January 2015 - Ffmpeg-devel-irc

ffmpeg-devel.log.20150105
by burek 06 Jan '15

06 Jan '15

[00:18] <ubitux> saste: it would be nice to have a sample accurate "aselect" :p [00:18] <ubitux> with eval it might be way too slow though [00:25] <wm4> why is asample _not_ sample accurate? [00:26] <wm4> err aselect [00:26] <wm4> seems like an achievement [00:26] <wm4> does it select by frames? [00:27] <ubitux> yes, afaik [00:27] <wm4> but frames have an arbitrary size [00:27] <ubitux> if you want to do it sample accurate you need to evaluate the expression per sample [00:27] <kierank> ubitux: can a libavfilter filter generate both audio and video at the same time? [00:27] <ubitux> kierank: yes [00:27] <kierank> next q [00:28] <ubitux> i think ebur128 actually does this [00:28] <wm4> (especially funny because ffmpeg.c/libavformat apparently makes PCM frames in some files arbitrarily huge) [00:28] <kierank> is it possible then to have a chain of filters and output an exact (but variable) number of samples per frame? [00:28] <ubitux> i guess so? [00:28] <ubitux> asetnsamples? [00:29] <kierank> I mean for each video frame I need n samples? [00:29] <kierank> basically I want to build my test pattern filter at some point [00:29] <kierank> with a lipsync test [00:29] <ubitux> i think you could do that [00:29] <ubitux> look how asetnsamples does it [00:30] <kierank> I need it to be variable though (for ntsc) [00:30] <ubitux> the filter calls ff_get_audio_buffer [00:30] <ubitux> to which you pass the number of sample you want [00:30] <ubitux> outsamples = ff_get_audio_buffer(outlink, nb_out_samples); [00:31] <kierank> the difficulty I have is how to sync my video and audio in ntsc-land [00:31] <kierank> because you don't sync by video frame in ntsc-land [00:31] <kierank> s/don't//g [00:33] <ubitux> well, you could probably write a "sync" filter [00:33] <ubitux> with an ntsc mode to start with [00:33] <kierank> it would be easy with a standalone filter, yes [00:33] <kierank> but chaining filters (to use drawbox etc) is hard [00:34] <ubitux> why? [00:34] <ubitux> also, do you need the video stream for this audio pattern? [00:34] <kierank> yes [00:34] <kierank> because I need to sync the drawing of a box or whatever with the sound [00:35] <ubitux> i mean, isn't what you want just a special cut? [00:36] <ubitux> like, you just apply a certain pattern of frame segmentation to the audio, and that's it? [00:36] <kierank> it's not that simple [00:36] <ubitux> typically, can't you hack something in asetnsamples with an eval expression? :p [00:36] <ubitux> (or just as a PoC) [00:37] <ubitux> otherwise yeah, feel free to make a filter that output both [00:37] <kierank> ubitux: afaik you can't do frame accurate drawbox on the cli [00:38] <kierank> at the moment in my code I have a moving message with sin(k*t) [00:38] <kierank> but that's not frame accurate [00:39] <ubitux> well you can add a "n" paramter in eval [00:39] <ubitux> it's a 2-3 line diff [00:40] <ubitux> it won't work with seeking (so you can't really test with ffplay), but it will work for a transcode [00:40] <ubitux> git grep 'VAR_N[^A-Z]' is your friend [00:42] <llogan> saste: Postprocessing wiki page doesn't exist [01:21] <KGB> [13FFmpeg] 15michaelni pushed 1 new commit to 06master: 02http://git.io/mca6TQ [01:21] <KGB> 13FFmpeg/06master 14907101e 15Reynaldo H. Verdejo Pinochet: ffserver: reflow compute_status()... [02:07] <KGB> [13FFmpeg] 15michaelni pushed 12 new commits to 06master: 02http://git.io/CJ50XA [02:07] <KGB> 13FFmpeg/06master 149e55130 15Reynaldo H. Verdejo Pinochet: ffserver: reindent compute_status()... [02:07] <KGB> 13FFmpeg/06master 14196bc03 15Reynaldo H. Verdejo Pinochet: ffserver: drop redundant else clause... [02:07] <KGB> 13FFmpeg/06master 14ac1940b 15Reynaldo H. Verdejo Pinochet: ffserver: reflow rtsp_cmd_setup()... [02:18] <KGB> [13FFmpeg] 15michaelni pushed 2 new commits to 06master: 02http://git.io/QmI_tA [02:18] <KGB> 13FFmpeg/06master 143e16065 15Johan Andersson: cmdutils: update copyright year to 2015.... [02:18] <KGB> 13FFmpeg/06master 141d41571 15Michael Niedermayer: Merge commit '3e160652219ff4da433f5672ae1e5f4956abb815'... [02:26] <KGB> [13FFmpeg] 15michaelni pushed 2 new commits to 06master: 02http://git.io/1tE5PA [02:26] <KGB> 13FFmpeg/06master 14f4d4e66 15Martin Storsjö: configure: Remap -L to -libpath for msvc... [02:26] <KGB> 13FFmpeg/06master 141ed5912 15Michael Niedermayer: Merge commit 'f4d4e66a24a5c9497a5b6d3c089ac58089a87428'... [02:34] <KGB> [13FFmpeg] 15michaelni pushed 2 new commits to 06master: 02http://git.io/Ls6VLQ [02:34] <KGB> 13FFmpeg/06master 14c4aa0f8 15Martin Storsjö: libavformat: Remove a misleading/incorrect comment... [02:34] <KGB> 13FFmpeg/06master 14dd4de4a 15Michael Niedermayer: Merge commit 'c4aa0f865fb2c3b28a309d1bb27fd1c24b75ba28'... [02:44] <KGB> [13FFmpeg] 15michaelni pushed 2 new commits to 06master: 02http://git.io/JKlKuw [02:44] <KGB> 13FFmpeg/06master 149f810a9 15Martin Storsjö: libavformat: Check for malloc failures in avformat_new_stream... [02:44] <KGB> 13FFmpeg/06master 14e7521d9 15Michael Niedermayer: Merge commit '9f810a9b374e0ff8e2a19fd8a7347afe2933e229'... [04:19] <KGB> [13FFmpeg] 15michaelni pushed 2 new commits to 06master: 02http://git.io/9TklZA [04:19] <KGB> 13FFmpeg/06master 1446808fd 15Martin Storsjö: movenc: Enable editlists by default if delay_moov is enabled... [04:19] <KGB> 13FFmpeg/06master 14fb22e5a 15Michael Niedermayer: Merge commit '46808fdf04ab113df374157b90b506eb3110daf2'... [05:03] <KGB> [13FFmpeg] 15michaelni pushed 1 new commit to 06master: 02http://git.io/eotSfQ [05:03] <KGB> 13FFmpeg/06master 14beedeb4 15Werner Robitza: doc/ffmpeg: mention both ffpreset/avpreset in documentation, remove superfluous example... [05:11] <KGB> [13FFmpeg] 15michaelni pushed 1 new commit to 06master: 02http://git.io/uCInAA [05:11] <KGB> 13FFmpeg/06master 14c9151de 15wm4: avcodec/dvdsubdec: fix out of bounds accesses... [06:42] <wm4> michaelni: would you still be interested in my sws_scale_frame() API patch I sent over a year ago? [06:45] <rcombs> wait since when do repo logs come from russian intelligence [06:47] <wm4> the main git repo is down [06:47] <wm4> and apparently videolan's infrastructure? [06:50] <rcombs> oh, so KGB is GitHub and work's being done directly on the mirror then [06:56] <Tim_G> rcombs: from the README of KGB (assuming one is available): "CIA.vc is dead, long live KGB.vc!" [06:57] <rcombs> if it works the way I assume it does, it's just GitHub's IRC hook on a custom nick? [06:57] <Tim_G> rcombs: I guess [06:58] <Tim_G> Libav is using https://github.com/ehamberg/simple-gitbot with the nick "lCIA" [06:58] <Tim_G> but they are equally ugly IMO [06:59] <rcombs> oh hah [06:59] <rcombs> there's the joke [07:04] <wm4> so is the KGB nick owned by someone related to ffmpeg, or is it some kind of shared service? [07:13] <rcombs> seems to be owned by v1adimir? [10:29] <ubitux> damn ffmpeg is so complex wrt timestamps [10:36] <compn> rcombs : we also use cia bot, but since videolan is down, cia is not updating... [10:37] <compn> easier to use kgb for github than reconfigure cia for github it seems [10:44] <saste> there are issues with the git server today? [10:44] <ubitux> saste: http://ffmpeg.org/pipermail/ffmpeg-devel/2015-January/167262.html [10:45] <saste> ubitux, thanks [12:27] <michaelni> wm4, do you have a link to the patch you speak of ? i dont exactly remember but i just woke up ... [12:29] <wm4> this one https://ffmpeg.org/pipermail/ffmpeg-devel/2013-September/148824.html (that's the thread) [12:30] <wm4> I didn't pursue it further, because I thought libavscale could actually happen [12:31] <nevcairiel> You should know better by now [12:46] <michaelni> wm4, yes, sure the patch looked/looks useful [15:15] <cone-342> ffmpeg.git 03Paul B Mahol 07master:91bdb244009c: libavfilter/vf_blend: do not unconditionally compile both filters [15:49] <durandal_1707> would it be okay to add (a)deinterleave filters? [16:03] <cone-342> ffmpeg.git 03Johan Andersson 07master:3e160652219f: cmdutils: update copyright year to 2015. [16:03] <cone-342> ffmpeg.git 03Martin Storsjö 07master:f4d4e66a24a5: configure: Remap -L to -libpath for msvc [16:03] <cone-342> ffmpeg.git 03Martin Storsjö 07master:c4aa0f865fb2: libavformat: Remove a misleading/incorrect comment [16:03] <cone-342> ffmpeg.git 03Martin Storsjö 07master:9f810a9b374e: libavformat: Check for malloc failures in avformat_new_stream [16:04] <cone-342> ffmpeg.git 03Martin Storsjö 07master:46808fdf04ab: movenc: Enable editlists by default if delay_moov is enabled [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:9bff052b51f2: avfilter/vf_sab: fix filtering tiny images [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:0f931b29f7e9: Factorize avpriv_mirror() out [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:58721388b89b: avformat/flvenc: omit more metadata elements with specific meaning [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:dc265e2b6b6b: avformat/rawdec: Make ff_raw_data_read_header() non static [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:d1781fd786de: avformat/rawdec: Add FF_DEF_RAWSUB_DEMUXER [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:e313735d7d7d: avformat: add dvbsub demuxer [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:ce1e10aa24cc: avformat/utils: add dvbsub to set_codec_from_probe_data() [16:04] <cone-342> ffmpeg.git 03Peter Ross 07master:f1098eb97d3c: avformat/aiffdec: only read codec tag when there is space in header [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:1d41571bbf0d: Merge commit '3e160652219ff4da433f5672ae1e5f4956abb815' [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:1ed5912e819d: Merge commit 'f4d4e66a24a5c9497a5b6d3c089ac58089a87428' [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:dd4de4a987f3: Merge commit 'c4aa0f865fb2c3b28a309d1bb27fd1c24b75ba28' [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:e7521d9771ea: Merge commit '9f810a9b374e0ff8e2a19fd8a7347afe2933e229' [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:fb22e5ae4585: Merge commit '46808fdf04ab113df374157b90b506eb3110daf2' [16:04] <cone-342> ffmpeg.git 03Werner Robitza 07master:beedeb4399fe: doc/ffmpeg: mention both ffpreset/avpreset in documentation, remove superfluous example [16:04] <cone-342> ffmpeg.git 03wm4 07master:c9151de7c425: avcodec/dvdsubdec: fix out of bounds accesses [16:04] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:0e1c827e842c: Merge remote-tracking branch 'origin/master' [17:41] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:3bf2b376e64b: avformat/dvbsub: check for 0xf0 earlier in probe, 30% faster [17:41] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:b0826b832465: avformat/dvbsub: fix indent after previous commit [17:41] <cone-342> ffmpeg.git 03Michael Niedermayer 07master:d02d4d0457f9: avformat/dvbsub: increase probe score [17:55] <durandal_1707> isn't first hunk of f929ab0569ff31ed5a59b0b0adb7ce09df3fca39 wrong? [17:57] <ubitux> how so? [18:19] <ubitux> rhaa ffs i'm so annoyed at these edit lists [18:20] <JEEB> le edit lists, such a feature, much headache [18:21] <ubitux> i started yet another approach to the problem: basically exporting the edts atom on demand [18:22] <ubitux> and building the filtergraph in app side (so cmdutils, shared with ffplay/ffmpeg) [18:22] <ubitux> (so people do not complain about the lavfi stuff in the libs) [18:23] <ubitux> the problem is that the unit of time is not in the stream unit [18:23] <ubitux> it's in the presentation unit, which is not exported [18:23] <ubitux> ...i need to add a time_base field in the AVFormatContext... [18:23] <ubitux> unless someone has a better idea [18:23] <wm4> does anyone know why id3v2.c does this retarded dance around "extra metadata", and where to add custom tag parsers that just want to write to s->metadata? [18:38] <wm4> kierank: now our problem is apparently that ffmpeg doesn't know how to deal with streams it's supposed to ignore (extra stuff like EPG etc.), any tips? [18:38] <kierank> why is ffmpeg not ignoring EIT? [18:39] <wm4> ignoring what? [18:39] <kierank> EIT is the EPG table [18:39] <kierank> in dvb [18:41] <wm4> apparently such streams are detected as unknown streams (codec "none"), and they slow down probing [18:41] <kierank> probably because of ffmpeg hacks [18:41] <kierank> when there is no pmt [18:41] <wm4> now there's a PMT (apparently) [18:41] <wm4> I know nothing about dvb [18:52] <ubitux> maybe exporting the filtergraph from the demuxer wasn't that a bad idea in the end [18:52] <ubitux> rhaaa i'm so fucking annoyed [18:54] <wm4> ubitux: be assured, it is [18:54] <ubitux> i'll need to pass a fucking tb in all the per-stream functions of the app [18:54] <ubitux> how insane is that [18:54] <wm4> an incredibly unclean hack to make things somehow work for ffmpeg.c, and only ffmpeg.c [18:54] <ubitux> wm4: i'm trying not to make a hack [18:55] <ubitux> wm4: as i said, i was basically just exporting the edit list [18:55] <ubitux> verbatim [18:55] <ubitux> so the app can implement the timeline on its own [18:55] <wm4> yes, that sounded ok [18:55] <ubitux> except it's fucking annoying because as is, it's not usable since you need the presentation [18:56] <ubitux> the time base* presentation [18:56] <ubitux> i'm thinking of just putting the time base into that side data as a first field though [18:56] <ubitux> won't be the exact atom, but it will relieve some pain for everyone [18:59] <wm4> how does L-SMASH export this stuff? [19:03] <wm4> also I'm going to implement this http://id3.org/id3v2.3.0#Synchronised_lyrics.2Ftext and everyone will hate me, including myself [19:18] <compn> wm4 : what you dont like cdg ? do we have id3 lyrics samples ? [19:18] <compn> oh wow that looks horrible :D [20:30] <cone-342> ffmpeg.git 03James Almer 07master:08810a889517: x86/flacdsp: remove unneeded ifdeffery [21:02] <cone-342> ffmpeg.git 03wm4 07master:bd7801040786: avformat/id3v1: strip trailing whitespace [00:00] --- Tue Jan 6 2015

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ffmpeg.log.20150105
by burek 06 Jan '15

06 Jan '15

[00:04] <zumba_ad_> hi folks, what's missing from my command, I couldn't hear the audio? ffmpeg -i output.mp4 -i song.aac -c:v copy -c:a copy output3.mp4 [00:04] <schiho> Is it possible to force constant delay when streaming? [00:05] <schiho> so the delay varys between 200ms and 300ms it is more important for me to have all the time 300ms rather then fluctuating delay [01:27] <chan1> chan1 [01:29] <chan1> sfan5: thanks. I tried "ffmpeg -i file1.MTS -c file1.mkv" but it doesn't work. [01:29] <chan1> ffmpeg -h tells me I have to give codec name after -c. [01:29] <chan1> How do you use -c option? (I want to bypass transcoding. want to just copy the video, audio, just different container file format) [01:32] <c_14> -c copy [01:33] <llogan> and add -map 0 if you want to include all streams [01:40] <chan1> c_14, llogan : Thanks a lot! I tried "ffmpeg -i 20121129133142.MTS -c copy 20121129133142.avi" and it worked! [02:56] <k_sze[work]> I have run into a tough problem. [02:56] <k_sze[work]> I'm trying to encode infrared video from Kinect for Windows v2. [02:56] <k_sze[work]> But the video is really huge as FFV1. [02:57] <k_sze[work]> (unsigned 16-bit grayscale) [02:57] <xreal> I want to compress a video using ffmpeg and play it by ffplay. Piping doesn't seem to work: "pipe:: Invalid data found when processing input" - what am I doing wrong? [02:57] <k_sze[work]> I'm trying to identify some other codec, even if that means lossy. [02:57] <k_sze[work]> but there seems to be none that supports 16-bit grayscale. [02:59] <k_sze[work]> (I know libopenjpeg does 16-bit grayscale and it can be lossy, but it seems to be bugged in the recent revisions of ffmpeg) [03:06] <zacarias> Hi. I want to burn subtitles onto a mpeg file, which has a birate of around 2200 kbps. But if I just use the default options (ffmpeg -i video.avi -vf subtitles=subtitle.srt out.avi) I get a low-quality video with a bitrate of around 900 kbps. How should I do this? [03:08] <c_14> Increase the bitrate? [03:13] <zacarias> c_14: Is the question for me? [03:13] <c_14> Less of a question, more of a suggestion. [03:14] <c_14> And yes, for you. [03:16] <zacarias> c_14: That's it. But I tried to add this (-b:v 2200k -bufsize 2200k ) just after the input filename and it gives me an error. The command was: ffmpeg -i video.mp4 -b:v 2200k -bufsize 2200k -vf subtitles=subtitle.srt out.avi) [03:17] <c_14> You don't need bufsize [03:18] <zacarias> c_14: ok... I'll try without it. I don't know if the automatic reduction of bitrate has anything to do with the fact that the original mp4 has a h264 encoding, and ffmpeg automatically encodes it with xvid [03:20] <c_14> You'll probably want to force h264 [03:20] <zacarias> c_14: Oh! Is it possible? [03:21] <c_14> If your version of ffmpeg has libx264 support, ye. [03:21] <c_14> Just add -c:v libx264 [03:22] <c_14> In which case you might also want to look into -crf. Something around 18 is usually considered visually lossless. [03:23] <zacarias> c_14: ok, I'll try it if the option with just increasing the bitrate doesn't fit my needs. Just another questions: -c:v libx264 or -crf should be put where (sorry for my ignorance)? [03:23] <c_14> somewhere in front of the output file [03:23] <c_14> but it has to be after the input filename[s] [03:26] <zacarias> c_14: thanks. So should I make something like: ffmpeg -i video.mp4 -c:v libx264 -crf 18 -vf subtitles=subtitle.srt out.avi? [03:27] <c_14> ye [03:28] <zacarias> c_14: Thanks for your precious help! [03:33] <Demon_Fox> What would I use to slice up a video file into smaller ones losslessly given I can get the exact times for everything? [03:33] <c_14> A lossless codec? [03:33] <Demon_Fox> I mean [03:33] <Demon_Fox> -vcodec copy [03:33] <Demon_Fox> (Not sure if that incurs loss though.) [03:34] <Demon_Fox> Well, it is a huge mpeg2 file. [03:34] <c_14> It shouldn't, but you can only cut on iframes [03:34] <Demon_Fox> key frames? [03:34] <c_14> ye [03:35] <Demon_Fox> Not exactly sure how the dvd people made it [03:36] <Demon_Fox> It would matter much c_14 [03:36] <Demon_Fox> It ends in black before starting the next one [03:36] <Demon_Fox> So it probably hits a keyframe [03:37] <Demon_Fox> c_14, How would I do it though? [03:37] <c_14> ffmpeg -ss [from] -t [duration] -i file [options] outfile [03:37] <Demon_Fox> Thanks c_14 [03:38] <Demon_Fox> c_14, By chance do you know if ffmpeg will copy multiple audio channels? [03:38] <c_14> add -map 0:a [03:38] <c_14> It doesn't by default. [03:39] <Demon_Fox> Will it also copy the subtitles over? [03:39] Action: Demon_Fox should probably check the format on those. [03:39] <c_14> It will copy 1 track by default. add -map 0:s and it'll copy them all [03:40] <c_14> You'll want '-c copy' [03:40] <Demon_Fox> Thanks c_14 [03:55] <k_sze[work]> Hmm, so jpegls supports gray16le [03:56] <k_sze[work]> Does ffmpeg's jpegls support near-lossless mode? [04:58] <codehero> today i learned that i shouldn't use -vcodec copy -acodec copy when trimming videos... [06:23] <ParkerR> Well my ffpmeg experience may get better after all [06:26] <tdr> codehero yeah, there's a differnet way to do the same thing that is more correct [06:55] <rcombs> codehero: you can do that for audio if you throw in `-copypriorss:a 0` [09:06] <Akagi201> Could someone help me to provide a ffmpeg static build script or options? [09:07] <Akagi201> I found http://ffmpeg.gusari.org/static/ and http://johnvansickle.com/ffmpeg/ but I don't find a working script [09:26] <Akagi201> relaxed: Could you provide a static build script? [09:26] <Akagi201> Or give me some ideas to do it. [10:35] <bahathir> Hi... I have problem playing youtube URL links, since several git versions. [10:35] <bahathir> It segfaulted. [10:36] <bahathir> ffplay https://www.youtube.com/watch?v=0j5ttFMqbFM [10:36] <bahathir> ffplay version N-53543-g32c836c Copyright (c) 2003-2014 the FFmpeg developers [10:36] <bahathir> built on Dec 23 2014 15:52:25 with gcc 4.9.2 (GCC) [10:36] <bahathir> configuration: --prefix=/usr --shlibdir=/usr/lib64 --libdir=/usr/lib64 --enable-shared --disable-static --enable-gpl --enable-pthreads --enable-nonfree --enable-postproc --enable-x11grab --enable-libmp3lame --enable-libdc1394 --enable-libbluray --enable-libquvi --enable-vdpau [10:36] <bahathir> libavutil 54. 15.100 / 54. 15.100 [10:36] <bahathir> libavcodec 56. 16.100 / 56. 16.100 [10:36] <bahathir> libavformat 56. 16.101 / 56. 16.101 [10:37] <bahathir> libavdevice 56. 3.100 / 56. 3.100 [10:37] <bahathir> libavfilter 5. 4.100 / 5. 4.100 [10:37] <bahathir> libswscale 3. 1.101 / 3. 1.101 [10:37] <bahathir> libswresample 1. 1.100 / 1. 1.100 [10:37] <bahathir> libpostproc 53. 3.100 / 53. 3.100 [10:37] <bahathir> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [10:37] <bahathir> /0 [10:37] <bahathir> 0B f=0/0 [10:37] <JuanDosSantos> Hi, is it possible to send output stream to a http socket? So that I can access the stream trought a http proxy like nginx? [10:37] <bahathir> q= 0B f=0/0 [10:37] <bahathir> 0KB sq= 0B f=0/0 [10:37] <bahathir> vq= 0KB sq= 0B f=0/0 [10:37] <bahathir> 0KB vq= 0KB sq= 0B f=0/0 [10:37] <bahathir> aq= 0KB vq= 0KB sq= 0B f=0/0 [10:38] <bahathir> Actually, previously, ffplay able to to playback the URL without any problem/ [10:40] <bahathir> Ok, thank you. I try to send the bug the ffmpleg mailing list later. :) [10:41] <bahathir> exit [10:43] <JuanDosSantos> If i try to create a http output I get this error: [tcp @ 0xa63600] Connection to tcp://127.0.0.1:8099 failed: Connection refused \n http://127.0.0.1:8099/test.ts: Input/output error [10:44] <JuanDosSantos> ffmpeg -i http://10.8.0.10:8000/stream.ts -c copy http://127.0.0.1:8099/test.ts [10:45] <Mavrik> JuanDosSantos, ffmpeg isn't going to create an HTTP server for you [10:45] <Mavrik> JuanDosSantos, ffserver does that [10:46] <Mavrik> what you're trying to do is send data to a server that listens over HTTP protocol for incoming video [10:47] <JuanDosSantos> ye but ffserver is no longer maintained.. [10:48] <Mavrik> if that's a concern, you'll have to get a streaming server then [10:48] <Mavrik> there's several out there, from commercial to opensource [10:48] <JuanDosSantos> Do you know a opensource project for that? [10:49] <Mavrik> Depends on your requirements. [10:49] <JuanDosSantos> need only a http listener, to send stream data to it.. [10:49] <Mavrik> But I didn't find anyting opensource that would work significantly better than ffserver, we used Wowza for pro streaming (which is probably too expensive for you) [10:49] <Mavrik> JuanDosSantos, that means nothing [10:49] <Mavrik> JuanDosSantos, the important part is: which protocol, which clients, who wants to watch and in what way [10:50] <Mavrik> also which video format [10:50] <JuanDosSantos> Mavrik: I need only mpegts... [10:55] <JuanDosSantos> Mavrik: Ok I will try again with ffserver... [12:09] <kaotiko> hi [13:35] <chan1> hi, all, how do we know the video and audio type of an avi file?(without converting it) [13:42] <Fjorgynn> chan1: what? [13:42] <Fjorgynn> ffmpeg -i file.avi [13:59] <Fjorgynn> lifeofguenter: :D [14:10] <Fjorgynn> it's more activity in a vacum than here [14:12] <d_ferdi> hello, i have a problem with ffmpeg and mp4: first run: ffmpeg insert many dupe frames and increase the playlenght by 15%, second run with other parameters video is ok for virtualdub, but other progs (vlc,zoomplayer) display same playlength as first time with video/audio async and 15% black videotime at the end [14:27] <d_ferdi> http://pastebin.com/pfxaeM5h [14:28] <d_ferdi> there is a difference of 20 minutes between input and output but both have a fps of 25 [14:46] <d_ferdi> how can audio be 20 minutes longer as video ? [14:47] <d_ferdi> http://pastebin.com/mHAbt1u3 [14:53] <d_ferdi> why says ffmpeg, that a 03:23:09.65 mp2-audio file is 03:53:53.28 long, if i demultiplay and save the audio ? [15:26] <d_ferdi> i convert the mp2-audio-channel with Duration 03:23:09.65 (ffprobe) to acc with time Duration: 03:40:08.73 (ffprobe), mux them with ffmpeg to video with Duration 03:23:09.65 (ffprobe) and video/audio are sync. this is impossible! [16:30] <eyal-ff> Hey, I have a question about monitoring stream. Does anyone knows if it's possible to monitor a live stream (that I publish with ffmpeg) and get information about current framerate, bitrate and more? [18:00] <Teilio> hi im trying to find a way of handling latm audio without reencoding it. is there a way to alter the bitstream so that it can be changed to adts/raw aac? [18:00] <Teilio> videoredo seems to be able to do this thanks to ffmpeg but i can't work out how to do this in ffmpeg itself [18:11] <xreal> I've got a ts stream, which is PAL already. Does ffmpeg do fast copy or a complete reencoding when using -target pal-dvd ? [18:13] <kepstin-laptop> by default it'll do a reencode; the main use of the 'pal-dvd' target is that it sets up the codec parameters correctly for dvd. [18:15] <xreal> kepstin-laptop: So I shoud do it manually and use -f vob ? [18:16] <xreal> it's a ts-stream from DVB-S btw [18:16] <kepstin-laptop> if your video and audio streams are both already in codecs supported on DVD, and are within the allowed bitrate for DVD, then that should work. [18:17] <xreal> kepstin-laptop: since it's DVB-S it should work, shouldn't it? [18:20] <xreal> Overall bit rate: 4 339 Kbps - should be okay? :) [18:20] <xreal> mmh, there's also -f dvd [18:21] <xreal> they're equal [18:21] <kepstin-laptop> i dunno, you'd have to check the codecs used. [18:22] <xreal> kepstin-laptop: http://todayiwantedtoprogram.tumblr.com/post/15142587796/what-does-ffmpegs-… [18:22] <xreal> Thanks for help :D [18:23] <xreal> Funny: Stream #0:2 -> #0:1 (mpeg2video -> mpeg2video) [18:23] <xreal> ffmpeg should make this "copy" automatically :( [18:24] <kepstin-laptop> assuming that the video in your stream is already mpeg2 and the audio is one of pcm, mp2, or ac3, you're probably ok with just a remux to vob. [18:25] <kepstin-laptop> copy is a bit of a special case, a lot of the ffmpeg functionality is disabled when you use it. so it doesn't normally make sense to use automatically [18:26] <xreal> okay, let me play around, thanks again [19:40] <bttf> question ... is it possible to capture stills from a video file using ffmpeg, and if so where is a good place to start to read more about it? [19:42] <bttf> nvm .. i found it in http://ffmpeg.org/ffmpeg.html#Video-Options [19:42] <bttf> sry, http://ffmpeg.org/ffmpeg.html#Video-and-Audio-file-format-conversion [20:48] <ChocolateArmpits> Are there any known compatibility problems with proxy profile prores? Premiere crashed when I tried to import some of the converted files. The "standard" profile is handled well. [21:53] <squeegily_> I have some MKV files with H.264 video and styled subs. Is there a way to convert them to MP4 (or some other PS3-compatible format), keeping the subtitles, but *without re-encoding the video*? I'm suspecting something along the lines of converting the subtitles to some sort of overlay..? Google didn't turn up anything useful find anything [21:54] <llogan> softsub support in MP4 isn't great. i doubt PS3 can play mov_text subs. [21:54] <squeegily_> llogan: Thanks. :( [21:54] <llogan> and i'm not sure what other formats the PS3 supports. [21:54] <c_14> http://manuals.playstation.net/document/en/ps3/current/video/filetypes.html [21:55] <squeegily_> llogan: I've seen it choke on some AVIs so I have pretty little faith in it [21:55] <c_14> Basically, nothing interesting. [21:55] <JEEB> the best bet for such devices is to re-encode with mpv which easily lets you keep the styled subtitles [21:55] <JEEB> unlike ffmpeg, with which you have to extract fonts and the subtitles first [21:55] <c_14> you don't have to [21:56] <c_14> the subtiles filter does all that [21:56] <llogan> ffmpeg -i input.mkv -vf subtitles=input.mkv [21:56] <llogan> i think... [21:56] <JEEB> so it registers fonts and gets the subtitles these days? [21:56] <c_14> yep [21:56] <c_14> Works for me anyways [21:56] <squeegily_> JEEB: yeah, right now if I want to watch anime on the PS3 I have to wait like an hour for the video to re-encode and burn in the subtitles [21:56] <JEEB> c_14, the first part is usually quite important :P [21:57] <JEEB> if ffmpeg can do it then it's sure great of course [21:57] <JEEB> the last I checked something like mpv was the quicker and simpler choice, though [21:57] <squeegily_> ffmpeg does a really good job - they're rendered the way they're supposed to be [21:57] <kepstin-laptop> squeegily_: yeah, no other option :/ even the crunchyroll ps3 app needs a special encode with the hardsubs burned in... [21:57] <JEEB> why does everyone actually ignore the one point I'd like the answer for :P [21:58] <JEEB> does it actually register the fonts included in the matroska file [21:58] <squeegily_> ouch. Oh well I guess that's what I get for going with a proprietary OS.. :/ [21:58] <squeegily_> JEEB: yes, it works [21:58] <JEEB> if it uses libass then it's pretty fucking simple to actually render the subs [21:58] <JEEB> ok, thank you [21:58] <JEEB> for whatever reason I didn't get any comments on the fonts side of things [21:58] <squeegily_> I'll test it out with the SAO II subs as the ultimate benchmark [21:58] <squeegily_> They've got so many particle effects it hangs any media player I try [21:58] <JEEB> anyways, then it's quite much more usable than it was [21:58] <JEEB> uhh [21:59] <squeegily_> Did it not used to be good? [21:59] <JEEB> it was not exactly simple to use :P [21:59] <JEEB> you had to register the fonts, then extract the subs [21:59] <llogan> what's "Memory Stick Video Format" as mentioned in c_14's PS3 link [21:59] <JEEB> then you could use the ass filter to use libass to render it [21:59] <squeegily_> I didn't realize it used to be inconvenient [21:59] <JEEB> so it could do the job, but it required all kinds of things the players doing the same thing didn't :P [22:01] <llogan> squeegily_: i guess you'll just have to use a faster encoding preset [00:00] --- Tue Jan 6 2015

1 0

ffmpeg-devel.log.20150104
by burek 05 Jan '15

05 Jan '15

[00:10] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:5c95de150f3b: avcodec/libxvid: check for av_malloc*() failures [00:27] <cone-644> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:907101eb8395: ffserver: reflow compute_status() [00:27] <cone-644> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:9e55130a9d84: ffserver: reindent compute_status() [00:27] <cone-644> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:196bc03a869a: ffserver: drop redundant else clause [00:27] <cone-644> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:ac1940b2ba7d: ffserver: reflow rtsp_cmd_setup() [00:27] <cone-644> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:997508847215: ffserver: reindent rtsp_cmd_setup() [01:31] <jamrial> the git repo seems to be down [02:03] <michaelni> jamrial, yes, dunno what the issue is, use the github mirror [08:32] <sth0R> is this a 0day vuln in ffmpeg? [08:32] <sth0R> http://www.openwall.com/lists/oss-security/2015/01/04/10 [08:32] <sth0R> plz check it asap.... [09:52] <aetasx> in the least, the source code he provides doesn't match [09:53] <wat2015> Does anyone have a copy of the dolby TrueHd/Atmos technical spec? [09:56] <aetasx> sth0R: that release is against 2.1.6 and we're up to 2.5.2 now as an fyi [09:58] <sth0R> aetasx, good. So it won't be an issue;-) [09:59] <aetasx> not that I saw, it looks like that whole section of code has been rewritten from what it was and the block hes referencing doesn't exist as it was anymore [10:56] <nevcairiel> there was also a fix commited to the 2.1 release branch 5 days ago for that [10:57] <nevcairiel> and 2.1.7 released [13:08] <cousin_luigi> bbl [19:23] <drwx> git down? [19:23] <drwx> err wrong channel, sorry :) [19:26] <wm4> correct channel [19:26] <wm4> the main git repo is still down [19:52] <kierank> what's the ffmpeg trac ftp password? [19:56] <kierank> compn ? [19:56] <kierank> michaelni: ? [20:26] <Daemon404> kierank, isnt upload anonymous [20:26] <kierank> yes but i want to read [20:26] <Daemon404> o [20:30] <kierank> these mp3 detections are pissing me off [20:33] <wm4> lol [20:34] <kierank> also mpv shouldn't be making ts files without a pmt [20:34] <wm4> it doesn't [20:34] <wm4> this is just straight from dvb [20:34] <kierank> "Patching mpv (which I use for recording) I did a recording of the same program, but only selecting the "0x68" and "0x69" streams. " [20:35] <wm4> yes, but that's just the byte stream straight from the kernel [20:36] <kierank> then the guy has a fucked stream [20:36] <kierank> but i can't tell until he uploads the actual thing [20:59] <kierank> wm4: ^ [20:59] <kierank> he blames mpv [21:01] <JEEB> lolwut [21:01] <JEEB> mpv has no such things afaik [21:01] <JEEB> it just uses lavf & lavc [21:03] <wm4> kierank: that's fine; it's old mplayer dvb code [21:03] <wm4> and it was written against mplayer's .ts demuxer (which is now gone from mpv) [21:03] <wm4> though I don't know how the fix is going to work... inject stuff into the byte stream? [21:04] <kierank> when you filter a PID, make sure it's PMT is included [21:04] <kierank> its* [21:04] <JEEB> also where the fuck do people come up with all these files without PMTs [21:04] <JEEB> >:| [21:04] <kierank> JEEB: +1 [21:06] <JEEB> I know of this one south american government sponsored STB, which lets you grab recordings, and there you have the PMT for whatever reason in a separate index file [21:06] <JEEB> oh, and then there's some shitty korean one [21:07] <JEEB> but those definitely can't be all of them [21:07] <JEEB> :V [21:18] <compn> kierank : yes ? [21:18] <kierank> 6:52 PM <"kierank> what's the ffmpeg trac ftp password? [21:21] <compn> anyone else want incoming access while i'm here ? [21:21] <compn> Daemon404 or wm4 ? [21:22] <wm4> sure [21:23] <compn> [vulnerability sample] [21:23] <compn> The sample is a porn video. [21:23] <compn> woo [21:24] <Daemon404> nah. [21:24] <compn> incoming is useful when telling your users to upload samples [21:24] <compn> so you dont have to fiddle with stupid file hosts [21:24] <compn> :) [21:24] <compn> it has http upload and ftp uploading capabilities [21:27] <Daemon404> well most of my samples are kinda already uploaded [21:27] <Daemon404> to us [21:28] <compn> no worries [21:29] <compn> libav probably has its own incoming as well [21:29] <Daemon404> hows that related? [21:29] <Daemon404> my samples are from work [21:30] <compn> well i am trying to remember which projects you work on , and libav popped into my head. mostly i was thinking about libav samples just because i like sample files [21:30] <compn> and maybe i want to browse their incoming [21:30] <compn> just me talking to myself really [21:30] <Daemon404> o ok [21:56] <aetasx> cause no one else is willing to talk to you? [22:34] <ubitux> wtf @ #4223 [22:38] Action: llogan pokes git.videolan.org [22:38] <llogan> ubitux: we also got a spanish language report today [22:38] <llogan> i don't know spanish [22:39] <pross> hi. is git.videolan.org down? [22:39] <ubitux> http://www.downforeveryoneorjustme.com/git.videolan.org [22:39] <llogan> since i a United Statesian i only know English and expect everyone else to know my language [22:39] <pross> thx [22:43] <iive> ubitux: is that one of these sites that count the number of visitors, to figure out if a site is down/inaccessible ? [22:44] <ubitux> i think it just make a request and see how it goes [22:45] <aetasx> you'd think they'd take this line out of the site source: // Merged from /Users/armiol/development/projects/bitbucket-estimate-counter/src/js/chrome/includes/content_kango.part.js [23:18] <compn> aetasx : probably its there so we can update that from git if need be [23:18] <compn> like if theres security violation [23:19] <jamrial> ubitux: that's mean :p [23:20] <ubitux> i know :( [23:22] <llogan> at least he wasn't using a ffmpeg binary from 2004 like in #4220 [23:26] <jamrial> anyway, he's running the command he wrote in bold on lubuntu 14.04 and getting tons of "ALSA buffer xrun" errors. The output file ends up containing either noise or garbled audio (recorded from a microphone) [23:28] <jamrial> ffmpeg 1.2.6 from the PPA linked from ffmpeg.org/download [23:29] <jamrial> he says it didn't happen on lubuntu 12.10. no idea which ffmpeg version, if any [00:00] --- Mon Jan 5 2015

1 0

ffmpeg.log.20150104
by burek 05 Jan '15

05 Jan '15

[01:42] <schiho> Guys, i was using windows 7 + blackmagic WDM Capture + FFMPEG and it was oke. For speed gain, i#ve installed Ubuntu and there is no support for ffmpeg + blackmagic :((( [01:42] <c_14> I thought Ubuntu was black magic. [01:44] <c_14> Did you try installing their funky "Desktop Video" thingymabob? [01:45] <schiho> yes i did it, then i got /dev/blackmagic/dv0 [01:46] <schiho> however ffmpeg is not recognizing under video4linux2 [01:48] <schiho> I've installed Desktopvideo and MediaExpress, in the mediaexpress i get a signal, but don't know how to get it with fmpeg [01:50] <troy_s1> schiho - See PM [01:56] <awilliam> any ffprobe experts around? I want to test whether a video is mpeg2 vs h264 and test the height. I've come up with "ffprobe -select_streams v:0 -show_entries stream=index,codec_name,height -hide_banner <file>", but it always wants to show the info twice, inside a [PROGRAM] and out [01:57] <awilliam> for example http://fpaste.org/165351/42033303/ [01:59] <awilliam> hmm, on an encoded file w/o a program it only shows it once [02:01] <awilliam> I can s/stream/program_stream/ on the show_entries to just get the stream within the program [03:06] <troy_s> awilliam: No idea, but I wonder if it reports once for the codec context and once for the stream [03:09] <awilliam> seems like I need to figure out if I have a program, then use program_stream, otherwise use stream [03:09] <awilliam> or I suppose I can use the flat print format and grep -v program [03:10] <troy_s> In the api [03:11] <troy_s> You almost always have to rely on the stream data [09:05] <chan1> Hi, I tried to convert MTS file (h264,ac3) to mkv file Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) [09:05] <chan1> Stream #0:1 -> #0:1 (ac3 (native) -> ac3 (native)) [09:06] <chan1> Sorry, I tried to convert an MTS file (h264,ac3) to mkv file. [09:06] <chan1> The command was ffmpeg -i file1.MTS -acodec ac3 file1.mkv [09:07] <chan1> But it takes some time and the log shows [09:07] <chan1> Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) [09:07] <chan1> Stream #0:1 -> #0:1 (ac3 (native) -> ac3 (native)) [09:08] <chan1> If I can use the same h264 I thouhgt maybe it will take very little time. [09:08] <chan1> Is there a way to do it? How can I specify h264 (native) codec for output? [10:04] <edulix> hi peope [10:05] <edulix> I'm having trouble recording audio when trying to use ffmpeg to do it. by the way, audacity works fine to do that. what options should I use? I'm using opensuse 13.2 [10:06] <edulix> when I try to use -i pulse, ffmpeg says cannot open audio device pulse (No such file or directory) [10:32] <sfan5> chan1: use -c copy to tell ffmpeg to copy the streams from the input file [12:58] <belgaat> hi, I have this 1080i tv cap. And its 50fields/s So I apply yadif filter -vf "yadif=0:-1:0". But I don't see the frames go to 25fps. [12:58] <belgaat> Is that normal? [13:41] <DelphiWorld> yo [13:42] <DelphiWorld> if someone know a good IPTV middleware let me know [14:17] <belgaat> This cropdetect doesn't work anymore since recently. I outputted this into a bash variable to input it directly in a -vf "crop". But computer says no. http://www.pastie.org/pastes [14:19] <belgaat> http://pastie.org/9812271 [18:18] <mmance> How do you troubleshoot video parameters incompatible with DNxHD, I am going from glc-play. [18:20] <mmance> http://pastebin.com/VAmc7y2F [18:20] <mmance> I have tried different bitrates [19:34] <q0_0p> how do i convert an mkv file to ISO Media, MP4 v2 [19:34] <q0_0p> when i convert i'm getting ISO Media, MP4 Base Media v1 [IS0 14496-12:2003] [19:52] <Fjorgynn> What is iso media? [19:53] <Fjorgynn> q0_0p: I'd wote for Handbrake [19:53] <Fjorgynn> or FFMPEG [19:53] <q0_0p> i guess the reason why I ask MP4 base media V1 audio will not load in chromecast [19:54] <q0_0p> but V2 ISO Media, MP4 V2 works [19:54] <Fjorgynn> http://handbrake.fr/ [19:55] <q0_0p> very cool a UI [19:55] <Fjorgynn> :) [19:55] <Fjorgynn> And mp4 is a container what I know. [19:57] <Fjorgynn> aha I was in the ffmpeg channel [19:57] <Fjorgynn> ;) [19:58] <Fjorgynn> but isn't it like ffmpeg -i file.mkv -c copy file.mp4 or something? [19:59] <Fjorgynn> q0_0p: [20:00] <Fjorgynn> but then you will lost all subtitles. And I don't know if it will downgrade the quality [20:01] <pzich> copy won't downgrade quality, but your new container has to support it [20:01] <Fjorgynn> pzich: okay [20:10] <Fjorgynn> I think someone is using debian [20:31] <sciho> Hi guys, pipes are really cool, but can i pipe streams to ffserver as well? [20:31] <c_14> afaik ffserver only accepts input via http [21:20] <Milos_SD> Hello to everyone [21:21] <Milos_SD> Does someone else have problems with cloning ffmpeg from git ( from git://source.ffmpeg.org/ffmpeg.git )? [21:22] <Milos_SD> it forwardes to git.videolan.org, and I am getting "connection reset by peer" [21:22] <JEEB> yes, use the github mirror for now [21:22] <Milos_SD> I can't even browse git sources from ffmpeg web site [22:21] <d_ferdi> hello, i have a big problem with ffmpeg: the source video lenght is 03:23:09.62 and the ffmpeg-output video is 03:53:53.31 with 46091 dupe frames inserted by ffmpeg [22:22] <c_14> Is the input and output framerate identical? [22:23] <d_ferdi> yes 25 fps [22:23] <d_ferdi> source Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [22:24] <d_ferdi> output Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 12800 tbn, 25 tbc [22:24] <d_ferdi> ffmpeg info output mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 2116 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc [22:28] <d_ferdi> i think problem have somthing todo with the audio because audio of the output is ok, only video is bad (dupe frames) [22:29] <c_14> Is the audio the same length in both? [22:30] <d_ferdi> yes audio have the same length as the video in both files, source and output [22:31] <d_ferdi> but the video is jerky and the audio seems ok [22:31] <d_ferdi> of the output [22:31] <q0_0p> Fjorgynn, solution i did was use acodec using libfaac [22:32] <q0_0p> ffmpeg -i video.mkv -vcodec copy -acodec libvaac -c:s mov_text output.mp4 [22:32] <d_ferdi> Stream #0:0 -> #0:0 (mpeg2video -> mpeg4) [22:32] <d_ferdi> Stream #0:1 -> #0:1 (mp2 -> libvo_aacenc) [22:33] <d_ferdi> i try libfaac, but this is not installed for ffmpeg [22:34] <c_14> try reencoding with -vf aresample=async=1000 [22:34] <d_ferdi> Unknown encoder 'libfaac' [22:35] <c_14> just use libvo_aacenc, or aac [22:37] <d_ferdi> don't work: Value inf for parameter 'time_base' out of range [0 - 2.14748e+09] [22:38] <Fjorgynn> aha [22:39] <q0_0p> d_ferdi, i had to compile ffmpeg with the faac support [22:39] <q0_0p> i forget what dependencies it brought in [22:39] <q0_0p> lame was one of them probably because i added mp3 support [22:40] <d_ferdi> i test codec aac, but it seems same problem [22:40] <q0_0p> for me it brought in libmp4v2 lame faac [22:41] <q0_0p> when building it with faac and mp3 [22:41] <llogan> q0_0p: if you compiled ffmpeg, you should have used libfdk_aac instead of faac [22:41] <d_ferdi> i use gentoo and have compiled in support for mp4 today [22:41] <llogan> better quality per bitrate [22:42] <c_14> d_ferdi: what ffmpeg version? [22:42] <q0_0p> ah the fdk flag [22:42] <q0_0p> fdk : Use external fdk-aac library for AAC encoding [22:42] <q0_0p> faac : Use external faac library for AAC encoding [22:43] <q0_0p> i was able to use libvaac compiling ffmpeg with faac [22:44] <q0_0p> d_ferdi, looks like a lot of people use gentoo these days [22:44] <q0_0p> ah fdk is better than faac i see [22:45] <d_ferdi> i check the use-flags for emerge at the moment [22:47] <d_ferdi> what settings are needed, to compile ffmpeg with fdk ? [22:47] <llogan> --enable-libfdk-aac and also --enable-nonfree if --enable-gpl is included [22:48] <d_ferdi> yes, but i must tell this emerge or must compile ffmpeg without emerge [22:48] <llogan> i know nothing of emerge [22:48] <llogan> refer to #gentoo? [22:49] <q0_0p> d_ferdi, echo "media-video/ffmpeg faac mp3" >> /etc/portage/package.use [22:49] <d_ferdi> is gentoo specific [22:49] <q0_0p> d_ferdi, you may need to go to #gentoo for help [22:49] <d_ferdi> i will try somthing, but i can test it tomorrow [22:49] <q0_0p> d_ferdi, try the command above will build with faac and mp3 [22:53] <d_ferdi> i find the flags on http://gentoobrowse.randomdan.homeip.net/package/media-video/ffmpeg [22:57] <q0_0p> d_ferdi, you can find the flags using "equery u ffmpeg" [22:59] <d_ferdi> thank you, this will help me in the future to find the flags without google [23:00] <q0_0p> d_ferdi, if you dont have it you might want to emerge gentoolkit [23:03] <d_ferdi> i have it, but i don't find the time, to check all commands and parameters for the portage tools. [23:03] <d_ferdi> most i use eix to find some packages because it is very fast [23:06] <d_ferdi> but i think, the problem is not the codec. this problem must have an other reason. i try to copy source audio to mp2-file and ffmpeg insert dupe frames [23:08] <d_ferdi> a short test gives ca. 15% more frames in output as in input [23:10] <d_ferdi> with -r 25 no dupe frames are inserted [23:12] <llogan> !pd d_ferdi [23:12] <llogan> i guess the bot is not here [23:12] <llogan> show your command and complete output. use pastebin [23:13] <llogan> oh, typo. duh. [23:14] <d_ferdi> but i must insert this parameter as first parameter [23:18] <llogan> no command, no console ouptut = no help [23:23] <d_ferdi> it seems, the problem is the frame-rate, but why use ffmpeg an other framerate for the input as given in the detected framrate by ffmpeg ? [23:23] <d_ferdi> i install pastebin and send you the log-file [23:23] <llogan> install? just use a pastebin site [23:24] <d_ferdi> moment please [23:25] <d_ferdi> http://pastebin.com/JgKbPQVm [23:26] <llogan> why use ffmpeg version 1.2.6 instead of current git master? [23:27] <llogan> also your command is missing [23:27] <d_ferdi> command line was: ffmpeg -i Ben_Hur-2014-12-23-20-00-Spielfilm_USA_1959_(Ben-Hur).vdr.mpg -c:a libvo_aacenc -b:a 128k -c:v mpeg4 -q:v 3 Ben_Hur-2014-12-23-20-00-Spielfilm_USA_1959_(Ben-Hur).vdr.mp4 [23:27] <llogan> instructions are hard. let's go shopping. [23:31] <d_ferdi> new encoding run's tonight and i will see tomorro if output is ok [23:31] <llogan> just encode a segment instead of the whole thing [23:32] <d_ferdi> i have make this for test and it was ok [23:33] <llogan> can you provide a short sample of the input? ffmpeg -ss 00:10:00 -i input -t 60 -c copy output.mpg [23:33] <llogan> can you test a recent ffmpeg? [23:33] <d_ferdi> i will yet shutdown my workstation. i have much to do tomorrow [23:33] <d_ferdi> i can test many things on my test-installation [23:34] <d_ferdi> or i can send you an input file, which make problems without the parameter -r for the input-frame rate [23:37] <d_ferdi> are you interested in a sample mpeg-file ? [23:39] <d_ferdi> you can send me a mail at dieter.ferdinand(a)gmx.de goodnight [00:00] --- Mon Jan 5 2015

1 0

ffmpeg-devel.log.20150103
by burek 04 Jan '15

04 Jan '15

[01:17] <llogan> non-CamelCase, long wiki URLs are ugly. [01:19] Action: llogan would have used wiki/Waveform or similar [01:25] <cone-375> ffmpeg.git 03Martin Storsjö 07master:355d01a1bf55: movenc: Factorize writing ftyp and other identification tags to a separate function [01:25] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:7ca10d10aae3: Merge commit '355d01a1bf55297b1d1f04e4bfbf0ddc93b6247e' [01:33] <cone-375> ffmpeg.git 03Martin Storsjö 07master:724cbea71939: movenc: Remove an unnecessary condition when flushing fragments [01:33] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:68a06a244c06: Merge commit '724cbea7193945fe5a5b4dea8ede344803572844' [03:07] <cone-375> ffmpeg.git 03Martin Storsjö 07master:c725faebda9a: movenc: Use start_dts/cts instead of cluster[0] for writing edit lists [03:07] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:0506f687e877: Merge commit 'c725faebda9a516766d94c33b07972ab0f70cf93' [03:07] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:504267fb56ba: avformat/movenc: Use cluster timestamps when available in edts [04:01] <cone-375> ffmpeg.git 03Martin Storsjö 07master:847bf5988fec: movenc: Add an option for delaying writing the moov with empty_moov [04:01] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:f38e2bcb72ee: Merge commit '847bf5988fec1d3e65c1d8cf0cdb8caf0cfd0c1b' [04:08] <cone-375> ffmpeg.git 03Martin Storsjö 07master:b3b0b35db2f3: movenc: Get rid of a hack for updating the dvc1 atom [04:08] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:b0ba8a23b615: Merge commit 'b3b0b35db2f3b61bf2f0f4fa85f5b6267d83c8fe' [04:26] <cone-375> ffmpeg.git 03Martin Storsjö 07master:c5e7ea13d2d4: dashenc: Use delay_moov [04:26] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:983778f1f68a: Merge commit 'c5e7ea13d2d4da0c5da91973a547afff6fe9e011' [04:51] <cone-375> ffmpeg.git 03Martin Storsjö 07master:7a1a63e34fa4: dashenc: Use pts for MPD timeline timestamps [04:51] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:3be67f33a5f2: Merge commit '7a1a63e34fa46af18311c2493fdaec9a93bdb750' [05:16] <cone-375> ffmpeg.git 03Martin Storsjö 07master:5cf6bda6e2ea: sidxindex: Don't adjust the Period start time depending on the track start time [05:16] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:e566667d8fd0: Merge commit '5cf6bda6e2eae496e8eb2bb06c96852d59a58b8a' [05:44] <cone-375> ffmpeg.git 03James Almer 07master:b8db25a3338b: configure: bump year [05:55] <cone-375> ffmpeg.git 03Martin Storsjö 07master:470c9db11ff2: sidxindex: Remove parsing that isn't necessary any longer [05:55] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:38c1bffd80c1: Merge commit '470c9db11ff2c3249e995e7ba68e87bb81bf778c' [09:57] <cone-644> ffmpeg.git 03Clément BSsch 07master:cc5f5e140cdc: build: require libxcb >= 1.4 [14:03] <cone-644> ffmpeg.git 03Martin Storsjö 07master:f2ad1495f233: avisynth: Use RTLD_LOCAL instead of RTLD_GLOBAL [14:03] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:4bcb58b3ad88: Merge commit 'f2ad1495f23376ce61542967f4fc14205f284d40' [15:10] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:00f9e51a2638: tests/regression-funcs: Remove hardcoded threads 1 [17:57] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:0daf60140bd0: avformat/movenc: fix DELAY_MOOV [21:35] <cousin_luigi> Greetings. [21:43] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:d56b0984f977: avformat/westwood_vqa: Remove unneeded cast [21:43] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:bd12aa2bc597: avcodec/ljpegenc: Check for av_malloc_array() failure [00:00] --- Sun Jan 4 2015

1 0

ffmpeg.log.20150103
by burek 04 Jan '15

04 Jan '15

[02:39] <squeegily_> ffmpeg -i \[gg\]_Joshiraku_-_04v2_\[EA549C43\].mkv -c:a copy -c:s copy -tune fastdecode 4.mkv should be enough to play a 1280x720 video on an Asus Eee PC 1000, right? [02:48] <squeegily_> Wow the -tune fastdecode looks beautiful AND plays at regular speed! [03:06] <bpye> I'm capturing some raw bayer video via a pipe, is there any format in ffmpeg that I can encode bayer_bgrg8 in? [03:06] <bpye> grbg* [03:50] <squeegily_> bpye: Are you wondering which *container format* can handle it, or which video *coded* you should use? [03:50] <squeegily_> *codec [03:51] <bpye> It was the former however I discovered that nut appears to handle it well, I guess that was somewhat it's design goal, to handle different pixel formats and such [03:51] <squeegily_> bpye: When in doubt about a container, use MKV [03:52] <squeegily_> It supports every codec and any number of streams [03:52] <squeegily_> It's literally the perfect container (which is what it was designed to be) [03:52] <squeegily_> The only cases where MKVs aren't the best thing ever are cases involving compatibility (like you can't play them on a PS3) [03:53] <bpye> bayer appears unsupported by mkv, at least, ffplay can't play the output file [03:53] <squeegily_> bpye: Did you try mplayer? [03:53] <bpye> Uh no, I assumed it'd have the same/similar compatiblity though... [03:54] <squeegily_> Thats' really weird that a player would choke on a specific container-codec combination [03:54] <bpye> "[rawvideo @ 00000000044cc440] Invalid pixel format." [03:54] <squeegily_> Did you accidentally encode it to raw video? [03:55] <bpye> Intentional, this is raw capture from the CCD of a camera [03:55] <bpye> Looking at http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2014-February/154802.html I think I see why nut works, it appears to be the only container for which support was added [03:56] <squeegily_> Wow that's weird. MKVs are supposed to be forward-compatible [05:32] <justinX> when adding a subtitle to a mp4, are there any special text format the subtitle need to be for extra picky players? [05:33] <justinX> should it be CRLF or LF formatted text, or shouldn't i matter? [05:34] <justinX> my Vestel TV didn't like my mp4 with subtitles so it must be something it didn't like :-/ [05:46] <c_14> Didn't like as in didn't play entirely or didn't show the subtitles? [05:52] <justinX> c_14: it didn't show the subtitles [05:53] <c_14> You'll probably have to hardsub. [05:54] <justinX> the TV can show other filmes with subtitles though. both ones with text subtitles inside the file, and as seprate subtitle files [05:54] <pzich> take a look at how those are encoded [05:54] <c_14> Those are probably idx/dvbsub [05:55] <justinX> I guess I should investigate more tomorow. I try make some mkv instead and see if that works. [05:55] <justinX> there is not much info about what the TV is capable in the manual. it just says "subtitles: .sub and .srt" [05:57] <justinX> for .mp4 it says "MPEG4, Xvid 1.00, Xvid 1.01, Xvid 1.02, Xvid 1.03, Xvid 1.10-beta1/2 H.264 with sound PCM/MP3" [05:58] <c_14> If your TV supports matroska, use that with subrip subtitles [05:58] <c_14> Otherwise, eh avi *ehem* [05:59] <justinX> :-) [06:01] <justinX> and yeah it supports mkv with these the manual says "video: H.264, MPEG1,2,4 sound:EAC3 / AC3" [06:02] <c_14> So your TV doesn't support AAC? [06:02] <c_14> How old is it? [06:03] <justinX> it doesn't say anything about AAC in the manual at least. That doesn't have to mean anything I guess [06:03] <c_14> Good point, still [06:04] <c_14> I just love TVs and video/audio support. [06:04] <justinX> and the tv is a "Andersson LED4240FDC PVR" I don't know the age really... maybe 2010? [06:06] <justinX> where "Andersson" is one of the many names turkish Vestel tvs are sold if I understand right. One name per reseller or something :-D [06:08] <justinX> It could be interesting to have a large set of small test files that could be tried on the tv to see what it really supports or not. [06:12] <c_14> You could probably try some of the relevant fate samples. [06:18] <justinX> c_14: fate samples? uh.. where? [06:20] <justinX> if I'd could gain knowledge of what things it supported, I guess I could make a command that could verify that a file was "ok" or needed some conversion. [06:21] <c_14> The fate samples are available via rsync at rsync://fate-suite.ffmpeg.org/fate-suite/ [06:22] <c_14> They're designed to test that commits don't introduce regressions. [06:22] <c_14> You probably won't need/want all of them. [06:22] <c_14> Check the list for codecs you're interested in. [06:22] <justinX> aha ok [06:36] <sasha-> Yop anyone know how I can ignore errors like 'Error while decoding stream #1:0: Invalid data found when processing input' ? [06:37] <sasha-> I tried '-err_detect ignore_err' but that didn't help [14:23] <Marcin_PL> Hello. I don't get registering e-mail from forum at Gusari. [14:26] <c_14> >If you did not receive an e-mail, you may have provided an incorrect e-mail address or the e-mail may have been picked up by a spam filer. If you are sure the e-mail address you provided is correct, try contacting an administrator. [14:31] <Marcin_PL> It's a regular mailbox on gmail& huh [14:31] <c_14> Check the spam? [14:31] <Marcin_PL> Empty. [14:32] <Marcin_PL> I'll try other one or type that withuot a dot. [14:42] <Marcin_PL> Clever. I cannot contact admin without a valid account. Huh. :( [14:43] <Marcin_PL> Well, maybe here somebody can help me? [14:45] <Marcin_PL> I got some problem with fps& The point is I got MJPEG file (with PCM sound) at 30 fps (easy Canon PowerShot) and I cannot make it the same duration time in my rotating script. I used -r 30 and it worked somehow (default is 25), but not well (movie is going to be longer). I learned some time ago to work on ffmpeg separately with audio and video, so if it's not the reason pls don't irritate about separating sound off. Here's the [14:45] <Marcin_PL> script: http://pastebin.com/mDi9NjD6 [14:46] <c_14> add -framerate 30 before -i $1_f_%d.jpg [14:52] <Marcin_PL> Thanks, I'll try it out [14:53] <Marcin_PL> So -r 30 is not neccesary now? [14:54] <c_14> Might not be, not 100% sure. [14:54] <Marcin_PL> OK, I'll practice with and without [14:56] <c_14> ffmpeg _should_ realize that the input stream has 30fps and use that as the output fps, but I can't remember if it actually does [14:57] <Marcin_PL> I wonder if it's the only method to rotate image (without reencoding&) [15:55] <Marcin_PL> c_14: It didn't take effect, I'll try without -r 30 now [16:15] Action: Marcin_PL is really dumb today the script will work better after saving changes in editor. Huh. [16:16] <schiho> Hi guys, i am running a blackmagic decklink studio 2 card on linux, what is the input for that card, it's apparently not v4l2 [16:16] <schiho> v4l2-ctl --list-devices > just returns: /dev/video0 which is my webcam [16:30] <c_14> Maybe you need the driver? [16:43] <loa> hello. how i can handle "can't write packet with unknown timestamp"? I tried -fflags +genpts with no luck. I am trying to convert .h264 to .mkv [17:03] <Marcin_PL> c_14: BIG thanks, now the script works properly. [17:05] <Venemo> hey guys [17:05] <veltz> hi [17:05] <Venemo> is it true that ffmpeg is deprecated? I've read some conflicting articles about it [17:06] <JEEB> ffmpeg the binary in libav the project is, which is a fork of ffmpeg the project [17:06] <Venemo> ah [17:06] <JEEB> in fffmpeg the project ffmpeg is still very much alive [17:06] <JEEB> *ffmpeg [17:06] <JEEB> so if you are getting that warning you probably are using a distro that packages libav [17:07] <JEEB> and in that case you should be using avconv with the same syntax [17:07] <JEEB> that said you are probably on something old'ish as far as libav goes if it still has the ffmpeg binary [17:08] <Venemo> I didn't get a warning from my distro-packaged ffmpeg, fortunately. [17:08] <Venemo> there was just a guy on twitter who tried to be a smartass, it seems [17:08] <justinX> Venemo: what linux dist do you use? [17:08] <Venemo> Fedora 21 [17:08] <JEEB> that doesn't package ffmpeg to begin with vOv [17:08] <JEEB> or libav [17:08] <Venemo> well, their "rpmfusion" repo does [17:09] <JEEB> it's not really "theirs" afaik [17:09] <JEEB> it's third party packages [17:09] <Venemo> not officially, at least, yeah [17:09] <justinX> are there something like the ppa's that linuxmint/ubuntu have for fedora? I mean extra repositories people make? [17:09] <Venemo> justinX: yeah [17:10] <Venemo> JEEB: at least they seem to be packaging the true ffmpeg, and a quite recent version of that [17:10] <minnesotags> A while ago, someone here gave me some helpful advice. I needed to encode a huge amount of wav files to flac through a directory tree. I got something better to share.... [17:11] <JEEB> well, both ffmpegs were "true", libav just renamed the tool when they rewrote parts of it and then after a release they removed it altogether [17:11] <JEEB> FFmpeg then merged those changes to its ffmpeg tool [17:11] <Venemo> I'm glad to see that the original project is still alive and rocking [17:11] <JEEB> (FFmpeg being the project, ffmpeg being the tool) [17:12] <minnesotags> First you have to apt-get install "parallel" (this is for Debian". Then "find . -type f -name "*.wav" | parallel flac {} -keep-foreign-metadata" (if keeping foreign metadata is the tag you want). [17:13] <minnesotags> I had over 7000 songs to encode. That cut the time from 6 hours to 2. [17:13] <minnesotags> Sorry, the tag shoud be "--keep-foreign-metadata". [17:14] <justinX> are there any specification of how linebreaks in subtitles are supposed to be when embedded inside an mp4? [17:15] <minnesotags> Point is if you have any other huge batch process to do, "parallel" is an awesome command/package. [17:15] <Venemo> justinX: as for your previous question (about ppa-like stuff for fedora), take a look at https://copr.fedoraproject.org/ [17:15] <Venemo> minnesotags: nice stuff [17:16] <minnesotags> 2-3 seconds for file adds up when you have 7500 files. [17:16] <Venemo> yeah [17:16] <Venemo> the -threads option for ffmpeg is also useful in this regards [17:16] <minnesotags> It will use as many cores on your processor as are available. [17:17] <justinX> aha [17:17] <minnesotags> I had seen tons of people suggesting batch files on stackoverflow and elsewhere. Seems many people don't know this package. [17:59] <Marcin_PL> How should I encode to MJPEG with highest quality? [18:17] <Marcin_PL> Hello? :) [18:20] <loa> Marcin_PL, i think you need to just start experiment... [18:24] <Marcin_PL> I started& Now i'm trying -q:v 0 [18:30] <Marcin_PL> [swscaler @ 0xf46460] deprecated pixel format used, make sure you did set range correctly [18:30] <Marcin_PL> huh [18:30] <loa> why you need mjpeg? [18:30] <Marcin_PL> Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj420p, 640x480, q=2-31, 200 kb/s, 30 tbn, 30 tbc [18:31] <Marcin_PL> Because it's just a source to further cuts [18:32] <Marcin_PL> And I think it's best compromise, because RAW bitmap is too large and anyway I got RAW from stabilising script. [18:45] <Venemo> I'm encoding some h264 and getting this message frequently: "Application provided duration: -7 / timestamp: 488588 is out of range for mov/mp4 format" and "pts has no value" [18:45] <Venemo> what do they mean? [18:46] <Venemo> note that the source format is also h264 [19:51] <jcay> hello, is there a possibility to do video filter only for certain time, not whole video, like 10 seconds and rest just copy? [19:56] <loa> Venemo, i found this pts errors in ffmpeg so annoing... [19:57] <Venemo> loa: sure, but what does it actually mean? [20:00] <loa> i think if it is not fatal and encoding is actully done it is ok for you. [23:18] <pagios> hello, tryin to install faac http://pastebin.com/kKPJXjwJ any idea? [00:00] --- Sun Jan 4 2015

1 0

ffmpeg-devel.log.20150102
by burek 03 Jan '15

03 Jan '15

[00:34] <kierank> Daemon404: have you used hopper? [00:34] <kierank> it's quite nice [01:00] <Daemon404> kierank, hopper? [01:00] <kierank> hopper disassembler [01:00] <Daemon404> oh that [01:00] <Daemon404> i trid it [01:00] <Daemon404> for 32-bit, it's not quite as nice as hexrays [01:01] <Daemon404> x64 is obviously better. [01:01] <kierank> it has a decompiler as well [01:01] <kierank> but to pseudocode [01:01] <Daemon404> yes [01:01] <Daemon404> hexrays has some nicer features, but is 32bit only [01:02] <Daemon404> (there supposedly also exists an arm version) [01:02] <Daemon404> i guess if youre trying to Be Legal (TM) then hopper is actually, you know, purchasable [01:03] <wm4> how much does this hexrays thing cost? [01:03] <Daemon404> doesnt matter [01:03] <Daemon404> they dont sell to you unless you are notable [01:03] <wm4> lolwut [01:03] <wm4> seems like a bad sales strategy [01:04] <Daemon404> not really [01:04] <Daemon404> theyre undispitedly the best, and every single firm uses them [01:04] <Daemon404> undesputedly* [01:04] <compn> they are e-l335 ! [01:04] <Daemon404> eh, screw spelling. [01:04] <compn> er el33t [01:45] <BBB> isnt hexrays the decompiler? [01:45] <BBB> you dont need hexrays, IDA is good enough [03:25] <michaelni> cousin_luigi, thx [03:33] <cone-515> ffmpeg.git 03Michael Niedermayer 07master:aeb36fd2072e: avcodec/mpeg12dec: Move user data debug code out of unrelated if (buf_size > 29) [03:33] <cone-515> ffmpeg.git 03Michael Niedermayer 07master:1010b36d8672: avcodec/mpeg12dec: Recalculate SAR unconditionally [03:33] <cone-515> ffmpeg.git 03Michael Niedermayer 07master:75cc57f73f9a: avcodec/mpeg12dec: Check actual aspect ratio instead of aspect_ratio_info [08:54] <anshul_mahe> michaelni: I have tried to complete all your comments in closed caption decoder patch, is something still missing [09:38] <nevcairiel> someone should delete #4219, its some disguised spam again [10:04] <anshul_mahe> closed it as invalid [10:06] <ubitux> it should be deleted [10:06] <ubitux> btw it's kind of interesting [10:06] <ubitux> it looks like a bot that tried to pick this: http://trac.sagemath.org/ticket/3101 [10:06] <ubitux> and it seems to have attempted the same thing in several other places: https://josm.openstreetmap.de/ticket/10915 https://www.mail-archive.com/tor-bugs@lists.torproject.org/msg67946.html [10:48] <anshul_mahe> ubitux: I dont know how to delete, please do the needful if you know how to delete [10:48] <ubitux> i don't have the rights [10:48] <ubitux> afaik [11:47] <cone-375> ffmpeg.git 03Clément BSsch 07master:55faf56c7256: avformat/mov: move edit list heuristics into mov_build_index() [11:47] <cone-375> ffmpeg.git 03Clément BSsch 07master:11201bbf7fc9: avformat/mov: reindent after previous commit [11:48] <ubitux> akira4: i'll try to reply to your mail tonight :) [11:48] <akira4> okay ! thanks :) [14:49] <datenwolf> Hi, I'm trying to understand the ffmpeg build system. Specifically I'd like to use libswscale (and just libswscale) in a small project of mine and for that transplant a copy of the libswscale sources into the source tree of my project (the rationale behind that is, that I don't want to have people install a full ffmpeg source build just for building a program which grabs webcam images and process [14:49] <datenwolf> es them (and may have to perform YUV->RGB transformation for that)). [14:50] <datenwolf> So essentially what I actually want to understand is how to build the libswscale sources without using the ffmpeg build system. [14:50] <wm4> libswscale is pretty bad for yuv->rgb conversions [14:50] <datenwolf> wm4: and the other way round? Because that will happen as well. [14:50] <wm4> but what's hard to understand? it's handwritten, a bunch of shell scripts and makefiles [14:50] <wm4> maybe even worse [14:51] <datenwolf> wm4: Well for example there's identically named source files in libswscale/ and libswscale/x86, so I wonder are those complementary or do they replace it. [14:52] <wm4> also, you can disable almost everything when building ffmpeg (though I'm not sure if you can actually disable everything except libswscale) [14:52] <datenwolf> wm4: Excuse me? Worse for colorspace transformation? So given the fact that you have to convert color spaces for about every lossy video codecs and swscale being used by ffmpeg for this, this statement surprises me. [14:52] <wm4> creating your own build system for libswscale sounds like maintenance nightmare [14:52] <JEEBsv> it is, and you really don't want to do it [14:52] <datenwolf> wm4: Well, actually I'm using CMake [14:53] <wm4> libswscale just has been dragged along from its beginnings... it was never particularly good [14:53] <wm4> oh, I'm sorry [14:53] <datenwolf> JEEBsv: So, a) why is it then still in ffmpeg and b) any suggestions for a different library. I'd really prefer not to write that stuff myself. [14:54] <wm4> Libav wanted to rewrite libswscale for ages, but nothing came of it [14:54] <JEEBsv> libz? [14:54] <wm4> yeah, zimg might be an alternative [14:54] <JEEBsv> not as optimized, but seems to be getting used @ vapoursynth [14:54] <wm4> datenwolf: a) because it's always been used in ffmpeg, and is not easy to replace [14:55] <datenwolf> So when you say "swscale is bad" what exactly do you mean by it? [14:55] <wm4> maybe libyuv is also viable? [14:55] <wm4> slow and low quality [14:55] <JEEBsv> you usually want only one of those at most [14:56] <JEEBsv> https://github.com/sekrit-twc/zimg [14:56] <JEEBsv> this is the one that seems to have most buzz around it atm [14:56] <datenwolf> JEEBsv: Well, essentially the problem boils down to: Webcam may not support the colorspace I need (I420 in particular), so whatever comes out of it, must be converted first (and fast). [14:56] <wm4> this is probably more for scaling and end-conversion [14:56] <datenwolf> So anything that gets the job done is fine to me [14:56] <wm4> not so much about handling the fucked up shit webcams can output [14:57] <wm4> like potentially packed formats etc. [14:57] <JEEBsv> seems to be planar YCbCr so far according to his words :D [14:57] <wm4> well, some webcams even output jpeg and stuff [14:57] <JEEBsv> yes [14:57] <JEEBsv> but in that case you don't only need swscale :P [14:57] <datenwolf> wm4: Actually the jpeg case is what I'm mostly dealing with, and I got that covered already. [14:58] <JEEBsv> you need avcodec and friends, or whatever you choise [14:58] <wm4> I'm just saying yuv conversion alone is not all of it [14:58] <JEEBsv> of course not, but that's what he seems to be wanting [14:58] <JEEBsv> scaling and colorspace conversion [14:58] <JEEBsv> :P [14:58] <datenwolf> But JPEG is quite allowing in the colorspaces that can be shoved through it. [14:59] <datenwolf> I'm using libuvc to access the webcams (deliberately bypassing any system drivers for webcam access here). [14:59] <JEEBsv> anyways, zimg can be quick with newer CPUs, swscale has some optimizations for various older CPUs [14:59] <JEEBsv> but you have to remember that swscale isn't exactly doing it all the best way [14:59] <cone-375> ffmpeg.git 03Michael Niedermayer 07master:4c7a1ccb366c: Changelog: Add cropdetect non 8bpp [14:59] <JEEBsv> since it's meant for quick n' dirty things [15:00] <datenwolf> JEEBsv: actually quick and dirty sounds nice to me :) [15:00] <wm4> it originates from mplayer, where they wanted to scale and convert 4:2:0 in realtime on pentiums, or something [15:00] <wm4> (I wonder if that ever actually worked...) [15:00] <JEEBsv> maybe with the resolutions in use back then... [15:00] <JEEBsv> well, I'm just saying that you might be able to use both [15:00] <wm4> datenwolf: also, libswscale depends on libavutil [15:01] <JEEBsv> also if you are going to use either, you will want to just make your cmake build system look for the libraries, or you use git submodules and use the ffmpeg/zimg build system from cmake [15:01] <datenwolf> The main reason why I don't want to use the full ffmpeg stack for the whole processing (which actually it could be used for) is, that I never managed to get its processing chain down to only a single frame of latency. [15:02] <JEEBsv> that's possible but requires poking at the buffering etc [15:02] <JEEBsv> never done it fully myself but I've seen people do it :P [15:02] <datenwolf> Yes, and I'd like to avoid that. ffmpeg is kind of a swiss army knife, but what I need is more like a specialized surgical scalpel right now. [15:02] <JEEBsv> input buffering, filtering and then making sure the encoder plays along as well [15:04] <wm4> yeah, ffmpeg has the surgical precision of as shovel [15:04] <wm4> (just wanted to say that) [15:05] <wm4> anyway, libswscale is so disappointing, people have written their own naive conversions in C, and were then surprised that it was faster than libswscale [15:06] <anshul_mahe> wm4: if u have some benchmark regarding what u ar speaking, why dont you make swscale faster [15:07] <wm4> because I'm neither an optimization or colorspace conversion expert, nor am I courageous enough to hack libswscale code [15:08] <wm4> and I think it's important not to promise the wrong things [15:08] <wm4> also, libavscale will liberate us all [15:09] <datenwolf> Maybe I should quickly explain, what I'm trying to do: Take two webcams in a stereoscopic configuration, mount them on a Oculus Rift. Fetch their images with as little latency as possible, encode/lossy compress the video stream in an error resilient fashion, adding the positional tracking data, send it over UDP (I don't care if packages are lost, just get it there fast). On the receiving side dec [15:09] <datenwolf> ompress into OpenGL textures and integrate it into Cube Maps, then view these through another Oculus Rift on the other end (using the viewing Rift's tracking data to control the view onto the cubemaps). And do that in both directions. [15:09] <datenwolf> Some of the code to do this is already in place (the whole cubemap integration and viewing for example). [15:10] <datenwolf> I originally wanted to use x264 in zerolatency mode, but then had the problem that apparently there's no "standalone" lightweight decoder for h.264 available. So my second choice fell on libvpx. [15:11] <JEEBsv> well, depending on your feature set used that cisco thing might be useful for you [15:11] <JEEBsv> although it probably doesn't support stuff like 4:4:4 chroma for example [15:12] <JEEBsv> that said, libvpx doesn't support it either in any sane configuration [15:12] <wm4> anyway, ffmpeg can be stripped down a lot [15:12] <datenwolf> libvpx expects a I420 format. I already got that part working, but only from a file with the images preconverted. [15:13] <wm4> while distros force you to use their version (or something), it doesn't matter if you redistribute your program as a blob [15:13] <datenwolf> Or YV12 which is kind of the same a I420 but with chroma planes swapped. [15:13] <wm4> and then I believe you could reduce ffmpeg to having a h264 decoder and libswscale, or so [15:13] <datenwolf> Well, actually I'd prefer not to distribute a blob, but a repository which kind of builds "stand alone". [15:15] <datenwolf> Maybe that's just habitual, but it comes with the line of work I'm earn my money with, where we have to produce reproducible builds and have tight control over every piece code that goes into the final program. [15:15] <datenwolf> s/I'm earn/I'm earning/ [15:18] <datenwolf> libyuv looks exactly what I need. [15:18] <datenwolf> have to take a glimpse at zimg yet, but so far it looks far better suited than swscale to me. [15:18] <datenwolf> Thanks! [15:20] <datenwolf> Okay, which zimg is zimg? If I type "zimg library" into Google I get a number of (orthogonal) results. [15:21] <JEEBsv> the twc-sekrit one on github [15:21] <datenwolf> JEEBsv: got it. [15:23] <datenwolf> Okay, both libyuv and zimg are implemented in C++; not that I'd complain, but I's prefer pure C (again habitual, due to the headaches I've to deal with at work). [15:24] <JEEBsv> the API for zimg should be C [15:24] <JEEBsv> so as such it shouldn't be a problem [15:24] <datenwolf> Yes it's C. [15:25] <datenwolf> Well for a stand alone project it doesn't matter. But the linkage problems involved when multiple C++ libraries built by different compilers collide, ugh...) [15:26] <datenwolf> Take one library that makes liberal use of G++ extension features and take one library that uses CUDA and try to use them in a Windows build. [15:26] <datenwolf> In Windows CUDA only works with MSVC++ [15:27] <wm4> C++ being "more portable" is part of why everyone is using it [15:27] <datenwolf> define "more portable"... [15:27] <wm4> though that's mostly MS actively sabotaging C, and the C standard committe being a bunch of old turdheads [15:28] <cone-375> ffmpeg.git 03Supraja Meedinti 07master:c6bb651bce2b: libavutil: Added Camellia symmetric block cipher [15:28] <wm4> datenwolf: these days you can have threads, atomic, and a number of things that work on all platforms in the core language [15:29] <wm4> while in C you must NIH your portable mechanisms [15:29] <datenwolf> wm4: C11 defines most of these as well. [15:29] <datenwolf> (or rather all of them, actually) [15:29] <wm4> except it's completely useless [15:29] <wm4> because for example MSVC doesn't implement C11 [15:30] <wm4> or even if they do, threads are most likely a toy implementation that doesn't interact well with "foreign" native threads [15:30] <wm4> just like "fopen" is completely useless in msvc [15:30] <datenwolf> wm4: Don't remind me (O_BINARY / O_TEXT...) [15:30] <wm4> the problem with fopen is that it doesn't do utf-8 [15:31] <datenwolf> wm4: You mean in filenames? [15:31] <wm4> yes [15:32] <datenwolf> Well, that's actually not a fopen problem, but a problem deep down in the Windows API and the assumptions made by legacy programs Microsoft doesn't want to break if they introduced proper UTF-8 support in filenames. The filesystems deal with it just fine. [15:32] <wm4> the main issue is that MS doesn't care [15:34] <datenwolf> A number of "shortcommings" of the Windows API are actually that Microsoft doesn't want to break legacy code. And fopen they consider a legacy API. They don't care? Maybe, but it's also part negligence in the original implementation and lock-in due to broken programs. [15:34] <datenwolf> Anyway, thanks for your help and suggestions. [15:34] <datenwolf> both libraries look fine, and I'll try each of them. [15:35] <wm4> since on Microsoft each program has its own (independent) stdlib anyway, there's no reason why MS couldn't provide a fixed, modern replacement [15:36] <datenwolf> wm4: Well, every version of MSVC comes with a new MSVCRTxxx.dll (and a bunch of statically linked and debug build counterparts). [15:37] <datenwolf> So it could be integrated in that quite effortlessly. [15:37] <wm4> I also blame mingw [15:37] <datenwolf> And the API to access UTF-8 filenames does exist in the Win32 API. [15:37] <wm4> and the terrible way they use an ancient msvcrt, and their own replacements for some functionality [15:37] <wm4> what where [15:37] <datenwolf> You mean, because MinGW uses the system default MSVCRT? [15:38] <wm4> yes [15:39] <datenwolf> That's probably to avoid licensing issues regarding the redistributable. [15:40] <datenwolf> but yeah, shitty situation. However the MSVCRT gets updated with every version of Windows as well. It retains the interface used up until VisualStudio-6 for compatibility (VS6 linked against the system MSVCRT as well). [15:40] <datenwolf> But it _gets_ updated. [15:42] <datenwolf> After all it's the CRT drivers and services are to be linked against. [16:51] <kierank> datenwolf: what's wrong with just disabling the parts of the ffmpeg build you don't need [16:52] <datenwolf> kierank: That then I've to include the whole ffmpeg sources into my source tree OR have people perform a tailored ffmpeg build themselves. [16:53] <datenwolf> kierank: Also I'm not sure just disabling the unneeded parts and using libavcodec would give me the required low latencies without hacking into the libavcodec sources themself. [16:53] <kierank> then you are wrong [16:53] <kierank> just because ffmpeg.c is not low latency [16:54] <kierank> doesn't mean libavcodec is suddenly high latency [16:55] <datenwolf> Well, a few years ago I had another project (without the latency requirements) where I had to fetch images from Webcams as well. And because I was lazy I just used libavcodec for that. But I never got the camera latency down as much as I managed with this project by using libuvc and accessing them directly. [16:56] <datenwolf> I made a small libavcodec wrapper for that then (terribly outdated, ffmpeg's API moved along some way) https://github.com/datenwolf/aveasy [16:59] <kierank> what kind of latency are you looking for? [17:15] <iive> imho, most of the latency is in the output muxers. libx264 could buffer a dozen frames, but it also have nice options to disable that and lower latency further. [17:16] <nevcairiel> libx264 has a low-latency option as well if that is desired [17:16] <nevcairiel> the decoders dont buffer frames needlessly in my experience, you just need to turn off frame-threading if latency is a concern [17:17] <nevcairiel> but a webcam is unlikely to spit out h264 anyway, rather some intra codec like mjpg, or even uncompressed [17:17] <nevcairiel> (or at the very least only i/p frames, not b frames) [17:17] <nevcairiel> (so no delay) [17:31] <datenwolf> iive: Well, in that case I mentioned we just grabbed the images from the Webcam. No encoding whatsoever. [17:32] <datenwolf> We were using libavcodec then to have some flexibility regarding the end system setup. [17:32] <iive> datenwolf: -c copy doesn't do any encoding either, but it does demuxing and mixing. [17:32] <datenwolf> Could have been a USB camera or a IEEE1394 [17:33] <datenwolf> iive: Well, look at my aveasy code; it's just that what we used and we has between 10 to 15 frames of latency. [17:33] <datenwolf> Not a big deal for that project then, but a huge roadblock if to be used on a VR system. [17:34] <datenwolf> iive: That's why I've decided to bypass as much as I could using libuvc to get the camera images. [17:34] <datenwolf> And the result speaks for itself. [17:34] <anshul_mahe> is there any filter which merge 2 programs to 1 [17:35] <iive> datenwolf: link? [17:35] <anshul_mahe> https://github.com/datenwolf/aveasy [17:35] <datenwolf> iive: that would be the "old" stuff. [17:36] <nevcairiel> its hard to judge where the delay is being created, v4l2 might add delay [17:36] <datenwolf> The low latency stuff is still in a rather unordered form. Needs a proper repository. Will setup a public Git repo as soon as its done (that was the goal from the beginning). [17:37] <datenwolf> nevcairiel: Ideed that may be the case. That's why I now use libuvc to get around anything that might introduce latency. [17:38] <datenwolf> The drawback is, that it's no longer truly "plug-and-play", because it requires raw USB access and at least in Linux you must add a few UDev rules to set the /dev/bus/usb/* nodes to rw for permitted users/groups. [17:41] <iive> i thought that v4l2 provides monotonic timestamps, so you should be well aware of the delay of the frames when you get them. [17:41] <iive> the code looks like simple wrapper, and it even uses swscale. [17:43] <datenwolf> iive: Yes, this is old code. I no longer use/maintain it. [17:44] <datenwolf> The timestamps are only good to know the latency. But if the frame's are "stuck" in the kernel side buffers you don't get them until you get them. And there's nothing you can do about that. Which means, you can't get less latency than the kernel allows you. [17:44] <iive> true [17:45] <datenwolf> iive: And yes, it uses swscale. That's why I was asking, because I already know how to use it. But then ffmpeg was just installed on the system and I had not to care that other people had to build it. [17:46] <anshul_mahe> datenwolf: you can try xenomai patch for kernel, and make the latency accurate [17:48] <datenwolf> anshul_mahe: Why should I bother? libuvc gives me extremely low latency (less than a frame in fact, measured it using a flashing LED). [17:49] <datenwolf> And installing custom kernels is even more inconvenient than just chmod 660 on /dev/usb/... or installing a udev rule, this would be kind of counterproductive. [17:51] <datenwolf> Apparently the Webcams I use use rolling shutter (no big surprise) and feed the scanlines to USB as they come in. [17:52] <iive> so, libuvc access the usb camera directly, without bothering with the kernel? [17:52] <anshul_mahe> no library can access kernel directly [17:53] <datenwolf> iive: it goes directly to the USB using the kernel interfaces. [17:53] <datenwolf> In Linux you have /dev/usb/... in Windows you have WinUSB. [17:53] <iive> like libusb [17:53] <datenwolf> iive: In fact libuvc uses libusb [17:55] <iive> i would expect it to :) [17:55] <anshul_mahe> libuvc seems to interface directly uvc webcam driver rather then using v4l2, just a guess [17:56] <datenwolf> andrewrk: libuvc does implement a UVC driver [17:56] <datenwolf> It's not using any OS drivers other than direct access to USB. [17:57] <datenwolf> anshul_mahe: UVC is simple enough [17:57] <anshul_mahe> yes I have started looking at it [17:57] <anshul_mahe> I am looking at https://github.com/ktossell/libuvc, this is the one [17:58] <datenwolf> Of course the latency also depends on the Webcam, how it controls the shutter, internal buffers, etc. etc. [18:04] <anshul_mahe> datenwolf: thnks it look good i will surely try to add libuvc support in ffmpeg, but may be some month later [18:04] <datenwolf> anshul_mahe: Take you time :) [21:23] <cone-375> ffmpeg.git 03Nicolas George 07master:55763b6f5eda: lavd/lavfi: allow to extract subcc. [22:34] <cone-375> ffmpeg.git 03Giorgio Vazzana 07master:88d19d240aee: avutil/camellia: fix documentation for av_camellia_crypt() [22:34] <cone-375> ffmpeg.git 03Giorgio Vazzana 07master:8e38b1539e7d: avutil/camellia: make LR128() more robust [22:35] <cone-375> ffmpeg.git 03Giorgio Vazzana 07master:fbb792f90fce: avutil/camellia: use K[2] instead of *K in generate_round_keys() [22:35] <cone-375> ffmpeg.git 03Giorgio Vazzana 07master:a3ab87427c53: avutil/camellia: cosmetic fixes [00:00] --- Sat Jan 3 2015

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ffmpeg.log.20150102
by burek 03 Jan '15

03 Jan '15

[00:07] <smo_> hi :) [00:07] <smo_> still loving ffmpeg more and more .... :p [00:08] <smo_> i m trying some visualisations thru ffmpeg with exemples on this page http://video.stackexchange.com/questions/9644/how-do-i-turn-audio-into-vide… [00:08] <smo_> is there some others viz availables? [00:09] <smo_> i search visualisation by band like html5 visualisations http://www.smartjava.org/sites/default/files/localhost_Dev_WebstormProjects… [00:09] <smo_> possible? [00:11] <c_14> The only other filter that does A->V that I know of is: ... showcqt A->V Convert input audio to a CQT (Constant Q Transform) spectrum video output. [00:11] <justinX> I normally don't use visualisations on music, but yesterday it could have been nice to have one that presents the music as fireworks :-) [00:11] <ubitux> smo_: ffmpeg -v 0 -filters|grep 'A->V' [00:12] <ubitux> also note that you have many option for these [00:12] <smo_> humm i use -i and -ss yet [00:12] <ubitux> http://ffmpeg.org/ffmpeg-filters.html#showspectrum [00:12] <smo_> i look thx [00:12] <ubitux> showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt this one from the doc is cool [00:18] <smo_> works but it really show a spectrum, nice but not what i search :p really search something likemy exemple image [00:18] <smo_> avectorscope seems right too but it only draw a thine line [01:17] <smo_> ok ubitux i ll use this finally , pretty cool [01:17] <smo_> ffmpeg -ss 00:00:00.00 -i /home/smo/T\u00E9l\u00E9chargements/Ideal J - Le Combat Continue 1998 FLAC/11-j'ai mal au coeur.opus -filter_complex [0:a]showwaves=mode=cline:rate=25,format=yuv420p[vid] -map [vid] -map 0:a -codec:v libx264 -crf 18 -preset ultrafast -codec:a libvorbis -threads 0 -copyts -sn -f matroska pipe:1 [01:18] <smo_> so i have visualisations in my node-webkit app [01:18] <smo_> LOOOOOOOOOOOOOVE ffmpeg :p [01:19] <smo_> thanks to the devs for all this work ! [01:20] <smo_> a "classic" visualisation by band (horizontals bars by frequencies ) might be a good idear :p [03:54] <Keshl> Thank you, whoever worked on hqdn3d! o_o.... I just fed it some footage shot at the equlivant of 128,000 ISO and it looks like it's around 800 now. o_o... [07:43] <fmax30> Hello, ffmpeg current is very slow even on the latest android devices. converting a 15 second 480p video takes on average 14-15 seconds. Here is the command that i am using. [07:45] <fmax30> -y -i input.mp4 -strict -2 -threads 4 -b:v 700k -s 640x360 -r 30 -vcodec libx264 -acodec aac -ac 1 -b:a 64k -ar 44100 -preset ultrafast -crf 24 -vf vflip [07:45] <fmax30> output.mp4 [08:28] <fmax30> anyone ? [12:25] <Mavrik> fflogger, and? [12:25] <Mavrik> bah. [12:25] <Mavrik> damn slow logs [12:52] <kaotiko> hi [13:46] <shevy> does anyone happen to know whether the ffmpeg filters also have an example of a filter that removes or blurs out a logo? [13:46] <shevy> ah they do [13:46] <shevy> T.. delogo V->V Remove logo from input video. [13:46] <shevy> T.. removelogo V->V Remove a TV logo based on a mask image. [14:52] <Marcin_PL> Hello! I got some problem with fps here& The point is I got MJPEG file (with PCM sound) at 30 fps (easy Canon PowerShot) and I cannot make it the same duration time in my rotating script. I used -r 30 and it worked somehow (default is 25), but not well (movie is going to be longer). I learned some time ago to work on ffmpeg separately with audio and video, so if it's not the reason pls don't irritate about separating sound off. [14:52] <Marcin_PL> Here's the script: http://pastebin.com/mDi9NjD6 [15:53] <Glanzmann> Hello; I tried to cut some parts out of h264/mp4 video and merge them together with copy. ffmpeg -i k1_last_day.mp4 -ss 0:00:00 -c copy -t 0:01:00 1.mp4 ... ffmpeg -f concat -i list.txt -c copy output.mp4 the resulting video does not open in google chrome. What did I do wrong or what do I do to reencode it. I tried ffmpeg -i output.mp4 -codec:v libx264 -profile:v high -threads 0 -f mp4 output2.mp4 which gave me Unable to parse option value "-1" as p [15:56] <c_14> Did you make sure that the videos you were trying to concat all had the same settings? (pixel format, codec, fps, etc) [15:58] <Glanzmann> c_14: Yes, they came from the same original video. [15:58] <Glanzmann> If I play it with mpv the picture hangs when I go to clip 1 to clip 2. [15:58] <Glanzmann> If I seek, everything is fine. [16:00] <Glanzmann> But even the first extract that I took using copy does not work in google chrome, so I assume it needs reencoding. [16:00] <c_14> If the original works, then the copied sub-portion should work as well. [16:01] <Glanzmann> It does not. [16:02] <Glanzmann> I try to do the same with a static build from the website. [16:03] <Glanzmann> Same problem. [16:04] <Glanzmann> Give me one second. [16:07] <Glanzmann> https://pbot.rmdir.de/2mbCXLirKwXiltAJoQrwWw [16:11] <c_14> try adding -pix_fmt yuvj420p as an input option to the last command [16:13] <Glanzmann> (lenovo) [~] ffmpeg -y -f concat -i list.txt -c copy -pix_fmt yuvj420p output.mp4 2>&1 | pbot [16:13] <Glanzmann> https://pbot.rmdir.de/rDBfz_JYKkeIfgednyxpkA [16:13] <Glanzmann> Same problem. [16:13] <c_14> input option [16:13] <c_14> before -i list.txt [16:14] <Glanzmann> Another error: (lenovo) [~] ffmpeg -y -f concat -pix_fmt yuvj420p -i list.txt -c copy output.mp4 2>&1 | pbot [16:14] <Glanzmann> https://pbot.rmdir.de/uUQI_MQhO-mzG4CtAQq0QQ [16:14] <c_14> -pix_fmt [16:14] <c_14> wait [16:15] <c_14> you did right -pix_fmt, why is it saying it can't find pixel_format... [16:15] <Glanzmann> My assumption is that when I do the extraction part, it screws up the extracted parts. [16:15] <Glanzmann> Because I used copy before and it worked. [16:16] <c_14> try adding -pix_fmt yuvj420p as an output option to all the ones before [16:16] <c_14> *commands [16:16] <Glanzmann> My GOPRO splits up in multiple 4 GB files, I used ffmpeg to reencode to h264 360p and than I used ffmpeg cocnat to merge it. [16:17] <Glanzmann> Okay. Where do I put the output option? [16:17] <c_14> somewhere before the 1.mp4 etc [16:17] <c_14> and after the k1_last_day.mp4 [16:19] <Glanzmann> I see. That worked. But same problem: https://pbot.rmdir.de/WqfljlZtfVAlpFgSR64Evg [16:23] <Glanzmann> If you cut with stream copy (-c copy) you need to use the -avoid_negative_ts 1 option if you want to use that segment with the #concat demuxer . [16:23] <Glanzmann> Source: https://trac.ffmpeg.org/wiki/Seeking [16:25] <c_14> mhm, try that? [16:26] <Glanzmann> Yes, did not fix the problem. [16:26] <Glanzmann> I got it. [16:26] <Glanzmann> Problem was that the offsets need to go in front of the input file. [16:27] <c_14> putting the -ss in front of the input file fixed it? [16:27] <Glanzmann> Yes, it did. [16:27] <c_14> that's weird [16:27] <Glanzmann> I now try to get a minimal modification. [16:27] <Glanzmann> It also now takes a lot longer to encode. [16:27] <Glanzmann> I read somewhere that one is better and the other one is faster. [16:28] <Glanzmann> c_14: Thanks for the help. [16:29] <Glanzmann> Minimal example is: https://pbot.rmdir.de/gnX26WXQj0vw9qG6vgBSKw [16:33] <Glanzmann> Sorry the last script had an error. https://pbot.rmdir.de/H2wxt7fkGcdQ0wYs-Wzh6A [16:36] <c_14> hmm, k. Still not sure why the -ss as an output option broke it, but I'll keep it in mind [16:40] <diffuse> hi, i am trying to write an app to decode any audio format and covert it to AV_CODEC_ID_PCM_S16LE [16:40] <Glanzmann> c_14: Thanks again for your help. [16:41] <Mavrik> diffuse, you might want to ask an actual question :P [16:41] <diffuse> getting there :-) [16:41] <diffuse> I have the mp3 decode working fine, and i am invoking the decoder, [pcm_s16le @ 0x6030a0] nb_samples (1152) != frame_size (0) (avcodec_encode_audio2) [16:41] <diffuse> and i get that error [16:42] <Mavrik> diffuse, um, you don't really have to encode PCM audio [16:42] <Mavrik> when you decode audio it's already in PCM format :) [16:42] <diffuse> well [16:43] <diffuse> is there a way to set specific flags when decoding the mp3 to make it decode to AV_CODEC_ID_PCM_S16LE? [16:44] <diffuse> i am trying to reproduce the results of what -acodec pcm_s16le -ar 16000 would produce [16:44] <Mavrik> hmm, not really because it'll decode to whatever the mp3 is in [16:44] <Mavrik> diffuse, use filters for that, not encoder [16:44] <Mavrik> diffuse, initialize a filter chain in form of "aformat=s16,aresample=16000" [16:45] <Mavrik> since filters are the ones that convert raw frames, not encoders per-se [16:45] <Mavrik> PCM encoder just does passthrough :) [16:46] <diffuse> Mavrik: ok, do you have a url to the api doc handy? [16:46] <Mavrik> diffuse, doc/examples/filtering or whatsit [16:47] <Mavrik> diffuse, also look at filtering documentation for ffmpeg binary, all filters you have available are listed there [16:47] <diffuse> ok [16:47] <diffuse> so, do i decode the mp3 then run it through the filter to get my desired output? [16:47] <Mavrik> yep [16:47] <Mavrik> filter chain actually, since you'll have to resample the audio and convert the format :) [16:48] <diffuse> so, i first run the mp3 through a filter, then run it through decode? [16:48] <diffuse> or can all this be done in a filter chain directly? [16:49] <Mavrik> ugh [16:49] <Mavrik> you filter decoded raw audio [16:49] <Mavrik> you can't filter encoded data :) [16:50] <Mavrik> you grab a raw PCM audio frame in whatever format the decoder gives you and you push it into filter chain [16:50] <diffuse> ok, that makes more sense [16:50] <Mavrik> then you grab it out of the filter chain filtered :) [16:50] <Mavrik> see the example code, that's probably the easiest to understand [17:34] <anshul_mahe> is there any filter which merge 2 program stream [17:36] <Mavrik> define merge. [17:39] <anshul_mahe> Mavrik: anything which takes 2 program (2 audio 2 video pair ) and give one 1 program in output [17:40] <Mavrik> anshul_mahe, ok, and how would that look like? Muxed together in a single TS, picture by picture, audios separate or mixed? [17:41] <anshul_mahe> I was trying to implement a filter which takes 2 programs and give 1 programs in output, putting some advertisement in video stream [17:41] <anshul_mahe> I wanted some filter for reference [17:42] <anshul_mahe> means taking 2 video of 1 hour and 10 minute and give output of 1.10 hr [17:42] <c_14> So basically concat? or trim + concat? [17:43] <anshul_mahe> c_14: thnks I will look at it [17:45] <anshul_mahe> I cant use anything already inside there, because timing are in 1 of the program, abt when to trim and when to concat, but that filter would be good start [17:47] <SuperSQL> Hi, I would like to extract a single frame from long video to a jpg file, I use this command: "ffmpeg -ss xx:xx:xx.xx -i filepath -vframes 1 output.jpg" it works great for most video files but on some file it takes way lot longer to extract frame, is there any faster way of seeking? is there a way to reach same speeds as VLC or any other player wh [17:47] <SuperSQL> en seeking? Percision is not important, I would just want to extract frame with rough timestamp. Can anyone help me with that? [17:49] <mcnesium> i accidentally quit an ffmpeg transcoding session. can i somehow resume it instead of starting over? it took quite a while so far& [17:51] <c_14> mcnesium: if you didn't output to mp4, you can check how far the output got, seek to that point in the input and keep encoding then concat the 2 files [17:51] <c_14> SuperSQL: afaik, that is the fastest method of seeking [17:52] <mcnesium> c_14: output file was .webm [17:52] <c_14> mcnesium: use ffprobe to check how long the webm is, then use the same command as before with a different output file and -ss [time] [17:52] <c_14> Then later concat them with -c copy [17:53] <c_14> should work [17:54] <SuperSQL> wow thanks for that [17:58] <mcnesium> c_14: ffprobe says Duration: N/A, start: 0.000000, bitrate: N/A - both firefox and vlc can play the file but neither can jump within the file. its broken, i guess [17:59] <c_14> mcnesium: try remuxing it into another webm [17:59] <c_14> that might recreate the timestamps [17:59] <mcnesium> like ffmpeg -i old.webm new.webm ? [17:59] <c_14> ie with ffmpeg -i webm -c copy -map 0 out.webm [17:59] <mcnesium> ok :) [18:06] <mcnesium> c_14: so my original command included concat:VTS_01_1.VOB\|VTS_01_2.VOB& and so on. will -ss time work with that? [18:06] <c_14> should [18:06] <mcnesium> ok i will try. thank you [18:18] <diffuse> Mavrik: i am making some progress here, but i am getting an error when setting up my filter [18:19] <diffuse> Mavrik: av_get_default_channel_layout(pDecCtx->channels) is returning a NULL [18:20] <Mavrik> what's the value of channels? [18:21] <diffuse> Mavrik: 0 [18:22] <Mavrik> well, thats your problem then [18:22] <Mavrik> did you probe input? [18:22] <Mavrik> or did you set it manually? [18:23] <diffuse> if(avformat_open_input(&fmt_ctx, filename, NULL, NULL) != 0 [18:23] <diffuse> i did that [18:25] <diffuse> Mavrik: ok, i was missing a call to find stream info [18:26] <SuperSQL> I guess that didn't fix :/ Video file is encoded in h264 and in mkv container, it looks like if I put normal -ss infront of input file it would still try to decode whole thing from the beginning. If I put -vcodec copy and output as mkv and then run ffmpeg again for converting first frame to jpg picture is then black, nothing there as for other form [18:26] <SuperSQL> ats it works great [18:27] <diffuse> Mavrik: ok, it works [18:27] <diffuse> Mavrik: i appreciate you pointing me in the right direction [18:27] <diffuse> thanks [21:39] Action: shevy points at a pit and looks at diffuse expectingly. [22:31] <diffuse> Mavrik: i had a chance to run some audio through and the audio produced is way to fast [22:35] <Mavrik> diffuse, then play it slower [22:35] <Mavrik> :P [22:35] <Mavrik> (you're messing something up with playback) [22:47] <shevy> hmm [22:47] <shevy> when trying the filters [22:47] <shevy> ffplay -f lavfi -i life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 [22:47] <shevy> I don't seem to get much at all? only a black video [22:48] <shevy> seems to be the case for all filters :( [22:48] <shevy> but the examples at https://trac.ffmpeg.org/wiki/FancyFilteringExamples looked so fancy! [22:55] <llogan> shevy: works for me [22:55] <shevy> hmm [23:27] <squeegily-OtherC> AWW YES I tried out the XVid encoding and NOW I CAN PLAY ANIME ON MY Asus Eee PC 1000! [23:27] <squeegily-OtherC> ffmpeg is the best [23:28] <squeegily-OtherC> I am unbelievably happy right now\ [23:30] <JEEB> -tune fastdecode libx264 should work as well, and probably be simpler to use / get better compression / possibly better encoding speed [23:31] <nasojlsu> ffmpeg = awesomeness [23:31] <JEEB> (fastdecode + lower bit rate is usually a good combo) [23:31] <llogan> and maybe better battery loife [23:31] <squeegily-OtherC> You guys don't get how old this thing is [23:31] <JEEB> I do [23:31] <JEEB> I had the eeepc 900 [23:32] <squeegily-OtherC> Hmm I'll give that a try then [23:32] <JEEB> I played normal H.264 encodes on it (barely, but I did). The 1000 IIRC has the first-gen atom so it's a wee bit slower, but SD -tune fastdecode encodes should still be fine [23:32] <JEEB> maybe even 720p [23:32] <squeegily-OtherC> As for the battery, I spilled coffee on it and fried it. It's essentially a desktop computer with a lame screen right now [23:32] <squeegily-OtherC> I had no idea -fastdecode existed [23:33] <JEEB> too bad I broke the hinge and decided to give my 900 back to the university after five years [23:33] <JEEB> not -fastdecode, -tune fastdecode [23:33] <JEEB> -tune is a libx264-specific option [23:33] <llogan> i still have a HP Mini. would still use it if i could find the power cord... [23:33] <llogan> hadly see anyone using netbooks these days. i can't stand tablets [23:34] <JEEB> ultrabooks pretty much took over [23:34] <Mavrik> yep [23:34] Action: llogan is now officially old [23:34] <JEEB> actual performance in a nice size [23:34] <JEEB> I don't really dislike tablets but I see the reasons for having small portable laptops [23:34] <Mavrik> and tablets as well - my Nexus 9 is faster and has better screen (+keyboard) than any of those notebooks ever had [23:34] <Mavrik> also those 11" chromebooks are cheap as hell [23:35] <Mavrik> I blame crappy desktop OSes for the reign of tablets tho [23:35] <squeegily-OtherC> Mavrik: I assume you use Mac? [23:35] <Mavrik> ? [23:35] <squeegily-OtherC> You mentioned desktop OSes being not good [23:36] <squeegily-OtherC> Which one do you use? [23:36] <Mavrik> I run pretty much everything lately :) [23:36] <squeegily-OtherC> I recently gave up on maintaining an Arch installation and switched to Xubuntu [23:36] <Mavrik> But, try teaching an elder person how to use a computer and you'll quickly see just how non-sensical alot of desktop OS stuff is [23:37] <JEEB> if you want rolling I'd go for either debian testing/unstable or opensuse tumbleweed [23:37] <Mavrik> like... explaining double-vs-single click :) [23:37] <Mavrik> "But why is this picture special?" [23:37] <nasojlsu> i like opensuse [23:37] <squeegily-OtherC> That is one irritation is I have to add a PPA for every little thing [23:37] <nasojlsu> just works [23:37] <JEEB> I've had opensuse break GNOME for me a few times in a VM [23:37] <JEEB> (tumbleweed) [23:37] <Mavrik> I run ElementaryOS on my Linux boxes, I like the fact they actually made a pretty desktop on an Ubuntu base [23:37] <squeegily-OtherC> Since I went with the LTS release to minimize my troubles, ironically, the PPA thing is a slight inconvenience [23:46] <llogan> i didn't think there was much to maintain in arch. doesn't give me any trouble. [23:46] <llogan> using it on a few servers too [23:46] <squeegily-OtherC> What's the mplayer flag for the framebuffer? -vo directfb bricked the TTY [23:48] <tdr> squeegily-OtherC, mplayer -vo help will list your video out options [00:00] --- Sat Jan 3 2015

1 0

ffmpeg-devel.log.20150101
by burek 02 Jan '15

02 Jan '15

[00:41] <thardin> gott nytt år! [00:58] <kierank> Happy new year [02:24] <michaelni> Happy new year to you and everyone else as well ! [03:15] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:18802bc81ce8: avfilter/vf_idet: Use frame_requested instead of prev [06:17] <cone-386> ffmpeg.git 03Michael Niedermayer 07master:d25d92960034: avfilter/vf_idet: Add 9, 12 and 14 bit pixel formats [12:31] <j-b> Happy New Year! [12:36] Action: compn wonders when the cone hat goes away [12:38] <j-b> compn: today [18:27] <cone-211> ffmpeg.git 03Michael Niedermayer 07master:5c37ffca59e7: avformat/flvdec: add several error messages to error conditions [18:27] <cone-211> ffmpeg.git 03Michael Niedermayer 07master:322f0f5960a7: avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream [18:27] <cone-211> ffmpeg.git 03Michael Niedermayer 07master:eb767a276bfd: avformat/flvdec: Increase string array size [20:19] <cone-211> ffmpeg.git 03Michael Niedermayer 07master:efc4bfc195da: avformat/flvdec: use named identifier instead of literal numbers as return values [20:34] <cone-211> ffmpeg.git 03Marton Balint 07master:8c1f5fb0ae19: ffplay: remove flushed state from decoder context [20:34] <cone-211> ffmpeg.git 03Marton Balint 07master:82928656b134: ffplay: remove unused no_background from videostate [20:34] <cone-211> ffmpeg.git 03Marton Balint 07master:5e7dcb0491cc: ffplay: add support for stream specifiers in -ast, -vst, -sst options [20:34] <cone-211> ffmpeg.git 03Marton Balint 07master:46aa75eea1bd: ffplay: dump format before selecting streams [20:34] <cone-211> ffmpeg.git 03Michael Niedermayer 07master:8009a51f7bbd: Merge remote-tracking branch 'cus/stable' [22:02] <cousin_luigi> Greetings. [22:04] <cousin_luigi> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/flvdec.c#L444 <- is this line supposed to be there? [22:05] <thardin> that doesn't look right [22:06] <compn> no it does not [22:07] <wm4> lol merge error? [22:07] <ubitux> no, regression from a few hours ago (5c37ffca) [22:08] <ubitux> didn't even notice while reading the log [22:08] <wm4> well this caught my eyes http://git.videolan.org/?p=ffmpeg.git;a=blobdiff;f=libavformat/flvdec.c;h=c… [22:08] <cousin_luigi> I'm trying to see if that fix will work on 2.3.5. [22:08] <wm4> returns named constants, but doesn't compare them anywhere [22:09] <cone-211> ffmpeg.git 03Clément BSsch 07master:6a29499b95bb: avformat/flvdec: remove duplicated line error [22:09] <ubitux> cousin_luigi: thx [22:10] <cousin_luigi> wm4: hopefully it will in the future? [22:25] <compn> maybe its a workaround for some braindead compiler [22:30] <ubitux> compn: what are you talking about? [22:30] <compn> what wm4 was talking about, named constants [22:31] <compn> the git hash url he posted [22:31] <compn> ubitux : did you contact that libavfilter patch author about including it upstream ? [22:31] <compn> i guess you found another patch too, how many are we collecting now? :) [22:32] <ubitux> no it's not a compiler workaround [22:32] <ubitux> >that libavfilter patch author [22:32] <ubitux> are you talking about nlmeans? [22:34] <compn> yes nlmeans [22:34] <ubitux> no i'm reading the paper(s) and reimplementing it [22:34] Action: compn falls into reimplemented well of doom [22:34] <compn> ok then [22:44] <aetasx> great game [00:00] --- Fri Jan 2 2015

1 0

ffmpeg.log.20150101
by burek 02 Jan '15

02 Jan '15

[02:04] <BlackBishop> I'm using ffmpeg -ss 00:56:13 -i file -vf "ass=..." ... but the subtitle doesn't seem to start from the right offset .. [02:04] <BlackBishop> the video is ok ... but the subtitle is from the start :/ [02:04] <BlackBishop> any ideas on what I'm doing wrong ? [02:24] <c_14> put the -ss as an output option [02:56] <justinX> The easiest way to quickly remove a sound channel and replace a subtitle, without changing the mp4 in any other way, would be with ffmpeg right? [02:59] <c_14> easiest way I know of [02:59] <c_14> you can probably also use something like mp4box [02:59] <c_14> or l-smash [03:01] <c_14> wait, remove a channel or a stream? [03:03] <c_14> If you want to remove a channel, use ffmpeg. Don't think those other 2 can do that. [03:06] <justinX> a stream is correct name I guess [03:07] <c_14> If it's a stream you can probably do it with ffmpeg, mp4box or l-smash. Whichever you prefer. [03:07] <justinX> the file have two audio language, and I want to remove the first one to give my poor TV less of a problem when playing it. (becuas I want the second language) [03:08] <justinX> and I also want to remove the subtitle and replace it with my own (translated one). [03:08] <c_14> yep, I'd just use ffmpeg with 2 inputs -c copy and some maps [03:10] <justinX> ok... I try see if I find out how to write :-) I'm not sure how I remove an audio channel though. [03:10] <c_14> it's (probably) just a stream, just don't map it [03:12] <justinX> is it possible to get man file for ffmpeg somewhere btw? I guess it will be same thing as "ffmpeg -h full" or something but.... [03:13] <c_14> https://ffmpeg.org/ffmpeg.html [03:13] <c_14> They're all there under documentation [03:13] <c_14> Though you'll probably only need that one. [03:14] <justinX> ok. yeah the web pages is easier to use anyway :-) [03:14] <justinX> if I only would add a subtitle I would have written ffmpeg -i file.mp4 -i file.srt outfile.mp4 right? [03:15] <c_14> add -c copy -map 0 -map 1 [03:15] <c_14> The copy will keep the quality from deteriorating, the maps will make sure it takes all streams [03:15] <c_14> (ffmpeg only copies 1 video, 1 audio, and I think 1 subtitle stream by default) [03:16] <justinX> oh [03:17] <justinX> in this case I would actually want it to only have 1 video,1 audio (the second in the input file),1 subtitle (the srt I provide) [03:17] <c_14> -map 0:v -map 0:a:1 -map 1 [03:18] <c_14> assuming the first file is the video+audio file, the second is the subtitle file and that you want the second audio stream [03:18] <justinX> ok. wonderful :-) [03:19] <justinX> but I must read more about the -map things, I never used them before. tomorow. [04:16] <shevy> guys I need confirmation... -t means duration, so -t 30 means a duration of "+30 seconds"? [04:16] <c_14> yep [04:17] <c_14> duration of "30 seconds" that is. Not sure what that plus is specifying since negative time wouldn't make much sense in this context. [04:17] <shevy> ah, it was meant in the context of setting a position in a file. like I set, let's say, at 153 seconds and then wonder if I would have to use -t 30, or rather -t 183 if you see what I mean [04:18] <c_14> yes, -t 30 [04:18] <shevy> from your answer I deduce it is -t 30, which helps clear my confusion - thanks :) [05:40] <justinX> btw, it looks like I have to specify another subtitle codec to match mp4 (and original encoding) so the command in the end went ffmpeg -i in.mp4 -sub_charenc LATIN1 -i in.srt -c copy -map 0:v -map 0:a:1 -map 1 -scodec mov_text out.mp4 [08:29] <scragglez> Is there a way to take mplayer output as input in ffmpeg? [12:00] <rue_more> how can I list codecs, I been searching thru the help for 15 mins now and cant find it... [15:38] <shevy> hey guys, I am studying https://trac.ffmpeg.org/wiki/FilteringGuide [15:39] <shevy> The first example shows "Resize a 640x480 input to a 320x240 output." [15:39] <shevy> ffmpeg -i input -vf scale=iw/2:-1 output [15:40] <shevy> essentially, the iw/2 just means a reduction towards 50% right? [15:40] <JEEB> input width divided by two [15:41] <JEEB> and the -1 is a shortcut to say "matching value according to the aspect ratio" [15:41] <JEEB> that said, that isn't always enough due to subsampling (usually of chroma) [15:42] <JEEB> the wiki should have an example of making sure the width and height meet the minimum requirements of subsampled planes [15:43] <shevy> yeah, the mention of the pixel confused me for just a moment there [15:44] <shevy> much easier to understand in percentage for me :) [16:13] <shevy> JEEB, do you perhaps happen to know if this here may work? [16:13] <shevy> -vf scale=iw*0.5 [16:13] <shevy> I think it would technically be the same as iw/2 ? [16:13] <shevy> I'll be trying on a video file, ffmpeg seems to accept the syntax [16:14] <JEEB> it will probably work :) [16:14] <JEEB> also, I will still have to note that the comment on subsampling is a worthwhile one and you should make sure that is applied accordingly to your things [16:15] <JEEB> otherwise your command line might work in some cases, and not in other [16:16] <shevy> yeah I mostly just started, slowly working my way through the examples of the filters. Been a little bit difficult to get into ffmpeg, I always forget what -y -i and so forth stand for [16:33] <shevy> hmm [16:33] <shevy> I think it has worked [19:34] <shevy> hmm [19:34] <shevy> is there a way to tell ffmpeg to try and preserve as much quality as possible? [19:35] <shevy> when I did the above -vf scale=iw*0.5, I noticed a big decrease in quality (of an .avi file... which is quite old and had bad pixels anyway, I guess I should try it on a better video file). I'd like to compare it to a variant where the original is preserved as much as possible [19:35] <pzich> try setting a CRF value [19:36] <pzich> if you're going to h264, that is: https://trac.ffmpeg.org/wiki/Encode/H.264 [19:36] <Mavrik> shevy, did you actually set the desired output video quality parameter? :P [19:36] <shevy> Mavrik probably not [19:36] <pzich> yeah, the default quality is generally not the best quality [19:36] <Mavrik> yeah, default settings suck and look terrible [19:37] <shevy> I guess the codec it uses is: "Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4)" [19:37] <Mavrik> eww. [19:37] <Mavrik> please do paste your command line and the ffmpeg output so we can help you get your quality up ;_ [19:38] <shevy> nothing fancy: ffmpeg -i foo.avi -vf scale=iw*0.5:-1 bar.avi [19:38] <Mavrik> ugh. [19:38] <shevy> looks like a blur! hahaha [19:38] <Mavrik> yeah, that will chose hideous defaults :D [19:38] <Mavrik> shevy, where do you want to play your output? [19:38] <shevy> I think I'll try on a better video to start with [19:39] <Mavrik> try rather something in line of [19:39] <shevy> Mavrik actually that is just a test altogether of ffmpeg filters... I am just considering to get this "ready" for one later day when I would try to modify smaller video chunks I think [19:39] <Mavrik> ffmpeg -i foo.avi -vf scale=iw*0.5:-1 -c:v libx264 -c:a copy bar.mp4 [19:39] <shevy> so I guess it will be played just on a normal regular desktop computer [19:39] <Mavrik> yeah, use H.264 then, the baseline profile MPEG4 your command line chooses is like 15 years obsolete :) [19:40] <shevy> hehe [19:40] <Mavrik> also, use -c:a copy to copy audio without reencoding it [19:40] <Mavrik> throw in -crf 24 as a quality setting and tweak that for size/quality :) [19:40] <Mavrik> and you're done. [21:20] <wachpwnski> how do I get the bit rate info for a video stream? Like if I wanted to use the same compression settings from an existing mkv file? [21:21] <BtbN> ffmpeg/ffplay should display the average bitrate when playing the stream. Buf if you want to use the very same settings, why transcode at all? [21:21] <BtbN> There is a lot more that determines the quality than just the bitrate. [21:21] <wachpwnski> no I have a compression setting from another video that I like, I want to use it on another source [21:22] <Mavrik> wachpwnski, ugh... that's usually a terrible idea [21:22] <Mavrik> wachpwnski, but ffprobe is meant for inspection and will output bitrate when able. [21:22] <BtbN> You won't be able to get the exact settings that were used to encode an existing video. [21:23] <wachpwnski> BtbN: Ok, so it's just trial an error then? [21:23] <BtbN> No idea what you are trying to achive. [21:23] <Mavrik> wachpwnski, usually when encoding you set visual quality with CRF if that's what you care about [21:24] <Mavrik> wachpwnski, since bitrate for the same visual quality depends on what is pictured in the video [22:33] <shevy> was there a predecessor to ffmpeg before it got started? [22:40] <klaxa> i don't think so [00:00] --- Fri Jan 2 2015

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