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October 2014

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62 discussions

ffmpeg-devel.log.20141020
by burek 21 Oct '14

21 Oct '14

[03:19] <gudu_khan> hi any one is thr need help [03:19] <gudu_khan> ? [03:35] <reynaldo> tions and no mentor please write to opw(a)ffmpeg.org [03:35] <reynaldo> damn.. too large [04:32] <cone-777> ffmpeg.git 03Michael Niedermayer 07master:d4e62ebae7f4: avcodec/snowenc: remove redundant FF_MPV_OFFSET() [04:32] <cone-777> ffmpeg.git 03Andrey Utkin 07master:9d86ce783bf4: avformat/http: Introduce ff_http_averror() [04:32] <cone-777> ffmpeg.git 03Andrey Utkin 07master:81ce3f8e8017: avformat/http: Use ff_http_averror() [05:46] <cone-777> ffmpeg.git 03Michael Niedermayer 07master:b5583fc3e002: avutil/cast5: Fix off by one error [07:34] <prachi> Hello,I am new and was trying to write a simple demuxer. I am not able to understand what AVFormatContext structure actually is. I've read the documentation and data variables etc in it.Still I am not able to get a solid idea of what it actuall is.I was hoping if someone could help me .Thanks in advance! [10:00] <amalia> Don't have a clue about writing an RMS-ALS decoder. Any ideas ? [10:10] <amalia> stephano Don't have a clue about writing an RMS-ALS decoder. Any ideas ? [10:11] <amalia> Stefano Don't have a clue about writing an RMS-ALS decoder. Any ideas ? [10:25] <amalia> Don't have a clue about writing an RMS-LMS decoder. Any ideas ? [10:34] <ubitux> where did you see this qualification task? [10:34] <ubitux> who is the mentor? [10:34] <ubitux> (hi)( [10:34] <amalia> I was given that by Thilo Borgmann [10:34] <amalia> To write an RMS-LMS decoder [10:34] <amalia> Hi ubitux [10:36] <ubitux> better contact him directly then [10:36] <ubitux> by mail, that is [10:36] <amalia> Hmm [10:36] <amalia> No tips... :( [10:36] <amalia> I am dying here [10:36] <ubitux> i don't know, your question is vague [10:36] <ubitux> did you have a specs? [10:36] <ubitux> do* [10:38] <amalia> Okay Fine ubitux [13:20] <saste> hi, can you update me with the latest instructions about opw? [13:21] <saste> is the candidate requested to send a qualification task *before* Oct 22? [13:24] <michaelni> saste, i know only whats written on the opw pages and they arent 100% clear, reynaldo did ask the admins afew days ago but got no reply or ive been lost in the CC [13:24] <saste> michaelni, also, do I need an account in the OPW admin system? [13:24] <saste> how can I get that? [13:25] <michaelni> for "https://opw.gnome.org/?q=manage_projects&prg=3" there was a way to "login" with your google account [13:26] <michaelni> if you want to mentor or rate your potential applicants or read the applications then you need an acount [13:26] <saste> that is, I need a google account? [13:26] Action: Jul13t asks about this on the #opw channel [13:27] <michaelni> Jul13t, thanks [13:28] <michaelni> saste, i dont know ... a google account was just what i used [13:31] <saste> michaelni, about the qualification tasks page, what's this complexity index? [13:31] <saste> also what's this "Date Given"? [13:32] Action: michaelni looks for reynaldo but doesnt see him [13:33] <saste> michaelni, do i need to apply as a mentor? [13:34] <michaelni> if you want to mentor you need to get an account for opw and click that "willing to mentor" button for the application you are willing to mentor [13:34] <amalia> Which channel is that Jul13t ? [13:36] <michaelni> #opw IRC channel on GIMPNet (irc.gnome.org) [13:38] <thardin> neat [13:38] Action: thardin joined as tomas [14:53] <cone-232> ffmpeg.git 03Kevin Mitchell 07master:ff68ceb1b55b: avfilter/vf_idet: add both multiple and single frame detection metadata [16:37] <cone-232> ffmpeg.git 03Brad Smith 07master:04a4fb81b3d1: v4l2: Make use of the VIDIOC_ENUM_FRAMESIZES ioctl on OpenBSD [18:58] <rhea> hey everyone i jus got to know about the opw programme from a friend is there any bug i could fix right now [19:01] <J_Darnley> You can have a look at trac for an unsorted dump of issues people report. [19:01] <J_Darnley> As for applying to OPW I reckon you are too late. [19:01] <J_Darnley> Your application has to be done by Wednesday. [19:03] <J_Darnley> Though we do still welcome anyone who wished to contribute. [19:03] <rhea> isn't there time till tomorrow if i fix a bug? [19:06] <Guest37454> hello all. there is a ffmpeg bug report i'd really need fixed but nothing happened for 21 months now. what can i do to get this solved? can i somehow do a bounty for getting this fixed? [19:08] <rhea> i am ready to fix a bug [19:22] <Guest37454> https://trac.ffmpeg.org/ticket/2220 [19:32] <cone-232> ffmpeg.git 03Thilo Borgmann 07master:f31445a82d6a: lavf/mov.c: Allocate buffer in case of long metadata entries. [20:14] <cone-232> ffmpeg.git 03Rong Yan 07master:a001db2e2c27: configure: fix bug to ensure that HAVE_VSX is only enabled when HAVE_ALTIVEC is enabled and in LE environment [20:15] <ubitux> ok so, the main blocking point with the text subtitles is now zvbi teletext support :( [20:24] <akira4> umm ubitux, I tried adding the stl test sample but I get nothing in the reference file. [20:27] <llogan> Guest37454: you can create a bounty here: https://www.bountysource.com/issues/1414464-ffmpeg-unable-to-stream-copy-so… [20:30] <akira4> nevermind. I got it :D [20:33] <Guest37454> bounty done, thanks [20:33] <akira4> just out of curiosity,how exactly are the reference files that are generated used ? [20:47] <ubitux> akira4: they're used to compare obviously [20:48] <ubitux> so if the code is broken later, the output will differ from the originally generated file [20:48] <ubitux> and raise an error [20:48] <akira4> compare with? [20:48] <akira4> oh [20:49] <ubitux> don't forget to git add the reference file [20:49] <ubitux> then you can test easily; break your code, then run make fate-sub-stl again [20:49] <ubitux> (without GEN=1) [20:50] <Yolande> Hello, My name is Amate Yolande, I just came across the OutreachProgramForWomen and though its is close to the deadline it would like to know if there is a chance i can start working on a project and submit a proposal before the deadline. [20:50] <akira4> hmm..I see. [20:54] <Compn> Yolande : we have a lot of applicants so far, what kind of skills do you have? anything to do with simd /asm ? :) [20:57] <akira4> ubitux, I've mailed the patch. [20:59] <ubitux> akira4: you're not working on the latest ffmpeg [20:59] <ubitux> you need to rebase, the tests have changed [21:00] <ubitux> we're not using a md5 system anymore for text subtitles [21:00] <Yolande> Compn: No i am not familiar with simd /asm and i have no actual experience with ffmpeg but have done programming/plugin development with gstreamer. [21:00] <ubitux> akira4: anyway, can i se the .stl file? [21:00] <akira4> sure. [21:01] <ubitux> !strncmp(ptr,"//",2) you forgot some spaces here [21:01] <akira4> http://pastebin.com/tBxTGZx4 [21:01] <ubitux> akira4: not bad [21:01] <ubitux> akira4: can you make a line with both " , " as separator? [21:02] <akira4> I'll add that. [21:03] <ubitux> otherwise it looks like a good case [21:03] <ubitux> anyway, git rebase this and re-gen the sample [21:03] <ubitux> you'll have to solve the conflict in subtitles.mak [21:04] <Compn> Yolande : well, to be honest, we have more applicants than we have slots, which means your chances of getting paid in the opw are very slim. but if you want to work on ffmpeg, we might be able to find a mentor and a project to work on :) [21:07] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:8ab3b71e4b34: idcin: fix return check [21:07] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:ab077b3d7a79: Merge commit '8ab3b71e4b343cec3fa369a880ab4496d7345154' [21:10] <michaelni> Compn, we have a single qualified applicant [21:10] <Compn> haha , no one filled out an application yet ? [21:10] <Compn> Yolande : good news.... fill out an application and you're in! :P [21:11] <michaelni> Compn, people did submit applications but only one submitted a qualification task that passed review and was applied [21:11] <Compn> ah [21:19] <akira4> ubitux, is this better? http://pastebin.com/0JRH59N3. [21:19] <ubitux> yeah sure [21:19] <ubitux> btw [21:20] <ubitux> i know i'm demanding [21:20] <ubitux> can you add more style change for later? [21:20] <ubitux> like changing multiple ColorIndex[1234] at once [21:21] <ubitux> and mixing bold and italic, then only disable bold [21:21] <ubitux> or stuff like that [21:22] <akira4> hmm...alright. [21:22] <akira4> oh and since the tests have changed..I'll have to rebase and make the changes again? [21:23] <ubitux> yes you need to rerun the fate test with GEN=1 [21:23] <ubitux> and fix the conflict you're gonna get in subtitles.mak [21:23] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:02484d1a9316: avio: fix sizeof argument [21:23] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:54112629f196: Merge commit '02484d1a9316a9309fc1830825345ab79f6a8d60' [21:23] <akira4> heh. and I thought this time I'll get the patch applied for sure :P [21:23] <ubitux> sorry :) [21:23] <ubitux> you're almost there [21:24] <akira4> yup. thanks :) [21:24] <akira4> oh and about the proposal for OPW [21:24] <akira4> what should I write? [21:24] <ubitux> mmmh [21:24] <ubitux> whatever you want basically [21:25] <ubitux> let me check what i said last time [21:25] <akira4> cool [21:26] <ubitux> akira4: [21:26] <ubitux> 23:32:34 <@ubitux> there are a lot of work on subtitles stuff [21:26] <ubitux> 23:32:52 <@ubitux> the final goal is to get a proper (at least text) subtitles support in ffmpeg [21:26] <ubitux> 23:33:02 <@ubitux> and that includes having them in libavfilter for hardsubbing [21:26] <ubitux> so, just word something along those line [21:27] <akira4> hmm...okay. [21:42] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:d7f530b0d67f: aviobuf: check context before using it [21:42] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:36552345d042: Merge commit 'd7f530b0d67fe3996dbfa990a47c865e0d1400a2' [21:52] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:9fbc613f0df1: wtv: check seek_by_sector return value [21:52] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:c04f125bb42a: Merge commit '9fbc613f0df1628e7e78bca791fa8833846f8210' [21:52] <akira4> ubitux, how does this stl file look? http://pastebin.com/MkHZpJUW [21:53] <ubitux> akira4: looks good :) [21:53] <akira4> phew. [21:53] <akira4> I'll run the fate-tests then [21:53] <ubitux> michaelni: can you upload this file in fate-suite/sub/STL_capability_tester.stl? [21:53] <ubitux> (http://pastebin.com/raw.php?i=MkHZpJUW) [21:57] <michaelni> ubitux, done [21:58] <ubitux> michaelni: thank you [21:59] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:7b48bf95242e: wtv: clean memory on error [21:59] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:6ef4e0c8b899: Merge commit '7b48bf95242ebf95366d24d32e4cfca6aa77c5e2' [22:02] <cone-232> ffmpeg.git 03Lukasz Marek 07master:e493814d6191: lavd: export all symbols with av_ prefix [22:07] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:d1afd3e1d6e4: vf_format: check input validity [22:07] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:2e0ac145d56f: Merge commit 'd1afd3e1d6e43f4d37ae147091f270124ac48e04' [22:19] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:322b571d5591: rtmpproto: remove dead code [22:20] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:20d3348e8a19: Merge commit '322b571d5591978efd6f6fee12181ac8900079fe' [22:34] <cone-232> ffmpeg.git 03Vittorio Giovara 07master:350ed1829268: rtpdec_hevc: drop unnecessary check [22:34] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:e162db66a224: Merge commit '350ed1829268d343b791208c8fd1cddd44d52a8e' [22:48] <akira4> ubitux, I've sent the patch. [22:49] <ubitux> akira4: cool, it looks fine to me so far [22:49] <akira4> that's great :) [22:49] <ubitux> akira4: i'll wait 24 hours to apply (to wait for the fate farm to synchronize for the sample) [22:49] <ubitux> so i'll probably apply tomorrow around the same hour [22:50] <ubitux> after some testing [22:50] <akira4> cool. [22:50] <akira4> thanks [23:03] <cone-232> ffmpeg.git 03Luca Barbato 07master:96bfb6774785: nutdec: Prevent a memory corruption [23:03] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:5753d9677145: Merge commit '96bfb677478514db73d1b63b4213c97ad4269e8f' [23:16] <cone-232> ffmpeg.git 03Luca Barbato 07master:d2771a1dc0a3: rtpproto: Free the addrinfo pointer on failure [23:16] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:cec4e1c7c712: Merge commit 'd2771a1dc0a3695e8873adc16d068077f2417eea' [23:18] <Jul13t> michaelni, Marina says prospective OPW applicants CAN continue working on their qualifications tasks right up to November 12. So I'll like to know if it's valid here? [23:19] <michaelni> Jul13t, yes it is [23:19] <Jul13t> and applications too [23:20] <Jul13t> michaelni, concerning the MPEG lossless encoder project, it's said the qualification task cannot be completed? [23:21] <michaelni> the applications submitted to OPW can AFAIK also be editing by applicants after oct22 yes [23:22] <Jul13t> so any prospective applicant still has a chance since it's possible to keep working on the qualification task before Nov 12 right? [23:24] <michaelni> before october 31, yes, before nov 12 maybe [23:25] <Jul13t> thanks michaelni was getting discouraged already :( [23:40] <michaelni> note though that we have only until the 31oct for applicants who would need a slot from general opw funding, as opw needs to know to which applicants this applies by oct 31 [23:48] <Jul13t> thanks michaelni that'll do [00:00] --- Tue Oct 21 2014

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ffmpeg.log.20141020
by burek 21 Oct '14

21 Oct '14

[00:54] <sim590> I have this file that is not recognized by my video editing software (blender). mpv can however read it fine. It outputs this information about the file : http://ix.io/eQ6. Should I put the data in a container like mkv ? [00:55] <sacarasc> What does blender support? [00:56] <sacarasc> (Also, isn't blender a 3D image renderer, not a video editor?) [00:58] <sim590> sacarasc: it can do video editing. [00:58] <sim590> I don't really know what it supports. I'm looking for the info [00:58] <sim590> I've found this page : http://wiki.blender.org/index.php/Doc:2.6/Manual/Render/Output/Video [01:00] <sacarasc> That's output, not input. [01:04] <sim590> I got this http://wiki.blender.org/index.php/Doc:2.6/Manual/Data_System/Files/Formats [01:06] <sacarasc> Try "-f h264 blah.h264" as the output. [01:11] <sim590> is h264 a container ? [01:12] <sacarasc> I think it puts it without a container, but not 100% sure. It didn't say what containers it supports other than AVI, and AVI sucks. [01:18] <sim590> sacarasc: -f h264 doesn't carry audio, does it ? [01:18] <sacarasc> No, but it didn't say it supported audio there. I am working on the info I have. [01:19] <sim590> yes. I understand. I'm sorry, I don't have lots of information. I'm a bit of a beginner. [01:21] <sim590> knowing that kdenlive does open the files, do you know if it could let me edit the video in the way that I could toggle the display between fullscreen video and having the video in the bottom left corner and having the slides of a lecture in the background ? [01:21] <sim590> if not, what other software would you recommend ? [03:18] <Gudu> hi all [03:19] <Gudu> any one is the [03:19] <Gudu> *thr [03:19] <Gudu> need help [13:06] <ruby_on_tails> i had encoded a video using mencoder from .ogv to .mp4 my video doesnt have voice so i never cared about audio codecs, now its creating a problem with online players. http://pasteboard.co/14Xc4ywO.png in this image the right one is from mencoder H264 at the end, it doesnt work with many players, the left one was encoded using ffmpeg and works fine, now i want to make the right one similar to the left one to make it work, can anyone help me with it? [13:07] <ruby_on_tails> i dont have the source file of the right one so i cant use ffmpeg in a fresh way now, have to use the mencoder output as the input for ffmpeg [13:08] <relaxed> why do you have to use mencoder? [13:08] <ruby_on_tails> i dont have to, i had used it earlier when i had some issues with ffmpeg [13:08] <ruby_on_tails> the output that mencoder gave doesnt work with some online players now, but what ffmpeg does works [13:09] <ruby_on_tails> so i pasted the information of both the videos in that image to tell you the specs about both, now i want to convert the flawed video into a working one [13:09] <relaxed> Oh, you mean using ffmpeg to convert it... [13:10] <ruby_on_tails> yes [13:10] <ruby_on_tails> i dont want to reencode the video [13:10] <ruby_on_tails> i just want the audio codec bug to be fixed [13:10] <ruby_on_tails> is that possible? [13:11] <ruby_on_tails> that audio codec issue fails the video from playing on mac quicktime, online video players, etc [13:11] <ruby_on_tails> only VLC can play that video [13:12] <relaxed> pastebin the output of "ffmpeg -i input" on the one that plays and the one that doesn't. [13:14] <ruby_on_tails> relaxed: plays: http://pastebin.com/aq8ywWcn doesnt play: http://pastebin.com/sTbvqr2P [13:16] <relaxed> the one that doesn't play isn't actually an mp4. "Input #0, avi, from '/var/www/df2.mp4'" [13:17] <ruby_on_tails> relaxed: yea i guess so, how do i fix it? [13:17] <relaxed> it's an avi with an mp4 extension. run ffmpeg -i df2.mp4 -c copy df2.fixed.mp4 [13:19] <ruby_on_tails> relaxed: it says Unrecognized option 'c' Failed to set value 'copy' for option 'c' [13:19] <ruby_on_tails> ffmpeg -i /var/www/df2.mp4 -c copy /var/www/df2.fixed.mp4 [13:19] <relaxed> ffmpeg -i df2.mp4 -vcodec copy df2.fixed.mp4 [13:20] <ruby_on_tails> a lot of error type messages are shown: [mp4 @ 0x239ddc0] pts has no value [NULL @ 0x239ce20] missing picture in access unit [13:20] <ruby_on_tails> but it finished [13:20] <ruby_on_tails> hmm ok this will help [13:20] <ruby_on_tails> looks like it worked [13:22] <ruby_on_tails> but what do these errors mean? [13:23] <relaxed> ffmpeg stops on errors, those are just informational [13:24] <ruby_on_tails> hmm [13:24] <ruby_on_tails> another questions [13:24] <ruby_on_tails> relaxed: i am making videos from screencasts, why isnt the video of same quality as that of what i see on the monitor? [13:24] <ruby_on_tails> it degrades, specially text and colors [13:25] <relaxed> poor color conversion [13:25] <ruby_on_tails> how can i fix it? [13:27] <relaxed> Did you see https://trac.ffmpeg.org/wiki/Capture/Desktop ? [13:28] <relaxed> eh, it doesn't give any hints there. [13:28] <ruby_on_tails> relaxed: i used recordmydesktop at the highest quality which uses ffmpeg i guess [13:30] <relaxed> see if using ffmpeg directly makes any difference. [13:30] <ruby_on_tails> k let me try [13:34] <ruby_on_tails> relaxed: i am using this [13:34] <ruby_on_tails> ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -r 25 -i :0.0 -sameq -f mpeg -ar 48000 -s wvga -y sample.avi [13:34] <ruby_on_tails> quality is horrible [13:34] <ruby_on_tails> got it from a tutorial online, coz the one on the link you gave throws some deprecated errors [13:36] <relaxed> ignore that tutorial [13:37] <ruby_on_tails> then what command do i use to record? [13:37] <relaxed> you could follow the guide I just pasted [13:38] <ruby_on_tails> relaxed: http://pastebin.com/78nr1UTR this is what happens [13:45] <ruby_on_tails> tried 2 commands from the url you gave, both give the same errors [13:47] <relaxed> what is your dedktop resolution? [13:47] <relaxed> desktop* [13:47] <ruby_on_tails> 1360*768 [13:48] <relaxed> ffmpeg -video_size 1360x768 -framerate 25 -f x11grab -i :0.0 output.mp4 [13:51] <ruby_on_tails> relaxed: that worked, but the quality is horrible [14:12] <relaxed> add -crf 16 after the input [14:45] <ruby_on_tails> relaxed: still bad quality [15:04] <edoardo> hello! ffpeg can't get my easycap [15:04] <edoardo> ffmpeg -f v4l2 -i /dev/video0 -vcodec libx264 -preset ultrafast -b:v 2000k -g 100 -acodec libvo_aacenc -b:a 1024k test.mkv [15:04] <edoardo> anyone know why? [15:05] <edoardo> i can see it in vlc, though [15:05] <rsevero> Hi, I have several webm streams produced by ffmpeg and distributed by ffserver. I can see them with vlc and ffplay but I can't see them with Firefox or Chrome. Both browsers show me the controls bar but I never see the video. I see no errors on any log. Where should I look for clues on whats wrong? How can I further troubleshoot this issue? [15:05] <edoardo> [video4linux2,v4l2 @ 0x7f5f140008c0] Cannot find a proper format for codec 'none' (id 0), pixel format 'none' (id -1) [15:05] <edoardo> Assertion *codec_id != AV_CODEC_ID_NONE failed at libavdevice/v4l2.c:805 [15:36] <BtbN> rsevero, browsers can't stream. They only download whole files and play them. [16:31] <rsevero> BtbN, thanks for your info. So i I want to see streams on the browers I would need some kind of flash or java applet? [16:31] <rsevero> So if I want... [16:34] <brontosaurusrex> rsevero: dunno, but pseudostreming works just fine http://1stdev.com/tremendum-transcoder/articles/seeking-videos-beyond-the-b… [17:21] <casper366> hi, i'd like to decode an .oma (atrac3plus) to mp3. I've got the atrac3-codec but not the atrac3plus-codec. Is there a way to implement this codec? [18:31] <casper366> 250 ppl here and noone can help [19:02] <storrgie> I've got some .cine footage (super high framerate stuff) that I want to convert to mp4. I'm getting very small files that just play as black for a couple of seconds. Has anyone worked with this type of file before? [21:07] <ChocolateArmpits> test test [21:16] <edoardo> ChocolateArmpits: yes sir [21:16] <edoardo> i hear you [21:16] <ChocolateArmpits> Has KVIRC log out of freenode another day [21:16] <ChocolateArmpits> Had* [21:16] <edoardo> meh [21:16] <edoardo> i don't like these permanent clients [21:17] <edoardo> you get like 200+ channels with all idles [22:11] <PovAddict> ChocolateArmpits: what is a DVD menu image supposed to have? [22:11] <PovAddict> I thought the main image was only used for dvdauthor to autodetect where the buttons are [22:11] <PovAddict> so I made big ugly red rectangles where the buttons are [22:12] <PovAddict> works fine in VLC, but in my set-top DVD player, the rectangles show up on top of the menu video :( [22:12] <ChocolateArmpits> uh dvd author doesn't detect button placement based on image [22:13] <ChocolateArmpits> There's another xml file that defines it and links according images [22:14] <ChocolateArmpits> It's hard for me to answer concretely, work files are at another computer [22:15] <PovAddict> it supports autodetection, with autooutline="infer" in the <spu> element [22:15] <PovAddict> spumux output says [22:15] <PovAddict> INFO: Autodetect 0 = 82x30-313x212 [22:15] <PovAddict> INFO: Autodetect 1 = 408x30-639x212 [22:15] <ChocolateArmpits> Oh I then haven't used that [22:15] <edoardo> why isn't ffmpeg default in opensuse? O_o [22:15] <sacarasc> Ask them. [22:15] <ChocolateArmpits> I don't think I ever saw that feature, PovAddict: Is that something new ? [22:15] <PovAddict> maybe I should switch to setting the button coords in the xml [22:15] <PovAddict> I don't know how new [22:15] <edoardo> you need to add an external ppa [22:16] <PovAddict> opensuse has ppas? [22:16] <edoardo> home:nickname [22:17] <ChocolateArmpits> Anyways, if you write out the file rather than let the program decide it will all work. To some degree of course, depending on the program used. VLC for example is horrible in dvd menu recreation [22:18] <PovAddict> what software do you recommend to test DVDs? [22:18] <ChocolateArmpits> Are you on Linux ? [22:18] <PovAddict> yes [22:18] <edoardo> mplayer? [22:19] <ChocolateArmpits> Damn, I actually used WMP :) [22:20] <ChocolateArmpits> But did some compensation targetting VLC playback so it looked good even there [22:20] <PovAddict> my problem is stuff looking good in VLC so I think it's fine, then I burn the disc and it looks like shit in a real DVD player :( [22:20] <edoardo> i know that feel bro [22:21] <ChocolateArmpits> I used simple outline buttons so had no problems [22:21] <ChocolateArmpits> or rather, outline graphics [22:21] <PovAddict> I'll find out the hard way how many times a DVD-RW can be rewritten :P [22:22] <ChocolateArmpits> As far as I remember MPC-HC had truthful playback [22:22] <ChocolateArmpits> At least for menus [22:23] <edoardo> never remove yast2 from opensuse [22:23] <edoardo> never [22:51] <zway> Can someone help me, I'm trying to decode uLaw, resample, and then encode in AAC, but I'm getting about 2 seconds of junk audio added to the file after resampling. The 2 seconds sounds like they're spread out randomly in the audio as noise. Here's my code: http://pastebin.com/um0sLrJu [22:52] <zway> the input file is 48kHz Mono uLaw, and I'm trying to output 16kHz 16bit AAC [22:53] <ChocolateArmpits> zway, did you try sox? [22:54] <zway> I haven't tried writing a sox wrapper no...ffmpeg command line works fine, but I'm trying to write a C++ wrapper that will do this so it can be called from a windows service without using PInvoke [00:00] --- Tue Oct 21 2014

1 0

ffmpeg-devel.log.20141019
by burek 20 Oct '14

20 Oct '14

[00:50] <kierank> ffmpeg is on tv [00:51] <amalia_> kierank Which TV is that ? [00:52] <kierank> bbc four, "the code" [01:02] <amalia_> kierank Thanks for your tips on the BBC I have just downloaded the ffmpeg code from github. How do I compile it ? [01:06] <amalia_> Compn J_Darnley How do I compile the ffmpeg code on my machine ? [01:06] <kierank> https://trac.ffmpeg.org/wiki/CompilationGuide/Generic [01:06] <amalia_> Thanks kierank [01:09] <pross> kierank: that is an australian show \o/ [01:09] <kierank> yup [01:09] <kierank> https://fbcdn-sphotos-d-a.akamaihd.net/hphotos-ak-xpf1/v/t1.0-9/1794677_585… [01:10] <kierank> and there are some historgrams [01:10] <kierank> which could be the waveform filter [01:13] <iive> kierank: ffplay when playing audio only shows histograms [01:15] <J_Darnley> kierank: neat [01:16] <J_Darnley> I wonder what decoder(s) are disabled in that configure line. [04:18] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:993a2487de7d: avcodec/nellymoserenc: use av_freep(), do not leave stale pointers in memory [04:18] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:c5e295cb9d57: avcodec/mpegaudiodec_template: use av_freep(), do not leave stale pointers in memory [04:18] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:765abb3b76a1: avcodec/libvpxenc: use av_freep(), do not leave stale pointers in memory [04:28] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:e23af4662dc6: avcodec/utils: Display pixel format information as none ... if unknown instead of omitting it [05:05] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:5c8eb16769e5: avcodec: add codec_whitelist [05:05] <cone-741> ffmpeg.git 03Michael Niedermayer 07master:057ea2a9823a: avformat: Add format_whitelist [05:16] <cone-741> ffmpeg.git 03Simon Thelen 07master:69071e8e9c04: doc/filters: fix localtime drawtext example. [05:51] <jamrial> wait so, the symmetric-key block ciphers OPW project had three objectives, implementing CAST-128, Twofish and Camellia. This means the student went and did a third of the project as the qualification task? [05:55] <wm4> jamrial: I guess the project objectives could be raised on demand [06:02] <jamrial> maybe aes-ni as kierank requested on trac. though the project requisites never stated asm skills, so probably not. [15:08] <amalia> I am new to github - Created a folder https://github.com/Ngassa/myFFmpeg for a copy of the ffmpeg code but it's empty. What do I do ? [15:09] <amalia> I am an OPW student new to github [15:09] <wm4> amalia: you don't need to use github, but if you want to, then just clone the official ffmpeg repo on github [15:10] <amalia> Okay. Just what command do I use ? [15:11] <wm4> every repo has a "fork" button [15:17] <Compn> wm4 : she clicked that , presented with 3 options, 'quick setup' 'create new repo on command line' 'push code from computer' [15:17] <Compn> i guessed 'create new repo on the command line' , i havent done github before :P [15:58] <cone-35> ffmpeg.git 03Michael Niedermayer 07master:a71a5c911eb0: avformat/riff: Add UGY2 [15:58] <cone-35> ffmpeg.git 03Kevin Mitchell 07master:9d51bad6255e: avfilter/vf_idet: add counts to frame metadata [17:35] <Compn> michaelni : seems rtmpdump-web is not updating http://rtmpdump.mplayerhq.hu [17:36] <kierank> jamrial: yes but she seems quite competent [17:54] <michaelni> Compn, what exactly is where not updating ? what file or content ? [18:19] <cone-35> ffmpeg.git 03Nicholas Robbins 07master:22cfa1f759ac: libavfi: added option to vf_perspective to specify transformation by giving destinations of corners of source. [18:25] <Compn> michaelni : erm, updating the index.htm is not being updated on the website [18:26] <Compn> index.html [18:26] <Compn> changed the mplayer link from a git repo to mplayerhq.hu (bottom of page) [18:26] <Compn> and it hasnt updated [18:33] <Compn> michaelni : nevermind. looks like its working somehow [18:33] <Compn> i must have done something wrong before , sorry for the noise [18:39] <wm4> so why is libutvideo being kept around [18:40] <wm4> didn't someone say the lib is unmaintained and abandoned [18:43] <Compn> wm4 : michael offered to maintain it [18:44] <Compn> or do you mean the libutvideo itself and not our wrapper ? [18:44] <wm4> itself [18:44] <Compn> i was unclear which part was unmaintained, good question [18:44] <JEEB> it seems like someone is still kind of poking it from the release tarballs of Ut Video and kind of keeping it compiling [18:44] <JEEB> it lacks the asm and the new features of Ut Video (10bit etc) [18:45] <JEEB> and I think no-one is making even sure that its output matches the VFW one [19:05] <cone-35> ffmpeg.git 03Kacper MichajBow 07master:f22cf88fd317: rtmpproto: Don't mistake app for playpath. [20:17] <cone-35> ffmpeg.git 03Michael Niedermayer 07master:f63d38138716: ffprobe: Add support for color_trc/color_pri/chroma_loc [20:39] <amalia> In which directory is the ALS decoder code found ? [20:40] <JEEB> it's not a container but an actual a/v/s/whatever format so it is found under libavcodec [20:40] <JEEB> containers and otherwise things to parse to get to actual a/v/s/whatever streams = libavformat (lavf) [20:41] <JEEB> actual a/v/s/whatever formats/streams and their decoding/encoding = libavcodec (lavc) [20:45] <amalia> What about the RLS-LMS encoder ? [20:46] <JEEB> unless it's a container/format to be parsed or muxed (often called an 'encoder' in lavc/lavf speak, even though it's not really encoding), it's in lavc [20:46] <JEEB> also git grep is your friend [21:02] <wm4> amalia: all decoders and encoders are in the libavcodec sub-directory, but yeah, grepping helps [21:02] <wm4> it will often happen that you have to track down something by reading code, grepping the sources, etc. [21:02] <amalia> Thanks very very much wm4 [21:03] <JEEB> wm4, saying that can get kind of hard because lavf loves calling muxers "encoders" :P [21:04] <amalia> There are quite a number of files in that library [21:04] <JEEB> yes, which is why you use tools like git grep to quickly find where exactly that thing is, or you just search within the file names something similar :P [21:05] <wm4> true, libavformat has "enc" and "dec" suffixes on muxer and demuxer implementation files [21:15] <Compn> would you rather mu and dem ? :P [21:15] <amalia> ? [21:15] <JEEB> _mux _demux [21:16] <Compn> amalia : just JEEB and i discussing the color of the bikeshed [21:16] <JEEB> yeh [21:16] <Compn> nothing important [21:16] <wm4> libavdevice is the most dumb here [21:16] <wm4> so reading from an alsa device is done by alsa-audio-dec.c [21:17] <amalia> JEEb So an examplery git grep command would be ? [21:18] Last message repeated 1 time(s). [21:18] <JEEB> git grep -i "RMS" [21:18] <amalia> That's much better [21:18] <JEEB> related documentation: http://git-scm.com/docs/git-grep [21:18] <amalia> Thank you [22:07] <ubitux> ok, now i need an avi sample, with ssa in it, but with multiple Dialogue at the same time [22:08] <nevcairiel> didnt my sample do that on the credits stream? [22:10] <ubitux> nevcairiel: ah? let me check... [22:16] <ubitux> nevcairiel: your sample only has 5 events [22:16] <ubitux> but actually, it seems to use the ass demuxer itself so& [22:16] <ubitux> it won't be a problem actually [22:17] <ubitux> ah there are other streams indeed [22:17] <nevcairiel> the credits stream only has like one bunch of events right at the start, the other ones have normal subtitle dialog [22:18] <ubitux> anyway, all of this is really evil [22:19] <ubitux> i'm going to drop the ssa packets support since it won't be necessary in this case actually [22:19] <ubitux> and since nothing in ffmpeg outputs these& [22:19] <ubitux> or maybe nut [22:20] <ubitux> but can anyone find me a sample for that? :P [22:34] <cone-35> ffmpeg.git 03Andrey Utkin 07master:65639e6bef80: avutil/error: Introduce new error codes for 4XX and 5XX replies from remote servers [23:07] <ubitux> lol @ canon storing jpeg thumbnail, into the jpeg itself [23:07] <ubitux> (in some custom exif tag) [23:09] <wm4> why is that surprising [23:10] <wm4> it only gets dumb if the thumbnail has exif and thumbnail too [23:10] <J_Darnley> Does ffmpeg read that instead of the actual image? [23:10] <ubitux> no [23:11] <J_Darnley> Oh I'm glad that ffmpeg isn't that silly yet. [23:11] <ubitux> and not the jpeg thumb doesn't have exif with a thumb [23:11] <ubitux> no* [23:26] <ubitux> i really won't be easy to test if i'm not breaking the ssa decoder, so does anyone object if i drop it? [23:26] <ubitux> (i guess no one care but...) [23:29] <cone-35> ffmpeg.git 03Clément BSsch 07master:3d258696057d: avcodec/ass: assume raw=0 in ff_ass_add_rect_bprint [23:29] <cone-35> ffmpeg.git 03Clément BSsch 07master:c669af0f248e: avcodec/ass: add ASS trailing \r\n in one single place [00:00] --- Mon Oct 20 2014

1 0

ffmpeg.log.20141019
by burek 20 Oct '14

20 Oct '14

[01:52] <luc4> Hello! I re-encoded a mpeg2 video to h265 but what I get is a larger video. Why can this happen? [01:52] <c_14> Magic. [01:52] <c_14> Otherwise known as your bitrate is too high. [01:53] <luc4> c_14: checked that. Bitrate was 10Mbit, now it is 6Mbit. [01:53] <luc4> Should I set it to a lower value? [01:53] <c_14> Audio tracks? [01:53] <c_14> What's your command? [01:54] <luc4> Yes, it was mp2. I transcoded to ac3. [01:54] <PovAddict> same bitrate? [01:56] <luc4> yes [01:56] <luc4> I mean, the audio track has the same bitrate as the original one. [01:56] <luc4> ffmpeg -i HPIM0758.MPG -vcodec libx265 -preset veryslow -x265-params crf=18 -acodec ac3 HPIM0758.mkv [01:57] <luc4> This is the command I used. [02:01] <c_14> Maybe the bitrate calculation isn't working, try encoding vbv ? [02:02] <luc4> c_14: vbv? [02:02] <c_14> variable-bitrate video [02:02] <c_14> ie -b:v [02:06] <luc4> c_14: you mean I should add -vbr to something like 3? [02:07] <c_14> Nah, just try switching from cbr to setting a target bitrate. [02:07] <c_14> Bitrate calculations have always been rather fishy. [02:07] <c_14> If you pick a bitrate manually, you can ensure that it'll be smaller. [02:07] <luc4> c_14: ah, so you mean the resulting file should be cbr correct? [02:08] <PovAddict> huh, was he using cbr? [02:08] <c_14> eh s/cbr/crf [02:08] <c_14> For my earlier one. [02:09] <PovAddict> is there anything like crf for mpeg2 video? [02:09] <c_14> luc4: that is what I'd probably try, yes [02:09] <luc4> c_14: trying that already. Ill see the result. Thanks then! [02:11] <luc4> What bitrate should I use. From what I read half or two third of the origin may be appropriate. Correct? [02:12] <PovAddict> you want the same quality as the original? [02:12] <luc4> If possible yes, more or less. [02:12] <PovAddict> or do you have a target file size? [02:12] <luc4> No target file size. [02:14] <luc4> The idea is just that transcoding those old files to h265 I may gain a little in size for the same quality. [02:14] <c_14> I'd probably start around 2/3 to 1/2, play around with it. [02:14] <luc4> c_14: ok, thanks! [02:20] <PovAddict> I transcoded a pile of videos to h264 and deleted the originals already :( [02:20] <PovAddict> well, the real-originals are 8mm video tape, but I'm *not* going back to those again :) [02:51] <PovAddict> man, making a dvd menu is such a pain in the ass [07:17] <evilwombat> Hello. I own a popular consumer video camera which produces MP4 files that are otherwise valid, but cause mplayer (well, actually, libavformat.so) to spew warnings of "Non-increasing DTS in stream blah-blahlbhah" for several minutes before playback actually begins. [07:18] <evilwombat> The warnings themselves are not that big a deal, but the associated delay in playback is rather annoying. Attempting to play a 5-minute clip results in a delay of several minutes while these warnings are spewed. Piping them to /dev/null doesn't help with the delay, either. [07:19] <evilwombat> What can I do here? Should I report a bug someplace, along with sample video? Or is there a flag that can suppress this check for monotonically-increasing time values? [11:39] <luc4> Hello! I have a few mpeg video files, which contain mpeg2 video and mp2 audio streams. I thought I may get better compression by using h265 but after transcoding I get a file size which is larger or equal to the original. Original bitrate is 10Mbit/s while h265 bitrate is 6Mbit/s. Any idea why compression is not higher? [11:41] <Mavrik> well, that's because you set the desired bitrate to be 6Mbit/s. [11:46] <luc4> Mavrik: should I set it lower? [11:46] <luc4> Mavrik: to get a more or less identical quality I mean. [11:50] <edoardo> hi! [11:51] <edoardo> i'm trying to stream a concert in a theatre to a near bar [11:51] <edoardo> i'm using ffmpeg as a client to connect to a mplayer istance listening for connections and it works allright [11:51] <edoardo> however, the quality is awful and uses a lot of bandwidth [11:51] <edoardo> ffmpeg -i web.mp4 -f mpegts tcp://192.168.1.23:2000 [11:52] <edoardo> is it possible to use h264 or libvpx to lower the bitrate? in 1080p the stream is unwatchable and it uses 4mbit of bandwidth [12:06] <ChocolateArmpits> edoardo: h264 will be a better choice in this case. You can try changing the profile to High, increase gop, setting the preset setting to "slower". For relatively static content at 1080p I would expect 2mbps to be a good starting point. If however there is a lot of motion, scene change happening more bandwith will be required anyways [12:07] <edoardo> ChocolateArmpits: but how do i stream in h264? [12:07] <edoardo> -f h264 doesn't work [12:07] <edoardo> neither libx264 [12:08] <ChocolateArmpits> "-f" defines wrapper format [12:08] <ChocolateArmpits> h264 is a codec format [12:08] <ChocolateArmpits> x264 is an encoder [12:08] <edoardo> oh! [12:08] <edoardo> so i need -f:v? [12:08] <ChocolateArmpits> to define the encoder you use "-vcodec" for video codec [12:08] <edoardo> or -v:f, can't remember [12:08] <ChocolateArmpits> "-acodec" for audio codec [12:09] <edoardo> perfect, what about libx264 vs libvpx? what's better? [12:09] <ChocolateArmpits> If you want to check available codecs and wrapper formats use "ffmpeg -encoders" and "ffmpeg -formats" respectively [12:09] <ChocolateArmpits> x264 by a huge margin, unless you're using a less system intensive baseline profile [12:10] <edoardo> i'll use x264 then [12:10] <edoardo> aac for audio? [12:10] <edoardo> i always use that for uploading to youtube, but i don't know if it's suitable for a stream [12:10] <ChocolateArmpits> Again, AAC is a codec standard, to encode in that standard you have multiple encoders with different names [12:12] <ChocolateArmpits> For example Zeranoe build of Ffmpeg contains two aac encoders. "AAC" and "Libvo_aacenc" [12:12] <ChocolateArmpits> If I believe right "AAC" requires experimental tags, so "Libvo_aacenc" will probably be more suited for this [12:14] <ChocolateArmpits> Your command line should probably look like this "ffmpeg -i $input -vcodec libx264 -profile:v high -preset slower -b:v 2500k -g 100 -acodec libvo_aacenc -b:a 384k -f mpegts $output" [12:18] <edoardo> ChocolateArmpits: why -preset slower? [12:19] <edoardo> nevermind, it works perfectly [12:19] <ChocolateArmpits> edoardo: http://mewiki.project357.com/wiki/X264_Settings#preset [12:21] <ChocolateArmpits> Compared to medium, the default one, compression efficiency should increase by about 15% [12:21] <edoardo> perfect [12:21] <edoardo> if the pc handles it, i'll use that one [12:21] <ChocolateArmpits> However if your system can't handle it try changing it to a faster one [12:21] <edoardo> what's -g 100? [12:21] <ChocolateArmpits> GoP size [12:21] <ChocolateArmpits> Group of Pictures [12:22] <ChocolateArmpits> http://en.wikipedia.org/wiki/Group_of_pictures [12:22] <edoardo> oh, the full image thing [12:22] <edoardo> each 100 images one if full, the other are based on the full one [12:22] <ChocolateArmpits> Sorta [12:22] <ChocolateArmpits> There are different frame types so they act differently [12:23] <ChocolateArmpits> Usually you should set the gop to 2 seconds when streaming, that is your framerate multiplied by 2 [12:24] <ChocolateArmpits> If your framerate is at 25 fps then setting gop to 50 would be appropriate from streaming standpoint [12:24] <edoardo> 30fps here [12:24] <ChocolateArmpits> However if you want to conserve bandwith increasing it past that is a good solution but only if you have a stable connection at both sides [12:24] <edoardo> the connection is like this: pc -> router -> pc [12:25] <edoardo> so it should be fairly stable [12:27] <ChocolateArmpits> edoardo: any more questions? [12:27] <edoardo> no :) [12:27] <edoardo> streaming right now a 1080p video and the computer that's struggling is the receiver [12:28] <edoardo> -f scale=1024:-1 solved everything :) [12:28] <edoardo> also, with a bitrate of 2m the image quality is very high with no visual artifacts whatsoever [12:30] <ChocolateArmpits> "-f scale "? ?? [12:30] <ChocolateArmpits> Are you sure it's not -vf scale / [12:30] <ChocolateArmpits> ? [12:30] <edoardo> yes, my bad [12:30] <edoardo> it was a typo [12:31] <edoardo> so the final command is [12:31] <edoardo> ffmpeg -re -i test.mp4 -vcodec libx264 -profile:v high -preset veryfast -b:v 2000k -g 100 -acodec libvo_aacenc -vf scale=1280:-1 -b:a 128k -f mpegts tcp://192.168.1.23:2000 [12:31] <edoardo> i get a delay of 6 secs, so that's fairly good [12:32] <ChocolateArmpits> If it's a concert I would increase audio bitrate [12:32] <edoardo> isn't aac very efficient? [12:32] <edoardo> also, is the -b:a total or per channel? [12:34] <ChocolateArmpits> Total, there's no such thing as "per channel", "per track" maybe [12:34] <edoardo> isn't a stereo stream a two channel stream? [12:35] <ChocolateArmpits> A track is composed of either two channels or a one mono channel [12:35] <ChocolateArmpits> In terms of formats it's one essence stream [12:36] <ChocolateArmpits> If you have multiple stereo tracks you have multiple audio essence streams [12:36] <ChocolateArmpits> Depending on the format each essence stream can have different settings, length, channel count, bitrate or even encoding [12:37] <edoardo> so, you can have multiple "tracks" each composed of one or more channels? [12:37] <edoardo> and these tracks can have different codecs, bitrates.... [12:37] <ChocolateArmpits> Or more depending on channel system, say 4.1 can have 5 channels in one track [12:38] <edoardo> cool [12:38] <ChocolateArmpits> Depending on the format the same 4.1 system can be separated into 5 mono tracks [12:39] <ChocolateArmpits> It's can get pretty messy and I'm not really that familiar with this yet [12:39] <ChocolateArmpits> it* [12:39] <ChocolateArmpits> As for the audio bitrate I would increase it more [12:39] <edoardo> yes, i'm testing right now and i can hear some losses [12:39] <ChocolateArmpits> Compared to how much you're using for video alone, trying to conserve on audio is not the best idea [12:39] <edoardo> i'm trying to transmit with opus [12:40] <edoardo> ...and it doesn't work :) [12:40] <edoardo> i'll stick to aac then [12:41] <ChocolateArmpits> There is a patch to libav for opus support in mpegts. It should probably get into ffmpeg sometime in the future [12:42] <ChocolateArmpits> Mpegts standard was also updated for opus support https://twitter.com/obencoder/status/482548498467155968 [12:42] <edoardo> yeah! [12:43] <edoardo> i would really like to use opus to stream, i can get amazing results with only 128k of bitrate [12:54] <luc4> Hello! Im transcoding to h265 and I see this output: x265 [info]: Main profile, Level-3 (Main tier). Can I use a higher level to improve compression? [12:54] <edoardo> luc4: -profile:v high maybe? [12:57] <luc4> edoardo: will that provide better compression? [12:58] <edoardo> luc4: yes [12:58] <edoardo> brb/afk [12:59] <luc4> edoardo: Ill try that, thanks. [13:13] <JEEB> edoardo, there is no high profile in HEVC [13:16] <JEEB> (also setting the profile higher than the encoder sets in libx264 does not lead to better compression either) [13:16] <ChocolateArmpits> Does Ffmpeg use any type of dithering when changing pixel format (for example from 16bpp to 8bpp) ? [13:20] <luc4> JEEB: I thought so. But I dont clearly understand how it is possible that transcoding mpeg2 to h265 results in a larger file& Probably because of bitrate, but I set 6000k from 10000k and yet the file is not smaller... [13:24] <JEEB> well, multiple possibilities there: 1) you miscalculated the overall bit rate of the original file 2) libx265 fails with ABR encoding (it's a very volatile project) [13:24] <JEEB> personally I would not use any HEVC encoder yet for real use cases, except if you are aiming for very, very low bit rate [13:27] <JEEB> because that's the only use case where in my testing libx265 can "beat" libx264. In which case the picture from both will suck, but HEVC sucks a bit less. [13:28] <JEEB> of course, this is with preset placebo and all those bells and jingles [13:28] <JEEB> so not exactly quick [13:29] <JEEB> as soon as you start wanting good compression on a nice picture quality level, libx265 is still not good enough with psychovisual optimizations [13:30] <luc4> JEEB: I see, but the same happened to me with h264. [13:30] <luc4> JEEB: Ill try with this sample and see the result. [13:31] <luc4> JEEB: thanks for the information. [13:31] <JEEB> sounds like you are just missetting or misusing something. Feel free to pastebin your full command line and terminal output of that command line and link it here [13:32] <JEEB> because libx264 is by now rather stable and it does exactly what you set [13:34] <luc4> JEEB: this is what Im testing now: http://paste.kde.org/pyb0pvwdr [13:34] <luc4> JEEB: but I previously tried not setting -b:v at all and setting to 6000. [13:35] <JEEB> not setting it at all is a default that is different for every encoder [13:35] <luc4> JEEB: by setting 6000k I got a file which was nearly as large as the original. [13:35] <JEEB> x264 had it at crf 23 [13:36] <JEEB> luc4, using bit rate based encoding, esp. with a single pass is probably rather broken in libx265 :P at least I definitely wouldn't be surprised if it was [13:36] <luc4> JEEB: in the h264 guide I real crf 18 should be almost no quality loss. [13:36] <JEEB> that is not a definitive value [13:36] <luc4> JEEB: ah I see. [13:36] <JEEB> you start from 23 and go up if it looks good already, and down if it looks bad [13:36] <JEEB> and thus you see which value is the highest still looking good for you for that kind of content [13:36] <JEEB> that is the idea of CRF [13:37] <luc4> JEEB: ok then. But should I keep -b:v or crf takes that into account? [13:37] <JEEB> they are incompatible settings, you can't set an average bit rate and a rate factor at the same time [13:38] <JEEB> also, the range of CRF values is completely different with libx265, so if you're going to be using that, you'll probably have to find the initial value from around 28 or so ? Not sure [13:41] <JEEB> anyways, I recommend going for x264 in general for this kind of usage [13:41] <JEEB> x265 is just too volatile for my use [13:41] <JEEB> not to mention that in all of my tests as soon as you leave the "very low bit rate" scenario [13:41] <JEEB> x265 is the one starting to lose :P [13:42] <JEEB> maybe in the future x265 is going to be better, but not now [13:42] <luc4> JEEB: sure, Ill stay with h264. I was just starting to test h265 as Im reading a lot about it and I was anxious to give it a try. [13:42] <JEEB> yeah, HEVC as a standard is nice [13:43] <JEEB> too bad it'll take time for the implementations to become good :) [13:43] <JEEB> or well, let's just say that libx264 being so good doesn't make them look good even if they were kind of alright now [13:43] <JEEB> because they'd still have to beat libx264, which is kind of awesomesauce as far as an implementation goes [13:43] <luc4> JEEB: but happened to me something similar with h264. mp2 video resulted smaller in size than h264 after transcoding. So I thought it was me again doing something wrong. [13:44] <JEEB> yes, with x264 that option is a bigger one [13:44] <JEEB> so you most probably miscalculated the original file's average bit rate and set the rate control to a too big of a value [13:44] <luc4> JEEB: ah possible, as I understand ffprobe does not always compute the bitrate correctly. [13:45] <JEEB> personally I haven't used bit rate based encoding in quite a while, and if one is to use such it's much better to do it wish -pass 1 and -pass 2 [13:45] <JEEB> *with [13:45] <JEEB> single pass ABR is only for streaming scenarios, and even there I'd rather use CRF ;) [13:46] <JEEB> (with VBV limitations) [13:46] <rcombs> CRF :D [13:46] <luc4> JEEB: so I should go with crf and also -pass 2? [13:46] <JEEB> no, CRF is single pass only [13:46] <luc4> ah ok [13:46] <JEEB> so you don't need to set any -pass there [13:46] <JEEB> you only set -pass 1 with the first pass and -pass 2 with the second when using bit rate based ABR [13:46] <JEEB> (and yes, you have to run ffmpeg twice) [13:47] <luc4> Ill try with crf then and see what I get. Thanks for the info! [13:47] <JEEB> well, as I said [13:47] <JEEB> crf is something you will have to tweak yourself [13:48] <JEEB> <JEEB> you start from 23 and go up if it looks good already, and down if it looks bad [13:48] <JEEB> <JEEB> and thus you see which value is the highest still looking good for you for that kind of content [13:48] <JEEB> <JEEB> that is the idea of CRF [13:48] <JEEB> and thus you will find the exact compression rate you can get with those compression settings to gain the quality level you want [13:48] <JEEB> and then you can use the CRF value with similar content [13:49] <JEEB> if you go for completely different content (resolution, frame rate etc.), then you might have to tweak the value somewhat [13:49] <JEEB> for example for higher resolution content you generally don't zoom it in as much when watching so you can use a higher crf value, and so forth [13:51] <rcombs> and if your audience is OCD you say "fuck it", set -crf 15, and end up with huge files [17:06] <BlackDream> I have created an application that takes the output of ffmpeg (mpegts/flv or any other output) and streaming it to a port, then the clients can connect to it [17:06] <BlackDream> The command is as follows [17:07] <BlackDream> ffmpeg -i "stream_link" -codec copy -f mpegts - | server -p 20000 [17:07] <BlackDream> It works fine, however, i'm unable to create duplicate outputs using this [17:07] <BlackDream> For example if i want to duplicate flv and mpegts how i will do it :/ [17:08] <BlackDream> I saw that: https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#no1 however this comes to pipes and it's a bit hard its not the same [17:14] <sacarasc> BlackDream: How about using a named pipe and using server with that instead... Something like mkfifo cheese.ts && ffmpeg -i "stream_link" -codec copy -f mpegts cheese.ts -f copy keeping.ts -f copy keeping.flv & server -p 20000 [17:15] <sacarasc> mkfifo cheese.ts && ffmpeg -i "stream_link" -codec copy -f mpegts cheese.ts -f copy keeping.ts -f copy keeping.flv & server -p 20000 cheese.ts [17:15] <sacarasc> That, even. [17:19] <BlackDream> wait to investigate it :P [17:33] <kaotiko> hi [17:58] <binni> How do I convert a video using the libx264 codec but NOT touch the audio, keep it as it is? [17:59] <binni> I have: avconv -i video1.mp4 -vcodec libx264 -b:v 3000k -maxrate 3000k -bufsize 3000k video1-quality.mp4 [17:59] <binni> but it decides to convert the audio aswell [17:59] <binni> Stream #0:1 -> #0:1 (aac (native) -> aac (libvo_aacenc)) [18:00] <foonix> -c:a copy [18:01] <binni> foonix, thank you! [18:13] <zenny> Hi, was reading this: "https://trac.ffmpeg.org/wiki/How to concatenate (join%2C merge) media". But my situation is a bit different than what the script at the bottom of the page describes. I need to add a video (including audio) each in the beginning and end of a certain video. What should be the tweak to achieve? Thanks! [18:14] <c_14> just use the concat demuxer? [18:14] <c_14> file video1\nfile video2\nfile video3 [18:19] <zenny> c_14: Thanks! [19:49] <pappi> hello. i'm trying to force the use of --hwaccel vdpau in ffmpeg. any tips on how to deal with this? [19:56] <BtbN> Pass --hwaccel vdpau on the command line. [20:42] <antranigv> Greetings, I'd like to convert an .webm video to .ogg audio, any CLI help ? thanks in advance! [20:44] <JEEB> webm files tend to have either opus or vorbis audio, and both in theory should go into the ogg container, I guess? [20:44] <JEEB> so ffmpeg -i welp.webm -map 0:a -c copy out.ogg [20:44] <JEEB> try this [20:45] <JEEB> -map 0:a selects the audio track(s) from the input [20:45] <JEEB> -c copy sets "codec" to copy, which copies the input track [20:45] <JEEB> out.ogg creates an ogg file [20:46] <antranigv> JEEB it worked! thanks ! I was looking for this for 2-3 hours :D [21:58] <ChocolateArmpits> Is there a way to check if a source is true 10 bit and not 8bit resampled to 10bits ? [21:59] <JEEB> no [22:00] <JEEB> because a "true 10bit source" might not be using the full range of values (as in, have similar steps to 8bit content), and 8bit content could be debanded after being bit shifted to 10bit range of values (in which case there would be steps using 10bit steps to all of their potential) [22:03] <ChocolateArmpits> JEEB: but it would be extremely unlikey for real life shot material to have 8bit stepping matching if it was downsampled from whatever the sensor captured down to 10 bits ? [22:03] <ChocolateArmpits> Unless shooting artifical charts [22:05] <JEEB> if specific to real life material straight out of a camera, then I guess. But I still wouldn't be able to trust the thing based upon a single picture, you'd have to push through a whole clip and have a heuristic based on it [22:06] <ChocolateArmpits> So I guess only a gradient chart could be used for truthful testing ? [23:44] <kevmitch> what's the difference between bbox and cropdetect filters? [23:58] <kevmitch> and what exactly does dejudder do? I thought pullup restored the original film frames. Is that not the case? [00:00] --- Mon Oct 20 2014

1 0

ffmpeg-devel.log.20141018
by burek 19 Oct '14

19 Oct '14

[00:00] <J_Darnley> Does FFmpeg have an example of increasing indentation in printed output somewhere? [00:00] <J_Darnley> What about ffprobe's various outputs? [00:03] <Compn> i dont think so [00:04] <Compn> screenshots and examples would look nice in the docs [01:32] <cone-574> ffmpeg.git 03Marton Balint 07master:9dac7977a981: ffplay: remove delays when paused from video and subtitle thread [01:32] <cone-574> ffmpeg.git 03Marton Balint 07master:0edf3e7a28d0: ffplay: remove manual bug option handling code [01:32] <cone-574> ffmpeg.git 03Marton Balint 07master:2c4d6a342d4f: ffplay: update documentation [02:14] <jamrial> "<ubitux> the 7 lines of vpx hacks could be replaced with a check_pkg_config "vpx >= 0.9.7" or something" <--- I disagree [02:14] <jamrial> the libvpx checks are for vp8 decoder and encoder, and vp9 decoder and encoder [02:16] <jamrial> a pkg-config for vpx >= 0.9.7 will only guarantee that vp8 enc/dec is available. you still need to check if vp9 enc/dec are available and new enough [02:17] <jamrial> in the end it will still be at least three checks [02:19] <jamrial> we can replace the checks for pkg-config ones, though. i'm not against that [02:19] <jamrial> it's just that the replacement will still be at least three separate checks [02:34] <cone-574> ffmpeg.git 03Supraja Meedinti 07master:0eabbe2101f1: libavutil: Add CAST128 symmetric block cipher [03:01] <cone-574> ffmpeg.git 03Benoit Fouet 07master:e06263ef1e0e: configure: add pkg-config support for libx264 [03:02] <jamrial> ^now we just need to remove the fallback hack :p [03:11] <cone-574> ffmpeg.git 03Uwe L. Korn 07master:f4cd8b80b9cb: rtmpproto: Track last received timestamp [03:11] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:178764e833b1: Merge commit 'f4cd8b80b9cb2a9a4112d8abb1d3b93cd3b4e467' [03:49] <cone-574> ffmpeg.git 03Uwe L. Korn 07master:9bec3ca2b878: rtmpproto: Add pause support [03:49] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:7891b4fa590b: Merge commit '9bec3ca2b878c83d2337bc6095bd51c3e3f8eef0' [11:43] <ubitux> fate is a bit yellow [11:46] <ubitux> and cast leaks [12:21] <cone-919> ffmpeg.git 03Anton Khirnov 07master:fb238f8230c1: urlprotocol: remove unused url_interrupt_cb declaration [12:21] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:83a897656aae: Merge commit 'fb238f8230c1b0314985482207add548c68f83cf' [12:46] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:7321fc2f4304: tests/fate/mpeg4: add -sws_flags +bitexact [13:07] <cone-919> ffmpeg.git 03Timothy B. Terriberry 07master:6cbbf0592f4f: resample: Avoid off-by-1 errors in PTS calcs. [13:07] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:17d16008befa: Merge commit '6cbbf0592f4f3940aac7f687850d1b726a2ea836' [13:52] <cone-919> ffmpeg.git 03Anton Khirnov 07master:a6e4ce9fd508: lavc: make rc_qsquish a private option of mpegvideo encoders [13:52] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:74128352047e: Merge commit 'a6e4ce9fd50897dc6d9c2ada4b6b8090de7de5bf' [14:03] <cone-919> ffmpeg.git 03Anton Khirnov 07master:946f95354ba7: lavc: make rc_qmod_* into private options of mpegvideo encoders [14:03] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:17a065cf204f: Merge commit '946f95354ba76ef73c9b66889d86ab5fba4fb486' [14:17] <cone-919> ffmpeg.git 03Anton Khirnov 07master:a75c2eb25a62: lavc: make rc_eq into private options of mpegvideo encoders [14:17] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:ec8f56efad2c: Merge commit 'a75c2eb25a62105c09b48521aef429dc8a231637' [14:28] <cone-919> ffmpeg.git 03Anton Khirnov 07master:66a68ddd1ac3: lavc: make rc_buffer_aggressivity/rc_initial_cplx into private options of mpegvideo encoders [14:28] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:fc67d937cc73: Merge commit '66a68ddd1ac376b24db8695058bc8fc28f5bada6' [15:27] <cone-919> ffmpeg.git 03Anton Khirnov 07master:59b55c494e01: lavc: deprecate unused me_threshold field [15:28] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:c48e8d9dca55: Merge commit '59b55c494e01de09dfd7cbf9cbbe7223265ac283' [15:37] <anshul_mahe> I have completed my closed caption decoder for ffmpeg [15:38] <kierank> anshul_mahe: how have you done it [15:38] <kierank> and is it 608 and 708? [15:39] <anshul_mahe> it is 608 now using external ccextractor lbrary [15:39] <anshul_mahe> my problem is that my work is based on base patchof nicolas [15:39] <kierank> libzvbi could also do it [15:39] <kierank> and ffmpeg already uses that for 608 [15:39] <kierank> i mean teletext [15:40] <kierank> the difficult part is how you decode captions inside h264/mpeg-2 [15:40] <anshul_mahe> using movie filter [15:41] <anshul_mahe> Just now for giving my commits on ffmpeg-devel list, I want my patch to be multi authored [15:41] <ubitux> speaking of this, libzvbi decodes as text like shit [15:42] <cone-919> ffmpeg.git 03Anton Khirnov 07master:685be3d1aecf: libvpxenc: add static-thresh private option [15:42] <ubitux> so if anyone is willing to fix that... [15:42] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:8be1120deb2b: Merge commit '685be3d1aecf965281128f3b631ed965bcfa754d' [15:42] <ubitux> (the zvbi wrapper i mean) [15:43] <wm4> ubitux: teletext has a more serious problem [15:43] <wm4> it has multiple pages at once or something [15:43] <ubitux> yeah& [15:44] <ubitux> it needs a "don't consume the whole packet at once" just like a/v [15:44] <wm4> uh [15:44] <wm4> please no [15:44] <wm4> because fuck that [15:44] <ubitux> then what do you propose? [15:44] <wm4> and it should be fixed for audio [15:45] <ubitux> and how are you going to do that? [15:45] <wm4> it makes the data flow much more complicated and awkward and has potential to introduce infinite loops if not careful [15:45] <wm4> dunno [15:45] <wm4> for teletext, the abstraction is wrong [15:45] <wm4> teletext is basically stateful as I understand it [15:47] <anshul_mahe> I would like someone to take authorship of my patch for closed caption using ccextractor library, is someone here willing to take [15:47] <anshul_mahe> then I would send my patch to ffmpeg-devel mailing list [15:47] <J_Darnley> What do you mean "take authorship"? [15:49] <anshul_mahe> I dont want to be author of those patch, i signed NDA for it [15:50] <anshul_mahe> I tried making whole new implentation, but I found my mind is working in same way [15:51] <wm4> anshul_mahe: ask kierank or j-b, I think they have a grasp on such issues [15:51] <wm4> I mean, don't ask them for taking ownership, but for advice what to do [15:51] <kierank> anshul_mahe: nda for what? [15:53] <anshul_mahe> ok, looking in mail, did not read that carefully, will come back soon with more detail [15:56] <anshul_mahe> Its just an statement here http://pastebin.com/HAUS5UKw [15:57] <cone-919> ffmpeg.git 03Anton Khirnov 07master:c4f7bee0872d: lavc: deprecate unused mb_threshold field [15:57] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:1a5871d1bec6: Merge commit 'c4f7bee0872d5e2e0e91673dd4b0f18de4a3d866' [16:08] <anshul_mahe> kierank: any idea's? [16:08] <kierank> then you can't [16:08] <kierank> but if that company distributes ffmpeg [16:08] <kierank> then they would have to send patches [16:08] <kierank> ianal [16:09] <anshul_mahe> they are using it, in there server they dont distributes [16:12] <kierank> ah then you can't [16:13] <anshul_mahe> ok, just now i am using whole ccextractor library for decoding, If I decompose ccextractor with small library of different decoder, then integrate it [16:14] <anshul_mahe> will that work [16:14] <J_Darnley> IANAL but I would side with kierank's opinion on this. If XYZ own the patch you've written then it is up to them whether they submit it. They are under no obligation to release source code if they don't distribute FFmpeg. [16:15] <J_Darnley> From that one sentence you posted, any work you do while being paid by XYZ is owned by XYZ. [16:21] <anshul_mahe> ok, I would rather implement decoder over here, I have its RFC, that was not I did when I was getting paid. [16:22] <anshul_mahe> and rather stay away from signing such document [16:24] <anshul_mahe> I am in dilemma, should I use decomposed ccextractor library or write a new decoder for ffmpeg [16:26] <anshul_mahe> or may be libzvbi [16:36] <BBB> omg more inline assembly [16:44] <cone-919> ffmpeg.git 03Anton Khirnov 07master:15ec053c4c0b: lavc: make border_masking into private options of mpegvideo encoders [16:44] <cone-919> ffmpeg.git 03Michael Niedermayer 07master:709ab39cb9a4: Merge commit '15ec053c4c0b198a2e93eb8e60c8f41e091e0c40' [16:48] <J_Darnley> BBB: what now? [16:48] <BBB> that vf_noise patch [16:48] <BBB> inline asm [16:48] <BBB> :( [17:02] <ubitux> BBB: it's just moved somewhere else [17:07] <BBB> I know, but its still inline asm [17:07] <BBB> lets convert it and make msvc users happy [17:07] <BBB> they get such a degenerate experience :( [17:11] <cone-919> ffmpeg.git 03Giorgio Vazzana 07master:b48c95b6c92e: libavutil: Add missing av_free() to cast5 selftest [17:13] Action: michaelni wonders if inline->yasm really fixes "MS users getting a degenerate experience" [17:17] <michaelni> wouldnt it be needed to replace the OS too ... [17:19] <kierank> sure, probably ~1 billion reinstalls needed (chrome+vlc) [17:33] <ubitux> i think vlc doesn't care about libavfilter [17:33] <ubitux> and chrome doesn't need it [17:34] <ubitux> that said, i'm all for having it in yasm [17:34] <ubitux> and that commit eases this transition [17:34] <ubitux> it shouldn't be that much work afaict [17:34] <ubitux> and this asm was just ported from mplayer, and it was kept as is to avoid the risk of bugs during the port [17:51] <hima> ubitux : hi [17:53] <hima> here is the modified code 1. removed get_pts_end 2. include event { .please have a look http://fpaste.org/143128/41364750/ [17:55] <ubitux> does it work with "event" with AND without "{"? [17:55] <ubitux> also, what if there is more spaces between "event" and "{" [17:55] <ubitux> what about a line break between "event" and the timestamp? [17:57] <hima> in all the examples i saw there was just a single space between event and { [17:57] <hima> ohh right i forgot that case [17:57] <hima> making that one [17:58] <hima> and line break...it does skip \n and spaces [18:05] <hima> ubitux : http://fpaste.org/143129/36482931/ [18:05] <ubitux> does it work? [18:07] <ubitux> there is nothing that prevent anyone from having spaces or \n between the "event" and { [18:09] <hima> yes succesfully did make it [18:09] <ubitux> can you make a patch with git so i can test? [18:09] <hima> okayy will add spaces and \n between event and { then [18:11] <hima> ok after this change it doesnt work again showing segmentation fault :| [18:11] <hima> ./ffmpeg -f kate -i sample_kate.kate out.srt [18:11] <hima> with make it did not give any error [18:11] <hima> though [18:12] <ubitux> i can help you with ffmpeg stuff, but you know you are supposed to know C and deal with pure C problems yourself [18:12] <ubitux> i'm not going to debug your code [18:12] <ubitux> i will just make sure it's not broken and point you out issues with it if there are [18:14] <hima> http://fpaste.org/143131/64875014/ there are no syntax problems ubitux logically i guess its fine. can you just help me how to read the output of the valgrind as to figure out the problem [18:14] <hima> thisis the valgrind output [18:14] <ubitux> at kate_read_header (katedec.c:60) [18:14] <ubitux> you are reading a byte you're not supposed to [18:15] <ubitux> that's all i can say [18:17] <hima> okay i will figure it out. [18:28] <hima> ubitux: do i have to make a change in libavcodec/avcodec.h too besides allformats.h and Makefile? [18:29] <ubitux> not you're exporting as text [18:29] <ubitux> so no need to touch libavcodec [18:29] <ubitux> you're supposed to add an entry in the Changelog and doc/general.texi, but you probably want to do that at the end, don't bother about it for now, it will brings problem when rebasing [18:30] <hima> okay [18:53] <hima> ubitux: on ./ffmpeg -f kate -i sample_kate.kate out.txt it gives http://fpaste.org/143139/51048141/ but the cursor just keeps on waiting there same for valgrind.no segmentation fault though [18:53] <hima> what could be the error? [18:54] <ubitux> infinite loop [19:08] <arwa> I tried writing a very naive code for neighborhood scaling. I am not getting the desired image. Can anyone help me debug it? http://pastebin.com/NMie9LNi [19:15] <J_Darnley> arwa: "This is a private paste. If you created this paste, please login to view it." [19:24] <arwa> The paste is public. [19:24] <arwa> Are you not able to view it? [19:24] Action: wm4 suggests http://www.sprunge.us [19:25] <wm4> pastebin.com is incredibly annoying slow and shitty, don't know why anyone would use it [19:29] <anshul_mahe> wm4, what you suggest show 404 in India [19:31] <wm4> oh, it's actually served on http://sprunge.us/ [19:34] <arwa> Actually, its giving me proxy error. How do I do the proxy settings? [19:35] <arwa> I tried the command line --> "cat try | curl -F 'sprunge=<-' http://sprunge.us" [19:36] <wm4> here's another random one that's IMO better that pastebin.com: http://privatepaste.com/ [19:38] <arwa> My code -> http://privatepaste.com/80b17a285f [19:38] <arwa> thanks :) [19:40] <wm4> you should restrict the pixel formats (PIXFMTS) it takes to the formats which work [19:41] <wm4> the (int) cast in the filter function are pointless... if j is int, then j/2 is also int [19:42] <arwa> okay. [19:42] <arwa> It wont work without restricting the pixelformat? [19:43] <wm4> it will, but there's the chance it gives you a pixelformat you didn't expext, and then random things will happen [19:43] <wm4> s/expext/expect [19:47] <arwa> okay [19:51] <arwa> I added the query formats [19:52] <arwa> still its not giving me the desired image [19:52] <ubitux> what pix fmt did you request? [19:54] <arwa> AV_PIX_FMT_RGBA, AV_PIX_FMT_BGRA, AV_PIX_FMT_ARGB, AV_PIX_FMT_ABGR, AV_PIX_FMT_RGB24, AV_PIX_FMT_BGR24, AV_PIX_FMT_RGB565BE, AV_PIX_FMT_BGR565BE, AV_PIX_FMT_RGB555BE, AV_PIX_FMT_BGR555BE,AV_PIX_FMT_RGB565LE, AV_PIX_FMT_BGR565LE, AV_PIX_FMT_RGB555LE, AV_PIX_FMT_BGR555LE,AV_PIX_FMT_NONE [19:54] <ubitux> do you really support all of these? [19:54] <ubitux> it looks like to me that you only support AV_PIX_FMT_RGB24 [19:57] <arwa> no [19:58] <arwa> but i have returned an avlog if we get any format diff than rgb24 [20:00] <ubitux> i mean your code seems to be able to cope with rgb24 only [20:00] <ubitux> and bgr24 [20:00] <ubitux> not so sure about the others [20:01] <wm4> the others are 4 bytes and 2 bytes per pixelformats, so they definitely won't [20:02] <arwa> yeah i am working on rgb24 only [20:02] <wm4> then don't allow the other formats [20:04] <arwa> i just requested all the pixel formats. But I am giving out a error message via avlog informing that the other pixel formats wont work [20:07] <arwa> and now when I am removing all the pixel formats, I am getting segmentation fault. Does this mean that the format I am giving does not belong to either rgb24 or bgr24? [20:13] <cone-105> ffmpeg.git 03Anton Khirnov 07master:530c1441fd14: lavc: make lmax/lmin into private options of mpegvideo encoders [20:13] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:ae6ba2db8ff8: avcodec/mjpegenc: add priv_class to encoders [20:13] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:25f4e9387d0e: Merge commit '530c1441fd1426b6a4bb33485ff3226e1ae0ad45' [20:13] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:bf301258b49f: avcodec/snowenc: add FF_MPV_COMMON_OPTS [20:13] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:f4445c7be45c: avcodec/wmv2enc: add priv_class [20:14] <wm4> arwa: then try to find out where it crashes, by using gdb or valgrind [20:19] <arwa> I was trying to do that. But I am not getting how to use it with ffmpeg executable file? [20:20] <arwa> if i am doing normal gdb, it is not able to compile the file itself. [20:22] <wm4> how are you trying to use it? [20:27] <arwa> I am doing gdb, then its giving the file to be compiled as input [20:27] <arwa> * I am giving the file to be compiled as input [21:43] <cone-105> ffmpeg.git 03Luca Barbato 07master:f401792595dd: vf_drawtext: Do not leak the mmapped textfile [21:43] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:0b2f860db53e: Merge commit 'f401792595dd7760f531e8a3bd2336e9033bd45a' [22:01] <cone-105> ffmpeg.git 03Vittorio Giovara 07master:6d96c7ea8960: vf_showinfo: Forward the av_image_get_linesize error [22:01] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:3a4c22f8067f: Merge commit '6d96c7ea89600655b1f18a3b12f3f041fa714152' [22:13] <cone-105> ffmpeg.git 03Vittorio Giovara 07master:0d989dbfc4bc: af_resample: check av_opt_set_dict return value [22:13] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:733db64ce0c6: Merge commit '0d989dbfc4bc5bc1d563e967449116a7a9865258' [22:20] <cone-105> ffmpeg.git 03Vittorio Giovara 07master:088eca28164c: avresample: prevent theoretical division by zero [22:20] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:9bcc430481d7: Merge commit '088eca28164c8cd3b72b0c3d3f9e3fe5ee5cb28f' [22:53] <cone-105> ffmpeg.git 03Luca Barbato 07master:963f76144897: avresample: Make sure the even check does not overflow [22:53] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:85cf049d5092: Merge commit '963f76144897d3f7684d82ec21e51dd50ea1106e' [23:37] <cone-105> ffmpeg.git 03Timothy Gu 07master:b7c77912b621: oss_audio: use a macro to simplify ioctl() error checking [23:37] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:837605704daf: Merge commit 'b7c77912b62163b3b46ce93fe42fff3c83604c82' [23:47] <cone-105> ffmpeg.git 03Vittorio Giovara 07master:fc1eda543c6e: swscale: fix sign extensions in yuv planar conversion [23:47] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:fa040c224703: Merge commit 'fc1eda543c6ef043300612db90da5cfd972af650' [23:56] <cone-105> ffmpeg.git 03Alexander Drozdov 07master:0034314a69e7: rtmp: Always call rtmp_close() on rtmp_open() failure [23:56] <cone-105> ffmpeg.git 03Michael Niedermayer 07master:52bf4ad6746a: Merge commit '0034314a69e76a53534a74cceef865cfcb7b42cc' [00:00] --- Sun Oct 19 2014

1 0

ffmpeg.log.20141018
by burek 19 Oct '14

19 Oct '14

[00:45] <Baked_Cake> is mpc a reliable way to check a vids format [00:45] <Baked_Cake> it's saying the format is yuv 4:2:0 but ctrl+j says yuy2 4:2:2 [13:04] <svvitch> how can I get the size of second input to use in crop's formula of first input ? [13:04] <svvitch> in ffmpeg params [14:43] <BlackDream> Hello, why when i compile FFmpeg staticaly the domain names are not working "Name or service not known" [14:43] <BlackDream> Only with IPs [14:44] <BlackDream> I have a working FFmpeg which i compiled statically and the domain names were working, however i don't remember how i did it :S [14:45] <c_14> Have you tried it with one of the builds from http://johnvansickle.com/ffmpeg/ ? [14:45] <BlackDream> yes the same happens [14:45] <BlackDream> try it yourself too you will see that domain name are not working in stream source [14:45] <relaxed> BlackDream: install nscd and see hostname resolution works. [14:45] <relaxed> see if* [14:46] <relaxed> Most likely what happened is you linked against eglibc, but your distro moved to glibc. [14:47] <BlackDream> if i install nscd and compile it again will work? [14:47] <relaxed> No, you shouldn't need to recompile ffmpeg. [14:47] <BlackDream> you are right [14:47] <BlackDream> it works [14:47] <BlackDream> lol [14:47] <BlackDream> -.- [14:48] <BlackDream> Thanks! [14:48] <relaxed> you're welcome. [14:50] <BlackDream> relaxed , --enable-librtmp is better than the native ffmpeg rtmp library? What do you think? [14:51] <relaxed> I really can't speak to that. [15:01] <BlackDream> Ok another question :(. I have an rtmp stream that IS NOT WORKING! However when i import it in the ffmpeg -i "rtmp_link_here" it just hangs [15:01] <BlackDream> and never end [15:01] <BlackDream> however, rtmpdump does the same, but rtmpdump has the argument -m which is the timeout connection so it exits [15:01] <BlackDream> is there something similar to ffmpeg? [15:03] <relaxed> -timeout [15:04] <relaxed> ffmpeg -h full |less +/'rtmp AVOptions' [15:10] <BlackDream> relaxed sorry to bother you again... ffmpeg -timeout 10 -i "rtmp_link_here" [15:10] <BlackDream> and doesnt work says an error [15:12] <relaxed> place it after the input [15:13] <BlackDream> still hangs, timeout didnt do anything [17:02] <svvitch> how to blend overlay video and png with different sizes ? [17:09] <anshul_mahe> svvitch: resize them or pad the video or image [17:34] <svvitch> anshul_mahe: I try this command but it said that sizes not corresponding https://dpaste.de/QQpp [17:45] <anshul_mahe> isn't it saying you need to make SAR same [17:59] <svvitch> anshul_mahe: yes I tried to set 'sar' but it output a red layer https://dpaste.de/9eQV [18:08] <anshul_mahe> are you getting red layer by just changing sar [18:10] <svvitch> anshul_mahe: yes [18:10] <svvitch> anshul_mahe: http://im9.eu/picture/de2203 [18:12] <c_14> I'm pretty sure that's the pad. [18:13] <c_14> change the red to blue and check if it changes [18:14] <svvitch> this works fine https://dpaste.de/u42f screen: http://im9.eu/picture/zu2203 [18:14] <svvitch> but I need to aply transparency to png image [18:17] <svvitch> c_14: it just changes color :) http://im9.eu/picture/kt2203 [18:21] <c_14> I have a strong feeling that the blend filter is resetting the opacity on the background of the overlay. [18:21] <c_14> Have you tried without the pad? [18:21] <c_14> I'm relatively sure overlay at least works with different sizes [18:23] <c_14> hmm, overlay works blend does not [18:26] <c_14> Have you tried making the input png transparent? [18:27] <c_14> yep [18:27] <c_14> I just used imagemagick to make the input png transparent and then used the overlay filter. [18:30] <svvitch> hah the main goal is to do this without external tools [18:31] <svvitch> thats why I'm searching for blend way [18:31] <svvitch> there must be a solution, for sure, but how [18:38] <c_14> I don't know of any good way to set transparency in ffmpeg... [18:42] <svvitch> suppose you know some bad ways :) [18:45] <c_14> alphaextract/alphamerge come to mind. You'll need a video/png with the correct alpha set which you could extract and then merge into your overlay png. [18:46] <svvitch> hmm, I've played with some alphamerge options, ok ok [18:47] <c_14> Then there's geq, you could use an expression to modify the alpha at each pixel [18:51] <c_14> But I never tested either method, ymmv [18:56] <svvitch> https://dpaste.de/J1cw something is wrong :( [19:00] <svvitch> heh ffmpeg docs wrote "If youre trying to apply an image as an overlay to a video stream, consider the overlay filter instead." [20:28] <anshul_mahe> how do we access temperal sequence number in frame got by av_decode_video2 [21:47] <PovAddict> topic should be updated: ffmpeg is now in debian unstable! [21:50] <PovAddict> can I ask questions about dvdauthor? I don't know any better place [23:07] <ChocolateArmpits> PovAddict, I can try answering, used it myself [00:00] --- Sun Oct 19 2014

1 0

ffmpeg-devel.log.20141017
by burek 18 Oct '14

18 Oct '14

[04:21] <rcombs> wm4: poking at that file from yesterday; mpv actually also has trouble seeking in the full version of it [04:22] <rcombs> it works, but it takes a while, probably because it's reading the entire file instead of using the index/cues to seek [04:30] <rcombs> looks like the demuxer just never finds any Cues [04:32] <rcombs> (mpv has no trouble using its internal matroska demuxer) [05:44] <cone-879> ffmpeg.git 03James Almer 07master:864f9326fb4d: x86/vf_noise: move asm code to a separate file [06:30] <cone-879> ffmpeg.git 03Moritz Barsnick 07master:bbd8c8526320: doc/filters.texi: fix several typos in the vidstab* sections [07:13] <rcombs> so, commenting out matroskadec.c:1371 fixes it, but I'm not sure if that has a reason to be there [07:34] <rcombs> apparently added in a commit from 2008; 13b350a3a4b2f9a04d2d2bcdee5b7f26d96fe34b [08:00] <jamrial> http://pastebin.com/1KY7kGm7 the power of bmi2's rorx instruction [08:05] <jamrial> gcc also generates vfm{add,sub} instructions instead of a combination of mulss/{add,sub}ss if you use -fma or -fma4 [08:05] <jamrial> really nice [08:41] <akira4> I'm trying to add a sample to the fate-suite. How exactly is it done? [08:48] <jamrial> if you mean locally, follow ubitux's instructions on his latest email. namely just drop the sample inside the folder you rsynced the fate suit, add the relevant entry to subtitles.mak, then run the test adding GEN=1 to create the reference file [08:48] <jamrial> if you meant to the rsync server, it will be uploaded after the test you wrote is reviewed and accepted [08:49] <jamrial> rcombs: you forgot to attach the patch :p [08:50] <akira4> I see. I get an error when I try to rsync http://pastebin.com/tFaiywAx. [08:50] <rcombs> jamrial: huh? It shows as attached here& [08:50] <jamrial> rcombs: http://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164181.html [08:52] <rcombs> jamrial: well that's certainly strange; just re-sent [08:54] <rcombs> AND failed again [08:54] <rcombs> no idea what's going on with that [08:55] <jamrial> try pasting the patch in the body. as long as your client doesn't break lines it should be ok [08:56] <rcombs> linked to a gist [08:57] <jamrial> akira4: try again. that shouldn't happen [08:58] <jamrial> it may be something with your connection. the rsync server seems to be working for me [09:00] <akira4> okay. thanks. [09:45] <j-b> 'morning [13:03] <wm4> rcombs: can you post the file? [13:13] <cone-104> ffmpeg.git 03Janne Grunau 07master:f29c226af0ec: fate: add mpeg4 tests for frame size changes [13:13] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:1aed82846cbc: avformat/m4vdec: mark as possibly containing discontinuities [13:13] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:a708689fdac6: Merge commit 'f29c226af0ecc34c417b646580a9acd7d8bbaf48' [13:13] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:c40a1bd253f6: tests/fate/mpeg4: fix samples path [13:24] <cone-104> ffmpeg.git 03Luca Barbato 07master:cf83c0173917: avplay: Always free find_stream_info options [13:24] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:019ff7a0c37a: Merge commit 'cf83c0173917a44b89a25ea2b435429bbf9ee767' [13:34] <cone-104> ffmpeg.git 03Luca Barbato 07master:e58a140cf91d: avplay: Always free opts [13:34] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:ee73a69020a8: Merge commit 'e58a140cf91d1a9cdfa3115d73c923dab0e9e7dc' [13:43] <cone-104> ffmpeg.git 03Luca Barbato 07master:aeb23fc4549a: avprobe: Remove a pointless check [13:43] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:9d1155dae303: Merge commit 'aeb23fc4549a25ef32ff085d2a76227f90caf403' [14:04] <cone-104> ffmpeg.git 03Vittorio Giovara 07master:c802a2e718fb: avconv: fix leak in filter error [14:04] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:1dd38e99fd12: Merge commit 'c802a2e718fb3619291f310f851f1a1cdcf4f581' [14:04] <prachi> i was wondering if anyone could tell me what [14:05] <prachi> if (sscanf(ptr, "%*d:%*d:%*d.%*d%c", &c) == 1 && strchr(": =", c)) [14:05] <prachi> means [14:05] <prachi> static int vplayer_probe(AVProbeData *p) [14:05] <prachi> 35 { [14:05] <prachi> 36 char c; [14:05] <prachi> 37 const unsigned char *ptr = p->buf; [14:05] <prachi> 38 [14:05] <prachi> 39 if (sscanf(ptr, "%*d:%*d:%*d.%*d%c", &c) == 1 && strchr(": =", c)) [14:05] <prachi> 40 return AVPROBE_SCORE_MAX; [14:05] <prachi> 41 return 0; [14:05] <prachi> 42 } [14:06] <prachi> in this function [14:06] <prachi> I was looking for some help [14:06] <J_Darnley> Woah! Use pastebin rather than pasting in the channel. [14:06] <wm4> prachi: use http://sprunge.us [14:06] <prachi> okay,soory,I am new.will take care in future [14:07] <J_Darnley> if scanf return 1 it means it read exactly one thing (number, character, string) [14:08] Action: J_Darnley really hates strings [14:08] <Daemon404> they're only bad if you use [14:08] <Daemon404> C [14:08] Action: Daemon404 runs [14:10] <prachi> okay! [14:11] <J_Darnley> Okay, I think this code is checking that the string in ptr has a colon, space, or equals sign following a series of numbers. [14:14] <J_Darnley> I also hate binary data formats, especially ones that have gone through several rounds of backward or forward compatible extension. [14:15] <J_Darnley> If I may quote a comment I read last night "size of extended data + 4 cause we fucked up". [14:21] <cone-104> ffmpeg.git 03Vittorio Giovara 07master:c92965dbfbb7: avconv: check return value [14:21] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:61c6e69709ca: Merge commit 'c92965dbfbb7e2b49df14db2fd0e23a29295730a' [14:40] <Daemon404> ARG [14:40] Action: Daemon404 bitten again by lack of fucking pkg config use in configure for various libs [14:40] <ubitux> :) [14:41] <Daemon404> not just x264 this time! [14:41] <ubitux> which one? [14:41] <Daemon404> libvpx [14:41] <ubitux> aaah libvpx [14:41] <ubitux> and the checks are awesome [14:41] <Daemon404> its especially misleading [14:41] <Daemon404> cause it checks for a header and then says "needs >=0.9.1" [14:41] <Daemon404> as if it had used pkg config [14:42] <cone-104> ffmpeg.git 03Luca Barbato 07master:0db6bbb24c1a: avformat: Make avformat_free_context handle NULL [14:42] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:7dba57d1fa6b: Merge commit '0db6bbb24c1a9f6b5527f460361a5a4132248eee' [14:43] <ubitux> the 7 lines of vpx hacks could be replaced with a check_pkg_config "vpx >= 0.9.7" or something [14:43] <Daemon404> yea [14:43] <ubitux> good luck with that [14:43] <Daemon404> sure ill send it to libav [14:43] <Daemon404> where carl isnt a huge asswad [14:43] <ubitux> right now i'm suppose to write a code that use pkg config for x264 and fallbacks on the hack [14:44] <ubitux> i'm actually not happy with that so i'll probably won't do it [14:44] <Daemon404> meanwhile at libav, pkgconfig for months [14:44] <Daemon404> :/ [14:45] Action: Daemon404 seems to remember some ffmpeg guys sayign somethign akin to "no patch is going to be held up by one guy over at ffmpeg" [14:45] <Daemon404> carl hath become diego [14:45] <nevcairiel> no, he is actually worse [14:45] <Daemon404> thats true. [14:49] <J_Darnley> What was his objection to pkg-config? [14:49] <J_Darnley> I mean I dislike it for depending on glib [14:49] <ubitux> "it breaks current configuration" [14:50] <iive> what does it break? [14:50] <ubitux> it breaks if you don't have pkg-config [14:50] <nevcairiel> presumably people who have giant hacks on their system to accomodate the old broken mess [14:51] <ubitux> and i maintain the idea that documenting a behaviour change is better than adding a shitload of fallback crap in the configure [14:51] <Daemon404> any sane person would [14:51] <nevcairiel> why are we listening to him again [14:51] <Daemon404> nobody ever said why. [14:51] <ubitux> it's like, "ok let's use pkg-config, but still specify -lx264 just in case" [14:52] <iive> the person who wrote most of configure is Mans, and he hated pkg-config [14:52] <iive> that's why it is almost never used... [14:54] <Daemon404> that's a load of shit, iive [14:54] <iive> you can blame him for the level of obfuscation in the configure [14:54] <Daemon404> and also a terrible reason even if it were true [14:55] <Daemon404> (also irrelevant to teh situation at hand) [14:55] <iive> it makes the code a lot smaller, but it is insanely hard to customize tests. [14:55] <Daemon404> not really true if youve actually looked [14:55] <Daemon404> i can spout cargo culted opinions from 8 years ago too. [14:56] <iive> you don't even know what that phrase means. [14:56] <wm4> time to overrule cehoyos? [14:56] <Daemon404> iive, perhaps you do not [14:56] <Daemon404> also, as i said, entirely irrelevant to the problem at hand: [14:56] <Daemon404> 1) a proper fix is there [14:56] <Daemon404> 2) carl wont let it in [14:57] <wm4> if, say, at least 5 devs agree to a patch replacing manual checks with pkg-config, and only cehoyos disagrees... [14:57] <wm4> does he have some magical veto-right? [14:57] <Daemon404> more than 5 have already agreed [14:58] <iive> well, the best solution would be to try pkg-config first and if it is not available, fall back to the current method [14:58] <iive> how easy is to do that? [14:58] <wm4> isn't this already done for some things [14:58] <Daemon404> iirc ubitux sent that already [14:58] <iive> then, there is no breakage. [14:59] <wm4> of course not [14:59] <Daemon404> carl had some insane argument along the lines of [14:59] <wm4> the problem is that cehoyos is a blockhead [14:59] <Daemon404> if you have two x264 installs [14:59] <Daemon404> and one is pkg config and one is not [14:59] <Daemon404> itll break for you by picking up the wrong one [14:59] <Daemon404> or something [14:59] <Daemon404> (dotn quote me) [14:59] <nevcairiel> define "wrong one" [14:59] <nevcairiel> ie. "whichever he wants it to use" [15:08] <iive> hum... [15:08] <iive> i don't see it. I found an x264 one [15:09] <iive> but the fallback is actually workaround, by asking the user to specify --pkg-config=true and setting the libraries manually [15:09] <iive> that's not falling back... that's hack. [15:10] <iive> (aug 28) [15:10] <nevcairiel> its fine to simply document that we require pkg-config, we already do for a bunch of other libs [15:11] <ubitux> iive: yes it's a hack to replace another hack [15:11] <ubitux> if you really don't want to use pkg-confiug [15:13] <iive> i don't get it... if you don't want to use pkg-config , shouldn't you specify it to `false` to emulate its absence? [15:13] <benoit-> ubitux: maybe you could use --enable-libx264 vs --enable-libx264-pkg-config, or --enable-libx264 vs --enable-libx264-no-pkg-config? [15:14] <benoit-> iive: I believe that its absence would make configure fail [15:14] <benoit-> or other things would have to be done to handle this too, and that would affect other libs configuration [15:14] <iive> well, the idea of fallback is that, you first check if pkg-config is available/working [15:14] <ubitux> it's stupid to duplicate that logic [15:15] <iive> if it is, then use it. [15:15] <ubitux> i just don't want to keep the hack [15:15] <benoit-> ubitux: yes [15:15] <ubitux> it's broken [15:15] <ubitux> and if we keep it people will use it [15:15] <iive> if it is not, then fallback to the current method. [15:15] <benoit-> iive: that was proposed, IIRC [15:15] <iive> was it implemented? [15:15] <benoit-> only it was not sufficient [15:16] <ubitux> i don't want to have the "-lx264" written in configure [15:16] <benoit-> as it would not allow to discriminate "easily" between x264 pkg-config vs without [15:16] <ubitux> it's the task of pkg-config [15:16] <Daemon404> not using pkg-config to begin with was wrong [15:16] <benoit-> ubitux: the point of iive is that it's its role when it's present [15:16] <Daemon404> benoit-, the library defines pkg-config as teh way to use it. [15:17] <ubitux> pkg-config is the only official supported way by libx264 period. [15:17] <Daemon404> if it's not present, anythign else is a hakc [15:17] <Daemon404> what ubitux said. [15:18] <ubitux> it's just like if someone had a project using cmake but prefered to gcc src/*.c [15:18] <ubitux> cmake might sucks but that's how it is [15:18] <ubitux> deal with it [15:18] <Daemon404> except pkg-config is actually useful. [15:18] <iive> ubitux: you see, it is not about code cleanness and purity [15:19] <iive> it is about users, and if users need this mess, you have to deliver. [15:19] <ubitux> they don't [15:19] <Daemon404> every user i have talked to massively prefrred pkg config [15:19] <ubitux> they're using the hack because that's the only solution available currently [15:19] <Daemon404> also that [15:19] <ubitux> we have tons of tickets in the trac complaining about that [15:19] <iive> think of it this way. you first have to introduce pkg-config an provide fallback [15:20] <iive> then a while, remove the fallback [15:20] <benoit-> just an historic point, though... when x264 support reached ffmpeg tree back in 2005, there was no pkg-config support [15:20] <iive> then after a while, remove the fallback [15:20] <benoit-> ubitux: and you already know I'd prefer having pkg-config in place [15:21] <j-b> Are you still discussing pkg-config? [15:21] <iive> or, if you demand code purity. rewrite pkg-config without glib [15:21] <ubitux> alright, so i'm going to had more code in the configure [15:21] <j-b> iive: there are already 20x of such projects [15:21] <ubitux> which makes sure the code is disabled at next bump [15:21] <cone-104> ffmpeg.git 03Luca Barbato 07master:47e3e1097e09: cmdutils: Use the correct guard [15:21] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:1eda312aca27: Merge commit '47e3e1097e09da7f973908e09df15e3bb7c5e1f4' [15:21] <j-b> Sorry guys, but pkg-config troll was fought and lost 8 years ago [15:21] <benoit-> j-b: no, they (we?) discuss the x264 configuration in ffmepg [15:21] <ubitux> j-b: we all agree with the pkg-config stuff here [15:21] <j-b> Use pkg-config. [15:22] <j-b> drop everything else. [15:22] <ubitux> j-b: we're just trying to workaround cehoyos [15:22] <j-b> rm -rf cehoyos. [15:22] <Daemon404> ^ +1 [15:22] <j-b> fuck him [15:22] <Daemon404> yes. [15:22] <j-b> http://sourceforge.net/projects/pkgconfiglite/ [15:22] <iive> j-b: is any of these rewrites good enough to put inside ffmpeg? [15:22] <ubitux> j-b: i'm actually willing to copy that .c in ffmpeg [15:22] <ubitux> instead of keeping that shitty hack [15:22] <Daemon404> j-b, freebsd also uses pkgconf [15:23] <Daemon404> iive, that's teh distro's job [15:23] <Daemon404> ln -s etc [15:23] <Daemon404> ffmepg is not mplayer [15:23] <Daemon404> it shouldnt be inside ffmpeg. [15:23] <j-b> And we come back to what I've been saying in years: SPLIT YOUR FUCKING REPOS [15:24] Action: j-b is now an old troll [15:24] <ubitux> let's not change subject :P [15:24] <iive> ubitux: do the ugly hack and be done with it. [15:25] <benoit-> Sorry j-b, but splitting repo troll was fought and lost years ago ;) [15:25] <ubitux> iive: no it sucks [15:25] <ubitux> if you want to, do it [15:25] <iive> ubitux: trust me, it is the better decision. [15:25] <Daemon404> iive? write code? LOL [15:25] <ubitux> well, anything can submit that patch, i don't care if it's iive or someone else [15:25] <j-b> benoit-: :) [15:26] <ubitux> i don't want to do deal with that anymore [15:26] <Daemon404> ubitux, if carl is involved i keep a distance. [15:26] <Daemon404> as do most i imagine [15:26] <benoit-> ubitux: can you ping your last version so that it's resurrected from where it is, please? [15:26] <ubitux> i can rebase it for you if you want [15:26] <ubitux> just a moment [15:27] <benoit-> ubitux: merci [15:28] <ubitux> benoit-: might actually take me some time [15:28] <ubitux> benoit-: http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2014-August/161979.html [15:28] <ubitux> i need to split and stuff [15:28] <ubitux> the idea is to make it more obvious that it's using pkg-config etc [15:29] <ubitux> i could probably start applying the other chunks [15:30] <ubitux> that was my latest iteration [15:33] <cone-104> ffmpeg.git 03Uwe L. Korn 07master:324b23dde1bc: rtmpproto: Add function to read a number response [15:33] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:6cf9d02cea5e: Merge commit '324b23dde1bc8638959eb32419c95a93906db272' [15:39] <cone-104> ffmpeg.git 03Uwe L. Korn 07master:e65c776d18dc: rtmpproto: Add getStreamLength call to query duration [15:39] <cone-104> ffmpeg.git 03Michael Niedermayer 07master:3dda0bbe4250: Merge commit 'e65c776d18dc14df8a279e017760862f9fc8763b' [16:05] <cone-104> ffmpeg.git 03Benoit Fouet 07master:ac1594bc179a: gitignore: add test_copy.ffmeta [16:23] <cone-104> ffmpeg.git 03Benoit Fouet 07master:242f8bb3a8a1: avformat/id3v2: support buggy id3v2.3 tag length in id3v2.4 [16:36] <benoit-> ubitux: just sent a patch with a "minimal" workaround for x264 [17:24] <ubitux> benoit-: missing versions check [17:25] <ubitux> but it's insane anyway [17:25] <ubitux> ah yeah you actually have version check [17:25] <ubitux> but not with pkg-config [17:25] <ubitux> derp, that's insane [17:25] <Daemon404> an insane patch to satisfy an insane person [18:51] <hawken> Got an answer from Jochen Britz, "My code is under the same license as FFmpeg itself." [18:51] <hawken> Referred me to this: https://groups.google.com/forum/#!forum/h264mvc [18:52] <hawken> Pointed me to koda's work [18:53] <myra> Can someone explain me what is fate folder ? What is it used for ? [18:53] <wm4> myra: FATE is the ffmpeg automatic test suite [18:54] <wm4> the scripts are in the ffmpeg git repo itself, the associated test data is on a separate rsync server [18:54] <wm4> there's some documentation about it [18:56] <myra> @wm4 : I'll look into it . Thank you !! [19:07] <Daemon404> dang kids and their out of place @s... [20:06] <hima> here http://fpaste.org/142955/35686551/ here we define the function and the argument is char ** type [20:07] <hima> ubitux : but when a value is passed its char* type is not it? [20:07] <hima> i am getting this error in my code [20:07] <hima> where i see it runs fine in vplayerdec.c [20:08] <hima> ubitux [20:08] <ubitux> that looks correct so far [20:08] <ubitux> p is a char* [20:09] <ubitux> and you send the address of p to the function [20:09] <ubitux> so it becomes a pointer on char*, thus a char** [20:09] <hima> yeah i know [20:09] <hima> http://fpaste.org/142958/69378141/ [20:10] <hima> i get this error again and again [20:10] <ubitux> can you show me the complete code? [20:10] <hima> yeah just a second [20:12] <hima> http://fpaste.org/142963/35695271/ [20:12] <hima> here [20:14] <ubitux> it's about the const [20:14] <ubitux> i think you have char *const *buf or something [20:14] <ubitux> or just char **buf [20:15] <hima> const [20:15] <hima> ohh i okkk [20:15] <ubitux> or you can just drop the const [20:16] <hima> i added the const corrects it..stupid mistake :| [20:28] <akira4> ubitux, I've attached the revised patch to the mailing list thread. I am getting some error when trying to rsync the fate-suite so couldn't add the stl file [20:28] <ubitux> akira4: i saw that [20:28] <ubitux> you can workaround it actually [20:28] <akira4> really? [20:28] <ubitux> akira4: try to mkdir -p /tmp/fake-fate-suite/sub [20:28] <ubitux> copy the sample in the directory [20:29] <ubitux> then make fate-sub-stl GEN=1 SAMPLES=/tmp/fake-fate-suite/sub [20:29] <ubitux> (assuming you created that rule [20:29] <ubitux> /tmp or wherever you want actually [20:29] <ubitux> you get the idea [20:29] <akira4> got it. [20:31] <akira4> is the identation and style problem solved or should I do it again? [20:32] <ubitux> i'll have a look later [20:32] <akira4> sure. [20:33] <ubitux> akira4: you still lack spaces around operators [20:33] <ubitux> <space>+=<space> [20:33] <ubitux> <space>=<space> ... [20:33] <ubitux> etc [20:33] <ubitux> while<space>( [20:34] <ubitux> <space>==<space> [20:34] <akira4> is there some way I could run a test for checking this? [20:34] <ubitux> not sure if patcheck does that [20:34] <ubitux> the indent in the "if (pts_start != AV_NOPTS_VALUE) {" block is wrong [20:35] <akira4> okay. I'll redo it. [20:35] <ubitux> len=0 and len>0 are wrong as well [20:35] <akira4> you wouldnt mind if it gets a little delayed right? My midterms are going on [20:36] <ubitux> i have no problem with that [20:36] <akira4> alright. I'll fix it. thanks [20:42] <cone-574> ffmpeg.git 03Omer Osman 07master:b01a2204b5cf: libfdk-aacdec: Enable Decoder Downmix including Downmix Metadata Support [20:42] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:5a1aa559054b: Merge commit 'b01a2204b5cff7bb920f42fda1bb0103f450fe93' [20:42] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:e18fc35a07f4: avcodec/libfdk-aacdec: use av_feeep() to avoid leaving stale pointers in memory [20:52] <cone-574> ffmpeg.git 03Omer Osman 07master:66e9f8395362: libfdk-aacdec: Enable Dynamic Range Control Metadata Support [20:52] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:45fed258ee6a: Merge commit '66e9f839536238945fbfe9d2041b6891cb150e45' [21:21] <cone-574> ffmpeg.git 03Vittorio Giovara 07master:bae557edcea9: dump: display codec tags when available [21:21] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:0b0d47e4b21e: Merge commit 'bae557edcea996328e8ff48b9e074a4210f7e6b4' [22:17] <cone-574> ffmpeg.git 03Mika Raento 07master:75c8d7c2b497: hlsenc.c, segment.c: propagate defaults to mpegts [22:57] <hima> can anyone help me with this error http://fpaste.org/143016/35793831/ in this code http://fpaste.org/143015/13579342/ (kate format demuxer) [23:00] <ubitux> hima: pts are 64 bits [23:02] <hima> ohh return type ok changing that [23:02] <J_Darnley> And if you haven't seen them yet, use int64_t or uint64_t [23:03] <J_Darnley> instead of long. [23:04] <hima> yeah done thankyou [23:07] <ubitux> hima: get_pts_start and get_pts_end function looks very much alike [23:07] <ubitux> you should refactor them [23:08] <hima> okay doing it. i calculated duration in read_header so okay will make that change [23:09] <ubitux> what if the timestamps are not on the same line as the event [23:09] <ubitux> does it work? [23:09] <ubitux> your probe function use strcmp functions for 1 char, just compare directy with the character instead [23:09] <hima> yes [23:09] <hima> find=strstr(ptr,"event"); [23:09] <hima> while(strcmp(find,"\n")==0 || strcmp(find," ")==0) [23:09] <hima> find++; [23:10] <ubitux> ok [23:10] <hima> its a while statement and the pointer goes on increasing [23:10] <hima> while it gets a \n or a " " [23:11] <ubitux> but you get only one line at once [23:11] <ubitux> and this is the probing [23:11] <ubitux> not the read_header function [23:11] <hima> yes so suppose its event \n \n \n x:y:z -> p:q:r [23:12] <hima> so it will run untill it finds the timestamp [23:13] <hima> and the sample for which u asked me to make it for minimal.kate there was no such ambiguity there [23:16] <ubitux> i think that code might crash if you add a line return after event { [23:16] <ubitux> or just not work [23:16] <hima> there is no { after event [23:17] <hima> http://fpaste.org/143022/41358062/ [23:17] <ubitux> ok [23:17] <hima> umm the sample did not have a { after event [23:17] <ubitux> what if you add a line break after the first "event"? [23:17] <hima> hm [23:18] <hima> line break you mean \n ? [23:19] <hima> find=strstr(ptr,"event"); here now find has the rest of the string after event [23:19] <hima> while(strcmp(find,"\n")==0 || strcmp(find," ")==0) [23:19] <hima> find++; [23:20] <ubitux> events can have { } [23:20] <hima> this takes care that we skip the line breaks and spaces [23:20] <hima> you mean event{ and then timestamp? [23:20] <ubitux> http://git.xiph.org/?p=users/oggk/kate.git;a=blob;f=examples/kate/z.kate;hb… [23:22] <hima> sorry i did not see that example the one you gave me did not have [23:22] <hima> http://www.filewatcher.com/p/libkate_0.3.7.orig.tar.gz.728239/libkate-0.3.7… [23:22] <hima> i will make the change then [23:22] <ubitux> look at http://git.xiph.org/?p=users/oggk/kate.git;a=tree;f=examples/kate;hb=HEAD [23:23] <ubitux> i'm just asking to parse the events, and get the text and timestamps out of them [23:23] <ubitux> so you ignore the rest [23:23] <ubitux> but they must all be supported [23:23] <wm4> oh god [23:23] <wm4> thus format looks almost as bad as ASS [23:23] <wm4> *this [23:23] <ubitux> wm4: haha yeah kinda [23:23] <ubitux> wm4: actually, more painful to parse [23:23] <hima> yeah i get it i will make the change for including the { after event then [23:25] <J_Darnley> ubitux: of course it is, it is ogg! [23:25] <hima> ubitux after including that should i submit this patch in the mailing list? [23:29] <ubitux> you can do that anytime [23:29] <ubitux> but i believe there is still work to do [23:30] <hima> yeah please tell me. [23:30] <ubitux> then start fixing what we said here [23:30] <ubitux> we'll iterate from here [23:31] <hima> okay i am fixing the existence of { after event first [23:31] <hima> ubitux: bothering you again about the application [23:32] <hima> are we supposed to propose any idea ? [23:32] <hima> as there is a section in the format of the application for tasks and you timeline [23:32] <hima> as you plan to do it [23:32] <ubitux> there are a lot of work on subtitles stuff [23:32] <ubitux> the final goal is to get a proper (at least text) subtitles support in ffmpeg [23:33] <ubitux> and that includes having them in libavfilter for hardsubbing [23:34] <hima> okay so this would be our goal if we qualify for OPW? [23:34] <ubitux> yes [23:36] <hima> ok thanks [23:40] <hima> ubitux in this example there is almost a nested loop of { and its not always "event" do i include that kind. or just a simple kind where there would be event { ? [23:41] <hima> http://git.xiph.org/?p=users/oggk/kate.git;a=blob;f=examples/kate/kate.kate… [23:41] <ubitux> i don't see anything textual you could extract here [23:41] <ubitux> it should not break as invalid data [23:42] <ubitux> but no recognized event should come out of it [23:45] <hima> hm i want to know like in the minimal.kate there was only event, but inthese examples there is motions {}curve{} points{} in a kind of nested fashion [23:45] <ubitux> skip them, properly. [23:45] <hima> okay [00:00] --- Sat Oct 18 2014

1 0

ffmpeg.log.20141017
by burek 18 Oct '14

18 Oct '14

[00:00] <llogan> (...not counting devices or decoders that may not be enabled if missing --enable-gpl) [00:00] <akiselev> If you distribute ffmpeg/libav* binaries it can get a bit hairy. Look here http://www.gnu.org/licenses/gpl-faq.html#LGPLStaticVsDynamic for some info on LGPL (ffmpeg's general license) but watch out for --enable-gpl and --enable-nonfree compile config flags [00:01] <pzich> if I ffprobe one of my videos I see "Duration: 00:00:46.03, start: 0.033333", is there something I can pass when encoding to reset the start to 0? [00:02] <sourpulse> llogan, akiselev: Thank you, that's how I was thinking too. The commercial product won't distribute FFmpeg, it will simple mention that utilities such as FFmpeg, x264, etc are available elsewhere. [00:04] <sourpulse> s/simple/simply [00:04] <llogan> sounds fine to me [00:05] <akiselev> pzich, you can use the a/setpts video filter: -vf setpts=PTS-STARTPTS [00:05] <akiselev> sourpulse, you can distribute LGPL static/dynamic libs as long as you "make the source code available" [00:05] <akiselev> *you can also [00:06] <sourpulse> llogan, akiselev: That's reassuring. Does FFmpeg has an official contact person for such questions? [00:06] <akiselev> Which means either responding to requests for source code of the LGPL parts you used to compile the binaries, or putting them online (most companies just bury zip files on some back page) [00:06] <sourpulse> The commercial program won't be using libs. It communicates merely by command-line pipe. [00:07] <akiselev> sourpulse, doesn't matter to the LGPL, they're both binaries [00:07] <akiselev> But as long as you don't distribute it doesnt matter [00:07] <llogan> sourpulse: no contact person, but you can ask the ffmpeg-user mailing list. [00:07] <pzich> akiselev: hmm, it still shows the same start time after I do that [00:07] <akiselev> What do you mean it shows the start time? [00:08] <sourpulse> Ah, the mailing list sounds like a good idea. Thanks. [00:08] <llogan> or refer to http://ffmpeg.org/legal.html [00:08] <akiselev> which basically says "you're on your own, get a lawyer" [00:08] <llogan> or ask a lawyer who knows about this stuff [00:08] <pzich> akiselev: if I export a video with that and run ffprobe on it, I still see "Duration: 00:00:46.07, start: 0.033333" [00:08] <sourpulse> I've read the FFmpeg legal page and didn't see any contact info. [00:09] <llogan> i meant as general info, not as a means of contact [00:11] <sourpulse> llogan: Ok, understood. Yes, asking the lawyer is usually wise. [01:42] <sourpulse> "ffplay -v verbose -f lavfi -i testsrc" --- That ffplay command auto-inserts a scaler conversion to yuv420p, causing color subsampling artifacts along the perimeter of the circle. Is there any way to avoid that conversion and get a sharper image? ffplay output: http://ffmpeg.gusari.org/viewtopic.php?f=26&t=1722 [01:46] <llogan> ffplay -v verbose -f lavfi -i testsrc,boxblur [02:20] <[2]Ian> Can someone tell me how I would modify this command to properly use the adelay filter to delay the second input by 10 seconds? [02:20] <[2]Ian> ffmpeg -i 1.opus -i 2.opus -filter_complex amix=inputs=2:duration=longesst conference.opus [02:22] <sourpulse> llogan; ??? That's still subsampling, and now blurry. I'm hoping for sharper. [02:23] <[2]Ian> misspelling of longest aside... [02:26] <llogan> sourpulse: it was a bad joke [02:27] <sourpulse> Oh ok! I get it now. [02:33] <llogan> [2]Ian: [0:a]adelay=1000[a0];[a0][1:a]am(ix|erge)...? [02:44] <[2]Ian> thanks [07:05] <Nosomy> vp9 only have bitrate ratecontrol method? [09:09] <Baked_Cake> ne1 know the correct refresh rate multiples for monitors [09:09] <Baked_Cake> i think they use upto 3 decimal points [09:09] <Baked_Cake> im trying to use 75hz [09:09] <Baked_Cake> khv* [09:09] <Baked_Cake> whatevs [11:01] <flavioberetti> hi. i want to pass the 'date' metadata to the libvorbis audio encoder, where do i find documentation on how the fields are named in ffmpeg? [11:01] <flavioberetti> ffmpeg -y -i Remark_Music_remix.wav -acodec libvorbis -metadata artist="Juha-Matti Hilpinen (AMJ)" -metadata genre="Chiptune" -metadata title="Remark Music (remix)" -metadata album="HVSC" -metadata date="1993 Side B/Topaz Beerline" -aq 8 ../C64Music-ogg/MUSICIANS/A/AMJ/Remark_Music_remix.ogg [11:02] <flavioberetti> gives me Unrecognized option 'metadata date="1993 Side B/Topaz Beerline"'. [11:02] <flavioberetti> Error splitting the argument list: Option not found [11:02] <flavioberetti> however, man oggenc describes the option as -d date, --date date [11:03] <flavioberetti> thinking about it, i can just use oggenc anyways... [11:03] <flavioberetti> still can i do it with ffmpeg? [11:11] <relaxed> flavioberetti: single quotes [11:11] <flavioberetti> relaxed: for the other keys it works with double quotes.. [11:12] <relaxed> oh, well, see if it works [11:13] <flavioberetti> relaxed: nope, get Unrecognized option 'metadata date='1993 Side B/Topaz Beerline''. [11:14] <flavioberetti> relaxed: i got it working with oggenc now, thanks for helping! :) [11:17] <c_14> -metadata date="1193 topaz/fooobar" <- this works for me [11:37] <rahul__> hii all there is any way to stream my video continouesly using ffplay [11:37] <rahul__> i am trying udp with ffplay to stream frames from server [11:37] <rahul__> i need to give continoues mouse clicks to update it [11:37] <rahul__> is there any way to do it fluently? [12:32] <Diogo> hi i use this command ffmpeg -threads 1 -i alice.avi -vcodec libx264 -crf 15.0 -acodec libfdk_aac -ab 64 3194 root 20 0 5932 632 532 S 0.0 0.0 0:00.00 /sbin/getty 38400 tty4 [12:32] <Diogo> k -ar 48000 -ac 2 -f mp4 alice.mp4 [12:33] <Diogo> i want that ffmpeg use only threads = 1 [12:34] <Diogo> http://postimg.org/image/lnvwow4bx/ ?? [12:38] <Diogo> http://pastebin.com/GSmWpBmz [16:17] <nick112234> hi everyone Q coming: [16:17] <nick112234> ffmpeg -i sbs.mp4 -vf stereo3d -vf scale=w=iw*2:h=ih -acodec copy -threads 10 -b:v 10000k -preset ultrafast -vcodec libx264 out.avi [16:18] <nick112234> does the 3d filter merging but not the rescaling... why is that? thanks [16:19] <Mavrik> you should probably do a single filter chain [16:19] <Mavrik> -vf stereo3d,scale=w=iw*2:h=ih [16:24] <nick112234> ok my mistake; -vf stereo3d,scale=w=iw*2:h=ih does the 3D filter but not the rescaling !! [16:27] <spaam> !!! [16:28] <nick112234> ok it does something as the output is said to be 1920x1080 (and 960x1080 if scale not present....) still i get a 960x1080 looking output with black left & right of the video, any clue? [16:32] <nick112234> anyone? thanks [16:35] <nick112234> anyone avail to help tuning a cmd line? thanks [16:36] <nick112234> a free beer is at stake! ;-) [16:43] <ChocolateArmpits> nick112234, Did you correct the aspect? [16:43] <ChocolateArmpits> setdar=dar=16/9 [16:43] <ChocolateArmpits> At the end of the filter chain [16:45] <ChocolateArmpits> It would also be good if you could ffprobe the output video file to be sure the combined picture has a resolution of 1920 by 1080 [16:49] <nick112234> no, i took it for granted the aspect ratio would be what ouput i required, but it truely seems the aspect ratio is kept so i got black adds on left & right.... [16:53] <nick112234> ok so the question is how do i set up the right cmd line "-vf scale=w=iw*2:ih" to get a 2x width same height ? [16:55] <nick112234> having the DAR or SAR whatever.... match the new 2*w:h format ? [16:58] <albator> hello [16:58] <albator> i m trying to restream an udp stream, but the audio is out of sync [16:58] <albator> so I tried the ifoffset method, but I get input/output error [17:00] <albator> http://pastebin.com/LLRwR6aL [17:01] <albator> I remember it use to work before.. [17:02] <albator> and this is the stream Stream #0:0[0x32a]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1440x1080 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [17:02] <albator> Stream #0:1[0x334](fra): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 123 kb/s [17:12] <nick112234> and the answer was : ffmpeg -i sbs.mp4 -vf stereo3d,scale=w=iw*2:ih,setsar=sar*2 -acodec copy -threads 10 -b:v 10000k -preset ultrafast -vcodec libx264 out.avi [17:12] <nick112234> thanks & bye [17:16] <nick112234> let's go for the last Q of the day... [17:17] <nick112234> what cmd line would you suggest to make a very ultra fast conversion from any format to a ??? format x264? xvid? (popular one easy to read on any platform) with good quality, big file size is ok (not huge) and ultra fast transcoding ?? thanks [17:18] <nick112234> what is the fastest and file size reasonnable encoding codec & option in ffmpeg ? [18:05] <kaotiko> hi [19:12] <Matina25> Hello, is anyone here with experience to tell me if --enable-librtmp is better than the native ffmpeg rtmp? [19:12] <Matina25> and what is the differences between these two [20:34] <albator> i d enable it yea [20:34] <albator> had problem with ustream for instance without [20:47] <Matina25> Another question, when i compile ffmpeg statically [20:47] <Matina25> it cant process domain names [20:48] <Matina25> only ips [20:48] <Matina25> :/ [22:24] <bofh> Hi all! Given an audio file (MP3) of length 30 seconds I need to cut off first 15 seconds, pad it with silence, then for next 5 seconds increase volume by 20% and cut off the rest 10 seconds to the end of the file. So the resulting mp3 will contain 15 seconds of silence, then 5 seconds of audio with increased volume, and of length 20 seconds in total. Is it possible to do with FFMPEG? [22:38] <kepstin-laptop> sure, would need to use -filter_complex with an avealsrc to get the silence, a concat, and some seeking on the file input or a trim filter. [22:39] <bofh> kepstin-laptop: so it's feasible by using a bunch of command-line arguments, or I have to chain/pipe multiple invocations of ffmpeg? [22:40] <kepstin-laptop> you can do it with a complex filter chain in a single ffmpeg invocation. [22:41] <bofh> nice, could you please suggest some how-to/guide/article to read? [22:41] <kepstin-laptop> http://www.ffmpeg.org/ffmpeg-filters.html is probably not a bad place to start [22:42] <bofh> kepstin-laptop: great, cam ffmpeg also mix several files? [22:42] <kepstin-laptop> define mix [22:42] <kepstin-laptop> there's several ways in which multiple files could be combined into one file [22:43] <bofh> kepstin-laptop: just "join" several files, combine/merge [22:44] <bofh> does it make sense? [22:44] <kepstin-laptop> one after the other? both playing at the same time? both syncronized, but the player can select between them? [22:44] <kepstin-laptop> you have to be more specific :) [22:44] <bofh> kepstin-laptop: well, the result would be the single MP3 file [22:45] <bofh> with all input streams combined, and the length of the file is the length of the longest input file [22:45] <kepstin-laptop> ok, so the audio from all streams played at the same time, mixed into a single audio stream. [22:46] <bofh> yep [22:46] <kepstin-laptop> that would be the 'amix' filter [22:47] <bofh> so if I have, say, 5 input files, and each of them needs to be transformed somehow, and then "merged" into a single file - I need to invoke 5 ffmpeg processes to prepare all files, and the invoke another ffmpeg process to apply the 'amix' filter? [22:47] <kepstin-laptop> you could do it all in one complex filter chain in one process if you want to. [22:48] <kepstin-laptop> command line gets more and more complex, obviously ;) [22:48] <bofh> that would be perfect [22:48] <bofh> kepstin-laptop: as soon as I'll generate the command-line, I don't care how complex it could be ) [22:49] <DarKnesS_WolF> a noob question, if i have a nvidia GT0640 with like 300 Core and wanna convert a video using that gpu, i have vdpau whatever is taht installed and i have -hwaccel auto enabled in the ffmeg command but i see only CPU us being used , what i am missing ? [22:50] <kepstin-laptop> I don't think ffmpeg currently supports doing any video encoding via gpu, only decoding. could be wrong. [22:50] <DarKnesS_WolF> decoding in that sense ? [22:50] <ubitux> long story short, gpus are useless [22:53] <DarKnesS_WolF> mmm [22:54] <kepstin-laptop> imo, the only reason to want to use a fixed-function video encoder on a graphics card is that you want to do live screencasting of something that's cpu-intensive, and can't use an external encoder box. [22:56] <kepstin-laptop> in thoery, a gpu could be used to accellerate portions of the code of something like x264, and x264 does have some opencl support for that, but I dunno if it really gains you much. [22:56] <kepstin-laptop> apparently it's buggy, reduces quality, and may be slower than cpu-only encoding atm :) [22:59] <DarKnesS_WolF> wow [22:59] <DarKnesS_WolF> lol [22:59] <DarKnesS_WolF> i have a big video and wanted to use GPU to convert it [22:59] <DarKnesS_WolF> no other tools ? [23:00] <kepstin-laptop> just use x264, you'll be happier with the result. [23:00] <DarKnesS_WolF> needed to try x265 [23:01] <kepstin-laptop> I haven't personally tried it, but apparently hevc/h.265 encoding is hella slow compared to h.264 if you actually want the result to be better quality [23:02] <JEEB> I'd say for most use cases HEVC encoding is not useful right now [23:02] <JEEB> for very low bit rates it can give a better result if you put in a *lot* of time [23:04] <JEEB> well, the result will kind of suck with both, but HEVC will suck a bit less [23:04] <JEEB> but it's still not good enough with psychovisual optimizations, so if you are trying to aim for good compression and a quality level higher than "lol low" [23:04] <JEEB> then x264 still has the upper hand [23:26] <DarKnesS_WolF> thx guys :) [00:00] --- Sat Oct 18 2014

1 0

ffmpeg-devel.log.20141016
by burek 17 Oct '14

17 Oct '14

[00:02] <cone-712> ffmpeg.git 03Martin Storsjö 07master:ced7238cd01c: rtpdec_hevc: Use av_realloc instead of av_malloc+memcpy [00:02] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:f09162c06bba: Merge commit 'ced7238cd01cc2199acf9225305628641a27c1d7' [00:09] <cone-712> ffmpeg.git 03Martin Storsjö 07master:79dd756e143a: rtmpproto: Fix a typo [00:09] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:917f86f47908: Merge commit '79dd756e143a54efc25d64e90505f0dca6cbc4ec' [01:04] <randomdude123> guys. [01:04] <Compn> yes. [01:04] <randomdude123> can anybody comment on this weird requirement by RCS spec: [01:04] <randomdude123> http://stackoverflow.com/questions/26393553/h264-rtp-and-packetization-mode… [01:04] <randomdude123> this is about h264 and STAP-A in RTP [01:06] <Compn> i see [01:06] <Compn> have to ask people who work on rtp h264 code [01:06] <Compn> which does not include me :) [01:06] <Compn> wait around for someone to asnwer [01:06] <Compn> or ask on list [01:06] <Daemon404> probably will be a bit [01:07] <Daemon404> most are european and asleep [01:07] <Daemon404> maybe. [01:43] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:2f7bd04afb29: avcodec/avcodec: fix non doxy comment [01:43] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:7b6a97edd1b5: avcodec/avcodec: more verbose documentation for time_base [01:43] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:46f52274f38d: avformat/oggparsedirac: use AVCodecContext.framerate [01:49] <rcombs> https://www.dropbox.com/s/bwv99lz0xtbckpn/Maken-Ki%21%20-%20S01E02.mkv?dl=0 <-- apparently ffmpeg can't seek in this file [01:50] <rcombs> `ffmpeg -ss 922 -i 'Maken-Ki! - S01E02.mkv' -c copy -map 0 -y out.mkv` gives output starting at the beginning of the file [02:11] <randomdude123> i'll hang around, untill they wake up :) would like to know what ppl familiar with that stuff can say about the spec [02:13] <wm4> rcombs: seeks perfectly well in mpv, even with the lavf demuxer [02:13] <rcombs> wm4: yeah, I'm confused about what's going wrong [02:14] <wm4> rcombs: so I conclude it's some kind of obscure ffmpeg.c problem [02:14] <rcombs> things go wrong when using either -copyts or -c copy [02:14] <wm4> ah [02:14] <rcombs> seems likely (which is why I filed the component undetermined, instead of avformat) [02:47] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:004f1c6cf13a: avformat/utils: Try to correct the codec_framerate in ff_compute_frame_duration() for the encoding case [02:47] <cone-712> ffmpeg.git 03Michael Niedermayer 07master:d267a0f8c1a8: avformat/util: Simplify av_guess_frame_rate() by using AVCodecContext.framerate [04:13] <rcombs> ^ just realized that didn't actually submit due to the description limit [04:22] <cone-712> ffmpeg.git 03Di Wu 07master:0e406aba14a8: mpegts: add the judgement if a new program is created successfully [09:20] <sophiejjj> reynaldo: there? [12:08] <cone-322> ffmpeg.git 03Carl Eugen Hoyos 07master:392b4b663c45: Fix initialisers in dct-test.c. [12:38] <cone-322> ffmpeg.git 03Michael Lynch 07master:460b509a34fb: rtsp: Check a memory allocation [12:39] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:293a8e426fc0: Merge commit '460b509a34fb5fad3bedac8429f53594d3923ea8' [12:54] <cone-322> ffmpeg.git 03Mika Raento 07master:979932378ae3: ismindex: use tfhd default duration if no sample duration [12:54] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:a9b613b60e51: Merge commit '979932378ae3fbf452e312eb759cc7ce175f78de' [13:29] <BBB> iive: https://wiki.videolan.org/X264_asm_intro/ [13:36] <iive> BBB: i think kierank already pointed that out. [13:45] <ubitux> kierank: aren't you looking for sine or aevalsrc audio filters? [13:46] <ubitux> won't be exactly accurate but that can be done with aevalsrc [13:46] <BBB> oh, sorry, missed that [13:47] <kierank> ubitux: it can be done but it's not guaranteed to be frame accurate [13:47] <kierank> which is the point of a lipsync test [13:51] <Daemon404> why would you use lavfi at all for something so simple? [13:55] <ubitux> kierank: ffplay -f lavfi "aevalsrc='st(1,mod(t,2));sin(440*2*PI*ld(1))*exp(-5*ld(1))|st(1,mod(t+1,2));sin(440*2*PI*ld(1))*exp(-5*ld(1))'" [13:55] <ubitux> (i know that's not what you want but well.) [13:56] <Daemon404> or you could write 3 lines of C [13:56] <kierank> Daemon404: that's what i did for a static tone [13:56] <kierank> and probably extend it for a lipsync test (ntsc is problematic) [13:57] <ubitux> ffplay -f lavfi "testsrc[out0];aevalsrc='st(1,mod(t,2));sin(440*2*PI*ld(1))*exp(-5*ld(1))|st(1,mod(t+1,2));sin(440*2*PI*ld(1))*exp(-5*ld(1))'[out1]" [13:57] <ubitux> now you include a showwaves and you have a cute thing [13:58] Action: Daemon404 wipes up the vomit ubitux just barfed into the channel [13:58] <ubitux> :( [13:59] Action: ubitux http://z0r.de/4098 [14:00] <Daemon404> cant view that on android [14:00] <ubitux> you really miss something [14:03] <espes__> maybe a long shot, but do these look like sane aac-eld packets? https://gist.github.com/espes/4dc4b3c0538beac12f68 [14:03] <espes__> or some obvious transport? [14:03] <espes__> this is from an rtp stream that is /supposed/ to be rfc3640, but clearly not :/ [14:40] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:411be72dcbc9: avfilter/vf_noise: fix high resolution support [14:40] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:aba61b22f70f: avfilter/vf_noise: move shift calculation to filter_frame() [15:04] <jeetu> @michaelni ,,,,,,hlo sir ,the task you gave me yestrdy about audio filter,i am working on it......... actly sir i am doing it on java platform,,,,so jst wanna confirm whether it's okay or i need to switch on some other language platform?????? [15:06] <jeetu> Or is there any particular language you would suggest??? [15:06] <J_Darnley> 1 comma, 1 full stop, and 1 question mark, please. [15:07] <J_Darnley> FFmpeg is a C project. We will want a filter that can easily be linked into the libavfilter library. [15:07] <michaelni> jeetu, FFmpeg is written in ANSI/ISO C [15:07] <michaelni> jeetu, you know C ? [15:08] <jeetu> yes sir ,i do [15:11] <iive> jeetu: and in case you might need something like fast fourier transform, there are already dsp routines for that. [15:16] <michaelni> jeetu, look at 21b092de7c1e2d7fae24ceca3aa2f8f9f1730cfc, thats the addition of the sine audio source filter, you could add DTMF numbers->PCM into that. The PCM->DTMF numbers would be a separate filter [15:18] <michaelni> also that commit is a good example of what is needed to add a new filter [15:23] <jeetu> okay sir , [16:33] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:ed3efbcd0c84: avfilter/vf_noise: use per component rand_shift [16:33] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:4f1a252fd3f0: avfilter/vf_noise: Use a separate seed for each plane [16:33] <cone-322> ffmpeg.git 03Benoit Fouet 07master:8bcf425d06a2: avformat/id3v2: silence a warning when CONFIG_ZLIB is unset. [17:17] <arwa> how do I malloc new space ? What does av_mallocz do?? [17:17] <kierank> it zeros out the array [17:17] <kierank> well the memory [17:17] <kierank> imagine it like av_calloc [17:17] <BtbN> it makes sure it's alligned and fills it with zeros. [17:18] <arwa> ummm...i dont know av_calloc [17:19] <arwa> if I want to malloc a chunk of memory of size double the previous image...what do i do? [17:24] <J_Darnley> Isn't there a realloc function? [17:26] <J_Darnley> arwa: yes, av_realloc [17:28] <J_Darnley> look at libavutil/mem.h for the various memory functions. [17:28] <ubitux> arwa: you're doing it wrong [17:28] <ubitux> arwa: you want to configure your outlink with 2x the dimensions [17:28] <ubitux> and use the get_buffer function [17:29] <ubitux> look at how hqx does it [17:29] <ubitux> (i'm assuming you're still in the filter) [17:29] <ubitux> if you want an extra buffer, the you have various alloc frame utils [17:32] <arwa> okay, but when I tried to follow the things done in hqx, I am getting this error --> lvalue required as left operand of assignment [17:32] <arwa> what does it mean? [17:33] <arwa> I am getting error for this line --> "out->data[0] + j*out->linesize[0] + i*3 = in->data[0] + (int)(j/2) * in->linesize[0] + (int)(i/2)*3;" [17:33] <arwa> Why can I not do this? [17:35] <wm4> never say "I get an error" [17:35] <wm4> say WHAT error you get [17:36] <arwa> "lvalue required as left operand of assignment" [17:36] <arwa> Sorry :/ But I already mentioned the error..!! [17:37] <wm4> I didn't read [17:37] <ubitux> arwa: this is not valid c [17:37] <wm4> yeah, you're writing something like "123 = variable;" [17:38] <arwa> okay [17:38] <wm4> instead of "pointer + 123 = 5;" you want "*(pointer + 123) = 5;" [17:40] <arwa> okay [17:40] <arwa> thanks [17:40] <arwa> :D [17:49] <J_Darnley> Oh come on! That should be pointer[123] = 5 [17:51] <J_Darnley> And in the specific example: out->data[0][j * out->linesize[0] + i * 3] = ... [17:52] <J_Darnley> I also wonder why you are casting i/2 to an int, is i not an integer type? [18:00] <randomdude123> can anybody comment on this weird requirement by a spec: [18:01] <randomdude123> http://stackoverflow.com/questions/26393553/h264-rtp-and-packetization-mode… [18:01] <randomdude123> this is about h264 and STAP-A in RTP [21:25] <DocC> anybody in here familiar with the dshow components of ffmpeg under windows? [21:26] <wm4> never ask to ask [21:27] <wm4> because that's how you ensure the highest chance you'll never get an answer [21:28] <DocC> fair enough. I'm encountering an incompatibility between capturing with dshow in ffmpeg under windows using the latest generation of blackmagic design 4k decklink cards [21:28] <BtbN> DirectShow is one huge incompatiblity. [21:29] <BtbN> You basicaly have to write code for every single device. [21:31] <DocC> good to know [21:31] <DocC> I have several c developers on staff I can assign to the project [21:32] <BtbN> ffmpeg only supports very basic stuff, like webcams. Capturecards are way too much. [21:32] <DocC> I see [21:32] <BtbN> They are poorly documented, if at all, and add proprietary extensions that couldn't even be used in ffmpeg. [21:32] <DocC> the old generation blackmagic cards work fine [21:33] <ubitux> we have indev for decklink [21:33] <DocC> I saw there was an option to --enable-decklink [21:33] <ubitux> yes [21:33] <DocC> that will bypass dshow and go directly to the BMD hardware? [21:33] <kierank> BtbN: not true [21:33] <kierank> the blackmagic api is gpl compatible [21:34] <BtbN> They have an entire _API_? [21:34] <DocC> our application is hd capture and realtime x264 encoding, btw [21:34] <BtbN> The ones for Roxio and Hauppauge are not. [21:34] <DocC> and the results are spectacular with 1080p SDI in [21:34] <kierank> BtbN: yes, known as capturing properly [21:34] <kierank> in a professional environment [21:35] <DocC> so it sounds like skipping the -f dshow is what to try next [21:35] <DocC> I'll test it with our 4k cards [21:35] <J_Darnley> I always read that as "deliberately more complicated than it has to be" [21:35] <kierank> DocC: what is broken about the 4k cards [21:36] <kierank> J_Darnley: feel free to explain how you would provide audio in sync on linux [21:36] <DocC> when capturing 1920x1080p at greater than 24 fps, the start time is erroneously large [21:36] <kierank> when alsa doesn't provide you timestamps [21:36] <DocC> like 7M seconds [21:36] <J_Darnley> Isn't it a continuous stream of samples at a given rate? [21:36] <kierank> J_Darnley: yes and linux gives you two file descriptors [21:36] <kierank> how do you open both at the same time [21:36] <kierank> and sync up audio and video when you have no timestamps [21:37] <J_Darnley> If fopen that slow? [21:37] <kierank> it's nondeterministic [21:37] <kierank> that's the problem [21:38] <J_Darnley> Okay, you have problem with that. I have a problem with TV or "studio" levels and interlacing. Neither of which shoule exist today. [21:38] <kierank> tv levels make perfect sense [21:39] <kierank> how do you deal with filter overshoots? [21:39] <kierank> in 0-255? [21:39] <wm4> J_Darnley: but why wouldn't you fix ALSA? [21:39] <kierank> wm4: they don't want to fix alsa [21:39] <kierank> I went to the summit last year [21:39] <J_Darnley> Saturating arithmetic? [21:39] <kierank> and so you end up with blotches on the screen where the filter overshoots [21:39] <rcombs> J_Darnley: how about overscan [21:39] <prabhy> hey,i wanted to know if i can convert gcc asm to yasm or nasm,or is it compulsory to convert it to yasm [21:40] <wm4> kierank: what was their answer as to how sync should be done? [21:40] <kierank> yasm [21:40] <kierank> wm4: "must be done in alsa" [21:40] <JEEB> prabhy, this project only takes in IA32 and x86_64 asm in yasm form [21:40] <J_Darnley> What else does a TV show when you get "overshoots"? [21:40] <JEEB> inline asm is only allowed on architectures where separate asm files are not possible [21:41] <J_Darnley> Don't most things clip to TV range? [21:41] <prabhy> ok [21:42] <DocC> http://pastebin.com/MQqV7JZd [21:42] <kierank> at the consumer yes but you have to bear in mind these signals pass through crazy infrastructure [21:43] <kierank> so you end up filtering again and again [21:43] <DocC> pastebin of the decklink capture bug I'm encountering [21:43] <kierank> and doing that on clipped values screws you [21:43] <J_Darnley> "crazy infrastructure" and you want to keep it going? [21:44] <J_Darnley> With enough craziness it sounds like you'll be clipping at 0-255 eventually. [21:44] <kierank> well yes, you are going to have your cable company processing the video [21:44] <kierank> and possibly a distributor along the way [21:44] <kierank> cnn for example is processed many many times [21:44] <kierank> before it gets to your home [21:45] <kierank> no because you're only allowed 2% or so of the picture as overshoot/undershoot iirc [21:45] <wm4> processed for what? [21:45] <kierank> wm4: framerate converted, re-encoded etc [21:45] <J_Darnley> To remove quality ofcourse [21:46] <wm4> "crazy infrastructure" I guess [21:49] <rcombs> interlaced, de-interlaced, re-interlaced, over-scanned, under-scanned, window-boxed, cropped, stretched, squeezed& [21:49] <wm4> NSA'd! [21:49] <rcombs> if it's a movie made on film, then along with over- and under-scanned it got outright scanned at some point [21:50] <rcombs> and telecined, inverse-telecined, pulled up, pulled down, and sped up if you're in Europe [21:51] <rcombs> I have a feeling one could write a highly amusing song about all this [21:51] <wm4> and then converted to analogue by sending it through ancient cable TV links [21:52] <wm4> it's ironic that my mom's new TV looks much worse than the old CRT she had [21:54] <J_Darnley> rcombs: in the style of the elements song or weird al's "hardware store"? [21:55] <rcombs> J_Darnley: either could be great [21:55] <rcombs> I was just thinking of xkcd's Every Major's Terrible, but that's the same tune as the elements song anyway [21:56] Action: J_Darnley hasn't heard that [21:56] <llogan> wm4: what's with these tvs lately? i don't know what postprocessing crap is going on, but it looks like shit. [21:56] <J_Darnley> 200Hz interpolation! [21:57] <llogan> early 2000 soap opera vision [21:57] <rcombs> https://www.youtube.com/watch?v=pRexBMPeRTo from http://xkcd.com/1052/ [21:58] <wm4> llogan: dunno, just cheap shit [22:36] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:080c846f5999: swresample: do not put multiple statements in one line [22:36] <cone-322> ffmpeg.git 03Michael Niedermayer 07master:f6bb2cd1b048: swresample/resample: fix invert_initial_buffer() after flush [23:43] <ubitux> can anyone comment on http://www.cccp-project.net/forums/index.php?topic=6826.0 that it should be fixed? [23:43] <ubitux> JEEB maybe? [23:43] <ubitux> (i'd actually like these people to test :p) [23:59] <J_Darnley> in case that JEEB isn't here, JEEBsv ^^ [00:00] --- Fri Oct 17 2014

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ffmpeg.log.20141016
by burek 17 Oct '14

17 Oct '14

[03:21] <fourkay> Hi [03:23] <fourkay> May i ask a user question here ? [03:26] <fourkay> Just in case some one is readind. Here it is. [03:27] <fourkay> I have a delay of 3.3 sec between the video stream and the audio stream [03:29] <fourkay> I'heve tried the parameter -itsoffset "3.3" and it didn"t correct the problem. Any other idea ? [03:36] <fourkay> Here is the compete command i used: [03:36] <fourkay> ffmpeg.exe -i "input.mp4" -metadata title=" " -ss "00:00:00" -t "23:00:00" -itsoffset "3.3" -s 640x360 -ab 128k -ar 44100 -ac 2 -acodec libfaac -r 25 -vcodec libx264 -fpre "libx264-medium.ffpreset" -fpre "libx264-baseline.ffpreset" -aspect 16:9 -f mp4 -b 1123328 -y "output.mp4" [04:06] <fourkay> thnks for your support. I'll try to apply each and every recommandation i received so far. thnks agan !! [08:16] <yoyowinwin> c_14: we found the root cause ! [08:17] <yoyowinwin> static int flv_read_packet(AVFormatContext *s, AVPacket *pkt) [08:17] <yoyowinwin> { [08:17] <yoyowinwin> if ((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_KEY || [08:17] <yoyowinwin> stream_type == FLV_STREAM_TYPE_AUDIO) [08:17] <yoyowinwin> av_add_index_entry(st, pos, dts, size, 0, AVINDEX_KEYFRAME); [08:17] <yoyowinwin> } [08:18] <yoyowinwin> thanks again for your note. [08:28] <polysics> hello! I am going mad over this very simple issue [08:29] <polysics> ffmpeg -i my_file.m2v -filter_complex "[0:v]colorbalance=rs=.3[vout0];" -map "vout" dest.mp4 [08:29] <polysics> https://gist.github.com/polysics/c047df1094aca7e09d76 [08:30] <polysics> No such filter: '' [08:30] <polysics> the above is simple enough I thought I would be able to spot it [08:30] <polysics> the error occurs with any filter, by the way [08:36] <xeoncore> polysics Are you sure your shell isn't detecting the `;` [08:42] <Baked_Cake> can avi files handle vbr aac [08:42] <Baked_Cake> it doesnt seem to want to convert the container to mkv [08:43] <polysics> xeoncore: on both bash and zsh? [08:44] <c_14> polysics: get rid of the ';' [08:44] <polysics> although removing the ; seems to fix it [08:44] <c_14> and it should be -map "[vout0]" [08:44] <polysics> yup, that's a typi [08:44] <polysics> typo [08:44] <polysics> but the ; is what was causing it [08:44] <c_14> If you put a ';' it expects another filter after it [08:44] <polysics> I think ffmpeg is trying to run a filter [08:44] <c_14> since you didn't write anything there it's expecting the filter [empty string] [08:44] <polysics> well, thanks to both of you, saved my morning :) [08:46] <xeoncore> Plorkyeran c_14 :) [08:46] <xeoncore> I knew it was the ; [08:46] <xeoncore> just didnt know why ;) [09:08] <zenny> Good morning! [09:10] <zenny> Can ffmpeg record and stream live the composite video? [09:12] <zenny> or is concurrent recording and streaming is possible only with the servers like red5? I love the simplicity of ffmpeg command line, but finding it difficult to figure out. Inputs appreciated! [09:27] <c_14> yoyowinwin: did you open a bug report? [09:40] <Baked_Cake> ffmpeg is like the most rewarding command line stuff ive learnt in my whole life [09:41] <Baked_Cake> tho i use mkvtoolnix to ez mux and avidemux to ez cut samples [09:42] <Baked_Cake> also https://trac.ffmpeg.org/wiki/StreamingGuide [09:45] <polysics> another question: I am getting the video clipped to the overlay size, which is smaller than the video source [09:45] <polysics> is htere a switch to avoid that or am I doing things in the wrong order? [10:01] <zenny> Baked_Cake: very truly said, but it whines on times ;-) when something that one wants cannot be achieved. (happening to me) [10:06] <Akagi201> LD_LIBRARY_PATH="/home/akagi201/libs-a5s/lib" ./configure --enable-cross-compile --cross-prefix=arm-none-linux-gnueabi- --arch=armel --target-os=linux --prefix=/home/akagi201/libs-a5s --enable-gpl --enable-libx264 --enable-nonfree --enable-libaacplus --enable-librtmp --extra-cflags="-I/home/akagi201/libs-a5s/include" --extra-ldflags="-L/home/akagi201/libs-a5s/lib" [10:06] <Akagi201> I got rtmp not found [10:06] <Akagi201> What's wrong? [10:08] <Akagi201> How can I modify the configure file and write the fixed librtmp lib path? [10:55] <mansi> hi ! i am trying to do ffmpeg -i image.jpeg. However i get : Operation not permitted. why is that? i have read/write permission for the image [11:01] <ubitux> just a return -1 in the code somewhere [11:12] <mansi> http://sprunge.us/LXcZ. ubitux . i am trying make and install again. [11:13] <ubitux> can you upload the jpeg? [11:13] <ubitux> (maybe that's actually a permission issue though) [11:14] <mansi> ubitux, i checked the permission read and write access to all users (which i gave and tried again) [11:14] <ubitux> then upload the file [11:16] <mansi> http://imgur.com/1mW9hx8 . not sure how much this will help. image pretty much summarizes my mood [11:16] <ubitux> ffprobe 'http://i.imgur.com/1mW9hx8.jpg' [11:16] <ubitux> no error here [11:22] <mansi> fresh install and it works for me too. ubitux thanks [11:27] <mansi> ubitux, after a change to a file, i do a make and i get this http://sprunge.us/SBSe. i should do a make clean again? [11:28] <ubitux> you have something broken in the permissions of your system [11:28] <ubitux> as if you build in root once or something [11:29] <ubitux> or maybe the binary is in use [11:29] <ubitux> no idea [11:29] <ubitux> that's not a ffmpeg issue, it's an issue on your end [11:29] <ubitux> fix your system [11:37] <zenny> mansi: which linux distro and version do you use? [11:39] <mansi> zenny, 3.12.24-1-MANJARO [11:56] <zenny> mansi: can you execute 'ffmpeg -report -i anyvideofile.ext' and share the log? [11:58] <mansi> zenny, http://sprunge.us/GQEg [12:00] <zenny> mansi: the log shows it works fine with the specified video. What are you trying to do? [12:01] <mansi> yeah. it showed me some permission errors previously when i edited a file and did make again http://sprunge.us/SBSe i keeping doing make clean and make then [12:01] <mansi> zenny, [13:43] <FunnyLookinHat> Is there any good documentation explaining the differences between FFMPEG 1.x and 2.x ? I just noticed that this PPA only has 1.x builds: https://launchpad.net/~jon-severinsson/+archive/ubuntu/ffmpeg [13:57] <Mavrik> FunnyLookinHat, this is as good as it gets: https://raw.githubusercontent.com/FFmpeg/FFmpeg/master/Changelog [13:59] <FunnyLookinHat> Mavrik, hehe - that works! [14:02] <FunnyLookinHat> Mavrik, Is it safe to say that 1.2.x is maintenance mode at this point? Only bugfixes? [14:02] <Mavrik> FunnyLookinHat, yes, pretty much anything that's not current master is in maintenance mode [14:02] <FunnyLookinHat> Mavrik, cool thx [14:02] <Mavrik> release are just "snapshots" at points of time where people decide ffmpeg kinda works OK [14:06] <Jenser> hi everyone. i don't get a filter problem solved... can s/o please tell me how to get the "interlace" filter into my ffmpeg installation. i run ffmpeg on linux wheezy and installed ffmpeg via deb-multimedia. ffmpeg is at version 1.0.10, libavfilter is at 3.17.100. [15:25] <Mavrik> Jenser, grab a new static build from links in topic [15:25] <Mavrik> the filter name is "yadif" [15:26] <c_14> Mavrik: he wanted to interlace, not deinterlace. [15:26] <c_14> The interlace filter is called 'interlace'. [15:33] <Mavrik> and my reading comprehension sucks arse [15:33] <Mavrik> thanks c_14 [15:38] <Jenser> @Mavric yadif is the deinterlacing filter, i need to deinterlace -> ... -> interlace [15:39] <Mavrik> yes, I misread you :) [15:40] <Jenser> i have a 2013 static build with libavfilter version 3.7xxx, which includes the filter "interlace" [15:41] <Mavrik> you probably have to do something in lines of -vf fps=50,interlace -x264opts=tff -codec:v x264 ... [15:41] <Jenser> yeah, that's what i want. problem is: error no such filter "interlace" [15:42] <Jenser> ffmpeg -filters outputs me that the interlace filter is not acutally installed [15:42] <Mavrik> Jenser, are you claling the proper ffmpeg binary? what does -filters say? [15:42] <Jenser> it says NO interlace [15:43] <Jenser> if i manually start the old 2013 build, theres 3.77.01 @ libav_version [15:43] <Jenser> the new one (installed yesterday via deb-multimedia is at 3.17.100 [15:44] <Mavrik> um [15:44] <Jenser> -filters @ old one HAS "interlace", the same @ new one has NOT [15:45] <Mavrik> I don't really understand the issue, if your build doesn't have interlace built in, why don't you take a new 2.4.2 build that certanly has it? [15:45] <Jenser> as static build? [15:46] <Mavrik> yes, you download it, unzip it and run it [15:46] <c_14> Jenser: grab http://johnvansickle.com/ffmpeg/ [15:46] <c_14> It has interlace [15:47] <Jenser> gonna give it a try [16:12] <Jenser> c_14 thank you so much!!! [16:12] <Jenser> this sources builds seem to be exactly what i need [17:53] <akiselev> When I transcode an MP4 file to m3u8 (using HLS muxer), the resulting video basically stutters/loops over a few frames every second. i.e., during the first second it will loop over frames #1-5 (approx), during the next second it will loop over frames ~25-30, etc. [17:54] <akiselev> Is this because I'm doing something wrong with the PTS/DTS or could this be cause by some other mistake? (I am using libavformat + codec, not ffmpeg) [17:55] <akiselev> I am using several filters and have memory pools as well, so could this be a problem with packets or frames deallocated early? [18:06] <BtbN> m3u8 is a playlist, not a video container [18:30] <akiselev> BtbN, yes, but the HLS muxer that writes the m3u8 also writes the .ts files directly with the mpegts muxer [19:40] <zenny> Does anyone know of a way to remove a known background from the foreground using ffmpeg? I tried many a ways, but did not work. Appreciate any input! Thanks [19:49] <DocC> Hi all. I'm encountering an issue when capturing with ffmpeg on windows using a blackmagic decklink 4k card and the -f dshow interface. FFmpeg gives a very large start time, on the order of 7M seconds, but only when the frame rate is set higher than 24 fps [19:50] <DocC> and ffmpeg drops frames rather than capturing while it counts down this high start time [19:51] <DocC> when frame rate is set to 24 fps, it captures fine. Resolution is 1920x1080p, vcodec is v210 [19:52] <DocC> the older generation blackmagic decklink (with no 4k support) captures fine at 24, 29.97, and 30 fps at 1920x1080p [20:20] <[1]Ian> hello [20:25] <[1]Ian> I'm trying to figure out how to take a bunch of ivf files and opus files representing a conference with the ivf files being screen shares and the opus files being the voices of the participents, and turn them into a single webm file. [20:26] <[1]Ian> something like this: [20:26] <[1]Ian> merging video: [20:26] <[1]Ian> black 10 seconds [20:26] <[1]Ian> video 1: [20:26] <[1]Ian> offset 10 seconds [20:26] <[1]Ian> length 300 seconds [20:26] <[1]Ian> black 50 seconds [20:26] <[1]Ian> video 2: [20:26] <[1]Ian> offset 360 seconds [20:26] <[1]Ian> length 300 seconds [20:26] <[1]Ian> black 40 seconds [20:26] <[1]Ian> total 700 seconds [20:26] <[1]Ian> merging audio: [20:26] <[1]Ian> silence 1 second [20:26] <[1]Ian> audio 1: [20:26] <[1]Ian> offset 1 seconds [20:26] <[1]Ian> length 320 seconds [20:26] <[1]Ian> audio 2: [20:27] <[1]Ian> offset 10 seconds [20:27] <[1]Ian> length 690 seconds [20:27] <[1]Ian> audio 3: [20:27] <[1]Ian> offset 240 seconds [20:27] <[1]Ian> length 420 seconds [20:27] <[1]Ian> total 700 seconds [20:44] <llogan> use a video editor. or a big filterchain involving trim, atrim, setpts, color, aevalsrc, and concat filters [20:45] Action: kepstin-laptop notes that he wrote some recording processing stuff for bigbluebutton which basically is just a ruby script that builds the filter chain for the latter [20:46] <kepstin-laptop> (but rather than using concat, it outputs to a bunch of mpeg-ts files that are concatenated) [20:55] <[1]Ian> My goal is to run this batch style on the command line from .net [20:57] <kepstin-laptop> so yeah, you'd have the .net app read the info about which video/audio goes where, have it build a filter chain, then run ffmpeg with that. shouldn't be too bad. [21:09] <DocC> anybody familiar with capturing via dshow in ffmpeg? [21:26] <RoyK> hi al. how can I package a DVD into an mkv file with ffmpeg? [21:27] <RoyK> s/al\./all./ [21:43] <llogan> RoyK: perhaps look at tools/dvd2concat [21:44] <RoyK> llogan: is that normally installed with an apt-get install, or just in the repos? [21:44] <llogan> i don't know anything about apt-get [21:48] <llogan> if it's a commercial disc then you'll probably have to use something that uses libdvdread first (tccat, vobcopy, etc) [21:49] <llogan> *libdvdcss [21:51] <RoyK> got that, but [21:51] <RoyK> Your lsdvd version does not print cell sectors. [21:51] <RoyK> Make sure your lsdvd version has the two following patches applied: [21:51] <RoyK> http://sourceforge.net/p/lsdvd/feature-requests/1/ [21:51] <RoyK> https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=603826 [22:07] <VLADIMER_PUTIN> putin here! [22:07] <VLADIMER_PUTIN> hi! [22:23] <timezone> hey guys, i have some binary files in ARGB but i'm not sure how to encode them, they are just raw pixel buffers, what flags do i need to hint ffmpeg? -pix_fmts shows IO... argb 4 32 [22:46] <[2]Ian> Putin, when are you going to deal with the feminist problem? [22:46] <ChocolateArmpits> timezone: Do you need them preserved? probabyl rawvideo and specify the pixel format [22:47] <[2]Ian> I admired what you did to those dirty punks, but... when are you going to take it to the next level? [22:49] <ChocolateArmpits> timezone: or 'bmp', supports rgb with alpha [23:15] <timezone> ChocolateArmpits: yeah got it working, cheers mate [23:51] <sourpulse> A commercial program (not GPL or LGPL) has the ability to launch any other program via user-configurable command line, and output raw video frames to it through a pipe. FFmpeg (such as a Zeranoe static build for Windows) is able to encode that pipe into a movie file. Does FFmpeg's license allow it to be used that way? [23:56] <llogan> i don't see how the license should restrict what inputs you use [23:58] <akiselev> sourpulse, IANAL but: it depends on how you package it. If the user of the software downloads ffmpeg binaries or builds it themselves and points your software to it, there is (basically) no restriction whatsoever on what you can do [00:00] --- Fri Oct 17 2014

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