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January 2015

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ffmpeg-devel.log.20150110
by burek 11 Jan '15

11 Jan '15

[00:03] <cone-671> ffmpeg.git 03James Almer 07master:da02ee127a21: x86/vf_pp7: port dctB_mmx to yasm [00:25] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:11d8fa5e9d69: ffserver: Use av_malloc_array() [00:36] <cone-671> ffmpeg.git 03wm4 07master:aab74a38b8ba: rm: fix memory leak on init failure [02:23] <cone-671> ffmpeg.git 03Arwa Arif 07master:4df01d5d44cf: avfilter: Factorize the use of norm_qscale in pp7 [04:03] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:bdb136d30218: avfilter/internal: add fixme to ff_norm_qscale() to document obvious scaling issue [04:03] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:7824dc5150c0: avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1" [05:00] <cone-671> ffmpeg.git 03Andreas Cadhalpun 07master:6108485cf70f: configure: enable vsx together with altivec for ppc64el [05:00] <cone-671> ffmpeg.git 03Andreas Cadhalpun 07master:f97f2a3527ea: doc/examples: fix lib math dep for decoding_encoding [05:00] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:6f838dee3cf8: avformat/movenc: move variable to a more local block [05:20] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:b089b582500f: avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1" [05:20] <cone-671> ffmpeg.git 03Andreas Cadhalpun 07release/2.5:07d508e4f55f: doc/examples: fix lib math dep for decoding_encoding [05:30] <cone-671> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.5.3': unknown revision or path not in the working tree. [05:30] <cone-671> Use '--' to separate paths from revisions [05:30] <cone-671> refs/tags/n2.5.3:HEAD: avformat/movenc: move variable to a more local block [14:13] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:35e559ae32f2: avcodec/cavs: Check for av_malloc* failure in ff_cavs_init_top_lines() [14:22] <ubitux> wtf @ #4240 [14:32] <cbsrobot_> ubitux: I guess spammers trying to blend in [14:34] <cone-577> ffmpeg.git 03Paul B Mahol 07master:241f858cf4c3: LICENSE.md: mention mpdecimate and not decimate to be GPL [15:12] <anshul_mahe> Do we have some linklist in ffmpeg utils for using in libavformat [15:24] <relaxed> @4245, "I bonk investigated and instrument prophesy you on the pros of body wraps." haha [15:27] <anshul_mahe> we should have something for these spams they will make searching bugs harder [15:31] <relaxed> maybe instead of searching for bugs you should try these amazing cleanse efx body wraps [15:50] <anshul_m> relaxed: haha [15:56] <aetasx> found this while digging around: http://siliconandlithium.blogspot.com/2014/01/ [16:19] <ubitux> cbsrobot_: but i see no link whatsoever [16:22] <aetasx> its in all the msvc folders within each of his git repos apparently [16:32] <nevcairiel> aetasx: the first line of that post is already wrong, ffmpeg doesnt use autotools :) [16:33] <aetasx> yeah I know I saw it but theres updated posts if you look under the newer ones [16:34] <nevcairiel> seems like an excercise in futility though, what would a generator for a msvc project offer that using make doesnt offer me [16:34] Action: nevcairiel shrugs [16:34] <aetasx> its for building against msvc/icl [16:34] <nevcairiel> so? [16:34] <nevcairiel> i can do that without any external tools [16:35] <aetasx> I doubt thats the case for all the libs hes using [16:35] <nevcairiel> we went through great lengths to actually make that possible [16:36] <aetasx> the little project builder thing builds ffmpeg as well as all its dependencies [16:39] <nevcairiel> i guess it may add some covenience for some use-cases, but its all possible without special tools either way [16:40] <aetasx> the project I work on does it within msys2 which gets to be a somewhat pain in the ass a lot of times [16:40] <aetasx> not to mention most of them are all build for gcc [16:40] <nevcairiel> i only use msvc builds for development/debugging anyway, since for production the performance impact is too big [16:40] <aetasx> yeah thats why Im doing it against icl. MSVC doesn't really perform that great otherwise [16:40] <nevcairiel> its unfortunate, but msvc builds are quite a bit slower [16:40] <nevcairiel> i dont trust icl to generate working code tbh :p [16:41] <aetasx> Im sure if it wasnt working for this, he wouldnt bother doing it :p [16:41] <compn> i think you guys may have misunderstood what the blog post does [16:41] <anshul__> does any one know about hls->start_pts, it looks like we have forgotten to update it, or may be I dont undestand its purpose [16:41] <aetasx> it lets me be lazy [16:41] <compn> it looks like he removes the dependency of using mingw/gcc on windows all together if you are just using vs [16:41] <aetasx> correct [16:42] <nevcairiel> aetasx: you need to disable a bunch of optimizations in icl which are on by default so it doesnt miscompile ffmpeg (configure does this for you) ... tells you a lot about their compiler :p [16:42] <nevcairiel> compn: you d ont need mingw/gcc on windows at all if you compile with vs [16:42] <nevcairiel> all you need is gnu make and a sh shell [16:42] <aetasx> if this guy couldnt get it to build correctly, he wouldnt continue to be updating it [16:43] <nevcairiel> the fact remains that for a very expensive compiler, icl is quite crappy :P [16:43] <nevcairiel> its popular in the "ricing" crowd, since apparently it claims to produce faster code [16:44] <aetasx> yeah, for intel platforms it does [16:44] <nevcairiel> .. at the cost of miscompiling occasionally [16:45] <nevcairiel> it may as well work well, but i wouldnt pay for it [17:09] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:1673436d85d6: avfilter/vf_pp7: use av_malloc_array() [17:09] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:0b52bdfb3b47: avfilter/vf_pp7: Add GBRP pixel format [19:06] <compn> nevcairiel : well unxutils project (sh and make .exe files, amongst others like grep) isnt very well known. [19:06] <compn> nevcairiel : also, if ffmpeg can be built like you say so, can you write down those instructions in our documentations ? [19:07] <compn> do we have a vs build guide ? [19:07] <JEEB> yes, in the documentation [19:07] <JEEB> and Daemon404 even made a vimeo clip [19:08] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:a4d3cf10b2ec: avcodec/diracdec: Check slices malloc and propagate error code [19:15] <BBB> if people still cant build ffmpeg with vs nowadays, they likely wouldnt be able to build it with gcc under linux either [19:16] <JEEB> it's the usual crowd that expects that they just need to double-click a visual studio solution and ctrl+alt+f7 (rebuild) [19:21] <wm4> JEEB: I'd expect that too [19:21] <wm4> it's equivalent to ./configure && make [19:21] <wm4> _if_ there's a vs project file [19:22] <wm4> though I never ever had a case where this actually worked with any vs project file with any project [19:22] <wm4> not even MS examples [19:22] <wm4> (confusing) [19:23] <nevcairiel> I wanted to say my project works like that, but i would be lying, since you need to compile ffmpeg first :D [19:24] <Daemon404> JEEB, the clip is outdated [19:24] <Daemon404> you dont need c99conv at all for msvc2013 [19:24] <JEEB> yeah [19:25] <Daemon404> also the only reason i made that clip is that one guy who kept emailing me was *so dumb* [19:25] <Daemon404> he couldnt follow step-by-step [19:25] <Daemon404> side note: if i google how to do something and i find videos but no text, i rage [19:25] <compn> nevcairiel : liar! :p [19:26] <compn> Daemon404 : you rage because you are basically watching a 5 minute video for the guy to read off 8 sentences of instructions? :P [19:26] <Daemon404> yea [19:26] <Daemon404> 10 seconds vs 5 minutes of my lie [19:26] <Daemon404> life* [19:31] <wm4> Daemon404: agreed, fuck video "tutorials" [19:35] <aetasx> how would you guys compare transcoding a movie in x264 vs ffmpeg speed-wise? [19:35] <aetasx> like to squeeze something that I've already run through ffmpeg [19:39] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:ebe3a41ea3b9: avcodec/dvbsubdec: Add some av_malloc() failure checks [19:39] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:1bf747ae84fc: avcodec/dvbsubdec: Return proper error codes from more functions [19:39] <cone-577> ffmpeg.git 03Michael Niedermayer 07master:0898a6d4e4b6: avcodec/ratecontrol: Check for av_malloc*() failures [20:25] <cone-577> ffmpeg.git 03wm4 07master:09b4ad15681b: vp9: avoid infinite loop with broken files [21:20] <cone-577> ffmpeg.git 03Matt Oliver 07master:87deebe11a3a: avcodec/libvpxenc.c: Fix linking with codec interface with msvc. [22:53] <michaelni> ubitux, is the closed caption patch from a few days ago ok to comit or was there some issue remaining ? [22:59] <wm4> where is that patch? I want to take a look too [23:46] <wm4> ubitux: what this cc patch would need it a way to realize the cc grid layout in ASS... [23:46] <wm4> don't know if there's a good way to enforce this [23:47] <kierank> does the cc patch match the test vectors? [23:48] <kierank> if not then it shouldn't be committed [23:53] <wm4> who knows [23:53] <wm4> and some advanced stuff like formatting is not supported (there's a "todo" comment in the patch) [00:00] --- Sun Jan 11 2015

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ffmpeg.log.20150110
by burek 11 Jan '15

11 Jan '15

[00:18] <grepper> damn, the fast seek doesn't work with avconv 11.1 , I get black frames from mpg. Too bad I decided to support both avconv and ffmpeg [00:18] <c_14> check the version header and fallback? [00:19] <c_14> If it says 'Copyright the FFmpeg Developers' it's FFmpeg [00:19] <c_14> I mean, you have the fallback anyway for old versions. [00:24] <grepper> yeah, I might just have to code yet another work-around [00:31] <grepper> 3 fallbacks needed really, 1. use transcode 2. use 'select' filter if new enough 3. use -ss -i and loops if not avconv 4. slow seek using loops [00:31] <grepper> heh, no wonder this script has become so long [00:41] <JEEB> fast seek as in container side seek? [00:41] <JEEB> that should work just fine in avconv as well, you just have to put it before -i instead of after IIRC [00:42] <c_14> JEEB: <grepper> damn, the fast seek doesn't work with avconv 11.1 , I get black frames from mpg. Too bad I decided to support both avconv and ffmpeg [00:42] <JEEB> yeah, mpeg-ts/ps can be problematic depending on what exactly you're doing. [00:43] <JEEB> if you want to get it fixed on the other side I recommend you report it there if you have the time [00:44] <JEEB> c_14, I mostly missed the "black frames with mpg (I guess mpeg-ps)" part [00:44] <JEEB> I think VFR Maniac or so added some timestamp stuff to raw video streams which might be helping with a container like that (which doesn't by itself contain hints for seeking) [00:45] <grepper> maybe I should stick with the select filter for mpg then ? [05:33] <bofh> Hello there! I've created the MP3 file using ffmpeg, and now I can see that "ffprobe file.mp3" returns: "start: 0.025057" [05:34] <bofh> what us "start" and why it is > 0? [05:46] <bofh> Duration: 00:00:54.00, start: 0.025057, bitrate: 128 kb/s to be more precise [05:46] <bofh> I don't really get what is that "start" [05:52] <BtbN> propably when the audio starts [05:53] <BtbN> before that, there's just silence [05:53] <bofh> well, but there's nothing like silence [05:53] <BtbN> 0.02 seconds isn't that long [05:54] <bofh> yeah, but when merging with video [11:52] <meorom> hi all! [11:52] <meorom> now I can compile ffmpeg for Windows success [11:53] <meorom> but in lib folder, I only see *.a, it doesn't create *.lib [11:53] <meorom> if I compile dynamic lib, it still create *.dll [11:54] <meorom> I don't know what I am missing ? [11:54] <meorom> I'm also thanks to c_14 [11:55] <meorom> I use config: --toolchain=msvc --target-os=win64 --arch=x86_64 --prefix=build [12:44] <Fjorgynn> :) [12:44] <Fjorgynn> merzo: [12:44] <Fjorgynn> wrong [12:48] <meorom> ? [12:48] <meorom> could you pls tell me more ? [12:49] <meorom> ah, sorry [12:49] <meorom> wrong person :) [12:51] <Mavrik> so what exactly are you trying to do? [12:52] <meorom> me ? [12:54] <Mavrik> yes. [12:55] <meorom> I follow http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=1309 [12:56] <meorom> now I want to build static lib, but it only create *.a, not *.lib [12:57] <meorom> I'm also check https://trac.ffmpeg.org/wiki/CompilationGuide/MSVC [12:57] <Mavrik> 1.) Those instructions are horribly out of date. [12:57] <Mavrik> 2.) You still didn't tell what your end goal is. Do you need the binary? Static lib? Dynamic lib? [12:57] <meorom> my goal is build static lib [12:57] <Mavrik> 3.) You pasted different guides and you didn't say which compiler do you want to use :) [12:58] <Mavrik> meorom, well, check if you have MSVC2013 first [12:58] <Mavrik> then add "--enable-static" to configure right? :P [13:00] <meorom> yes, actually I'm using MSVS 2013, and compiler MSYS MingW [13:01] <Mavrik> ugh. [13:01] <Mavrik> No, either you're using one or the other. [13:01] <Mavrik> Anything else will cause you a huge pain. [13:01] <Mavrik> so either mingw/gcc or msvc compiler [13:02] <meorom> I mean I follow exactly in https://trac.ffmpeg.org/wiki/CompilationGuide/MSVC, only different thing I'm using MSVC 2013 [13:03] <meorom> I've just finished build with --enable-static, but it's still not create *.lib, it's only has *.a [13:04] <Mavrik> .a is a static lib [13:04] <Mavrik> it's created by gcc compiler. [13:04] <Mavrik> if you get .a instead of .lib you're using the wrong compiler. [13:05] <Popara> Hello, why with -f mpegts ... the bitrate is visible like bitrate=2400kbps under stats and when i output it via HLS it says bitrate=N/A [13:05] <meorom> so, can I check which compiler is using ? [13:06] <meorom> and you said that this document is out of date, could you show me newer doc ? [13:13] <Mavrik> Popara, probably because HLS muxer doesn't have that implemented. [13:13] <Mavrik> meorom, the MSVC compilation on Wiki or FFmpeg on homepage should be up to date [13:13] <Mavrik> not the zeranoe link [13:26] <meorom> Mavrik: but if I use "--enable-shared", it can create *.dll and *.lib [13:27] <Mavrik> because gcc creates .dlls when using shared configuration and the glue library. [13:28] <meorom> okey, so could you show me a way to check this ? [13:29] <meorom> in doc "FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility and wrapper, or with MSVC 2013 natively." [13:29] <meorom> so, how can I build "MSVC 2013 natively", could you tell me more ? [13:30] <meorom> maybe here I'm using MSVC 2013, so fail result. [13:32] <meorom> I'm a noob, so you shouldn't angry :) [13:35] <Mavrik> Ok, again. MSVC Linker will create a .lib static library, gcc will create a .a [13:35] <Mavrik> if you get a .a, your ffmpeg is being compiled with gcc not msvc [13:35] <Mavrik> I can't really help you more than that because I'm not at your machine. [13:35] <Mavrik> If you're getting a .dll you're compiling in shared library configuration. [13:39] <Popara> Mavrik how hard it would be to add the bitrate into the hls? :/ [14:23] <ribasushi> greetings [14:24] <ribasushi> is there something like "matroska for streaming" protocol? the goal is to be able to take a random source and -c copy it to something that will take any stream and wrap it apropriately [14:24] <ribasushi> mpeg-ts and rtp are both limited in what they will take [14:33] <ribasushi> the silence is unnerving, hopefully it's simply too early, instead of "no such thing" :) [14:38] <Mavrik> hrmf [14:38] <Mavrik> don't remember any, MPEG-TS is the standard for streaming [14:41] <sfan5> ribasushi: it looks like matroska does support streaming http://matroska.org/technical/streaming/index.html [14:41] <ribasushi> let me step back a bit for a bigger picture [14:42] <ribasushi> the current setup I have is: ffmpeg -i <some hardware> -c copy -f matroska <diskstore.mkv> -c copy -f matroska /dev/stdout | ffmpeg -i /dev/stdin -f rtp rtp://127.0.0.1:30321 [14:43] <ribasushi> the rationale is to be able to "tune into" what is currently being recorded, while using virtually 0 cpu otherwise (hence the copy's) [14:43] <ribasushi> the problem comes when <some hardware> spews out non strict-conformant streams [14:44] <ribasushi> what I am looking for is a reliable "tune-in" mechanism [14:44] <ribasushi> neither rtp nor mpeg-ts give me that [14:45] <Mavrik> well. [14:45] <Mavrik> Nothing much you can do unless you tighten up your requirements. [14:47] <ribasushi> hm hm hm... but that's so uncool ;) [14:47] <ribasushi> there got to be some generic format to consume "random frames from the aether" [14:48] <Mavrik> you're not streaming random frames [14:48] <Mavrik> you're streaming encoded data which can be delimited in bunch of ways and multiplexed in bunch of ways [14:49] <Mavrik> and that encoded data tends to have predispositions, like headers carrying data that's critical for decoding ;) [14:50] <ribasushi> nod [14:51] <ribasushi> aight will ponder more, thanks [14:51] <Mavrik> but I'm rather curious what kind of hardware outputs stuff that's not MPEG or H.264 :) [15:33] <bofh> Hello again [15:33] <sfan5> hi [15:33] <bofh> I need some advice of how to remove that non-zero "start" something in generated mp3 [15:34] <bofh> ffprobe 1.mp3 [15:34] <bofh> Duration: 00:00:54.00, start: 0.025057, bitrate: 128 kb/s [15:34] <bofh> what is that "start" and why it is non-zero? [15:36] <shadowchancellor> Hey guys I've just managed to convert some LPCM to WAV, however, it seems the LPCM is muxed with some kind of video. How would one go about extracting and encoding the video portion? [15:37] <relaxed> bofh: ffmpeg -i input.mp3 -ss 0.025057 -c copy out.mp3 [15:38] <relaxed> shadowchancellor: why not encode the video and copy the audio without extracting anything? [15:38] <sfan5> shadowchancellor: ffmpeg -i input_file -an -c:v video_codec output_file [15:38] <sfan5> video_codec can be "copy" if you just want to extract the video [15:38] <bofh> relaxed: already tried that [15:39] <bofh> the resulting file has exactly the same start [15:39] <bofh> relaxed: not really the same, Duration: 00:00:53.94, start: 0.011995, bitrate: 128 kb/s [15:43] <relaxed> can you put it up somewhere? [15:44] <bofh> sure [15:47] <bofh> relaxed: http://s3.amazonaws.com/animatron-snapshots-dev/8db3b0540b6f95cdef44ae07.mp3 [15:48] <Popara> Hey guys can i ask something, for a bug that WASN'T exists in FFmpeg 2.4.x (it exists in FFmpeg 2.5.x) how much time it will take to be fixed in the new version? It should be something simple. just asking nothing more :) [15:48] <rcombs> link to the ticket? [15:49] <Popara> https://trac.ffmpeg.org/ticket/4247#ticket [15:49] <rcombs> it's not really possible to provide an estimate on when something will be fixed without knowing anything about the issue [15:49] <Popara> and this one [15:49] <Popara> https://trac.ffmpeg.org/ticket/4247 [15:49] <Popara> i made them now [15:49] <Popara> yes i'm just saying it because in ffmpeg 2.4.x it was fine [15:51] <rcombs> it'll be kinda hard to work on that without a media sample [15:52] <Popara> A sorry wrong ticket ( the second one is this: https://trac.ffmpeg.org/ticket/4243 ) [15:52] <Popara> yeahi write in the ticket so its fine [15:52] <Popara> About the sample, i can't make it, because when i get the sample with the new version it closes, when i get it with ffmpeg 2.4.x it seems that the sample is already fixed so it works on ffmpeg 2.5.x as well [15:53] <Popara> Also, for me it appears to be a problem with the -re argument mostly [15:54] <Popara> I also tried with FFmpeg 2.5.0 sam results. So something changed between 2.4.5 and 2.5.0 and i don't know what [15:55] <rcombs> why are you passing -analyzeduration 15000000? [15:55] <Popara> it just general solution because i use it for all of my streams [15:55] <Popara> some streams require larger analyze duration [15:55] <Popara> but as i said even in this case, in ffmpeg 2.4.x was fine [15:55] <rcombs> provide logs with -v verbose [15:55] <Popara> ok [15:56] <Popara> i will do that now [15:57] <rcombs> and yeah, you need to provide a sample that reproduces the issue [15:57] <rcombs> otherwise nobody will be able to do anything about it [15:57] <relaxed> bofh: hmm, I'm not sure how to set it to zero. Are you editing something with it? [15:58] <Popara> i will try maybe to do something using wget to get the stream since it's over HTTP [15:58] <Popara> and then parsing into ffmpeg i will see it now [15:59] <bofh> relaxed: I'm adding it to the video stream [15:59] <bofh> and somehow it is not in sync [16:03] <xreal> Is it good to store anything mkv, if mkv is readable from my player? [16:03] <xreal> anything in* [16:03] <relaxed> bofh: what about, ffmpeg -i 8db3b0540b6f95cdef44ae07.mp3 -ss 0.025057 -y output.flac [16:04] <Popara> rcombs that is really so strange. I grabbed a sample using WGET, i add it into FFmpeg to re-parsing it into MPEGTS. With OR Without -re it works (because i guess it's a file) . SO, what i understand is that happens when the connections is remotly and when i use -re as well [16:05] <relaxed> bofh: without the -ss it will remove the start delay, too [16:05] <Popara> So somthing happens with the connection when i use -re and messes the things up. Now the only way to provide the sample is to give the direct link [16:06] <bofh> hm [16:06] <bofh> let me try something [16:07] <anshul_mahe> in ffmpeg hls_wrap is not working [16:08] <c_14> anshul_mahe: what is it doing, what are you expecting it to do, what version are you running, full commandline and output on a pastebin please [16:10] <xreal> relaxed: do you know, if mp3->flac is done in 32-bit mode? mp3 is bitless, flac can handle up to 32 "only". [16:11] <anshul_mahe> I here is http://pastebin.com/JhFT7HHc [16:11] <bofh> relaxed: do you know if I could play videostream from mp4 and audiostream from external file, and start playing of audio from some time? [16:12] <anshul__> sry for noise I wanted hls_list_size i was using hls_wrap [16:18] <relaxed> bofh: where is the delay happening with the audio/video? does the audio start late? [16:19] <relaxed> xreal: mp3 and flac support s16 [16:20] <relaxed> (and s32 too) [16:28] <xreal> relaxed: MP3 is floating, you can push 32 bit into it without a problem. [16:31] <bofh> relaxed: audio seem to start late [16:31] <bofh> so I just want to cut off that initial delay of 0.02 seconds and see what will happen [16:31] <xreal> The FLAC format supports from 4-bit to 32-bit sample sizes. [16:31] <xreal> libFLAC does not support encoding or decoding anything above 24-bit. [16:33] <relaxed> ffmpeg's flac encoder supports 32bit [16:33] <xreal> relaxed: but 32-bit decoding is only supported ifmid/side is disabled due to bitstream reader limitations. [16:34] <xreal> relaxed: so, mp3->flac will be done in 32bit? nice. [16:36] <relaxed> bofh: ffmpeg -i video.mp4 -itsoffset -0.02 -i audio.mp3 -map 0:v -map 1:a output [16:36] <relaxed> that might work [16:39] <xreal> Damn, my reported bug is reproduced by developers, but open for 9 months now! [16:39] <relaxed> xreal: it's done in 16 bits. [16:40] <xreal> relaxed: that really suxx. I'll better use sox for this or do it with piping. [16:40] <relaxed> I don't know what you're going on about, it's completely lossless [16:41] <xreal> relaxed: I talked about it on hydrogenaudio about 10 years ago. They suggested to do it in 24 or 32 bit. [16:43] <relaxed> I still don't know what you're saying. If there's an issue with ffmpeg you should file a bug report, or point to one if you've already filed it. [16:44] <xreal> relaxed: I don't file bugs anymore. They don't get resolved after all. [16:47] <relaxed> If you're against filing them then there's no point in discussing them here. [16:48] <durandal_1707> xreal: which bug? [16:48] <relaxed> there is no bug [16:48] <bofh> relaxed: hmm, it says something about the AAC encoder [16:48] <bofh> which isn't my case [16:49] <relaxed> pastebin [16:51] <xreal> durandal_1707: https://trac.ffmpeg.org/ticket/3576 [16:53] <bofh> relaxed: http://pastie.org/private/oxmmgebszp8ygtyarisdq [16:55] <relaxed> ok, do you want the video from the first input and the audio from the mp3? If so, -map 0:v -map 1:a only use those streams and omit the aac stream. [16:55] <bofh> it seems that mp4 contains the AAC stream [16:55] <bofh> may it happen that the AAC stream is different from mp3 in terms of timing? [16:56] <relaxed> I have no idea, play it and see [16:57] <relaxed> but you should be able to sync the mp3 with itsoffset [16:58] <bofh> yeah [16:59] <bofh> ffmpeg -i 993db15476e8b2cd1674d064.mp4 -itsoffset -0.02 -i 993db15476e8b2cd1674d064.mp3 -map 0:v -map 1:a test.mp4 [17:00] <bofh> [aac @ 0x7f26b37bbb40] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [17:00] <xreal> Now durandal is gone :D [17:00] <bofh> weird [17:01] <xreal> bofh: -c:a copy [17:01] <relaxed> oh, it's converting the mp3 to aac. Use -c copy to avoid that. [17:01] <xreal> relaxed: -c copy? isn't it -c:a copy? [17:01] <relaxed> well, -c copy if he want to copy both the audio and video [17:01] <relaxed> wants* [17:01] <xreal> relaxed: does -c also do -c:s ? [17:02] <relaxed> -c copy will copy everything you have mapped from the input [17:02] <xreal> ah ok [17:06] <bofh> okay, that worked [17:06] <bofh> still there's some sync problem, but perhaps not related to ffmpeg [17:08] <relaxed> try playing with the itsoffset number. it's time for me to hit the sack. [17:56] <bofh> relaxed: ok, jfyi - I've found the problem and it is not related to ffmpeg anyhow :) [17:57] <bofh> anyway, thanks for the help! [17:57] <bofh> it helped me to narrow the scope of the issue and eventually nail down the real issue [18:00] <allengates> hi [18:56] <xreal> Children are like Windows Vista: not that far developed and hard to handle. [19:01] <shadowchancellor> Anyone know what the 'muxing overhead' info that gets logged when converting pcm to wav is? [19:01] <shadowchancellor> For instance, mine logs 'muxing overhead: 0.026929%' [19:02] <justinX> it is how much extra space a file takes compared to the involved video and sound [19:03] <shadowchancellor> so I can assume that ~.02% is extraneous and could be removed? [19:03] <justinX> space consumed by headers and the way it mixes the parts togheter so sound is near the video for the same time in the file etc... [19:04] <justinX> it can't be removed unless you want to play the video separately from the sound :-D [19:05] <justinX> in videos I mean. for pure sound files... I guess it is just headerS? [19:05] <justinX> telling what sound file format it is I suppose? [19:07] Action: justinX imagines a world without that info.. where everything would have to be packed into the filename instead [19:30] <fvjfadskj> hi [19:30] <fvjfadskj> i'm piping an image stream to ffmpeg using image2pipe [19:30] <fvjfadskj> and taking timestamps with wall clock option [19:30] <fvjfadskj> and meanwhile taking sound from directshow [19:30] <fvjfadskj> sound and video are out of sync [19:30] <fvjfadskj> how to fix? [20:09] <fvjfadskj> nobody knows? [00:00] --- Sun Jan 11 2015

1 0

ffmpeg-devel.log.20150109
by burek 10 Jan '15

10 Jan '15

[00:00] <compn> tell that to cygwin1.dll :P [00:01] <compn> >also 20 other cygwin dlls [00:07] <nevcairiel> this is why noone uses cygwin for shit anymore [00:07] <aetasx> exactly [00:08] <nevcairiel> and most people link the mingw extra-crt staticly [00:08] <nevcairiel> although i've seen that not be the case, and thats ugly as hell <.< [00:10] <aetasx> vlc ships with libgcc dll [00:10] <aetasx> at least their nightlies do [00:11] <j-b> releases don't [00:11] <j-b> of course [00:12] <aetasx> dunno, its been awhile since I've used one. there was some feature or something I upgraded to 2.2 to get but cant recall what [00:13] <aetasx> oh right, it couldnt play x265 within mp4s [00:15] <aetasx> that was before I realized it was too damn slow to encode to justify using it [00:15] <j-b> it got better, no? [00:16] <aetasx> I cant recall but I remember doing medium encodes and it only going at something like under 60fps [00:17] <aetasx> for video thats 320x240 [00:19] <aetasx> x264 could hit the same size, same quality much faster [00:19] <nevcairiel> x265 is still under fast development, so any opinions formed half a year ago may already be quite invalid :d [00:20] <aetasx> this was months ago [00:20] <aetasx> but I still consider it in active development so its not like I can expect the same speed results [00:20] <nevcairiel> half a year is "6 months", so it would qualify! [00:21] <aetasx> would 2 months still qualify? :p [00:21] <nevcairiel> dunno, possibly, i havent been checking on its progress that much over the last time [00:22] <aetasx> x264 has been out for quite awhile so its not really unexpected that it would be much slower [00:23] <aetasx> it may just end up being for more professionally done video than for the random stuff I need [00:23] <jamrial> the bulk of optimizations on x265 has been fox avx2, so unless you have a haswell you're out of luck [00:23] <aetasx> nah, nehalem [00:50] <cone-173> ffmpeg.git 03Carl Eugen Hoyos 07master:cd3405282c70: Skip Exif data when auto-detecting jpeg images. [04:04] <ac_slater_> sigh. Hey all. My libavformat shared library output is missing `ff_isom_write_hvcc`. Oh, ffmpeg 2.5 [04:08] <ac_slater_> I've tried enabling all of the HEVC features but no go. I'll paste a config.log [04:12] <ac_slater_> config.log ... http://sprunge.us/FhIF [04:26] <jamrial> as you were told last night that function is not related to either hevc decoder or hevc de/muxer. it's only needed for mov, matroska, webm and rtp muxers [04:27] <ac_slater_> jamrial: I lost connection for a bit last night so I must have missed it [04:28] <ac_slater_> jamrial: sorry mate. [04:28] <ac_slater_> jamrial: so I narrowed it down to one VERY specific thing. If you have `--enable-protocol=rtp` but not a muxer or demuxer for RTP, this will happen [04:28] <ac_slater_> is that intended? [04:29] <jamrial> look for CONFIG_RTP_MUXER in libavformat/Makefile and add hevc.o at the end of the list of objects it should compile [04:30] <jamrial> unlike in git master, that's not in ffmpeg 2.5 [04:30] <jamrial> it's probably the source of your problem [04:33] <jamrial> see if it builds after that. if it does then fccfc22d1 should be backported [04:35] <ac_slater_> jamrial: awesome ill report back [04:40] <ac_slater_> jamrial: hmm well that didnt do it. I'll find the difference with this. But for now (since Im in a rush) adding matroska mux/demux fixes the build for me [04:45] <jamrial> eh, that commit should be backported anyway [04:45] <jamrial> michaelni_: ok with the above? [04:46] <michaelni_> jamrial, fccfc22d1, sure [04:47] <jamrial> ok [04:51] <cone-161> ffmpeg.git 03Martin Storsjö 07release/2.5:93cbdcd4d30c: libavformat: Build hevc.o when building the RTP muxer [05:05] <cone-161> ffmpeg.git 03Peter Ross 07master:624384503643: avformat/aiffdec: improve readability [05:05] <cone-161> ffmpeg.git 03Michael Niedermayer 07master:d1c6b7b2d1ef: configure: fix convoluted shlib code [12:59] <cone-671> ffmpeg.git 03wm4 07master:b88e80589bd1: vp9: fix parser return values in error case [14:09] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:2959a612326c: avformat/utils: Use avio_closep() to avoid leaving stale pointers in memory [14:09] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:643dee846add: avformat/sapenc: Use avio_closep() to avoid leaving stale pointers in memory [14:09] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:ef3a6a48b56d: avformat/tee: Simplify code by using avio_closep() [15:21] <anshul_mahe> is there any way in command line to disable bug of perticular codec [15:24] <anshul_mahe> Its hard to find my codc(what I implementing) related debug message when other codecs overflow my terminal [15:28] <wm4> well, the log callback has a context [15:28] <wm4> you can get the AVCodecContext from that I think [15:29] <wm4> but you might miss messages, because av_log is often called with no context [15:29] <wm4> or with a different context, even if it should be related to the codec you're debugging [15:31] <compn> anshul_mahe : you can force your codec ? -vcodec codecname [15:31] <compn> or not sure what you want [16:07] <anshul__> wm4: I am greping my codec context name, but this thing is taking long time, I was looking for some better trick [16:09] <anshul__> its not that I am missing them on my screen, I redirect and take them in file, but this long procedure is slowing me down [16:55] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:323355413dbe: tools/uncoded_frame: Use avio_closep() to avoid leaving stale pointers in memory [16:55] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:04091552616e: ffmpeg: remove unneeded NULL pointer check [16:55] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:b1d223301f03: avformat/hdsenc: Remove redundant NULL pointer checks [17:18] <cone-671> ffmpeg.git 03Anton Khirnov 07release/2.4:883795fb0fac: h264: restore a block mistakenly removed in e10fd08a [17:18] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.4:33c8bb2c6a87: Merge commit '883795fb0face00a07349fbd32f4775431a9f30d' into release/2.4 [17:32] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:bcd7f35717db: avformat/segment: Use av_freep() avoid leaving stale pointers in memory [17:32] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:68c63470895d: avformat/cdxl: Fix integer overflow of image_size [17:32] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:eac0451e4738: avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream [17:32] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:5bb31e856d7c: avformat/flvdec: Increase string array size [17:32] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:252ba4a9259f: avfilter/vf_sab: fix filtering tiny images [17:32] <cone-671> ffmpeg.git 03wm4 07release/2.5:e0a12b3dc3a2: avcodec/dvdsubdec: fix out of bounds accesses [17:32] <cone-671> ffmpeg.git 03Dale Curtis 07release/2.5:022bfd3dd47c: mov: Avoid overflow with mov_metadata_raw() [17:32] <cone-671> ffmpeg.git 03Dale Curtis 07release/2.5:9143ab0e5a75: mov: Fix overflow and error handling in read_tfra(). [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:25312a427bda: avformat/mov: fix integer overflow in mov_read_udta_string() [17:33] <cone-671> ffmpeg.git 03Dale Curtis 07release/2.5:0ce35b8ce8b9: mov: Fix negative size calculation in mov_read_default(). [17:33] <cone-671> ffmpeg.git 03Paul B Mahol 07release/2.5:c7a2ac6b6bd5: lavfi: check av_strdup() return value [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:0d3a07852c3c: avformat/mov: use av_freep() to avoid leaving stale pointers in memory [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:2791eba1d78a: avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:918ed73b70a9: avformat/utils: Clear pointer in ff_alloc_extradata() to avoid leaving a stale pointer in memory [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:b3c082412cda: cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file() [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:d0041dc8c4d7: avformat/mov: Fix mixed declaration and statement warning [17:33] <cone-671> ffmpeg.git 03James Almer 07release/2.5:b834dc14da08: configure: bump year [17:33] <cone-671> ffmpeg.git 03wm4 07release/2.5:3d71024f8ab3: avcodec/dvdsubdec: error on bitmaps with size 0 [17:33] <cone-671> ffmpeg.git 03wm4 07release/2.5:b895e29941f0: avcodec/dvdsubdec: fix accessing dangling pointers [17:33] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:f9bbc26e691f: ffmpeg: Clear error message array at init. [17:33] <cone-671> ffmpeg.git 03wm4 07release/2.5:3032291b3a88: vp9: fix parser return values in error case [17:35] <cone-671> ffmpeg.git 03Arwa Arif 07master:a299cd5ab356: lavfi: port mp=pp7 to libavfilter [18:19] <cone-671> ffmpeg.git 03Michael Niedermayer 07release/2.5:a7dd37169c9a: Update for 2.5.3 [20:46] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:727cb2bda9df: avcodec/ass: Use av_realloc_array() [20:46] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:aefb99f2c055: avcodec/ass_split: use *alloc_array() [21:25] <cone-671> ffmpeg.git 03Anton Khirnov 07master:7e4e010b80e7: mpeg4audio: check the init_get_bits() return value [21:25] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:1cf0f2798596: Merge commit '7e4e010b80e76862e83afbd41c25d50e72f0b44c' [21:36] <cone-671> ffmpeg.git 03Anton Khirnov 07master:51da7d02748c: matroskaenc: refuse to write AAC without valid extradata [21:36] <cone-671> ffmpeg.git 03Michael Niedermayer 07master:f95cd5a235ba: Merge commit '51da7d02748cc54b7d009115e76efa940b99a8ef' [22:52] <compn> whoa [22:52] <compn> intel and amd both contributing to ffmpeg at the same time :D [22:53] <compn> plus mips before. how many other cpu manf have submitted patches? :) [00:00] --- Sat Jan 10 2015

1 0

ffmpeg.log.20150109
by burek 10 Jan '15

10 Jan '15

[00:43] <mikos_> hi there [00:43] <mikos_> i m running into an issue i fugured i would ask here in case anyone knows why: [00:44] <mikos_> so i m generating jpg and piping them into a named pipe [00:44] <mikos_> and have it piping to ffmpeg this way: [00:44] <mikos_> ffmpeg -y -f jpeg_pipe -s $SIZE -r 24 -i $FIFO -c:v copy out.mp4 [00:45] <mikos_> it works but for some reason it waits to generate ~100 jpg images before ffmpeg starts rendering [00:45] <mikos_> does anyone know why it doesn t start rendering as soon as the first jpg gets piped? [00:46] <mikos_> help would be apreciated :) [01:08] <c_14> x264 ? [01:14] <c_14> http://mewiki.project357.com/wiki/X264_Encoding_Suggestions#Encoder_latency [03:44] <grepper> https://trac.ffmpeg.org/wiki/FilteringGuide needs to use an array rather than 'eval' to keep #bash happy. [03:52] <c_14> hmm, not sure exactly what you mean [03:55] <c_14> Can you give me an example or something of what you mean? [03:55] <grepper> codec=(-vcodec libx264 -pix_fmt yuv420p -b:v 700k -r 25 -maxrate 700k -bufsize 5097k); ffmpeg -i "$infile" "${codec[@]}" ... etc [03:58] <c_14> http://sprunge.us/HJCe <- like this? [04:00] <grepper> quoting the array will keep the quoted elements separate, which is what you want in case of paths or file names with spaces etc. [04:01] <grepper> though it is not as relevant there [04:02] <c_14> Right, but other than that it's fine? [04:02] <grepper> not sure why you escaped the enclosing quotes [04:03] <c_14> The enclosing quotes around the filtergraph? [04:04] <grepper> yeah [04:04] <c_14> Right, just didn't get rid of those. [04:04] <c_14> Did now. [04:04] <grepper> okay [04:07] <c_14> Updated the pagee [04:07] <c_14> -e [04:07] <grepper> alternatively I guess you could escape the single quotes and put it in a plain var, then use the array for the full command [04:08] <grepper> probably that would be more common, ie. using the array for the full command [04:08] <grepper> but either is better than eval :) [04:09] <grepper> c_14: do you know how to use shell variable within a filter that needs to be single quoted? [04:10] <grepper> like: -vf "select='lt(mod(n\, 8991), 60)',setpts='N/(29.97*TB)'" <- using vars for the interval (8991) , # of frames (60) and fps [04:10] <grepper> obviously the vars won't expand with single quotes, tried swapping the single and double quotes unsucessfully [04:11] <c_14> drop the quotes and escape everything with '\' [04:11] <grepper> tried escaping the brackets and using double quotes [04:11] <grepper> oh [04:11] <grepper> hm, maybe I messed that up then [04:11] <grepper> thanks for the confirmation, I'll try again :) [04:12] <c_14> escaping can be very funky sometimes [04:12] <c_14> I've had filtergraphs with 5 layers of escapes before... [04:13] <grepper> heh [04:13] <grepper> yeah, I hate reading bash that is escaped all to hell, its like reading a complicated sed line [04:14] <relaxed> can't you call filtergraphs from a file? [04:14] <c_14> You won't get variables there though [04:15] <relaxed> ugh, right [04:16] <c_14> I've actually sometimes resorted to small python scripts over bash. Just because it saves me a layer of escapes. [04:24] <relaxed> yeah, or run sed/awk on the file but then you're using bash [04:25] <relaxed> I guess making filters useful and easy to use with a shell are mutually exclusive [04:29] <grepper> select1=("lt(mod(n\, $interval), $frames)") ; setpts1=("N/(${fps}*TB)"); ffmpeg -i "$infile" -vf select="'${select1[@]}'",setpts="'${setpts1[@]}'" -y %08d.png [04:29] <grepper> that worked with arrays okay [04:30] <c_14> mhm right, I keep forgetting that variables inside single quotes inside double quotes get expanded [04:32] <c_14> So you could probably -vf "select='lt(mod(n, $interval), $frames),setpts='N/(${fps}*TB)' [04:33] <c_14> probably anyway [04:36] <grepper> nope, tried that one [04:36] <c_14> mhm, k [04:39] <grepper> oh, it does work, was missing a quote mark [04:39] <grepper> -vf "select='lt(mod(n, $interval), $frames)',setpts='N/(${fps}*TB)'" [04:40] <c_14> oh, missed that [04:40] <grepper> bah, computer languages are soo picky, spoiled brats [04:41] <c_14> Computers do what you tell them to do, not what you want them to do. [04:42] <grepper> shame that [04:52] <grepper> cmd=(ffmpeg -i "$infile" -vf "select='lt(mod(n\, $interval)', setpts='N/($fps*TB)'" -y %06d.png) [04:56] <grepper> oops [04:56] <grepper> cmd=(ffmpeg -i "$infile" -vf "select='lt(mod(n\, $interval), $frames)', setpts='N/($fps*TB)'" -y %06d.png) [04:56] <grepper> echo ${cmd[@]} [04:56] <grepper> ffmpeg -i monkey.mpg -vf select='lt(mod(n\, 8991), 60)', setpts='N/(29.97*TB)' -y %06d.png [05:16] <xreal> Does ffmpeg has an "internal" benchmark ? [05:18] <grepper> -benchmark ? [05:18] <grepper> shows with ffmpeg --help long [05:21] <xreal> grepper: nice :) [05:23] <xreal> ffmpeg -benchmark_all doesn't do anything :) [05:27] <grepper> looks like it prints timing for each frame [05:29] <xreal> Pff, it's totally useless to compare performance. Okay, I'll do it the old way using "time" :) [06:03] <vivan> dunno what is -benchmark_all but -benchmark is real benchmark [06:03] <vivan> https://www.ffmpeg.org/ffmpeg-formats.html#null [06:08] <xreal> vivan: I don't get the sense of benchmark :) [06:08] <xreal> it does a normal decoding in this example to NULL [06:09] <c_14> -benchmark should (according to docs) print CPU time and max memory consumption after the encode finishes [06:10] <xreal> That's nothing I would expect from a benchmark :) [06:10] <xreal> but okay, thanks for the hint. [06:10] <c_14> What are you expecting? [06:10] <vivan> xreal: yes, it measures decoding speed [06:11] <xreal> That's the result: bench: utime=161.851s | bench: maxrss=36100kB [06:11] <c_14> ye, it took 161.851s user cpu time and used max 36100kB RAM [06:11] <xreal> /usr/bin/time is better, I think :D [06:18] <xreal> Can ffmpeg switch x264's constant quality to fixed bitrate while running? I mean, can you enter "zones" ? [06:23] <c_14> I don't think so. [06:23] <xreal> That would be great :( [06:24] <xreal> Highest bitrate for fast action scenes and low bitrate for end credits. [06:24] <c_14> crf should do that [06:25] <xreal> yeah, but i'd like to get to crf 40 for credits... [06:25] <c_14> you can also split the source up where you want to the nearest I-frame, encode each separately and concat with copy [06:25] <xreal> c_14: sounds good [06:26] <xreal> What range does crf work in ? [06:26] <xreal> like +/-4 ? [06:27] <c_14> hmm? [06:27] <c_14> What do you mean range? [06:28] <xreal> c_14: CRF raises or lowers QP for complex or non-compexe parts. [06:28] <c_14> Oh, that. erm [06:32] <c_14> I can't find any finite numbers, but I'd guess around +-4 [06:39] <xreal> Since libfaac is gone, is "-strict experimental -c:a aac" okay to use? [06:40] <grepper> c_14: backslashing the array quotes isn't going to work, you will end up with literal quotes around each element [06:40] <grepper> I'd put the full command into an array, like: http://pastebin.ca/2901272 [06:42] <meorom> hi all! [06:43] <meorom> I'm building ffmpeg for Windows follow link: https://trac.ffmpeg.org/wiki/CompilationGuide/MSVC [06:43] <meorom> I use last rev of ffmpeg today is 73851 [06:45] <meorom> but I still get error [06:45] <meorom> e:\programming\c++\ffmpeg\libavutil\common.h(34) : fatal error C1083: Cannot open include file: 'inttypes.h': Invalid argument [06:45] <meorom> make: *** [doc/print_options.o] Error 1 [06:45] <meorom> in very many source file, why this file get error ? (other files not) [06:45] <c_14> grepper: fixed [06:45] <grepper> great [06:46] <meorom> could you please help me ? [06:46] <grepper> $in_file still not quoted, not needed in example, but if its for newbies ... [06:46] <c_14> meorom: can you pastebin your config.log ? [06:48] <c_14> grepper: fixed that as well [06:48] <meorom> I'm still a newbie of IRC, so...you mean paste all config.log content here ?? [06:48] <c_14> no [06:48] <meorom> so, what should I do ? [06:48] <c_14> pastebin it to pastebin.com or pastie.org or sprunge.us or ix.io [06:48] <c_14> and paste the link here [06:48] <meorom> ah, okey [06:48] <meorom> thanks [06:48] <xreal> c_14: Since libfaac is gone, is "-strict experimental -c:a aac" okay to use? [06:49] <c_14> It's fine [06:49] <c_14> libfdk-aac is better, but as long as you use a decent bitrate it should be ok [06:50] <xreal> the official binaries don't have libfdk-acc in Windows, I think ? [06:50] <c_14> No, building with libfdk-aac makes the build non redistributable [06:50] <meorom> c_14: http://pastebin.com/download.php?i=7VHvkX4b [06:50] <xreal> c_14: ;..-( [06:52] <c_14> meorom: did you install msinttypes ? [06:52] <meorom> yes [06:53] <meorom> msinttypes-r26 [06:54] <meorom> I don't know if I miss something, could you pls give a way to try ? [06:55] <grepper> finally the summary is dated now :) Where it starts with "Note that" [06:56] <c_14> meorom: try reading this: https://www.ffmpeg.org/platform.html#toc-Microsoft-Visual-C_002b_002b-or-In… [06:57] <c_14> If that doesn't help, send as much information as you can (including all output etc) to the -user mailing list, or wait around a bit here for somebody with more windows knowledge. [07:00] <c_14> grepper: reworded [07:01] <grepper> "The command_line array" [07:02] <meorom> thanks c_14, I will try [07:02] <c_14> grepper: Yeah, I'll reword that later. [07:02] <grepper> cool [07:02] <grepper> just trying to contribute, not be picky [07:04] <c_14> It's fine. I just decided to check over the whole article later. Check if any info is outdated or could be reworded. [07:40] <mr_lou> http://stackoverflow.com/questions/27834184/making-a-blu-ray-video-album-fr… [08:15] <Naji> mr_lou this is annoying... you posted your problem on stackoverflow, so please stop the broadcasting [08:33] <Orly> hello? [08:34] <Orly> hello, hello? [08:39] <anshul__> hello [08:39] <tmh1999> Orly : just ask your question, someone will answer if they know/ when they're back to the keyboard. [08:41] <Guest32443> doesnt matter [08:42] <Guest32443> wondershare converter - metadata overwriter - fix Gamma Shift Bug in MOV files!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! [08:42] <Guest32443> bye [08:56] <anshul_mahe> Guest27353: I was out of network, did u asked some question in between [09:09] <mr_lou> Naji, You should take a vacation if you get annoyed by someone posting the same url twice in two days. [09:18] <Naji> twice? [10:35] <cajone> HI Guys, I sped up a mp4 from 2hrs to 30mins using "setpts=0.25*PTS" which worked well but the converted video still contains the old time line and things its 2hrs long instead of 30mins, anyway I can fix this? [12:13] <Nebukadneza> heho [12:14] <Nebukadneza> i'm trying to extract all keyframes, and all frames-one-before-keyframe from a x264 video using ffmpeg. i've already succeeded in -vf select'ing keyframes, but the one-before-frame is giving me a little headache ;/ [13:29] <aleek> hi! Is there a way to change fps of a video file? I mean: I have video with 25fps, and I want to make it 50fps with times shorter time (total number of frames is the same) [13:39] <relaxed> aleek: 50 fps and half the duration? [13:41] <relaxed> try, ffmpeg -r 50 -i input -c copy output [13:42] <mr_lou> I used to use ffmpeg for scaling down videos to fit my cellphone, meaning lower resolution and less fps. Then suddenly it didn't work for some reason, so I began using HandbrakeCLI instead, with ffmpeg as parameter. </useless info> [13:44] <mr_lou> What I wanted to say with that, was that HandbrakeCLI could be another option: HandBrakeCLI -i input.mp4 -o output.mp4 -e ffmpeg -r 50 [13:45] <relaxed> ffmpeg should do it without issue. [13:46] <mr_lou> Yes [13:46] <mr_lou> There's a lot of that "should". ;-) [13:48] <relaxed> your cellphone issue is not likely ffmpeg's fault [13:49] <mr_lou> I'm not talking blame. Just saying, the command suddenly didn't work. [13:49] <mr_lou> Maybe some lib was broke, or what do I know. [13:50] <mr_lou> But whatever it was, HandbrakeCLI apparently didn't care. [13:51] <mr_lou> And ffmpeg is working for me fine now. Using it a lot to create blu-ray files these days. [13:51] <termos> Hi. I am using av_read_frame and sometimes I get an AVPacket that is empty (size == 0 and data == NULL), how should I handle such a situation? [13:51] <relaxed> yeah, it's the cat's meow [13:51] <termos> Right now I'm just skipping it [14:38] <Mavrik> termos, hmm, what's your input? [14:39] <termos> it's an rtmp that contains two streams: h264 video and aac audio [14:40] <Mavrik> hmm [14:42] <termos> If I don't skip the empty packages and pass them into decode+encode functions I get a very unstable output [14:43] <termos> I only have empty input packages on one of my streams though, the rest are working fine [14:45] <termos> I tried not ignoring the empty packages and pass them through a AVThreadMessageQueue and after poping the queue before decode+encode there are no empty packages, I wonder if there is some magic in this thread queue [14:46] <termos> The thing is I have implemented by own thread queue that has some more features that I like (like reading the number of elements), but of course it doesn't magically remove the empty packages [14:57] <Mavrik> mhm [16:39] <Fjorgynn> :D [18:36] <Phlarp> Is there anyway to control character spacing / font kerning in FFMPEG drawtext filters? [18:54] <c_14> None I know of besides editing the font. [18:55] <relaxed> probably through fonts.conf [18:56] <relaxed> if you have fontconfig support [18:56] <c_14> If there is, I couldn't find any documentation for it... (not that that means anything) [19:02] <relaxed> using ass subs might be easier if you need that much control [19:21] <JEEB> aegisub for actual subtitle creation and then the libass filter for rendering them on top of video [19:50] <justinX> anyone heard of a (tv) player that shows some added random character at the end of the last line in a subtitle when viewing? [19:51] <justinX> maybe I have to add an extra CRLF at the last line in the .srt file... [20:05] <smo_> hi, is it possible to covert and pipe a stream in hls ? [20:05] <smo_> -i myhttpurl -> to hls [21:07] <grepper> wow, looping through with -ss before the -i INFILE is 25 times faster than using the 'select' filter to output a chunk of frames (pngs) at intervals. [21:14] <grepper> is there any reason to prefer the select filter looping if I am just outputting pngs ? [21:15] <c_14> select is frame-exact [21:15] <c_14> -ss is usually timestamp-exact [21:15] <grepper> but probably not off by more than a frame or 2 ? I could live with 'close' at that speed. [21:16] <grepper> I guess if the timestamps are messed up ... [21:16] <c_14> If you don't use -codec copy, it should get the exact frame at that timestamp [21:17] <c_14> -ss will be fine most of the time [21:18] <grepper> okay, thanks. Now to figure out a way to programmaticly determine if the user's ffmpeg is new enough to benefit for the new accurate seeking with -ss on input. [21:18] <grepper> ffmpeg -h long 2>&1|grep -q accurate_seek [21:18] <c_14> You could just check the version string [21:18] <grepper> if their ffmpeg has -accurate_seek it is probably okay? [21:19] <grepper> yeah, tough with version string if they use git [21:19] <c_14> mhm, lemme check [21:20] <c_14> ye, if it has -accurate_seek it should be fine [21:29] <grepper> c_14: awesome, thanks so much. Been hoping this would be added/possible for many years, haven't really checked recently. [21:29] Action: grepper gotta go out, bbiab [21:39] <jirido> Hi.. I would need a test picture film for to set corect color adjustment in vlc and cant find any.. So i got the idea to make one with ffmpeg from ONE image i have.. Could you say if it is posible and eventualy how? [21:42] <c_14> You have the picture and want to make a video out of it? [21:43] <c_14> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images… [21:43] <jirido> Yes! [21:44] <jirido> I found some PM5544 test image films but they all blury [21:45] <jirido> Thanx [22:47] <bofh> Hello all. Does anybody know what is the way to override limitation of 32 channels for amix? I need to merge 20-50 audio streams into one [22:47] <c_14> Merge 32 into one, then merge the output from that with the remaining streams? [22:48] <bofh> yeah, that is what I was thinking about - merge chunks of streams [22:48] <bofh> but it's not possible to do without an intermediate file? [22:48] <c_14> nah, should work [22:49] <bofh> you mean piping? [22:49] <c_14> -filter_complex '[0:0][0:1][0:2]amix[foo];[foo][0:3][0:4][0:5]amix[out]' [22:49] <bofh> hm! [22:50] <bofh> I already have a very complex filter [22:50] <bofh> perhaps I could split it into several chunks [00:00] --- Sat Jan 10 2015

1 0

ffmpeg-devel.log.20150108
by burek 09 Jan '15

09 Jan '15

[00:24] <cone-947> ffmpeg.git 03Alexandre Colucci 07master:5a1addd7c1d8: xsub: Support DXSA subtitles [00:24] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:3ed80bd95e1d: Merge commit '5a1addd7c1d8ff218ed4b84f4f02fdb83980094c' [02:07] <Timothy_Gu> wbs: to recap the ac_slater's issue: 1. Android API Level <= 17 contains a broken math.h header that does not declare log2(). 2. The issue is fixed in API Level 18, but only in NDK version 9b+. 3. He could not compile the thing he was building with API Level 19, only 17. 4. He tried building with both, and only 19 works. 5. I don't know the *exact* NDK he is using although theoretically as long as it is >=9b it doesn't matter. 6. log2 seem [02:08] <Timothy_Gu> wbs: see https://developer.android.com/tools/sdk/ndk/index.html: `Updated include/android/*h and math.h for all Android API levels up to 18` [02:08] <Timothy_Gu> (9b changelog) [02:09] <Timothy_Gu> 9 apparently supported API level 18 as well, but still with the broken math.h [02:10] <cone-947> ffmpeg.git 03wm4 07master:bcaa9099b364: avcodec/dvdsubdec: error on bitmaps with size 0 [02:10] <Timothy_Gu> wbs: see the broken header: https://github.com/android/platform_bionic/blob/android-4.2.2_r1.2/libm/inc… [02:11] <Timothy_Gu> wbs: commit that fixed it: https://github.com/android/platform_bionic/commit/a0ee07829a9ba7e99ef68e8c1… [02:13] <Timothy_Gu> and the ndk headers are synced to bionic's in https://android-review.googlesource.com/#/c/68014/ [03:18] <cone-947> ffmpeg.git 03Lou Logan 07master:919e038a95fd: doc: fix typos [03:20] <aetasx> have you guys seen any issues with linking against an ffmpeg lib built with --enable-lto and using -ffat-lto-objects? [05:29] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:1d2410639d9d: doc/examples/muxing: Use avio_closep() in examples as it avoids leaving stale pointers [05:29] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:c61b84ccfda1: doc/examples/transcode_aac: Use avio_closep() in examples as it avoids leaving stale pointers [05:29] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:a63c17cec723: idoc/examples/remuxing: Use avio_closep() in examples as it avoids leaving stale pointers [05:30] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:a85f033149a6: doc/examples/transcoding: Use avio_closep() in examples as it avoids leaving stale pointers [05:37] <Timothy_Gu> aetasx: not really, but that's probably because nobody here uses it [07:11] <ac_slater> hey guys. I'm working through some ./configure issues. I would spend a bunch of time narrowing this down, but maybe you guys know. When building with --enable-programs, ffmpeg.o is looking for the symbol `ff_isom_write_hvcc` but it's undefined in my libavformat. I tried enable hecv parsers, etc. Any clue? [07:12] <ac_slater> ie - trying to find what provides that symbol [07:14] <ac_slater> oh... matroska [07:16] <Timothy_Gu> ac_slater: i think that's a known bug [07:16] <ac_slater> Timothy_Gu: Im trying to just enable matroska mux/demux and parser to see [07:17] <ac_slater> Timothy_Gu: thanks! [07:17] <Timothy_Gu> ac_slater: wait no [07:18] <Timothy_Gu> ac_slater: can you confirm libavformat/hevc.o is built? [07:18] <Timothy_Gu> ac_slater: and also, what release are you using? [07:19] <Timothy_Gu> ac_slater: libavformat/hevc.c supplies that symbol [07:21] <ac_slater> Timothy_Gu: release 2.5 [07:22] <ac_slater> Timothy_Gu: and no, I'm not building HEVC ... hmm. [07:22] <Timothy_Gu> ac_slater: no, the hevc.o object file doesn't mean hevc decoder [07:22] <ac_slater> Timothy_Gu: I guess I need it ;). Parser? Mux/Demux? [07:22] <Timothy_Gu> ac_slater: the muxer need it [07:23] <Timothy_Gu> ac_slater: libavformat/hevc.c file contains some muxer routines that are not related to the decoder [07:23] <Timothy_Gu> ac_slater: so try to see if hevc.o is built [07:23] <ac_slater> Timothy_Gu: yea it doesnt get built [07:42] <ac_slater> Timothy_Gu: well. I dont really know enough ffmpeg' [07:42] <ac_slater> Timothy_Gu: to solve this. I tried enable just the hevc muxer and the symbol still isnt there. What do you think I should enable? [07:45] <jamrial> check libavformat/Makefile, search for hevc.o [07:46] <jamrial> matroska, webm, mov or rtp muxers [07:48] <ac_slater> jamrial: interesting. I do need the RTP muxer [07:48] <Timothy_Gu> jamrial: hevc.o is in matroska muxer [07:49] <Timothy_Gu> s/(in)/included \1/ [07:49] <jamrial> i know, that's why i mentioned it [07:55] <Timothy_Gu> looking at http://sprunge.us/cRbQ he did enable matroska [07:55] <Timothy_Gu> ac_slater: that's odd [07:56] <Timothy_Gu> ac_slater: check libavformat/Makefile and find the line with OBJS-$(CONFIG_MATROSKA_MUXER) [07:56] <Timothy_Gu> ac_slater: can you see "hevc.o" there? [07:58] <ac_slater> Timothy_Gu: that paste didnt have the matroska enabled since, well, I dont need it ;) [07:59] <ac_slater> Timothy_Gu: oh wait, that paste doesnt disable anything, it just added the android flags, etc. [08:00] <ac_slater> So the question remains, if I want to build the ffmpeg binary, should I just enable matroska ? [08:07] <jamrial> you shouldn't need to, really. not sure why it told you ffmpeg.o needed ff_isom_write_hvcc. ffmpeg.c doens't call it [08:08] <Timothy_Gu> ac_slater: jamrial is right. Can you show the **exact** error? [08:38] <ac_slater> Timothy_Gu: uploading paste now [08:39] <ac_slater> Timothy_Gu: http://paste.debian.net/plain/139668 [08:39] <ac_slater> I guess this isnt for ffmpeg.o [08:39] <ac_slater> it's obviously not since the error is for libavformat.so [08:40] <Timothy_Gu> ac_slater: can you try make distclean and reconfigure with static libs instead of shared libs? [08:41] <Timothy_Gu> two things might be happening: 1. just a configure remnant from a previous build causing the issue; 2. there's a problem with the build system and using static helps debugging this kind of issue [08:44] <ac_slater> Timothy_Gu: gotcha. I create the complete new tree every build. And even if static worked I could use it since I'd violate GPL :( [08:46] <Timothy_Gu> ac_slater: I think you meant LGPL [08:46] <ac_slater> Timothy_Gu: yes. [08:46] <Timothy_Gu> ac_slater: but anyways this is just for debugging [08:46] <ac_slater> Timothy_Gu: true. And it did work with a static build [08:47] <Timothy_Gu> interesting. can you try shared again from a clean tree? [08:53] <Timothy_Gu> gotta sleep now [08:54] <Timothy_Gu> bye [14:21] <Daemon404> i wonder how many bugs can be summed up as "i dont understand how vsync works" [14:31] <aetasx> on that note, I tried finding a tool that would let me resync things the way ffplay and vlc seems to play them but wasnt able [14:42] <kierank> Daemon404: more like people don't understand vfr [14:42] <nevcairiel> vfr is evil [14:43] <nevcairiel> our screens are not vfr, why would you make video that way :D [14:43] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:558cef886054: ffmpeg_opt: Use avio_closep() to avoid leaving stale pointers in memory [14:43] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:78cb3c069d15: ffmpeg: Use avio_closep() to avoid leaving stale pointers in memory [14:43] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:eb3e661babfd: ffserver: Use avio_closep() to avoid leaving stale pointers in memor [14:43] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:e5993aeb606e: avdevice/lavfi: Use avio_closep() to avoid leaving stale pointers in memory [14:43] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:b565fd44642c: avformat/hdsenc: Simplify code by using avio_closep() [14:44] <wm4> <nevcairiel> our screens are not vfr, why would you make video that way :D <- they're about to fix this [14:44] <kierank> hopefully will do it properly too [14:44] <nevcairiel> if you refer to GSYNC or Adaptive Sync, its a gaming feature, dont expect TVs to ever have that [14:44] <kierank> timestamp + duration [14:45] <Daemon404> kierank, not a fan of matroska? [14:45] <Daemon404> ;) [14:45] <kierank> of course not [14:45] <kierank> timestamp alone is moronic [14:45] <wm4> matroska also has a fps field (as float) [14:46] <kierank> that's just metadata [14:46] <nevcairiel> matroska has a BlockDuration element, so you can have timestamp + duration [14:47] <kierank> it should be mandatory [14:47] <wm4> matroska has no dts though [14:47] <nevcairiel> its probably mostly used for subtitles [14:48] <aetasx> can only see vfr as saving storage for crappy video [14:49] <aetasx> which I have lots of :/ [14:49] <nevcairiel> well in theory it could make sense to reduce the number of frames when you dont need them as the video is not moving, but on the other hand with predicted frames encoding those takes practically no space so why bother with the complexity [14:49] <kierank> ricing [14:50] <aetasx> nevcairiel: mostly realtime streaming being the problem [14:52] <aetasx> if you get trashed sync, is there a way to fix it the way ffplay manages to? [15:07] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:50bb9b87ab6f: avformat/img2enc: Use avio_closep() to avoid leaving stale pointers in memory [15:07] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:26c72d2941d9: avformat/img2dec: Use avio_closep() to avoid leaving stale pointers in memory [15:07] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:bf3126db6a60: avformat/hlsenc: Use avio_closep() to avoid leaving stale pointers in memory [15:09] <wm4> I keep wondering what the fuck the point of these commits is, when the functions they're used in are called before deallocation anyway [15:10] <Daemon404> Code Defensively (TM) [15:20] <aetasx> you trademark that? [15:21] <aetasx> Daemon, how familiar are you with --enable-lto for ffmpeg? [15:22] <Daemon404> it's broken on mostly everything because of asm symbols iirc. [15:22] <Daemon404> also, why me [15:23] <TimNich> you Know Everything (") ;) [15:23] <aetasx> because I had an impression you would know and it wasnt the first time I had asked [15:25] <aetasx> it seems to work fine internally but including a lto-linked ffmpeg library in this case within x264 does break within the configure script due to the asm issues [15:26] <aetasx> I chalked it up to mingw64 stupidity though [15:27] <Daemon404> if you made a .a, it is not lto linked. [15:27] <Daemon404> the emited bytecode is just added to teh ar archive. [15:27] <Daemon404> no linking is done [15:28] <aetasx> I know, but -flto is passed into those objects, the actual flto errors out during the x264 configuration conftest part [15:28] <Daemon404> i would check if lto is even properly being used in the conftest checks. [15:29] <aetasx> yeah they are, it comes out with that typical "asm" constraints error [15:29] <aetasx> for some mmx function wihtin swscale [15:29] <Daemon404> lto is not very useful anyway [15:30] <aetasx> it strips quite a lot of code out of a static build [15:31] <Daemon404> it is limited as to what it can do to ffmpeg, due to things like av_register_all() existing [15:31] <Daemon404> (sometimes ive seen output be larger) [15:32] <aetasx> the ffmpeg I built through linking in -flto extra libraries and running --enable-flto on the ffmpeg built is somewhat around 2-4mb less in size [15:32] <aetasx> without the external libraries, thats probably the case [15:33] <Daemon404> using lto-built external libs does nto require lto-build ffmpeg fwiw [15:36] <aetasx> you'd have to pass in -fuse-linker-plugin and -flto to ldflags and I wanted to apply flto to the ffmpeg objects regardless [15:36] <aetasx> it works fine in binary builds of ffmpeg, just screws with the libraries when getting included into other things [15:38] <aetasx> right now though I think the only lto build problems are a result of mingw64's recent idiotic pthreads commits [15:39] <nevcairiel> lto never worked for me, and i dont use pthreads [15:39] <aetasx> nevcairiel: you know that commit I showed you before? [15:40] <aetasx> Heres what it is now: http://sourceforge.net/p/mingw-w64/mingw-w64/ci/5eb74fa49b3b34e21c98f85c6db… [15:40] <aetasx> all the flto built errors are: input/lavf.o (symbol from plugin):(.text+0x0): multiple definition of `asctime_r' [15:41] <nevcairiel> did you like disable inline asm? mine fails much earlier on that [15:41] <aetasx> nah, didnt need to [15:42] <aetasx> when I build ffmpeg with lto, Im passing in -ffat-lto-objects though so non-lto builds still build properly [15:44] <aetasx> within the configure script for x264 though, even if you don't pass in flto or fuse-linker-plugin, it is still invoked when it hits lto-built objects due to some configure script wrapper I think [15:45] <aetasx> but these asctime_r errors are the exact same type of crap I was experiencing before so Im guessing its still more mingw64 screwyness [15:53] <aetasx> Daemon404: you said you saw flto builds bigger in size when running them before. are you sure this isn't due to not adding in -ffat-lto-objects in the ffmpeg's --enable-lto options? the gcc docs say without it, the builds can end up larger [15:53] <aetasx> or do you think there's something else going on? [15:53] <Daemon404> i personally do not care about lto, and thus cannot be arsed to look. [15:54] <aetasx> thats pretty close to my own feeling ;) [16:12] <wm4> why would you use lto anyway [16:13] <aetasx> [09:31] <aetasx> the ffmpeg I built through linking in -flto extra libraries and running --enable-flto on the ffmpeg built is somewhat around 2-4mb less in size [16:14] <wm4> you're probably better off by disabling all the dumb unneeded shit [16:14] <wm4> like game format decoders [16:15] <aetasx> thats not really the purpose of this project, its for a build-script system that builds and links multimedia tools. making assumptions about what they do or don't need isn't really my place [16:17] <aetasx> its like zeranoe's build scripts [16:20] <aetasx> plus I get more experience having gone through the nightmare itself ;) [16:30] <Snaggle1> Is the AVFoundation indev on ffmpeg supposed to have an OS X min version for building? When autodetected to build, it tries to link to the CoreGraphics framework, but that only exists as a standalone framework from 10.8 and up. However, the detection code only checks for the presence of the AVFoundation.h header. Compilation of libavutil.54.dylib on 10.7 fails with "ld: framework not found CoreGraphics" [16:44] <wm4> michaelni_: for me, fate-sub2video fails even before my patches [16:45] <wm4> oh, I guess it uses system ffmpeg, wtf [16:46] <Daemon404> wtf? [16:48] <wm4> hm, double-checking it, this is not the case [16:51] <wm4> but the test still fails [16:53] <wm4> the only difference is actually that my output is missing stream 0 [17:05] <cone-173> ffmpeg.git 03Carl Eugen Hoyos 07master:26ac6d543598: lavfi/boxblur: Readd 8bit colour spaces on big-endian. [17:14] <wm4> updating magically fixed it [17:15] <nevcairiel> magic is always a good solution [17:20] <saste> nevcairiel, only when it works [17:27] <cone-173> ffmpeg.git 03Martin Storsjö 07master:71ec3d36fa61: doc: Add initial documentation about the OpenH264 encoder wrapper [17:27] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:de8f70b8e02d: Merge commit '71ec3d36fa6122b3b6c5c68a21b9c70cb2e668e8' [17:35] <cone-173> ffmpeg.git 03Martin Storsjö 07master:3852e2c926dd: libopenh264enc: Fix a typo and some nitpicks [17:35] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:93870f0a0b71: Merge commit '3852e2c926ddb166c7aa69c4644a86100ea144d9' [17:43] <aetasx> *sigh* http://sourceforge.net/p/mingw-w64/mailman/message/33212333/ [17:43] <aetasx> wm4: you're right, these people are idiots [17:44] <cone-173> ffmpeg.git 03Martin Storsjö 07master:2dbd35b00c64: libfdk-aacdec: Make sure decoding doesn't add any extra delay in the latest version of fdk-aac [17:44] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:3fd3647fdc5b: Merge commit '2dbd35b00c6433e587d5f44d5dbc8972ebbaa88e' [17:48] <wm4> aetasx: did I say that? I think it was someone else [17:49] <wm4> in any case, mingw itself is a somewhat idiotic project, because it hogs on an ancient unsupported msvcrt dll [17:52] <aetasx> actually you did say it was run by idiots ;) [17:53] <funman> tmatth: no idea how hard it would be to decode bpg [17:55] <aetasx> mingw-w64 specifically [17:55] <wm4> meh [17:55] <aetasx> oh no you said it was a bad commit, it was nevcairiel that said it was run by morons ;) [18:01] <Daemon404> i wouldnt consider them morons [18:01] <Daemon404> they provide a massive effort wrt CRT and the only working and update FOSS compiler for windows [18:02] <Daemon404> protip: shitting on the hand that feeds is a terrible idea [18:02] <Daemon404> and you come off as an as. [18:02] <Daemon404> ass* [18:02] <aetasx> nah, we were just having a discussion of their breaking things a number of times a few in the crt and headers [18:03] <Daemon404> protip 2: thats why they have releases [18:03] <Daemon404> its youre own fault if you use HEAD [18:03] <aetasx> where did I mention using HEAD? [18:03] <Daemon404> most of the breakage in headers that have been linked here were never in any release. [18:04] <Daemon404> and compaining about them is thsu pretty silly. [18:04] <cone-173> ffmpeg.git 03wm4 07master:816577716bc6: avcodec/dvdsubdec: fix accessing dangling pointers [18:05] <aetasx> msys2 only does releases based on that, Alexey being the dev of that project [18:05] <Daemon404> msys2 is unrelated to toolchains. [18:05] <Daemon404> msys2 is an environment [18:05] <aetasx> an environment that provides build releases [18:06] <Daemon404> they HAPPEN to have a toolchain package, and its their fault for using HEAD [18:06] <Daemon404> its the packagers fault i your toolchain is broken then [18:06] <Daemon404> if* [18:06] <aetasx> and the project maintainer is on the mingw-w64 group [18:06] Action: wm4 still hopes for midipix [18:06] <Daemon404> he is not a mingw-w64 core devel [18:06] <Daemon404> or even a contributer [18:06] <Daemon404> he is a thrid party mostly. [18:08] <aetasx> why is this even relevant? I didn't say they were morons [18:08] <Daemon404> [16:43] < aetasx> wm4: you're right, these people are idiots [18:08] <Daemon404> orly [18:09] <aetasx> nev was referring to the management of their pthreads built-in library [18:09] <Daemon404> in context, you were not referring to that [18:09] <Daemon404> see your link. it is not about their pthreads lib. [18:09] <aetasx> no, what I was doing was referrencing someone elses comment [18:10] <Daemon404> right. so you link to some unrelated thing, and then comment implicitly about pthreads. [18:10] <Daemon404> yea seems legit. [18:10] <Daemon404> anyway /care [18:10] Action: Daemon404 gets coffee [18:11] <aetasx> except its the pthreads lib that gets broken due to POSIX defines [20:09] <sectroyer> Hello. I have raw video with nv16 pixel format but when I try to play it with ffplay I get this error: [20:10] <sectroyer> Impossible to convert between the formats supported by the filter 'ffplay_buffer' and the filter 'auto-inserted scaler 0' [20:31] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:e360d030b362: avformat/rtsp: Use avio_closep() to avoid leaving stale pointers in memory [20:31] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:f9ddaa35c7d9: avformat/rtpenc_chain: Use avio_closep() to avoid leaving stale pointers in memory [20:32] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:07e4ed5637f5: avformat/mov: Use avio_closep() to avoid leaving stale pointers in memory [20:32] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:edeaf6f3a4d7: avformat/mlvdec: Simplify code by using avio_closep() [20:32] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:d1d8f866b540: avformat/mlvdec: Use avio_closep() to avoid leaving stale pointers in memory [20:51] <wm4> sectroyer: libswscale apparently can't handle this format [20:52] <sectroyer> wm4: hmmm than what should I do? I have rawvideo data that I would like to play/rencode... [20:52] <wm4> nothing you can do, other than hacking libswscale, or hoping that someone else does, or converting the format yourself [21:02] <kierank> iirc swscale does support nv16 [21:02] <kierank> apparently not [21:02] <wm4> at least it doesn't report it [23:10] <cone-173> ffmpeg.git 03Michael Niedermayer 07master:6d1a2efb8ac3: ffmpeg: Clear error message array at init. [23:14] <nevcairiel> Daemon404: at least their pthreads lib is quite moronic :P it doesn't work for shit [23:15] <nevcairiel> And just because it's the only thing we have doesn't make many of their actions any better [23:15] <wm4> I made my own pthread wrapper now [23:15] <nevcairiel> But I stick to my own builds using releases and generally don't run into big problems [23:16] <wm4> it was pretty simple, though no XP compat [23:16] <nevcairiel> For FFmpeg I use w32threads, don't really build anything elaey [23:16] <nevcairiel> Else* [23:16] <jamrial> i missed the msys2 chat. was it about how their mingw64 packages got an update that broke ffmpeg yesterday or so? [23:17] <jamrial> because i had that issue. fortunately they fixed it already [23:17] <nevcairiel> I didn't try msys2 yet, but everything I hear about it makes me not want to [23:17] <jamrial> i wonder why they use mingw64 HEAD, though [23:17] <jamrial> things break too often [23:18] <nevcairiel> Even if I would use msys2, I would probably just use their msys parts and stick to my own toolchain [23:18] <nevcairiel> I like control :) [23:19] <jamrial> same. i'm thinking about doing that since their mingw64 toolchain is not 100% static, and i don't like having to rely on dlls [23:20] <jamrial> especially the winpthreads one that i don't use. i use w32threads [23:20] <nevcairiel> I have pthreads-w32 in my toolchain if I need one implementation for something else, but yeah [23:22] <nevcairiel> Using pthreads-w32 together with my MSVC code leaks thread handles though, there is some weirs incompatibility there [23:22] <nevcairiel> But w32threads works just fine so I don't care [23:25] <wm4> I think pthreads-w32 requires you to "register" new threads not created by it? [23:25] <wm4> with pthread_win32_thread_attach_np() and pthread_win32_thread_detach_np() [23:26] <wm4> needed when doing static linking [23:27] <aetasx> ah, yeah I dont use MSVC for any of this stuff [23:28] <aetasx> jamrial: you're right about that. the project that I use and help with for building multimedia libs just goes through and renames all .dll.a files to something else if it can find a static lib [23:29] <wm4> what project is that? [23:29] <aetasx> https://github.com/jb-alvarado/media-autobuild_suite [23:29] <wm4> ah [23:29] <aetasx> it builds maybe like 30 different libs/binaries for multimedia stuff [23:30] <jamrial> if you mean renaming the .dll.a so the linker uses the .a files instead, you could just use -static when linking [23:30] <aetasx> ffmpeg, mp4box, mplayer, mpv, mkvtoolnix, f265, x264, x265 etc [23:30] <wm4> I guess it's easier for single projects with specific own solutions [23:30] <aetasx> that doesn't work so well actually, some projects screw it up when building [23:32] <aetasx> theres a lot of libraries that are only built as shared which I'm not wild about [23:33] <wm4> windows is strangely the only OS left where static linking is preferred over dynamic for dependencies [23:34] <aetasx> bitter memories over the old dll hell days [00:00] --- Fri Jan 9 2015

1 0

ffmpeg.log.20150108
by burek 09 Jan '15

09 Jan '15

[00:02] <llogan> xreal: why not just use -target instead of following that link? [00:02] <xreal> llogan: I've checked it in current source, it's still valid. [00:02] <xreal> llogan: also, I tried to combine it, doesn't work. wait [00:02] <xreal> http://pastie.org/9818794 => target instead of preset. [00:02] <llogan> where is the rest? [00:04] <xreal> llogan: that's some hours old. Wait. It said something about a missing format. [00:08] <xreal> llogan: http://pastie.org/9819149 [00:09] <llogan> mbrittsess: maybe something like colorlevels=rimax=0.5:gimax=0.5:bimax=0.5 [00:09] <xreal> llogan: The preset dvd can't write into mkv, I think. [00:10] <mbrittsess> llogan: I tried that one already, it didn't work [00:12] <llogan> then adjust the *imax values [00:12] <t4nk240> hello anyone [00:12] <t4nk240> that give a help on ffmpeg h265 [00:12] <t4nk240> please [00:13] <mbrittsess> llogan: I tried both 0.9 and 0.1 and it didn't work [00:13] <llogan> if you have photoshop, and you can get curves filter to do what you want then you can export the curves preset file and use it in the ffmpeg curves filter [00:14] <mbrittsess> I don't have photoshop [00:14] <t4nk240> no one [00:14] <llogan> then use gimp and interpolate the values [00:15] <mbrittsess> Maybe there's something wrong elsewhere in my cmdline and I'm not noticing it? If you don't mind looking at it, just to make sure [00:15] <mbrittsess> ffplay -f lavfi -i amovie=Typing_cleaned.flac,asplit[out0],showwaves=320x180:p2p:1,curves=negative,colorlevels=rimax=0.5:gimax=0.5:bimax=0.5,super2xsai[out1] [00:16] <mbrittsess> (that's a single-channel audio file, by the way, if it matters) [00:17] <mbrittsess> The audio file drives 'showwaves', I invert the colors so it's black-on-grey, then I'm *trying* to adjust that to black-on-white, then running it through super2xsai [00:17] <llogan> t4nk240: you didn't ask a question yet, so we can't give you an answer [00:18] <t4nk240> i logan [00:18] <t4nk240> i am using h264 to transcode some streaming to flv [00:18] <t4nk240> and work fine [00:18] <t4nk240> i am triyng h265 [00:18] <t4nk240> and i can get this working [00:18] <llogan> why? [00:19] <t4nk240> it give a error off bitrate [00:22] <xreal> llogan: did you read my pastebin? [00:23] <llogan> xreal: "-c:v:1 -target pal-dvd" doesn't make any sense [00:23] <t4nk240> llogan can you help please? [00:24] <llogan> t4nk240: you didn't provide your command and output [00:24] <t4nk240> hum [00:24] <t4nk240> sorry [00:26] <t4nk240> llogan: ffmpeg -i input -vcodec libx265 -vprofile baseline -vb 650k -vf scale=720:404,setdar=16:9 -acodec libvo_aacenc -ab 128k -ac 2 -f flv rtmp:// [00:26] <t4nk240> this what i am trying to use [00:28] <llogan> you didn't include your console output, but i am unsurprised [00:28] <xreal> llogan: That's why I used the normal settings. [00:28] <t4nk240> console output [00:28] <t4nk240> just a seconf [00:29] <xreal> llogan: with "decoded" target settings, it works as expected. [00:29] <t4nk240> [flv @ 0xa7e12e0] Tag [4][0][0][0]/0x00000004 incompatible with output codec id '86016' ([0][0][0][0]) Output #0, flv, to 'rtmp://istoeprivado.com:1939/live5/tvi': Stream #0:0(por): Video: hevc (libx265), yuv420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc56.13.100 libx265 Stream #0:1(por): Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, 192 kb/s Stream mappin [00:29] <t4nk240> fuck [00:29] <t4nk240> lol [00:29] <t4nk240> Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input [00:29] <llogan> you have a hard time following instructions [00:30] <llogan> no command + no console output = no help [00:30] <t4nk240> i dont understand llogan [00:30] <t4nk240> i put where all [00:30] <t4nk240> what i need more? [00:30] <llogan> ok, someone else can help you [00:31] <t4nk240> llogan [00:31] <t4nk240> that is nice [00:31] <t4nk240> heheh [00:31] <xreal> t4nk240: please try to write in ONE line :) [00:32] <t4nk240> i will try to find on google x265 any suport [00:32] <t4nk240> yet all post i read irc etc etc help [00:32] <t4nk240> only blah blah blah [00:32] <t4nk240> help [00:32] <t4nk240> hehehhe [00:32] <xreal> t4nk240: 1 line... [00:33] <t4nk240> iff i put 1 line anyone will help me xrreal? [00:33] <xreal> t4nk240: no, but it's more polite to do so. [00:34] <justinX> dvir: is the resulted mp4 supposed to have variable framerate too, or be more normal, filling up with do-nothing frames or something? [00:34] <t4nk240> ok xreal so no one help in where what begin the irc is better, no you are by your one find..... [00:34] <mbrittsess> If you're having trouble catching the console output, you can type your command and then put "| clip" after it to put the output to the system clipboard [00:34] <mbrittsess> Then you can simply put that on pastebin [00:35] <xreal> or use this one: https://clbin.com/ [00:35] <llogan> mbrittsess: why not set the size in showwaves instead of using super2xsai? [00:36] <mbrittsess> I do set the size in showwaves. I set it at half of what I want the final size to be, because super2xsai doubles the resolution by necessity [00:36] <t4nk240> i will gooogle about it x265 will talk with me [00:36] <dvir> justinX: the result mp4 should mimic the GIF, by playing it as a video instead. This means that yes, it should fill in that time with do-nothing frames :) [00:37] <xreal> llogan: Do you understand my problem? [00:37] <dvir> justinX: a similar technique would be to extract the frames from the gif, let's say 10 frames with variable timings, and then for the required FPS, copy each frame FPS*time_for_frame, and then stitch them together into a video [00:37] <llogan> mbrittsess: what i mean is, why not just set it to your desired output frame size and not even use super2xsai [00:38] <dvir> of course, when creating a video and knowing all that you don't have to do it like that [00:38] <justinX> dvir: yeah while mp4 can do variable framerate I guess it is better to not do it that way. I assume you should have to put a parameter telling what the resulting files framerate will have then. [00:38] <llogan> xreal: no [00:38] <mbrittsess> Because I specifically wanted it to be smoothed with super2xsai, the output from showwaves is fairly pixellated and it looks a bit nicer this way [00:38] <llogan> ah [00:39] <dvir> justinX: the interesting thing is that the rest of the video has the correct frame rate [00:40] <dvir> justinX: just the first and last frame that are a bit messed up [00:40] <justinX> dvir: and yeah if you can't get it work in another way, I guess you can manually duplicating the frames and use these as input to ffmpeg instead. [00:40] <llogan> mbrittsess: where is the gray? it looks white to me [00:41] <mbrittsess> http://i.imgur.com/IuOYjq5.png [00:41] <mbrittsess> That's the output I'm getting [00:41] <justinX> dvir: yeah there is support for variable framerate and, animated gifs was added 2013 or something I think. [00:41] <mbrittsess> The background is a uniform (235,235,235) instead of (255,255,255) [00:41] <xreal> llogan: I want to transcode the incoming mpeg2ts stream to pal-dvd compatible and put it in MKV. [00:41] <dvir> justinX: I did that, the issue with that is that it is a 'stupid' conversion - if you have duplicate frames, ffmpeg isn't taking advantage of it and would create a huge video file, compared to the optimized way [00:42] <justinX> dvir: ah [00:42] <t4nk240> one question does the x265 suport flv? [00:43] <llogan> mbrittsess: i'm not getting the gray. tested both ffplay and ffmpeg output [00:43] <justinX> dvir: if the only the first and last frame is problematic, I guess you can somehow circumvent the problem by asking ffmpeg to stitch together 3 files... one with only yoru first frame, then the mp4 you have made, and then the last frame [00:43] <mbrittsess> Something else I tried, based on how I'd do it with image-editing stuff, was using: curves=all="0/0 0.4/0 0.6/1 1/1" [00:43] <mbrittsess> Hmmm [00:43] <dvir> justinX: so I will assume it's just a bug, even though seems to be a pretty big one. Unless the GIF conversion of FFmpeg is being completely not worked on [00:44] <mbrittsess> Let me try it with ffmpeg and see if the resulting video file is gray, or white [00:44] <justinX> dvir: an alternative is investigating the bug. may the source be with you :-) [00:44] <llogan> mbrittsess: ffmpeg -i input -filter_complex showwaves=320x180:p2p:1,negate -ss 5 -vframes 1 output.png [00:46] <vivan> >uniform (235,235,235) [00:46] <dvir> justinX: can't wait to dive in ;) thanks by the way! [00:46] <vivan> that sounds like tv range [00:47] <vivan> is black 16,16,16 instead of zeroes? [00:47] <mbrittsess> This may sound stupid, but do you want me to replace the token 'input' with some particular file, in order to test that? [00:47] <llogan> yes [00:48] <xreal> Does anyone remember tools like "DVD shrink"? They resized DVD-R9s very fast to DVD-R. How did it work? [00:48] <mbrittsess> vivan: You are exactly correct [00:48] <mbrittsess> I thought it might've been a TV color range issue until I saw showwaves was producing rgb24 output, so I assumed it was working on a normal computer, uh...range [00:49] <mbrittsess> I think "range" is not the right word, but I'm not sure what *is*, sorry [00:50] <xreal> Is it possible to switch to another bitrate between minutes 20 and 30? [00:51] <mbrittsess> llogan: I'm getting yellow text "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)" [00:51] <mbrittsess> Sorry if there was some other obvious thing I need to replace, this is my first day using ffmpeg [00:52] <llogan> it's possible your -ss value is longer than the input duration. i only added that so the first frame wasn't used in case it was silent so it would be less boring [00:52] <mbrittsess> (It's pretty wonderful software, like ImageMagick for videos, I will be using it more in the future!) [00:53] <mbrittsess> Er...actually, that was my fault :P [00:54] <mbrittsess> Output file is (128,128,128) on top of pure white [00:55] <mbrittsess> Except for a few pixels where the two waveforms intersect, which are full-black [00:55] <llogan> maybe SDL/ffplay acts differently between Windows and Linux. not sure why it shows gray for you and white for me [00:56] <llogan> or you're using an outdated version with an old bug [00:56] <mbrittsess> Hmm...maybe the default video encoding can't accommodate the color extremes? [00:56] <mbrittsess> Either that or a new bug, I downloaded this a few hours ago [00:57] <llogan> i tested current git head [00:57] <mbrittsess> It says...it's mpeg1video in yuv420p, I guess that would do it [00:57] <mbrittsess> Let me see if I can coax it into an RGB format [00:58] <mbrittsess> Erm...although if you already know how to do that, I wouldn't mind saving the trip to the documentation :P [00:59] <xreal> Is it possible to switch to another bitrate between minutes 20 and 30? [01:09] <mbrittsess> llogan: I tried exporting video to an actual format in a different format, and it gave me pure-white [01:10] <mbrittsess> So I guess it was a final-encoding issue [01:10] <mbrittsess> Although I'm curious to know what kind of awful video format ffplay is using by default that can't handle the extreme ends of grayscale [01:12] <mbrittsess> Well, thanks for the help, everyone! [01:12] <mbrittsess> Have a good night [02:54] <xreal> Can I make ffmpeg a streaming server, which sends to tcp://localhost:2222, until a clients opens and acceses this stream? [02:57] <c_14> add ?listen [02:58] <c_14> tcp://localhost:2222?listen [03:00] <xreal> let me try [03:10] <xreal> hmm... I can't get the stream to work. Let me try other things. [03:17] <xreal> finally! [03:17] <xreal> thanks :) [04:17] <grepper> I know about the -vf fps technique for outputting a single frame at intervals (ie. fps=1/300 for every 5 minutes), but I'm wondering if there is to output a range of frames at intervals, like every 5 minutes output 300 frames. ? [04:18] <grepper> s/there is/there is a way/ [04:19] <grepper> the only way I've been able to do it is to seek from the beginning of the file for each range of frames, which is obviously slowish. [04:41] <radiator> hi gius. been looking for thiis everywhere. got a bunch of mov files, need to covert them to audio. whats options do i need in ffmpeg? [04:42] <radiator> like mp3 i guess [04:57] <datajerk> tried the docs and google, no luck. is there a way to blur out a section of a video? not by time (figured that out), but by x,y-x1,y1? thanks [04:58] <datajerk> trying to blur out passwords on demo videos [05:02] <grepper> I know about the -vf fps technique for outputting a single frame at intervals (ie. fps=1/300 for every 5 minutes), but I'm wondering if there is a way to output a range of frames at intervals, like every 5 minutes output 300 frames ? [05:02] <grepper> oops, sorry [05:12] <grepper> radiator: Something like: -vn -b:a 128000 -c:a mp3 -ar 44100 [05:18] <xreal_alt> Is there a fast way to fix "moov atom not found" only with ffmpeg ? [05:22] <radiator> grepper: that works exactly :D [05:29] <xreal_alt> anyone? [07:32] <mr_lou> http://stackoverflow.com/questions/27834184/making-a-blu-ray-video-album-fr… [15:51] <techtopia> hello [15:51] <techtopia> this is my encoding line http://pastebin.com/JGprvWbx [15:51] <techtopia> how do i sat the colormatrix correctly [15:51] <techtopia> it worked fine before i added colormatrix=bt709 into the -vf video filters [15:54] <techtopia> [Parsed_colormatrix_2 @ 042d3e40] Unspecified destination color space [15:54] <techtopia> [AVFilterGraph @ 06b31f80] Error initializing filter 'colormatrix' with args 'bt709' [15:54] <techtopia> Error opening filters! [15:55] <techtopia> thats the error i get when running it atm [15:56] <Mavrik> um, colormatrix=dst=bt709 ? [15:56] <Mavrik> also, do you have new ffmpeg? [15:56] <techtopia> it's about a 2 weeks old [15:57] <techtopia> looks like it's working [15:58] <techtopia> trying it on a 30 sec clip to test [16:00] <techtopia> hmm the encode worked [16:00] <techtopia> but it's not showing up as bt709 in media info [16:04] <techtopia> http://i.imgur.com/pJUIqJ6.jpg [16:04] <techtopia> on the right is what im getting, on the left is what i want [16:10] <vivan> try -x264opts colormatrix=bt709 [16:11] <vivan> video filters are video filters, I believe that they don't set metadata [16:13] <JEEBsv> there's also a global setting [16:13] <JEEBsv> that is specific to libx264 [16:14] <techtopia> Matrix coefficients : BT.709 [16:14] <techtopia> got it [16:14] <techtopia> thank you very much vivan [16:14] <techtopia> and Mavrik, JEEBsv [16:14] <techtopia> :) [16:15] <Mavrik> yep, x264 opt will set metadata [16:15] <Mavrik> colormatrix filter will convert the colorspace [16:17] <claz> how do i apply fadein and fadeout in the same filtergraph? I've tried "afade=t=in:ss=0:d=5 [main]; [main] afade=t=out:ss=10:d=5" but it doesn't work as expected ( fadein first 5 then play normally 5 then fadeout last 5 [16:28] <coolusernameXX> Hello, I'm trying to use ffmpeg's HLS segmenter (full command here: http://pastebin.com/7aiHcfud ) but I can't get metadata to copy between input and output. Has anybody know how to copy metadata (language names etc) to output files? [16:29] <coolusernameXX> If I try to use ssegment ant write directly to .ts files metadata transfers, but with HLS it doesn't. [17:36] <johnboy> Hi I have an issue when I try to merge a mkv file, here's the mediainfo output : http://sebsauvage.net/paste/?c9d0e92401e49d46#fPGEY6Ok5ly0VSS6LXXEx6NLLpXvD… is something wrong with my settings ? [17:38] <techtopia> best way is not to try and merge mkv in ffmpeg [17:38] <techtopia> either use mkvtoolnix [17:38] <techtopia> or with ffmpeg, encode to ts, concat as ts (merge), transcode to mkv [17:38] <techtopia> or [17:39] <techtopia> transcode mkv to ts in ffmpeg, then concat, then transcode back to mkv [17:41] <johnboy> I used mkvmerge techtopia [17:42] <johnboy> tried mediacoder too but it does the same mistake which is : it adds 10 seconds of black screen in the beginning of the video [17:43] <techtopia> are you appending and not adding [17:44] <techtopia> in mkvmerge if you hit add instead of append it adds all video and audio files to one file, instead of apending them one after the other [17:46] <johnboy> yes I know that but when I merge the mkv file with a ass subtitle it still does not make a proper file [17:46] <johnboy> so I tried without the .ass file and same results [17:46] <johnboy> I don't get why it adds 10 second of black screen in the beginning of my file [17:51] <techtopia> oh i see [17:51] <techtopia> you messed up your frame rate [17:52] <techtopia> Frame rate : 23.976 fps [17:52] <techtopia> Original frame rate : 23.810 fps [17:52] <techtopia> you should encode to the source fps which in this case is 23.810 so you should add to your encoding line before the output -r 23.810 [17:52] <techtopia> but your source video is probably bad [17:53] <techtopia> and was probably 23.976 to begin with [18:00] <johnboy> ok techtopia I'll try that thank you [18:58] <sectroyer> Hello. I have raw video with nv16 pixel format but when I try to play it with ffplay I get this error: [18:58] <sectroyer> Impossible to convert between the formats supported by the filter 'ffplay_buffer' and the filter 'auto-inserted scaler 0' [19:07] <llogan> it's listed in -pix_fmts but not as a supported input or output format. i don't know why though. [20:09] <sectroyer> llogan: is it a bug ? :) [20:10] <Fohlen> anyone knows a simple snippet to "make a song louder"? [20:11] <Fohlen> I effectively need to change the dB or similliar because my targeting device is really quiet [20:11] <llogan> sectroyer: i don't know [20:11] <sectroyer> llogan: okay, thanks :) [20:12] <Fohlen> oh got it, there's a simple -vol option [20:12] <Fohlen> :D [20:13] <llogan> Fohlen: the volume filter values make more sense to me [20:13] <llogan> http://ffmpeg.org/ffmpeg-filters.html#volume [20:25] <johnboy> techtopia, ok so I managed to get a proper file. Basically, forcing the FPS to the original FPS wasn't enough because the sound got also 10 seconds so I encoded the original sound in ogg and I muxed it with the encoded video file plus the sub [20:25] <johnboy> thanks a lot techtopia ;) [20:35] <Fohlen> llogan: thanks [00:00] --- Fri Jan 9 2015

1 0

ffmpeg-devel.log.20150107
by burek 08 Jan '15

08 Jan '15

[00:03] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:15ea222778ca: vf_interlace: merge FIELD_LOWER check [00:03] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:82b1bcd7f9b6: Merge commit '15ea222778caaec0877b3f9938140b707c931d96' [00:13] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:4438c256387a: vf_interlace: remove the warning tag in the message [00:13] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:ed97963bdbf3: ulti: invert the order of parameters of ulti_decode_frame() [00:13] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:f2f88b952d66: Merge commit '4438c256387a5845e5cdc5c14c43f4047b68301b' [00:13] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:40e123bc67cc: Merge commit 'ed97963bdbf3bb17fca4f9ea0aa1a97722dec907' [00:33] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:1c6183233d56: msmpeg4: check memory allocations and propagate errors [00:33] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:89388cf60bd5: Merge commit '1c6183233d56fb27a4a154e7e64ecab98bd877f1' [00:39] <cone-150> ffmpeg.git 03Justin Ruggles 07master:8e03ca2fff40: isom: add 'mp1v' fourcc [00:39] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:4009ee3eccac: Merge commit '8e03ca2fff40ce065ea5a516200e777c96cd785e' [00:46] <cone-150> ffmpeg.git 03Martin Storsjö 07master:e581e88cbdb3: movenc: Readd an accidentally removed condition [00:47] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:0d0a905152f6: Merge commit 'e581e88cbdb3323ca0026a54b39a9716a3686e9f' [01:24] <cone-150> ffmpeg.git 03Martin Storsjö 07master:2a1500fb5523: movenc: Fix a typo in a comment [01:24] <cone-150> ffmpeg.git 03Martin Storsjö 07master:8a3d9ca603f4: libavcodec: Add an OpenH264 encoder wrapper [01:24] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:cb03d29dec64: Merge commit '2a1500fb552364488cfbee3e29e296788a3a6bdd' [01:24] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:c09eeca5468c: Merge commit '8a3d9ca603f4d15ecaa9ca379cbaab4ecaec8ce4' [01:24] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:ddae03f69bc1: avcodec/allcodecs: prefer x264 over openh264 if both are available [04:01] <jamrial> german spambot? [04:05] <Timothy_Gu> lol [04:07] <Timothy_Gu> lol x265 is classified by GitHub as being an assembly project [04:21] <jamrial> Isn't x265 a mercurial project hosted on bitbucket? what are you looking at? [04:36] <Timothy_Gu> jamrial: https://github.com/mirror/x265 [04:43] <jamrial> ah, interesting [04:45] <jamrial> but yeah, x265 doesn't like to use yasm macros so their .asm files are huge, apparently making up 56% of total lines of code [06:26] <ac_slater_> hey guys. I'm making a series of patches to fix the stagefright android example. Issue is, when I do a clean build against bionic (libc) via `./configure` ... apparently log2() isnt available. If I disable most codecs then it works. [06:26] <ac_slater_> any hints? [06:26] <ac_slater_> (even though log2 says it's available in config.log) [06:40] <ac_slater_> also, ffmpeg 2.5 [06:54] <Timothy_Gu> ac_slater_: you should find out 1. if log2 is supposed to work, and 2.why log2 doesn't work, and then fix it in configure [06:55] <Timothy_Gu> ac_slater_: seems like it is supposed to work by reading bionic source [06:56] <ac_slater_> Timothy_Gu: exactly ... [06:56] <Timothy_Gu> ac_slater_: what's the error? [06:56] <ac_slater_> Timothy_Gu: http://paste.debian.net/plain/139519 [06:58] <Timothy_Gu> ac_slater_: so apparently log2 is not defined in a header [06:58] <ac_slater_> Timothy_Gu: and I dont really believe that [07:00] <Timothy_Gu> ac_slater_: seems like it is [07:00] <Timothy_Gu> ac_slater_: can you run `make V=1` and paste config.h and config.log? [07:00] <Timothy_Gu> the decl is guarded with __BSD_VISIBLE [07:01] <ac_slater_> oooh intersting [07:01] <ac_slater_> working on those pastes [07:03] <Timothy_Gu> the exact line is "#if __BSD_VISIBLE || __ISO_C_VISIBLE >= 1999 || __XSI_VISIBLE" [07:04] <ac_slater_> Timothy_Gu: stupid question, should I just force one? [07:05] <Timothy_Gu> ac_slater_: you can for this build, but it would: a. be stupid and b. not benefit other users [07:06] <ac_slater_> agreed [07:10] <Timothy_Gu> ac_slater_: any progress? [07:11] <ac_slater_> Timothy_Gu: yea interesting. config.log is showing the same thing... log2 and log2f implicitly defined [07:11] <ac_slater_> (I have to scrub my paths since my work policy says I have to :( ) [07:12] <Timothy_Gu> ac_slater_: sed -i 's,/my/path,/censored,g' config.log [07:12] <Timothy_Gu> ac_slater_: you work at Raytheon? [07:16] <ac_slater_> Timothy_Gu: a subsidiary, but yes ;) [07:19] <ac_slater_> alright, almost there... apparently some sites dont like pasting 400+kb [07:21] <Timothy_Gu> curl -F 'sprunge=<file' http://sprunge.us | should work [07:21] <ac_slater_> Timothy_Gu: http://sprunge.us/cRbQ [07:21] <Timothy_Gu> haha [07:21] <ac_slater_> haha yea, I couldnt remember the name of it! [07:22] <ac_slater_> so yes, this isnt as simple as `./configure -os=android` ... but I hope it's close enough to debug [07:24] <Timothy_Gu> ac_slater_: yeah I think it's fine but not everybody here thinks so, so don't be surprised when you meet a person not as nice as me ;) [07:25] <ac_slater_> I've probably met them all haha. I'm usually here asking ridiculous things [07:26] <Timothy_Gu> ac_slater_: stupid question: what android sdk are you using [07:26] <Timothy_Gu> oh 4.2.2 saw that [07:26] <ac_slater_> well, 4.2.2 is directory I made ... I'm using NDK r10 [07:27] <Timothy_Gu> nvrmd [07:30] <ac_slater_> If it'd help, I can the minimal config [07:31] <Timothy_Gu> that's dumb, bionic doesn't map _ISOC99_SOURCE to it's very own __ISO_C_VISIBLE or _C99_SOURCE [07:31] <ac_slater_> interesting [07:32] <Timothy_Gu> *its [07:33] <Timothy_Gu> ok try this [07:33] <Timothy_Gu> Add [07:33] <Timothy_Gu> add_${pfx}cppflags -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 [07:33] <Timothy_Gu> to after [07:33] <Timothy_Gu> eval ${pfx}libc_type=bionic [07:35] <Timothy_Gu> _POSIX_C_SOURCE implies __ISO_C_VISIBLE == 1999 so it *should* work [07:36] <ac_slater_> cool, testing now [07:37] <anshul_mahe> ubitux: for add_insert_timeline_graph is it possible to add it in parser instead of cmdutil [07:42] <ubitux> what do you mean? [07:42] <ubitux> what parser? [07:43] <ac_slater_> Timothy_Gu: :( [07:44] <ac_slater_> no go [07:46] <ac_slater_> and I could see the cppflags added [07:46] <ac_slater_> can * [07:46] <Timothy_Gu> ac_slater_: can you paste the preprocessor output? [07:46] <Timothy_Gu> of a file that contains #include <math.h> [07:47] <aetasx> you guys know if theres a way to pull gcc's default optimization param values? (the ones passed in with --param) [07:50] <Timothy_Gu> aetasx: man gcc | grep default ? [07:50] <ac_slater_> Timothy_Gu: (899 lines) [07:51] <Timothy_Gu> aetasx: do you mean like `gcc -fthis-is-an-op`? [07:51] <Timothy_Gu> aetasx: 768 here ;) [07:51] <aetasx> nah theres some other parameters get passed in and set that change how those options work. lemme grab an example [07:52] <Timothy_Gu> grepping for "Options That Control Optimization" in man page might help [07:53] <aetasx> so like "gcc -fgcse --param max-gcse-memory=1024..." [07:53] <ac_slater_> Timothy_Gu: http://sprunge.us/GgDi [07:54] <aetasx> yeah the manual lists some but the manual also lists other optimization options I don't have so its obviously different versions and most of the actual params don't state their default value in the manual anyway [07:54] <ac_slater_> Timothy_Gu: shit... doesnt have the #define statements [07:54] <anshul__> ubitux: in libavcodec in file allcodec.h there is provision to register any parser, to make any file and then register there [07:54] <anshul__> exact file path is libavcodec/allcodecs.c [07:54] <aetasx> https://gcc.gnu.org/onlinedocs/gcc/Optimize-Options.html#Optimize-Options if you just hit END and skip to the bottom of the page for instance [07:55] <Timothy_Gu> ac_slater_: try -D_POSIX_C_SOURCE=200112 [07:55] <Timothy_Gu> ac_slater_: that will be more realisitic like FFmpeg [07:55] <ubitux> anshul__: no it doesn't belong here at all [07:56] <ac_slater_> Timothy_Gu: yea no difference [07:56] <Timothy_Gu> aetasx: I don't see any undocumented params [07:56] <Timothy_Gu> aetasx: or you can use man page which should be corresponding to your local gcc [07:56] <Timothy_Gu> ac_slater_: hm [07:57] <ac_slater_> Timothy_Gu: this is really odd [07:57] <aetasx> no, those are documented but I don't know which ones are available in my gcc version since there's no listing for them and I was trying to find out what their default "sane" value is set at [07:57] <aetasx> before I go and set it to something ludicrous [07:58] <anshul_m> is that libavcodec parser only for some specific purpose? [07:58] <Timothy_Gu> ac_slater_: do you have permission to modify ndk headers? [07:59] <Timothy_Gu> aetasx: yes that's why I told you to read your *local* man page [08:00] <aetasx> Timothy_Gu: thanks but it looks like I found the flag. Looks like you can list them through "gcc --help=params" [08:00] <ac_slater_> Timothy_Gu: yea. Im just curious why it's not there [08:03] <anshul_m> ubitux: is that libavcodec parser only for some specific purpose parser? [08:04] <ac_slater_> Timothy_Gu: I found some things on stackoverflow that point to this being intentional [08:04] <Timothy_Gu> ac_slater_: can you compile with gcc -E -dM /tmp/c.c [08:04] <Timothy_Gu> ac_slater_: that's odd [08:05] <Timothy_Gu> ac_slater_: this cmd will show the predefined variables [08:05] <ubitux> anshul_m: yes [08:05] <anshul_m> they could only used by libavcodec or something like that, just for information I am asking [08:06] <ubitux> this code is parsing format data, and constructing a lavfi filtergraph, it doesn't belong in a codec parser [08:06] <ac_slater_> Timothy_Gu: http://sprunge.us/PNjR [08:07] <anshul_m> ubitux: is it possible to have same parser register feature in libavformat [08:07] <ubitux> what for? [08:10] <Timothy_Gu> ac_slater_: apparently -D_POSIX_C_SOURCE didn't work because a) it's not shown in the output and b) __POSIX_VISIBLE is 200809 which is the default [08:10] <ac_slater_> Timothy_Gu: would it be weird to disable log2 detection and rely on libavutil/libm.h to fix it? [08:10] <ac_slater_> Timothy_Gu: I see, that makes sense [08:10] <Timothy_Gu> ac_slater_: why would you not use a (potentially optimized) log2() and depend on a fallback? [08:11] <anshul_m> ubitux: we have parser which works in avformat, even in my scte case my parser need to be share by different muxer's [08:11] <ac_slater_> Timothy_Gu: cause it's not there ;) at least not in a header [08:11] <Timothy_Gu> ac_slater_: sorry to bother you again but this time try: gcc -E -D_POSIX_C_SOURCE=200112 -dD /tmp/c.c [08:11] <Timothy_Gu> ac_slater_: it is there just somehow not enabled [08:12] <ac_slater_> no bother [08:12] <ubitux> anshul_m: well then just write shared code [08:13] <anshul_m> ubitux: where should I register that file with shared code [08:13] <ac_slater_> Timothy_Gu: ooooo [08:13] <ac_slater_> Timothy_Gu: http://sprunge.us/EHPC [08:13] <Timothy_Gu> aha [08:13] <ubitux> anshul_m: "register"? you just add it wherever you want in lavf and reference it in the makefile? [08:13] <ac_slater_> Timothy_Gu: so what does that actually mean? [08:14] <Timothy_Gu> ac_slater_: nothing [08:14] <Timothy_Gu> ac_slater_: now try: gcc -E -D_POSIX_C_SOURCE=200112 /tmp/c.c [08:14] <Timothy_Gu> it just means that a log2() exists [08:14] <Timothy_Gu> wait [08:15] <ac_slater_> that works too... ie - log2 is exposed [08:15] <Timothy_Gu> oh I think I gave the wrong instructions [08:15] <Timothy_Gu> <@ Timothy_Gu> ac_slater_: try -D_POSIX_C_SOURCE=200112 <--- did you *really* follow this? [08:15] <anshul_m> ubitux: In libavformat each reference in Makefile is related to some muxer_opt, is same file in different muxer_opt is possible [08:16] <Timothy_Gu> ac_slater_: btw your c.c is *really* broken btw ;) casting pointer to int [08:16] <ac_slater_> Timothy_Gu: it's taken from the ffmpeg configure script :p [08:17] <ubitux> anshul_m: see OBJS? these are not (de)muxers [08:17] <Timothy_Gu> ac_slater_: hehe [08:18] <Timothy_Gu> ac_slater_: now try to compile an FFmpeg file [08:18] <Timothy_Gu> with -D_POSIX_C_SOURCE=200112 [08:18] <ac_slater_> Timothy_Gu: hmm well. I upgraded to ndk r10 with android-19. Let me put it back [08:18] <anshul_m> ubitux: thanks [08:19] <Timothy_Gu> ac_slater_: the ndk version shouldn't matter [08:19] <anshul_m> ubitux: is it also possible to share that code in libavfilter [08:19] <Timothy_Gu> I'm looking at the source of android-4.0.1_r1 [08:19] <ac_slater_> Timothy_Gu: interesting [08:19] <ac_slater_> Timothy_Gu: one sec [08:19] <ubitux> anshul_m: that code could be in libavfilter, but there is no point, and it means exporting the symbol [08:20] <ubitux> i don't want that, because it's app specific, and thus belongs in the tools sources [08:21] <anshul_m> ubitux: I understand your patch now, from my side no prob where it is, for your patch [08:21] <ac_slater_> Timothy_Gu: hmm well, ndk with the android-19 sysroot requires no posix std hacks... but my code breaks. I brought this issue up with android-17... so that's what I need to fix [08:23] <anshul_m> ubitux: I was aking it for my scte case, i was writing a filter that take 2 program and give one prog in output, which also use the same code shared by muxers [08:23] <ac_slater_> Timothy_Gu: I think the last two pastes I gave you were from android-19... let me verify [08:23] <anshul_m> ubitux: should I write same code again in that filter or export it [08:23] <anshul_m> export it from libavformat [08:24] <ac_slater_> Timothy_Gu: yup... my mistake.... I miss led you. This `gcc -E` calls we from android-19 .... I redid them with android-17 and no log2 functions were exposed. [08:25] <ac_slater_> misled * [08:25] <Timothy_Gu> darnit [08:26] <Timothy_Gu> android 17 is 4.2* right? [08:26] <ac_slater_> yea [08:26] <ac_slater_> Timothy_Gu: yea it's ok. I was merely trying to fix libavcodec/libstagefright.cpp ... the buildscript in the ffmpeg tree is soooo broken [08:26] <ac_slater_> I managed to get it to build and work with 4.3+ [08:26] <ac_slater_> but not 4.2 [08:27] <ac_slater_> but sadly to use libstagefright (and link it) ... you need some of the android source tree. So only about ~300mb of shit to get to build and link 1 file [08:28] <Timothy_Gu> I made a mistake too [08:28] <Timothy_Gu> I was looking at sys/cdefs.h from 4.0.1 but not math.h [08:29] <ac_slater_> I wonder what issues will come up building my libs against the 4.3 sysroot and using them in 4.2 .... [08:29] <Timothy_Gu> yeah the 4.2 math.h is completely broken [08:29] <Timothy_Gu> see https://android.googlesource.com/platform/bionic/+/a0ee07829a9ba7e99ef68e8c… [08:31] <ac_slater_> lol +doublelog2(double); [08:31] <ac_slater_> as in, I see the definition added [08:31] <Timothy_Gu> and the commit is included in 4.3 or api level 18 [08:32] <Timothy_Gu> so yes, the best work around is using libavutil/libm.h [08:32] <Timothy_Gu> after everything we've been through [08:32] <Timothy_Gu> what a standard-compliant libc [08:33] <ubitux> anshul_m: you can "private export" with avpriv_* [08:34] <ac_slater_> Timothy_Gu: so would you force libavutil/libm.h or upgrade to level 18 if you could? [08:34] <Timothy_Gu> ac_slater_: for you guys I don't know cuz I'm no android dev [08:34] <ac_slater_> Timothy_Gu: me either ;) [08:34] <Timothy_Gu> ac_slater_: but for FFmpeg we'll of course try to support both [08:35] <ac_slater_> well now I know what the deal is. I'll make a note and submit a patch in a few days [08:35] <ac_slater_> hmm but I guess it might be hard to detect bionic version [08:35] <Timothy_Gu> ac_slater_: you don't really need to I guess [08:35] <Timothy_Gu> remember the implicit decl error? [08:35] <Timothy_Gu> it is originally a warning [08:36] <ac_slater_> yea it still wouldnt continue though.. -Werror must have been on [08:36] <Timothy_Gu> but then turned into an error with -Weror=implicit-function-declaration [08:36] <Timothy_Gu> yes [08:36] <Timothy_Gu> but the check for log2() did not have the -Werror on [08:37] <ac_slater_> I see. Interesting [08:37] <Timothy_Gu> so we can either move the entire l. 5267 (master) block up [08:38] <Timothy_Gu> or we can manually add -Werror=implicit-function-declaration -Werror=missing-prototypes to check_mathfunc (l. 4851) [08:38] <Timothy_Gu> need to sleep now [08:38] <Timothy_Gu> school tomorrow [08:38] <ac_slater_> thanks for the help mate! [08:38] <Timothy_Gu> ac_slater_: good luck [08:38] <wbs> ac_slater_: fwiw, you shouldn't need to use the bionic libc headers at all, the normal ndk headers for that should work just fine [08:38] <ac_slater_> take it easy [08:38] <ac_slater_> Timothy_Gu: thanks! [08:38] <Timothy_Gu> yeah wbs should know more about this stuff than I do [08:38] <wbs> ac_slater_: just take the frameworks/av, system/core and hardware/libhardware headers from whatever android version you need [08:38] <ac_slater_> wbs: I think that's what I point it to anyways [08:39] <ac_slater_> oh *those* [08:39] <anshul_m> ubitux: thanks I will look at that aproch [08:39] <ubitux> anshul_m: that is, only if you need to share it between libs [08:39] <ac_slater_> wbs: I was assuming the sysroot the ndk produced would be enough for stdlib stuff [08:39] <ubitux> otherwise just use ff_ [08:40] <wbs> ac_slater_: yes, it should, and the math.h from the NDK should be identical for all android-3 to android-19 [08:41] <ac_slater_> yea that's what I was going. Timothy_Gu and I figured out that android-17 has some funky libm definitions [08:42] <wbs> hmm, no, indeed, apparently android-18 in the NDK headers has got a change in math.h that may or may not be broken [08:42] <Timothy_Gu> wbs: -18 changed it: https://android.googlesource.com/platform/development/+/master/ndk/platform… [08:43] <Timothy_Gu> and -18 is the non-broken one [08:43] <Timothy_Gu> now really gotta go [08:44] <Timothy_Gu> it is identical for 13-17 [08:44] <Timothy_Gu> then 18 changed it [08:44] <Timothy_Gu> then 21 changed it [08:48] <ac_slater_> wbs: it's working with 18 and 19 [08:49] <ac_slater_> but all of this came from trying to find out what was causing the symbol "ff_isom_write_hvcc" for not being available for me with my reduced configure call. I'm sure that's a separate topic though (and probably not for this channel) [08:49] <ac_slater_> s/reduced/reduced feature/g [09:02] <wbs> ac_slater_: so to recap your issue (I didn't read all of it properly), configure does detect the log2 function in android-17 even though it isn't available in math.h in that version? what exact NDK version is this? [11:58] <Loriker> does anyone know the reason for closing this ticket (#4219) ? [12:37] <rcombs> Loriker: because the report is extremely vague, and outright states that it can't be reproduced on any other machine? [12:39] <rcombs> &and it was closed by the reporter [13:35] <akira4> ubitux, would you please have a look at this diff? http://pastebin.com/tfvpDGwv. Still no subtitle stream :( [13:45] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:69ee915e1c62: avcodec/parser: add fuzzy mode to ff_fetch_timestamp() [13:45] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:ad465e789721: avcodec/mpeg12: Try to fetch pts/dts from both the packet associated with the picture start code as well as earlier start codes [13:56] <wm4> ubitux: I'd rather call it "more lazy" than "less risky" (because you seem to plan to break it anyway) [14:58] <aetasx> what was it that brought about adding the --enable-random/--disable-random configure flags? automated testing? [14:59] <wm4> holy fuck it exists [14:59] <wm4> pretty fun [14:59] <aetasx> lol, those flags? [15:00] <Daemon404> wm4, what does [15:00] <thardin> do they enable random features? [15:00] <wm4> --enable-random randomly enable/disable components [15:00] <wm4> --disable-random [15:00] <thardin> or just make the binary deterministic? [15:00] <thardin> lol [15:00] <Daemon404> o [15:01] <Daemon404> good for fuzzing maybe [15:01] <thardin> I wonder how many combinations are possible. probably billions upon billions [15:01] <Daemon404> indeed [15:02] <aetasx> I dunno, I saw that and was like.....what possible need could've drove this feature into here [15:02] <Daemon404> its found quite a number of build failures [15:03] <aetasx> wouldn't it be more thorough to just bruteforce the combinations though? you wouldn't have repeats at least [15:03] <Daemon404> uh no [15:04] <saste> and now we have this silly reference.ppm with food instead of the beautiful lena... [15:04] <Daemon404> saste, because lena wast "free" enough for debian [15:04] <Daemon404> and for some reason we care [15:04] <Daemon404> (i was against removing it) [15:04] <wm4> it was also sexist [15:04] <Daemon404> oh that too. supposedly. [15:04] <wm4> you should be ashamed [15:04] <aetasx> lol [15:05] <saste> wm4, I'm vegetarian and I'm a bit offended by the meat-food in reference.ppm ;-) [15:05] <aetasx> Daemon404: were you a part of the project when those flags got added? [15:05] <Daemon404> probably [15:05] <wm4> saste: surely you agree that women have more rights than cattle [15:05] <aetasx> interesting [15:05] <thardin> that or start using burly men in image compression tests, see how long it takes people to feel uncomfortable [15:06] <saste> wm4, I don't eat women, at least not literally [15:06] <kierank> use a picture of a certain religious figure [15:06] <Daemon404> aetasx, if you were to try and build every combination of: config options, enabled muxers, demuxers, parsers, filters, etc [15:06] <Daemon404> i would probably outlive the universe [15:06] <Daemon404> too many to try [15:06] <wm4> use a picture of allah eating pork, get ffmpeg bombed [15:06] <thardin> aetasx: as I said, there's probably billions of combinations. there's no hope of even ./configuring them all, let alone compiling and linking [15:07] <aetasx> true, but handling the various tests outside of the script would at least give you more control over it. was just curious what the history was behind it [15:07] <thardin> it would perhaps be better to have it outside configure, but whatevs [15:07] <Daemon404> i dont really follow what 'more control' means [15:08] <thardin> Daemon404: how do you repeat a full configure + build + run thing when it just does it randomly? [15:08] <Daemon404> thardin, log it bro [15:08] <Daemon404> its in config.log [15:08] <aetasx> meaning you can generate the random lists without the internal randomizer...anyway I wasn't debating its purpose, was just curious of its history [15:08] <Daemon404> and e.g. FATE keeps all config logs [15:08] <Daemon404> for all runs [15:08] <Daemon404> so there is no repeatability issue. [15:08] <thardin> ah, so it "translates" the config line or somesuch [15:08] <wm4> y4m for chrome? wtf [15:08] Action: kierank wonders whether to send a chrome patch to decode vp9 with ffmpeg [15:09] <Daemon404> aetasx, it was probably so it could be hooked up to FATE [15:09] <aetasx> gotcha [15:09] <Daemon404> kierank, will be rejected [15:09] <Daemon404> libvpx has better ER [15:09] <wm4> kierank: ffmpeg still can't decode vp9 in webm correctly [15:10] <Daemon404> all of 0 people care too [15:10] <Daemon404> because nobody outside of youtube can use it [15:10] <kierank> i respect the strategy though [15:10] <kierank> not playing codec wars [15:11] <kierank> since atm hevc is just noise [15:11] <compn> whas incorrect about it ? [15:11] <Daemon404> only annoying since they periodically contact us [15:11] <Daemon404> with bs info [15:11] <Daemon404> on how much It Is Great [15:11] <Daemon404> and try to get us to switch [15:11] <compn> standard vp9 trolling [15:11] <wm4> compn: ffmpeg drops timestamps for a certain type of frame [15:11] <compn> er standard on2 trolling* [16:10] <aetasx> hey Compn, looks like those K-S-V patches fixed all that PTS nonsense I was getting [16:12] <compn> aetasx : of course thats why i told you to use ksv build :p [16:15] <aetasx> I know. It was totally my fault for assuming the official repo was the best place to get it :p [16:17] <compn> only took you a month to test? :P [16:17] <compn> ehe [16:17] Action: compn kidding [16:17] Action: compn goes afk [16:19] <aetasx> yeah I know, theres like 300gigs of video on my disk that I didn't go through [16:58] <iive> A user seeks help with raw image format. the image seems to be variety of nv12, but with bigger chroma channel (aka nv12->420, nvXX->422). Does somebody know if such format is supported by ffmpeg/swscale ? Or at least how it is named? [17:00] <iive> hum. nv16 seems like it. [17:01] <nevcairiel> nv16 is 422, but if swscale likes that i couldnt tell you [17:04] <kierank> I think it does [18:23] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:dfffc4aadc7c: avformat/dvbsub: remove unused variable [18:23] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:e00499eb4c80: avcodec/libxvid: fix potential integer overflow [18:23] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:2d7fd4cc33d5: ffmpeg_opt: Use av_realloc_array() [18:23] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:7efabffc2899: ffmpeg_opt: Check for av_realloc_array() failure [20:13] <akira4> uhm ubitux , are you free right now? [20:14] <ubitux> akira4: hi [20:14] <ubitux> sorry, in a moment [20:14] <ubitux> akira4: btw, i'm going to provide you a dvd sample [20:14] <ubitux> which i need to upload (might take a while) [20:14] <ubitux> so we can test on the same data [20:14] <akira4> that'd be great. [20:18] <ubitux> akira4: ETA 25 minutes [20:18] <ubitux> i'll be back then [20:19] <akira4> alright. I'll stick around till then [21:00] <ubitux> wm4: deprecating a side data is way complex and annoying that a whole C structure api with headers [21:00] <ubitux> you honestly want a full MOV/MP4 specific field in AVFormatContext and AVStreams? [21:00] <ubitux> with all the structures that it means? [21:08] <wm4> I just don't want an idiotic API that tells me "fuck you" [21:09] <wm4> though it might still be slightly better than nothing [21:10] <wm4> but that's the only thing that speaks for it [21:56] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:86e7d20ac2e7: avcodec/aic: Use av_malloc_array [22:47] <cone-947> ffmpeg.git 03Luca Barbato 07master:b3f04657368a: segment: Fix the failure paths [22:47] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:2a741baeff35: Merge commit 'b3f04657368a32a9903406395f865e230b1de348' [22:47] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:e2404096b02e: avformat/segment: Clear contexts after deallocation [23:02] <cone-947> ffmpeg.git 03Luca Barbato 07master:01168bf1405d: mpeg: Remove unused field [23:03] <cone-947> ffmpeg.git 03Michael Niedermayer 07master:0e1bb03cb034: Merge commit '01168bf1405ddbf671f1074a927b951b8777adbb' [00:00] --- Thu Jan 8 2015

1 0

ffmpeg.log.20150107
by burek 08 Jan '15

08 Jan '15

[01:28] <xreal> I'm decoding AC3 to WAV. Why - the heck - is this such slow? it uses 25-30 % cpu only :( [01:28] <xreal> Perhaps I should spread it to multiple cores? [01:50] <xreal> can ffmpeg decode Dolby Surround ? [01:50] <xreal> Prologic 2, I mean [02:30] <xreal> Ah DAMN. drc doesn't work for DTS :( [07:37] <hjjkhjhjhj> Why am I getting not a desired result after "ffmpeg -vsync throughput -i input.mkv -vf -mpdecimate=max=35000 -c:v libx264 output.mkv" ? Timecodes shifted in a some weird way because after comparing with blend +-1 frame difference. [07:39] <hjjkhjhjhj> *-vsync passthrough [11:05] <baran> hey guys [11:05] <baran> do you know where i could find scaning type in ffmpeg data structures? [11:05] <baran> i mean interlaced vs progressive [11:06] <c_14> look at what the idet filter does [11:09] <SirDarius> baran: maybe you are looking for this: https://ffmpeg.org/doxygen/trunk/structAVFrame.html#a7f761518da738c704676c6… ? [13:18] <pagios> anyone familiar with Set top boxes? [13:22] <Mavrik> asking a concrete question helps. [13:24] <pagios> hi Mavrik [13:24] <pagios> well ok i am trying to stream to STBs like 10 channels, how will the user instruct the stb to read a certain stream? do i need to broadcast the 10 streams individually to my server and then publish those on the server for the user to select from? [13:24] <Mavrik> Yes, that's how it's usually done in IPTV installations [13:24] <Mavrik> You broadcast 10 streams on multicast on the network [13:24] <Mavrik> and the STB then "tunes-in" to the right multicast address for that channel [13:25] <Mavrik> STBs usually have JavaScript or somesuch interface where you tell the player which multicast (or unicast or whatever) address to read [13:27] <pagios> Mavrik: the STB establishes a kind of VPN to the main cloud server? [13:27] <pagios> and listens to a private multicast address like 224.0.0.0 ? [13:27] <Mavrik> ugh [13:27] <Mavrik> no [13:28] <Mavrik> STBs are dumb [13:28] <pagios> tell me more about it [13:28] <Mavrik> How about you actually explain what you're trying to achieve. [13:28] <pagios> ok sure [13:28] <Mavrik> You're asking about a huge field with alot of solutions :) [13:29] <pagios> well i have 10 signals received on my main server, i would like to broadcast those to STB clients [13:29] <pagios> i already have a WOWZA server [13:29] <pagios> to broadcast to STBs [13:29] <pagios> but want to understand the workflow of how to design and implement [13:29] <Mavrik> those clients are on your local network or do you want to stream over public internet? [13:29] <pagios> public internet , Mavrik [13:30] <Mavrik> did you already buy STBs? Which? [13:30] <pagios> nop didnt buy the STBs yet [13:30] <Mavrik> pagios, so basically, over internet you'll probably do unicast streaming [13:30] <Mavrik> currently most elegant is the HLS protocol, RTMP works as well (both are supported by Wowza) [13:31] <Mavrik> the "standard" IPTV STBs are just dumb browsers essentially... you preupload a HTML/JS app (the GUI essentially) that controls what you're playnig [13:31] <pagios> yes WOWZA provides an HLS stream [13:32] <Mavrik> but essentially what you'll do on STB is something in line of .play("http://your.wowza.server/channel5.m3u8") [13:32] <pagios> Mavrik: how will you handle subscription in this case? and make sure only authorized STBS get the streams? [13:32] <Mavrik> that's the hard part :) [13:33] <Mavrik> I've seen tons of solutions, mostly done via STBs sending their serial/MAC over a secure HTTPS connection to Wowza and wowza then uses auth module to verify them [13:33] <Mavrik> but thats only one of the ways, that's hard to get right :/ [13:34] <Mavrik> pagios, but essentially your authentication/subscriptions have to be done on HTTP(S) layer or via DRM like on a usual webapp [13:36] <pagios> Mavrik: so mainly i broadcast 10 channels over wowza and the stb client just switches hls urls on his side right? [13:37] <Mavrik> yep [13:37] <Mavrik> or RTMP or plain MPEG-TS over HTTP, depending on what your STBs will support [13:37] <Mavrik> HLS is just an example of tech that works rather well for that kind of streaming [13:38] <pagios> is the switch so fast usually on the STB side? [13:38] <pagios> or do you just provision manually the 10 channels first on stb and then start switcing? [13:39] <Mavrik> switch time heavily depends on how the stream is encoded and the network latency [13:39] <Mavrik> but usually you have little choice, since I haven't seen alot (or indeed any) STBs that could stabily decode two streams at once [13:39] <pagios> rtmp is faster as i recall [13:40] <pagios> right? [13:40] <Mavrik> as I said, it depends heavily on network infrastructure and encoded stream (we got it to under 500ms, but it required some really heavy work) [13:40] <Mavrik> mostly be prepared for zap times of about 1 sec [13:40] <pagios> zap time means when you press the remote control button to switch a stress right [13:41] <Mavrik> you press the button -> you see the new channel [13:41] <Mavrik> as I said, STBs are mostly really dumb and crappily programmed [13:42] <pagios> i see [13:43] <pagios> but mainly i need a very strong connection to be able toupload the 10 channels from source site to WOWZA right? specially if content is in HD [13:44] <Mavrik> yes [13:44] <Mavrik> e.g. a well encoded FullHD stream is about 15Mbit/s [13:45] <Mavrik> meaning you need 150Mbit + overhead, that is 200Mbit/s link from your source to wowza to keep it stable [13:45] <Mavrik> and remember, you need to provider bandwidth from wowza to EACH user :) [13:45] <Mavrik> (note that you can get away with well-encoded 720p sterams at 3-5Mbit as well :) ) [13:47] <pagios> Mavrik: i need 150mbit/sec for each user from wowza to the user [13:47] <Mavrik> no, remember, each user will be connected to only one stream [13:47] <Mavrik> so if you'll be streaming 10Mbit/s streams, you need users * 10Mibit from wowza to users [13:48] <pagios> Mavrik: for a best SD signal i need how much ? ;/ [13:48] <pagios> if it is 720x576 which is the best you can get with an SD i gyess [13:49] <Mavrik> we usually do 1Mbit for good quality SD signals [13:49] <Mavrik> 3Mbit for 720p, 6-8 for 1080p [13:49] <Mavrik> it's not as pretty as the 30Mbit broadcast, but looks good enough on TV :) [13:51] <pagios> Mavrik: 1mbit/sec for a 720x576? [13:51] <pagios> thats around 100KB/sec [13:51] <Mavrik> yep [13:52] <Mavrik> there's alot of people wanting to watch videos on 1Mbit DSL lines :) [13:52] <pagios> Mavrik: do some STB provide adaptive bitrating? [13:52] <pagios> so they switch to a low res of the HD signal [13:52] <Mavrik> yes, mostly over HLS [13:52] <pagios> thats a good option [13:52] <Mavrik> but remember, you need to provide both then yourself :) [13:53] <Mavrik> some of them are buggy while switching over though [13:53] <pagios> on wowza i would provide the transcoded streams [13:53] <pagios> and the user would choose one [13:53] <Mavrik> ah, that would work yes [13:53] <pagios> so i am same on bandwidth for the user :) [13:53] <pagios> even less sometimes [13:54] <pagios> my big question is now authentication of stb toward the server [13:55] <pagios> Mavrik: cant you send the full 10 channels to the user STB for him to choose locally? [13:56] <pagios> or that would require a huge connection? it is not the usual way of doing it right? [13:56] <pagios> like traditional TVs.. [13:56] <Mavrik> pagios, that would require the user to have the internet fast enough to download all channels at once :) [13:56] <pagios> Mavrik: the traditional tv does that.. [13:56] <pagios> how much b/w does the tv receive handle? [13:56] <pagios> huge i think right? [13:58] <Mavrik> according to Wikipedia, DVB-T is up to 30Mbit/s, DVB-C up to 40 [13:58] <pagios> the switching does not happen locally right? [13:58] <Mavrik> it does [13:58] <pagios> 30mbit/sec means around 30 channels at most [13:58] <pagios> sd [13:58] <Mavrik> you get a whole bucket of channels on a single TS stream [13:58] <Mavrik> pagios, usually you get like 5 [13:59] <pagios> 5 HD or 30 SD [13:59] <Mavrik> since they encode at 3-5 Mbit/s for SD [13:59] <pagios> some cable providers provide 100 channels, how are they handled? [13:59] <Mavrik> multiple frequencies [13:59] <Mavrik> coaxial cable can carry several 100s of Mbit :) [13:59] <pagios> oh so the receiver receives multiple freq [13:59] <pagios> each having liets say 10 channels at most [13:59] <Mavrik> you switch between "buckets" of channels [13:59] <pagios> lovely [14:00] <Mavrik> also called "multiplexes" in DVB-T terminology [14:00] <pagios> so mainly a receiver handles 30mbit/sec x number of freq [14:00] <Mavrik> but the catch is, RTMP or HLS don't support that... and I haven't seen STBs that could demux MPEG2-TS streams with multiple programs :/ [14:00] <Mavrik> pagios, no, receiver tunes into one freq [14:00] <Mavrik> and handles 30Mibt/s, then throws away programs it's not showing [14:00] <pagios> ok so 30mbit/sec at a time [14:01] <pagios> thats why when you are switching from a channel on a different freq you experience more zapping time [14:01] <Mavrik> yep [14:03] <pagios> Mavrik: how can i implement the authentication for the STB [14:03] <Mavrik> *shrug*, can't help you much there [14:04] <Mavrik> I know we track them via serial numbers and use HLS/DRM protection, but can't really tell you much more [14:04] <Mavrik> the whole HTML UI / auth layer is done by another team, I just deliver video :) [14:04] <pagios> any specific STB model you advise? [14:04] <pagios> something popular stable flexible enough [14:09] <Mavrik> pretty much anything I've dealt with was crap (Motorola, Albis, Amino, Sagem) [14:09] <Mavrik> just check if they support your protocols and resolution [14:10] <pagios> Mavrik: how about the multicast you talked about? [14:10] <pagios> it is not applicable over public internet right? [14:10] <Mavrik> yep [14:10] <Mavrik> that's for IPTV for ISPs where they have a separate network to users :) [14:11] <pagios> thanks Mavrik [14:11] <pagios> i will do my researc for authentication [14:16] <pagios> Mavrik: Roku is considered a good STB? [14:17] <Mavrik> no idea, Roku isn't available in EU [14:20] <pagios> Mavrik: hmm i think i will point the STBs to an initial page where they get to authenticate with a user/pass using an html form [14:20] <Mavrik> pagios, well, just remember that users can extract URL to wowza [14:20] <Mavrik> and you still need something there [14:21] <pagios> Mavrik: your STBs authenticate with a user providing a user/pass by the remote control? [14:28] <pagios> Mavrik: the STBs require any cards to be inserted? [14:29] <Mavrik> pagios, no to both [15:17] <sectroyer> Hello I have raw video data of size 960x540. For single grame size in bytes is: 1036800. I am able to play it using ffplay as rawdata but I don't get correct colors. How to detect/find correct pixel_format? [15:52] <kruug> Is there a preset for WinFF that just does the `-codec copy` option? [15:53] <kruug> or am I restricted to the cli for that? [15:56] <relaxed> this is not the place for WinFF support [15:57] <kruug> Well, there isn't a WinFF channel, and when I searched for "WinFF community" in Google, it brought me to the FFMpeg community page, so... [15:57] <kruug> I think someone in the FFmpeg community did some fun SEO on that :) [16:01] <relaxed> kruug: https://groups.google.com/forum/#!forum/winff [16:01] <relaxed> found on the fromtpage of http://winff.org [16:01] <relaxed> frontpage* [18:29] <bigzed> Hello guys, I have a file input.mp4. If I encode it like this 'ffmpeg -i input.mp4 out.mp4' the file has two more frames after encoding even though the framerate is the same. Is this intended behaviour? [18:33] <Mavrik> um [18:35] <bigzed> I can provide you with a sample if you want to try it out :) [18:35] <Mavrik> doing what you did makes no sense anyway [18:35] <Mavrik> ffmpeg did whatever it felt like it... hard to talk about "intenteded behaviour" [18:36] <Mavrik> because you just said "do whatever" :P [18:38] <bigzed> I use ffmpeg to encode incoming files (of variuous codecs) to mp4 proxies of the same resolution and bitrate including a burned in timecode. I just noticed that this one mp4 gets additional two frames. So the timecode between original and proxie is of. [18:39] <bigzed> So, yeah encoding mp4 to mp4 seems senseless. But generally we use ffmpeg to encode various formats. Just this example is mp4 to mp4 [18:39] <bigzed> debug output of ffmpeg while encoding shows these lines [h264 @ 0x5499a00] no picture [18:39] <bigzed> [h264 @ 0x54dec80] no picture. Maybe those are the two additional frames? [18:43] <Mavrik> bigzed, no. [18:44] <Mavrik> I suggest you report that on bugtracker with proper parameters (at least codec + quality) and a sample and ffmpeg version / commandline output [18:45] <bigzed> thanks [18:52] <dowdle> How does one specify a 2.39:1 aspect ratio? [18:53] <dowdle> -aspect ? [18:53] <Popara> Hello, i'm trying the -progress argument into ffmpeg to report me statistics about the process encoding [18:54] <Popara> IT seems that it doesnt work at all, it does POST, i debug it using PHP, but it onyl writes at the end of the encoding but the post array is empty [18:54] <Popara> so it does only a single request without anything, only at the end [18:54] <dowdle> Nevermind... -aspect 2.39 seems to work. [19:05] <Popara> ah it works with a static file, i put a php file thought it does post [19:42] <t4nk081> anyone here with ffmpeg and xcode development experience? I am pretty new to ffmpeg dev and have difficuties compiling a sample source. [19:50] <c_14> I have no idea about xcode, but if you state your problem in a bit more detail somebody might be able to help [20:17] <xreal> I wonder if it's possible to store a whole transponder in mkv and make the single streams readable? :) [20:22] <xreal> Doesn't seem to work: ffmpeg -threads 16 -f mpegts -i tcp://localhost:2345 -c:v:0 copy -map 0:16 -c:v:1 copy -map 1:23 -f dvd -y test.mkv [20:23] <c_14> you can't map 1, you only have 1 input [20:24] <c_14> and -f dvd doesn't really make sense if you're trying to produce an mkv [20:24] <xreal> lol, got it :) [20:24] <xreal> I wanted to "route" 16 to 0 and 23 to 1 :) [20:24] <llogan> and you should always use pastebin to show your command and the complete console output [20:24] <xreal> llogan: even1 liners? [20:24] <llogan> everything [20:25] <llogan> what i mean is you need to always include your console output too [20:26] <xreal> llogan: ok. but it was my fault. Thanks for help. [20:28] <xreal> Nice, I know can record the whole transponder and capture it in MKV :) [20:35] <xreal> This won't work, will it? http://pastie.org/9818794 [20:36] <xreal> ah, it's target. [20:37] <xreal> no, it's also not target :D [20:45] <xreal> I can do it the manual way :) http://todayiwantedtoprogram.tumblr.com/post/15142587796/what-does-ffmpegs-… [20:46] <llogan> why are you using a dvd target for MKV container? [22:33] <dvir> hello! I'm using FFmpeg to convert a GIF into an MP4, with the following command: "ffmpeg -f gif -i ~/transparent.gif -c:v libx264 -f mp4 ./transparent.mp4". This is the input GIF: http://media.giphy.com/media/lXiRla43JDySJhTLW/giphy.gif and this is the output mp4: https://vid.me/aGv6 [22:33] <dvir> it seems like FFmpeg is skipping the first frame of the GIF, and skips the timing for the last frame (the first and last frame are 200ms long, yet it doesn't halt at the end like it does in the beginning). [22:33] <dvir> 2000ms* [22:34] <dvir> anyone has an idea why would that happen? am I missing a parameter? [23:07] <mbrittsess> Hello, everyone...I hate to have to ask this but I've been trying for 90 minutes now with no success... [23:07] <mbrittsess> I've got a video with some uniform light-grey areas and uniform pure-black areas [23:07] <mbrittsess> I'd like to change this to uniform pure-white areas and uniform pure-black areas [23:08] <mbrittsess> I've tried experimenting with colorbalance, colorlevels, curves, and lutrgb without success, the grey just stays grey [23:08] <mbrittsess> Although it's quite possible I'm using one or more them wrong [23:09] <mbrittsess> So, in summation: Using filters, how can I map black and grey to black and white? [23:16] <justinX> dvir: oh so some of the frames in the animated gif are longer? [23:21] <dvir> justinX: yup! if you have ImageMagick installed, you can get it easily by running "identify -format '%T|' transparent.gif" [23:21] <dvir> the first one is 2000ms, second is 50ms, etc etc [23:22] <dvir> (200|5|5|5|5|10|10|10|10|10|10|10|200, separated by | for each frame, multiply by 10ms) [23:43] <dvir> justinX: if you said anything after what I said, I got disconnected apparently :\ [23:50] <xreal> llogan: for archive use. [23:50] <xreal> llogan: I can play MKV anywhere and "output" the scream easily to VOB when I need it. [00:00] --- Thu Jan 8 2015

1 0

ffmpeg-devel.log.20150106
by burek 07 Jan '15

07 Jan '15

[00:35] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:42411a85b78c: avfilter/vf_yadif: detect telecine content [01:26] <nevcairiel> videolan git down again? at least the website seems unreachable [01:33] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:a79ac73b631a: avfilter/vf_idet: Add analyze_interlaced_flag mode [02:17] <jamrial> the repository seems to work. it's just the website [05:08] <compn> wow [05:08] <compn> amd getting in on that opencl action [05:09] Action: compn remembers everyone hating on opencl support [05:12] <jamrial> well, it is kind of broken, in a way [05:13] <jamrial> works wonders for lavfi filters, but the installed header is broken for API users [05:16] <jamrial> since it tries to #include config.h :p [05:18] <compn> :p [05:18] <compn> there is only ffmpeg, there is no api users! :P [05:18] <compn> is/are [05:44] <cone-342> ffmpeg.git 03Dale Curtis 07master:db42d93a61be: mov: Fix overflow and error handling in read_tfra(). [05:44] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:3859868c7531: avformat/mov: fix integer overflow in mov_read_udta_string() [05:44] <cone-342> ffmpeg.git 03Dale Curtis 07master:3ebd76a9c575: mov: Fix negative size calculation in mov_read_default(). [05:56] <jamrial> there, patch sent [08:13] <wm4> kierank: in addition to mplayer, there's also szap-s2 (some dvb tuning utility), which will result in ts streams without PMT [10:21] <ffbuilder> Hi [10:27] <durandal_1707> ubitux: see f929ab0569ff31ed5a59b0b0adb7ce09df3fca39 for dvdsubdec and first hunk [10:28] <ubitux> ah, dvdsub [10:28] <ubitux> yes it looks wrong [10:29] <ubitux> but it's changed [10:29] <ubitux> i mean, here at least [10:30] <nevcairiel> it was fixed in the merge apparently [10:40] <ffbuilder> I have question about libfdk-aac [10:42] <ffbuilder> If libfdk-aac build with --enabled-shared, no license problem on ffmpeg? [11:11] <rcombs> you'll get nonredistributable binaries [11:12] <cone-150> ffmpeg.git 03Paul B Mahol 07master:145a84717b62: lavfi: check av_strdup() return value [11:15] <ffbuilder> hm.. thanks i will find non redistributable [11:16] <ubitux> > The 1.02 scale factor is black magic, but produces correct results. I don't know what nvenc is doing [11:16] <ubitux> + nvenc_ctx->nvenc_cfg.darHeight = 1024 * avctx->height * 1.02; [11:16] <ubitux> haha wat :D [11:17] <wm4> lol [11:18] <wm4> darHeight is float? [11:23] <ubitux> wm4: looks like it's u32 [11:25] <wm4> weird [11:26] <ubitux> maybe it should be 1.024 [11:39] <nevcairiel> i imagine the 1024 is there to make it kind-of fixedpoint [11:40] <nevcairiel> both factors in the AR contain 1024, so it just scales the value up, doesnt change it [11:51] <ubitux> Past duration 1.000000 too large [11:51] <ubitux> Last message repeated 4273 times [11:51] <ubitux> mmh. [11:58] <saste> ubitux, do you know how to add duration information in AVFrame? [11:58] <saste> right now we have pts, pkt_pts, and pkt_duration [11:59] <ubitux> no idea [11:59] <saste> in lavfi we only use the pts information, and we ignore pkt_pts [11:59] <ubitux> cast it in AVSubtitle [11:59] <ubitux> @_@ [11:59] <ubitux> but yeah no, i have no idea [11:59] <ubitux> you usually just diff with the next frame [12:00] <saste> ubitux, yes, but sometimes it's usefult to have the duration information [12:00] <saste> it is a main limitation at the moment [12:00] <wm4> if you add duration, be sure not to make it int [12:01] <wm4> and to make a difference between 0 duration and no duration [12:01] <wm4> 2 past API mistakes [12:01] <saste> especially considering that filters such as setpts doesn't know the next PTS and can't cache [12:01] <wm4> though in general I'd consider the duration info lavf outputs useless [12:01] <wm4> better don't rely on it? [12:01] <wm4> the buffersink filter could emulate it [12:02] <nevcairiel> yeah duration info for video is 90% useless [12:02] <nevcairiel> and for audio its already defined [12:02] <nevcairiel> (nb_samples and samplerate, etc) [12:02] <saste> nevcairiel, why, because it is unreliable? [12:02] <wm4> I wonder when will they invent variable samplerate audio [12:02] <nevcairiel> wm4: you mean aac ssr? [12:03] <wm4> really? stupid reality [12:03] <nevcairiel> saste: unreliable, inexistant, hence not useful [12:03] <nevcairiel> the only w ay to get reliable duration info is by diffing the frames [12:03] <nevcairiel> frames timestamps* [12:05] <nevcairiel> wm4: well technically it uses different sample rates for different frequency bands, somewhat similar to how opus splits music and speech coding .. but yeah. [12:05] <nevcairiel> a decoder would probably just resample that to the highest possible rate [12:06] <nevcairiel> luckily aac ssr is barely used anywhere [12:06] <wm4> using different samplerates for different channels would also be a great way to break absolutely everything [12:07] <ubitux> do we support odd sample rates? [12:09] <mattlea> Hi all, I'm trying to create a filter that does a FFT followed by an IFFT, but I get horrible artifacts. Here's the code: http://pastebin.com/7M5ma2hf [12:13] <mattlea> If anyone has any hints that'd be great :) [13:38] <ubitux> mattlea: your query_format is correct? [13:40] <ubitux> mattlea: also use the public api (av_fft_init()) and check return value [13:41] <ubitux> same for other functions, just use the public api [13:41] <mattlea> ubitux: ah, yep, I'm using floats, I need to cast extended_data [13:41] <mattlea> ubitux: will do, thanks! [13:41] <ubitux> what did you request in query_format? [13:43] <mattlea> AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE [13:46] <ubitux> ok [15:28] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:62eafb6a0dd6: avformat/mov: Clear array to prevent potential out of array read from av_dlog() [15:28] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:383c6a3a07b7: avformat/mov: use av_freep() to avoid leaving stale pointers in memory [15:28] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:6e70e4aca506: avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory [15:28] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:59ca6e258694: avformat/hlsproto: Use av_freep, to avoid leaving stale pointers in memory [15:29] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:da0dadbee4e9: avfilter/buffer: use av_freep() to avoid leaving stale pointers in memory [15:29] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:bbfca8e84b0e: avformat/utils: Clear pointer in ff_alloc_extradata() to avoid leaving a stale pointer in memory [15:29] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:369b4cd4120b: cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file() [17:41] <user123> Hello [17:41] <user123> I have some question [17:42] <compn> user123 : then ask them. [17:43] <compn> this is the developer channel, if you have user questions ask in #ffmpeg :) [17:43] <user123> I want build ffmpeg with libfdk_aac [17:43] <compn> ok [17:43] <user123> I heard libfdk_aac have license problem [17:43] <kierank> I don't understand how the yadif telecine stuff works [17:44] <kierank> it just detects soft pulldown? [17:44] <user123> Somepeople said if distribute with enable-shared, then no license issue [17:44] <kierank> user123: wrong [17:44] <user123> oh [17:44] <user123> wrong? [17:44] <compn> there is a license conflict [17:44] <compn> i dont remember the details however. [17:44] <compn> kierank : you dont understand if it works, or how it works ? [17:44] <kierank> compn: both [17:45] <iive> yadif have inv-telecine support? [17:45] <user123> Also this is license conflict right ? http://oss.netfarm.it/mplayer-win32.php [17:45] <kierank> yes [17:46] <iive> doesn't look like it. [17:46] <compn> kierank : you are referring to avfilter/vf_yadif: detect telecine content ? [17:46] <user123> If purely for free application, also license conflict? [17:46] <kierank> compn: yes [17:47] <kierank> user123: yes licence conflict [17:47] <user123> ugh.. ok thank you answer... [17:47] <compn> kierank : user123 is wondering if theres a way to build against libfdk but not distribute the dll and let the user download the .dll himself ? [17:47] <cone-150> ffmpeg.git 03wm4 07master:ea7af58fc6b3: avformat/id3v2: support USLT tags [17:47] <kierank> compn: dunno [17:48] <compn> user123 : you may want to ask the other builders of ffmpeg (like zeranoe or sherpya ) about libfdk-aac [17:48] <compn> and how they handle it [17:48] <compn> or ask on mailinglist [17:48] <compn> or stick around here [17:48] <iive> https://trac.ffmpeg.org/wiki/Encode/AAC#fdk_aac [17:49] <iive> I would speculate that it conflicts with gpl, but it "might" be ok with LGPL [17:49] <user123> for example... I just distribution ffmpeg (enabled-libfdk aac and shared) exclude fdk aac dll file [17:50] <user123> then no problem? [17:50] <Daemon404> iive, it is not [17:50] <Daemon404> i have confirmed with SFLC in the past [17:50] <Daemon404> it conflicts with all LGPL and GPL [17:50] <compn> kierank : i think the telecine commit only 'fixes' the sample in that it detects telecine, and then disables yadif , http://www.mail-archive.com/ffmpeg-devel@ffmpeg.org/msg07592.html [17:50] <iive> x264 is gpl, so you might not have both at the same time. [17:50] <compn> but i could be misreading the commit and thread there. [17:50] <iive> also, what Daemon404 said. [17:51] <iive> user123: this only matters if you want to distribute the build. you can use it yourself. [17:51] <compn> Daemon404 : bonus points if someone puts the sflc opinion in the wikipage for fdk ? :P [17:51] <user123> hm.. [17:51] <wm4> what resolution do DVB subs have? I have a sample here that seems to use 720x576 or so, but the DVBSUB_DISPLAYDEFINITION_SEGMENT indicates 1920x1080 [17:53] <user123> some people enable libfdk-aac with shared, and build with disable-nonfree(maybe configure modified?) and don't distribution libfdk .dll file. [17:53] <user123> it also conflicts license right? [17:53] <compn> user123 : i think what we are saying is that you are in violation of fdk's license if you distribute fdk with gpl software. [17:54] <compn> or gpl license, i am not sure which one conflicts... [17:54] <wm4> ah no I was wrong [17:54] <compn> although i dont think ffmpeg goes after such cases :P [17:54] <user123> libfdk aac with gpl [17:54] <wm4> sometimes DVBSUB_DISPLAYDEFINITION_SEGMENT is just missing [17:55] <user123> ok... last question... [17:56] <user123> I build ffmpeg with enable libfdk aac and libfdk aac with enable-shared... when I build done, and execute ffmpeg [17:56] <user123> then get error "can not find libfdk-aac-1.dll" [17:57] <JEEB> not in library load path then :P [17:57] <user123> I want remove dependency [17:57] <iive> hum... is the problem with patents? [17:58] <user123> http://oss.netfarm.it/mplayer-win32.php - this mplayer build ffmpeg with libfdk aac, enable shared [17:58] <Daemon404> the problem is the license has restrictions based on patents which are incompatible with (L)GPL [17:58] <user123> but no problem when do not have libfdk-aac-1.dll [17:58] <Daemon404> any build linking to fdk is illegal to distribute [17:58] <Daemon404> full stop. [17:58] <JEEB> user123, then that person has patched in a loader :P [17:58] <user123> whats that? [17:58] <JEEB> dynamic loading of a library [17:58] <iive> in theory, gpl also forbids distribution of code covered by patents... [17:59] <user123> any reference data? [17:59] <JEEB> user123, but even that is not correct license-wise :P [17:59] <user123> oh.. [18:00] <iive> but at least it requires that limitation to be imposed by court. [18:00] <nevcairiel> well its rather opaque if a simple loader is actually still wrong as long as you dont distribute the libfdk binary with it .. you get into areas that noone can really verify [18:01] <user123> aw.. ok. [18:01] <user123> recently how about experimental aac codec? improved quality or same? [18:01] <JEEB> I'm not sure what would be the case if you made your software use a completely LGPL-licensed lavf/lavc, and had a separate module in your application that uses fdk-aac, and you have fully aligned yourself with the fdk-aac license [18:02] <user123> ok...hm... [18:02] <user123> too low quality another aac codecs :( [18:03] <JEEB> because if you can use LGPL lavf/lavc in a proprietary application I guess as long as you are adhering to the fdk-aac license you should be able to actually use it? But I don't know these things and at this point you are already getting into lulzy territory [18:03] <JEEB> of course if you want to use LGPL lavf/lavc and corporate licensed libx264 I think you have to do something similar? [18:03] <JEEB> because lavc libx264 wrapper makes you enable gpl [18:04] <user123> I'm so sad hm,,, [18:05] <wm4> blame Fraunhofer [18:06] <nevcairiel> switch to opus [18:06] <nevcairiel> :D [18:06] <user123> yea.. [18:06] <user123> opus? [18:06] <user123> Opus have aac? [18:06] <JEEB> no [18:06] <JEEB> it was in relation to the fact that Opus was standardized for broadcast [18:07] <JEEB> and there'll be a mp4 spec for it soon as well [18:07] <user123> ok.. I need good quality aac codec... [18:07] <nevcairiel> it was in relation that opus is decent quality and free to use [18:07] <JEEB> yes, that too [18:07] <JEEB> but if you need AAC you are pretty fucked :P [18:07] <JEEB> I mean, the lavc encoder isn't /that/ bad any more [18:07] <JEEB> but it's not exactly spectacular [18:08] <JEEB> for lower rates the vo-aacenc apachev2 one might be "alright" [18:08] <JEEB> but in general I've seen the lavc encoder beat vo-aacenc [18:08] <nevcairiel> vo-aacenc is terrible [18:08] <nevcairiel> as are all the other options in lavc, the native one is already the best short of fdk of course [18:09] <nevcairiel> should poke the mega ticket again [18:09] <JEEB> yeah [18:09] <user123> sorry I'm beginner... [18:09] <user123> lavc encoder is acodec=aac ? [18:11] <JEEB> yup [18:12] <user123> but maybe much worse than libfdk aac ? [18:13] <JEEB> it is worse then fdk-aac, but there is no simple way for you to actually use it :P [18:14] <user123> yea.. [18:14] <JEEB> you can forget about it pretty much [18:14] <nevcairiel> you can use it yourself just fine, you just cannot distribute it [18:14] <nevcairiel> i think its weird that debian of all places actually has it packaged [18:15] <nevcairiel> strong lobbying going on i guess? [18:15] <user123> my object is distribute [18:15] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:d2838f09d6dc: cmdutils.c: Use av_realloc_array() [18:15] <user123> make good ffmpeg based encoder like megui [18:16] <user123> I need think about use qaac with pipe [18:16] <user123> clearly qaac better quality than libfdk-aac.. pretty many difference.. [18:17] <Daemon404> it is not better actually [18:17] <user123> or maybe just my feel :) [18:17] <Daemon404> some ABX tests were done and it more or less matches it [18:17] <Daemon404> it juts sounds slightly different [18:17] <Daemon404> but not in a better or worse way [18:17] <Daemon404> or so im told. [18:17] <Daemon404> i dont have the golden ears. [18:17] <nevcairiel> qaac is at least similar, both have pretty good behaviour, but both are practically unusable for open-source [18:17] <user123> yea... [18:18] <nevcairiel> psycho-acoustics are hard [18:19] <user123> If use pipe, can't utilize with ffmpeg's trim feature (-ss , -to) . sad about that [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:05bc6f8ba653: avcodec/h264_slice: Clear table pointers to avoid stale pointers [19:35] <cone-150> ffmpeg.git 03Carl Eugen Hoyos 07release/2.3:d4c70c8b502a: lavc/utils: Make pix_fmt desc pointer const. [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:6ed35a66745e: avcodec/options_table fix min of audio channels and sample rate [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:b89b136c001b: avcodec/utvideodec: fix assumtation that slice_height >= 1 [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:66261cfa7789: avcodec/wmaprodec: Fix integer overflow in sfb_offsets initialization [19:35] <cone-150> ffmpeg.git 03Lukasz Marek 07release/2.3:bb2c09310cbd: lavu/opt: fix av_opt_get function [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:693d0d3ac520: avformat/hlsenc: Free context after hls_append_segment [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:17ff5d3f8826: doc/APIchanges: Fix some wrong versions [19:35] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:dca70c593178: swscale/x86/rgb2rgb_template: handle the first 2 lines with C in rgb24toyv12_*() [19:58] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:8327bef1c920: Update for 2.3.6 [19:58] <cone-150> ffmpeg.git 03Michael Niedermayer 07release/2.3:db27f50e0658: avformat/mov: Fix mixed declaration and statement warning [20:10] <cone-150> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.3.6': unknown revision or path not in the working tree. [20:10] <cone-150> Use '--' to separate paths from revisions [20:10] <cone-150> refs/tags/n2.3.6:HEAD: cmdutils.c: Use av_realloc_array() [21:36] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:a0a7b154bb49: tools/ismindex: Keep count fields consistent with arrays in case of errors [21:36] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:dede2f7fc444: tools/ismindex.c: Use av_realloc_array() [21:44] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:07a0c0f00050: opt: check memory allocation [21:44] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:b20e79dbe998: Merge commit '07a0c0f0005072d115ace61e60f46be68582cc3a' [21:51] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:1dd797e3c9f1: swscale: check memory allocations [21:51] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:90de28befd88: Merge commit '1dd797e3c9f179f957316a0becbec048b42df8aa' [22:01] <cone-150> ffmpeg.git 03Diego Biurrun 07master:ca09effb01e1: ffv1: Drop unnecessary casts and const qualifiers to match function signatures [22:01] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:f4cc19cdd082: Merge commit 'ca09effb01e126b0ac74ff3de70a475423ddee82' [22:12] <cone-150> ffmpeg.git 03Diego Biurrun 07master:0352ff102d62: ffv1: const correctness for encode_rgb_frame() [22:12] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:3b3782d74e5a: Merge commit '0352ff102d62ee94e79e0baaf64d5ad4e66f907b' [22:12] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:9198397115bf: avcodec/ffv1enc: Fix incompatible pointer type warning [22:31] <cone-150> ffmpeg.git 03Stefano Pigozzi 07master:2cef68da69a1: vda: error out if decoded CVPixelBuffer is empty [22:31] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:2ab6deb36f19: Merge commit '2cef68da69a17ed09c313ba3c3850ec1cc0a80e0' [23:40] <cone-150> ffmpeg.git 03Agostino Sarubbo 07master:fe0829986833: build: Use -Werror=format-security [23:40] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:696141e89819: vf_interlace: use image width rather than linesize [23:40] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:30cf755548a9: Merge commit 'fe082998683334e7f573cfe3876810fb80c5249c' [23:40] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:f931b1ef450e: Merge commit '696141e898193311c994b399a8dc60713709092f' [23:40] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:ac3f6429ba5b: vfilter/vf_tinterlace: Fix issues with linesize and cols [23:51] <cone-150> ffmpeg.git 03Vittorio Giovara 07master:046f75a97070: vf_interlace: also assert for height [23:51] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:6279cb3597cc: Merge commit '046f75a970701b4c947d38bfd2186dcc5f2ddae2' [00:00] --- Wed Jan 7 2015

1 0

ffmpeg.log.20150106
by burek 07 Jan '15

07 Jan '15

[01:47] <pagios> how can i merge two mp4 files? [02:10] <c_14> define "merge" [02:10] <c_14> Do you mean concatenate? [02:11] <xreal> Is MP4 a streaming format? When writing to MP4, I can't open the file the same time in Windows. When writing to MKV, it works without a problem. [02:14] <pagios> c_14: yes concat [02:14] <pagios> c_14: i have 2 files that i would like to combine into [02:14] <pagios> 1 [02:15] <c_14> https://trac.ffmpeg.org/wiki/Concatenate [02:16] <c_14> xreal: not sure what you mean with streaming format, but you can only play an MP4 once the entire file has been written [02:16] <c_14> The link was for pagios [02:16] <xreal> c_14: With "streaming format" I mean: read while writing to it. I thought, MP4 is capable of it. [02:16] <c_14> nope [02:16] <xreal> c_14: So MKV is a better way to use? [02:17] <c_14> yep [02:17] <c_14> Matroska is better at everything, except for compatibility. [02:19] <pagios> c_14: is aac installed by default with ffmpeg? [02:19] <pagios> the file is h264/aac [02:19] <xreal> c_14: Yeah, that's my main problem... my DVD player can't play MKV, but my Raspberry can :) [02:20] <c_14> pagios: yep, ffmpeg has an internal encoder/decoder [02:20] <pagios> oh [02:20] <pagios> libav does not [02:20] <muken> xreal: mp4 "can" be streamable [02:21] <c_14> muken: are you talking about fragmentation? [02:21] <muken> c_14: yes [02:21] <xreal> muken: gimme more details :D [02:21] <pagios> you can do progressive download with mp4 only [02:21] <pagios> unless you have a smart player [02:21] Action: c_14 has never really looked into fragmentation, but does know that many clients don't support it [02:23] <muken> mp4 can be streamable without fragmentation if optimized for progressive download, i.e. moov atom precedes mdat atom [02:23] <xreal> frag_duration works, thanks! [02:23] <muken> but cannot be __live__ streamable without fragmentation [02:24] <xreal> let me play with -movflags +faststart [02:25] <c_14> xreal: that won't help for live streaming/playing while it's still encoding [02:25] <c_14> And I'm not sure what that option does when fragmentation is enabled. [02:25] <xreal> c_14: let me try :) [02:25] <pagios> why not HLS? [02:25] <c_14> pagios: I think he just wants to start watching a video while he's encoding it. [02:25] <c_14> And then wants the file to be around afterwards as well. [02:26] <xreal> pagios: c_14 is right. [02:26] <xreal> Perhaps I can use -f tee and pipe it to ffplay ? [02:26] <pagios> you can encode to hls at the same time and watch the fragments [02:27] <xreal> pagios: does HLS need a running webserver? [02:27] <pagios> yes [02:27] <xreal> pagios: that's overkill for my use. [02:27] <pagios> or youcan do an avplay playlist.m3u8 [02:27] <pagios> if locally stored [02:27] <c_14> pagios: libav does have an aac encoder/decoder [02:27] <c_14> xreal: you could just use the tee pseudo muxer, yes [02:28] <c_14> either with a pipe or a udp stream or something [02:31] <xreal> HLS also works, but I don't like the small .ts files which get created :D [02:48] <pagios> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2461ac0] Invalid SampleDelta -18810 in STTS, at 22616 st:1 [02:48] <pagios> [mp4 @ 0x254dfc0] Non-monotonous DTS in output stream 0:1; previous: 61529079, current: 61505344; changing to 61529080. This may result in incorrect timestamps in the output file. [02:49] <pagios> any idea about these errros [02:49] <c_14> That looks like a warning, not an error. [02:50] <pagios> ignorable? [02:50] <c_14> If the encoding process doesn't die and the output is playable, ignore it. [02:50] <muken> input timestamp is invalid. libavformat interpolates the timestamp [02:51] <c_14> ^ if you wanted a more technical explanation [02:52] <pagios> timestamp invalid can be due to slow uplinks right [03:16] <xreal> msg NickServ identify fewih32ihfid [03:16] <c_14> xreal: you might want to change your NickServ password [03:17] <xreal> c_14: I already did, this was just a joke :) [03:17] <xreal> When downmixing AC3 to stereo, the effects are always too loud. I hate this :( [03:17] <c_14> AC3 is not a channel layout [03:17] <xreal> I know... [03:18] <c_14> Did you mean, downmixing 5.1 (or whatever) to stereo? [03:18] <xreal> yeah, sure. AC3 5.1 sound from DVD [03:19] <c_14> What command are you using? [03:19] <xreal> c_14: 1 sec [03:19] <xreal> -strict experimental -c:a aac -b:a 240k -ac 2 [03:21] <c_14> Find the channels that have the effects on them, then use the pan filter to downmix while making sure those channels are quieter [03:21] <c_14> or use the compand filter to apply drc [03:21] <xreal> c_14: that's not easy at all :( [03:23] <c_14> I believe effects are normally on FL,FR and BL,BR so pan='stereo|FL<0.5*FL+FC+0.4*BL|FR<0.5*FR+FC+0.4*BR' might do it/help [03:25] <xreal> c_14: okay, let me try this. DRC works like this? -drc_scale between >= 0 ? [03:26] <c_14> I'd probably start around 0.5 and increase/decrease as needed [03:27] <xreal> ok [03:45] <chan1> Fjorgynn: I mean what is the ffmpeg command to see the video and audio codec type of a file? (without converting it) [03:45] <c_14> chan1: if memory serves, his command was correct. you can also use ffprobe [03:46] <xreal> Can I use "q:v" with x264? Is it an alias of qp ? [04:01] <c_14> eeeeeeeeh, -qscale is ignored, -crf is recommended. [04:01] <c_14> -q:v being an alias for -qscale [04:53] <xreal> Can I use "q:v" with x264? Is it an alias of qp ? [04:53] <xreal> and what about crf? :D [04:57] <c_14> eeeeeeeeh, -qscale is ignored, -crf is recommended. [04:57] <c_14> -q:v being an alias for -qscale [04:57] <c_14> https://trac.ffmpeg.org/wiki/Encode/H.264 [05:02] <chan1> my Samsung TV supports avi file containing h.264 and ac3. [05:02] <chan1> I converted my Sony Camcoder output mts file containing h.264 and ac3 [05:02] <chan1> into avi file containing h.264 and ac3 [05:02] <chan1> using command 'ffmpeg -i file1.MTS -c copy file1.avi (output codec is just copy) [05:02] <chan1> and the conversion is fast as expected but my TV cannot play it. [05:02] <chan1> I don't know why. Using ffprobe, I see some differences in the file codec format [05:02] <chan1> Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc [05:02] <chan1> => Video: h264 (High) (H264 / 0x34363248), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 14783 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn, 59.94 tbc [05:02] <chan1> Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s [05:02] <chan1> => Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/s [05:02] <chan1> As you see, some values changed (in parentheses) and maybe that's the reason. [05:02] <chan1> I'm not a video/audio/ffmpeg expert.. [05:02] <chan1> Can anybody suggest any additional option to preserve as many parameters as possible? [05:02] <chan1> Thanks in advance and sorry for the long question. [05:02] <xreal> c_14: okay, qscale for xvid/divx and crf for x264/x265 ? [05:03] <c_14> xreal: yep [05:06] <c_14> chan1: does your tv support the high profile? [05:06] <c_14> might only support main (or if it's terrible baseline) [05:06] <chan1> c_14: yes I'm sure. [05:06] <c_14> Do you have a video that you know works? [05:07] <c_14> Can you pastebin the ffprobe from that video? [05:08] <chan1> sorry I'm at work and I have go to a meeting in a minuite. I'll get back to you later (hope you don't miss it :)) [05:10] <xreal> c_14: Is there an easy way to figure out, if source is interlaced? [05:11] <c_14> use the idet filter [05:13] <xreal> c_14: good idea, thanks [05:39] <xreal> c_14: When opening my DVB-S stream in ffplay, the resolution seems to be smaller. The same input on ffmpeg looks great. [05:40] <xreal> even when piping from ffmpeg to ffplay, it's looking ugly. [05:44] <c_14> What about other players? [06:49] <xreal> c_14: you mean piping ? [06:50] <c_14> Try playing the stream/ffmpeg output with another player. [06:50] <xreal> hmm, need to try how to do this :) [06:57] <xreal> c_14: looks nice in VLC [06:57] <c_14> Maybe it's a bug in ffplay, what version are you running? [06:58] <xreal> c_14: ffplay version N-68826-g504267f [06:58] <xreal> let me try mplayer [07:02] <chan1> c_14: sorry to interrupt, but this is the ffprobe info of a file that can be played at my TV. [07:02] <chan1> ffprobe version N-68826-g504267f Copyright (c) 2007-2014 the FFmpeg developers [07:02] <chan1> built on Jan 2 2015 22:18:10 with gcc 4.9.2 (GCC) [07:02] <chan1> configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --ena ble-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --e nable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme [07:02] <chan1> --enable-lib gsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencor e-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enabl e-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-l ibtheora --enable-libtwolame [07:02] <chan1> --enable-libvidstab --enable-libvo-aacenc --enable- libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-l ibwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --ena ble-lzma --enable-decklink --enable-zlib [07:02] <chan1> libavutil 54. 16.100 / 54. 16.100 [07:02] <chan1> libavcodec 56. 19.100 / 56. 19.100 [07:02] <chan1> libavformat 56. 16.102 / 56. 16.102 [07:02] <chan1> libavdevice 56. 3.100 / 56. 3.100 [07:02] <chan1> libavfilter 5. 6.100 / 5. 6.100 [07:02] <chan1> libswscale 3. 1.101 / 3. 1.101 [07:02] <chan1> libswresample 1. 1.100 / 1. 1.100 [07:02] <chan1> libpostproc 53. 3.100 / 53. 3.100 [07:03] <xreal> c_14: mplayer looks gr8 [07:04] <c_14> chan1: first of all, please use a pastebin service for anything more than 2 lines, second that's only stderr. Where's stdout? [07:06] <c_14> xreal: might be an ffplay bug then [07:06] <xreal> c_14: okay, let me try different versions. [07:21] <xreal> c_14: how do I pipe audio? rawvideo does video only :) [07:22] <c_14> just pick a format that supports video + audio, mpegts or matroska or something [07:23] <xreal> doesn't work for pipe, I think ? [07:23] <c_14> It does. [07:24] <xreal> c_14: I hate you :) [07:24] <xreal> why do you know anything? [07:25] <c_14> Practice makes perfect. [07:26] <xreal> c_14: it looks like ffplay playbacks the content in half the resolution [07:26] <xreal> and scales it up then [07:27] <c_14> If you can get a sample, submit a bug report. [07:29] <xreal> okay, I'll do it tomorrow [07:38] <xreal> c_14: when using HLS, the small files automatically get created. Can I create them on demand when accessing the playlist? [07:38] <xreal> c_14: or is the HLS function about generating those files? [07:47] <chan1> c_14: here's the original and two converted files' ffprobe info. http://pastebin.com/yRYBV9pY [07:47] <chan1> The TV can play only the .mkv file converted using -acode ac3_fixed and cannot play .avi file that was converted with -c copy. .MTS is the original. [07:48] <chan1> c_14:hope you could guess the problem.. [09:06] <Elirips> Is it normal that every ffmpeg instance uses exactly 528m virtual size or should I worry about ffmpeg loosing memory? [09:07] <Elirips> I'm calling ffmpeg like this'ffmpeg -i $URL -an -c:v libx264 -b:v 128K -x264opts keyint=$X264_KEYINT -r $FPS -f flv $NGINX 2> $LOGFILE &' [10:14] <Cas> Hi, I try to send my Decklink input to v4l2loopback and my audiocard but the two slowly drift away from each other. [10:14] <Cas> Any tips how I can help this? [10:15] <Cas> my command is: `ffmpeg -f decklink -i 'Decklink SDI' -f v4l2 /dev/video0 -f alsa -ar 4800 -ac 2 -acodec pcm_s16le hw:0` [10:16] <sfan5> I have no idea whether this will help, but try -async 1 [10:17] <Cas> sfan5: thanks for the suggestion, tried that by didn't help [10:40] <rhagu> hi, when I install ffmpeg using this guide: http://ask.xmodulo.com/compile-ffmpeg-ubuntu-debian.html will there be support for h264 and vc-1 mkvs? [10:44] <Mavrik> encoding or decoding? [10:51] <zap0> i have some webm, i'd like to make a small(thumbnail) 3-frame GIF.. how? [10:58] <rhagu> Mavrik I will use this command: ffmpeg -v warning -i "$X" -map 0:v -map 0:a -threads 3 -f null - [11:04] <Mavrik> rhagu, that doesn't answer my question. [11:04] <Mavrik> zap0, ffmpeg -i blah.webm -vframes 3 test.gif ? [11:05] <zap0> thanks, shall try in a moment. [11:06] <rhagu> Mavrik I think this commands mean decoding and encoding to /dev/null, not sure whether ffmpeg really encodes if it goes to null [11:07] <Mavrik> You still didn't answer the question if you want to decode or encode those formats. [11:07] <Mavrik> And that command line also doesn't tell anything abou it. [11:07] <Mavrik> (it's also hideous, what are you even trying to do with that? O.o) [11:08] <rhagu> I want to decode and encode both h264 and vc-1 [11:21] <Fjorgynn> Good morgnin [11:21] <Fjorgynn> morning [11:42] <feliwir> sorry that i need to ask this here, but does someone know if there's a program with that i can download ultraviolet videos on my pc? [11:46] <Fjorgynn> :) [12:41] <c_14> chan1: I'm sorry, I have no idea. Your TV is probably looking for something very specific, but I don't know what. [12:42] <Mavrik> using AAC and Main video profile is probably a better idea [13:02] <eyal-ff> How can I know in real-time what is my bitrate? in live streaming [13:04] <Fjorgynn> Welcome [13:11] <Mavrik> eyal-ff, em, it shows up in the output line. [13:15] <eyal-ff> I know that, I was wondering if there is a way to know it from ffprobe or the output line is the only way? [13:23] <c_14> You can ffprobe the stream and it will give you the average bitrate for the samples it collected [13:38] <eyal-ff> Thanks, I'll try [14:20] <Popara> Guys i have noticed huge differences in FFmpeg 2.5.x even in Simple Streams whereas in the FFmpeg 2.4.x everything was perfect [14:21] <Popara> it is hard for me to explain what my current problems are with details but in a few words some streamings are being closed without a reason in 2.5.x, some others say some errors and some other just lags [14:21] <Popara> in v2.4.x the exactly the same streams are working very well [14:21] <Popara> and i tried that in the same time, it can't be wrong [14:21] <Popara> what so you suggest me to do [14:22] <Popara> i stream via HLS [14:39] <ubitux> Popara: can you try to git bisect? [14:46] <Popara> @ubitux i grabbed the latest static binary today [14:46] <Popara> and same happened [14:49] <Popara> I would really like to report the bugs, but in this case what i will say, that it just lags? Nobody will reply. But i'm sure that this happens [14:49] <ubitux> that's why you can try to git bisect yourself [14:49] <ubitux> to point out the commit that caused this [14:51] <Popara> aaa git bisect now i get it [14:57] <Popara> i have completely no idea how to use it :( [15:19] <c_14> Popara: http://git-scm.com/book/en/v2/Git-Tools-Debugging-with-Git#Binary-Search http://git-scm.com/docs/git-bisect [15:51] <Cas> How do I know if I use the XCB screengrabber? I did compile without --x11grab but its a little unclear what is grabbing my display now [16:59] <shevy> is there a short-variant of ffmpeg -i that just tells me which audio codec is in use? e. g. whether I have AAC or .mp3 in a .mp4 file? [17:01] <Ultima> im not sure with ffmpeg but I know mediainfo will tell you [17:02] <Mavrik> shevy, ffprobe [17:02] <Mavrik> also accepts formatting options so you can get output in JSON and other formats for machine consumption [17:02] <shevy> ah [17:04] <schiho> Is there a list what options are set when using -preset ultrafast -tune zerolatency ? [17:05] <SirDarius> shevy: with mediainfo, use something like `mediainfo --Inform="Audio;%Format%" file.mp4` this is as short as you can get [17:05] <Mavrik> schiho, you'll probably have to look at x264 source :/ [17:07] <shevy> hmm mediainfo is an external command? [17:07] <Mavrik> it's a separate project not related to ffmpeg :) [17:09] <schiho> Marvik: Oh i see, i am just searching for a way to support more "seekability" of videos, pushing more intraframes... i can set the gopsize to 1 but i think the option "keyint" is the right thing but i don't know how to type this to the ffmpeg command [17:10] <Mavrik> uhh [17:10] <Mavrik> you do know that if you set keyint to 1 you'll get video with only IDR frames? [17:10] <Mavrik> meaning it's going to be huge / ugly? [17:11] <iive> shevy: ffprobe ? [17:11] <shevy> iive yeah, I think I'll stick to ffmpeg-related commands [17:12] <schiho> Mavrik: yes you are right but marvik when i want to enable seeking i thought it is essential to have as many i frames as possible? [17:12] <schiho> Mavrik: What happens, when i click on a P Frame in the timeline, it has to go back and recalculate till where the Frame was clicked? [17:12] <Mavrik> yes. [17:13] <schiho> Is this something players do fast? [17:13] <Mavrik> But the whole POINT of video codecs is the P and B frames. [17:13] <Mavrik> If not you can just send list of JPEGs to people. [17:13] <Mavrik> You'll get in range of 100x worse compression. [17:13] <Mavrik> I frames are 30-150x as big as P and B frames. [17:13] <schiho> ah you say, sending jpegs will have better compression? [17:13] <schiho> so is one I frame bigger than a compressed jpg image? [17:14] <Mavrik> ... [17:14] <Mavrik> No. [17:14] <Mavrik> It's about the same. [17:14] <schiho> yes, and that is, instead of sending jpg's i can send i frames, i am in a local network [17:14] <schiho> so bandwith is not a big problem [17:16] <schiho> so changing gop size made the file only 2 times bigger...thats okey [17:16] <schiho> -g set to 1 [17:17] <Mavrik> well. [17:17] <Mavrik> remember to also set min-keyint [17:17] <Mavrik> or gop size will get overriden [17:18] <schiho> frame I:2035 it seems to be not overriden [17:23] <iive> keyframes are like jpeg pictures. they don't need any other info to be decoded, but they are big. very big. [17:23] <iive> that's why you want as little as possible of them. for rips, it is usually 1 keyframe per 10 seconds (or 250 frames) [17:23] <schiho> :) what does very big mean in terms of MB ? So a 1920 x 1080 image with jpg compression is about 240kb in average [17:24] <schiho> for real-time streaming it would be bad to have such a big interval? or am i overseeing an advantage? [17:24] <iive> [libx264 @ 0x346dc00] frame I:760 Avg QP:19.34 size: 47999 [17:24] <iive> [libx264 @ 0x346dc00] frame P:19845 Avg QP:22.26 size: 9884 [17:24] <iive> [libx264 @ 0x346dc00] frame B:43028 Avg QP:25.66 size: 1558 [17:25] <iive> that's SD encode. [17:26] <iive> so, yes, if you want to stream, you need to have frequent keyframes, or people would have to wait for the next keyframe to get normal image. [17:26] <iive> usually 1 per second. [17:28] <schiho> Okey, there are to problems with streaming 1) Latency 2) Wait for next clean image the latter one isn't interesting for me [17:28] <iive> h264 provides another trick... partial keyframes. aka you use intra macroblocks to refresh just a portion of the image. this way you distribute the keyframe over a number of P frames. You'd have to discard n frames before you can show image... [17:35] <BtbN> How is it possible that the same input video(a concat list) with -c copy produces 44.1kHz audio when muxed to flv, but 4800kHz when muxed to mp4? The original .ts files all show 48kHz. [17:36] <BtbN> The flv output video has weird audio issues, with it jumping and cracking a bit all the time, so i think the 44.1kHz rate is just wrong [17:37] <JEEB> flv the container limits the rate to that, but it just means "look at the stream's rate instead" with AAC [17:37] <JEEB> if you look at the flv container's specs [17:37] <JEEB> and flash seems to do exactly that, too [17:38] <BtbN> the youtube transcoder doesn't aparently. [17:38] <shevy> quick question - I can convert from .aac to .mp3 just fine? quality loss is no problem; command I use: 'ffmpeg -i test.aac test.mp3', so far it seems to work [17:38] <JEEB> yes, many implementations didn't read the spec too well [17:38] <JEEB> so they instead just encode in 44.1kHz [17:38] <BtbN> http://youtu.be/HMA-m3-k6h4?t=2h56m40s that's what it sounds like [17:39] <BtbN> it's not off by much, but doesn't sound completely right [17:39] <JEEB> anyways, ffmpeg should show it as the proper rate if it can access the stream itself [17:40] <JEEB> if it doesn't and/or something else is blindly trusting the container ratio (which is wrong), then of course there can be fuck-ups [17:40] <JEEB> but well, I have no idea what the fuck you've done and which of all the moving parts has fucked up :P [17:40] <schiho> @live so what is best for low-latency-streaming i want to have as fast as possible the image which is at the receivers side [17:40] <JEEB> I'm just saying that if you are only looking at the container audio data, then that will tell you 44.1kHz [17:40] <JEEB> because that's what the container metadata will say with AAC in flv [17:41] <BtbN> JEEB, recording a twitch livestream. So i have a massive ammount of 4 seconds .ts files, and combine them with the -f concat input. [17:41] <BtbN> For some reason that needs "-bsf:a aac_adtstoasc" [17:41] <BtbN> But works fine otherwise [17:41] <JEEB> (because the spec says "write this value, the demuxer/decoder combo will then ignore it and use the actual AAC stream" [17:41] <BtbN> ffmpeg -f concat -i "${inp}" -c copy -bsf:a aac_adtstoasc -movflags +faststart -y "${outp}" [17:41] <BtbN> is the commandline i use [17:41] <BtbN> (without the movflags for flv) [17:42] <JEEB> is the bitstream filter required or does it even work correctly for FLV output? [17:42] <BtbN> it's required, ffmpeg aborts without it [17:42] <JEEB> ok [17:42] <BtbN> and tells me to use it [17:42] <JEEB> if you decode the output with mpv or something else sane does the result work? [17:43] <BtbN> i'll have to download one part for that [17:43] <BtbN> https://bpaste.net/show/1c883ed97490 [17:43] <JEEB> well you created the files [17:43] <BtbN> on my sever [17:43] <JEEB> you should have them on hand then :P [17:43] <BtbN> *server [17:43] <JEEB> well, then grab one of them and check [17:44] <BtbN> yeah, i'm on it right now. Just takes a moment, as they are 10GB each [17:44] <JEEB> with something relatively sane like mpv, and if you are willing enough, with flash player [17:44] <JEEB> because the latter is the "reference" for FLV [17:44] <BtbN> so ffprobe reporting it as 44.1 is ok, or a bug? [17:44] <JEEB> which part of ffprobe [17:44] <BtbN> see my paste [17:44] <JEEB> container only or does it actually decode [17:45] <JEEB> because if it touches the stream then it should get the actual rate [17:45] <JEEB> if container only, then 44.1kHz should be gotten (and ignored by the player/whatever) [17:45] <JEEB> I have no idea how much ffprobe does in that case [17:46] <JEEB> I will guess that it doesn't actually decode, or there is a bug [17:46] <JEEB> also have you tried with a newer ffmpeg? [17:46] <JEEB> 2.2 series is pretty old [17:46] <BtbN> Not yet, 2.2.11 is already the "unstable" version on gentoo [17:46] <BtbN> stable is 1.2.6 [17:47] <JEEB> yes, because they haven't updated everything under gentoo to be compatible with the newer APIs [17:47] <shevy> huh [17:47] <JEEB> that said, pretty much everything a normal user uses is updated [17:47] <shevy> I have 2.5.1 here [17:47] <JEEB> so updating to a newer ffmpeg shoud be relatively problemless [17:47] <BtbN> 2.5 is available, but hardmasked [17:47] <shevy> I thought gentoo was source based [17:47] <JEEB> yes, they're all hardmasked [17:47] <JEEB> shevy, basically they're not updating their "stable" shit until _everything_ in their repos gets updated [17:48] <JEEB> there's some really minor things or whatever there, which are taking really long [17:48] <JEEB> so even debian has now been quicker at updating libav/ffmpeg than gentoo :P [17:48] <shevy> yeah. arch too https://www.archlinux.org/packages/extra/i686/ffmpeg/ [17:48] <BtbN> debian even has better support, as it can install libav and ffmpeg ad the same time [17:48] <BtbN> on gentoo it's mutualy exclusive [17:48] <shevy> I guess what someone else said is true; like when the gentoo wiki died [17:49] <BtbN> the gentoo wiki died? [17:49] <shevy> yeah [17:49] <JEEB> the last time I poked the libav person who was maintaining libav/ffmpeg in gentoo [17:49] <BtbN> looks very alive to me [17:49] <JEEB> and asked what the fuck is taking time with he just yelled at me and ragequit [17:49] <JEEB> because as always the libav/ffmpeg maintainers get all the shit [17:49] <JEEB> even though it's the projects that use those projects' libraries that are not being updated [17:49] <shevy> BtbN http://www.gentoo-wiki.info/ "Gentoo-Wiki.com has been lost and it will be rebuilt from scratch." [17:50] <BtbN> shevy, that's an inofficial wiki, and it's been down forever [17:50] <BtbN> http://wiki.gentoo.org/ is the Gentoo wiki [17:50] <JEEB> anyways, you can always build a simple ffmpeg build in your homedir :P [17:50] <JEEB> for testing [17:50] <JEEB> and trying ffprobe/remuxing with it [17:50] <BtbN> I'm not aware of any software on that machine which links against ffmpeg [17:50] <BtbN> i just use the cli tools [17:50] <BtbN> so i'll just bump it [17:50] <JEEB> k [18:02] <shevy> Unknown encoder 'libx264' - question. seems I forgot to compile that support in, will recompile later. Is there an alternative to a .mp4 codec? I am not sure how to find out which codecs will work and which not with my current ffmpeg [18:03] <c_14> mpeg4 should work [18:03] <sfan5> shevy: ffmpeg -codecs tells you which codecs are compiled in; mpeg4 is an alternative for .mp4 (it has a lower quality though) [18:04] <shevy> aha [18:05] <shevy> well the quality is fine, I only wanted to add an image to a mp3 file and call that a "video" [18:08] <sfan5> if you want to add artwork to an mp3 file use mjpeg [18:08] <sfan5> IIRC it only works with mjpeg [18:20] <kepstin-laptop> on the other hand if you want to make a "video" file that just shows a static image to e.g. upload to youtube, you're on the right track. [18:22] <shevy> hmm [18:22] <shevy> the .mp4 is much larger than the .mp3 [18:23] <shevy> can that be? if you add like only one small .png to a .mp3 file. I have the feeling I must have done something wrong... from 4 MB to 12 MB [18:24] <BtbN> did you re-encode the audio? [18:26] <shevy> hmmm [18:26] <BtbN> Paste your full command (and output) please [18:58] <kruug> Hello, I'm converting all of my video media files to MP4, and I see a lot of them have the YUV420p codec (is it a codec?) being used. My question is this: does this mean that it's not HD? As in, 420p vs 720p vs 1080p. Using WinFF, I'd like to keep the files HD if they originally were HD, but how can I tell? [18:59] <ezekiel> kruug: that's a color space designation [19:00] <ezekiel> https://en.wikipedia.org/wiki/YUV [19:00] <ezekiel> it's not necessarily linked to "HD" which is a quality designation [19:00] <kruug> How can I use WinFF without losing definition then? I don't want to have to manually find the "definition" for each individual file [19:17] <nasojlsu> ffmpeg -i filename [19:18] <nasojlsu> will give you specs [19:18] <nasojlsu> or you could use mediainfo [20:25] <swillits> I have a movie file with one video and one audio track in it. What I need to see is the start times of the individual tracks. I suspect that the audio starts before the video and need to confirm it. I'm new to ffmpeg surely somewhere in its 8 million capabilities it can show me this. [20:39] <ezekiel> swillits: my suggestion may not be the simplest answer, but I have done what you're talking about [20:39] <ezekiel> I used filters to show the "Presentation Time Stamps" (PTS) [20:39] <swillits> I just figured out that ffprobe -show_streams did the trick. I looked at the start_pts/time of each stream [20:40] <swillits> I had no idea streams could have negative start times. [20:40] <ezekiel> nice [23:43] <jookiyaya> does ffmpeg support h263 encoding? [23:44] <JEEB> I think yes [23:44] <jookiyaya> which one [23:45] <JEEB> ? [23:45] <JEEB> I think there's just one ITU-T H.263 :P [23:45] <jookiyaya> which h263 encoder [23:45] <JEEB> I would guess lavc has its own [23:45] <JEEB> libavcodec/h263enc or something along those lines? [23:45] <jookiyaya> i see [23:46] <JEEB> hmm or actually [23:46] <JEEB> I see a h263dec [23:47] <JEEB> but not h263enc [23:47] <JEEB> anyways, it would be much simpler for you to just run `ffmpeg -codecs:v |grep "h263"` [23:48] <JEEB> ah yes [23:48] <JEEB> it's in mpegvideo_enc [23:48] <JEEB> of course [23:48] <JEEB> the horribly big monster :P [00:00] --- Wed Jan 7 2015

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