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May 2013

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ffmpeg-devel.log.20130510
by burek 11 May '13

11 May '13

[00:23] <ubitux> michaelni: http://pastie.org/7824488 [00:30] <michaelni> ubitux, is there a bug or you just want me to silcene the warnings ? [00:31] <ubitux> i don't know if there is a bug, i just like it when compilation is not cluttered with all kind of warnings hiding others [00:31] <ubitux> and those look recent [00:31] <ubitux> maybe they were missed for some reason (already too much warnings?), so i was just pointing them out [01:47] <cone-954> ffmpeg.git 03Michael Niedermayer 07master:8ef453ff830b: sws/output: silence warnings about uninitialized use of variables [01:47] <cone-954> ffmpeg.git 03Michael Niedermayer 07master:500220a8e84a: mimic: Fix race condition [02:43] <cone-954> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.2.1': unknown revision or path not in the working tree. [02:43] <cone-954> Use '--' to separate paths from revisions [02:43] <cone-954> refs/tags/n1.2.1:HEAD: mimic: Fix race condition [03:03] <ubitux> http://pastie.org/7824987 am i doing something stupid, or is it the compiler? [03:24] <Skyler_> ubitux: does it get better if you use -ffast-math? [03:25] <ubitux> mmh that's indeed better [03:25] <ubitux> well, only with gcc though :) [03:27] <ubitux> Skyler_: any idea why this optim is not done by default? [03:27] <ubitux> maybe some precision issue or something? [04:00] <Skyler_> because it's not a valid optimization [04:00] <Skyler_> (x*x)*(x*x) != ((x*x)*x)*x [04:00] <Skyler_> float arithmetic is not associative. [04:00] <Skyler_> if I remember right, at least [04:12] <iive> ubitux: what is the problem exactly? [04:23] <iive> well, later... [09:45] <cone-267> ffmpeg.git 03Luca Barbato 07master:ee9a4dff4ad7: mmf: K&R formatting cosmetics [09:45] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:c475a58dd60c: Merge remote-tracking branch 'qatar/master' [10:40] <cone-267> ffmpeg.git 03Stefano Sabatini 07master:eeb9a4928dcc: lavu/mem.h: fix typo [10:40] <cone-267> ffmpeg.git 03Stefano Sabatini 07master:e1332ff324e9: lavfi/avfilter.h: fix typo [10:40] <cone-267> ffmpeg.git 03Stefano Sabatini 07master:5c4c929878f4: lavfi/avfiltergraph: fix style [10:41] <cone-267> ffmpeg.git 03Carl Eugen Hoyos 07master:0f05b72f19c0: Fix configure --help output for some autodetected features. [10:53] <cone-267> ffmpeg.git 03Carl Eugen Hoyos 07release/1.2:33d699a4e73d: Fix configure --help output for some autodetected features. (cherry picked from commit 0f05b72f19c0a07cf48c912d2a71c3fb5f7bc325) [10:55] <durandal_1707> saste: select filter have changed PICT_TYPE_I to just I at some point in time, is this expected? [10:56] <saste> durandal_1707, no [10:59] <durandal_1707> so should i add PICT_TYPE_I/.. back for backward compatibility? [10:59] <saste> durandal_1707, if you want, yes seems a good idea [11:02] <nevcairiel> whats with the lazy fate boxes [11:04] <durandal_1707> perhaps some matchines have been disconnected/disabled [11:15] <durandal_1707> i get this when seeking with ffplay with keys: [matroska,webm @ 0x29e37100] File is broken, keyframes not correctly marked! [11:16] <durandal_1707> and same if i seek with mpv before the start ... [11:18] <durandal_1707> saste: any comments on avectorscope?, i gonna apply it soon [12:13] <ubitux> saste: s#"tcp://localhost:5555"#"tcp://localhost:5555" if len(sys.argv) < 2 else sys.argv[1]# [12:14] <ubitux> well i could improve the script :p [12:15] <saste> ubitux, any way to generate auto docs, like as in perl scripts [12:15] <ubitux> yes [12:15] <saste> (check tools/plotframes)? [12:16] <ubitux> basically you put some kind of doxy into scopes with ''' blabla ''' [12:16] <ubitux> then you can pydoc the thing [12:29] <durandal_1707> ^ looks like parser issue to me (that code is nightmare) [12:36] <durandal_1707> dint filter is useful? [13:02] <cone-267> ffmpeg.git 03highgod0401 07master:63e8fc4d8f08: lavfi: modify names of deshake_kernel.h and unsharp_kernel.h [13:04] Action: ubitux still thinks opencl prefix would have been better but well.. [13:09] <durandal_1707> move to separate directory so it can be removed with simple rm -rf opencl/ [13:17] <durandal_1707> there is no example how to use ffmeta files to add metadata when muxing? [13:51] <ubitux> ok, got a better vignette filter. [13:52] Action: durandal_1707 wants 3dlut [13:53] <ubitux> durandal_1707: it's also done [13:53] <ubitux> i'll submit soon, i just need to do something first [15:17] <ubitux> any idea what could have cause such broken blocks? http://ubitux.fr/pub/pics/_broken-blocks.png (8x8, jpeg blocks) [15:18] <ubitux> and if there is a way to fix them without denoising [15:31] <saste> ubitux: do you have a command to show the chroma subsampling artifacts fixed by your patch? [15:32] <ubitux> saste: it's mostly untested, so no :) [15:32] <ubitux> ask durandal_1707, he did a few tests [15:33] <saste> durandal_1707, ^^ [15:37] <ubitux> saste: we need more creepy tests like this btw [15:40] <saste> ubitux, testsrc=s=223x103,format=yuv410p,hue=s=0,format=yuv410p [15:40] <saste> but in general the effect is not very visible, and thus pass unnoticed [15:42] <ubitux> heh. [15:43] <ubitux> s/testsrc/mandelbrot/ [15:49] <ubitux> i'll apply the timeline patchset slightly differently soon [16:15] <ubitux> http://lucy.pkh.me/old-scrat.webm [16:15] <ubitux> ./ffmpeg -ss 10 -i ~/samples/GoneNutty.avi -vf "curves=vintage,vignette=PI/4+random(1)*PI/50:eval=frame,format=yuv420p,noise=c0s=5:c0f=t" -b:v 5000000 -t 20 -y old-scrat.webm [16:15] <durandal_1707> saste: you sure that is testsrc bug? tried generating pngs and playing them with non-ffplay app? [16:15] <ubitux> noise is not really doing what i'm expecting, but good enough for npw [16:15] <ubitux> now* [16:16] <durandal_1707> what kind of noise you want? [16:16] <ubitux> i'm trying to make an "old" effect [16:17] <durandal_1707> ubitux: stop using vp8 [16:17] <ubitux> no idea what kind of noise i should be trying [16:17] <saste> durandal_1707, #2555 is a testsrc bug [16:17] <ubitux> durandal_1707: playable in browsers, so handy [16:18] <durandal_1707> saste: yes, you are correct, i thought it may be ffplay issue [16:19] <durandal_1707> ubitux: mkv should be playable too [16:19] <ubitux> durandal_1707: no [16:19] <durandal_1707> and i do not play videos in brauzer [16:20] <ubitux> mpv in a tty on your freebsd? :) [16:20] <durandal_1707> saste: the ubitux's chroma subsampling patch should fix issues when width is not multiple of subsampled chrome values (log_chrome_v/h << 1) [16:21] <durandal_1707> *width/height [16:21] <ubitux> want me to apply it without extensive testing? [16:21] <durandal_1707> ubitux: that noise in sample you uploaded is little repeating too much [16:22] <saste> ubitux: i'm fine assuming that FATE is fine with it [16:22] <ubitux> saste: sure [16:22] <saste> as I wrote it LGTM [16:22] <ubitux> ok, will apply soon then [16:23] <durandal_1707> you could try other flags: c0f=t+a+u:c0s=13 [16:25] <durandal_1707> well that combination needs higher strength [16:25] <ubitux> yes i tried with average [16:26] <ubitux> but it makes a gummy effect [16:30] <ubitux> it would be awesome if testsrc could keep the circle round when playing with its ar [16:31] Action: durandal_1707 wonders from where fish effect is [16:32] <ubitux> you could adapt the vignette filter to make a fish effect [16:32] <ubitux> well i guess i'll have to write the doc and submit it [16:33] <saste> durandal_1707, VHOOK [16:48] <durandal_1707> from where ami get all those game files... [16:51] <kierank> Amiga stuff I guess [17:00] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:a079ed63d3c0: avidec: Enable full parsing for potencoder mpeg1/2 [17:07] <ubitux> saste: my patch seems to fix the right border, but not the botton line [17:07] <saste> ubitux, what's the problem with the bottom line? [17:08] <ubitux> still green [17:08] <durandal_1707> ubitux: what filter? [17:08] <saste> in my test i couldn't notice any artifact in the bottom line... [17:08] <saste> ubitux: may be an SDL issue [17:08] <cone-267> ffmpeg.git 03Paul B Mahol 07master:d751a2526f9b: lavfi/anull: remove redundant lines [17:08] <saste> uhm not so sure about that... [17:10] <ubitux> only reproducible with ffplay though [17:10] <ubitux> http://b.pkh.me/hue-old.png http://b.pkh.me/hue-new.png [17:10] <ubitux> (ffmpeg ^) [17:12] <ubitux> saste: and ffplay: http://b.pkh.me/ffplay-hue-old.png http://b.pkh.me/ffplay-hue-new.png [17:12] <ubitux> but crop/pad might need fixing as well [17:14] <durandal_1707> swscale may have own bugs, so use extractplanes [17:15] <saste> ubitux, which command are you using? [17:15] <ubitux> saste: yours, ffplay -f lavfi testsrc=s=223x103,format=yuv410p,hue=s=0 [17:17] <ubitux> well anyway, that's at least better with the patch [17:17] <ubitux> but i think we'll need to have a deeper look into at least pad, crop, and possibly sws [17:18] <ubitux> (and the other(s?) filter(s?) i mentioned) [17:20] <saste> i cannot observe the issue with ffplay, only if i use ffmpeg [17:20] <durandal_1707> ubitux: with your patch and ffmpeg -f lavfi -i testsrc=s=801x603,format=yuv410p,hue=s=0 /tmp/1.png its looks fine in mpv [17:21] <ubitux> i'm going to push the patch in a moment so we can have a better basis [17:22] <durandal_1707> same with ffplay [17:23] <ubitux> i still have a bottom green line here [17:24] <saste> ubitux, what if you convert to png and display it? [17:25] <ubitux> saste: i can only reproduce with ffplay [17:25] <durandal_1707> sdl bug probably [17:25] <saste> ubitux, yes [17:26] <saste> does SDL supports yuv410p? [17:26] <ubitux> ./ffmpeg -f lavfi -i testsrc=s=801x603,format=yuv410p,hue=s=0,format=yuv420p -f sdl - [17:26] <ubitux> completely broken :)) [17:27] <durandal_1707> oh yes, i got that yesterday (forgot about it completely) [17:27] <cone-267> ffmpeg.git 03Clément BSsch 07master:50e66726a237: lavfi: use ceil right shift for chroma width/height. [17:28] <saste> is that another SDL bug? the same one affecting ffplay? [17:28] <ubitux> no, a different bug :) [17:29] <durandal_1707> or avdevice bug [18:45] <saste> ubitux: what I need is a lightweight dynamic buffer API [18:46] <saste> a single function should be enough, and is more or less equivalent to the alloc_elem() function, but more flexible [18:46] <saste> indeed I'm surprised we don't already have something similar [18:49] <ubitux> we already have avbprint and avio_dyn_buf things [18:49] <ubitux> for this purpose [18:50] <ubitux> the dynamic array is useful for other purposes, and we already have various use cases [18:52] <saste> uhm i'm confused [18:59] <ubitux> ETOOMUCHFILTERS [18:59] <durandal_1707> ? [19:00] <ubitux> durandal_1707: 3 pending filters :) [19:00] <ubitux> dctdnoiz, vignette and lut3 [19:00] <ubitux> lut3d* [19:01] <ubitux> with avectorscope and zmq we're going to have quite a bunch of nice filters [19:01] <ubitux> comments welcome on timeline btw [19:03] <durandal_1707> how is lut3d used? [19:03] <ubitux> lut3d=foobar.3dl [19:04] <durandal_1707> and how i get foobar ? [19:04] <ubitux> it's hard to find samples though [19:04] <durandal_1707> what such file actually contains? [19:04] <ubitux> parse error [19:04] <ubitux> ah [19:04] <ubitux> it contains... r,g,b values [19:04] <ubitux> see the pandora example [19:05] <ubitux> (not yet supported in this filter though) [19:05] <ubitux> we need a slow motion filter :( [19:06] <ubitux> anyway, i have a few hours for me right now [19:06] <ubitux> should i port a mp filter? [19:07] <ubitux> (if so, which one?) [19:07] <durandal_1707> dint? [19:07] <ubitux> i don't like the name [19:07] <ubitux> reminds me bad nightmares [19:08] <durandal_1707> ow [19:09] <ubitux> okay [19:09] <ubitux> durandal_1707: please suggest a name [19:09] <ubitux> :) [19:09] <ubitux> wldnoiz? [19:10] <durandal_1707> ovrcmpltwvltdnisr [19:10] <durandal_1707> ovrcmpltwvltdnsr [19:11] <ubitux> hell yeah [19:11] <durandal_1707> just do eq? [19:12] <ubitux> too late i started ow [19:12] <ubitux> feel free to do eq2 [19:12] <ubitux> so we can drop two filters at the same time [19:13] <ubitux> nice, unused mode in ow [19:13] <ubitux> and unused delta [19:16] <durandal_1707> well i'm more interested in this http://docs.gimp.org/en/gimp-tool-levels.html (something like (auto)levels in avisynth) , deflicker and mergeplanes [19:16] <ubitux> i think you got lost in your quest to kill mp wrapper [19:17] <durandal_1707> i can remove non-functional softpulldown, crashing ilpack, and broken/flawed fil right now [19:18] <ubitux> fine with me [19:18] <ubitux> but i find it more fun when there is a new filter :) [19:20] <ubitux> no one motivated to add gimp presets support to vf curves btw? [19:24] <cone-267> ffmpeg.git 03Paul B Mahol 07master:005ee7a5867e: lavfi: avectorscope filter [19:25] <ubitux> :) [19:41] <cone-267> ffmpeg.git 03Paul B Mahol 07master:4868855b1305: lavfi/select: set forgotten VAR_PICT_TYPE_BI [19:44] <durandal_1707> AV_PICTURE_TYPE_S is not set by any decoder/parser [20:43] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:7ded0179832b: sws/output/yuv2gbrp_full_X_c(): silence warning about uninitialized variable [20:43] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:b2cf655d6e3d: sws: add casts to silence pointer type mismatch warnings [20:46] <ubitux> michaelni: thx ^ [20:48] <durandal_1707> michaelni: you fixing bogus warnings of stupid compilers [20:49] <michaelni> all compilers are stupid [20:49] <durandal_1707> some less [20:52] <ubitux> durandal_1707: such as? [20:52] <durandal_1707> anyone have amarok or banshee installed, or foobar2k? [20:55] <llogan> why those? [20:55] <durandal_1707> for id3v2 chapters [21:07] <BBB> Yuvi: hey, want to write a vp9 decoder? ;) [21:34] <ubitux> yay, ow ported. [21:55] <Daemon404> 'ow' ? [21:55] <ubitux> Overcomplete Wavelet denoiser. [21:55] <ubitux> i'm sure you already love the name [21:56] <Daemon404> it certainly makes me skeptical :) [21:56] <ubitux> i'm not sure from where the "overcomplete" comes from though [21:56] <ubitux> since there is only 3 option [21:57] <durandal_1707> and one not implemented [21:57] <ubitux> i counted only the implemented ones [21:57] <ubitux> the two strength and the depth [21:58] <durandal_1707> if depth is too big it cause crash iirc [21:58] <ubitux> i've fixed that [21:58] <durandal_1707> so is it better than dtcdntz? [21:58] <ubitux> no idea [21:58] <ubitux> (careful, "dtc" means "iya" in french) [21:59] <durandal_1707> dctdnoiz, i'm not one who make up such names [21:59] <ubitux> :) [21:59] <ubitux> dctdnoiz really is slow [22:00] <ubitux> at least if you don't play with overlap [22:10] <ubitux> okay, ow submitted [22:10] <ubitux> still have some time [22:10] <ubitux> durandal_1707: which one now? [22:11] <durandal_1707> qp? [22:11] <ubitux> erh [22:11] <ubitux> no thx [22:11] <durandal_1707> phase filter looks nice (syntax) [22:12] <ubitux> michaelni: you wanted vf spp at some point, no? [22:12] <durandal_1707> i think this it is obsoleted by pp [22:13] <ubitux> nope, it's more effective afaik [22:13] <durandal_1707> beating complete libpostroc? [22:13] <ubitux> it had a pretty decent output iirc [22:14] <durandal_1707> sab looks trivial [22:14] <ubitux> go for it [22:15] <durandal_1707> the only big monster left is mcdeint [22:16] <michaelni> ubitux, yes, i want spp at some point [22:16] <ubitux> michaelni: any other filter in priority? [22:17] <michaelni> all that are popular with the movie/anime/pr0n encoding communities [22:17] <durandal_1707> that is none [22:17] <durandal_1707> except pullup all others should be removed, and better filters written from scratch [22:18] <michaelni> i didnt think of libmpcodecs here ... [22:18] <ubitux> my question was porting to libmpcodecs [22:18] <ubitux> since i want to get done with that stuff [22:18] <ubitux> so we can focus on some more important stuff [22:18] <michaelni> porting TO is a interresting idea [22:18] <ubitux> TO? [22:19] <michaelni> from libavfilter to libmpcodecs ;) [22:19] <ubitux> ah haha sorry [22:19] <ubitux> anyway, porting from mp [22:19] <michaelni> but actually, some libavfilter wraper would make sense for mplayers libmpcodecs [22:20] <ubitux> mpv has one [22:20] <ubitux> mplayer too afaik, but broken iirc [22:20] <durandal_1707> talk about that on #mplayer [22:20] <ubitux> anyway, so next mp to port would be spp? [22:20] <durandal_1707> or pp777 [22:21] <ubitux> no one wants to port some other? :( [22:21] <durandal_1707> if i get really bored may write supereq [22:21] <ubitux> michaelni: so if i port spp, you ok for dropping mp={fspp,pp7,spp,uspp}? [22:22] <michaelni> i think some of these are faster than spp so i dont think they should be lost [22:23] <ubitux> don't want to help porting them then? :( [22:26] Action: michaelni doesnt see a major problem with using them through vf mp [22:27] <michaelni> so i would probably rather work on adding things that can currently not be used at all [22:27] <ubitux> mp is supposed to be a temporary solution :( [22:27] <ubitux> michaelni: can you work on a fast dct 2d then? ;) [22:28] <ubitux> so we can have a faster dctdnoiz [22:29] <michaelni> is there no lgpl compatible fast dct2d code available that we could use ? [22:31] <michaelni> It would be less work to reuse existing code than write it new [22:31] <ubitux> possibly [22:32] <ubitux> well anyway, ETOOMUCHWORK today [22:32] <ubitux> i'll go watch some animes waiting for reviews before i do anything else [22:32] <ubitux> cya [22:32] Action: ubitux & [22:36] <michaelni> the different pp and wavelet filters could also be compared before porting, that is if any is always worse qiality & speed it could be skiped [22:56] <cone-267> ffmpeg.git 03Luca Barbato 07master:72fbc9685c08: avconv: use current syntax for the scale filter [22:56] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:fc94b68eb53f: Merge commit '72fbc9685c08baae7ab9996642707eaab873b6f1' [23:04] <cone-267> ffmpeg.git 03Anton Khirnov 07master:16a4a18db089: af_asyncts: fix offset calculation [23:04] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:2a7d71984805: Merge commit '16a4a18db089af8c432f1cdec62155000585b72c' [23:09] <Daemon404> so uh [23:09] <Daemon404> can someone regenerate the site docs [23:09] <Daemon404> ffmpeg.org/filters.html is 5 months old [23:13] <durandal_1707> can't be, force refresh? [23:13] <ubitux> ffmpeg-filters.html [23:13] <Daemon404> o [23:13] <ubitux> filters.html should not exist anymore [23:13] <Daemon404> well it does [23:13] <ubitux> then it should be removed, or symlinked [23:14] <ubitux> the documentation page should not reference it anymore anyway [23:18] <michaelni> filters.html deleted [23:19] <michaelni> i can setup a redirect if something off site points to it still ? [23:20] <ubitux> did you delete codecs.html etc? [23:20] <ubitux> (if any) [23:21] <cbsrobot> one thing that could be improved is the site title [23:21] <cbsrobot> it's always "FFmpeg documentation : : " [23:21] <cbsrobot> even if it's ffplay ... [23:21] <michaelni> ubitux, didnt delete anything else [23:21] <cbsrobot> but I know I should send a patch - so ... [23:23] <michaelni> ubitux, if you want a directory listing to check the files there, i can provide one [23:23] <ubitux> michaelni: ls *.html welcome [23:23] <durandal_1707> ubitux: ow is not LGPL [23:24] <ubitux> durandal_1707: oh oups [23:24] <ubitux> will fix asap [23:25] <michaelni> ubitux, list sent [23:25] <ubitux> thx [23:26] <cone-267> ffmpeg.git 03Christophe Gisquet 07master:2c299d4165cd: x86: sbrdsp: implement SSE2 qmf_pre_shuffle [23:26] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:5e1278c6401a: Merge commit '2c299d4165cd9653153e12270971c2368551b79e' [23:27] <ubitux> michaelni: 'seems like a nice mess :)) [23:28] <ubitux> michaelni: i think decoders.html bitstream_filters.html eval.html protocols.html muxers.html encoders.html demuxers.html indevs.html syntax.html outdevs.html metadata.html can be dropped [23:29] <ubitux> michaelni: according to the owner/group and date, they are not updated anymore [23:29] <ubitux> and likely replaced by the ffmpeg-*.html [23:32] <cone-267> ffmpeg.git 03Janne Grunau 07master:94235f2ba2eb: indeo3: avoid writes without necessary alignment in copy_cell() [23:32] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:496f2df9084b: Merge remote-tracking branch 'qatar/master' [23:34] <michaelni> ubitux, done [23:34] <ubitux> :) [23:35] <ubitux> michaelni: so what lavc wavelet code in particular were you thinking about? [23:37] <michaelni> *dwt*.{c,h} [23:38] <ubitux> no dwtutils? : [23:38] <ubitux> :( [23:50] <cone-267> ffmpeg.git 03Clément BSsch 07master:a9705e4de9c0: lavfi/curves: avoid ctx pointer indirection. [00:00] --- Sat May 11 2013

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ffmpeg.log.20130510
by burek 11 May '13

11 May '13

[00:27] <Dettorer> hi there, I have a problem with ffmpeg trying to stream to a justintv service (twitch.tv) [00:27] <Dettorer> all frames are droped after a few seconds of streaming [00:38] <Dettorer> strangely, on my laptop (same network, same system, same kernel, just a slightly better hardware), it works fine [00:39] <Dettorer> (oh yes I'm on archlinux) [00:40] <Dettorer> I use these options http://paste.awesom.eu/0qs [01:07] <schtinky> i'm back. using the following to grab still frames from a digital TV tuner device on a raspberry pi: [01:08] <schtinky> "gnutv | ffmpeg -skip_frame nokey -i inputfile.mpg -an -r 1 -vsync vfr outputfiles%9d.tiff" [01:09] <schtinky> ... and that command works except the raspberry pi is a hair too slow [01:09] <schtinky> can do about 27 images in 30 seconds [01:09] <schtinky> generates dvr overflow errors due to being unable to keep up [01:09] <schtinky> I know tiff is faster than jpg, png and pam [01:10] <schtinky> is there anything else I can try, either format-wise or via the ffmpeg parameters to squeeze it out just a little faster? [01:10] <Mavrik> not really [01:11] <schtinky> i've tried overclocking the pi, but it effs the SD card [01:11] <schtinky> looks like I"m going to have to upgrade to some sort of atom-based machine [01:11] <schtinky> yes I said "upgrade to atom" [01:13] <Mavrik> yeah well [01:13] <Mavrik> RPi does have an exceptionally crappy CPu [01:14] <Mavrik> with obsolete architecture [01:14] <Mavrik> no NEON support, nothing [01:14] <Mavrik> I'm not sure who keeps persuading people it's capable of doing anything useful related to video above 320x240 >( [01:14] <Mavrik> :) [01:23] <schtinky> I have to buy potentially hundreds of machines that can grab still frames from OTA TV [01:23] <schtinky> if I can do it with $100 machines instead of $500 machines, that makes a big difference [01:24] <schtinky> oh man, don't look now. Just got 29/30 frames to work [01:24] <schtinky> overclocked the pi, turned off xorg [01:24] <schtinky> SOOoooo close [01:24] <schtinky> still probably very far away from production-worthy stability, though [01:25] <schtinky> one hiccup in the digital tuner would probably wreck the whole thing [01:26] <schtinky> and I still have to run some sort of program to upload the resulting images [01:26] <schtinky> which will take its own chunk of cpu [01:26] <schtinky> I think I'm beaten on this [01:39] <Zarx> if anyone could help me, I have been having a really hard time getting things to work right when I build ffmpeg using patched versions of x264. It doesn't want to accept the new commandline options that are added by the patches [01:54] <llogan> Dettorer: you should show an actual ffmpeg command and the complete console output. [01:56] <llogan> Dettorer: ...minus your stream key of course [02:00] <Zarx> i gotta go pick someone up, ill paste it in just a bit [02:15] <Mavrik> schtinky, how bout getting one of those boards with actual 1GHz + multicore Cortexes A15 with NEON? [02:24] <highgod> Hi, I want to ask a question,we want to use ffmpeg to upload the stream to ffserver, and use a nother ffmpeg to receive the stream from ffserver, add save as a file, can ffmpeg do that, what is the command line? thanks [02:26] <gajbooks> I'm using sort of a yucky frontend for FFMPEG called FFsplit to record my screen to H.264 and I'm noticing that any text that is truly black is weirdly multicolored. Is this fixable by a flag, or is that just the way H.264 works? [02:32] <Mavrik> gajbooks, that would look like a colorspace conversion wart [02:32] <Mavrik> gajbooks, are you using x264 and encoding to yuv420? [02:33] <gajbooks> I don't know what I'm technically encoding from, but I'm encoding to x264 [02:34] <gajbooks> Oh, it is grabbing from yuv420p [02:34] <gajbooks> Excuse me, to. [02:34] <gajbooks> Mavrik: Yes. [02:35] <Mavrik> gajbooks, yeah, going from RGB to YUV420 can cause color artifacts :) [02:35] <gajbooks> Is there any way to fix it? [02:35] <Mavrik> gajbooks, usually setting colormatrix flag on x264 fixes it [02:35] <Mavrik> try either bt709 or bt470bg [02:35] <Mavrik> I always forget which is used by ffmpeg [02:37] <Mavrik> gajbooks, so to recap, use x264 parameters and pass in "colormatrix=bt709" :) [02:38] <gajbooks> It doesn't like that argument. [02:38] <Mavrik> are you passing it to x264 not ffmpeg? [02:39] <gajbooks> How would one do that through a sucky front end? [02:39] <Mavrik> no idea [02:39] <Mavrik> by not using sucky frontend usually ;) [02:39] <Mavrik> or consulting docs :P [02:43] <gajbooks> This sucky front end seems to be one of the things with the right DirectShow filters to record screens for free. [02:49] <llogan> ffmpeg can record your screen too [02:49] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20%28sc… [02:49] <llogan> ugly ass url. why no camelcase? [02:50] <llogan> wiki/Screencast would have been nice [02:51] <gajbooks> Now that I found the right thing, it's giving me an "error splitting the argument" [02:51] <llogan> highgod: did you refer to the tee muxer? http://ffmpeg.org/ffmpeg-formats.html#tee [02:58] <highgod> record screen ffmpeg -f dshow -i video="screen-capture-recorder" -vcodec libx264 -x264opts bframes=0:ref=1 -preset fast out.m2ts [02:58] <highgod> it works, I work on it too [03:03] <ubitux> http://pastie.org/7824987 am i doing something stupid, or is it the compiler? [03:03] <ubitux> oups bad chan [03:58] <Hans_Henrik> if i want to COPY the audio from src to destination, how can i do that? does this seem right? -acodec copy [04:04] <Hans_Henrik> seems so yes [09:47] <ankr> How to convert MP4 => OGV on a mac? [09:47] <ankr> I tried "ffmpeg -i input.mp4 -acodec libvorbis -vcodec libtheora -f ogv output.ogv" but got the error "[NULL @ 0x7fe2a901ee00] Requested output format 'ogv' is not a suitable output format" [09:48] <ankr> I also tried "ffmpeg -i input.mp4 -acodec vorbis -vcodec libtheora -f ogg output.ogv" but got the error "Unknown encoder 'libtheora'" [09:54] <relaxed> ankr: -f ogg [09:54] <relaxed> oh [09:54] <JEEB> then you don't have libtheora built into your ffmpeg build [09:54] <relaxed> ankr: http://goo.gl/DPrRY [09:55] <RSDRSDRSD> I want to encode user submitted videos, I am using the following command: http://pastebin.com/1vYhFL48 but the videos seem rather shocky, what can go wrong? [09:56] <retard> the videos seem shocky? your users might be trolls i guess? [09:56] <RSDRSDRSD> quality of a still image is rather good, but when it is playing it doesn play smooth [09:57] <RSDRSDRSD> sorry for my wrong language [09:57] <RSDRSDRSD> it doesn play smooth ;-) [09:57] <relaxed> are you sure their input is interlaced? [09:58] <relaxed> because using yadif on non-interlaced input would do that [09:59] <RSDRSDRSD> no not sure about that [09:59] <RSDRSDRSD> how can I detect that? [10:01] <RSDRSDRSD> if a video is interlaced, does it give problems when I don;t make it non-interlaced? [10:03] <JEEB> you probably meant something else when writing that [10:03] <JEEB> you will have combing and the clip loses the "I'm interlaced yo" flag if you re-encode interlaced content as-is as progressive [10:03] <ankr> relaxed: can you point me in a direction on how to get the libtheora? [10:04] <relaxed> ankr: install libtheora-dev and recomplile ffmpeg with --enable-libtheora, or use my static binary [10:05] <RSDRSDRSD> hmm, don understand that [10:06] <RSDRSDRSD> I have user submitted video content [10:06] <RSDRSDRSD> I want to encode it to webm/mp4 [10:06] <JEEB> also it all depends on what you mean "doesn't play smooth" [10:07] <RSDRSDRSD> Well on scene changes or fast scenes it just looks like it stands still and goes further [10:09] <relaxed> RSDRSDRSD: Install mediainfo and test with --> mediainfo --Inform='Video;%ScanType%' user_video.mpg [10:09] <relaxed> If it returns "Interlaced", use yadif [10:11] <RSDRSDRSD> ok thanks, will try that [10:11] <relaxed> there's also ffmpeg's idet filter [10:11] <relaxed> I'm not sure which is better. [10:12] <relaxed> Seems like there should be an option for yadif to test for interlaced input and have the option to pass through if it's not. [10:15] <RSDRSDRSD> doesn ffprobe have some info on interlaced content? [10:16] <relaxed> RSDRSDRSD: try yadif=deint=interlaced [10:17] <relaxed> that might be your silver bullet ^^ [10:18] <RSDRSDRSD> shouldn it be tdeint ? [10:19] <relaxed> "interlaced, only deinterlace frames marked as interlaced" [10:22] <RSDRSDRSD> oh it wasn a typo? [10:22] <relaxed> man ffmpeg-filters| less +/^' 'yadif [10:26] <RSDRSDRSD> if I read at http://sonnati.wordpress.com/2012/10/19/ffmpeg-the-swiss-army-knife-of-inte… [10:26] <RSDRSDRSD> point 3 [10:30] <RSDRSDRSD> what if I do yadiff=0:-1:1 sendframe, autodetect,only frames which are interlaced [11:38] <RSDRSDRSD> deint=interlaced is also working good, but I donŽt know what really happens when using that one, canŽt find the command in de the docs [11:50] <saste> RSDRSDRSD: please file a ticket (about the missing docs) [11:55] <relaxed> it's in my man page, I quoted it above. [11:59] <RSDRSDRSD> ah now i see, but shouldnŽt I send the mode and parity also? [11:59] <praveenmarkandu> hi is there a way to generate WEBVTT [11:59] <praveenmarkandu> closed questions [11:59] <praveenmarkandu> *closed captions [12:02] <RSDRSDRSD> ah okey yadif=0:-1:1 is the same as yadif=deint=interlaced, since mode and parity are by default 0 respectively -1 [12:02] <RSDRSDRSD> thanks [12:04] <ubitux> praveenmarkandu: no, only demuxing so far [12:04] <ubitux> patch welcome [12:12] <praveenmarkandu> @ubitux: any way to insert soft subs into a hls stream? [12:18] <ubitux> praveenmarkandu: possibly [12:18] <ubitux> with mov_text maybe [12:30] <durandal_1707> RSDRSDRSD, ubitux: deint have been removed [12:32] <ubitux> durandal_1707: huh?? [12:32] <ubitux> why? [12:32] <ubitux> oh, merge... [12:33] <ubitux> ? [12:34] <durandal_1707> oh. deint is what? [12:34] <ubitux> it looks present [12:34] <ubitux> just undocumented [12:34] <durandal_1707> where it is? [12:35] <ubitux> yadif filter option [12:35] <ubitux> to make it deinterlace only interlaced frames [12:35] <durandal_1707> ah, i though user was talking about filter, so I replaced it with something else .... [12:42] <relaxed> why were the other deinterlacing filters ported? I thought yadif was by far the best. [13:05] <Gabriel_Blake> Could someone help me out ? I'm trying to convert .dv (from minidv camcoder) to .mkv (x264/AAC). After demuxing the original stream, audio and video are out of sync, but in the original file, they're ok. Can ffmpeg correct the demuxing ? [13:17] <luc4> Hi! I have an h264 raw file with variable framerate. Can I wrap this into an mp4 with ffmpeg? [13:18] <klaxa> <relaxed> why were the other deinterlacing filters ported? I thought yadif was by far the best. [13:19] <klaxa> i thought so too until yesterday [13:19] <klaxa> yadif actually introduced some stuttering (frames shown slightly too long) which mplayers pulldown filter did not [13:19] <klaxa> *mplayer's [13:20] <klaxa> *pullup [13:20] <klaxa> god what's wrong with me today [13:23] <durandal_1707> yadif is inverse interlacer and pullup is inverse teleciner [13:23] <klaxa> hmm... right [13:25] <klaxa> nvm me... [14:10] <RSDRSDRSD> what is best to use for deinterlacing now? [14:11] <JEEB> I think current ffmpeg has two good filters for deint [14:11] <JEEB> yadif (for general deinterlacing, and handles true interlacing) [14:11] <JEEB> and vivtc if ubitux got that in [14:12] <JEEB> latter is very specific for a certain use case, so it isn't useful for automation [14:12] <luc4> Hi! I have an h264 raw file with variable framerate. Can I wrap this into an mp4 with ffmpeg? [14:12] <JEEB> how are the timestamps for it stored (if at all)? [14:13] <JEEB> if it's raw H.264 I don't think it has any timing information whatsoever [14:13] <ubitux> JEEB: fieldmatch is in yes [14:13] <luc4> JEEB: oh... I created using Android classes, I given the encoder the timestamp. [14:13] <luc4> JEEB: but it is possible it is ignored... [14:13] <JEEB> luc4, and if it's a raw H.264 stream that doesn't keep the information at all ^^; [14:14] <JEEB> it probably isn't ignored during the encoding [14:14] <ubitux> JEEB: http://ffmpeg.org/ffmpeg-filters.html#fieldmatch [14:14] <JEEB> but the raw Annex B just doesn't contain timestamps [14:14] <JEEB> so they got used during encoding and then thrown out [14:14] <JEEB> ubitux, nice, what about decimation? [14:14] <luc4> oh... so from that file there is no way I can get a "correct" mp4 right? [14:15] <JEEB> yes, unless you bring in the timestamps somehow else [14:15] <ubitux> JEEB: http://ffmpeg.org/ffmpeg-filters.html#decimate [14:15] <JEEB> \o/ [14:16] <JEEB> also as far as I know ffmpeg's annex B frame types etc. -> PTS calculation is still borked, so I recommend using L-SMASH for annex B -> mp4 [14:17] <luc4> JEEB: ok, so I'll have to change this entirely. Thanks! [14:18] <JEEB> I would recommend you mux your encoded stream into some container (mkv, mov/mp4) when capturing if you want to keep the timestamps [14:18] <luc4> JEEB: that seems not to be supported with the hardware encoder unfortunately... [14:19] <luc4> so I thought I could mux after the entire video was stored in raw format. I thought that nal units did carry timestamps... but I guess I was wrong... [14:19] <JEEB> luc4, you take the NAL units and use something to mux it? [14:19] <JEEB> and the timestamps should have to be passed somehow [14:20] <JEEB> also there might be a way of putting timestamps into the NAL units, but I'm pretty sure most parsers will just ignore it, even if it would exist :s [14:21] <luc4> JEEB: I encode raw frames from the camera (vfr) to h264 with the hardware encoder, and then I thought I could use ffmpeg to create an mp4 or a mov to correctly play in the android player. The timestamps are passed to the hardware encoder for each frame. [14:22] <JEEB> you could use libavformat yes [14:22] <luc4> JEEB: but as the hardware encoder API are generic, timestamps could be ignored I suppose. [14:22] <JEEB> does the encoder give you timestamps from the output side? in the API? [14:22] <luc4> JEEB: the problem is that I see the result has an incorrect frame rate after muxed via ffmpeg. So either my h264 does not have the timestamps, or I'm muxing wrong using ffmpeg. [14:23] <luc4> no, I give the timestamps to the encoder, but it returns only a buffer [14:24] <JEEB> so you get no PTS from the encoder in addition to the buffer? [14:24] <JEEB> that sucks :s [14:25] <luc4> JEEB: it seems not... http://developer.android.com/reference/android/media/MediaCodec.html [14:25] <luc4> or maybe yes... [14:26] <luc4> JEEB: dequeueOutputBuffer also filles a bufferinfo, that might be what you are talking about... [14:26] <luc4> in case those were the timestamps, how could I provide to ffmpeg to get a correct mp4? [14:27] <JEEB> yeah, the presentationTimeUs sounds like PTS [14:27] <JEEB> not sure what the offset is :s [14:28] <luc4> but can I provide those to ffmpeg somehow then? [14:28] <JEEB> the library, yes [14:29] <luc4> and from the command line? I could test with that first, might be simpler. [14:30] <JEEB> you could output the timestamps onto a "ogm timecode file" that is a text file [14:30] <JEEB> *v2 timecode file [14:31] <JEEB> and then mux that Annex B file with L-SMASH's muxer first [14:31] <JEEB> and then use the timelineeditor binary in L-SMASH to modify the timestamps [14:31] <JEEB> according to the v2 timecode file [14:33] <luc4> JEEB: ok, I'll try to study this and see if I can follow your instructions :-) thanks! [14:34] <JEEB> example of the v2 timecode format btw https://trac.bunkus.org/browser/examples/example-timecodes-v2.txt [14:34] <JEEB> these do actually ignore b-frames and such so you might be able to just grab the timestamp information from the input side [14:35] <JEEB> (as in, these are the timestamp values for the input/after decoding clip) [14:35] <JEEB> or put more simply, "in presentation order" [14:38] <luc4> But then I should provide that v2 file to ffmpeg together with my h264? [14:39] <JEEB> yes, but I'm not sure if ffmpeg reads that, which is why I mentioned the tools from the L-SMASH toolset [14:39] <JEEB> http://code.google.com/p/l-smash/ [14:41] <luc4> JEEB: ah ok, I see. But in case I wanted to do it in code with libavformat, should this be possible directly with ffmpeg? [14:41] <JEEB> well yes, you would be directly using libavformat then, and giving the encoder's decoding order timestamps to the muxer [14:41] <JEEB> ffmpeg the tool wouldn't be called [14:42] <JEEB> (of course if you end up using mp4 as the output, L-SMASH's muxer is another alternative, but libavformat can generally output to various formats so it can be closer to your needs, possibly) [14:43] <luc4> I'll have to study then :-) thanks! [14:49] <luc4> JEEB: I read this: Note that FFmpeg's mp4 muxer does not support vfr output. Do you know if this is correct? [14:50] <JEEB> no, it is not [14:50] <JEEB> it works on timestamps and thus VFR works just fine [14:50] <JEEB> the only problem I know of the command line application's muxing capabilities is the PTS generation from a frame rate and the frame types [14:51] <luc4> JEEB: I was thinking of trying to use libavformat directly :-) [14:52] <JEEB> I wish thee luck, you should first get current ffmpeg built for whatever architecture you're using android under [14:52] <JEEB> and then you shall open the documentation [14:53] <JEEB> and read where available :) [14:55] <luc4> JEEB: I am a little familiar with this: http://ffmpeg.org/doxygen/trunk/doc_2examples_2muxing_8c-example.html. I suppose that is what I should refer to. I cross-compiled ffmpeg for some arhitectures in the past and it worked perfectly, maybe there is something done already for android... [14:56] <luc4> JEEB: this seems nice! http://sourceforge.net/projects/ffmpeg4android/ [15:47] <MortenB> Hi there. I'm converting Canon Ixus digital camera videos with 11024 Hz mono pcm_u8 sound into mp4 files using libfdk_aac for the sound side. If I don't upsample I get "Unable to encode frame: Encoding error". If I do (using -ar 44100) I get a weird hiss, that is noticeably badder sound quality, irrespective of VBR setting. How do I go about this? [15:52] <Peace-> hey guys i did this but i got a video that is too much fast ... /home/shared/ffmpeg-04-23-13/./ffmpeg -s $(xrandr | awk '/, current /{print $8}')x$(xrandr | awk '/, current /{gsub(/\,/,"");print $10}') -f x11grab -r 25 -i :0.0 -vcodec h264 -y $HOME/output.avi [15:53] <Peace-> i have changed codec but no way [15:53] <Peace-> the video it's always too much fast [16:37] <luc4> Hi! I'm trying to compile a cpp source file but it keeps complaining undefined reference to `avcodec_get_name', and a couple of others... shouldn't this be included in the static libs? [16:38] <JEEB> luc4, this? http://ffmpeg.org/faq.html#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-… [16:39] <luc4> JEEB: done that already yes [16:40] <luc4> JEEB: I'm trying to compile muxing.c to be precise. [16:40] <JEEB> that'd be C, not C++... [16:40] <JEEB> and yes. those should be in the libraries [16:40] <luc4> JEEB: yes, but I put those sources in a c++ file [16:41] <luc4> I had to redefined a symbol which was not compiling [16:41] <luc4> but still for 6 functions I get undefined reference [16:42] <JEEB> also you should check the api changelog documentation regarding those undefined ones, make sure the example (if that was it) was up-to-date [16:42] <JEEB> and do check the exports of that library to make sure what is being exported [16:44] <luc4> JEEB: those are in the header... by exported you mean __attribute__((__visibility__("default")))? [16:45] <luc4> JEEB: for examples, I don't understand why avcodec_get_name should not be visible... [16:46] <JEEB> no, I mean check with tools what symbols the libraries export [16:46] <JEEB> the libraries you built [16:47] <luc4> JEEB: seems to be there yes: avcodec_get_name [16:48] <JEEB> ok, then make sure you are linking to that library [16:48] <luc4> I used pkgconfig line: -pthread -L/usr/local/ffmpeg-1.2/lib -lavformat -lavcodec -ldl -lva -lXfixes -lXext -lX11 -ljack -lasound -lSDL -lx264 -lvorbisenc -lvorbis -ltheoraenc -ltheoradec -logg -lspeex -lrtmp -lgnutls -lopencore-amrnb -lz -lavutil -lm -lswscale [16:49] <JEEB> if the library is there, and gets linked to, and you have included all the headers correctly as C stuff, I don't see a reason for it to fail :P [16:49] <JEEB> so make yourself a checklist and see where you could have failed [16:49] <luc4> maybe... the linker is taking the libs from my system?... [16:50] <luc4> I mean those from the system path instead of those I told it to. [17:00] <luc4> JEEB: ok, that was it. For some reason it was ignoring the -L and was taking the system libs. Thanks. [17:02] <JEEB> luc4, I think the -L spots are added to the end of the search path, no? [17:03] <luc4> JEEB: I thought it could override the system paths... [17:04] <JEEB> that would happen if the -L flags set directories before the search path [17:04] <JEEB> I /think/ they set them /after/ [17:18] <xlinkz0> how do i use -report? [17:18] <xlinkz0> how do i set the filename? [17:19] <xlinkz0> tried -report:file=name [17:24] <trose> xlinkz0: i'm not an ffmpeg dev but i found this [17:24] <trose> http://stackoverflow.com/questions/11241878/ffmpeg-report-generation [17:24] <trose> hopefully it helps? [17:25] <xlinkz0> thanks [18:05] <ov3rmind> hello guys is a pleasure undertand a litle to try stream audio & video thanks all for any comment ? ! :) [18:05] <ov3rmind> "understand" [18:25] <ov3rmind> http://iworks.srv.br/~leo/relar%C3%B3rio_Stream-server.html i can't connect webcam on debian 7 any help are greatly welcome [18:55] <ov3rmind> any can help? [19:54] <ov3rmind> how i can see the formats suported by one codec in ffmpeg ? [19:54] <ov3rmind> thanks for any advice ! [19:56] <ubitux> ./ffmpeg -h encoder=ffv1 [19:56] <xlink> how do i put a watermark on a video? [19:56] <xlink> i tried ffmpeg -y -i video.mp4 -vf "movie=photo.png[watermark];[in][watermark]overlay=10:10 [out]" out.mp4 [19:56] <ubitux> 'doesn't work? [19:56] <xlink> but i get a lot of [png @ 0435fbe0] Missing png signature [19:57] <xlink> ubitux, http://codepad.org/0l6Glxf5 [19:58] <ubitux> xlink: can you share that png9$? [19:58] <ubitux> -9$ [19:59] <smj> do you know a music player that tracks the contents of a directory? so that I don't have to add the files manually to the playlist when playing on shuffle [19:59] <xlink> ubitux, http://www.iconspedia.com/icon/stopwatch-icon-20052.html [20:00] <ubitux> xlink: which one? [20:00] <xlink> the png one [20:00] <xlink> hover over the big picture [20:00] <xlink> ubitux, full download http://www.iconspedia.com/dload.php?up_id=110917 [20:02] <ubitux> mmh [20:02] <ubitux> looks like that png is not correctly supported [20:03] <durandal_1707> that palette looks to have alpha [20:04] <ubitux> xlink: can you open a ticket attaching that sample? [20:04] <xlink> idk how.. [20:04] <xlink> so the command is good? [20:04] <ubitux> yes, it's a bug [20:04] <xlink> thanks i just needed to find a command [20:05] <ubitux> it seems converting the png to gif works as a workaround [20:05] <ubitux> ffmpeg -i /tmp/Stopwatch-256.png test.gif [20:05] <ubitux> then use test.gif in your filtergraph [20:12] <ubitux> xlink: https://ffmpeg.org/trac/ffmpeg/ticket/2556 [20:12] <xlink> thanks [23:21] <luc4> Hi! I'm trying to mux a h264 coming from a variable frame rate source. When I open my mp4, I get that portions are slower and other are faster. I suppose this means that the timestamp was not taken into consideration when muxing right? [23:31] <relaxed> luc4: Try muxing with MP4Box. If that works then file a bug report against ffmpeg and provide your sample. [23:32] <luc4> But can I provide timestamps along to h264 stream? [23:32] <luc4> I mean, can I do that with this MP4Box? [23:34] <Zarx> if I want to load a video file (that may contain an arbitrary number of audio streams) and then also load a seperate audio file, then copy ONLY the seperate audio file along with the video into my output, what would that commandline look like? [23:35] <Zarx> basically, ignore all audio streams from first input [23:37] <relaxed> Zarx: ffmpeg -i video.mkv -i audio.flac -map 0:v -map 1:a ... [23:39] <Zarx> ok, let me try this... [23:45] <Zarx> seems like its doing something different than before. Thanks, ill play with this for a bit. [23:56] <luc4> When muxing with libavformat a h264 stream with vfr, is it correct to create a AVacket like this http://paste.kde.org/741548/ to call the av_interleaved_write_frame afterwards? Or maybe something is missing? Because at the end of my muxing vlc seems also unable to get the entire length of my video. [00:00] --- Sat May 11 2013

1 0

ffmpeg-devel.log.20130509
by burek 10 May '13

10 May '13

[00:00] <michaelni> FF_CEIL_RSHIFT() OTOH might be understood by a random reader as it refers just to iso C terminology and r close to "shift" likely would get correctly guesseed to right [00:01] <ubitux> fine with me [00:03] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:df9f9caba58a: avfilter: Make enabled/disabled state available as a field of the AVFilterContext [00:03] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:1e1015f330e3: avfilter/af_apad: Support AVFILTER_FLAG_SUPPORT_TIMELINE [00:03] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:410905f7fd0d: ffmpeg: use av_freep() for forced_kf_pts [00:03] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:33ff943f70f0: ffmpeg: add -apad option which injects a apad filter only in the presence of video streams [00:03] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:e4723a82f708: avfilter: fix 2 warnings about discarding const [00:06] <ubitux> ah, not a bad idea [00:06] <ubitux> it could help simplifying overlay [00:08] <ubitux> michaelni: maybe the passthrough callback could be removed then [00:08] <ubitux> and replaced with another flag [00:08] <ubitux> like "AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL" [00:10] <ubitux> #define AVFILTER_FLAG_SUPPORT_TIMELINE_INT AVFILTER_FLAG_SUPPORT_TIMELINE | (1 << 17) [00:10] <michaelni> maybe, i dont know which way its better [01:26] <ubitux> michaelni: https://github.com/ubitux/FFmpeg/commit/61bd2b341ea579c55d85b823fb53aaa8139… [01:26] <ubitux> that's definitely sexier [01:27] <ubitux> also allows: https://github.com/ubitux/FFmpeg/commit/c44bdcf1998a9193cb33aad4e0d8b1ff7ff… [01:43] <llogan> looks like fflogger forgot his meds today [02:06] <ubitux> "The alpha channel is encoded separately and the output is placed in AVPacket's side_data" [02:06] <ubitux> huh?? [03:01] <highgod> Hi, all. I want to ask two questions, I have use ffmeg dshow to capture the desktop. First is, does ffmpeg has comand line to use dshow to capture a specific window? The second is, dose it can capture a specific area of desktop? Thanks [07:26] <burek> hi, can anyone, familiar with the topic "HLS with encrypted stream", help to answer this: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=897 [07:28] <burek> also, this one, about using vf inside preset files: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=917 [11:20] <cone-764> ffmpeg.git 03Diego Biurrun 07master:87614667606b: x86: dsputil: Move ff_pd assembly constants to the only place they are used [11:20] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:3469c24a1084: Merge commit '87614667606b42476f9017d79faf12b45a0bd77c' [11:34] <cone-764> ffmpeg.git 03Diego Biurrun 07master:ed880050edf0: x86: dsputil: Group all assembly constants together in constants.c [11:34] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:5747d835c775: Merge commit 'ed880050edf061b38d3e39e25657c59ad9108b27' [11:42] <cone-764> ffmpeg.git 03Diego Biurrun 07master:71469f3b636f: x86: dsputil: Move constant declarations into separate header [11:42] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:164899c6e858: Merge commit '71469f3b636fbe06b6aca5933f9fdebddd8d5f57' [11:51] <cone-764> ffmpeg.git 03Diego Biurrun 07master:3d40c1ee742d: x86: dsputil: Move TRANSPOSE4 macro to the only place it is used [11:51] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:bda5487d92fc: Merge commit '3d40c1ee742db5f13ebcf53c2d1fa4bf4f39bcd2' [11:55] <cone-764> ffmpeg.git 03Diego Biurrun 07master:6e9f8d6a7d73: x86: vf_yadif: Remove stray dsputil_mmx #include [11:55] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:696f5f98e282: Merge commit '6e9f8d6a7d7392a236df19fef6f4eba41f18167e' [12:36] <cone-764> ffmpeg.git 03Diego Biurrun 07master:f243bf7aa27f: x86: dsputil: Remove unused argument from QPEL_OP macro [12:36] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:769efe56b145: Merge remote-tracking branch 'qatar/master' [12:43] <durandal_1707> michaelni: there is no way to workaround issue for le/be pixformats and pixfmts filter tests? [12:43] <durandal_1707> because filter would for highger bitdepths pick native endian format [12:49] <cone-764> ffmpeg.git 03Paul B Mahol 07master:d1cb559a6a35: fate: add test for lut filter [12:54] <ubitux> durandal_1707: do you need review of anything? [12:55] <durandal_1707> ubitux: chapter encoding for id3v2 [12:56] <durandal_1707> test with some program that does not use lavf but can read & show chapters [12:56] <ubitux> anything in lavfi ? [12:57] <durandal_1707> want to extend colorkeymask ? [12:57] <ubitux> what's the usage of such filter? [12:57] <ubitux> s/usage/common case/ [12:58] <durandal_1707> http://en.wikipedia.org/wiki/Chroma_key [12:58] <ubitux> oh [12:58] <ubitux> that will be pretty nice [13:02] <durandal_1707> well, i did not compared but its current funcionality should be better or same as avisynth internal filter of same name [13:02] <durandal_1707> i also added yuv support locally [13:04] <ubitux> should i wait for the next version for a review? [13:05] <durandal_1707> nope, you are free to extend it and/or write documentation [13:05] <ubitux> i'm busy with my own filters :) [13:05] <ubitux> i was just asking if you need a ok or anything for some lavfi-related work [13:05] <ubitux> can't help you much with id3 chaptering [13:06] <durandal_1707> nobody here have amarok/songbird installed? [13:06] Action: ubitux uses mpd [13:06] <durandal_1707> is vlc using lavf for id3 tags? [13:07] <funman> no [13:07] <funman> we use taglib [13:11] <durandal_1707> ubitux: you can comment/try avectorscope (added zoom locally) [13:12] <ubitux> is your remote up-to-date? [13:16] <ubitux> oh it's pretty with colors [13:25] <ubitux> durandal_1707: the zoom creates some kind of "maggots" effect, it's fun [13:29] <durandal_1707> what ? [13:32] <ubitux> my imagination certainly :) [13:32] <ubitux> if you zoom, there is some kind of little arc circle [13:33] <ubitux> not strong, but enough with the motion to give a swarm feeling [13:33] <durandal_1707> you mean different color? [13:34] <durandal_1707> what zoom ? [13:39] <ubitux> durandal_1707: http://lucy.pkh.me/avectswarm.webm [13:39] <ubitux> it's a very mild effect, and most likely in my head ;) [13:41] <ubitux> (zoom=10) [13:42] <durandal_1707> it happens with zoom 1 too, its because fade/contrast differs, creating such effect [13:45] <ubitux> why +1 for x? [13:48] <durandal_1707> because value on the left is in -1, +1 range [13:54] <durandal_1707> funman: and vlc displays id3 chapters? [13:54] <funman> never heard of id3 chapters so no clue :D [13:55] <funman> i doubt it does [13:56] <durandal_1707> http://forum.videolan.org/viewtopic.php?f=7&t=104274 [13:57] <durandal_1707> i downloaded songbird for windows (run under wine) and it looks to not show chapters too [14:02] <funman> https://trac.videolan.org/vlc/ticket/7485 apparently someone wanted to support it [14:05] <durandal_1707> the files linked does not have text null terminated, as described in spec [14:09] <funman> can you add a good sample ? [14:09] <durandal_1707> one created by ffmpeg? [14:13] <durandal_1707> looks like only apps that can display it need mono or java [14:14] <durandal_1707> but taglib should export chapters as CHAP frames [14:14] <durandal_1707> so it should be trivial to add in vlc [14:21] <funman> i'd guess especially since mkv/mp4 chapters are already supported [14:22] <durandal_1707> https://github.com/taglib/taglib/issues/81 [14:27] <Compn> what? [14:32] <Compn> ehe [14:32] Action: Compn comes up with solution for michaelni and fate crc/md5 [14:47] <Compn> michaelni : does fate disable video (-vn) for audio tests and audio (-an) for video testing ? [14:48] <ubitux> why does "MurMur" makes me think of "Moon Moon"? [14:49] <Compn> course i didnt see much speed up with -f framecrc -an anyways... [14:50] <michaelni> Compn, i think so but didnt check, also running both v& a can be a good thing as more is tested [14:53] <Compn> michaelni : i'm just looking for ways to save time in fate runs :) [14:53] <Compn> maybe have a fate-fast with these alternates [14:54] <Compn> like fate-fast would only tests the first 10 frames in each sample, instead of the entire sample? [14:55] <Compn> unless the entire ref is checked... maybe too difficult [14:56] <ubitux> sounds like a maintainance burden [14:57] <ubitux> i wonder if moving lavf-regressions.sh to .mak wouldn't help [14:57] <ubitux> (all the successive if in that file for each test might be a bit too much) [14:59] <ubitux> also, md5.sh might be slow to call [14:59] <Compn> what about optimizing md5.c ? [15:00] <Compn> i dont see any yasm, cpu opts, gpu offloading, etc [15:00] <Compn> :) [15:01] <ubitux> we could switch to sha and use git implementation [15:02] <michaelni> we also could use md5 or sha1 with fewer rounds [15:02] <michaelni> but i doubt that would win the speed record either [15:05] <Compn> are you running complete fate tests when the code you changed only affects video decoding ? or is it compartmentalized enough already ? [15:05] Action: Compn afk [15:32] <michaelni> Compn, i tend to run full fate tests [15:49] <funman> michaelni: hm swscale's init_hscaler_mmxext is problematic for Windows Store [15:50] <funman> VirtualAlloc (mmap) is forbidden and i doubt there's another way to have self modifying code [15:50] <funman> that's why mozilla complained Microsoft was doing a favor for IE by forbidding other browsers to use javascript JIT [15:54] <michaelni> funman, only the fast bilinear scaler uses self modifying code, this shouldnt be hard to disable ... [15:56] <funman> add some configure option ? [15:56] <michaelni> possible [15:56] <michaelni> if it cant be autodetected [15:57] <michaelni> i dont know if theres anything special set when compiling for "Windows Store" [15:59] <funman> hmm well yes and no [16:00] <funman> in our case we don't set anything special [16:00] <funman> there is some other code we need to remove also [16:00] <funman> like GetProcessAffinityMask [16:01] <funman> i think j-b is working on this [16:23] <michaelni> ok, if someone needs help with making that optional in sws ... but i think it should be fairly straightforward [16:24] <durandal_1707> what to make optional in sws? [16:28] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:3d66ce17b7f8: tools: Add adler32 checksum tool [16:28] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:a11593189b26: ffadler: add multifile support [16:30] <michaelni> durandal_1707, self modifying code as windows store doesnt like it [16:32] <durandal_1707> sws have that? [16:34] <funman> durandal_1707: see call to mmap in init_hscaler_mmxext [16:39] <durandal_1707> michaelni: am I correct that there is not way for user(via ffmpeg) to add chapters and chapter metadata? [16:44] <JEEBsv> I think there are ways to add metadata? [16:44] <JEEBsv> saw on the documentation pages [16:45] <durandal_1707> but not chapter one... [16:50] <JEEBsv> yes, true [16:50] <JEEBsv> missed that chapter word in the second one [16:52] <ubitux> michaelni: any comment on the timeline patchset? [16:53] <cone-764> ffmpeg.git 03Paul B Mahol 07master:88e0e2054d91: lavfi: remove alphaextract special code [16:54] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:d9cb1e0e15e5: ffadler: fix duplicate \n on errors [16:54] <ubitux> durandal_1707: and the dep? [16:55] <durandal_1707> it does not need one [16:58] <michaelni> ubitux, needing 2 flags is somehow comlpex otherwise i see nothing bad on the patch [17:00] <cone-764> ffmpeg.git 03Clément BSsch 07master:570d63eef3f9: lavu: add FF_CEIL_RSHIFT and use it in various places. [17:00] <cone-764> ffmpeg.git 03Clément BSsch 07master:f0250cc46366: lavfi/overlay: simpler up-rounded w/h computations. [17:00] <cone-764> ffmpeg.git 03Clément BSsch 07master:4db84bac1353: lavfi/opencl: replace SHIFTUP with FF_CEIL_RSHIFT. [17:00] <ubitux> michaelni: do you have a better suggestion? [17:02] <ubitux> btw, it seems we do link->[wh] >> xsub in various filters [17:02] <ubitux> michaelni: mmh i'll push a fix in 1.2 [17:03] <durandal_1707> ubitux: what filters? [17:06] <ubitux> durandal_1707: git grep 'link->[wh].*sub' [17:06] <ubitux> possibly variants with width/height and log2_chroma [17:08] <cone-764> ffmpeg.git 03Clément BSsch 07release/1.2:3d6219b6dbf4: lavfi/subtitles: zero-init AVSubtitle. [17:08] <ubitux> ok i think that's all i want to backport :) [17:13] <michaelni> ubitux, no, not really, the 2 flags just dont look pretty ... [17:16] <durandal_1707> ubitux: going to fix that hsub/vsub issues? [17:17] <ubitux> durandal_1707: maybe a few ones [17:17] <ubitux> i need a test case [17:17] <durandal_1707> i have one - do not use ffplay - it pads [17:22] <cone-764> ffmpeg.git 03Reimar Döffinger 07release/0.11:d12294f447f4: matroska: set "done" only during resync fail. [17:22] <cone-764> ffmpeg.git 03Reimar Döffinger 07release/1.0:9fef5d55538c: matroska: set "done" only during resync fail. [17:22] <cone-764> ffmpeg.git 03Reimar Döffinger 07release/1.1:2922ab7e6fec: matroska: set "done" only during resync fail. [17:22] <cone-764> ffmpeg.git 03Reimar Döffinger 07release/1.2:a857811c755b: matroska: set "done" only during resync fail. [17:22] <cone-764> ffmpeg.git 03Michael Niedermayer 07release/1.2:84df0dc40bd6: avcodec/mpegvideo: Fix block height for lowres 3 interlaced blocks [17:22] <cone-764> ffmpeg.git 03Michael Niedermayer 07release/1.2:3ca6253beb12: avcodec/mpegvideo: Fix edge emu with lowres [17:45] <cone-764> ffmpeg.git 03Carl Eugen Hoyos 07master:43b6482d0a8f: Add fate test for solid colour lagarith frames. [18:00] <ubitux> durandal11707: http://b.pkh.me/0001-lavfi-use-ceil-right-shift-for-chroma-width-height.pat… [18:00] <ubitux> let's break everything \o [18:00] <ubitux> (possibly fixing issues though) [18:04] <funman> btw the external swscale repo has been abandoned? [18:05] <durandal11707> there was such thing? [18:05] <durandal11707> ubitux: missing vflip [18:05] <kierank> funman: you not mean postproc? [18:05] <ubitux> durandal11707: it's using a weird hack so i didn't touch [18:05] <ubitux> same for pad, where there is likely something to fix btw [18:06] <funman> kierank: hum right, i'm confused [18:06] <ubitux> swscale repo was outside too a while ago [18:06] <nevcairiel> a long while now [18:06] <kierank> oh yeah ages ago [18:06] <kierank> 3yrs or so ago [18:06] <funman> yeah i meant postproc indeed [18:07] <ubitux> postproc is maintained in ffmpeg [18:07] <durandal11707> ubitux: whatever hack does its not working for "weird" dimensions for subsamples yuvs [18:07] <ubitux> the external library is just an excuse for the fork to drop it afaik [18:08] <ubitux> durandal11707: possibly in another commit then [18:08] <ubitux> cause i think this one might break a few things already [18:09] <durandal11707> what it can break? [18:09] <durandal11707> it should fix weird green lines with odd dimensions and normal are unaffected [18:10] <funman> i'll wait for j-b to be back and work on a patch to disable the self-modifying code [18:10] <funman> ./configure --disable-compiler :) [18:10] <ubitux> we should have more tests with odd dimensions [18:12] <durandal11707> ubitux: push/send to ml/ [18:12] <ubitux> yeah, i'll send the patch [18:18] <ubitux> michaelni: maybe a less ugly macro could be AVFILTER_FLAG_INTERNAL_TIMELINE? [18:18] <ubitux> i could remove the internal timeline implying the first flag too [18:28] <durandal11707> ubitux: adding more tests is makeing michaelni wait more [18:29] <ubitux> i'd be pleased to have a faster fate [18:30] <ubitux> and more tests ;) [18:31] <durandal11707> ubitux: so what i should do to get you to push that (chroma w/h) patch? [18:31] <ubitux> test if it fixes anything maybe [18:31] <ubitux> or encourage me by fixing lut/vflip/pad [18:32] <ubitux> or finding moar similar issues in lavc/lavu [18:32] <durandal11707> ubitux: that lut code is also wrong, i get pink color [18:33] <durandal11707> ffmpeg -f lavfi -i smptehdbars=1600x903,negate -f sdl - [18:35] <durandal11707> hmm it may be other way around ... [18:36] <ubitux> nothing in particular here [18:36] <durandal11707> but il/interlace/blend/extractplanes are ok [18:38] <durandal11707> -f sdl shows black last line, mpv shows green, so i will use mpv to test this [18:39] <ubitux> right, i noticed the black line [18:43] <durandal11707> ubitux: it could be bug in source filter... [19:18] <durandal11707> ubitux: interlace have same bug, as lines get / 2 later [20:59] <michaelni> latest news: trac.ffmpeg.org finally works, no more /trac/ffmpeg needed [21:11] <llogan> nice. [21:11] <llogan> any plans to update trac to 1.0? [21:15] <ubitux> btw, fate moved to the new machine? [21:19] <michaelni> ubitux, hmm, judging from the ip id say not yet [21:19] <michaelni> llogan, any features in that that we would want / need ? [21:24] <llogan> michaelni: for me, mostly a nicer looking theme to make wiki look nicer. [21:24] <llogan> "Usability improvements for the tickets, with a better support for conflict detection and resolution" might be interesting [21:25] <llogan> carl might like the now integrated BatchModifyPlugin [21:25] <llogan> i guess nothing too important since you have better things to do [22:05] <durandal11707> oh noes [22:08] <llogan> please indicate which ticket # it is a duplicate of when closing (although it is obvious here). [22:19] <Compn> doesnt it leave a comment ? [22:19] <Compn> bugzilla did... [22:25] <llogan> Compn: not that i know of but it can be manually added in the comment box when closing. i guess that's the only good thing about bugzilla. [22:30] <michaelni> can you elaborate what bugzilla does better than trac? I dont think i understand [22:30] <Daemon404> bugzilla is way better at being ugly [22:30] <Daemon404> than trac [22:38] <cone-954> ffmpeg.git 03Paul B Mahol 07master:13cb6ed3c449: takdec: simplify use of scalarproduct_int16 in decode_subframe() [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:54f390239316: rmdec: flush audio packet on seeking [22:52] <cone-954> ffmpeg.git 03ArnoB 07release/1.2:298e03d1023d: dpxenc: fix data offset [22:52] <cone-954> ffmpeg.git 03Dale Curtis 07release/1.2:f8d5f3dff5d6: Fix heap-buffer-overflow in matroska_parse_block [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:4d0090f90a97: xbmdec: fix off by one error in scanf() [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:1ce3f736d2aa: src_movie: fix scanf string [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:840931e76608: mjpegdec: fix overlapping memcpy with upscale_v [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:7fa6db254564: mmvideo/mm_decode_inter: check horizontal coordinate too [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:039f6921c276: mmvideo/mm_decode_intra: check horizontal coordinate too [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:0baa0a5a02e1: vmdav: Try to fix unpack_rle() [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:cc0dd86580b3: avutil/log: Fix context pointer used for get_category() [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:ec89046fa110: avutil/intfloat_readwrite: avoid comparission with INFINITY, use isinf() [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:9f7baf139f72: avutil/intfloat_readwrite: include common.h for isinf() [22:52] <cone-954> ffmpeg.git 03Clément BSsch 07release/1.2:91f04e741096: cmdutils: avtool -> fftool (cherry picked from commit 7d8ad6c1fa11ec548fc63427656989e0e7c6af8b) [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:524d0d2cfc7b: sanm: Check dimensions before use [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:7ef2dbd2392e: avcodec/cdgraphics: check buffer size before use [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:7ee5e97c46e3: gifdec: reset previous Graphic Control Extension disposal type [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:c5948b472b7d: update for 1.2.1 [22:52] <cone-954> ffmpeg.git 03Michael Niedermayer 07release/1.2:f166a02b67a2: exr: silence warning "libavcodec/exr.c:351:9: warning: variable ret set but not used" [22:59] <saste> bugzilla is very good [22:59] <saste> at confusing people [22:59] <saste> it prevents bad user reports, since only masochistic users are able to complete the bug registration process [00:00] --- Fri May 10 2013

1 0

ffmpeg.log.20130509
by burek 10 May '13

10 May '13

[00:03] <denysonique> whats is the ffmpeg screencast guide link? [00:03] <denysonique> popular on this channel [00:17] <llogan> denysonique: http://ubuntuforums.org/showthread.php?p=8746719#post8746719 [00:17] <llogan> or https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20%28sc… [02:43] <highgod> Hi, all. I want to ask two questions, I have use ffmeg dshow to capture the desktop. First is, does ffmpeg has comand line to use dshow to capture a specific window? The second is, dose it can capture a specific area of desktop? Thanks [03:29] <BtbN> dshow is directshow, it doesn't have screen recording capabilitys. [03:32] <sacarasc> You can use it in ffmpeg to do just that. [03:36] <highgod> just capture the desktop? [03:47] <highgod> I think I can add a filter like crop to implement it [05:58] <sacredchao> Can ffmpeg/avconv decode mbaff encoded h.264 streams? [09:18] <Threefifty> Hello, I am not to savvy with linux but I do have a ReadyNas Pro 6 that I am logged in through ssh. A Thirdparty app called headphones can use ffmpeg to convert flac files into 320kbps mp3's. All it needs is the directory where ffmpeg is installed. I know i have it installed because in ssh i can type ffmpeg and the options for ffmpeg will show. As I stated before I dont know much about linux and if anyone can help me find a way to [09:18] <Threefifty> would be greatly appreciated [09:22] <relaxed> Threefifty: run `which ffmpeg` [09:23] <Threefifty> -bash run command not found [09:24] <relaxed> type this --> which ffmpeg [09:24] <Threefifty> awesome [09:24] <Threefifty> thank you so much [09:26] <relaxed> Threefifty: you're welcome [17:49] <schtinky> I'm attempting to grab iframes from an mpegts digital TV stream with "-vsync vfr -vf 'select(pict_type\,I)'"... what build config options should I have enabled to make this possible [17:49] <schtinky> when I execute it on a machine with an ffmpeg yum package installed (and all of its config options), the command works fine [17:50] <schtinky> when I execute it on a machine with an ffmpeg build with "./configure" and no options, the command produces just a few frames and basically fails [17:50] <schtinky> no real error message though [17:53] <tclarke> I'm trying to setup ffserver to broadcast an rtp stream of an .mp4 and to loop it continuously (I'm trying to simulate a live camera feed). I'm not sure how to loop the file and I'm having trouble getting the rtp working at all...vlc just errors out...unable to connect to the feed...looks like it might be getting the rtp url from the rtsp address but the url is pointing to port 0 [17:53] <tclarke> any suggestions would be helpfull [18:09] <Emmanuel_Chanel> Hello! Is https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide still useful also for Ubuntu 13.04? [18:14] <dextro_> https://gist.github.com/anonymous/0e30d615fbd363471caf [18:14] <dextro_> The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [18:14] <dextro_> i used ffmpeg -i Django.Unchained.2012.DVDSCR.XVID-NYDIC.mdi.flv Django.Unchained.2012.DVDSCR.XVID-NYDIC.mp4 -strict -2 [18:14] <dextro_> :/ [18:22] <dextro_> nvm [18:22] <dextro_> man pages ftw [18:39] <paradoxical> Hey, are there any examples of how to do the equivalent of a filter_complex using the avfilter API directly? I'm basically using the code from here: http://ffmpeg.org/doxygen/trunk/doc_2examples_2filtering_video_8c-example.h… but trying to change the filter_descr string to do some more complex filtering. If I change it to something like "[in]scale=474:316[a];[a]pad=1050:632:25:0[out]" it succeeds. But then if I try to do somet [18:39] <paradoxical> hing like "[in]scale=474:316[a];[in]scale=474:316[b];[a]pad=1050:632:25:0[c];[c][b]overlay=551:0[out]" the call to avfilter_graph_config errors out with a -22. [18:40] <paradoxical> The only examples of using avfilter API seem to not cover how to do anything more complex than just doing scaling, or doing a crop, but nothing for wanting to do overlays or anything more complex. [18:41] <schtinky> Hey guys, I'm trying to grab an iframe every second from a USB digital tuner card... and do it on a raspberry pi [18:42] <schtinky> the command I've been using (which works great on my quad-core laptop, using very little cpu) is: [18:43] <schtinky> ffmpeg -i input.mpg -vsync vfr -vf "select=eq(pict_type\,I)" -r 1 outputfiles%9d.jpg [18:44] <schtinky> unfortunately, even when overclocked, the raspberry pi takes 1m46s to process 30 seconds of recorded video [18:44] <durandal11707> that does not take picture every second, but every keyframe [18:45] <schtinky> -r 1 forces 1 per second [18:45] <durandal11707> what video codec in in that mpg? [18:45] <durandal11707> schtinky: -r 1 sets output frame rate [18:45] <schtinky> ffmpeg says mpeg2video [18:45] <schtinky> it's a digital mpegts stream from a digital tuner card [18:46] <durandal11707> if you really need to grab key frames only there is very fast way for some codec(s) [18:46] <durandal11707> ffmpeg -skip_frame nokey [18:46] <durandal11707> use above and remove filter code -vf ... and try it [18:47] <schtinky> keep vsync? [18:47] <durandal11707> yes [18:47] <durandal11707> but set vsync -vfr to output, after -r 1 [18:48] <schtinky> "Codec AVOption skip_frame (skip decoding for the selected frames) specified for outputfile #0 (outputfile%9d.jpg) is not an encoding option [18:49] <durandal11707> skip_frame option goes after -i input.mpg [18:49] <durandal11707> its for decoding not encoding... [18:49] <schtinky> same error [18:50] <durandal11707> ah, before -i ... [18:51] <schtinky> what do you mena "set vsync vfr to output"? [18:52] <durandal11707> that you apply it for output part, so set it after "-r 1" [18:53] <schtinky> dude I think that worked [18:54] <schtinky> 20 seconds to process 30 seconds of video [18:54] <schtinky> here's what i used: tell me if you think it's optimal [18:54] <schtinky> ffmpeg -skip_frame nokey -i inputfile.mpg -r 1 -vsync vfr "outputfile%9d.jpg" [18:56] <schtinky> and let me see if I understand that command in plain english: [18:57] <durandal11707> try if -vsync 0 is faster? [18:57] <schtinky> "video encoding failed" [18:57] <schtinky> is the -r in the right place? [18:59] <durandal11707> sorry... [19:00] <schtinky> unfortunately, that's going to be tough [19:01] <schtinky> I need to reboot the raspberry into graphical mode [19:01] <schtinky> what is it you need to see? [19:05] <durandal11707> error you get [19:06] <schtinky> http://pastebin.mozilla.org/2388131 [19:07] <schtinky> two commands there, FYI [19:07] <schtinky> first one works with vfr, second one gets the error with -vsync 0 [19:08] <schtinky> btw, is there an image output format that might take less encoding power? [19:08] <schtinky> or is jpeg pretty good on the cpu? [19:09] <durandal11707> you could try: png, tiff, pam [19:09] <durandal11707> but they usually take more space are they are lossless [19:17] <schtinky> png took much longer [19:18] <schtinky> I think jpeg is the best because mpeg and jpeg both use yuv420p [19:18] <schtinky> png requires a conversion to rgp [19:18] <schtinky> rgb [19:18] <durandal11707> you could use rawvideo, no encoding at all, but no compression too [19:19] <schtinky> ding ding ding [19:19] <schtinky> tiff is the winner [19:19] <schtinky> 17s on 30s of video [19:19] <schtinky> beats jpg by 5 seconds [19:20] <durandal11707> but it writes yuvs, which are not support by lavc yet... [19:20] <durandal11707> but if you use libtiff for tiff decoding it will work [19:20] <schtinky> what's lavc? [19:21] <durandal11707> libavcodec, ffmpeg uses that library to encode/decode [19:21] <schtinky> well in any case, I think I'm set [19:21] <schtinky> thanks a ton for your help [19:43] <dextro_> schtinky use alix [19:43] <dextro_> rpi is for chumps [19:52] <dextro_> frame=185843 fps= 32 q=29.0 size= 419733kB time=01:43:20.91 bitrate= 554.5kbits/s dup=1 drop=0 [19:52] <dextro_> is that pretty slow? [19:53] <llogan> depends on your input, your encoder and its settings, and your cpu. [19:53] <llogan> and how you compiled ffmpeg [19:54] <llogan> among other things im probably forgetting [19:56] <TheSchaf> io speed :p [20:02] <llogan> moon phase and planet alignment [22:10] <dextro_> does ffmpeg lock the files [22:19] <durandal11707> nope [23:23] <hendry> can i make a audio mp3 stream using ffmpeg without shoutcast? [00:00] --- Fri May 10 2013

1 0

ffmpeg-devel.log.20130508
by burek 09 May '13

09 May '13

[00:04] <llogan> Compn: no, i wasn't questioning the "n" (i looked that up once but forgot the details...). just the significance of "0.8". [00:17] <xlinkz0> what are the reasons for getting Operation not permitted on av_seek_frame? [00:19] <ubitux> return -1 [00:20] <xlinkz0> yeah but why wouldn't the seek work? [00:21] <ubitux> that's all i can answer with the amount of info you provided [00:21] <xlinkz0> http://codepad.org/g8PietjZ [00:23] <xlinkz0> i don't think the whole code is relevant.. basically i'm trying to seek to video_stream->duration - 1500ms [00:25] <ubitux> reproducible with all formats? [00:26] <xlinkz0> Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], 3245 kb/s, 6.68 fps, 45.42 tbr, 90k tbn, 180k tbc [00:26] <ubitux> that's a codec [00:26] <xlinkz0> mp4 [00:26] <ubitux> so, only mp4? [00:26] <xlinkz0> that's what i'm testing on [00:27] <ubitux> then compare with something else [00:27] <xlinkz0> any suggestions? [00:27] <ubitux> % ./ffmpeg -v 0 -formats | wc -l [00:27] <ubitux> 254 [00:28] <xlinkz0> but what do i have to do to make the seek work on mp4? [00:28] <xlinkz0> i need it on mp4.. [00:28] <ubitux> i'm suggesting a way of getting a clue about the issue [00:28] <ubitux> not a workaround [00:29] <xlinkz0> ok so i should transcode the video to a different format and try again? [00:38] <xlinkz0> ubitux: i added pos = pos - (pos % 90000); and now it works [00:38] <xlinkz0> it seems it will only seek to specific seconds [00:38] <xlinkz0> not a huge problem for me.. but i think it's a bug in ffmpeg [01:18] <Compn> llogan : isnt 0.8 the 0.8 branch ? [01:21] <llogan> i guess so. that's what is seems to show by default, as in "n0.8-21930-gbc0cbac" [01:21] <llogan> *that's what "git describe" seems to show... [01:22] <llogan> oh, i think i did something dumb [01:22] <llogan> because 21930 isn't right either [01:23] <Compn> no, i think what happeend is your git checkout is for the 0.8 branch [01:23] <Compn> :P [01:23] <Compn> you have to git voodoo master or something to switch it back to master [01:26] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:bca50e5cd522: avcodec/mpegvideo: Fix block height for lowres 3 interlaced blocks [01:26] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:c67bca2b5a94: avcodec/mpegvideo: Fix edge emu with lowres [01:44] <cone-605> ffmpeg.git 03Stefano Sabatini 07master:c773adee3fda: lavu/mem: extend documentation for av_dynarray_add() [01:44] <cone-605> ffmpeg.git 03Stefano Sabatini 07master:ee9794ed2052: lavu/mem: fix potential int overflow and crash in av_dynarray_add() [01:44] <cone-605> ffmpeg.git 03Stefano Sabatini 07master:ae0ce99ca425: lavfi/unsharp: use verbal form in help message options [03:00] <highgod> Hi, does anyone ever use ffmpeg rtsp? Is there any sample in ffmpeg? thanks. [03:27] <Compn> highgod : sending rtsp or recieving rtsp ? [03:27] <Compn> highgod : there are rtsp sample urls on http://wiki.multimedia.cx (search for rtsp) [03:31] <highgod> both of them [03:31] <highgod> Compn:thanks, I will read it [03:39] <cone-605> ffmpeg.git 03Christophe Gisquet 07master:fc37cd43338b: x86: sbrdsp: force PIC addressing for Win64 [03:41] <Compn> http://wiki.multimedia.cx/index.php?title=RTSP#Samples_in_the_wild [03:44] <highgod> Compn: so great, thanks, another question, do you know whether ffserver can by use as a transcode server? [03:49] <highgod> Compn: hwo can I get the code? [04:05] <Compn> you mean use it like an ffmpeg youtube backend ? [04:06] <Compn> there are some projects that use ffmpeg in that way [04:06] <Compn> i dont remember the names offhand [04:09] <highgod> oh,hehe [04:10] <highgod> yes [04:10] <highgod> use the libs, not use command lines of ffmpeg.exe [04:18] <Compn> let me google some projects up [04:19] <Compn> i dont think many projects use ffserver to serve lots of users [04:19] <Compn> there are more dedicated servers for rtsp and such [04:20] <Compn> its hard to google. seems like every single web host now has ffmpeg hosting support :) [04:25] <highgod> Thanks Compn [09:53] <highgod> ffmpeg is so great [09:56] <av500> of course [10:30] <durandal_1707> the older versions of select filter used PICT_TYPE_I instead of I and etc... [11:15] <Bor0> how would I go implementing buffering state in ffplay? I have something like buffering=1; av_read_frame(...); buffering=0; but when I display the variable it always shows buffering = 0 even when it's buffering [11:29] <av500> Bor0: how much time is spent inside the av_read_frame() call? [11:31] <Bor0> it looks like it's working now though, seems that I wrongly printed the value [11:31] <Bor0> av500: I have no idea. it's a libff* internal function [11:32] <av500> yes, but you can instrument around it [11:32] <Bor0> how do you mean [11:32] <Bor0> by timing it? [11:33] <av500> indeed [11:33] <av500> but it looks like you found your error [11:33] <Bor0> seems I did, now it only remains to know if this is the right way to implement such a thing [11:34] <Bor0> assuming that my stream is a http [11:34] <av500> depends on what you want to achieve [11:34] <Bor0> when I get the same picture displayed for a couple of seconds, I want the user to be notified that his player did not freeze but it is downloading from the slow http server [11:35] <Bor0> "a couple of seconds", it could be instantly [11:39] <av500> on slow connections you might in fact implement some buffering yourself [11:39] <Bor0> how would I go for that? [11:39] <av500> so read_frames() and store them for a while, then start playing [11:46] <cone-474> ffmpeg.git 03Janne Grunau 07master:45a1694fde25: jpeg2000dec: cleanup properly after decoding errors [11:46] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:aecb3421f641: Merge commit '45a1694fde251ee56e23aec073d73172c00aa470' [11:47] <durandal_1707> michaelni: merging again? [11:54] <cone-474> ffmpeg.git 03Janne Grunau 07master:70f9661542a5: jpeg2000dec: initialize static data just once [11:54] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:1bad40ef2723: Merge commit '70f9661542a581dfe93b636b1c55b5558e4a4e3c' [11:55] <durandal_1707> michaelni: say when you are done so i can push little bomb [11:58] <Bor0> av500: could you show me some example implementation of that? I understand how it would be implemented but it'd be great if I can see an example [11:59] <av500> sorry no [11:59] <av500> also, its totally not about the channel topic :) [12:00] <Bor0> okay, thanks about your help. yes I understand, but this channel seems to be far more active regarding development than #ffmpeg (where mostly ffmpeg command line stuff is discussed) :) [12:01] <Bor0> I guess it's the devel keyword :p [12:01] <av500> if you look at ffplay, there is already some buffering of frame in queues [12:04] <Bor0> yeah, I am using ffplay. I see that buffering is already implemented, I just need to show an indicator to the user when the buffer is empty and it is waiting for a buffer fill [12:04] <cone-474> ffmpeg.git 03Anton Khirnov 07master:a97d8cc16e0d: indeo3: use unaligned reads on reference blocks. [12:04] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:e077ba801644: Merge commit 'a97d8cc16e0da30c9ffefa1ede2a0adf3db5f3f8' [12:06] <rkrishna> Hi, any plans to add support for AVCi50/100 to ffmpeg? [12:06] <JEEB> is it one of those crazy spec-incompliant "AVC Intra" formats? [12:07] <JEEB> because any sane'ish AVC will get decoded just fine in pretty much all cases [12:07] <Compn> rkrishna : got a sample of a file that doesnt work in ffmpeg ? [12:08] <durandal_1707> see c5142a95a51320e65d80d6bca0eb69fcff05508f [12:08] <rkrishna> I am actually looking for Encoder support, if we can implement it in some form [12:09] <JEEB> x264 is used for AVC encoding [12:09] <JEEB> the problem is that there is no single specification for AVC Intra [12:10] <JEEB> there's a "nice" mailing list thread about AVC Intra on x264-devel [12:10] <rkrishna> ok, so I should be talking to x264 devel guys. [12:12] <rkrishna> my understanding is we can intra encode with x264. We probably need to handle the bit stream so that it is compatible with 3rd party softwares [12:14] <cone-474> ffmpeg.git 03Anton Khirnov 07master:481575d25519: avconv: remove -deinterlace [12:14] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:42f5fc716e93: Merge commit '481575d255195c6c2f32e04686b94de984ae78d4' [12:14] <Compn> rkrishna : did you check ffmbc project ? it is tailored for avc broadcasting support [12:15] <JEEB> rkrishna, there was a thread in jan/feb 2012 @ http://mailman.videolan.org/pipermail/x264-devel/2012-January/009048.html , http://mailman.videolan.org/pipermail/x264-devel/2012-February/009078.html [12:15] <JEEB> it seems like there just is not a single specification for AVC Intra :P [12:15] <Paranoialmaniac> what is AVCi50/100? [12:17] <Paranoialmaniac> afaik AVC-Intra itself is not an issue, A fatal issue i know is that AVC-Intra-in-MOV generated by FCP doesn't contain any SPS and PPS. [12:17] <rkrishna> I checked ffmbc it does not say any thing about AVCi50/100 [12:17] <j-b> wow. so how do you decode those? [12:18] <JEEB> j-b, I guess they are hardcoded in the decoder... sounds as awful as it is :D [12:20] Action: j-b 's head explodes [12:20] <j-b> JEEB: you got to be kidding me [12:21] <av500> like mjpeg with fixed tables :) [12:25] <cone-474> ffmpeg.git 03Anton Khirnov 07master:bce70f817059: configure: make avconv select (a)trim filters [12:25] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:bf0547504b73: Merge commit 'bce70f8170592599a4f2f5f02606c1246e88df57' [12:40] <michaelni> mjpeg with fixed tables is not that bad, its just one or 2 kinds of tables, in h264 its the whole set of global headers that are missing [12:40] <michaelni> they would set more than 1 table [12:41] <michaelni> and its not one fixed value, IIRC avc intra uses different values depending on resolution and such [12:42] <cone-474> ffmpeg.git 03Anton Khirnov 07master:9779f93066ee: configure: add dependencies to lavfi .pc file [12:42] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:28397de4df4b: Merge commit '9779f93066ee5ea6fba6b76f001d358cc2857891' [12:43] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:1aed0513a1c2: configure: use shorter variable name for "libavdevice_pc_deps" [12:50] <durandal_1707> why was drawgrid rejected? [12:53] <cone-474> ffmpeg.git 03Anton Khirnov 07master:cd4bfe3a50ee: FATE: use a less ambiguous end time for filter-trim-time test [12:53] <cone-474> ffmpeg.git 03Luca Barbato 07master:e0c53c340818: nut: use meaningful error values [12:53] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:9818bbde5303: Merge commit 'e0c53c3408181d2e7d544eb65fc7355a79438dfe' [13:37] <cone-474> ffmpeg.git 03Luca Barbato 07master:e55e8b2c362a: lavc: do not leak frames in avcodec_decode_audio4 [13:37] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:fe6a36e7687b: Merge commit 'e55e8b2c362a60bad66929d83d8c83b0d7f5ffc9' [13:41] <cone-474> ffmpeg.git 03Luca Barbato 07master:6f98508e2b86: oma: return meaningful errors [13:41] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:d13bc208da3b: Merge commit '6f98508e2b86dcb79614d8c9b8e97c5e83a21cb0' [13:42] <michaelni> durandal_1707, it was rejected ? [13:55] <cone-474> ffmpeg.git 03Luca Barbato 07master:db9aee6ccf18: oma: properly forward errors in oma_read_packet [13:55] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:f302ba4ddab7: Merge commit 'db9aee6ccf183508835acc325f5ad87d595eacc4' [13:59] <cone-474> ffmpeg.git 03Diego Biurrun 07master:7edaf4edb5c3: x86: rnd_template: Eliminate pointless OP_AVG macro indirection [13:59] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:bf18810a215c: Merge commit '7edaf4edb5c3c04f34ad1242680cbc32d11f4087' [14:05] <cone-474> ffmpeg.git 03Diego Biurrun 07master:92f8e06ecb43: x86: dsputil hpeldsp: Move shared template functions into separate object [14:05] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:2f9ef60c9765: Merge commit '92f8e06ecb431a427ea13d794e5a6bc927a034d2' [14:12] <cone-474> ffmpeg.git 03Diego Biurrun 07master:63bac48f734f: x86: dsputil: Move rv40-specific functions where they belong [14:12] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:69d2eff5af67: Merge commit '63bac48f734fc69cca2ef2cfada92cd9a222734d' [14:17] <cone-474> ffmpeg.git 03Diego Biurrun 07master:3334cbec0aa1: x86: dsputil: Remove unused MOVQ_BONE macro [14:17] <cone-474> ffmpeg.git 03Diego Biurrun 07master:1b343cedd7cd: x86: dsputil: Remove unused ff_pb_3F constant [14:18] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:e3869dd17ee1: Merge commit '1b343cedd7cd68e7865aa5280d1568c7e5d79917' [14:31] <cone-474> ffmpeg.git 03Diego Biurrun 07master:bd392785a75d: fate-run: Drop unnecessary regtest() indirection [14:31] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:63208c034bf6: Merge commit 'bd392785a75d7d51414a9788d50396a8d9f29d77' [14:41] <cone-474> ffmpeg.git 03Martin Storsjö 07master:06122c253394: msvc: Move linker pragma from config.h to libavutil/internal.h [14:41] <cone-474> ffmpeg.git 03Michael Niedermayer 07master:8d9537228f03: Merge remote-tracking branch 'qatar/master' [14:41] <michaelni> durandal_1707, merge done, fire at will [14:49] <cone-474> ffmpeg.git 03Paul B Mahol 07master:5c057433ccd3: libswscale: GBRAP input & output and GBRAP16 input support [14:49] <cone-474> ffmpeg.git 03Paul B Mahol 07master:1f2baec7bd60: lavfi: add gbrap support to some filters [14:49] <cone-474> ffmpeg.git 03Paul B Mahol 07master:94ff68ccb3d1: lavfi/lut: add support for YUVA422P and YUVA444P [14:49] <cone-474> ffmpeg.git 03Paul B Mahol 07master:443181c0a9a8: libswscale: simplify isAnyRGB macro [14:49] <cone-474> ffmpeg.git 03Paul B Mahol 07master:8e2972274e1b: lavfi/null: remove redudant lines [15:29] <kurosu_> is there a macro to round a value A to the nearest multiple of N being >A ? (something like (A+N-1) & -N if I'm not mistaken) [15:30] <nevcairiel> there is FFALIGN, but it only works for N being a power of two [15:30] <kurosu_> yeah, what I meant, sorry [15:31] <nevcairiel> well then thats what you want :) [16:07] <Hywan> Hi all. I would like to know if it is possible to record a Mac OS X screen without using VLC or another software. I read this post (from 2010) http://ffmpeg.org/pipermail/ffmpeg-devel/2010-May/095605.html that gives some clues. Are they any developments in this direction currently? [16:15] <xlinkz0> Hywan: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(scree… [16:16] <Hywan> xlinkz0: Mac OS is not listed here. [16:16] <xlinkz0> doesn't mac also use x11? [16:17] <Hywan> xlinkz0: With XQuartz, yes, but you will be able to only grab X11 windows, not the other (native) windows [16:18] <xlinkz0> idk [16:18] <xlinkz0> doesn't apple provide its own software for doing so? [16:19] <Hywan> Yes. Quicktime Player, but I can't use it as an input of ffmpeg (except if I do it in 2 steps: recording at first and then processing with ffmpeg). And Quicktime Player can not be used in CLI. [16:22] <durandal_1707> ubitux: FF_RIGHT_SHIFT_ROUNDED_TOWARD_INFINITY() [16:23] <durandal_1707> but FF_DO_NOT_USE() is less scary [16:26] <ubitux> :/ [16:32] <Compn> Hywan : install qt_tools , command line interface for quicktime [16:32] <Compn> its very useful :) [16:33] <Compn> Hywan : http://omino.com/sw/qt_tools/ [16:33] <Hywan> Compn: Does it allow me to record a specific region of the screen? I didn't see this on the website. [16:35] <Compn> actually i'm not sure if it will work for you [16:35] <Compn> lets see if i can find something better [16:37] <Compn> looks like a two step process so far , http://superuser.com/questions/164892/screen-capture-command-line [16:38] <Hywan> Compn: I would like to precisely specify a region of my screen. [16:38] <Compn> Hywan : this is what you probably want > https://github.com/csexton/captured [16:40] <Compn> http://www.tuaw.com/2007/05/17/terminal-tip-6-easy-ways-to-capture-your-scr… [16:40] <Hywan> Compn: "captured" only capture images. [16:40] <Compn> i wonder if you can set screencapture command line program to pipe png files to ffmpeg for outputting as a movie or stream [16:41] <Hywan> Compn: Wow, it would be very slow :-/, no? [16:41] <Compn> mkfifo lotsofpng && capture | ffmpeg -f image2 -i lotsofpng out.mov [16:41] <Compn> for encoding png ? shouldnt be that slow [16:41] <Compn> depending on encoding codec [16:41] <Compn> depending on what you are recording [16:42] <Compn> are you screen recording video or just regular programs ? [16:42] <Hywan> regular programs [16:43] <Compn> should be fine, but i'm not sure the interval possible with those programs [16:44] <Hywan> Compn: Very low :-( [16:46] <ubitux> http://pastie.org/7818302 any suggestion before i add this to the wiki? [17:05] <Hywan> ubitux: xwininfo --root would be better no? [17:12] <ubitux> the idea is to record a window you clic on [17:15] <durandal_1707> then add that explanation to wiki too [17:51] <d```> is there a list or page of all the hardware supported by ffmpeg? I'm trying to find an ARM platform that's supported [17:55] <funman> d```: if it runs linux it is supported [17:56] <d```> funman: as far as I can tell, there's no support for marvell vmeta on ARM [17:57] <d```> funman: http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2012-June/000708.html [17:57] <d```> and yes, I realize that was from almost a year ago [17:58] <funman> that's for some specific hardware decoder afaiu [17:58] <funman> cpu decoding works fine [17:59] <d```> that's what I'm looking for, what hardware is supported [17:59] <d```> sorry if I was unclear [18:00] <Compn> ubitux : i dont know what this is ... but its interesting. http://ebit.lbl.gov/~arun/ffmpeg/index.html [18:00] Action: Compn searching for 'ffmpeg' on .gov sites [18:00] <funman> d```: i'm fairly sure, 'none' [18:01] <cptspiff> d```: no omx in ffmpeg afaik. which typically is the interface to these decoders (or properietary nda'd shit). [18:02] <d```> funman cptspiff: thanks for the info, this makes me sad [18:04] <Compn> d``` : you mean you want hardware acceleration support list ? [18:04] <Compn> there should be one in ffmpeg, just grep hwaccel [18:04] <d```> Compn: that's exactly what I'm looking for [18:04] <cptspiff> np. you can have a look at https://github.com/xbmc/xbmc/tree/master/xbmc/cores/omxplayer for a player using avformat for demux and omx for decode [18:05] <Compn> d``` : unfortunately, all the arm apps i've seen using ffmpeg are closed source and no one felt like sharing code with us [18:05] <Compn> theres a few ffmpeg-android projects which may have optimizations. but i havent done an exhaustive search to find out [18:05] <cptspiff> droid needs stagefright love.. [18:06] <Compn> d``` : its quicker just for us to say ffmpeg doesnt have any arm based hwaccels iirc [18:07] <funman> Compn: vlc is open source ;) [18:07] <funman> we use omx decoder on android [18:07] <Compn> :) [18:07] <Compn> funman : why u no share with us your code then ? [18:07] <Compn> :P [18:07] <funman> maintained by wbs [18:07] <funman> Compn: just for evil :P [18:08] <d```> Compn: thanks for all the info, I wish I knew all this a few days ago [18:08] <cptspiff> funman: is omx avail on all hw? [18:08] <Compn> http://limoa.sourceforge.net/docs/0.3/group__limoi__component__ffmpeg__omx.… [18:08] <Compn> whats this limoa thing [18:08] <Compn> it looks useful to us... [18:09] <Compn> cptspiff : not a chance in hell [18:09] <Compn> all these arm devices have different hw [18:09] <Compn> and api [18:11] <Compn> am curious what people use ffmpeg on a phone for anyway [18:11] <Compn> streaming video ? [18:11] <cptspiff> phones, not so interesting. pads, more interesting. [18:11] <funman> cptspiff: omx is just an api [18:11] <funman> which loads real decoders dynamically if present [18:12] <cptspiff> yes, i know this. point is, we fell down on stagefright since if you use omx, there is no guarantee there are sw decoders around. with stagefright you have a minimum set of features that MAY be hw backed. [18:12] <funman> and yes every device as its own decoder [18:13] <Compn> but yeah, having xbmc on android is amazing... [18:14] <JEEBsv> I still have my first-gen Xbox with XBMC on it :D [18:15] <cptspiff> heh mine still lives as well. which is a damn miracle consider how i have treated it over the years [18:37] <durandal_1707> http://imageshack.us/download/826/27283552.png [19:08] <durandal_1707> one can use overlay with timeline? [19:11] <ubitux> yes [19:13] <durandal_1707> then i could add it to blend [19:16] <durandal_1707> i get warnings, and wrong few frames, guess because relevant code is missing [19:16] <ubitux> you need to use the passthrough callback [19:18] <durandal_1707> ugh, that code in overlay is not nice [21:38] <cone-463> ffmpeg.git 03Paul B Mahol 07master:deef4b1ac944: lavfi/drawutils: add 16 bit rgb and planar rgb to ff_fill_rgba_map() [21:38] <cone-463> ffmpeg.git 03Paul B Mahol 07master:8245520b8800: lavfi/colorchannelmixer: remove now redudant code [21:38] <cone-463> ffmpeg.git 03Paul B Mahol 07master:c4a5499d259b: lavfi/extractplanes: packed rgb support [22:09] <ubitux> durandal11707: how did you do the dep thing? [22:09] <ubitux> added #if CONFIG... around the specific extractplanes code? [22:10] <durandal11707> ubitux: in configure [22:10] <ubitux> what if you want only alphaextract? [22:10] <durandal11707> also i moved #if CONFIG to right place (it was wrong) [22:11] <durandal11707> that is little akward [22:11] <ubitux> why? [22:11] <durandal11707> well, removing deps from configure should make that possible [22:12] <durandal11707> i could add ifdev around options if that would silence warnings [22:13] <durandal11707> *ifdef [22:13] <durandal11707> and unused class.. [23:15] <ubitux> durandal11707: could we use extract/merge planes filter to make some filter work only on specific planes? [23:17] <durandal11707> not yes, as there is not mergeplanes filter [23:17] <durandal11707> *yet [23:18] <ubitux> i wonder what you are waiting for! [23:18] <durandal11707> i want to make good chroma key filter [23:19] <durandal11707> i already sent link to some experiment i had with colorkeymask and overlay [23:20] <durandal11707> results is pretty good for such trivial implementation, but i can be improved (adding extra modes ...) [23:22] <durandal11707> for merge planes filter you just want to be able to merge gray single planes? [23:23] <durandal11707> well i can do it if only merging is going to be made (hence that name....) [23:42] <ubitux> should i just name the macro FF_SHIFTUP()? [23:56] <michaelni> thats possible but i think noone will understand what the macro does without reading the #define [00:00] --- Thu May 9 2013

1 0

ffmpeg.log.20130508
by burek 09 May '13

09 May '13

[00:04] <Sashmo_> anyone have an idea why vlc wouldnt open a UDP unicast in linux, but it opens in windows?? [00:21] <Zarx> im trying to build ffmpeg with patched x264 (--fade-compensate patch), and I'm not sure if I'm having a problem with the patch itself, or if i am using the wrong syntax in my commandline when I encode. [00:21] <Zarx> this is my commandline: [00:21] <Zarx> ffmpeg.exe -i "inputfile.avs" -c:v libx264 -x264opts fade-compensate=0.6 "outputfile.mp4" [00:24] <smj> if :0.0 and :1.0 are X servers, what are :0.0, :0.1, :0.2,...? [00:30] <ezekiel> smj: "Screen"s [00:32] <ezekiel> each server can have an arbitrary number of 'Section "Screen"' definitions, which are individually handled via the use of DISPLAY variables like =:0.2 [03:11] <dregan> what is the link for the precompiled ffmpeg binaries for centos ? [03:12] <highgod> Hi, I want to ask a question, does anyone ever use ffmpeg rtsp, is there any sample I can reference? thanks [03:14] <reindeer> Q:Is it possible somehow to stream multiple tracks? What stream format could support it, like i could have two inputs and then stream it out, so that receiver (ffplay) could choose track1, or track2? [03:15] <reindeer> highgod: check this out http://sonnati.wordpress.com/2011/08/30/ffmpeg-%E2%80%93-the-swiss-army-kni… [03:17] <reindeer> i bet that will get you streaming..not multichannel though, like what i am after :-/ [03:18] <klaxa> reindeer: shouldn't be impossible [03:18] <reindeer> highgod: and this http://ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide [03:18] <klaxa> i barely use ffserver though... [03:18] <klaxa> not sure if that is configurable [03:20] <reindeer> klaxa: yeah, i think im trying to find the correct codec and stream type that would support it [03:20] <reindeer> just a bit tired and trying to get it work.. [03:21] <klaxa> container would be most important [03:23] <reindeer> hmm..libx264 and aac.. [03:24] <klaxa> both codecs, both (more or less) irrelevant to containers [03:25] <reindeer> yeah..see ...tires 8-) [03:25] <reindeer> tired [03:26] <reindeer> basically anything that both ends understand should work [03:26] <klaxa> well with streaming, most people use flv as a container [03:26] <klaxa> however, its supported codecs are limited [03:26] <klaxa> i'm not even sure it supports multiple tracks for either audio or video [03:27] <highgod> reindeer:thanks [03:27] <reindeer> highgod: your welcome [03:27] <reindeer> im trying to didgest the same pages.. [03:28] <klaxa> somewhere in my TODO list is to implement a matroska http server... but that's still far away [03:28] <reindeer> that would rock [03:29] <klaxa> i haven't seen a lot of open-source software that can stream matroska [03:29] <klaxa> until today i counted exactly: 0 [03:29] <klaxa> even though matroska is designed to be streamed (apparently) [03:30] <highgod> reindeer:is there any sample which using libs of ffmpeg to use the rtsp? [03:31] <reindeer> klaxa: i've taken 5 http streams and output to mkv with 5 tracks, done. but now i would just want to listen to those, without outputing to file. thats why im trying to figure this streaming out, so i could at least stream to locally to me [03:32] <klaxa> can you elaborate your setup a bit further? how did you mux the http streams, where does the mkv go? [03:32] <reindeer> highgod: if by sample you mean "examples of command line code" your best source from me are still those links :) [03:32] <klaxa> does the mkv even go anywhere? [03:33] <klaxa> if you gather the input with: ffmpeg -i <input 0> -i <input 1> [...] -i <input n> streams.mkv you could use pipes [03:33] <klaxa> like this: ffmpeg -i <input 0> -i <input 1> [...] -i <input n> -f matroska pipe: | ffplay - [03:33] <reindeer> klaxa: just multiple "-i http://stream" then "-map 0:0 -map 1:0" and output file [03:33] <klaxa> yeah you could just pipe that to ffplay if that helps [03:34] <highgod> reindeer:I mean some codes using libs of ffmpeg to implement the rstp, thanks [03:34] <reindeer> yeah definetly helps.. [03:35] <reindeer> highgod: what you mean by "codes" or "sample" .. :-/ [03:36] <reindeer> klaxa: hmm..im gonna try that [03:40] <highgod> reindeer:sorry, I mean some code calling ffmpeg API to implement the rtsp [03:41] <reindeer> highgod: sorry no info about that [03:42] <reindeer> maybe google ffmpeg API [03:44] <reindeer> klaxa: it worked, great! [03:44] <klaxa> :) [03:45] <highgod> reindeer:thanks [03:45] <reindeer> i pushed to vlc, so i can switch the tracks, haven't used ffplay at all..yet :( [03:45] <reindeer> highgod: good! [03:45] <reindeer> :) [03:46] <klaxa> you can also use mplayer or in general any player that can work with mkv [03:47] <reindeer> yeah, thats what i figures, but i thought that i would need to specify, into what the stream is piped to, but seems not [03:48] <reindeer> not just to change name "and it works" [03:48] <reindeer> its readind from "fd://" at the moment [03:48] <reindeer> actually "fd://0" [03:49] <reindeer> but hey it works, so awsome [03:49] Action: reindeer really needs to get some sleep..soon..in few hours :) [03:50] <reindeer> yeah, ill make something more usefull from this tomorrow..gnite! 8-) [06:15] <Technician> Hello. . . will someone please help me with converting files? . . . this thread: < http://superuser.com/questions/230538/how-to-convert-3gp-audio-file-to-mp3 > explains what I am trying to do . . . the problem I am having is that partway through the conversion the process fails. How can I resolve this? [06:16] <Magicking> There is a shortcut for that ! [06:20] <Technician> http://pastebin.ca/2376183 [06:20] <relaxed> Read what fflogger said again until you understand it completely. [06:23] <relaxed> From your command can I tell your ffmpeg version is ancient. [06:24] <Magicking> relaxed: He just c/p from the link above [06:25] <relaxed> huh? [06:25] <Magicking> http://superuser.com/questions/230538/how-to-convert-3gp-audio-file-to-mp3 <= It's the same command line [06:25] <relaxed> oh [06:28] <relaxed> Technician: so? [06:30] <Technician> relaxed; I am trying it again to capture the output messages. [06:33] <Technician> http://pastebin.ca/2376195 [06:36] <relaxed> I don't see where it failed. [06:37] <relaxed> you need to change it to "-acodec libmp3lame" [06:37] <relaxed> remove "-f wav" [06:40] <relaxed> also remove "-ar 22050" unless you really need it. [06:45] <Technician> I see . . . it seems like I just don't know what I am doing. [06:46] <relaxed> If you want a high quality encode to mp3 use- ffmpeg -i input -c:a libmp3lame -q:a 0 output.mp3 [10:33] <kvz> Can anybody help me make sense of this message: "MS Video-1 warning: stream_ptr out of bounds" ? I've tried googling but that mostly leads to the source code that throws it. [10:34] <kvz> It happens when I try to convert a "Video: msvideo1, rgb555le, 1548x1080, 3 fps, 3 tbr, 3 tbn, 3 tbc" to "Video: theora, yuv420p, 960x480 [PAR 1:1 DAR 2:1], 3 fps, 3 tbr, 3 tbn, 3 tbc" [10:35] <kvz> Encoding is slow, and every once in a while it throws the stream_ptr message [10:35] <Magicking> kvz: Your error msg seems to be truncated no ? [10:35] <kvz> Let me give you full stderr [10:35] <Magicking> Wait [10:36] <Magicking> It works \o/ [10:38] <kvz> Magicking: thanks :) this is it: http://pastebin.com/WZHe8BKx [10:41] <kvz> Magicking: it seemed to run forever so I ended up killing it (hence the Received signal 15: terminating.). Between the frame updates, you can see the "stream_ptr out of bounds" warnings. I was wondering what those warnings signify. And if I can/should deal with them on my end [10:43] <durandal_1707> kvz: you are using very old ffmpeg, from 2011 [10:47] <kvz> durandal_1707: true. upgrading will cause bc-breakage for other jobs, so that would be my last resort. trying to understand if that's my only option at this point though [10:49] <Magicking> kvz: But can't you at least try I don't know in a VM or whatever to see if it's a solved bug ? [10:49] <Magicking> (If it's a bug) [10:49] <kvz> Magicking: sure [10:54] <robertzaccour> How do I record from both my mic and the sound source on my laptop? [10:54] <robertzaccour> I use ffmpeg -f alsa -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -vcodec libx264 -preset ultrafast -threads 4 -y -sameq regrunner.mkv(MP4 OUTPUT) and it captures from my mic, but how do I capture from the laptop at the same time? [10:56] <relaxed> robertzaccour: remove -sameq [10:56] <robertzaccour> relaxed: What will that do? [10:56] <JEEB> sameq is not "same quality" [10:57] <JEEB> it tries to copy the quants from the source, which makes no sense in 99% of all cases [10:57] <relaxed> robertzaccour: for your audio problem, I think you need to use pavucontrol to mix system sound and mic input. [10:58] <relaxed> JEEB: it's been removed in recent version. elenril found it actually did nothing. [10:58] <JEEB> yes [10:58] <JEEB> it seemed to do /something/ in some random cases [10:59] <JEEB> but not what it was supposed to do IIRC [10:59] <JEEB> so in that way it hadn't seemingly worked correctly for ages [11:00] <robertzaccour> relaxed: How do I do that? [11:00] <robertzaccour> relaxed: So the key to mixing both isn't in the command? [11:01] <relaxed> install pavucontrol and set ffmpeg's setting I think. I don't use pulse myself. [11:01] <highgod> Hi, I want to ask a question, how can I capture desktop in linux? [11:01] <highgod> using ffmpeg [11:01] <relaxed> highgod: there are examples in the docs [11:01] <durandal_1707> highgod: read documentation here: https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(scre… [11:01] <relaxed> look at the wiki in the topic [11:02] <highgod> OK thanks [11:02] <Magicking> robertzaccour: Last time I had to do that, I think I used a monitor or a mixer [11:15] <Bor0> how would I go implementing buffering state in ffplay? I have something like buffering=1; av_read_frame(...); buffering=0; but when I display the variable it always shows buffering = 0 even when it's buffering [11:19] <xlinkz0> is it possible to use the concat demuxer without a file? [11:19] <xlinkz0> like with a string? [11:19] <durandal_1707> is it mentioned in documentation? [11:21] <xlinkz0> no [11:22] <xlinkz0> durandal_1707: it does say something about the shell substitution thing but i'm launching it from another program not from command line [11:23] <xlinkz0> with fork and execvp [11:25] <durandal_1707> xlinkz0: it it is not mentioned that its not possible, if you want such feature open feature request on our bug tracker [11:40] <xlinkz0> durandal_1707: if i want to concat the last minute of video 1.mp4 and the first minute of video 2.mp4 i first need to make the cuts and then the concat right? [11:45] <relaxed> xlinkz0: you may want to look into MP4Box for this. [11:45] <xlinkz0> relaxed: is it any faster than the method i said or more convenient? [11:47] <relaxed> I going with both [11:48] <xlinkz0> thanks [11:49] <relaxed> I'm* [11:50] <sybariten> oh hai [11:51] <sybariten> i'm on FFmpeg version 0.6.5, running 'ffmpeg -i in.mov -vf "transpose=1" out.mov' in order to rotate a vid, and get unrecognized option vf .... any other options i could use to rotate a vid? [11:52] <durandal_1707> you use extremly old ffmpeg version [11:52] <sybariten> yeah, i know [11:52] <sybariten> but didnt the rotate come about in 2007, btw? [11:53] <durandal_1707> which appears to not have any filtering ability, prior to libavfilter [11:53] <sybariten> ah ok [11:53] <sybariten> dang [11:53] <durandal_1707> perhaps you need -filter ... [11:53] <durandal_1707> but that is soooooooo oooold version that you should not really use it [11:54] <durandal_1707> actually transpose filter was added in 0.8 [11:54] <sybariten> no, no thats undersood [11:56] <durandal_1707> and that was in 2010 [11:58] <durandal_1707> while you use ffmpeg somewhere from 2009 [12:00] <sybariten> yes, yes, yes, i know... now, the question was this [12:00] <sybariten> any other options i could use to rotate a vid? [12:02] <durandal_1707> install latest ffmpeg release [12:17] <kvz> Latest ffmpeg build throws "MS Video-1 warning: stream_ptr out of bounds" on an .avi: http://pastebin.com/0hVG646P. Does anybody know what this means? [12:18] <kvz> The same warning is thrown using an old sastes ffmpeg build on ubuntu (vs 1.2 on osx) and when trying to target different formats (e.g. theora). So it seems to be an issues with the input file. Tried googling but only getting source-code hits and my c isn't what it used to be ;) [12:19] <durandal_1707> so you just get warning, or nothing useful at output? [12:21] <kvz> durandal_1707: just that warning, every 10 seconds or so as ffmpeg shows frame / fps / etc information [12:22] <kvz> durandal_1707: trying to understand (the severity of) it [12:22] <kvz> durandal_1707: the same files also throws "max_analyze_duration 5000000 reached at 5016000 microseconds" as can be seen in the pastebin. Maybe it's related. [12:26] <durandal_1707> that last whan is irrelevant spam [12:27] <kvz> ok good to know [12:27] <durandal_1707> if there is nothing wrong in decoded output you should not worry about such warnings [12:27] <durandal_1707> but if error happens it will interupt decoding and that would be more visible in output [12:39] <kvz> durandal_1707: thank you [12:45] <xlinkz0> can i inspect the keyframe interval of a movie? [12:48] <xlinkz0> the seek documentation warns against -vcodec copy but if i use -c copy and have no audio streams i should not encounter problems right? [12:53] <xlinkz0> why do i get a 13 second video with this command? ffmpeg -y -ss 00:00:03 -i 1.mp4 -t 10 -c copy out.mp4 [12:55] <durandal_1707> because you can not cut sample accurately with -c copy [12:57] <xlinkz0> thanks [12:58] <xlinkz0> durandal_1707: same with -vcodec copy right? [13:01] <durandal_1707> video is different [13:01] <durandal_1707> audio have multiple samples in frames [13:03] <xlinkz0> all my containers have only video streams [13:03] <xlinkz0> i don't care about audio, i just want fast splitting [13:03] <relaxed> did you try MP4Box? [13:04] <xlinkz0> my boss told me to first try it with ffmpeg and if it's too slow try something else... [13:05] <xlinkz0> durandal_1707: but with -vcodec copy i still get 13 second video :( [13:06] <retard> of course you do [13:07] <xlinkz0> so -ss and -t will not work together with copy? [13:07] <retard> they work just fine [13:07] <retard> they won't break your video stream [13:07] <xlinkz0> but they do.. [13:07] <retard> which is the correct functioning [13:07] <durandal_1707> perhaps your boss should find another job? [13:07] <relaxed> xlinkz0: move -ss after the input. [13:09] <relaxed> meet the new boss, same as the old boss [13:09] <xlinkz0> i get http://superuser.com/questions/140899/ffmpeg-splitting-mp4-with-same-quality [13:09] <xlinkz0> wrong paste sry [13:09] <xlinkz0> Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) [13:09] <xlinkz0> on ffmpeg -y -i 1.mp4 -ss 3 -t 10 -vcodec copy out.mp4 [13:10] <relaxed> use MP4Box already and stop trying to shoehorn ffmpeg into everything. [13:11] <xlinkz0> ok.. [13:11] <retard> what are you trying to achieve [13:11] <xlinkz0> cut from second 3 to 13 [13:11] <retard> do you want to get a video file that is the exact length with exact start point you define [13:11] <retard> is there a keyframe at second 3 [13:11] <retard> because otherwise you will need to reencode [13:12] <durandal_1707> ahh, yes, that is video problem with that kind of cutting [13:12] <xlinkz0> i'm not sure if there is a keyframe there [13:12] <durandal_1707> i completly forgot that you are not dealing with intra only video [13:13] <retard> http://pastebin.com/1u96HKuz this is what i use to clip at keyframes without desynching audio in certain players [13:13] <xlinkz0> i want to get a video file that starts at second 3 and is of length 10 seconds [13:13] <durandal_1707> showinfo will dump bunch of info... [13:14] <durandal_1707> without reencoding exact cut is not always possible [13:14] <retard> rarely really [13:14] <xlinkz0> so.. i should use mp4box :D [13:15] <durandal_1707> well you could cut with mp4box, causing first frame not to be keyframe but that would give you bad output for first seconds [13:15] <retard> i don't know, i have never used mp4box [13:15] <JEEB> afaik everything cuts at RAPs [13:15] <JEEB> Random Access Points [13:16] <JEEB> you can't cut on random frames [13:16] <JEEB> period [13:16] <JEEB> if there is no RAP on that point of time, the closest is selected [13:16] <JEEB> thus, you should always see if there's a RAP picture around *before* trying to cut [13:17] <retard> what the script i pastet above does (very very slowly) is find the keyframes and cut from close to their exact timestamps in order to minimize audio/video desynch, it is not a solution for precise cuts [13:17] <retard> but it works for me [13:17] <xlinkz0> i have no audio [13:17] <JEEB> you still can't cut closer if there are no RAP frames [13:17] <JEEB> s/frames/pictures/ [13:18] <retard> you can look at how i extract a list of the frames in the video [13:18] <xlinkz0> i don't know bash script and never used ffprobe :( [13:18] <JEEB> thus, as I have noted, first of all go look at the picture types :P [13:18] <retard> since what i do there is look for keyframes [13:18] <xlinkz0> JEEB: do you know how to use the library to seek? [13:19] <xlinkz0> i think i'd do it faster with the api [13:19] <xlinkz0> but i've had problems in the past with seeking.. [14:26] <xlinkz0> relaxed: so if my camera is set to have a GOP length of 100 and fps is 25 it means that i can only seek to pts's divizible by 4 right? [14:38] <abhilashtt01> Hello, I have installed ffmpeg in our server [14:39] <abhilashtt01> it is taking a long time to convert small videos less than 5MB size [14:39] <abhilashtt01> any idea why it is taking long time [14:44] <xlinkz0> maybe you don't have a good encoding library [14:44] <xlinkz0> or bad proccesor [15:28] <DX099> hello [15:28] <DX099> I'm lost between ffmpeg & avconv [15:28] <DX099> can someone explain ? [15:30] <JEEB> ffmpeg is the binary name for the main binary in the ffmpeg project, avconv is the name for the main binary in the libav project [15:30] <JEEB> libav did have the ffmpeg binary as well up until version 0.8 or so, and then it get removed because no-one was updating it (it was a separate piece of code in that project by then, all of the updates and rewrites went to avconv) [15:31] <JEEB> ffmpeg project has merged most if not all of the changes that went into avconv into its ffmpeg binary [15:31] <JEEB> I hope this explains it a bit [15:32] <JEEB> and this is the reason why distributions that have an old version of libav that still had the ffmpeg binary as well tend to have that weird message there >_> (because _within_that_project_ it was deprecated) [15:32] <DX099> alright. So avconv is a sort of testing pool for ffmpeg new features ? [15:32] <JEEB> not really, it's just the name of the app that a fork of the original ffmpeg project uses [15:33] <DX099> alright [15:33] <DX099> so... which one should I compile ? [15:34] <JEEB> ffmpeg merges so much of libav that it shouldn't really lack anything that libav has. It also has some other stuff and changes. I generally don't want to take too much part in pointing you towards either project, you should see what you want to grab, and grab that. [15:36] <DX099> hmm... Ok. I'll think I'll just stick with ffmpeg [15:44] <DX099> thanks [16:16] <abhilashtt01> configure: error: ffmpeg headers not found. Make sure ffmpeg is compiled as shared libraries using the --enable-shared option [16:16] <abhilashtt01> Guys, I am trying to install ffmpeg-php [16:17] <abhilashtt01> getting above error [16:17] <abhilashtt01> ffmpeg is already compiled as shared [16:17] <abhilashtt01> any ideas? [16:18] <JEEB> ffmpeg-php is not in any way related to ffmpeg, and as far as I know it's been left to bit rot for several years now [16:25] <abhilashtt01> ok [16:28] <DX099> hello again. I compiled fdk-aac from git with appropriate ./configure --disable-shared, but now ffmpeg is telling it doesn't see "libfdk_aac" ? [16:29] <JEEB> look at config.log [16:31] <DX099> I don't understand what it says [16:32] <t4nk188> hi [16:32] <JEEB> DX099, then pastebin the whole config.log and link it here? [16:34] <t4nk188> i'm making a gui for ffmpeg in java, i can choose a file in a listbox and play it with ffmpeg, it's streaming it with rtmp to justin tv, the problem is that i need to launch alway a new ffmpeg for a new video file and it closes the rtmp connection, is it possible to play many video files with ffmep without closing the connection and without the need of using another program [16:34] <JEEB> use the API? nîn [16:34] <t4nk188> are you talking to me ? [16:34] <DX099> JEEB, http://paste.ubuntu.com/5644793/ [16:35] <JEEB> t4nk188, no -- to the gentleman drinking coffee sitting right next to you [16:35] <t4nk188> i am the one drinking a kenya black coffee [16:35] <JEEB> DX099, first of all you don't need both faac and fdk :P [16:36] <JEEB> DX099, also it seems like it just doesn't find the library [16:36] <DX099> ok [16:36] <JEEB> where did you install the library? [16:36] <DX099> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [16:36] <DX099> I just followed this [16:37] <JEEB> I have no idea what exactly that leads to, but basically that error means that there was no such library under that name found in the default library search paths (/usr/bin/ld: error: cannot find -lfdk-aac), and thus the function that was being tested against that should be in the library was not found (aacEncOpen wasn't found) [16:38] <JEEB> I would guess by default it installs to /usr/local ? No idea though [16:38] <t4nk188> any idea for my problem ? [16:38] <JEEB> check your /lib and /usr/local/lib for libfdk-aac.a or libfdk-aac.so [16:39] <JEEB> t4nk188, using the API would still most probably be the easiest way, or use udp or http input that you keep filling with blackness or something until you get the new input pushed there :P [16:39] <JEEB> still plenty of work for you to do [16:40] <t4nk188> the api ? i did not know about any api.... i can use it with java ? What do you mean by blackness ? can i use ffmpeg for the udp input and another one ffmpeg process for the rtmp connection and it will work ? is that what you mean ? [16:41] <JEEB> use libavformat/-codec and friends via JNI? [16:41] <JEEB> is the API part [16:42] <JEEB> the second thing is to just keep another application giving you stuff via http or udp locally, and that input never ends [16:42] <JEEB> but of course you'd have to make the application that does that :D [16:42] <t4nk188> so i can't use 2 ffmpeg process to achieve that ? [16:43] <t4nk188> about that -codec and jni... i really don't know that [16:43] <JEEB> nope, ffmpeg can't serve http or udp in that way as far as I know? it can just push stuff out with http post, not actually serve http requests [16:44] <t4nk188> ho... ok :/ it seems more tricky than i thought [16:44] <JEEB> in any case you'd end up writing some kind of application of your own for at least some part of it [16:44] <t4nk188> it's so weird that ffmpeg can't just open another file without disconnecting [16:44] <t4nk188> yes the gui is working to stream with rtmp [16:45] <t4nk188> but it's not fun when the connection is diconnected everytime it opens a new video file [16:45] <t4nk188> anyway may be i'll be able to do what you're saying but a this time i am not sure i understand it [16:46] <JEEB> you could make a playlist file or whatever and have that be read as input, but that means that you know all your input files before hand. Also I don't know if switching input will make ffmpeg reset the connection [16:46] <t4nk188> thanks for your help [16:46] <JEEB> anyways, the hard way would be to make your app use the libraries that ffmpeg uses in the way that you want, and the second one would just make you "cheat" around the limitation in ffmpeg [16:47] <JEEB> which would be you making some kind of UDP or HTTP server that keeps feeding raw video to ffmpeg [16:47] <JEEB> and which wouldn't stop the connection [16:47] <JEEB> and then you just make ffmpeg read from that [16:47] <t4nk188> well at this time my program is just making a command line to launch ffmpeg with the correct options, just don't know how to make it works with a playlist without launching a new command line [16:47] <JEEB> but yeah, plenty of work in either case :P [16:47] <t4nk188> but everything else works fine [16:48] <t4nk188> with my gui i can choose many options and it's fun i can choose a directshow device as a source but i wanted to make it works without any other device and a playlist [16:50] <DX099> JEEB, http://paste.ubuntu.com/5644844/ [16:51] <JEEB> time for you to check where the linker checks by default :D [16:51] <JEEB> and if it doesn't check in /usr/loca/lib [16:51] <JEEB> *local [16:51] <JEEB> then you have to add it to the linker search path [16:51] <JEEB> it seems like the headers are found just fine, though [16:51] <JEEB> also the fact that you have both shared and static libs could be a problem [16:51] <DX099> how do I check that ? [16:52] <JEEB> http://stackoverflow.com/questions/9922949/how-to-print-the-ldlinker-search… [16:52] <JEEB> first check with the answer here if /usr/local/lib is getting checked for libraries [16:52] <t4nk188> it seems even more tricky with linux... [16:54] <t4nk188> may be i'll share my gui program if people are interested... [16:57] <DX099> /usr/local/lib is checked [16:57] <t4nk188> when you say : "you could make a playlist file or whatever and have that be read as input, but that means that you know all your input files before hand. Also I don't know if switching input will make ffmpeg reset the connection [16:58] <t4nk188> how do you switch inputs ????? [16:58] <JEEB> uhh, you give ffmpeg the playlist and switching means going to the next one in that :P [16:59] <t4nk188> ???? ffmpeg support playlist ??? i was looking for that but it seems it does not support it [16:59] <JEEB> well, I've seen quite a few times here someone saying that you could set some text file as input(s) :P [17:00] <t4nk188> o_O how ???? [17:01] <t4nk188> i've been looking for that for 2 days ... [17:01] <t4nk188> i found only someone talking about pipes but it doesn't even seem to work [17:02] <JEEB> DX099, then run ldconfig with admin rights and try again [17:02] <DX099> JEEB, that I have already done [17:03] <JEEB> then re-run the configure script and see if it still fails, and if it does -- then post another pastebin of config.log [17:05] <DX099> ah.. just spotted that locate lied [17:05] <DX099> http://paste.ubuntu.com/5644891/ [17:05] <DX099> there's no .so file... [17:06] <JEEB> that's fine [17:06] <JEEB> that means that there are only static libs [17:07] <DX099> http://paste.ubuntu.com/5644895/ [17:25] <DX099> JEEB, --extra-ldflags="-L/usr/local/lib" adding this to ./configure makes it work. However, I don't get why even though /usr/local/lib is the /etc/ld.so.conf, and that I made an ldconfig I still can't just run configure... [17:25] <DX099> I also tried LD_LIBRARY_PATH and stuff with no success [17:25] <DX099> beats me... [17:43] <DX099> then can I require the Ubuntu compilation guide to be completed with "--extra-ldflags="-L/usr/local/lib" option for configure ? [18:11] <JEEBsv> DX099: weird. In general the linker *should* check /usr/local/lib just like the compiler checked for /usr/local/include and found your headers :D [18:12] <JEEBsv> and in the few debian-based systems I've seen it had worked :3 [18:12] <JEEBsv> LD_LIBRARY_PATH is for runtime loading of shared libraries [18:16] <DX099> JEEBsv, ok [18:17] <DX099> but for some reason it didn't make it... [18:17] <JEEBsv> yeah, it's weird [00:00] --- Thu May 9 2013

1 0

ffmpeg-devel.log.20130507
by burek 08 May '13

08 May '13

[01:08] <ubitux> http://pastie.org/7810717 [01:08] <ubitux> ./record.sh bla.webm [01:08] <ubitux> *clic* [01:34] <cone-588> ffmpeg.git 03Matt Wolenetz 07master:004779cc59ae: avcodec/utils: Fix leak by dereferencing audio frame when side data causes whole frame skip [03:07] <llogan> is there a chat log for the GSoC feedback? [03:32] <Compn> llogan : ffmpeg had a private feedback [03:33] <Compn> did you want that or the public channel log ? [03:33] <Compn> (i dont have either) [03:47] <llogan> the private of course. like the log from last year. [03:47] <llogan> Compn: ^ [03:48] <llogan> i could not attend, unfortunately. [03:48] <Compn> i forgot who handled it [03:48] <Compn> saste maybe [03:56] <llogan> why is the tag shown as n0.8 with "git describe"? [04:10] <Compn> you are questioning the 'n' specifically then ? [05:35] <cone-270> ffmpeg.git 03Reimar Döffinger 07master:762d4335aec2: matroska: set "done" only during resync fail. [05:35] <cone-270> ffmpeg.git 03Reimar Döffinger 07master:399f6ef971eb: Add FATE test for matroska error recovery. [06:39] <Zeranoe> I see vid.stab was added, and I'm getting a few people asking for it to be included. I tired to compile it from source, and everything worked, but no .pc pkgconfig file was generated. It keeps failing at configure with "check_pkg_config vidstab vid.stab/libvidstab.h vsMotionDetectInit" which is probably because I didn't include the path, plus no .pc file. So has anyone managed to compile vid.stab into a FFmpeg build? [09:31] <durandal_1707> michaelni: so you did not review my gbrap patch? :( [09:36] <highgod> Hi, I want to ask a question, where can I find any code to using rtsp for reference? Thanks [10:02] <michaelni> durandal_1707, i wanted but it breaks fate [10:03] <durandal_1707> it just missing new entries, should not break old ones [10:05] <durandal_1707> but yes, i updated and running fate for 2nd time [10:06] <michaelni> ok, thanks [10:57] <cone-270> ffmpeg.git 03Janne Grunau 07master:74652af821d7: pixdesc: add function to return pixel format with swapped endiannes [10:57] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:60fd88059a62: Merge commit '74652af821d7e526a6541876a637fbf81261fa8e' [11:10] <cone-270> ffmpeg.git 03Janne Grunau 07master:c1eb3e7fecdc: swscale: add support for endianness only conversion [11:10] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:2060d944090f: Merge commit 'c1eb3e7fecdc270e03a700d61ef941600a6af491' [11:20] <cone-270> ffmpeg.git 03Janne Grunau 07master:9cfa21c8c2b8: swscale: support endianness conversion for AV_PIX_FMT_XYZ12 [11:20] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:dce12f694864: Merge commit '9cfa21c8c2b81146aab574f55c93f27e5c2d435c' [11:32] <cone-270> ffmpeg.git 03Janne Grunau 07master:5f87c277bd5c: vf_scale: add endianness conversion pixel format in query_formats [11:32] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:fe5b08959663: Merge commit '5f87c277bd5caa09cc4f9061d4ccdd58dc121110' [11:41] <cone-270> ffmpeg.git 03Janne Grunau 07master:851465f9d9ed: fate-jpeg2000-dcinema: force output to xyz12le [11:41] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:9276467bcec5: Merge remote-tracking branch 'qatar/master' [11:48] <michaelni> durandal_1707, fate-filter-pixfmts-scale fails for me [11:48] <michaelni> -gbrap fb463f07681f099871db2309dd438123 [11:48] <michaelni> +gbrap 7d4452269fab4e29b4a9458be112d03c [11:50] <durandal_1707> only that one? [11:52] <michaelni> apparently but let me doubel check [11:55] <michaelni> yes fate passes if i chnage this one [12:05] <durandal_1707> michaelni: perhaps the A thing you wrote in yuv2gbrp_full_X_c is wrong? (overflows) [12:20] <durandal_1707> but that can't be as same is in packed rgb [12:24] <durandal_1707> lol, i see problem [12:24] <durandal_1707> alpha plane somehow get wrong [13:10] <durandal_1707> michaelni: the alpha plane width get divided by two for some reason [13:20] <durandal_1707> the commit 7b41c24c5fa221b55a9302efaf1a9 mentions planar RGB but set only >8 bit planar RGB [13:21] <durandal_1707> same for 3c2ecfcc24761a [13:25] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:2efbbc464458: avutil/av_pix_fmt_swap_endianness: simplify and fix code [13:32] <michaelni> durandal_1707, does adding 8bit there fix anything ? [13:36] <durandal_1707> but why would it be needed, only alpha gets fcked up [13:36] <durandal_1707> and only when scaling [13:37] <durandal_1707> so its somewhere is scaling code [13:42] <durandal_1707> michaelni: it does not help [13:45] <durandal_1707> michaelni: http://oi41.tinypic.com/juh30n.jpg [13:47] <cone-270> ffmpeg.git 03Gunther Piez 07master:acc4a7e5b350: Ticket #2540 Enable threading when using -c:v libx264rgb [13:51] <michaelni> durandal_1707, dunno what is causing this assuminh that represents the alpha plane, have you tried the various flags ? [13:52] <durandal_1707> i'm trying other scalers, gauss/bitexact/... does not help [13:57] <michaelni> i was mainly thinking of the flags related to half resolution [13:59] <durandal_1707> ffmpeg -v debug -i /tmp/1.nut -vf format=gbrap,scale=800x600:flags=+gauss-bilinear+full_chroma_inp+full_chroma_int,format=gbrap,extractplanes=a -vcodec png /tmp/x.nut [13:59] <durandal_1707> same [14:25] <Bor0> I'm trying to implement a "buffering" state to my player. so I added a new integer in the VideoState struct, and here's what I do: is->buffering = 1; av_read_frame(...); is->buffering = 0; but this always shows that is->buffering is 0, even when it's buffering. what am I doing wrong? [14:27] <av500> its not buffering, its blocking [14:27] <av500> er [14:29] <av500> I mean, if av_read_frame(...) is fast enough, another thread wont see buffering == 1 [14:32] <cone-270> ffmpeg.git 03Carl Eugen Hoyos 07master:f381dc1d6186: Support decoding of Lagarith frame type FRAME_SOLID_COLOR. [14:32] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:144baf427a6b: Merge remote-tracking branch 'cehoyos/master' [14:34] <Bor0> what's a proper way to implement this? [14:34] <Bor0> I'm reading from http, so buffering can take some time [14:39] <av500> first make sure you are reading your flag correctly [14:40] <Bor0> I'm constantly printing it [14:41] <av500> yes, from a different thread [14:41] <av500> so the compiler might assume its constant [14:41] <Bor0> hmm, why does that matter? my structure is global [14:41] <durandal_1707> michaelni: actually bugs happen when doing gbrap->subsampled yuv conversion [14:44] <durandal_1707> eg, chroma subsampling get leaked into alpha [14:48] <durandal_1707> actually it happens with 444p too [15:27] <cone-270> ffmpeg.git 03Clément BSsch 07master:c3dcd3a255a0: sws: factorize rgb 9+ planar funcs declarations. [15:32] <durandal_1707> michaelni: i'm out of ideas where is error [15:39] <cone-270> ffmpeg.git 03Clément BSsch 07master:5043d8d213f1: lavfi/alphaextract: copy width and not linesize. [15:44] <cone-270> ffmpeg.git 03Clément BSsch 07master:d3f7b98f87ff: lavfi/overlay: reindent fix. [16:16] <ubitux> michaelni: swscale.c:const int chrSrcSliceH = -((-srcSliceH) >> c->chrSrcVSubSample); [16:16] <ubitux> isn't this supposed to be HSub? [16:16] <ubitux> const int chrSrcSliceY = srcSliceY >> c->chrSrcVSubSample; [16:16] <ubitux> const int chrSrcSliceH = -((-srcSliceH) >> c->chrSrcVSubSample); [16:18] <michaelni> SliceH = slice Height, VSubSample = Vertical sub sample, HSub would be Horizonbtal [16:20] <ubitux> aah ok i see [16:20] <ubitux> thx [16:23] <durandal_1707> michaelni: its looks like alpha plane is not read when scaling, should i add readAlpPlane? [16:28] <michaelni> durandal_1707, iam not sure what you want to add where, you seem to assume i know where the issue is and what you plan to do ... [16:29] <durandal_1707> michaelni: see hyscale in swscale.c [16:29] <durandal_1707> alpToYV12 is never set for gbrap [16:35] <michaelni> maybe i miss something but why should it be set ? the data is already in 8bit planar format [16:36] <durandal_1707> perhaps because it wants it in 16bit [16:48] <durandal_1707> i added some code, and also shift alpha by 8 and get something sane, but with less detail [16:52] <durandal_1707> actually shifting by 6 give correct output [16:53] <michaelni> yes its a bit annoying, i had not realized the rgb 16bit issue [17:03] <cone-270> ffmpeg.git 03Michael Niedermayer 07master:d23b8462b5a4: gifdec: reset previous Graphic Control Extension disposal type [17:30] <durandal_1707> fflogger is duplicating stuf [17:46] <durandal_1707> hmm, neigther id3info nor id3v2 list id3 tags created by ffmpeg [17:53] <durandal_1707> michaelni: should i pass table when calling function, or just call it with NULL arg? [17:54] <michaelni> doesnt matter currently [17:54] <durandal_1707> yes, but what you prefer? [18:01] <michaelni> null [18:06] <xlinkz0> how do i transform the return value of av_seek_frame into a string to know why it failed? [18:06] <xlinkz0> or more simply, how do i find out why it failed? [18:07] <ubitux> michaelni: any name suggestion? [18:09] <ubitux> ROUNDED_INF_RSHIFT is a bit long [18:09] <durandal_1707> xlinkz0: some return values are defined in libavutil/error.h [18:11] <xlinkz0> thanks [18:17] <michaelni> ubitux, no good ones atm [18:26] <cehoyos> __gb__: Ping [18:34] <durandal_1707> cehoyos: why you do not want to send patch from #2527 to ml? [18:35] <cehoyos> (Why do you think that my answer changed in the last 24 hours?) I don't know if there is a bug or not. If you know that there is a bug, please (review it and) push my patch. [19:18] <ubitux> cbsrobot: 'just submitted the true/sample peak thing [19:19] <ubitux> no visual yet, but av_log'ed and metadata injected [19:19] <ubitux> should be good enough [20:06] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:89d998f1c1ed: shorten: allocate space for padding [20:24] <cbsrobot> ubitux: I tried it - but in what unit is it ? [20:24] <ubitux> double :p [20:24] <ubitux> [0;1.0] [20:25] <durandal_1707> 0 too? [20:25] <cbsrobot> would be nice to have it in dBTP [20:36] <ubitux> mmh true [20:37] <cbsrobot> and if you plan to print the values to the summary - which would be very usedful - I think there is also some channel weighting going for multichannel files [20:41] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:f96e0eb2387f: dxa: check vectors of 4x4 motion blocks [20:41] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:ead590c25619: dxa: check vectors of 2x2 motion blocks [20:46] <durandal_1707> Compn: have any sbc samples? [21:11] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:ad002e1a13a8: avcodec/cdgraphics: check buffer size before use [21:43] <Compn> durandal_1707 : are they in trac ? [21:43] <Compn> i know aac sbr [21:43] <Compn> what sbc ? [21:55] <nevcairiel> the only sbc i know is a subband codec for bluetooth audio devices, not sure you can even put that into a file [21:55] <gnafu> Anything is possible ;D. [21:56] <Compn> http://wiki.multimedia.cx/index.php?title=Bluetooth_SBC [21:56] <Compn> yeah i think theres a rfc for sbc [21:57] <Compn> durandal_1707 : so probably no samples [21:58] <durandal_1707> i just created some files ... [21:59] <durandal_1707> i just got interested because it got mentioned on ha [21:59] <Compn> if you put them in some http i can put them on samples repo [22:00] <durandal_1707> no need, i just wanted to know is it popular enough to be included in lavc [22:04] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:9dd04f6d8cdd: sanm: Check dimensions before use [22:12] <durandal11707> michaelni: what interger overflow 9dd04f6d8cdd fixes? [22:32] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:a527e692592b: avformat/vqf: check number of channels before use. [22:41] <Sergey> Hello. About the bug #2425 ([regression] Smacker audio played with clicks). It was introduced by lib av, should I report it to their bugtracker instead? [22:43] <durandal_1707> no, join #libav-devel, and explain them bug [23:10] <Sergey> durandal_1707: Ok, thanks, trying... [23:53] <cone-605> ffmpeg.git 03Nicolas George 07master:449704b47500: ffmpeg: move a local variable definition later. [23:53] <cone-605> ffmpeg.git 03Nicolas George 07master:78d16b6a5109: lavfi: fix filter format negotiation loop. [23:53] <cone-605> ffmpeg.git 03Nicolas George 07master:4582e4c086bc: lavfi: add comments to explain the negotiation loop. [23:53] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:bc0cbaca62b6: Merge remote-tracking branch 'cigaes/master' [00:00] --- Wed May 8 2013

1 0

ffmpeg.log.20130507
by burek 08 May '13

08 May '13

[04:09] <Giri> Hello [04:09] <Giri> How are you [04:10] <Giri> I want to cross compile the ffmpeg source code arm-linux [04:10] <Giri> which is on imx35 and arm-none-linux-gnueabi complier [04:11] <Giri> please tel me the ./configure command [04:11] <Giri> i could able to compile it for the host [04:11] <Giri> but i am want to compile it and use it for the targe [04:11] <Giri> please help me [06:09] <hunterp> Hey: What does this mean: ld: error: cannot find -lvorbisenc [08:24] <vdm> hi [08:24] <vdm> I have video frame raw data in the PIX_FMT_YUYV422 format and need to save it to the JPEG file using LIBJPEG library. With swscale library i convert the PIX_FMT_YUYV422 frame to the PIX_FMT_RGB24 to save it as JPEG. I have analyzed YUYV422 (http://i.imgur.com/Oa0whCn.jpg) and RGB24 (http://i.imgur.com/kb1UHTX.jpg) buffers using YUVTools software - it shows both buffers correctly. The problem is that saved JPEG image is distorted (http://i.imgur. [08:24] <vdm> com/bT5NR7B.jpg). Here is the function which save PIX_FMT_RGB24 format buffer to the JPEG file, using LIBJPEG library: (http://pastebin.com/K3NdJE8X) I saw that the same distorted effect happens if i choose YUV444 format for the RGB24 buffer in the YUVTools (http://i.imgur.com/ooMgrtH.jpg) Any ideas, why the buffer saved with LIBJPEG is distorted? [08:29] <IanWizard-Cloud> I'm trying to compress an mp3 (by stripping to mono, dropping sample rate, etc), but each time, it comes out bigger than it went in. [08:30] <IanWizard-Cloud> I compressed one with audacity, and got it down to ~300Kb, but with ffmpeg (avconv), it ends up larger. Even when using the compressed version, it comes out as 1.7Mb [08:30] <IanWizard-Cloud> My cmdline is some variation of: avconv -i input.mp3 -ac 1 -ab 16000 output.mp3 [08:37] <Mavrik> avconv isn't really ffmpeg now is it? :) [08:37] <Mavrik> and I very much doubt the mp3 encoder will want to encode your audio to 16kb/s [08:37] <IanWizard-Cloud> Mavrik: that was my (possibly flawed) understanding. [08:38] <IanWizard-Cloud> I wanted to use wav, but that ended up with a flat 8mb file. [08:39] <IanWizard-Cloud> admittedly, I don't have the experience here that I should, and to a large degree, am winging it. [08:43] <vdm> Mavrik, maybe you know, is it possible with ffmpeg to save RGB24 buffer or file to the JPEG? [08:45] <Mavrik> vdm, should be [08:45] <Mavrik> IanWizard-Cloud, well. [08:45] <Mavrik> let's start at the beginning [08:50] <IanWizard-Cloud> Mavrik: http://paste2.org/n7UFt3WJ [08:50] <IanWizard-Cloud> The notice there is what led me believe that avconv was replacing ffmpeg [08:52] <Mavrik> IanWizard-Cloud, yeah, that's outdated and misleading. [08:52] <Mavrik> IanWizard-Cloud, do note that maybe some things we tell you here won't work on avconv [08:52] <Mavrik> and we're not avconv support. [08:53] <Mavrik> IanWizard-Cloud, um, you're trying to recompress that file to even less bitrate? [08:53] <IanWizard-Cloud> Mavrik: The file I used there is one that I've already mucked about with in Audacity [08:54] <IanWizard-Cloud> I should have used a virgin file probably. [08:54] <Mavrik> IanWizard-Cloud, well, that file is such a low qualith [08:54] <Mavrik> that I'm surprised it still resembles anything at all [08:54] <Mavrik> going lower than that with mp3.... [08:54] <Mavrik> not really recommended :) [08:58] <IanWizard-Cloud> Mavrik: here's some progress, and a real file. Just used the first song tab complete brought me to :P http://paste2.org/eHXbxh2e [08:59] <IanWizard-Cloud> It does get it slightly smaller here, which is some progress, but I must be doing something wrong. Is this the best I can expect? [09:00] <IanWizard-Cloud> Audio is, admittedly, not my field, but with Audacity I got that amazing grace mp3 down to 300k, with out too much degradation. [09:00] <Mavrik> huh? [09:00] <Mavrik> you must be half deaf or have broken speakers if you don't hear degradation at 16kHz sampling rate O.o [09:01] <Mavrik> most decent quality sound uses 44100Hz - 48000Hz and is compressed to 192kB/s+ [09:01] <Mavrik> IanWizard-Cloud, what exactly is your goal here? [09:01] <IanWizard-Cloud> no, sorry, the first track, amazing_grace.mp3, sounds worse than the original, but it's still reasonable [09:01] <Mavrik> why such needed compression? have you thought of better formats? [09:02] <Mavrik> woult HE-AACv2 fit better to your needs? [09:06] <IanWizard-Cloud> Mavrik: I'm trying to compress it on the fly, depending on a users connection speed. And trying to get decent browser compatability, so sadly, it's a bit of a mess. [09:07] <IanWizard-Cloud> Compounded by the fact that I don't know nearly enough about the pros and cons of the various codecs [09:07] <Mavrik> IanWizard-Cloud, yeah, educating yourself will help :) [09:07] <Mavrik> IanWizard-Cloud, for low-bitrate streaming the HE-AACv2 (also called HE-AAC+) would be the best [09:07] <Mavrik> since you get practically free stereo [09:07] <IanWizard-Cloud> Mavrik: heh, trying to, but also trying to get it done quickly to keep boss happy :( [09:08] <Mavrik> and it sounds noticably better and very low bitrates [09:08] <IanWizard-Cloud> I remember when i did this for nobble reasons, and not a paycheck :( [09:08] <Mavrik> but please, keep samplerate high [09:08] <Mavrik> or you'll destroy your sound [09:08] <Mavrik> so keep it at least 22050 [09:08] <Mavrik> try something in terms of 22050, 32kb/s, mono with SBR on [09:08] <Mavrik> encoded to AAC high profile [09:09] <Mavrik> and go no lower [09:09] <IanWizard-Cloud> ok. [09:09] <Mavrik> it's really pointless to compress to 200kb audio [09:09] <IanWizard-Cloud> I know -ar 22050 [09:09] <IanWizard-Cloud> the rest is lost :P [09:09] <Mavrik> people have fast connections, don't rape their ears [09:09] <IanWizard-Cloud> but RTFM I will [09:09] <IanWizard-Cloud> :) [09:09] <Mavrik> your clients won't be happy ;) [09:09] <Mavrik> IanWizard-Cloud, libfdk_aac is the keyword [09:09] <Mavrik> and updating from old avconv to something new is a must ;) [09:10] <IanWizard-Cloud> IDK why it's outdated. I may just compile it myself. [09:10] <IanWizard-Cloud> Or find an oft-updated repo. [09:10] <Mavrik> IanWizard-Cloud, or grab a static pre-compiled build with libfdk built :) [09:13] <IanWizard-Cloud> Mavrik: downloading now. [09:13] <IanWizard-Cloud> Thanks for taking the time. [09:13] <IanWizard-Cloud> I'll let you know how I manage :) [10:52] <Zeeflo> ive been noticing that ffmpeg doesnt consume all the force of the cpu.. It consumes all threads in the cores, but only at.. around 7-80% per thread.. [10:52] <Zeeflo> is there an option to make ffmpeg utiliza all cpu power? [10:52] <Zeeflo> utilize* [11:21] <elkng> don't you think its strange ? [11:24] <elkng> no [12:02] <keyzs> little offtopic but, was planning to convert mov to avi, seen some converters super, format factory, media coder, whats the best way and tool to do this via win and mac? is there any video channel on freenode? [12:52] <renevolution> Hi there, i'm trying to connect to an icecast/shoutcast server from ffmpeg. As far as i understand i "just" need to set the correct http headers for the connection. Has anyone tried to do this? [13:20] <iMath> I use the following python code to split a FLV video file into a set of parts ,when finished ,only the first part video can be played ,the other parts are corrupted.I wonder why and Is there some correct ways to split video files http://stackoverflow.com/questions/16417598/use-python-to-split-a-video-fil… [15:13] <renevolution> Does anybody know if it is possible to stream to an icecast2 server using ffmpeg? [15:17] <stf> where can i add the libdir to ./configure? I have installed libfaac instead of system folder into a seperated one. [15:19] <Yulth> Hi everyone! I'm having troubles transcoding FLAC (even MP3) to 64kbps HE-AAC. Although the quality of the resultant HE-AAC file is high enough, some distortions in the basses are present, and I don't understand why this is happening.... [15:19] <Yulth> Does anyone have some experience? [15:21] <klaxa> stf: LDFLAGS="-L/path/to/lib/dir" [15:21] <klaxa> Yulth: what encoder are you using? [15:21] <Yulth> sorry..., I'm using libfdk_aac [15:21] <klaxa> if you are not using libfdk-aac, try it [15:21] <klaxa> mmh... no idea then :/ [15:22] <Yulth> exactly: -acodec libfdk_aac -profile:a aac_he [15:24] <Yulth> When transcoding from mp3 to he-aac, some distortions are supposed to will happen, but when transcoding from high-quality FLAC files..., conversion must be nearly "perfect" [15:24] <Yulth> and these distortions in the basses still happening... [15:25] <JEEB> uhh, HE-AAC is not magic [15:25] <stf> klaxa: thx seems to work :-) [15:25] <JEEB> there will be some distortions naturally, but you might want to try enabling afterburner first [15:26] <JEEB> basically -afterburner 1 [15:26] <JEEB> with fdk-aac [15:27] <Yulth> afterburner? [15:27] <JEEB> it's a setting in fdk-aac [15:27] <stf> klaxa: hm i was to uncaution [15:27] <klaxa> hm? [15:27] <JEEB> slower encoding, more quality is how the documentation notes it. Not that it should be *slow* . [15:28] <JEEB> Anyways, basically HE-AAC's whole point is to start eliminating stuff that is harder to compress, and that does indeed lead to audible differences, whether or not you perceive that as "distortions" as you listen to the audio is a whole separate discussion. [15:28] <Yulth> where Can I find more information about this option? In the ffmpeg manual? [15:29] <Bor0> I'm trying to implement a "buffering" state to my player. so I added a new integer in the VideoState struct, and here's what I do: is->buffering = 1; av_read_frame(...); is->buffering = 0; but this always shows that is->buffering is 0, even when it's buffering. what am I doing wrong? [15:29] <JEEB> Yulth, http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/libfdk-aacenc.c;h… [15:30] <Yulth> Should this option be specified during compilation time? [15:30] <Yulth> I'm on FreeBSD [15:31] <Yulth> options "afterburner" is new in version 1.2? [15:32] <JEEB> I think it's there as long as you have fdk-aac built in [15:32] <stf> have i to link 'LDFLAGS="-L/path/to/dir"' to the lib dir or is it enought to give the dir above this folders? [15:32] <JEEB> it should also be shown in fdk-aac's help [15:32] <JEEB> and no, it's not that new, it just needs fdk-aac to be built in [15:33] <JEEB> (and used, naturally) [15:33] <Yulth> ok, I'm going to investigate it. Thank you!! :) [16:10] <stf> how to use --extra-cflags right? [16:11] <JEEB> --extra-cflags="-I/wat/is/this/i/dont/even/" [16:11] <JEEB> for example [16:11] <stf> hm okay thx [16:15] <Yulth> JEEB: -afterburner 1 fixed the problem with the basses, but in general terms, the audio quality has been decreased [16:16] <JEEB> Yulth, that's HE-AAC. It's made for relative low bandwidth audio and it is in no way trying to be transparent [16:17] <Yulth> what pitty! [16:18] <Yulth> however, at the same bandwidth OPUS is amazingly brilliant... [16:20] <JEEB> could try the HE-AACv2 profile of course, I think you were using the normal he-aac profile [16:20] <Yulth> yes [16:20] <JEEB> but in general, HE-AAC was first officially publicized in '03, and HE-AACv2 was in '06 [16:20] <JEEB> you can compare that to Opus [16:20] <Yulth> but generally, at 64kbps HE-AAC works better than HE-AAC,v2 [16:22] <JEEB> well, in any case use whatever you need. I generally use LC-AAC and just push out more rate if I need to get closer to transparency, and need something supported by certain hardware decoders. [16:29] <Yulth> JEEB: How LC-AAC could be specified as profile in ffmpeg? [16:31] <JEEB> the usual way? Although you could just raise rate and not specify a profile and it should use LC :P [16:32] <JEEB> it should show which profile got used during encoding [16:42] <Yulth> ams ok, its true :) [17:32] <elkng> http://imagebin.org/index.php?mode=image&id=256816 why its ends on number 7 ? and why 7 is in red ? [17:34] <smj> Why is -f alsa -i pulse always 48000Hz even when PulseAudio is running at 44100Hz? [17:35] <smj> Can I fix that to save some CPU and audio quality? [17:35] <Mavrik> smj, alsa probably doing some conversion [17:35] <Mavrik> is audio speed good? [17:35] <smj> yes [17:36] <Mavrik> you can always try setting put samplerate (-ar before -i) or resample later (-ar after -i) [17:36] <smj> could that work? [18:41] <vdm> [swscaler @ 003638E0] Warning: data is not aligned! This can lead to a speedloss - i have this error using sws_scale method. How to fix it? [18:42] <tmatth> vdm: allocate your buffer with av_malloc? [18:42] <elkng> how to convert gif to video ? [18:42] <elkng> can ffmpeg do it ? [18:43] <ubitux> just like any other format [18:43] <vdm> tmatth, i'll try [18:46] <vdm> tmatth, thanks that was the solution. [18:46] <tmatth> vdm: np [18:51] <QT-BOSS> why does ffmpeg take so long to encode for mp4 adobe, x264 but for IPhone mp4, it encodes 1.5gb in 14 mints ? [18:51] <QT-BOSS> is there any way to make IPhone mp4 work on adobe flash ? so i dont spend lots of time on encoding ? [18:55] <QT-BOSS> ok [18:57] <QT-BOSS> here for IPhone, which encodes really fast http://pastie.org/7814074 [18:57] <QT-BOSS> how to make that work on adobe flash player too >? but keep the encoding speed ? [18:59] <durandal_1707> QT-BOSS: that is not full uncut output, without it i can't help, but just play lottery [19:01] <QT-BOSS> i don't have any problems with the ffmpeg, both cli command lines work, im trying to figure how to make the pastie i just gave work on adobe flash player [19:02] <QT-BOSS> ffmpeg version http://pastie.org/7814105 [19:17] <durandal_1707> QT-BOSS: that is still incomplete and thus useless output [19:18] <durandal_1707> QT-BOSS: have you read faq? [19:18] <QT-BOSS> which one sorry [19:18] <QT-BOSS> in the ffmpeg faq or the ffmpeg website ? [19:19] <durandal_1707> faq on website [19:20] <QT-BOSS> ok im reading @http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [19:20] <durandal_1707> that is not faq [19:22] <durandal_1707> but last paragraph may be relevant for you [20:34] <Jofironses> Hello, I was trying to execute this command: 'avconv -i video.mp4 -acodec copy -vcodec copy -map 0:0 -map 0:1 -map 0:2 -i English.ass -map 1:0 -scodec copy output.mkv' but it gives me this error ' Application provided invalid, non monotonically increasing dts to muxer in stream 0: 8500 >= 8500 [20:34] <Jofironses> av_interleaved_write_frame(): Invalid argument'. Does any one got a clue what the problem could be? [20:38] <Jofironses> I should add that if I execute without the (.ass) it works just fine, that is 'avconv -i video.mp4 -acodec copy -vcodec copy -map 0:0 -map 0:1 -map 0:2 output.mkv' It works just fine [21:48] <tonsofpcs> hi, how can I tell ffmpeg that an input stream is raw pcm 24bps 48kHz stereo? [21:51] <tonsofpcs> pcms24le is the audio format but how do I tell it that? trying to pass it to -f says 'unknown input or output format' [21:51] <durandal_1707> -formats [21:51] <tonsofpcs> oh, wait, it's just s24le! [21:51] <tonsofpcs> I may have it working. Thanks durandal_1707 :) [21:51] <niemand> Can I say rtmpdump to don't save the received data? without /dev/null, because I want to use rtmpdump on Windows [21:53] <durandal_1707> Jofironses: you are not using ffmpeg, this channel is for ffmpeg and not avconv/Libav fork [21:55] <klaxa> niemand: http://stackoverflow.com/questions/313111/dev-null-in-windows [21:56] <klaxa> also, i compiled ffmpeg and installed it, but pkg-config doesn't recognize it because there are no .pc files, do i generate them somehow or what? [21:57] <niemand> klaxa, thanks! [22:02] <klaxa> k nvm figured it out [22:17] <benbro> I have RTMP stream [22:17] <benbro> what encoding should I use to be able to stream it to adnroid and ios html5 [22:21] <Mavrik> HLS probably. [22:30] <Jofironses> durandal_1707: Ok, but whenever I run ffmpeg directly it tells me: *** THIS PROGRAM IS DEPRECATED *** [22:30] <Jofironses> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [22:31] <durandal11707> Jofironses: because you are using ffmpeg from Libav [22:31] <JEEBsv> that's within the libav project that debian and gentoo now use by default, and the ffmpeg binary was later removed from libav [22:32] <Jofironses> durandal_1707: I see, probably something about ubuntu apt-get. I will try compiling it then [22:32] <JEEBsv> basically the binary to use with libav-using distros is 'avconv' [22:32] <JEEBsv> but yes, building ffmpeg from source should do it as well :) [22:32] <Jofironses> k :0 [22:32] <Jofironses> k:) [23:15] <benbro> Mavrik: thanks. I'll check HLS [23:22] <Jofironses> Just to report back with my problem: I had no problems using a similar command using compiled ffmpeg :) [23:42] <IanWizard-Cloud> Mavrik: I don't think I ever told you last night, but thank you for the help. Everything's running smoothly now :) [00:00] --- Wed May 8 2013

1 0

ffmpeg-devel.log.20130506
by burek 07 May '13

07 May '13

[00:17] <kierank> 22:24:25 <"Daemon404> i still find it amusing that rff flags have to be hadled by an external filter. --? [00:17] <kierank> why not [00:23] <Daemon404> because applyign rff flags is the correct thing to do before returning video to users [00:23] <Daemon404> part of 'decoding' imho [00:23] <cehoyos> durandal_1707: I live in PAL-land and don't know much about sof-ttelecine, I believe there are several samples that use the feature on samples.ffmpeg.org [00:23] <Daemon404> without making them jump through hoops [00:23] <Daemon404> (and lavfi is a BIG hoop_) [00:25] <durandal_1707> so what is done in filter should be instead done in decoder? [01:27] <megaTherion> Is there an function which creates an flat duplicate of an AVFrame struct? [01:29] <Plorkyeran> memcpy? [02:24] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:dc35b77b28f4: ff_h264_decode_picture_parameter_set: use the correct sps [02:24] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:ccdaa31d8647: ff_h264_decode_picture_parameter_set: fix return code [02:24] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:26482ae716ba: h264_sei/decode_picture_timing: do not depend on h->sps being set [02:24] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:edabbfbaf2c9: ff_h264_decode_seq_parameter_set: dont set h->sps [02:24] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:5a9e3760495e: h264: check mb_width/height [02:52] <michaelni> ubitux, why is the "--enable-shared + libs " fate test failing? [02:53] <michaelni> gcc 4.8 ? [02:53] <michaelni> iam asking cuz i couldnt reproduce with older gcc [02:59] <cone-652> ffmpeg.git 03Paul B Mahol 07master:f5846dc98cee: id3v2: stop ignoring text encoding for chapter titles [03:01] <michaelni> ubitux, seems the ones failing are 20130425 gcc and the ones passing are 20130411 [03:17] <ubitux> michaelni: i think that's simply because i added libaacplus [03:18] <ubitux> and the fate test might not explicit the codec or something [03:18] <ubitux> so it might select libaacplus instead of the native decoder [03:18] <ubitux> (libaacplus and libfdk-aac) [03:19] <durandal_1707> its shared build? [03:21] <michaelni> ubitux, its failing in a vc1 test [03:22] <ubitux> ah, then i dunno [03:22] <ubitux> i'll try tomorrow [03:22] <michaelni> also http://fate.ffmpeg.org/report.cgi?time=20130505101638&slot=x86_64-darwin-gc… [03:22] <michaelni> fails similarly [03:23] <michaelni> both 4.8 [03:23] <michaelni> in absence of other ideas a bisect might be interresting [03:24] <ubitux> mmh im on a slow laptop right now, but i have 4.8, give me a few minutes [03:24] <michaelni> no hurry, it can wait till tomorrow or when you have time [03:27] <ubitux> 'got a gcc upgrade, possibly fixing that bug [03:37] <ubitux> fate-vc1-ism passes here with 4.8.0 20130425 [03:38] <ubitux> i'll have a look later if it's still reproducible after the upgrade i did [03:40] <ubitux> ah and i'm going to clean the ccache as well, just in case [03:41] <durandal_1707> why you setup .0 compiler at first place? [03:41] <ubitux> because i can [04:00] <highgod> Hi, all, I want to ask a question, what does ffserver do? I just know it is a server, but I don't konw what the capability it is, thanks all [04:01] <durandal_1707> there is documentation [04:06] <highgod> @durandal_1707:oh thanks, is there any application use it? [04:07] <durandal_1707> highgod: just durandal_1707 [04:09] <durandal_1707> highgod: http://ffmpeg.org/ffserver.html [04:09] <highgod> :)thanks [04:09] <durandal_1707> michaelni: i have hard time understanding asm in swr [04:10] <durandal_1707> is there any asm still to be written? [04:12] <michaelni> theres always more and better asm that could be written [04:15] <michaelni> also with every generation of cpus new instructions get added, so asm has to evolve too [09:51] <BuxiNess> Morning [09:52] <michaelni> morning BuxiNess [10:27] <cone-833> ffmpeg.git 03Diego Biurrun 07master:7abd35a1ffae: avformat: Add av_cold attributes to init functions missing them [10:27] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:5c2ccf83df3b: Merge commit '7abd35a1ffaecfd79fa07b801621ee11ab595c43' [10:34] <cone-833> ffmpeg.git 03Diego Biurrun 07master:bd8ac882140a: avcodec: Add av_cold attributes to end functions missing them [10:34] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:b50ab3a4c7a1: Merge commit 'bd8ac882140a38868c33c000a430a1292a352533' [10:42] <cone-833> ffmpeg.git 03Diego Biurrun 07master:2d5fd7886a29: bgmc: Do not mark ff_bgmc_decode_init() as av_cold [10:42] <cone-833> ffmpeg.git 03Luca Barbato 07master:46c1917350f5: nut: use a define for the nut version [10:42] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:77c85cbd9d4d: Merge commit '46c1917350f58dfab84e41919e6c02d43950db8c' [10:55] <burek> is there a recommended way to add dynamic text to a video using ffmpeg? something like gps data, provided with external application and such things [10:55] <durandal_1707> burek: drawtext supports process_command [10:56] <burek> great :) will look into that :) thanks man :) [10:56] <cone-833> ffmpeg.git 03Luca Barbato 07master:ac87eaf856e0: wav: Always seek to an even offset [10:56] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:5cb9093aa848: Merge remote-tracking branch 'qatar/master' [10:57] <durandal_1707> michaelni: when you gonna merge rsd/redspakr/.. ? [11:08] <michaelni> durandal_1707, I dont remember anyone asking for it to be merged. But i can anytime, is it jamrial/rsd on github ? [11:09] <durandal_1707> michaelni: i can merge it too... [11:09] <durandal_1707> i prefer to not commit as merget though - cleaner log [11:25] <durandal_1707> michaelni: so pick it: me or you [11:42] <michaelni> durandal_1707, i can do it once the extradata issue is fixed [11:44] <durandal_1707> michaelni: extradata is changed in each packet [11:47] <michaelni> durandal_1707, the data is only changed in rsd_read_header() which is called just once per file [11:47] <michaelni> for rsd [11:48] <durandal_1707> indeed [11:49] <michaelni> and same for redspark_read_header() and redspark [11:49] <durandal_1707> but not for brstm... [11:50] <michaelni> yes that needs to use something else than avctx.extradata [12:03] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:09602dbe7a64: wav_seek_tag: fix integer overflow [12:03] <cone-833> ffmpeg.git 03highgod0401 07master:9a09a9fc43e0: lavfi/unsharp: fix opencl crash on 64bit linux [12:14] <kierank> 23:23:28 <"Daemon404> part of 'decoding' imho --> no it's not [12:14] <kierank> applying rff is part of *display* [12:15] <kierank> such things are well defined in mpeg [12:17] <cone-833> ffmpeg.git 03highgod0401 07master:41aa19999858: lavu/opencl_internal: use av_opencl_errstr [12:17] <cone-833> ffmpeg.git 03highgod0401 07master:e215cb5e9670: lavfi/unsharp_opencl: use av_opencl_errstr [12:43] <cone-833> ffmpeg.git 03Carl Eugen Hoyos 07master:76f43cbe2648: Remove a superfluous include from libavcodec/options_table.h. [12:43] <cone-833> ffmpeg.git 03Michael Niedermayer 07master:0b5f4fdc381f: Merge remote-tracking branch 'cehoyos/master' [13:59] <xlinkz0> so there is a way to take a normal video and make it 3d? [13:59] <av500> yes [13:59] <av500> lots of TVs do that [13:59] <av500> its crap of course [14:00] <av500> a useless feature trying to sell 3D TVs [14:05] <durandal_1707> you can't make normal video 3d [14:08] <av500> you can fake it [14:14] <michaelni> you can hire a director, actors, cameraman and so on and turn any 2d movie in a 3d one [14:16] <durandal_1707> and compare that to recording movie first time with right camera [14:16] <iive> they actually do it with already filmed 2d movies, they cut out sections and displace them to create depth. [14:17] <iive> the first few movies after avatar were done this way. [14:17] <xlinkz0> so what is this stereoscopic thing in ffmpeg? [14:18] <xlinkz0> a fake filter? [14:22] Action: durandal_1707 omg! its fake! [14:38] <durandal_1707> seriously, why would filter be fake? [14:39] <av500> there is so much evil in the world these days [14:40] <durandal_1707> what is evil in this specific case? [14:40] <av500> no idea [14:40] <av500> I dont even know what that filter does [14:42] <xlinkz0> i meant to ask if it is similar to what tv's do and if it is any better [14:45] <iive> xlinkz0: there are videos that provide 2 separate pictures for each eye, but there are a number of different ways to encode the, side by side, top/bottom pictures, interlace (fields) and whatever mvc provides. so there is need to convert from one format to the other. [14:46] <iive> the, -> them, [14:48] <xlinkz0> i see [14:52] <Compn> hopefully 3d is just a fad [14:52] <Compn> because idiot movie directors only use it to put things in your face, instead of using it to enhance the experience [15:12] <ubitux> maybe xwininfo could be mentionned in that wiki page [15:12] <ubitux> eventually with a small 3-lines shell script to extract window position & run a capturing ffmpeg session [16:10] <durandal_1707> michaelni: working on gbrap input/output? [16:31] <michaelni> durandal_1707, not ATM [16:31] <michaelni> so you can work on it if you want ... [16:33] <durandal_1707> michaelni: i could try, but i'm not very happy what become of RGBA64 output patch [16:33] <durandal_1707> *already trying [16:34] <durandal_1707> what i need to do to read alpha? nothing? [16:36] <cone-125> ffmpeg.git 03highgod0401 07master:b02f073ad492: lavfi/deshake_opencl: use ff_opencl_set_parameter [16:40] <michaelni> iam not aware of anything special being needed for alpha, we already support packed rgba and planar yuva [16:43] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:d2bf8aecd35a: j2kdec: frame multithreading support [16:47] <durandal_1707> michaelni: looks like i got output working for rgba->gbrap, but it crashes for rgb24->gbrap case [16:52] <durandal_1707> there is code that fills one plane with 255 [16:59] <michaelni> thats for the case where input has no alpha but output has IIRC [17:00] <durandal_1707> yes, i have still that amount of brain to conclude that [17:00] <durandal_1707> thing is i get black for alpha plane, when i should get white [17:05] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:f4341c64be19: avfilter: add av_cold to uninit() [17:05] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:8cdf95d9b8f9: snow: add av_cold where appropriate [17:07] <durandal_1707> fixed segv, now only black issue remains [17:27] <durandal_1707> hmm gbrap output looks good and conversion from gbrap to 444p is ok, but to subsampled its wrong [17:39] <durandal_1707> huh gbrp16 sets output func [17:40] <durandal_1707> michaelni: sent patch, awaiting your review [18:15] <michaelni> durandal_1707, make fate fails with the patch [18:16] <michaelni> also when you update the checksums please look at if the newly added entries "look" ok [19:38] <Compn> http://gynvael.vexillium.org/ext/ffmpeg_log.txt [19:40] <Compn> they've been working hard :) [19:54] <saste> Compn, 666, the beast inside ffmpeg [19:55] <saste> trac is ill [20:52] <cone-588> ffmpeg.git 03Diego Biurrun 07master:9e5e76ef9ea8: x86: More specific ifdefs for dsputil/hpeldsp init functions [20:52] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:32cc7dacde6d: Merge commit '9e5e76ef9ea803432ef2782a3f528c3f5bab621e' [20:58] <cone-588> ffmpeg.git 03Diego Biurrun 07master:dc1b328d0df6: x86: hpeldsp: Only compile MMX hpeldsp code if MMX is enabled [20:58] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:a6e782434a21: Merge commit 'dc1b328d0df6e5ad5ff0ca4ae031e08466624f9c' [21:06] <cone-588> ffmpeg.git 03Diego Biurrun 07master:110796739ab3: x86: hpeldsp: Move avg_pixels8_x2_mmx() out of hpeldsp_rnd_template.c [21:06] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:108e2ae8299a: Merge commit '110796739ab32854dc0b6b0a1c95e6ae98889062' [21:38] <cone-588> ffmpeg.git 03Nicolas Bertrand 07master:28816f9db877: jpeg2000: Fix uninitialized data errors pointed out by valgrind/memcheck [21:38] <cone-588> ffmpeg.git 03Martin Storsjö 07master:6c5eac9a603a: swscale: Remove unnecessary backslashes [21:38] <cone-588> ffmpeg.git 03Nicolas Bertrand 07master:a06f0cef5bd4: jpeg2000: fix for uninitialized data errors in valgrind/memcheck [21:38] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:050b48f39783: Merge commit '6c5eac9a603a91a707d030d96ba97e2ef6b3ce50' [21:39] <michaelni> BuxiNess, your incomplete valgrind patch was applied to libav, so i applied&merged it too [21:39] <michaelni> ill post a patch that fixes the length==1 case in a moment [21:46] <michaelni> BuxiNess, patch posted, please review [21:52] <cone-588> ffmpeg.git 03Martin Storsjö 07master:6e293d111fca: swscale: Use alpha from the right row in yuva2rgba_c [21:52] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:b3e9fd1fe578: Merge remote-tracking branch 'qatar/master' [22:11] <michaelni> BuxiNess, posted a second patch that fixes the memleaks, please review [22:31] <cone-588> ffmpeg.git 03Rainer Hochecker 07master:06d8fdb660fd: Fix vdpau vc1 interlace modes also when using decoder vc1_vdpau. [22:31] <cone-588> ffmpeg.git 03Carl Eugen Hoyos 07master:8b8fa09b654e: vc1dec: Call ff_vdpau_vc1_decode_picture() once for each field. [22:31] <cone-588> ffmpeg.git 03Carl Eugen Hoyos 07master:905e1e4b521b: Cosmetics: Fix indentation and add braces after last commit. [22:54] <cone-588> ffmpeg.git 03Clément BSsch 07master:a82cc54fb5ce: lavf/mpsub: remove sscanf from probing. [22:56] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:974420a701c7: jpeg2000dec: fix memleaks in error cases in jpeg2000_decode_frame() [00:00] --- Tue May 7 2013

1 0

ffmpeg.log.20130506
by burek 07 May '13

07 May '13

[00:04] <mapreduce> On that same question, I've now recompiled ffmpeg with mjpeg enabled, and avcodec_find_decoder_by_name("mjpeg") still gives NULL. How can I do something like ffmpeg -codecs from code? [00:05] Action: mapreduce examines ffmpeg.c [00:06] <ubitux> i think it's in cmdutils.c [00:10] <durandal_1707> mapreduce: "jpeg" also gives null? [01:45] <mapreduce> durandal_1707: Correct. [01:46] <mapreduce> I copied the cmdutils.c code into my program, and it only gives the headers; it shows no decoders at all. [01:46] <mapreduce> yet ffmpeg -codecs shows what I expect, including mjpeg. [01:47] <BtbN> Is it possible to use lame presets when transcoding to mp3 with ffmpeg? [02:12] <BtbN> Hm, and it is possible to keep the metadata of a file when transcoding m4a -> mp3? [02:12] <BtbN> "-map_metadata 0" does not work [02:15] <BtbN> The output #0 ffmpeg gives looks ok and contains all the metadata. But the resulting mp3 file does not contain any metadata. [02:26] <BtbN> Hm, seems like the tags are copied, but somehow broken. At least VLC can't read them. [02:26] <BtbN> ffplay displays them correctly [05:17] <mapreduce> If anyone cares about my earlier problem, it was that I had not called avcodec_register_all(); [05:17] <mapreduce> and so no codecs had been registered. [05:49] <Astin> Um .. [05:50] <Astin> How can I split rtmp to file for every 7 seconds? [05:51] <Astin> when rtmp with live T.T [05:55] <Fjorgynn> loops? [06:05] <Astin> yes [06:05] <Astin> i will loop [06:05] <Astin> but what is option for ffmpeg [06:05] <Astin> I tried -t [06:06] <Astin> but I think -t donen't [06:06] <Astin> work [06:06] <Astin> I want to chunk rtmp to files. [07:55] <smj> How do I use x264's vbv-maxrate and vbv-bufsize options with ffmpeg? [07:58] <smj> -x264opts "vbv-maxrate=5000:vbv-bufsize=200:slice-max-size=1500" there it is [08:02] <smj> hm? "[libx264 @ 0x2c925e0] bad option '"vbv-maxrate': '800'" [08:05] <smj> oh nevermind, too many " in the script [08:08] <JEEB> smj, for just maxrate and bufsize there are the -bufsize and -maxrate mappings just fyi [08:08] <JEEB> if you need slice-max-size and stuff, then yes -- you will have to use x264opts [08:08] <smj> thanks [08:39] <smj> wasn't it supposed to work like ABR? it's encoding at about half the maxrate [08:46] <JEEB> maxrate and bufsize just make sure that you do not go over those limitations [08:46] <JEEB> so that if a decoder buffers bufsize amount of video, and has at least the bandwidth of maxrate, it should only need to buffer once during the whole playback session [08:51] <smj> so if I set bufsize of 1600 and maxrate 800, shouldn't the buffer of two seconds have an average bitrate of 800? [08:52] <JEEB> yes, but it will only limit the encoder's other rate control functions, not blow up the bit rate :) [08:53] <JEEB> so if your source doesn't need much bits or you have instructed it to keep at some other bit rate, it will just keep a check on the stream to be under those limits [08:54] <smj> so should I use CRF in addition if that's what I want? [08:54] <JEEB> yes, if you want to set your rate control via crf (closest to "constant quality" we currently have), then that's the way [08:55] <JEEB> if you haven't set any rate control yet, then the default is crf 23 [09:16] <smj> What happens if I use -crf 1 with the limiting options? [09:16] <JEEB> x264 should keep it all within the maxrate/bufsize [09:18] <JEEB> (and it will yell at you loudly if it couldn't, although that should never happen with relatively new x264 versions, as about a year+ ago an emergency mode was added for things that wouldn't compress no matter what that filters stuff to keep it within limits, so you could encode /dev/urandom and still keep VBV alright) [09:18] <JEEB> in general I just recommend using the highest crf value that still looks good to you, so you don't overuse rate and leave room for possibly harder-to-compress pictures [09:19] <smj> okay [09:35] <EvilTengil> Hello everyone! I'm still looking for a way to send data (6 doubles) bundled with each frame. Im using JMPEG as a codec but I could use any container. I'm writing both the sender and receiver of the video so it doesn't have to be a proper standard way of doing it. [11:07] <kornerr> hi [11:23] <Astin> Hello ~~ [11:23] <Astin> Can I ask that how can I split rtmp to files ? [11:24] <Astin> When I read the data from rtmp, ffmpeg wait till stop. [11:25] <Astin> I just want to chop the rtmp to file . [11:25] <Astin> Somebody help me . [11:42] <Astin> ^^ [11:49] <kornerr> there's nobody here to help [11:49] Action: kornerr wants help too [11:49] <kornerr> this is a god forsaken channel [11:51] <JEEB> (says someone who doesn't even note his problem or what he wants help with) [11:51] <kornerr> hehe [11:52] <kornerr> i want to stream to android like here: http://ffmpeg.org/pipermail/ffmpeg-user/2013-January/012825.html [11:52] <JEEB> and no Astin, I have no idea how to split output between files of a certain length in ffmpeg as you highlighted me elsewhere probably trying to bring it up :P [11:52] <kornerr> and it doesn't work [11:52] <kornerr> either ffmpeg is developing fast, but almost none of the examples i find on the net works [11:53] <kornerr> or my setup is different [11:53] <JEEB> at the point that you are using ffserver I can't help you [11:53] <JEEB> sorry [11:53] <kornerr> yeah, ffserver [11:53] <kornerr> curses! [11:53] <JEEB> try VLC's rtsp encoding or something? [11:53] <JEEB> it has command line interface as well [11:53] <kornerr> well, actually my basic task is to display UDP on Android [11:53] <JEEB> while the encoder settings are at times crazy, setting up just an rtsp stream with VLC was rather simple [11:53] <kornerr> so i thought ffmpeg could convert it to something Android understands [11:53] <JEEB> tried it locally the other day [11:54] <kornerr> hm [11:54] <kornerr> ok [11:55] <JEEB> also that sample seems surprisingly sane at the mailing list, but to be honest I don't really want to go into ffserver stuff :s [11:55] <JEEB> it seems like it's one of those Dark Tomes from the Cthulhu Mythos [11:56] <JEEB> mostly unmaintained thing that someone made back in the days [11:56] <JEEB> stuff that ffmpeg can do by itself is awesome tho :D [12:04] <kornerr> :) [12:16] <kornerr> looks like every open source project needs moar ppl [13:15] <Astin> Thanks for answering [13:15] <Astin> JEEB, [13:15] <Astin> some of part, I did it [13:15] <Astin> but [13:16] <Astin> I wanna .. Um .. [13:16] <Astin> Okay anyway thanks [13:16] <Astin> I want to http to client from rtmp and ffmpeg by chopping the rtmp live protocol. [13:17] <Astin> I have to see the source code more I think [14:18] <abhilashtt01> Hi Guys [14:18] <abhilashtt01> I am tring to install ffmpeg [14:19] <abhilashtt01> getting the following error [14:19] <abhilashtt01> /usr/bin/ld: /usr/local/lib/libvorbisenc.a(vorbisenc.o): relocation R_X86_64_32S against `a local symbol' can not be used when making a shared object; recompile with -fPIC [14:19] <abhilashtt01> /usr/local/lib/libvorbisenc.a: could not read symbols: Bad value [14:19] <abhilashtt01> collect2: ld returned 1 exit status [14:19] <abhilashtt01> make: *** [libavcodec/libavcodec.so.52] Error 1 [14:19] <abhilashtt01> any ideas? [14:20] <JEEB> abhilashtt01, static library being used to build shared? Re-compile it with PIC enabled (the dependency that is) [14:23] <abhilashtt01> how to enable PIC [14:23] <abhilashtt01> ? [14:23] <JEEB> if the configuration script for that library doesn't have --enable-pic or something as an option, then you would have to add the -fPIC option to the cflags used during compilation [14:27] <abhilashtt01> ok [15:00] <abhilashtt01> @JEEB , I had recompiles with --enable-pic --extra-cflags=' -fPIC' [15:00] <abhilashtt01> Still same error [15:02] <zozo> Good day! I have many wav files with phone conversations and I want to convert all this files to mp3. But after I did it with ffmpeg or lame (tried many different parameters) I've got an mp3 with strange noise like UFO :) Does anybody know what it could be? Thanks in advance! [15:04] <retard> zozo: what is the sampling rate and bit depth of the files you begin with? [15:05] <zozo> You mean source wav files? [15:05] <retard> yes [15:05] <zozo> how can I get it? [15:05] <retard> ffprobe should do it [15:06] <zozo> anyway if I listening source wav files in player - no this kind on noise [15:06] <zozo> https://gist.github.com/anonymous/bfbe2b9c61947cd60022 [15:06] <retard> yeah [15:07] <retard> this is what i thought [15:07] <retard> notice the 8000 hz [15:07] <cbsrobot> zozo you are using a very old ffmpeg [15:07] <zozo> yes - I see. What could it mean? [15:07] <cbsrobot> and you are using the fork version [15:08] <zozo> I use it from reps in linux mint 13 (ubuntu 12.04) [15:08] <cbsrobot> compile ffmpeg on ubuntu yourself [15:08] <cbsrobot> or see: [15:09] <retard> try resampling to something higher [15:09] <zozo> as I understand, ffmpeg now is just an alias to avconv in ubuntu? [15:09] <zozo> how can I resample it? [15:10] <retard> --resample in lame [15:10] <retard> for instance [15:11] <zozo> one second... [15:11] <retard> not sure what the option is in ffmpeg [15:11] <durandal_1707> zozo: ffmpeg is not alias for avconv, you are not using ffmpeg from FFmpeg but ffmpeg from Libav [15:12] <zozo> ok, so If I install them from https://launchpad.net/~jon-severinsson/+archive/ffmpeg it should be not so old? [15:13] <ubitux> these packages are still extremely old [15:14] <zozo> ok, so the reason is the old version? or that I need to resample? [15:15] <retard> strictly speaking you are in the wrong channel for the software you're using [15:16] <durandal_1707> zozo: you can download/use static builds [15:17] <JEEB> abhilashtt01, let me guess... you compiled ffmpeg with that instead of the actual dependency [15:18] <zozo> I didn't compile it myself - I've just used from reps. Downloading static... [15:18] <retard> this will most likely not do anything for the problem you're describing [15:19] <abhilashtt01> JEEB: its fixed now. recompiled libvorbis with --enable-shared --with-pic [15:19] <JEEB> yes, that is what I said you should do [15:19] <abhilashtt01> Ok, ty. [15:19] <abhilashtt01> I have a doubt about ffmpeg versions [15:19] <abhilashtt01> FFmpeg version SVN-r26402, [15:19] <abhilashtt01> FFmpeg version 0.6.5, [15:19] <abhilashtt01> which one is latest? [15:19] <JEEB> both are old as hell [15:20] <JEEB> latest is 1.2 release-wise [15:20] <abhilashtt01> ffmpeg version git-2013-05-03-f51aa92 [15:20] <durandal_1707> lame supports 8000 sample rate... [15:21] <JEEB> abhilashtt01, that sounds like newer :P [15:21] <abhilashtt01> our clients are using clipbucket and others s/w with latest ffmpeg those softwares not have compatibility issues [15:22] <zozo> this commands fix the problem with my current (old) ffmpeg, but files size is not so small: sox test.wav -r 14400 test1.wav resample && lame test1.wav test1.mp3 [15:23] <retard> i think the problem is related to the psychoacoustics [15:24] <retard> i would consider using a different codec altogether if possible, like say speex [15:30] <zozo> ok, so as I see the problem is in the 8KHz. If I resample with 14400Hz and after that convert with lame - everything is ok. If I resample with 8000Hz - the same noise. [15:31] <durandal_1707> use 44100 [15:31] <zozo> If I use '-b 32' in lame, file size is enough small and sound is enogh good [15:32] <zozo> If I use 44100 it will increase result file size as I understand [15:32] <retard> there's not much point in using 44100 [15:32] <zozo> I need this convertion to store many phone conversation [15:32] <zozo> so the files size and enough quality - is two main requirements [15:32] <retard> if the format is arbitrary, convert to speex codec [15:33] <durandal_1707> use opus instead [15:33] <retard> it is made for saving space and maintaining quality on human speech [15:34] <zozo> actually, I just need format that has good compression and possible to easy play (googling about speex...) [15:34] <durandal_1707> and lame does not suppoort 14400, but it supports (among others) 16000 and 11025 [15:34] <zozo> and one more requirements - easy integration :) [15:35] <zozo> lame resample to 16kHz automatically [15:35] <durandal_1707> than just zip wavs? [15:35] <zozo> I'll try 11025 - thanks [15:35] <durandal_1707> zozo: thing is you do upsampling twice [15:35] <zozo> seems so [15:35] <zozo> zip - is not good compression as I tested [15:36] <durandal_1707> and you do not need sox to upsample, -ar 16000 should do same [15:37] <durandal_1707> thought that old version may do it slowly... [15:38] <zozo> seems like 11025 and 16000 gives almost the same file size at result [15:39] <zozo> as I inderstand speex also need to be compiled manually under ubuntu? [15:56] <hughmanwho> Hello, would anyone know why I'm getting this output when I try to stream h264 via rtsp and save the first 60 seconds as a file? http://pastebin.com/j6N0WVX0 [16:01] <klaxa> unreliable server? [16:03] <hughmanwho> Works fine when I use VLC player [16:30] <hughmanwho> Is there a way to tell it to request PPS frames or something? Just not sure why I'm getting all these error messages: http://pastebin.com/j6N0WVX0 [16:35] <JEEB> seems like you are missing packets and thus not being able to get enough of the H.264 stream to be able to start decoding it [16:36] <JEEB> [h264 @ 00000000023cefc0] RTP: missed 158 packets [16:50] Action: durandal_1707 swscale can't get uglier than it is [16:51] <hughmanwho> Yeah, but it doesn't always miss packets [16:51] <hughmanwho> And every once in say 10-20 times it'll actually work [16:52] <hughmanwho> Also it works fine for mjpeg [17:31] <xlinkz0> hughmanwho: try adding -rtsp_transport tcp before the input parameter [17:32] <xlinkz0> might also run it higher verbosity [17:32] <xlinkz0> something like -report [17:51] <hughmanwho> Hey xlinkz0, nope still the same thing with '-rtsp_transport tcp' [17:51] <xlinkz0> supply -report and see if you can find anything useful in the detailed output [17:54] <hughmanwho> Here it is with -report, shows a little bit more about the SDP http://pastebin.com/Rf7CrSTv [18:00] <xlinkz0> hughmanwho: run again with tcp and "-c copy" to out.mp4 [18:04] <hughmanwho> Like this: C:\Tools\ffmpeg-20130428\bin>ffmpeg -report -rtsp_transport tcp -i rtsp://admin:admin@192.168.14.25/defaultPrimary?streamType=u -t 30 -analyzeduration 25000000 -c copy out.mp4 [18:06] <hughmanwho> Here's the pastebin: http://pastebin.com/Rf7CrSTv same thing basically [18:08] <xlinkz0> so you say it works with vlc? [18:09] <xlinkz0> did you compile ffmpeg yourself by any chance? [18:10] <hughmanwho> It does work with vlc [18:11] <hughmanwho> No this version if the latest version off of the zeranoe website, compiled on Apr 28th about a week ago [18:15] <xlinkz0> idk [18:16] <hughmanwho> Ok well thanks xlinkz0, I appreciate the suggestions!! [18:17] <hughmanwho> Also what is DTS in regard to h264? When I run with loglevel debug I get a whole bunch of errors along the lines of: [rtsp @ 00000000023a0420] Non-increasing DTS in stream 0: packet 2 with DTS 4294966984, packet 3 with DTS 4294966984 [18:17] <hughmanwho> [rtsp @ 00000000023a0420] Non-increasing DTS in stream 0: packet 3 with DTS 4294966984, packet 4 with DTS 4294966984 [18:17] <hughmanwho> [rtsp @ 00000000023a0420] Non-increasing DTS in stream 0: packet 4 with DTS 4294966984, packet 5 with DTS 4294966984 [19:08] <xlinkz0> DTS means decoding time-stamp [19:08] <xlinkz0> read this if you want to learn a bit about video low level stuff : http://dranger.com/ffmpeg/ffmpeg.html [19:09] <xlinkz0> @ hughmanwho [19:26] <hughmanwho> Thanks again xlinkz0 [19:43] <hughmanwho> Anyone know if ffmpeg supports all h264 profiles? [19:44] <JEEB> I'd say it supports everything you could come up with in the Real World, except MVC (the other views being encoded in a specific way from the main view, the main view being normal AVC/H.264) [19:45] <hughmanwho> Is there a way I could determine which profile an incoming stream is? [19:45] <JEEB> by parsing specific NAL units I would guess [19:45] <JEEB> which would have that information [19:46] <JEEB> also just going to note that if VLC worked with it, that uses the libavcodec library as well for decoding :P [19:48] <hughmanwho> hmm.. so I guess I must be passing in different parameters than VLC? [19:57] <hughmanwho> What does probesize change? Could that be causing problems? [19:58] <JEEB> probesize changes the amount of data libavformat reads to check what's coming in [20:00] <hughmanwho> hmm.. ok well I'm off to a meeting.. thx I'll bbl [20:00] <xlinkz0> hughmanwho: just a final thought, you should try sniffing the network traffic with both vlc and ffmpeg and see where they differ in the rtsp request [20:01] <xlinkz0> the request is formatted similar to http so it's human readable [21:02] <cbsrobot> is it known fact, that pure black won't be encoded as black in h264 ? see http://i.imgur.com/EvCnhHI.png (image with gain applied) [21:02] <cbsrobot> ffmpeg using libx264 encoder [21:04] <iive> cbsrobot: yes, yuv formats have dead-zones of about 10-16 values at each end [21:05] <hughmanwho> Sounds like a good idea xlinkz0, how would I sniff the network traffic? [21:05] <cbsrobot> because my projector setup was badly configured I spotted some artefacts in a bluray [21:06] <xlinkz0> hughmanwho: wireshark on windows [21:06] <cbsrobot> iive: I encode prores bure black to bluray and back to prores and I can see these artefacts on the filnal image aswell [21:07] <cbsrobot> I'd had guessed, that pure black will remain pure black - even for a lossy encoding like h264 [21:07] <cbsrobot> pure black in the sens of RGB 0 = YUV 16 [21:07] <iive> do you have idea what dct is used by prores? [21:08] <cbsrobot> no - where can I find it out &. ? [21:08] <iive> probably in the source [21:09] <cbsrobot> could be a problem in yuv422p10 -> yuv420p .. no ? [21:11] <iive> no idea... cutting bits is easy task, but if dither is involved then it could introduce some pattern, thanks to bad rounding. [21:13] <iive> h264 have looseless mode, it should provide binary identical image [21:15] <iive> afair h264 didn't had any magic rounding constants, so this is not likely to be problem. [21:18] <cbsrobot> iive: good idea - i'll try that tomorrow [21:19] <iive> i mean,t h264 (i)dct ... [21:20] <cbsrobot> the values for pure black to one dot is just 1 - meaning RGB: 0 0 0 -> 0 0 1 [21:21] <iive> entirely reasonable for rounding error. [22:31] <hughmanwho> Turns out the company that made the camera I'm trying to stream from uses UDP instead of TCP.. still hasn't fixed my problem but they sound like they will be helpful and have encountered this problem before, have gotten it to work with ffmpeg, and are willing to look at my code/give me some.. sweet! Thanks again for the help, especially xlinkz0! [22:33] <xlinkz0> np [23:46] <trose> hey so I'm looking to make a video that is mostly static with only a fixed area containing animation. Is it possible to use a static image and then overlay each frame of a smaller video in a fixed location. I did some research into -vf but it appears that ffmpeg will not allow me to use an image as the primary input. Any ideas? [23:47] <Mavrik> um, ffmpeg should allow image as primary input :) [23:52] <trose> hmm okay [23:52] <trose> so i already tried this.. [23:53] <trose> Mavrik: http://pastebin.com/vHrNfPDH [23:53] <trose> i'm most likely missing something.. [23:54] <Mavrik> hmm [23:54] <Mavrik> usually "-loop 1" is used as well [23:54] <Mavrik> so it "loops" input [23:56] <trose> ah... [23:56] <trose> hmmm maybe i need to specify the length to loop [23:58] <trose> oh snap! yeah i needed to specify a -t for my input image [23:58] <trose> ty Mavrik :P [00:00] --- Tue May 7 2013

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