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August 2014

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ffmpeg-devel.log.20140809
by burek 10 Aug '14

10 Aug '14

[00:11] <cone-822> ffmpeg.git 03Carl Eugen Hoyos 07master:4b63bcef907b: Autodetect jpeg-ls files. [02:45] <kierank> nevcairiel: sent a patch to do afd in h264 [02:45] <kierank> (I think it was you who asked me) [03:34] <cone-495> ffmpeg.git 03Mark Reid 07master:d6af706eee48: avformat/movenc: write reel_name metadata to tmcd atom [05:21] <cone-495> ffmpeg.git 03Michael Niedermayer 07master:64d029de41ed: avformat/matroskaenc: fix MAX_CUEPOINT_SIZE calculation [06:38] <Compn> mythtv has many modifications to ffmpeg, including teletext mpeg-ts changes ? nice [06:39] <Compn> i vote we include the whole demuxer as a secondary demuxer in ffmpeg [06:39] <Compn> call it mpegts2 or something [06:39] <Compn> maybe it will demux some files better or worse than our demuxer [06:53] <nevcairiel> Luckily no one listens to you. :) [06:56] <Compn> so its better just to ignore possibly better code ? [06:56] <Compn> we do nothing > nothing improves [06:56] <Compn> we do code duplication > someone gets annoyed enough to port changes [06:57] <Compn> troll development 101. [06:57] <Compn> i remember some complaints about mpegts stuff over the years [06:57] <nevcairiel> Just copying some code doesn't improve anything [06:58] <nevcairiel> Find stuff that doesn't work and improve what we have [07:00] <wm4> makes me want to grab mythtv's ffmpeg and diff it [07:00] <Compn> nevcairiel : you looked at mpegts.c ? :) [07:00] <nevcairiel> Often [07:00] <nevcairiel> Have you? [07:01] <Compn> no not at all [07:01] <Compn> wm4 : do it. post giant diff :) [07:04] <wm4> how many libraries did they fork [07:04] <wm4> libmythbluray... [07:13] <Compn> well when you start collecting internal libs.... and the externals are either a) dead or b) change api or c) hate downstream ... its a tough habit to break [07:13] <Compn> also when you modify them and optimize or fix and upstream rejects. [07:19] <nevcairiel> I don't remember any rejected back ports on the ML, which means their last attempt to must be more than 2-3 years ago. A lot had changed in that time, and by now they are just a bitter uncooperative downstream [07:20] <Compn> we have to make friends with all of those downstreams we burned years ago [07:20] <Compn> cant abandon them now nevcairiel [07:20] <nevcairiel> They abandoned us :P [07:20] <wm4> oh no [07:20] <wm4> they're cherry-picking [07:21] <Compn> wm4 : are you saying theres no git magic to save us now ? [07:21] <Compn> ehe [07:22] <wm4> probably will lead to a huge diff, depending on stuff [07:23] <wm4> I found this mysterious README.sync: http://sprunge.us/HCKZ [07:23] <wm4> instead of leading it to git, they made some sort of changelog [07:24] <Compn> closed caption support [07:24] <Compn> didnt someone just ask about that [07:25] <Compn> yes they did [07:25] <Compn> now you know i'm going to have to post this ... [07:25] <Compn> :) [07:26] <j-b> they should use libvlc </compn> [07:32] <wm4> this is the diff: http://sprunge.us/gTcY [07:32] <wm4> from mythtv git head to ffmpeg 2.2.4 [07:33] <wm4> I think my ffmpeg git tree wasn't entirely clean, I hope no crap ended up in it [07:34] <wm4> ff_codec_id_string [07:34] <wm4> ???? [07:34] <j-b> I hope this is cleanly integrated and not rushed-in because "MOAR FEATURES" [07:34] Action: Compn trolls j-b extreme [07:35] <Compn> one suggestion and everyone freaks out :\ [07:35] <Compn> j-b : we are already reviewing the diff for useful items! [07:35] <j-b> Compn: sorry, but seeing some features getting merged... [07:36] <Compn> when exactly did everyone become so against new features ? [07:36] <Compn> its a strange argument [07:36] <wm4> vlc prefer to NIH the easy bits [07:36] <j-b> wehn you started merging EVERYTHING [07:36] <j-b> wm4: right. [07:37] <j-b> wm4: and your ref for this is? [07:37] <j-b> Compn: when you arrive to get 3 prores decoders and 2 prores encoders... [07:37] Action: Compn lols [07:37] <j-b> Compn: or merging code producing non-standard files [07:38] <Compn> the prores thing has reasons [07:38] <Compn> good reasons! [07:38] <wm4> j-b: such as you being against subtitle support, because it just adds potential for bugs and because vlc already has its own subtitle code [07:38] <j-b> wm4: I'm clearly not against subtitles support [07:38] <j-b> I'm against bad subtitles support [07:38] <wm4> :) [07:38] <j-b> and I'd love to get rid of VLC subtitles code [07:39] <j-b> the less code I need to maintain, the best it is. [07:39] <wm4> cue to ubitux blaming Libav for "needing" to use the bad subtitle API [07:39] <j-b> but, seriously, subtitles decoders are very simple [07:39] <j-b> issue is rendering [07:40] <wm4> there's a library with a nice name for that [07:40] <j-b> really ? [07:40] <j-b> I doubt it. [07:41] <wm4> (libass) [07:41] <j-b> There is a library that depends on fontconfig [07:41] <j-b> aka, not an option [07:41] <wm4> libass is working on it [07:41] <j-b> sure [07:41] <j-b> it's not there yet [07:41] <wm4> blame the lazy contributors [07:41] <wm4> for OSX there's a working branch but with problems [07:42] <j-b> yeah, so, there is nothing I can use now [07:42] <j-b> 07:36 <+wm4> vlc prefer to NIH the easy bits [07:42] <wm4> for windows there's an idea, but the one working on it has time for it only in 10 years [07:42] <j-b> and this is not NIH [07:42] <j-b> it was in VLC before [07:42] <wm4> sure, that was a sarcastic joke at best [07:42] <wm4> (same as the 10 years) [07:42] <Compn> now mpv dev is trolling vlc dev :P [07:42] <Compn> ehe [07:43] <j-b> Compn: but yeah, so no libavfilter for me [07:43] Action: wm4 slaps Compn [07:43] <Compn> j-b : fine, have fun reinventing stereo3d :P [07:44] <Compn> wm4 : i'm trying to get libavfilter into vlc :P [07:44] <j-b> Compn: maybe one day, you will split the repos like I suggested a long time ago [07:44] <j-b> and stop violating libav* boundaries [07:45] <Compn> those boundaries are more like guidelines [07:45] <Compn> is there an actual rule about them somewhere ? [07:45] <wm4> so yeah, the largest addition in the mythtv patch is a new mpeg-ts demuxer, probably based on the old one [07:46] <wm4> plus some dvb stuff, plus random hacks and fixes all over the place [07:46] <wm4> +libavformat/mov.c // Fix http streaming of iPhone videos (Fix from xbmc) [07:46] <wm4> that one is pretty funny [07:46] <j-b> Compn: making more libraries depend on libavcodec. is not fun [07:47] <wm4> a fix that traveled from xbmc's internal copy to mythtv's... [07:47] <j-b> wm4: and will get merged into mplayer's copy, then merged into ffmpeg, then backported in libav [07:47] <wm4> j-b: IMO filters would be better implemented as a standard API (think avisynth/ladspa/etc.), rather than a repo that tries to accumulate everything [07:47] <wm4> lol [07:47] <Compn> mplayer doesnt have internal copy anymore... [07:47] <Compn> git copy now [07:47] <j-b> wm4: I agree, of course [07:48] <nevcairiel> My fork probably has more changes than theirs, pfff amateurs [07:48] <wm4> Compn: it's still built as internal copy [07:48] <wm4> right, nevcairiel has his own fork [07:48] <j-b> wm4: and resampler shouldn't be there either [07:48] <Compn> i wonder what changes ffmbc has... [07:48] <wm4> with ordered chapters [07:48] <wm4> j-b: that's needed for automatic conversion filters, but I find them evil too [07:48] <j-b> Compn: ffmbc has mostly mov, mp4 and mxf massive changes [07:48] <wm4> just how many fucking forks are there [07:48] <j-b> wm4: so? why does libavcodec or so need conversion filters? [07:49] <Compn> wm4 : tons. [07:49] <nevcairiel> ffmbc abused the GPL to make back porting impossible though [07:49] <wm4> j-b: oh, thought you talked about lavfi [07:49] <wm4> lavc depends on resampling? [07:49] <Compn> j-b : would you be happy if we duplicated all code around so each libav* was easily seperated ? [07:49] <Compn> pfft not abusing gpl but whatever [07:49] <nevcairiel> wm4: opus decoder uses av/swresample [07:50] <wm4> meh. [07:51] <nevcairiel> Compn: that clause of the LGPL is BS. You can fork a LGPL project and then make it so that the original project can't benefit from your changes. Good free software! [07:52] <wm4> j-b: what I envision is a set of API definitions to allow freestanding filters (and maybe more), plus a support library for basic data structures (ffmpeg-independent AVFrame replacement) [07:52] <Compn> nevcairiel : just more ifdef enable_gpl .... i dont see problem with it :P [07:52] <wm4> the incompatibilities between the various GPL licenses are the greatest failures of the GPL [07:53] <Compn> also talking with baptiste a few times, he seems willing to go lgpl [07:53] <Compn> or maybe i have misunderstood his replies [07:54] <j-b> wm4: libavfi depends on libavcodec, libavcodec depends on resample, lavf depends on lavc (this cannot be done differently) [07:55] <Compn> j-b :we combine it all and call it libffmpeg , similar to libvlc no ? [07:55] <j-b> Compn: but that's not what you do. [07:55] <j-b> if you did, maybe [07:55] <Compn> i mean would it be ok ? [07:55] <Compn> ok [07:55] <nevcairiel> These days lavfi at least shouldn't have a hard dep on lavc anymore [07:55] <nevcairiel> Only if you use one of the specific filters that use it [07:56] <j-b> Compn: it was my major point when I said you should split the gits [07:56] <wm4> on the other hand, I see no big point in actually separating lavc and lavf [07:56] <Compn> i probably forgot your solution j-b [07:57] <j-b> wm4: well, agreed [08:00] <wm4> also, I guess one "problem" with split repo is that FATE is also part of the repo [08:00] <wm4> I guess it would have to be part of the repo that houses ffmpeg.c [08:00] <Compn> talking about just splitting lavc into its own git repo? what would it do ? [08:01] <wm4> it would bring peace and harmony [08:01] <Compn> oh [08:01] <Compn> somehow i doubt [08:02] <Compn> maybe if someone took over as admin and referee/babysitter [08:02] <wm4> aka maintainer? (??) [08:08] <Compn> thanks for the diff wm4 [08:08] <Compn> wonder what xbmc's fork diff looks like [08:09] <nevcairiel> They keep individual patches in a folder, iirc [08:09] <j-b> wm4: ffmpeg.c should be in a different repo [08:10] <Compn> probably i agree with that [12:23] <cone-860> ffmpeg.git 03Diego Biurrun 07master:d35b94fbabd8: avcodec: Rename xvidmmx IDCT to xvid [12:23] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:3841f2ae665d: Merge commit 'd35b94fbabd8beb5d566c0b5d01688aff62c3b36' [12:52] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:25eeff960030: avcodec/avdct: add "xvid" alias AVOption and use FF_IDCT_XVID [14:55] <kierank> 6:56 AM <"j-b> Compn: it was my major point when I said you should split the gits --> you'd have to make libav do that [14:55] <kierank> otherwise ffmpeg can't [14:58] <wm4> lol [15:12] <viperfx_> Is there any way to improve the speed of the avformat_open_input() call for an HTTP stream? Does FFmpeg need to read all of it before it can proceed? [15:14] <michaelni> viperfx_, submit a bugreport to trac if opening http takes unreasonably long [15:14] <wm4> viperfx_: first I'd try to confirm how much data it really reads [15:15] <wm4> it can vary; e.g. I had cases where the probing ruins web radio streams, while in other cases it's minimal [15:19] <viperfx_> Sorry I did not mean stream as in live. I meant its a file. But its a mp4a in DASH container [15:19] <viperfx_> It seems to take 4-6s to complete that call. That time suggests that it is reading the whole file. [15:19] <wm4> there's some issue in the mov demuxer that makes it read dash files completely [15:20] <wm4> yeah [15:20] <wm4> AFAIK you can work it around by disabling indexing [15:20] <wm4> I haven't bothered to track down why it happens, though [15:20] <viperfx_> how can I do that? [15:21] <wm4> unfortunately, I forgot [15:21] <wm4> it's some weird avoption [15:21] <viperfx_> ok I will track it down. [15:22] <wm4> I think a generic one (not specific to mov/mp4, just interpreted by it) [15:39] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:f75786f3bc88: avformat/avio: Fix "warning: struct AVBPrint declared inside parameter list" [17:45] <ubitux> beastd: ping [19:12] <cone-860> ffmpeg.git 03Anton Khirnov 07release/0.10:187cfd3c13a1: eamad: use the bytestream2 API instead of AV_RL [19:12] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:d60f680fa741: Merge commit '187cfd3c13a1deb47661486824a5b8f41e158a7a' into release/0.10 [19:19] <cone-860> ffmpeg.git 03Diego Biurrun 07release/0.10:e4fdfdf65d52: vf_select: Drop a debug av_log with an unchecked double to enum conversion [19:19] <cone-860> ffmpeg.git 03Bernhard Übelacker 07release/0.10:277103e07fbe: video4linux2: Avoid a floating point exception [19:19] <cone-860> ffmpeg.git 03Diego Biurrun 07release/0.10:28f2d3c5a5a3: cmdutils: Conditionally compile libswscale-related bits [19:19] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:c4dabc38a3a2: Merge commit 'e4fdfdf65d520ce3af13a21ff8a3649e37757af8' into release/0.10 [19:19] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:f3ae90621e03: Merge commit '277103e07fbe22fc8e4361bacd5c6b48133f3ba5' into release/0.10 [19:19] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:cdfd61b78ba6: Merge commit '28f2d3c5a5a3a3c14a68cf691054f15e4f23355a' into release/0.10 [19:33] <ubitux> kierank: avr is way less tested than swr in ffmpeg [19:34] <cone-860> ffmpeg.git 03Diego Biurrun 07release/0.10:976f2e0a542e: x86: Fix linking with some or all of yasm, mmx, optimizations disabled [19:34] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:a465ed5707f5: pgssubdec: Check RLE size before copying [19:34] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:184c79729d40: h264_sei: check SEI size [19:34] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:2c10833d5e4c: Merge commit '976f2e0a542e47aaf68ddbe001fb70a00bf96d99' into release/0.10 [19:34] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:5404bf29c3b7: Merge commit 'a465ed5707f5cbc9713d5e9629d424cd2d46e038' into release/0.10 [19:34] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:330791c2ae2b: Merge commit '184c79729d4011f33027bcdc61a63d521017ebc1' into release/0.10 [19:38] <ubitux> kierank: did you find your issue with opus/swr? [19:38] <kierank> no [19:38] <kierank> it's something to do with float_dsp I think [19:38] <kierank> not sure if swr uses float dsp, need to check [19:39] <kierank> ubitux: i mean avr should be enabled to allow people to use it [19:39] <kierank> certainly the assembly is much better than swr [19:39] <ubitux> how so? [19:41] <kierank> I guess it got fixed but there was a lot of inline asm [19:42] <ubitux> there isn't anymore yes [19:42] <ubitux> did you make any benchmark? [19:45] <nevcairiel> swr has some more assembly though, it should be faster when resampling [19:45] <kierank> well i need to figure out how i can get a crash dump that is usable from it [19:49] <cone-860> ffmpeg.git 03Vittorio Giovara 07release/0.10:7585a6254bbb: h264: prevent theoretical infinite loop in SEI parsing [19:49] <cone-860> ffmpeg.git 03Janne Grunau 07release/0.10:3e60501f311c: h264: slice-mt: check master context for valid current_picture_ptr [19:49] <cone-860> ffmpeg.git 03Mans Rullgard 07release/0.10:50493f1f7d22: twinvq: fix out of bounds array access [19:49] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:0a83007ceee4: Merge commit '7585a6254bbb38148e4467793fc34211b79d5f7d' into release/0.10 [19:49] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:b5ae0e349adc: Merge commit '3e60501f311c50bf234033f206c19d34d889df01' into release/0.10 [19:49] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:2ec8e46550dc: Merge commit '50493f1f7d2235db811d2991b9e5b330baf7c05a' into release/0.10 [19:51] <nevcairiel> Libav did bump now, if someone has some quick ABI changes, now is the chance :p [19:52] <jamrial> i already sent some patches with the objective of cleaning the .v files so we stop exporting internal functions [19:52] <nevcairiel> The constness change we used yo hide behind a define is probably one thing that can be enabled after the bump [19:53] <jamrial> what we need to check is which of the deprecated stuff we have that libav doesn't we want to postpone or let die right now [19:54] <nevcairiel> Their goal was to keep API in place and only do a few ABI changes [19:54] <ubitux> ./ffmpeg -i ~/samples/audio/april02.flac -ar 8000 -f null - 4.61s user 0.10s system 312% cpu 1.508 total [19:54] <ubitux> ./ffmpeg -i ~/samples/audio/april02.flac -ar 192000 -f null - 9.92s user 0.21s system 156% cpu 6.480 total [19:54] <ubitux> ./avconv -i ~/samples/audio/april02.flac -ar 8000 -f null - 5.91s user 0.03s system 99% cpu 5.942 total [19:54] <ubitux> ./avconv -i ~/samples/audio/april02.flac -ar 192000 -f null - 15.39s user 0.03s system 99% cpu 15.425 total [19:54] <nevcairiel> They postponed like all API changes [19:54] <ubitux> (not a proper benchmark, alright) [19:54] <jamrial> i know at least one in lavc we have that libav doesn't (and thus didn't postpone) that's needed for on libmpcodecs filter [19:55] <jamrial> *on a [19:55] <ubitux> ./ffmpeg -i ~/samples/audio/april02.flac -ac 6 -f null - 3.90s user 0.11s system 380% cpu 1.054 total [19:55] <ubitux> ./ffmpeg -i ~/samples/audio/april02.flac -ac 1 -f null - 4.19s user 0.11s system 451% cpu 0.952 total [19:55] <ubitux> ./avconv -i ~/samples/audio/april02.flac -ac 6 -f null - 2.78s user 0.03s system 99% cpu 2.806 total [19:55] <ubitux> ./avconv -i ~/samples/audio/april02.flac -ac 1 -f null - 2.69s user 0.01s system 99% cpu 2.709 total [19:55] <nevcairiel> Still some libmpcodecs things left huh [19:56] <ubitux> maybe the difference is not due to swr/avr, someone needs to make proper benchmarks [19:56] <nevcairiel> I'm surprised the mixing would be so different. AFAIK only resampling is more optimized [19:56] <ubitux> but i wouldn't say for sure avr asm is better than swr one [19:56] <ubitux> maybe [19:57] <jamrial> avresample has optimized 6 channel mixing afaik [19:57] <ubitux> proper benchmark should be made [19:57] <jamrial> whereas swr doesn't [19:58] <ubitux> actually, decoding looks faster in ffmpeg, that might be reason [19:59] <nevcairiel> FFmpeg has mt audio decoding for some codecs [19:59] <ubitux> ah, threading :) [20:00] <jamrial> and if that's a 24bits flac, pengvado added x86 asm earlier this year [20:01] <ubitux> no, s16 here [20:01] <nevcairiel> I never got the point of mt audio though, but apparently it can help a bit [20:01] <ubitux> it definitely helps yeah [20:02] <kierank> as usual when i want opus/swr to crash I can't make it crash [20:02] <nevcairiel> I disable it on principle though :P [20:12] <ubitux> ok avconv is faster @ -ac 6 but fails at other mostly (slightly faster at -ac 1 but might be per chance): http://pastie.org/pastes/9458271/text [20:12] <ubitux> i should probably use a longer audio [20:17] <jamrial> you could instead use ffmpeg compiled first with lavr and then swr to make sure differences in decoding are not affecting the benchmarks [20:17] <ubitux> i'm not sure how to test avr properly in ffmpeg [20:18] <ubitux> we don't use the same filters [20:18] <ubitux> ideally some C code should actually be written :p [20:18] <ubitux> anyway that's not a serious benchmark [20:19] <ubitux> the resampling in avr is disturbing though [20:22] <ubitux> mmh, threading dctdnoiz is actually going to be a bit tricky.. [20:29] <cone-860> ffmpeg.git 03Diego Biurrun 07release/0.10:4a6622550a4a: huffyuv: Check and propagate function return values [20:29] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:e17dc0a254ac: mmvideo: check horizontal coordinate too [20:29] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:a1804df66a40: huffyuvdec: check width size for yuv422p [20:29] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:7d42ede8fec4: Merge commit '4a6622550a4a4bf4690ea7d9fe42210a30a67936' into release/0.10 [20:29] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:e719bfc403df: Merge commit 'e17dc0a254ac8d3c33887a114a66e2b659ba0bc5' into release/0.10 [20:29] <cone-860> ffmpeg.git 03Michael Niedermayer 07release/0.10:90241187ceb6: Merge commit 'a1804df66a4064aa30554a11e4fd6cdac3ed89c0' into release/0.10 [20:34] <cone-860> ffmpeg.git 03Clément BSsch 07master:d7594beedee3: avfilter/dctdnoiz: remove a few indirections in idcts [20:41] <cone-860> ffmpeg.git 03Anton Khirnov 07master:4d1ff2a489f4: hevc: calculate the dbf strength in hls_pcm_sample() only if dbf is enabled [20:41] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:adba796eb4a5: Merge commit '4d1ff2a489f4c60501b1a6a2d1f3874e61a77df9' [20:45] <michaelni> if avr has some asm that swr is missing then it should be ported to swr [20:57] <cone-860> ffmpeg.git 03Anton Khirnov 07master:65b8b6c47645: hevc_filter: drop redundant checks [20:57] <cone-860> ffmpeg.git 03Anton Khirnov 07master:550197157857: hevc_filter: drop more redundant checks [20:57] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:ca80c65726db: Merge commit '65b8b6c476454d201348737527a1d9471f689278' [20:57] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:73e9d4cd6fa6: Merge commit '55019715785790836f60870180e1764b06e6591c' [20:58] <ubitux> so, can anyone test & suggest nice freq denoising expression for dctdnoiz? [21:01] <ubitux> currently it's just either a high pass or a custom expression [21:01] <ubitux> if one or two expression appears to be interesting i could add them in a mode so it's faster than calling the eval api [21:06] <michaelni> one could from a series of test images of "signal" and a series of noise images both representative of the actual signal and noise calculate the optimal scalar function for each coefficent [21:08] <cone-860> ffmpeg.git 03Anton Khirnov 07master:70211539a39c: hevc: deobfuscate slice/tile boundary handling for DBF [21:08] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:3f2495d98c26: Merge commit '70211539a39ca3854f8a9e97d51dc27caa079943' [21:31] <cone-860> ffmpeg.git 03Anton Khirnov 07master:a7a17e3f1915: hevc_filter: move some conditions out of loops [21:31] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:8d7c4cc08238: Merge commit 'a7a17e3f1915ce69b787dc58c5d8dba0910fc0a4' [21:45] <ubitux> is the jobnr in the filter_slice callback supposed to be ordered? [21:45] <ubitux> mmh my bad [21:45] <ubitux> stupid question [22:00] <michaelni> kierank, do you provide security support for avr and do maintain it ? [22:01] <michaelni> if not i think you shouldnt suggest people to use it [22:01] <michaelni> because i dont [22:01] <michaelni> and iam not sure who else in ffmpeg does [22:01] <michaelni> if someone does, then thats ok of course [22:04] <michaelni> kierank, also if you have an issue with swr, please tell me more about it so i can fix it [22:10] <kierank> michaelni: daemon404 uses it [22:10] <kierank> For many millions of viewers every day [22:10] <kierank> I can't because the crash entirely smashes the stack frame [22:10] <michaelni> did you try asan ? [22:12] <kierank> No [22:12] <michaelni> also is it known if any & which asm function is responsible ? [22:12] <michaelni> asan might be worth a try [22:13] <kierank> It crashed in emms [22:13] <michaelni> did it work with cpuflags 0 ? [22:14] <kierank> Yes [22:14] <kierank> It was on a live stream using the api [22:14] <jamrial> didn't you say that in the end the crash was in floatdsp and not swr? [22:15] <kierank> Not sure but it was in the opus decoder somewhere [22:15] <kierank> Which uses both [22:15] <michaelni> if it works with cpuflags 0 then one can "bisect" which asm function causes it [22:16] <michaelni> by disabling asm functions [22:16] <michaelni> isnt 100% reliable though if theres a random factor [22:16] <cone-860> ffmpeg.git 03Anton Khirnov 07master:52a2c17ec006: hevc_refs: drop the handling of negative coordinates in ff_hevc_get_ref_list() [22:16] <cone-860> ffmpeg.git 03Anton Khirnov 07master:7acdd3a1275b: hevc_filter: avoid excessive calls to ff_hevc_get_ref_list() [22:17] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:5a3a83f01df0: Merge commit '52a2c17ec006282f388071a831dfb21288611253' [22:17] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:499ff6a05259: Merge commit '7acdd3a1275bcd9cad48f9632169f6bbaeb39d84' [22:17] <jamrial> floatdsp doesn't seem to use mmx at all, only sse or 3dnow. swr uses mmx and calls emms only if you have pre sse2 cpus (or force mmx with cpuflags) and only with 32 bits builds [22:18] <michaelni> kierank, if you suspect its swr, you also could also put a fprintf(stderr before the swr calls in opus and see what is the last printout before it crashes [22:18] <michaelni> maybe that points to something [22:18] <nevcairiel> The emms may be a red herring [22:19] <michaelni> kierank, also enable asserts at level2 [22:20] <michaelni> --assert-level=2 [22:20] <kierank> How do I do that on the api? [22:21] <nevcairiel> Its a build option [22:21] <kierank> Oh [22:21] <kierank> Didn't crash at work but I will try at home in 1hr [22:26] <cone-860> ffmpeg.git 03Anton Khirnov 07master:f4c444e17d13: Postpone API-incompatible changes until the next bump. [22:26] <cone-860> ffmpeg.git 03Michael Niedermayer 07master:3e41d2e61221: Merge commit 'f4c444e17d137c786f0ed2da0e5943df505d5f9e' [22:44] <jamrial> michaelni: will we postpone any of the ffmpeg-only FF_API deprecations? [22:46] <j-b> Anyone heard anything about DxVA breakage lately? [22:48] <j-b> kierank: FYI, libavcodec-ffmpeg plugin is 2MB more than libavcodec-libav plugin for VLC 2.2.0-pre1 [22:55] <Daemon404> j-b, wtf adds 2 mb [22:56] <kierank> Libavcodec only... [22:56] <kierank> Wtd [22:56] <kierank> Wtf [22:56] <ubitux> iirc ffmpeg has something like 100.000 lines of additional code, but that's not lavc specific [22:57] <ubitux> also, http://lucy.pkh.me/diff/diff-codecs.html [22:57] <Daemon404> thats not 2mb [22:57] <michaelni> jamrial, feel free to suggest it on the ML [22:58] <kierank> ubitux: what is up with ac3 fixed end? [22:59] <kierank> Enc? [22:59] <ubitux> dec [22:59] <ubitux> "ENCODER" "ENCDEC" [23:00] <kierank> Yeah hard to read on my phone [23:02] <j-b> Daemon404: I do not know. [23:02] <j-b> Daemon404: I will check if iconv got in, maybe? [23:02] <Daemon404> ah. [23:02] <j-b> kierank: and I will do the usual testing of both :) [23:02] <Daemon404> why would you build wiht iconv if you dont use libav* for subs [23:02] <Daemon404> or lavfi or w/e [23:03] <j-b> Daemon404: exactly, that would be braindead [23:03] <j-b> Daemon404: nope, I pass --disable-iconv [23:03] <Daemon404> o ok [23:03] <Daemon404> beats me [23:03] <j-b> Daemon404: but it could have been the case, because of pkg-config-somethign [23:04] <Daemon404> lol. [23:04] <ubitux> given the number of additionnal codecs we have, it doesn't sounds that surprising to have 2 additionnal MB [23:06] <ubitux> about 8k lines of assembly too, but that might not be that huge on the final size [23:06] <ubitux> (x86) [23:07] <j-b> ubitux: like which codecs? [23:07] <ubitux> 22:57:08 <@ubitux> also, http://lucy.pkh.me/diff/diff-codecs.html [23:07] <j-b> lol [23:07] <j-b> xvmc, vdpau [23:07] <jamrial> hevcdsp_init.o alone is 220kb after stripping [23:07] <jamrial> in libav it's a couple kb at most [23:07] <j-b> because some people refuse to read the hwaccel doc [23:07] <j-b> 3prores decoders [23:08] <ubitux> no [23:08] <j-b> jamrial: that is a better point, indeed. and probably DSD has new tables [23:08] <j-b> jamrial: plus some encoders [23:10] <ubitux> j-b: there are 2 decoders, and 2 encoders [23:10] <nevcairiel> I agree about the hwaccel stuff, but some people just can't let go [23:10] <j-b> ubitux: that too much [23:10] <ubitux> it appears there are 3 encoders, but "prores" is aliased on "prores_aw" [23:11] <nevcairiel> And those people can't even maintain it them self since they are not coders :P [23:11] <ubitux> j-b: licenses & compatibility [23:11] <Daemon404> nevcairiel, we know exactly who "those people" are [23:11] <Daemon404> it's always the same people. [23:11] <Daemon404> for everything [23:11] <nevcairiel> Person in this case [23:11] <Daemon404> :P [23:11] <Daemon404> same source [23:13] <nevcairiel> Speaking of DSD, I need to get a DSD DAC for work. Wonder if there is something not costing $1000 and above [23:14] <Daemon404> lol DSD [23:14] <Daemon404> why wuld you bother [23:14] <nevcairiel> Because users care, and I get paid to care about what the users care about [23:15] <iive> i kind of don't get your hwaccel rant. [23:16] <nevcairiel> j-b: dxva works fine for me still [23:52] <kierank> as usual when i need to debug the crasb the crash goes [00:00] --- Sun Aug 10 2014

1 0

ffmpeg.log.20140809
by burek 10 Aug '14

10 Aug '14

[00:14] <elfer> best dvd author software on linux suggestions? [00:15] <spectralsun> is there any kind of test you can do to determine if a video has been qtfaststarted? [00:23] <sacarasc> spectralsun: Copy the first half of it somewhere and see if it playsd. [00:24] <Mavrik> spectralsun, "qtfaststarted"... a better term is "has MOOV atom at the start of file" [00:24] <Mavrik> spectralsun, which also tells you the answer to your question :) [01:07] <spectralsun> Mavrik: cool, thanks [02:04] <luc4> Hello! I have a variable frame rate mp4 with h264. Can I convert that to fixed frame rate? I know this seems pointless, but it is for development purposes. [03:44] <active8> refresh my memory, please. Was the copy codec ( e.g., -c:a copy) deprecated in favor of -q:a and -q:v ? Maybe I'm just mixed up and remembering that -qscale is deprecated and -q:x is the preferred option; or that -qscale is misunderstood (and I don't remember why [03:45] <relaxed> -q is shorthand for -qscale [03:46] <relaxed> -c:a copy is shorthand for -acodec copy [03:46] <active8> yeah. i think the output of ffmpec to the command line just suggests using -q: [03:47] <Dark-knight> what are you trying to do? [03:48] <active8> oh I just stripped the audio from barney miller and trimed the guitar solo in one shot. just trying to remember things i learned from previous trials [03:48] <active8> good learning. -vn vorbis to mp3 -ss and -t --- one shoit [03:48] <active8> shot [03:49] <active8> it was trimming off a second of applause where i used the copy codec which ... [03:50] <active8> wait. is it needed when just trimming a bit of track from one file to another using the same format - like mp3 or does it decode and encode, thus losing quality. [03:50] <active8> ? [03:52] <relaxed> active8: you can copy the stream or use a lossless codec like flac [03:53] <Dark-knight> does mp3 support flac? [03:54] <active8> but would ffmpeg -i clip.mp3 -ss 00:01 -t 30 trimmed.mp3 cause it to decode/encode? That would force a lossy compression. So I ask if I should use the copy codec. [03:56] <active8> seems to me ffmpeg is smart enough to pick the codec based on the output file extension and would use the codec if I don't -c:a copy [03:57] <relaxed> active8: yes, try ffmpeg -i clip.mp3 -c copy -ss 00:01 -t 30 trimmed.mp3 [03:58] <active8> both work but can't hear the difference. just askin' 8) [04:00] <relaxed> it's lossy unless you use -c copy [04:03] <active8> ok. looks like the output text says "encoder : Lavf55.13.101" for both ways but it's different in other ways [04:03] <active8> Stream mapping: Stream #0:0 -> #0:0 (mp3 -> libmp3lame) [04:03] <active8> vs [04:03] <active8> Stream mapping: Stream #0:0 -> #0:0 (copy) [04:04] <active8> great! I meant to ask about that and almost forgot. thanks for you help [04:11] <Dark-knight> can ffmpeg do everything mp4box can? [04:11] <Dark-knight> http://gpac.wp.mines-telecom.fr/mp4box/ [04:15] <Baked_Cake> well let me just quote the late, great, colonel sanders [04:15] <Baked_Cake> he said: im too drunk, to taste this chicken [04:15] <Baked_Cake> oops wrong chat again [04:15] <Dark-knight> lol [08:12] <Dark-knight> according to this from 2008, ps3 supports ALAC [08:12] <Dark-knight> http://www.avforums.com/threads/flac-and-apple-lossless-alac-on-ps3.778932/ [08:13] <Dark-knight> is this still true? [08:14] <Dark-knight> it seems there are differences in how the alac audio is made that affects the way its played [08:20] <Dark-knight> so what is the difference between [08:20] <Dark-knight> taking a .mkv with FLAC audio and converting it into .m4a with ALAC audio and then renaming the file ext. to .mp4 [08:20] <Dark-knight> and using -f ipod [08:20] <Dark-knight> ffmpeg -i input -c copy -c:a alac -f ipod output.mp4 [08:20] <Dark-knight> ? [08:21] <Dark-knight> renaming the extension vs. using -f ipod [09:06] <Dark-knight> yes, im still here waiting [09:06] <Dark-knight> i can read you mind [09:19] <Dark-knight> seriously what is the difference? [09:19] <Dark-knight> http://pastebin.com/FpBBccLk [09:19] <Dark-knight> http://pastebin.com/27UVCBHM [09:40] <relaxed> Dark-knight: what is your question? [09:40] <Dark-knight> scroll up? [09:41] <relaxed> the -f ipod question? [09:41] <Dark-knight> yes [09:42] <relaxed> -f ipod enables some options in mp4 that are apple centric [09:42] <relaxed> We've been over this. Look at the source if you want to see the difference. [09:43] <relaxed> And since apple created alac, it makes since that it works with -f ipod [09:43] <relaxed> Do you have a real problem you need help with? [09:43] <Dark-knight> i looked at both files using mediainfo and also used ffmpeg to get more info and i honestly can't see what the difference is [09:44] <relaxed> Why do you give a shit? [09:44] <Dark-knight> well i need make sure that i can play this file on the ps3 because some instances of ALAC dont work on ps3 [09:45] <relaxed> maybe the ps3 only supports 2 channel alac, and not 5.1 [09:52] <relaxed> I'm sure all these details are on the interweb [09:52] <Dark-knight> i looked [09:52] <Dark-knight> that why im asking here [09:52] <Dark-knight> i've learned to google before asking dumb questions [09:54] <Dark-knight> some people have converted flac to alac themselves but it wouldn't play on their ps3, but when they used itunes to convert the flac to alac, it worked [09:54] <Dark-knight> what i want to know is, why. and if that is that same as rename vs -f ipod [10:03] <relaxed> make small some samples and see what's what [10:03] <relaxed> some small* [10:06] <Dark-knight> i already poked around with those 2 i converted already, i dont know where else to look inside of them? [10:07] <sfan5> you could convert two files [10:07] <sfan5> one with -f ipod, one without [10:07] <sfan5> and compare them in a hex editor [10:07] <Dark-knight> -_- i all ready did [10:07] <Dark-knight> oh hex editor [10:08] <Dark-knight> and what would i look for? [10:08] <sfan5> everything that differs [10:08] <Dark-knight> ok [10:08] <Dark-knight> got a link for hex editor? [10:08] <sfan5> which OS? [10:08] <Dark-knight> windows [10:09] <sfan5> nope [10:09] <sfan5> google is your fried [10:09] <sfan5> friend* [10:14] <Dark-knight> which one is best? [10:14] <Dark-knight> http://en.wikipedia.org/wiki/Comparison_of_hex_editors [11:28] <luc4> Hello! I have an h264 variable frame rate stream in an mp4 file. Can I convert it to a fixed frame rate? [11:29] <sacarasc> You can, but it might look jerky. [11:30] <sacarasc> -r sets the frame rate and will make it fixed. [11:32] <luc4> sacarasc: tried that& but it doesnt seem to work& [11:33] <relaxed> try -vsync 1 [11:33] <luc4> I mean that the fps is written in the container probably, ffprobe sees it. But if I set something like 20 or 40 I see no difference. [11:34] <luc4> relaxed: you mean in addition? Or should I replace it? [11:39] <relaxed> use it alonf with -r, man ffmpeg| less +/' -vsync' [11:39] <relaxed> along* [11:40] <relaxed> you are trying to re-encode the stream, correct? [11:46] <luc4> relaxed: yes, I suppose that is necessary to change the frame rate. [11:50] <luc4> relaxed: I tried 40fps but I see no difference& [11:50] <relaxed> do you want to see a difference? [11:50] <luc4> relaxed: yes, maybe ffmpeg replicates the frames? [11:50] <relaxed> look at ffmpeg's console output for dup frames [11:51] <luc4> dup=3493 [11:51] <luc4> so I suppose yes [11:52] <luc4> Any way to avoid this? I would like to see the frames one after the other with fixed frequency. [11:52] <luc4> I know it makes no sense usually, but it is for development purposes. [11:55] <relaxed> I'm not sure how to do that. You could output all the frames to png(s) anf then use them as input using an arbitrary freame rate. [11:57] <luc4> relaxed: ah ok, Ill try that, thanks. [11:58] <relaxed> luc4: wait, I believe you can force decoding at a specific frame rate. [11:59] <relaxed> ffmpeg -r $whatver -i input [12:00] <relaxed> from the fine manual, "As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps." [12:02] <relaxed> luc4: try that with stream copying [12:03] <luc4> relaxed: Ill try this! Thanks! [12:27] <Dark-knight> not a SINGLE hex editor for windows 8 [12:27] <Dark-knight> F*** [12:28] <Dark-knight> yeah im done looking for tonight [12:43] <viperfx_> Hi all. I have posted a question on SO regarding compilation flags. If someone here could take a look and offer some help, that would be great. https://stackoverflow.com/questions/25217628/detecting-the-needed-decoder-m… [12:48] <relaxed> viperfx_: -enable-demuxer=matroska --enable-decoder=vorbis [12:49] <viperfx_> Would you be able to help me understand what the purpose of demuxer/decode etc is. Or point me to a link? [12:49] <viperfx_> Thanks, I will try those flags. [13:03] <viperfx_> It works! Thanks relaxed [13:05] <relaxed> viperfx_: you're welcome [13:25] <viperfx_> relaxed: What about this one? http://pastebin.com/RrYn6We7 [14:11] <viperfx_> Is there any way to improve the speed of the av_open_input_file() call for an HTTP stream [14:12] <viperfx_> Does FFmpeg need to read all of it before it can proceed? [14:22] <viperfx_> *avformat_open_input() [14:55] <anshul_mahe> How to set thread_count in avcodec_decode_video2 for libavcodec? [14:55] <anshul_mahe> I am using libav library [15:07] <anshul_mahe> I want to debug some part of code, if i put scheduler locking on in gdb then all threads hang to pthread_cond_wait, looks like deadlock cant exactly say since not verified all condition [15:09] <anshul_mahe> so I am searching for some av_dict_set to change thread_count [15:10] <anshul_mahe> !log [15:10] <anshul_mahe> log! [15:17] <anshul_mahe> where is ffmpeg-irc publicly logged? [15:18] <Chaz6> /dev/null [15:19] <anshul_mahe> ok, Chaz6 [15:20] <anshul_mahe> ohh i was missing 's' in !logs [15:21] <anshul_mahe> thanks, sacarasc [16:28] <phelps> how do arguments work in the expansion for a filter [16:28] <phelps> %{pts}, how do I add hms as an argument [16:28] <phelps> "The text between the braces is a function name, possibly followed by arguments separated by :." [16:29] <phelps> %{pts} %{pts:hms} %{pts hms} %{pts}:hms etc.. none work [16:58] <fajung> I'm trying to convert an avi/mp3 to mp4/aac, but I get this error:"Unknown encoder 'libfaac'". I'm on ubuntu 14.04, ffmpeg version 2.3 [17:00] <Mavrik> fajung, your ffmpeg isn't built with libfaac support. [17:00] <fajung> how can I do to make it support it ? [17:03] <sacarasc> fajung: You'd have to compile yourself, but if you do that, you might as well use libfdk_aac instead, because it's a lot better than libfaac. (Just a moment, getting a link.) [17:03] <sacarasc> https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu to compile. [17:03] <sacarasc> https://trac.ffmpeg.org/wiki/Encode/AAC about AAC encoders. [17:03] <Mavrik> fajung, you either use another AAC encoder (since libfaac is utter shit) [17:04] <Mavrik> fajung, or you recompile your ffmpeg with libfaac support [17:04] <Mavrik> if you're using ubuntu and you typed "apt-get install ffmpeg" you don't have ffmpeg installed anywa [17:04] <Mavrik> y [17:04] <fajung> ok, i''m going to try to compile it [17:05] <Mavrik> fajung, take fdk-aac instead of libfaac then. [17:05] <sacarasc> Yes, libfdk_aac is much better. [18:49] <fajung> now i'm getting this error: Unknown encoder 'libfdk-aac' [18:52] <fajung> ~/bin$ ./ffmpeg -i ./input.avi -acodec libfdk-aac -b:a 128k -vcodec mpeg4 -b:v 978k -flags +aic+mv4 out.mp4 [19:04] <nicholaswyoung> fajung: it's libfdk_aac [19:05] <nicholaswyoung> assuming you've installed libfdk, anyway [19:05] <fajung> I'm doing all the guide again to see where I did wrong [19:08] <nicholaswyoung> fajung: what OS are you using? [19:09] <fajung> ubuntu 14.04 [19:09] <fajung> I'm doing this: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [19:09] <fajung> trying... [19:10] <fajung> in the ffmpeg step, I have to add the --enable-libfdk_aac & --enable-libfdk-aac right? [19:11] <nicholaswyoung> looking at that page now, to see if it's current [19:12] <c_14> Just one of the two iirc. [19:12] <c_14> Just --enable-libfdk-aac [19:12] <nicholaswyoung> it looks like --enable-libfdk_aac is the correct formation: https://trac.ffmpeg.org/wiki/Encode/AAC [19:13] <c_14> Not according to ./configure --help [19:13] <nicholaswyoung> you might have to --enable-nonfree too [19:13] <fajung> --enable-libfdk-aac --enable-libfdk_aac --enable-nonfree [19:13] <fajung> just in case [19:14] <fajung> and gpl [19:16] <fajung> I got this: ERROR: libfdk_aac not found [19:19] <c_14> Are you sure you're running the ffmpeg that you just compiled and that it's loading the correct libraries? [19:20] <fajung> I deleted I've to recompile it again [20:13] <fajung> now it is working, I dont't know how or why, but it is working with lib_aac [20:13] <fajung> libfdk_aac [00:00] --- Sun Aug 10 2014

1 0

ffmpeg-devel.log.20140808
by burek 09 Aug '14

09 Aug '14

[00:18] <Zeranoe> To whoever decided to allow wildcards in --enable and --disable options: Thank you, you just saved me a lot of time. [00:20] <wm4> Author: Måns Rullgård <mans(a)mansr.com> 2009-08-06 17:45:22 [00:20] <wm4> that guy added it [01:54] <cone-128> ffmpeg.git 03James Almer 07master:5c3c67126feb: lavf/avio: rename url_feof() to avio_feof() [01:54] <cone-128> ffmpeg.git 03James Almer 07master:d34ec64a2290: replace calls to url_feof() with avio_feof() [02:56] <Compn> ubitux : lwn still going? one reason i had to stop talking to diego [07:17] <ubitux> wm4: you need to -show_streams to have the summary of the frame counting [07:18] <ubitux> Compn: dunno, 4 more messages since last time i checked [07:23] <ubitux> rcombs: yeah, the subtitle and data support in libavfilter is something we considered since years [07:23] <ubitux> Nicolas (cigaes) mentioned the data part several time, and he might have submitted a PoC already somehow [07:23] <rcombs> ubitux: have there been tickets tracking them that I'm duplicating? [07:23] <ubitux> i was looking for it but i think it's mostly on the ml [07:24] <ubitux> still, there is https://trac.ffmpeg.org/ticket/1305 and https://trac.ffmpeg.org/ticket/2067 [07:24] <rcombs> I looked around a bit, but it's kinda hard to search for, and I didn't turn anything up immediately [07:25] <ubitux> note that adding subtitles support in libavfilter means moving the AVSubtitle & friends to libavutil [07:25] <rcombs> shouldn't that second one be closed? [07:25] <ubitux> #2067? [07:26] <rcombs> yeah; there's a parameter in the filter to seek now, isn't there? [07:26] <ubitux> well, it's kind of the "Support subtitles in libavfilter" ticket ;) [07:26] <rcombs> or am I just thinking of Plex's [07:27] <ubitux> no seek is not supported, and even if it was it would be a hack [07:27] <ubitux> we need proper subtitles support [07:27] <rcombs> yeah, agreed [07:28] <rcombs> cehoyos seems to agree as well [07:28] <ubitux> but yeah again we need to move AVSubtitles API to libavutil, and this is tricky because we wanted to take the opportunity to review the structure itself [07:28] <ubitux> (of AVSubtitles, rects, etc) [07:29] <wm4> ehhhh [07:29] <ubitux> like question such as; do we need to support other pixel formats for rgb bitmap subs [07:29] <wm4> I'm pretty skeptic against subtitles in lavfi [07:29] <ubitux> or how are we supposed to store text subs, etc [07:29] <rcombs> I'd like to see AVSubtitles stop including the timestamps in the text :3 [07:29] <ubitux> rcombs: i "fixed" that, somehow [07:30] <ubitux> at least for matroska [07:30] <wm4> you know, most things in audio/video can be modeled as uni-directional data flow in a graph [07:30] <wm4> and lavfi looks like it's doing that [07:30] <wm4> but lavfi is way too closely coupled and special-cased to what it's supposed to be to be useful as generic framework for such things [07:31] <wm4> it's also too inflexible [07:31] <rcombs> wm4: "input frames -> output frames", with some initialization and configuration functions, basically [07:32] <wm4> it would be nice if we had a filter API that doesn't want to make you kill yourself, and that also allows exteral filters [07:32] <wm4> the "killing yourself" part is mostly about format negotiation [07:33] <wm4> and I think if we hack subs into lavfi in its current state, we will be very sad in a few years [07:34] <ubitux> except that hardcoding will not require to specify files in the filtergraph [07:35] <ubitux> and we'll be able to have one single blending process, in overlay [07:35] <wm4> if you want to solve that in the context of ffmpeg.c, you might as well add a dirty hack to build an appropriate filter graph with the needed options [07:35] <ubitux> i want it in ffplay too ;) [07:35] <rcombs> wm4: whaddayaknow, that's how Plex did it [07:35] <wm4> ubitux: nobody cares about ffplay [07:36] <ubitux> like, i want to be able to hardsub the subtitles of my mkv which specifying the mkv twice, and have a weird caching behaviour [07:36] <ubitux> wm4: well, ffplay == any similar user app [07:36] <wm4> not really [07:36] <rcombs> yeah, not really [07:37] <rcombs> mpv, for instance, (at default settings) renders subtitles internally at display resolution and has OpenGL handle the overlaying [07:37] <wm4> anyway, I really like the idea of a filter API that 1. allows external filters, and 2. allows much more things in the data flow than audio and video [07:37] <rcombs> wm4: what stops lavfi from allowing external filters? [07:38] <wm4> its API [07:38] <rcombs> how so? [07:38] <wm4> it even allowed external filters once, but it was so horrible and messy that it was made private [07:38] <ubitux> libav deprecated the public symbols to ease internal changes [07:38] <wm4> I mean, compare it with something like vapoursynth [07:38] <ubitux> and never made them public again [07:39] <wm4> anyway, if you want external apps to heavily rely on a ffmpeg filter graph thing, please don't make it lavfi, but something more flexible [07:39] <rcombs> I might be thinking of something different than you when I think of "external filters" [07:39] <wm4> lavfi is restricted to ffmpeg filters, ffmpeg formats, audio/video only (even if you add subs later) [07:40] Action: ubitux still wants proper subtitles support in ffplay [07:40] <rcombs> ah, I was thinking of a hypothetical lavfi filter that passed input video through a vapoursynth script, or something similar [07:40] <rcombs> ubitux: how does ffplay do scaling for display? [07:40] <ubitux> badly [07:40] <wm4> rcombs: that's a use-case, although that filter could just be part of lavfi [07:41] <wm4> rcombs: it uses SDL [07:41] <ubitux> not even sdl2 btw, which might fix a few problems we have with it [07:41] <wm4> "problems" [07:41] <ubitux> especially since sdl2 has a possible gl backend, right? [07:42] <wm4> well they still have their own retarded video texture formats [07:42] <wm4> hardcoded to bt.601 or whatever [07:42] Action: rcombs points at FFDraw [07:43] <ubitux> that's what we use to blend subtitles currently [07:43] <rcombs> ubitux: yes [07:43] <wm4> so what I envision is a ffmpeg-style directshow/gstreamer-like interface, except that it OF COURSE will not have the failure of those two [07:43] <ubitux> but not in overlay; and i'd love to use overlay instead [07:43] <wm4> or maybe mentioning these is not very constructive [07:43] <rcombs> and last I checked, it converts RBG subtitle colors to YUV using BT.601, regardless of the video's color space [07:44] <wm4> ubitux: wth is the point? [07:44] <rcombs> *RGB [07:44] <wm4> ubitux: overlay would just require blending the whole video surface, and it'd always force you to do video resolution rendering [07:44] <ubitux> wm4: ffmpeg -i in.mkv -vf '[0:0][0:2]subtitles' [07:44] <wm4> not to mention that vf_overlay format negotiation is fundamentally broken [07:44] <wm4> uh [07:44] <wm4> wqhat does that mean [07:44] <ubitux> wm4: ffmpeg -i in.mkv -vf '[0:0][0:2]overlay' [07:44] <ubitux> sorry [07:45] <wm4> I see horrible syntax [07:45] <ubitux> with [0:0] video stream, and [0:2] sub stream [07:45] <rcombs> ubitux: so, you want the sub2video hack to support text subtitles? [07:45] <ubitux> rcombs: exactly, but not as a hack [07:45] <wm4> basing everything on lavfi will lead to misery [07:45] <rcombs> ubitux: I'd rather that be in the "subtitles" filter tbh [07:46] <rcombs> rather than making overlay into a hydra [07:46] <ubitux> dunno [07:46] <wm4> ubitux: the concept sounds nice, but the result will be not so good [07:47] <ubitux> -vf rasterize @_@ [07:47] <wm4> mpv for example _pulls_ the subtitles from the subtitle renderer, as a list of surfaces that have to be blended [07:47] <wm4> not pushing a giant alpha surface [07:48] <ubitux> interesting [07:49] <ubitux> but what do you suggestion to avoid -vf subtitles=/home/foo/bar/Sub... ? [07:49] <ubitux> s/suggestion/suggest/ [07:49] <wm4> make ffmpeg/ffplay insert that filter? [07:49] <wm4> *groan* [07:49] <ubitux> haha, yeah right [07:50] <wm4> I'm not very fond of all these hacks that are added to lavfi and lavf to work around ffmpeg deficiencies [07:50] <wm4> like the concat demuxer... [07:50] <ubitux> but that's just hiding shit under the carpet [07:50] <wm4> isn't that what ffmpeg.c is best at [07:50] <ubitux> o u [07:50] <ubitux> :D [07:50] <rcombs> ubitux: I'd suggest having the subtitles filter take a subtitles track as input, as you're suggesting, but just leaving the drawing internal rather than converting everything to full frame-sized bitmaps [07:50] <wm4> seriously [07:51] <ubitux> rcombs: alright [07:51] <ubitux> so we agree that we need subtitles in lavfi anyway [07:51] <rcombs> well, as wm4 says, I don't know if that'd be sane with the current API [07:51] <ubitux> it will solve a bunch of things [07:51] <wm4> I'm arguing that adding it to lavfi will lead to failure eventually [07:51] <wm4> and there should be a light-weight filter API instead [07:52] <ubitux> like ffmpeg -i bla.avi -ss 12.34 -i bla.srt -vf subtitles ... [07:52] <wm4> that isn't as insanely closely-coupled and closed as lavfi [07:52] <ubitux> (s/subtitles/hardsub/ if you prefer) [07:53] <wm4> something that flexible could perhaps also include demuxers, decoders, bitstream filters, and such things [07:54] <ubitux> btw, somehow related to light-weighting, i'm actually secretly working on a python binding, just to try if it's do-able [07:55] <ubitux> you'll probably hate me for what i'll submit as a PoC but well.. [07:56] <wm4> probably [07:57] <wm4> the current APIs don't allow for great things IMO [07:57] <wm4> e.g. lavfi would need to be turned inside-out before it could have the flexibility of avisynth [07:57] <wm4> because lavfi wants to do its own thing once the graph is setup [11:32] <ubitux> so, hashes are highlighted in the trac, but "Repository '' not found" [12:26] <ubitux> btw, we need to find a solution to test filters using floats [12:29] <wm4> ubitux: what's hard about it? [12:31] <ubitux> thresholding on the resulting frames [12:31] <ubitux> i suppose [12:31] <wm4> on that matter, I'm wondering how libass should be tested [12:32] <wm4> because libass sure needs a testsuite [12:32] <wm4> but there the problem is way worse [12:32] <ubitux> you need a "bitexact" mode in freetype i suppose [12:33] <ubitux> and force a freetype version [12:33] <wm4> I'm not even sure if libass still uses freetype for rasterization [12:33] <wm4> I think not [12:34] <wm4> and how do you get bitexact fontconfig ;) [12:34] <wm4> anyway, most calculations are done with floats [12:47] <ubitux> i wonder if we could add a simple bitmap font in ffmpeg to test the drawtext filter [13:08] <ubitux> so we don't have any filter test making of this fuzz thing? [13:28] <cone-822> ffmpeg.git 03Justin Ruggles 07release/1.1:7997acee0542: Check if an mp3 header is using a reserved sample rate. [13:28] <cone-822> ffmpeg.git 03Justin Ruggles 07release/1.1:d7dbc687e312: Check mp3 header before calling avpriv_mpegaudio_decode_header(). [13:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:244a58fff0ad: Merge commit '7997acee0542f6e0bb9ea42ff783f80b70878a2f' into release/1.1 [13:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:cf7f798984a9: Merge commit 'd7dbc687e312a91ef2ccf797d57b95c61d0e8a2f' into release/1.1 [13:46] <cone-822> ffmpeg.git 03Martin Storsjö 07release/1.1:744e7eea5d81: adpcm: Avoid reading out of bounds in the IMA QT trellis encoder [13:46] <cone-822> ffmpeg.git 03Martin Storsjö 07release/1.1:21d3e0ac9e17: adpcm: Write the proper predictor in trellis mode in IMA QT [13:46] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:5fa56e6e6239: Merge commit '744e7eea5d815efea777b6179d96e8d94b63ccfa' into release/1.1 [13:46] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:6333c6c17d1f: Merge commit '21d3e0ac9e1719d8444b3f5466983587ac0ad240' into release/1.1 [13:59] <cone-822> ffmpeg.git 03Reinhard Tartler 07release/1.1:5e8eaa26b227: Prepare for 9.14 Release [13:59] <cone-822> ffmpeg.git 03Reinhard Tartler 07release/1.1:3ecbd911ff91: Update Changelog for v9.14 [13:59] <cone-822> ffmpeg.git 03Anton Khirnov 07release/1.1:e8ff79720646: eamad: use the bytestream2 API instead of AV_RL [13:59] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:53c3abc1083d: Merge commit '3ecbd911ff9177097820e5d00401c9bf29e5d167' into release/1.1 [14:00] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:c074feed292e: Merge commit 'e8ff7972064631afbdf240ec6bfd9dec30cf2ce8' into release/1.1 [14:18] <cone-822> ffmpeg.git 03Diego Biurrun 07release/1.1:3a6bc3e38164: vf_select: Drop a debug av_log with an unchecked double to enum conversion [14:18] <cone-822> ffmpeg.git 03Bernhard Übelacker 07release/1.1:d16515ae5fe7: video4linux2: Avoid a floating point exception [14:18] <cone-822> ffmpeg.git 03Diego Biurrun 07release/1.1:58d7b835e3ce: fate: Add dependencies for dct/fft/mdct/rdft tests [14:18] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:eaf64192d832: Merge commit '3a6bc3e381647bb4434317113f131f7e0ab5bf83' into release/1.1 [14:19] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:a4de70df20f0: Merge commit 'd16515ae5fe7daa6327d903cafb9a5ee43477b1e' into release/1.1 [14:19] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:6e83c266201a: Merge commit '58d7b835e3cec48ab5a2393405fe82dee72c06a0' into release/1.1 [14:30] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:00915d3cd2ce: pgssubdec: Check RLE size before copying [14:30] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:01f954032027: h264_sei: check SEI size [14:30] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:1ee5e2ce3d6d: Merge commit '00915d3cd2ce61db3d6dc11f63566630a9aff4ec' into release/1.1 [14:30] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:0bcf514198bf: Merge commit '01f9540320279954b2764645ab7136847d53d89f' into release/1.1 [14:43] <cone-822> ffmpeg.git 03Vittorio Giovara 07release/1.1:512354191328: h264: prevent theoretical infinite loop in SEI parsing [14:43] <cone-822> ffmpeg.git 03Diego Biurrun 07release/1.1:43d676432740: huffyuv: Check and propagate function return values [14:43] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:ae7ea2eabf35: Merge commit '512354191328c559fcff56070dab897ee2a1b4c1' into release/1.1 [14:43] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:451bc8ee2fce: Merge commit '43d676432740c6d5e5234ed343f13902909fd124' into release/1.1 [15:02] <cone-822> ffmpeg.git 03Luca Barbato 07release/1.1:146b187113e3: lavc: Check the image size before calling get_buffer [15:03] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:36d8914f1b94: wmalosslessdec: fix mclms_coeffs* array size [15:03] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:02018a359e05: Merge commit '146b187113e3cc20c2a97c5f264da13e701ca247' into release/1.1 [15:03] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:a88236f3d59c: Merge commit '36d8914f1b94e4731d2fc67162902839c106e72e' into release/1.1 [15:42] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:ede738880032: mmvideo: check horizontal coordinate too [15:42] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:c53effc41b93: huffyuvdec: check width size for yuv422p [15:42] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:0d6ed2f13faf: Merge commit 'ede738880032db62b7dc5b3712f769d3826f5974' into release/1.1 [15:42] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:91437631d76f: Merge commit 'c53effc41b9359261b17c8da3b7062369cafd686' into release/1.1 [16:01] <cone-822> ffmpeg.git 03Anton Khirnov 07release/1.1:8cd67ddde46a: cdgraphics: switch to bytestream2 [16:01] <cone-822> ffmpeg.git 03Anton Khirnov 07release/1.1:80c268eaaee4: cdgraphics: do not return 0 from the decode function [16:01] <cone-822> ffmpeg.git 03Anton Khirnov 07release/1.1:af9b62654d5a: svq1: do not modify the input packet [16:01] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:420f63984b39: Merge commit '8cd67ddde46a42a33149e7d42a2ab47852ff2a83' into release/1.1 [16:01] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:06f7e87e15a6: Merge commit '80c268eaaee402695a74d14acf76063100692a99' into release/1.1 [16:01] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:e4b1cffadef2: Merge commit 'af9b62654d5aa023a96906215365532d18541a09' into release/1.1 [16:02] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:09e3fe79fc09: avcodec/svq1dec: Fix multiple bugs from "svq1: do not modify the input packet" [16:16] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:52254067b312: error_concealment: avoid using the picture if not fully setup [16:16] <cone-822> ffmpeg.git 03Reinhard Tartler 07release/1.1:ecda9b90eccc: Update Changelog for v9.15 [16:17] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:b65c290f7fec: Merge commit '52254067b312e78d30bbe79fc33dbdf995b22b4e' into release/1.1 [16:17] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:b52952c6e96b: Merge commit 'ecda9b90eccc687202fe9fa20f7ca61d92d816b4' into release/1.1 [16:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:fe461238d33e: avformat/utils: do not wait for packets from discarded streams for genpts [16:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:e5fcc16a1fa7: avcodec/dvdsub_parser: never return 0 when the input isnt 0 [16:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:1298aa83180e: avcodec/dvdsub_parser: Check buf_size before reading 32bit packet size [16:28] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:a04bb8d6e7f7: avcodec/dvdsub_parser: print message if packet is smaller than the packet size field [16:30] <michaelni> Timothy_Gu, do you want to write a changelog/release notes for 1.1.13 ? [16:30] <michaelni> ill make the 1.1.13 release later today or tomorrow depening on when ive time [17:23] <cone-822> ffmpeg.git 03Carl Eugen Hoyos 07master:e4d983e2db0e: configure: Check if libwebp compilation will succeed. [17:23] <cone-822> ffmpeg.git 03Carl Eugen Hoyos 07master:f73d75384f1d: Autodetect webp files. [18:42] <cone-822> ffmpeg.git 03Marek Fort 07master:d1e750cd512a: avformat/mov: Support reading Avid's metadata for DNXHD codec. [19:00] <cone-822> ffmpeg.git 03Muhammad Faiz 07master:94494dab9101: avfilter/showcqt: add and extend tlength and volume options [20:00] <cone-822> ffmpeg.git 03Clément BSsch 07master:aaf82dc0fa47: avfilter/dctdnoiz: make color [de]correlation less clumsy [20:00] <cone-822> ffmpeg.git 03Clément BSsch 07master:eb16a6d2294c: avfilter/dctdnoiz: remove forward declarations after previous commit [20:00] <cone-822> ffmpeg.git 03Clément BSsch 07master:1ba7c6ead24f: avfilter/dctdnoiz: use 32-bit (float) operations instead of 64 (double) for DCTs [20:10] <ubitux> eval api is not thread safe, right? [20:19] <michaelni> ubitux, iam not sure i understand, what can be (not) thread safe on it ? [20:20] <ubitux> calling av_eval_expr() on the same context from different thread [20:20] <ubitux> i think there might be concurrent access in state variables or something [20:20] <ubitux> but maybe i'm wrong? [20:22] <michaelni> dunno, id have to RTFS [20:23] <michaelni> is it a problem to use seperate contexts ? [20:43] <ubitux> michaelni: yeah i thought of this but don't remember why i dismissed it [20:44] <ubitux> i think the idea that there was a state to keep between calls was the reason, but it's actually wrong when i think again [21:17] <cone-822> ffmpeg.git 03Justin Jacobs 07master:87dc8b3af913: avformat/matroskadec: Check avpriv_new_chapter() for failure [21:37] <cone-822> ffmpeg.git 03Diego Biurrun 07master:454697603e4e: mpegts: Use av_free() to free memory allocated by av_strdup() [21:37] <cone-822> ffmpeg.git 03Luca Barbato 07master:e4c9e59a4547: mpeg: K&R formatting cosmetics [21:37] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:fd1e4d66f183: Merge commit '454697603e4efdfc04fadec40518d56c7dc1e5dd' [21:37] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:a698efbc9df3: Merge commit 'e4c9e59a4547adaaa0ce9f25b0d0c5b91ae15472' [21:45] <cone-822> ffmpeg.git 03Diego Biurrun 07master:0026e356d044: configure: Die if gas is unavailable under aarch64 as well as ARM [21:45] <cone-822> ffmpeg.git 03Diego Biurrun 07master:444c73583d28: configure: Only run gas checks on ARM and PowerPC [21:45] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:0472c5f8a1f1: Merge commit '0026e356d044e72b6e743b234708b8b8af457ac0' [21:46] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:eb2def0ff2c4: Merge commit '444c73583d2848a542330c03949e1f933ac68f53' [22:00] <cone-822> ffmpeg.git 03Diego Biurrun 07master:6f1960ab71b4: idct: cosmetics: Drop one unnecessary if-block level [22:01] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:5ff2b33401ec: Merge commit '6f1960ab71b4f18551243ce22d01913108265233' [22:01] <Timothy_Gu> michaelni: not really. I'm not really interested in such an old branch. [22:14] <michaelni> Timothy_Gu, ok [22:26] <cone-822> ffmpeg.git 03Diego Biurrun 07master:84d173d3de97: xvididct: Ensure that the scantable permutation is always set correctly [22:26] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:0dcebb9f6345: Merge commit '84d173d3de97c753234ab0c0b50551d51413d663' [23:01] <cone-822> ffmpeg.git 03Michael Niedermayer 07release/1.1:41e859151543: update for 1.1.13 [23:53] <cone-822> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.1.13': unknown revision or path not in the working tree. [23:53] <cone-822> Use '--' to separate paths from revisions [23:53] <cone-822> refs/tags/n1.1.13:HEAD: Merge commit '84d173d3de97c753234ab0c0b50551d51413d663' [00:00] --- Sat Aug 9 2014

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ffmpeg.log.20140808
by burek 09 Aug '14

09 Aug '14

[00:00] <phelps> jrgill -c:a copy [00:01] <phelps> should tell it to copy the audio codec and not transcode [00:04] <jrgill> phelps, it seems to conflict, which was why I tried -c:v. "Filtergraph 'pan=stereo:c0=c0' was defined for audio output stream 0:1 but codec copy was selected." [00:05] <phelps> jrgill: hmm, dunno then, I've never used filters [00:06] <jrgill> Ultimately trying to invert that right channel. I think the only way is to multiply by -1 in pan audio. [00:07] <jrgill> Will just try some other ways. Was hoping it could be simple. :/ [00:19] <llogan> jrgill: filters require encoding [00:40] <jrgill> llogan, same with map_channel? With -c copy it just ignores my mapping while -c:v copy transcodes audio. ffmpeg -i in.m4v -map_channel 0.1.0 -map_channel -1 -c copy out.m4v [01:02] <jrgill> llogan, http://privatepaste.com/download/590d7e8605 [01:08] <jrgill> Guess I could map the channels in QuickTime and use ffmpeg to switch from mov to m4v. [01:40] <jrgill> Thinking maybe there's a way to use channel_layout to do the equivalent of QuickTime assigning the channels. Any suggestions? [03:01] <Dark-knight> Zeranoe: hey man [03:01] <Dark-knight> Got some question to ask you? [03:02] <Zeranoe> Whats up? [03:02] <Dark-knight> just a sec [03:12] <Dark-knight> I copy'd these from my notes. [03:12] <Dark-knight> I was wondering if you could add the ALAC tag to the .mp4 container and add support for setting the default stream/track for a container? [03:13] <Dark-knight> ...to the next build [03:17] <Zeranoe> I try to avoid making any modifications to the source code for FFmpeg, so if it's something that requires patching the source it would be best to submit a feature request ticket [03:20] <Dark-knight> those two things i suggested would be very beneficial to people that need a solution to certain problems. [03:21] <Dark-knight> you might have more pull in the community and your request tickets might be taken more seriously. [03:25] <Zeranoe> Are there already tickets out for those issues? [03:25] <Dark-knight> as it stands right now. support for setting the default stream is very high on my want list, and Im sure that it would benefit many people who needed the feature, (as i have seen on google) [03:26] <Dark-knight> if i had to choose between the 2, i would choose the feature to change the default stream [03:30] <Zeranoe> Dark-knight: Where does that apply? [03:32] <relaxed> Dark-knight: did you try -atag ALAC [03:34] <Dark-knight> what do you mean? [03:35] <relaxed> what did you mean by "add the ALAC tag to the .mp4 container"? [03:37] <Dark-knight> FLAC audio doesn't like the .mp4 container. so I converted the .mkv container to .m4a and the FLAC audio to ALAC. Then I changed the name from .m4a to .mp4 [03:37] <Dark-knight> What im suggesting is, adding the alac tag to .mp4, so making all those steps pointless. [03:38] <relaxed> .m4a and .mp4 are the same thing [03:38] <Zeranoe> (except one is audio only) [03:44] <Dark-knight> it would be a simple thing to change [03:45] <Dark-knight> but im more concerned about the default stream. [03:46] <relaxed> what kind of stream? [03:51] <relaxed> you can use the .mov container with alac [03:51] <Dark-knight> changing the default audio or subtitle stream [03:52] <relaxed> The default audio stream probablu defaults to the first stream [03:52] <relaxed> probably* [03:53] <Dark-knight> it doesn't always [03:53] <Dark-knight> which is why manual change is necessary [03:56] <relaxed> do you know of another tool that can do these things? Like mp4box, l-smash, or something? [04:02] <relaxed> Dark-knight: ffmpeg -i input -c:a alac -f ipod output.mp4 [04:04] <Dark-knight> i heard rumors that mp4box could change the default stream, but i found evidence to the contrary. [04:05] <Dark-knight> I believe the flag that tells the container which stream is the default is here [04:05] <Dark-knight> http://matroska.org/technical/specs/index.html Down by "Track" it's called "FlagDefault". [04:05] <relaxed> matroska is not mp4 [04:05] <Dark-knight> i know this [04:05] <relaxed> maybe it depends on the player [04:05] <Dark-knight> is should be the same [04:06] <relaxed> anyway, the command above sovles one your problems [04:06] <Dark-knight> interesting [04:06] <Dark-knight> does it have any loss in quaility? setting -f ipod? [04:07] <relaxed> it controls the container [04:07] <Dark-knight> or is that the same as changing the name from .m4a to .mp4 [04:13] <relaxed> it probably sets some apple centric things, since they insist on doing things their own way. [04:13] <Dark-knight> so is it the same a a name change? [04:13] <relaxed> No [04:14] <relaxed> but it should still work on whatever console you're targeting [04:14] <Dark-knight> how so? [04:14] <Dark-knight> thanks [04:15] <Dark-knight> now that, that is out of the way. how about we work on that default flag? [04:16] <relaxed> go read the mp4 spec, find out what has to be done or if it's even possible, and then file a bug report. [04:16] <relaxed> supply a sample of an mp4 with these features too [04:17] <relaxed> first make sure there's not already a feature request for it. [04:18] <Dark-knight> where do i check? [04:18] <relaxed> google [04:18] <relaxed> and the ffmpeg bug tracker [05:26] <jjohn> Ahoy, guys. Can you help me out with two little questions that I got for you? [05:26] <jjohn> I wrote it down here: http://pastebin.com/9PMsZ908 [05:29] <relaxed> jjohn: flac is lossless and it well supported, but the output will be larger than the original audio stream. [05:30] <relaxed> it is* [05:30] <jjohn> relaxed: It's not only flac, it's actually a wild mix of everything. And I basically just want to copy the audio stream. [05:32] <relaxed> right, but wouldn't it be nice to have it all the same format for consistency? [05:33] <jjohn> Actually, preserving the original audio quality is even more important than consistancy. :) [05:33] <jjohn> consistency* [05:33] <relaxed> you can achieve both by encoding everything to flac [05:33] <jjohn> Yeah, well, I once tried that out as well. [05:34] <jjohn> Turned a 200 MB video into a 800 MB audio file. m4a was about 123 MB. [05:34] <jjohn> So flac is even worse. [05:35] <jjohn> You already said that before, but I wanted to make the point that flac is really bad, and I have no real gains from it. [05:38] <Zeranoe> jjohn: flac will provide the best "original audio quality". The file size will be the sacrifice [05:39] <Dark-knight> is there more then one audio stream? [05:40] <jjohn> Zeranoe flac is cool if the quality is to be preserved. But I just want to copy the audio stream, no more, no less. [05:40] <Dark-knight> just "ffmpeg -i input -c:a copy output" should suffice [05:40] <jjohn> Dark-knight no there isn't. [05:40] <Zeranoe> ^ [05:41] <jjohn> Dark-knight that wasn't my question. ;) [05:41] <Dark-knight> but it is your answer [05:41] <jjohn> My question was: do you guys know any other good pure audio container formats? [05:42] <Dark-knight> let my quote my notes [05:42] <Dark-knight> FLAC and ALAC are lossless audio formats. (the files is compressed and no data is lost.) [05:42] <Dark-knight> WAV and AIFF are uncompressed lossless audio formats. (the files are uncompressed and take up lots of space. not recommended) [05:42] <Dark-knight> APE is a highly compressed lossless audio format (the files are more compressed and no data is lost. It's not very compatible.) [05:42] <Zeranoe> jjohn: What are you trying to do? [05:42] <Zeranoe> jjohn: and why [05:43] <jjohn> As I stated on pastebin: I have video files with audio, and the videos suck, so I want to GET the AUDIO stream and put it into a new file, no encoding, no decoding, into an appropiate audio container format. [05:44] <Zeranoe> jjohn: what is the format of the audio stream in the original? [05:44] <Dark-knight> again [05:44] <Dark-knight> just "ffmpeg -i input -c:a copy output" should suffice [05:44] <jjohn> Getting, for example, the entire AAC stream, and put it into m4a. Worked fine, but I wanna know if I can improve that. [05:44] <Dark-knight> improve how? [05:44] <Zeranoe> jjohn: why? is it not working? [05:45] <Zeranoe> if your input is AAC, your containers are m4a and aac [05:45] <Dark-knight> Zeranoe: i didn't think aac was a container, i thought it was a format? [05:46] <jjohn> Zeranoe: I am fairly new with all that encoding stuff, and I was actually sure my naive solution would not provide the best results AND saving all encoding and decoding stuff. [05:47] <jjohn> But if you tell me that if I got a raw AAC stream I should put it into a m4a container, as I did from the very beginning, I am glad. Because then someone who knows more about it told me I did OK. [05:47] <Zeranoe> Dark-knight: doesn't look like FFmpeg supports it anyway, muxing that is. It can demux [05:48] <Dark-knight> ah [05:48] <Zeranoe> jjohn: ffmpeg -i input -c:a copy -vn output.m4a will dump the exact AAC stream to a m4a file. [05:48] <jjohn> Zeronoe that's what I did. I have been telling you this three times now. I just wanted to know if there is a better container for AAC. [05:49] <jjohn> Or for pure audio in general. [05:49] <Dark-knight> i really dont see how thats different from mine Zeranoe. :/ [05:49] <jjohn> Because, as I stated, I am absolutely not sure about all the features ffmpeg provides, and I was sure I did something stupid. [05:50] <Zeranoe> Dark-knight: it's not [05:50] <jjohn> Like, using a container which is outdated or not appropiate. [05:50] <Dark-knight> jjohn: !google containers compatible with aac audio [05:51] <Zeranoe> jjohn: there isn't a audio only container for every type of audio stream if that's what you mean. m4a is the container for a AAC stream, and is the best container for your purpose. [05:52] <jjohn> Zeranoe: yep, that's what I meant. And since you told me it's the best for my purpose, I will accept that. [05:52] <jjohn> And what is about the second question? [05:54] <Dark-knight> probably not [05:55] <jjohn> probably? [05:55] <Dark-knight> 99.9% [05:56] <Dark-knight> how about you just do it and listen to it for yourself [05:56] <Dark-knight> should answer your own question [05:56] <jjohn> So I can ignore the missing flag? Dark-knight I did that but it's 5:56 AM here and I am a bit tired, I don't trust my hearing right know. And again I wanted to be sure. :) [05:57] <Dark-knight> paste an output of the error that you a receiving [05:57] <jjohn> It sounded good, but I could just imagine any affect subconsciously. [05:59] <Dark-knight> are* [05:59] <Dark-knight> sometimes i forget to type out entire words and just type the first letter by accident [05:59] <jjohn> ffloger I did so half an hour ago. Not the exact output, but what I used to extract the audio stream (first question) and what it gave me as flag on the audio output: http://pastebin.com/9PMsZ908 [06:00] <Zeranoe> jjohn: Time for bed, unless you woke up at that time, in which case, time for bed [06:00] <jjohn> It's not an error Dark-knight. As I said it's just a *missing* flag. [06:01] <Dark-knight> Zeranoe: lol [06:02] <jjohn> And I wondered if that's OK. Because, as I stated like three times *again* that I am not experienced with ffmpeg and codecs. [06:02] <Dark-knight> jjohn: can you post an output of the missing flag? [06:06] <jjohn> Dark-knight: sure: http://pastebin.com/KRui4wNV [06:07] <Dark-knight> looks fine [06:07] <Dark-knight> just do it and listen [06:08] <jjohn> OK, if you say so ... that missing flag has no meaning? [06:08] <Dark-knight> i've never seen it when i converted/remuxed shit [06:08] <Dark-knight> so nope [06:09] <jjohn> OK ... then I'll do it. Thanks for the help, and sorry if I was hard to understand. [06:09] <jjohn> Didn't mean to. [06:09] <Dark-knight> nah you were fine [06:10] <jjohn> Thanks again. And good night/morning/whatever. Bye [06:30] <Dark-knight> just a quick question [06:32] <Dark-knight> do i need to specify -map 0:v if there is only one video track or will -c:v copy suffice? [06:33] <relaxed> the latter [06:34] <Dark-knight> this is what i had [06:34] <Dark-knight> ffmpeg -i input -map 0:v -map 0:1 -map 0:2 -map 0:4 -map 0:6 -c:v copy -c:a alac -c:s mov_text -f ipod output.mp4 [06:34] <Dark-knight> will this work just the same [06:34] <Dark-knight> ffmpeg -i input -map 0:1 -map 0:2 -map 0:4 -map 0:6 -c:v copy -c:a alac -c:s mov_text -f ipod output.mp4 [06:37] <relaxed> do you only want one video stream in all your output? [06:38] <Dark-knight> yes [06:39] <relaxed> and is it always the first video stream? [06:39] <Dark-knight> video track is stream 0:0 [06:40] <Dark-knight> if i specify -map once, do i have to include everything i want? [06:41] <Dark-knight> using more -map [06:42] <relaxed> what do you want? [06:42] <relaxed> only one video stream and all audio and subtitle streams? [06:44] <Dark-knight> http://pastebin.com/Lr3QJ9Lg [06:44] <Dark-knight> contents of the file [06:47] <relaxed> Both of those commands should work. Is there a problem? [06:49] <Dark-knight> just trying to make it as simple as possible [06:49] <Dark-knight> just wondering for future reference if leaving -map 0:v out of that command line will still copy over the video stream [06:51] <relaxed> if there's more than one video stream it will copy them all [06:51] <Dark-knight> i know that [06:52] <relaxed> use -map 0:v:0 if you always want one, and you assume the first video stream is what you're after [06:54] <Dark-knight> im wondering. because i already specified a lot of other streams using -map and was wondering if not including -map 0:v will still allow the video stream to be copyed [06:56] <relaxed> oh yeah, you do have to specify all the streams if you specify any :) [06:56] <Dark-knight> ok thanks [06:56] <Dark-knight> ill ad that to my nots [06:56] <Dark-knight> add* [06:56] <Dark-knight> notes* [07:03] <Dark-knight> wait... does that mean if i specify any of these -map 0:v or -map 0:a or -map 0:s. that i have to all the other streams [07:05] <relaxed> you -map what you want in the output [07:14] <Dark-knight> not always [07:14] <Dark-knight> sometime -map is un-nessasry [07:15] <relaxed> example? [07:21] <Dark-knight> ffmpeg -i input -c copy output [07:21] <Dark-knight> when there is only 1 of each stream [07:21] <relaxed> yes, there are sane defaults [07:22] <Dark-knight> so like i said [07:22] <Dark-knight> this file has 1 video, 3 audio, 3 sub. Do i have to specify the 1 video file or can i just specify everything else [07:23] <relaxed> as I said earlier, once you map one stream you have to map them all [07:23] <Dark-knight> ok just making sure. i gotta cover all my bases [07:24] <relaxed> or else how would ffmpeg know what you wanted? [07:27] <Dark-knight> so the second command wouldnt do what i wanted [07:28] <relaxed> oh, right [07:28] <relaxed> too many numbers [07:28] <Dark-knight> not what i meant [07:29] <Dark-knight> by removing -map 0:v from the first command line, then the video stream wont be copyed [07:29] <Dark-knight> right? [07:29] <relaxed> correct [07:30] <Dark-knight> even thought i have -c:v copy? [07:32] <relaxed> try it [07:38] <Dark-knight> im busy, just tell me [07:40] <Dark-knight> nvm [07:40] <Dark-knight> im just try it [07:46] <Dark-knight> ahh shit [07:47] <Dark-knight> your right [07:48] <spectralsun> where did qt-faststart go? [07:48] <spectralsun> it is not in tools anymore :( [07:49] <relaxed> spectralsun: use ffmpeg -i input -movflags faststart ... output.mp4 [07:52] <Aero> Hai! [07:52] <Aero> Is this some scamware? [08:03] <Dark-knight> ok relaxed i have a file with 3 audio, 2 of them are flac and 1 is aac. i want to change the two flac into alac and keep the aac the same [08:03] <Dark-knight> whats the command line? [08:06] <relaxed> pastebin the ffmpeg -i input [08:07] <Dark-knight> http://pastebin.com/Lr3QJ9Lg [08:07] <relaxed> video too? [08:08] <Dark-knight> just -c:v copy for the video [08:10] <relaxed> ffmpeg -i input -map 0:v -map 0:a -map 0:s -c:v copy -c:a:0 alac -c:a:1 alac -c:a:2 copy -c:s mov_text -movflags faststart output.mp4 [08:10] <Dark-knight> Ahh thank you so much [08:11] <Dark-knight> didnt know you could do that [08:11] <Dark-knight> ill add that to my notes [08:11] <relaxed> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [08:11] <Dark-knight> Aero: What are you talking about [08:11] <relaxed> http://ffmpeg.org/ffmpeg.html#Advanced-options [08:17] <Dark-knight> what does ffmpeg -movflags faststart do? [08:18] <relaxed> moves the index to the beginning of the file. [08:37] <Dark-knight> index? [08:39] <relaxed> google moov atom [08:39] <ParkerR> Dark-knight, AVI files larger than 4GB usually have to have an index built in order to skip through properly [08:39] <ParkerR> Is my guess. may be talking out of my ass though :/ [09:08] <[A3G1S]> hey guys, I am trying to hardcode srt subtitles using ffmpeg, but it isn't working (I've build a static ffmpeg build and trying to use it for hardcoding subs) [09:08] <[A3G1S]> any 1 can help ? [09:42] <jrgill> Do the aac encoders each give some special treatment to the bitrate for mono streams? Seems native aac sees >=289k as invalid and libvo_aacenc just forces 64k. [10:20] <termos> how is ffmpeg handling multithreaded programs? I'm doing av_write_frame calls from different threads but to different rtmp streams and I'm getting some weird bugs. Could it be related? [11:59] <joe90k> Can anyone help me remove a compiled version of ffmpeg or totally remove it from my CentOS box? I have the original compile instructions and have tried removing the dir that was created, but when I run 'ffmpeg' I get '-bash: /usr/local/bin/ffmpeg: No such file or directory ' [12:07] <joe90k> Sorted with reboot.. [15:04] <zlice> what am i looking for? when ffmpeg is given an input file - it normally shows config and then says 'input #0 type, from 'file.type' ' [15:04] <zlice> i do not get there, just stops at the config [15:07] <zlice> hm...let me see what i can do [15:08] <zlice> f it, i'll have to type it, h/o [15:17] <zlice> http://pastebin.com/e6K7z0X6 phew [15:18] <zlice> i've tried other file formats, hang different places on the loglevel but all just hang after build config. so idk what i'm looking for really [15:23] <c_14> zlice: What exactly is the problem? Does the command freeze? Or do you just want the codec/format information? [15:26] <zlice> c_14 : pretty sure it freezes [15:27] <c_14> Can you try attaching an strace to the process or checking in something like htop what state the command is? Is it still taking cpu cycles? [15:27] <zlice> i'd have to cross-compile that...probably couldn't til this afternoon/night [15:28] <zlice> hm, opening up another ssh looks like ffmpeg is running at 100% - could it but stuck? trying to calc something? i didn't look at the code, where does it say 'input file #0' ? [15:29] <c_14> It should be the next line after the '[mp3' lines [15:30] <zlice> (running on qnx bb q10 btw) [15:30] <c_14> Can you play the mp3? [15:30] <c_14> Heck, can you cat it? [15:30] <zlice> i'll try that [15:30] <zlice> yyyep [15:30] <c_14> ffprobe mymp3.mp3 ? [15:31] <c_14> Oh, you disabled ffprobe... [15:31] <zlice> ya [15:32] <zlice> i guess, is there an easy way to tell if it's actually doing something? i thought verbose would help, maybe i'm not thinking of something simple [15:34] <zlice> it could just be qnx, idk [15:34] <c_14> You can try pressing h while ffmpeg is running. [15:35] <c_14> That gives out more debug info. [15:35] <c_14> Well, it starts dumping packets/hex which I guess counts as more debug info. [15:36] <zlice> that's what i'm looking for :) but i don't see anything yet :( [15:36] <c_14> What does `ps -o stat `pidof ffmpeg`' return? [15:37] <c_14> If you had something like gdb, or strace you could try that as well. [15:40] <zlice> pidin(ps) shows PID# ffmpeg 10r(the state) RUNNING [15:40] <zlice> i know qnx has gdb, haven't used it yet really... [15:41] <c_14> So, it's not stuck in IO-wait... [15:43] <c_14> I'm guessing it might be an issue with the configuration/architecture. [15:47] <zlice> hm ok, i will see if i can't get ntoarm-gdb working and try to look at it further, thans c_14 [15:48] <Neppy> ohi~ trying to loop a video+audio with ffmpeg... not haivng much luck though, ffmpeg -i "input.mkv" -c:v copy -c:a copy -loop 20 "output.mkv" but the output i sjust 3 seconds, same as input.. or can i not use "copy" with that? [15:49] <c_14> The loop option doesn't really do what you want it to. [15:50] <c_14> Your best bet is to use the concat demuxer and just concat the video 20 times. [15:51] <Neppy> ah :/ [15:59] <Neppy> hmm that works but messes up the audio xD [16:01] <Neppy> though when cutting out the file i want to loop out of a bigger file i also had some weird audio issues? seems audio is in chunks too and it cant cut "subchunks" or so.. [16:01] <Neppy> since i had to increase ms by .050 or it would cut out the audio early [16:02] <c_14> yep, if you use -codec copy you can only cut in "chunks" [16:03] <Neppy> hmm what if i use -c:v copy and -c:a libvorbis? [16:11] <c_14> It should be able to cut precisely. [16:12] <Neppy> hmm doesnt seem to :< [16:17] <bunniefoofoo> in mpeg2video, does a b frame have to be between two p-frames for example for a 4-frame closed gop, IPBB is invalid but IPBP is OK ? [16:18] <bunniefoofoo> the reason, I'm asking is because ffmpeg is generating files with a B frame at the end when using strict gop mode [16:30] <Chaz6> Hey there, just wanted to pop by and say, I got a Logitech C930e as it's supposed to be a step up from the C920, but it does not support h264 encoding with directshow :/ [16:30] <Chaz6> So I'll stick with the C920 [17:00] <DdoubleU> Hey, is there any way to encode a video file and have the output name the same as input , so just replacing the original? [17:01] <Chaz6> DdoubleU: i don't know but perhaps looking to COW (copy on write), e.g. btrfs/vss? [17:01] <bunniefoofoo> should work on linux if you pipe the file into ffmpeg, on windows probably not since typically you can't overwrite an open file [17:02] <DdoubleU> ok thanks [17:03] <DdoubleU> Another thing, Is there a way to encode multiple inputs all to their own outputs in one command? [17:06] <Bombo> DdoubleU: why not rename it before? source.mkv source_old.mkv then encode source_old.mkv to source.mkv, del old... [17:06] <DdoubleU> ^ thats what im going to end up doing [17:07] <Bombo> DdoubleU: you mean multiple inputs, as in 'encode *.mkv'? [17:08] <bunniefoofoo> doubleU: you don't save any drive space either way... if the file is open for reading and you "overwrite" it as on linux, the file is not free'd up until you are done anyways [17:09] <DdoubleU> yes, I have the paths of the inputs [17:09] <DdoubleU> and i want each to encode seperately into its own output [17:10] <Bombo> DdoubleU: just script it in bash or batch or whatever ;) [17:10] <DdoubleU> ok [17:26] <Bombo> how do i extract subtitles with ffmpeg? this is what i'm trying: http://bombo.jpe.gs/sub/ff.txt what codec parameters are missing? and why is the operation not permitted? [17:26] <ubitux> there is no vobsub muxer in ffmpeg yet [17:27] <ubitux> see http://trac.ffmpeg.org/ticket/2391 [17:27] <ubitux> Bombo ^ [17:28] <ubitux> dvd subtitles can only be extracted in a vobsub (sub+idx) [17:28] <ubitux> .sub is the extension for MicroDVD subtitles, which are text (while dvd sub are bitmaps) [17:30] <Bombo> ubitux: ah ok so it isn't possible right now... [17:30] <ubitux> feel free to +1 the ticket, there is a vote system [17:30] <Bombo> i just thought if ff is able to copy it it should be able to save it somehow [17:31] <ubitux> it also needs to index it in an idx file [17:31] <ubitux> it's not trivial, DVD are a real mess [17:31] <ubitux> and vobsub is a nice pile of crap as well [17:32] <ubitux> writing a demuxer for vobsub was a few orders of magnitude harder than what i expected [17:32] <ubitux> so i'm not really motivated to write a muxer right now ;) [17:33] <iive> i think you want to say that writing a demuxer for vobsub in theory is easy, but writing one that works correctly with all the available vobsub is nightmare :) [17:36] <ubitux> well vobsub is an home made split file format based on one format that was unknown [17:36] <ubitux> there is duplicated information, sometimes inconsistent, the segmentation is completely chaotic, and as no one understands any of the two, there are indeed all kind of broken files all around [17:37] <ubitux> and i don't feel like i understand it enough to take the responsibility to write a muxer that will output more broken files [17:38] <ubitux> but, i would love to have one ;) [17:39] <Bombo> +1'ed ;) [17:40] <Bombo> ok thx for the explanation [17:40] <JEEB> just find a bastard who is ready to take the responsibility and then not do anything about it [17:40] <JEEB> :3 [17:40] <ubitux> Nicolas seemed to be motivated [17:40] <ubitux> but i think seeing me derping with the demuxer warned him about the curse [17:55] <Bombo> hm doesn't work with mkvextract either [17:55] <Bombo> probably this https://trac.ffmpeg.org/ticket/2035 [17:56] <Bombo> i think i'll give up now [18:58] <LtHummus> I'm trying to encode a file for DVD, but DVD Studio Pro only allows for elementary mpeg streams...is there a way i can output both an m2v and an ac3 elementary file instead of doing run of ffmpeg to encode and an additional one to demux? now i'm running ffmpeg -i source.whatever -target ntsc-dvd <more random flags> output.mpg [19:04] <amkrankruleuen> Hello i have problem with ffmpeg, someone alive? [19:23] <azk> amkrankruleuen: Hi, your best bet is to state your problem and stick around [19:25] <amkrankruleuen> azk: Ok, I have problem with android player, i download mp4 file to my linux on laptop and convert mp4 file to mp3 file using ffmpeg, when i download mp3 file to my phone from laptop i jave problem with moving file [19:25] <amkrankruleuen> i have* [19:25] <amkrankruleuen> all android version, on laptop is okay, on old symbian is okay but on android is not okay [19:26] <azk> So, you're having problems moving the file within the android filesystem? [19:27] <amkrankruleuen> Nooo, i have problem with moving audio in player [19:27] <azk> Oh, seeking. I see. [19:27] <amkrankruleuen> for example i move from 0:01 to 1:00 but player moving to random time ex: 0:10 [19:28] <dequid> Hello all, i'm having trouble encoding mov's with prores and pcm audio. [19:28] <amkrankruleuen> azk: My coomand http://pastebin.com/3EjSerGP [19:29] <azk> amkrankruleuen: Let me see if I can reproduce it [19:29] <dequid> ffmpeg/ffprobe recognizes the audio as 'pcm_s16be', but when i examine the file in quicktime player inspector, it says for the audio channel: 32 bit, signed, big endian [19:30] <dequid> the resulting audio (even when just playing with ffplay) sounds weirdly slowed down, like slow-mo, with click sounds [19:31] <amkrankruleuen> azk: Ok [19:31] <c_14> dequid: Have you tried forcing the codec to pcm_s32be? [19:31] <dequid> Here is my output of ffprobe: http://pastebin.com/QQ1Ffbrh [19:34] <dequid> c_14: Yes, here's my command: http://pastebin.com/45YrcPER [19:35] <c_14> try your command with ffmpeg -c:a pcm_s32be -i [file] [options] [19:35] <dequid> c_14: ok [19:36] <c_14> That'll force the codec detection in case the audio really is pcm_s32be. [19:37] <azk> amkrankruleuen: I can't reproduce it. [19:38] <dequid> c_14: http://pastebin.com/Td1bp2KC [19:39] <dequid> c_14: No change, audio still not good. Only change is resulting file is half size. [19:39] <dequid> c_14: resulting file still recognized as s16be by ffprobe [19:41] <c_14> Without the first -i input.mov, but not sure that'll change much in this specific use-case. How did you create the video/audio track? [19:45] <dequid> c_14: Ops, my fault. Now we're getting closer. With: ffmpeg -c:a pcm_s32be -i input.mov -vn output.aiff the 'slo-mo' effect is gone, audio seems in the right pitch, but it plays very choppy, with click sounds [19:46] <dequid> c_14: The files come straight from the Convergent Design Odyssey 7Q 422 recorder [19:48] <c_14> Try adding -c:a pcm_s32be as an output option as well. ie: ffmpeg -c:a pcm_s32be -i input.mov -vn -c:a pcm_s32be output.aiff [19:49] <c_14> The clicks _could_ be originating from the bit depth change from the 32bit source to the 16bit output. Not sure though. [19:49] <c_14> (on my system the aiff muxer picks s16be by default) [19:50] <dequid> c_14: Same thing with ffmpeg -c:a pcm_s32be -i input.mov -vn -c:a pcm_s32be output5.aiff just that filesize doubles again [19:51] <c_14> The filesize is due to the bit depth. Do you have a player that can play the source without the clicking and without the slow-mo? [19:52] <dequid> c_14: Strange thing is, that Quicktime can play it without problem (original file) and Adobe media encoder also [19:53] <dequid> c:14: When i try to play it in VLC, picture works but i get a message that a decoder module for " " is not found (empty string) [19:54] <c_14> I'm guessing that's because ffmpeg can't correctly detect the audio format. [19:55] <c_14> Does this happen with every file from that recorder or just that one? [19:55] <dequid> c_14: With every one unfortunately. [19:55] <c_14> Might be a bug. [19:55] <c_14> I'd probably report it on the trac. [19:55] <c_14> If you can provide a sample, that would be great as well. [19:57] <dequid> My smallest file is 80 MB, i tried to make it smaller with 'dd' as described in the bug report howto, but the resulting file is unreadable for every player/ffmpeg. [19:57] <amkrankruleuen> azk: I don't understand [19:57] <amkrankruleuen> reproduce what? [19:57] <azk> Your error [19:58] <amkrankruleuen> maybe i show file? [19:58] <azk> That exact command produced a seekable mp3 on my android device [19:58] <azk> sure [19:58] <c_14> dequid: ffmpeg -t duration -i input.mov -codec copy out.mov [19:59] <dequid> c_14: duration = time in seconds? [19:59] <c_14> ye [19:59] <dequid> ok [19:59] <amkrankruleuen> azk: http://193.187.65.6/audio.mp3 [19:59] <amkrankruleuen> not seekable [20:00] <c_14> dequid: you could also use -fs limit_size , not sure how well it works though [20:01] <dequid> c_14: Funny: when i do that, the audio is now corrupted even in QT [20:01] <dequid> I think i'll best upload the whole thing to our server. [20:02] <c_14> If you can just get it somewhere that the devs will have easy access to, it should be fine. [20:02] <dequid> ok [20:02] <c_14> Just say that you didn't upload it to the ftp because it was large and cutting it introduced additional errors. [20:04] <azk> amkrankruleuen: Seems like it's the default Android [20:04] <azk> 'Music' app that's not seeking [20:04] <azk> Other players work fine. [20:06] <spectralsun> what happens if you faststart a video multiple times? [20:07] <amkrankruleuen> azk: any idea how to solve? [20:07] <azk> No idea at all. [20:07] <c_14> spectralsun: Nothing should happen. It should notice that the moov atom is already at the beginning and not do anything. [20:07] <azk> Short term solution would be to use another player amkrankruleuen [20:08] <amkrankruleuen> azk: But your file mp3 from mp4 work correctly? [20:08] <amkrankruleuen> on 'Music' app [20:08] <amkrankruleuen> ? [20:09] <spectralsun> c_14: cool, thanks. [20:09] <azk> Didn't actually check with that since it isn't my default. [20:10] <azk> amkrankruleuen: Yeh, I can seek in Music app [20:13] <dequid> c_14: Thanks for your help anyway [20:19] <amkrankruleuen> azk: show me your command ffmpeg [20:19] <azk> The exact same as your. [20:19] <azk> yours [20:19] <amkrankruleuen> :| [20:27] <amkrankruleuen> azk: hmm, indeed your is seekable but my not [20:29] <azk> pastebin your ffmpeg --version [20:30] <amkrankruleuen> http://pastebin.com/1x8NxYsB [20:45] <azk> Hmm, I'm running on 2.2.2, not sure if that has anything to do with it. [21:07] <bunniefoofoo> is there a way to feed ffmpeg separate audio & video stream, using -acodec copy, truncate the audio stream to match the video? [21:22] <relaxed> bunniefoofoo: the audio is longer? [21:22] <bunniefoofoo> yeah [21:22] <relaxed> ffmpeg -i input -shortest ... [21:23] <bunniefoofoo> I am trying to use ffprobe to get the frame count, then use ffmpeg -t <duration> [21:27] <relaxed> bunniefoofoo: -shortest [21:29] <bunniefoofoo> is that going to use estimated duration? [21:30] <relaxed> it will end encoding when it reaches the end of the shortest stream [21:32] <bunniefoofoo> shortest based on the packet pts then... [21:59] <MykeBates> I've always had issues with doing this properly. Hoping someone can help me out here! Trying to convert MKV to MP4. Simply running -i input.mkv output.mp4 works but the quality is poor. Every option combination I seem to try -vcodec copy -acodec copy, etc do not seem to work for me. Any pointers out there for how to convert mkv to mp4 while maintaining quality? [22:00] <Mavrik> -codec copy is what you're looking for [22:00] <Mavrik> (just remuxing without reencoding) [22:00] <Mavrik> so... I suggest fixing your "does not work" issue. [22:01] <MykeBates> so, for example - ffmpeg -i input.mkv -codec copy out.mp4 ? [22:02] <Mavrik> yes. [22:07] <MykeBates> Damn yes! Forgive my ignorance on this....0_o (mac specific question)would you happen to be able to explain why it would play fine in a video player(vlc) but not in Mac's preview? [22:08] <Mavrik> no idea [22:08] <Mavrik> Quicktime (mac default player) can be quite shitty with format support [22:08] <Mavrik> and you really didn't provide any info for me to know what you have :P [22:11] <MykeBates> I certainly can, didn't want to oberstay my welcome with bombarding with too much. Check it: [22:11] <MykeBates> Converting with just -i input.mkv output.mp4 results in a poor quality but playable in preview and here are some specs on the video - https://cloudup.com/cUZeWk3LD9Y [22:11] <MykeBates> When running with -codec copy - mac preview(space bar on file) results in https://cloudup.com/cNbo6browLN as opposed to playing the video and when opening in the same file viewer app - https://cloudup.com/ct457dLuKSn does not seem to want to show the preview or video information. [22:13] <Mavrik> uh [22:13] <Mavrik> yeah, that's not helpful [22:13] <Mavrik> do this: [22:15] <MykeBates> http://pastie.org/9456510 [22:17] <Dark-knight> wow [22:18] <Dark-knight> umm "EXACT ffmpeg command and the COMPLETE console output." [22:18] <Mavrik> MykeBates, output too, it's the important bit - it shows what your source is and what's it encoding into ;) [22:18] <MykeBates> damn, my bad -- friday -- brain fried. [22:18] <MykeBates> one sec [22:20] <MykeBates> http://pastie.org/9456521 [22:21] <Mavrik> ah [22:21] <Dark-knight> and what is the problem with the output? what are you trying to accomplish? [22:21] <Mavrik> MykeBates, my guess is preview/quicktime doesn't like the AC3 audio [22:22] <Mavrik> Dark-knight, he has problems with remuxed mp4 files not playing in OS X preview [22:22] <Dark-knight> let me check my notes, i got something for that [22:22] <Mavrik> MykeBates, AC3 isn't really standard as long as MP4 wide support goes [22:22] <Dark-knight> brb [22:22] <Mavrik> also, audio is 5.1 :) [22:23] <bunniefoofoo> MP4 proper needs to use AAC audio [22:23] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a libfaac output.mp4 [22:23] <Dark-knight> that should do it [22:24] <MykeBates> nailed it! [22:24] <Dark-knight> you're welcome [22:24] <MykeBates> Quite obvious I need some brushing up on a/v codecs... sorry for my ignorance and thanks a million for the help :) [22:25] <Dark-knight> np [22:26] <Mavrik> ugh [22:26] <Dark-knight> what [22:26] <Mavrik> well that just murdered all your audio quality :P [22:26] <MykeBates> yeah? [22:26] <Dark-knight> haha he could just try "-c:a aac" instead [22:27] <Dark-knight> delete the libf [22:27] <Mavrik> MykeBates, yes, since ffmpeg took defaults and libfaac doesn's support multichannel [22:27] <Mavrik> so your audio probably went from 5.1 high-quality to shitty stereo :D [22:27] <Dark-knight> well lets see [22:27] <Dark-knight> post the output [22:28] <MykeBates> cool, one sec [22:28] <MykeBates> http://pastie.org/9456535 [22:29] <Dark-knight> looks like 5.1 to me [22:29] <Mavrik> oh, libfaac now supports 5.1, nice [22:30] <Dark-knight> so it seems that worked, but ill post this anyway [22:30] <MykeBates> bonus! [22:30] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a aac output.mp4 [22:30] <Dark-knight> If that doesn't work the first time use [22:30] <Mavrik> anyway, you might wanna use the internal aac encoder (libfaac really is the worst of all possible AAC encoders)\ [22:30] <Dark-knight> ffmpeg -i input.mkv -c copy -strict -2 -c:a aac -b:a 32k output.mp4 [22:30] <Mavrik> and set bitrate to something like 256kbit [22:30] <Baked_Cake> mine only works if i set it to libvo_aacenc [22:30] Action: Mavrik slaps Dark-knight. [22:30] <Dark-knight> what? [22:30] <Mavrik> WHY are you giving him misleading advice [22:30] <Mavrik> coding movie audio to 32kbps?! [22:31] <Dark-knight> well it works for me [22:31] <Baked_Cake> lol [22:31] <Mavrik> jesus fuck [22:31] <Mavrik> that's 1950s telephone quality :D [22:31] <Baked_Cake> dark knight like to have his ears shredded by the devil [22:31] <Dark-knight> fine then dont listen [22:31] <Mavrik> Baked_Cake, yeah, that's because your ffmpeg is built with libvo_aacenc instead of libfaac [22:32] <Baked_Cake> what do you guys think about the vbr function on libfdk_aaac [22:32] <Mavrik> libfaac < libvo_aacenc < internal aac < fdk-aac [22:32] <Baked_Cake> typo [22:32] <Mavrik> that's pretty much the quality scale [22:32] <Baked_Cake> ic [22:32] <Dark-knight> Mavrik: mind posting the command line using the internal aac encoder then? [22:32] <Mavrik> fdk is currently still pretty much the best [22:33] <Mavrik> Dark-knight, take what Dark-knight said but change -b:a to something sensible, 256k should probably keep enough of 5.1 quality for you to not notice the difference :) [22:33] <Mavrik> er [22:33] <Mavrik> yes, that made no sense [22:33] <MykeBates> ffmpeg -i input.mkv -c copy -strict -2 -c:a aac -b:a 256k output.mp4 <--- worked really well [22:34] <Dark-knight> i actually want Mavrik to answer my question [22:34] <MykeBates> can notice the crispness in the audio :) [22:34] <Baked_Cake> ive been giving -vbr 3 a try on my 5.1 flacs i think it comes out to about 224k on average [22:34] <Baked_Cake> on 2 channel i get about 132k [22:35] <Mavrik> mhm, that's ok [22:35] <Mavrik> most AAC audio is about ~192k for nice quality [22:36] <Mavrik> we went as low as 64k for mobile audio, but just with HE-AACv2 profile there [22:37] <Dark-knight> Mavrik: mind posting the command line using the internal aac encoder then? [22:37] <Mavrik> em. [22:38] <Dark-knight> and do one with the fdk-aac while your at it [22:38] <Mavrik> didn't you read what I wrote couple of lines above? [22:38] <Baked_Cake> u have to get a pirateed version of ffmpeg to get the libfdk_aac codec [22:38] <Baked_Cake> or build it urself [22:39] <Dark-knight> so just [22:39] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a internal aac output.mp4 [22:39] <Dark-knight> will do? [22:40] <Baked_Cake> dark u can run ffmpgeg -codecs [22:41] <Baked_Cake> erm [22:41] <Dark-knight> just a simple yes or no [22:41] <Baked_Cake> ffmpeg -codecs [22:41] <Baked_Cake> pause [22:41] <Mavrik> Dark-knight, you need to set desired bitrate [22:41] <Mavrik> to something sensible [22:42] <Baked_Cake> and that will give you a list of the codecsthat u can try [22:42] <Dark-knight> what happens if i leave it blank? [22:42] <Mavrik> also, you need the strict option for experimental codec [22:42] <Mavrik> it'll default to something awful [22:42] <Baked_Cake> mine never worked if i just put -c:a aac [22:42] <Mavrik> Baked_Cake, mhm, but it's usually worth building your own ffmpeg if you do something n volume :) [22:42] <Mavrik> yeah, because aac is experimental [22:45] <Baked_Cake> i should take the time to learn how to do it [22:46] <Mavrik> well, if you're on Windows you have a problem [22:46] <Baked_Cake> ya thats my problem [22:46] <Mavrik> on linux it's mostly just ./configure --enable-fdk-aac [22:46] <Baked_Cake> i know almost nothing about linux [22:46] <Mavrik> on OS X it's just brew install ffmpeg --enable-fdk-aac :) [22:47] <Baked_Cake> i tried doing it myself with this guide that used a linux shell terminal in windows, but i didnt get vvery far [22:47] <Baked_Cake> im probly better off trying to set up my virtual machine again [22:49] <Dark-knight> i did ffmpeg -codecs but it gave me a list of all of them. is there a command that just shows me the audio codecs [22:50] <Baked_Cake> i think generally the video codecs r listed first [22:50] <Baked_Cake> then oudio, the availible aac codec-s shouldbe clumped together [22:53] <Dark-knight> i know there is a command that just shows the audio codecs, i used it once before but now i can't remember [22:54] <Mavrik> hmm, I don't think there is - probably something like "ffmpeg -codecs | grep A" [22:55] <Dark-knight> i did a quick google but couldn't find the command [22:56] <Baked_Cake> u can always right click + find [23:05] <Dark-knight> ahhhh it was -encoders [23:05] <Dark-knight> made it easy [23:06] <Dark-knight> what is this [23:06] <Dark-knight> libfaac is clearly supported but it is not shown in the output [23:07] <Baked_Cake> is probly not enabled [23:07] <Mavrik> it only shows up if you compiled ffmpeg with libfaac [23:07] <Baked_Cake> or that [23:08] <Baked_Cake> if ur not linux or compiler savy u can find nonfree versions of ffmpeg but they are outdated and god knows what else [23:08] <Dark-knight> ok i just converted a video with ac3 audio using libfaac as a test, and it worked [23:08] <Dark-knight> but its not showing up in the list [23:08] <Baked_Cake> o [23:56] <bunniefoofoo> is there some trick to get ffprobe to show me the duration of an mpeg2video stream (.m2v) file? [23:56] <bunniefoofoo> it says duration N/A [00:00] --- Sat Aug 9 2014

1 0

ffmpeg-devel.log.20140807
by burek 08 Aug '14

08 Aug '14

[01:12] <cone-140> ffmpeg.git 03James Almer 07master:8dfb8190d768: libavcodec/rawdec: remove old ff_find_pix_fmt() function [01:56] <cone-140> ffmpeg.git 03Anton Khirnov 07master:a1599f3f7ea8: cdgraphics: switch to bytestream2 [01:56] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:91b4ed09180c: Merge commit 'a1599f3f7ea8478d1f6a95e59e3bc6bc86d5f812' [02:04] <cone-140> ffmpeg.git 03Anton Khirnov 07master:c7d9b473e282: cdgraphics: do not return 0 from the decode function [02:04] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:c62862799575: Merge commit 'c7d9b473e28238d4a4ef1b7e8b42c1cca256da36' [02:12] <jamrial> michaelni: regarding the patch i just sent, there are other internal functions and tables being exported the same way, like ff_raw_pix_fmt_tags[] and ff_codec_get_id() [02:30] <michaelni> maybe ff_raw_pix_fmt_tags should be accessed through a function [02:30] <michaelni> from outside lavf [02:31] <michaelni> lavc i meant [02:31] <jamrial> yeah, like we're doing with avformat_get_riff_video_tags for ff_codec_bmp_tags[] [02:34] <cone-140> ffmpeg.git 03Anton Khirnov 07master:7b588bb69164: svq1: do not modify the input packet [02:34] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:e55e09949e59: Merge commit '7b588bb691644e1b3c168b99accf74248a24e3cf' [02:34] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:4213fc5b9eeb: avcodec/svq1dec: Fix multiple bugs from "svq1: do not modify the input packet" [02:38] <michaelni> about ff_codec_get_id() all uses are of the form "ff_codec_get_id(avformat_get_riff_video_tags()...", iam not sure if that could/should or not be used instead of making ff_codec_get_id avpriv [02:40] <jamrial> can those calls to ff_codec_get_id be replaced with av_codec_get_id instead? [02:41] <cone-140> ffmpeg.git 03John Stebbins 07master:d9432789bd11: movenc: remove pointless loop around BITEXACT test [02:42] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:71f33051e5d0: Merge commit 'd9432789bd119f0e37bcf65cebda05d36aafd4ed' [02:48] <jamrial> something like http://pastebin.com/XvXMRVgj [02:50] <Chillance> ok, can anyone of you guys look at my issue? [02:50] <Chillance> http://pastebin.com/qWkNPrKP - this works, but colors aren't as good. especially with fading, and the file is huge (17MB). [02:50] <michaelni> jamrial, it needs a NULL after the point in an array but yes it should work [02:51] <Chillance> http://pastebin.com/hrpaavJT - this works better with colors, but have a flickering line on the right side [02:51] <Chillance> you can actually see the issue here: http://tempsend.com/02C98306C8 [02:51] <Chillance> check the .gif [02:53] <cone-140> ffmpeg.git 03John Stebbins 07master:da9cc22d5bd5: movenc: add track title to tracks [02:53] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:403367d5a95c: Merge commit 'da9cc22d5bd5f59756c2037b02966376da2cf323' [03:01] <michaelni> Chillance, probably best if you open a bug report on trac [03:01] <Chillance> ok [03:07] <cone-140> ffmpeg.git 03John Stebbins 07master:b50173a4dd47: movenc: fix QT chapter track character encoding [03:07] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:1c59419419d2: Merge commit 'b50173a4dd47b9c3c89845b781fa958ccf860929' [03:19] <jamrial> michaelni: what do you mean with that? [03:21] <Chillance> damnit [03:21] <Chillance> file is 2.6Mb, limit 2.5Mb [03:24] <Chillance> oh wait, I can split it [04:10] <michaelni> jamrial, it takes a NULL terminated array, so {ptr, NULL} would be needed [04:30] <cone-140> ffmpeg.git 03James Almer 07master:5b12b4fc4b22: lavf/mpegts: rename ff_mpegts_parse_*() to avpriv_mpegts_parse_*() [09:57] <cone-401> ffmpeg.git 03Carl Eugen Hoyos 07master:7c73d215231a: lavf/mpegts: Do not return the result of a (void) function from a void function. [12:32] <cone-401> ffmpeg.git 03Carl Eugen Hoyos 07master:ef17a0c7ba00: Do not return a probe score from set_codec_from_probe_data() if the codec was ignored. [12:32] <cone-401> ffmpeg.git 03Michael Niedermayer 07master:4c404611dae4: Merge remote-tracking branch 'cehoyos/master' [13:00] <BBB> lol the joys of havong non-coders defend your codebase (diego@lwn) [13:02] <cone-401> ffmpeg.git 03James Almer 07master:0ab00a75e41f: avcodec/raw: add avpriv_get_raw_pix_fmt_tags() [13:18] <cone-401> ffmpeg.git 03James Almer 07master:1fd880cef305: lavd/dshow: use av_codec_get_id() [13:34] <cone-401> ffmpeg.git 03James Almer 07master:dd2b0c1cde15: lavc: remove nonexistent functions from the version file [13:43] <cone-401> ffmpeg.git 03James Almer 07master:350f4be12dbc: lavf: remove nonexistent functions from the version file [14:52] <cone-401> ffmpeg.git 03Carl Eugen Hoyos 07master:7d585bc94c49: Do not automatically map subtitles if the decoder and encoder properties are different. [14:52] <cone-401> ffmpeg.git 03Michael Niedermayer 07master:8fa695a9e379: Merge remote-tracking branch 'cehoyos/master' [15:46] <cone-401> ffmpeg.git 03Michael Niedermayer 07master:75bd83d448fe: avdevice/lavfi: Simplify non-AV stream check [17:19] <ubitux> "What features are you missing? I'm certainly missing nothing, as - arguably - are most people since either library versions are very featureful" lol Diego [17:49] <kierank> that article was always going to end well [18:03] <Compnn> ubitux : paste me the comments? i cant read lwn without an account :P [18:05] <iive> Compn: check the log of this channel, there is url that allows reading it. [18:08] <Compn> oh thanks [18:09] <kierank> lwn.net/SubscriberLink/607591/7190d6c0b7274eb0/ [18:25] <Compn> didnt know libav had problem with libavresample [18:27] <Compn> libavresample isnt listed on the (new) download page [18:30] <Compn> oh misread it, the guy wants avresample to be listed [18:41] <ubitux> Compn: the guy is probably koda, please read correctly what's being [18:42] <ubitux> Compn: what is your latest post for? it's unrelated to the issue.. [18:44] <Compn> ubitux : i could write 'no plans to put this info on the download page' if you like ? [18:45] <ubitux> i believe i explained enough [18:45] <ubitux> in my comment [18:45] <Compn> ok then [18:45] <ubitux> i'd like my answer not to be lost in dozens of messages where you try to figure out the issue if you don't mind ;) [18:46] <Compn> how did i get here, i am not good with computer... [18:46] <Compn> ubitux : so you know thats koda eh? [18:46] <ubitux> projectsymphony is the name of his blog [18:47] <ubitux> i'm currently answering a bunch of messages on lwn, since diego decided it was the main playground [18:48] <wm4> drama! [18:49] <Compn> ubitux : did you mention fabrice owns the domain and trademark of ffmpeg yet? :P [18:49] <Compn> ehe [18:49] <Compn> but yes drama [18:50] <Compn> lol @ mpv shilling in a bunch of these articles [18:50] <Compn> wm4 : ordered chapter support is duuumb! :P [18:50] <Compn> ehe [18:50] <ubitux> no it's not [18:51] <wm4> Compn: "mpv shilling"? [18:51] <wm4> pretty sure none of the core members are posting there [18:51] <Compn> users shilling, not devs :P [18:51] <Compn> "And mpv is currently the best video player in Linux" [18:51] <Compn> ehe [18:52] <wm4> that's OBVIOUSLY nothing but the truth [18:53] <ubitux> :D [18:54] <wm4> anyway, it turned out that mplayer2 doesn't compile with Libav 10 anymore [18:55] <Compn> could ping uau about it :P [18:55] <kierank> who from ffmpeg is going to vdd? [18:55] <Compn> me [18:55] <Compn> you can look forward to me trolling about how good mplayer is :P [18:56] <Compn> and trying to pronounce your nickname as ky-rank again [19:00] <kurosu> ubitux, you forgot mentioning various removals of copyrights when replying to the "without credit or thanks" :D [19:00] <ubitux> gives me the latest commits to our fixes and i'll answer [19:00] <ubitux> -s [19:00] <ubitux> but they will probably say it has no value ;) [19:01] <kurosu> ubitux, I mean from some libav developers [19:01] <kurosu> I don't want to judge whether these removals had merit, but at least I saw them restored on occasion in commits to ffmpeg [19:01] <ubitux> yes we're talking about the same thing [19:01] <ubitux> i was just asking for commits to show "facts" [19:01] <ubitux> anyway, you can create an account for free and answer [19:02] <kurosu> ah ok [19:02] <Compn> oh i think the 'credits' diego means is the 'qatar' thing ;D [19:02] <ubitux> ah i forgot that [19:02] <kurosu> it was mostly a tongue-in-cheek remark from me [19:03] <ubitux> qatar is a better name for the project, at least it's not confusing with our libraries ;) [19:03] <Compn> it is easier to grep... [19:04] <Compn> maybe there is too much petty infighting still around :\ [19:05] <kurosu> ubitux, http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=restore.*… <- what I meant [19:06] <ubitux> well, paste the link [19:06] <wm4> it's nice how this discussion is so centered around accusing the other side of stuff [19:07] <wm4> rather than any kind reconciliation [19:07] <kierank> wm4: makes a change [19:07] <wm4> I guess this is totaler krieg between the multimedia libs [19:07] <kierank> wm4: as a said in berlin - it's like the protestants and the catholics [19:07] <wm4> lol [19:07] <kierank> both consider themselves the true origin [19:07] <wm4> even they are able to work together sometimes [19:16] <funman> kierank: we all know who's the true upstream [19:16] <kurosu> after 3 years and the situation crystallizing into such strong opinions, I don't see a reconciliation happening without a few departures [19:18] <kierank> they'll be some kind of good friday agreement [19:18] <kierank> where each side acknowledges the right to exist [19:18] <kierank> in roundabout terms [19:19] <Daemon404> iwshful thinking [19:20] <Daemon404> some people wont even say 'ffmpeg' or 'libav' [19:20] <funman> 'vlc' [19:22] <Daemon404> 'dolby' [19:23] Action: kierank ENOCARRIER [19:24] <kierank> thardin: https://tech.ebu.ch/news/ebu-hosts-open-source-meetup-at-ibc-2014-07aug14 [19:24] <kierank> (if you are going to ibc) [19:24] <thardin> nope [19:24] <thardin> thanks for the tip though [20:03] <cone-128> ffmpeg.git 03Clément BSsch 07master:06362ab7509e: avfilter/dctdnoiz: rewrite [f/i]dct [20:20] <cone-128> ffmpeg.git 03Marek Fort 07master:054c506e3da3: avformat/mov: XMP metadata suppor. [20:20] <cone-128> ffmpeg.git 03Marek Fort 07master:4e70104aba42: avformat/movenc: XMP metadata support. [20:39] <cone-128> ffmpeg.git 03Felix Abecassis 07master:a7e541c9926d: h264: fix interpretation of interleved stereo modes [20:40] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:2bf87dcc2d30: Merge commit 'a7e541c9926d531a100ba0d36f4e56956dd84651' [20:43] <llogan> anyone interested in being backup mailing list maintainer? you probably won't have to do anything unless i am travelling [20:44] <llogan> such as on aug 10-17. [20:52] <cone-128> ffmpeg.git 03Diego Biurrun 07master:ad1ee5fa75a3: avcodec: Undeprecate reordered_opaque [20:52] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:a561662fb722: Merge commit 'ad1ee5fa75a3d69d54c0691048939e94ab5a7f0d' [21:03] <cone-128> ffmpeg.git 03Diego Biurrun 07master:bb789016d423: swscale: Undeprecate sws_getContext() [21:03] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:4917fecf4a00: Merge commit 'bb789016d423d2cfacd2904ac66257bdf7f0964e' [21:03] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:c6a596f62685: doc/APIchanges: Correct function reference. [21:22] <cone-128> ffmpeg.git 03Diego Biurrun 07master:b7b1bf9166ac: mpegts: K&R formatting cosmetics [21:22] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:ec0b08d20b73: Merge commit 'b7b1bf9166ac3102c401295fdd5d4933c512aa50' [21:32] <cone-128> ffmpeg.git 03Clément BSsch 07master:cec59eb63fef: avfilter/dctdnoiz: add 8x8 dct and make it the default [21:47] <ubitux> michaelni: i jsut went on your blog and went wtf "let's study vibrations of the toothbrush with an oscilloscope... oh hey noodles!" :))) [21:48] <ubitux> (that's awesome, don't change it) [21:58] <cone-128> ffmpeg.git 03Diego Biurrun 07master:6d6bd3a3db24: mpegts: Drop some unnecessary parentheses [21:58] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:066d77650173: Merge commit '6d6bd3a3db24747ae5e10716f4782129c38951f6' [22:00] <wm4> | + Default duration: 0.042ms (24000.384 frames/fields per second for a video track) at 421 [22:00] <wm4> what the heck [22:01] <wm4> maybe this is the wrong channel [22:06] <Compn> whats the question ? [22:06] <Compn> :P [22:06] <cone-128> ffmpeg.git 03Diego Biurrun 07master:f8ab9f2fe373: mpegts: Avoid unnecessary variable shadowing [22:06] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:3224a319c0b8: Merge commit 'f8ab9f2fe3730a0e8c3c1eb7c486f7c0041fb3fc' [22:41] <wm4> "ffprobe file.mkv -count_frames" prints nothing [22:41] <wm4> wtf? [22:41] <wm4> it just takes 3 minutes or so fully decoding (and discarding) the whole file [22:44] <cbsrobot-> what about ffprobe -count_frames file.mkv [22:45] <wm4> same [23:06] <cone-128> ffmpeg.git 03Diego Biurrun 07master:5b220e1e19c1: mpegts: Fix memory leaks and related crashes in mpegs_write_header() [23:06] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:efcf1fcad325: Merge commit '5b220e1e19c17b202d83d9be0868d152109ae8f0' [23:06] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:92deb28945a5: avformat/mpegtsenc: use the correct deallocation functions [23:07] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:5f8300afc653: mpegtsenc: Check dynarray_add() for failure [23:07] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:12b59e57f3d7: avformat/mpegtsenc: Use correct deallocation code on failure of pids array allocation [23:07] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:31eaecfee9d8: avformat/mpegtsenc: Free services array completely on failure [23:07] <cone-128> ffmpeg.git 03Michael Niedermayer 07master:92a28e9f5621: avformat/mpegtsenc: Use av_freep() for pids [23:54] <cone-128> ffmpeg.git 03Mark Reid 07master:f1e626a3570c: avformat/mov: read reel_name metadata from tmcd atom [00:00] --- Fri Aug 8 2014

1 0

ffmpeg.log.20140807
by burek 08 Aug '14

08 Aug '14

[02:29] <Baked_Cake> 16:47] Baked_Cake: can any one give me so possible options for encoding mpeg2 interlaced video, im using ffmpeg [02:29] <Baked_Cake> 7 Baked_Cake: so far im trying -flags +ilme+ildct [02:29] <Baked_Cake> 7[16:48] Baked_Cake: i cant seem to fid these optionss in the ffmpeg guide [02:30] <c_14> If you want to interlace the video, you could try the interlace filter. [02:34] <Baked_Cake> i was hoping to find a guide that autline each function and the syntax but i cant seem to find it [02:34] <c_14> https://trac.ffmpeg.org/wiki/FilteringGuide [02:34] <c_14> https://ffmpeg.org/ffmpeg-filters.html [02:35] <Baked_Cake> thx ill take alook [03:19] <Baked_Cake> do these commands on http://ffmpeg.org/ffmpeg-filters.html#Filtergraph-description work for windows [03:19] <Baked_Cake> im using batch files to run ffmpeg but im not quite getting the syntax right for stuff like yadif [03:19] <ersatzbeardr> i'm using them [03:20] <ersatzbeardr> on windows [03:21] <ersatzbeardr> for example http://dpaste.com/0N9A745.txt (bat file) [03:23] <Baked_Cake> ah i needed to add the "" stuff that came after -vf [09:43] <termos> I get a weird seeminly random crash in av_interleaved_write_frame when writing flv format to rtmp streams, seems to be crashing in avio_write when flushing some buffer. [09:43] <termos> Anything I should watch out for here? [10:20] <slowguy> i have an mkv(h264) video..i have extracted two segments out of it [10:21] <slowguy> 1. a segment from non key-frame to a keyframe...which was supposed to be re-encoded [10:21] <slowguy> 2. segment from a key frame to a key frame [10:21] <slowguy> both of them play fine individually [10:23] <slowguy> but when i join the two using concat muxer...combined video first plays fine but as soon as it reaches second part only green or grey or artifacts are seen [10:23] <slowguy> someone please tell me what is wrong here [11:16] <Lord> hi, i'm trying to record the audio from my computer with alsa (no pulseaudio). Is it possible to capture an output of alsa ? [11:17] <DopeLabs> yes http://trac.ffmpeg.org/wiki/Capture/ALSA [11:20] <Lord> DopeLabs arecord -l only shows capture devices from alsa, no output devices. [11:25] <DopeLabs> alsamixer might be of use [12:12] <[A3G1S]> Hey I am trying to statically link libass with FFmpeg, and getting this error ass_font.c:function ass_font_get_glyph: error: undefined reference to 'FT_GlyphSlot_Oblique' [12:12] <[A3G1S]> can any help me with this ? [12:37] <coalado> Hi there, I'd like to extrat an audiostream from a video without a special output format. I try: ffmpeg.exe, -i, bla.audio.tmp, -acodec, copy, -vn, bla.audio, -y [12:38] <coalado> ffmpeg says: [12:38] <coalado> Stream #0:0: Video: h264 (High), yuv420p, 400x224, 26 kb/s, 30.33 fps, 29.97 tbr, 1k tbn, 59.94 tbc [12:38] <coalado> Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 128 kb/s [12:38] <coalado> and Unable to find a suitable output format for 'bla.audio' any idea what might be wrong? [12:40] <sfan5> tried ffmpeg -i audio.in -map 0:a -c copy -f rawvideo audio.out ? [12:41] <DopeLabs> done use ,'s in your commands? [12:41] <DopeLabs> *dont [12:41] <DopeLabs> or does the windows version use commas [12:44] <sfan5> I don't think you need commas [12:45] <DopeLabs> ffmpeg -i input -vn -c:a copy output should really do te trick though.. if you dont specify a standard file extension such as output.m4a than you should prob specify the output format with -f [13:50] <coalado> DopeLabs: I copy&pasted the commands from the ide debugger.. the commas just are the ids string array representation [14:30] <termos> when calling av_copy_packet the AVPacket content does not seem to be copied. My copy is empty with AV_NOPTS_VALUE [15:27] <slowguy> i have a key frame at 3.136 so i give this command [15:27] <slowguy> G:\contribs\ffmpeg-20140731-git-d76675d-win64-shared\bin\ffmpeg.exe -ss 3.136 -i C:\wamp\www\manas/vids/dexter/dexter.mkv -to 3.3785 -c:v copy -an C:\wamp\www\manas/vids/dexter/dexter_2.mkv [15:28] <slowguy> the extracted video starts from correct time i.e. 3.136 sec [15:28] <slowguy> but its length is around 3 seconds [15:29] <slowguy> but according to command it was supposed to be only 3.3785-3.136 = 0.2425 sec only [15:29] <Mavrik> termos, see the source of that call, it's very simple [15:29] <Mavrik> it's possible that your types aren't really correct :) [15:29] <slowguy> can you help me ? [15:31] <Mavrik> slowguy, do you have a keyframe at 3.3785? [15:31] <slowguy> no [15:32] <slowguy> next is at 5.405 [15:32] <DopeLabs> slowguy: http://superuser.com/a/141343 [15:32] <sacarasc> slowguy: Maybe try using -vframes [15:33] <Mavrik> DopeLabs, that answer doesn't really help him when ffmpeg overruns the duration :) [15:33] <Mavrik> yeah, -vframes could help [15:33] <slowguy> actually i have read this post but it says about seek to a specific point but nothing is said about precision of --to option [15:33] <Mavrik> but the probable issue is that ffmpeg just does the conservative thing when doing -copy and seeks to next IDR frame [15:34] <iive> i think -to works like -t if the start time of the output file is 0 [15:35] <iive> i might be wrong. [15:35] <Mavrik> of course it does :) [15:36] <slowguy> --to option is not accurate? [15:37] <slowguy> then how do we cut to a precise time? [15:38] <iive> what i am trying to say is that the -to is not end-time in the input, it is end-time in the output, so when the output starts at 0s time, then it works just like -t , aka it works as duration. [15:38] <iive> you set it to 3.3s and it gives you 3 seconds. [15:38] <slowguy> oh i see [15:38] <iive> try -t 0.2425 and see if that works. [15:39] <slowguy> give me a min please [15:40] <slowguy> looks like it works [15:42] <slowguy> so -t or -to options is always precise in any case with re-encoding or without re-encoding? [15:44] <slowguy> actually what i was trying to do was cut a video from any location to any location without re-encoding the whole thing [15:44] <slowguy> so i probed all key frames [15:45] <slowguy> never mind [17:53] <brontosaurusrex> interesting: "MPD uses a flat file database to maintain the basic music file information when it is not running. Once the daemon is started, the database is kept completely in-memory and no hard disk access is necessary to look up or search for local audio files" [17:53] <brontosaurusrex> from http://en.wikipedia.org/wiki/Music_Player_Daemon [17:54] <sacarasc> And..? [17:55] <brontosaurusrex> wrong channel, sorry [19:11] <br1> How do I silence the log lines avery few frames while encoding, while leaiving the resto of the info level lines? [19:37] <DopeLabs> your looking for loglevel [19:37] <DopeLabs> https://ffmpeg.org/ffmpeg.html#Generic-options [19:40] <lovesax> h264 live streaming over http with ffserver... is it possible? I seem to be running into container-issues with all options. [19:43] <DopeLabs> command? [20:20] <br1> DopeLabs: I have loglevel = info, If I lower it one notch to warning I lose some messages about the format of the input that I want. [20:38] <DopeLabs> what/how much information do you need about your input that you cant decreas to warning [20:40] <llogan> michaelni: i'm going to be gone from Aug 10-17, so it will probably be up to you to approve messages in ML queue (since compn always forgets). [20:40] <llogan> unless someone else wants to volunteer [20:41] <Dark-knight> !builds [20:41] <llogan> i'll have no internet or mobile connection [20:41] <Dark-knight> is there a command to show the link to different builds? [20:41] <llogan> ah, i thought i was in -devel... [20:42] <Dark-knight> nope [20:42] <llogan> Dark-knight: /msg fflogger !help [20:42] <Dark-knight> thanks [20:52] <RedSpartan> Where do you find the ffserver download? [20:55] <RedSpartan> I downloaded ffmpeg for windows but it didnt come with ffserver....is it the same thing? and google wont help me [20:56] <Dark-knight> omg [20:57] <Dark-knight> can i please have the windows link? [20:58] <RedSpartan> http://ffmpeg.zeranoe.com/builds/win32/static/ [20:59] <Dark-knight> thanks [21:00] <Dark-knight> dont know why fflogger couldnt give me that [21:01] <RedSpartan> its on a different server than the linux static [21:13] <llogan> that's an old anchor name... [21:15] <RedSpartan> that just took me to the downloads page lol [21:21] <RedSpartan> g2g [21:25] <Dark-knight> why isn't there one that links straight to Zeranoe's dl page? [21:33] <DopeLabs> its the very first hit googling 'ffmpeg windows builds' [21:36] <llogan> http://ffmpeg.org/download.html#build-windows [21:37] <llogan> i guess that didn't work like i expected. [21:38] <Dark-knight> no it worked, i got the the windows dl page [21:39] <Dark-knight> i just thought it should link directly to the windows page instead of going a roundabout way [23:41] <phelps> just seen the new site, it really looks nice [23:58] <jrgill> Trying a simple pan audio here by dropping the right channel but seems to be transcoding. Shouldn't the operation be lossless? ffmpeg.exe -i in.m4v -af pan="stereo:c0=c0" -c:v copy out.m4v [23:58] <jrgill> cf. https://ffmpeg.org/ffmpeg-filters.html#Remapping-examples [00:00] --- Fri Aug 8 2014

1 0

ffmpeg-devel.log.20140806
by burek 07 Aug '14

07 Aug '14

[00:03] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:70cd3b8e659c: mmvideo: check horizontal coordinate too [00:03] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:3ce55d2e32ce: Merge commit '70cd3b8e659c3522eea5c16a65d14b8658894a94' [00:14] <jamrial> the new cpu-test is also failing in quite a few slots [00:22] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:a7153444df90: huffyuvdec: check width size for yuv422p [00:22] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:296b01d7f91f: Merge commit 'a7153444df9040bf6ae103e0bbf6104b66f974cb' [00:42] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:1b168e3bcf6f: avutil/cpu: fix cpu-test to work with ffmpegs cpuflags syntax [00:42] <michaelni> jamrial, fixed, anything else ? (as you say "also") [00:43] <jamrial> i said also after what Daemon404 mentioned about pixelutils [00:44] <jamrial> but for that matter, fate-cpu is not outputting the list of supported instruction sets like it used to before we merged the libav version of the test [00:45] <jamrial> i tried making it use runecho() again instead of run(), but it didn't work [00:48] <michaelni> ubitux, pixelutils is failing on ppc, are you awake/have time ? or should i take a look ? [00:50] <Daemon404> its failing on windows too [00:50] <Daemon404> on some compilers [00:51] <Daemon404> both icl [00:58] <jamrial> icl and msvc. gcc is fine [00:58] <jamrial> so probably alignment. again [01:17] <michaelni> jamrial, posted patch for cpu echo issue [01:40] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:84ac2f93ca11: avutil/pixelutils: avoid on stack arrays [01:40] <michaelni> ubitux, maybe fixed [01:41] <Daemon404> unchecked mallocs [01:45] <jamrial> michaelni: it works now with both patches, thanks [01:46] <michaelni> ok, is there any reason that iam missing why they used stderr ? [01:50] <jamrial> using run() instead of runecho() makes the test "fail" if outputting to stdout [01:51] <jamrial> libav doesn't have runecho() and probably also doesn't need/care about outputting the values to stdout anyway [01:56] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:6552b0558a15: avutil/pixelutils: check for malloc failure [02:07] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:a8689ba87207: tests/fate/libavutil: switch cpu-test back to runecho so its results are vissible [02:07] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:9101ef6757e9: avutil/cpu: output cpu data to stdout [02:08] <michaelni> applied the 2 cpu test fixes, i assume noone knows why it went to stderr [02:56] <jamrial> has anyone seen plepere lately? [04:00] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:a32e306be3a2: doc/muxers: document which applications are known to need disable_chpl. [04:28] <cone-118> ffmpeg.git 03Timothy Gu 07master:9f02a2b22400: transcode_aac: fix const return value [05:44] <Compnn> [23:23] <runeks> Is visualizing motion vectors still supported? mpv crashes for me when I try to play a file with the option --vd-lavc-o=vismv=1 [05:44] <Compnn> another vismv user [05:44] <Compnn> thanks michaelni for keeping these features around [05:45] <Compnn> i didnt know vismv was such a used feature... :) [09:40] <ubitux> michaelni: thanks for the fixes [12:46] <cone-167> ffmpeg.git 03Christophe Gisquet 07master:6786848585dc: hevc_deblock: change tc type [14:38] <cone-167> ffmpeg.git 03Michael Niedermayer 07master:efc4fe9d74a5: avutil/cpu: add aarch64 entries to 2nd table [14:38] <cone-167> ffmpeg.git 03Michael Niedermayer 07master:fe0157a19ac0: avutil/cpu: Make cpu flag names match between cpu-test and av_parse_cpu_caps() tables [14:38] <cone-167> ffmpeg.git 03Michael Niedermayer 07master:6b1df5544e8d: avutil/cpu: check av_parse_cpu_caps() table during cpu-test [15:59] <cone-167> ffmpeg.git 03Michael Niedermayer 07master:305f72aee77b: avcodec: Change get_pixels() to ptrdiff_t linesize [16:18] <ubitux> michaelni: thx for ptrdiff_t [16:19] <michaelni> np [16:46] <michaelni> mraulet, is the "Christophe Gisq (1.3K) [FFmpeg-devel] [PATCH 0/4] Exploit compile-time constant" patchset ok or could this conflict with future things in hevc ? [16:47] <mraulet> at the moment there is a conflict with arm mc code [16:47] <mraulet> I need to check it out with the main developper of this code [16:49] <michaelni> ok, ill wait, please ping me if there are news [16:52] <mraulet> yes [18:33] <mark4o> lwn article on ffmpeg in Debian http://lwn.net/SubscriberLink/607591/7190d6c0b7274eb0/ [18:40] <thardin> three months to go only? huh [18:58] <Daemon404> heh [18:58] <Daemon404> the article is a bit uh... one sided [18:59] <Daemon404> "he mplayer video player application has run into trouble due to its being primarily written and tested for ffmpeg rather than libav" [18:59] <Daemon404> tl: it canibalizes and rapes teh API [19:00] Action: kierank struggles with ffmpeg asm crashes [19:01] <nevcairiel> Still on that problem kierank? [19:02] <kierank> managed to comment out some asm init in float dsp to fix it at work [19:02] <kierank> but not at work today [19:02] <kierank> so need to do it again at home [19:02] <wm4> Daemon404: yep [19:02] <wm4> Daemon404: most of mplayer's problems with libav is because it's not using it in clean ways [19:03] <nevcairiel> At least you seem to be close to know which part exactly is broken [19:03] <wm4> cannibalizes the build system, accesses API internals, ... [19:04] <wm4> it's also using some deprecated (and removed in Libav API), as well as explicitly ffmpeg-only API [19:04] <wm4> I can only imagine the dirty hacks distro maintainers used to make it somehow work on libav [19:05] <JEEB> they just haven't updated it in years :P [19:05] <JEEB> at least if we are talking about debian-kei [19:06] <Daemon404> thats fine, its not like mplayer is maintained [19:06] Action: Daemon404 runs [19:07] <wm4> reimar even fixes mencoder bugs [19:16] <kurosu> kierank, did you try to hack the cpu flag reporting so that it doesn't report mmx ? sounds like an emms issue [19:16] <kurosu> (alternatively, if you know where dsp or float code is used, insert emms_c() appropriately) [19:17] <Daemon404> i think he went /care and use avresample [19:17] <kierank> can't use avresample for various reasons [19:17] <kierank> doesn't seem to be an avr problem anyway [19:17] <kierank> seems to be float dsp [19:17] <Daemon404> ah [19:17] <kierank> swr* [19:18] <kurosu> another thing with missing emms is sometimes that the thread/program slows down to a crawl [19:44] <cone-140> ffmpeg.git 03Christophe Gisquet 07master:4e128ab0b1b3: x86: vpx/h264/hevc/mpeg2: share constants [19:44] <cone-140> ffmpeg.git 03Christophe Gisquet 07master:71db2d08b1f1: x86: better share ff_pw_2 [19:44] <cone-140> ffmpeg.git 03Christophe Gisquet 07master:6622a6cff30f: x86: dwt: better share constants [19:45] <cone-140> ffmpeg.git 03Christophe Gisquet 07master:51dd80e7510e: x86: diracdsp: reuse constants [19:45] <cone-140> ffmpeg.git 03Christophe Gisquet 07master:75837e9add4e: x86: sbrdsp/fft: reuse ps_neg constant [20:36] <cone-140> ffmpeg.git 03Vittorio Giovara 07master:cbc808d726af: jpeg2000: enable 4 component pixel formats [20:36] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:5836fe20c686: Merge commit 'cbc808d726afdf53d866264722785c1304c17390' [20:56] <cone-140> ffmpeg.git 03Martin Storsjö 07master:ed6d9ce914d5: configure: Include the armcc build number in the compiler identification [20:56] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:40a820d6d865: Merge commit 'ed6d9ce914d552eeda16af857da97c4b1aea1e3f' [22:08] <ubitux> > All of the code from libav is merged into FFmpeg without credit [22:08] <ubitux> lol @ diego [22:10] <ubitux> and then advertising for a dead project (mplayer2) geez [22:12] <jamrial> is this about debian? [22:12] <ubitux> jamrial: Diego's long comment on lwn.net [22:17] <ubitux> about the "All of the code from libav is merged into FFmpeg without credit or thanks", if anyone with a lwn account could tell him that the commit merges keep full authorship of everything and that we even mention them in the 2nd paragraph of our release notes ? [22:18] <Daemon404> im not touching this article with a 10 foot pole [22:18] <Daemon404> idiots, idiots everywhere, on all sides. [22:18] <wm4> ubitux: diego only wrote "Just check out the homepages of mpv http://mpv.io/ and mplayer2 http://www.mplayer2.org/ and see for yourself." [22:18] <ubitux> wm4: see further [22:19] <wm4> I'm not sure, maybe the mpv one is too hipster [22:19] <ubitux> > Forget MPlayer, its successor mplayer2 and mpv, the successor of mplayer2, are much better. [22:19] <wm4> aha [22:19] <wm4> if I read that right, he advertises mpv over _two_ dead projects [22:20] <ubitux> he's saying mplayer2 (and mpv) are better because they build with libav :) [22:20] <beastd> hi [22:20] <wm4> have you ever seen the shit MPlayer does [22:20] <ubitux> he just forget to mention that mplayer2 is not maintained while mplayer still is [22:21] <wm4> I know you made some patches [22:21] <wm4> but did you ever go deep down the sources [22:21] <ubitux> kind of [22:21] <wm4> there are things that can scare a man for life [22:21] <ubitux> i'm not saying it's the best software in the world [22:21] <ubitux> the problem is just that mplayer2 is dead and not maintained [22:21] <beastd> And that makes mplayer2 much better, of course :) [22:21] <wm4> I mean not to belittle the greatness of mplayer, but it's a thing from the past [22:22] <wm4> and mplayer2 is just dead [22:22] <beastd> wm4: that is exactly the point [22:22] <beastd> wm4: mplayer is mplayer [22:22] <ubitux> there is still at least one developer on mplayer which you can poke if you have a security issue or a regression and get it fixed [22:22] <ubitux> so it's maintained [22:22] <Timothy_Gu> where's Diego's post? [22:22] <ubitux> Timothy_Gu: lwn.net [22:22] <wm4> on the bottom [22:23] <ubitux> beastd: do you have an lwn account? [22:23] <Timothy_Gu> oh shit on lwn's title page? [22:23] <beastd> no, but i have read the lies from DonDiego is spreading there [22:23] <ubitux> Timothy_Gu: http://lwn.net/SubscriberLink/607591/7190d6c0b7274eb0/ [22:23] <beastd> and the really good article of Jonathan Corbet [22:24] <ubitux> well just a summary [22:24] <ubitux> i just would like to fix the "All of the code from libav is merged into FFmpeg without credit or thanks" which is totally wrong [22:24] <wm4> ubitux: well it's somewhat true [22:24] <beastd> Interesting, I would say git merge usually preserves author ship of the commits [22:24] <ubitux> i'm repeating myself but someone needs to tell him the commit merges keep full authorship of everything (even when we don't merge) and that we even mention them in the 2nd paragraph of our release notes [22:25] <wm4> ffmpeg will announce support for feature X, but never tell that feature X was in fact implemented by Libav [22:25] <ubitux> wm4: look at our release notes [22:25] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=RELEASE_NOTES;hb=489d066 [22:25] <jamrial> yeah, you could say every other point is subjective, but that comment is objectively and absolutely wrong [22:25] <ubitux> we do, and every external commit is a merge [22:25] <wm4> ubitux: ok [22:25] <ubitux> er last sentence badly worded [22:25] <wm4> I'm pretty sure this didn't always happen [22:26] <wm4> does ffmpeg still change the copyright headers of new files merged from libav? [22:26] <ubitux> i think we don't really care about this [22:26] <ubitux> libav does it at least [22:26] <Timothy_Gu> ubitux: thx [22:27] <nevcairiel> Libav removes copyright lines because its considered trivial files. :P [22:27] <wm4> yeah [22:27] <wm4> except when it's their own copyright [22:27] <jamrial> both projects change "ffmpeg" with "libav" and vice versa for every file. libav however sometimes removes author names (especially when moving code to new files), whereas ffmpeg doesn't [22:28] <wm4> and Diego is also right in that Libav has done massive cleanups [22:28] <wm4> they've broken the API a lot, but in turn we have things like AVFrame refcounting [22:28] <wm4> I see no such developments on ffmpeg [22:28] <ubitux> aside dropping feature and reindenting, you have the ref counting and what else? [22:29] <wm4> in fact, mini even asked on the ML whether to merge AVFrame refcounting at all [22:31] <ubitux> i'm fine admitting ref counting is kind of a good thing (at least if you omit a few details such as squashing everything in a single commit), but can you mention much of other things? [22:31] <beastd> wm4: asking wether to merge the ref counting was of course the right thing to do. what is wrong with asking your development community about opinions??? [22:31] <ubitux> wm4: there isn't that much "cleanups" [22:31] <ubitux> wm4: i remember BBB cleanup swscale on libav side though [22:32] <wm4> beastd: nothing, I'm just saying that ffmpeg was being quite conservative against such large changes [22:32] <nevcairiel> K&R all the way! [22:32] <wm4> <ubitux> wm4: there isn't that much "cleanups" <- it all adds up [22:32] <ubitux> wm4: it's not like it was exactly trivial :) [22:32] <beastd> nevcairiel: yes and that is hell! never call that a real code cleanup [22:32] <ubitux> wm4: yes, it makes 2 [22:32] <wm4> part of why ffmpeg is still "better" than Libav is _because_ ffmpeg merges everything from Libav [22:32] <wm4> otherwise there's a good chance that Libav would have "won" [22:33] <ubitux> and because we add some great things above too maybe? [22:33] <jamrial> i thought making big changes (like ref counting was) on our side alone while libav doesn't means breaking (or making difficult) the drop-in compatibility [22:33] <jamrial> which is why such big changes never start in ffmpeg [22:33] <wm4> ubitux: some... I'm fond of subtitle and filter additions, which allowed me to delete scary mplayer code [22:34] <jamrial> we basically let libav take the API wheel to make downstream life easier [22:34] <beastd> jamrial: unfortunately :( that is exactly the downside of merging everything from Libav [22:34] <ubitux> wm4: note that we are blocked about api evolution somehow, not because we are "conservative" but because it can introduce conflicts with Libav in the future [22:34] <wm4> things like vlc do much better with their own subtitle support etc., though [22:34] <ubitux> wm4: that is one of the reason we can't really do evolution to subtitles [22:34] <ubitux> typically [22:35] <wm4> well Libav subtitle support is a complete joke, so I'm not sure why to keep compatibility for _that_ [22:35] <ubitux> because we still have people using AVSubtitle [22:35] <nevcairiel> If everyone does their own though, upstream never improves. So kinda it sucks every time vlc develops something internally instead of contributing instead. :p [22:35] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:7a34b7d80f35: avcodec/dca: Make ff_dca_convert_bitstream() available to libavformat, needed for dts_probe() [22:35] <cone-140> ffmpeg.git 03Michael Niedermayer 07master:6a4469974650: avformat/dtsdec: check more of the dca headers in dts_probe() [22:36] <ubitux> wm4: but you did keep subreader.c no? [22:36] <ubitux> (for compat with libav) [22:36] <wm4> ubitux: yes, also utf-16 (I know...) [22:36] <ubitux> :) [22:37] <ubitux> wm4: one good example for api limitation is the matroska demuxer [22:37] <ubitux> you know that we still don't export them? [22:37] <ubitux> i mean, verbatim [22:37] <wm4> you mean the ASS packet format? [22:37] <ubitux> i'm pretty sure vlc for example will break as soon as it will uses avformat demuxer for mkv [22:37] <ubitux> because it has parsing for it last time i looked [22:37] <wm4> wut, vlc has its own mkv demuxer [22:38] <wm4> for a reason [22:38] <ubitux> wm4: it also has avformat (optional) support somehow iirc [22:38] <jamrial> nevcairiel's lav filters also use its own demuxer (wrapper for mkvparser afaik) [22:38] <wm4> sure, vlc probably uses it for obscure formats where it didn't NIH something yet [22:38] <jamrial> nobody seems to like lavf's matroska demuxer :P [22:38] <nevcairiel> Speaking of breaking things, libav wants to bump. Perfect time to change the output of ass ;) [22:39] <wm4> jamrial: I think it got better recently, but it still lacks support for some weird matroska features [22:39] <jamrial> afaik all the new developments were added [22:39] <wm4> nevcairiel: AFAIK it's bound to the bump [22:39] <ubitux> nevcairiel: except they won't [22:39] <jamrial> matroska v4 elements are all supported [22:39] <wm4> jamrial: ordered chapters, editions [22:39] <nevcairiel> It's old features that are missing [22:39] <jamrial> ah, i see [22:40] <wm4> these are not so easy, because they break the demuxer abstraction [22:40] <ubitux> don't we have stuff like hls demuxer [22:40] <ubitux> that basically do the same thing? [22:40] <wm4> I'm not very fond of how HLS is implemented [22:40] <wm4> it breaks player caching [22:41] <nevcairiel> Haalis mkv parse code is a terrible mess as well. It uses alloca and setjmp all the time. But it parses the right things and was easy to wrap, so meh [22:41] <jamrial> if libav wants to bump we should take the chance to make some cleaning asap [22:41] <wm4> ubitux: and in fact, there was a patch years ago that makes lavf implement ordered chapters in a way HLS works today [22:41] <ubitux> yes [22:42] <ubitux> i would love ordered chapters in mkv [22:42] <wm4> anyway, I'd be fine if lavf would just export the list of segments, and leave the rest to the player... then I could drop my internal demuxer [22:42] <wm4> elenril even had a patch for something similar [22:43] <ubitux> i think i already asked but i supposed the AVChapter are not enough? [22:44] <wm4> currently, no [22:44] <nevcairiel> There is a whole boatload of special conditions in that stuff, and no spec (other than it works in Haali Splitter), it took quite a while to get it to a point where I don't get complaints anymore [22:45] <ubitux> AVChapter can't be extended to export more information? [22:45] <wm4> nevcairiel: I guess it's easier if you do it on a higher level [22:45] Action: ubitux wonders if having a filename in AVChapter->metadata would do it :-° [22:45] <wm4> matroska can have per-chapter metadata [22:45] <ubitux> oh well. [22:45] <nevcairiel> Well I kinda did, by wrapping multiple of the haali parser lib [22:45] <wm4> you'd get clashes [22:46] <ubitux> ok [22:46] <wm4> ubitux: http://git.khirnov.net/cgit.cgi/libav/log/?h=playlists [22:46] <wm4> especially http://git.khirnov.net/cgit.cgi/libav/commit/?h=playlists&id=bb611e5a6620da… [22:46] <wm4> though that ignores editions and all that stuff [22:49] <ubitux> on a totally unrelated note, i can't fucking find a way to transform "a*x + a*y + z" into "a*(x + y) + z" with any py library [22:50] <ubitux> and that blocks me to generate a sane dct code >:( [00:00] --- Thu Aug 7 2014

1 0

ffmpeg.log.20140806
by burek 07 Aug '14

07 Aug '14

[02:10] <home> hey guys! [02:10] <home> how do I read a ffmpeg RTMP stream? [02:10] <home> vlc doesn't work :/ [02:16] <rjp421> -f flv -re -i "rtmp://jblive.videocdn.scaleengine.net/jb-live/play/jblive.stream" [02:17] <rjp421> maybe -re before -f flv, if it matters [02:31] <fling> I have a bunch of videos from a security camera. [02:31] <fling> I want to keep only parts of videos containing movement. [02:31] <fling> How to do so? [02:32] <fling> the stupid option is to split videos into separate images and then compare these images, remove dupes and arrange them back. [02:32] <fling> but I will loose sound this way. [02:32] <fling> What is the smart solution? :P [02:37] <c_14> The "smart" solution is writing a motion detection filter for FFmpeg, but you could try messing around with the select filter and scene detection. It might work. ymmv [02:40] <fling> hmm mhmmmm [02:40] <fling> c_14: thanks :> [03:07] <home> rjp421, I am on my rpi [03:07] <home> gonna try something with nginx [03:07] <home> and hopefully I can point it to there :/ [03:16] <jrgill> How would you go about center panning the left channel in an m4v? I encoded a stereo recording of a balanced signal and got an inverted right channel. [03:17] <DopeLabs> https://trac.ffmpeg.org/wiki/AudioChannelManipulation [03:26] <jrgill> Ah, I think I just had the stream since video is 0. [08:02] <chunyu> always get "input/outut error" when converting a amr file to a mp3 file [08:02] <chunyu> I used the latest static binary on both Linux and Mac and all failed. [08:03] <chunyu> Here's the detailed log. [08:03] <chunyu> $ ffmpeg -i se_t630.amr se_t630.mp3 [08:03] <chunyu> ffmpeg version 2.3.1 Copyright (c) 2000-2014 the FFmpeg developers [08:03] <chunyu> built on Aug 5 2014 23:26:16 with Apple LLVM version 5.1 (clang-503.0.40) (based on LLVM 3.4svn) [08:03] <chunyu> configuration: --prefix=/usr/local/Cellar/ffmpeg/2.3.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb [08:03] <chunyu> libavutil 52. 92.100 / 52. 92.100 [08:03] <chunyu> libavcodec 55. 69.100 / 55. 69.100 [08:03] <chunyu> libavformat 55. 48.100 / 55. 48.100 [08:03] <chunyu> libavdevice 55. 13.102 / 55. 13.102 [08:04] <chunyu> libavfilter 4. 11.100 / 4. 11.100 [08:04] <chunyu> libavresample 1. 3. 0 / 1. 3. 0 [08:04] <chunyu> libswscale 2. 6.100 / 2. 6.100 [08:04] <chunyu> libswresample 0. 19.100 / 0. 19.100 [08:04] <chunyu> libpostproc 52. 3.100 / 52. 3.100 [08:04] <chunyu> [amr @ 0x7fecc2021800] Estimating duration from bitrate, this may be inaccurate [08:04] <chunyu> Input #0, amr, from 'se_t630.amr': [08:04] <chunyu> Duration: 00:00:05.57, bitrate: 5 kb/s [08:04] <chunyu> Stream #0:0: Audio: amr_nb (samr / 0x726D6173), 8000 Hz, mono, flt, 5 kb/s [08:04] <chunyu> Output #0, mp3, to 'se_t630.mp3': [08:04] <chunyu> Metadata: [08:04] <chunyu> TSSE : Lavf55.48.100 [08:04] <chunyu> Stream #0:0: Audio: mp3 (libmp3lame), 8000 Hz, mono, fltp [08:04] <chunyu> Metadata: [08:04] <chunyu> encoder : Lavc55.69.100 libmp3lame [08:04] <chunyu> Stream mapping: [08:04] <chunyu> Stream #0:0 -> #0:0 (amr_nb (amrnb) -> mp3 (libmp3lame)) [08:04] <chunyu> Press [q] to stop, [?] for help [08:04] <chunyu> se_t630.amr: Input/output error [08:04] <chunyu> size= 6kB time=00:00:05.62 bitrate= 8.6kbits/s [08:04] <chunyu> video:0kB audio:6kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 4.531250% [08:05] <ghospich> Lol, there is pastebin for that. [08:05] <ghospich> Dunno what's wrong though. [08:07] <chunyu> oh. forgot about pastebin. here's the url http://pastebin.com/qPZ4BrtM [08:12] <chunyu> just tried some old version binaries, still no luck. [09:27] <Binbo> what are the configure flags required to compile ffmpeg if I want to decode h264 stream only? [09:28] <Binbo> my shared library size is as high as 42MB due to libavformat [09:35] <Mavrik> Binbo, --disable-everything --enable-demuxer="demuxers,you,need" --enable-decoder="h264" [09:35] <Mavrik> demuxers are dependant on formats of input files [09:35] <Mavrik> also, remember, this will only decode video [09:36] <Binbo> Mavrik : what if I only decode raw h264 stream? [09:36] <Mavrik> you still need a demuxer that lets you do that [09:36] <Binbo> from network [09:36] <Mavrik> and input protocol that lets you read from network [09:36] <Mavrik> pipeline is [09:36] <Mavrik> [Input protocol] ----> [Demuxer] ---> [Decoders] [09:37] <Binbo> Mavrik : I developed my input protocol. I only use ffmpeg to decode nal frame. so that means I only need decoder rite [09:38] <Binbo> I construct the nal myself [09:38] <Mavrik> Binbo, yes, if you only use calls to avcodec.h stuff [09:38] <Mavrik> that means you don't even have to use anything from libavformat or link against it [09:38] <Mavrik> (unless there's some internal dependency) [09:38] <Mavrik> so you're right, you probably don't need any demuxers [09:39] <Binbo> Mavrik : av_register_all wont compile if I remove avformat header [09:39] <Mavrik> Binbo, yes, because that's the initialization function for libavformat :P [09:39] <Mavrik> Binbo, https://www.ffmpeg.org/doxygen/trunk/group__lavf__core.html#ga917265caec45e… [09:39] <Mavrik> you DID just tell me you don't use ffmpeg input protocols, muxers and demuxers? :) [09:41] <Binbo> I have full control of the sender. The stream is send using pure tcp socket. [09:41] <Mavrik> That's not what I asked - av_register_all initializes libav protocols, muxers and demuxers. [09:42] <Mavrik> If you don't use any of those (and just call out to codecs) then you don't need to call it right :) [09:42] <Binbo> Mavrik : I see [09:42] <Binbo> lemme try [09:42] <Mavrik> bbl [09:45] <Binbo> Mavrik : I removed av_register_all and error " no codec provided to avcodec_decode_video2" was shown [10:37] <dreamcat5> how to find when was removed 'attribute_deprecated int avcodec_decode_audio3' ??? [10:38] <dreamcat5> the removal isn't specifically-enough mentioned (not searchable) in APIchanges [10:38] <dreamcat5> can't git blame something that isn't there anymore. binary search ? [10:45] <ubitux> doc/APIChanges git show 0eea212 [10:45] <ubitux> "Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4()." [10:49] <dreamcat5> ubitux: yeah nevermind, [11:16] <tilak> Hello group! I have a question regarding ffserver delays and av_read_frame() processing.. We have a rendering application but we experience delays at the interface between ffserver and the webclient (uses vpx/webm/matroska) [11:18] <tilak> Latency of 300-400 msec is observed from end to end.. I wonder if the server code can be optimized. I have some logs which shows that ffserver goes to WAIT_FEED for a considerable time and it causes some delay. In our use case, it would be ideal to flush the packets immediately with optimized frame processing or reading.. [11:20] <tilak> Any thoughts on this would be appreciated. Thanks :) [12:34] <ghospich> When i am encoding some segment of large video, before encoding start there is huge delay. Looks like this delay is depends on time skipped before segment in original video. Is there any way to fix this? [12:35] <ghospich> I mean `-ss 1000.0` is very slow. [12:49] <ghospich> Looks like mkvmerge can cut segment faster. Thanks anyway. [12:53] <c_14> ghospich: use -ss as an input option instead of an output option [12:57] <slowguy> is it possible that an I-frame is NOT a keyframe? [12:57] <slowguy> i did ffprobe on a video and some frame are I-type but keyframe=0 [12:58] <slowguy> in that case i was wondering whether -ss fast seek will seek to a I-type frame or a frame for which keyframe=1? [12:59] <ghospich> c_14, thanks, i'll try [13:03] <ghospich> c_14, it's really starts faster that way! But subtitles are got screwed up. How to fix subtitles in that method? [13:03] <c_14> Can you pastebin your commandline? [13:03] <ghospich> ok. [13:07] <ghospich> it's pretty messy. http://pastebin.com/z7qD3mD0 [13:14] <ghospich> So, if this http://ffmpeg.org/ffmpeg-filters.html#subtitles-1 had option "ss" everything would be nice. [13:15] <c_14> Hmm, ye. I've had the problem before. Trying to remember how I fixed it. [13:16] <ghospich> Would be nice to have some workaround. [13:17] <c_14> iirc there are 2 options, either use the slow seeking which will seek both, or extract the subs from the mkv, delete all the sub lines up to the point you want and then adjust the times downward by your seek time. But that isn't all _that_ fun. [13:19] <ghospich> Maybe there is some easier option? [13:20] <c_14> hmm, what seems to work for me is cutting the mkv with ffmpeg -ss $time -t $time -i mkv -map 0 -codec copy out.mkv and then running ffmpeg -i out.mkv -vf subtitles=out.mkv [other stuff] [13:21] <ghospich> Okay, that also can be done by mkvmerge. But what about external subtitles? [13:22] <c_14> You mean if you have an ass or srt file and want to hardsub? [13:22] <ghospich> Yep [13:23] <ghospich> Pretty common for me, because russian subtitles usually come in different directory, not even inside mkv. [13:24] <c_14> Either mux the subs into the mkv while seeking and do the second step as above, or use slow-seeking. Not many good options to be honest. You could try opening a feature request for an ss option for the subtitles filter, not sure how hard it would be but it can't hurt. [13:25] <ghospich> Yep, i think i'll just mux subs into mkv, and then cut needed segment with mkvmerge. [13:26] <ghospich> I'll look into ffmpeg bugtracker then. Thanks! [13:26] <c_14> You can also mux with mkvmerge, so you might be able to cut and merge in one step. Not sure though. [13:26] <ghospich> Two steps is okay too, as far as they are fast. [14:28] <slowguy> suppose a group of picture is from 10sec to 15sec and i cut a video with -to 12 -c copy.... will it cut to 15 sec or 12 second? [14:45] <klaxa|work> depending on the codec that could output 12 seconds [14:45] <klaxa|work> have you just tried running it and see what happens? [15:00] <Ganimede> Hi all! [15:01] <Ganimede> anyone can help me ? [15:02] <sfan5> just ask your question [15:02] <sfan5> if someone can help, they will [15:03] <Ganimede> ok so... [15:04] <Ganimede> if i use ffmpeg to extract keyframes with the option -vstats, in the output vstats file there's a voice named "time"... but this is not the time of the extracted keyframe. What kind of time is that ? [15:08] <Ganimede> i just want to know the timestamp of the extracted keyframe [15:16] <Mavrik> Ganimede, I don't exactly know what vstats outputs, but perhaps it's PTS? [15:17] <Ganimede> vstats give to me some informations about the generated keyframes [15:17] <Ganimede> and the "time" seems to be a time slice instead of a timestamp [15:18] <Ganimede> now i've tried to out a debug file during extraction [15:18] <Ganimede> here is PTS and T [15:18] <Ganimede> T is the param that i should use [15:18] <Ganimede> i've discover it now [15:19] <Ganimede> but the strange thing is that i must use the debug output to know the timestamp o single keyframes [16:54] <Ganimede> goodbye! see ya! [16:55] <Fjorgynn> good [18:22] <ghospich> How to make audio tracks in mkvmerge to start from 0? Like in mpv, separate order for each type of track. [18:23] <ghospich> Video tracks: 0; Audio tracks: 0, 1, 2; Subtitle tracks: 0, 1, 2, 3 [18:44] <ghospich> Looks like there is no way, except of parsing output of `mkvmerge -i file` [18:45] <Bombo> hi i got a dvd.vob with subtitles, encoded it with ffmpeg to h264.mkv, played in vlc i see the subtitles are not centered, when i use mplayer the subs are centered [18:45] <Bombo> is it a bug in vlc or maybe ffmpeg? [18:47] <ghospich> softsubs? in which format? i guess you can check these subs with aegisub, and maybe problem would be spotted there. [18:48] <sfan5> I'm pretty sure dvds don't use SSA/ASS subtitles [18:48] <sfan5> aegisub will probably not load them [18:48] <sacarasc> They're probably still vobsubs. [18:49] <ghospich> okay, i'm silly [18:51] <Bombo> Stream #0:3: Subtitle: dvd_subtitle (default) [18:55] <Bombo> how can i extract them 'ffmpeg -i the.mkv -vn -an -c:s copy -map 0:3 the.sub' -> 'Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted' [19:00] <Bombo> hm '-vn -an -codec:s:0.3 srt sub.srt' -> 'Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height' [19:01] <Bombo> Stream #0:3 -> #0:0 (dvdsub -> srt) [19:07] <sfan5> Bombo: did you try 'ffmpeg -i the.mkv -map 0:3 -f rawvideo the.sub' ? [19:11] <Bombo> sfan5: Stream mapping: Stream #0:3 -> #0:0 (dvdsub -> ?) Encoder (codec none) not found for output stream #0:0 [19:13] <sfan5> try adding -c copy [19:14] <Bombo> sfan5: did dump something now, aegisub says 'unknown format' [19:14] <Bombo> 'unknown file type' [19:14] <sfan5> thats what I thought [19:14] <sfan5> aegisub only opens ASS/SSA [19:16] <Bombo> $ vobsub2srt the.sub [19:16] <Bombo> VobSub: Can't open IDX file [19:17] <sfan5> does the 'the.sub' file actually contain something? [19:17] <Bombo> data [19:17] <Bombo> $ file the.sub [19:17] <Bombo> the.sub: data [19:17] <Bombo> 212Kthe.sub [21:10] <Tarjaizaid> Hello community, i've MP4 file, i've reencode it on MP4 for to add ass subtitle. Now, i want add two subtitle file, (can i use MP4 container ?) do you have any suggestion to help me ? [21:12] <c_14> The MP4 container only supports 3GPP subtitles. If the subtitles are in that format or you can convert to that format, then yes. [21:13] <Tarjaizaid> ass format, MKV is a solution ? [21:13] <c_14> Ye [21:13] <Tarjaizaid> do you know ffmpeg option ? xD [21:14] <c_14> ffmpeg -i video -i sub1 -i sub2 -codec copy out.mkv [21:14] <Tarjaizaid> Ohhh ! so KISS [21:14] <Tarjaizaid> thx [21:14] <Tarjaizaid> :) [21:16] <Tarjaizaid> c_14: only one subtitle are in MKV :( [21:16] <Tarjaizaid> is* [21:17] <c_14> -map 0 -map 1 -map 2 [21:17] <c_14> Add those before out.mkv [21:18] <llogan> and read http://ffmpeg.org/ffmpeg.html#Stream-selection [21:29] <Tarjaizaid> yes, i've read, but... [21:32] <Chillance> Im trying to convert a movie into a animated gif, and it works. however, I get this wierd line on the right when using 2.x. It works fine in 1.2.7, but then the compression isn't as good, and it uses rgb8/rgb16 [21:33] <Chillance> anyone here know about this [21:33] <Chillance> ? [21:34] <Chillance> the line looks like the image isn't stretched properly, covering the whole area [21:36] <llogan> Chillance: can you show your command and complete console output for both 1.2.7 and 2.x? [21:36] <llogan> (use a pastebin service) [21:36] <Chillance> sure [21:36] <Chillance> sec [21:36] <Chillance> not sure if it helps, but sure [21:37] <Chillance> move is 220x220 [21:37] <Chillance> *movie [21:41] <Maverick|MSG> any idea what would cause a video to look like this after being encoded by ffmpeg: https://www.youtube.com/watch?v=22Nw8_buHX0 [21:42] <Maverick|MSG> only happens on 480p encodings. 720 and 1080p encodings work fine. [21:42] <Chillance> llogan, http://pastebin.com/qWkNPrKP - this works, but colors aren't as good. especially with fading, and the file is huge (17MB). [21:45] <Chillance> llogan, http://pastebin.com/hrpaavJT [21:46] <Chillance> that works too, but have this flickering line on the right.. like if the image didn't cover up the 220x220 properly [21:46] <Chillance> also, last one is better quality. and 1.2Mb [21:51] <Chillance> llogan, does it help? [21:55] <llogan> Maverick|MSG: does it look like that with ffplay or other players? [21:57] <Maverick|MSG> I'm piping in the tga images into ffmpeg [21:57] <Maverick|MSG> haven't touched ffplay or anything else [22:01] <llogan> Chillance: i don't see (or notice) a line when i tried it. does the line appear only with a certain input? [22:04] <Chillance> well, actually, the line is like 5px wide from the right side [22:05] <Chillance> just flickering [22:06] <Chillance> it really seems like its missing image information [22:06] <llogan> Maverick|MSG: what i mean is does the video only look like that on YouTube, or locally too? [22:07] <Maverick|MSG> oh locally as well [22:08] <Chillance> llogan, and I just use that input mentioned in pastebin [22:08] <llogan> can you provide that input file? [22:09] <Chillance> sure, where to send it? [22:09] <llogan> dropbox, google drive, datefilehost, mediafire, your server? [22:09] <Chillance> sec [22:11] <Chillance> http://tempsend.com/02C98306C8 [22:12] <Chillance> llogan, included the flickering .gif there [22:14] <ghospich> rghost.net is nice in my opinion. [22:15] <ghospich> (not advertise, mercy me) [22:25] <Chillance> llogan, how is it going? [22:27] <llogan> Chillance: sorry, i've been working on boring work stuff. i'll try to take a look later after my lunch break. [22:27] <llogan> i mean during [22:27] <Chillance> so, in 4 hours? [22:28] <llogan> usually i'm taking lunch now, but i'm still not done with something. [22:29] <llogan> you can try ffmpeg-user mailing list if you want additional input if you don't want to wait (and I may not end up having an answer for you). [22:29] <Chillance> are you one of the ffmpeg developers? [22:29] <Chillance> I would like to have some feedback/confirmation from you if its in ffmpeg or not [22:30] <Chillance> seems like it.. or in one of the used libraries [22:52] <Maverick|MSG> llogan: ignore the above issue. I'm certain this issue is occurring outside of ffmpeg. sorry about that [23:52] <Chillance> llogan, any progress? [00:00] --- Thu Aug 7 2014

1 0

ffmpeg-devel.log.20140805
by burek 06 Aug '14

06 Aug '14

[00:06] <Zeranoe> Is there a list of GPL only decoders? [00:12] <J_Darnley> What about the list of GPL files in LICENSE (I think)? [00:38] <cone-891> ffmpeg.git 03Michael Niedermayer 07master:700bf1fa4e78: swscale/swscale_unscaled: fix ya16 input [00:48] <Timothy_Gu> ubitux: huh? [01:01] <Timothy_Gu> ubitux: regarding my review for dctdenoiz, the lack of patch context confused me, so never mind. [01:15] <cone-891> ffmpeg.git 03Vittorio Giovara 07master:086a54185781: fate: png-suite [01:15] <cone-891> ffmpeg.git 03Michael Niedermayer 07master:6b646fb9e2ce: Merge commit '086a541857812b2e0435db38fbabfddc7d19e53c' [01:20] <cone-891> ffmpeg.git 03Diego Biurrun 07master:990e2f3555bb: avcodec: Suppress deprecation warnings from DTG code scheduled for removal [01:20] <cone-891> ffmpeg.git 03Michael Niedermayer 07master:debb88776c3d: Merge commit '990e2f3555bb0adae122d3599938ddcfee46e2d1' [01:36] <cone-891> ffmpeg.git 03Diego Biurrun 07master:eee813eec7d3: fate: Only generate tests/pixfmts.mak if some pixfmts fate test is run [01:36] <cone-891> ffmpeg.git 03Michael Niedermayer 07master:76bf39ab4b87: Merge commit 'eee813eec7d3c0b0689f80665d3f796401742935' [01:37] <BBB> I have the feeling I missed some responses [01:37] <BBB> ohwell [01:42] <cone-891> ffmpeg.git 03Thilo Borgmann 07master:86b7821e29b4: lavd/avfoundation: Add device category. [01:51] <cone-891> ffmpeg.git 03Michael Niedermayer 07master:aee99a5d4406: swscale/swscale_internal: fix ya16 -> rgba [02:04] <cone-891> ffmpeg.git 03Bernd Kuhls 07master:bf6c84d7eb27: Fix compile error on bfin. [03:01] <cone-891> ffmpeg.git 03James Almer 07master:de417982e897: x86/vp9lpf: use fewer instructions in SPLATB_MIX [03:45] <BBB> Im going to say this many times again, this coefficient coding in hevc is a total travesty [03:47] <jamrial> hevc spec, or ffhevc? [03:49] <BBB> ffhevc, primarily [03:54] <BBB> ffhevcs main coeff decoding function is 330 LOC, whereas VP9s is ~100 LOC [04:01] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:a1f7844a1101: pgssubdec: Check RLE size before copying [04:01] <cone-891> ffmpeg.git 03Vittorio Giovara 07release/2.2:2273e5ed9926: h264: prevent theoretical infinite loop in SEI parsing [04:01] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:7fc9c7c35b17: Merge commit 'a1f7844a11010d8552c75424d1a831b37a0ae5d9' into release/2.2 [04:01] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:188ce941ec01: Merge commit '2273e5ed992661e0c4b37208e792e2253d5a0b5b' into release/2.2 [04:13] <cone-891> ffmpeg.git 03Diego Biurrun 07release/2.2:744b406ff347: huffyuv: Check and propagate function return values [04:13] <cone-891> ffmpeg.git 03Reinhard Tartler 07release/2.2:07015d9f913d: Update Changelog for v10.3 [04:13] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:9ed1aa0465b3: Merge commit '744b406ff3474e77543bcf86125a2f7bc7deaa18' into release/2.2 [04:13] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:77b789b40635: Merge commit '07015d9f913d63bdc4495e75f4603c586553796e' into release/2.2 [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:c9f1456a41f5: avfilter/f_select: avoid using doubles for equals checks and casts to enums [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:21f6b07a977d: avfilter/f_select: Set var_values[VAR_KEY] correctly [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:f99b17bd32ca: avfilter/f_select: avoid double->int in debug output [04:58] <cone-891> ffmpeg.git 03Nicolas George 07release/2.2:f7b147548e76: ffmpeg_filter: refuse to configure input without a decoder. [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:a4e1532ee78d: avcodec/hevc_ps: do not loose all reference to pointers still in use [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:51dd23c4488b: avcodec/hevc_ps: prevent stale pointer in malloc failure case [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:815d3225e32b: avcodec/hevc: treat current_sps like sps_list [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:2105f046f571: avformat/dv: implement fallback in dv_extract_pack() [04:58] <cone-891> ffmpeg.git 03Chris \"Koying\" Browet 07release/2.2:6120ad315bc3: avcodec/h264_mp4toannexb_bsf: fix issue when sps/pps are already in the bistream (cherry picked from commit ad91bf854b559f1afc42fc06f8a7dfbb75def5c8) [04:58] <cone-891> ffmpeg.git 03Benoit Fouet 07release/2.2:cf41ff488978: h264_mp4toannexb_bsf: account for consecutive IDR pictures. [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:cdaf9fb2a054: libavcodec/h264_parser: Increase parse_history, fix huge resolutions [04:58] <cone-891> ffmpeg.git 03Anshul Maheswhwari 07release/2.2:0aee436728db: v4l2enc: adding AVClass [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:81d8bad78681: avcodec/mpeg12dec: fix support for interlaced mpeg2 with missing last slice [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:8eaefbe1beb5: avcodec/dvdsub_parser: never return 0 when the input isnt 0 [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:ce248bf7ee4e: avcodec/dvdsub_parser: Check buf_size before reading 32bit packet size [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:ad13a5c8fa09: avcodec/dvdsub_parser: print message if packet is smaller than the packet size field [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:01c4fe7ee7b3: avformat/tee: flip assigment direction [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:c13e38bac7e7: avcodec/wavpackenc: Fix log2sample() result value [04:58] <cone-891> ffmpeg.git 03Michael Niedermayer 07release/2.2:119131fcbfdc: ffserver: initialize pbuffer in prepare_sdp_description() [05:00] <michaelni> Timothy_Gu, if you want to write some Changeog / release notes for the next 2.2 release, i intend to release it tomorrow unless some other work delays it [06:14] <Timothy_Gu> michaelni: - bug fix in wavpackenc (you can go into more details for this one -- I only know something's wrong, don't know what) - Fix infinite loop in dvbsub parser - fix some interlaced MPEG-2 videos [06:23] <Timothy_Gu> - dv fallback - fix v4l2 and v4l2enc crashes - fix theoretical librtmp crash - fix theorectical eamad crash - support dimension change in g2meet [09:26] <ubitux> https://ffmpeg.org/pipermail/ffmpeg-user/2014-August/022835.html wut [09:34] <nevcairiel> who sells ffmpeg docs for 9.95? :D [09:35] <ubitux> i was wondering if it wasn't the random book from the guy who had the twitter/ffmpeg [09:36] <ubitux> which we actually reference in our documentation page [09:36] <ubitux> it's just a book full of examples iirc [09:37] <ubitux> the prices doesn't match exactly ($9.50) [09:37] <ubitux> -s [12:32] <michaelni> Timothy_Gu, thanks [12:55] <cone-861> ffmpeg.git 03Ben Avison 07master:db7f1c7c5a1d: h264: Move start code search functions into separate source files. [12:55] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:77aafadc5655: Merge commit 'db7f1c7c5a1d37e7f4da64a79a97bea1c4b6e9f8' [13:14] <cone-861> ffmpeg.git 03Ben Avison 07master:adf8227cf4e7: vc-1: Add platform-specific start code search routine to VC1DSPContext. [13:15] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:bf7ed956ffcd: Merge commit 'adf8227cf4e7b4fccb2ad88e1e09b6dc00dd00ed' [13:45] <cone-861> ffmpeg.git 03Ben Avison 07master:701e8b42e12a: vc-1: Optimise parser (with special attention to ARM) [13:45] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:9988899993ee: Merge commit '701e8b42e12ad625c64ceae2252acb1de390278c' [13:51] <cone-861> ffmpeg.git 03Luca Barbato 07master:89616408e38a: mpegts: Define the section length with a constant [13:51] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:94d4f9d40507: Merge commit '89616408e38ac7257e36976723df0e23d6ee1157' [13:57] <cone-861> ffmpeg.git 03Femi Adeyemi-Ejeye 07master:2601a9447efb: mpegts: Add HEVC definitions [13:57] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:2e2e6b1be591: Merge commit '2601a9447efb778930b01ac58d2cc1d02498db4e' [14:06] <cone-861> ffmpeg.git 03John Stebbins 07master:0897d2fdc775: movenc: Add option to disable nero chapters [14:06] <cone-861> ffmpeg.git 03Michael Niedermayer 07master:a53a107a2f04: Merge commit '0897d2fdc7755849c3ae58d8b543ef734c200d3c' [14:22] <cone-861> ffmpeg.git 03mrskman 07master:6f3e15af84eb: avfilter/avf_showwaves: New p2p mode for showwaves filter [17:26] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:256922768926: avfilter/vf_noise: make constant things const [18:23] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:592a854f8c8b: avformat/iff: Fix "source comment" [19:21] <cone-118> ffmpeg.git 03Janne Grunau 07master:d395895cdb2a: fate: generate tests/pixfmts.mak for all targets requiring it [19:21] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:0160fe454c38: Merge commit 'd395895cdb2ac8c95bd488549e7f893bd4dcc248' [19:40] <cone-118> ffmpeg.git 03Janne Grunau 07master:15201e256035: configure: check $as first before using $gas as GNU as [19:40] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:00d234688512: Merge commit '15201e256035a3e8f9d3d7b96fc327467e1a8ead' [19:55] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:849b9d34c7ef: wmalosslessdec: fix mclms_coeffs* array size [19:55] <cone-118> ffmpeg.git 03Michael Niedermayer 07master:6931d1274568: Merge commit '849b9d34c7ef70b370c53e7af3940f51cbc07d0f' [20:11] <ubitux> "t", ///< first PTS in seconds [20:11] <ubitux> >f_select.c [20:11] <ubitux> this comment looks wrong, am i right? [20:46] <ubitux> michaelni: i added the min/max tests to pixelutils [20:47] <ubitux> should i push? [20:49] <ubitux> ah, and the asm [20:50] <ubitux> should i make the sad 16x16 improvements? [21:02] <michaelni> ubitux, iam always for pushing improvments out [21:02] <ubitux> ok [21:10] <cone-118> ffmpeg.git 03Clément BSsch 07master:28a2107a8d61: avutil: add pixelutils API [21:10] <cone-118> ffmpeg.git 03Clément BSsch 07master:f50ec59751f8: build: remove wrong avcodec dep in decimate [21:10] <cone-118> ffmpeg.git 03Clément BSsch 07master:def03d34391f: avfilter/deshake: remove avcodec dependency [21:10] <cone-118> ffmpeg.git 03Clément BSsch 07master:164b7ccf2112: avfilter/mp_decimate: remove avcodec dependency [21:10] <cone-118> ffmpeg.git 03Clément BSsch 07master:eaeb61c3a693: avfilter/select: remove optional avcodec dependency [21:15] <cone-118> ffmpeg.git 03Clément BSsch 07master:97478ef5fe7d: avfilter/select: fix "t" eval variable comment [22:14] <cone-118> ffmpeg.git 03Michael Niedermayer 07release/2.2:2cbdbc367089: update for 2.2.6 [22:18] <Timothy_Gu> ubitux: why not just move the entire me_cmp to libavutil? Now there are code duplication between avcodec and avutil. [22:19] <Timothy_Gu> michaelni: please fix the "theorectical" typo in 2.2 changelog. [22:28] <cone-118> ffmpeg.git 03Michael Niedermayer 07release/2.2:f543d32455a3: remove VERSION file [22:29] <cone-118> ffmpeg.git 03Michael Niedermayer 07release/2.2:1f4d779e8705: version.sh: Print versions based on the last git tag for release branches [22:29] <cone-118> ffmpeg.git 03Michael Niedermayer 07release/2.2:1b9966700515: Changelog: fix typo [22:29] <michaelni> Timothy_Gu, done, thanks [23:03] <ubitux> Timothy_Gu: because if you make an API public you need to think about it and have multiple use [23:03] <ubitux> Timothy_Gu: we don't have any use-case for me currently [23:03] <ubitux> SAD was used in multiple filters and solve the issue we were trying to fix (dsputil dep) [23:10] <cone-118> ffmpeg.git 03Clément BSsch 07fatal: ambiguous argument 'refs/tags/n2.2.6': unknown revision or path not in the working tree. [23:10] <cone-118> Use '--' to separate paths from revisions [23:10] <cone-118> refs/tags/n2.2.6:HEAD: avfilter/select: fix "t" eval variable comment [23:26] <jamrial> why does cone choke like this when creating new tags? [23:47] <Compnn> [06:55] <cone-861> ffmpeg.git Ben Avison master:db7f1c7c5a1d: h264: Move start code search functions into separate source files. [23:47] <Compnn> did that break mplayer compilation ? [23:47] <Compnn> http://pastebin.com/CLFEyhYg [23:47] <Compnn> ^ not my build , i didnt test... [23:48] <Compnn> jamrial : probably need a simple filter in git send mail to add "--" to commit messages with tags [23:48] <Daemon404> Compnn, it's almost like mplayer shouldnt canibalize the build system [23:49] <Compnn> Daemon404 : huh, i thought we set it up to call ffmpeg configure itself ... guess not [23:50] Action: Compnn wonders if we throw mplayer build box on ffmpeg fate server :P [23:51] <michaelni> Compnn, did you svn up ? [23:51] <Compnn> michaelni : like i said, wasnt my build. could just be an svn problem. [23:51] <michaelni> i pushed some fix for CONFIG_STARTCODE to mplayer when i pushed the stuff to ffmpeg [23:52] <Compnn> oh ok, the user left already [23:52] <Compnn> but yea it looked like a strange error to get stuck there, it would have been caught when building ffmpeg stuff earlier. [23:52] <michaelni> and yes if mplayer would call ffmpegs configure that would simplify/avoid the occasional update [23:55] <Daemon404> hooray pixutils-test crashes on some windows builds [23:57] <Compnn> michaelni : thanks for keeping mplayer working :) [00:00] --- Wed Aug 6 2014

1 0

ffmpeg.log.20140805
by burek 06 Aug '14

06 Aug '14

[00:13] <TD-Linux> GPUs don't magically make your code faster. [00:14] <TD-Linux> many parts of video encoding are highly serial which works poorly on a GPU. [01:31] <voip1__> TD-Linux, i tink for parallel transcoding (for example 50 streams) it will be fine. [01:32] <TD-Linux> I highly doubt it. More CPU cores would be far more cost effective. [01:36] <voip1__> You realy tinking 1 intel xeon with 8 cores able transcode more streams than 1 Radeon card with 2816 Shading Units? [01:36] <TD-Linux> yes. [01:37] <voip1__> interesting [01:38] <TD-Linux> if you want to fit 2816 "cores" on a chip the same size as an 8 core xeon, obviously those cores are going to have compromises [01:38] <TD-Linux> one is that usually groups of cores need to execute in lockstep [01:39] <TD-Linux> in amd case it's 32 threads bundled [01:39] <voip1__> as you know for example adobe premiere can use GPU during encoding, its works fine [01:40] <TD-Linux> yes. but it only partially uses the GPU and produces encodes of worse quality than x264. [01:41] <llogan> and not any faster [01:42] <llogan> i prefer to use a frameserver instead of AME https://trac.ffmpeg.org/wiki/Encode/PremierePro [01:42] <TD-Linux> for video encodes I use amazon c3.8xlarge instances, with 16 physical xeon cores (32 threads). It is by far the most cost effective solution. [01:43] <TD-Linux> well, maybe not "by far". but useful for my use case. [01:45] <voip1__> i dont know [01:45] <TD-Linux> voip1__, write some code for a GPU and then come back with your own opinion :) [01:46] <voip1__> guys, im not developer :) [04:45] <MarcelvanLeeuwen> Morning! [04:46] <MarcelvanLeeuwen> Very early morning& [04:50] <MarcelvanLeeuwen> I have a little script which is encoding movies placed in a directory. Now would it be nice if movies with two audio track also be correct encoded.. [04:51] <MarcelvanLeeuwen> the script can only encode movies with one audio track [04:51] <MarcelvanLeeuwen> is it possible to adjust this? [06:25] <rjp421> when playing a file and streaming to flv over rtmp, the play time clock goes fast and skips ahead, while the video slowly speeds up until it crashes [06:37] <rjp421> floods with then ends at *** dropping frame 22208 from stream 0 at ts 22206 [06:37] <rjp421> No more output streams to write to, finishing. [08:51] <termos> is there some way to set an exact gop in ffmpeg when doing x264 encoding? [08:56] <termos> or at least set it so that it's not varying by a lot [08:57] <termos> the problem is that I will sometimes get very small hls segments when doing hls muxing [10:24] <zenderz> anyone familiar with the following error: [10:24] <zenderz> [ismv @ 0x9ba7d60] Malformed AAC bitstream detected: use audio bitstream filter 'aac_adtstoasc' to fix it ('-bsf:a aac_adtstoasc' option with ffmpeg) av_interleaved_write_frame(): Operation not permitted [10:24] <ubitux> maybe you need to do what the error message suggests? [10:25] <zenderz> i did and it works. however i want to know whats wrong with the input file and how i can fix it before hand [10:25] <zenderz> http://pastebin.com/BhFMTm5h [10:26] <zenderz> is there a way I can apply the bitstream filter to the original file [10:27] <ubitux> i don't think it's problematic to have ADTS in mpeg [10:28] <zenderz> the thing is that i want the input file to have exactly the correct codecs so that no transcoding/filtering is needed. this will make the smoothstreaming conversion process easier [10:29] <zenderz> so can i do something like: ffmpeg -i testinput codec:a copy -bsf:a aac_adtstoasc -codec:v copy inputfixed.mpg [10:33] <roee> Hi Guys, I'm trying to capture raw data from my webcam and analyze it using node.js so i created the following code just for debuging and understanding the content of the data generated by ffmpeg [10:33] <roee> http://pastebin.com/mUVgBBPf [10:34] <roee> I than opened the test file i created and saw that ffmpeg creates some kind of header and only after that the raw data [10:34] <roee> how can i find out what [10:34] <roee> what's the size of that header [10:35] <roee> * [10:35] <Mavrik> hmm [10:35] <Mavrik> MPEG-TS doesn't have a header per-se [10:35] <Mavrik> but it does inject PAT/PMT packets which describe content of streams [10:35] <Mavrik> which is what you're probably seeing now [10:35] <roee> i took mpeg-ts from a sample [10:35] <roee> could you suggest something better? [10:35] <Mavrik> well... depends on what do you want to achieve really [10:35] <roee> i want to detect the dominate color in each frame [10:36] <roee> a simple image analysis [10:36] <Mavrik> hmm [10:36] <Mavrik> thing about video is that you really need at least some kind of container around it [10:36] <Mavrik> lemme see if ffmpeg can do some kind of delimited raw output [10:36] <roee> thanks [10:38] <zenderz> so if i try fix the input file i get the following: [10:39] <zenderz> http://pastebin.com/EHD9rBqF [10:39] <zenderz> [mpegts @ 0x98ee640] AAC bitstream not in ADTS format and extradata missing [10:39] <zenderz> which is strange, because when using the same filter for smoothstreaming format, it works fine [10:39] <Mavrik> roee, -f rawvideo should give you raw video output [10:40] <roee> ok i'll try that and let you know [10:40] <roee> [NULL @ 0x9ebd5e0] Requested output format 'rawvideo' is not a suitable output format [10:41] <roee> I replace the mpegts with rawvideo [10:41] <roee> replaced* [10:42] <roee> do i need to rebuild ffmpeg with this capability? [10:42] <Mavrik> hmm, it seems that rawvideo muxer still hasn't bene implemented. [10:42] <zenderz> roee: have a look at http://zulko.github.io/blog/2013/09/27/read-and-write-video-frames-in-pytho… [10:42] <zenderz> its for python but its similar to what you need [10:42] <Mavrik> mhm [10:43] <Mavrik> it seems that you have to leave out -f and explicitly say codec:v rawvideo [10:43] <Mavrik> never did that tho [10:43] <Mavrik> zenderz, well you ARE explicitly converting ADTS audio to ASC audio :) [10:43] <roee> ok [10:43] <roee> i'll try [10:45] <zenderz> Mavrik, are they not just different containers for aac? [10:46] <Mavrik> yes [10:46] <Mavrik> they're a way of coding AAC [10:46] <Mavrik> and MPEG2-TS standard DEMANDS that AAC is encapsulated in ADTS packets inside [10:46] <Mavrik> hence muxer aborts encoding of invalid format [10:47] <zenderz> ah okay I see. so there is no way to trans-mux from mpegts to smooth streaming format without any codec changes then? [10:57] <termos> I am now setting AVFrame->pict_type = AV_PICTURE_TYPE_NONE before calling avcodec_encode_video2 and I get perfect stable GOP size. Is there a less hacky way to do this? [10:59] <roee> Unable to find a suitable output format for 'pipe:' pipe:: Invalid argument [11:00] <roee> cp.spawn("ffmpeg", [ "-f", "video4linux2", "-s", "176x144", "-r", "15", "-input_format","yuyv422", "-i", "/dev/video0", "-vsync", "drop", "-vcodec", "rawvideo", "-movflags", "faststart+frag_keyframe", "-" [11:00] <roee> am i missing something? [12:24] <zenderz> look at https://www.ffmpeg.org/ffmpeg-protocols.html#pipe [12:25] <zenderz> roee [12:50] <rjp421> when playing a file and streaming to flv over rtmp, the play time clock goes fast and skips ahead, while the video slowly speeds up until it crashes [13:08] <Nickwiz> Is it possible to use fifo for audio input? [13:10] <Nickwiz> ffmpeg ... -i /tmp/fifo_audio ... [13:12] <Nickwiz> Reason is that my soundcard does not support capture, but I can use pacat. Then, if possible, I could pipe audio to fifo file from pacat and use ffmpeg -i /fifofile [13:13] <Nickwiz> The idea is to record screen with audio. [13:13] <Nickwiz> in one go. [14:09] <Hello71> !tias [14:17] <DopeLabs> =] [14:20] <klaxa> Nickwiz: you are using pulseaudio then, right? [14:21] <klaxa> have you tried ffmpeg -f alsa -i pulse ? [14:21] <Nickwiz> klaxa: I have various "modes" I operate in, depending on what I do. Pure ALSA, Pulse or JACK. [14:21] <Nickwiz> yes. I have tried that one. Only silence. [14:22] <klaxa> have you tried using pavucontrol to change the stream that is being recorded? [14:22] <klaxa> if you want to record your internal audio change the recording stream to something like "Monitor of Analog Stereo" or something [14:24] <Nickwiz> The recording does not have any entries. I thought that was because my soundcard does not have any audio record support. [14:24] <Nickwiz> It si typically only mic and line in. [14:25] <klaxa> did you enable showing all streams? [14:25] <Nickwiz> Yes. No diff. [14:25] <klaxa> there is a drop-down switcher at the bottom where you can set it to "All input streams" [14:25] <klaxa> *devices even [14:26] <klaxa> that's pretty weird, how can pacat do it then [14:28] <Nickwiz> Not sure. Thought is hooked directly into the sink. But, yes I find it weird it is not in pavucontrol. Only recall I messed with this earlier. Then I found that the soundcard did nto have any "record" option. Do not recall waht exactly it is called. Ended up with using pacet. [14:28] <Nickwiz> *pacat [14:28] <Nickwiz> That is; I can record mic and line of course, but not e.g. streaming audio. [14:29] <Nickwiz> Some limitations on soundcard HW. [14:29] <klaxa> most stuff in pulse is software though [14:29] <klaxa> i don't think using pacat should be necessary [14:30] <klaxa> you can use named pipes though if you really need to [14:30] <klaxa> you will have to set the input format as an ffmpeg option though [14:32] <klaxa> like: ffmpeg -c:a pcm16le -ar 44100 <whatever the command for two channels is> -i <named pipe> ... [14:37] <Nickwiz> klaxa: Thanks, that's great. I'll look at it. I ended up recording audio and video separately in two terminals and it is now encoding (pos processing). I have to do a compile job after that, but I'll look at it later. Once again. Thank you. [14:37] <Nickwiz> s/pos/post [14:38] <Nickwiz> I also need to upgrade my HW, :/ [14:53] <jhon> Hello i'm trying to convert video to x264, i'm getting this error Unkown encoder 'libfaac'. Where I can find libfaac encoder i'm on windows [15:02] <JEEBsv_> you compile a binary yourself with fdk-aac instead; both faac and fdk-aac are "nonfree" (incompatible with GPL, binary distribution not possible), and fdk-aac is much better [15:02] <sacarasc> jhon: To get libfaac, you'd have to compile ffmpeg yourself as it is a non-free codec. There are a few other AAC encoders you can use, though. The one just called 'aac' is the better quality one allowed for a distributable ffmpeg. [15:02] <sacarasc> To use aac, though, you have to also use -strict experimental in your command line. [15:03] <sacarasc> I wonder how much of what I said got through there. :( [15:05] <jhon> sacarasc: any guide for compiling it. I'm new to this [17:33] <dreamcat4> hello! is this patch needed for 55.52.102 ??? [17:33] <dreamcat4> https://github.com/dreamcat4/tvheadend/commit/d18bc8dc65eea6f7a2e094379a6e4… [17:33] <c_14> That's a patch for libav. Ask #libav for information. [17:34] <JEEB> seems general lavc, and that's just an update for a newer API [17:34] <dreamcat4> c_14 even the libav provided by ffmpeg ? [17:34] <JEEB> and no idea about the random library version :P [17:35] <JEEB> if that version number is from FFmpeg then it's different than Libav, and it should be rather obvious which defines you have [17:35] <JEEB> herp derp [17:35] <dreamcat4> JEEB i guess i should dig the code [17:36] <dreamcat4> if says 'ffmpeg 2.2.4' [17:36] <JEEB> dreamcat4, then it has a relatively new API [17:36] <sacarasc> dreamcat4: There's libav then there's libav*, so there can be some confusion. :D [17:36] <dreamcat4> looks like it's been downloaded from http://ffmpeg.org/releases/ [17:37] <dreamcat4> (the FreeBSD port Makefile) [17:37] <dreamcat4> i can look for those defines in the tarball [17:37] <JEEB> I'm pretty sure it uses those newer namings there [17:38] <JEEB> and if that project is made with libav only in mind, then it just doesn't have a libav vs ffmpeg configure check [17:38] <JEEB> the version numbers are not the same in libraries, and so forth [17:39] <t4nk001> Does AVCHD format work with ffmpeg ? [17:39] <dreamcat4> JEEB: yep. in 2.2.3 tarball it use the newer AV_ defines. So that's needs the ".56" patch in URL above then [17:40] <dreamcat4> i guess the configure check should exclude versions =< 55.0.0 too, (since they seem to break each other) [17:41] <JEEB> dreamcat4, no [17:41] <JEEB> libav and ffmpeg have different points [17:41] <JEEB> the version numbers are not the same [17:42] <JEEB> if you just lower the number it will enable one to build with an older libav [17:42] <JEEB> which doesn't have it [17:42] <JEEB> thus you have to add a check for libav or ffmpeg [17:42] <JEEB> and then have separate version checks for both :P [17:47] <dreamcat4> JEEBus, that's potentially version confusing! [17:48] <dreamcat4> grateful to be made aware of that [17:48] <JEEB> for an example of how to check you can check ffms2's configuration [17:49] <JEEB> https://github.com/FFMS/ffms2/blob/master/configure.ac#L113 and forward [17:51] <dreamcat4> thanks JEEB. if only the project used autotools properly and actually had a configure.ac file. [17:52] <t4nk001> Does AVCHD format work with ffmpeg ? [17:52] <JEEB> t4nk001, in general yes [17:52] <dreamcat4> i think maybe pkg_config can output if ffmpeg is installed (*maybe*) [17:52] <JEEB> hahaha [17:53] <JEEB> just look at the code I linked :P [17:53] <JEEB> it does use pkg-config first that the libraries are there [17:53] <JEEB> then it checks a few things [17:54] <dreamcat4> ah ok JEEB - thanks i guess at least some of it can be usef [17:56] <t4nk001> Thanks JEEB , It is not working with FFmpeg version 0.6.5, Do not know the problem with specific version [17:56] <JEEB> welcome to ancient FFmpeg versions [17:56] <JEEB> in other word, update [17:57] <t4nk001> :) Can we update it with yum or manually ? [17:58] <JEEB> your distro most probably doesn't give you newer versions. for quick testing you can try grabbing a static build from the places linked under FFmpeg's downloads section [17:59] <JEEB> and if those just don't work and/or you want to update, then I recommend you compile it yourself [18:00] <dreamcat4> just shitting myself now. trying to figure out how i'm gonna explain all that to the other people on my project [18:01] <JEEB> "There are two projects, their version numbers are not compatible. We have to check for which we are dealing with first, and then check accordingly" [18:01] <JEEB> herp derp [18:07] <dreamcat4> JEEB: can't i just parse the output of 'ffmpeg -version' instead of using such autoconf macros ? [18:08] <dreamcat4> it would be easier for us (since not using autoconf) [18:08] <sacarasc> What if they installed the libraries only, and not the executable? [18:08] <JEEB> uhh, some of those checks are rather simple :P [18:08] <JEEB> sacarasc, not to mention that it could be a random ffmpeg binary in your PATH [18:08] <JEEB> not even a system packaged one [18:10] <dreamcat4> JEEB: thanks for trying to help (its another thing to know which version the source code in the project expects) [18:11] <JEEB> the "100" version check is the general way of determining libav vs ffmpeg btw :P [18:11] <JEEB> which was in the ffms2 configure stuff, too [18:12] <dreamcat4> ah right... that 100 check is a C MACRO not an autoconf macro... i can use that [18:12] <JEEB> a lot of what I linked you to is usable by you :P [18:12] <JEEB> a _lot_ [18:13] <JEEB> i you know what it does in general [18:14] <dreamcat4> JEEB: sure thing. i just know nothing about ffmpeg. finding it quite the stretch for simpletons like myself [18:22] <dreamcat4> presumably if CODEC_ID_G2M doesn't exist then compilation will fail, meaning it's libav, and not ffmpeg [18:23] <JEEB> that's a more specific check [18:25] <dreamcat4> JEEB maybe it's not needed anymore, if everyone is on 55 by now [18:25] <dreamcat4> doesn't hurt though [18:28] <dreamcat4> JEEB: https://gist.github.com/dreamcat4/42013f8fe6e637aa856a [19:48] <sirnfs> Hi guys. I have done a simple transcoding of an mp4 video using ffmpeg -i input.mp4 output.mp4, and I have noticed that in certain players (VLC, HTML5 player), the last couple of frames are not shown and the audio cuts off. Playing the video in Quicktime or Windows Media Player produces the correct output. I am using libx264 with the ffmpeg command line. Any ideas? [19:49] <sirnfs> The video can be seen here: https://www.dropbox.com/s/qkuh2w7f9y25p0l/jesse_outTRANSNOSEEK.mp4 [19:55] <bazurbat> Hello, have anybody tried to use afade filter programmatically? [19:56] <brontosaurusrex> sirnfs, what version of vlc is that, what html5 player? [19:57] <sirnfs> brontosaurusrex: 2.1.5 Rincewind. Not sure about the html5 player [19:57] <sirnfs> I'm using the latest version of google chrome [20:00] <sirnfs> brontosaurusrex: here is the original video file: https://www.dropbox.com/s/yo1v2jxk639hffj/jesse_in.mp4 [20:01] <ramu> hi, trying to stream continous set of images encoded from ffmpeg (command line)via ffserver (command line). THe video is received in chrome [20:02] <ramu> everything with in syste,m, but video is jerky [20:02] <brontosaurusrex> sirnfs, can't spot a problem, but here are some files with burned in timecode http://sourceforge.net/projects/ffdrop/files/video/syncTestBunny/24and25.7z… [20:02] <brontosaurusrex> if you want to do some easily spotable tests [20:02] <ramu> using vp8 encoding [20:03] <sirnfs> brontosaurusrex: at the very end, when he says "next LEVEL" [20:03] <brontosaurusrex> 24fpsSyncTester.mp4 < is what you want sirnfs [20:03] <brontosaurusrex> sirnfs, yeah works fine here in vlc and in chromium [20:03] <sirnfs> You compared it with the original? [20:04] <brontosaurusrex> sirnfs, nope [20:08] <brontosaurusrex> sirnfs, they sounds the same to me, out point is edited out a bit too soon [20:08] <brontosaurusrex> i'd give it 5 frames more at the end [20:08] <bazurbat_> Hello again, does somebody know how to make filtering_audio example work with afade filter? [20:08] <sirnfs> brontosaurusrex: yeh, it seems to cut off at the end [20:08] <sirnfs> brontosaurusrex: would you do that with the -ss command or ? [20:09] <brontosaurusrex> sirnfs, yeah, but the original is the same [20:09] <sirnfs> brontosaurusrex: for me, his lips seem to end in a different position [20:09] <brontosaurusrex> sirnfs, i'd do it it in the video editing app [20:10] <sirnfs> brontosaurusrex: I normally would, but these videos come from users. Need something more automated [20:13] <brontosaurusrex> sirnfs, i'd test various x264 switches first on a good video [20:13] <brontosaurusrex> (the link i gave you) [20:14] <brontosaurusrex> so that # of frames and playback behaviour is constant [20:14] <brontosaurusrex> on all platforms [20:14] <brontosaurusrex> else if that fails, try some other (older) x264 version [20:15] <brontosaurusrex> until everything works [20:15] <sirnfs> brontosaurusrex: ok, thanks for the feedback. [20:15] <brontosaurusrex> another idea might be too add some black at the end of videos (but really you shouldn't) [20:17] <sirnfs> I have an unrelated question: Is it possible to request certain chunks of a file from a users hard drive through some type of upload script that runs on a server? [20:17] <sirnfs> Seems that the sandbox will prevent that, but maybe there is a workaround [20:18] <brontosaurusrex> sirnfs, possibly with something like node.js [20:19] <sirnfs> Thanks [20:23] <dreamcat4> how can i find when the #defines became prefixed with AV_* in libavcodec ? [20:24] <dreamcat4> i know they are AV_ in 2.2.4, but how far back. maybe look in git history on src code git repo ??? [20:25] <dreamcat4> or the change was pre-github era ? i should instead download random old tarballs and grep them ? [20:26] <c_14> Well, the git history goes back to 2001, so it should be in there somewhere. [20:27] <c_14> Assuming the change was properly documented. [20:29] <ghospich> How to pass filename with spaces to subtitles filter? I can do `ffmpeg -i video.avi -vf subtitles=subtitle.srt out.avi`, but how to do the same, except with spaces in filename? [20:29] <llogan> dreamcat4: APIchanges may mention it, but I didn't look [20:31] <sacarasc> ghospich: Put "" around the file names. [20:31] <c_14> dreamcat4: You can also try running a git blame over the lines and seeing what commit added them. [20:32] <llogan> he's gone, jim. [20:32] <llogan> i mean dead [20:32] <ghospich> c_14, not working for me [20:33] <ghospich> Though if i pass same string to ls, it's confirming that file exists [20:34] <ghospich> okay [20:35] <llogan> dreamcat4: what you missed: "APIchanges may mention it, but I didn't look" [20:35] <llogan> <c_14> dreamcat4: You can also try running a git blame over the lines and seeing what commit added them. [20:41] <ghospich> Ohh, i found what's wrong. I passed -map 0:a:1 and -an to ffmpeg. [20:41] <ghospich> Silly me. [20:48] <brontosaurusrex> what machine was used with x264 threads=60 ? [21:06] <dreamcat4> thanks llogan ! [21:14] <Carlos-Riper> hola [21:14] <Carlos-Riper> me pueden ayudar porfavor :D [21:17] <Carlos-Riper> I need to mount a streaming with ffserver and ffmpeg, webcam and my web browser, please. [21:20] <llogan> Carlos-Riper: i'm not familiar with ffserver. this may help: https://trac.ffmpeg.org/wiki/Streaming%20media%20with%20ffserver [22:24] <FrEaKmAn_> hi all.. I'm trying to understand iframes more [22:25] <FrEaKmAn_> so to be able to have a better seek (video is played with OSMF), I should put an iframe every 1 second [22:25] <FrEaKmAn_> but based on https://lists.ffmpeg.org/pipermail/ffmpeg-user/2011-September/002349.html [22:25] <FrEaKmAn_> I'm not pretty sure if I must define each x frame or each second? [22:41] <brontosaurusrex> FrEaKmAn_, also decoding complexity [22:41] <FrEaKmAn_> I followe this tutorial https://support.octoshape.com/entries/25126002-Encoding-best-practices-usin… [22:42] <FrEaKmAn_> just instead of 48 I used 24 [22:42] <FrEaKmAn_> so every second [22:42] <FrEaKmAn_> seems to work [22:43] <Mavrik> oh god, why is he trying to make people use static keyframe ordering [22:44] <brontosaurusrex> to get reliable seeking? [22:45] <Mavrik> that makes no sense - setting GOP size will get you 1 sec (or less) spacing keyframes always [22:45] <Mavrik> without compromising quality due to forcing them to EXACTLY 1 sec spacing [22:46] <Mavrik> on fast / action movies you'll be burning alot of bitrate for pointless full-frame storage [22:46] <FrEaKmAn_> so what should I do? [22:47] <brontosaurusrex> Mavrik, right [22:48] <Mavrik> FrEaKmAn_, just do whatever, keyframe intervals are always passed in frame numbers [22:48] <brontosaurusrex> althought CBR is even worse idea there [22:48] <Mavrik> mhm, unless video is prepared for streaming [22:49] <FrEaKmAn_> it's for streaming [22:49] <FrEaKmAn_> in my case [22:49] <brontosaurusrex> and how is CBR good idea for streaming? [22:49] <Mavrik> brontosaurusrex, overlooked the -mirate [22:50] <Mavrik> *minrate [22:50] <Mavrik> that's madness. [22:50] <brontosaurusrex> yeah [22:50] <Mavrik> but yes, I have gotten requests from my customers to implement full CBR streaming [22:50] <Mavrik> Explaination: "When using Envivio the bitrate graph looks prettier." [22:50] <brontosaurusrex> Mavrik, any "whys" ? [22:50] <brontosaurusrex> oh [22:51] <Mavrik> and then they showed me a MPEG2 stream that was stuffed for exact 20Mbit CBR meant for DVB-T broadcasting [22:51] <Mavrik> which needs exact bitrate :P [22:51] <brontosaurusrex> well, not such a horrible idea when you have the reserved channel i guess [22:52] <brontosaurusrex> but even then [22:52] <Mavrik> Well, they didn't. [22:52] <Mavrik> Luckly I managed to explain to them to stop allowing management people to look at technical data. [22:53] <brontosaurusrex> FrEaKmAn_, so you are using osmf and octoshape? [22:53] <FrEaKmAn_> only osmf [22:53] <FrEaKmAn_> and wowza for streaming [22:56] <brontosaurusrex> FrEaKmAn_, examples here http://www.wowza.com/forums/content.php?192-How-to-encode-video-on-demand-c… [22:57] <brontosaurusrex> "-g" = gop size [22:58] <brontosaurusrex> this are 1pass vbr [23:05] <FrEaKmAn_> ok [23:05] <brontosaurusrex> why are you using osmf? isnt that like a dead project? [23:06] <FrEaKmAn_> thanks,, will check first thing in the morning [23:06] <FrEaKmAn_> yes, but works great [23:13] <brontosaurusrex> ok, how about mobile devices? [00:00] --- Wed Aug 6 2014

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