17:48] <An_Ony_Moose> saste: haha [17:55] <An_Ony_Moose> https://github.com/georgmartius/vid.stab now if only this were an ffmpeg plugin [17:58] <espr3ss0> Q: anyone know how to fix the "libiconv" problem ? [18:06] <Mista_D> Is the fatest way to decode x264 "-f null"? Can I force decoder to read every 10th frame or so? [18:07] <JEEB> no [18:07] <JEEB> if you want proper decoding that is [18:07] <JEEB> you could make it skip everything but IDR frames, but no idea how to do that :P [18:10] <Mista_D> JEEB: thanks, I'll see if I can read IDR only. [18:23] <grepper> An_Ony_Moose: transcode in debian is badly broken due to leaving some modules out (license issues). Dunno about ubuntu. [18:44] <Fjorgynn> what does -f null do? [18:45] <Fjorgynn> what do you use it for? Decode to what? [18:46] <Fjorgynn> Just read what Mista_D wrote [18:50] <extractor> Hello, do anyone know how to extract only the last image of a video? [19:31] <ubitux> Fjorgynn: -f null demux and decode, and output to nothing [19:36] <Fjorgynn> aha? [19:46] <seece> can ffmpeg stream any codec with UDP? or does it need to be in a mpegts container? [19:50] <ubitux> should not be container specific [19:50] <ubitux> it's a protocol like another [19:50] <klaxa> udp is just the protocol you should be able to send anything over udp [19:51] <seece> ok great [20:12] <seece> i get "real-time buffer 258% full! frame dropped!" errors even though i can see the CPU load is really low. any ideas what could be causing it with x264? [20:46] <espr3ss0> Q: "libiconv" requires enabling, is this done by "--enable-static" ? [20:49] <klaxa> doubt it [20:49] <espr3ss0> how can i enable it ? o_O [20:49] <Mavrik> what's libiconv? [21:12] <espr3ss0> Q: am i ok with "/usr/local/" or only "/usr/local" or even "usr/local" ? :-) [21:13] <espr3ss0> i have a feeling it's what cause "/usr/local//lib/.new" [21:14] <sacarasc> Just /usr/local should work. If you do /usr/local/ it will put do /usr/local//bin, which is the same as /usr/local/bin, but looks ugly. [21:14] <espr3ss0> ok thanks [21:21] <llogan> espr3ss0: what distro? [21:21] <llogan> ...not that it matters too much [21:22] <llogan> eh, nevermind. i misread your question [21:28] <espr3ss0> llogan: distro of ? [21:28] <klaxa> gnu/linux distribution [21:28] <klaxa> but doesn't matter regarding the question [21:29] <espr3ss0> os x 10.6 [21:37] <luc4> Hi! I'm getting a segfault when using the avcodec_decode_audio4 function. What are the possible reasons? I'm running ffmpeg on an armv6 hardfp platform. [21:39] <llogan> luc4: not many API answers here. try the libav-user mailing list if you don't get an answer. [21:39] <llogan> http://ffmpeg.org/mailman/listinfo/libav-user [21:48] <jadams> hello [21:48] <jadams> I'm a noob trying to stream video with as little lag as possible over the network [21:48] <jadams> I can only get it down to around 2s lag with this: `ffmpeg -framerate ntsc -video_size 320x240 -f x11grab -i :0.0 -vcodec libx264 -tune zerolatency -crf 26 -b 900k -probesize 1000 -analyzeduration 0 -g 10 -f mpegts udp://aye-seven.local:5003` [21:48] <jadams> opening the stream in vlc (ffplay is laggier by far) [21:49] <rkt> which is the best codec to use with ffserver for 8bit graphics? i mean vmv1 feels hardcore for an application only running at 8bit anyway? :D [21:56] <jadams> it would appear the problem was in the client in fact (caching) [21:56] <jadams> thanks for the help [22:01] <hub2> Hi,I'm making a program which uses ffmpeg in C++,and when I'm compiling it I get UINT64_C was not declared in this scope error. Is there any way to remove it? [22:04] <Mavrik> hub2: IIRC it's -D__STDC_CONSTANT_MACROS to the compiler to declare those macros also in C++ [22:07] <hub2> Thanks Marvik. That solved the problem [22:19] <luc4> Hi! Is ffmpeg supposed to work on armv6 hardfp? [23:18] <sesam> Hi [23:19] <sesam> What application do u guys to split a quicktime before using two or more instances of ffmpeg? [23:21] <sacarasc> I didn't know people did that. [23:21] <sesam> and what is the fastest distro for ffmepg? arch maybe the way to go but it takes some time to get up and running, maybe its worth it? [23:23] <sesam> I make large lots of large files that needs to compress into a h264, and a faster way would be to chop it up and process each slice on different computers and then stich them together again. [23:32] <sesam> best distro for ffmpeg? [23:38] <sesam> guess ill compile it on centos then =) [23:38] <luc4> Hi! Anybody who knows what attribute_align_arg means? [23:40] <llogan> sesam: the fastest distro is the one you're most familiar with [23:40] <Mavrik> luc4: if you grep through the code you'll find out it's defined as force_align_arg_pointer [23:41] <Mavrik> compiler macro. [23:41] <sesam> llogan: thx [23:41] <llogan> but arch usually does keep their repo ffmpeg more up to date than any other distro that i've checked [23:41] <luc4> Mavrik: yes, I found that. My problem is that I'm getting a crash in a strange point of avdec_decode_audio4, I'm trying to understand if that might be related. [23:42] <llogan> not that it's hard to compile ffmpeg (with the compile guides if you need help) [23:42] <sesam> llogan: then ill go for arch instead. I want it to be as small as i can get it. [23:42] <llogan> centos or arch would work just fine. [23:43] <Mavrik> luc4: what does the debugger say about the crash? [23:43] <luc4> Mavrik: the crash I'm getting is when avctx->codec->type != AVMEDIA_TYPE_AUDIO is executed. [23:43] <Mavrik> hmm [23:43] <Mavrik> are you passing a correct context pointer? [23:43] <luc4> Mavrik: I'm working on an embedded platform. Really hard to get a gdb debugger working. [23:43] <Mavrik> did you try to break and see the struct? [23:43] <sesam> llogan: im currently installing a centos minimal on my proxmox node. i will try arch as well. [23:44] <klaxa> sesam: splitting up a file and running multiple instances of ffmpeg doesn't increase encoding time [23:44] <luc4> Mavrik: I'm doing the same instruction just before invoking that function in my code, and I get no segfault. [23:44] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide [23:44] <klaxa> libx264 is multithreaded already [23:44] <klaxa> and should use all available cores [23:44] <sesam> llogan: thx [23:44] <klaxa> if you are spreading the split files across multiple computers it's a different story of course [23:45] <klaxa> i wrote some code for that in python, it only works with matroska containers though, you'd also have to modify it heavily to match your usecase [23:45] <sesam> klaxa: im spreadding them acress multible computers. Need the speed. [23:46] <sesam> does libx264 use the GPU as well? [23:46] <klaxa> in that case it makes sense to do that yeah [23:46] <klaxa> not that i know of [23:46] <klaxa> there have been approaches to use OpenCL with x264 [23:46] <klaxa> not sure if anything official has been released yet [23:46] <luc4> Mavrik: oh wait wait sorry, seems it is segfaulting also in my code. [23:47] <sesam> that would have been awsum if they could get opencl running [23:47] <Mavrik> not really [23:47] <klaxa> from what i read the usefulness is somewhat limited [23:47] <Mavrik> the quality of GPU encoding is pretty much very bad across the board [23:47] <sesam> ive tried the most commercial apps that are on the market and only episode pro does it but its crazy expensive [23:48] <sesam> bad quality in like picture quality? [23:48] <klaxa> Mavrik: huh? i thought they are using the GPU for general purpose computations [23:48] <Mavrik> hard to be general-purpose with a highly specialized core [23:48] <klaxa> mmh true that [23:48] <Mavrik> memory barriers still hurt quality of any video encoding done on GPUs [23:48] <Mavrik> which is the reason why x264 efforts have stalled [23:49] <Mavrik> you can't do efficient searches when you're locked into small image areas [23:49] <sesam> then what is the fastest way to encode a 264 file? [23:50] <sesam> and still ensure the quality [23:50] <Mavrik> a 4-core Intel CPU with huge clock [23:50] <sesam> sayway? [23:50] <sesam> hugh clock [23:50] <sesam> ? [23:51] <Mavrik> frequency. [23:51] <Mavrik> as much GHz as you can get [23:51] <sesam> =) [23:51] <sesam> im swedish, sorry for not understanding [23:51] <Mavrik> ^^ [23:51] <Mavrik> no problem [23:52] <sesam> why 4 and not 8? is it because the frequency gets lower as more cores there are crampt into a single cpu? [23:52] <klaxa> are there 8-core intel CPUs? [23:52] <Mavrik> because you get diminishing returns when paralelizing video encoding [23:52] <Mavrik> more than 4 cores (with HT) doesn't bring significant speed-ups anymore [23:53] <sesam> more like 8 threads =) [23:53] <Mavrik> klaxa: yeah, you can get 8-core Xeons with HT (16 threads total) [23:53] <klaxa> oh nice [23:53] <Mavrik> expensive beasts though [23:53] <luc4> Mavrik: I suspect I found it. I'm missing the codec init part. Sorry. [23:53] <luc4> Mavrik: thanks [23:53] <Mavrik> luc4: no problem, glad you found a bug :D [23:53] <sesam> i have 40x i7 3.2ghz computers that ive built. [23:54] <sesam> so splitting a file up in 40 peaces will go fast? [23:54] <klaxa> lol they have 10-cores [23:54] <sesam> hehe [23:54] <Mavrik> hmm [23:55] <Mavrik> sesam: rather run 10 files at once :) [23:55] <sesam> true that [23:55] <klaxa> sesam: do you know of map-reduce? [23:55] <klaxa> it's exactly for this usecase [23:55] <sesam> klaxa nope never heard of it [23:55] <sesam> ill do a search [23:56] <klaxa> it's a well... design-technique? for extremely scalable distributed computing [23:56] <klaxa> i don't know how well existing software can be used to optimize your usecase [23:56] <Mavrik> yeah, trouble with map-reduce with video is that you have to move those huge files around [23:56] <sesam> sounds interesting, not sure if it will work with ffmpeg though [23:56] <Mavrik> I've found that wastes alot of time :\ [23:57] <klaxa> mmh [23:57] <klaxa> Mavrik have we discussed this earlier last year? because i remember i was talking to someone from #ffmpeg about this [23:57] <Mavrik> hmm, don't remember [23:57] <klaxa> and i got some more or less usable code working for that [23:58] Action: t4nk637 knocks, asks if there is anyone home who is familiar w/ ffmpeg build in space-constrained settings & for a narrow band of conversions... [23:58] <Mavrik> we have a similar use case and just do a full file encode per 4 cores of a machine [23:58] <sesam> maybe with a fast raid-0 system with 2-4 nics working in link-aggregation it will run goot [23:58] <Mavrik> t4nk637: details! :) [23:58] <Mavrik> but we have about 1-2 hr long files that have to be encoded to several formats [23:58] <t4nk637> thanks @ Mavrik [23:58] <t4nk637> Question to power-users and those familiar w/ embedded contexts: I want to use ffmpeg for a narrow subset of conversions, -- only converting jpeg images to a time-lapse mp4 file -- [23:59] <sesam> mavrik: u work at a post-house? [23:59] <sesam> mavrik: do u do all conversions in ffmpeg? [23:59] <t4nk637> What is the minimal subset/choice of muxers, demuxes, decoders, encoders, filters, etc that I'd need to do this? [23:59] <t4nk637> just jpegs to mp4 timelapse [23:59] <Mavrik> sesam: post-production (cutting, etc.) in Premiere, release encodes in ffmpeg yeah [00:00] --- Sat Jan 5 2013

1 0

ffmpeg-devel.log.20130103
by burek 04 Jan '13

04 Jan '13

[00:00] <michaelni> its like i want filter foo but it should not be enabled [00:00] <ubitux> yeah sure [00:00] <ubitux> what about deint=all or deint=marked? [00:00] <saste> mp=kerndeint has such an option [00:01] <saste> it allows to "dynamically" disable the filter [00:01] <saste> i like deint=all|marked [00:01] <saste> since auto=1|0 is not very descriptive [00:03] <michaelni> deint=all|interlaced [00:04] <ubitux> :) [00:05] <ubitux> you may want to add a ref to the setfield filter btw [00:07] <saste> what about printing "CONSULT YOUR LOCALLY INSTALLED DOCUMENTATION FOR OLDER VERSIONS!!!" in red, bold, blinking, bleeding, 300% sized characters in documentation.html? [00:08] <ubitux> you forgot "moving" with the help of <banner> [00:08] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:740e74089555: av_rescale: support passing MIN/MAX through [00:08] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:98db90589651: avformat_seek_file: Dont attempt to rescale INT64_MIN/MAX [00:09] <beastd> saste: it would be better if we could archive the versions for the recent releases and make the accessible or similar [00:10] <saste> beastd, similar to what we did with doxygen? [00:10] <saste> yes that may be a good idea [00:12] <beastd> how is it done for the doxygen? [00:12] <cone-140> ffmpeg.git 03Clément BSsch 07master:f263426eedd3: lavu/math: fix than/then typo, and add a comma. [00:13] <beastd> i am running out of time for today. see you tomorrow [00:13] <beastd> bye... [00:13] <ubitux> cya [00:17] <cone-140> ffmpeg.git 03Clément BSsch 07master:43adc62e7022: lavf/mpegtsenc: fix missing word in error message. [00:28] <ubitux> saste: you should have said the faq will get updated in the next hours [00:28] <ubitux> otherwise the guy is going to rage and re-open the ticket saying it's not in the online doc [00:28] <ubitux> :D [00:28] <saste> let it be, let's have some fun [00:35] <saste> the it would be nice to have an ffmpeg-faq manual [00:35] <ubitux> wm4: i started looking at making the matroska demuxer output "ass" packet instead of "ssa", so introducing a new codec id [00:36] <ubitux> it's not really a problem, except that the whole demux/decode/encode/mux chain needs some little updates [00:36] <wm4> ok [00:36] <ubitux> also, the decoded subtitles will still be in "ssa" format (in the ass field of the rects) [00:37] <ubitux> that's the next step, but AFAIK all the mplayer forks just care about the demux for that, right? [00:37] <ubitux> saste: i'm not against [00:38] <saste> ubitux, problem is that faq references html pages [00:38] <saste> so we definitively need indexing [00:39] <saste> i plan to dive into haskell and possibly contribute to pandoc and add a texinfo reader and indexing [00:39] <saste> but that may take time [00:39] <ubitux> wm4: so basically, what will change for you is that you should be able to get rid of the various hacks for muxed subtitles, but if you're doing some subtitles decoding at some point (like you should do if you're planning to support the standalone subtitles files), it will still be using the ssa format [00:39] <ubitux> pandoc is in haskell? :) [00:39] <saste> (haskell is paradise for a computer scientist ;-)) [00:40] <wm4> does ssa not have the same issues? [00:40] <ubitux> by ssa i mean the current form [00:41] <ubitux> basically timings, clrf after each line, no read order, .. [00:42] <ubitux> so if you're demuxing and decoding a standalone file (.srt, .ass or anything else), the markup in the sub->rects[i].ass will still be in that format [00:42] <ubitux> i'm still unsure about what to do with the ass demuxer (wheither it should output ASS like in matroska, or still keep the lines so in SSA format) [00:43] <ubitux> but it should change anything for you in the end [00:43] <ubitux> since all external subtitles (.ass or no) should be handled exactly the same [00:43] <ubitux> (and you'll need a decode step) [00:44] <ubitux> wm4: and btw, if you plan to *decode* the (even new) ass packets from the matroska demuxer, you'll also end up with "ssa lines" in sub->rects[i].ass [00:45] <wm4> ubitux: so, why not proper ASS? does matroska actually use this ssa form? [00:45] <ubitux> what i'm doing right now is just add the non altered packets with ASS codec, mainly for the mkv demuxer [00:45] <ubitux> wm4: that's the next step, but that's a bit tricky from a retro compat PoV [00:45] <iive> \o/ [00:46] <ubitux> wm4: and no, it's not using the ssa form at all [00:46] <ubitux> it's just that the decoding process of subtitles, whatever the codec of the packet, leads to a common markup [00:46] <ubitux> with is currently ssa, stored in the ass field of the rects [00:46] <ubitux> currently, for ass decoding, it's just copied in the field [00:47] <ubitux> what will happen now is that like every other codecs, it will be "encoded" into these kind of ssa lines [00:48] <ubitux> wm4: do you expect ass markup from other containers? [00:49] <wm4> ubitux: what other containers? [00:49] <ubitux> AFAIK, only MKV, ASS and NUT demuxers can output these packets [00:49] <ubitux> mkv will now output proper ass packets with the new codec id [00:49] <wm4> and ASS? [00:49] <ubitux> ass, i'm still uncertain but it shouldn't affect you [00:50] <ubitux> and nut, i need to check what's going on here, but i'm pretty sure it will need various other work to be sane [00:51] <ubitux> wm4: as i said, the thing with the ass demuxer (i'm talking about when you demux a .ass file), is that you likely need exactly the same processing chain as the other subtitles [00:51] <ubitux> .srt, .sub, .aqt, etc [00:51] <ubitux> so you will NEED a decode step, filling the rects->ass fields with the "broken" markup [00:51] <ubitux> that is currently the only way to get a consistent output between all these formats [00:52] <wm4> I don't see much of a point in having two packet formats for the same thing... so ASS should definitely output the proper format [00:52] <wm4> the "decoded" rects are not packets, though [00:52] <ubitux> and you're supposed to decode everything at startup, and feed it to libass, and then don't care about [00:52] <ubitux> (just like vf subtitles) [00:52] <ubitux> you're not supposed to use the "chunked" mode with a .ass file [00:52] <ubitux> like you do with matroska [00:53] <ubitux> or well the "normal" mode i would say, sorry [00:53] <ubitux> wm4: well, that's just another packet [00:53] <wm4> that entirely depends how you feed it to libass, but the way vf_ass does it seems sort-of fine [00:54] <ubitux> not with .ass file [00:54] <ubitux> you're not really supposed to do that [00:54] <ubitux> just handle the .ass files just like every other standalone subtitles files [00:54] <wm4> also I don't see the point of the decoder interface here [00:55] <wm4> can't it just read and output full files? [00:55] <ubitux> consistent markup between all the subtitles [00:55] <ubitux> the demuxer subtitles split the the markup from the timing level [00:55] <ubitux> the subtitles decoders "encode" the markup in the ass field of the rects [00:57] <ubitux> note that this will change (basically we will make the decoders output a common abstract style sub instead of a ssa line markup, and provide helpers to get a ass markup to be piped to libass) [00:57] <wm4> the way vf_subtitles is written obviously won't really work with demuxed subtitles of other types, unless you do the silly reset-on-seek thing [00:57] <ubitux> it will [00:57] <ubitux> since you preload the whole file [00:58] <wm4> I mean with files that interleave video and subs [00:58] <ubitux> yes, vf subtitles is for standalone files [00:58] <ubitux> standalone subtitles files [00:58] <ubitux> .srt, .ass, .sub, .aqt, ... [00:59] <ubitux> it can work with video+sub though [00:59] <ubitux> (it will just demux the whole thing first) [00:59] <wm4> nit acceptable for a video player [00:59] <wm4> *not [00:59] <ubitux> yes, but i was not asking you to do that :) [01:00] <ubitux> as i said, you're only interested in clean ASS packets out of matroska, right? [01:00] <ubitux> since that's the only case where you pipe directly the packets to libass [01:00] <ubitux> wm4: i'm proposing to do like vf subtitles for the standalone subtitles, where it is perfectly appropriate [01:01] <ubitux> (and that's exactly what mplayer does manually with subreader) [01:01] <ubitux> of course, you'd better wait for the charset and utf16 thing before implementing it [01:02] <ubitux> i need to rush this stuff in the next days [01:18] <ubitux> https://bugs.gentoo.org/show_bug.cgi?id=447838 is this something we need to fix? [01:19] <ubitux> or it looks like a problem that affect both projects? [01:23] <cone-140> ffmpeg.git 03Clément BSsch 07master:491ca0e89ff9: Replace references to "que" with the appropriate word. [01:24] <ubitux> saste: did you look at the ffprobe & xface issues? [01:24] <ubitux> for ffprobe, i'm really surprised about the logic behind the code [01:25] <saste> ubitux, no my gcc is not complaining [01:25] <ubitux> althought the warning is likely to be fixed with a consistent usage of int & enum [01:25] <ubitux> saste: what about the logic of the code? (for both) [01:26] <saste> ubitux: i have to find you pastebin [01:26] <saste> it's not like it looks high priority since it worked fine so far... [01:27] <ubitux> yeah sure but& [01:27] <ubitux> the code really looks fishy to me [01:28] <durandal_1707> ubitux: release 2.0 [01:28] <ubitux> oh i think i understand& [01:28] <ubitux> durandal_1707: yes? [01:30] <ubitux> saste: well, it might not be that important then :) [01:30] <ubitux> the ffprobe aliasing warnings are present in arm boxes, and the xface with gcc 4.7 [01:31] <saste> ubitux, did you understand what is it complaining about? [01:40] <ubitux> saste: i believe that's because children_ids is not of type enum SectionID [01:41] <saste> ubitux, ok [03:57] <ubitux> wtf [03:57] <ubitux> #2102 "The input line is too long. [03:57] <ubitux> :" [03:57] <ubitux> what is the guy smoking [04:01] <Compn> ubitux : oh i got a great idea for a filter [04:02] <Compn> epilepsy filter, removes bright light flashes from video [04:09] <Compn> it will replace the bright flash frame with the previous frame before the bright light. so it will make the video stutter a little [04:13] <ubitux> Compn: good idea, now you can implement it [04:40] <cone-842> ffmpeg.git 03Michael Niedermayer 07master:52334f5be2ef: tiff: print error for old style LZW [07:30] <cone-842> ffmpeg.git 03Clément BSsch 07master:3fa642d60f42: subviewer: sanitize packets. [07:48] <cone-842> ffmpeg.git 03Clément BSsch 07master:dc8b36746a9d: lavf/webvtt: use ff_subtitles_read_chunk(). [08:06] <ubitux> why is my subtitles todo list still huuuge? :( [08:07] <ubitux> ass codec, charset, utf16, stylesubtitles, teletext, blending api, subtitles in lavfi, & [08:07] <ubitux> :( [08:33] <ubitux> mmh [08:35] <ubitux> wtf [08:39] <ubitux> it's like if the sources were updated, but the code still old [08:39] <ubitux> i guess i'll wait for the next run [09:33] <ubitux> saste: sorry, by macro i meant the enum [09:33] <ubitux> like checking for YADIF_PARITY_TFF instead of 0 [09:33] <ubitux> in the code [09:33] <saste> ubitux, uhm not for the moment [09:34] <saste> for merging reasons and also because i wanted to keep the patch short [09:34] <ubitux> okay [10:23] <saste> ^ the start_number saga continues... [10:35] <wm4> "Ubuntu does not provide ffmpeg from the FFmpeg project, but a fake version from a third-party fork which we can not support here." [10:35] <wm4> awesome [10:37] <JEEB> wm4, just like mplayer2 provides/provided a "fake mplayer binary" ^q^ [10:38] Action: wm4 waits for the 3rd ffmpeg fork [10:38] <wm4> or 2nd, depending on how you count [11:43] <kierank> ubitux: you're going to implement teletext in ffmpeg [11:43] <kierank> wow you're nuts :) [11:43] <ubitux> not now, as you can see there are several other entries in the TODO list [11:43] <kierank> doesn't change the fact that you're nuts :) [11:43] <ubitux> and by teletext i mean decoding from the sane packets [11:43] <ubitux> not from the vbi [11:43] <kierank> oh [11:43] <kierank> still [11:44] <kierank> teletext rendering is a total pita [11:44] <kierank> ask j-b [11:44] <ubitux> i'll start with decoding as normal text, and i'll see later [11:44] <ubitux> i guess you're talking about the black background etc? [11:45] <kierank> it's just a lot of work in general [11:45] <kierank> cf libzvbi [11:45] <ubitux> there is a guy working on this [11:45] <ubitux> he sent a draft patch a few days ago [11:45] <j-b> teletext is impossible to do right [11:45] <ubitux> i'm more focused on something else right now :) [11:46] <ubitux> j-b: yeah but being able to decode it to put it back as srt/ass/whatever is important for users when they transmux/transcode [11:46] <ubitux> and i think that's something relatively easy to do [11:46] <ubitux> especially when we'll be done with the StyleSubtitles API :) [11:47] <kierank> ubitux: what's the subject of the patch [11:47] <ubitux> mmh it was in an unrelated thread, just a min [11:49] <ubitux> kierank: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/136677.html [11:49] <kierank> oh i remember that actually [11:49] <kierank> he used libzvbi thankfully [11:49] <ubitux> yup [11:51] <ubitux> anyway, there are other priorities right now [11:51] <ubitux> i think we're going to get something really nice with subtitles [11:51] <ubitux> we already have an almost sane working chain [11:55] <pross-au> ubitux: is this also going to be able to carry eia-608 closed captions? [11:56] <ubitux> i'd like to [11:56] <ubitux> it should be flexible enough [11:56] <ubitux> ideally, someone should write a decoder [11:57] <ubitux> so i get an idea about what's necessary ;) [11:57] <ubitux> basically we can extend the styles, and the chunk "types" (we can insert timing chunks for karaoke etc) [11:57] <ubitux> and we can basically store any kind of information [11:58] <pross-au> decoder is on my todo list. last time i looked at subtitles, FFmpeg didn't have a good 'internal format'. We were had the choice of 'bitmaps' or 'ass' text string. [11:59] <ubitux> yes [11:59] <ubitux> eia-608 is text only, right? [12:00] <ubitux> just with a particular encoding [12:00] <ubitux> iirc the character drawing is free [12:00] <ubitux> pross-au: ass is used for the style [12:00] <ubitux> you can ignore it at first [12:01] <ubitux> and just create text packets for codec id text [12:01] <ubitux> basically you would just write a demuxer [12:01] <pross-au> IIRC, there are colors [12:01] <ubitux> but mmh it's already demuxed [12:01] <ubitux> well then just write a decoder like the text dec :p [12:02] <ubitux> pross-au: currently you have to print ass tags [12:02] <ubitux> that's what i'm going to change with the stylesubtitles api [12:02] <ubitux> but it would be really nice to have a basic decoder ;) [12:03] <ubitux> so we can do some eia-608 to srt [12:03] <ubitux> or to ass [12:03] <pross-au> i have the beginnings of one (it worked with the .wtv bastardised variant of eia-608). [12:04] <ubitux> beginning of a decoder? [12:04] <pross-au> yes, it works with a subset of eia-608 that .wtv uses [12:04] <ubitux> awesome [12:05] <ubitux> :) [12:05] <ubitux> michaelni: did you see that: http://fate.ffmpeg.org/log.cgi?time=20130103073615&log=compile/201301022249… ? [12:05] <pross-au> (kinda long story, but Windows Media Centre extracts the eia-608 from the ts, strips some stuff, and 're-encodes' it into a separate .wtv stream) [12:05] <ubitux> :/ [12:35] <kierank> pross-au: it drops 708? [12:44] <pross-au> kierank: i forget, there was a bug tracker entry on it. [12:45] <kierank> 608/708 is annoying - there are so many ways of storing it [12:47] <pross-au> https://ffmpeg.org/trac/ffmpeg/ticket/1482 [12:47] <pross-au> Yes [12:48] <pross-au> well there is the big problem of 608/708/TELETEXT packets embedded inside AV_MEDIA_TYPE_VIDEO streams. I don't think that one has been solved yet. [12:48] <kierank> not teletext [12:48] <kierank> 608/708 yes [12:49] <kierank> elenril said his buffer ref stuff will fix that [12:49] <pross-au> Okay. Cool. [12:50] <kierank> well it won't fix [12:50] <kierank> it'll just export [12:52] <pross-au> ubitux: is there a description of your sytlesubtitles system somewhere? i am a bit unclear on how its going to work [13:03] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2012-November/134607.html [13:04] <ubitux> pross-au: also, related: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-September/130474.html [13:07] <pross-au> this is really good :) [13:07] <ubitux> it's yet to be completed [13:08] <ubitux> i've still some pending stuff [13:09] <pross-au> i'll dig out the cea-608 stuff i was working on. also have a lrc (music lyrics) decoder somewhere too. might help to shake down the required chunks. [13:10] <michaelni> ubitux, didnt see the warning but well icc complaining about doing calcuilations with enums is a bit limited [13:10] <michaelni> adding casts wont make the code better [13:10] <ubitux> michaelni: i'm not saying that's justified, i just happened to see that :p [13:11] <michaelni> maybe that warning should be disabled [13:11] <ubitux> pross-au: there are now quite a bunch of subtitles demuxers and decoders [13:11] <ubitux> which you can use as base [13:59] <Compn> some video game drivers and other things have anti-epilepsy features [14:36] <divVerent> there are filters for that? How do they even work [14:37] <Compn> brightness / flicker detection [14:37] <divVerent> and that works? [14:37] <divVerent> https://www.youtube.com/watch?v=hNRhvHIk5C8 - what would they do here? ;) [14:38] <divVerent> black screen? ;) [14:39] <Compn> more curious about this video https://www.youtube.com/watch?v=1m8bBP1l1_c [14:39] <divVerent> the filter would have to change the volume to zero ;)+ [14:40] <Compn> lol [14:42] <Compn> http://forum.doom9.org/archive/index.php/t-132264.html [14:42] <Compn> dark_shikari , that name sounds familiar [15:20] <cone-971> ffmpeg.git 03Reinhard Tartler 07master:60e60d99ad86: forgotten changelogs for 9_beta2 [15:20] <cone-971> ffmpeg.git 03Janne Grunau 07master:9ac44ad9d06f: h264: check SPS entries directly to detect pixel format changes [15:20] <cone-971> ffmpeg.git 03Johan Andersson 07master:7768a635c4ea: cmdutils: update copyright year to 2013 [15:20] <cone-971> ffmpeg.git 03Michael Niedermayer 07master:1894302a44f3: Merge remote-tracking branch 'qatar/master' [15:24] <DEgITx> Question about AVSEEK_FLAG_BYTE. Is it seeking method really working, and is it need some recalculation of keyframes? Because after i have tryed it on h264 video i was getting such errors when decoding frame: [15:24] <DEgITx> [h264 @ 090c0060] Missing reference picture [15:24] <DEgITx> [h264 @ 090c0060] decode_slice_header error [15:24] <DEgITx> ... [15:24] <DEgITx> Even packet position didn't change =\ What i'm doing wrong?)) [16:35] <cone-971> ffmpeg.git 03Nicolas George 07master:be01d1a24ddc: fate: disable globbing when calling shell functions. [17:05] <saste> michaelni, do you want to comment on yadif named constants/options patches? [17:33] <saste> can someone confirm that the equivalent of tinterlace=4,format=yuyv422,mp=kerndeint,format=yuyv422 [17:33] <saste> doesn't work in mplayer? [17:33] <saste> also what if i drop support for yuyv422p? [17:35] <divVerent> doesn't work? as in, kerndeint does not support yuyv422 format? [17:35] <saste> divVerent, as in: the output is borken [17:36] <divVerent> hehe, I always found kerndeint's output broken ;) [17:36] <divVerent> and mpv even removed the filter [17:36] <divVerent> so, can't check [17:37] <saste> divVerent, no I mean that you get only the left side of the image actually written [17:37] <saste> i don't know how to force the format with mplayer [17:37] <divVerent> haha [17:37] <divVerent> scale,format=yuyv422,kerndeint,scale [17:37] <divVerent> if the format name is right in mplayer [17:39] <teratorn> divVerent: yuvv422? what is that ? [17:39] <teratorn> oh yuyv [17:39] <saste> well it ain't broken in mplayer [17:40] <saste> better not to know why [17:40] <divVerent> m- {IMGFMT_YUY2, PIX_FMT_YUYV422}, [17:40] <divVerent> - {IMGFMT_YUY2, PIX_FMT_YUYV422}, [17:40] Last message repeated 1 time(s). [17:40] <divVerent> sorry [17:40] <divVerent> scale,format=yuy2,kerndeint,scale [17:40] <divVerent> is this broken? [17:41] <saste> oh wait it is even in mplayer [17:41] <divVerent> apparently yuy2 is mplayer's format name for this [17:41] <saste> i mean, it is *broken* even in mplayer [17:43] <divVerent> from reading the code in the last version before mpv removed the filter: kerndeint IS broken for that format [17:43] <divVerent> #it assumes 8bpp on each plane [17:43] <divVerent> but it also doesn't claim support for YUY2 in that version [17:44] <saste> fixing it should be easy, but also it is worth it? [17:44] <divVerent> wait, it once did [17:44] <divVerent> given the filter sucks anyway [17:44] <divVerent> just remove its YUY2 support [17:44] <saste> michaelni, ^ ? [17:45] <saste> divVerent, in what sense does it suck? [17:45] <divVerent> it looks bad [17:45] <saste> divVerent, compared to what? yadif? [17:45] <divVerent> for example, yes [17:45] <divVerent> in my opinion it even looks worse than pp=lb [17:46] <divVerent> it's the one that looked like a weird sharpener [17:46] <divVerent> and the image shivers all the time [17:47] <divVerent> still, fixing it shouldn't be too hard [17:59] <michaelni> saste, ? [18:00] <saste> michaelni, don't bother, i already fixed yuyv422 [18:00] <michaelni> ok [18:00] <saste> still waiting for yadif patches [18:00] <michaelni> iam looking at them right now :) [18:00] <michaelni> well 1 patch that is [18:49] <saste> what's the mplayer equivalent name for "yuv420p"? [18:50] <saste> well without the ending "p" [18:52] <saste> -vf scale,format=argb,kerndeint,scale [18:52] <saste> this doesn't want to work [19:19] <durandal_1707> can i swap frame.data[x] pointers? [19:32] <michaelni> durandal_1707, in principle yes, you might have to also swap extended_data pointers though and there of course might be code ive forgotten that assumes things it shouldnt assume [19:49] <Compnn> the great format wars [19:49] <Compnn> pirates chose mkv. apple, sony, microsoft, adobe, android and others chose mp4. [19:53] <durandal_1707> pirates choose rar [19:53] <Prottey> DRM is money, money is everything [19:53] <durandal_1707> and nothing [19:54] <Prottey> yet Android (Google that is) chose WebM as well [19:54] <Prottey> and everybody forgot Dirac :P [19:55] <Prottey> except maybe FGPA nerds who do h/w accels [19:55] <michaelni> Tjoppen, can you look at ticket/1916 it seems a regression since a commit from you [19:57] <Tjoppen> right, forgot about that one [19:58] <Tjoppen> there's a bit of a start-of-the-year lull at work, so I should have time to look at it [19:58] <durandal_1707> no full time ffmpeg dev? [20:00] Action: durandal_1707 ^ becoming boring [20:14] <cone-971> ffmpeg.git 03Michael Niedermayer 07master:bbb11f383ee8: lavf: add return to silence compiler warning [20:17] <kierank> Compnn: rather strange thread [20:18] <kierank> "Not only is Linux the most widely used operating system on the planet" [20:18] <kierank> interesting assertion [20:18] <kierank> i guess if you include embedded [20:18] <Compnn> well, lots of droid phones :P [20:18] <Compnn> microwaves, etc [20:18] <Compnn> yeh [20:18] <Compnn> dvd players [20:19] <kierank> strange rant though [20:19] <kierank> especially the secret ffmpeg [20:19] <Compnn> :D [20:20] <Compnn> lol more comments [20:21] <kierank> mp3 is definitely allowed in mp4 [20:21] <durandal_1707> it is about m4v m4a [20:22] <Compnn> is there really a diff between m4v and mp4 ? [20:22] <Compnn> i thought it was all the same damn thing [20:22] <durandal_1707> and m4v/m4a are only used with ipod muxer [20:22] <Compnn> mj2 mov mp4 m4v m4a m4p etc [20:22] <kierank> m4v/m4a is renamed apple crap [20:22] <durandal_1707> ipod muxer write some extra shit [20:22] <Compnn> ah [20:23] <Yuvi> it's about the same as the difference between mp4 and 3gp [20:23] <Yuvi> different allowed codecs and atoms [20:23] <kierank> oh [20:23] <durandal_1707> the guy thinks that mp4 does not allow multiple a/v streams [20:24] <Compnn> he means ipod wont play multilple stream [20:24] <Compnn> or something similar... [20:24] <kierank> i don't see why the patch supporting dts in mp4 can't be committed [20:24] <durandal_1707> nope, he wants to mux dts-ma+ac3 and play that on ipod [20:24] <kierank> it's standards compliant [20:24] <Compnn> Yuvi : yeah thats what i was thinking. webm vs mkv :P [20:25] <durandal_1707> kierank: dts is already supported in mp4 [20:25] <Compnn> the patch is for muxer only ? [20:25] <durandal_1707> i dunno what apple m4a/m4v crap supports [20:25] <kierank> he's getting confused like me then [20:25] <kierank> between mp4 and m4v [20:27] <kierank> durandal_1707: why is he saying dts isn't allowed by ffmpeg in mp4 [20:27] <kierank> i am confused by this guy [20:28] <durandal_1707> he is confused too, but he is not interested in mp4 at all [20:28] <durandal_1707> only in m4a/m4v [20:29] <ubitux> speaking about mp4, does anyone have in mind the website with the list of all what is allowed in it [20:29] <ubitux> with the different fourcc etc [20:29] <ubitux> ? [20:29] <kierank> ubitux: it's in the ticket [20:29] <kierank> http://www.mp4ra.org/specs.html#MLP [20:29] <ubitux> oh, thx [20:29] <ubitux> exactly what i was looking for [20:30] <durandal_1707> and you know correct fourcc? [20:30] <durandal_1707> and all extra info? [20:30] <durandal_1707> if there is mlp in mp4 sample in wild, would we already know about it? [20:31] <kierank> since we're not going to get the mlp standard any time soon we will have to wait [20:36] <cone-971> ffmpeg.git 03Paul B Mahol 07master:0c8943939eb0: gifdec: use log level that have more common sense [20:38] <cone-971> ffmpeg.git 03Paul B Mahol 07master:48340bbb366a: aiffenc: metadata support [20:43] <llogan> nicholas' tee psuedo-muxer looks interesting. users ask for something like this on occasion. [20:43] <llogan> is he ever in this channel? [20:44] <Compnn> nicolas george ? [20:46] <Compnn> dnt think so [20:47] <llogan> trac has become the new ffmpeg-devel IRC channel. [20:48] <durandal_1707> lock that ticked [20:50] <Compnn> we finally got it out that his files work on google tv [20:50] <Compnn> now we're asking for a sample of the working file :P [20:50] <Compnn> so close to solving the ridddddle [20:50] <Compnn> why does he have to fake out an ac3 track however? [20:52] <durandal_1707> wtf, he use ipod "preset" so that he do not need to use faststart [20:52] <durandal_1707> but he is probably not aware of -movflags faststart [20:57] <Compnn> yes! more comments! [21:00] <durandal_1707> ^no because it cant work always [21:01] <Compnn> orly [21:01] <durandal_1707> if seeking is not possible [21:02] <Compnn> someone broke it [21:02] <Compnn> fate i mean :P [21:02] <durandal_1707> what fails? [21:03] <durandal_1707> shit stupid moronic seek test [21:07] <cone-971> ffmpeg.git 03Paul B Mahol 07master:28d75441ec2e: fate: update seek-lavf-aiff after previous commit [21:08] <durandal_1707> it is pointless because there is bunch of demuxers that use same function for seeking [21:11] <Compnn> durandal_1707 : are you working on evrc? [21:11] <llogan> durandal_1707: did you make that ffmpeg logo for the g+ page? [21:12] <durandal_1707> Compnn: was, i'm on 1787 written lines [21:13] <durandal_1707> still ~200 - ~600 to write [21:13] <Compn> wow :) [21:14] <durandal_1707> llogan: i googled it [21:14] <durandal_1707> file was named ffmpeg so used it [21:16] <Compn> without knowing the copyright!? :P [21:16] <durandal_1707> it is public domain [21:49] <cone-971> ffmpeg.git 03Reimar Döffinger 07master:def01739c136: mxfdec: Set AV_FIELD_PROGRESSIVE, needed for AVC-intra [21:49] <cone-971> ffmpeg.git 03Reimar Döffinger 07master:c5142a95a513: Support more AVC-Intra files [23:27] <saste> ubitux: is there a way to add an offset to a subtitle stream? [23:28] <saste> would it make sense to add an offset option to the subtitles filter? would that be possible at all? [23:48] <cone-971> ffmpeg.git 03Michael Niedermayer 07master:9e36d9e4ed2a: pngdec: fix decoding of right column for 2/4bpp [00:00] --- Fri Jan 4 2013

1 0

ffmpeg.log.20130103
by burek 04 Jan '13

04 Jan '13

[02:39] <abara> hey guys [02:40] <abara> someone knows how can i convert .mkv to .avi? [02:40] <sacarasc> The easiest way is ffmpeg -i input.mkv output.avi [02:41] <sacarasc> You'd have to give more details to get a proper command line. [02:42] <abara> the mkv file is a bluray rip, 1080p? and i want to play this avi at my panasonic tv :D [02:42] <abara> with .srt :D [02:42] <sacarasc> You'd have to give more details than that. Like what the TV supports. [02:43] <abara> ok [02:44] <abara> let me check manual [02:47] <abara> ah [02:48] <abara> my tv is a panasonic model TC-L32U5B [02:48] <abara> im looking at codecs there are supported [02:49] <abara> codecs for avi : H.264 BP/MP/HP and MPEG4 [02:50] <abara> codecs for mkv : H.264 BP/MP [02:51] <abara> sacarasc: .-. [02:53] <sacarasc> And for audio? And subtitles? [02:54] <sacarasc> And bitrates? [02:54] <sacarasc> Also, resolutions. [02:56] <abara> audio codecs are the same for both : mp3, mpeg layer 1/2, dolby digital, aac, he-acc [02:56] <abara> he-aac* [02:57] <abara> resolution for this codecs are not especified [02:57] <abara> but tv is fullhd [02:58] <sacarasc> What audio does the input file have? [02:59] <abara> http://pastebin.com/8BgHXBMd [02:59] <abara> mediainfo output for this mkv file :D [03:01] <sacarasc> ffmpeg -i illegallydownloadedmovie.mkv -c:v libx264 -profile:v main -preset:v slow -crf 22 -c:a copy -c:s srt output.mkv [03:01] <sacarasc> That might work, but I am a little tired and so not fully with it at the moment. [03:01] <abara> sacarasc: ok bro! no problem :D [03:01] <abara> i'll try it :D [08:17] <wieshka> i am streaming as pirmary output RTMP stream from my Blackmagic Decklink capture card with FFMmpeg, as secondary output i would like to have each 30 sec output of image for picture analysis. Having constant framerate (25) [09:40] <sesam> So guys, what's your workflow with videofiles and ffmpeg? [09:41] <ubitux> alias ffmpeg=rm [09:45] <sesam> any post-house guys here? [10:27] <sesam> So guys, what's your workflow with videofiles and ffmpeg? [10:34] <saste> sesam: it's like asking a baker, so what's your workflow with bread? [10:57] <torbjorn> can I use ffmpeg to just report audio information on a media file? [10:58] <torbjorn> or video info too for that matter [11:41] <kodaws> so what happens when a company sells you their product (closed source video app) that contains a modified version of ffmpeg libraries? [11:42] <JEEB> if it's an LGPL ffmpeg, you can ask them for the sources of that ffmpeg [11:42] <JEEB> if it's a GPL ffmpeg, you can ask for their full sources for the thing that links to it :P [11:42] <JEEB> (that is with dlls/dynamic linking) [11:43] <kodaws> umh how can i tell if it's gpl or lgpl? [11:43] <kodaws> and what happens if they say no? [11:43] <JEEB> strings some.dll > derp.txt [11:43] <JEEB> then crawl through the derp.txt [11:44] <JEEB> or well, since ffmpeg can either be GPL or LGPL [11:44] <JEEB> strings some.dll |grep -i -C3 "GPL" > derp.txt [11:59] <kodaws> libavcodec license: LGPL version 2.1 or later [11:59] <kodaws> well good enought [11:59] <kodaws> -t [11:59] <JEEB> yeah, seems to be doing it the Right Way [12:00] <JEEB> (I've seen plenty of people fail and then run away (and possibly change it to be correct for newer versions) [12:01] <kodaws> well not really right, they should be providing the sources for their modified version [12:01] <JEEB> yes, naturally [12:02] <JEEB> (well, they'd really be wrong if they said "no u" if you asked them for it, it is OK to have someone actually ask for it) [12:02] <JEEB> although IIRC LGPL makes you note somewhere that you are using it [12:02] <JEEB> not sure of the exact wording [12:03] <JEEB> which reminds me I asked for the source code for my zyxel [12:03] <JEEB> still haven't gotten a response [12:04] <kodaws> yes they should be saying "we're using ffmpeg" somewhere [12:04] <kodaws> and they aren't -.- [12:04] <klaxa> don't worry, apple will sue them [12:05] <klaxa> meh... this joke has gone too far [12:05] <klaxa> it's not doing it for me anymore [12:06] <kodaws> i am wondering... [12:07] <kodaws> since they are kind enough to provide the dlls [12:07] <kodaws> can't them be used again with some ffmpeg version? [12:17] <kodaws> actually since it's (l)gpl they can be easily reverse engineered [12:17] <kodaws> decompilation to the rescue! [12:17] <kodaws> s/easily/allowed to be/ [13:36] <Fjorgynn> Good afternoon [15:55] <_dr> hi, can someone tell me what "auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out'" means? what does the scale 0 filter do? [16:01] <saste> _dr: convert formats, so that the destination filter can processe the frames from the input filter [16:27] <Nick-S> hey guys, straighy question, does image2 supports .psd?(photoshop) [17:29] <Sashmo_> has anyone used the sharpen filter with ffmpeg?? I cant see to make it work [17:37] <Sashmo_> looking for a PAL to NTSC specialist....... [17:37] <Sashmo_> I'm giving up on my trial and error :p [17:38] <Mavrik> hmm [17:38] <Mavrik> what are you trying to achieve? [17:38] <Sashmo_> I'm converting interlaced 25fps PAL, to either interlaced or non interlaced 30fps NTSC [17:39] <Sashmo_> and the video output on the left to right motion/text is choppy [17:39] <Sashmo_> and fuzzy [17:39] <Mavrik> hmm [17:39] <Mavrik> that's probably from deinterlacing [17:40] <mpfundstein_home> Sashmo_: what i learned from ntsc to pal is: dont use ffmpeg for it. better check avisynth and get some scripts on doom9. [17:41] <Mavrik> that too :) [17:41] <Sashmo_> hmm, interesting, so for live conversion, I have to pipe? [17:41] <mpfundstein_home> Sashmo_: ffmpeg does just duplicate frames, thats where your stuttering come from [17:41] <Sashmo_> mpfundstein_home: is there a work around? [17:41] <mpfundstein_home> Sashmo_: Oh, pipe and avisynth, i dont know how that works sry. i use it from whole files only (until now) [17:42] <mpfundstein_home> Sashmo_: but i pipe avisynth output in ffmpeg to get mp4's [17:42] <Sashmo_> mpfundstein_home: and the quality is good? [17:42] <Sashmo_> on a scale from 1 to 10 ? [17:42] <mpfundstein_home> Sashmo_: the best until now. i tried month to get it right. [17:42] <Sashmo_> care to share your settings? [17:42] <mpfundstein_home> Sashmo_: But the very best quality i achived by using hardware [17:43] <Sashmo_> snell wilcox [17:43] <Sashmo_> cant afford [17:43] <Sashmo_> :( [17:43] <mpfundstein_home> Sashmo_: We neither, we had a trial from the terranex :D [17:43] <mpfundstein_home> no problem to share my settings. but lets do it in private. and later. im off in a bit [17:43] <Sashmo_> that would be great [17:44] <Sashmo_> thank you [17:44] <Sashmo_> just PM me, im always on line [17:44] <Sashmo_> hang on teranex is BM now? [17:44] <Sashmo_> wow...... I didnt see that one coming..... [17:44] <mpfundstein_home> yeah [17:45] <mpfundstein_home> the Teranex had such an incredible conversion quality from ntsc to pal (i dont know about the other way around). it was shocking [17:45] <Sashmo_> wow.... [17:45] <Sashmo_> I use all the BM stuff now [17:45] <Sashmo_> I was considering this [17:45] <Sashmo_> but [17:46] <Sashmo_> still expensive for what Im doing [17:46] <mpfundstein_home> but ajk from #avisynth actually made me some handcrafted scripts for each video reference and the results were great too [17:46] <mpfundstein_home> he is really helpful, maybe you can ask him more [17:46] <Sashmo_> that would be great to play with, so see what you have done, you seem to know your stuff [17:46] <mpfundstein_home> well [17:46] <mpfundstein_home> ima noob still [17:46] <mpfundstein_home> im in broadcasting for 6 month now. and i learn everyday. [17:47] <Sashmo_> haha just like me [17:47] <mpfundstein_home> 6 month ago i wrote my first ffmpeg script :D [17:47] <Sashmo_> where do you work? [17:47] <Sashmo_> what type of scripts? [17:47] <mpfundstein_home> Xite in Amsterdam. A music channel with its own it department [17:47] <Sashmo_> I'm using all the BM sdi stuff for custom palyout [17:47] <mpfundstein_home> my main job is to build the backend transcoding pipelines, so that the marketing team can easily decode and deploy videos [17:48] <Sashmo_> we should chat [17:48] <mpfundstein_home> this week i learned that batch transcoding sucks. [17:48] <mpfundstein_home> :D [17:48] <Sashmo_> LOL [17:48] <Sashmo_> I rolld my own UDP transfer software [17:48] <divVerent> I wonder BTW [17:48] <Sashmo_> to transmit UDP transport streams around the world [17:48] <divVerent> is there even ANY good way to transcode PAL/NTSC? [17:48] <mpfundstein_home> oh niiiceee [17:48] <divVerent> the frame rates differ just too much [17:49] <mpfundstein_home> divVerent: well one thing i found out is, most NTSC stuff is actually Film with one frame added [17:49] <divVerent> the only somewhat ok way seems to slow down 25fps->24fps in PAL->NTSC [17:49] <divVerent> then if needed, telecine to 30fps [17:49] <divVerent> but when keeping the speed, you just can do wrong [17:49] <divVerent> this is BTW why PAL versions of movies tend to be shorter [17:49] <mpfundstein_home> Sashmo_: transport streams? mpegTS? [17:50] <divVerent> it's not the censorship, it's the NTSC -> PAL speedup [17:50] <divVerent> and when doing the speeding up, the hard part is the audio :P as you don't want to change pitch, you also don't want pitch shift artifacts [17:51] <divVerent> German movie versions have it easy, they are dubbed anyway, can dub to the right speed, problem solved [17:51] <mpfundstein_home> divVerent: i am working with music videos, no chance to speed it up [17:51] <divVerent> hehe [17:51] <divVerent> the problem is, there is absolutely no way to handle the frame rate right [17:52] <divVerent> some do a weird telecine variant for 23.98 -> 25 fps conversion [17:52] <divVerent> a 2222222222...2223 telecine [17:52] <divVerent> that still stutters, but a lot less than when doing it for whole frames [17:52] <divVerent> wikipedia on telecine explains some of this, and mentions the patterns people use [17:53] <divVerent> "The method used is 2:2:2:2:2:2:2:2:2:2:2:3 (Euro) pulldown (see below[citation needed])." [17:53] <divVerent> you may have no choice but to do this [17:54] <divVerent> which of course makes the video then play badly on non interlaced outputs :P [17:55] <Sashmo_> divVerent: so what do you suggest? [18:54] <Fjorgynn> is -f rawvideo the best thing to do pass 1 when you do two pass? [18:58] <xintron> I have a few 720p videos I want to scale down and compress to put on my phone (Samsung Galaxy S 2, resolution 480x800). What would be a good starting point for this where quality is important but filesize < 150MB would be great? [19:02] <Fjorgynn> Try something like ffmpeg -i input.mp4 -c:v libx264 -crf 22 -s hd480 -c:a libvo_aacenc -b:a 128k output.mp4 [19:02] <Fjorgynn> xintron: [19:05] <xintron> Fjorgynn, ah, missing libvo_aacenc it seems [19:06] <Fjorgynn> xintron: don't you have the last ffmpeg? ;) [19:06] <Fjorgynn> latest [19:07] <Fjorgynn> libfaac works if you have Linux [19:07] <Fjorgynn> I think [19:07] <JEEB> if you want to have the best aac encoder, then use fdk-aac :V [19:07] <JEEB> vo_aacenc is on the level of the internal aac encoder [19:09] <Fjorgynn> I don't know if -s hd480 should be -vf scale=480:720 [19:09] <Fjorgynn> JEEB: do you know? [19:10] <Fjorgynn> instead of hd480 I mean [19:10] <JEEB> uhh, no idea what you mean with the latter :V [19:10] <klaxa> 480p is actually 854x480, 720x480 isn't 16:9 [19:11] <Fjorgynn> I see [19:11] <Fjorgynn> klaxa: but the -s vs the -vf scale? [19:11] <klaxa> dunno i think i'd use -s [19:11] <Fjorgynn> ok [19:12] <Fjorgynn> then -s hd480 is fine [19:12] <klaxa> but i don't know if the scale video filter is better quality wise [19:12] <Mavrik> you get scale video filter in any case [19:12] <Mavrik> (if your ffmpeg is new enough) [19:13] <Fjorgynn> I have no idea. I learned about hd480 yesterday xD [19:15] <xintron> Trying some different alternatives and checking the output on the phone to see which is good enough (contra filesize) [19:16] <Fjorgynn> also you can lower the bitrate instead of doing a -crf [19:16] <Fjorgynn> set the bitrate manually. [19:17] <xintron> yeah, tried -crf 22 and 28 and 28 works fine on the phone. How can I compress the audio (audio quality isn't that important for these videos)? [19:18] <JEEB> -ac 1 (make it mono) and bring down -b:a from the 128k [19:18] <Fjorgynn> so that's what ac does [19:19] <Fjorgynn> ^^ [19:23] <Fjorgynn> -preset slow will make the bitrate smaller I think. But it takes long time [19:24] <Fjorgynn> filesize* [19:24] <sesam> anyone uses a ffmpeg cluster? [19:24] <Fjorgynn> inte jag [19:24] <sesam> hepp [19:24] <Fjorgynn> stay in here, someone probably does [19:24] <Mavrik> Fjorgynn, preset doesn't really change bitrate [19:25] <sesam> vad använder du ffmpeg till? inom jobbet? [19:25] <Mavrik> it just increases the quality at the same bitrate [19:25] <emerica_> MEC but i think dev slowed/stopped [19:25] <emerica_> http://www.encodingcluster.com/trac/ [19:25] <Fjorgynn> Mavrik: I added the -preset slow and went from 8,68 MB to 8,3 MB [19:26] <klaxa> recently i encoded a 1080p 90 seconds video with placebo preset [19:26] <Fjorgynn> 23 second clip [19:26] <klaxa> took 3h 42m [19:26] <sesam> emerica_: thx [19:27] <Fjorgynn> klaxa: xD [19:28] <Mavrik> Fjorgynn, it was a side effect, it's not a rule to do that [19:28] <Mavrik> it only keeps to set bitrate better [19:28] <Fjorgynn> I see [19:29] <Fjorgynn> how much slower is -preset slow? [19:29] <klaxa> video looks very good, filesize was decreased slightly from original, original was high 10 profile, encoded to high profile [19:30] <Fjorgynn> klaxa: and it was woth 3,5 hrs? [19:30] <klaxa> absolutely :P [19:30] <JEEB> Fjorgynn, you should encode 2500-5000 frames or so and test per preset :V [19:30] <JEEB> to see which preset is the slowest that's still OK for you [19:30] <JEEB> and 2500-5000 frames shouldn't be too much [19:31] <JEEB> use -ss to seek somewhere and -t to encode ~10-20 seconds [19:32] <klaxa> Fjorgynn: lemme just upload both files... [19:32] <klaxa> may take a few mins, i'll come back to you [19:32] <Fjorgynn> nah [19:33] <divVerent> 17:55:52 Sashmo_ | divVerent: so what do you suggest? [19:34] <divVerent> Sashmo_: if you can't edit the video manually, use this "Euro Pulldown" as described on Wikipedia [19:34] <divVerent> if you CAN edit it, though [19:34] <divVerent> a better choice may be manually deleting frames before and after all those scene cuts in the video [19:34] <divVerent> so the beats match up [19:34] <divVerent> to achieve a better change from 25fps to 23.98fps [19:34] <divVerent> and then using the standard 3:2 pulldown [19:35] <divVerent> but before you do any of this - is your input footage really 25 progressive frames? [19:35] <divVerent> if it is ALREADY europulldown'd, you may rather want to invert that process and get back the original 24fps [19:36] <divVerent> also... don't actually DO the 3:2 pulldown, encode at 23.98fps and set the necessary flags ("RFF") in the MPEG-2 stream to let the DVD player do it [19:39] <klaxa> Fjorgynn: in case you are interested regardless: http://klaxa.in/Renai_Circulation_orig.mkv and http://klaxa.in/Renai_Circulation.mkv [19:52] <Fjorgynn> :) [19:53] <Fjorgynn> hm [19:56] <Fjorgynn> xintron: how's it going? [20:24] <creep> hahaha what is this film? http://static1.fjcdn.com/thumbnails/comments/Didn+t+both+of+them+died+alrea… [20:25] <leoj3n> dumb and dumber? @creep [20:25] <creep> thanks i must watch that ^^ [20:38] <crolab> i'm trying and only get sound - is it my system or the file? ffplay -i -x 640 -y 350 http://s01.webport.tv/ilive/z050236.stream/playlist.m3u8?wowzasessionid=123… [21:13] <seablade_mbp> Ok, tougher question for me at least. I am trying to duplicate the results in this threadhttp://atemuser.com/forums/atem-vision-mixers/blackmagic-atems/atem-tvs-and… [21:13] <seablade_mbp> I have the encoder set up and streaming via USB the stream, which I believe is just the raw h.264 video and AAC audio if I understand the thread correct [21:14] <seablade_mbp> I attempt to connect to the stream via `echo "receive -id 1 -transport tcp" | nc -v -v 127.0.0.1 13823 | ffmpeg -i - -vcodec copy -acodec copy test.ts` [21:14] <seablade_mbp> And I get the error 'pipe:: Invalid data found when processing input' [21:14] <seablade_mbp> Is this possible, or am I screwing something up very basic "again" ;) [21:18] <seablade_mbp> fflogger, You pretty much have it [21:29] <crolab> seablade_mbp: that is their way of making believing they help [21:36] <crolab> iive might be the only one seems to know things here :( [21:43] <llogan> without any context we can only make guesses. [21:44] <llogan> since you did not provide much information there is little to work with and my crystal ball is broken [21:46] <crolab> llogan: i think both of us gave more then enough info - why did you bother with this last line? [21:46] <llogan> i only see commands, but i've only been here for less then 2 hours so maybe i missed something [21:49] <crolab> lol i think we both would love to have someone _else_ run the commands and see if they _also_ get the mentioned problems [21:50] <crolab> those commands are giving us problems - and what more is there besides the commands to work with? [21:54] <llogan> the complete console output shows useful information. this is a standard request. [21:54] <llogan> but you expect answers with little investment [21:54] <crolab> don't you get any console output on your test? [21:55] <llogan> it doesn't matter what I'm doing. you're the one experiencing issues. [21:55] <crolab> your are kidding right? [21:55] <llogan> nevermind. good luck. [21:56] <crolab> do you even have ffmpeg installed? [21:59] <crolab> for the next time you _want_ to make believe you can be a helper - do you want a link to how to cutnpaste commands to cli? [22:00] <crolab> well it works with vlc so more need for your expert guidance [22:03] <seablade_mbp> llogan, In my case the only difference in the standard output is the standard build information. There is no other output [22:03] <seablade_mbp> llogan, http://pastebin.com/w2a10S0R [22:04] <seablade_mbp> llogan, For reference [22:04] <seablade_mbp> llogan, I wasn't joking when I said you pretty much have it [22:04] <llogan> it shows that you're not using something 3 years old with can often be the case. [22:04] <seablade_mbp> In my case as well I don't expect many people will be able to do what I am doing specifically as it requires specific hardware in order to do so [22:05] <seablade_mbp> llogan, A simple question to answer yes, but even if it was something 3 years old, there isn't much clue as to what is causing the issue that I see. Either way moot point at this time [22:06] <llogan> since development is so active the first step is to always see the version [22:07] <seablade_mbp> Also for the record, running it with -v debug only adds a single line of output just before the error, '[AVIOContext @ 0x7fd568c152c0] Statistics: 54 bytes read, 0 seeks' [22:14] <llogan> seablade_mbp: what are your input formats? what are you doing, exactly? [22:18] <seablade_> llogan, The input encoding is h.264/aac [22:18] <seablade_> llogan, I believe it is MPEG2 TS [22:19] <seablade_> llogan, The end result is to take it and restream it out to a streaming server in the cloud, or that is the goal [22:19] <seablade_> llogan, But for the time being I just want to get ffmpeg to write it to a file so that I Know ffmpeg can handle it at all [22:20] <seablade_> llogan, The stream I am trying to read/restream is generated by a hardware h.264 encoder, and is realtime [22:22] <seablade_> llogan, The scripts mentioned in the thread I linked to are Perl scripts and write the stream to a file that is an MPEG2 TS encapsulated h.264/aac stream. The thread then uses the file to restream via ffmpeg but inputting the file as it is being recorded, I am trying to skip the file step if I can as it seems to make more sense to me to do it that way [22:23] <llogan> if you output to a file instead can ffmpeg decode it properly? [22:23] <seablade_> llogan, That is the impression I got from that thread, I can check again in a few minutes(In my office away from that computer at the moment) [22:51] <seablade_mbp> llogan, Good catch '[mp4 @ 0x7fa2d9828600] malformated aac bitstream, use -absf aac_adtstoasc [22:51] <seablade_mbp> av_interleaved_write_frame(): Operation not permitted' [22:53] <seablade_mbp> llogan, Sadly when processing the live stream that flag did not make a difference. Adding the flag to the command line for the recorded file however allows ffmpeg to process it fine [22:54] <seablade_mbp> llogan, So short version is yes ffmpeg can process the recorded file fine from what it seems [22:54] <seablade_mbp> llogan, And assuming that I can read perl halfway decent, the recorded file is literally just a stream dump of the stream that I am trying to feed to ffmpeg [22:55] Action: seablade_mbp prefers python, never bothered much with perl [23:12] <espr3ss0> Q: ffmpeg "./configure", "pkg-config not found" ? [23:13] <klaxa> espr3ss0: you are missing pkg-config [23:13] <klaxa> install it, that should fix it? [23:16] <espr3ss0> I downloaded "freedesktop.org - Software/pkg-config" [23:16] <espr3ss0> but no idea how to install it o_O [23:17] Action: Fjorgynn is upscaling [23:19] <klaxa> espr3ss0: what distribution are you running? [23:19] <espr3ss0> I used "git clone git://anongit.freedesktop.org/pkg-config pkg-config" [23:19] <espr3ss0> not sure which version [23:19] <klaxa> just install the pkg-config provided by your distro's repos [23:24] <espr3ss0> says glib not found [23:27] <espr3ss0> is there no version of ffmpeg to skip all this ? [23:28] <klaxa> if you want to compile from source, no [23:28] <klaxa> otherwise there are static builds [23:28] <espr3ss0> will probably make life easier [23:28] <klaxa> http://ffmpeg.gusari.org/static/ [23:29] <espr3ss0> i just went with "git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg" [23:30] <klaxa> espr3ss0: then you get the source code to compile it [23:30] <Sashmo_> ok I give up!...... I will pay someone to find me the best possible command line solution to convert from PAL to NTSC from a UDP interlaced PAL source....... Anyone??? [23:33] <espr3ss0> ok, thanks klaxa [00:00] --- Fri Jan 4 2013

1 0

ffmpeg-devel.log.20130102
by burek 03 Jan '13

03 Jan '13

[00:12] <cone-159> ffmpeg.git 03Nicolas George 07master:102cf964ddc9: ffmpeg: sub2video: set resample size. [00:12] <cone-159> ffmpeg.git 03Nicolas George 07master:98ce9b84681b: fate: merge mapchan and options into ffmpeg. [01:24] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:3a6b6f425ee3: lavf: move force_codec_ids() up [01:24] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:341e40f1e1ee: lavf: Fix codec id forcing with probed codecs [02:38] <cone-159> ffmpeg.git 03Janne Grunau 07master:e9fd51b0d6dc: h264: check SPS entries directly to detect pixel format changes [02:43] <cone-159> ffmpeg.git 03Michael Niedermayer 07release/0.11:5502b073ec29: lavf: factor codec id forcing out [02:43] <cone-159> ffmpeg.git 03Michael Niedermayer 07release/0.11:bf19d4c6fa12: lavf: Fix codec id forcing with probed codecs [02:43] <cone-159> ffmpeg.git 03Michael Niedermayer 07release/1.0:bc1e72c38ead: lavf: move force_codec_ids() up [02:43] <cone-159> ffmpeg.git 03Michael Niedermayer 07release/1.0:ff08767817c2: lavf: Fix codec id forcing with probed codecs [04:03] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:1e28fa21de23: rawdec: fix NV12 [04:07] <Compn> ubitux : ever talk to virtualdub, see if they are going to incorporate ffmpeg filters? :P [09:34] <cone-140> ffmpeg.git 03Clément BSsch 07master:5a2f3f0bca5a: lavf/vobsub: do not count trailing NUL char in extradata. [09:58] <ubitux> does anyone know if it is possible to be put in cc automatically in the trac when participating to a ticket? [09:58] <ubitux> i sometimes forget to follow a ticket after asking or saying something [10:34] <cone-140> ffmpeg.git 03Clément BSsch 07master:8bc74221f8ef: lavf: remove generic index flag from text subtitles. [11:58] <ubitux> what is this incremental cluster parsing in matroska? [11:58] <ubitux> /* File has SSA subtitles which prevent incremental cluster parsing. */ [11:58] <ubitux> int contains_ssa; [11:58] <ubitux> any hint about this? [12:06] <nevcairiel> ubitux: its a trick to reduce latency on parsing, where it doesnt parse the full cluster but only as much as it needs to for the next packet, but that has some issues with ass subs apparently [12:07] <nevcairiel> it greatly complicates the whole parser for a minor latency gain on seeks, but oh well :) [12:13] <ubitux> ok [12:13] <ubitux> thx :p [12:36] <burek> Happy New Year guys :) [12:39] <ubitux> hey burek [12:39] <ubitux> burek: how is the new fate going on? [12:40] <burek> well, i paused it since then, when i asked if anyone can help about those logs [12:40] <burek> because im not quite familiar with git and stuff [13:08] <michaelni> burek, what do you need from git ? [13:10] <burek> well, to be more precise, I need directions on what the exact output should be for each link clicked on the fate [13:11] <burek> for example, it's not helpful to say "see on old fate and just copy/paste" because some diffs I just don't understand why they were created at all [13:12] <burek> so, it would be the best if someone could tell me (besides configure.log and compile.log) what the other outputs should look like and what is their purpose [13:12] <burek> so I could optimize things properly [13:17] <ubitux> you mean the compile diff? [13:17] <ubitux> it's used to spot new warnings [13:17] <burek> or, let me simplify this more.. given the configure.log, compile.log and tests.log (for each fate client upload), what are the outputs that developers need [13:18] <ubitux> compile.log is the only one where we need a diff afaik (and that's the only one where it's done iirc) [13:18] <burek> i mean, don't get me wrong, it might be obvious to you, but it's not that obvious for me.. [13:18] <ubitux> configure.log and tests.log are useful to debug, but no need to diff [13:19] <ubitux> and the diff for compile.log, as i said, is just for spotting new warnings [13:19] <ubitux> look in the current fate for details on the diff: if it's red, new warnings, if it's green, less warning, if it's blue, nothing changed [13:20] <ubitux> i think there is also a "bold" attribute [13:20] <ubitux> for cases like one less warning, and one more [13:20] <ubitux> also note that diff can be relatively huge because of threading compilation [13:20] <ubitux> (warnings don't appear in the same place) [13:21] <nevcairiel> yeah the diffs are somewhat useless with threaded compilation [13:21] <burek> yes, i noticed that [13:21] <ubitux> |sort! [13:21] <ubitux> sorted warnings [13:21] <ubitux> (that is a joke, don't do that) [13:21] <burek> :) [13:22] <ubitux> (also note that a new line in the compile.log does not mean a warning, it can be a new file) [13:23] <burek> you mean in a diff? [13:23] <ubitux> yes [13:24] <burek> i'm thinking, you might even be right about |sort :) [13:24] <ubitux> no [13:24] <ubitux> what you can do is filtering only the warnings from the compile.log [13:25] <ubitux> and do a diff on that output [13:25] <ubitux> (which you could sort, with great care about multiline warnings) [13:25] <ubitux> (sort before diff of course) [13:25] <ubitux> BUT [13:26] <burek> well, yes, if the point is to spot new warnings, it might be enough to just sort / diff and later ctrl+F for that warning [13:26] <ubitux> warnings output differs between compiler [13:26] <ubitux> so you need a very smart parser :) [13:27] <burek> btw, is there a way i could get a copy of the directory on current fate server (just the part with uploaded client stuff) [13:27] <burek> so that I can have real data to test the outputs on [13:35] <michaelni> burek, that directory is big [13:36] <burek> hm [13:36] <burek> how big [13:36] <michaelni> i started a "du -m" [13:38] <burek> oh yes, one more question, i've noticed that history for one fate client goes back to 15 and more days [13:38] <burek> what is that used for? [13:45] <ubitux> burek: useful for --enable-random, or the threading failing test for example [13:45] <ubitux> or anything that "recently" (on a time scale of 15 days) failed [13:58] <burek> > [mpegts @ 0x9e16580] first pts value must set [13:59] <burek> is this spell checked [14:00] <ubitux> yes, missing a "be" [14:03] <burek> i'm not sure if this is a bug or not: http://pastebin.com/qjBu2BeQ [14:03] <burek> but it looks like ffmpeg can't remux h264 into ts [14:04] <michaelni> burek, 23gb [14:04] <burek> nice :) [14:04] <burek> is there a way i could get just one subdir (1 fate user)? :) [14:05] <michaelni> probably [14:05] <michaelni> any specific one you want ? [14:05] <burek> no preferences really [16:34] <cone-140> ffmpeg.git 03Stefano Sabatini 07master:fd44dfb29dc7: doc/muxers: fix typos in the segment chapter [16:34] <cone-140> ffmpeg.git 03Stefano Sabatini 07master:8bbe9d90fa47: doc/muxers: add a dedicated section for segment examples [16:34] <cone-140> ffmpeg.git 03Stefano Sabatini 07master:82deb0c42e5b: doc/muxers: adopt new -codec:SPEC syntax in segment example [17:13] <cone-140> ffmpeg.git 03Nicolas George 07master:b99bef17b44c: lavfi/avfiltergraph: check pick_format return code. [17:44] <cone-140> ffmpeg.git 03Nicolas George 07master:e4f14c32b95a: examples/muxing: improve error messages. [18:03] <cone-140> ffmpeg.git 03Nicolas George 07master:b252d9e77763: fate: add sub2video test. [19:35] <wm4> saste, ubitux: have there been any plans to multithread lavfi? [19:42] <durandal11707> wtf is ipod muxer and how it relates to m4a/m4v [19:44] <durandal11707> it is just different extensions [19:45] <nevcairiel> its all mp4 files, apple just likes naming it differently sometimes for audio (m4a) and video (m4v) [19:45] <durandal11707> so what is point in having ipod muxer? [19:46] <michaelni> ubitux, saste sub2video test fails http://fate.ffmpeg.org/report.cgi?time=20130102174823&slot=x86-opensolaris-… [19:48] <nevcairiel> ipod only supports h264/aac or something? tell them to use a mp4 file with the mp4 muxer :P [19:48] <michaelni> also http://fate.ffmpeg.org/report.cgi?time=20130101175054&slot=powerpc-linux-gn… [19:48] <durandal11707> nevcairiel: he wants m4a/m4v [19:48] <nevcairiel> but its the same format [19:49] <nevcairiel> just a different name [19:49] <durandal11707> i know that already [19:51] <durandal11707> anyway adding ipod/ipad/crap muxers is brain dead idea [19:51] <llogan> ask bcoudurier [19:53] <durandal11707> hmm ipod muxer write different stuff to file [19:53] <durandal11707> it is not plain extension thing [19:54] <durandal11707> so i'm totally lost [19:55] <llogan> "Write uuid atom: Needed to make file play in iPods running newest firmware"...although i've never had to use it to get video working on the few iDevices i've tested. [19:56] <durandal11707> does ipod supports dts? [19:59] <llogan> if you have a sample i can maybe test [20:02] <durandal11707> you can create one with ffmpeg [20:16] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:ccb7f203099e: fate: add missing bitexact flag to scale filter [20:44] <durandal11707> http://www.vapoursynth.com/2012/12/positive-thinking-how-to-convince-yourse… <<<< lol [21:32] <nevcairiel> there is some failure here [21:33] <ubitux> arg [21:35] <ubitux> seems the bitexact flag wasn't enough [21:40] <ubitux> maybe there is an automatic scale inserted [21:43] <nevcairiel> shouldnt the log say that [21:44] <ubitux> [auto-inserted scaler 0 @ 0x23f32a0] w:720 h:480 fmt:yuv420p sar:0/1 -> w:720 h:480 fmt:yuva420p sar:0/1 flags:0x2 [21:44] <ubitux> ow. [21:46] <ubitux> so overlay only supports yuv420p for main, but not for overlay [21:47] <ubitux> ah right, the overlay needs a transparency layer all the time [21:47] <ubitux> fun [21:59] <ubitux> beastd: btw, what happened to the better resampling settings? [21:59] <beastd> hi [22:00] <beastd> happy new year [22:00] <beastd> ubitux: about the resampling settings [22:01] <beastd> i need to figure out fate before. a lot of tests get changed. seems like some sample that is created is used in lots of other tests or something like that. [22:01] <ubitux> sure [22:01] <ubitux> make -k|patch -p1 [22:02] <ubitux> make -k fate|patch -p1 [22:02] <beastd> ubitux: nice trick. haven't thought of that [22:03] <ubitux> doesn't come from me :p [22:04] <michaelni> ubitux, theres also a inetger oveflow caused by subtitle.c [22:04] <beastd> still i want to make sure that i understand why so many tests change. it is a bad idea to update regression test references carelessly; and i need to grep documentation too [22:04] <michaelni> i do have a fix for the overflow already [22:05] <michaelni> will push in a moment [22:05] <michaelni> unless you want to see/review ? [22:05] <llogan> we'll fix it in post [22:06] <ubitux> michaelni: i don't mind looking at it right now [22:06] <ubitux> is it causing problems? [22:06] <ubitux> or that's from coverity or something? [22:08] <michaelni> ubitux, http://pastebin.com/FB57sadL [22:08] <michaelni> ubitux, fate client failure [22:09] <michaelni> this here was what found it: http://fate.ffmpeg.org/report.cgi?time=20130102195020&slot=x86_64-freebsd8.… [22:09] <ubitux> oO [22:10] <ubitux> illegal instruction? because of this? [22:10] <michaelni> :) [22:10] <ubitux> nit: please add a space after the if [22:10] <beastd> that sounds bad (like control of instruction pointer) [22:12] <ubitux> oh [22:12] <ubitux> i guess that's because it's doing some magic in the av_rescale_rnd [22:13] <ubitux> michaelni: av_rescale_rnd seems to be checking for INT64_MIN [22:13] <ubitux> shouldn't such kind of test added to that function? [22:13] <michaelni> what test ? [22:14] <michaelni> what function and what should it do ? [22:14] <ubitux> i see a "a != INT64_MIN" [22:14] <michaelni> yes the code cant handle INT64_MIN [22:14] <ubitux> might be completely unrelated though [22:15] <ubitux> well, if there is a check already, that's where the check for INT64_MAX would belong [22:15] <ubitux> no? [22:16] <ubitux> i don't think the caller should do such sanity check (it's not like it's a div by zero or something) [22:16] <ubitux> it's more like an implementation specific issue in that function [22:16] <michaelni> no [22:16] <ubitux> so IMO the check belongs there [22:16] <ubitux> what am i missing? [22:16] <michaelni> INT64_MIN * 90 / 50 gives what ? [22:17] <nevcairiel> EOVERFLOW :P [22:17] <ubitux> an overflow? [22:17] <michaelni> yes [22:18] <nevcairiel> * 50 / 90 would however work [22:18] <michaelni> yes [22:18] <michaelni> not with INT64_MIN and the current code though [22:18] <michaelni> i could fix that but [22:18] <ubitux> michaelni: which one is causing the illegal instruction? [22:18] <michaelni> it wouldnt avoid the need for checks outside that way [22:19] <michaelni> ubitux, clang [22:19] <ubitux> INT64_MIN or INT64_MAX? [22:19] <michaelni> MIN [22:19] <ubitux> then what is the INT64_MIN check for in the rescale function? [22:19] <ubitux> if it's not to prevent such thing [22:20] <michaelni> -INT64_MIN == INT64_MIN [22:20] <michaelni> the code handles negative by -a [22:20] <michaelni> so it cant handle INT64_MIN taht way [22:22] <ubitux> i'm still a bit concerned about adding such test in the caller [22:23] <ubitux> it sounds like an issue which could be present in a lot of places [22:24] <michaelni> we could add a ==INT64_MIN) return INT64_MIN [22:25] <michaelni> but that would be "wrong" if someone really wanted to rescale that [22:28] <cbsrobot> wow @ https://ffmpeg.org/trac/ffmpeg/ticket/2101 [22:28] <ubitux> right now, if we seek to INT64_MIN, it will fail the same way, no? [22:28] <ubitux> cbsrobot: oh sounds fun. [22:29] <michaelni> seeking to INT64_MIN will surely fail dunno how exactly [22:31] <ubitux> i'm not sure where to put the check [22:31] <michaelni> neither am i [22:31] <michaelni> but i want to put it somewhere :) [22:31] <michaelni> nowhere is bad [22:31] <michaelni> either its in the callers or rescale passes MIN and MAX through [22:32] <michaelni> if its in the callers then rescale can be made to rescale MIN correctly [22:32] <michaelni> if its in rescale then things may be simpler but rescale cant rescale MIN/MAX [22:33] <michaelni> maybe we can make rescale scale down correctly and limit on upscale [22:33] <michaelni> but still this wont fix all callers [22:33] <michaelni> AV_NOPTS_VALUE must not be scaled [22:34] <michaelni> so either check outside or passthrough inside rescale [22:35] <ubitux> the AV_NOPTS_VALUE could be check in the seek caller [22:36] <ubitux> in the av seek function, before the callback [22:38] <ubitux> also, since the if stream_index==-1 case looks like a code that could be common to a lot of demuxer seeking callback [22:38] <ubitux> i'd better make some kind of lavf utils helper and use it appropriatly [22:38] <ubitux> then you could put whatever sanity check you want in [22:39] <ubitux> basically everything related to overflows, div/0, AV_NOPTS, .. [22:40] <ubitux> but well, afaict currently [22:40] <ubitux> only SGB demuxer and all the text subtitles are using the seek v2 api [22:41] <ubitux> but i'm pretty sure the option will rise for others [22:41] <ubitux> assuming someone would use it at some point [22:50] <michaelni> ubitux, the stream==-1 code hardcodes stream 0 so [22:50] <michaelni> its wrong if theres more than 1 stream [22:51] <cone-140> ffmpeg.git 03Clément BSsch 07master:fc86f863536b: fate/sub2video: move sws flags globally in the filtergraph. [22:52] <ubitux> i meant a function taking a timebase, and min_ts/ts/max_ts [22:52] <ubitux> i know that can't be done currently in the caller [22:53] <ubitux> also, i don't know enough about the different seeking callbacks to be able to tell what kind of helper would be welcome [22:53] <michaelni> if tehres just 1 stream theres no sense in passing stream=-1 on [22:53] <michaelni> so no code would need to handle it [22:53] <ubitux> that's currently the case at least :p [22:53] <michaelni> i can fix that i think [23:00] <michaelni> ubitux, patch sent [23:03] <ubitux> michaelni: i'm fine with adding any sanity check at that level now :p [23:03] <ubitux> of course if it can be done in the rescale function that's even better, but if you say that's not possible& [23:06] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:aa86d2d88461: lavf: move stream==-1 handling from ff_subtitles_queue_seek() to avformat_seek_file() [23:24] <cone-140> ffmpeg.git 03Clément BSsch 07master:49a78e6b8cbd: lavu/eval: handle div by zero in mod(). [23:31] <ubitux> we need a mac os ffmpeg developer :p [23:31] <ubitux> i have no idea where to look for #2100 [23:34] <beastd> ubitux: i would like to buy and install the os but it seems they want me to buy their hw for getting at the os... [23:35] <michaelni> ubitux, patches to fix the overflow posted [23:36] <saste> ubitux, would it help if you get a mac for testing purposes? [23:37] <ubitux> saste: no, because i have no idea what to test [23:37] <ubitux> i never actually used a mac, i don't know how it works [23:38] <saste> ubitux, it's not like you need to be a genius to use one [23:38] <saste> but i know i prefer not to have one since that would mean more work for me ;-) [23:39] <ubitux> you want to buy me an iPad? [23:39] <saste> uh.. for testing compilation it's not a big idea [23:39] <ubitux> then 2 iPad? [23:40] <saste> then you can use one as a keyboard? [23:40] <ubitux> compilation cluster [23:41] <ubitux> well anyway, no i don't think i would make use of a mac if i had one [23:41] <ubitux> if i haven't one currently, that's not because of the money :p [23:41] <saste> then we need a mac geek [23:41] <ubitux> exactly my point :) [23:41] <saste> and while at it, a windows nerd [23:41] <saste> or a mac nerd and a windows geek [23:42] <ubitux> michaelni: not afraid about libav adding a rescale flag? [23:42] <ubitux> (with the value=8) [23:42] <ubitux> we have at least one windows guy :) [23:42] Action: ubitux look at nevcairiel [23:42] <ubitux> the other rage quitted [23:42] <ubitux> +one [23:42] <ubitux> (d404) [23:43] <michaelni> ubitux, ill change it to some other less likely value than 8 [23:43] <saste> where is llogan and his start_number patch? [23:44] <ubitux> saste: no idea [23:45] <ubitux> michaelni: so how sane are the rescaled min/max ts after the function call now? [23:46] <michaelni> min/max will stay min/max with the flag [23:47] <michaelni> its like rescaling infinity [23:47] <saste> it was already oked [23:47] <saste> i'm going to push it [23:49] <saste> michaelni, you ok with the yadif patches? [23:52] <michaelni> saste, should be ok [23:53] <saste> i'm just not sure about the name of the "enable" option [23:55] <cone-140> ffmpeg.git 03Stefano Sabatini 07master:69d75dc4dd10: lavu/base64: extend/clarify doxy for the base64 API [23:55] <cone-140> ffmpeg.git 03Lou Logan 07master:091ce6bcb21e: doc/faq: add -start_number example [23:56] <michaelni> saste, i think calling it auto is better than enable [23:56] <michaelni> liek auto=1 -> automatic enabling [23:56] <saste> michaelni, i plan to add named constants [23:57] <ubitux> enabled=yes/no/auto? [23:57] <saste> so that would be enable=always|interlaced [23:58] <ubitux> what about deinterlace=all|interlaced ? [23:59] <ubitux> deinterlace=all|marked maybe [23:59] <michaelni> saste, auto=1/0 or yes/no [23:59] <ubitux> or just "deint" [23:59] <michaelni> ubitux, enabled=no makes no sense :) [00:00] --- Thu Jan 3 2013

1 0

ffmpeg.log.20130102
by burek 03 Jan '13

03 Jan '13

[00:00] <JEEB> I don't know random people's random shit [00:00] <misterno> apparently there is option CRF enable or disable [00:00] <JEEB> in that guy's random GUI shit [00:00] <JEEB> with x264 you pick a rate control mode [00:00] <JEEB> it's not an on/off switch [00:01] <JEEB> it's either constant quant (for development only), constant rate factor (crf) or average bit rate (bitrate-based) [00:01] <misterno> interesting comparing filesize [00:01] <JEEB> well you have NO FUCKING IDEA WHAT HE IS COMPARING AGAINST [00:01] <JEEB> that thing makes my head hurt so much so I just fucking ask you to forget about it [00:02] <misterno> alright don't get mad :) [00:02] <klaxa> since he has "[X] CRF (overrides video bitrate settings)" in the GUI i'd assume the "fallback" is bitrate based encoding [00:02] <klaxa> but yeah still retarded not using radio buttons or some shit [00:02] <JEEB> with the same settings, 1pass crf and 2pass average bit rate should give the same visual quality level, except for the fact that bit rate distribution can end up different [00:02] <JEEB> (with the same end bit rate) [00:03] <JEEB> the latter part gets really important with really, really short samples [00:03] <misterno> is this guide good http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [00:03] <JEEB> because 2pass generally tends to do worse there [00:04] <JEEB> in general that guide goes over whatever I was saying as well [00:04] <JEEB> 1) deal with your source (deinterlacing/IVTC, cropping, possible resize) [00:04] <JEEB> 2) use -ss to seek to some part of the clip where you know it should be like it is generally in the source, and set -t to like 10-15 seconds [00:05] <JEEB> 3) encode with some crf, starting from 23 which is the default [00:05] <misterno> wait wait, possible resize? [00:05] <misterno> so normally you dont even resize? [00:05] Action: JEEB facepalms [00:05] <JEEB> jesus freaking christ... [00:05] <misterno> why you facepalm on me [00:05] <misterno> i came up forward saying i dont know this stuff [00:05] <misterno> why do that? [00:05] <JEEB> let me scroll up and copy what I already IIRC said about that [00:05] <JEEB> and paste it here [00:06] <misterno> that is mean [00:06] <misterno> that only makes me not want to ask anymore questions [00:06] <JEEB> 1) you crop the source [overcropping is better than undercropping] 2) you resize the video as you see fit [features that might be reasons to downscale: bad quality of source, lack of detail and so forth and so forth] [00:06] <JEEB> sorry, I just get derpy when I have to repeat myself [00:06] <JEEB> don't take it too personally [00:06] <JEEB> 2) includes "do not resize" [00:07] <JEEB> every source I deal with is different, and depending on what kind of source I am dealing with I shall either resize or not resize [00:08] <misterno> ok but i ask why you wouldn't resize, resizing affects quality too much? [00:08] <misterno> because in my mind is lower the res = smaller the file size [00:08] <misterno> and i want to keep some files small [00:09] <misterno> and i have seen some excellent quality videos with reduces res [00:09] <misterno> meh nvm , i grilled you too much for today, happy new year! [00:09] <JEEB> sure was a quick leave [01:54] <espr3ss0> is this for Development or general help, I might be @ the wrong place [01:54] <espr3ss0> ? [01:56] <sacarasc> This is general help. [01:59] <espr3ss0> thanks sacarasc :-) [01:59] <espr3ss0> can i ask at the moment ? [01:59] <sacarasc> You can ask whenever you want. [02:00] <espr3ss0> i'm not sure the easiest way to achieve what I'm wanting [02:01] <espr3ss0> specifically at the moment, I have an ".avi" file & the video track is "mpeg-4", and without transcoding make the container ".mp4" [02:03] <espr3ss0> running "OS X Snow Leopard" [02:04] <espr3ss0> i've tried using "ffmpeg -i infile -vcodec copy -an -f mp4 outfile.mp4" [02:05] <espr3ss0> but playback is jumpy and only shows the first several minutes [02:10] <espr3ss0> oh btw, ... [02:11] <espr3ss0> do i need to re make / install ffmpeg if i alter "./configure" ? [02:12] <sacarasc> Yes. [02:23] <espr3ss0> checked x264 and an error came up, "Warning: PIX_FMT_RGB is missing from libavutil", is there a good place for library downloads ? [02:23] <espr3ss0> i'm not exactly running the latest n greatest ;-p [03:09] <defaultro> hey folks, why am I getting this error? I also have the latest x264 compiled/installed. http://pastebin.com/GYca5JqT [03:11] <sacarasc> Do you have multiple instances of x264 installed? [03:12] <defaultro> looks like [03:12] <defaultro> but not a package for sure [03:12] <defaultro> i have an install back in 2010 [03:12] <defaultro> what's the ideal way of cleaning it? I also installed it from source. I still have the old dir. Will make uninstall work? [03:14] <sacarasc> Should work. [03:14] <defaultro> k [03:14] <sacarasc> Then in the ffmpeg directory, you'll have to make clean && ./configure again. [03:14] <defaultro> k [03:15] <defaultro> ran it. It deleted x264.h, libx264.a, x264 binary and x264.pc [03:16] <defaultro> should I use --enable-static or shared [03:16] <defaultro> i used static earlier [03:22] <defaultro> make is running in ffmpeg [03:24] <defaultro> still failed with the same error [03:26] <defaultro> i am not sure if it's needed but I don't see libx264.a [03:28] <defaultro> i'm going to regit x264 [03:41] <defaultro> i see I have x264-lib and x264-devel as package installed [03:41] <defaultro> i uninstalled it. Hopefully it works this tie [03:41] <defaultro> time [03:41] <defaultro> yes, it worked! [04:04] <pinkette> what is the difference between 5.1 and dolby surround [10:04] <gxk> i am started to use ffmpeg to stream h264 over mpeg2ts/udp. it works well. i need to low latency, so the cache was disabled in vlc. this is causes to frame lost. What can be done in ffmpeg? my command line http://pastebin.com/u51jRUZq [10:11] <klaxa> hmm not really a lot i think [10:12] <gxk> just seconds before found something like '-tune zerolatency' is it helps? [10:19] <klaxa> i think you would have to re-encode the video stream then [10:29] <gxk> is there a way to measure latency inside ffmpeg? [11:00] <yellabs-r2> hello all [11:01] <yellabs-r2> can i use ffmpeg to turn an colour mp4 to black and white movie ? [11:10] <ubitux> yellabs-r2: does -vf format=gray works? [11:10] <ubitux> you might want to play with pix format encoding etc [11:12] <yellabs-r2> i had this one , but it did not work [11:12] <yellabs-r2> ffmpeg -i input.mp4 -strict -2 -pix_fmt yuvj420p -vcodec libx264 -acodec aac output.mp4 [11:13] <ubitux> ? [11:13] <ubitux> and where is the convert to grayscale supposed to be done with your cmd line? [11:14] <yellabs-r2> its wrong , i guess [11:14] <ubitux> yellabs-r2: -pix_fmt gray may help [11:14] <yellabs-r2> [format @ 0x9a0ba60] Unknown pixel format: grey [11:14] <yellabs-r2> Error initializing filter 'format' with args 'grey' [11:14] <yellabs-r2> Error opening filters! [11:14] <ubitux> read again [11:15] <yellabs-r2> i see [11:15] <ubitux> note: it's stupid to use -vf format=gray -pix_fmt yuvj420p [11:15] <ubitux> use -pix_fmt gray direct [11:15] <ubitux> directly [11:15] <yellabs-r2> trying [11:16] <ubitux> since you're not touching audio, -c:a copy instead of -strict -2 -acodec aac would be wise too [11:17] <yellabs-r2> hmm, i am to much of an newbee .. [11:17] <yellabs-r2> ffmpeg -i '/home/yellabs-r2/Bureaublad/ewo-mp4/snt1.mp4' -pix_fmt gray output.mp4 [11:17] <yellabs-r2> tried this, still colour [11:18] <ubitux> Incompatible pixel format 'gray' for codec 'libx264', auto-selecting format 'yuvj420p' [11:18] <ubitux> hum. [11:19] <ubitux> well, then just keep the -vf format=gray [11:20] <yellabs-r2> testing [11:20] <ubitux> and drop the -pix_fmt from your cmd line [11:20] <yellabs-r2> it works [11:20] <yellabs-r2> :) [11:21] <yellabs-r2> thanks for the help [11:21] <ubitux> of course it works [11:21] <yellabs-r2> :) [11:21] <ubitux> np [11:32] <Max__> How can I cut yuv video files with ffmpeg? None of the obvious google hits turned up anything useful. [11:52] <buhman> Max__: man ffmpeg [11:52] <buhman> Max__: line 1471,1478 [12:11] <Max__> buhman: thanks but I can't find the relevant lines. I looked at three separate versions of the man pages and I can't seem to find it. [12:12] <buhman> 04:52 < buhman> Max__: line 1471,1478 [12:12] <buhman> how can you not find them when I told you *exactly* which lines are of interest [12:12] <Max__> Because line 1471 in my man file is talking about ac3 audio codec. [12:13] <buhman> you must have some ancient ffmpeg then [12:14] <Max__> My mistake, perl metadata. [12:14] <Max__> version 0.10.6 [12:14] <Max__> ^ffmpeg [12:17] <ubitux> we are in 1.0+ [12:19] <buhman> Max__: http://ffmpeg.org/ffmpeg.html#Main-options paragraphs 6+8 [12:19] <Max__> 0.10.6 built Nov 13 2012, pretty recent. Nothing newer available for my distro. :\ [12:19] <buhman> no that is not recent [12:19] Action: Max__ submits package request [12:19] <klaxa> compile from source or get a static build [12:20] <Max__> Will do, thanks. [12:21] <buhman> Max__: for latest manpages: git clone git://git.videolan.org/ffmpeg.git && cd ffmpeg/doc && make doc/ffmpeg [12:21] <buhman> takes all of 30 seconds [12:22] <buhman> hrm I guess you need to configure first :P [12:22] <buhman> and not go into doc/ [12:23] <ubitux> Max__: built doesn't mean the sources aren't ancient [12:24] <ubitux> Max__: https://www.ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_… [12:24] <Max__> ubitux: I know, I just couldn't find the right line in the man page. [12:25] <ubitux> all the doc is available only, but is for git/master/HEAD [12:25] <ubitux> s/only/online/ [12:26] <buhman> well, in his case it *should* be applicable :P [12:26] <ubitux> segment format has evolved quite a bit recently [12:26] <ubitux> i'm not even sure it was in 0.10 [12:26] <Max__> 0.10 is only a month older than 1.0... [12:26] <ubitux> 12:23:12 <@ubitux> Max__: built doesn't mean the sources aren't ancient [12:27] <ubitux> 0.10 is about 1 year old [12:27] <buhman> Max__: ^ was released 27-Jan-2012 00:00 [12:27] <Max__> I just did -version and it said 0.10.6 Nov 13 2012. [12:27] <ubitux> January, 27, 2012, FFmpeg 0.10 [12:27] <ubitux> We have made a new major release (0.10) [12:27] <buhman> Max__: that's the *BUILD* date [12:27] <Max__> Oh. [12:28] <buhman> Max__: as you've been told multiple times now [12:28] <ubitux> we release 0.11 in the meantime [12:28] <ubitux> and 0.1 [12:28] <ubitux> erm [12:28] <ubitux> and 1.0* [12:29] <buhman> Max__: but really there's no reason to not use git://git.videolan.org/ffmpeg.git [12:29] <ubitux> and we might release another soon [12:29] <buhman> as far as I'm concerned, anything prior to 8bc74221f8ef6fe97801025866943013b94f5e2f might as well have been released in 40,000 B.C. [12:30] <Max__> buhman: there is a very good reason. It's my wife's laptop and I can't always pull and build the newest versions of everything she needs for her research. [12:30] <buhman> I have no idea what that means, but ok [12:30] <ubitux> buhman: haha [12:32] <buhman> also, why does https://github.com/FFmpeg/FFmpeg exist? it's ~8 hours behind at the moment [12:33] <Max__> Anyway, the segmenter seems to be what she needs. Thanks. [12:33] <ubitux> buhman: a relic of the past certainly [12:34] <ubitux> a legendary mirror [12:34] <buhman> heh [12:40] <buhman> wow the segmenter is really cool [12:41] <ubitux> :) [12:41] <buhman> much easier than -ss and -t :D [12:45] <buhman> ubitux: why is https://ffmpeg.org/ffmpeg.html much shorter than doc/ffmpeg.1 ? [12:47] <ubitux> buhman: it's not [12:47] <ubitux> maybe you're still living in the past [12:47] <buhman> O.o [12:47] <ubitux> it's been split recently [12:47] <ubitux> see latest news [12:47] <buhman> ffmpeg(1) is 8502 lines [12:48] <ubitux> 1685 ./doc/ffmpeg.1 [12:48] <ubitux> [~/src/ffmpeg]- ls doc/*.1 [12:48] <ubitux> doc/ffmpeg.1 doc/ffmpeg-codecs.1 doc/ffmpeg-filters.1 doc/ffmpeg-protocols.1 doc/ffmpeg-scaler.1 doc/ffplay.1 doc/ffserver.1 [12:48] <ubitux> doc/ffmpeg-bitstream-filters.1 doc/ffmpeg-devices.1 doc/ffmpeg-formats.1 doc/ffmpeg-resampler.1 doc/ffmpeg-utils.1 doc/ffprobe.1 [12:49] <buhman> wat [12:49] <buhman> what commit was that O.o [12:50] <ubitux> there are several [12:50] <ubitux> 12:47:37 <@ubitux> see latest news [12:51] <buhman> somewher between November 14 and December 30 I guess [12:51] <ubitux> b448677e6ea734b9f7a57970560f600a2a893a9b was the final commit [12:52] <buhman> so I guess you do some magic to convert texi to that html format [12:53] <ubitux> that was already the case [12:53] <ubitux> Stefano improved a bit the script [12:55] <NuxRo> hello guys, I've got a dd copy of a dvd, basically a iso9660 4GB img file i guess, how could I extract the video from it into some more usable format? [12:59] <buhman> NuxRo: mount -o loop iso9660-4gb.iso /mountpoint && dvdunauthor --magic-arguments && ffplay -i whatever-combined.vob --more-magic-arguments output.desired-container-and-so-on [12:59] <buhman> s/ffplay/ffmpeg/ [13:00] <NuxRo> aha, need to look up dvdunauthor, cheers [13:00] <buhman> http://dvdauthor.sourceforge.net/ [13:27] <gxk> if ffmpeg converts h264 to mpeg2video while streaming mpeg2ts? http://pastebin.com/Gw1RVSLd [13:28] <burek> gxk, what is your question exactly [13:31] <gxk> how to run ffmpeg without transcoding from h264 to mpeg2video? [13:31] <burek> -c:v copy [13:33] <gxk> hmm, got error "Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument" [13:34] <gxk> got error - http://pastebin.com/UpAh9TkV [13:36] <burek> gxk, try removing -bsf h264_mp4toannexb [13:39] <burek> if that doesn't help, can you try updating your ffmpeg to latest version? [13:39] <gxk> 1 sec :) [13:50] <gxk> ffmpeg fresh from git, same problem [13:56] <gxk> http://pastebin.com/qjBu2BeQ [13:58] <burek> try adding back -bsf [14:00] <gxk> http://pastebin.com/2JJbsHD0 [14:01] <burek> you don't need -b:v 4M [14:01] <burek> since you are doing -c:v copy [14:01] <burek> but that shouldn't matter in this error [14:03] <gxk> same problem - http://pastebin.com/yiFiR0SU [14:06] <burek> it might be a bug, i guess [14:06] <burek> could you report it on our bug tracker? [14:07] <burek> you could provide both these logs in your report: http://pastebin.com/qjBu2BeQ and http://pastebin.com/yiFiR0SU [14:11] <ubitux> -fflags +genpts? [14:11] <ubitux> or specify a rate or something? [14:11] <burek> with -c:v copy? [14:11] <ubitux> that's the codec [14:11] <ubitux> timing is handled at container level [14:12] <burek> couldn't it be figured out from input [14:12] <ubitux> input looks like raw h264 [14:12] <burek> so fps is missing? [14:12] <ubitux> i don't know if it has any usable timing info [14:12] <ubitux> i'd try to -r 25 or something as input option [14:12] <burek> gxk, you've heard the man :) [14:13] <gxk> burek: full attention [14:16] <gxk> tried all of below - http://pastebin.com/MAMwaaWe [14:18] <burek> try ffmpeg -re -r 25 -i ./transformers/trans3-720p.h264 [14:18] <burek> ... [14:20] <gxk> burek, no. do you need another pastebin? [14:21] <burek> how did you generate that .h264 file [14:22] <gxk> downloaded mkv of Transformers3, extrackted video es by mkvtools [14:26] <burek> could you try with the original mkv [14:26] <burek> or [14:26] <burek> when you extract the video [14:26] <burek> try extracting it to mkv container [14:26] <burek> to preserve all the needed info [14:26] <AL13N> i'm using ffmpeg in cli and i notice that whatever -r option i give, be it 25 or 12 or 10 or 6; i have only about 80% of the framerate... since i'm using it with output file '-' (inside a php passthru) i would like some kind of way to encoding into webm/ogg that would be faster than it's output rate, so that the browser can continue [14:27] <burek> this way you've got a raw .h264 video, without some needed info, which causes problems [14:27] <AL13N> can anyone help me pass options to make it's encoding go faster? [14:28] <burek> btw, you don't encode things in php page man.. [14:28] <AL13N> http://pastebin.com/dMNr1PqV [14:29] <AL13N> this gives about fps=10; if i specify -r 25, i have fps=20 if i specify -r 6; i have fps=5 [14:29] <burek> also, that pastebin is not what was asked from you [14:29] <AL13N> i tried lowering bitrate, but it doesn't help [14:29] <burek> read more carefully [14:29] <AL13N> ah [14:33] <AL13N> http://pastebin.com/7SsbddfX [14:33] <AL13N> burek: ^^ [14:33] <AL13N> now, i know that this is an older PC, but i'd like someway of speeding things up [14:34] <burek> you might update your ffmpeg [14:35] <burek> also [14:35] <burek> your fps depends on video encoding parameters mostly [14:35] <burek> which you haven't specified except for the -g [14:35] <burek> try using -b:v [14:35] <AL13N> k [14:36] <burek> http://ffmpeg.org/ffmpeg-codecs.html#libtheora [14:38] <AL13N> burek: i tried alot of settings: http://pastebin.com/9eYHQprh [14:38] <AL13N> but nothing helps [14:38] <burek> no need to focus on audio encoding options [14:38] <burek> audio encoding is usually fast [14:38] <burek> focus on video [14:38] <AL13N> oic [14:38] <AL13N> any other options i can use? [14:39] <burek> try something like -b:v 1M [14:39] <burek> to make it encode video faster [14:40] <gxk> burek: created new ticket 2098. welcome to call me for testing the issue. BTW, it is my 1st bug with ffmpeg, so hope it created well. [14:40] <burek> gxk :) :beer: thanks [14:41] <AL13N> burek: i also tried this: http://pastebin.com/Cvz4tCSV which didn't help either [14:41] <AL13N> burek: 1M also has 10 fps btw [14:41] <AL13N> burek: this is what is annoying [14:42] <burek> maybe your cpu is just not fast enough [14:42] <burek> can you run screen [14:42] <burek> and in one screen run ffmpeg [14:42] <burek> in another run top [14:42] <burek> to see the cpu usage [14:42] <AL13N> k [14:42] <burek> or run ffmpeg in the background [14:42] <burek> and type top [14:42] <AL13N> maybe the stats are slowing down the output? [14:43] <burek> (put -t 20) to end encoding after 20 seconds [14:43] <burek> while you are testing [14:43] <AL13N> 99.3% ffmpeg [14:43] <AL13N> all the time [14:43] <AL13N> (single core machine) [14:43] <burek> arm? [14:43] <AL13N> no [14:43] <AL13N> Athlon 64 [14:44] <burek> well, 99% tells you the reason [14:44] <AL13N> 2.4Ghz [14:44] <AL13N> burek: right, so, is the problem the input quality? [14:44] <AL13N> that's what my last thought was [14:44] <AL13N> considering whatever -r:v i gave, i got about 80% of it [14:45] <burek> try again this example http://pastebin.com/Cvz4tCSV [14:45] <burek> without -b:v 16k [14:45] <AL13N> k [14:45] <burek> or put -b:v 2M [14:45] <AL13N> ok [14:47] <AL13N> burek: http://pastebin.com/gMrgKNsM [14:47] <AL13N> strange [14:48] <burek> remove -r 6 [14:48] <AL13N> do i leave the scale to 80 ? [14:48] <burek> yes [14:50] <AL13N> http://pastebin.com/LELQ8tgR [14:50] <burek> fps= 50 [14:50] <AL13N> since input rate is 60, now fps=50 [14:50] Action: AL13N sighs [14:50] <burek> did you check your cpu [14:50] <AL13N> it's quite strange [14:51] <AL13N> same [14:51] <AL13N> 99.3% [14:51] <burek> now, try with real file output [14:51] <burek> and after that type ffmpeg -i outputfile.webm [14:51] <burek> to see what's the real frame rate in the file [14:52] <burek> and if you want to do live re-encoding, for viewers to watch the video while you are encoding [14:52] <burek> then use ffmpeg -re -i /media... [14:53] <AL13N> http://pastebin.com/dFFE1exb [14:54] <AL13N> the output file has 60 as well, it's just produced at 50fps ... sigh [14:54] <AL13N> i'll try -re [14:54] <burek> this output is good [14:54] <burek> 59.94 fps [14:54] <AL13N> yes [14:54] <burek> -re will read the input in "real-time" [14:55] <AL13N> i'll give it a shot [14:55] <burek> i.e. ffmpeg will not try to do everything as fast as it can [14:56] <AL13N> http://pastebin.com/R0961EJH [14:56] <AL13N> 99.3% cpu again [14:56] <AL13N> it seems to be a small bit better [14:56] <AL13N> but not 60 fps [14:56] <burek> well, I'm not sure, but everything is telling me that either something is not well optimized for your hardware [14:56] <AL13N> should i try to lower bitrate? [14:56] <burek> or that cpu is slow [14:56] <burek> try to lower the output frame rate [14:57] <burek> add -r 30 before -f ogg [14:57] <AL13N> ok [14:57] <AL13N> fps seems to be 26 now before it's finished [14:58] <burek> that fps which is shown during the encoding is the approx. curr. fps of the encoding [14:58] <AL13N> 27fps [14:58] <burek> also, why do you use -g 0 [14:58] <AL13N> http://pastebin.com/7XcnzS8e [14:59] <AL13N> burek: some docs told me that -g 0 would disable me to make it faster [14:59] <burek> -g 0 means "make every frame a key frame" [14:59] <burek> why would you want that [15:00] <AL13N> ah, ok, losing that [15:00] <burek> you are wasting (cpu) time on every frame, to encode all the info from the frame, instead to encode only differences from the next frame [15:01] <AL13N> http://pastebin.com/pi0PevjT [15:01] <AL13N> seems to be 27fps still [15:01] <burek> doesn't have to be [15:01] <burek> try file output [15:01] <burek> and ffmpeg -i output [15:02] <AL13N> k [15:03] <AL13N> http://pastebin.com/vpTvPFsU [15:04] <AL13N> still 27 [15:04] Action: AL13N sighs [15:05] <burek> Input #0, ogg, from 'file.ogg': [15:05] <burek> Stream #0:0: Video: theora, yuv420p, 80x45 [SAR 1:1 DAR 16:9], 30 fps, 30 tbr, 30 tbn, 30 tbc [15:06] <burek> where did you see 27? [15:06] <AL13N> at the end of encoding, the speed at which it encoded [15:06] <burek> that's not accurate [15:06] <burek> i told you already [15:06] <burek> it's approx [15:06] <AL13N> yes [15:06] <AL13N> but even though the output is correct at 30fps [15:07] <AL13N> i would like that the speed would be 27 fps as well, so that i have no delay when people play it in a <video></video> tag [15:07] <AL13N> err, i mean 30fps [15:07] <burek> it IS 30 [15:07] <AL13N> the output rate is 30 [15:08] <AL13N> but the speed at which the frames are produced by ffmpeg is approx 27fps [15:08] <burek> loose -re [15:08] <AL13N> which means the output of this in a pipe will be coming in at around 27fps, thus playing a bit and then pausing to get more data and then playing [15:08] <burek> no, it doesn't mean that [15:09] <AL13N> ok, loosing -re [15:09] <burek> it just means ffmpeg took some time to start up [15:09] <burek> initialize encoders, pipes and filters [15:09] <burek> which, when calculated in current fps, makes it 27 [15:09] <gxk> burek: original mkv file works good. you was right! [15:09] <burek> because of that (idle) time when it didn't encode [15:09] <burek> gxk, well, ubitux told you, your h264 file is just raw video frames [15:10] <burek> without info about frame rate or timestamps [15:10] <AL13N> burek: right, but is there a way for me to get the output rate lower than the approx fps it produces the frames? [15:10] <burek> which creates problems eventually :) [15:10] <burek> mkv keeps all that info, that's why it works [15:10] <burek> AL13N just try a live test [15:10] <burek> and you'll see what im telling you [15:11] <burek> it is working well, it's just the start of the encoding that makes it appear as less than 30 fps [15:11] <burek> which it actually isn't [15:11] <burek> and with live encoding/viewing, you can't send more data to viewer [15:11] <burek> than it is already available [15:12] <gxk> burek: how can ffmpeg extract video track with needed info from mkv? [15:12] <burek> that would seriously undermine einstein's relativity theory :) [15:12] <AL13N> burek: ok [15:12] <burek> gxk, just extract it like: ffmpeg -i original.mkv -an -c:v copy justvideo.mkv [15:13] <burek> -an = no audio [15:13] <burek> so, you are just copying the video streams inside original mkv file into a new mkv file, which will contain just video stream(s) [15:14] <burek> AL13N, you might check if the viewers/players have any caching options [15:14] <burek> which gives you what you are after [15:14] <AL13N> k [15:17] <AL13N> burek: well, i'm using html5 video tags, so i donno if there is something... however, it seems the speed is not really enough... i would have thought it to be possible to encode something with bad quality faster than it actually takes to play (like double) [15:18] <burek> you'll have to test more [15:18] <burek> im not sure if it's the cpu [15:18] <burek> if you were using libx264 instead of libtheora [15:18] <burek> there would be more options to fine-tune your output [15:18] <burek> and cpu usage [15:18] <burek> with libtheora im not that much familiar :/ [15:20] <AL13N> it seems MP4 is not streamable, if i have output file '-' than -t mp4 gives error like that [15:24] <sacarasc> You need to do special things with MP4 to make it streamable. [15:31] <AL13N> burek: i retried on another machine, and that one can have approx fps much higher than the output rate [15:31] <AL13N> sacarasc: any idea what? [15:33] <sacarasc> Moving the moov atom to the start of the file using qtfaststart for non-live streaming. For live streaming, I think you have to do that and break the files up into multiple little ones. [15:37] <ambro718> Hi. Has someone implemented using OpenMAX in ffmpeg to offer transparend hardware acceleration? [15:37] <gxk> which option removes metadata builtin stream like http://pastebin.com/pUUDT6jD [15:48] <Agiofws> hello [15:48] <Agiofws> i have this kind of media file which a dvd player of mine will not play http://www.pasteall.org/38473 can anyone advise to re encoded it with what options thank you ? [15:49] <burek> sacarasc: http://ffmpeg.org/ffmpeg-formats.html#MOV_002fMP4_002fISMV (especially -movflags faststart) [15:50] <burek> AL13N, then your cpu is too old.. [15:50] <burek> or your ffmpeg was not compiled for your architecture [15:50] <burek> try compiling the latest git version [15:50] <Agiofws> anyone ? [15:50] <burek> and latest libtheora of corse [15:50] <burek> course* [15:51] <AL13N> my cpu is old-ish, but still... [15:51] <burek> <burek> try compiling the latest git version <burek> and latest libtheora of course [15:51] <AL13N> yes, but this is a production server, so it's not that easy [15:51] <burek> or try static builds [15:51] <AL13N> but thanks for the help, i guess i know where to look [15:52] <AL13N> i was hoping there was something that could drop frames when reading or something [15:52] <AL13N> so that at least that would be faster read [15:52] <burek> Agiofws, try typing: ffmpeg -i yourfile [15:52] <burek> and copy the output to a pastebin-like site (like www.pastebin.com) [15:53] <burek> AL13N, you can forse the input frame rate [15:53] <burek> force* [15:54] <AL13N> i tried with -r before -i , but i got: invalid "frame_rate" property [15:54] <burek> but still, the frames sent to the player/viewer will be with the same timestamps [15:54] <AL13N> in this MP4 [15:54] <burek> you won't be able to send "more data" in advance [15:54] <AL13N> i'm not particularly worried about that [15:54] <burek> (with live video of course) [15:54] <burek> with pre-recorded video, well, just offer a .flv file or .mp4 [15:54] <AL13N> burek: i'm not using a streaming protocol [15:55] <burek> and it will work, depending on the bandwidth the viewer has got [15:55] <AL13N> if there was a way to produce less cpu when reading the file, it might go better [15:55] <AL13N> so that's what i'm thinking [15:55] <burek> AL13N, what is your overall (logical) goal? [15:56] <burek> gxk, did you read about metadata in ffmpeg docs? [15:56] <AL13N> atm, i'm thinking that this server is to slow to read the input file at 60 fps; so i'm trying to get it to use less cpu when reading the file, which is likely the bottleneck atm [15:57] <burek> gxk, maybe something like -map_metadata -0 [15:57] <burek> AL13N, if that's the case [15:57] <burek> you'll need to re-encode your input video [15:57] <burek> to lower the frame rate [15:57] <gxk> burek: checking... [15:58] <Agiofws> burek, http://pastebin.com/8RhLJ7JV [15:58] <Agiofws> thnak you [15:58] <burek> oh man [15:58] <burek> i think you have the first version of ffmpeg [15:58] <burek> ever compiled [15:58] <burek> :D [15:59] <burek> you really need to update your ffmpeg Agiofws :) [15:59] <burek> however, Agiofws, do you have a sample media file that your dvd CAN play? [16:00] <AL13N> burek: myes, i had figured that, but trying to avoid [16:00] <Agiofws> burek, no i don't sorry ... [16:00] <AL13N> burek: if all else fails, i'll just hold off outputting until done (buffering in server) [16:00] <burek> Agiofws, well there's no way we can predict what your dvd can play :( [16:01] <burek> AL13N, what is your overall (logical) goal? [16:01] <burek> what are you trying to accomplish? [16:03] <Agiofws> but generally it does play avi's i think it just does not play movies which are in a blue ray ? format .. if you can give me some pointers ? maybe its the not normal resolution ? [16:04] <burek> Agiofws, avi is a container, like a box, containing media streams (audio/video/subtitles) [16:04] <burek> so your dvd doesn't play avi's, but what's inside of it [16:05] <Agiofws> burek, i assume its the AAC sound codec ? [16:06] <burek> that's why i asked if you have any other media that you can play in your dvd [16:06] <burek> to run an info on it and see what it is [16:06] <burek> brb [16:08] <Agiofws> agiofws@ixus:/media/usb0/chris/clouldAtlas$ file 0001-247812.avi [16:08] <Agiofws> 0001-247812.avi: RIFF (little-endian) data, AVI, 720 x 304, 23.98 fps, video: FFMpeg MPEG-4, audio: MPEG-1 Layer 3 (stereo, 44100 Hz) [16:26] <wieshka> guys help out with -vf scale - I need to scale 16:9 video (actual video area is 4:3, with black margins) to 4:3 with cuted out black area [16:26] <wieshka> input 1920x1080, taget ouput: 800x600 [16:27] <Mavrik> you'll have to use crop filter to cut away the black bands before scaling [16:28] <Mavrik> so something like [16:28] <wieshka> -filter:v "crop=800:600:560:0" ? [16:28] <Mavrik> depending on the order of filters [16:29] <Mavrik> if you want to crop before scaling you do something like [16:29] <Mavrik> crop=1440:1080:240:0,scale=800:600 [16:29] <Mavrik> that will crop the frame first to 4:3 and then scale it down to 800x600 [16:31] <wieshka> Mavrik, thx, figured out [16:31] <wieshka> only i got: [swscaler @ 0x27f3de0] Warning: data is not aligned! This can lead to a speedloss [16:31] <wieshka> as warning [16:32] <wieshka> setdar=4:3 ? [16:32] <Mavrik> nah, that's something else [16:32] <Mavrik> (image frame data is not aligned to 64-bit boundaries) [16:32] <Mavrik> it has nothing to do with images :) [16:32] <Mavrik> probably a bug that has to be addressed [17:02] <canaux> Hi all [17:16] <canaux> does ffmpeg permits to manage the aac encoder delay in another way than assuming a fixed arbitrary delay ? I need to encode files with ffmpeg that will be well "understood" by Apple tools. [17:16] <canaux> Here are more information about encoder delay: http://developer.apple.com/library/mac/#documentation/QuickTime/qtff/QTFFAp… [18:49] <Diogo> hi anyone know a good repository of video in different formats to test encodings? [18:54] <saste> Diogo, the ffmpeg FATE samples suite [19:03] <Diogo> to encode video with audio aac and Bit rate mode: Variable [19:13] <Diogo> saste: but this samples i view a webm , mp4 file, a wmv ..examples in different formats [19:13] <Diogo> ? [19:21] <Diogo> this commando put me audio bitrate a constant why ? : /servers/ffmpeg/bin/ffmpeg -i '.$file.' -r 25 -s 580x326 -c:v libx264 -vprofile baseline -b:v 700k -keyint_min 50 -pass 2 -acodec libfaac -ar 44100 -ab 128k -ac 2 -f mp4 '.$to [19:33] <llogan> Diogo: because you used "-ab 128k" [19:33] <llogan> try with "-q:a 100" when using libfaac for VBR [19:34] <llogan> it is similar to "faac -q 100" [19:35] <llogan> but VBR audio doesn't really make sense for a two-pass video encode [20:00] <tmkt> Afternoon [20:00] <tmkt> any idea how to fix "stream 0, error opening alias: path='/Users/'," [20:00] <tmkt> on a .mov file i received from a user? [20:01] <llogan> probably one of those mov files don't actually contain video but point to it [20:02] <tmkt> http://pastebin.com/QdaLp3F3 [20:12] <tmkt> guessing it was created/exported incorrectly from that info [20:37] <llogan> tmkt: ask them to send you an actual video file instead of a referenced mov [20:38] <tmkt> they aren't smart enough to know what they did [20:38] <tmkt> they told me they renmaed the file [20:38] <tmkt> and re-uploaded it [20:38] <tmkt> and that version worked [20:38] <tmkt> the first file was 5k..the 2nd one was 1.8M [20:39] <llogan> i love clients like that [20:43] <llogan> tmkt: tell them to check the "make movie self-contained" box [20:47] <tmkt> k...thanks [20:53] <Sashmo_> for the life of me, I cant figure out how to get ffmpeg to play a range of files. Is it possible with out having to script it? Is there a way to play all files in a directory? [20:59] <klaxa> ffplay * ? [20:59] <klaxa> mmh... [20:59] <klaxa> doesn't look like it [21:01] <klaxa> you can always do: [21:01] <klaxa> for i in *; do ffplay "$i"; done [21:01] <klaxa> only if your shell supports that of course [21:02] <klaxa> or use a player for playback, i think ffplay isn't supposed to be used for anything else than testing files [21:04] <llogan> consider adding -autoexit to klaxa's example [21:09] <espr3ss0> hi again ! ;-p [21:10] <espr3ss0> do i need to specify "-vpre" for ffmpeg with "libx264" every time ? [21:11] <Sashmo_> klaxa: thats what I want to avoid.... [21:11] <klaxa> avoid what? using a for-loop or using a mediaplayer? [21:11] <klaxa> just get mplayer or mplayer2 [21:11] <klaxa> then you can run mplayer * [21:16] <llogan> espr3ss0: it depends on your ffmpeg version [21:16] <llogan> recent ffmpeg will use -preset instead of -vpre, and default is "-preset medium" if you do not declare a preset. [21:19] <celesteh> i've got a commandline argument string that will record audio via jack, which is fantastic. i have to connect the inputs via the JACK Control GUI. Is there a way to specify them in the command line? [21:21] <klaxa> celesteh: maybe some JACK cli? [21:23] <llogan> try #jack [21:32] <maujhsn209_> Hi I'm new to kdenlive and want to know which version of ffmpeg is best compatible with kdenlive...I've used this site as a guide: https://launchpad.net/~sunab/+archive/kdenlive-svn [21:34] <Sashmo_> klaxa: yeah, I could.... I'll give it a try later.... [21:37] <espr3ss0> llogan: "FFmpeg SVN-r26402" on Mac's "Snow Leopard" [21:37] <ubitux> wat [21:38] <ubitux> svn, really? [21:40] <llogan> espr3ss0: that's incredibly old [21:41] <espr3ss0> llogan: i know ;-p [21:41] <espr3ss0> thanks ! [21:42] <llogan> maujhsn209_: are you familiar with ffmpeg? [21:43] <maujhsn209_> llogan Yes! [21:45] <maujhsn209_> llogan I am concerned with compatibility issue w/ kdenlive! [21:46] <maujhsn209_> llogan See this: https://launchpad.net/~sunab/+archive/kdenlive-svn [21:50] <Sashmo_> anyone have experience converting PAL to NTSC ? [21:51] <llogan> maujhsn209_: then i recommend you look at the kdenlive presets and see if their syntax will work with recent ffmpeg...or just ask at #kdenlive [21:53] <Sashmo_> in my conversion I am getting some bad looking text in the news scrolls at the bottom of the screen, anyone know if there is some special filters that I should add? [21:54] <maujhsn209_> llogan...So it okay to just dowload ffmpeg like I would do normally & worry about the preset stuff later...is this correct? [22:02] <llogan> maujhsn209_: i suppose. you can always make your own presets too. [22:05] <maujhsn209_> Okay thanks! [22:09] <Sashmo_> http://pastebin.com/bwUgSbH2 [22:09] <Sashmo_> thanks guys [22:12] <llogan> Sashmo_: do you get the bad looking text if you omit "-r 29.97"? [22:12] <llogan> why -vsync 1? [22:12] <Sashmo_> llogan: yes [22:13] <Sashmo_> llogan: I found that the vsync helped [22:13] <llogan> then it's probably due to interlacing [22:13] <Sashmo_> llogan: a bit [22:13] <Sashmo_> I tried yadif filter [22:13] <Sashmo_> it helped a bit, but the text was still fuzzy and choppy [22:14] <llogan> and you should use -r ntsc or -r 30000/1001 instead of 29.97 [22:14] <llogan> and you didn't include the complete output [22:14] <Sashmo_> let me try [22:15] <llogan> it won't make a difference [22:15] <Sashmo_> llogan: complete output http://pastebin.com/YK4YULkj [22:29] <Sashmo_> llogan: what do you think? [22:40] <celesteh> when somebody starts ffmpeg at the command line with "ffmpeg -f jack -ac 2 -i ffmpeg -f x11grab ... etc" can somebody explain what's going on with the syntax? Is -f x11grab a couple of arguments to ffmpeg or is x11grab the name of a separate program that ffmpeg invokes, such that the user would be able to specify additional arguments to x11grab or to jack? [22:54] <Sashmo_> so here is what I have been able to do, its better, but the quality is still fuzzy on the text, and the scrolls are choppy not smooth, any ideas guys? http://pastebin.com/8SBjBhgH [22:56] <llogan> celesteh: x11grab is an input device that can accept various options [23:22] <maujhsn209> I have encountered a build error I have included the ffmpeg config.log...pastebin.com/baZMfDVj need help thanks! [23:25] <maujhsn209> I have encountered a build error I have included the ffmpeg config.log...http://www.pastebin.com/baZMfDVj need help thanks! [23:26] <klaxa> wha- [23:26] <klaxa> can you... paste the file raw without the nano editor stuff? [23:27] <maujhsn209> klaxa I'll get back to you in just a few moments...thanks! [23:32] <maujhsn209> klaxa Here it is: http://www.patebin.com/n4hgte82 [23:34] <klaxa> install libvpx-dev [23:34] <ubitux> patebin :) [23:34] <klaxa> that too :P [23:35] <maujhsn209> klaxa http://www.pastebin.com/n4hgte82 :) [23:36] <klaxa> maujhsn209: install libvpx-dev [23:44] <maujhsn209> klaxa sudo apt-get -y remove the "previous" libvpx-dev....cd ~/libvpx make clean git pull...Yes or No? [23:44] <klaxa> wait... you already have libvpx-dev? [23:44] <maujhsn209> It was one of the steps [23:45] <maujhsn209> In the build process! [23:45] <Sashmo_> the current yadif filer in ffmpeg, is that interpolating or blending? [23:45] <klaxa> hmm... [23:46] <maujhsn209> klaxa I am not making this stuff up! [23:46] <klaxa> yeah well it would probably be best then to compile from source [23:47] <maujhsn209> No thanks! [23:48] <maujhsn209> Bye! [23:49] <klaxa> what i said was a yes to: "sudo apt-get -y remove the "previous" libvpx-dev....cd ~/libvpx make clean git pull...Yes or No?" [00:00] --- Thu Jan 3 2013

1 0

ffmpeg-devel.log.20130101
by burek 02 Jan '13

02 Jan '13

[00:07] <ubitux> http://prog21.dadgum.com/161.html [00:58] <cone-112> ffmpeg.git 03Carl Eugen Hoyos 07master:67abe21c0bdc: mov: fix handling of odd sized yv12 [00:58] <cone-112> ffmpeg.git 03Michael Niedermayer 07master:f105fe5c22c5: rawdec: fix decoding of odd dimensioned raw yv12 [01:05] <nevcairiel> didnt someone tell these people that 4:2:0 with odd dimensions is rather invalid? yet they still create such files :( [01:06] <wm4> the jpeg decoder creates such output, and these are definitely valid [01:07] <nevcairiel> how can you subsample chroma by a factor of two if you didnt have an even number of pixels to start with? [01:08] <nevcairiel> do you create one chroma too much or one too few? [01:08] <wm4> one too much [01:09] <wm4> and the last luma row or column is simply missing, I suppose [01:10] <nevcairiel> isnt jpeg also macroblock based, so its always mod16 coded? [01:11] <wm4> maybe, but the image dimensions can obviously be arbitrary [01:11] <wm4> I wonder if ffmpeg has a policy how odd sizes should be handled? [01:11] <nevcairiel> if you look at those commits up there, they decided to just cut one pixel :) [01:12] <wm4> the jpeg decoder still outputs odd sizes [01:26] <iive> nevcairiel: the visible pixels could be subset of the coded pixels [01:26] <iive> this is why these are separate dimensions. [02:02] <cone-112> ffmpeg.git 03Nicolas George 07master:2eb2e1798eee: lavfi: add avfilter_get_audio_buffer_ref_from_arrays_channels. [03:07] <Compn> 4k v210 [03:07] <Compn> sounds like ... high quality [05:31] <pross-au> michaelni: do you have any thoughts on what to do with bayer pixel formats that use 10 or 12 bits per pixel and are *NOT* padded out to 16. Such formats come from webcams (v4l) and still cameras. Does it make sense to have AV_PIX_FMT_BAYER_xxx for these variants, or better to process them in libavcodec, e.g. get_bits(gb,12) and shifting them to make 16-bits per pixel. [05:33] <michaelni> pross-au, iam in favor of aligning the MSB and using just a 16bit pix_fmt and storing the true bpp elsewhere [05:34] <pross-au> ok [05:48] <cone-750> ffmpeg.git 03Michael Niedermayer 07master:13131685f187: ffmpeg: avoid negative timestamps for DVD [11:08] <cone-159> ffmpeg.git 03Clément BSsch 07master:c1abfcd7d640: doc/resampler: remove .html from link text. [12:17] <durandal_1707> ping for my fate patch [12:40] <durandal_1707> looks like nobody use w64 format [13:03] <Compn> ehe [13:12] <ubitux> saste: http://pastie.org/private/flb6qqrsnw8iqvbydaf7q [13:12] <ubitux> did you see this? [13:13] <ubitux> saste: also, michael spotted an aliasing problem in ffprobe, which actually sounds fishy for some reason [13:13] <saste> ubitux, how do you get that warning? [13:14] <ubitux> "make" [13:14] <ubitux> :) [13:14] <saste> which aliasing problem [13:14] <ubitux> (i'm using gcc 4.7) [13:15] <ubitux> saste: http://fate.ffmpeg.org/log.cgi?time=20130101081420&log=compile&slot=armv7l-… [13:15] <ubitux> look for ffprobe.c [13:16] <ubitux> and this led me to: 03:31 >>> ubitux wonders where these children_ids are init& [13:16] <ubitux> but i didn't dig further [13:26] <durandal_1707> why nobody cares about #1163? [13:27] <ubitux> i didn't see the patch on the ml [13:28] <durandal_1707> some kind of patch is in ticket [13:29] <ubitux> :/ [13:32] <ubitux> erm [13:32] <ubitux> there is some crap going on here [13:34] <cone-159> ffmpeg.git 03Clément BSsch 07master:d079d1d368a9: ffprobe: do not escape '"' between single quotes. [13:38] <ubitux> durandal_1707: that code looks mostly broken and inefficient [13:38] <durandal_1707> #1539 can be closed [13:39] <durandal_1707> than fix it :/ [13:39] <ubitux> sounds like another bug for saste ;) [13:40] <ubitux> i need to fix one of mine now& [13:55] <durandal_1707> how typos can happen? [13:58] <cone-159> ffmpeg.git 03Martin Storsjö 07master:c76daa89ab91: rtmp: Return a proper error code in handle_invoke_error [13:58] <cone-159> ffmpeg.git 03Martin Storsjö 07master:33f28a3be309: rtmp: Add a function for writing AMF strings based on two substrings [13:58] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:c047a41bc5ff: Merge commit '33f28a3be3092f642778253d9529dd66fe2a014a' [13:59] <saste> durandal_1707, the invalid UTF-8 codes are stored in the metadata, fixing them at the application level is not a good idea [13:59] <saste> we should have some validation/fixing function in the demuxer maybe [13:59] <durandal_1707> +1 [14:00] <cone-159> ffmpeg.git 03Paul B Mahol 07master:8e4e532bb28e: lavfi/avcodec: fix typo [14:09] <cone-159> ffmpeg.git 03Martin Storsjö 07master:08225d01262b: rtmp: Add support for adobe authentication [14:09] <cone-159> ffmpeg.git 03Martin Storsjö 07master:c1ea44c54d83: rtmp: Add support for limelight authentication [14:09] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:9336a980bb26: Merge remote-tracking branch 'qatar/master' [14:50] <ubitux> oh. [14:50] <ubitux> nevcairiel: i think i found a better way :) [14:51] <ubitux> and btw, your patch doesn't really fix the problem [14:59] <ubitux> i think i'm going to simplify that whole timecode stuff [15:01] <ubitux> mmh wait your patch might be ok :) [15:02] <ubitux> almost [15:05] <cone-159> ffmpeg.git 03Paul B Mahol 07master:8c33cb5c77b9: fate: add nistsphere demuxer test [15:18] <nevcairiel> ubitux: well it fixed my problem, of chap nolonger being parsed :) [15:18] <cone-159> ffmpeg.git 03Hendrik Leppkes 07master:765158dd82a2: mov: fix parsing of the chap atom. [15:18] <ubitux> nevcairiel: yes it is good [15:18] <ubitux> thanks :) [15:19] <nevcairiel> the context variables are named a bit oddly [15:19] <ubitux> yes [15:19] <nevcairiel> should rather be tmcd_count or something like that [15:19] <ubitux> i was planning to change this [15:22] <ubitux> i don't think i can reference more than one timecode track per stream anyway so& [15:27] <cone-159> ffmpeg.git 03Nicolas George 07master:82541d833016: lavfi: merge all filtering code into ff_filter_frame. [15:36] <ubitux> is there a function to get a stream from its stream->id or we have to write the loop all the time? [15:53] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:9b53306178ca: cmdutils: we have 2013 [15:54] <durandal_1707> ubitux: what is id used for? [15:58] <ubitux> it helps me identify the references [16:06] <saste> ubitux: you confused nicolas with the AV_BASE64_SIZE thing! [16:06] <ubitux> haha [16:06] <ubitux> my stupidity is so evil! [16:07] <saste> maybe a longer name AV_BASE64_ENCODE_SIZE was better [16:10] <ubitux> saste: any objection to ffprobe stream->id? [16:11] <saste> ubitux, no but let me read the patch [16:17] <cone-159> ffmpeg.git 03Nicolas George 07master:69a1667b064b: ffmpeg: mark attachment streams finished. [16:22] <cone-159> ffmpeg.git 03Clément BSsch 07master:8d398f40b4a1: lavf/mov: simplify timecode track ref. [16:22] <cone-159> ffmpeg.git 03Clément BSsch 07master:c14f8a52b0aa: lavf/mov: fix timecode track reference usage. [16:43] <cone-159> ffmpeg.git 03Clément BSsch 07master:1dd4a21cb787: fate: re-enable fate-sub-mpsub. [16:43] <cone-159> ffmpeg.git 03Clément BSsch 07master:7d66bc792024: lavc/movtextdec: keep the min size instead of max to fix overread. [17:11] <saste> ubitux: ffprobe already prints the stream id [17:12] <ubitux> oh& [17:12] <ubitux> i'm stupid [17:12] <ubitux> thanks :P [17:17] <saste> ubitux, no, you're working too much ;-) [17:18] <ubitux> if only& [17:24] <cone-159> ffmpeg.git 03Clément BSsch 07release/1.0:68e48ed72e05: lavc/movtextdec: keep the min size instead of max to fix overread. [17:40] <cone-159> ffmpeg.git 03Hendrik Leppkes 07release/1.0:5f95b300c121: mov: fix parsing of the chap atom. [17:45] <saste> int len = strlen(p) * 6 / 8; [17:45] <saste> ^ seriously?? [17:46] <wm4> let me guess, this code calculates how large a buffer allocation is? [17:47] <saste> 6/8 baffles me [17:47] <durandal_1707> where? [17:47] <ubitux> simplifying fraction can be hard [17:47] <ubitux> durandal_1707: libavformat/rtp_asf.c [17:47] <ubitux> base64 stuff [17:47] <saste> yes [17:48] <saste> i'm not sure that size is correct [17:48] <ubitux> 6/8 looks bigger than 3/4 so it should be safe [17:49] <wm4> I like how it doesn't check the return value of the malloc either [17:49] <durandal_1707> like? [17:49] <saste> ubitux, lol [17:54] <saste> but yes the use of the function is correct [17:55] <saste> on the other hand the API is inconsistent [17:55] <saste> decode writes a not nul-terminated buffer [17:55] <saste> encode writes a nul-terminated buffer [17:56] <saste> AV_BASE64_SIZE() macro name is confusing [17:56] <saste> if you just want to encode a buffer (no null termination) you're out of lack [17:58] <saste> ^^ that's what happens when you "steal" VLC code :-) [18:06] <cone-159> ffmpeg.git 03Piotr Bandurski 07master:fde13052fb82: aiffdec: set block duration for QDM2 [18:34] <cone-159> ffmpeg.git 03Clément BSsch 07release/0.10:c3d7c805bc9c: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:34] <cone-159> ffmpeg.git 03Clément BSsch 07release/0.11:8bc0127b3d59: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:34] <cone-159> ffmpeg.git 03Clément BSsch 07release/0.7:20940780c726: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:34] <cone-159> ffmpeg.git 03Clément BSsch 07release/0.8:80b8dc30dc96: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:34] <cone-159> ffmpeg.git 03Clément BSsch 07release/0.9:42abae1125d7: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:38] <cone-159> ffmpeg.git 03Clément BSsch 07release/1.0:20c121c00747: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes(). [18:54] <durandal_1707> omg, why is bitrate int [18:57] <cone-159> ffmpeg.git 03Paul B Mahol 07master:ba1cbf40cebf: lavfi/lut: use ff_fill_rgba_map() [19:38] <cone-159> ffmpeg.git 03Nicolas George 07master:ecda1d3cbe1c: lavf: add data: URI scheme. [19:39] <cone-159> ffmpeg.git 03rogerdpack 07master:10a927756235: better filter error message when accidentally reusing outputs [20:00] <cone-159> ffmpeg.git 03Nicolas George 07master:951a03b3a8f2: ffmpeg: move stream_copy flags handling after attachments. [20:00] <cone-159> ffmpeg.git 03Nicolas George 07master:5bab99baea4f: ffmpeg: apply flags to attachments too. [20:00] <cone-159> ffmpeg.git 03Nicolas George 07master:8dbbaf568ed9: lavf/matroskaenc: respect bitexact for attachments. [20:17] <cone-159> ffmpeg.git 03Nicolas George 07master:5ed5e90f2ae2: lavc/dvdsubenc: improve color distance function. [20:20] <durandal_1707> anyone use PA-AF format ? [20:21] <kierank> what is that [20:22] <kierank> oh some mpeg thing [20:22] <kierank> never heard of it [20:22] <kierank> durandal_1707: i think pa-af is just a metadata store for other containers, no? [20:23] <durandal_1707> http://www.mb-1.jpn.org/paaf/ [20:25] <durandal_1707> it some crap that is misdetected as mov [20:26] <durandal_1707> it looks to be withdrawn [20:27] <durandal_1707> i found it on page of als [20:56] <kierank> withdrawn? [20:57] <kierank> looks active to me [20:57] <durandal_1707> but i cant get spec [20:57] <kierank> iso specs cost $$$$ [21:00] <durandal_1707> right, i ignored newer version here: http://www.iso.org/iso/home/store/catalogue_ics/catalogue_detail_ics.htm?cs… [21:46] <Compn> ack durandal left [21:46] <Compn> wanted to ask him how far he got with evrc [00:00] --- Wed Jan 2 2013

1 0

ffmpeg.log.20130101
by burek 02 Jan '13

02 Jan '13

[02:15] <pinkette> i have a video that video/audio is out of sync, how do i sync it permenantly [02:22] <ubitux> -async option, aresample filter with async option [05:41] <pinkette> i have a video that video/audio is out of sync: audio is ahead 500ms ; what is the command to sync this video [05:49] Last message repeated 1 time(s). [07:47] <zap0> what have you got so far? [10:17] <pinkette> i have a video that video/audio is out of sync: audio is ahead 500ms ; what is the command to sync this video [10:55] <ubitux> pinkette: 02:22:39 <@ubitux> -async option, aresample filter with async option [10:55] <ubitux> pinkette: 07:47:49 < zap0> what have you got so far? [10:55] <ubitux> please read instead of just asking the same question again and again& [10:58] <pinkette> ffmpeg.exe -i input.mp4 -async 500 output.mp4 is this right? [11:03] <ubitux> well if you want to shift the audio, maybe just -af asetpts=PTS+0.5/TB might be better [11:04] <ubitux> also see https://ffmpeg.org/ffmpeg-filters.html#aresample-1 [11:11] <pinkette> ubitux -af ? i thought you said to use -async [11:12] <ubitux> internally -async is doing a -af aresample=... [11:16] <pinkette> why is it taking so long [11:16] <pinkette> all i am doing is shifting [11:20] <pinkette> why is it reencoding audio/video [11:20] <pinkette> -af asetpts=PTS-0.5/TB did sync the video though [11:28] <ubitux> using filters means re-encoding [11:28] <pinkette> i cannot accomplish this without reencoding video and audio? [11:29] <ubitux> it would be nice, but i can't think of something like this right now [11:29] <divVerent> you MAY accomplish it by remuxing using some tools, but I am not aware of any [11:29] <divVerent> BTW, you only want to shift pts by that little? [11:29] <divVerent> a half time base? [11:29] <pinkette> okay can i accomplish this by reencoding audio only then and not touching the video [11:30] <divVerent> oh sorry, 0.5 seconds [11:30] <divVerent> if your output format is Matroska, mkvmerge can do this [11:30] <ubitux> seems to be mp4 [11:30] <ubitux> maybe mp4box? [11:30] <pinkette> i don't care if output is mp4 [11:30] <divVerent> it does it "somehow" without reencoding [11:30] <pinkette> i don't care if output is mkv [11:30] <divVerent> MP4Box may be able to do it too, see manpage [11:30] <ubitux> pinkette: -c:v copy to avoid re-encoding the video [11:31] <divVerent> in mkvmerge, see manpage for the keyword "delay"! [11:31] <divVerent> sorry, exclamation mark unintended ;) [11:31] <divVerent> is next to quote [11:32] <pinkette> i have mkvmerge , where is delay settings [11:32] <divVerent> option is -y [11:32] <divVerent> -y trackid:delay [11:32] <divVerent> track IDs may need guessing, but it'll either be 0 or 1 [11:33] <divVerent> guessing, or running mkvinfo on the mkvmerge'd mp4 :P [11:33] <divVerent> the delay amount is in milliseconds BTW [11:34] <ubitux> hey btw [11:34] <ubitux> why don't you seek into the audio stream? [11:34] <ubitux> by 500ms [11:34] <divVerent> because that tends to require reencoding [11:34] <divVerent> is my guess [11:35] <divVerent> e.g. in opus, it'd be REALLY evil as opus actually has a sort of rolling keyframe concept [11:35] <divVerent> basically, you can start decoding anywhere, but you need to decode a certain preroll amount of data to make the output correct [11:35] <pinkette> wow mkvmerge does have delay option and it works without reencoding [11:35] <ubitux> pinkette: can you try ffmpeg -i in.mp4 -ss:a 0.500 -c copy out.mp4? [11:36] <divVerent> ubitux: actually [11:36] <divVerent> I now see why it won't work :P [11:36] <divVerent> or rather, shouldn't work [11:36] <divVerent> it won't make the pts match up [11:36] <divVerent> but encode so it has 0.5 sec of video before the audio stream starts [11:36] <ubitux> i'd suggest to try anyway [11:36] <divVerent> or not? [11:36] <divVerent> sure, worth trying it is [11:36] <divVerent> IF it works, it's better than mkvmerge's shifting [11:37] <ubitux> not sure if -ss:a will seek only the audio thought =) [11:37] <divVerent> because the video will then not start with 0.5 sec of audio before video comes ;) [11:38] <pinkette> ffmpeg -i in.mp4 -ss:a 0.500 -c copy out.mp4 did not work [11:38] <ubitux> ok :( [11:38] <pinkette> but mkvmerge worked [11:38] <divVerent> just out of interest [11:38] <divVerent> is the mkv file bigger or smaller than the mp4? [11:39] <pinkette> 4M smaller [11:39] <divVerent> I noticed mkv'ing files save quite a lot of space (near one percent) on AVIs [11:39] <ubitux> :) [11:39] <divVerent> no idea how they achieve that [11:39] <pinkette> mkv has one 2 negative things [11:39] <pinkette> mkv has 2 negative things though [11:39] <divVerent> as "binary XML" doesn't sound very promising if you want to conserve space [11:40] <pinkette> aac with mkv ,you cannot see bitrate-audio where aac with mp4 can [11:40] <divVerent> pinkette: one is EBML, and the other is the (by default) 1ms granularity? [11:40] <divVerent> that may be a mere issue of the tools, though [11:40] <divVerent> and the other? [11:41] <pinkette> in windows-explorer i cannot see detail video on mkv but i can with mp4/avi/wmv [11:42] <pinkette> such as fps/resolution [11:43] <pinkette> how does mkvmerge fix audio shifting without reencoding [11:43] <ubitux> well since the ts are stored at container level& [11:44] <pinkette> so if i do ffmpeg -i mkvmergedfixedfile.mkv output.mp4, would the output.mp4 go out of sync ? [11:44] <ubitux> i don't think so [11:44] <ubitux> even with -c copy [11:45] <pinkette> i should try it; just in case [11:46] <ubitux> can you share the sample btw? [11:46] <ubitux> also, how come you ended up with shifted audiO? [11:47] <ubitux> audio* [11:47] <pinkette> i have no clue [11:47] <pinkette> okay output.mp4 did NOT go out of sync [11:47] <pinkette> wow, mkvmerge is powerful [11:48] <pinkette> i have a video where tint/hue is way off, can ffmpeg fix this [11:49] <ubitux> -vf hue [11:50] <pinkette> that requires reencoding the video. right? [11:50] <ubitux> of course [11:51] <pinkette> i wonder what causes to "audio out of sync" and tint/hue being way off [11:53] <ubitux> what's your source? [11:53] <pinkette> camera [12:10] <retard> true fact: almost all video comes from camera [12:10] <ubitux> not animes [12:11] <retard> ubitux: depends [12:11] <ubitux> not good animes [12:11] <retard> s/good/new [12:12] <pinkette> retard exactly [12:39] <pinkette> ffmpeg -i input.avi -vcodec copy -acodec copy output.mkv does not work [12:40] <pinkette> [matroska @ 03439600] Can't write packet with unknown timestamp [12:40] <pinkette> av_interleaved_write_frame(): Invalid argument [12:40] <Yulth> Happy new year for everyone! [12:41] <Yulth> Could anybody help me please with the error returned by this command? :) http://pastebin.com/UwjXT9JV [12:57] <ubitux> pinkette: try -fflags +genpts [12:57] <pinkette> what is that [13:03] <ubitux> generate presentation timestamps [13:03] <ubitux> because avi hasn't any afaik [13:04] <ubitux> (or missing some for b-frame stuff or something) [13:04] <ubitux> there is an opened trac issue to make it default when necessary [13:05] <ubitux> Yulth: [libaacplus @ 0x805d41620] libaacplus doesn't support this output format! [13:05] <ubitux> libaacplus: bad aac setting: br:40000, AACch:65536, AACsr:22050 [13:05] <ubitux> -ab 40k you sure that is correct? [13:06] <ubitux> AACch:65536 this is funny :) [13:16] <Yulth> ubitux: of course, it works using libfdk_aac -profile:a aac_he. So, is not the same for libaacplus? [13:17] <ubitux> maybe these settings are just not supported [13:17] <Yulth> mmm, perhaps libaacplus doesn't suport HE-AAC encoding... T_T [13:18] <Yulth> so, is libfdk_aac the only way to encode HE-AAC? [13:18] <Mavrik> um. [13:18] <Mavrik> "AAC+" == "HE-AAC" [13:18] <Mavrik> what makes you think a library named "AAC+" wouldn't be able to encode that format? :P [13:19] <Mavrik> the glue code is kinda buggy [13:19] <Mavrik> try removing channel spec or the profile spec [13:20] <Yulth> let my try it [13:20] <Mavrik> since the lib obviously thinks you're trying to set 65535 channels. [13:21] <Mavrik> FDK-AAC is more stable and provides output for wider range of audio though [13:22] <Yulth> it doesn't work neither removing -ac nor -profile:a [14:12] <gxk> i am beginning project which streams live h264 video by mpeg2ts/udp. the streaming with mpeg2ts is what i am missing. i don't have audio, but there is metadata. is ffmpeg the right project to start? [14:13] <Mavrik> yes, ffmpeg can stream MPEG2-TS/UDP with H.264/AAC in it [14:17] <gxk> is there demo/example for it? may be not exact but close enough [14:18] <Mavrik> hmm, not really [14:18] <Mavrik> ffmpeg -i <file> <h264 encoding parameters> -bsfs h264_mp4toannexb -f mpegts udp://127.0.0.1:5000 [14:18] <Mavrik> that usually does the trick [14:30] <gxk> is there example how to add ffmpeg for live h264 and from file? [14:33] <gxk> Mavrik: the option '-bsfs' or '-bsf' or '-vbsf'? [14:33] <Mavrik> er, yea, sorry [14:34] <Mavrik> any of -bsf, -vbsf, -bsf:v [14:34] <Mavrik> depending on your ffmpeg version [14:37] <gxk> Which encoding parameters to use, if i want to send h264 from file? [14:42] <Mavrik> um [14:42] <Mavrik> whichever you want depending on quality and bitrate you have? [14:44] <BtbN> works great for me without any special parameters except my desired bitrate. [14:44] <BtbN> and resolution [14:52] <gxk> is there option to start from beginning of a file when reach end of a file? [14:54] <sacarasc> There might be a -loop [14:55] <gxk> sacarasc: nope :( [15:41] <gxk> are there well known h264 files to test with? [16:17] <espr3ss0> hi [16:20] <espr3ss0> anyone here ? [16:28] <espr3ss0> what's the "--enable-nonfree" option ?, does ffmpeg cost ? o_O [16:29] <sacarasc> Some licences are not 100% free. [16:29] <sacarasc> To use the libraries with these licenses, you have to use that option. [16:29] <espr3ss0> I'm assuming that I'm installing libraries to be able to work with files e.g. audio track 'mp3' i need "Lame" & so on & so forth [16:33] <espr3ss0> is there a location for DivX Library ? [16:39] <sacarasc> Depends where you installed it. [16:39] <espr3ss0> fair enough [16:41] <espr3ss0> I have a "MacBook 13" White" & have installed "OS X SnowLeopard" ... do i need to set "--arch=x86_64" ? [16:42] <espr3ss0> I'll set it anyway [16:42] <klaxa> hmm i guess set --arch=native [16:42] <klaxa> if that's possible [16:42] <sacarasc> You shouldn't need to do that at all. [16:42] <klaxa> unless you want to cross-compile [16:43] <espr3ss0> noo [16:43] <espr3ss0> is this missing anything ? ... "./configure --enable-gpl --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libmp4v2 --disbale-mmx" [16:43] <espr3ss0> wanting to only setup for common .avi's & mp4's [16:44] <espr3ss0> or more specifically from ".avi" to ".mp4" [16:44] <creep> how do guys run their ddr3 1866 with i5 ? [16:59] <espr3ss0> "unknown option '--enable-libmp4v2'" ? o_O [16:59] <sacarasc> ffmpeg doesn't need it. [16:59] <sacarasc> It has its own MP4 writing library. [17:00] <espr3ss0> oh [17:00] <espr3ss0> i only assumed because when I tried to "no transcode" an avi to mp4 the mp4 video was very jumpy :-/ [17:03] <espr3ss0> brb, afk [17:22] <gxk> q [17:25] <espr3ss0> hmm, once again ... where can i download "libxvid" from ? [17:25] <espr3ss0> i will paste some info but not yet [17:27] <gxk> while streaming from h264 file, got in command line status with fps=298. the comand line: ffmpeg -i ./test.h264 -s 1280x720 -b:v 2M -r 30 -bsf h264_mp4toannexb -f mpegts udp://127.0.0.1:5004 [17:55] <Mavrik> gxk, add "-re" at the start of parameters [17:55] <Mavrik> gxk, this will tell ffmpeg to stream with playback speed [17:57] <gxk> Mavrik: thanks [18:31] <gxk> Mavrik: vcl shows statistics, but still no video. it also shows frame rate 60, while i am sending 30. is this a problem? [18:34] <Mavrik> well, probably. [18:56] <gxk> Mavrik: it was vlc problem on my Ubuntu. not installed properly. [18:59] <espr3ss0> if i'm wanting to extract "h.264" video track, do I need the x264 library "libx264" ? [19:00] <JEEB> no [19:00] <espr3ss0> also seen a method of extracting to "video.h264", but no idea how to use that after for an input [19:00] <JEEB> x264 is just for encoding [19:01] <JEEB> ffmpeg -i input -an -sn -c copy -f h264 out.h264 [19:01] <JEEB> if it's mp4, you will need the mp4-to-annexb filter [19:01] <JEEB> because mp4 contains the H.264 stream in a diff. way [19:01] <espr3ss0> oh [19:02] <JEEB> (and raw H.264 is in a format called Annex B because it's defined there in the specification) [19:02] <JEEB> http://ffmpeg.org/ffmpeg-bitstream-filters.html#h264_005fmp4toannexb [19:02] <JEEB> this one [19:02] <JEEB> you don't need it for other formats [19:02] <JEEB> just mp4 [19:03] <JEEB> (I don't know exactly why ffmpeg doesn't do this conversion automatically when needed) [19:04] <espr3ss0> i'm confused [19:04] <JEEB> ask away [19:04] <espr3ss0> if i'm wanting to simply copy from the ".avi" container to an ".mp4", how can i achieve this ? [19:05] <JEEB> oh [19:05] <JEEB> I thought you wanted to really /extract/ the H.264 track :D [19:05] <JEEB> which means demuxing from a container to a raw Annex B stream [19:05] <espr3ss0> well, i do want to be able to demux [19:05] <JEEB> basically you want to remux the video, what about the audio? [19:06] <espr3ss0> mainly only common rips [19:06] <JEEB> ... [19:06] <espr3ss0> but also some mp4's arn't done in the right way, which causes me some problms [19:06] <JEEB> I asked what you wanted to do to the audio [19:06] <espr3ss0> my main aim is no transcoding [19:06] <JEEB> ok [19:07] <JEEB> ffmpeg -i input -c:v copy -c:a copy out.mp4 and it should happily copy stuff over in most cases, except when you have a video or audio track that isn't such that can be put into "MP4" [19:08] <espr3ss0> oh, including both Video & Audio [19:08] <JEEB> that of course doesn't select all tracks from the input clip, only the "best" video and "best" audio track [19:08] <espr3ss0> i tried ... ffmpeg -i file -vcodec copy -acodec copy outfile.mp4 [19:09] <JEEB> that should work too I guess [19:09] <JEEB> (that's the older syntax) [19:09] <espr3ss0> the video comes out slow n jumpy [19:09] <espr3ss0> :-/ [19:09] <JEEB> what are you playing it in? [19:09] <JEEB> also... inb4 PTS values need to be generated if it's H.264 in avi [19:10] Action: JEEB checks the logs, IIRC there was something about that today [19:10] <espr3ss0> QuickTime, VLC works fine [19:11] <JEEB> QuickTime is known to be problematic, but you could try adding -fflags +genpts in case of AVI + H.264 [19:11] <espr3ss0> will be playing the mp4's on iPad (gen 3) [19:11] <JEEB> the iPad might actually play it better than QuickTime [19:11] <JEEB> funny enough :P [19:11] <espr3ss0> strangly does :-) [19:11] <JEEB> QuickTime is just like that, they've always had more or less problems with H.264 and friends [19:12] <JEEB> although lately IIRC they did somewhat better after they switched mostly to hardware decoding [19:12] <JEEB> which of course is limited in some ways [19:12] <JEEB> but should still be better in other ways [19:12] <JEEB> (than what QT used to be) [19:12] <espr3ss0> heh [19:13] <JEEB> also Apple is known to ignore a lot of the bug reports they get about these issues [19:13] <JEEB> so meh [19:13] <espr3ss0> :-/ [19:13] <espr3ss0> erm, quick Q ... [19:13] <espr3ss0> is this type of error normal ? "libavformat/metadata_compat.c:143: warning: language is deprecated (declared at libavformat/avformat.h:602)" [19:13] <espr3ss0> when making ffmpeg [19:13] <JEEB> it's a warning, not an error [19:14] <espr3ss0> oh, does it matter ? [19:14] <JEEB> and yes, it's OK -- it just tells you that that file uses something that is declared deprecated in the public API [19:14] <JEEB> and since it's within ffmpeg itself [19:14] <JEEB> it will get updated/removed when the deprecated API is removed [19:15] <espr3ss0> depricated mean ignored /rejected ? [19:15] <JEEB> no, it means old [19:15] <espr3ss0> ooooh i c [19:15] <JEEB> that there (generally) is a new way to do it [19:15] <JEEB> as the file name says _compat (compatibility) [19:15] <JEEB> I would guess it contains some old stuff [19:15] <JEEB> basically with one version you deprecate [19:15] <espr3ss0> gonna have to be bbs, thanks 4 ur help :-) [19:15] <JEEB> and then in the next version you remove [19:15] <JEEB> np [19:16] <espr3ss0> cool [19:33] <Sashmo> doese anyone know if there is a way to add the soruce name to the file name output?? http://pastebin.com/D7GXGYSx [19:35] <Sashmo> I'm assuming theres a bash expression that can do it..... no? [19:51] <grepper> Sashmo: its unclear what you want - could you give an example of an output name you would want ? [19:51] <Sashmo> basicly, I want my source file name to be the same as the screenshot file name, but not have to make a bash script to make it work [19:58] <grepper> no idea what you are talking about without an actual example, as I asked [19:59] <sacarasc> They want to have the output name have the same basename as the input name. [19:59] <sacarasc> Without using a whole script, it seems. [20:00] <Sashmo> here http://pastebin.com/UhqU918c [20:03] <grepper> Sashmo: ${source##*/} would be the basename of the file without the path [20:04] <grepper> "${source##*/}" (should be quoted in case of spaces in name) [20:05] <Sashmo> grepper: so like this? http://pastebin.com/84h2b9dA [20:10] <grepper> source=/home/sashmo/foo.avi ; ffmpeg -i "$source" ... -y "${source##*/}_%d.jpg" [20:10] <grepper> or somesuch [20:10] Action: grepper bbl [20:10] <Sashmo> grepper: thanks [20:10] <Sashmo> will try [20:14] <grepper> if you don't want the extension in the filenames, (ie. just foo_01.jpg instead of foo.avi_01.jpg) you can do: source=${source##*/}; source=${source%.*} , before feeding the variable to the ffmpeg command line [20:15] <Sashmo> I was just trying that now.... the problem is that the source is a URL..... [20:16] <espr3ss0> JEEB: u still there ? [20:40] <espr3ss0> put an mp4 track (x264) and an aac together with "copy" but audio has 0 channels & should have 2 [20:41] <espr3ss0> h.264 ! 8-/ [20:44] <espr3ss0> still need xvid & divx to work [20:45] <espr3ss0> ... ping ! ... pong ! ... o_O [21:46] <espr3ss0> can anyone say if xvid = divx ? [21:48] <espr3ss0> hmm guess not [21:48] <espr3ss0> all i can see is enabling "libxvid" & no divx ! :-( [21:49] <creep> a girl starts to turn into a dragon after 17-th birthday ? [21:53] <espr3ss0> ffmpeg, after "./configure etc." is it necessary to "make" & install every time ? [22:23] <octocpp> can ffmpeg use multiple threads to encode with libmp3lame ? I use -threads 0 or threads 8 and it does the same thing, I only see one process in htop? [22:29] <BtbN> lame is only single threaded. [22:33] <octocpp> ok, thanks, Im using winff ans a gui, would be cool if it ran a script for every song i was trying to convert, or maybe ran 4 seperate scripts and split the songs up 4 ways. That would nice. [22:35] <sacarasc> octocpp: Try using dbpoweramp for Windows... It is multithreaded. [22:36] <octocpp> arg, im in linux right now though. [22:36] <octocpp> soundKonverter worked really good, but it was not able to get the tags right for some reason. [22:37] <octocpp> Basically lost them all [22:40] <espr3ss0> Q: can ffmpeg correct "B Frames" ? ... "[mpeg4 @ 0x101024400] Invalid and inefficient vfw-avi packed B frames detected" [22:49] <espr3ss0> QuickTime "invalid sample description", fix with ffmpeg ? [22:49] <espr3ss0> plays fine in VLC [22:58] <espr3ss0> ".avi mpeg-4 xvid" to "m4v(mpeg-4) .mp4", same thing ? vcodec copy [22:58] <espr3ss0> this is too complicated ! ;-p [23:14] <misterno> what is absolute best quality encoding? [23:15] <sacarasc> Lossless. [23:16] <misterno> what is absolute best quality encoding for around 800 MB size movie? [23:17] <sacarasc> Two pass x264? [23:17] <misterno> there were some movies in MKV container around only 400MB that had amazing quality, how is that possible? [23:18] <sacarasc> Some codecs are better at compressing than others. [23:18] <misterno> do you have any idea which codec might be used in those 400 MB MKVs? [23:18] <sacarasc> Probably H264 encoded by x264. [23:18] <misterno> i can find that out using media info right? [23:18] <sacarasc> But if you have the files, you could see. [23:20] <misterno> i don't want to talk about piracy but i must just touch it because i want to understand quality.... you are saying probably one of best is x264 twopass, how come groups release xvids is it considered good quality or not? [23:21] <sacarasc> No sane person uses XviD or DivX any more. [23:21] <Adys> I have an mkv movie with two audio tracks (one french, one english); I want to delete the french one. any idea how? [23:22] <misterno> ok, are there tools you can recommend to me to use to get this x264 two pass encoding, handbrake? [23:22] <leoj3n> misterno: x264 is in all the tools, so it 2 pass [23:22] <sacarasc> Adys: ffmpeg -i blah.mkv -map 0:0 -map x:1 -c copy output.mkv. Replace x with the track number of the English audio. [23:22] <Adys> i see, thanks :) [23:22] <misterno> ok, and for container you guys recommend mkv? [23:23] <misterno> even though it is not supported by standalone dvd players [23:23] <misterno> mkv is superior because it can contain subtitles and multiple audio tracks right? [23:25] <JEEB> yes, it's superior for such usage because of its capability of keeping various types of streams within it. Unlike the "MP4" family of containers which is unfortunately quite limited by the fact that not much in the end has been specified for it [23:26] <misterno> how do you guys go about setting up resolution for dvd movie, and then resolution for lets say bluray [23:26] <misterno> is it better to have lower res and higher quality or higher res [23:27] <JEEB> uhh, I just use x264's crf rate control to more or less control the "general level of quality" [23:27] <JEEB> I don't encode for set file sizes :s [23:27] <misterno> but do you manually set resolution yourself? [23:28] <misterno> i haven't encoded yet so im a bit blank on it, although i did try in the past then i realized so many options [23:28] <JEEB> for DVDs I usually just deal with interlacing if there's any, crop and set aspect ratio, for blu-rays I generally try to see how much actual detail there is in the source [23:28] <misterno> do you do differently for action movie vs drama? [23:29] <JEEB> no [23:29] <misterno> doest it matter if picture is moving fast vs slow [23:29] <JEEB> no [23:29] <misterno> hmm i was quite sure that mattered in the past for quality settings [23:30] <misterno> perhaps with xvid [23:30] <JEEB> yes, x264's crf will take into notion how "fast" the stuff is going in the scene and somewhat higher quants get used for fast scenes... but uhm... that has NOTHING to do with resolution [23:31] <misterno> yes but if you look at file size then lower resolution alows more space for quality [23:31] <misterno> *fixed file size [23:31] <JEEB> yes, if you are running out of bits (aka have set your file size lower than it'd be appropriate for the content [23:31] <klaxa> meh, either you speak bitrate or quality [23:32] <misterno> oh you reminded me, what do you consider ok minimum bitrate? [23:32] <JEEB> there is no such thing [23:32] <misterno> you dont have personal preferences? [23:32] <misterno> i always have personal preferences on anything [23:32] <JEEB> I stopped throwing numbers at a wall with x264 [23:32] <JEEB> about four years ago [23:32] <misterno> ok [23:33] <JEEB> the minimum bitrate for a given source is the bit rate you get with the highest crf value that still looks good to you [23:33] <JEEB> that is personal, because your eyes are your eyes [23:33] <JEEB> if you do not have a strict limit you have to abide to, you use crf. Simple as that, and doesn't need a second pass [23:33] <misterno> yes i understand thats why i asked for your own personal preference :) [23:33] <misterno> i know you dont want to make claims this is better than that or so [23:34] <misterno> can i just ask, is there much difference in quality between lets say full blown adobe premiere vs some open source tol? [23:34] <misterno> *tool [23:35] <JEEB> well, adobe stuff uses mainconcept [23:35] <JEEB> and mainconcept came second or so after x264 in MSU's tests [23:35] <misterno> oh man theres so much info on this for someone new to this it is mind bogling getting grasp of it [23:36] <JEEB> it's not really that hard nowadays [23:36] <misterno> so i have to google these MSU tests, this is first time i hear on this [23:37] <JEEB> grab a thing you want to encode, encode around 5000 frames of it with various crf values after doing your thing (handled interlacing in one way or another, cropped, possibly resized if the source is of lower detail than the source media [f.ex. Japanese animation often is 540p-720p area, not the 1080p that's on the Blu-ray discs f.ex.]), starting from crf 23 [23:37] <JEEB> if it looks good, go up [23:37] <JEEB> if it looks bad, go down [23:38] <JEEB> then when you've found the highest crf value that still looks good for you [23:38] <misterno> jeeb what is the story with interlacing, is that because of TVs? and then we have to deinterlace for pcs? [23:38] <JEEB> oooh boy [23:38] <misterno> ok nvm ill just google that [23:38] <JEEB> nah [23:40] <JEEB> anyways, let's just keep to progressive content for now [23:41] <leoj3n> misterno: http://www.youtube.com/watch?v=j29b7B6CSNM [23:42] <JEEB> 1) you crop the source [overcropping is better than undercropping] 2) you resize the video as you see fit [features that might be reasons to downscale: bad quality of source, lack of detail and so forth and so forth] [23:43] <misterno> how come movies get nuked when they crop? again im not promoting piracy i just want to understand quality [23:43] <JEEB> aka you define the settings for these two, these have nothing to do with the actual encoding, this is basically handling the source before feeding it to the encoder [23:43] <foonix_> sometimes i try to find good resolution (on upscaled sources) by doing eye test: resize original->lower->original a compare how much detail is lost [23:44] <foonix_> misterno: on things like 960x544 crf 16 is awesome, but final size might not be ok for everyone.. so just test whats ideal for you [23:45] <JEEB> foonix_, STOP TELLING PEOPLE TO JUST USE A RANDOM NUMBER [23:45] <JEEB> THAT IS BAD [23:45] <JEEB> PLEASE [23:45] <JEEB> FOR THE LOVE OF DERP STOP THAT [23:45] <misterno> whoa [23:45] <misterno> no need to get angry jeeb :) [23:45] <foonix_> i told him some idea, ok it included numbers [23:45] <misterno> first im complete newb to this, i dont understand half what you guys are saying and will have to google it [23:45] <JEEB> you gave him a number that might look good from the get go [23:46] <JEEB> that means he might NEVER TRY HIGHER NUMBERS [23:46] <JEEB> 23 is better off because it generally should lead to better compression and might not look good from the beginning [23:46] <misterno> so you prefer to stick to 570p 720p? [23:46] <foonix_> if its ok for him, i dont have problem with that [23:46] <JEEB> oh for fuck's sake [23:46] <misterno> or you talking cfr? [23:46] <JEEB> we're talking about crf values [23:47] <misterno> alright [23:47] <JEEB> misterno, I have no idea the procedures the "scene" has but proper cropping is at max a few pixels over if the area of the black borders changes around [23:47] <JEEB> and that should be just fine unless you are encoding for some plastic boxes, although I would guess that'd work in most cases as well [23:47] <JEEB> foonix_, do you not see at all why giving a newbie a too low crf value to start with might be problematic? At all? [23:48] <JEEB> It's good if the person ends up finding that for a given source X crf 19 is the highest value that still looks good [23:48] <JEEB> I mean, you can test various crf values with like 2500-5000 frames' worth of content [23:48] <foonix_> well then dont give him 5k frames sample too [23:48] <JEEB> also x264 has a default, 23 [23:49] <foonix_> kinda absurd [23:49] <JEEB> well generally I tell people to cut a few hundred frames' samples from various parts of the source, but I have no idea how you can cut easily with ffmpeg like that [23:49] <misterno> yes but wait, if i have fixed file size then CRF will adjust to that wont it? [23:49] <JEEB> so I just tell them to ss to some part of the movie that is generally like the source is [23:49] <JEEB> no [23:49] <JEEB> CRF is "constant quality" [23:49] <JEEB> has no file size limitations set to it at all [23:49] <JEEB> other than possibly vbv [23:50] <JEEB> but that's a separate discussion [23:50] <misterno> so if i use fixed file size then no crf [23:50] <JEEB> yes, but you seemingly have no idea to what bit rate you want to limit yourself [23:50] <JEEB> thus I see no reason to keep your to bitrate-based encoding [23:50] <JEEB> *you to [23:50] <misterno> i can get that idea by looking at bitrates of movies i see from others [23:50] <JEEB> ... [23:51] <JEEB> > copycat'ing scene "rules" [23:51] <JEEB> let me facepalm here [23:51] <JEEB> hard [23:51] <JEEB> unless you have to participate in the "scene" [23:51] <misterno> dont face palm just because someone is interested in learning more [23:51] <JEEB> you have no reason to limit yourself to the shit they use [23:52] <misterno> and i probably wont but i wanna try everything [23:52] <JEEB> because the most I see of those "scene" folk is being utter retards who have to be limited to not the most crappiest settings because everyone wants to be the first and thus people would encode with the worst settings possible otherwise [23:52] <misterno> i wanna try both fixed file size and not [23:53] <misterno> ok can i ask a bit of different question, i would like to encode file best i can for upload to youtube, dont want to waste bandwidth by uploading uncompressed file [23:53] <misterno> do i still go for x264 two pass? [23:53] <JEEB> foonix_, also I see your comment about "Giving 5000 frame sample" absurd being weird in itself, you should be able to grasp the general quality that you will be getting from a certain crf value with that. 500 frame parts from various parts of the source would be better, but as I said I have no fucking idea how to do that with ffmpeg [23:54] <misterno> i have noticed some people make excellent youtube quality videos and some dont [23:54] <JEEB> misterno, youtube is going to rape your thing anyways so just use the lowest crf that doesn't seem to be ending too big, lossless being the biggest thing. x264's lossless should still be way smaller than uncompressed video. [23:54] <misterno> alright [23:54] <JEEB> also audio could be something lossless if you really want to give youtube the best thing to rape [23:57] <misterno> what do you think of this comparison http://www.tools4movies.com/2012/03/crf/ [23:57] <misterno> and that tool [23:58] <JEEB> what the flying fuck [23:58] <JEEB> why the fuck is he saying with "CRF" off [23:59] <misterno> don't ask me :) [23:59] <misterno> you're the expert here [00:00] --- Wed Jan 2 2013

1 0

ffmpeg-devel.log.20121231
by burek 01 Jan '13

01 Jan '13

[00:09] <cone-681> ffmpeg.git 03Clément BSsch 07master:faa94061dd72: Add SubViewer v1 subtitles demuxer and decoder. [00:17] <michaelni> <ubitux> btw, sub/MicroDVD_capability_tester.srt can be removed <--- was it never used ? [00:17] <ubitux> it was badly named at first, i don't remember if it was part of the tests, maybe at some point, but that's a long time ago [00:18] <michaelni> if it was used then it cannot be removed [00:21] <ubitux> ah? :( [00:21] <michaelni> consider bisect with fate ... [00:22] <ubitux> it's really supported? [00:23] <ubitux> i guess you would fate run a specific test for bisect [00:23] <ubitux> of course it might be a pain for tracking microdvd regressions [00:44] <cone-681> ffmpeg.git 03Clément BSsch 07master:1f265f52050e: microdvd: sanitize AVPackets. [00:44] <ubitux> here we go [00:44] <ubitux> DONE. [00:46] <wm4> ok how do I use this shit [00:46] <wm4> is converting to ass still done by "decoding"? [00:47] <ubitux> yes [00:47] <ubitux> again, you can look at vf subtitles [00:48] <wm4> right [00:48] <ubitux> wm4: also note that charset/encoding/codepage/etc is not done [00:48] <ubitux> (fribidi, ..) [00:48] <wm4> why would fribidi be needed here? [00:49] <ubitux> no idea, i didn't look into this stuff yet [00:49] <wm4> from what I understand, just passing the text to libass should be enough [00:49] <wm4> though there may be weird issues with the default base direction... [00:49] <ubitux> mmh yeah right, if libass is able to deal with it [00:50] <ubitux> note that if the input is not utf-8, it will likely be broken [00:50] <ubitux> a lot of subtitles are not in utf-8, and since we don't do any convert to utf-8, libass might not like the input [00:51] <wm4> subreader.c in mplayer does conversion [00:51] <ubitux> yes i know [00:52] <wm4> the fribidi stuff there is only needed for the internal sub renderer, I think [00:52] <wm4> while with -ass it's not needed and/or causes corruption (I don't quite remember) [00:52] <ubitux> see http://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/136506.html for the next plan [00:53] <ubitux> ok, i'll focus on charset convert then [00:53] <wm4> libass also has some charset stuff AFAIK, but you'll have to read the libass source, heh [00:54] <ubitux> ah? that's interesting if so [01:02] <ubitux> michaelni: so, do you have sub/MPSub_capability_tester.sub? [01:12] <Compnn> is that a sample ? [01:12] <Compnn> what links to it ? [01:13] <ubitux> i sent him a link a while ago in private [01:13] <ubitux> i think. [01:13] <Compnn> its not on ffmpeg box [01:13] <Compnn> at least not that i see [01:14] <michaelni> ubitux, dunno, locate doesnt find it on my disk [01:14] <ubitux> yes that's why i'm asking [01:14] <ubitux> ok [01:14] <Compnn> ubitux : i mean, who linked to it in the first place ? [01:14] <ubitux> http://b.pkh.me/MPSub_capability_tester.sub [01:14] <ubitux> if you guys could put this in fate-samples/sub/... :) [01:15] <Compnn> that looks familiar [01:15] <ubitux> Compnn: yes that's the default sample from mplayer [01:15] <Compnn> http://www1.mplayerhq.hu/DOCS/tech/mpsub.sub [01:15] <Compnn> bah [01:15] <ubitux> i may have done slight changes, i don't remember [01:16] <ubitux> Compnn: anyway, it's used for one test, but since it was forgotten i disabled the test [01:16] <michaelni> ubitux, uploaded [01:16] <ubitux> michaelni: thanks a lot :) [01:16] <ubitux> i'll re-enable the test in about two days [01:16] <michaelni> np [01:16] <michaelni> ok [01:41] <cone-681> ffmpeg.git 03Clément BSsch 07master:21b6991cd530: lavfi/Makefile: make FFLIBS list consistent. [01:41] <cone-681> ffmpeg.git 03Clément BSsch 07master:8d06e83d16eb: lavfi/Makefile: add missing dependencies to FFLIBS. [01:44] <saste> - av_basename and av_dirname [01:45] <saste> since when we add new functions in the changelog? [01:45] <ubitux> & [01:45] <ubitux> i didn't even realize [01:45] <ubitux> feel free to move it to APIChanges [02:33] <cone-681> ffmpeg.git 03Clément BSsch 07master:fe5a2fc46c55: Changelog: remove av_{base,dir}_name addition. [02:48] <ubitux> 'Enca' (Extremely Naive Charset Analyser) [02:48] <ubitux> 'wasn't aware what that meant :) [02:49] <ubitux> btw, too bad we don't have subtitles in lavfi [02:49] <ubitux> we could use it to rescale the subtitles pts with a setpts filter [03:09] <ubitux> what's the point of check_lib instead of check_lib2? [03:09] <ubitux> it seems check_lib2() is testing the function using the include [03:10] <ubitux> while check_lib is doing a first test with the include [03:10] <ubitux> and a second one with the function [08:14] <cone-668> ffmpeg.git 03Paul B Mahol 07master:9a742826443a: Sony Wave64 muxer [09:37] <ubitux> the charset conversion was simpler than expected [09:37] <ubitux> at least a naive one [10:03] <durandal_1707> our news page should be in separate html [10:04] <durandal_1707> currently i cant link to news from g+ because it will show description of project and not actual news [10:10] <cone-668> ffmpeg.git 03Clément BSsch 07master:e3c25860dc6e: lavc/utils: reindent in avcodec_decode_subtitle2(). [10:11] <cone-668> ffmpeg.git 03Clément BSsch 07master:72ff8ee158c7: lavc: correctly set AVSubtitle format for text based subtitles. [10:23] <ubitux> wm4: do you want the charset convertion to support enca as well when available? [10:57] <ubitux> wm4: enca is going to be a bit complicated [10:57] <ubitux> i suggest that you keep that at mplayer level [10:58] <ubitux> i may add it to the subtitles support [10:58] <ubitux> saste: do you mind if i rename vf_ass.c to vf_subtitles? [10:59] <saste> ubitux, no i don't [11:00] <ubitux> ok :) [11:00] <durandal_1707> why are there two filters with same description? [11:01] <saste> yes it should be fixed [11:01] <durandal_1707> in that file [11:02] <saste> adding "ASS subtitles" to ass should be enough [11:41] <ubitux> arg utf16 [11:41] <ubitux> fuck that shit :( [11:42] <nevcairiel> lol [11:43] <nevcairiel> iconv doesnt do utf16? [11:43] <durandal_1707> michaelni: apad pads silence at end only? [11:44] <durandal_1707> it could be nice to pad start and/or in middle [11:46] <saste> durandal_1707, you can use aresample for that [11:49] <durandal_1707> how? [11:55] <saste> durandal_1707, aresample=min_comp=0.001:min_hard_comp=0.100000 [11:56] <saste> min_comp enable compensation, and min_hard_comp enable HARD compensation [12:01] <ubitux> nevcairiel: i guess it does, the problem is that i'm using strlen() [12:01] <ubitux> (and assuming ascii compatible) [12:01] <ubitux> and afaik utf-16 files are most of the time half composed of 0x00 [12:02] <ubitux> wtf these guys were thinking.. [12:03] <saste> ubitux, UTF-16 is not that broken [12:03] <nevcairiel> if you have normal ascii chars in utf-16, there are indeed a lot of 0x00 in there [12:03] <ubitux> just half of it [12:04] <saste> it is an extended ASCII, with a simple encoding (well an identical encoding indeed), and who cares about text files size anyway? [12:04] <ubitux> who cares about using strlen() [12:04] <ubitux> :p [12:04] <nevcairiel> i'm just used to dealing with utf16 because i deal with windows, which deals with wchars =p [12:04] <durandal_1707> there is code that reads 16 utf and converts it to 8utf [12:05] <saste> ah strlen() [12:05] <saste> i assume there are specific functions to deal with that [12:05] <ubitux> saste: and any other string functions& [12:05] <saste> like wstrlen() or such [12:05] <ubitux> yes [12:05] <nevcairiel> wcslen i think [12:05] <ubitux> special cases [12:05] <ubitux> nothx [12:05] <ubitux> i'd better demux directly that stuff into utf-8 [12:05] <saste> ubitux, can you convert to UTF-8 and always assume UTF-8 internally? [12:06] <ubitux> i'm assuming !utf16 now [12:06] <ubitux> saste: the patch adds charset convert at codec level [12:06] <ubitux> the problem is that it works only with ascii compliant charset [12:06] <saste> wait, i have to actually read the patch... [12:07] <ubitux> now what i'd like to do is make the text demuxers actually do a first pass from utf-16 to utf-8 if necessary [12:07] <ubitux> so the following charset convert and any normal string processing will work [12:08] <durandal_1707> hmm, why you need strlen at all... [12:08] <ubitux> rects[i]->ass [12:09] <ubitux> decoded text subtitles [12:09] <ubitux> and actually, we are dealing with text subtitles [12:09] <ubitux> we are assuming ascii compliant string all over the code [13:07] <cone-668> ffmpeg.git 03Clément BSsch 07master:87350d6b44b2: lavc/textdec: fix subviewer1 preproc check. [13:17] <cone-668> ffmpeg.git 03Diego Biurrun 07master:015da965a68b: libavcodec/utils: Add braces to shut up gcc warnings [13:17] <cone-668> ffmpeg.git 03Anton Khirnov 07master:185c2ef8495c: cavsdec: cosmetics, reformat [13:17] <cone-668> ffmpeg.git 03Anton Khirnov 07master:64bb3fa6b7f5: cavs: cosmetics, reformat top part [13:17] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:6ddda83c3f93: Merge commit '64bb3fa6b7f5b2db22b7577585d5861be24eebd8' [13:38] <cone-668> ffmpeg.git 03Anton Khirnov 07master:7d8482640058: cavs: deMpegEncContextize [13:38] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:ca95d1b32df1: Merge commit '7d8482640058d66867602a6f58446c9342d9d146' [13:40] <durandal_1707> who owns darwin clang2.9 x86_32 fate machine? [13:41] <durandal_1707> michaelni: that machines that where red all the time have been removed? [13:44] <michaelni> durandal_1707, ask the owner of these boxes i dont know, i didnt remove any [13:44] <michaelni> but it can be a bug in fate too, ive seen machies disappear previously with noone doing anything [13:44] <durandal_1707> yes, but interface is so missing that i cant find such info [13:46] <durandal_1707> saste: are you against porting noise and instead want noise source? [13:46] <saste> durandal_1707, i'm not against porting noise [13:46] <saste> why should I? [13:47] <durandal_1707> well you did some work on some noise filter iirc [13:47] <saste> durandal_1707, it was eons ago [13:47] <saste> and my implementation was not efficient at all [13:48] <saste> now it would be easier to start from scratch (even if i don't like mp=noise syntax very much) [13:49] <saste> i had a file source reading from /dev/urandom at some point [13:49] <saste> in case you need "perfect" noise [13:55] <nevcairiel> hrm, sometimes i find the whole decode API rather limiting ... I'm trying to fix the dvd sub decoder to properly deal with "animated" subtitles (dvd subs allow fade-in/fade-out by changing their alpha), but it requires output of several subtitles from one input packet, but the API doesn't particularly like that... [13:56] <nevcairiel> i suppose i could somehow try to put the fade info into some metadata... [13:57] <durandal_1707> doesnt vf_lut do rgba map thing which is done by spec function? [13:59] <cone-668> ffmpeg.git 03Anton Khirnov 07master:fb0cb11f09c4: cavsdec: export picture type in the output frame [13:59] <cone-668> ffmpeg.git 03Anton Khirnov 07master:5e6ee38bd3ce: FATE: add cavs test [13:59] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:98275283d526: Merge commit '5e6ee38bd3cef0dd05f1dd7977c71f3479eb6d01' [13:59] <saste> wbs: is hls_wrap option compliant with M3U8 spec? [14:00] <saste> wbs: from the spec: The sequence number of a segment is equal to the sequence number of the segment that preceded it plus one. The EXT-X-MEDIA-SEQUENCE tag indicates the sequence number of the first segment that appears in a Playlist file. [14:10] <wbs> it only wraps the sequence number in the filenames, not the EXT-X-MEDIA-SEQUENCE numbers - they never wrap [14:23] <saste> wbs: yes but the second part states "the EXT-X-MEDIA-SEQUENCE tag indicates the sequence number of the first segment that appears in a Playlist file" [14:24] <cone-668> ffmpeg.git 03Diego Biurrun 07master:117f79de2d0c: truemotion2: Sanitize tm2_read_header() [14:24] <cone-668> ffmpeg.git 03Diego Biurrun 07master:bcb8d9eb8ff6: Drop unnecessary 'l' length modifier when printfing double values. [14:24] <cone-668> ffmpeg.git 03Hendrik Leppkes 07master:506409776c49: configure: suppress -fPIC in msvc builds [14:24] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:08196c785a1d: Merge remote-tracking branch 'qatar/master' [14:24] <saste> so if the segment index wraps, the EXT-X-MEDIA-SEQUENCE should wrap as well... or not? [14:24] <wbs> saste: for the hls spec part, the sequence number is only what is in EXT-X-MEDIA-SEQUENCE [14:24] <wbs> the hls spec doesn't know that the file names by conincidence happen to have a number in them [14:24] <wbs> they could just as well be named a.ts, b.ts, etc [14:25] <saste> wbs: ahhh... ok [14:25] <saste> thank you [14:47] <durandal_1707> michaelni: rgba<->abgr is not lossless here. is unscaled variant disabled recently? [14:48] <michaelni> i dont remember anything having been disabled [14:48] <durandal_1707> playing with format filter changes md5 [14:55] <durandal_1707> hmm it appears to work with images and ffmpeg [14:55] <durandal_1707> but with histogram filter i get different results while in fact i should not [15:02] <durandal_1707> nvm, guess this is caused by using frong format in filter [15:17] <durandal_1707> wtf, i should have black frame both times, but i get black frame with 2 different hashes [15:22] <durandal_1707> perhaps i left something not initialized [19:09] <cone-668> ffmpeg.git 03Nicolas George 07master:5f12d5ad78f7: lswr: improve rematrixing error message. [19:09] <cone-668> ffmpeg.git 03Nicolas George 07master:11ab2c25b6b2: lavfi/avfiltergraph: print query_formats errors. [19:22] <cone-668> ffmpeg.git 03Nicolas George 07master:9e28e0035a15: lavc/flacenc: improve error return. [21:01] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:343900a14777: mpeg12: print error for invalid skip cases [21:01] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:49ec0d8ed246: mpeg12: print invalid cbp value [21:01] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:6e8df250fd98: mpeg12: fix frame_pred_frame_dct handling in field pictures [00:00] --- Tue Jan 1 2013

1 0

ffmpeg.log.20121231
by burek 01 Jan '13

01 Jan '13

[00:00] <mickkie> Can someone please explain how this works? [00:13] <saste> mickkie, you have an input with geometry (0, 134) -> (720, 710) and you're trying to put it in a padded area with size (0, 0) -> (720, 576) [00:13] <saste> so it cannot be contained in the padded area [00:13] <saste> we should probably improve the error message, or support overlapping areas [00:28] <mickkie> thank u @saste, I think I managed to make it work - I used "-vf scale=0:306,pad=720:576:0:135:black" and it seems to have worked ... although when I play it with vlc or mplayer and maximise the window on a 1920x1080 monitor, I get padding on the sides too. Hmmm ... [00:29] <mickkie> TBH I find it quite confusing - I do not really understand where it reads that my input has a geometry of (0, 134) [00:30] <mickkie> :-/ [00:32] <mickkie> Thank you again for your help, got to run ... [00:47] <creep> for those who are bored :) "2x2 and 3x3 matrix operations in C with example" http://pastebin.com/Zq93TvNA [00:52] <taylanub> gagan662: Did you try -idx or -forceidx ? [00:52] <taylanub> (MPlayer CLI flags.) [01:21] <luc4> Hi! Can someone suggest a way to extract the audio stream from a mov container? I extracted the h264 stream, but how do I extract the AAC stream? [01:23] <klaxa> ffmpeg -i somefile.mov -vn -c:a copy somefile.aac [01:26] <luc4> klaxa: -c:a? Is 'a' the codec? [01:26] <klaxa> a is the stream identifier [01:26] <klaxa> a means audio [01:30] <luc4> klaxa: I'll study why it is working... but seems to be working :-D thanks [02:55] <level15> hi all. i am trying to transcode an avi (DivX) to avi (XviD). I am using "-vcodec libxvid -acodec copy output.avi" but it fails with error "[libxvid @ 0x151bc60] Invalid pixel aspect ratio 0/1 [02:55] <level15> Video encoding failed". Any hints? [03:11] <iive> level15: you know that xvid and divx implement same codec standard, don't you? [03:11] <level15> iive: yes, but somehow my blue ray player plays xvid but no divx [03:12] <iive> you can try to change the fourcc only. [03:12] <iive> the above is still bug, maybe it should be reported on the bug tracker. [03:12] <level15> iive: that's what sacarasc suggested in the afternoon... the result is that the file plays but i only get audio, no video [03:12] <iive> if you do use recent version of ffmpeg. [03:13] <iive> well, divx had a program to certify what capabilities are supported by hardware players... [03:14] <iive> let me take a look of the source, maybe there is aspect option. [03:14] <iive> level15: what ffmpeg version are you using btw? [03:15] <level15> iive: 0.8.4-4:0.8.4-0ubuntu0.12.04.1 [03:16] <iive> are you sure it is not libav the ffmpeg fork? it is packaged in ubuntu. [03:16] <level15> iive: dunno... how can i check that? [03:16] <level15> i think it is... it says Copyright (c) 2000-2012 the Libav developers [03:16] <level15> hm [03:22] <iive> i should upgrade ffmpeg, but v1.0 just worked with the above options. [03:22] <iive> i assume the input is .divx/.avi file. [03:22] <level15> yes, it is avi with divx codec [03:23] <level15> right now i am looking for a way to replace that fake ffmpeg then [03:23] <level15> there seems to be no package... so i guess i'll have to compile it [03:23] <iive> check the ffmpeg.org website. [03:23] <iive> I have to leave, sorry. [03:24] <level15> iive: that's ok, thanks for your help [03:24] <iive> if it still have same problem, be sure to report the bug, to both projects :) [09:48] <mpfundstein_home> hello, when i use yadif with libx264, i get a green line on the bottom of the video. it is not visible on tv but when the video is played in VLC or a web browser it is. -deinterlace doesnt generate this green line. anyone know what this causes? [10:52] <elkng> http://sprunge.us/KJKb thats a video file I trying to convert and there is a ffmpeg command and the output is: "Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height", but if I use [10:53] <elkng> ffplay it plays file normally [10:55] <elkng> thats the full output of the command http://sprunge.us/YHLS [11:00] <elkng> same problem with mplayer [11:00] <saste> elkng, timebase 1/90000 not supported by MPEG 4 standard, the maximum admitted value for the timebase denominator is 65535 [11:03] <elkng> how to fix it ? [11:04] <elkng> it plays with "ffplay version 0.10.2" and "mplayer-export-2012-04-22" but doesnt converts with ffmpeg or mencoder [11:05] <elkng> why ? [11:05] <saste> elkng, try to specify the output rate [11:06] <saste> also you didn't show the command [11:06] <saste> and 0.10 is oold (-> not very supported here) [11:07] <elkng> the comman was in the link, this "ffmpeg -i input.mp4 -vcodec mpeg4 -vb 1500k -s 650x361 -ab 128k -ar 44100 -ac 2 -acodec libmp3lame -y -f avi output.avi" [11:08] <elkng> saste: "output rate", frame rate ? [11:08] <saste> yes [11:09] <saste> then i don't remember how the timebase is computed from the frame_rate [11:09] <saste> i suspect there is an option to set the timebase directly [11:52] <brontosaurusrex> hello, what would be a bestest upscaling -sws-flag? [11:53] <elkng> why is this rate value 29.970 so specific ? why not to set it to 29 or 30 ? is there something special about it being so precise ? [11:57] <brontosaurusrex> i have mostly aliasing problems [11:57] <saste> elkng, yes the MPEG standards [11:57] <saste> but i'm not so sure about that without looking at the code [12:00] <brontosaurusrex> and another thing: yadif is almost certainly broken [12:00] <JEEB> elkng, it's actually 30000/1001 [12:00] <saste> brontosaurusrex, ->bug report [12:00] <JEEB> it comes from the old TV stuff [12:01] <JEEB> which was 60000/1.001 or so fields per second [12:01] <JEEB> (two fields merged give you a "frame" [not so with truly interlaced content tho, because it's two separate pictures]) [12:24] <Nolaan> hey guys! [12:25] <brontosaurusrex> testing spline, experimental and bicubic, certain parts of image are aliased, how to solve that? [12:25] <brontosaurusrex> I'am upsizing 720x576 to 1920x1080 [12:25] <Nolaan> when i try to compile the latest version with the option "--enable-thumb" somewhat i get the error cannot find libmp3lame.... [12:25] <brontosaurusrex> and i can't belive that ffmpeg uses bilinear by default .... [12:25] <Nolaan> *-lmp3lame [12:41] <Olive6767> Hi There, I have upgraded my conversion app to the latest version of ffmpeg, the problem is that the latest ffmpeg doesn't seem to accept anymore the following parameters: -cmp +chroma -directpred 1 -flags2 +bpyramid+mixed_refs+fastpskip , I'm wondering what are the equivalents? [12:41] <mpfundstein_home> hello, when i use yadif with libx264, i get a green line on the bottom of the video. it is not visible on tv but when the video is played in VLC or a web browser it is. -deinterlace doesnt generate this green line. anyone know what this causes? [12:43] <mpfundstein_home> Olive6767: i think -directpred must be -direct-pred [12:43] <brontosaurusrex> -sws_flags experimental -vf hqdn3d,scale=1920:1080 < seems to win so far [12:43] <mpfundstein_home> Olive6767: or -direct_pred , i had some issues with that as well [12:43] <Olive6767> trying now [12:46] <Olive6767> mpfundstein_home: -direct-pred works indeed :-) ... 2 more 2 go: -cmp +chroma -flags2 +bpyramid+mixed_refs+fastpskip [12:46] <mpfundstein_home> Olive6767: I just checked my working presets. i cmp=+chroma works, flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip works as well [12:47] <mpfundstein_home> Olive6767: are you using a preset or are you typing it directly into the cmd line ? [12:47] <Olive6767> directly into cml [12:47] <mpfundstein_home> Olive6767: mh, than i dont know. you could try writing all your stuff into a preset file [12:48] <Olive6767> about flags2, i have found this:http://forums.creativecow.net/thread/291/900#900 , however no exemple is given for the flags2 param I use, would you know what to do? [12:49] <mpfundstein_home> link doesnt work [12:49] <Olive6767> http://forums.creativecow.net/thread/291/900#900 [12:50] <mpfundstein_home> have you tried: -x264opts flags2=mixed_refs [12:50] <mpfundstein_home> ? [12:51] <mpfundstein_home> or flags2=+mixed_refs [12:51] <Olive6767> trying now ;-) [12:51] <mpfundstein_home> otherwise run (like said at your forum post) x264 --fullhelp [12:51] <mpfundstein_home> than you can see how it is called [12:51] <mpfundstein_home> :-) [12:53] <Olive6767> -x264opts flags2=+bpyramid+mixed_refs+fastpskip , doesn't work, the prob is that I use a static ffmpeg I don't have any x264 exe on hand [12:53] <brontosaurusrex> mpfundstein_home: mediainfo yourfgenerated.mp4 and pastebin and post url [12:53] <Olive6767> but will get one and see [12:53] <brontosaurusrex> -f [12:54] <mpfundstein_home> brontosaurusrex: regarding my yadif question? [12:54] <brontosaurusrex> yes [12:54] <mpfundstein_home> brontosaurusrex: thanks, i will get a file [12:54] <Olive6767> as for your other suggestion, do you have a good doc explaind how to convert cml params to a preset file ? [12:55] <brontosaurusrex> wow, dnhq3d looks like magic [12:55] <brontosaurusrex> thats pretty amazing [12:55] <brontosaurusrex> or maybe my sample just fits ...... [12:55] <mpfundstein_home> Olive6767: check /usr/local/share/ffmpeg for examples [12:55] <Olive6767> thx :) [12:56] <mpfundstein_home> brontosaururex: i dont have a file here at the moment. will take a bit [12:56] Action: mpfundstein_home is starting virtual box [13:01] <mpfundstein_home> one question in between: shall i place -deinterlace before -i or after? i am confused because the engineer before me put it before. my instinct says it should be after -i [13:09] <saste> mpfundstein_home, -deinterlace is deprecated, use yadif instead [13:10] <mpfundstein_home> @saste: yes, but yadif gives me a green line on the bottom [13:10] <mpfundstein_home> i will [13:10] <saste> mpfundstein_home, that's a bug and should be reported in case [13:10] <saste> also what's your ffmpeg version? [13:11] <mpfundstein_home> @saste: pastebin http://pastebin.com/uXvCv8Rn [13:11] <brontosaurusrex> mpfundstein_home: from my tests, yadif is broken [13:12] <saste> mpfundstein_home, SVN-r26402 [13:12] <brontosaurusrex> i mean yadif in ffmpeg [13:12] <saste> that's two years old [13:12] <brontosaurusrex> and i have git compile thats only few days old [13:13] <saste> brontosaurusrex, that's likely a regression, or we should have noticed [13:13] <mpfundstein_home> @saste: this is how the output looks like: http://postimage.org/image/pbnva4vdd/ [13:13] <brontosaurusrex> saste: all i can write at the moment is: yadif is borken and it brings bunch of aliasing in ...., so ..... [13:14] <mpfundstein_home> brontosaurusrex: shall i still run mediainfo on the file? [13:14] <brontosaurusrex> saste: its not, its broken all the time [13:14] <brontosaurusrex> mpfundstein_home: yes, becouse green line is not something i noticed [13:14] <saste> brontosaurusrex, depends, what's your definition of "broken"? [13:15] <saste> also yadif perform a lossy transform, it's not like magic, it works fine for me but then i don't know what you mean by "aliasing" in this context [13:15] <brontosaurusrex> saste: aliasing, aliasing on frames that are strictly progressive (those should be skipped), maybe best bet is to compare it to avisynth version [13:15] <mpfundstein_home> brontosaurusrex: mediainfo: http://pastebin.com/jYxZL6Hs [13:15] <saste> mpfundstein_home, understand that we can't support a two old version of ffmpeg, you should try with latest git [13:15] <saste> *two years old [13:16] <mpfundstein_home> @saste: yes i understand this :-) dont worry. i will upgrade (actually i thought i have a new one. i compiled from source) [13:16] <saste> brontosaurusrex, you have progressive and interlaced frames mixed up in the same source? [13:16] <brontosaurusrex> saste: sure [13:17] <saste> brontosaurusrex, that's sick [13:17] <brontosaurusrex> its rare, but it happens [13:17] <brontosaurusrex> i get material from all possible and impossible sources [13:17] <saste> how is the deinterlacer supposed to understand while a frame is interlaced or progressive? [13:17] <saste> the only reliable way is an human operator to tell, after watching the content [13:18] <saste> we have some filters for that, but not as reliable [13:18] <saste> if you mix progressive and interlaced, you're in trouble [13:18] <brontosaurusrex> saste: yadif , avisynth filter is capable of that [13:19] <brontosaurusrex> also, afaik yadif should skip progressive stuff (or skip in 87% of the cases) [13:19] <brontosaurusrex> in any case, added blinking and aliasing is not what is supposed to happen [13:19] <brontosaurusrex> perhaps its os dependant, iam on osx [13:19] <saste> brontosaurusrex, no [13:20] <brontosaurusrex> saste: ok, the yadif is great .... and it works, lol [13:20] <brontosaurusrex> then* [13:22] <durandal_1707> arent progressive frames markder as such? [13:22] <brontosaurusrex> no, but smart deinterlacers are supposed to detect them [13:23] <saste> brontosaurusrex, read what i already wrote, you need a human brain to perform interlacing detection [13:23] <mpfundstein_home> how shall one handle batch encodes when having multiple sources, some interlaced some progressive ? [13:24] <durandal_1707> saste: really is that so hard? [13:24] <saste> mpfundstein_home, if the frames are properly marked as I/P, then you can script, and auto-insert yadif when required [13:25] <saste> i don't know if yadif will act in no-op when frames are properly marked as progressive, that could be implemented [13:25] <Shu> Probably unrelated here, but I'm trying to port an SWF/flash video to an actual.. well, video. Any tips on the best way to do this? Other than using screen recording software to attempt to rip a copy of the stream [13:25] <mpfundstein_home> @saste: ok, so it depeneds on the source and if they decoded it right :-) [13:25] <Shu> Something to actually play the file back and get a raw copy of the stream somehow, that I can encode however I want. [13:25] <saste> brontosaurusrex, also we have a filter which does interlacing detection, idet iirc [13:25] <JEEB> Shu, if it contains actual frames within itself, you'll have to hack around various Flash tools to see if that is the case and possibly extract the audio / frames [13:26] <saste> but i can't say how reliable it is [13:26] <JEEB> if it's something like what flash should properly be used for [13:26] <JEEB> (vector stuff) [13:26] <Shu> Yeah. I'm pretty sure it is. [13:26] <JEEB> then you have really no alternative but to capture it :P [13:26] <Shu> Bleh >_< I figured. Just trying to figure out the best way TO capture it [13:27] <Shu> Was thinking there might be some method or piece of software out there that would let you capture a stream directly. [13:27] <JEEB> there was some aviutl thingy that did it somewhat well methinks [13:27] <brontosaurusrex> saste: in worst case when deinterlacer deinterlaces progressive frame is that becomes slightly softer, do we agree on that? [13:27] <JEEB> I haven't used it and the documentation is in Japanese so it'll probably be pretty much poking dance for you in that case tho :P [13:27] <saste> brontosaurusrex, let me check [13:29] <JEEB> the last time I did flash vector stuff -> video, I just used a random windows screen capping system (amareco[co] in my case as it's free and didn't contain too much crap) [13:31] <saste> brontosaurusrex, it's possibly "softening" the input, but i can't notice a big difference with the original source [13:31] <brontosaurusrex> exactly, did you use yadif? [13:32] <Shu> I can, at least, use ffmpeg to rip the audio from the files. So I guess I might just end up doing that, and can easily re-attach the auido without losing quality [13:32] <saste> brontosaurusrex, exactly, where are you trying to bring me? ;-) [13:33] <saste> brontosaurusrex, anyway feel free to report a bug if you think that yadif is borken or missing features [13:33] <brontosaurusrex> saste: actually i will, but i need to some tests firsts, prepare test footage, ect [13:33] <brontosaurusrex> it will be some work [13:34] <saste> brontosaurusrex, what i can't tell is that with my limited testing i can't see anything obviously broken [13:34] <brontosaurusrex> saste: what os? [13:34] <saste> also keep in mind that yadif is widely used, so obvious bug should have been already noticed [13:34] <brontosaurusrex> yeah, thats the funny part [13:34] <saste> brontosaurusrex, oh... [13:35] <saste> brontosaurusrex, try to --disable-optmizations and check if it does any difference [13:35] <saste> although from what I see yadif has optimizations only for X86 [13:36] <brontosaurusrex> thats a compile-time switch? [13:36] <saste> brontosaurusrex, no configure [13:36] <brontosaurusrex> ok [13:40] <brontosaurusrex> is thera an addgrain filter in ffmpeg? [13:41] <saste> brontosaurusrex, frei0r? [13:41] <brontosaurusrex> id have to compile it with that? [13:42] <saste> brontosaurusrex, make sure it has something like that [13:42] <saste> or you could try to use mp=noise to add random noise [13:44] <brontosaurusrex> something to counterbalance dnhq3d [13:45] <brontosaurusrex> so that i get the path: denoise > upscale > add-some-salt-or-grain [13:50] <brontosaurusrex> (whats freaking annoying is that i shot this project in hd, but now only sd master is left ..., damn ..) [14:04] <gagan662> taylanub: < taylanub> | gagan662: Did you try -idx or -forceidx ? [14:04] <gagan662> taylanub: no [14:06] <Nano_> hi everyone [14:07] <Nano_> how can i pass a byte array instead of a filename to avformat_open_input(...) ? [15:14] <elkng> http://www.ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 is that version stable one ? [15:14] <espr3ss0> hi ? [15:14] <elkng> espr3ss0: are you in 2012 yet arnt you ? [15:15] <espr3ss0> no [15:22] <durandal_1707> stable in what sense? [15:24] <espr3ss0> is this showing ? o_O (newbie to Limechat ;-p) [15:24] <elkng> durandal_1707: its latest version, latest arnt stable, is there stable version to download or that is stable ? [15:25] <espr3ss0> can i ask some easy enough questions for FFMPEG ? :-) [15:25] <durandal_1707> elkng: i cant find that link, from where you got it? [15:26] <elkng> durandal_1707: guess [15:26] <elkng> http://www.ffmpeg.org/download.html [15:27] <durandal_1707> than you mistyped it [15:27] <elkng> isnt it obvious captain ? [15:27] <elkng> where ? [15:27] <humbolt> can x264 be adapted to work on a gpu computing cluster? [15:27] <durandal_1707> that is link to git snapshot and that is latest to some sense [15:27] <durandal_1707> elkng: you are probably more interested in releases [15:27] <elkng> what is wrong with link, it doesnt downloaded ? [15:28] <elkng> I need working stable version [15:28] <elkng> I have "ffmpeg version 0.10.2" [15:28] <durandal_1707> that is old [15:28] <durandal_1707> try this: http://ffmpeg.org/releases/ffmpeg-1.0.1.tar.bz2 [15:28] <elkng> someone suggested to update it, those the link I found [15:29] <elkng> 1.0.1 is it pun ? [15:29] <elkng> neo lived in a room 101 [15:29] <elkng> when it becomes 1.33.7 ? [15:30] <elkng> I cant find book specific to ffmpeg, found that one: http://www.amazon.com/Real-World-Video-Compression-Beach/dp/0321514696/ref=… is it worth to read ? [15:30] <durandal_1707> nope [15:31] <espr3ss0> durandal_1707: do u know how old "FFmpeg version SVN-r26402" ? or is that not the version :-/ [15:31] <durandal_1707> that is dead, please leave it to rip [15:32] <iive> elkng: i think the original reference is from the book "1984" [15:32] <durandal_1707> elkng: it is just coincidence [15:33] <elkng> "You're a detective now, son. You're not allowed to believe in coincidence anymore." -- Commissioner Gordon [15:44] <espr3ss0> hi ? [15:45] <elkng> espr3ss0: try speaking, and someone will notice you [15:46] <espr3ss0> just did ;-p [15:46] <elkng> just noticed [15:46] <espr3ss0> need some help installing ffmpeg [15:46] <elkng> espr3ss0: compiling it from source ? [15:47] <espr3ss0> no idea :-) , i guess [15:47] <espr3ss0> using ./conf... make etc. ? [15:47] <elkng> espr3ss0: try this guide: http://slackbuilds.org/slackbuilds/14.0/multimedia/ffmpeg/ffmpeg.SlackBuild [15:47] <durandal_1707> there is guide on ffmpeg wiki [15:47] <espr3ss0> i've tried several sites but need to ask some Q's [15:48] <espr3ss0> might have done it backward [15:48] <durandal_1707> espr3ss0: so would you finally ask? [15:48] <espr3ss0> lame was first & then ffmpeg, but faac & other stuff after making it [15:49] <espr3ss0> Q: ... [15:49] <espr3ss0> do I need to re-configure it ? [15:49] <espr3ss0> last trying to get libx264 / x264 from VLC (i think), not working [15:50] <espr3ss0> config shows libx264 as disabled still [15:50] <elkng> espr3ss0: are you using ubuntu ? [15:50] <espr3ss0> no no, Mac :-) [15:50] <cbsrobot> espr3ss0: http://ffmpeg.org/trac/ffmpeg/wiki/MacOSXCompilationGuide [15:51] <espr3ss0> cbsrobot: tried that one ;-p [15:51] <cbsrobot> the homebrew way ? [15:52] <espr3ss0> maybe not, missed a bit ... [15:52] <espr3ss0> should I uninstall stuff, then reinstall ? [15:52] <cbsrobot> espr3ss0: it's a simple copy paste [15:52] <cbsrobot> espr3ss0: yes you could try that [15:52] <espr3ss0> ok [15:53] <cbsrobot> espr3ss0: and plese let me know if it worked [15:53] <espr3ss0> just tried "make un-install" for ffmpeg [15:53] <cbsrobot> "make uninstall" [15:53] <espr3ss0> using ./ffmpeg now does nothing [15:53] <cbsrobot> and the dependencies ? [15:54] <espr3ss0> how can i check that ? [15:54] <cbsrobot> did you install x264, yasm, openjpeg etc manualle [15:54] <cbsrobot> s/manualle/manually/ [15:55] <espr3ss0> did x264 yasm lame faac faad, but lame was the only one first [15:56] <espr3ss0> funny thing is ... it says use gpl ? instead of libx264. complains if not specified in the right way [15:56] <espr3ss0> i checked the config and still showed libx264 as disabled [15:57] <espr3ss0> on "Snow Leopard" btw [15:57] <elkng> is it correct: someone using Mac and trying to build software from source at thesame time ? arnt Mac people choose Mac because they dont want "build from source" stuff ? [15:57] <mpfundstein_home> espr3ss0: did you configured x264 with --enable-static --enable-shared ? [15:57] <espr3ss0> no ? [15:58] <mpfundstein_home> elkng: bullshit. i am on mac osx also at work and i build a lot stuff from source. [15:58] <espr3ss0> mpfundstein_home: does it not do 1 @ a time ? [15:58] <mpfundstein_home> espr3ss0: maybe you should try it [15:58] <elkng> so that was a prejudice [15:58] <espr3ss0> will do ! mpfundstein_home [15:58] <mpfundstein_home> elkng: pretty much yes. mac osx is unix [15:59] <mpfundstein_home> elkng: more or less [15:59] <mpfundstein_home> elkng: ;-) [15:59] <elkng> less [15:59] <mpfundstein_home> elkng: well. the core is unix. and you can feel it while developing. thats why i love it. [15:59] <espr3ss0> gone afk (cigarette ! ;-p) [16:00] <cbsrobot> lol [16:01] <elkng> "gone afk", because of "cigarette" some day it could be just "gone" [16:05] <mpfundstein_home> elkng: thats will happen to everyone :-) [16:15] <mpfundstein_home> i transcode a ntsc video to pal and getting really good looking results when combining avisynth and ffmpeg, playing the output with VLC. if i play the output on my playout system however, the horizontal movement stutters like crazy. does anyone recognize this problem. this are the scripts + the ffmpeg output: http://pastebin.com/KTjTjbpE [16:16] <mpfundstein_home> the playout system is JustOut from ToolsOnAir + an aja.io xt [16:22] <espr3ss0> elkng: :-P [16:23] <espr3ss0> concern, does this mean anything ... x264, "ld: warning: -read_only_relocs cannot be used with x86_64" [16:25] <espr3ss0> also won't do "make install", permission denied [16:29] <espr3ss0> use sudo ? [16:30] <elkng> espr3ss0: try "make install DESTDIR=/tmp" [16:34] <espr3ss0> sudo worked fine [16:37] <espr3ss0> do i need nasm for lame ? [16:41] <espr3ss0> hmm, lame "make" showing "nothing to be done for 'all'", for everything :-/ [16:45] <elkng> there should be README or INSTALL [16:47] <espr3ss0> good plan ! [16:47] <elkng> Building the software on *NIX platforms without configure: make -f Makefile.unix [16:55] <espr3ss0> what's the latest libmp4v2 download ? o_O [17:35] <espr3ss0> what's the difference between "libmp4v2-1.5.0.1.tar.bz2" & "mp4v2-2.0.0.tar.bz2" ? [17:37] <elkng> second can be standalont tool and first one is the library to use in another tools ? [17:37] <espr3ss0> thanks elkng [17:38] <espr3ss0> is that the same for libx264 & x264 ? [17:38] <elkng> could be [17:39] <espr3ss0> doh [17:40] <espr3ss0> does that mean it will still register as libx264 ? [17:40] <espr3ss0> o_O [18:03] <espr3ss0> is this a bad sign ? keeps coming out a lot "warning: deprecated conversion from string constant to 'char*'" [18:04] <espr3ss0> or can i ignore it ? [19:07] <taylanub> gagan662: So, try them. :P [19:38] <espr3ss0> i must ask ... when installing libs n such for ffmpeg to use, if my main aim is for no transcoding from .avi to .mp4 (of course it's necessary contents), what will I need ... "libmp4v2" / "lame" / "x264" and of course "yasm" ? [19:56] <nelson8874> I'm really new to ffmpeg. Want to know if it is possible to take the 5.1 ac3 audio from a vob and rip it into 6 separate mono wave files. [20:00] <espr3ss0> I'm new at this too nelson8874, all I can say is, it is possible ! :-) [20:01] <espr3ss0> also to my last Q if anyone can answer my needs, is there a specific "./configure ..." for the setup i'm trying to achieve ? [20:01] <nelson8874> espr3ss0, i new to linux to. so i don't know thing on how to use ffmpeg.... can you recomend a guide? [20:15] <espr3ss0> nelson8874: no, soz :-( [20:15] <espr3ss0> i'm using a Mac, so it's essentially the same, but still different :-/ [20:16] <espr3ss0> this could be a start, really not sure ... http://jungels.net/articles/ffmpeg-howto.html [20:18] <gagan662> taylanub: din't help. [20:19] <Sashmo_> is yadif the only deinterlace filter? [20:19] <JEEB> the only good one [20:19] <JEEB> (and maybe the only one if the crappy -deinterlace thingy was finally removed) [20:19] <Sashmo_> I was reading the -vf yadif=1, are there any other options there? [20:20] <Sashmo_> or is that option depreciated? [20:20] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#yadif [20:20] <Sashmo_> gracias [20:22] <Sashmo_> im trying some pal to ntsc conversions, and for the life of me, I am not able to get rid of this jitter in the video, only really noticeable on the text news scrolls.... any ideas? [21:23] <Sashmo_> im trying some pal to ntsc conversions, and for the life of me, I am not able to get rid of this jitter in the video, only really noticeable on the text news scrolls.... any ideas? http://pastebin.com/XfhMuW41 can anyone point me int he right direction? [00:00] --- Tue Jan 1 2013

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