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November 2012

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ffmpeg-devel.log.20121129
by burek 30 Nov '12

30 Nov '12

[00:04] <cone-797> ffmpeg.git 03Anton Khirnov 07bb6c67bb36b1: lavfi: remove vf_slicify [00:04] <cone-797> ffmpeg.git 03Michael Niedermayer 078c1f98d9542d: Merge commit 'bb6c67bb36b136de10256f0999128df4a42f9ffc' [00:13] <cone-797> ffmpeg.git 03Anton Khirnov 07267290ce3b4d: vflip: switch to filter_frame [00:13] <cone-797> ffmpeg.git 03Anton Khirnov 07a42b89910b07: vf_drawbox: switch to filter frame [00:29] <ubitux> huh [00:29] <ubitux> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?p=ffmpeg.git;a=commitdiff;h=… [00:29] <ubitux> i don't understand the fate update [00:34] <michaelni> ubitux, the old code analyzed the pixels of the input during start frame [00:34] <michaelni> but the pixels only become available during draw_slice [00:34] <michaelni> for each slice [00:34] <michaelni> and all then during end_frame [00:35] <ubitux> oh. [00:35] <ubitux> ok [00:37] <ubitux> mmh sampling dithering in lavr& [00:38] <cone-797> ffmpeg.git 03Anton Khirnov 0771f82c3805eb: vf_crop: switch to filter_frame [00:38] <cone-797> ffmpeg.git 03Anton Khirnov 07aa61728d0af9: vf_cropdetect: switch to filter_frame [00:38] <cone-797> ffmpeg.git 03Anton Khirnov 078f21cfc6b39d: vf_aspect: switch to filter_frame [00:38] <cone-797> ffmpeg.git 03Anton Khirnov 070a767ad796e4: vf_blackframe: switch to filter_frame [00:55] <cone-797> ffmpeg.git 03Clément BSsch 078a12c96d2712: lavfi/mptestsrc: add FLAGS to AVOption array. [00:55] <cone-797> ffmpeg.git 03Clément BSsch 07a5b765236bc5: lavfi: add priv_class for some forgotten filters. [00:55] <cone-797> ffmpeg.git 03Clément BSsch 070b70ffa4acc1: lavfi/sendcmd: add FLAGS to AVOption array. [00:55] <cone-797> ffmpeg.git 03Clément BSsch 073860e34b0865: lavfi/sendcmd: expose the options for both filters. [01:24] <cone-797> ffmpeg.git 03Michael Niedermayer 0794fdef818e27: vf_scale: switch to filter_frame [01:24] <cone-797> ffmpeg.git 03Anton Khirnov 0769d4420aeac4: libavfilter/split: switch to filter_frame [01:45] <cone-797> ffmpeg.git 03Clément BSsch 0724f425319d79: lavfi/thumbnail: switch to filter_frame. [01:45] <cone-797> ffmpeg.git 03Clément BSsch 07782993d9e424: lavfi/thumbnail: use avfilter_unref_bufferp() where appropriate. [01:45] <cone-797> ffmpeg.git 03Clément BSsch 075d796270c537: lavfi/thumbnail: re-use ctx instead of inlink->dst. [01:56] <cone-797> ffmpeg.git 03Anton Khirnov 07b5ecfa1d8d8d: buffersink: switch to filter_frame [01:56] <cone-797> ffmpeg.git 03Anton Khirnov 0760e50dd96023: buffersrc: switch to filter_frame [02:08] <ubitux> michaelni: if the filter is manually querying a new buffer for writing using ff_get_video_buffer in filter_frame(), it doesn't need AV_PERM_WRITE as minimum permissions for the input pads, right? [02:09] <ubitux> (classic scenario with a filter modifying each frame passing by) [02:13] <ubitux> i just made the change in vf colormatrix, and i was wondering about that flag [02:14] <michaelni> WRITE is only needed if you write into that buffer [02:14] <ubitux> into the input buffer passed to filter_frame() right? [02:15] <wm4> why does vf_colormatrix even exist, when there's libswscale [02:15] <michaelni> ubitux, yes if i understand you correctly [02:15] <ubitux> alright, then i can push [02:15] <ubitux> wm4: not the same thing [02:15] <ubitux> bt701 etc [02:16] <cone-797> ffmpeg.git 03Michael Niedermayer 07015c2b406662: libavfilter: default to filter_frame when neither it nor start/slice/end is set. [02:16] <wm4> ? [02:16] <cone-797> ffmpeg.git 03Anton Khirnov 07ece5decbe052: vf_null: switch to filter_frame [02:16] <cone-797> ffmpeg.git 03Anton Khirnov 077c42814782e3: vf_copy: switch to filter_frame [02:16] <cone-797> ffmpeg.git 03Anton Khirnov 0788f8af26a92b: vf_format: switch to filter_frame [02:17] <ubitux> wm4: sws doesn't handle bt709,bt601,fcc etc convert [02:17] <ubitux> afaik. [02:18] <cone-797> ffmpeg.git 03Clément BSsch 07269cd07702de: lavfi/colormatrix: switch to filter_frame. [02:20] <ubitux> oups& [02:21] <cone-797> ffmpeg.git 03Clément BSsch 07502ecc9cc232: lavfi/colormatrix: 10l fix forgotten buffer unref. [02:22] <wm4> ubitux: at least its interface would allow for it, and don't you agree it would be better to have this functionality to swscale instead? [02:22] <ubitux> no idea, maybe yes [02:22] <ubitux> patch welcome? :) [02:26] <michaelni> sws has partial support for yuv type [02:26] <michaelni> see the get/set colorspace functions [02:33] <ubitux> huh [02:34] <cone-797> ffmpeg.git 03Anton Khirnov 079178235ffb2d: avfilter: mark start_frame/end_frame/draw_slice as deprecated [02:35] <wm4> removing documentation for deprecated functions is a nice way to say "fuck you" to whoever has to deal with code using these functions and is trying to port it to current API [02:35] <wm4> just saying [02:37] <ubitux> what are you talking about? [02:37] <ubitux> that latest commit? [02:38] <wm4> yes [02:38] <ubitux> it's internally, it's just a fuck you to ffmpeg developers :) [02:38] <cone-797> ffmpeg.git 03Clément BSsch 07031d6448782b: lavfi/ass: switch to filter_frame. [02:38] <wm4> ubitux: mplayer uses them [02:39] <ubitux> ??? [02:39] <iive> haven't followed development. is slice functionality completely removed? [02:39] <ubitux> iive: in progress [02:39] <ubitux> should be done soon [02:39] <iive> why? too complicated? [02:39] <wm4> ubitux: why "???"? it's as I'm saying, see vf_lavfi [02:40] <wm4> though I wonder if mplayer's vf_lavfi even compiles still [02:40] <ubitux> oh it's not a problem [02:40] <ubitux> Nicolas is a ffmpeg developer ;) [02:40] <ubitux> (and he is the author of that filter) [02:40] <ubitux> oh [02:41] <ubitux> i just got a great idea. [02:43] <iive> still, somebody care to answer why is slice functionality removed? [02:43] <iive> or at least prove meaningful link. [02:45] <ubitux> " Any alleged performance benefits gained from the split are purely [02:45] <ubitux> mythological and do not justify added code complexity. [02:45] <ubitux> " [02:45] <wm4> did anyone actually measure it? [02:46] <iive> probably somebody run sd video on his i7 [02:49] <iive> ubitux: link? [02:50] <ubitux> http://git.libav.org/?p=libav.git;a=commitdiff;h=565e4993c63f797e2d50ad2f1e… [02:50] <ubitux> not yet merged since we are porting our filters [02:52] <cone-797> ffmpeg.git 03Anton Khirnov 07565e4993c63f: lavfi: merge start_frame/draw_slice/end_frame [02:52] <cone-797> ffmpeg.git 03Michael Niedermayer 0787b9dc098240: Merge commit '565e4993c63f797e2d50ad2f1e8f62fdbe299666' [02:54] <ubitux> i guess it is now :) [02:55] <iive> honestly, ffmpeg should stop merging all the crap from libav. [02:56] <ubitux> i disagree [02:56] <wm4> just like libav stopped merging all that crap from ffmpeg? [02:56] <ubitux> :) [02:57] <ubitux> iive: i will be maintainance mess if we stop merging [02:57] <ubitux> especially that kind of work [02:57] <ubitux> also, it actually simplifies a lot of things, and the filters are a lot more accessible now [02:57] <iive> well, either you maintain it, or ;et them maintain it. in the latter case, better go work for libav, because your changes will be overwritten anyway. [02:59] <ubitux> that's the burden ffmpeg developers have to live with [02:59] <ubitux> but i'd better live with this than working for a project 2 years late in comparison to ffmpeg [03:02] <cone-797> ffmpeg.git 03Anton Khirnov 0749dd71a6f1bc: vf_fieldorder: reindent [03:02] <iive> ubitux: not sure what you mean... probably i'm not quite clear either. [03:02] <cone-797> ffmpeg.git 03Anton Khirnov 074c973de9a5b6: vf_fieldorder: require write permissions [03:02] <cone-797> ffmpeg.git 03Michael Niedermayer 071eb8809a4178: Merge remote-tracking branch 'qatar/master' [03:02] <iive> but here we have a feature removal, that probably makes ffmpeg measurably slower in non-trivial cases. [03:03] <michaelni> iive do you have a benchmark ? [03:03] <iive> and it is just merged. It probably also removes the work of few developers. [03:03] <iive> michaelni: do YOU have a benchmark? [03:03] <michaelni> i tried it with vf_scale but it wasnt any faster with slices [03:03] <michaelni> that is today before the merging [03:04] <iive> i hope you tried with video chain that doesn't fit in the cpu cache. [03:05] <wm4> the advantage of slices may show up better with several consecutive slice based filters too [03:05] <iive> and, even if you did, then it could have been a sign of bug, not proof that it doesn't work. [03:06] <iive> because it did provide measurable difference in mplayer. [03:06] <iive> wm4: my point exactly. [03:06] <wm4> mplayer can't even use the slices path in most cases [03:06] <michaelni> iive, i know it was faster in mplayer at the time of single core cpus [03:07] <wm4> because ffmpeg's slice callback is not thread-safe [03:07] <wm4> (or actually the other way around, anyway, you get it) [03:07] <iive> wm4: yes, and it still delivered. [03:07] <michaelni> iive i tried 1920x1024 and 4096x4096 i do think they didnt fit in the cache [03:08] <wm4> iive: also, what's even your use case? the more complicated filters can't even do slices [03:09] <michaelni> iive, but lets imaging someone comes along tomorrow and throws a patch on the table that fixes a terrible bug that prevented slices from being faster ... [03:10] <michaelni> what would happen, is we just would revert the patches if the difference and use case is significant [03:10] <michaelni> worst would be that we would look stupid ... [03:11] <iive> 1. nobody would look into fixing bugs in non-existing code. [03:11] <iive> 2. you won't be able to remove it, because you would have built other stuff on top of it. [03:13] <iive> I don't see saste around. Have somebody asked him about it? [03:17] <iive> michaelni: you should spend less time merging stuff and more time writing your own code. [03:17] <iive> n8 ppl. [03:18] <Compn> ya we talked to saste [03:18] <ubitux> iive: i strongly disagree [03:18] <wm4> iive: the forkers said that too :) [03:18] <ubitux> we can't keep up with libav if we don't merge with them [03:18] <ubitux> IMO you better go complain/discuss with them about that drop [03:19] <ubitux> don't expect much more than here though [03:19] <iive> then merge only the good parts. not everything. [03:19] <ubitux> not merging this is really problematic [03:19] <ubitux> it will make the later merges way more complicated [03:20] <iive> well, there is no point of ffmpeg if it is libav with just little dressing over it. [03:21] <ubitux> we have thousands of bugfixes [03:21] <iive> as I said, either you rely on them to do the heavy lifting, or do it on your own. [03:21] <ubitux> dozens of formats/codecs/filters [03:21] <ubitux> and we have several others differences [03:21] <wm4> iive: you should make your own fork [03:21] <michaelni> iive, if slices are faster we should revert if not we should not [03:21] <wm4> that never merges anything [03:21] <michaelni> noone showed them faster on a modern cpu [03:21] <iive> wm4: there are enough of forks. [03:22] <iive> michaelni: you were the best person to find how to make something faster. [03:22] <iive> you would have never removed a acceleration feature without having through and through test. [03:23] <iive> right now you just don't know why it is not faster. [03:23] <iive> and you are too busy merging stuff to find out why. [03:23] <wm4> by the way, clean, simple and maintainable code > fast shitcode [03:23] <wm4> *faster [03:24] <iive> wm4: so.. lets rewrite ffmpeg on C# ... [03:24] <wm4> iive: why is ffmpeg written in C, if assembler is "faster" than C? [03:25] <iive> wm4: very few projects have more assembler code than ffmpeg. everything that matters for speed is asm. [03:25] <cone-797> ffmpeg.git 03Marton Balint 07fc38bbcd6ab3: ffplay: disallow seeking before the start of the file [03:25] <cone-797> ffmpeg.git 03Marton Balint 072efd01a32f0c: ffplay: fix updating external clock after seeking [03:25] <cone-797> ffmpeg.git 03Marton Balint 07f7eb50f3c070: ffplay: increase maximum frame duration to 1 hour for streams without TS discontinuity [03:25] <cone-797> ffmpeg.git 03Michael Niedermayer 0755a5ded67e0e: Merge remote-tracking branch 'cus/stable' [03:26] <iive> and I need to get some sleep. [03:27] <wm4> iive: then C# + asm should work out fine (assuming C# code is cleaner and more maintainable than C code) [03:27] <wm4> though I'd suggest something like Rust rather than C# [03:28] <iive> wm4: so it is win-win. go for it. [03:28] <wm4> why does he compare rewriting thousands of lines of code to removing a minor source of complexity [03:35] <ubitux> major source of complexity :p [03:35] <ubitux> this function merge really simplifies everything [03:40] <wm4> you could have the slice path without complicating normal frame filtering [03:40] <wm4> like mplayer actually does [03:48] <cone-797> ffmpeg.git 03Clément BSsch 079236e9f1e114: lavfi/ebur128: use ff_filter_frame() everywhere. [05:53] <cone-797> ffmpeg.git 03Michael Niedermayer 073fd8e07265d2: vsrc_mandelbrot: switch to filter_frame [10:23] <cone-348> ffmpeg.git 03Stefano Sabatini 07989c6a4943e3: doc/ffmpeg-codecs: add short description [10:23] <cone-348> ffmpeg.git 03Stefano Sabatini 07cf56c2076180: doc/Makefile: rework component configuration logic [10:23] <cone-348> ffmpeg.git 03Stefano Sabatini 07605f1d9865d4: lsws: define version in SWScaler class [10:54] <saste> how can i know if a muxer supports negative timestamps? [10:55] <durandal_1707> isn't there flag? [10:56] <saste> avoid_negative_ts? it's an option [11:04] <durandal_1707> saste: i mean .flags [11:38] <ubitux> saste: i didn't understand what you said for CONFIG_PROTOCOLS/ffmpeg-protocols [11:39] <saste> ubitux: i'm not yet sure I want to disable protocols doc generation if the user disabled protocols [11:40] <ubitux> ah, ok [11:40] <saste> also that will complicate the build logic [11:40] <saste> i might reconsider that in the future though [11:40] <saste> but doesn't seem crucial at the moment, so i just did the lazy thing and pushed as it was [11:41] <ubitux> sure ok [11:47] <cone-348> ffmpeg.git 03Clément BSsch 079262f13269b1: lavfi/show{spectrum,waves}: use ff_filter_frame(). [11:48] <saste> what's the name of the technique meant to generate corrupted video by altering the encoded bitstream? [11:48] <saste> it was something like smooshing [11:49] <ubitux> fuzzing? [11:49] <durandal_1707> there is noise bitstream filter [11:49] <saste> durandal_1707, no i read some time ago but can't remember [11:50] <saste> that consisted of tweaking a few bits of an MPEG video so that the result was corrupted [11:50] <saste> but generating funny effects on the played video [11:50] <ubitux> that's exactly what zzuf does, and it's called a fuzzer :p [11:51] <nevcairiel> its meant to expose code flaws like crashes though, and not produce funny effects in the resulting image .p [11:51] <saste> ubitux, but it was acting on *specific* bits, and altering it in a deterministic way [11:52] <saste> it was not meant as a tool device, but as a "recreational" effect [11:52] <saste> tool device -> debugging device [11:53] <saste> alright I'll have to dig in my mail archive [11:54] <ubitux> michaelni: why did you remove the unref_buffer in 3fd8e07265d247fd502bc7c9fcf09fa922f82335? [11:56] <ubitux> oh i'm stupid you add a reference in the original code [11:58] <saste> http://www.youtube.com/watch?v=tYytVzbPky8 [11:59] <saste> any taker? :) [11:59] <nevcairiel> somehow i dont find watching movies with missing reference frames particularly recreational [11:59] <nevcairiel> thats really all required to make that effect :P [11:59] <ubitux> oh yes [11:59] <ubitux> it reminds me something [11:59] <ubitux> in ruby maybe [12:00] <ubitux> saste: http://ucnv.github.com/aviglitch/ [12:00] <ubitux> reminds me this [12:01] <ubitux> heh it seems to link to your video ;) [12:07] <ubitux> saste: so you want to write such filter? [12:07] <saste> ubitux, bitstream filter [12:07] <saste> seems easy and fun [12:07] <saste> also useless [12:08] <ubitux> :)) [12:09] <ubitux> saste: there are a few remaining filters using start_frame/draw_slice/end_frame, maybe some you maintain [12:09] <ubitux> care to move them to ff_filter_frame? [12:10] <saste> yes, give me a few days [12:10] <ubitux> i can do it right now otherwise :) [12:13] <saste> only if you have fun doing so [12:13] <saste> which are left? [12:13] <ubitux> ~20 [12:14] <ubitux> afaict [12:14] <saste> damn why do we have so many filters? [12:15] <av500> instagram [12:53] <cone-348> ffmpeg.git 03Clément BSsch 07ea3bad0e9e3e: lavfi/smartblur: switch to filter_frame. [12:54] Action: ubitux wonders about an filter_frame_alloc_out() [12:57] <cone-348> ffmpeg.git 03Clément BSsch 07b99f1303ad9c: lavfi/concat: switch to filter_frame. [12:57] <cone-348> ffmpeg.git 03Clément BSsch 07a7eabbb20dc0: lavfi/concat: prefer av_asprintf() over stack allocated buffer. [13:04] <ubitux> saste: would be nice to have a fate test for decimate [13:04] <saste> yes [13:40] <cone-348> ffmpeg.git 03Diego Biurrun 0789145fbbfecf: x86: h264dsp: Fix linking with yasm and optimizations disabled [13:40] <cone-348> ffmpeg.git 03Diego Biurrun 079534e0f552a6: fate: lossless-audio: Add dependencies [13:40] <cone-348> ffmpeg.git 03Diego Biurrun 075116ac777424: fate: real: Add dependencies [13:40] <cone-348> ffmpeg.git 03Diego Biurrun 071f3f89656450: fate: Add dependencies for Vorbis, ProRes, QTRLE, utvideo tests [13:40] <cone-348> ffmpeg.git 03Michael Niedermayer 077dc0ed80e877: Merge commit '1f3f896564501c23b44fcf605567c78ce066b539' [13:50] <cone-348> ffmpeg.git 03Diego Biurrun 079b15c0a9b38e: x86: dsputilenc: port to cpuflags [13:50] <cone-348> ffmpeg.git 03Diego Biurrun 07a1d1fc9b4a8f: fate: Fix wavpack-matroskamode test dependencies [13:50] <cone-348> ffmpeg.git 03Diego Biurrun 07db9dbfb72a65: fate: vpx: Add dependencies [13:50] <cone-348> ffmpeg.git 03Justin Ruggles 07cdaa1f84fb6f: lavf: move "MP3 " fourcc from riff to nut [13:50] <cone-348> ffmpeg.git 03Justin Ruggles 07bfe5454cd238: lavf: move ff_codec_get_tag() and ff_codec_get_id() definitions to internal.h [13:50] <cone-348> ffmpeg.git 03Michael Niedermayer 07076300bf8b43: Merge commit 'bfe5454cd238b16e7977085f880205229103eccb' [14:05] <cone-348> ffmpeg.git 03Justin Ruggles 07261e9348ef37: lavf: add a common function for selecting a pcm codec from parameters [14:05] <cone-348> ffmpeg.git 03Justin Ruggles 075c7bf2dddee5: lavf: move nuv fourcc audio tags from riff to nuv [14:06] <cone-348> ffmpeg.git 03Michael Niedermayer 07d7b20bfbb582: Merge commit '5c7bf2dddee5bdfa247ff0d57cb8a37d19077f66' [14:14] <cone-348> ffmpeg.git 03Justin Ruggles 07c74f81786d43: nuv: cosmetics: pretty-printing [14:14] <cone-348> ffmpeg.git 03Michael Niedermayer 0752066bdb300b: Merge commit 'c74f81786d434dfaf9b3dff06aa96bfd23d0127b' [14:21] <cone-348> ffmpeg.git 03Justin Ruggles 07838ed296df91: nuv: use the stream indices generated by avformat_new_stream() [14:21] <cone-348> ffmpeg.git 03Justin Ruggles 07ab87d9b6677c: nuv: check for malloc failure when allocating extradata [14:21] <cone-348> ffmpeg.git 03Diego Biurrun 072c4593dd132e: rtpenc_chain: Remove unused variable [14:21] <cone-348> ffmpeg.git 03Diego Biurrun 0747e7fb881584: fate: Do not unconditionally run libavutil tests [14:21] <cone-348> ffmpeg.git 03Diego Biurrun 07d2f576bd4991: fate: ea: Add dependencies [14:21] <cone-348> ffmpeg.git 03Diego Biurrun 07e4d349b4014e: fate: h264: Add dependencies [14:21] <cone-348> ffmpeg.git 03Michael Niedermayer 079f8e2e92ae88: Merge commit 'e4d349b4014ee2a03f521027e0bd1ace4a9e60bd' [14:24] <cone-348> ffmpeg.git 03Justin Ruggles 0795682d8cd210: avconv: fix variable shadowing in configure_input_audio_filter() [14:24] <cone-348> ffmpeg.git 03Anton Khirnov 07e2718e7a7006: avplay: Do not use removed av_get_int() [14:24] <cone-348> ffmpeg.git 03Michael Niedermayer 070ecfcf862136: Merge remote-tracking branch 'qatar/master' [14:27] <cone-348> ffmpeg.git 03Paul B Mahol 0726f1b1a0fa6b: fate: add ADPCM 4XM test [14:27] <cone-348> ffmpeg.git 03Paul B Mahol 07a9236b87b75f: fate: add tak dependencies [15:28] <cone-348> ffmpeg.git 03Michael Niedermayer 073ae610451170: roqvideodec: check dimensions validity [15:28] <burek> i can't find in the irc logs now, but is there a way to make stderr output of ffmpeg more frequent [15:28] <burek> like 25 log lines / sec [15:30] <Daemon404> -loglevel verbose [15:30] <Daemon404> oh [15:30] <Daemon404> you mean flushing? [15:30] <ubitux> burek: someone asked for this a while ago, and iirc it was for the wrong reason [15:31] <Daemon404> flushing tons isnt really a greate idea [15:31] <burek> Daemon404 yes, more frequent flushing and ubitux i remember also that we talked about it recently but i cant find that in irc logs :S [15:31] <Daemon404> what do you need it for? [15:31] <ubitux> look on the user channel [15:31] <burek> for #ffmpeg channel, a guy wants to get the info more frequently [15:31] <ubitux> Daemon404: iirc it was a guy willing to get "real time" status for a java app or something [15:32] <burek> i guess he displays that in his app or something [15:32] <Daemon404> ubitux, "real time" [15:32] <Daemon404> lol wat [15:32] <Daemon404> the amount stderr flushes is more than enough for any progress meter [15:33] <ubitux> maybe he was counting frames using that [15:33] <ubitux> i don't remember exactly [15:33] <ubitux> it was a stupid thing anyway [15:33] <burek> yes, for most of normal use cases :) but is it possible to set the desired time period for that flush? [15:33] <Daemon404> burek, i cnat help but think you are doing something evil [15:33] <Daemon404> and wrong [15:33] <Daemon404> :) [15:33] <burek> not me :) [15:33] <burek> im just trying to help a guy :) [15:33] <ubitux> yes, even at 25 refresh seconds it would have been wrong [15:34] <Daemon404> either way [15:34] <burek> can ffprobe do this maybe [15:34] <ubitux> burek: try to figure out why he wants this [15:34] <Daemon404> i cant htink of any reason you need ro force flushing [15:34] <Daemon404> not 1 valid reason :V [15:34] <burek> [15:07:58] <TwisteR> can I somehow increase the rate of stdout messages from ffmpeg during capture? [15:34] <burek> [15:08:56] <TwisteR> I use ffmpeg's console output to generate subtitles [15:34] <burek> [15:09:52] <TwisteR> and I want 1 line of console output every 40 ms (@ 25 fps) [15:34] <Daemon404> what the hell? [15:34] <burek> :))) [15:34] <nevcairiel> subtitles....? [15:34] <Daemon404> also why not just update teh dang subtitles [15:35] <Daemon404> when theyre ready [15:35] <Daemon404> i sense hes doing soemthing -really- dumb here [15:35] <Daemon404> like one subtitle per frame [15:35] <burek> well, people get that ffmpeg is all-in-one free tool for multimedia and they try to use it for a lot of things, i guess :) [15:36] <Daemon404> burek, im syaing is entire appraoch is wrong :P [15:36] <burek> i could see that coming :) [15:37] <ubitux> he is right [15:38] <ubitux> burek: i found the log [15:38] <ubitux> "< Marlinc> So my original question was if it is possible to get those status messages more often" [15:38] <ubitux> look for this [15:39] <burek> oh thanks [15:39] <burek> that was it [15:39] <burek> :beer: :) [15:39] <burek> -progress right? [15:40] <ubitux> he was likely looking for a showinfo filter [15:40] <ubitux> but well. [15:40] <burek> maybe he is playing the video and in the background analyzing stderr and parsing/displaying something [15:40] <burek> dunno [15:41] <burek> i redirected him to read the docs about those 2 i hope he'll find what he is looking for :) [15:41] <burek> thanks again :beer: :) [15:42] <Daemon404> sounds like theyre writing something powered by duct tape and hope [15:42] <Daemon404> :/ [15:43] <burek> :D [15:43] <ubitux> powered by ducks [16:04] <cone-348> ffmpeg.git 03Michael Niedermayer 0727eada287af5: tiffdec: better checks for bitstream offsets, fixes out of array reads [16:04] <cone-348> ffmpeg.git 03Michael Niedermayer 076abb9a901fca: huffyuvdec: check width more completely, avoid out of array accesses [16:07] <ubitux> strange, i was expecting more opposition to the subtitles filter [16:07] <ubitux> then great, i guess i'll push it in a few days then :) [16:52] <simonec77> I all, I'm confused, what is the list of experimental codec? [16:52] <TimNich> ubitux: might give it a try, are there any libass version dependencies? [16:52] <ubitux> yes it uses libass [16:53] <ubitux> ah you mean a particular version? [16:53] <ubitux> i don't think so [16:55] <TimNich> I'll pull in the latest version and give it a try, once I've finished my current project let. [17:02] <simonec77> Anyone of us is try to encode using the experimental AAC encoder in FFMPEG? [18:00] <teratorn> anyone know what might be wrong seeing error like this, [libvpx @ 0x2ac7f8007cc0] Provided packet is too small, needs to be 27 [18:01] <teratorn> I encode a buncha frames then all of a sudden it starts to spit this out... where 27 is always different.. the one before was 16464 [18:22] Action: TimNich has just noticed that sometimes ffprobe puts [SAR &DAR] and sometimes leaves the brackets off... [18:23] <Daemon404> TimNich, use -of [18:23] <Daemon404> its your friend [18:26] <TimNich> but it consistently does it one way on one set of files, and the other way on another set. [18:27] <Daemon404> im sure there's a reason [18:27] <Daemon404> but ffprobe has -of for a reason :P [18:27] <Daemon404> (for pragmatic usage) [18:30] <TimNich> you can prefix -of default and its still different... [18:30] <Daemon404> you want -show_streams as well [18:31] <TimNich> no I don't, I only need the summary, but I cannot see why its inconsistent... [18:31] <TimNich> ffmbc is the same too... [18:33] <Daemon404> -show_streams is basically a summary of the streats like what youre viewing... [18:34] <Daemon404> but not using whatever weird way ffmpeg.c and ffprobe.c do it usually [18:34] <TimNich> yes, I know, but the insertion of the [] must mean something, perhaps inconsistent data or somesuch. [18:35] <Daemon404> i gave up on trying to figure out these sorts of cosmic mysteries a long time a ago [18:35] <Daemon404> :/ [18:36] Action: TimNich decides to have a beer instead... [18:36] <Daemon404> good choice [18:36] <Daemon404> i might do that too [18:36] <Daemon404> cause im trapped in UK maybe [18:36] <Daemon404> due to flooding [18:36] <Daemon404> \o/ [18:37] <TimNich> trapped where? [18:37] <Daemon404> exeter [18:37] <TimNich> Ahh, yes. Trains still not running then? [18:37] <Daemon404> nope. [18:38] <TimNich> beer it is then! [18:39] <Daemon404> yup. [19:59] <ubitux> oh, animated gif strike back! [20:07] <cone-186> ffmpeg.git 03Paul B Mahol 079a31997938c9: BRSTM demuxer [20:11] Action: Compn throws a gif at ubitux [20:12] Action: ubitux gif a throws at cone-186 [20:12] <ubitux> haem [20:12] Action: ubitux gif a throws at Compn [20:19] <cone-186> ffmpeg.git 03Piotr Bandurski 070b14c197f1ff: iff: mention all decoders [20:22] <Compn> dont worry cone-186 , i'll protect you [20:38] <llogan> why is it named "cone"? [22:13] <cbsrobot> llogan: it's the irc bot from the coneheads [22:14] <cbsrobot> meaning vlc [23:07] <llogan> cbsrobot: thanks [23:24] <cone-186> ffmpeg.git 03Michael Niedermayer 0710416a4d56fa: id3v2: check index against buffer size. Fix out of array access [23:24] <cone-186> ffmpeg.git 03Michael Niedermayer 070b28abf903cd: vble: check packet size. [00:00] --- Fri Nov 30 2012

1 0

ffmpeg.log.20121129
by burek 30 Nov '12

30 Nov '12

[00:11] <samek__> burek looks like it's working (not sure tho) but i ran it several times without a problem (from recorded file to rtmp) [02:11] <coneone> hi #ffmpeg [02:11] <sacarasc> Hello, coneone. [02:15] <coneone> just say "rtfm" if i can find my answer @ ffmpeg.org/ffmpeg.html (because i am currently reading it) -> is there an option to add _all_ available audio-streams and subtitles without specifying each one seperately? [02:16] <coneone> ok -map 0 [02:20] <coneone> i can't find the answer to my subtitles question especially the one for "how can i dump _all_ available subtitles to a seperate file?" [02:21] <coneone> i dont want the subtitles rendered into the video-stream [02:26] <coneone> one SRT file for each language would be nice! does anyone know if i can include a SRT file into MKV and if this works with say vlc? [02:27] <coneone> i read that it is possible to attach a font, did anyone try to include SRT files as attachment? [02:28] <mykul> ffmpeg is making my life so easy. that is all [02:30] <coneone> mykul that sure is true once you have your presets polished. [02:35] <coneone> also either i did something wrong or ffmpeg really is more efficient compared to mencoder, i managed to get better quality (better deinterlacing) at lower bitrate than with mencoder but i am a total noob@recoding, so this means nothing. [02:39] <coneone> according to the manual and google i will have to resort to other software to be able to cope with subtitles, please someone tell me that i'm wrong. [02:42] <wm4> coneone: subtitles are not added as attachment [02:42] <wm4> coneone: you mux them as subtitle track [02:42] <wm4> at least with mkv [02:42] <wm4> coneone: attached fonts typically won't be uswd with SRT, I think [02:42] <wm4> *used [02:45] <coneone> wm4 okay you say i can embed the subtitle track directly into the stream without having the text rendered directly into each video-frame, thats good but am i just too blind to find the according options in ffmpeg? or is it not possible to add multiple subtitle-languages to mkv? [02:46] <wm4> I don't know if ffmpeg can, but you can use mkvmerge to add subs to an existing mkv file [02:49] <coneone> wm4 thanks, your comment lead me to mkvtoolnix [02:56] <coneone> eh eh eh "i want to have my presets polished, swedished and finished!" =) [07:24] <lake> hello, i am trying to write a jack audio stream to a file with ffmpeg. [07:25] <lake> ffmpeg -f jack -i ffmpeg foo.mp3 [07:26] <lake> the audio file that is output does not have any sound [07:52] <burek> also, did you configure your jack audio properly? [09:24] <Element9> i am capturing video (from screen) with audio and the sync isn't good. here are the parameters i am using: [09:24] <Element9> -c:v libx264 -b:v 512000 -crf 0 -preset UltraFast -tune ZeroLatency -r 20 -c:a vorbis -async 44100 "c:\asdf.mp4" [09:25] <Element9> -async didn't help. is there anything else I can try? [09:26] <Element9> (ignore the -b option :) ) [10:34] <Rapper_skull> Ciao a Tutti ;) [10:34] <Rapper_skull> hello [10:34] <Rapper_skull> I am triyng to include subs into a video [10:34] <zap0> try harder [10:34] <Rapper_skull> but ffmpeg can't load fonts [10:35] <Rapper_skull> this is my command: [10:35] <Rapper_skull> ffmpeg -threads 2 -i input.mkv -vf ass=subs.ass -vcodec libx264 -acodec libfaac -profile high -level 4.1 -sn -preset ultrafast output.mkv [10:36] <klaxa> you will have to make the fonts available globally [10:36] <klaxa> i.e. extract them from the matroska file to something like ~/.fonts/ [10:36] <Rapper_skull> fonts are included in the input.mkv file and I extracted them in the same folder of input.mkv [10:37] <klaxa> <klaxa> you will have to make the fonts available globally [10:37] <klaxa> they are not available globally if you extract them to the current folder [10:37] <Rapper_skull> so there is no way to load them from a custom path? [10:37] <klaxa> none that i know of, i'd be pleased of someone can prove me wrong [10:38] <Rapper_skull> ok thanks [10:39] <Rapper_skull> there is no ~/.fonts/ in my ubuntu installation [10:40] <klaxa> Rapper_skull: https://wiki.ubuntu.com/Fonts [10:56] <Rapper_skull> I opened my /etc/fonts/fonts.conf [10:57] <Rapper_skull> and added <dir>./</dir> after <dir>~/.fonts</dir> [10:57] <Rapper_skull> it works with fonts in the current directory [10:57] <klaxa> ah nice [10:58] <Rapper_skull> onother question [10:58] <Rapper_skull> is there any way to make ffmpeg faster [10:59] <klaxa> no [10:59] <Rapper_skull> or must I change my computer? [10:59] <Mavrik> Rapper_skull: depends on which encoder you're using and what your settings are :) [10:59] <Mavrik> but yeah, if you want faster ffmpeg without loss of quality you have to buy faster cores :) [10:59] <klaxa> from what i read, ffmpeg is one of the most advanced and fastest utility available (for free) [10:59] <klaxa> *utilities [11:00] <Rapper_skull> actually it is encoding at 12 fps, not enough for transcoding [11:00] <Mavrik> Rapper_skull: I just came, can you paste your ffmpeg command and it's output so we see if there's anything you can do? :) [11:00] <Rapper_skull> just a minute [11:01] <klaxa> he already uses the ultrafast preset for libx264 [11:01] <Mavrik> oh [11:01] <klaxa> i don't think it gets any faster with h.264... right? [11:01] <Mavrik> yeah, you really can't get any faster than ultrafast :\ [11:01] <Mavrik> downscaling image helps [11:02] <Rapper_skull> i'm using [11:02] <Rapper_skull> -vcodec libx264 -acodec libfaac -profile high -level 4.1 [11:02] <klaxa> or using another codec [11:02] <klaxa> but that might increase filesize and reduce quality [11:02] <Rapper_skull> because I have 10 bits video and I must convert them to 8 bit [11:02] <Rapper_skull> + they are level 5.0, not supported by my tv [11:03] <klaxa> hah more retarded anime encoders [11:03] <klaxa> (what group if i may ask?) [11:03] <klaxa> i mean the level 5.0 high 10p release [11:03] <divVerent> how is it retarded to use features of modern codecs? [11:03] <klaxa> isn't level 5.0 for 720p tv releases totally overkill? [11:04] <divVerent> these encodes don't work on any non-computer hardware ANYWAY, due to embedded subtitles [11:04] <divVerent> klaxa: it depends on the quality you want [11:04] <Rapper_skull> the group is Task-Force [11:04] <Rapper_skull> (Italy) [11:04] <divVerent> for my purposes it's bitrate waste [11:04] <divVerent> but some people like the excessive quality [11:04] <Mavrik> divVerent: hmm, there's alot of new TVs/TV STBs that easly play MKV with subtitles but don't handle 10bit input [11:04] <divVerent> and they shall have it [11:04] <Mavrik> and that's annoying [11:04] <klaxa> you don't get that quality from TV rips [11:05] <divVerent> klaxa: "depends" [11:05] <divVerent> ;) [11:05] <klaxa> but yeah depends [11:05] <divVerent> klaxa: I'd say, if the mkv is bigger than the .ts, then they did something wrong [11:05] <divVerent> wiothout that, it may be ok [11:05] <klaxa> there is a lot to discuss, everyone has different views [11:05] <klaxa> mine might be wrong ;) [11:05] <Rapper_skull> actually they are BD-Rip [11:05] <divVerent> half .ts size is probably the largest I wouldn't call total overkill [11:05] <klaxa> oh, nvm then [11:05] <divVerent> there is no such thing as a "wrong view" there [11:06] <divVerent> you just need less quality than some peopöle want [11:06] <divVerent> same with me [11:06] <klaxa> actually i do want high quality [11:06] <divVerent> I only call it overkill if the enocded file actually encodes more visual information than the original [11:06] <klaxa> but i want it properly encoded [11:06] <divVerent> i.e. if you spend bandwidth on properly encoding the original MPEG artifacts as close as possible ;) [11:07] <klaxa> anyways, battery dying [11:07] <klaxa> bye [11:07] <divVerent> likme, if your input audio is 48kHz stereo, and you encode as 96kHz flac, then it may be smaller, but it's STUPID STUPID OVERKILL [11:07] <divVerent> if your input audio had 320kbps, and your encode uses flac, OVERKILL [11:07] <divVerent> Hi10p is borderline [11:08] <divVerent> as it often even makes smaller files than 8bit [11:08] <Rapper_skull> obliviously you must encode at the same quality, overscaling is stupid [11:08] <divVerent> due to less quantization noise [11:08] <divVerent> and it reduces banding on decode side [11:08] <divVerent> so even 8bit to 10bit encodes may actually make sense [11:09] <divVerent> for audio though, there is no similar situation [11:11] <Rapper_skull> do you think a raspberry pi can handle transcoding with my current command of ffmpeg? [11:12] <burek> no [11:12] <Rapper_skull> it is not enough powerfull, right? [11:12] <burek> it's not [11:12] <burek> i tested a lot of them lately [11:12] <burek> they suck at anything cpu intensive [11:12] <Mavrik> Rapper_skull: I very much doubt RPi is capable of encoding anything over 320x240 resolution [11:12] <Mavrik> it's just that slow [11:13] <Mavrik> and HW Encoder libraries are far from being usable :\ [11:14] <Rapper_skull> so do you know some x64 mini PCs enough powerfull? [11:14] <Mavrik> practically anything that has a quad i5+ in it can take everything you throw at it [11:15] <Rapper_skull> ok thanks [11:15] <Rapper_skull> this pc is 8 years old [11:15] <Rapper_skull> it is a miracle it still works [11:18] <Rapper_skull> do you think atom processors wold have a chance? [11:19] <Rapper_skull> *would [11:20] <Mavrik> as I tried them they were rather slow and not really meant for this use [11:20] <Mavrik> but I have no idea how the new generations perform [11:20] <Mavrik> rule of thumb is to always avoid the low-power versions of CPUs for something that intensive [11:22] <Rapper_skull> the problem is: I have my pc that is shit a this 8 years old pc, but I can substitute only one of them 2 [11:28] <burek> Rapper_skull, video encoding is a really cpu/ram intensive process.. [11:28] <burek> don't expect to get away with cheap/slow hardware utilizing some "tweaks" or something [11:29] <burek> whatever you save on buying, you'll pay later on unexpected results :) [11:29] <divVerent> even if you get it done in the time you want [11:29] <divVerent> quality will be BAD [11:31] <Rapper_skull> thaks for the advice [12:57] <samek_> Hi, is it possible to compile ffmpeg and then change one of the .c files and recompile without recompiling everything ? [13:16] <juanmabc> ? as it is not a header should be the way it's done, still linkings needs to be redone, fcourse [13:34] <DonGnom> samek_: thats what make is for. it takes care what needs to be recompiled and only compiles that. so answer is yes. [13:40] <samek_> dongon [13:41] <samek_> the funny thing is that if i change rtmpproto.c and do make after it it doesnt [13:41] <samek_> but if i change mov.c it does? [14:09] <mdsh> I need to output the average quantizer scale for every frame of an MPEG2 file, along with whether the frame is an I, P or B frame. I cannot see the qscale in FFprobe output, have I missed a parameter for FFprobe? [14:18] <gxk_> how to get information about h264 es from mpeg2ts file ? [14:20] <saste> mdsh: patch [14:20] <siganderson> hi, I'm trying to stream a file with ffserver but there is a problem: when I launch ffmpeg -i filename.mp4 http://localhost:port/filename.swf, I can see the video in the browser only while ffmpeg is working. When it finishes the video is nomore available [14:24] <saste> siganderson, as expected [14:25] <siganderson> saste, is there a way to sinchronize ffmpeg's start with the client's get request? [14:26] <saste> siganderson, no idea [14:28] <siganderson> let's suppose the video to stream is only 1 or 2 secs... how could a client see it, if while he asks for the video ffmpeg already did its work? [14:30] <mdsh> saste: thanks Stefano - I take a look [14:30] <Tjoppen> configure your http server [14:30] <gxk_> can ffmpeg show SPS and PPS from TS file ? [14:31] <Tjoppen> oh, ffserver. nm [14:32] <Guest9497> hi [14:34] <Guest9497> hhhhiiiiiii lollll [14:34] <Guest9497> lmao [14:35] <burek> siganderson, you need video on demand [14:35] <burek> take a look at the ffserver's "File" directive [14:35] <siganderson> burek, thanks [14:43] <Element9> I capturing from the screen (with audio) but the result is out of sync. I asked for help earlier already, but I didn't prepare the log. Here is the whole command line and the output: [14:43] <Element9> pastebin.com/GxA7Um0Y [14:44] <Element9> any hints or ideas would be much appreciated [14:47] <burek> Element9, try this: ffmpeg -y -f dshow -i audio="virtual-audio-capturer" -f dshow -i video="UScreenCapture" -c:v libx264 -crf 0 -preset ultrafast -r 20 -c:a libvorbis -ar 44100 asdf.mp4 [14:47] <relaxed> Element9: Try capturing at a lower framerate and maybe without vsync/async. [14:47] Action: relaxed forks burek [14:48] <burek> :) [14:48] <Element9> I'll try and get back to you [14:48] <burek> also -r 20 might be better if put before -i video="UScreenCapture" [14:49] <Element9> so, the order of inputs is important? [14:50] <burek> shouldn't be in theory, but my experimenting with cameras/desktop captures showed that it's better to first ask for the audio [14:50] <burek> donno why [14:51] <relaxed> Element9: with -r 20 after the input you're capturing at 25fps and dropping frames to achieve 20fps [14:52] <relaxed> while having it before captures at 20fps [14:53] <relaxed> So that could account for the desync. [14:54] <Element9> burek's line fixed the problem already [14:54] <Element9> test just once, but bad sync was pretty consistent so far [14:56] <Element9> relaxed: that makes sense. does input have to support that or will ffmpeg ask it for a frame on a set interval? [14:56] <Element9> burek: why did you set -ar? any logic behind it or just from the experience again? :) [14:57] <burek> well you wrote -async 44100 so I presumed you meant -ar 44100 [14:57] <burek> which is also better to put before -i audio="virtual-audio-capturer" [14:57] <burek> since there's no need to use -async when capturing both video/audio [14:58] <Element9> it's just when you mux from two separate files? [14:58] <burek> ffmpeg -y -f dshow -ar 44100 -i audio="virtual-audio-capturer" -f dshow -r 20 -i video="UScreenCapture" -c:v libx264 -crf 0 -preset ultrafast -c:a libvorbis asdf.mp4 [14:58] <burek> well yes, in your case ffmpeg is generating timestamps for both [14:58] <burek> so, why syncing? [14:59] <Element9> got it [14:59] <Element9> so seems like the order of inputs fixes it [15:05] <TwisteR> hello! [15:06] <TwisteR> can I somehow increase the rate of stdout messages from ffmpeg during capture? [15:07] <TwisteR> I use ffmpeg's console output to generate subtitles [15:08] <TwisteR> and I want 1 line of console output every 40 ms (@ 25 fps) [15:09] <Element9> burek: looks like moving -ar in the last command line you pasted made it go out of sync again. but I'll have to test some more. [15:09] <Element9> burek: thanks for the help! [15:10] <relaxed> Element9: don't set -ar unless you need to. [15:11] <Element9> good point [15:16] <blez> burek any ideas how to get the uscreecapture stream with max quality [15:18] <gxk_> how to show SPS and PPS from H264 file? [15:19] <killown> I converted a 1.5GB video for 231MBwebm and it has a good 1080p quality!! webm is amazing [15:24] <burek> blez -crf 0 -preset ultrafast [15:24] <burek> i.e. capture in lossless [15:25] <burek> gxk_ did you try ffprobe ? [15:26] <burek> TwisteR, there was some talk not that far ago about that same thing [15:41] <burek> TwisteR, it was suggested that you take a look at -progress option [15:41] <burek> or read about showinfo filter [15:46] <TwisteR> burek, ok, thanks for tip [15:47] <samek_> is there a "central" file which is feeding packets to muxers ? [15:47] <samek_> i need to debug something and i want to set up some debbuging there also. [15:50] <Tjoppen> look for ->write_packet in utils.c [15:52] <gxk_> burek: can you help with command? [15:55] <burek> gxk_, try man ffprobe [15:55] <burek> but usually -show_format, -show_streams, etc should help [15:55] <burek> i don't use it that much so i don't know precisely [15:57] <burek> -show_packets and -show_frames maybe [15:58] <samek_> tnx [15:59] <apo> Hi [16:00] <apo> I'm doing this: ffmpeg -nostats -nodump -nohex -v quiet -y -i "$song" -acodec libvorbis -aq 6 "$filename", but ffmpeg still calls do_pkt_dump for some reason. Ideas? [16:01] <apo> ... and only if I call that from a script [16:01] <apo> as I just found out [16:16] <gxk_> burek: i'll try [16:17] <apo> (The script does not export any variables) [16:29] <apo> ... [16:29] <apo> it was because ffmpeg was reading the stdin that was supposed to be looped over by the script [16:32] <mdsh> saste: I think I'm right in thinking that if frame->quality/(float)FF_QP2LAMBDA returned ~0 then I've got a lot more work to do [16:33] <ubitux> (saste isn't here) [16:59] <mistym> I can't seem to get ffmpeg to build statically on OS X. I provided --disable-shared and --enable-static to ./configure, and all of the optional libraries I'm using have static libraries available, but the resulting binary is still dynamically linked. [17:06] <relaxed> mistym: you need --extra-ldflags="-static" too [17:09] <mistym> relaxed: Hm, that gives me "gcc is unable to create an executable file." [17:10] <relaxed> try again without any --enable-* [17:11] <mistym> relaxed: With --extra-ldflags="-static" as the only argument, it fails. [17:11] <relaxed> you still need --disable-shared --enable-static [17:12] <mistym> It fails with those too. [17:12] <relaxed> hmm, I don't have any experience with building on OS X, only Linux. [17:12] <relaxed> Are you using a modern version of gcc? [17:14] <mistym> Ah, apparently -static is simply not supported on OS X because the system libs are dynamic-only. It should be possible to link to most libraries statically while linking to the system libraries dynamically, not sure how to tell ffmpeg to do that though. [18:03] <PMacDiggity> Hello, I can't seem to find a list of OS X SDI video input devices supported by ffmpeg, does anyone know what these are/know a list (the list at http://ffmpeg.org/general.html#Input_002fOutput-Devices only seems to include Windows and Linux devices) [19:14] <xroberx> hi [19:15] <xroberx> does anyone know if it is possible to somewhat "sign" a video so I am the only one able to open it ? [19:15] <retard> you could encrypt it in some kind of archive for which only you have the key? [19:16] <xroberx> mmm [19:18] <xroberx> the situation is this: I've developed an application for Android which uses a series of videos I've created. The application is generic enough that a "pirate" could change the videos and make a copy of the application with his own videos... [19:19] <xroberx> retard: so your solution wouldn't work, because I need to somewhat lock the video playback in my application to only "signed" videos [19:19] <sacarasc> Time to invent your own DRM! Which is usually a failure... [19:19] <xroberx> i.e I want my application to ony play the videos I've created [19:20] <JEEBsv> xroberx: you could sign them and have the app check the signature, but... it too could be easily circuimvented in the end [19:20] <xroberx> JEEBsv: you mean using something like MD5 or SHA-1 ? [19:20] <JEEBsv> no [19:20] <JEEBsv> having a file signed with a certificate [19:21] <retard> or you could employ a stream cipher [19:22] <xroberx> retard: yeah, but that would be too much for a mobile CPU to handle [19:23] <JEEBsv> with a certificate someone could just overwrite the cert you have there, or if you read the cert from the web they could just change the URI [19:23] <JEEBsv> tl;dr it can be in any case circuimvented [19:24] <JEEBsv> I think the only way to do it is to not have the videos in the application, but only available on a server that you connect to and ask for credentials... of course that means saying goodbye to offline viewing [19:24] <JEEBsv> tl;dr DRM is always defective and there are better ways of putting in your development time into something [19:24] <xroberx> JEEBsv: what about using a hash (MD5, SHA-1, etc) ? How that could be circunvented if I only allow a set of known hashes ? [19:25] <JEEBsv> same way like with the certificate [19:25] <JEEBsv> the string could be changed [19:26] <xroberx> JEEBsv: I could hide the hash string or encode it somewhat... [19:27] <JEEBsv> still doesn't change the fact that you can still edit the binary and get what you want [19:27] <xroberx> sure [19:27] <JEEBsv> in that case the person might just reverse engineer the dalvik byte code [19:27] <JEEBsv> for the generic features [19:27] <JEEBsv> and that's it [19:28] <xroberx> but it's just a matter of making it more difficult to crack [19:28] <xroberx> not impossible of course [19:28] <JEEBsv> suit yourself, but personally that just seems like a really dumb thing to put development effort into [19:29] <xroberx> I've never done something like that, but I know there are some chinese pirates out there that take your application, change some strings and publish it under their brand name [19:30] <xroberx> and Google won't help me if that happens [19:31] <JEEBsv> then they might as well change the hashes or the cert... puh-re-tteh futile [19:31] <xroberx> ok [19:31] <JEEBsv> if it happens, it happens [19:31] <xroberx> i'll see then, thanks for your time! :) [19:32] <JEEBsv> np [21:05] <danverde> in a video player I am writing I call av_register_all and avcodec_register_all to register the available codecs. Can I only register specific codecs instead of all? [21:07] <danverde> nm, I'll try to do what av_register_all does for only the codecs I want [23:34] <lake> if someone could help me undertand why ffmpeg/v4l2 breaks arecord, that would be AWESOME! http://pastie.org/5455090 [23:49] <llogan> lake: try a simpler command and see if the behavior continues [23:49] <llogan> ffmpeg -f alsa -i hw:6,0 -acodec pcm_s16le -ac 2 -ar 48000 out.wav [23:49] <llogan> or maybe: ffmpeg -f alsa -i hw:6,0 -acodec copy out.wav [23:52] <llogan> makybe i should have kept it as mkv [00:00] --- Fri Nov 30 2012

1 0

ffmpeg-devel.log.20121128
by burek 29 Nov '12

29 Nov '12

[00:01] <burek> yeah, fast and without a reload, which slows it down [00:02] <ubitux> burek: nit/style: i have a right shift of the content on hover [00:02] <ubitux> would be nice if a padding was filled instead [00:02] <ubitux> (the vertical black bar on the left seems to be in cause) [00:02] <burek> i see, i had to put some indicator of which column is highlighted for colorblinds and similar people [00:03] <burek> but i can remove it [00:03] <ubitux> the hilight is fine, but it should just be a color change [00:03] <ubitux> not an addition [00:03] <ubitux> like it seems to be [00:04] <ubitux> border-left: 2px solid black put that border all the time, with the default color [00:05] <ubitux> i also have an overflow on the tooltip [00:05] <burek> can you refresh and try now [00:06] <saste> burek: link? [00:06] <burek> http://ffmpeg.gusari.org/fate/ [00:06] <ubitux> burek: sounds good [00:06] <ubitux> burek: still that overflow: http://ubitux.fr/pub/pics/_fate-burek-tooltip-overflow.png [00:07] <burek> oh i see [00:07] <cbsrobot> ubitux: stop using netscape 4.7 ! [00:07] <burek> :D [00:07] <ubitux> cbsrobot: haha [00:08] <saste> burek: cool :) [00:08] <cbsrobot> cfg = configure was successful & ? [00:09] <cbsrobot> shouldnt it be cws ? [00:09] <burek> well, i'll need some help about that part [00:09] <burek> since im not a developer and i dont know what you need [00:09] <burek> i put something that looked reasonable to me [00:23] <ubitux> btw, still no one to improve our gif encoder? [00:24] <ubitux> it's pretty still sad we can't generate gif with a decent size for the web with ffmpeg [00:24] <ubitux> pretty <-> still [00:25] <saste> ubitux: add it to your todo ;-) [00:26] <durandal_1707> it is all about pallete and there was already some code [00:28] <ubitux> durandal_1707: i think it's all about non-intra :p [00:29] <durandal_1707> really? you cant get useful gif encoder without pal output [00:30] <ubitux> i'm not sure what you mean there [00:31] <durandal_1707> you need scaling [00:31] <ubitux> i get fairly good colors with our encoders if i do add format=rgb8,format=rgb24 at the end of the filtergraph [00:32] <ubitux> encoder* [00:33] <durandal_1707> that is far from optimal [00:35] <ubitux> http://lucy.pkh.me/bbb-default.gif http://lucy.pkh.me/bbb-sws-rgb8-rgb24.gif [00:35] <ubitux> both are almost 3MB unfortunately [00:35] <ubitux> :( [00:40] <durandal_1707> no compression? [00:40] <ubitux> it's intro only afaik, no-reuse of the previous frame [00:40] <ubitux> intra* [00:40] <ubitux> which is imo the reason they are so huge [00:41] <durandal_1707> but gif is soo prev century so i'm no motivated to write Encoder [00:42] <ubitux> it's just improving the existing one ;) [00:42] <ubitux> gif are nice for loops [00:42] <burek> ubitux, is it better now? [00:42] <ubitux> when i see how buggy and inconsistent the html5 support is& [00:42] <durandal_1707> better finally get proper demuxer&decoder [00:43] <ubitux> burek: yep [00:43] <burek> cool [00:44] <ubitux> durandal_1707: wasn't the guy supposed to re-send a new version soon"®©? [00:44] <durandal_1707> ubitux: bug him more [00:45] <ubitux> :) [00:45] <ubitux> i'm already busy bugging myself getting things done [00:47] <Compn> you guys are working too hard [00:47] <Compn> dont you have lives!? :) [00:47] <ubitux> michaelni: would it be ok to seek in the subtitles as it is now, but adding on top of the queue for read packet all the subtitles which are on current overlap (and eventually with the start ts at the requested time)? [00:48] <burek> do you developers need the same kind of information, that is on the current fate web page, i.e. should i just copy/paste all the current info, or do you need some more info or in a different format or something.. what is the usual way a developer analyzes that page? [00:48] <ubitux> Compn: define:lives [00:48] <Compn> non ffmpeg related time :P [00:48] <ubitux> Compn: please provide examples, i can't guess what you are talking about [00:49] <Compn> hehe [00:49] <ubitux> burek: i'm not interested in working on this personally, sorry [00:49] <ubitux> Compn: you sound like another clueless user! [00:49] <burek> ok, im just asking for a clue how to know if the data presented would be enough or not :) [00:50] <llogan> burek: http://pastebin.com/q8bLZFgk [00:50] <llogan> might be slightly ugly but i looked at it too long [00:50] <burek> :) ok, will test it :) [00:51] <ubitux> anyway, what are you going to do with this frontend now burek > [00:51] <ubitux> ? [00:52] <burek> i intend to place 1 trigger inside the current script on the fate server, which will make possible to generate the data for this layout (preserving the old fate cgi stuff) [00:52] <burek> so, you'll have both [00:52] <burek> this one should be a lot faster, since it won't constantly decompress things before display [00:53] <ubitux> ok [01:00] <michaelni> ubitux, suggested queeuing behavior sounds ok [01:02] <ubitux> maybe it could implemented easily with setting a "nb_skip" along with a "nb_cached" in the context [01:02] <ubitux> so read_packet returns first nb_cached, then skip nb_skip and then goes back to normal [01:03] <ubitux> i'm unsure about changing the initial pts though [01:03] <ubitux> (since the renderer might get duplicates?) [01:03] <michaelni> burek, about how i use fate, 1. i try to find out if anything got worse (cfg, compile, tests) 2. i try to find out what newly failed exactly, and sometimes when it last worked and when it first failed and any additional details about the failure and the box that failed [01:05] <michaelni> currently the new fate shows no hint of what failed at all just that there are 4 failures for a client for example [01:06] <michaelni> when is meant in the sense of git revission not time [01:06] <michaelni> with 2 git revissions one can bisect [01:07] <burek> if the fate client sends its configure/compile/tests log, is that enough info to get all that you mentioned right now? [01:10] <burek> llogan, is it better now? [01:12] <michaelni> old fate has all the info so it should be possible for new too [01:12] <burek> ok, for a start, we can copy all that was there, and remove/add things after that [01:12] <michaelni> yes [01:13] <durandal_1707> michaelni: looks like encoding pcm_s24le in nut is broken [01:13] <durandal_1707> it is misdetected as s16le [01:13] <michaelni> for example look at old fate and click on ond of the black trianngles in a cell that shows the count of failed tests [01:14] <llogan> burek: maybe blue was better. [01:14] <llogan> just kidding. looks good to me. [01:14] <burek> :) ok, it's more in the fashion of ffmpeg this way imho [01:15] <ubitux> now put some black in background [01:15] <ubitux> ;) [01:15] <burek> it will look scary that way :) [01:15] <ubitux> that's the spirit [01:15] <durandal_1707> michaelni: it is because tag is picked from wav [01:16] <durandal_1707> and this cant work because nut does not store bits per coded sample [01:16] <ubitux> http://ubitux.fr/pub/pics/_fate-burek-stylish.png [01:16] <ubitux> much better that way, but i loose the line color switch :( [01:16] <llogan> burek: .sortable th {... color: #333; ...} [01:17] <burek> I think I'll create presets :D [01:18] <llogan> let's make it a flash page. [01:18] <burek> some icons in the top, which change the colors :) [01:18] <burek> :)) [01:18] <ubitux> and again, we need a good gif encoder to add some on the page [01:18] <llogan> "ubitux's ugly ass gray" [01:18] <ubitux> :) [01:18] <Compn> people still use gif for forum avatars [01:18] <Compn> and memes [01:18] <durandal_1707> michaelni: that is because you did not merged libav hunk which fixed this [01:19] <cbsrobot> Compn: and gifs ! [01:19] <Compn> and gifs! [01:19] <Compn> you check out gifsound yet ? [01:19] <durandal_1707> this is sorry state, becuase fate do not covers such scenarios at all. but really should [01:20] <durandal_1707> michaelni: what is your solution? [01:20] <michaelni> durandal_1707, ill fix it ASAP [01:21] <michaelni> the hunk btw was buggy i must have missed that it did fix this [01:22] <llogan> ubitux: want to write the "A Beginner's guide to git bisect" article? [01:22] <ubitux> here it is: "man git-bisect" [01:23] <ubitux> you can retweet it [01:23] <llogan> i don't "tweet". [01:23] <ubitux> you just did [01:23] <llogan> and luser's don't read man pages [01:23] <ubitux> it's < 140 char, it's a tweet [01:23] <Compn> i'll tweet your tweet! [01:23] <ubitux> llogan: here is the fancy html one: http://git-scm.com/docs/git-bisect [01:24] <ubitux> it's hard to make it simpler [01:24] <ubitux> it starts with an example [01:24] <llogan> forget i axed [01:34] <durandal_1707> ubitux: i want spectrogram filter like one in sox, that creates spectrogram image of whole file [01:35] <ubitux> showspectrum isn't good? [01:36] <cbsrobot> there is still time to celebrate and join the oxford dict. word of the year 2012 party ! [01:36] <cbsrobot> http://blog.oxforddictionaries.com/2012/11/us-word-of-the-year-2012/ [01:36] <ubitux> durandal_1707: i think saste has some lavfi sox stuff [01:36] Action: Compn wonders what the word of the year is [01:36] <Compn> oh lawd [01:36] <cbsrobot> Compn: you'll never guess [01:37] Action: ubitux wonders what year it is [01:37] <Compn> no, i wouldnt have guessed that haha [01:37] <ubitux> still 2012? [01:37] <Compn> oxford is a few years old [01:37] <Compn> i mean, books ? who has books anymore ? [01:37] <Compn> i havent touched a book in months [01:37] <Compn> fithly disgusting germy things [01:42] <llogan> Compn: look at your keyboard [02:06] <cone-624> ffmpeg.git 03Michael Niedermayer 072754dbb6e151: nut: put nut PCM audio tags in front of avi tags. [03:28] <durandal_1707> ubitux: this ebur128 filter is killer [03:46] <Compn> whats ebur128 mean [03:46] <Compn> do i mean [03:46] <Compn> great hes gone [03:46] Action: Compn talks to himself [03:47] <Compn> audio loudness filter ? [03:54] <burek> people are getting more and more complex ideas for ffmpeg usage :) http://ffmpeg.gusari.org/viewtopic.php?f=16&t=743 [03:54] <burek> the guy wants to create "virtual environment" with 4 cams in sync :) [03:58] <Compn> yes, once you introduce filters, theres no going back [03:59] <Compn> watch in a few months someone will come in asking if there is a limit to the number of inputs that can be added at once :D [04:09] <burek> :)) [05:26] <cone-834> ffmpeg.git 03Michael Niedermayer 072c923983b6e5: qpeg: reset palette on seeks [10:22] <burek> michaelni, I've added configure/compile/test logs in the popup on mouse over those cells that display errors/warnings etc. so, is it better that way or should we embed the data directly into the html? http://ffmpeg.gusari.org/fate/ [10:22] <burek> (this way it loads only on mouse over) [10:23] <burek> the previous fate page had data embeded into the html table, which required all logs to be processed too, in order to generate the data [10:25] <burek> also, I configured apache to server gzipped pages, to save space on hdd, which means, all those pages you see loaded in your browser are actually stored on the server as xxxx.html.gz (no plain-text htmls), which should additionally save space on the server [10:25] <burek> to serve* gzipped [11:20] <durandal_1707> is there any format that use half-float pcm? [12:00] <cone-797> ffmpeg.git 03Paul B Mahol 071793128cf7d9: avrdec: fix typo in .long_name [12:10] <cone-797> ffmpeg.git 03Paul B Mahol 07b0d9b011c752: avrdec: sample rate takes 3 bytes not 4 [13:04] <michaelni> burek, the mouse over doesnt show what failed, it shows the whole output of each stage raw [13:05] <michaelni> the complete raw output should be on seperate pages [13:06] <michaelni> the mouse over should show just what failed [13:09] <ubitux> durandal_1707: oh? do you use it? :) [13:10] <durandal_1707> half-float? [13:10] <ubitux> no ebur128 [13:11] <durandal_1707> i tried video output and it is fantastic [13:11] <ubitux> burek: the over isn't really handy to "use" [13:11] <ubitux> durandal_1707: haha ok :) [13:11] <ubitux> glad you like it [13:11] <durandal_1707> ubitux: though it hangs for apes with BIG packets [13:12] <ubitux> oh? :( [13:12] <ubitux> oh here we go, l merged the callbacks [13:13] <ubitux> i guess i'll have to rework some of my filters [13:31] <cone-797> ffmpeg.git 03Martin Storsjö 0754fd593a0e63: configure: Enable avconv filter dependencies automatically [13:31] <cone-797> ffmpeg.git 03Diego Biurrun 0728e1cf19aa3c: x86: h264_weight: port to cpuflags [13:31] <cone-797> ffmpeg.git 03Justin Ruggles 077f534d11ed37: lavr: do not pass sample count as a parameter to ff_audio_convert() [13:31] <cone-797> ffmpeg.git 03Justin Ruggles 07d2f9f8e094d9: lavr: correct the documentation for the ff_audio_resample() return value [13:31] <cone-797> ffmpeg.git 03Justin Ruggles 071d86aa8b0f49: lavr: do not pass consumed samples as a parameter to ff_audio_resample() [13:31] <cone-797> ffmpeg.git 03Piotr Bandurski 07f5fa03660db1: vble: Do not abort decoding when version is not 1 [13:31] <cone-797> ffmpeg.git 03Michael Niedermayer 07264441715b13: Merge commit 'f5fa03660db16f9d78abc5a626438b4d0b54f563' [13:38] <cone-797> ffmpeg.git 03Diego Biurrun 07490df522c732: x86: cpu: Drop unused HAVE_RWEFLAGS condition [13:38] <cone-797> ffmpeg.git 03Diego Biurrun 072e89aeed6543: x86: h264_idct: port to cpuflags [13:38] <cone-797> ffmpeg.git 03Anton Khirnov 07af7d13ee4a4b: asink_nullsink: plug a memory leak. [13:38] <cone-797> ffmpeg.git 03Michael Niedermayer 0742d3fea65f78: Merge commit 'af7d13ee4a4bf8d708f9b0598abb8f6e22b76de1' [14:08] <cone-797> ffmpeg.git 03Paul B Mahol 0723101ceaa226: avrdec: implement seeking [14:11] <burek> michaelni, how to generate what has failed? [14:11] <burek> ubitux, why? [14:15] <michaelni> burek, see old fate [14:16] <burek> i did but i dont quite get it.. [14:18] <ubitux> burek: the popup goes outside the visible area, and it's uneasy to scroll [14:18] <burek> ubitux, you really need to update your browser :) [14:19] <burek> it's standard css + little bit of js, nothing else [14:19] <burek> i tried in chrome+ff+ie and it's the same [14:19] <ubitux> it's firefox 17& [14:19] <burek> did you update it? [14:19] <ubitux> update to what? [14:19] <ubitux> it's the latest one [14:19] <burek> latest.. [14:20] <burek> oops, my bad [14:20] <burek> sorry [14:21] <ubitux> http://ubitux.fr/pub/pics/_fate-burek-popup-scroll.png [14:21] <ubitux> bottom scroll get extended on hover [14:21] <ubitux> (full bottom on screenshot) [14:21] <burek> yes yes, sorry, i just checked one more time [14:21] <burek> and saw the same [14:22] <burek> last changes broke it [14:37] <durandal_1707> michaelni: there are samples on fate that are never used [14:37] <durandal_1707> somene forget to apply patch? [14:45] <burek> ubitux, should be fine now [14:45] <burek> different javascript implementations across browsers.. :/ [14:47] <burek> michaelni, would it be enough to just auto-scroll the log to the bottom in order to show what went wrong? [14:56] <ubitux> burek: doesn't work that well for the ones at the bottom [15:02] <leandrosansilva> Hello to all. Why is the field format of AVFrame a int? It should be an AVPixelFormat, shouldn't it? [15:02] <leandrosansilva> Is it safe to cast it (format) to an AVPixelFormat? [15:25] <ubitux> leandrosansilva: can't be it be an AVSampleFormat in case of audio? [15:25] <ubitux> enum are int anyway [15:27] <leandrosansilva> ubitux, yes I know. I just want to know if it can change in the future. But I hadn't thought about the question of audio, because I'm using only video :-) [15:27] <leandrosansilva> thx for the help [15:34] <durandal_1707> michaelni: actually some of them are pointing to bug in decoder [16:42] <cone-797> ffmpeg.git 03Reimar Döffinger 078aa29f063c8d: atrac3: do not use init_static_data to init VLC data. [16:42] <cone-797> ffmpeg.git 03Reimar Döffinger 0792947c6d7278: Use err_out label for error-case cleanup. [16:42] <cone-797> ffmpeg.git 03Reimar Döffinger 0716af29a7a6de: vc1dec: add hacks to ensure static tables are initialized in open(). [16:49] <cone-797> ffmpeg.git 03Michael Niedermayer 07cd7febd33f20: lavfi: replace filter_samples by filter_frame [16:49] <cone-797> ffmpeg.git 03Michael Niedermayer 07f6dd145533ee: avfilter: check that filter_frame and slice variants are not both used [16:49] <cone-797> ffmpeg.git 03Michael Niedermayer 07a0b8eec71932: default_filter_frame: fix argument name [16:49] <cone-797> ffmpeg.git 03Michael Niedermayer 073ed483cdfa5b: libavfilter: Support using filter_frame for video [16:49] <cone-797> ffmpeg.git 03Michael Niedermayer 078227a0b7df7b: libavfilter: update doxy for filter_frame [16:49] <cone-797> ffmpeg.git 03Anton Khirnov 071b43fc127cc3: vf_hqdn3d: switch to filter_frame [16:59] <cone-797> ffmpeg.git 03Paul B Mahol 0701eed8c60762: iff demuxer: 16SV support [17:23] <cone-797> ffmpeg.git 03Paul B Mahol 076549a9b75333: pgssubdec: remove unused variable [17:30] <Daemon404> humm what does dup=1 drop=0 mean in ffmpeg cli? [17:30] <Daemon404> reading teh code has proved to be less than useful to figure it out [17:30] <JEEB> one duplicated frame (For frame rate differences or whatever), and zero dropped frames [17:32] <Daemon404> i see [17:32] <Daemon404> god im so tired of dealign with teh weird shit 'pro' apps do [17:37] <cone-797> ffmpeg.git 03Anton Khirnov 0705945c7f8827: libavfilter/fifo: switch to filter_frame, the filter did not support slices. [17:37] <cone-797> ffmpeg.git 03Anton Khirnov 07250b122e3ab1: vf_boxblur: switch to filter_frame, boxblur did not support slices [17:45] <Daemon404> Duration: 00:08:31.45, start: -2.622623, bitrate: 13400 kb/s [17:46] <Daemon404> ^ like this stuff [18:12] <cone-797> ffmpeg.git 03Michael Niedermayer 0746a4e72bc518: vf_delogo: fix order of operations in apply_delogo() [18:12] <cone-797> ffmpeg.git 03Michael Niedermayer 07ed547e2ce5d5: vf_hqdn3d: fix permissions [18:12] <cone-797> ffmpeg.git 03Michael Niedermayer 07384641468b60: vf_delogo: switch to filter_frame, this filter did not support slices [18:12] <cone-797> ffmpeg.git 03Michael Niedermayer 077bfa2f25c021: vf_drawtext: switch to filter_frame, this filter did not support slices [18:17] <ubitux> michaelni: i can care of the filters i wrote if you don't want to do the switch for them [18:29] <michaelni> ubitux, feel free to switch all that are not in l and that at the same time currently do not support real slices anyway [18:29] <michaelni> iam still a bit undecided what to do about filters that do fully support slices currently [18:31] <ubitux> does it look like it will affect performances? [18:45] <michaelni> ubitux, slices certainly made a speed difference at the time of single core cpus, they maybe dont make one now anymore, and if you want you surely can switch to filter_frame for all filters that arent in l or that you maintain [18:46] <michaelni> the main reason for supporting both ATM is to ease switching filters one by one and keeping git bisectability and all that, ive no real oppinon about droping slices entirely ATM [18:54] <michaelni> note, "time of single core cpus" i also meant libmpcodecs and mplayer as testbed, that was before libavfilter [18:56] <ubitux> ok [19:00] <Compn> mplayer has -noslices but i dunno if it affects lavfi or filters [19:35] <cone-797> ffmpeg.git 03Michael Niedermayer 07b5572ab861d7: vf_boxblur: cosmetics to reduce diff to qatar [19:36] <cone-797> ffmpeg.git 03Michael Niedermayer 0759907340e1e1: libavfilter/fifo: fix indent [19:36] <cone-797> ffmpeg.git 03Anton Khirnov 07e67fdbffe588: vf_fieldorder: switch to filter_frame, this filter did not support real slices [19:36] <cone-797> ffmpeg.git 03Anton Khirnov 0792b57e8dac76: vf_fps: switch to filter_frame [20:04] <cone-797> ffmpeg.git 03Clément BSsch 0757d9c86bf34e: lavfi/edgedetect: switch to filter_frame. [20:07] <cone-797> ffmpeg.git 03Anton Khirnov 076325957b2885: vf_frei0r: switch to filter_frame, the filter did not support slices [20:07] <cone-797> ffmpeg.git 03Anton Khirnov 072a06bc10e7cb: vf_gradfun: switch to filter_frame, this filter did not support slices [20:09] <ubitux> the out_buf thing was pretty nice actually [20:09] <ubitux> any idea why it wasn't kept? [20:21] <cone-797> ffmpeg.git 03Clément BSsch 075d170b09d3e6: lavfi/geq: switch to filter_frame. [20:26] <cone-797> ffmpeg.git 03Anton Khirnov 071f94f86ab560: vf_libopencv: switch to filter_frame, this filter did not support slices [20:26] <cone-797> ffmpeg.git 03Michael Niedermayer 07e42b7ff2b2fd: vf_pixdesctest: switch to filter_frame [20:34] <wm4> why is Libav still called "qatar" in commit messages? [20:35] <ubitux> Voldemort syndrom [20:43] <llogan> wm4: maybe so people don't equate it to libav* [20:56] <durandal_1707> incoming.... [21:17] <cone-797> ffmpeg.git 03Daniel Verkamp 07a0958949ebe1: Vivo demuxer [21:24] <cbsrobot> \o/ for #132 closed [21:25] <cbsrobot> not that I care much about viv demuxer ... [21:28] <llogan> originally roundup 99 [21:28] <llogan> drv and durandal_1707 are zombie killers [21:30] <wm4> vivo demuxer is in mplayer, but was not in libavformat [21:30] <durandal_1707> llogan: zombies are still there viv 2 is not supported at all [21:31] <durandal_1707> because there are no decoders [21:35] <durandal_1707> there is one big demuxer left TiVo(ty, tivo), but mplayer ty code and any code on net is hardly readable so I abondoned it for now [21:35] <wm4> would it be possible to add a decoder for "Hauppauge Macroblock" raw video? I don't really know whether this belongs into libavcodec though; seems to be a mplayer internal thing used for TV input (covered by vd_hmblck.c) [21:36] <cone-797> ffmpeg.git 03Michael Niedermayer 076be0df508019: vf_pad: switch to filter_frame [21:36] <cone-797> ffmpeg.git 03Anton Khirnov 070663a9eab8b8: vf_showinfo: switch to filter_frame [21:36] <cone-797> ffmpeg.git 03Anton Khirnov 072c3b665379de: vf_select: switch to filter_frame [21:37] <durandal_1707> wm4: what commit introduced it? [21:38] <wm4> durandal_1707: it was added in 2003 [21:39] <wm4> this thing http://repo.or.cz/w/mplayer.git/blob/HEAD:/libmpcodecs/vd_hmblck.c [21:46] <wm4> (unfortunately I don't even have a sample file) [21:47] <durandal_1707> video4linux have support for that [21:48] <durandal_1707> it is more pixel format that codec [21:48] <durandal_1707> *than [22:03] <cone-797> ffmpeg.git 03Michael Niedermayer 07afded1816b12: vf_transpose: switch to filter_frame [22:03] <cone-797> ffmpeg.git 03Anton Khirnov 0745eed9b19751: vf_unsharp: switch to filter_frame, this filter did not support slices [22:03] <cone-797> ffmpeg.git 03Anton Khirnov 07c262e8cff633: vsink_nullsink: switch to filter_frame [22:03] <cone-797> ffmpeg.git 03Anton Khirnov 07f7f6f281dd8e: vsrc_testsrc: switch to filter_frame [22:03] <cone-797> ffmpeg.git 03Michael Niedermayer 07eb1424b4fef2: src_movie: switch to filter_frame [22:04] <wm4> so, is this the removal of the slice rendering path? [22:04] <ubitux> yes [22:05] <saste> well for sure it simplifies things a bit [22:05] <Compn> cbsrobot : everyone cares about vivo! :P [22:06] <Compn> saste : is libav more open to video filters now ? [22:07] <saste> do we have a list of missing formats/codecs somewhere? [22:07] <Compn> i dont remember much enthusiasm for them previously [22:07] <wm4> does libavfilter still has DR similar to mplayers? [22:07] <Compn> wiki under category 'formats missing in ffmpeg' [22:07] <saste> Compn, no idea, why do you ask? [22:07] <Compn> saste : just seeing more filter work [22:07] <Compn> just curious [22:07] <ubitux> Compn: http://lucy.pkh.me/diff/diff-filters.html long path to go :) [22:08] Action: cbsrobot hands Compn the blue ribbon of "I want to vivo" [22:08] <Compn> http://wiki.multimedia.cx/index.php?title=Category:Formats_missing_in_FFmpeg [22:08] <saste> Compn, they seem mostly care about the framework, don't know if they plan to implement features/filters [22:08] <Compn> i'm still waiting for mvi2 ... [22:08] <Compn> motion pixels movie cd avi cdoec :) [22:08] <saste> they never seemed to care about porting ffmpeg filters [22:09] <ubitux> btw, i'm doing the same thing as 568c70e7 for our filters [22:09] <saste> but maybe that's because of NIH syndrome [22:09] <ubitux> (and what a pain&) [22:09] <Compn> saste : oh, i thought someone was posting patches to libav [22:09] <Compn> or merging from ffmpeg [22:09] <Compn> for new filters [22:09] <ubitux> no [22:09] <ubitux> they don't care [22:10] <Compn> makes my brain hurt heh [22:10] <saste> ubitux, what's the gain for that? [22:10] <llogan> they never seemed to care about porting ffmpeg * [22:10] <saste> i mean i believe it is related to some compiler support or what? [22:10] <ubitux> saste: consistency, and seems to have a portability issue [22:10] <Compn> i thought we were going to have some collaboration :( [22:10] <ubitux> saste: seems to, according to the commit desc [22:11] <saste> ok, i'll wait for the water to calm down before touching filters again [22:12] <ubitux> i miss out_buf :( [22:13] <uglyoldbob> how hard would it be to modify the code to add a "fallback" file to a feed on ffserver? ffserver would use data from this file when the primary source is not present [22:14] <Compn> uglyoldbob : you mean, like a background movie that plays no matter what ? [22:14] <Compn> good question [22:15] <uglyoldbob> Compn: yeah it would only be visible when the feed is not being fed [22:15] <wm4> ubitux: does libavfilter support this thing that a filter can get a buffer from a subsequent filter? (also known as direct rendering) [22:16] <saste> wm4, yes [22:16] <saste> although buffer handling in lavfi is a bit involved [22:16] <wm4> always striked me as very complicated and of little actual use [22:16] <saste> "very complicated" -> true [22:16] <saste> "little actual use" -> false [22:17] <saste> it saves pointless memcpies (and allows DR) [22:18] <wm4> how often does it save memcpies? only case I can think of is a filter that adds black bars (vf_expand and vf_ass in mplayer, dunno about lavfi) [22:18] <saste> uglyoldbob, feature request is welcome, patch is even more welcome [22:18] <wm4> for all other cases, handing down the buffer works well enough [22:18] <saste> uglyoldbob, problem is that ffserver these days is almost unmaintained [22:19] <saste> wm4, padding, cropping, in general hepls whenever you have in-place processing [22:21] <wm4> cropping and in-place can be handled by having the ability to pass down mutable buffers [22:22] <saste> wm4, yes, and talking about design is much simpler than implementing it [22:23] <wm4> well, I did this on the mplayer filter chain (in an experimental branch)... [22:23] <wm4> and the concept is quite simple, isn't it [22:24] <saste> wm4, "mutable buffers", how is that different from the current design? [22:24] <saste> check doc/filter-design.txt [22:25] <durandal_1707> Compn: you have more mvi2 samples? [22:25] <saste> which reminds me that we have to update that... [22:26] Action: ubitux wonders why the "avfilter_" was added all the time [22:31] <Compn> durandal_1707 : theres bunch on ebay , search for "movie cd" :) [22:31] <Compn> i'll have to dig, i may have another sample cd [22:35] <durandal_1707> Compn: usually i do not work on something i have single sample [22:44] <cone-797> ffmpeg.git 03Anton Khirnov 07bb56436cfda8: f_settb: switch to filter_frame [22:44] <cone-797> ffmpeg.git 03Anton Khirnov 07dabb993c44cf: f_setpts: switch to filter_frame [22:44] <cone-797> ffmpeg.git 03Anton Khirnov 07eb7e7e82e17b: vf_hflip: switch to filter_frame [22:44] <cone-797> ffmpeg.git 03Anton Khirnov 07bff576c77947: vf_lut: switch to filter_frame [22:51] <ubitux> michaelni: http://lists.freedesktop.org/archives/pixman/2012-November/002353.html [22:51] <ubitux> you might want to look if there are some interesting things not in sws [23:19] <cone-797> ffmpeg.git 03Clément BSsch 072d9d4440519f: lavfi: convert remaining input/output list compound literals to named objects. [23:20] <ubitux> here we go. [23:23] <burek> ubitux, any screenshots so i can see what is wrong? [23:25] <ubitux> burek: scroll to the bottom of the page, and try to hover in the latest 10-20 lines [23:26] <ubitux> (on the right, with the popup) [23:27] <burek> oh i see, scrolling [23:27] <burek> again, some javascript params are different across browsers [23:27] <burek> ill fix it right away [23:30] <burek> ubitux should be working better now, i guess [23:32] <ubitux> looks fine [23:32] <ubitux> saste: got a strange bug, interested? :) [23:33] <ubitux> i'm adding the priv_class where missing the in AVFilter's filters [23:33] <ubitux> and it seems that with mptestsrc, the options don't appear in the -help full [23:34] <ubitux> there is just "mptestsrc AVOptions:" with nothing below [23:39] <ubitux> it works for the other (ebur128, delogo, edgedetect, mandelbrot), but not this one [23:39] <ubitux> there is another problem with sendcmd, but it's something else [23:39] Action: ubitux doesn't get it [23:42] <saste> ubitux: what's the prob with sendcmd? [23:43] <ubitux> i got an infinite loop [23:43] <ubitux> but it's likely it was because i specified it twice [23:43] <ubitux> but i removed it for now [23:43] <ubitux> i try to understand the problem with mptestsrc [23:43] <ubitux> and& i just don't get it [23:43] <ubitux> :D [23:43] <saste> ubitux, commandline? [23:43] <ubitux> ffmpeg -help full [23:43] <ubitux> and look for mptestsrc [23:44] <ubitux> after adding: [23:44] <ubitux> + .priv_class = &mptestsrc_class, [23:44] <ubitux> in libavfilter/vsrc_mptestsrc.c [23:46] <saste> ubitux: no flags in options [23:46] <ubitux> ah! [23:46] <ubitux> thanks :) [23:46] <saste> i can fix it if you want [23:47] <saste> what about the sendcmd issue (I was asking about that before) [23:47] <ubitux> 23:43:05 <@ubitux> i got an infinite loop [23:47] <ubitux> 23:43:20 <@ubitux> but it's likely it was because i specified it twice [23:47] <ubitux> 23:43:25 <@ubitux> but i removed it for now [23:47] <ubitux> doing something similar [23:47] <ubitux> .priv_class = &sendcmd_class, for both [23:48] <ubitux> (asendcmd and sendcmd) [23:48] <ubitux> and same command [23:48] <saste> yes, you can't specify the same class for two instances [23:48] <saste> it's silly, but it's true [23:49] <saste> see the trick in vf_lut.c [23:50] <ubitux> hehe ok i see :) [23:50] <ubitux> thanks [23:50] <cone-797> ffmpeg.git 03Anton Khirnov 079a65b8c0a221: vf_fade: switch to filter_frame [23:50] <cone-797> ffmpeg.git 03Anton Khirnov 075f648ce43eeb: vf_yadif: switch to filter_frame, this filter did not support slices [23:51] <saste> ubitux: about the yadif ticket, didn't you report a similar issue? [23:52] <ubitux> mmh [23:52] <ubitux> i remember a ticket with yadif + timecode [23:52] <ubitux> but nothing more [23:53] <ubitux> https://ffmpeg.org/trac/ffmpeg/ticket/1469 ? [23:54] <cone-797> ffmpeg.git 03Stefano Sabatini 075ddc072d0ddc: doc: add ffmpeg-bitstream-filters.texi file [23:55] <saste> ah ok [00:00] --- Thu Nov 29 2012

1 0

ffmpeg.log.20121128
by burek 29 Nov '12

29 Nov '12

[00:14] <bakers> I have a video with two audio streams, and a subtitle track. I want to strip out everything except the video and the first audio track [00:14] <bakers> I'm trying to do -map 0:0 -map 0:1 but I get errors [00:14] <bakers> http://www.fpaste.org/H8zR/ [00:14] <bakers> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [00:15] <JEEB> libvo_aacenc doesn't support >2ch afaik, and is a crappy encoder [00:15] <bakers> Oh the source is 5.1 [00:15] <bakers> I didn't even notice [00:15] <JEEB> it seems to have picked the right tracks tho [00:15] <bakers> what's the "best" aac encoder then [00:15] <JEEB> fdk-aac is the best you can link to ffmpeg [00:16] <JEEB> (too bad you can't distro it because of fraunhofer being dumb with their license) [00:16] <JEEB> (they ended up with something very close to GPL, but then added a line about you not being able to take money from the piece of software, which GPL lets you do -> nonfree) [00:16] <JEEB> but the input is already AAC it seems [00:17] <JEEB> you could just do -c:a copy ? [00:17] <bakers> true [00:17] <bakers> ok [00:17] <bakers> I didn't notice the 5.1 [00:17] <JEEB> heck, you could do c:v copy and c:a copy [00:17] <JEEB> because you're not setting any settings :) [00:17] <bakers> That was just a test to see if I mapping was correct [00:17] <bakers> I need to make this smaller [00:17] <JEEB> k [00:17] <bakers> is -acodec aac [00:17] <bakers> bad? [00:17] <bakers> it's still "Experimental" [00:18] <JEEB> yes, it's better than vo_aacenc but still needs at least ~192kbps or so [00:18] <JEEB> better to just install fdk-aac if you need AAC [00:18] <JEEB> another alternative is to build aotuv's vorbis [00:18] <bakers> I'm running a "static" version that doesn't seem to have that [00:19] <JEEB> well fdk-aac can't be redistributed [00:19] <JEEB> as it's "nonfree" [00:19] <bakers> k [00:19] <JEEB> aotuv's vorbis can be, but I'm not sure if the guy who builds the static linux builds with aotuv or official [00:20] <JEEB> official is still somewhat worse off than aotuv [00:21] <bakers> This is x264 so that's also non-free [00:21] <JEEB> no, x264 is GPL [00:21] <bakers> oh ya? [00:21] <JEEB> yes [00:21] <JEEB> really [00:22] <JEEB> you /can/ license x264 commercially if you wish, but by default it is GPL. Nonfree means something is not GPL-compatible. [00:22] <JEEB> (or LGPL) [00:26] <bakers> what's the status on libfaac? [00:26] <JEEB> that was found to be using the reference source code [00:27] <JEEB> -> nonfree [00:27] <JEEB> same for libaacplus [00:27] <bakers> are there any "free" aac encoders? [00:27] <JEEB> vo_aacenc is based on other reference code, but properly licensed as Apache v2 [00:27] <JEEB> but it's crappier than the libavcodec ffaac one [00:27] <JEEB> (aka -c:a aac) [00:28] <JEEB> fdk-aac is actually is open source and properly licensed, but the license is not GPL compatible -> nonfree in ffmpeg [00:28] <JEEB> which is a real shame because unlike the other nonfree things, it actually is properly licensed :< [00:28] <bakers> What exactly does -strict experimental do [00:29] <bakers> just says use these "not fully tested things" [00:29] <JEEB> it just enables stuff that is labled "experimental" [00:29] <JEEB> the internal AAC encoder is OK if you use bit rates of 192kbps+ [00:29] <bakers> and the -acodec aac is still experimental? [00:29] <bakers> ah ok [00:29] <JEEB> yes [00:29] <bakers> got it [00:29] <JEEB> it would be better if someone finished it :D [00:29] <JEEB> too bad no-one did [00:30] <JEEB> it currently has features turned off in the code because they just don't work [00:30] <bakers> Awesome, thanks for the explanation [00:30] <JEEB> the case of fdk-aac is really, really sad tho :< [00:31] <JEEB> brought in by android just like vo_aacenc [00:31] <JEEB> -> "lol we made the license not compatible with GPL" [00:31] <JEEB> of course if you compile yourself that's not a problem [00:31] <JEEB> or possibly if you only compile LGPL things, but I think ffmpeg doesn't even let you build it with fdk even in that case :) [00:31] <JEEB> --enable-nonfree is global [00:39] <bakers> What was brought in by android? [00:39] <bakers> fdk-aac and what? [00:40] <JEEB> vo_aacenc [00:40] <chrisballinger> we are having trouble using the built in aac encoder because it expects floats [00:41] <JEEB> well it isn't exactly good for low bitrate targets either [00:41] <chrisballinger> we are going for 64kbps mono [00:42] <JEEB> well, everything good is pretty much nonfree so you'll probably just have to talk on the -devel channel and see if you can get someone looking over you making a non-float version of/fixing the internal [00:42] <JEEB> unless you don't distribute any binaries :) [00:43] <chrisballinger> :( [00:43] <chrisballinger> it used to accept floats [00:43] <chrisballinger> er i mean ints [00:43] <chrisballinger> 16-bit [00:43] <JEEB> you should be able to convert within ffmpeg's libraries [00:44] <JEEB> I thought you meant you can't have float math [00:44] <JEEB> lol [00:44] <JEEB> libavresample or libswresample [00:44] <JEEB> one of these [00:44] <chrisballinger> we are getting a crash using swr_convert [00:44] <ubitux> how to reproduce? [00:44] <JEEB> get a backtrace and poke -devel / the trac bug tracker [00:45] <chrisballinger> this is JNI so it's hard to debug / get a trace [00:45] <bakers> man that "aac" encoder is crap... [00:45] <bakers> even at 128k [00:45] <JEEB> yes [00:45] <JEEB> which is why I said 192k [00:45] <JEEB> + [00:45] <bakers> I know I know [00:45] <bakers> This is only for testing [00:45] <bakers> and even for testing it's crap [00:46] <chrisballinger> why can't we distribute binaries that use fdk-aac again? [00:46] <JEEB> because it's not GPL-compatible [00:46] <chrisballinger> not even LGPL compatible? [00:46] <bakers> any chance on getting that changed? [00:46] <JEEB> it could be LGPL compatible, but the nonfree limitation is global [00:46] <JEEB> so derp [00:47] <JEEB> bakers, find someone who wants to read the spec and is interested in psychoacoustics [00:47] <chrisballinger> That seems rather silly [00:47] <JEEB> well ask someone related to the project about that... [00:48] <JEEB> might either get an answer that it's not OK, or that it's too hard to implement in the current configure way, or that you'll just get it within some hours [00:48] <JEEB> bakers, and someone who wants to work on the encoder [00:48] <JEEB> it's bad because a lot of features are turned off as they don't work f.ex. [00:48] <JEEB> (it's better than the WMA2 encoder tho, lol) [00:48] <JEEB> also did you downmix to 2ch? [00:49] <JEEB> 128k and 5.1 is WHOOPS [00:49] <bakers> libvo_aacenc is decent [00:49] <bakers> just can't convert 5.1 to 2.0 [00:50] <JEEB> ...so you didn't even downmix when testing the internal one? [00:50] <JEEB> because IIRC in a Japanese test the internal one was pretty much on the level of the vo_aacenc with 2ch [00:50] <JEEB> not sure what bit rate it was, but it really sounds that you didn't downmix [00:50] <bakers> how do I tell it to downmix? [00:50] <JEEB> ... [00:51] <bakers> I probably didn't [00:51] <bakers> -ac 2? [00:51] <bakers> I'll try that [00:52] <JEEB> yeah, that is the old way IIRC, not sure if there's a new way to call the swresample based audio thingy [00:52] <JEEB> the internals probably work on the same swresample base [00:52] <bakers> Or I could just a clip that's not 5.1 to start with :) [00:53] <JEEB> anyways, yeah -- -ac 2 makes sure you only have 2 channels in the output [00:53] <JEEB> yeah, seems like the correct way still :) [00:57] <bakers> Oh hey I meant to ask [00:57] <bakers> is ffprobe -print_format json supposed to output anything [00:57] <JEEB> never used that, so no idea :3 [00:57] <ubitux> along with -show_streams/-show_format/etc yes [00:58] <joecool> seems like ffmpeg in the past 2 days broke spek, any major changes that would cause it to throw "unsupported file format"? [00:58] <ubitux> what's spek? [00:58] <bakers> ubitux: when I run that I get the standard ffmpeg output to stderr and { } to stdout [00:58] <joecool> spectral analyser [00:58] <ubitux> bakers: what command? [00:58] <bakers> ubitux: http://www.fpaste.org/ze77/ [00:59] <ubitux> 00:57:55 <@ubitux> along with -show_streams/-show_format/etc yes [00:59] <bakers> OH! [00:59] <bakers> Got it [00:59] <bakers> I expected there to be SOME default [00:59] <ubitux> joecool: i don't know what you are talking about [01:00] <ubitux> please provide a sample/command line/... [01:05] <joecool> ubitux: it's graphical, https://xwaretech.info/spek.jpg [01:05] <joecool> normally that would generate a spectrogram, it doesn't [01:06] <llogan> JEEB: were you referring to this? http://d.hatena.ne.jp/kamedo2/20120729 [01:06] <llogan> i never did try with -cutoff [01:06] <JEEB> llogan, yeah [01:08] <ubitux> joecool: i don't know how that work [01:09] <ubitux> it can be a badly compiled ffmpeg, using libs or the tool, can even not be ffmpeg, etc [01:10] <joecool> ubitux: it does not bundle ffmpeg, it was working fine until about 2 days ago, i build ffmpeg nightly [01:11] <JEEB> if you build it, you could start going backwards in revisions [01:11] <JEEB> and see which one was the one that borked something for you [01:11] <joecool> i can bisect it, but it takes time [01:11] <joecool> i was hoping someone would say "oh yeah we did this, they need to fix their shit" or something to that effect [01:11] <ubitux> use ccache, build only the minimum, bisect with 2 days [01:11] <ubitux> should be pretty fast [01:12] <JEEB> yup, but since no-one else has an idea how to replicate it or what's the problem, you can't really get an answer like that :P [01:12] <JEEB> so bisect it is [01:12] <ubitux> if you can reproduce with ffmpeg directly that could help [01:15] <funyun> hi. can ffmpeg repair a corrupt .mov file? [01:16] <JEEB> if you have no index in it, nope.avi [01:17] <funyun> JEEB: how can i check? [01:18] <JEEB> if you do ffmpeg -i derp.mov and it tells you it can't find something, that means it can't be automatically opened with a generic demuxer [01:19] <funyun> JEEB: http://pastebin.com/0ir23Mez [01:20] <JEEB> well, at least it seems to have an index [01:20] <JEEB> also man that is old [01:20] <JEEB> build a newer one [01:20] <funyun> JEEB: you mean ffmpeg? [01:20] <JEEB> also when did clang get to 4.0 o_O [01:20] <JEEB> funyun, yes [01:21] <Plorkyeran> apple clang [01:21] <JEEB> oh [01:21] <Plorkyeran> completely different set of version numbers from normal clang [01:21] <JEEB> they have a separate versioning system? [01:21] <JEEB> ah [01:21] <JEEB> funyun, ok your problem wasn't the one most people have with mov/mp4 etc. files [01:21] <Plorkyeran> there was a really confusing point where they almost but didn't quite line up [01:21] <JEEB> what's the problem, or is it that other timestamp funnyness? [01:22] <funyun> JEEB: it plays fine in quicktime. but it won't play in vlc or mplayer and most importantly, i can't edit the video in final cut pro [01:23] <JEEB> first build current ffmpeg from the git repo, secondly try ffmpeg -i input.mov -map 0 -c:a copy -c:v copy out.mp4 or so [01:23] <JEEB> and see if it can copy the streams [01:23] <JEEB> although that might or might not affect this if this is the problem "Seems stream 0 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 24.00 (24/1)" [01:24] <JEEB> (the video track and the container have different maximum timescales) [01:24] <JEEB> aanyways, I really need sleep now o/. [01:24] <funyun> alright. thanks anyway [01:26] <cbsrobot> funyun: it could also be the audio ... [01:26] <funyun> cbsrobot: in fcp, the video is corrupted [01:26] <funyun> i get a green screen [01:26] <cbsrobot> ah ok [01:27] <funyun> even tho when i play that same time with quicktime, it plays fine [01:28] <funyun> a little back story if it might help. i recorded this video on my iphone and it recorded until the phone ran out of battery [01:28] <funyun> so maybe it just didn't save correctly? [01:30] <cbsrobot> funyun: maybe it wont play it because of vvs [01:30] <funyun> vvs? [01:30] <cbsrobot> http://www.youtube.com/watch?v=Bt9zSfinwFA [01:32] <funyun> cbsrobot: lol [01:32] <funyun> i hate vvs as well [01:32] <funyun> so that's not the problem [01:33] <cbsrobot> ok - lucky you [01:33] <cbsrobot> try what jeeb said [02:16] <joules> hi, what is the ffmpeg equivalent to mp=mcdeint:0:0:10 and mp=pp:hb/vb/dr ? [02:18] <joules> theres no mcdeint so i can forget about that... [03:06] <joules> hi, i'm trying to seek to a position in this vob but only -sb works -ss gives me EOF @ 100secs [03:08] <joules> damn -ss is not working for me. any suggestions? [04:30] <joecool> ubitux: working on bisecting it now, i'm at this commit and it works https://github.com/FFmpeg/FFmpeg/commit/a13148f63351db5f8283a23de9a22e940d2… [05:31] <joecool> ok i think i got it [05:32] <joecool> https://github.com/FFmpeg/FFmpeg/commit/59b68ee8870ad7b831321ff39638660ac59… << this commit breaks spek (just won't show graphs at all, claims unsupported format on anything) [05:32] <joecool> https://github.com/FFmpeg/FFmpeg/commit/a13148f63351db5f8283a23de9a22e940d2… << this commit works just fine, last working commit [05:40] <joecool> actually that commit half-works, still going back to see when it 100% works [05:55] <joecool> https://github.com/FFmpeg/FFmpeg/commit/86270236d5ef97263858a576fab2c366fb2… << fully works on this [07:38] <JEEB> joecool, if it starts failing on the commit where certain decoders moved to planar audio input (multiple pointers and multiple buffers instead of everything being interleaved into a single buffer) then whatever is utilizing it probably still tries to get interleaved? [07:39] <JEEB> which it can't [07:39] <JEEB> if you use the libraries you would have to use libswresample or libavresample to interleave it to the format you need [07:40] <joecool> JEEB: dunno enough about it, it generates visual spectrographs, not sure how it does it [07:41] <JEEB> it probably assumes that everything outputs interleaved [07:41] <JEEB> which is how it was before that change [07:41] <JEEB> now I think all audio decoders output planar [07:42] <JEEB> and if you don't like planar audio, you have to interleave it with, say, libswresample or libavresample [07:42] <JEEB> that's it [07:43] <JEEB> and "half-works" probably means that some formats were planar, some interleaved at that point [07:44] <JEEB> so you're pretty much getting <joecool> i was hoping someone would say "oh yeah we did this, they need to fix their shit" or something to that effect [07:45] <joecool> JEEB: cool, i'll see if i can get in contact with the developer [07:45] <joecool> thanks [07:45] <JEEB> np [10:39] <bizulk> Hi ! I would like to port my app on the new ffmpeg API. But I do not understand the options params with avcodec_open2. May I put NULL for options, if no how can I fill it before calling options. [10:42] <JEEB> http://ffmpeg.org/doxygen/trunk/group__lavc__core.html#g11f785a188d7d9df716… [10:43] <JEEB> even has an example [10:43] <JEEB> :) [10:46] <bizulk> Yes thanks. But on this exemple we set arbitrary options. But if there a not particular options ? [10:46] <JEEB> the decoding example just passes NULL to that just fine [10:46] <JEEB> it seems [10:46] <JEEB> http://ffmpeg.org/doxygen/trunk/doc_2examples_2decoding__encoding_8c-exampl… [10:49] <bizulk> that's what I expected. I shall take some time to read these relevant short example. ffplay confuses me. [10:50] <bizulk> JEEB: thks [10:50] <JEEB> ffplay is a "barely working example" :) for example, it fails if the video is not 4:2:0 YCbCr [11:13] <Mavrik> hmm, why does filtering with avfilter change PTS of my frames? [12:32] <JEEBsv> Mavrik: you might want to detail on that some more :D [12:32] <JEEBsv> which filter, how etc. [12:32] <Mavrik> yeaah, basically I'm doing yadif,scale on video and aresample,aconvert on audio [12:33] <Mavrik> and I'm seeing the PTS change on both accounts to... something [12:33] <Mavrik> and I can't seem to find any documentation/mention of filters adjusting PTS of frames [12:33] <Mavrik> which (obviously) causes me some sync problems [12:34] <Mavrik> I can gladly provide more information... but I have to know what :D [12:34] <JEEBsv> well, yadif can change the pts naturally, as it can bob [12:34] <JEEBsv> scale shouldn't [12:34] <JEEBsv> audio naturally could change PTS with resampling [12:34] <Mavrik> mhm [12:34] <Mavrik> [2012-11-28 12:34:28,735][TRACE][../src/transcode/streams/VideoStreamProcessor.cpp:219] [VideoStreamProcessor] - Frame [720x576] Timestamp 7779355407 [12:35] <Mavrik> [2012-11-28 12:34:28,737][TRACE][../src/transcode/streams/VideoStreamProcessor.cpp:237] [VideoStreamProcessor] - [F] Frame [640x360] Timestamp 15558703614 [12:35] <Mavrik> this is what happens - the change is pretty big (not just adjustment for deinterlace, resample) [12:35] <JEEBsv> make sure you give out the timescale too [12:35] <Mavrik> a. [12:35] <JEEBsv> PTS is a x/y value after all [12:35] <Mavrik> I only set time_base on the buffersrc, not src [12:35] <Mavrik> *not sink [12:36] <Mavrik> this is how I setup the filter chain: http://pastebin.com/dxVPe3j1 [12:36] <JEEBsv> I have absolutely no idea what you're doing and what you're not doing, but I would check what timescales you are getting on all sides, there are also functions to then match PTS to a certain timescale [12:37] <Mavrik> JEEBsv: as far as I know the frame object have no timescale property attached, only PTS. Where can I check that? [12:38] <Mavrik> as I said, I have no idea how timing is handled within libavfilter and I can't seem to find any documentation on that [12:39] <JEEBsv> and I have only used libavcodec from the inside, aka implemented an encoder. I only know limited amount of stuff on the outside APIs [12:40] <JEEBsv> all I can say is check your timescale and that you're handling both the X and the Y of your PTS correctly [12:40] <Mavrik> mhm, I'll go bug on libav-users or something [12:40] <JEEBsv> and not taking an X value and putting it into another timescale [12:40] <JEEBsv> which naturally will end up with a/v desynch [12:41] <Mavrik> JEEBsv: yeah, I'm handling those correctly outside the filter chain, but as I said, my PTS gets chained while frames are being filtered and I can't seem to find the timebase to which it gets converted [12:41] <JEEBsv> because that's what it sounds like [12:41] <Mavrik> *changed [12:41] <Mavrik> anyway, thanks :) [12:41] <Mavrik> I'm gonna continue to go through the source to find out what the heck is going on [12:50] <Mavrik> ok got it, I need to rescale timebase from output buffer sink filter's timebase to the codec's timebase [13:30] <brontosaurusrex> any clues what quantel is using as a master codec? [13:30] <brontosaurusrex> is that some i-frames type of system? [13:30] <brontosaurusrex> i know, iam offtopic [14:03] <samek_> Hi, can anyone help me debug why rtmp streaming is not stable ? It stalls after sometime or it issues FCunpublish. Source of the video is Live rtp multicats. [14:33] <samek_> Is it possible to debug only output RTPM packet communication ? [14:34] <samek_> I have a problem with streaming from rtp multicast source to rtmp. ffmpeg connection normaly just hangs or it issues FCunpublish. I have a log with debug but's it's 400+Mb [16:05] <samek_> Hi, can anyone help me debug rtmp publishing from an rtp multicast source ? [16:06] <samek_> It stops after a a certain time (ffmpeg "hangs") [16:58] <^BB^> question: i have an MKV file with a video and audio stream. the audio stream must have a delay, because when i demux the streams and remake an MKV without audio delay, the audio is out of synch (it runs about 3 seconds behind the video). so, how do i found out the audio delay? ffmpeg -i file.mkv doesn't show any delay.. :s [16:59] <^BB^> mediainfo will say it's at 1mn 5sc, which to me sounds like a 655xx value (signed bit problem?) [17:00] <^BB^> however, any media player will play it correctly, so there must be another way to obain the correct delay value [17:02] <JEEB> <Mavrik> ok got it, I need to rescale timebase from output buffer sink filter's timebase to the codec's timebase <- called it :) [17:02] <Mavrik> JEEB: yeah :D [17:02] <Mavrik> had to read through ffplay.c to find out where time_base is stored [18:46] <samek__> anyone here ever had ffmpeg hand during rtmp streaming ? [18:47] <samek__> hand = hang [18:51] <Laputa> hi, i'm wondering if ffmpeg support mp4 container ? i wanna wrap h.264 video and aac audio to a mp4 container. [18:51] <sacarasc> Laputa: Yes. [18:54] <Laputa> sacarasc: any url for that ? i converted some video to mp4, but the flash player could read the video's info untill the video is all loaded. [18:54] <Laputa> sacarasc: i used to use mp4box to wrap the video again, i wonder if i can do this in ffmpeg [18:55] <sacarasc> You'd have to run qtfaststart on the video, which comes with the ffmpeg source. [19:03] <samek__> could someone suggest me on how to debug the rtmp session which hangs when streaming to FMS server. http://pastebin.com/jXzExPit [19:25] <Laputa> sacarasc: thanks . i've tried that , and works great ! [20:06] <PMacDiggity> Hello, how can I identify in MacOS 10.8 what the correct name for an input device is for video capture (Thunderbolt BlackMagic Deign UltraStudio Express) for use with the '-i' flag? [21:49] <zuloyd> hihi [21:49] <zuloyd> (just once, sorry) [21:50] <zuloyd> can someone please tell me what's the problem here? http://pastebin.com/Xfz7PPwV [21:50] <zuloyd> i'm trying to generate a thumbnail from a video [21:51] <zuloyd> is it because it's too short? [21:54] <llogan> zuloyd: input duration is 3.9 seconds. your -ss is 4 seconds. [21:54] <zuloyd> is it possible to extract a random frame? [21:55] <zuloyd> or at least something from the middle of the film [21:55] <zuloyd> i don't want it to be the first frame in every movie [21:55] <zuloyd> that is, without knowing the duration [22:02] <llogan> zuloyd: ignoring any scripting you might be able to also do it with the select filter, but i don't know [22:02] <llogan> however your ffmpeg is too old [22:03] <zuloyd> I know, but I have no chance of updating it since it's a managed server [22:03] <zuloyd> and the hoster is very conservative [22:03] <llogan> you can use a static build [22:03] <llogan> http://ffmpeg.org/download.html#LinuxBuilds [22:04] <zuloyd> okay, a related but different questions: Can I somehow use static libraries? [22:04] <zuloyd> i.e. codecs [22:04] <llogan> sorry, i don't understand the question [22:04] <zuloyd> I need a video codec that's not installed on the system and I can't install it since I don't have root access [22:05] <zuloyd> but is there a way to get this codec from somewhere, perhaps place it in the same directory as the static ffmpeg build and then use it? [22:05] <llogan> use it with ffmpeg? [22:05] <zuloyd> yes [22:05] <llogan> it doesn't work like that [22:06] <llogan> what format are you trying to encode or decode that you can't now? [22:06] <zuloyd> trying to encode WebM [22:06] <llogan> (better links for static builds) [22:08] <llogan> the static builds both support libvorbis and libvpx which are the audio and video formats used in webm container [22:12] Action: llogan needs to make a non root/su version of the ubuntu compile guide [22:16] <llogan> ...and i should say "encoders" instead of "formats" [22:41] <giany> hello [22:42] <giany> anyone is able to play properly mp4->webm files on firefox browser? [22:51] <bakers> Is there way to make ffprobe NOT output the compile information to stderr? [22:52] <saste> bakers, 2&> /dev/null? [22:52] <bakers> saste: Other than that :) [22:52] <bakers> I think -loglevel quiet might work [22:52] <saste> and does it work? [22:57] <bakers> Testing now [22:57] <bakers> Does apple trailers use ffmpeg to encode? [22:57] <bakers> I see lavf in the encoding header section, isn't that ffmpeg [22:59] <sacarasc> I would presume they use the QuickTime encoders... [23:03] <bakers> If I'm joe-bob-tube... some small youtube startup, and I want to use (legally) ffmpeg to encode x264 and aac audio what would I need to license [23:03] <bakers> and from whom? [23:06] <burek> bakers, license applies to (re)distribution afaik [23:06] <burek> not usage [23:06] <bakers> so I can encode all I want [23:06] <bakers> but if I have actual visitors is when you have to pay [23:06] <burek> as long as nobody arrests you :) [23:07] <burek> hm [23:07] <burek> not that [23:07] <burek> http://ffmpeg.org/legal.html [23:07] <bakers> who wrote libx264 [23:07] <bakers> where did that come from? [23:08] <burek> videolan [23:08] <bakers> the VLC guys? [23:09] <bakers> libx264 is a pretty solid encoder I'm impressed... considering we can't get a solid GPL compatible AAC encoder [23:13] <samek__> could someone suggest me on how to debug the rtmp session which hangs when streaming to FMS server. http://pastebin.com/jXzExPit [23:13] <samek__> I tried even with static build posted here on this channel moments ago .. it hangs. [23:17] <burek> why do you use -re with live input [23:18] <samek__> burek i've tried everything until now .. [23:18] <samek__> with and without -re [23:18] <samek__> I can create another without -re it will hang [23:19] <burek> your "Stream mapping:" says you have 3 video + 3 audio encodings for the live stream [23:19] <burek> are you sure your cpu/ram can handle all that in real time? [23:20] <samek__> It should [23:21] <samek__> dual Intel(R) Xeon(R) CPU X5550 @ 2.67GHz [23:21] <burek> did you check your cpu usage during your tests [23:21] <burek> also -threads 0 is advised [23:22] <samek__> i'm running it right now [23:23] <samek__> top shows .. Cpu(s): 9.2%us, 0.3%sy, 36.2%ni, 54.0%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st [23:23] <burek> next thing, try removing rtmp outputs and use file outputs [23:24] <burek> to see if it is the encoder(s) issue [23:24] <samek__> ok I'll do it. [23:27] <samek__> What if it's working on files ? how can we debug the rtmp session ? [23:27] <burek> if it does [23:28] <burek> then either rtmp muxer is buggy [23:28] <burek> or your network is unstable [23:28] <burek> or somebody cursed you :D [23:29] <samek__> Since i belive i've tried with recorded file couple of days ago and it didn't go off on the same timestamp (when sending to rtmp) [23:30] <samek__> so it's the other way arround what I'm doing right now not streaming to rtmp [23:32] <samek__> It's still writing to files .. but than again somethimes it streams also for several minutes. [23:36] <samek__> burek may i ask where are you from cause of the nick - It's a food. [23:36] <samek__> I'm from slovenia btw. [23:42] <burek> serbia :) [23:42] <samek__> Cool [23:42] <burek> :beer: :) [23:44] <samek__> burek i guess it's not the encoder since it's still running. How could i home in where the problem is network or muxer then ? [23:45] <burek> hmh... try another test.. save your input in a file (1-2 minutes long) and then use that file instead of live input and use your rtmp outputs [23:46] <burek> (don't forget -re) [23:46] <burek> to see if the sync between input/outputs causes issues [23:46] <samek__> Ok [00:00] --- Thu Nov 29 2012

1 0

ffmpeg-devel.log.20121127
by burek 28 Nov '12

28 Nov '12

[02:44] <durandal_1707> lol i found what BRSTM stands for by accident [02:51] <llogan> durandal_1707: dispell my ignorance [02:52] <tg2> hey [02:52] <llogan> hej [02:52] <tg2> is there any easy way to get ffprobe to output only I frames? [02:52] <tg2> without parsing through grep [02:54] <tg2> --show-frames scans for and shows all frames, audio and video, however I only need pict_type I frames [02:55] <tg2> just modify show_frame in ffprobe.c? [02:55] <tg2> or is there a more efficient way [02:58] <durandal_1707> llogan: what is ^ [02:59] <llogan> durandal_1707: an article about frameserving from premiere to ffmpeg because Adobe Media Encoder is butt. [02:59] <llogan> also, you never told me what BRSTM is and I'm too lazy to search [02:59] <durandal_1707> Binary Revolution Stream [02:59] <llogan> ah [03:00] <llogan> ...but first i will have to motivate myself to use the windows partition. [03:01] <tg2> adobe dilemma [03:02] <llogan> and i already found a typo [03:59] <cone-819> ffmpeg.git 03Michael Niedermayer 07407921072f6d: file: fix file_check() [05:08] <cone-819> ffmpeg.git 03Michael Niedermayer 07465becbc4f1f: configure: check for access() [05:08] <cone-819> ffmpeg.git 03Michael Niedermayer 0716ef48c37fad: file: Fallback to stat() based file_check() when access() or its named flags are unavailable. [05:08] <cone-819> ffmpeg.git 03Michael Niedermayer 078aa6d899c01f: cafdec: fix seeking with adpcm [08:28] <burek> does this look like an acceptable layout for the fate html page: http://ffmpeg.gusari.org/fate/ [08:29] <burek> (try also hovering the mouse over maintainers) :) [08:29] <burek> btw, it's pure html + css, without javascript :) [08:57] <cbsrobot> burek: I like the avatar of barabra streisand [08:57] <cbsrobot> where does she live again ? [09:01] <burek> :))) [09:01] <burek> somewhere.. [09:01] <burek> over the rainbow i guess :) [09:44] <llogan> burek: looks pretty good to me. "ffmpeg green" might be nice though but that's an ignore-a-nit [10:48] <cone-677> ffmpeg.git 03Mark Himsley 0713c625253625: lavfi/overlay: enable yuva420p as overlay background [11:18] <cone-677> ffmpeg.git 03Paul B Mahol 0710c8f913410e: astdec: improve probing [11:23] <cone-677> ffmpeg.git 03Paul B Mahol 0751d6879939bf: AFC demuxer [13:09] <michaelni> burek, looks cool, it lost the ability to sort by clicking on the column header though [13:27] <michaelni> burek, also current fate server shows a little black triangle on the main page with every client that has failures, clicking on that reveals more info about the failures [13:36] <cone-677> ffmpeg.git 03Michael Niedermayer 07257196209fe7: file: fix 10l error in access() check [13:47] <cone-677> ffmpeg.git 03Mans Rullgard 07dd3b73f3905c: base64: fix signed overflow in shift [13:47] <cone-677> ffmpeg.git 03Mans Rullgard 07edd80ec7e32b: aacdec: fix signed overflows in lcg_random() [13:47] <cone-677> ffmpeg.git 03Michael Niedermayer 071e276553886a: aacenc: use the correct output buffer [13:47] <cone-677> ffmpeg.git 03Justin Ruggles 07284ea790d894: dsputil: move vector_fmul_scalar() to AVFloatDSPContext in libavutil [13:47] <cone-677> ffmpeg.git 03Michael Niedermayer 072684d2e3ea8a: Merge commit '284ea790d89441fa1e6b2d72d3c1ed6d61972f0b' [14:14] <cone-677> ffmpeg.git 03Justin Ruggles 07947f933687b9: x86: float_dsp: add SSE version of vector_fmul_scalar() [14:14] <cone-677> ffmpeg.git 03Diego Biurrun 07d89f2fa970ed: configure: Add separate list for libraries and use where appropriate [14:14] <cone-677> ffmpeg.git 03Justin Ruggles 073c370f5abc55: riff: only warn on a bad INFO chunk code size instead of failing [14:14] <cone-677> ffmpeg.git 03Michael Niedermayer 07b4d4e51027e0: Merge commit '3c370f5abc55739a261534b9f9bdc739cedbbbb9' [14:36] <cone-677> ffmpeg.git 03Justin Ruggles 07259d8c564741: riff: do not add empty metadata tags in INFO chunk [14:36] <cone-677> ffmpeg.git 03Michael Niedermayer 072fe0094e0bf9: wavenc: write fact chunk sample count at the correct file position [14:36] <cone-677> ffmpeg.git 03Michael Niedermayer 075d47850bbd9e: dcadec: skip QMF on unused channels [14:36] <cone-677> ffmpeg.git 03Janne Grunau 07150b2361ca08: h264: add missing new line to log message [14:36] <cone-677> ffmpeg.git 03Janne Grunau 076e5cdf262819: h264: check ref_count validity for num_ref_idx_active_override_flag [14:36] <cone-677> ffmpeg.git 03Michael Niedermayer 07577b39aea221: Merge commit '6e5cdf26281945ddea3aaf5eca4d127791f23ca8' [14:41] <cone-677> ffmpeg.git 03Janne Grunau 07d5e83122bdc3: h264: set Picture.owner2 to the current thread [14:41] <cone-677> ffmpeg.git 03Michael Niedermayer 074063bb212e18: Merge remote-tracking branch 'qatar/master' [15:06] <cbsrobot> burek: No! See http://en.wikipedia.org/wiki/Streisand_effect [16:00] <ubitux> if we want to play streams from different files, i guess we have to keep track of multiple AVFormatContext, right? [16:00] <ubitux> or is there another trick? [16:04] <vdurand> Hi. My question concerns HE-AAC decoding in ffmpeg. It is currently too CPU intensive to work smoothly on fixed-point architectures in embedded devices. The Fraunhofer SDK provides a fixed-point implementation that proves much more efficient. Since ffmpeg 1.0, Fraunhofer is used for aac encoding. Is there any reason to not use it for aac decoding? [16:05] <JEEBsv> you can't distro binaries with it [16:05] <JEEBsv> that's all [16:05] <JEEBsv> I'm not sure about LGPL, but its license is not compatible with GPL (if you, f.ex. have libx264 in there) [16:06] <JEEBsv> in any case, it seems to need --enable-nonfree with both LGPL and GPL [16:06] <JEEBsv> (the only reason why it's not GPL-compatible is because it adds the additional definition that you can't take money for things with it, which GPL doesn't prohibit) [16:08] <vdurand> I see. I'll need to review its license. Do licenses apply differently for encoding and decoding? [16:08] <JEEBsv> no [16:09] <JEEBsv> vdurand: license is over @ https://github.com/mstorsjo/fdk-aac/blob/master/NOTICE [16:09] <vdurand> Then I don't get why ffmpeg uses Fraunhofer for encoding and not decoding [16:09] <cone-677> ffmpeg.git 03Michael Niedermayer 07c2c1726847fe: ffserver remove CONFIG_NOCUTILS check [16:10] <JEEBsv> because no-one wrote a decoder that uses the library? And because the libavcodec AAC decoder in general is in much better standing than the internal encoder [16:12] <vdurand> OK thanks. I'm about to integrate Fraunhofer into ffmpeg for aac decoding, hence my questions. [16:13] <JEEBsv> it's really sad that fraunhofer had to write their own license, which was pretty much GPL, but then not exactly :P [16:13] <JEEBsv> and that "not exactly" then makes it GPL-incompatible because it doesn't limit you in taking money from the software, while the fraunhofer license does [16:57] <Daemon404> nevcairiel, where did you get LAVF_DISPOSITION_SUB_STREAM from [16:57] <Daemon404> MagicNumbers(TM) [16:59] <Daemon404> hmmm only for audio [16:59] <Daemon404> guess its not my problem [17:36] <nevcairiel> Daemon404: thats my own definition :p [17:36] <nevcairiel> i even used a special prefix! [17:37] <Daemon404> i see it [17:38] <nevcairiel> it also just flags the embeded ac3 stream [17:38] <Daemon404> yeah i sae [17:38] <Daemon404> saw [17:38] Action: Daemon404 is adding pid switching [17:38] <nevcairiel> sounds like fun [17:38] <Daemon404> now if only i could figure out why teh 2nd pid plays linearly but crashes [17:38] <Daemon404> :V [17:39] <Daemon404> while seeking [17:39] <JEEBsv> * Daemon404 is adding pid switching <- zomg [17:40] <Daemon404> JEEBsv, for d2vsource [17:40] <Daemon404> dont get your hopes up [17:40] <JEEBsv> :< [17:40] <JEEBsv> looked how marumo's does it? [17:40] <Daemon404> no [17:40] <Daemon404> because there are no comments [17:40] <JEEBsv> k [17:41] <Daemon404> or theyre in jp [18:17] <cone-677> ffmpeg.git 03Michael Niedermayer 0793b89868e139: h264: support invalid annex B in mp4 [20:09] <cone-677> ffmpeg.git 03Paul B Mahol 07761aba6811ba: fate: add ADPCM IMA Dialogic OKI decoder test [20:12] <Daemon404> durandal_1707, did you ever hear back from the tak dev? [20:12] <Daemon404> re: his pascal source [20:12] <durandal_1707> not (yet) [20:13] <Daemon404> might be the lazy way to add coding type 0 support [20:14] <durandal_1707> Daemon404: what happened with your uscaled rgb<- (?) ->gbrp ? [20:15] <Daemon404> its still sitting on the mailing list [20:15] <Daemon404> waiting for me to be unlazy and implement the ugly YUV path [20:15] <Daemon404> which swscale requires [20:15] <Daemon404> (you cant have only an rgb<->rgb path) [20:16] <durandal_1707> does it works for rgb->gbrp if you do not set output for gbrp? [20:17] <durandal_1707> there should be unscaled rgb<->gbrp path but it may be blocked by stupid dependency on yuv -> gbrp path [20:17] <Daemon404> yes [20:17] <Daemon404> it's exactly that [20:17] <Daemon404> there is no yuv->bgrp path [20:17] <Daemon404> my patch only adds rgb->gbrp [20:18] <durandal_1707> but it still should work if output flag is not set for gbrp [20:19] <durandal_1707> michaelni: ^ similar is with rgb48->rgb64 [20:20] <ubitux> (funny patches incoming!) [20:21] <cone-677> ffmpeg.git 03Michael Niedermayer 07de9fc0e9d67e: mjpegdec: make sure ret is set before its used. [20:21] <cone-677> ffmpeg.git 03Michael Niedermayer 071c16483cc0da: mjpegdec: replace returns by goto fail in frame decode. [20:21] <cone-677> ffmpeg.git 03Michael Niedermayer 07541efe4120bd: mjpegdec: reset got_picture on errors, successful return, init and flush. [20:21] <cone-677> ffmpeg.git 03Michael Niedermayer 072cfa6fd025f7: mjpegdec: dont set got_picture to 0 per frame. [20:21] <ubitux> (not yet ready though) [20:25] <durandal_1707> ubitux: ? [20:26] <ubitux> i added the support for text subtitles in ffplay :p [20:26] <ubitux> (muxed & external) [20:26] <ubitux> seems to work, mostly [20:26] <ubitux> i get duplicated events sometimes though [20:34] <durandal_1707> ubitux: l is removing slicify filter [20:34] <ubitux> i saw that [20:35] <ubitux> i have no particular opinion [20:35] <ubitux> the simplification looks appealing like every simplification, and i didn't check the no-performance-benefit assertion [20:35] <ubitux> s/and/but/ [20:37] <ubitux> we will have to update all our filters though [20:38] <wm4> removing slices from lavfi or what? [20:39] <ubitux> wm4: yes [20:40] <wm4> but what about UNCOMPROMISED PERFORMANCE [20:40] <Daemon404> orite [20:40] <Daemon404> that reminds me [20:40] <Daemon404> i remember you guys pimping lavfi because 'it can direct render' [20:40] Action: Daemon404 added direct render support for lavc to his vapoursynth plugin =p [20:41] <ubitux> wm4: as i said, i haven't benchmarked, and i have no opinion on the matter, except that the simplification looks appealing [20:41] <wm4> I wonder why libavcodec can't do buffer management on its own [20:41] <wm4> instead it's messed into cmdutils.c [20:42] <Daemon404> a lot of the 'interesting' logic is in cmdutils or ffmpeg.c [20:42] <Daemon404> News at 11 [20:46] <durandal_1707> when should i add new codec and do not abuse demuxer? [20:47] <michaelni> wm4, libavcodec can and does, but if you want it to use your buffers you have to provide callbacks to allocate and release them [20:48] <durandal_1707> there is format that packs 10 24bit samples into 32 bytes [20:48] <wm4> michaelni: I think the use case where you just want to keep a decoded image around for a while is the most common [20:49] <wm4> michaelni: for that, the current interface is complicated and messy, and right now you even have to work around decoder bugs (see cmdutils.c) [20:50] <wm4> michaelni: it'd be much more useful if libavcodec just gave the user a buffer of a decoded image, and a function to give it back to libavcodec again [20:54] <michaelni> wm4 that shouldnt be hard to implement [20:55] <wm4> well, consider it a feature request :) [20:58] <saste> wm4, consider it a "patch welcome" ;-) [20:58] <nevcairiel> i heard rumors anton was working on reference counting for avframes, or was that avpackets.. i forget [20:59] <ubitux> saste: i see no av_free_packet() in doc/examples/demuxing.c; is this normal? [20:59] <saste> ubitux, maybe, or maybe not [20:59] <durandal_1707> nevcairiel: hmm, what about switching to c++ ? [21:00] <saste> durandal_1707, no, let's designa new language [21:00] <ubitux> based on ffmpeg eval()? [21:01] <durandal_1707> web [21:11] <durandal_1707> bunch of mp3 failures recently [22:12] <ubitux> now vobsub subtitles are gonna get ugly [22:14] <wm4> does anyone know how vobsubs are positioned over the video? [22:15] <nevcairiel> they have x/y coordinates in the bitstream, how else? :d [22:16] <nevcairiel> the only problem is that it doesnt come with a video frame size in the bitstream, so you either need to guess that (based on the video stream that comes with it), or have a vobsub index file with that info [22:17] <wm4> I know [22:17] <wm4> and then there's the question how anamorphic video is handled in the first place (my own attempts of testing anamorphic DVD rips with vobsubs embedded give a "stretched" result) [22:18] <wm4> (still slightly better what mplayer/mplayer2 do right now, though) [22:19] <nevcairiel> the most annoying part about vob subs is that the bitstreams fails to carry a palette, so if you just open a .vob file, you get ugly ugly subs [22:44] <saste> ubitux: subtitles rendering in ffplay, yeah :) [22:44] <ubitux> :D [22:45] <ubitux> i think it's going to get a bit hacky when i'll add support for external bitmap subtitles (such as vobsub) [22:46] <saste> ubitux, this reminds me that nicolas posted a patch for showing subtitles text via ffprobe [22:47] <saste> it was not committed because ffprobe couldn't support arbitrary section nesting, now it should be fairly possible [22:47] <ubitux> feel free to nudge him :) [23:40] <saste> ubitux: i have an updated psnr diff filter in one of my local repo, but it is still incomplete IIRC [23:40] <ubitux> based on luca's one? [23:41] <saste> it should have more features than compare [23:41] <ubitux> the biggest problem with compare was the 10 alloc per frame [23:41] <saste> no it was based on another filter posted some time ago by roger pau monnè [23:41] <ubitux> which was -really- stupid [23:41] <saste> IIRC [23:41] <ubitux> ok [23:41] <saste> i can send you psnr if you want to work on it [23:42] <ubitux> no i don't have time for this sorry [23:42] <ubitux> but i'm pretty interested in the feature though :) [23:42] <saste> there was an optimization problem also, it used generic routines but we wanted to implement optimized format-specific ones [23:44] <cone-624> ffmpeg.git 03Stefano Sabatini 072ac6a3d1bfd0: lavfi/drawtext: add support to expansion of generic expressions [23:44] <cone-624> ffmpeg.git 03Stefano Sabatini 078ddeae5ae7b8: doc: add libavformat.texi and ffmpeg-formats.texi files [23:44] <cone-624> ffmpeg.git 03Stefano Sabatini 07d82331721ee6: doc: add libavdevice.texi and ffmpeg-devices.texi files [23:44] <cone-624> ffmpeg.git 03Stefano Sabatini 079c492271a8e4: doc: move syntax and eval chapters from libavutil to ffmpeg-utils [23:44] <cone-624> ffmpeg.git 03Stefano Sabatini 073a227f5d9042: doc: add ffmpeg-protocols.texi manual [23:53] <burek> michaelni, it's sortable now [23:53] Action: ubitux wonders if he shouldn't wait for the crowd-funding thing before doing true peaks in ebur128 [23:53] <burek> (without page reload) [23:56] <cbsrobot> ubitux: no [23:57] <ubitux> cbsrobot: why? :D [23:57] <ubitux> you want to profit from my work for free?! you evil! [23:58] <ubitux> :)) [23:58] <cbsrobot> well I do not want you to think about it too much [23:58] <ubitux> well i should have time for this pretty soon anyway [23:58] <cbsrobot> ubitux: sure and become a millionair [23:58] <burek> llogan, if you could provide a sample screen shot of "ffmpeg green" that would be great :) [23:59] <burek> cbsrobot :))) +1 [23:59] <michaelni> burek, the sorting is cool [00:00] --- Wed Nov 28 2012

1 0

ffmpeg.log.20121127
by burek 28 Nov '12

28 Nov '12

[00:01] <perryg> howdy, could i get some help writing a command converting a 2 channel, 16 bit, 48000hz, wave file to a 16 bit 16000 hz headerless (raw) PCM file? [00:01] <perryg> the PCM file being mono [00:02] <perryg> i've got ffmpeg -i audio_006.wav -ar 16000 -acodec pcm_s16le test.pcm [00:02] <perryg> but that throws an error [00:03] <llogan> ffmpeg -formats | grep -i pcm [00:03] <llogan> "-f s16le" for example [00:03] <llogan> i guess [00:04] <llogan> also, you forgot "-ac 1" or equivalent audio filter [00:06] <perryg> ffmpeg -i audio_006.wav -ar 16000 -f s16le -acodec pcm_s16le -ac 1 test.pcm [00:06] <perryg> that at least runs [00:08] <perryg> i'm trying to create an API that takes these wave files and runs them through dragon dictation's API. they're very picky about how the audio is formatted [00:08] <perryg> that goes through but the caption is garbage [00:09] <llogan> a friend recently attempted to use dragon to IM me, and all it said was, "Saw from saw for the. The fed could." [00:09] <llogan> ok! [00:10] <perryg> ha, yeah, since the output file is headerless, i'm a little blind as to how dragon is interpreting it, but I can't get a good conversion [00:10] <perryg> er a good response from them [00:11] <perryg> yeah, dragon might just be garbage [00:14] <llogan> i don't the other guy "trained" it enough [00:15] <perryg> yeah, this is intended to work without training, but I think the API might just not be there yet [00:17] <llogan> did ffmpeg make a good output for you? you can play it back with ffplay: ffplay -ar 16000 -f s16le -ac 1 test.pcm [00:21] <perryg> i forgot about ffplay, i was loading it in audacity, checking now [00:22] <perryg> foo, used the brew installer. didn't include ffplay [00:23] <perryg> loading now [01:08] <lakcaj> llogan, thanks for the info on the patch [01:27] <tg2> guys is there an easy way to dump the info about keyframes (duration + bytelocation) in a video? [01:28] <tg2> ffprobe with -show_frames ? [01:33] <tg2> goal being to manually generate an flv manifest of keyframes and their locations (in the timeline and at the byte location) [02:18] <Aristide> Hi ! [02:19] <Aristide> What is a lightest video and audio codec for ffmpeg ? (for CPU ressources) [02:38] <llogan> Aristide: "-c copy" [02:41] <tg2> is there a way to limit ffprobe's --show-frames output to I frames only? [02:41] <tg2> without parsing through grep? [07:25] <defaultro> good evening folks, what would be the best encoding so that when we upload to youtube, the quality isn't much affected? [08:23] <llogan> defaultro: you upload the highest quality that is practical for you. that usually either means the original content, or re-encoded with libx264 encoder [08:23] <llogan> since youtube is going to re-encode whatever you give it anyway [08:25] <llogan> possibly starting with something like: ffmpeg -i input -c:v libx264 -preset slow -crf 18 -c:a copy output.mkv [09:00] <defaultro> thanks llogan [15:24] <leandrosansilva> Hello to all. Do I need do call av_free() in a AVCodecContext after I close it? [15:24] <leandrosansilva> or ... do avcodec_close frees the AVCodecContext which it closes? [15:26] <ubitux> look at the doxy of avcodec_alloc_context3 [15:28] <leandrosansilva> ubitux, the answer is yes... thx :-) [15:29] <juanmabc> if you need to alloc, open, close, seems expected behaviour to free [15:29] <juanmabc> too [15:30] <juanmabc> makes it reusable between open/closes maybe [15:37] <Mavrik> hmm, using libavfilter for filtering audio/video frames change my PTS to some wierd values [15:38] <Mavrik> is there a way to control that? [15:56] <juanmabc> there is a setpts filter [16:58] <dericed> i'm looking for a suggested command to produce IMX50 using ffmpeg. I see an example here: http://www.itbroadcastanddigitalcinema.com/ffmpeg_howto.html#Encoding_D10 but it looks out-of-date. [17:11] <mateo`> dericed: dericed: hi, something like this should be ok: http://pastie.org/5443193 [17:45] <dericed> mateo`: thanks [17:45] <dericed> i also need to at a pad filter to ensure the output is 720x512 [17:52] <DelphiWorld> Hey everyone [17:52] <mateo`> dericed: sure :) [17:53] <DelphiWorld> it's pocible to create a Multi track MPEG2TS using FFMPEG? [17:53] <mateo`> dericed: also you may have to merge the audio tracks to generate a proper D10 layout [17:54] <DelphiWorld> any *clue* ? [17:55] <mateo`> dericed: you also may have to use the fieldorder filter if you're dealing with ntsc input content [17:57] <Mavrik> DelphiWorld: what's a "multi-track" mpeg2-ts? [17:57] <DelphiWorld> Mavrik: multi audio stream, encapsulated in a MPEG2TS container [17:57] <Mavrik> btw, anyone here with libavfilter experience? is there a documentation about what happens to PTS when frames go through filters? [17:57] <DelphiWorld> Mavrik: like a tv channel that have multi stream [17:58] <Mavrik> DelphiWorld: IIRC it is possible [17:58] <DelphiWorld> Mavrik: how please? :) [17:58] <Mavrik> you define several inputs and then use "-map" to map them all to a single output [17:58] <Mavrik> as different streams [17:59] <DelphiWorld> Mavrik: then the end user can chouse ? [17:59] <Mavrik> that depends on the player :) [17:59] <Mavrik> but usually yes [17:59] <Mavrik> :) [18:00] <DelphiWorld> Mavrik: :) [18:00] <DelphiWorld> Mavrik: i may bugs you soon, ok ? [18:10] <dericed> mateo`: this is turning into a huge command. Compared to 'ffmbc -i file -target imx50 outfile' [18:11] <dericed> btw can someone point me to where the target options are defined in ffmpeg. i'm interested in developing one for imx. [18:13] <mateo`> dericed: yes this is a huge command ... i use some tools to generate it automatically [18:13] <ubitux> dericed: opt_target() in ffmpeg_opt.c [18:27] <dericed> ubitux: thanks [20:28] <danverde> Is this the correct forum to ask development questions about using libavfilter and libavcodec in my own applications? [20:37] <DelphiWorld> danverde: this is not a forum:P [20:41] <danverde> my bad. channel :) [21:04] <Mavrik_> danverde, yeah [21:04] <Mavrik_> to some extent :) [21:05] <sacarasc> DelphiWorld: Strictly speaking, it is a forum. [21:05] <DelphiWorld> :) [21:22] <elkng> can ffmpeg do something with *.caf files ? [00:00] --- Wed Nov 28 2012

1 0

ffmpeg-devel.log.20121126
by burek 27 Nov '12

27 Nov '12

[00:50] <cone-97> ffmpeg.git 03Diego Biurrun 07ddb009425cdf: riff: Make ff_riff_tags static and move under appropriate #ifdef [00:51] <cone-97> ffmpeg.git 03Sean McGovern 07f2ce813e3760: configure: suncc: Add compiler arch support for Nehalem & Sandy Bridge [00:51] <cone-97> ffmpeg.git 03Marcus Stollsteimer 072bf794b69820: doc: avtools-common-opts: Fix terminology concerning metric prefixes [00:51] <cone-97> ffmpeg.git 03Anton Khirnov 074632abc7a3a6: avconv: fix copying per-stream metadata. [00:51] <cone-97> ffmpeg.git 03Paul B Mahol 07b491da18d1df: yop: fix typo [00:51] <cone-97> ffmpeg.git 03Diego Biurrun 0728c8e288fa03: x86: h264_chromamc: port to cpuflags [00:51] <cone-97> ffmpeg.git 03Michael Niedermayer 07bf2f93cdbf15: Merge commit '28c8e288fa0342fdef532a7522a4707bebf831cc' [01:02] <cone-97> ffmpeg.git 03Daniel Kang 07ad01ba6ceaea: x86: h264: Remove 3dnow QPEL code [01:02] <cone-97> ffmpeg.git 03Michael Niedermayer 0766c3bac2b9de: Merge commit 'ad01ba6ceaea7d71c4b9887795523438689b5a96' [01:52] <cone-97> ffmpeg.git 03Paul B Mahol 077d2156aee8b6: adpcm/thp: make use of ADPCMChannelStatus to store prev samples [02:22] <cone-97> ffmpeg.git 03Daniel Kang 07610e00b3594b: x86: h264: Convert 8-bit QPEL inline assembly to YASM [02:22] <cone-97> ffmpeg.git 03Janne Grunau 07e96d90eed66a: remove #defines to prevent use of discouraged external functions [02:22] <cone-97> ffmpeg.git 03Michael Niedermayer 077b29b07394c6: Merge remote-tracking branch 'qatar/master' [02:23] <Compn> they beat 3dnow support out of diego ? [02:25] <michaelni> Well, if diego wants to maintain 3dnow support iam happy to put it back and have him maintain it in ffmpeg ... [02:26] <durandal11707> diego wanted to remove 3dnow, no? [02:29] <Compn> i dont know [02:29] <Compn> i remember he had the k6-3 [02:30] <cone-97> ffmpeg.git 03Michael Niedermayer 07262402743911: lavu: drop time redefinition as it causes some problems in some uncommon configurations. [02:52] <iive> is the code actually broken? [03:00] <Compn> we just dont have k6-3 fate box [03:00] <Compn> :P [03:01] <michaelni> Compn, no [03:01] <michaelni> we have a 3dnow fate box [03:01] <michaelni> we do not have a maintainer for the 3dnow code [03:01] <Compn> oh :) [03:01] <Compn> ok [03:01] <michaelni> its a duron not a k6-3 though strictly speaking [03:02] <michaelni> so it supports mmx2 too, but its disabled for the test [03:02] <michaelni> so it should behave like a k6-3 [03:03] <michaelni> also that commit just disabled part of the 3dnow code in ffmpeg not all [03:04] <iive> still, is the code broken? [03:08] <michaelni> iive, no, it worked [03:09] <michaelni> but there was a subsequent commit that moved some of it to yasm so to leave 3dnow in there one would have to port it to yasm [03:10] <michaelni> i dont volunteer for doing that ... [03:14] <iive> also, have somebody benchmarked the new code. I understand it is just rewrite of the old code... but everything is possible. [04:33] <cone-97> ffmpeg.git 03Michael Niedermayer 0769d45d2ecb94: avfiltergraph: add aresample_swr_opts [04:33] <cone-97> ffmpeg.git 03Michael Niedermayer 074bd6d73fc8d4: ffmpeg: fix passing swresample options to the auto inserted swr [04:33] <cone-97> ffmpeg.git 03Michael Niedermayer 07b5e0e327ceb7: ffmpeg: fix swr filter_type for auto inserted aresample filters [07:32] <llogan> "-metadata creation_time=now" with mp4 container appears to set correct time in console output, but probing with 'ffmpeg -i' shows a time 24 years in the future. [07:43] <kcm1700> what do you people think about making segment muxer to support cutting output video file by its file size? [10:02] <juanmabc> that darwin is being a b**ch [11:19] <ubitux> meh drawutils are lavfi specific :( [11:43] <cone-99> ffmpeg.git 03Paul B Mahol 07c5607dfddcdf: doc/general: improve AST description [13:04] <cone-99> ffmpeg.git 03Paul B Mahol 075a3370816fc9: av_get_audio_frame_duration: add ADPCM_AFC [13:14] <cone-99> ffmpeg.git 03Peter Ross 07f2dc82b90ffe: dpx: perform width/height upgrade and av_image_check_size earlier to prevent segfault on malformed input [13:30] <cone-99> ffmpeg.git 03Peter Ross 07929a24efff9a: iff decoder: merge iff ilbm and byterun1 decoders [13:30] <cone-99> ffmpeg.git 03Peter Ross 07f737e7602649: iff decoder: reindent [13:30] <cone-99> ffmpeg.git 03Peter Ross 0776f60788e796: iff demuxer: don't bother checking bitmap compression type; iff decoder does this now [14:03] <cone-99> ffmpeg.git 03Michael Niedermayer 0786270236d5ef: dsputil_mmx: ff_put_dirac_pixels depend now on yasm. [14:36] <cone-99> ffmpeg.git 03Justin Ruggles 072d3993ce8c38: x86: h264 qpel: use the correct number of utilized xmm regs in cglobal [14:36] <cone-99> ffmpeg.git 03Justin Ruggles 073ffed68c2afa: mpegaudiodec: use planar sample format for output unless packed is requested [14:36] <cone-99> ffmpeg.git 03Justin Ruggles 078e134e5104e9: lavc: clarify get_buffer() documentation [14:36] <cone-99> ffmpeg.git 03Michael Niedermayer 07a13148f63351: Merge commit '8e134e5104e99a69cd4cea10540a7ce9c3682a2c' [15:46] <blez> hello, since the #ffmpeg is silent I'll ask here (also it might be a code problem) [15:46] <blez> I'm converting wav stream to .flac, but the output is not seekable [15:46] <durandal_1707> what player? [15:47] <blez> vlc, winamp [15:48] <durandal_1707> what about ffplay? [15:48] <durandal_1707> or mplayer [15:48] <blez> I'll check. [15:49] <blez> ffplay doesn't play it well [15:49] <blez> it's all slowed down [15:49] <blez> invalid sync code [15:50] <durandal_1707> pastebin full ffplay console output [15:51] <blez> http://pastie.org/pastes/5437530/text [15:52] <blez> that's with this cmd line: ffmpeg -f dshow -i audio="virtual-audio-capturer" something.flac [15:54] <durandal_1707> try another container [15:55] <durandal_1707> is this regression? [15:59] <blez> it works with .wav [16:00] <durandal_1707> it may be flac muxer bug [16:00] <durandal_1707> blez: you use flac in .wav ? [16:01] <blez> I think the input stream is wav and I want to output flac [16:01] <durandal_1707> input stream is dshow [16:01] <durandal_1707> use -acodec flac something.wav [16:02] <cone-99> ffmpeg.git 03Justin Ruggles 073d3cf6745e2a: aacdec: use float planar sample format for output [16:02] <cone-99> ffmpeg.git 03Michael Niedermayer 0759b68ee8870a: Merge commit '3d3cf6745e2a5dc9c377244454c3186d75b177fa' [16:05] <blez> vlc can't even play it, winamp plays it but it's not seekable [16:05] <blez> (winamp plays it when renamed to .flac) [16:08] <durandal_1707> blez: is it seekable with ffplay? [16:10] <blez> no, it's slow [16:11] <blez> I can't tell if it's seekable cause the sound is screwed [16:11] <blez> nope, not seekable, I saw the output [16:13] <blez> so, is it fixable? Cause I know there are formats that can't be converted well from streams [16:15] <durandal_1707> hmm, try nut [16:17] <cone-99> ffmpeg.git 03Diego Biurrun 077ee40713621d: x86: fix build without inline asm [16:17] <cone-99> ffmpeg.git 03Janne Grunau 07706acb558a38: h264: enable low delay only if no delayed frames were seen [16:17] <cone-99> ffmpeg.git 03Janne Grunau 075ae72f545329: flashsv: check for keyframe before using differential coding [16:17] <cone-99> ffmpeg.git 03Michael Niedermayer 07a3f30f2e995c: Merge commit '5ae72f54532960cb9eae82a1c9e8d505106c022b' [16:18] <blez> is that no [16:20] <durandal_1707> try -acodec flac something.nut [16:20] <durandal_1707> does it seeks/plays correctly with ffplay? [16:22] <blez> Invalid data found while processing input [16:23] <durandal_1707> it plays nothing? [16:24] <blez> yes [16:24] <blez> it just prints this line [16:24] <blez> "Invalid data found when processing input" [16:24] <cone-99> ffmpeg.git 03Janne Grunau 077709ce029a7b: lavf: avoid integer overflow in ff_compute_frame_duration() [16:24] <cone-99> ffmpeg.git 03Michael Niedermayer 07386fc67c79b7: Merge remote-tracking branch 'qatar/master' [16:25] <durandal_1707> looks like you cant ecode directly to flac [16:25] <blez> blah. [16:26] <durandal_1707> you could try other codecs, perhaps only pcm ones will work [16:30] <blez> fine, thanks [16:31] <blez> also, is there a way to redirect the visual output of ffplay to a window? [16:33] <cone-99> ffmpeg.git 03Michael Niedermayer 074b5a12a2cb02: af_aresample: fix rounding of negative timestamps [17:58] <cone-99> ffmpeg.git 03Sergio Garcia Murillo 07ec51b3308eff: h264: Fix minor bug in h264.c error trace [18:05] <Compn> anyone find any good deals on 3tb hds today ? [18:05] <Compn> ehe [18:23] <michaelni> Compn, whats good ? [18:24] <Skyler_> Thre's a ~$130 refurbished one on Newegg I think, for internals, but I'm pretty sure I remember Black Friday having better deals [18:24] <michaelni> i see this one: http://geizhals.at/eu/686485 <-- 117.9 euro [18:26] <michaelni> but thats more expensive than US :( [18:29] <michaelni> amazon seems to list some 3tb at 130usd too [18:38] <Compn> heh [18:38] <Compn> 4TB $170 http://www.amazon.com/Touro-Desktop-External-Drive-0S03396/dp/B007K4HA0W/re… [18:38] <Compn> go go go... [18:38] <nevcairiel> probably some noname branded drive inside. :p [18:39] <Compn> hitatchi :) [18:39] <tg2> likely a hitachi ;) [18:39] <nevcairiel> somehow i feel confirmed in my opinion [18:39] <tg2> yeah hgst [18:39] <tg2> its a deathstar [18:39] <Compn> hitachi is what i've been buying [18:40] <Compn> after wd , seagate, maxtor shenanigans [18:40] <tg2> hitachi is great actually [18:40] <nevcairiel> you never had ibm shenanigans? :p [18:40] <tg2> I have 36 of the regular deskstars in raid, not 1 failure yet [18:40] <Compn> i didnt even know ibm still sold drives haha [18:40] <nevcairiel> they sold their hdd line to hitachi [18:40] <tg2> and hundreds of ultrastars [18:40] <tg2> hgst is owned by WD now though I believe [18:41] <tg2> © 2012 HGST, a Western Digital company [18:41] <tg2> yep [18:42] <nevcairiel> i run mostly WD drives, and dont remember any serious failure. got a DOA once, but that can happen to anything, so i don't think much about it [18:42] <JEEB> I just run stuff through badblocks, if it passes that it'll work for a couple of months at least [18:43] <tg2> zfs [18:43] <tg2> fucks given about dead blocks: 0 [18:43] <JEEB> badblocks is the testing tool :P [18:43] <JEEB> that runs a write/read pass through it all [18:43] <tg2> ^ [18:44] <JEEB> aanyways, it's just a general test before formatting and putting the HDD into use [18:44] <JEEB> :P [18:44] <tg2> like burn in test? [18:44] <JEEB> yes [18:44] <tg2> makes sense, probably takes a while on a 4tb though :D [18:44] <tg2> esp when you have 180 of them [18:47] <JEEB> haha, yeah [18:47] <JEEB> took like 28h for a 1TB drive [18:48] <tg2> could probably use some optimization [18:49] <tg2> at drive throughput should take ~20hrs to read/write [18:49] <tg2> a raid-z replacement of a 4tb ultrastar takes about 36 hrs [20:27] <ajiekcahdp> hi all,can anyone help me? rtmp [21:01] <cone-99> ffmpeg.git 03Paul B Mahol 0767be7fb43259: adpcm: thp, afc: support multichannel decoding [21:06] <burek> sorry for bugging you guys, but, can anyone explain what does line 51 in fate-recv.sh, which says "exec <report" actually do? (in fateserver repo) [21:07] <burek> report is a text file [21:07] <burek> i.e. not a script/executable [21:09] <AJIeKCaHDp> Is it possible to ignore messages coming from the server? [21:09] <AJIeKCaHDp> rtmp [21:09] <burek> 2>/dev/null ? [21:09] <cone-99> ffmpeg.git 03Michael Niedermayer 0789d5262289ae: mpeg4videodec: use ROUNDED_DIV for pts calculation [21:09] <cone-99> ffmpeg.git 03Michael Niedermayer 078d231e37af0c: af_aresample: consistently use ROUNDED_DIV [21:10] <AJIeKCaHDp> I think that the RTMP server sends ffmpeg command abort and streaming video stops [21:42] <cone-99> ffmpeg.git 03Michael Niedermayer 07109268c826ae: get_dc: fix rounding [21:56] <burek> if anyone finds some spare time to check this, i would really appreciate it: http://pastebin.com/nWTgymYv [21:56] <burek> i dont understand the meaning of "exec <report" [22:12] <michaelni> burek, it makes stdin suck from a file named report [22:15] <burek> so, those 2 lines [22:15] <burek> exec <report [22:15] <burek> head -n2 >summary [22:16] <burek> are actually equivalent to head -n2 report > summary? [22:16] <burek> oh i get it now.. [22:16] <burek> thanks [22:16] <burek> i thought he was executing something like variables assignments or such :S [22:55] <saste> ffserver badly needs a maintainer (or more love) [00:00] --- Tue Nov 27 2012

1 0

ffmpeg.log.20121126
by burek 27 Nov '12

27 Nov '12

[09:05] <zap0> if i have a YUV codec and i output in a r16.g16.b16 format is the YUV to RGB conversion 1st going to happen in 8bit, then the output simple expanded to 16bit? or the YUV will be done with all 16bits of resolution ? [09:06] <zap0> please ping me if you answer, back in 10. [09:35] <bimbis> hi all, I have the same problem as described in http://ffmpeg.org/pipermail/ffserver-user/2012-September/000248.html [09:35] <bimbis> ffmpeg version 1.0, also tried from git N-46531-g7693543 [09:36] <bimbis> os: linux, x86_64 [09:39] <bimbis> ffmpeg -i http:src -map 0:0 -map 0:2 -c copy smth.ts - then writing to file, source maping works [09:40] <bimbis> ffmpeg -i http:src -map 0:0 -map 0:2 http://127.0.0.1:8090/feed.ffm - ffserver takes 0:0 and 0:1 stream, but not 0:0 and 0:2 [10:00] <blez> for max mp3 I should do: -acodec libmp3lame -ac 2 -ab 320k right? [10:01] <juanmabc> freq? [10:04] <blez> oh yes [10:04] <blez> what else? [10:04] <blez> and how to fix frequency [10:06] <juanmabc> -ar 44100 [10:06] <blez> http://svnpenn.github.com/2012/ffmpeg-encode-mp3/ [10:06] <blez> what does: -b:a 320k do ? [10:06] <juanmabc> newer syntax for -ab [10:07] <blez> looks like mp3 has also 48000 [10:08] <juanmabc> f course, was an exmpl [10:08] <blez> I'm looking for the most cleanest sound [10:09] <juanmabc> flac is loosless [10:11] <blez> I know, I'm using it too [10:11] <blez> what should I use -f mp3 or -acodec libmp3lame [10:16] <blez> after encoding the original flac is way louder than my generated mp3 [10:16] <blez> why is that? [10:18] <blez> even if I do -vol 384 [10:21] <blez> hm, my generated flac is not seekable [10:22] Action: blez help? [10:43] <bimbis> i need help with ffserver too :) can someone help [10:43] <bimbis> ? [10:47] <zap0> anyone help with my problem? [11:35] <elkng> how can I run ffplay with more than one video file as option ? [11:36] <elkng> if I run it "ffplay 1.avi 2.avi" it says: "Argument '2.avi' provided as input filename, but '1.avi' was already specified." [11:39] <Tjoppen> you can't [11:44] <elkng> why its this way ? any player should get as many filenames parameters as user wants to open [11:53] <Mavrik> becuase it's a rare feature needed and it's not trivial to implement? [12:03] <bimbis> is someone using audio stream mapping to pick one language audio stream and then feed ffserver? [12:06] <elkng> Mavrik: "not trivial to implement", why not to look at how "mplayer" implements it and copy code to ffplay ? [12:12] <EricAhn> I configured using " --enable-muxers \ [12:12] <EricAhn> --enable-muxer=h264 \ [12:12] <EricAhn> --enable-muxer=mpeg4 \ [12:12] <EricAhn> --enable-muxer=rtp \ .... ", but I found "#define CONFIG_RTP_MUXER 0" config.h. It's weird? It's full configuration, http://pastebin.com/nB6yx3WB. [12:59] <Tjoppen> elkng: ffplay isn't meant as a fully featured player, AFAICT. it's more of an example program [14:14] <StaRetji> howdy folks [14:15] <StaRetji> can ffmpeg be comiled with live555 support? [14:16] <StaRetji1> so I thought if I could compile ffmpeg with live555 support, I might actually play and re-stream to rtmp [15:18] <blez> somebody? [15:18] <blez> my generated flac is not seekable [15:45] <blez> sooomeboooooody [15:46] <relaxed> blez: ffmpeg version? [15:47] <blez> ffmpeg version N-46936-g8b6aeb1 Copyright (c) 2000-2012 the FFmpeg developers [15:47] <blez> built on Nov 20 2012 19:34:37 with gcc 4.7.2 (GCC) [15:47] <relaxed> did you try a diffferent player? mplayer or vlc? [15:47] <blez> yes, same with vlc and winamp [15:49] <relaxed> I'm building git now. Give me a second and I'll test too. You could wrap it in matroska if you need a quick fix. [15:59] <blez> did you tried it? [16:06] <relaxed> let me do that now [16:08] <relaxed> I'm not seeing the issue here. mplayer seeks through it normally. [16:09] <durandal_1707> relaxed: problem it that it is using streamed input [16:10] <durandal_1707> direct files works fine [17:49] <DelphiWorld> hey all [17:49] <DelphiWorld> burek: :-P [17:53] <Iandeb> Was thinking I would encode my ripped music with FLAC. In the past I used lame with ffmpeg. What would good settings be for music that I am going to listen to on my Android? [17:55] <Iandeb> Using FLAC that is. Normally I am happy with something like -acodec libxxxx -b:a 128k -ac 2 [17:57] <Iandeb> not sure what settings for a lossless codec are more or less the same qulity above. [18:03] <DelphiWorld> it's pocible to get DVBS stream using FFM? [18:03] <relaxed> Iandeb: going from mp3 -> flac makes zero sense. [18:05] <relaxed> But, if you're going to rerip and do wav -> flac, "ffmpeg -i input.waw output.flac" will do it. [18:06] <relaxed> input.wav* [18:09] <burek> Iandeb, use aac+ [18:09] <burek> DelphiWorld :beer: :) [18:10] <DelphiWorld> burek: ;) [18:25] <smj> I'm trying to capture some video from a game with x11grab but enabling VSync, disabling triple buffering and flipping doesn't seem to help with screen tearing [18:25] <smj> is there some x11grab settings I could try? [19:06] <bimbis> is someone using audio stream mapping to pick one language audio stream and then feed ffserver? [19:07] <bimbis> version 1.0 ? [19:08] <bimbis> for example in ffserver.conf in feed section "launch ffmpeg -i source -map 0:2 -map 0:0" or in Stream section AVOptionVideo map 0:2 ... ? [19:10] <bimbis> to me, ffserver everytime picks only first audio stream (ignores my arguments) [19:26] <tg2> lol ffserver [20:22] <ajiekcahdp> hi all,can anyone help me? rtmp [20:25] <ajiekcahdp> hello? [20:44] <uglyoldbob> is it possible to have a Feed of ffserver use a file when no other input is present? EG: I want the feed to have a looping video when I am not feeding it a webcam stream (without having the streams interrupted). [20:49] <retard> ajiekcahdp: you should get better at asking for help [20:49] <ajiekcahdp> retart: how? :) [20:52] <retard> ajiekcahdp: http://catb.org/~esr/faqs/smart-questions.html [20:52] <ajiekcahdp> Who can help me: I bring streaming video over rtmp, after 60 seconds, it stops without error, what could be wrong? [20:55] <ajiekcahdp> ffmpeg -report ololo.log -vstats_file MFRfile.txt -re -f x11grab -s "1366x768" -r "25" -i :0.0+0,0 -itsoffset 00:00:01 -t 1000000000 -vcodec libx264 -vpre "medium" -s "1024x768" -qscale 5 -f flv ("RTMP server") [20:56] <ajiekcahdp> a command like I use [21:05] <AJIeKCaHDp> Is it possible to ignore messages coming from the server? [21:09] <bimbis> AJIeKCaHDp: 2>/dev/null at the end of command [21:10] <AJIeKCaHDp> I think that the RTMP server sends ffmpeg command abort and streaming video stops [21:11] <bimbis> someone use opensource rtmp server in production systems ? [21:12] <bimbis> other than crtmpserver/red5 ? [21:12] <AJIeKCaHDp> links are not allowed here? [21:14] <AJIeKCaHDp> http://kiwi.kz/broadcast/ [21:15] <AJIeKCaHDp> Here's a site that I want to send the video stream [21:16] <AJIeKCaHDp> I do not know what kind of server it is installed [21:19] <burek> read their documentation [21:21] <AJIeKCaHDp> Streaming works in them with XSplit and Flash media live encoder [21:23] <burek> AJIeKCaHDp, please ask a specific question related to ffmpeg [21:23] <burek> if you want a specific answer, that is [21:24] <burek> bimbis, why are you launcing your ffmpeg from ffserver [21:24] <burek> is there a particular reason for it? [21:24] <bimbis> no, it doesnt matter I launch it from ffserver or not [21:24] <burek> because launching it from cmd line would give you a useful output log [21:24] <bimbis> the result the same - ffserver ignores audio mapings [21:24] <burek> which you could paste using pastebin like site [21:25] <burek> and we might be able to help you [21:25] <bimbis> with loglevel debug I get the same logs :) [21:25] <bimbis> ok, I'll paste soon [21:25] <AJIeKCaHDp> I bring streaming video over rtmp, after 60 seconds, it stops without error, what could be wrong? [21:26] <burek> smj, did you read our wiki? [21:26] <burek> AJIeKCaHDp, one more time, this channel is related to ffmpeg project, so only questions related to ffmpeg are welcome here [21:27] <Garglfluz> does anyone know if there is a way to set specifically the tbc of either the input mpg as read or the output mpg as written? [21:27] <AJIeKCaHDp> burek, I stream the video via ffmpeg [21:27] <Garglfluz> I'm getting a file that reads as tbc 50 and outputs as tbc 25, and somehow it's dropping half the bitrate on the output file [21:29] <burek> Garglfluz, afaik it is related with -r (setting fps) and if the video is interlaced or not [21:29] <burek> but, did you take a look at the docs first? [21:30] <Garglfluz> yes I've been looking through the docs for the last few hours but have made no real progress [21:30] <Garglfluz> pastebin command line and output here [21:30] <Garglfluz> http://pastebin.com/hzASerAX [21:30] <Garglfluz> The file was encoded with "Progressive sequence" set to ON in the Mainconcept MPEG encoder [21:31] <Garglfluz> note in my pastebin that it reads the mpg infile as 38039 kb/s and writes it out at 21305.2kbits/s. I would expect some drop as it's not copying the 1536kbps of audio but that's a heck of a drop! [21:32] <burek> there were no drops man [21:32] <burek> your log is clean [21:32] <Garglfluz> setting an -r 25 for the infile or outfile or both doesn't seem to help. It's reading tbr properly, it's tbc that seems to be the problem (unless it's something else) [21:33] <Garglfluz> sorry I didn't mean a dropped frame, I mean that the bitrate has been reduced to approx half of the original bitrate despite that -vcodec is set to copy [21:33] <burek> nothing was reduced [21:33] <Garglfluz> this is not behavior I would normally anticipate with -vcodec copy [21:33] <burek> the stream is copied verbatim [21:33] <burek> remuxed that is [21:33] <smj> burek: which page do you mean? [21:33] <burek> the bitrate shown is just a statistics of the process [21:34] <burek> which might be dependendent on your cpu/ram or such [21:34] <burek> type ffmpeg -i outfile.m2v [21:34] <Garglfluz> yes I understand that this is the intent, I certainly would expect -vcodec copy to work that way, but my infile is 6GB, the audio portion of it is 230 MB, and the outfile .m2v is 3.2GB [21:34] <burek> and see if the bitrate is correct [21:35] <burek> smj, try ctrl+f "captur" [21:35] <Garglfluz> Stream #0:0: Video: mpeg2video (High), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 1200k tbn, 50 tbc [21:35] <Garglfluz> that's from outfile [21:35] <Garglfluz> so you might be right if you're thinking that it's not tbc and it's something else [21:35] <smj> http://ffmpeg.org/trac/ffmpeg/search?q=captur&noquickjump=1&milestone=on&ti… ? [21:35] <Garglfluz> really my question is why is my outfile so much smaller than my infile [21:36] <AJIeKCaHDp> burek, http://pastebin.com/rcxLSZMs [21:37] <smj> burek: do you mean https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20%28sc… ? it doesn't say a thing about screen tearing [21:37] <burek> Garglfluz, it might as well be a bug [21:38] <burek> could you consider reporting it to our bug tracker? [21:38] <bimbis> burek: http://pastebin.com/EMu5WS7Z [21:38] <Garglfluz> burek, here is a pastebin of infile vs outfile http://pastebin.com/QG9EbD4u [21:38] <bimbis> test1: writing directly to file [21:39] <bimbis> test2: feeding to ffserver [21:41] <burek> bimbis Format flv [21:41] <burek> no go [21:41] <burek> you used -f mpegts [21:41] <AJIeKCaHDp> burek, I've tried a bunch of options, does not help anything [21:42] <burek> bimbis, try: ffmpeg -i http://10.2.3.14:9050/233.239.0.66 -map 0:2 -map 0:0 http://127.0.0.1:1200/stream.ffm [21:43] <burek> Garglfluz, yes, something is fishy there.. [21:43] <burek> I would go for a bug report [21:43] <Garglfluz> eee i've never made a bug report before I feel like I'm growing up [21:44] <Garglfluz> slash... that doesn't help me for today! ha ha ha [21:44] <burek> :) [21:45] <burek> any particular reason why do you remux to m2v? [21:45] <bimbis> burek the same :( [21:46] <burek> bimbis http://ffmpeg.org/trac/ffmpeg/wiki/Streaming%20media%20with%20ffserver [21:47] <bimbis> read 10 times ;) [21:47] <bimbis> I've read 10 times ;) [21:48] <burek> so it works? :) [21:48] <bimbis> no :) [21:48] <burek> how can you tell? [21:48] <burek> any outputs? [21:49] <AJIeKCaHDp> burek, You can not help me? [21:50] <bimbis> in pastebin i pasted 1. ffmpeg writing to file 2. ffserver config 3. ffmpeg feeding ffserver [21:50] <burek> AJIeKCaHDp, first, remove "-re" it's used only with file inputs [21:50] <burek> (read the docs!) [21:50] <burek> second, why "-itsoffset" ?? [21:50] <burek> (again, docs) [21:50] <bimbis> output are ok in both cases, but audio stream is incorrect from ffserver [21:51] <burek> and the worst thing, AJIeKCaHDp, you're not even using ffmpeg [21:51] <burek> ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers [21:51] <burek> that's a fake ffmpeg [21:51] <burek> the real one says Copyright by FFmpeg developers [21:51] <burek> bimbis, what does it mean "audio stream is incorrect from ffserver" [21:51] <Garglfluz> burek, I need to switch out the audio. I get the same problem going to a new .mpg [21:52] <burek> Garglfluz, did you try simply using -an [21:52] <AJIeKCaHDp> burek, ajiekcahdp@notebook:~$ svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg [21:52] <burek> AJIeKCaHDp, svn is deprecated long time ago [21:52] <burek> where did you get the info [21:52] <burek> to use svn anyway? [21:52] <Garglfluz> burek, that command line is with -an [21:54] <burek> Garglfluz, can you try first: ffmpeg.exe -i "V:\infile.mpg" -vcodec copy -an "V:\outfile.mpg" [21:54] <burek> just to see if you get the correct thing [21:55] <bimbis> burek: any ideas ? :) [21:55] <Garglfluz> burek, same problem http://pastebin.com/9GbLG44z [21:58] <burek> Garglfluz your input might be damaged [21:58] <Garglfluz> burek, there is that possibility. starting to wonder if the bug report should really be to mainconcept [21:58] <burek> <burek> bimbis, what does it mean "audio stream is incorrect from ffserver" [21:59] <bimbis> this mean that audio language track is incorrect [21:59] <burek> Garglfluz, well, during that process you have an opportunity to provide a sample to be checked for validity [21:59] <burek> bimbis, any log outputs? [21:59] <bimbis> if mapping is 0:2 and 0:0 in output i get 0:1(not 0:2) and 0:0 [22:00] <bimbis> burek in pastebin are logs, did you see ? [22:01] <burek> bimbis, i dont understand you [22:01] <burek> in your cmd [22:02] <burek> you said -map 0:2 which is Stream #0:2[0x73f](eng): Audio: mp2 [22:02] <bimbis> sorry for my english :) [22:02] <burek> and -map 0:0 which is Stream #0:0[0x73b]: Video: mpeg2video [22:02] <burek> that means in the output you should have [22:02] <burek> 0:0 Stream #0:2[0x73f](eng): Audio: mp2 [22:02] <burek> 0:1 Stream #0:0[0x73b]: Video: mpeg2video [22:02] <burek> which you have [22:02] <burek> so, what's the problem? [22:04] <bimbis> burek its then i write to file [22:05] <bimbis> bellow is example what happens then i feed ffserver [22:05] <burek> i see [22:05] <burek> so ffserver ignores your -maps [22:05] <bimbis> ffserver ignores any arguments [22:05] <bimbis> including maps :) [22:05] <burek> can you try this (and paste the logs): ffmpeg -i http://10.2.3.14:9050/233.239.0.66 -map 0:2 -map 0:0 http://127.0.0.1:1200/stream.ffm [22:06] <bimbis> they are pasted, but i can do it again :) [22:10] <bimbis> Stream mapping: [22:10] <bimbis> Stream #0:1 -> #0:0 (mp2 -> libmp3lame) [22:10] <bimbis> Stream #0:0 -> #0:1 (mpeg2video -> libx264) [22:10] <burek> please paste the full output [22:10] <bimbis> here, o private? there is many lines :) [22:12] <bimbis> Input #0, mpegts, from 'http://10.2.3.14:9050/233.239.0.69': [22:12] <bimbis> Duration: N/A, start: 62592.609278, bitrate: 384 kb/s [22:12] <bimbis> Program 1860 [22:12] <bimbis> Stream #0:0[0x745]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [22:12] <bimbis> Stream #0:1[0x748](lit): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16, 192 kb/s [22:12] <bimbis> thats i was afraid of :) [22:14] <bimbis> burek: http://pastebin.com/3mnXFCYP [22:22] <burek> just a sec [22:23] <bimbis> ok [22:23] <burek> bimbis, what commands exactly did you use for this output [22:24] <bimbis> ffmpeg -i http://10.2.3.14:9050/233.239.0.69 -map 0:2 -map 0:0 http://127.0.0.1:1200/stream.ffm [22:25] <bimbis> last mcast group octet from 66 changed to 69, because its unreachable [22:26] <Tayeb__> yo saste_ [22:29] <burek> bimbis, i see.. it seems it's a bug.. [22:29] <burek> could you report it please? [22:30] <bimbis> ok burek [22:30] <bimbis> maybe you can suggest opensource flv stream server besided red5/rtmpd ?? [22:31] <bimbis> protocol: http or rtmp [22:31] <burek> :) [22:32] <bimbis> there is no answer [22:32] <bimbis> mommoth is absolete [22:34] <maister> in libavcodec, how to make codecs more "verbose". For example, the CLI app outputs this: http://pastebin.com/qs11ern4, but I don't see the last output line in the API. How to enable that? [22:34] <maister> when using the API* [22:46] <bimbis> burek: bug reported :) [22:51] <dbro> Ahoy! I'm trying to convert AV_SAMPLE_FMT_S16 audio samples to AV_SAMPLE_FMT_FLT for use with ffmpeg's native aac encoder. Can anyone point me in the right direction. http://ffmpeg.org/doxygen/0.11/group__lavc__resample.html#g21439d23385e1536… seems close, but I have no idea what to provide for the filter related parameters& Thanks! [22:57] <dbro> hmm, looks like avresample_convert() in libavresample could be what I want [22:57] <maister> Thought it was libswresample? [23:01] <dbro> maister: that looks promising. Thans! [23:03] <dbro> seems there's too ways about this: libswresample with SwrContext and libavresample with http://ffmpeg.org/doxygen/0.11/structAVAudioResampleContext.html#4d60b0d0fa… [23:03] <dbro> *two [23:35] <lakcaj> I'm writing the rtsp from my security camera to file using ffmpeg. Is it possible to view the video at the same time? [23:56] <llogan> lakcaj: you can try the tee psuedo muxer patch [23:57] <llogan> http://ffmpeg.org/pipermail/ffmpeg-devel/2012-November/133744.html [00:00] --- Tue Nov 27 2012

1 0

ffmpeg-devel.log.20121125
by burek 26 Nov '12

26 Nov '12

[00:02] <cone-864> ffmpeg.git 03Nicolas George 07ca4872e8878d: lavf/sbgdec: use proper constants for av_log(). [00:29] <cone-864> ffmpeg.git 03Stefano Sabatini 07fa4ead1d6692: doc: move codec options and descriptions to a dedicated manual page [00:29] <cone-864> ffmpeg.git 03Stefano Sabatini 07f62abbf3b794: doc: move filters documentation to dedicated manual page ffmpeg-filters [00:29] <cone-864> ffmpeg.git 03Stefano Sabatini 07e903cb419473: doc: add libswresample.texi and ffmpeg-resampler.texi files [01:20] <cone-864> ffmpeg.git 03Michael Niedermayer 07e6305f763131: mov: set flags to indicate that byte wise seeking is not supported. [02:45] <cone-864> ffmpeg.git 03Michael Niedermayer 0748cbdaea1576: smacker: fix smacker_decode_header_tree() empty vlc table init [13:24] <durandal_1707> if read_header fails is read_close still called? [13:49] <michaelni> durandal_1707, no, its not, but i think it was discussed a few time that this should/could be changed [14:47] <cone-97> ffmpeg.git 03Michael Niedermayer 078f507feecbe9: avfiltergraph: add AVOption table [14:47] <cone-97> ffmpeg.git 03Peter Ross 07fdd71cf04c4f: iff decoder: initialise HAM line decoder with first palette entry [15:05] <cone-97> ffmpeg.git 03Stefano Sabatini 07c70ec631c928: lavu/opt: add av_opt_ accessors for pixel/format/image size options [15:05] <cone-97> ffmpeg.git 03Stefano Sabatini 07be2c0bc94974: configure: prefer "resampling" in the libswresample pkg-config description [15:59] <cone-97> ffmpeg.git 03Stefano Sabatini 0726c531cc223d: lavu/opt: fix range shown in set_format() log message [17:31] <ubitux> anyone willing to make a proper api for subtitles encoding? :( [17:36] <durandal11707> you can't? [17:37] <ubitux> i don't want to [17:37] <ubitux> it's a pain :p [17:37] <ubitux> but i need it :( [17:37] <ubitux> no one helps me :'( [17:38] <durandal11707> why it is pain? [17:39] <ubitux> you need to change the encoders to use an api such as bprint because they can't guess the final size [17:39] <ubitux> you also need to keep the old api working [17:39] <ubitux> and stuff like that [17:40] <wm4> what is subtitle "encoding" other than muxing? [17:40] <ubitux> markup encoding [17:40] <ubitux> wm4: codecs deal the styles/markup [17:41] <ubitux> formats with the timing and stuff [17:41] <wm4> wait, is this about having a generic internal subtitle representation with styling? [17:41] <ubitux> yes that's what i'm working on [17:41] <wm4> oh dear [17:41] <ubitux> right now the internal subtitle representation is ASS [17:41] <ubitux> i need to keep compat with this btw, i think i found a way [17:42] <ubitux> but with the current state of the encoding api, it's a real pain [17:42] <ubitux> :( [17:43] <ubitux> wm4: related: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-September/130474.html and more recently: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-November/134607.html [17:46] <wm4> ubitux: why not try to avoid all this useless complexity, and add converters for formats where it makes sense (like srt->ass), and use pure text for the rest? [17:47] <ubitux> because there are too much different markups [17:47] <ubitux> and there will be more [17:48] <ubitux> we already have some "converters" [17:48] <ubitux> the more we add, the more messy it gets [17:49] <wm4> what's the purpose of subtitle conversion at all? [17:50] <ubitux> what's the purpose of video conversion at all? [17:51] <ubitux> most of the use case i see right now is convert from [anything] to ASS or SRT [17:51] <ubitux> but webvtt comes into play, and we might need one [17:52] <ubitux> if you want to convert some advanced ass file to a webvtt file for the web for instance [17:52] <ubitux> webvtt has a few things ass doesn't have, and the reverse is true as well [17:53] <wm4> and what will you do in this case? [17:53] <ubitux> the encoder will ignore the styles it doesn't know how to honor [17:53] <ubitux> the abstract subtitles is pretty handy for this [17:54] <ubitux> we have already something re-parsing the ass markup to do this [17:54] <ubitux> but it really is hard to maintain, and doesn't work that well [17:54] <ubitux> especially to manage global styles for instance [17:56] <wm4> I'd just convert everything to flat text (possibly with minor markup) and leave it at that [17:57] <wm4> not like trying to handle more complicated format specific feature will actually work [17:58] <ubitux> it will ease encoders work [17:58] <ubitux> if at some point libass is adding support for a new awesome subtitles markup, or just add new things [17:58] <ubitux> (such as ruby char like we have in webvtt) [17:59] <ubitux> we would be able to make the encoder output them easily [17:59] <ubitux> we won't be able to do that in the current form [17:59] <ubitux> since the packet are decoded to ass and directly sent to libass by the apps [17:59] <wm4> I don't really think general format conversion is the job of ffmpeg; this is a bit too high level [17:59] <wm4> what's next, converting BD menus to DVD menus? [17:59] <ubitux> general format conversion is exactly the job of ffmpeg IMO [18:01] <wm4> video formats [18:01] <wm4> and audio [18:01] <ubitux> why not subtitles? [18:02] <ubitux> if players want to use ffmpeg to decode subtitles, it's nice if they are decoded with a maximum of information kept from the source [18:02] <wm4> because it opens a can of worms [18:02] <ubitux> and also, to offer them the ability to encode them to ass or something else to be sent to libass/whatever is nice [18:03] <ubitux> and we need to deal with encoding too, i hope nicolas will work on this :p [18:04] <ubitux> wm4: so you'd better keep a miserable support for subtitles? :) [18:04] <wm4> ubitux: well, correct ass demuxing would be a nice start to better support [18:04] <ubitux> i would already have done if i could [18:04] <ubitux> unfortunately, that's far from easy for several reasons [18:05] <ubitux> main one being compat, and the other one the fork which won't be able to follow [18:05] <ubitux> and thus app adding #ifdery [18:05] <nevcairiel> stop worrying about them, they dont worry about you =p [18:05] <ubitux> nevcairiel: i worry about apps [18:06] <nevcairiel> give apps reasons to use ffmpeg by providing superior features [18:06] <nevcairiel> ignore politics [18:06] <ubitux> breaking things is not going to give any reason to use ffmpeg [18:06] <wm4> ubitux: whatever you do, please do not try to implement a ffmpeg native subtitle rasterizer; it won't end well [18:06] <ubitux> i just can't change the current output of subtitles decoders [18:07] <saste> wm4: why? [18:07] <ubitux> wm4: why? [18:07] <ubitux> :)) [18:07] <ubitux> wm4: it's almost already done you know [18:07] <ubitux> i just have a headaches to not break anything during integration [18:07] <wm4> oh well... [18:08] <saste> ubitux: "integration"? are you porting libass code? [18:08] <ubitux> no no [18:08] <ubitux> i mean make the decoders rasterize is one thing, but the output must not change [18:08] <ubitux> i mean, they must keep to fill the ass field as well [18:08] <ubitux> otherwise it will break apps [18:08] <ubitux> (that's one thing) [18:09] <wm4> so there'll be text and bitmaps at the same time? [18:09] <ubitux> no [18:09] <ubitux> well, you can. [18:09] <ubitux> rasterize wasn't the correct word sorry [18:10] <ubitux> forget this, i meant make the text subtitles output an AST [18:10] <ubitux> not making any rasterization [18:10] <ubitux> see graph on http://ffmpeg.org/pipermail/ffmpeg-devel/2012-September/130474.html [18:10] <nevcairiel> rasterize is the word that means converting text into images, so not that? :p [18:10] <ubitux> yeah not that. [18:10] <ubitux> i meant abstract representation of the event, instead of a string [18:10] <ubitux> (the ASS line like currently) [18:16] <wm4> ubitux: so, does that mean the application can create an encoder to convert AVSubtitle2 subtitles from one format to another? [18:16] <uau> ubitux: subtitles are different enough that anything relevant to them (other than the lowest level of demuxing) can be handled in a library separate from libavcodec without losing any significant "synergy" [18:16] <uau> and IMO they should, rather than bloating everything together in the same project [18:22] <saste> uau, adopting the same logic we should have two distinct libavcodec, one for audio and one for video [18:23] <saste> subtitles -> image rendering is another matter, and maybe it should stay in a separate lib [18:23] <saste> (libavsubrend?) [18:24] <nevcairiel> avrasterizer [18:25] <durandal11707> why av in it? [18:25] <saste> branding reasons [18:25] <nevcairiel> could make it swrasterizer, even if i never understood the sw* prefix :p [18:26] <saste> SoftWare? [18:27] <uau> saste: separate libraries for audio and video might not be a bad idea [18:27] <uau> but it's a bit late to change that now [18:27] <uau> while it's not too late for subtitles [18:30] <saste> uau, I'm not sure, from an high level point of view A/V/S codecs are not very different, so they benefit from a unified interface [18:32] <uau> that's too high level to be relevant to practice [18:33] <cone-97> ffmpeg.git 03Stefano Sabatini 07252746d05265: lavu/imgutils: add consistency checks to av_image_copy_plane() [18:40] <durandal11707> so too support subtitles one would need to start from 0 ? [18:46] <cone-97> ffmpeg.git 03Peter Ross 072d954ccd84e6: rawdec: use av_image_check_size [18:46] <cone-97> ffmpeg.git 03Peter Ross 07a246a603bf81: avrndec: use av_image_check_size [18:46] <cone-97> ffmpeg.git 03Peter Ross 0733181975b5ac: mpsmpeg4: use av_image_check_size [19:07] <ubitux> wm4: decoders would output the "AVSubtitle2", and then the app would encode it to whatever markup its rasterizer needs (so plain text if mplayer osd or printf, ass if libass, maybe nothing if it has its own one, etc) [19:11] <ubitux> now it's likely the decoders will continue to output also ass for a while, since most apps might not want to do the "encoding step" [19:12] <wm4> what decoders output ass for non-ass subtitles? [19:12] <ubitux> all of them [19:13] <ubitux> remember, that's the internal representation. [19:13] <ubitux> even the raw text decoder is actually outputing ass subtitles [19:14] <ubitux> because ass is supposed to be the format with the most features, and because apps can just send them to libass directly [19:17] <uau> i wonder if zgreg would be willing to drop support for the mangled ffmpeg packet format from libass [19:18] <ubitux> uau: i really don't mind changing this you know [19:18] <ubitux> but it will break everything. [19:19] <ubitux> michaelni: btw, any comment on the ass demuxer? (and the patches it depends on) [19:19] <ubitux> (afaik you're the maintainer) [19:21] <ubitux> uau: it will be fairly easy to just make the ass demuxer output real ass packets, the lavf demuxer output them verbatim as well, and make the muxers deal with "muxed" ass [19:21] <wm4> uau> i wonder if zgreg would be willing to drop support for the mangled ffmpeg packet format from libass <- this doesn't make any sense [19:21] <uau> wm4: how so? [19:21] <ubitux> the problem is, if apps are using ass_process_data(), on pkt->ass it will break them [19:21] <ubitux> (i mean unconditionnally) [19:21] <wm4> even mplayer originally used the function you're probably referencing for some purpose other than ffmpeg interop [19:22] <wm4> and why would this function be dropped? not like it causes trouble [19:22] <ubitux> wm4: ass_process_data() isn't that useful [19:22] <ubitux> it's useful if you're reading line by line an .ass thought [19:22] <ubitux> but i can make the ass demuxer output proper packets instead so& [19:23] <ubitux> uau: so anyway, what do you propose? [19:24] <wm4> hm actually I'm not so sure what libass originally provided this function for [19:24] <ubitux> maybe add a demuxer option? [19:24] <uau> wm4: aurelien added ass_process_data to libass explicitly because he wanted to add his stupid mangled subtitle format to ffmpeg [19:24] <uau> it was not there for any "natural" reason [19:25] <ubitux> maybe i'll add an option to the mkv demuxer [19:25] <ubitux> (and ass demuxer) [19:25] <wm4> still, it's a rather small function, why remove it? [19:26] <wm4> just to accelerate politics inside of ffmpeg? [19:26] <ubitux> wm4: to force ffmpeg to change its format [19:26] <wm4> :/ [19:26] <ubitux> again, i'm fine changing it, but i see no easy way without breaking things [19:26] <ubitux> and of course if you want your player to also work with old ffmpeg or libav, you'll be fucked as well [19:27] <ubitux> i'm not sure libav can change it that easily btw, since they are lacking quite a bunch of timing changes [19:27] <cone-97> ffmpeg.git 03Stefano Sabatini 07cf6c6134cd61: doc: add libswscale.texi and ffmpeg-scaler.texi files [19:52] <michaelni> ubitux, i dont think iam ass demuxer MAINTAINER, except by lack of anyone else actively doing it. [19:52] <ubitux> aren't you the author? :p [19:52] <michaelni> maybe :( [19:53] <ubitux> :) [19:53] <michaelni> if you or uau want to take over maintainership of the ass demuxer i sure have no objections [19:54] <ubitux> i think uau concern is about something else [19:54] <ubitux> it's all about the fact that our subtitles decoders output a ass markup as if they were line from a ass file [19:54] <wm4> ubitux: in general, I think it'd be preferable to keep all subtitles in their "most native" format as possible [19:55] <ubitux> wm4: then just demux them. [19:55] <wm4> ubitux: if there's a separate library that can do generic subtitle conversion, that can be separate [19:55] <ubitux> ffmpeg is all about conversion [19:55] <ubitux> if you don't want convert, just demux the subtitles [19:55] <ubitux> it will do exactly what you want [19:55] <wm4> ubitux: and creating a generic internal format that is a superset of all other subtitle formats sounds like it'll be a big mess [19:55] <ubitux> it's already the case [19:56] <ubitux> it's all "FFmpeg's ASS" [19:56] <wm4> ubitux: most attempts to convert subs using more special features will probably fail to produce usable results [19:56] <ubitux> it works here, and it will be better with what i'm doing [19:56] <ubitux> what you want is just a demuxer [19:56] <wm4> ubitux: what I'm trying to say: if there's such an internal format, it should be actually internal of a separate subtitle conversion library [19:56] <ubitux> and we already do this [19:57] <wm4> ubitux: which can take packets in proper subtitle formats as in/output [19:57] <wm4> k [19:57] <ubitux> our demuxers split the subtitles into events, put the timing & duration info in the packet, the text/markup is unmodified [19:57] <ubitux> that's what you want [19:58] <ubitux> now i also want people to be able to convert properly these subtitles, that's why i'm writing this [19:58] <ubitux> instead of our current ass-hacks [19:58] <ubitux> wm4: moving the internal representation outside is complicated to make the link with codecs [19:59] <wm4> what I can see happening is that everyone is suddenly using (or somehow supposed to be using) that internal representation, even if that wasn't planned for [19:59] <ubitux> it supposed to. [20:38] <durandal11707> ubitux: i cant disable metadata [20:38] <cone-97> ffmpeg.git 03Stefano Sabatini 0751e9f58e1c9c: lavu/opt: add support for reading pixel and sample format through av_get_int() [20:38] <cone-97> ffmpeg.git 03Stefano Sabatini 07e55c3857d20b: lavc/utils: check return value of avcodec_fill_audio_frame() for < 0 [20:45] <ubitux> durandal11707: mmh ? [20:50] <durandal11707> ubitux: the only way to disable encoder tag is to use bitexact flag [20:50] <ubitux> oh, isn't that one of the few fields added by FFmpeg? [20:52] <ubitux> (the only one?) [22:40] <cone-97> ffmpeg.git 03Michael Niedermayer 07579d21f777ea: tga: check palette size earlier. [23:05] <cone-97> ffmpeg.git 03Michael Niedermayer 07b5b9686615e2: imc: flush decoder [23:06] <cone-97> ffmpeg.git 03Carl Eugen Hoyos 07d643dd5c55c9: Support switching field order when decoding frwu. [23:18] <cone-97> ffmpeg.git 03Ivan Pozdeev 07329b8f85b048: doc/encoders: add a note for x264 options that use colon [23:57] <cone-97> ffmpeg.git 03Stefano Sabatini 07b473c9937ebe: lavu/samplefmt: return the size of the allocated samples buffer at the next bump [00:00] --- Mon Nov 26 2012

1 0

ffmpeg.log.20121125
by burek 26 Nov '12

26 Nov '12

[00:11] <maurer> Hey, when trying to use the ffmpeg command line tool, I'm getting Unable to set CBR to 128000: not supported [00:12] <maurer> I'm not particularly attatched to that bitrate, but I can't seem to figure out how to change this [00:12] <maurer> (this is in the vorbis audio output) [00:36] <freezway> so ffplay can play a file but ffmpeg can't convert it [00:36] <freezway> what gives [00:40] <freezway> nvm [00:40] <freezway> im basically using ffmpeg to fix metadata so id3tag can tag it [00:40] <freezway> is there a way to not convert but just redo the metadata [00:41] <JEEBsv> use the 'copy' codec? [00:41] <freezway> ok ill try [00:41] <freezway> how do I do that? [00:42] <JEEBsv> -c:a copy for example for audio, c:v copy for video, c:s for subtitles [00:47] <freezway> what containers can contain aac [00:48] <JEEBsv> mp4/mkv/flv ? [00:49] <freezway> Unknown decoder 'copy' [00:49] <JEEBsv> uhhh [00:50] <JEEBsv> before -i = decoder/input settings [00:50] <JEEBsv> after -i = encoder/output settings [00:57] <freezway> hmm [00:57] <freezway> weird [00:58] <freezway> after copying to a new container, I can play it [00:58] <freezway> then I run id3tag on it [00:58] <freezway> cant play it anymore [00:59] <freezway> BUT, if I convert to mp3 instead of copying, then i can play it [00:59] <JEEBsv> I have no idea what id3tag is or what it does :P [01:00] <JEEBsv> if you want mp4 you probably want to use the m4a extension that is apple's extension to the ISO Media Container [01:00] <JEEBsv> and that has tags, and afaik ffmpeg nowadays can write them [01:01] <freezway> id3tag sets metadata [01:01] <freezway> like artist [01:01] <JEEBsv> sounds like it supports mostly idv3 :P [01:01] <freezway> yup [01:01] <JEEBsv> well, uhh. idv3 doesn't exactly work as-is in mp4/m4a [01:02] <JEEBsv> and ffmpeg can do tagging now too [01:02] <freezway> oh [01:02] <freezway> it can? [01:02] <freezway> awesoem [01:02] <JEEBsv> yes [01:02] <JEEBsv> elenril wrote some nice code [01:02] <freezway> ooh, this makes my program so much simpler [01:04] <JEEBsv> http://ffmpeg.org/ffmpeg.html#Metadata methinks [01:04] <JEEBsv> although it doesn't have to be a file as far as I know [01:06] <freezway> ffmpeg -i KnockinOnHeavensDoor.mp4 -metadata title="Knocking on Heaven's Door" temp.mp4 [01:06] <freezway> gives me [01:06] <freezway> KnockinOnHeavensDoor.mp4: Invalid data found when processing input [01:07] <freezway> wait [01:07] <freezway> nvm [01:08] <freezway> how do you get ffmpeg to write to the same file it reads from [01:08] <freezway> i.e. modify it rather than copy? [01:08] <JEEBsv> you can't really [01:08] <JEEBsv> you'd have to rewrite the whole index and thus the file's length changes [01:09] <JEEBsv> and other stuff [01:09] <freezway> meh [01:10] <freezway> ill just do an mv and rm [01:10] <freezway> no biggie [01:39] <freezway> sall good [01:39] <freezway> i modded pianobar to download music and apply metadata [01:56] <a|3x> hi [01:57] <a|3x> i can't seem to figure out how to build ffserver with ffmpeg, the executable isn't being built, any suggestions? [02:15] <aixenv> hey guys - it seems that aconv is taking over for ffmpeg, is that correct or not? [02:17] <ubitux> it's not [02:17] <aixenv> can you explain the msg i get when i run ffmpeg ? [02:17] <ubitux> avconv si the tool name from the fork [02:17] <ubitux> aixenv: yes, it's the fork propaganda [02:18] <aixenv> is this related toe debian? [02:18] <aixenv> how would get int he ffmpeg code to not use it that it's deprecated [02:18] <ubitux> yes, debian is distributing libav under the ffmpeg name because the maintainer is one of the libav folks [02:18] <ubitux> aixenv: somehow related: http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html [02:19] <ubitux> a|3x: ffserver_deps="avformat ffm_muxer fork rtp_protocol rtsp_demuxer" [02:19] <ubitux> a|3x: maybe you don't support one of these? [02:19] <aixenv> ok thanks reading, this was really surprising to me, ive always been a big ffmpeg user [02:20] <ubitux> look for "packaging" for a quick answer about your deprecated message [02:21] <a|3x> ubitux: what do you mean i don't support? how do i check if i support these? [02:22] <ubitux> a|3x: maybe you disabled one of these in your configure line [02:22] <a|3x> none of these are in my config line [02:22] <ubitux> how did you run the configure? [02:23] <a|3x> ./configure --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -pipe' --extra-cflags='-O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi [02:23] <a|3x> --disable-vdpau --enable-libmp3lame --enable-libvo-aacenc --enable-libx264 --enable-libxvid --disable-indev=v4l --disable-indev=oss --disable-indev=jack --enable-x11grab --enable-libpulse --disable-outdev=oss --enable-libfreetype --enable-libvorbis --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --enable-hardcoded-tables --enable-librtmp --enable-ffserver [02:23] <ubitux> --optflags='-O2 -pipe' --extra-cflags='-O2 -pipe' please don't do that [02:24] <ubitux> --enable-avfilter is redundant [02:24] <ubitux> --disable-network do you expect ffserver to work without network? [02:24] <ubitux> why are you disabling all the optims manually? [02:24] <ubitux> --enable-ffserver is redundant [02:25] <a|3x> whats wrong with -O2 -pipe ? [02:26] <ubitux> -O2 is the default [02:27] <ubitux> you can add -pipe if you really want to though [02:27] <ubitux> --optflags looks uneeded [02:28] <ubitux> --enable-postproc is also redundant afaik [02:28] <ubitux> --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon this really bother me; why do you do that? [02:29] <ubitux> also, if you really insist on doing this, i think you are looking for --disable-asm [02:29] <ubitux> but it's really going to get slow [02:29] <a|3x> actually a lot of these flags are from gentoo emerge scripts [02:29] <ubitux> :/ [02:30] <aixenv> what an interesting read [02:30] <aixenv> devs acting like children how sad [02:32] <aixenv> can someone critique this line? i wrote this script a while ago - and wonder if it's still accurate : "ffmpeg -y -i "$file" -vcodec libx264 -r 30000/1001 -deinterlace -s 1024x576 -crf 21 -maxrate 2M -bufsize 6M -vpre slow -threads 0 -acodec libfaac -ar 48000 -ab 128k ${file%.*}-1.mp4 >> "${file%}"-encoding.out 2>&1 [02:33] <aixenv> second - ive got a ISO Media, MPEG v4 system, version 1, that im trying to encode to avi (to play on my xbox360, the current format isn't compatible), i usually do my video camera MTS files to mp4, so im not exactly sure what's a good line for mp4 to avi [02:33] <ubitux> what's the purpose of -threads 0? [02:34] <aixenv> i am not going to take credit for that line good sir, i got that from one of the guys in this channel about a year or so ago [02:34] <ubitux> ok [02:34] <aixenv> Darksomething, i forget his name, im not normally on irc [02:34] <aixenv> but usually pretty active contributor to the irc channel whenever i happen to join [02:35] <aixenv> and im not saying that line is "broken" just wanted to make sure it didnt need to be optimized [02:35] <ubitux> looks pretty sane [02:36] <aixenv> ok great , and whats a good line recommendation to keep the quality 1:1 (as close as possible) and go mp4 v1 to avi (that'd work on the xbox360?) [02:36] <ubitux> we tend to use -c:v instead of -vcodec, -c:a instead of -acodec, -b:a instead of -ab though [02:36] <ubitux> but it won't affect your output [02:37] <aixenv> ah ok gotcha - [02:37] <ubitux> maybe using -q:v 0 ? [02:37] <ubitux> look at the -qscale documentation [02:37] <aixenv> whats the vcodec to use for avi? [02:38] <ubitux> isn't it selecting mpeg4 by default? [02:38] <aixenv> well all i can tell you is this particular movie file doesnt play on the 360, and 99% of my movies are avi [02:39] <aixenv> so i was going to try and encode it wih avi -but normally im doing everythign to x264, so this is a bit dif than what im used to doing [02:39] <ubitux> can you ffprobe the file? [02:39] <aixenv> sure one sec [02:39] <ubitux> (pastebin please) [02:40] <aixenv> yes sir one sec [02:42] <ubitux> blindly, i'd say you may try ffmpeg -i in.mp4 -q:v 0 -c:a libfaac -q:a 0 -y out.avi [02:42] <aixenv> http://pastebin.com/4PrDCU3U [02:42] <ubitux> oh you can codec copy the audio actually [02:43] <ubitux> ffmpeg -i in.mp4 -q:v 0 -c:a copy -y out.avi then [02:43] <ubitux> the file might get quite bigger though :P [02:44] <aixenv> thats ok its home network streaming so that should be ok [02:44] <aixenv> unless you mean 2Gb to 200Gb :P [02:44] <aixenv> im not sure if you have a 360 or not, but any idea why that one doesnt work? i have had other mp4's work so i found this one curious [02:45] <aixenv> running now - so we'll see [02:45] <aixenv> i greatly appreciate the line, let me see if it runs w/o error [02:45] <aixenv> ah i might need to update my ffmpeg [02:46] <ubitux> no i don't have a 360 [02:46] <aixenv> Unrecognized option 'q:v' [02:46] <ubitux> well you're using libav so& [02:47] <ubitux> i need to go, gl [02:47] <aixenv> bah i thinkk my upgrade to wheezy blaste away my svn version of ffmpeg [02:47] <aixenv> so i need to just get it updated i bet [02:47] <ubitux> we don't use svn anymore [02:47] <aixenv> ffmpeg version 0.8.4-6:0.8.4-1, [02:47] <aixenv> git ? [02:47] <ubitux> this is not ffmpeg [02:47] <ubitux> yes, git [02:47] <aixenv> whats the git clone url? can you give me that rq before going? [02:48] <ubitux> https://ffmpeg.org/download.html [02:48] <aixenv> ok sweet ty [02:50] <aixenv> this a good configure line (this is what i used to use) ; configure --prefix=/usr --enable-vdpau --enable-libx264 --enable-libtheora --enable-libvorbis --disable-stripping --enable-x11grab --enable-libfaac --enable-nonfree --enable-shared --disable-static --enable-runtime-cpudetect --enable-gpl --enable-libmp3lame --enable-libspeex --enable-xvid --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-postproc --enable-libvpx --enabl [02:50] <a|3x> ubitux, its still skipping ffserver [02:51] <aixenv> i think he left - [02:52] <aixenv> it seems xvid is now libxvid, ok that line configured let's see if it makes heh [03:02] <aixenv> here we go - that line configured once i changed --enable-libxvid ; ffmpeg version N-47057-g48cbdae [03:04] <aixenv> ok that line you gave me is running -ty, will letcha know if that does the trick [03:05] <aixenv> tyvm #ffmpeg, always a great channel (helpful) [03:34] <aixenv> eh not too bad 2.1 to 5.4 [03:34] <aixenv> testing to see if it works [03:36] <okhattab> I am having a hard time building omxplayer from source for the fedora remix on raspi. The issue is with ffmpeg. does any one know if this package has been released for the remix? or for f17? [03:40] <aixenv> hrm this still didnt work - Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1920x804 [SAR 1:1 DAR 160:67], 23.98 fps, 23.98 tbr, 23.98 tbn, 24k tbc [05:42] <aixenv> hrm i got that one to work but the quality is crappy [05:42] <aixenv> seemed pretty basic - ffmpeg -i file.mp4 -vcodec mpeg4 -vtag XVID -acodec ac3 -ar 48000 -ab 192k output.avi [05:44] <aixenv> this mp4 is really crisp [05:44] <aixenv> but the avi looks bleh [05:45] <aixenv> file.mp4: ISO Media, MPEG v4 system, version 1 [05:45] <aixenv> vs [05:45] <aixenv> output.avi: RIFF (little-endian) data, AVI, 1920 x 804, 23.98 fps, video: XviD, audio: Dolby AC3 (stereo, 48000 Hz) [05:54] <aixenv> whats the sub for 'slow' ? - File for preset 'slow' not found [06:08] <aixenv> so if im trying to take a nice h264 movie and make it as good as possible in xvid/avi - any recommendations? [06:31] <aixenv> bah this isnt working for the 360 [06:31] <aixenv> ffmpeg -y -i file.mp4 -vcodec mpeg4 -vtag XVID -r 30000/1001 -filter:v yadif -q:v 0 -q:a 0 -crf 21 -threads 0 -ss 30 -t 60 -acodec ac3 -ar 48000 -ab 128k output-test.avi [06:31] <aixenv> but this did, just crummy quality - ffmpeg -i file.mp4 -vcodec mpeg4 -vtag XVID -acodec ac3 -ar 48000 -ab 192k output.avi [06:38] <aixenv> found one that looks good [06:38] <aixenv> ffmpeg -y -i file.mp4 -vcodec mpeg4 -vtag XVID -r 30000/1001 -q:v 0 -q:a 0 -crf 21 -threads 0 -ss 30 -t 60 -acodec ac3 -ar 48000 -ab 128k output-test.avi [06:39] <aixenv> trying the whole movie now [08:37] <dgbaley27> Hey. I have an avi with an mpeg4 video stream (and mp3 audio stream). I'm trying to include a separate .srt file and mux into an mkv. I get an error with matroska about not being able to write packet with unknown timestamp. Is there a way to regenerate the timestamps for the mpeg4 video? [11:21] <XORi_> Hello. I'm capturing video from a new frame buffer device and ffprobe does not recognize the output. I try to pass it as raw video to ffmpeg but it fails. [11:21] <XORi_> How can I track that ? [11:58] <burek> guys, try to always provide your pastebin links to outputs so we can see what you are asking about [11:58] <burek> it will speed up the time between answers [12:52] <luc4> Hi! Has the symbol ::avfilter_poll_frame been removed recently? [13:09] <ubitux> luc4: the api was made private [13:10] <ubitux> but you're eventually looking for av_buffersink_poll_frame [13:10] <ubitux> look at the filtering examples [13:10] <ubitux> doc/examples [13:13] <luc4> ubitux: I'm trying to compile a player which seems to be using both av_buffersink_poll_frame and avfilter_poll_frame. Was it replaced with ff_poll_frame? [13:13] <ubitux> ff_poll_frame is private [13:13] <ubitux> you can't use that anymore [13:14] <ubitux> the player is not supposed to use it [13:15] <ubitux> what was that player doing? [13:15] <luc4> ubitux: since I'm not the author, I'm just trying to recompile, would changing the function to public make that work? [13:16] <ubitux> you can only disable that code [13:18] <luc4> ubitux: I've never used ffmpeg, I'll try that. Thanks! [16:52] <function1> when i give ffmpeg a file with commas as -i "some,file.flac" it chokes and 'cannot find the file'. 0.10.3. nothing in bug tracker that i could find... anyone know anything about this? [16:53] <function1> or can anyone confirm? [16:54] <klaxa> use multiple -i parameters i guess? [16:54] <ubitux> funyun: yes, the ',' is a special char [16:54] <ubitux> i think i read a thread about this on ffmpeg-user last month [16:54] <function1> any workaround? quotes or escape \, doesnt help [16:55] <ubitux> are you sure the \ is passed? [16:55] <klaxa> oh misread the question, nvm [16:56] <klaxa> did you try 'some,file.flac'? [16:57] <ubitux> function1: you may need to put two of them depending on your shell [16:57] <ubitux> and your command line [16:57] <function1> ubitux: two \ doesnt work [16:57] <function1> \\, [16:58] <function1> klaxa: 'some,file.flac' doesnt work [16:58] <ubitux> mmh [16:59] <saste> ubitux: that's not like filters, it's a whole new mess, some filename just cannot be expressed [16:59] <ubitux> function1: try "file: ..." [17:00] <saste> yes file: could be a valid workaround [17:00] <ubitux> 'file:some,file.flac' should work [17:00] <function1> -i "file: some,file.flac" ? [17:00] <function1> ah [17:02] <function1> yeah that does it. what exactly have i dont there? [18:05] <blez> is it possible to output into http/mms while converting? [18:06] <retard> one does not simply output into http/mms [18:06] <retard> (yes) [18:08] <blez> actually I'm using ffmpeg to capture the audio directly [18:08] <blez> and I'm trying to detect silence [18:08] <blez> any way to do that? [18:08] <retard> i think ecasound can do that, so i guess you could pipe it through that [18:08] <retard> or use ecasound for actually capturing the audio [18:08] <retard> depending on source [18:09] <blez> can ffplay show the current audio level? [18:09] <retard> i don't know [19:01] <iksik2> hello, i'm trying to grab some jpeg images from rtsp stream (one image per 30 seconds running ffmpeg in a loop - just for tests) http://scr.hu/0u6x/7b15k [19:01] <iksik2> command which i'm using here is: ffmpeg -i rtsp://my_stream/ -y -f image2 -vsync 1 test.jpg [19:02] <iksik2> once in a while (not all of them) jpg image seems to be 'broken' like this one attached under url above [19:03] <iksik2> any hints? :( [20:30] <aixenv> ubitux - thanks for your help yesterday, i was able to get a nice quality copy of my mp4 in avi, i basically used my normal mts -> mp4 and made it work for xvid/mpeg4 [20:30] <aixenv> this was the line if anyone is interested - ffmpeg -y -i file.mp4 -vcodec mpeg4 -vtag XVID -r 30000/1001 -q:v 0 -q:a 0 -crf 21 -threads 0 -acodec ac3 -ar 48000 -ab 128k [20:30] <aixenv> well then of course the $outfile.avi [20:34] <conG> Hello everyone [20:35] <conG> I ask your help to make working ffmpeg with downloadhelper (mp3 on) on ubuntu 12.10 [20:38] <conG> I tried this http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [20:39] <conG> but don't know what specifiate in download helper ( ~/root) [20:40] <iksik2> http://scr.hu/0u6x/7b15k [20:40] <iksik2> uh, sorry, to fast click ;-) [21:19] <y_nk> hello [21:20] <y_nk> i've a question related to audio format, no directly to ffmpeg but i don't know where to ask [21:20] <y_nk> i've a network audio device that streams audio. I found my way through soap and i'm now able to get the stream [21:21] <y_nk> i found a kind of pattern that revealed the header... so i think i have the audio message now, but i don't know how to play it properly [21:21] <y_nk> does anyone have any clue on how to do it ? [21:24] <blez> what's the proper way to convert mp3 to flac? [21:24] <blez> I mean, do I need any additional args [21:24] <fatpony> mp3 to flac? [21:24] <fatpony> you mean flac to mp3, right? [21:26] <blez> no [21:26] <blez> anyway, how to convert to flac? [21:27] <blez> at maximum quality (if there is such thing in flac) [21:27] <klaxa> that is... if i may say, stupid [21:27] <klaxa> you will increase filesize without gaining quality [21:27] <klaxa> flac is lossless also, there are no quality settings [21:27] <blez> ok [21:27] <klaxa> rather compression settings [21:28] <blez> for max mp3 I should do: -acodec libmp3lame -ac 2 -ab 320k right? [21:31] <iksik2> 19:02 < iksik2> hello, i'm trying to grab some jpeg images from rtsp stream (one image per 30 seconds running ffmpeg in a loop - just for tests) http://scr.hu/0u6x/7b15k [21:31] <iksik2> anyone? :P [22:02] <cong> Help [22:03] <cong> toujours le même problème, j'ai suivis la procédure ici pour compiler ffmpeg avec mp3 sur ma ubuntu 12.10 (http://doc.ubuntu-fr.org/tutoriel/comment_compiler_ffmpeg) [22:03] <cong> mais rien ne marche et quand je tape ffmpeg en console, il ne m'affiche pas le codec lib...mp3 que j'ai pourtant bien compilé avec ... [22:05] <cbsrobot> cong: on est pas a Saint-Tropez ici [22:05] <cong> rapport ? [22:05] <cong> Je suis sur un chan irc et je viens demander de l'aide. [22:06] <cbsrobot> essaye de parler une autre langue [22:06] <cong> ah ouai pas bête sa par contre [22:07] <cong> Maybe someone can help me, i tried to do working ffmpeg with mp3 support (i tried this http://doc.ubuntu-fr.org/tutoriel/comment_compiler_ffmpeg) [22:07] <cong> but when i type ffmpeg in console mode, anything about lame and mp3 support ... [22:08] <cbsrobot> cong: did you first run: sudo apt-get remove ffmpeg x264 libx264-dev lame libmp3lame-dev [22:08] <cbsrobot> just to make sure it's not installed anymore [22:09] <cong> i retried about 5 or 6 times like all the step on the website [22:09] <cbsrobot> and then type ffmpeg [22:09] <cong> (and yes i remove all of this before) [22:09] <cbsrobot> so it is not there anymore [22:09] <cbsrobot> ok [22:09] <cbsrobot> so in your ffmpeg dir [22:09] <cbsrobot> when looking at config.log [22:09] <cbsrobot> what does it say when searching for libmp3lame ? [22:10] <cbsrobot> did configure find it ? [22:11] <cbsrobot> or wait ... [22:11] <cbsrobot> whan you added --enable-limp3lame to configure [22:11] <cbsrobot> it should have failed if it did not find it [22:11] <cbsrobot> what do you type in console to find a hint for mp3 ? [22:12] <cong> "which ffmpeg" said /usr/local/bin/ffmpeg [22:12] <cong> but i can't go to ffmpeg because it's not a folder ? am I crasy ? [22:12] <cong> no failed no [22:12] <cbsrobot> ffmpeg is not a folder [22:12] <cbsrobot> ffmpeg dir I mean the git clne dir [22:13] <cbsrobot> *clone [22:13] <aixenv> ie: git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg [22:14] <cbsrobot> cong: whats your output for: [22:14] <cbsrobot> ffmpeg -codecs | grep mp3 [22:14] <cong> i use downloadhelper (add on for firefox), so what i should write for the dir ? [22:14] <aixenv> try following this cong, i wrote this, and just even tested it out yesterday albeit on wheezy [22:14] <aixenv> do it from the CLI - follow this [22:15] <cong> D.A.L. mp3 MP3 (MPEG audio layer 3) (decoders: mp3 mp3float ) D.A.L. mp3adu ADU (Application Data Unit) MP3 (MPEG audio layer 3) (decoders: mp3adu mp3adufloat ) D.A.L. mp3on4 MP3onMP4 (decoders: mp3on4 mp3on4float ) [22:15] <cong> for ffmpeg -codes grep mp3 [22:15] <cong> codecsù [22:15] <cong> codes* [22:15] <cong> codecs* [22:15] <cbsrobot> well [22:15] <cbsrobot> no libmp3lame encoder &. [22:16] <cong> no --' [22:16] <cong> what i sould i do now ? [22:16] <cbsrobot> cong: you followed http://doc.ubuntu-fr.org/tutoriel/comment_compiler_ffmpeg ? [22:16] <aixenv> http://pastebin.com/5xkXArgA <== follow that [22:17] <cong> lame != libmp3lame ? [22:17] <cong> yes cbsrobot [22:17] <cbsrobot> step 9 in this [22:17] <cbsrobot> 2nd item [22:17] <cbsrobot> git clone git://git.videolan.org/ffmpeg.git [22:17] <aixenv> follow the doc i linked, it'll work cong [22:17] <aixenv> i just did it yesterday [22:17] <cbsrobot> wut wait [22:18] <cong> ok aixenv thanks i will try [22:18] <cbsrobot> thats just insane [22:18] <cong> i don't understand cbsrobot what you want i do ? [22:18] <cbsrobot> cong: don't follow this tutorial you sent [22:18] <cong> ok [22:18] <aixenv> like i said, follow what i did, it works - [22:18] <cbsrobot> maybe aixenv 's tutorial is better [22:18] <aixenv> it works :) [22:18] <cbsrobot> hehe [22:18] <cbsrobot> thats a trademark ! [22:18] <cong> for ubuntu 12.10 ? for sure ? [22:19] <aixenv> it works on wheezy.. unbuntu/debian are same family, it'll be fine, just grab the latest stable of all the various packages [22:19] <cong> ok i [22:19] <cong> will try [22:19] <aixenv> the main thing to understand is to have all the pre-reqs done, then do the ffmpeg config line [22:19] <cbsrobot> aixenv, cong: why not http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide ? [22:20] <aixenv> cbs- that's just the doc i wrote that we use @ work - [22:20] <aixenv> and like i said it works [22:20] <cbsrobot> ok - I guess the one on ffmpeg trc works aswelll [22:20] <aixenv> that one on ffmpeg looks good too, less stuff on it than mine [22:21] <aixenv> i would definitely imagine they both work - ya, depends on the functionality you need i guess [22:21] <cbsrobot> yeah and no php related things init too [22:21] <cbsrobot> and it is copy paste friendly :-) [22:21] <cong> which one i should use ? [22:22] <cbsrobot> I'd go with ffmpeg trac [22:22] <aixenv> whichever you want - that one is directly for ubuntu the versoin you have and obvoiusly official since it's on the site, so i'd use that first [22:23] <cong> ok i try this now [22:23] <cbsrobot> I guess aixenv 's is better suited for a specific server implementation [22:23] <cbsrobot> aixenv: whats debian_patches_build-fix.patch ? [22:24] <cbsrobot> is it in faac ? [22:27] <aixenv> thats old, not needed anymore - i need to udpate the doc, only thing i updated last night was the ffmpeg config line which def works [22:27] <aixenv> wont be needed in unbuntu or squeeze [22:27] <aixenv> or wheezy [22:28] <aixenv> @cong im running through the ffmpeg trac doc, letcha know if there are any problems [22:35] <aixenv> @cong ok just finished running through that doc, it works too [22:36] <aixenv> i adjusted my config line to: [22:36] <aixenv> ./configure --prefix=/usr --enable-vdpau --enable-libfdk-aac --enable-libmp3lame --enable-libx264 --enable-libtheora --enable-libvorbis --disable-stripping --enable-x11grab --enable-libvpx --enable-libfaac --enable-nonfree --enable-shared --disable-static --enable-runtime-cpudetect --enable-gpl --enable-libmp3lame --enable-libspeex --enable-libxvid --enable-pthreads --enable-x11grab --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --en [22:36] <aixenv> --enable-version3 [22:36] <aixenv> oops i duped x11grab [22:37] <aixenv> and libvpx heh [22:37] <aixenv> long ass configure line ,other than those dupes looks good [22:42] <cong> yes guys, it works great, thank you [22:45] <aixenv> np - happy to help [22:47] <mistergibson> Open question: is there an option that will allow me to do auto-audio-leveling on a conversion? Thanks in advance. [22:48] <ubitux> you can use ebur128 filter to get a loudness evaluation of the audio, and then use volume audio filter to adjust it [22:49] <mistergibson> ubitux: ok, thanks [23:13] <fatpony> is there a default root password? i just managed to boot my new install and if i could avoid chrooting again i'd be glad... [23:13] <fatpony> oop] [23:13] <fatpony> wrong chan [23:16] <cbreak> sudo -s? [23:19] <fatpony> no user configured yet :) [23:21] <cbreak> isn't that part of the installation with linux? [23:21] <cbreak> ah well... user friendlyness... :D [23:24] <fatpony> to each their own, i'm looking for fun, not user-friendliness :p [23:48] <aixenv> just boot into singleuser and reset your root p/w [23:52] <fatpony> nah that didn't work, still required the root pass [23:52] <fatpony> i rebooted with a live usb anyway [00:00] --- Mon Nov 26 2012

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