Ffmpeg-devel-irc

Download

ffmpeg-devel-irc@ffmpeg.org

September 2014

1 participants
60 discussions

ffmpeg-devel.log.20140919
by burek 20 Sep '14

20 Sep '14

[00:00] <jamrial> so you may want to check the list archive [00:02] <cone-389> ffmpeg.git 03Michael Niedermayer 07master:d9b141c05f32: postproc/postprocess: Remove unused argument from reallocAlign() [00:02] <cone-389> ffmpeg.git 03Michael Niedermayer 07master:859d7d4640c3: postproc/postprocess: Use FF_ARRAY_ELEMS() in pp_free_context() instead of hard-coding their size [00:02] <cone-389> ffmpeg.git 03Michael Niedermayer 07master:5a9ca68e15fe: postproc/postprocess: remove redundant casts [00:03] <llogan> weirddan455: indeed. since you already got two replies. https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2014-September/163099.html [00:06] <J_Darnley> He is CCed but unfortunately not everyone or every client correctly CCs back [00:09] <weirddan455> llogan, yea i didn't get the emails... i just subscribed though [00:10] <rcombs> llogan: why doesn't the list allow digest mode, btw? [00:12] <nevcairiel> rcombs: because people reply to the digests, and its horrible .P [00:19] <llogan> weirddan455: you can reply via the archives to preserve the in-reply-to hash. [00:19] <llogan> (click the author's email link) [00:20] <weirddan455> llogan, woops too late already sent without that [03:02] <cone-289> ffmpeg.git 03Michael Niedermayer 07master:7f8d28f68678: postproc: Replace CLIP by av_clip_uint8 [09:53] <cone-236> ffmpeg.git 03Gianluigi Tiesi 07master:59af5383c18c: avcodec/libilbc: support for latest git of libilbc [11:51] <kriegerod> could anybody experiences review ticket https://trac.ffmpeg.org/ticket/3823 and give some clue what happens and how it should be fixed? [16:36] <cone-495> ffmpeg.git 03Pascal Massimino 07master:4fd21d58a72c: libavcodec/webp: treat out-of-bound palette index as translucent black [17:05] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:5ccd08d26dda: postproc/postprocess: Avoid some ifs in do_a_deblock_C() [19:31] <cone-232> ffmpeg.git 03James Almer 07master:6edd6a4f002d: avcodec/dv_profile: deprecate internal function that shouldn't be public [20:36] <cone-232> ffmpeg.git 03Michael Niedermayer 07master:adaa274fa8a2: tests/utils: make arguments const that arent changed [20:46] <wm4_> ubitux: another filter suggestion: hrtf [20:46] <wm4_> it seems some people want this [20:47] <ubitux> doesn't this belong in swr? [20:47] <wm4_> dunno [20:48] <ubitux> it just sounds like a set of matrix stereo encoding [20:48] <ubitux> but i might be wrong [20:48] <wm4_> something like this, but I have no clue what exactly it's doing [20:49] <ubitux> i have a tons of stuff pending as usual [20:49] <wm4_> " Experimental audio filter that mixes 5.1 and 5.1 with matrix encoded rear channels into headphone signal using FIR filtering with HRTF." [20:49] <ubitux> i'd better focus on text subtitles right now since i strangely feels motivated on that currently [20:49] <wm4_> well it's just a suggestion [20:49] <ubitux> and that might not last long [20:49] <wm4_> heh [20:49] <wm4_> yes subtitle stuff needs to be fixed [20:50] <ubitux> libavformat/assdec.c | 25 ++++++++++++++++++------- [20:50] <ubitux> 1 file changed, 18 insertions(+), 7 deletions(-) [20:50] <ubitux> mmh i like this better [20:50] <ubitux> thanks for the suggestion wm4_ :p [20:51] <wm4_> did you merge the header parse code? [20:51] <ubitux> yeah [20:51] <wm4_> nice [20:51] <ubitux> i sent it a moment ago [20:55] <ubitux> wm4_: did you drop it from mpv or something? [20:55] <wm4_> ubitux: drop what? [20:55] <ubitux> hrtf [20:55] <wm4_> no, but I'm debugging it right now [20:56] <ubitux> ah, i see [21:07] <ubitux> wm4_: Comments being timed is so evil :D [21:07] <wm4_> the format is not called ASS for no reason [21:07] <ubitux> it's ugly but it has benefits? [21:08] <wm4_> they show up in aegisub I guess [21:08] <wm4_> so an editor can comment and uncomment them [21:09] <ubitux> you can't actually mux them in mkv [21:09] <wm4_> hm really [21:09] <ubitux> so they only matters when transmuxing (and eventually transcoding) [21:09] <ubitux> yes, mkv only supports Dialogue lines [21:09] <wm4_> hm really [21:09] <ubitux> it's even dropping the "Dialogue:" part [21:10] <ubitux> so you can't make a distinction between the two [21:10] <wm4_> mkv might put comments into the extradata [21:11] <wm4_> not sure [21:11] <ubitux> note they're in the events section [21:11] <ubitux> (since timed) [21:12] <wm4_> I'm relatively sure I've seen "Comment:" lines in ass files extracted from mkv [21:12] <ubitux> actually i think we might have a mechanism to transfer them [21:12] <ubitux> ah? [21:12] <ubitux> can you find me one? [21:12] <ubitux> i'm very interested in this [21:12] <wm4_> ah yes they're in the CodecData aka extradata [21:12] <wm4_> so basically they get sorted to the top [21:12] <wm4_> of the Events section [21:13] <wm4_> how odd [21:13] <ubitux> show me the magic :) [21:14] <wm4_> well, take any random mkv file with ass tracks, they're likely to contain some comments [21:18] <ubitux> i just tried a few, nothing else than Dialogues [21:18] <ubitux> at least mkvextract doesn't extract any [22:32] <cone-232> ffmpeg.git 03Daniel Bomar 07master:af1818276ef2: avformat/os_support: Add _DEFAULT_SOURCE to hide warning about _SVID_SOURCE depreciation [00:00] --- Sat Sep 20 2014

1 0

ffmpeg.log.20140919
by burek 20 Sep '14

20 Sep '14

[00:34] <active8> http://pastebin.com/jAnmAzAa - x11grab - Non-monotonous DTS in output stream 0:0; 4 examples with the multitude of warnings snipped to a few. 1. no audio playing. 2. VLC playing audio. 3. Audio coming from a browser video. 4. Full screencast with the browser video playing. [00:38] <llogan> what if you use -c:a copy instead? i wonder if using libpulse instead of alsa would make any difference. [00:40] <llogan> you would need to build with --enable-libpulse though [00:42] <relaxed> it appears he is using pulse [00:42] <llogan> i meant use -f pulse, not -f alsa [00:43] <okokokno> Hmm... Even though I now did avformat_find_stream_info, I still don't see some info I need for mp3 files [00:43] <okokokno> For example, how many packets there are in the file [00:43] <active8> it's not in the command output so I uess I would. ffmpeg -formats would output libpulse? [00:44] <llogan> no, because your build configuration lacks --enable-libpulse [00:44] <llogan> http://ffmpeg.org/ffmpeg-devices.html#pulse [00:44] <okokokno> Or can that be computed from the duration somehow? [00:44] <active8> but also notice that in the full screencap output, the message is all about the video stream: [matroska @ 0x2224460] Non-monotonous DTS in output stream 0:1; [00:45] <okokokno> (AVStream::nb_frames is still 0) [00:45] <llogan> active8: all this time i was thinking of "xruns". [00:46] <llogan> trying to do too much as once and i suck at multitasking and reading [00:46] <active8> that's what I meant by "output" --enable-libpulse isn't there [00:46] <relaxed> active8: you're using a 2013 build, which is pretty old [00:48] <active8> relaxed: roughly how often do they update ffmpeg. I'm not exactly ready to rebuild. Just asking. [00:48] <relaxed> every three months - or in git everyday [00:49] <active8> ffmpeg version git-2013-08-04-61af627 built on Feb 19 2014 04:33:30 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4) [00:49] <active8> funny. i pulled down those sources in feb and they were 6 months old [00:50] <relaxed> active8: try again with output.flac [00:51] <active8> same command, diff extension? [00:51] <relaxed> yes [00:51] <relaxed> loose -threads 0 [00:51] <relaxed> lose* [00:53] <active8> unsupported codec [00:54] <relaxed> pastebin [00:55] <active8> http://pastebin.com/JKx2rApg [00:56] <relaxed> active8: ffmpeg -f alsa -ac 2 -i pulse output.flac [00:58] <active8> now it's a red message : http://pastebin.com/6a0PK8Ji [00:59] <relaxed> what was the exit status? [01:00] <Hfuy> hihi [01:01] <Hfuy> ack [01:01] <Jan-> Problem: we have some AVC-Intra MXF files which seem to have eight streams of audio. [01:01] <Jan-> We're converting them (successfully) to ProRes quicktimes but the output files only have one mono audio channel in them. [01:02] <Jan-> Here's the ffmpeg analysis of the files, input first: http://pastebin.com/jwiSR6SR [01:02] <active8> relaxed: 2 if $! is exit status - can't remember [01:03] <active8> i mean $? [01:03] <relaxed> exit status is $? [01:03] <active8> linux [01:03] <Jan-> there should presumably be some way we can create stereo output files (using map?) but I'm not sure what the syntax would be. Any suggestions [01:03] <Jan-> ...would be most appreciated :) [01:03] <active8> the exit is at the bottom of the paste [01:03] <c_14> Jan-: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [01:03] <active8> exit is 2 if $? is exit status [01:03] <kepstin-laptop> Jan-: http://www.ffmpeg.org/ffmpeg-filters.html#amerge also [01:04] <relaxed> ok [01:04] <pzich> Jan-: have you been through https://trac.ffmpeg.org/wiki/AudioChannelManipulation ? [01:04] <relaxed> Jan-: you want to include all the audio streams and change them to stereo? [01:04] <active8> IIRC. I had this, or at least a messed up seek on VLC when making a slideshow. The time in VLC would be 00:00 - but later conversion from mpeg to mp4 or just mpeg to mpeg through c:v copy fixed it. I'd still like to not see the meaasage and don't really know what it means. Seems ffmpeg would need to guess the timestamps from the sample rate [01:04] <Jan-> As far as we know the input should only be stereo. [01:05] <Jan-> So there are probably only two useful channels of interest [01:05] <relaxed> Jan-: you're build is too old and that could be the problem [01:05] <relaxed> your* [01:05] <Jan-> we'd like to figure out which they are (probably by spitting out wave files to analyse which input streams we want) then put those into the stereo output. [01:06] <pzich> Jan-: sounds like you need -map_channel [01:06] <Jan-> uhhuh [01:07] <Jan-> The problem we found is that a lot of the examples deal with muxing wave files into videos to create multipoint surround etc [01:07] <llogan> or perhaps play in VLC and choose each individual stream. i imagine they would be in the same order as shown by ffmpeg [01:08] <Jan-> Oh hm you can do that [01:08] <Jan-> wait one we'll figure it out [01:08] <Jan-> I would be amazed if we didn't just want 0 and 1 [01:08] <Jan-> or rather 1 and 2, as 0 seems to be the video [01:08] <Jan-> MXF is such a PITA :/ [01:09] <Jan-> yeah, it's just 1 and 2 [01:10] <Jan-> for some reason VLC is interpreting the eight soundtracks as independent mono tracks, as opposed to assuming 1 and 2 are the stereo pair. [01:10] <Jan-> ffmpeg does the same [01:10] <llogan> -filter_complex "[0:a:0][0:a:1]amerge" -ac 2 [01:10] <llogan> or something like that [01:10] <relaxed> it could be a decoding issue. are you sure there are stereo streams? [01:11] <Jan-> wait one [01:12] <pzich> it may need to be [0:a][1:a] [01:12] <Jan-> llogan: Perfect [01:12] <pzich> oh nm, misread the output [01:12] <Jan-> man MXF Is horrible [01:12] <Jan-> also, how would we ever have figured that out :/ [01:13] <llogan> you may want to compare using -ac 2 with the pan example in https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a2stereostereo [01:13] <Jan-> Honestly it's just a simple channel assignment issue [01:13] <Jan-> your filter_complex line does it for a wave file [01:13] <Jan-> we just have to integrate it into a script [01:14] <llogan> i assumed pan and -ac 2 would provide the same output, but they don't for me [01:14] <Jan-> C:\Users\CobaltUser>c:\ffmpeg -i "E:\Roll 1\XDROOT\Clip\120_0017.MXF" -filter_co [01:14] <Jan-> mplex "[0:a:0][0:a:1]amerge" -ac 2 d:\test.wav [01:14] <Jan-> ...has the desired effect [01:14] <Jan-> where does 0:a:0 come from [01:15] <pzich> 0:a:0 would be channel 0 of the first audio stream [01:15] <Jan-> oh, right [01:15] <kepstin-laptop> in your ffmpeg output, there's a bunch of Stream # lines [01:15] <Jan-> there is [01:16] <kepstin-laptop> I think using [0:1] and [0:2], matching the stream lines in the output, would work as well? [01:16] <Jan-> fyi this MXF is the output of a Sony PXW-X180 camcorder in AVC-Intra 1080p25 mode. [01:16] <llogan> yes, but it's lazier to use 0:a:0, etc. [01:17] <Jan-> What does it mean by [mxf @ 00000000024d77e0] could not resolve sub descriptor strong ref [01:17] <Jan-> is that some weird mxf container stuff [01:17] <llogan> i need a new video camera... [01:17] <Jan-> don't get a pxw-x180 :) [01:18] <llogan> maybe i'll just start renting, but that's a bitch where i live [01:18] <Jan-> this is a review loaner. [01:18] <Jan-> and we scratched the display on one of the radio mic receivers :/ [01:19] <llogan> it was like that when you got it [01:19] Action: Jan- nods vigorously [01:20] <Jan-> sony's much-advertised "AVC Intra" just seems to be 10-bit h.264 with all I frames [01:20] <Jan-> at craptons of bitrate [01:21] <Jan-> what does it mean by "input channel layouts overlap" here: http://pastebin.com/Vz7xp2S9 [01:22] <kepstin-laptop> Jan-: both input channels are 'mono', rather than one being 'left' and one 'right' [01:22] <kepstin-laptop> Jan-: in this case it does the right thing, it assumes since there's two, that you meant stereo [01:22] <Jan-> fair enough [01:23] <Jan-> I wasn't aware a an audio channel could be marked with a position [01:23] <kepstin-laptop> well, an audio stream is marked with a layout [01:23] <Jan-> does this imply that the MXF isn't properly marking its channels [01:23] <Jan-> (and wtf is it doing with 8 channels of audio anyway?!) [01:24] <llogan> http://ffmpeg.org/ffmpeg-filters.html#amerge describes the input layout stuff [01:24] <Jan-> ffmpeg is a very very big piece of software isn't it... [01:27] <Jan-> Ah. I can't have -c:a copy with this audio filtering [01:28] <kepstin-laptop> no, but -c:a pcm_s24le will pass it through with no further encoding. [01:28] <Jan-> will just omitting the -c:a do the same [01:28] <kepstin-laptop> depends on the output format, some may autoselect a lossy codec. [01:29] <relaxed> probably not, be spcific [01:29] <Jan-> hm no I get AAC in that case :) [01:29] <Jan-> snksnksnk a prores quicktime with 128kbps AAC audio [01:29] <Jan-> that's hilarious [01:29] <kepstin-laptop> using ffmpeg's not particularly good aac encoder too, I bet ;) [01:29] <Jan-> woah, this camera uses 24bit audio!? [01:30] <llogan> Jan-: did you ever get to mess around with a Canon EOS C100 to C500? [01:31] <Jan-> a C300, and not me, my boyfriend [01:31] <Jan-> why d'you ask [01:32] <llogan> what did he think of it? i may rent one for a week for a project (and learn how to use it on the plane). [01:32] <Jan-> he says "incredibly nice sensor, shame about the gimped 8 bit output" [01:32] <Jan-> and "what're you gonna do for glass" [01:34] <llogan> more rental, but i haven't decided which lens(es) yet. i need to find more about the actual project if it's mostly interviews or more outdoor. [01:35] <Jan-> canon know how to make nice sensors [01:36] <Jan-> this fault had us worried though [01:36] <Jan-> suddenly both channels appear to be the same audio and neither of them is the radio mic [01:36] <Jan-> eeeeek [01:37] <Jan-> By the way, who's this famous Kostya guy who did the prores encoder? [01:37] <llogan> there are 2 prores encoders. [01:38] <Jan-> Yeah. [01:38] <Jan-> prores_ks seems to work quite a lot better, but it's a bit slower [01:38] <Jan-> the normal "prores" one creates files that go way over bitrate (though few things seem to care) [01:39] <kepstin-laptop> http://codecs.multimedia.cx/?p=388 has some discussion on speed [01:39] <Jan-> we even put some of the non-ks prores files on hardware recorders [01:39] <Jan-> and they played them fine [01:43] <Jan-> "probesize" tells it how much of the video to look forward at when estimating bitrate, right? [01:44] <Soul_keeper> I use it for streams that arent detected at the begging [01:45] <kepstin-laptop> and in some formats (mpeg ts in particular) to find streams that don't start at the start of the file [01:45] <Jan-> oh hm [01:45] Action: Jan- runs "convert.bat" and goes to bed [01:45] <Jan-> is "could not resolve sub descriptor strong ref" bad? [04:47] <fluke_> Hi every one. I want to ask how to calculate network speed while playing http video with ffplay? [04:47] <fluke_> My thought is to calculate in http.c [11:47] <fswings> Morning, how do I extract the DTS core from a DTS-HD MA track using ffmpeg? Or are there alternative Linux based tools which are recommended? [11:47] <fswings> I'm running via command line. [13:29] <okokokno> Can I somehow get the info from the XING header in an mp3 via libavformat? [13:38] <krasnayarsk> Hi. I have an avi video and its been converted from mp3. There is no image when listening/watching the avi file. What is the correct/best way to add a still image, perhaps a series of images, to the sound in the avi file? [13:40] <c_14> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images [13:42] <krasnayarsk> c_14: Thank you. [13:43] <krasnayarsk> part [15:27] <phw> Can ffmpeg play one file with video being redirected but audio playing locally? [15:28] <ubitux> try to map a stream to an alsa output or something [15:28] <ubitux> and map the video stream to another output [15:29] <ubitux> your question is weird though [15:30] <phw> well i am using piwall to display a video, while playing the same videos audio [15:32] <phw> This results in "ffmpeg -re -i big_buck_bunny_720p_surround.avi -vcodec copy -f avi udp://239.0.1.23:1234" to broadcast the video via stream [15:33] <phw> but i want ffmpeg to simultaniously play the audio of the file [18:04] <Joryn> I'm looking to get some of the functionality in the AC3 decoder from FFMPEG into a gstreamer application that I'm working on (specifically recent dialnorm and DRC changes), but not sure how compatible the current version is with gstreamer-1.0. What do I need to know? Thanks! [18:19] <lipizzan> is there a way to "tee" ffmpeg output to both write to ts stream to file disk and STDOUT (or socket?) as input to mplayer? [18:24] <kepstin-laptop> sure, have ffmpeg write to stdout, and pipe it to the 'tee' command to save a copy to disk [18:24] <kepstin-laptop> then pipe that to mplayer. [19:35] <c_14> lipizzan: https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Teepseudo-muxer [20:30] <jelle__> I have a origin.mkv with 1 fps and I would like to keep all the frames, but play them 60 fps. How do I do this? Keep all frames, but speed them up? [20:36] <__jack__> strange idea [20:40] <__jack__> you can still extract all frames as image, then copy all of them 60x, and recreate the movie [21:12] <jelle__> have a origin.mkv with 1 fps and I would like to keep all the frames, but play them 60 fps. How do I do this? Keep all frames, but speed them up? [21:13] <c_14> ffmpeg -i origin.mkv -r 60 out.mkv ? [21:14] <jelle__> no filters needed? [21:15] <c_14> shouldn't [21:19] <jelle__> ok, Ill fiddle with that, thanks [21:42] <Diogo> Hi this is possíble to generate hls m3u8 with fixes extinf? [21:49] <jjohn> Ahoy, guys. Just a quick question. [21:50] <jjohn> I got raw pcm_s16le streams and want to put them into a lossless container format. I read that FLAC might be appropiate, but the bitrate of the converted files is lower than that of the original ones. [21:50] <jjohn> I thought FLAC was lossless, wasn't it? Why is the bitrate lower then? [21:50] <c_14> pcm is uncompressed [21:50] <c_14> FLAC is compressed [21:51] <jjohn> But still lossless? And it's the ONLY difference? [21:51] <c_14> It's not the _only_ difference, but it is still lossless. [21:51] <c_14> Ever take a big file and compress it in rar/zip/7z/gz/bz2/xz ? [21:52] <c_14> Same concept. [21:52] <jjohn> No differences in quality either? [21:52] <jjohn> OK, I got it. Thank you very much then. [21:52] <c_14> No difference in quality (as long as the encoder doesn't start doing weird shit) [21:52] <jjohn> Thanks. You have been always a great help to me. [21:52] <jjohn> Bye then. [22:52] <ackwood> Hi everyone. What is "AVCODEC_MAX_AUDIO_FRAME_SIZE" known as these days? [22:52] <ackwood> Last time I remember being able to use it I had ffmpeg 0.8 [23:04] <Plorkyeran> it was removed entirely, not renamed [23:05] <ackwood> Plorkyeran, that sucks. I have a framework that depends on the existence of that variable. Is there a program way of coming up with that number? [23:05] <ackwood> I'm not exactly an expert on multimedia encdec, I'm just trying to maintain something useful [23:06] <Plorkyeran> it was removed because anything that used it was inherently broken and needs to be rewritten to do things correctly [23:07] <Plorkyeran> iirc doing so was fairly easy [23:07] <Plorkyeran> other than the utter lack of useful documentation... [23:09] <goulard> anyone know the rule of thumb on how many AAC ADTS packets to put in one TS packet? [23:14] <Sokolio> You mean PES packet? [23:14] <goulard> yeah [23:15] <goulard> I put one ADTS packet per PES, and quicktime and vlc wont play [23:15] <Sokolio> I guess there's no rule, even a rule of thumb [23:18] <Sokolio> PTS however must be set right according to sample rate [23:18] <Sokolio> does vlc complain in any way? [23:18] <goulard> it just says there is nothing to play [23:18] <goulard> i suppose the pts is correct, as the duration in quicktime is correct [23:19] <goulard> and media info reports properly [23:19] <goulard> pts = frame# * 1024 / samplerate [23:21] <Sokolio> does quicktime actually support adts packaging? [23:21] <goulard> yes [23:21] <goulard> you have to contain AAC in ADTS in TS [23:21] <Sokolio> in qt there is a 2 byte config [23:21] <Sokolio> and raw AAC [23:21] <Sokolio> yes, in TS that;s true [23:22] <Sokolio> well, I have no idea [23:23] <Sokolio> I've seen a single ADTS packet per PES and it sorta played [23:24] <goulard> do you know what buffer fullness is for adts? [23:24] <goulard> I had wondered if that had something to do with it [23:24] <Sokolio> you mean the header field [23:25] <Sokolio> well, I usually hardcoded it [23:25] <Sokolio> dunno if it wasnt all 1's in my case [23:27] <Sokolio> 7ff to be specific [00:00] --- Sat Sep 20 2014

1 0

ffmpeg-devel.log.20140918
by burek 19 Sep '14

19 Sep '14

[00:46] <cone-913> ffmpeg.git 03Vitor Sessak 07master:55d11d277bf5: swscale/x86: do not expect registers to be preserved across inline ASM blocks [00:50] <compn> Ffmpeg made Debian? Yay [00:53] <wm4> yes [00:53] <nevcairiel> well, experimental [00:54] <nevcairiel> until its at least promoted to unstable, i won't celebrate [00:55] <wm4> true [00:55] <wm4> well I have no clue about debian politics [00:55] <wm4> aren't they promoted semi-automatically? [00:55] <nevcairiel> i doubt it has any chances to still get into testing (ie. next stable), so it will be a while until it is wide spread [00:59] <cone-913> ffmpeg.git 03Andreas Cadhalpun 07master:739f179dd6a2: vf_deshake: rename Transform.vector to Transform.vec to avoid compiler confusion [01:00] <J_Darnley> 2.4 will gestate for 9 months till it gets birthed into unstable... [01:00] <J_Darnley> then it will mature over a few years to make stabele [01:00] <J_Darnley> *stable [01:11] <cone-913> ffmpeg.git 03Katerina Barone-Adesi 07release/2.4:b45ab61b24a8: apetag: Fix APE tag size check [01:11] <cone-913> ffmpeg.git 03Michael Niedermayer 07release/2.4:ab02548c8a71: Merge commit 'b45ab61b24a8f2aeafdd4451491b1b30b7875ee5' into release/2.4 [01:35] <iive> nevcairiel: going through the ftp masters is the hardest part. [01:41] <cone-913> ffmpeg.git 03Michael Niedermayer 07release/2.4:d694ab846cf1: avcodec/x86/vp9lpf: Always include x86util.asm [01:41] <cone-913> ffmpeg.git 03Andreas Cadhalpun 07release/2.4:dc4e34a2f059: vf_deshake: rename Transform.vector to Transform.vec to avoid compiler confusion [02:10] <cone-913> ffmpeg.git 03Jörg Krause 07master:3f2c70355ab7: configure: Use the right variables in check_host_cpp [02:10] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:343643fdca4f: Merge commit '3f2c70355ab722bc9f741bd3ed8224c7cfb62379' [02:33] <cone-913> ffmpeg.git 03Martin Storsjö 07master:2bb2c2bd75e5: rtpenc_chain: Pass the initial time_base hint on to the chained muxer [02:33] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:986f63d17fa1: Merge commit '2bb2c2bd75e5f4b28a945511cda77e0a1a44c758' [03:15] <kepstin> ooh, nice, the fate tests on my k6 finished. [03:16] <kepstin> assuming I'm interpreting the results correctly, they passed too. [03:19] <kepstin> took a little over 10 hours total for one run. [03:20] <kepstin> I suppose I should set it up to run automatically and submit the results, now :) [03:43] <pross> kepstin: k6-2 or iii? [03:48] <kepstin> pross: no [03:48] <kepstin> "AMD-K6tm w/ multimedia extensions" (i.e with mmx only) [04:17] <jamrial> kepstin: I'd say disable hevc, vp9 and similar codecs that are not relevant for pre sse processors if you really intend to run fate on that k6 [04:17] <jamrial> it will shave an hour or two of runtime [04:52] <cone-913> ffmpeg.git 03James Almer 07master:77f9a81ccaad: x86/me_cmp: combine sad functions into a single macro [05:05] <kepstin> yeah, I suppose without sse, there's no assembly in any of those that would run on this chip [07:28] <ubitux> michaelni: how did you spot that issue in vp9lpf? [07:47] <wudi> lpf? [07:47] <wudi> loop filter? [07:52] <ubitux> wudi: yes [08:01] <jamrial> nevcairiel: interesting commit http://git.1f0.de/gitweb?p=ffmpeg.git;a=commitdiff;h=4cc6c06fb7c298dbf081cc… :p [08:07] <sunye_> hi , everyone, I am Sun Ye ^ ^ [11:44] <michaelni> ubitux: for f in libavcodec/x86/*.o ; do echo $f ; readelf -S $f | grep GNU-stack || break ; done [11:44] <ubitux> noted [11:44] <ubitux> thank you [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:074eef56ef4a: fate: add fate-filter-pp1 [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:9f9ebe631dc1: libpostproc/postprocess: avoid some if() [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:44dabf1f422b: postproc: simplify forwarding return codes [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:921caf620386: postprocess: make some variables in pp_get_mode_by_name_and_quality() const [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:9e8be462521e: postproc/postprocess: use av_strtok() [14:55] <cone-41> ffmpeg.git 03Michael Niedermayer 07master:27f936eca8a1: Merge branch 'postprocwork' [14:55] <michaelni> j-b, please add *.mak to the files which may contain tabs in the ffmpeg git hook. We have tabs in *.mak files due to Makefile syntax and it requires each change to such files to be done as a merge currently, this is ugly [14:56] <michaelni> or tests/fate/*.mak if *.mak would only cover the "root" directory [15:00] <ubitux> heh [15:00] <ubitux> second time this happens :) [15:44] <Daemon404> cd .. [15:44] <Daemon404> woops [16:05] <kierank> wow that guy's first name is guerilla [16:05] <kierank> oh gorilla [20:25] <ubitux> do we enable PIC by default with shared builds? [20:27] <ubitux> (it seems to be the case after a simple test) [20:29] <wm4> I suppose most systems need PIC for shared libs [20:51] <wm4> how can I make ffprobe output as many CodecContext fields as possible? [20:52] <wm4> and formatcontext [21:01] <ubitux> what does that mean? [21:01] <wm4> there are lots of AVCodecContext fields that might be interesting somehow, and ffprobe should print them [21:02] <nevcairiel> invent some sort of reflection based code to automatically print the entire context [21:02] <wm4> also, oh god, mp3dec updates AVCodecContext->skip_samples on seeking [21:03] <nevcairiel> should send side data [21:03] <wm4> maybe utils.c actually does [21:03] <wm4> yep it does [21:03] <nevcairiel> thats ok then [21:03] <nevcairiel> not the cleanest solution, but it works [23:43] <weirddan455> hey is there anyone here who can approve me for the mailing list? i tried submitting a patch and it says i need approval [23:44] <J_Darnley> Someone will get to it. [23:45] <J_Darnley> If you're not subscribed emails get sent for moderation, or something. [23:51] <llogan> weirddan455: approved [23:53] <llogan> if you don't want to wait for the lazy moderator in a weird timezone then you can subscribe (but it could be considered a high volume list). [23:53] <weirddan455> llogan, ok thank you :) [23:59] <jamrial> weirddan455: you will not get the replies delivered to your email unless you're CCed (or you subscribe to the list) [00:00] --- Fri Sep 19 2014

1 0

ffmpeg.log.20140918
by burek 19 Sep '14

19 Sep '14

[00:00] <c_14> Yes, not sure if that's done automatically though... [00:01] <quebre> ok... [00:02] <kepstin-laptop> the -target option is already rescaling the video to the correct 720x480/5XX for the dvd; i'd expect that ffmpeg is probably guessing something reasonable for aspect by default to tell the player if stretching/squishing is needed. [00:02] <kepstin-laptop> but -aspect could override that if something goes wrong... [00:05] <quebre> alright [00:05] <quebre> how can i enable CUDA for ffmpeg? this conversion takes really long time ;P [00:11] <quebre> seems it got converted [00:11] <quebre> is it ok that original file had 1.73gb and output is 1.02gb ? [00:13] <kepstin-laptop> your original was a 720p h.264 video, the output is a standard definition mpeg2 video; sounds about right. [00:13] <quebre> oki [00:22] <quebre> is it ok that the resulotion after conversion is 1280x720 ? [00:23] <kepstin-laptop> huh, I thought the target option would set that.... I guess I was wrong [00:23] <quebre> ;X [00:25] <llogan> it should scale to 720x576 or 720x480 [00:25] <kepstin-laptop> yeah, the docs say it should be scaling [00:25] <llogan> (see ffmpeg_opt.c) [00:25] <kepstin-laptop> the -target option does the equivalent of setting -s, which will insert a scale filter if needed [00:25] <quebre> i did: ffmpeg cartoon.mkv -target pal-dvd dvd cartoon.mpg [00:25] <quebre> (im using 64bit ffmpeg uner windows 7) [00:26] <quebre> i meant ffmpeg -i (...) [00:26] <c_14> eh [00:26] <llogan> remover that superflous dvd [00:26] <c_14> ^that [00:27] <quebre> ffmpeg cartoon.mkv -target pal-dvd cartoon.mpg [00:27] <quebre> good ? [00:27] <llogan> sure [00:27] <quebre> ok [00:27] <quebre> here we go again [00:27] <quebre> that's the only downside. speed [00:28] <quebre> or lack of speed [00:29] <quebre> i know ffmpeg for longer time, i guess it's best converted in the world :p [00:29] <quebre> converter* [00:30] <llogan> kepstin-laptop: is the k6 done yet? [00:33] <Suchiman> quebre: just get a frickin 36 core system, ffmpeg nicely scales with CPU cores ;) [00:33] <quebre> yeah, one day.. when i get better job :PPP [00:34] <kepstin-laptop> llogan: still going. [00:34] <sacarasc> You're converting to MPEG2, I wouldn't be surprised if most of the CPU is being used for decoding. :p [00:34] <quebre> 95% cpu usage :D [00:34] <quebre> the ammount of ram does not help much i think [00:34] <kepstin-laptop> it's onto the filter-pixfmts tests now... [00:35] <Suchiman> quebre: you just need a motherboard that holds two of them: http://ark.intel.com/products/81061/Intel-Xeon-Processor-E5-2699-v3-45M-Cac… [00:37] <quebre> The Xeon E5-2699 V3 is listed for 3,960.26 EUR (...) [00:37] <quebre> right. [00:37] <Suchiman> i wonder how much cores x264 can utilize at maximum [00:42] <foonix> maybe video height / 48 [00:46] <quebre> ok the resolution is now 720x576 [04:18] <molikto> Hi, anyone here? [04:18] <molikto> question: Can I get metadata for a stream? because I want to get the rotation of a video but the metadata does not appears if I use ffmetadata [04:35] <molikto> is this channel dead? [04:36] <EvolE_> quebre: probably all are sleeping [04:36] <EvolE_> ... [08:46] <anshul_mahe> how to set explicitly maximum_number_of_frames in x264 [09:04] <K4T> frame=3407399 fps= 50 q=30.0 size=37510276kB time=18:55:47.97 bitrate=4509.1kbits/s DUP=14 DROP=0 [09:04] <K4T> what is that DUP and DROP? [09:04] <K4T> it is not always showing [09:05] <BtbN> duplicated and dropped frames. [09:05] <K4T> it means that 14 frames is duplicated in every second, yes? [09:06] <sacarasc> Total, I thought. [09:06] <K4T> ow, from ffmpeg start, yes? [09:06] <sacarasc> Yes, I think so. [09:07] <Mavrik> yes [09:07] <Mavrik> dup= - number of frames that were duplicated to keep up with desired framerate [09:07] <K4T> so, nothing to worry, 14 duplicated frames per 19h streaming [09:07] <Mavrik> drop= - number of frames tropped to keep up with farmerate [09:07] <K4T> Mavrik, thank you [09:07] <BtbN> more like dropped because of bad network connection [09:08] <BtbN> accidently encoding too many frames doesn't happen [09:08] <K4T> video card is input for ffmpeg, so I thjink it is not related to network environment [09:08] <BtbN> The input doesn't matter at all for dropping frames [09:28] <K4T> what will happen when I run ffmpeg on machine which have 2 CPUs, each with 4 cores? All CPUs and cores (8) will be used by that ffmpeg instance? [09:28] <sacarasc> Depending on the encoder, yes. [09:31] <anshul_mahe> I am using libx264 when I encode the video it sets log2_max_frame_num to 4 which means maximum frames as 2^4+4 = 256, since i concat that video with other large video [09:31] <K4T> sacarasc, I will use libc264 for video and aac for audio [09:31] <K4T> libx264* [09:31] <anshul_mahe> I need to set the value of log2_max_frame_num to 6 means number of frame 1024 [09:32] <sacarasc> K4T: That should use all 8 cores. [09:33] <anshul_mahe> If I directly use x264 then there is option called --frames and how to use that option in ffmpeg, I tried -x264opts but that fails for this option [09:47] <K4T> anshul_mahe, try -x264opts frames [09:48] <anshul_mahe> K4T: not working [09:48] <K4T> paste error [09:48] <K4T> and command line [09:51] <anshul_mahe> here is all you wanthttp://pastebin.com/NYbe9u7e [09:51] <anshul_mahe> here is all you want http://pastebin.com/NYbe9u7e [09:54] <K4T> what frames is supposed to do? I can not find doc [09:56] <anshul_mahe> from x264 help --frames <integer> Maximum number of frames to encode [09:57] <K4T> maybe you can use ffmpeg argument like -r or fps vilter? [09:57] <K4T> filter* [10:01] <K4T> anshul_mahe, pm [10:20] <relaxed> anshul_mahe: -frames:v 1024 [10:26] <okokokno> Hi. I'm trying to use libavformat but are noticing that for many supported formats, there's a lot of missing file information [10:27] <anshul_mahe> relaxed: not working, in actual I dont want to limit my output, I want to set log2_max_frame_num to 6 in SPS of H264 [10:28] <okokokno> for example, for mp3 files libavformat doesn't provide channel count and samplerate [10:28] <okokokno> do i have to parse the formats myself? [10:31] <okokokno> *The format's header info [10:38] <myubuntu__> Can I disable faac's three frames [10:38] <myubuntu__> Can I disable faac's three frames's buffering? [10:57] <K4T> is RAM important for ffmpeg enconding? [10:59] <__jack__> not really [10:59] <K4T> I think I will have max 8 instances of FFMPEG, each of them will create 4 streams. Inputs for each instance will be video card [10:59] <__jack__> depending on what you're encoding (source, size, format, codec etc), you 'll require a min of ram, but increase that value won't make it faster [10:59] <K4T> 8GB will be enough or I should buy 16GB? [11:00] <K4T> 720p quality [11:00] <__jack__> codec ? options ? you may test [11:01] <K4T> x264 for video and aac for audio [11:01] <__jack__> you should test [11:01] <__jack__> you must test ? [11:01] <__jack__> hum [11:02] <K4T> ok [11:03] <__jack__> If you want a really answer: get 16GB, it's not really more expensive, and may be usefull in the future [11:04] <K4T> yeah, 16 should be enough [11:04] <K4T> now I have one ffmpeg with one streamd and it takes 300MB [11:04] <K4T> I will get probably 16 streams [13:20] <xtruder> when i sent multi bitrate streams to my rtmp server, it recognizes only one. When i do the same with FMLE it recognizes them correctly. Is this due to ffmpeg not able to align keyframes? [13:34] <sekon> if i do ffmpeg -i /path/to/ip/file [13:35] <sekon> I get .... Stream #0:0(eng): Audio: pcm_s16le (sowt ..... [13:35] <sekon> so what does sowt mean ? [13:36] <c_14> > 'sowt' ('twos' spelled backwards) also denotes signed linear PCM. However, 16-bit data is stored in little endian format. [13:36] <c_14> AFAIK it's just a QuickTime identifier. [13:36] <sekon> c_14: thanks [13:37] <c_14> It's not like the le in pcm_s16le already stands for little endian or anything. Nah, we need more identifiers& [13:37] <Dark-knight> kerploop [13:39] <sekon> c_14: your help was greatly appreciated [13:39] <sekon> not just by me .. but by others here too [14:24] <wysaid> Anybody have the static release library of the latest ffmpeg? Please send me a link~~~~~~thanks a lot [14:28] <c_14> wysaid: the gusari builds haven't been updated since july, just stick with the johnvansickle builds. [14:29] <wysaid> sorry, I forgot to say that I am asking for the builds of windows platform... [14:30] <c_14> zeranoe [14:30] <c_14> http://ffmpeg.zeranoe.com/builds/ [14:31] <c_14> https://ffmpeg.org/download.html [14:31] <c_14> They're all linked there. [14:31] <c_14> For future reference. [14:34] <wysaid> I know that, I'm using the package for Dev, but it doesn't include the dll files. The program always report an error: " missing avcodec-56.dll" [14:34] <wysaid> The other downloads are 'exe' files [14:35] <wysaid> Can you tell me which one is for developing? Thanks a lot~ [14:35] <wysaid> I am wasting a lot of time here [14:37] <__jack__> that's so kind, thank you [14:37] <c_14> If I'm reading this correctly, you need the shared and the dev package. [14:39] <wysaid> Yes, I used to think the three downloads "static", "shared" and "dev" are for the same usage... [14:39] <wysaid> I'm downloading the shared version... [14:39] <wysaid> thanks~ [14:45] <wysaid> I found the DLLs! @_@b [14:47] <lipizzan> What would be an efficient console video player (framebuffer) to "tee" output to. [14:48] <lipizzan> .-? [14:48] <lipizzan> .->? [15:00] <c_14> lipizzan: I remember mplayer having framebuffer support. [15:05] <lipizzan> c_14: yes, i'm testing with it. I have an old version though, and i cant re-position the video on the screen (-geometry=x:y). [17:21] <lipizzan> when encoding dv->h.264->ts container file, is there some switch I should use to maintain audio\video sync? video races ahead of audio on my fresh compiled mplayer when playing the ts file (linux console, X-less). [17:25] <klaxa|work> how are you rendering the video? [17:25] <klaxa|work> is your computer fast enough to render the video without X? [17:26] <Mavrik> hmm [17:26] <Mavrik> lipizzan, are you somehow clobbering the timestamps on frames? [17:29] <lipizzan> klaxa|work: It;s <2KB/sec bitrate (-preset faster... when generated) , and cpu is < 30% during playback. [17:29] <lipizzan> Mavrik: Not sure. How would I find out? [17:29] <Mavrik> are you piping stuff around without containers? [17:30] <lipizzan> The work flow is dvgrab(dv stream)->ffmpeg(h.264->ts)->ts file. [17:32] <lipizzan> should I be using a bitsteam filter on the audio to when creating the h.264\ts? I use one on the video. [17:33] <lipizzan> should I be using a bitsteam filter on the audio also when creating the h.264\ts? I use one on the video. [17:43] <excalibr> HOw do you specify milisecond with -ss and -to param? [17:44] <Ess4_> -ss 1.23 [17:46] <Ess4_> excalibr, you can use -ss param like that : -ss 00:00:00.123 [17:48] <excalibr> thank you sir [19:36] <tab1293> if I am transcoding a mp3 file to aac what should I use as my output format specified with the -f flag? [19:40] <Mavrik> tab1293, mp4 [19:42] <okokokno> Hi. I've been here earlier and asked a question but later disconnected so I'll ask again, sorry. [19:43] <tab1293> Mavrik, I am getting this error Could not write header for output file #0 (incorrect codec parameters ?) [19:44] <tab1293> okay hold on [19:46] <okokokno> avformat seems to, for some formats like mp3, not provide any info about the file (e.g. channels, sample rate) [19:47] <Mavrik> okokokno, are you streaming the file? [19:48] <okokokno> Is there a way to get this info via the av libraries or do I have to parse the format header myself? [19:48] <Mavrik> also, how are you reading that data? [19:49] <okokokno> I wish to stream or otherwise read the file later, but first I wish to get its info [19:50] <Mavrik> you're not answering my questions. HOW are you trying to read the files that you don't get that info? [19:52] <tab1293> Mavrik here is the command and error http://pastebin.com/LrnedP9h [19:53] <Mavrik> tab1293, you need to write mp4 / m4a to file [19:53] <Mavrik> not pipe it somewhere. [19:53] <Mavrik> also. [19:53] <tab1293> Mavrik, why can't I pipe it? [19:53] <Mavrik> [libfdk_aac @ 0x29d2b20] Note, the VBR setting is unsupported and only works with some parameter combinations [19:53] <okokokno> I'll take a look and answer, thanks, Marvik [19:53] <Mavrik> tab1293, because index cannot be written before file is complete. [19:53] <Mavrik> tab1293, see, you get this error: [mp4 @ 0x29c42e0] muxer does not support non seekable output [19:54] <tab1293> yeah [19:54] <tab1293> so I can't transcode on the fly [19:54] <tab1293> with aac at least [19:54] <Mavrik> of course you can transcode on the fly. [19:54] <Mavrik> you just can't write it to MP4/M4A container. [19:54] <Mavrik> use something else. [19:55] <tab1293> Mavrik, let me explain what I am trying to do [19:56] <tab1293> I am trying to use python to transcode an mp3 to aac. So I have it set up that the mp3 file data is being passed to stdin using python's subprocess command. I would like to then take the stdout and write it to an amazon s3 bucket [19:56] <tab1293> So I am trying to work with the raw data in python so I can upload ffmpeg's output to amazon [19:57] <tab1293> what container supports on the fly transcoding? [19:57] <Mavrik> you're asking the wrong question. [19:57] <Mavrik> which containers must you support for delivery to the player? [19:57] <Mavrik> how are you going to play stuff? [19:57] <Mavrik> are you going to remux on delivery? [19:57] <Mavrik> when you have those answers, then you can choose a container :) [19:59] <tab1293> Mavrik, I was planning on using a container that is supported in most browsers hence the mp4 [19:59] <kepstin-laptop> and if the answer ends up being 'mp4', you're going to have to buffer the complete file before uploading it, for example by saving a temporary file on disk. [20:00] <tab1293> Mavrik, does mp3 support being written to a pipe? [20:00] <Mavrik> no idea [20:00] <tab1293> kepstin-laptop yeah I guess I can just make a temp file [20:00] <Mavrik> make a temp file and upload that [20:00] <tab1293> yeah alright [20:00] <Mavrik> browsers really don't support anything else than mp4 / mp3 [20:01] <Mavrik> so your choices are very limited in that regard [20:01] <Mavrik> sadly people are dragging their asses as hell with HLS support which uses MPEG2-TS container [20:01] <kepstin-laptop> mp3 in its "native" format, when not using vbr encoding, doesn't need a seek index, so you can stream it fine. If you use vbr, I think there's an optional xing header to add seeking, I'm not sure if it can be written at the end of the file or needs space reserved at the start. [20:02] <tab1293> Mavrik, I was actually thinking about trying to write a hls client in javascript. It seems like it could be done using aurora.js [20:02] <Mavrik> yuck :) [20:02] <Mavrik> we just used flash [20:02] <Mavrik> way more supported, way better experience for users :) [20:03] <tab1293> Isn't flash a nono nowadays? [20:35] <Mavrik> tab1293, why would it be? [20:35] <Mavrik> mobile platforms support HLS natively [20:35] <Mavrik> desktop people can use Flash because it still isn't as shitty as 5000 lines of JS decoder :) [20:36] <tab1293> Mavrik, mobile browsers support HLS natively? [20:36] <tab1293> how would you embed the HLS stream for mobile? [20:37] <Mavrik> ? [20:37] <Mavrik> you just point the video tag to m3u8 [20:39] <tab1293> Mavrik, hmm okay I didn't know that. So are there any open source flash hls players? [20:39] <Mavrik> I think JWPlayer has HLS module [20:39] <tab1293> but you developed your own player? [20:40] <okokokno> Hi Mavrik. I'm opening the mp3 file via avformat_open_input, providing a path. Is this what you meant by how I'm reading the data? [20:40] <okokokno> (Sorry it took so long to answer) [20:41] <Mavrik> okokokno, yes, and you're getting the data form AVStream right? [20:41] <Mavrik> tab1293, honestly I do backend work so I don't know which option the frontend guys took at the end :) [20:41] <Mavrik> I just deliver video ;) [20:42] <okokokno> I'm not getting any data before I'm trying to get the format's info [20:42] <okokokno> E.g. via <codec context>->channels, which is 0 [20:43] <Mavrik> uhm [20:43] <okokokno> I need the codec's/format's info long before I want to start reading the actual audio samples [20:44] <okokokno> Do I need to do something different? [20:45] <Mavrik> okokokno, you need to call avformat_find_stream_info on the codec context [20:45] <Mavrik> to probe the input and get all that data populated [20:45] <Mavrik> that will read your input up to probesize / analyzeduration and extract data [20:46] <okokokno> Ah [20:46] <okokokno> I'll try that, thanks [20:48] <EvolE> tab1293: for hls you can use http://www.flashls.org/ with videojs for example (or with any osmf player) [20:49] <tab1293> Mavrik, do you know if you use ffmpeg to create a HLS live stream with the +live flag do the segments get automatically deleted? [20:49] <Mavrik> I doubt it [20:49] <okokokno> Thanks, it worked! [20:49] <tab1293> EvolE, awesome thanks [20:50] <tab1293> Mavrik, so what does the +live flag change? [20:50] <Mavrik> hmm, I don't even see the option here: https://www.ffmpeg.org/doxygen/trunk/hlsenc_8c_source.html [20:51] <Mavrik> ah [20:51] <Mavrik> 00573 { "live", "enable live-friendly list generation (useful for HLS)", 0, AV_OPT_TYPE_CONST, {.i64 = SEGMENT_LIST_FLAG_LIVE }, INT_MIN, INT_MAX, E, "list_flags"}, [20:51] <tab1293> Mavrik, https://www.ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_… under options [20:51] <Mavrik> https://www.ffmpeg.org/doxygen/trunk/segment_8c-source.html [20:52] <Mavrik> it seems to just add this to m3u8: 00203 "#EXT-X-TARGETDURATION:%"PRId64"\n", seg->time / 1000000); [20:53] <tab1293> hmm okay [20:54] <tab1293> do you think adaptive bitrate streaming is useful when just streaming audio? I am trying to decide if my app should stream with hls or just transcode to m4a and serve them statically [20:55] <Mavrik> I doubt it's useful [20:55] <Mavrik> I even doubt the audio does enough bitrate difference for players to even switch [20:55] <tab1293> Both formats are supported by the popular browsers (hls with flash) and mobile so I am trying to decide if hls will give better performance [20:56] <tab1293> or any benefit over using m4as [20:57] <tab1293> so you think not? [21:34] <EvolE> tab1293: using hls should be ok if you plan to live stream and have a lot of concurrent listeners, because it's easy to cache on the server or cdn. not sure how it works with audio, but i suppose it should be ok [21:44] <Suchiman> is it just me or does ffmpeg rarely match the timestamps other players report? e.g. when i cut something with the timestamps from the players, the result will not be exactly what i've targeted [21:45] <jehar_work> Are you cutting h264 videos? There's the keyframes to keep in mind whenever you're slicing without doing a re-encode. [21:46] <Suchiman> jehar_work: the difference is up to 30 seconds so i don't believe it's just about keyframes, is it? i also tried moving -ss before and after the Input file [21:46] <Suchiman> but in this case it's slicing a vob file [21:46] <Suchiman> so mpeg2 [21:46] <jehar_work> Hmm, that'd be outside my head then. [21:50] <kepstin-laptop> vob files are mpeg-ps; a quick search reveals that at least some players (vlc and maybe gstreamer stuff?) have had issues with timestamps in those. Fixed by using the avformat (ffmpeg) demuxer. [21:50] <kepstin-laptop> so, make of that what you will :/ [21:59] <Suchiman> kepstin-laptop: well right now i try to repair a broken DVD Video, 2 titles won't read by MakeMKV but Play fine in VLC. copying them with mplayer (dvdnav just stops with error, dvd works) will break the titles even more. But i managed to put all vobs together using ffmpeg -i "concat:..." -c copy huge.vob (which plays fine) and now i'm trying to slice it by [21:59] <Suchiman> adding the title lengths 1-3 reported by mplayer to get the start Position of title 4 and then stream copy that Segment with fflags +genpts and -ss ... -t ... result.mkv which produces a playable mkv again. [21:59] <Suchiman> all fun stuff lol [22:02] <okokokno> I see AVCodecContext::block_align tells me the size of each packet read from the file but not decoded yet [22:03] <okokokno> Is there a similar value that tells me the decoded size of each packet, if known? [22:03] <okokokno> (For codecs where this is constant) [22:59] <active8> is there a syntax error here? ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1920x1080 -i :0.0 -a:c pcm_s16le -v:c libx264 -preset ultrafast -crf 0 -threads 0 09-14-2014-cast.mkv [22:59] <active8> Unrecognized option 'a:c'. Error splitting the argument list: Option not found [23:00] <lipizzan> active8: think that should be -c:a [23:00] <active8> yeah - copy pasted from article, duh! [23:00] <lipizzan> -c:a aka, codec:audio [23:02] <active8> right. now this: Invalid loglevel "libx264". $ffbin -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1920x1080 -i :0.0 -c:a pcm_s16le -v:c libx264 -preset ultrafast -crf 0 -threads 0 09-14-2014-cast.mkv Invalid loglevel "libx264". Possible levels are numbers or: [23:02] <active8> i tried it without the preset, too. I might not have the preset, but even without it, same error [23:03] <active8> $ffbin is a shortcut to a recent build [23:03] <active8> = ffmpeg [23:06] <Suchiman> active8: -c:v not v:c [23:07] <Suchiman> c:v = codec:vide [23:07] <active8> oh my. I must really need a shot to the head. lemme try that. [23:09] <active8> that's better. sorry, but thanks. [23:09] <active8> why, when there's no audio playing do I get: Non-monotonous DTS in output stream 0:1 ? [23:09] <active8> I sometinmes get that even with audio playing [23:28] <jehar_work> I'm doing some initial tests with x265 - it looks like I'm getting the same filesize with 1 pass as I am with 2 - is there any implication that I can get everything done with 1? [23:34] <kepstin-laptop> jehar_work: nah, that just means that their 1-pass bitrate algorithm is really strong. You should still prefer 2-pass, because it uses knowledge of future frames to create a file that's roughly constant quality [23:35] <kepstin-laptop> rather than dropping quality in hard bits and wasting file size on easy bits [23:36] <llogan> jehar_work: use -crf, one pass. or use -b:v with two passes if you are targeting a specific output file size. [23:36] <Suchiman> *thought that crf is better than two pass and two pass should only be used if e.g. constant file size is desired [23:36] <jehar_work> gotcha [23:37] <kepstin-laptop> with x264, crf and 2-pass are exactly equivalent, given the same output file size. I don't know if that holds for x265 [23:37] <Suchiman> what x_X [23:37] <llogan> i read 26*4* [23:37] <llogan> damned similar names... [23:37] <jehar_work> Me neither - I'm just scoping out the current time/filesize footprint currently. [23:38] <kepstin-laptop> x264's 2-pass algorithm works by doing one pass, using the results to calculate a crf value, and using that for the second pass. I have no idea what x265 does for that :/ [23:38] <jehar_work> Fortunately I can likely go "there's no huge gain at the moment, let's check again in 2 months" and get on with my day :D [23:39] <kepstin-laptop> well, to see the difference between 1- and 2- pass, you'd have to watch the output and see how the quality is [23:39] <kepstin-laptop> both should be able to get very close to the target bitrate in a good codec. [23:40] <jehar_work> Yeah, I've been having trouble checking the visual quality with the vlc nightly - is there a decoder that's working well? [00:00] --- Fri Sep 19 2014

1 0

ffmpeg-devel.log.20140917
by burek 18 Sep '14

18 Sep '14

[00:07] <cone-798> ffmpeg.git 03Lou Logan 07master:9c9aec33c0cf: doc/filters: add geq gradient examples [02:44] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:3c020b695959: avformat/avienc: Dont search for startcodes in inserted empty avi frames [03:22] <cone-798> ffmpeg.git 03Michael Niedermayer 07release/2.4:cc73b4f574b6: tools/crypto_bench: fix build when AV_READ_TIME is unavailable [04:52] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:9ffa705edd77: ffplay: try to fix build failure on apple [09:44] <cone-798> ffmpeg.git 03James Almer 07master:d1c49bcae9b7: avfilter: remove obsolete FF_API_ACONVERT_FILTER cruft [09:56] <cone-798> ffmpeg.git 03Paul B Mahol 07master:f1a8943ddd15: doc/filters: remove aconvert [10:25] <abaudoux> Hello [10:29] <abaudoux> I run into an error when using the hls format [10:29] <abaudoux> I think it may be a bug [10:30] <abaudoux> Here is the command:http://pastebin.com/JmUXNhWz [11:04] <ubitux> abaudoux: please provide the file(s) in the ticket [11:04] <ubitux> so it can be reproduced [11:12] <abaudoux> Its quite big, can I upload it somewhere? [11:15] <ubitux> see http://ffmpeg.org/bugreports.html [11:15] <ubitux> "Submitting Sample Media" [11:15] <abaudoux> thx [11:22] <abaudoux> uploaded on ftp as ticket_3957.ts [11:39] <abaudoux> Regarding #3957, I am in fact capturing a live udp stream, to generate HLS. But ffmpeg always stops after a few seconds and complains always with this same error. [11:41] <cone-798> ffmpeg.git 03James Almer 07master:0456d169c469: x86/me_cmp: port mmxext and sse2 sad functions to yasm [13:12] <ubitux> eerr wait [13:12] <ubitux> michaelni: you sure about the me_cmp stuff [13:12] <ubitux> im sorry i hadn't had a look [13:13] <ubitux> but typically the sad8 is not height variable [13:13] <ubitux> why it was originally [13:13] <ubitux> the h arg is ignored [13:15] <ubitux> (that's one of the reason it was reimplemented differently in pixelutils) [13:23] <michaelni> ubitux, i missed that the 8x8 functions lost h :( [13:23] <ubitux> he probably forked it from the pixelutils [13:23] <ubitux> where the width are fixed [13:26] <ubitux> the number of xmm reg looks off in sad16_y2 [13:26] <ubitux> well i should probably review it a bit [13:29] <ubitux> mmh actually no [13:29] <ubitux> it's in the mmx scope so shouldn't matter [14:10] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:85f2c0124d89: avcodec/x86/me_cmp: fix sad8xh [14:52] <ubitux> oh there is also a sad for alpha? [15:05] <Daemon404> ubitux, oh, you mean the arch? [15:05] <ubitux> yes [15:05] <Daemon404> i was confused for a sec thinking "why is alpha channel SAD special" [16:05] <J_Darnley> Which moon is out of alignment today? [16:05] <J_Darnley> (Why is ffmpeg just recieveing black frames? What did I change to cause this?) [16:33] <someone-noone> How can I use ffmpeg to capture UTF8 window title with gdigrab device? [17:24] <J_Darnley> Ha! What an idiot I am. I was trying to draw lines of zero width [18:02] Action: kepstin-laptop waits for the fate tests to finish running on his 266mhz k6. [18:28] <J_Darnley> how long has it been running? [18:59] <kepstin-laptop> J_Darnley: started it about 4 hours ago [20:08] <cone-913> ffmpeg.git 03Diego Biurrun 07master:2dd09ebf5597: cmdutils: Print a more sensible message in show_filters() w/o libavfilter [20:08] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:9ac08d93b2aa: Merge commit '2dd09ebf5597fd5e4a573bee9cccf21ae821f286' [20:21] <cone-913> ffmpeg.git 03James Almer 07master:fffc9a077d39: avfilter: remove obsolete FF_API_FILL_FRAME cruft [20:26] <llogan> whay was aconvert removed from doc/filters? f1a8943 [20:31] <jamrial> because the filter was also removed [20:31] <jamrial> check d1c49bca [20:37] <llogan> jamrial: i see. i missed that. [20:38] <cone-913> ffmpeg.git 03Katerina Barone-Adesi 07master:44caf99ecae7: fate: Add tests for RealAudio 1.0 (14.4) and RealAudio 2.0 (with 28.8) [20:38] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:a4cefc25ed83: Merge commit '44caf99ecae7bc1b907fab849ecaa72dd340ba2e' [20:38] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:43633c5388d3: fate: disable fate-ra4-288, the code uses floats and does not produce the same results on different platforms [20:49] <cone-913> ffmpeg.git 03Katerina Barone-Adesi 07master:c5560e72d0bb: apetag: Fix APE tag size check [20:49] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:8c71de762b84: Merge commit 'c5560e72d0bb69f8a1ac9536570398f84388f396' [21:20] <ubitux> wtf is the guy doing with this libffmpeg.a :D [21:34] <ubitux> < ChocolateArmpits> Ok, made a few tests myself, the speed increase is by about 4%-5%. [...] [21:34] <ubitux> < koda> ok, hardly noticeable :) [21:34] <ubitux> ??? [21:45] <iive> ubitux: yeh, that is a lot for overall speed increase. [22:16] <ubitux> http://www.hydrogenaud.io/forums/index.php?showtopic=106911 [22:25] <iive> ubitux: these test seems way too perfectly grouped to be real... [22:26] <ubitux> i like the confidence interval of faac [22:26] <ubitux> :D [22:29] <J_Darnley> Good. More evidence that lame -b 320 is idiotic. [22:30] <microchip_> J_Darnley: pfff! they want lossless now because they clearly hear the difference ;) [22:31] Action: J_Darnley uses flac [22:34] Action: ubitux as well [22:34] Action: microchip_ pities... [22:40] <J_Darnley> I'm pleased by the conclusions further down the page. [22:41] <J_Darnley> Some of which can be summed up with "stop fucking with the settings and it will give a good result" [23:30] Action: kepstin-laptop continues to wait for his k6 box to finish running fate. [23:58] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:41d82b85ab0e: avcodec/x86/vp9lpf: Always include x86util.asm [00:00] --- Thu Sep 18 2014

1 0

ffmpeg.log.20140917
by burek 18 Sep '14

18 Sep '14

[00:40] <lipizzan> llogan: I've done some console-only testing. There does seem to be some modest benefit, in that networking and all that other cruft isn't running. [00:42] <lipizzan> llogan: However, when i run the full command that deinterlaces and overlays a logo and some other things that I'm not certain of, the counters at the bottom of the screen become strange. [00:43] <lipizzan> llogan: even with -preset fast , I think I'm still cpu bound when ovelayong and deinterlacing. [00:55] <rickbol> llogan: http://pastie.org/9560289 [01:08] <rickbol> added the closing stats: http://pastie.org/9560308 [01:26] <rp__> Hey, can I make ffmpeg ignoring all input errors? I've tried -er/-err_detect options with vaious flags with no success. I'm trying to run couple of input streams and bandwidth goes over maximum of my connection giving "Invalid data found when processing input". Is it possible to skip errors caused by poor connection while repeating (a:copy, v:copy) stream? [01:26] <rp__> Any ideas? [01:44] <relaxed> rickbol: did you add the #comments afterwards? [01:54] <lipizzan> relaxed: they're from the perl script that runs this process. I left them for reference. [01:55] <RyanV> why does even a successful response from running ffprobe or ffmpeg write to sterr? [02:04] <relaxed> lipizzan: I assume you saw "/home/rickbol/vids/cfs/img/cfsbug-x1.png: Input/output error" [02:05] <Timothy__Gu> michaelni: can you op me on #ffmpeg? [02:10] <lipizzan> relaxed: I saw that, but I don't know what to make of it. The png exists, and is successfully overlayed into the output ts file. [02:11] <michaelni> Timothy__Gu, done [02:11] <Timothy__Gu> michaelni: thanks! [02:14] <Hello71> i think you have room now for the log link [02:15] <c_14> I think it should be "FFmpeg is in Debian experimental now thanks to Andreas Cadhalpun and Niv Sardi! \o/" [02:15] <c_14> There's a "now" too many. [02:15] <Timothy__Gu> c_14: fixed thanks! [02:16] <Suchiman> there are way too many things that are like ffmpeg but aren't ffmpeg [02:16] <Suchiman> such confuse [02:22] <relaxed> lipizzan: then what's the problem? fps is the encoding speed [02:31] <lipizzan> relaxed: but I'm inputing a 29.97fps dv stream, so how is it processing 45fps? if I remove the logo overlay, it processes ~30fps and maintains a lower bitrate. [02:41] <relaxed> you can attribute the higher bitrate to the image, but I'm not sure about the fps difference [02:41] <llogan> it's yadif [02:42] <llogan> mode 1 will output one frame for each field [02:43] <llogan> you can try mode 0 and see which one looks best for your content [02:43] <llogan> using mode 1 will effectively double the frame rate [02:44] <llogan> hence that's why it looks like it is encoding "faster" [02:44] <lipizzan> I'll try mode 0. [02:45] <llogan> and filtering can definitely slow things down depending on the filtergraph, the input, the encoder, etc [02:47] <lipizzan> is mode the first or second parm? yadif=x:y ? [02:47] <llogan> http://ffmpeg.org/ffmpeg-filters.html#yadif [02:53] <lipizzan> well fps=30, and cpu is not as consumed. [02:53] <llogan> why scale to 640x480? [02:56] <lipizzan> llogan: it's DV, and I know DV pixels aren't square, but in the old ffmpeg I used, I had to do that to make circles round. It's an issue now, cause my logo isn't round anymore. What should it be? [02:56] <lipizzan> llogan: just don;t scale at all? [02:57] <llogan> your player didn't display it correctly? [02:57] <llogan> i mean the native size and aspect [02:58] <lipizzan> vlc [02:58] <llogan> if you wan tto scale the i recommend using the scale filter instead of -ss. maybe right after yadif. that way you can control when and where it actually performs the scaling [02:59] <lipizzan> wow, removing scaling gave me back another 25% cpu! [03:00] <llogan> you may not need hqdn3d. it's a denoiser and it's slow. [03:01] <lipizzan> llogan: yea, I wondered what that was. [03:05] <lipizzan> actually, the hqdn3d seems to have to reduce cpu a little with my current image (which is dark and noisy). [03:06] <llogan> you can use it if you like the result [04:32] <jpleau> oo, ffmpeg into debian experimental, shiny [07:36] <tacocat> hi, i'm trying to do a cross compile for 64-bit windows from a ubuntu 14.04 install, but i'm having a hard time getting configure to detect any external libraries [07:36] <tacocat> the command i'm running is: ./configure --prefix="$HOME/ffmpeg_build" --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-libmodplug --enable-libgsm --enable-libtwolame --target-os=mingw32 --arch=x86_64 --cross-prefix=x86_64-w64-mingw32- --enable-shared [07:37] <tacocat> right now it's failing with ERROR: libass not found, but if i take that off, it just fails on to the next one (ERROR: libfdk_aac not found) and so on [07:38] <tacocat> the libraries (libass-dev, libfdk-aac-dev, etc.) I installed from ubuntu's repositories, along with the mingw packages [07:38] <tacocat> i'm not sure what's wrong, googling hasn't been very helpful [07:40] <ubitux> tacocat: can you look at the end of the config.log? [07:40] <tacocat> yeah, one sec [07:40] <ubitux> maybe you just lack pkg-config [07:42] <tacocat> http://pastebin.com/EZi2f8np i take off --enable-libass and it just fails with the next library [07:42] <tacocat> pkg-config is installed according to apt [07:42] <ubitux> ah missed the x compile part [07:43] <ubitux> yeah well you need these libs for windows [07:43] <tacocat> oh okay [07:43] <ubitux> no idea how that's supposed to work [07:43] <ubitux> don't ask me more :) [07:44] <tacocat> oh, so all the dependency libraries have to be cross compiled too? :s [07:44] <tacocat> gah [07:51] <tacocat> i'll try again tomorrow, good night :) [07:57] <rcombs> https://drive.google.com/file/d/0B6MFALddgIQ_MTVGaGhvUkkteGc/edit?usp=shari… <-- so, ffmpeg and mediainfo disagree on this file's frame rate, and my instinct is to call mediainfo buggy, but I'd like to double-check; anyone who knows the format (mpeg2 video in .mpg) better than me want to take a look? [08:28] <Nosomy|off> mediainfo says 24000/1001 for progressive, 30000/1001 if read as interlace. [08:28] <Nosomy> rcombs [08:29] <rcombs> Nosomy: hmm, interesting [08:29] <rcombs> so sounds like their bug, then [08:30] <Nosomy> nop [08:30] <Nosomy> isn't a bug. [08:30] <rcombs> oh? [08:30] <Nosomy> is little diff of detects [08:31] <rcombs> well, which is correct? Or is it encoded using some weird telecine hinting or something? [08:34] <Nosomy> is the telecine [08:35] <Nosomy> look: http://pastebin.com/sW8v7ZHH [10:14] <tedy> could any one here enlighten me how avcodec_register_all(); registers or specifically finds the codecs available please? [10:19] <tedy> sorry i got that one all codecs are listed in allcodecs.c [10:21] <tedy> what i seem not to understand is how avcodec_find_encoder() finds codecs especially if codec an external one like libx264 [10:32] <abaudoux> Hello [10:32] <anshul_mahe> hello [10:32] <abaudoux> I run into an error when using the hls output format with ffmpeg [10:33] <abaudoux> I think it may be a bug [10:33] <abaudoux> http://pastebin.com/JmUXNhWz [10:34] <abaudoux> I tried with -bsf:a aac_adtstoasc [10:34] <abaudoux> but It doesnt change anything [10:36] <anshul_mahe> is your inpu file play fine with ffplay [10:36] <anshul_mahe> *input [10:40] <abaudoux> i trid with vlc and it plays fine, I can try with ffplay too [13:50] <rsdrsdrsd> I am using yadif for deinterlacing, but I don't want to deinterlace the non interlaced videos, but I can't detect if a video is interlaced or not.... should I use as filter yadif=0:-1:0 or yadif=0:-1:1 ? [13:51] <rsdrsdrsd> What I read from the docs is that I should use yadif=0:-1:1 , but from some tutorials online I read yadif=0:-1:0 ? [14:02] <Taconut> Hi! I've run into a problem with FFMPEG. I'm converting an MP4 to an MP3, but I need to know the length of the MP3 first, so I can set the content-length header. If it helps, I have the bitrate. [14:02] <Taconut> *Audio Bitrate [14:05] <wodim> another youtube-to-mp3 site? [14:07] <Taconut> Yes and no [14:07] <__jack__> bitrate nor content-lenght will tell you the "length" (= time) of the track [14:07] <__jack__> s/will/won't/ [14:07] <Taconut> I need content-length [14:08] <__jack__> what is the output of ffmpeg -i source.mp4 ? [14:08] <Taconut> Also using regex in conversations = super badass [14:08] <Taconut> It reports incorrect lengths [14:08] <Taconut> The duration is like a couple ms [14:08] <__jack__> anyway, it will be hard for you to guess the exact mp3 overhead [14:08] <Taconut> I got the length in seconds and the bitrate, but I'm like +/-10 seconds off [14:09] <__jack__> the most simple is to process the file on a temporary file, then get the size etc [14:09] <Taconut> Is that any faster than writing the file to a buffer, and getting the size of that? [14:09] <Taconut> Because that's around 60x slower [14:10] <__jack__> what is slower than what ? file vs buffer ? [14:10] <__jack__> you are encoding, or just remuxing ? be sure to set the codec to "copy", right ? [14:10] <Taconut> I'm encoding [14:11] <Taconut> M4A->MP3 [14:11] <__jack__> owh, yep [14:11] <Taconut> Streaming the file directly to the client/browser is 60x faster than proccesing it into a buffer first [14:12] <Taconut> Does FFMPEG -i give me the size? [14:12] <Taconut> I don't think I saw it anywhere [14:12] <Taconut> But I do know the length isnt working [14:12] <__jack__> It will never give you the encoded length before encoding [14:12] <Taconut> *facepalm* that was a stupid question sorry [14:13] <Taconut> Hmmm... Are there any workarounds? [14:14] <__jack__> how works your project ? [14:15] <Taconut> I'm getting an M4a-encoded video as a stream or url in Node.js and using FFMPEG to convert that into an mp3 and pipe it out to the browser [14:15] <Taconut> I /could/ do something with the streams so I'm left with just urls [14:16] <Taconut> But FFMPEG reports incorrect information when using remote URLS with -i [14:16] <Taconut> And downloading anything into a temp file slows it way down [14:16] <__jack__> you have a sample ? [14:16] <__jack__> (sample uri) [14:16] <Taconut> K one sec. I'll find one [14:19] <Taconut> Hmmm... Now its stopping half-way [14:19] <Taconut> At least one output file must be specified [14:20] <Taconut> Its not even giving me the duratio [14:20] <Taconut> I'm doing something stupid [14:23] <anshul_mahe> How to check whether h264 video is interlaced or progressive [14:25] <Taconut> Okay [14:25] <Taconut> The following video reports 00:00:10.56 as the duration [14:26] <Taconut> http://download.wavetlan.com/SVV/Media/HTTP/H264/Other_Media/H264_test7_voi… [14:26] <Taconut> This was the first MP4 I could find [14:26] <Taconut> I dont even know wtf it is [14:29] <__jack__> Taconut: the lenght is 10s 565ms, really :) [14:29] <Taconut> Hey it is working! [14:29] <Taconut> lol sorry. I didn't see that [14:30] <Taconut> Well i guess I'm good then. Multiplying this by (128 * bitrate) should give me the bytes? [14:30] <Taconut> 128 = 1024/8 = bytes/kilobit [14:31] <__jack__> I don't think so, bitrate are not constant [14:31] <Taconut> So then what can I do? [14:31] <__jack__> and you may want the real and exact value [14:31] <Taconut> That I do [14:31] <Taconut> I'm assuming trying to flatten the bitrate wuld have searious quality impacts [14:48] <fling> Hello. [14:48] <fling> -f video4linux2 -s 864x480 -r 15 -i /dev/video0 [14:48] <fling> but I have 25 fps! [14:49] <fling> http://dpaste.com/1QQ43F7 [14:49] <__jack__> try ffmpeg -i blabla -r 15 output.avi, nah ? [14:51] <fling> __jack__: yay! working [14:52] <fling> http://dpaste.com/3YS28M8 [14:52] <fling> __jack__: does this look right? [14:52] <fling> __jack__: should not I change any other args order? [14:57] <__jack__> -threads 0 is strange [14:58] <__jack__> the rest is fine [14:58] <__jack__> forcing the aspect may be useless (native aspect is betterà [14:58] <__jack__> ) [14:58] <__jack__> depending of your needs [15:04] <fling> __jack__: how much threads do I need for 6 cores [15:15] <fling> __jack__: I'm encoding theora :D [15:16] <fling> __jack__: because I'm playing captured video over sshfs on an old pentium3 :< [15:18] <__jack__> fling: remove the -threads, it will take all usable core [15:18] <__jack__> dunno if such encoder is multithreaded, anyway [15:19] <fling> but atleast it plays on pentium3 [15:19] <fling> I removed crf and threads [15:19] <fling> just -c:v theora \ [15:19] <fling> __jack__: hmm hmm how to capture fullhd from the camera? [15:20] <fling> __jack__: do I need to enable x264 in v4l2 somehow? [15:20] <__jack__> v4l2 is the source format, x264 may be the output format [15:20] <__jack__> if you cpu is not power enough, you must not encode at runtime [15:21] <__jack__> may require a lot of disk space & IO [15:21] <__jack__> you've to try [15:21] <fling> __jack__: encoding cpu is phenom x6 1100T [15:21] <fling> __jack__: so how to get fullhd off a webcam? [15:22] <__jack__> ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video0 output.mkv ? [15:22] <__jack__> said the doc [15:22] Action: fling is asking v4l guys [15:25] <anshul_mahe> is there any good text about croma in yuv [16:23] <JodaZ_> has anyone taken a look at what changes PLEX has in their customized ffmpeg? [16:33] <someone-noone> How can I use ffmpeg to capture UTF8 window title with gdigrab device? [17:35] <deu__> Hi, anybody here? I am having hard time compiling one application which uses ffmpeg libraries. It complains that theres no r_frame_rate in AVStream. [17:35] <deu__> Is it safe to replace it with avg_frame_rate? [17:51] <Nykee> \0/ [17:51] <Nykee> | [17:51] <Nykee> \0/ [17:51] <Nykee> | [20:29] <llogan> JodaZ_: rcombs may know [20:30] <llogan> oh, he did explain some to me, now that i remember. trawl the ffmpeg-devel IRC logs from the last month or so [20:49] <llogan> JodaZ_: http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2014-August/002308.html starting around [20:13] [20:56] <Nomad145> Why did avconv fork from ffmpeg? [20:59] <llogan> Nomad145: http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html [20:59] <JEEB> Nomad145, you mean Libav. And it's because Old FFmpeg sucked. Both Libav and FFmpeg are better than Old FFmpeg [21:52] <Phlarp> is it possible to do font smoothing with ffmpeg drawtext? [21:53] <llogan> Phlarp: perhaps ft_load_flags option? http://ffmpeg.org/ffmpeg-filters.html#drawtext [21:54] <Phlarp> no_bitmap? [21:54] <llogan> i don't know. i've never used any of those. [21:55] <Phlarp> alright, I'll fiddle with them. Thanks [21:55] <llogan> "For more information consult the documentation for the FT_LOAD_* libfreetype flags. " [23:12] <zzz_> There seems to be a couple different ways to get rid of telecine. What is the most recommended method? [23:21] <kepstin-laptop> zzz_: a bunch of the detelecine filters in ffmpeg are really new, so I dunno if there's any good recommendations :) [23:22] <kepstin-laptop> zzz_: that said, i'd probably go with the pullup filter followed by dejudder filter, if there's no mixed interlaced content (i.e. it's pure telecine) [23:27] <quebre> hello, how can i burn BDRip to DVD so i can play the DVD CD in my DVD player that is connected to TV ? [23:28] <c_14> Use ffmpeg with -target dvd and then write it with some dvd writing tool of which there are a plethora. [23:41] <quebre> c_14: could you suggest me one good writting tool ? ... [23:44] <c_14> I've never used one before. [23:44] <c_14> I'd probably just pick whatever you can find in your repos. [23:44] <quebre> btw if i do the ffmpeg -target dvd, it will be converted to the DivX format? my DVD player have the "DivX" icon printed on front.. [23:45] <kepstin-laptop> quebre: no, it'll be converted to a format compatible with standard dvds (which is mpeg2) [23:45] <Suchiman> *completed reading the ffmpeg libav situation, facepalms* [23:45] <quebre> kepstin-laptop: so hmm, if my player says DivX then it will handle mpeg2 ? [23:47] <kepstin-laptop> if your player plays dvds, it will handle mpeg2, because that is required in order for it be called a dvd player. [23:47] <quebre> alright... ;) [23:47] <quebre> i think :P [23:49] <quebre> so full command will be: ffmpeg -in cartoon-720p-BluRay.x264.mkv -target dvd -out cartoon.mkv ? [23:51] <c_14> eh [23:51] <c_14> ffmpeg -i cartoon.mkv -target dvd cartoon.mpg [23:51] <quebre> oh ok [23:51] <c_14> Oh, if you need pal or ntsc, preface dvd with pal- or ntsc- [23:51] <quebre> how can i preface ? [23:52] <quebre> im not good with all this ;( [23:53] <c_14> -target pal-dvd or -target ntsc-dvd [23:53] <c_14> preface as in put in front [23:53] <c_14> prepend so to say [23:53] <quebre> ok thanks ! [23:54] <quebre> seems it working :D [23:55] <quebre> i will use CDburnerXP to record it, hope it works... [23:57] <quebre> i read http://www.kevssite.com/2009/02/22/converting-a-video-file-to-dvd-with-ffmp… , there is also '-aspect' option, if i provide 16:9 or 4:3, i mean if i select wrong option will my dvd cd wont play ? or it will play whatever option i choose ? [23:58] <c_14> I'm not sure you need it. Worst case if you don't supply it or supply the wrong one your video will either be stretched or squished. [23:58] <sacarasc> Technically, you're meant to have it in 4:3 and anamophically make it 16:9. [00:00] --- Thu Sep 18 2014

1 0

ffmpeg-devel.log.20140916
by burek 17 Sep '14

17 Sep '14

[00:03] <iive> so, ffmpeg have been 6 months in the queue, but it shows it have been there for 3 hours debian time. how long have the oldest package been there if it shows 1 year in debian time? [00:03] <ubitux> actually probably 1yr because there is only one version for those [00:05] <ubitux> most packages only have one version so it's fairly safe to assume that ffmpeg is part of the oldest [00:06] <iive> i'm just kidding... but yeh... 1 year. [00:06] <ubitux> i count less than 10 packages with multiple versions, and most of them have very close micro numbers [00:07] <ubitux> so ffmpeg is probably between pos 5 and 10 [00:07] <iive> well... there is nothing showing that... is there? [00:08] <llogan> anyone planning on making a news entry for 2.4? [00:11] <llogan> these new headphones aren't as good as i hoped (ATH-M40x). guess it doesn't really matter since they're just for field monitoring. [00:13] Action: llogan goes back to the Alessandro MS-1 [01:14] <michaelni> llogan, i was planning it but it appears i forgot [01:23] <michaelni> llogan, just posted a patch for 2.4 news, dont hesitate to comment or correct, ill probably push it soon [01:27] <llogan> michaelni: 14th July? [01:29] <cone-192> ffmpeg.git 03Michael Niedermayer 07master:b76d6132e385: avformat/network: move sockaddr_union after sockaddr_storage compatibility code [01:30] <michaelni> September ... :) [01:30] <llogan> otherwise, LGTM. [01:31] <llogan> thanks. i was thinking of writing one, but I'm at work now. color correcting/grading is so boring. [01:32] <llogan> ...if you were wondering why i asked but didn't just do it [04:10] <cone-192> ffmpeg.git 03Michael Niedermayer 07master:aaeae281a898: avformat/mpegtsenc: More verbose warning messages in case of missing video ES startcodes [04:10] <cone-192> ffmpeg.git 03Michael Niedermayer 07master:688a40b4ed7d: avcodec/h264_sei: ff_h264_decode_sei: dont try to parse trailing zeroes [07:14] <ubitux> oooh [07:15] <ubitux> that's a good morning news :o [07:20] <Timothy__Gu> Good night :) [07:33] <jamrial> Timothy__Gu: why did you drop the "Discussions about..." part from the channel topic? [07:33] <Timothy__Gu> jamrial: it's too long. Plus it's already covered in "FFmpeg dev channel" [07:34] <Timothy__Gu> I really need to sleep now do don't ask me any more questions please. Thanks [07:36] <Timothy__Gu> https://tracker.debian.org/news/570904 is the key upload BTW [11:24] <ubitux> should we consider the interfaces with cli (and similar AVOption stuff) as API and bump accordingly the "FFmpeg bundle" when they change? [11:25] <nevcairiel> isnt the whole avoption stuff specifically designed to avoid issues that API would usually have [11:25] <ubitux> it's still an API [11:26] <ubitux> it's not C, but still a user interface that can be used for programming [11:26] <ubitux> typically scripts [11:26] <ubitux> and we actually try to keep it by not renaming/delete these options randomly [11:27] <ubitux> having a 2.4 3.0 bump could mean that some options somewhere (in whatever libs) got changed [11:27] <ubitux> of course, we would need to document it [11:28] <ubitux> (changed = renamed or deleted) [11:32] <ubitux> it doesn't happen often; the "draw" option in drawtext is one example [11:38] <cone-798> ffmpeg.git 03Michael Niedermayer 07release/2.4:8ac3b2cdb751: RELEASE_NOTES: add H.261 & HEVC RTP support [11:39] <ubitux> BTW, unrelated, is there a special tag or whatever marker for "backport relevant"? [11:44] <michaelni> no but could make sense to add such a thing, what i do is generally backporting stuff locally when i do the work in master [11:47] <iive> ubitux: imho, bump only when option is removed or changed/renamed. if old scripts work, no need for bumps. [11:48] <ubitux> iive: that's what i suggested [11:49] <iive> but... what number are you going to bump and why? [11:51] <ubitux> we are currently at 2.4. When we decide to make a new release, we check if we renamed some option or anything user interface relevant (like ffprobe output change or whatever). If so, it will be 3.0, if not 2.5. [11:52] <ubitux> if we have a FF_API compat layer, we don't need to, until the next bump [11:58] <iive> so, you propose a special number for cli options? [11:58] <ubitux> a propose a non-random way of defining the version for the bundle [11:58] <ubitux> i* propose [11:59] <ubitux> 1.x 2.x was arbitrary [11:59] <ubitux> like, hey that's cool let's bump [11:59] <ubitux> but no meaning whatsoever [12:00] <ubitux> 2.3 2.4 no major bump even though we major bump the libraries leading to a API change in drawtext ("draw" option dropped) [12:09] <nevcairiel> i think any version change in the package is considered "major", as the numbering is arbitrary [12:10] <cone-798> ffmpeg.git 03Diego Biurrun 07master:2d589273dd36: configure: Split adding of CFLAGS and CPPFLAGS for hardened toolchain [12:10] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:5ddfac81c4a9: Merge commit '2d589273dd36c5eb271a035ea0e669b64dae257f' [13:01] <wm4> ubitux: I'd consider AVOptions API [13:01] <wm4> definitely [13:02] <ubitux> we already do, in the library themselves [13:02] <ubitux> but for example, a major bump might *not* cause an api user bump from the CLI PoV [14:09] <wm4> so ffmpeg 2.4 made it into debian experimental [14:09] <iive> yes it finally did. [14:25] <ubitux> libav distributing ffmpeg tshirts? we need to print libav tshirts to trade with them! [14:26] <wm4> they're violating your trademark rights, you must start thermonuclear war with them [14:32] <iive> well send polite green men first. they are very polite. [14:59] <cone-798> ffmpeg.git 03Stefano Sabatini 07master:3a0c70f1e374: doc/examples/transcoding: use av_packet_rescale_ts() [16:08] <thardin> does anyone in here know what kind of convention DFD is? [16:17] <J_Darnley> No. Any more context? [16:18] <av500> http://en.wikipedia.org/wiki/Dog_Fashion_Disco [16:19] <J_Darnley> "Donald Fauntleroy Duck, the full name of Donald Duck" [16:19] <nevcairiel> he has a middle name o.o [16:19] <nevcairiel> fascinating [16:26] <thardin> J_Darnley: not really. a friend of mine heard of the acronym [16:27] <nevcairiel> in multimedia i only know DSD, DXD and DVD [16:32] <cone-798> ffmpeg.git 03Steven Liu 07master:2c21e8b1a490: lavf/hls: add hls_ts_option [16:57] <cone-798> ffmpeg.git 03Clément BSsch 07master:de0e49c2450f: avformat/flacenc: use av_clip() instead of av_clip_c() [16:57] <cone-798> ffmpeg.git 03Clément BSsch 07master:d469aa8cfaa9: sws: use av_clip() instead of av_clip_c() [18:26] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:4a99134f1a71: tools/crypto_bench: fix build when AV_READ_TIME is unavailable [18:59] <cone-798> ffmpeg.git 03Carl Eugen Hoyos 07master:7c3ba1b83aeb: Add a closing LF to Webm-manifest files. [18:59] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:a07b0ffdc342: Merge remote-tracking branch 'cehoyos/master' [19:07] <abaudoux> Hello, I have the following error when demuxing a ts file : http://pastebin.com/JmUXNhWz [19:55] <compn> Abaudoux [19:55] <compn> Oh he gone [20:19] <cone-798> ffmpeg.git 03Mikulas Patocka 07master:9030c58a780a: configure: Disable i686 for i586 and lower CPUs [20:19] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:9bf260b2d2fc: Merge commit '9030c58a780a02fb8256cb75cd39275eaad786d1' [20:38] <cone-798> ffmpeg.git 03Gabriel Dume 07master:45ff7c93dd84: dca: K&R formatting cosmetics [20:38] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:7d43fbe3aef8: Merge commit '45ff7c93dd84a254cc96acc589e5ac3d7bd16bce' [21:24] <cone-798> ffmpeg.git 03Clément BSsch 07master:a87527ad6737: avfilter/ass: make sure the log level are in available range [21:24] <cone-798> ffmpeg.git 03Clément BSsch 07master:8a9c5db29e84: avfilter/ass: better log level mapping [21:24] <cone-798> ffmpeg.git 03Clément BSsch 07master:4b58349bc8ff: avfilter/ass: add shaping option [21:25] <ubitux> JEEB: saw that https://trac.ffmpeg.org/ticket/3949#comment:2 ? [21:26] <JEEB> nope, but I will check it when I'm not tired as fuck [21:30] <ubitux> ok :) [21:36] <ubitux> llogan: i don't understand the second example [21:37] <llogan> ubitux: would an image displaying what (I expect) it does help? [21:43] <llogan> ffmpeg -f lavfi -i testsrc=s=hd720:d=5 -filter_complex "geq=lum=255*(Y/H),format=gray[grad];[0:v]boxblur=4[blur];[blur][grad]alphamerge[alpha];[0:v][alpha]overlay" output.mp4 [21:44] <llogan> it just gradually gets blurry. not blurry on top, blurry on bottom. [21:44] <ubitux> say that in the description please [21:46] <llogan> words hard. [21:47] <llogan> i'll come up with something [21:47] <llogan> how about? "In this example the video will progressively become more blurry along the y-axis as defined by the linear gradient." [21:48] <llogan> or "...more blurry from top to bottom." [21:48] <llogan> if it's a shitty example i can remove it [21:51] <ubitux> alright, so this says what it will look like, now add how it's done [21:51] <ubitux> and i'll be fine with that ;) [21:51] <llogan> alright. thanks for taking a look at it. [21:54] <llogan> wm4: elementary appears to be the least ugly of the ugly claws themes [21:54] <wm4> claws as themes? [21:54] <wm4> *has [21:54] <llogan> http://www.claws-mail.org/themes.php [21:55] <wm4> ... [23:43] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:474d858fd955: Revert "avformat/rtpproto: fix strict aliasing violations with sockaddr" [00:00] --- Wed Sep 17 2014

1 0

ffmpeg.log.20140916
by burek 17 Sep '14

17 Sep '14

[00:01] <Mikflut> Thanks llogan & fflogger :) I'll come back when I have the full outputs. [00:01] <ubitux> fflogger did all the job [00:02] <Mikflut> any particular settings I should use on pastebin? syntax highlighting? (it's going to be a massive log....) [00:03] <t4nk221> EvolE: found it in C:/windows/ [00:03] <t4nk221> but still getting that error [00:04] <t4nk221> Output file #0 does not contain any stream Assertion v>0 && v<=(1 ? 32 : 16) failed at libavutil/mem.c:226 [00:04] <llogan> Mikflut: you can trim multiple repeating lines [00:05] <EvolE> t4nk221: trying to get some paramenets for both mencoder and ffmpeg, kinda tricky :) [00:06] <t4nk221> well i was thinking if mencoder outputs raw video, i guess we can achive that thru ffmpeg also [00:19] <edoceo> rickbol: http://avconv.com/#convert [01:31] <EvolE> t4nk221: getting uncompressed video but after compressing it's sliding and blinking... probably some pixel format is wrong when converting in ffmpeg. don't know how to fix it [01:32] <idetuxs> Hi everyone [01:35] <idetuxs> nevermind, got it. [01:36] <idetuxs> bye bye [01:36] <idetuxs> good luck [01:40] <t4nk221> damn [01:40] <t4nk221> no luck [01:40] <t4nk221> EvolE: its ok, i appreciate for your help [01:41] <EvolE> np, just got very interested in doing this xD [01:41] <t4nk221> i m not sure if i record a video with ffmpeg i would get the same lag.. [01:44] <t4nk221> EvolE: can you suggest me some other software for recording? [01:52] <EvolE_> t4nk221: can't action record directly to h264 ? [01:52] <EvolE_> t4nk221: did you try OBS ? [01:57] <t4nk221> EvolE_: my friend found Mirillis, so i guess if there was an option for h264 he would use it, but i will check again [01:57] <t4nk221> EvolE_: also OBS looks promising [01:57] <EvolE_> t4nk221: yeah, I live OBS because it's free. many streamers use it [01:58] <t4nk221> EvolE_: i guess i wont run on the same problems with OBS :) [01:58] <t4nk221> EvolE_: thanks for the suggestion [01:59] <EvolE_> t4nk221: yeah, OBS can write directly to h264. im sure Action should do it also... want to check it now) [02:00] <EvolE_> t4nk221: if you have some video you want to convert though, you can try to convert with mencoder. I just couldn't make it to write directly to ffmpeg, but if you write first to avi, then you can read avi from ffmpeg [02:01] <EvolE_> t4nk221: only if game video is big enough it can produce large temporary avi file (because mencoder will write uncompressed file) [02:03] <t4nk221> well i tried to produce an avi from mencoder under lines but getting some errors [02:04] <t4nk221> linux* [02:04] <EvolE_> you must to do it in windows [02:04] <t4nk221> same under windows [02:05] <t4nk221> that mem.c error occurs all the time [02:05] <t4nk221> even with the 32 bit dll file [02:05] <t4nk221> i will try on other machines [02:06] <t4nk221> nvm i think its enough ffmpeg and mencoder for today [02:06] <EvolE_> t4nk221: can you try with this command "mencoder.exe spidermanlovefixmytiles.avi -o test -of avi -ovc lavc -lavcopts vcodec=ffvhuff" ? [02:06] <EvolE_> t4nk221: ok))) [02:07] <t4nk221> will try tomorrow [02:07] <t4nk221> also is there any chance to read the history of this channel somewhere? [02:08] <c_14> t4nk221: http://ffmpeg.gusari.org/irclogs/ [02:08] <t4nk221> thanks c_14 [02:10] <t4nk221> thanks again for the help, cya [02:10] <EvolE_> np. cya) [03:35] <khtresa> hellow can some one help me with ffmpeg multi-Threaded version [03:43] <rickbol> when concatenating mp4 files (created with libx264) w/o re-encoding (i.e., -c copy) does "-crf xx" have to match? How about "-preset xxxxx"? [03:45] <bryno> rickbol: i don't think so [03:46] <rickbol> is there a way of determining what the -preset and -crf values were by interrogating a file with ffprobe? [07:21] <LanDi> hey guys [07:23] <LanDi> hey guys, I want to start 2 ffmpeg commands at the same time... how can I do that ? [07:25] <Nosomy> Example: ffmpeg -i video1.mp4 [etc] output | ffmpeg -i video2.mp4 [etc] output2 [07:26] <Nosomy> for runs 2 ffmpeg instances. [07:37] <LanDi> ok [07:39] <LanDi> thanks [07:40] <LanDi> Nosomy: one question.... does your name refers to the Nosomy from Shenmue ? [07:40] <LanDi> lol [07:43] <Nosomy> Dunno, what's Shenmue? [07:46] <LanDi> Nosomy: https://www.youtube.com/watch?v=W8RpDgxAyKI [07:49] <Nosomy> hmm, never hear about this [07:51] <LanDi> lol [07:51] <LanDi> thanks dude [08:10] <ruby_on_tails> http://pasteboard.co/eTlBjfi.png the video on the left doenst work with jwplayer/quicktime/windows media player, why? the one on the right works [08:11] <Nosomy> make test with ffplay [08:11] <ruby_on_tails> Nosomy: means? [08:12] <Nosomy> well, wmp runs only with codecs. [08:13] <ruby_on_tails> i can ignore wmp for now [08:14] <ruby_on_tails> even quicktime [08:14] <ruby_on_tails> jwplayer is web based [08:14] <ruby_on_tails> the video on the right works, the one on the left doesnt [08:14] <Nosomy> another thing [08:14] <ruby_on_tails> can tell what the AVC means in the brackets in the video on the right? [08:14] <Nosomy> forcc can be the problem. [08:15] <ruby_on_tails> Nosomy: whats forcc? [08:15] <Nosomy> *fourcc [08:15] <ruby_on_tails> Nosomy: whats that? [08:15] <Nosomy> the thing in the brackets, probably refers to fourcc. [08:16] <ruby_on_tails> oh [08:16] <ruby_on_tails> but i had simply copied the audio and video codecs while converting the video on the right to the one on the left [08:16] <ruby_on_tails> weird [08:16] <Nosomy> http://en.wikipedia.org/wiki/FourCC [08:26] <ruby_on_tails> Nosomy: i set the fourcc manually as avc1, now the info is same for both videos http://pasteboard.co/eWsxf5O.png but still right one works with jwplayer, left one doesnt [08:28] <Nosomy> should be return the original fourcc and analyze the video with ffprobe.. [08:29] <Nosomy> sone thing is wrong.... [08:29] <Nosomy> *something [08:29] <ruby_on_tails> hmm [12:04] <Rathann> hi [12:04] <Rathann> can anyone recommend a good VM (KVM-based preferably) hosting provider in EU? [12:30] <__jack__> Rathann: which contry ? doesn't matter ? online is pretty good [12:30] <__jack__> I dunno if they are kvm-based [12:38] <spaam> Rathann: linode in lindon is great. using xen. i think digitalocean is using kvm and they have things in amsterdam. [12:38] <spaam> london.. [12:49] <AlexC_> Morning [12:56] <AlexC_> I'm trying to convert an Apple prores video to h264, which I can do but I seem to loose my audio. This is what I'm doing: avconv -i in.mov -acodec copy -an -vcodec libx264 -crf 23 out.mov [12:57] <relaxed> remove -an [12:58] <AlexC_> Ah, I see :) Blindly copied that [12:58] <__jack__> use ffmpeg! [13:01] <AlexC_> I'd like to, though not in Ubuntus repos and I cba with compiling it right now. Just needed to get it done [13:03] <AlexC_> Thanks relaxed [13:18] <joules> lol avconv [13:19] <joules> can ubuntu's use dmm? [13:20] <joules> I see satanic.church isn't even available pre registration (along with other OBVIOUS domain names, why bother!). [13:27] <ubitux> Announce for all the people who missed it [13:27] <ubitux> https://packages.debian.org/experimental/ffmpeg [13:27] <ubitux> EOA [13:30] <iive> ubitux: ? [13:31] <ubitux> ffmpeg is back in Debian [13:31] <__jack__> you're dreaming alive [13:32] <iive> I don't see it in the queue, but the link you gave also says no such thing. [13:32] <__jack__> there is a lot of talk about that return, but you it is not done yet [13:32] <__jack__> ffmpeg package is an ugly lie [13:33] <ubitux> iive: "ffmpeg 2.4" what do you need more? [13:33] <__jack__> aptitude show ffmpeg [13:33] <__jack__> .. [13:33] <__jack__> Libav is a complete, cross-platform (blablabla) [13:33] <ubitux> __jack__: experimental [13:33] <__jack__> yep [13:33] <gerion> iive: now the link works and points to the right package [13:33] <ubitux> __jack__: maybe your mirror isn't sync [13:33] <ubitux> or something along those lines [13:34] <__jack__> owh, my bad! [13:34] <__jack__> you're right [13:34] <__jack__> let's buy some beers [13:34] <iive> probably not on the mirror i get : "Error Package not available in this suite." [13:34] <ubitux> you might want to wait 24h [13:34] <ubitux> it was added something like 6-7 hours ago [13:35] <iive> i'll put the champagne in ice [13:35] <ixian> Hi, I'm trying to extract the chapter markers with ffprobe from a bluray; using "ffprobe bluray:/mnt/bluray -show_chapters", however, the output does not contain any chapter information [13:36] <iive> \o/ [13:36] <ixian> I'm sure the bluray has chapter markers because I have extracted them successfully with eac3to [13:36] <ixian> but I would like to use ffmpeg for this [13:37] <aditsu> hi, I'm quite perplexed about theora and the ogv format.. I have an ogv file, and if I convert it to avi, using -vcodec copy and -acodec copy, it takes a long time and blows up more than 10 times in size, what's up with that?! [13:37] <ubitux> ixian: our "protocol" might not support exporting the chapters [13:37] <iive> the mirror updated for me too!! [13:37] <ubitux> ixian: you might want to open a feature request on our trac [13:38] <ubitux> ixian: bluray support is added through the protocol layer of ffmpeg, and i would guess it needs a more high level interface, such as a "format" so the chapters can be added, so yeah open a bug report please [13:39] <ixian> ubitux: ok, thank you [13:41] <aditsu> ubitux: http://dpaste.com/2BRY338 [13:42] <ubitux> 1.2.6 is really old, please git/master or at least 2.4 [13:42] <ubitux> +test/try [13:42] <aditsu> hmm, the latest I can install easily is 2.2.7 [13:43] <ubitux> better than nothing [13:43] <ubitux> anyway, is it the audio or the video who make the file so big? [13:43] <aditsu> I guess the video [13:43] <ubitux> (can we really remux theora/vorbis in avi?) [13:44] <ubitux> aditsu: confirm it by mapping only the video (-map 0:v) [13:45] <aditsu> ubitux: oops, that was instant and the result was tiny [13:45] <ubitux> so it's the audio? [13:45] <aditsu> it seems so.. [13:46] <aditsu> I don't really need the audio for that clip so I guess the problem is solved :p but I was wondering what's going on [13:46] <ubitux> all i see is this: [13:46] <ubitux> /* HACK/FIXME: Does Vorbis in WAV/AVI have an (in)official ID? */ [13:46] <ubitux> { AV_CODEC_ID_VORBIS, ('V' << 8) + 'o' }, [13:46] <ubitux> :p [13:47] <ubitux> aditsu: can you share a sample? [13:48] <aditsu> ubitux: it's created with recordmydesktop [13:48] <ubitux> ok [13:48] <ubitux> (you know you can record with ffmpeg right?) [13:48] <aditsu> does it have a gui? :) [13:48] <ubitux> yes [13:48] <ubitux> xterm [13:49] <aditsu> hahaha [13:49] <ubitux> why do you want to remux in avi btw? [13:50] <aditsu> no special reason.. just thought it would be easier for editing [13:56] <aditsu> so avi doesn't like vorbis? I'm getting similar results with ffmpeg 2.2.7, it's just faster [13:59] <ubitux> that's indeed very weird [14:00] <ubitux> i'll open a bug [14:04] <aditsu> seems to be fine when I use libmp3lame [14:04] <aditsu> anyway, thanks for the help [14:06] <ubitux> aditsu: http://trac.ffmpeg.org/ticket/3954 [14:06] <ubitux> feel free to upvote [14:07] <aditsu> you have a communist symbol in your prompt? o_O [14:09] <ubitux> depends [14:09] <ubitux> when the command fails it becomes [14:09] <aditsu> haha [14:09] <aditsu> I guess I need to register to vote [14:10] <ubitux> probably yeah [14:12] <aditsu> done, trac is simple enough [14:13] <aditsu> have a good day [18:09] <willwh> hi guys, does ffmpeg support fcsubscribe? [18:16] <tlhiv_work_> i have an AVI file that i was capturing that had to be "killed" and therefore it was "truncated" ... when I try to do anything with it with FFMPEG, I get "Invalid data found when processing input" ... is there anything that I can do to "repair" this AVI? [18:40] <sine0> ffmpeg -i youtubevideo.mp4 -vn -ac libmp3lame -ab 192 [18:40] <sine0> can i do that for audio bitrate ? [18:40] <willwh> to answer my own question; ffmpeg -i "rtmp://server/app/stream live=1" -o whatever.flv seems to work just fine [18:48] <Nosomy|off> lol [18:48] <Nosomy> ffmpeg to recovert youtube audio [18:51] <Mavrik> willwh, you might wanna add "-codec copy" to that :) [18:52] <Nosomy> best thing to do in case is make a copy of the audio: ffmpeg -i youtubevideo.mp4 -vn -acodec copy [18:53] <Nosomy> audioyt.aac [18:54] <abaudoux> Hello [18:55] <abaudoux> Is this the right place to ask for help about en ffmpeg error? [18:55] <abaudoux> *an [18:55] <Mavrik> most likely :) [18:56] <ChocolateArmpits> tlhiv_work_, did you try specifying the format of the input video with "-f avi" before the input ? [18:58] <tlhiv_work_> ChocolateArmpits: "Operation not permitted" [18:59] <tlhiv_work_> http://pastebin.tlhiv.org/5l8JYoCL [18:59] <c_14> You don't have read permissions an that file. [19:00] <c_14> eh, no wait [19:00] <c_14> You _really_ don't want to specify the same output file as your input file. [19:00] <Nosomy> thefirst "-f avi" isn't necessary [19:00] <c_14> That will _not_ grok. [19:01] <Nosomy> like this, should be work: ffmpeg -i foo.avi -vcodec copy -acodec copy foo.avi [19:01] <ChocolateArmpits> tlhiv_work_, try ffprobe on the file [19:01] <abaudoux> I have an error while trying to use the hls format on a .ts input file : http://pastebin.com/JmUXNhWz [19:02] <abaudoux> [mpegts @ 0x7fec2403f200] AAC bitstream not in ADTS format and extradata missing [19:02] <abaudoux> av_interleaved_write_frame(): Invalid data found when processing input [19:02] <tlhiv_work_> http://pastebin.tlhiv.org/2CKwtsCg [19:04] <ChocolateArmpits> abaudoux, try "-bsf:a aac_adtstoasc" [19:05] <c_14> tlhiv_work_: Any reason you're using an ancient version of FFmpeg? [19:05] <abaudoux> still the same erro [19:05] <abaudoux> error [19:05] <tlhiv_work_> c_14: same things on a new version [19:05] <tlhiv_work_> i have two versions installed [19:06] <c_14> ffprobe -analyzeduration 1G -probesize 1G -f avi foo.avi [19:07] <c_14> If that doesn't work, it's probably too broken to be fixed. [19:38] <ChocolateArmpits> Is there any filter to blur video vertically? [19:56] <wintershade> hey guys [19:56] <wintershade> after a few days of experimenting, I have some more questions. [19:56] <K4T> ye? [19:56] <wintershade> first of all, which do you guys think is the best sws_scaler for downscaling video? [19:57] <wintershade> i.e. I want to downscale from 1080p HD video to an old SD-video, like 720x480 or 720x576. which scaler is best to use for that? [20:08] <Nosomy> any scaler serves to downscaling, but i prefer Lanczos for this [20:08] <ChocolateArmpits> Lanczos slightly sharpens the image upon downscaling [20:09] <ChocolateArmpits> Spline doesn't [20:09] <wintershade> okay, thanks. I thought as much. which leads me to another question... [20:10] <wintershade> I'm experimenting with libxvid and mpeg4 video codecs. currently a bit more inclined to use libxvid for this purpose. however, my video gets very "blocky" at times, especially in dark scenes, and especially with anime/cartoons. any ideas on that? [20:13] <jehar_work> Heyo - is there a windows build available that has globbing support? [20:14] <jehar_work> It seems the Zeranoe build doesn't have that in it's libavformat [20:14] <Nosomy> wintershade, uses libx264 10bits, lol [20:15] <jehar_work> sweet [20:15] <Nosomy> ? [20:15] <jehar_work> Oh, I thought you were referring to an alternative build [20:15] <wintershade> Nosomy: I would love to. however, the aim is to share some of my HD films with my parents, who still have an old home theater which plays only DVDs and DivX/XviD stuff. [20:15] <wintershade> nothing better than mpeg4-asp will work [20:16] <ChocolateArmpits> wintershade, try to look into lumi_aq and variance_aq [20:16] <ChocolateArmpits> I suppose those should work to adapt matrix to low luminance scenes [20:17] <JEEB> or just in general up the bit rate :P [20:17] <JEEB> and/or lower the quantizer [20:17] <Nosomy> jehar_work, sure? [20:17] <wintershade> JEEB: I am currently using the 2-pass 2000kbit/s. Tried with q:v 3 and q:v 2. same thing. [20:17] <jehar_work> Yeah, attempting to run an input with -pattern-type glob results in this message: " Pattern type 'glob' was selected but globbing is not supported by this libavformat build" [20:17] <wintershade> ChocolateArmpits: thanks, will look into that. which of the two would you suggest for this? [20:17] <JEEB> then you either go lower with quants or higher with bit rate [20:18] <ChocolateArmpits> both, they are bolean, so try setting to 1 as 0 is default [20:18] <JEEB> it's not like 2000kbps or q:v 2 are the maximum/minimum [20:18] <ChocolateArmpits> you can check encoder commands with "ffmpeg -help encoder=[encoder_name]" [20:18] <ChocolateArmpits> get encoder list with "ffmpeg -encoders" [20:18] <wintershade> JEEB: q:v 2 is actually the minimum for libxvid. q:v 1 will produce more artifacts. [20:19] <JEEB> well that's libxvid being retarded then, or the libxvid wrapper [20:19] <wintershade> ChocolateArmpits: yes, I know. the documentation says I shouldn't use both at the same time. [20:19] <JEEB> libxvid might be wanting a zero of course, but I have no idea about MPEG-4 Part 2 by now [20:19] <ChocolateArmpits> oh [20:19] <JEEB> the bottom line is the same though [20:19] <JEEB> UP THE BIT RATE [20:19] <JEEB> AQ might help, but it most probably would lead to the same result [20:20] <wintershade> JEEB: hehe, okay... but like I said. I've tried with 3mbit/s, same thing. q2, same thing. q1 even, same thing. q0 doesn't exist with libxvid. [20:20] <wintershade> JEEB: AQ? you mean adaptive quantisation? [20:20] <JEEB> yes [20:20] <wintershade> JEEB: how do I turn that on? [20:21] <JEEB> the hell I know, I just remember that libxvid got it at one point, and I wouldn't be surprised if libavcodec's encoder had it, too [20:21] <JEEB> also you could try maximizing out the ME etc algorithms [20:21] <JEEB> which are fully encoder-side [20:21] <ChocolateArmpits> ok, just try "lumi_aq 1" [20:21] <wintershade> JEEB: ChocolateArmpits suggested lumi_aq and variance_aq. that's the only AQ I see in the documentation. I've maximised the ME's already. [20:21] <ChocolateArmpits> don't turn on variance_aq [20:21] <wintershade> ChocolateArmpits: doing that now. will see how it works. [20:22] <wintershade> ChocolateArmpits: why not variance_aq? [20:22] <ChocolateArmpits> "When combined with lumi_aq, the resulting quality will not be better than any of the two specified individually. In other words, the resulting quality will be the worse one of the two effects. " [20:22] <ChocolateArmpits> If we can trust documentation [20:22] <wintershade> ChocolateArmpits: I know about that. I'm asking, why lumi_aq before variance_aq? [20:23] <ChocolateArmpits> The problem is with the luminance, no ? [20:23] <wintershade> ChocolateArmpits: hmm... okay, that's a sound logic :D [20:23] <JEEB> anyways, my guess is that AQ probably isn't going to help too much (it might have some effect), in the end you will just have to push the bit rate further by using a lower quantizer or just setting a higher bit rate with 2pass [20:23] <ChocolateArmpits> It will adapt based on lightness, while the variance_aq is probably a more general, based on frame complexity [20:23] <JEEB> mpeg-4 part 2 sucks, and in the end you just have to heave-ho it with bit rate :P [20:23] <JEEB> it's not magic and all that jazz [20:24] <wintershade> JEEB: I know, I know... [20:24] <ChocolateArmpits> ALSO, denoise your video to get the best out of low bitrate video [20:24] <JEEB> lol [20:24] <wintershade> ChocolateArmpits: hqdn3d? anything else? [20:24] <JEEB> welcome to early 2000s hints [20:24] <wintershade> JEEB: lol, true that. [20:25] <ChocolateArmpits> hqdn3d is good, just be sure to have temporal denoise set higher, than spatial [20:25] <ChocolateArmpits> Also deringing [20:25] <ChocolateArmpits> if you have any sharpness artifacts [20:25] <ChocolateArmpits> Basically lowpass the video as much as you can bear [20:26] <JEEB> I'm so glad that I have no hardware that I care about that only supports MPEG-4 Part 2. Because trying to find out what the decoder actually supports is already a damn chore (because no-one marked the level/profile/etc anywhere) [20:26] <JEEB> "what divx outputs, more or less" ended up being the "spec" for many devices [20:26] <wintershade> JEEB: heh, I know. if it was me, I'd be using theora only :P just to spite the x264 :D [20:27] <wintershade> JEEB: mainly because all of my own devices play basically anything that I can throw at them. if it fits in MKV, it plays on my Android. [20:27] <JEEB> I take that as offense because x264 related people are some of the best open source developers I've seen [20:27] <wintershade> JEEB: oh geez, take a joke man. [20:27] <JEEB> and if you are going to blabber about H.264 "not being open" you should go poke the google door [20:28] <JEEB> because VPx are developed in an even more air-tight environment [20:28] <wintershade> JEEB: I've been using FLOSS for the past 10 years, almost purely. [20:28] <wintershade> JEEB: I know who these people are and they have my full respect. [20:28] <JEEB> if _I_ can join a mailing list that is about developing the next MPEG video format, and all the drafts are public I think it's pretty damn open [20:29] <JEEB> yeah, I just see too many people going "But AVC/H.264 is not an 'open format' nah nah" or so [20:29] <wintershade> JEEB: besides... x264 existed as a codec long before any of those "wise-men" from Apple or Google decided to call it their own. [20:29] <JEEB> uhh [20:29] <wintershade> ...iirc. [20:29] <wintershade> same was with AAC audio. [20:30] <wintershade> anyway... back to topic. [20:30] <JEEB> x264 is an encoder that implements the ISO/IEC 14496-10 specification, more readably noted as MPEG-4 Part 10: Advanced Video Compression (AVC) [20:30] <JEEB> which is also called H.264 [20:30] <JEEB> and Apple or Google have not "called it their own" [20:30] <wintershade> yeah, yeah. I know all that. and yes they did, at least Apple did with their DRM-friendly container formats. [20:30] <JEEB> uhh [20:30] <wintershade> nevermind. I get what you mean. [20:30] <JEEB> you mean MPEG-4 container [20:30] <JEEB> which is based on Apple's MOV [20:31] <JEEB> and which is not any more or less "DRM-friendly" than the other container next to it [20:31] <JEEB> my comment regarding google was towards those idiots who call the AVC or HEVC "closed formats", while the irony is that the video formats developed by Google are much more closed [20:31] <wintershade> JEEB: uhh. OGM and MKV are not exactly DRM-friendly, iirc. [20:31] <JEEB> they are just as DRM-friendly [20:32] <wintershade> *shrug* ah well. my mistake then. I've never seen an ogm or mkv file that was cluttered with any drm. [20:32] <JEEB> I'm pretty sure divx did matroska with drm [20:32] <JEEB> but that's not what I wanted to say [20:32] <wintershade> in any case, I don't really give a damn about politics (as you might have noticed). if it works, I use it. [20:32] <JEEB> I wanted to say that all of those formats are as "DRM-friendly" as the other [20:33] <wintershade> ah well. [20:33] <wintershade> moving back to... 2000s tips, as you called it. [20:34] <wintershade> ChocolateArmpits: what would you suggest as the hqdn3d options? last time I tried it, I used 10:10:15:15 (so, 10 for spatial and 15 for temporal). is this enough? or too much? how high can it go, anyway? [20:35] <wintershade> oh, 255. I see. [20:50] <wintershade> oh gosh, I feel so stupid now. [20:50] <wintershade> JEEB: you were right... [20:51] <wintershade> JEEB: pulling the bitrate up DID solve the problem... as well as changing the video player I used for testing. [20:51] <wintershade> I tested everything with VLC and somehow it produced giant ugly blocky artifacts... with mplayer (vdpau video driver) everything is totally smooth. for some reason. [20:52] <ChocolateArmpits> VLC doesn't render the video gamma right [20:52] <wintershade> *facepalm* [20:52] <wintershade> today I learned a lot... [20:52] <ChocolateArmpits> Making blacks more gray [20:52] <wintershade> do they know about this? seems like a serious bug to me... [20:52] <ChocolateArmpits> VLC wasn't conceived as a media player [20:53] <wintershade> huh? what is it then? [20:53] <ChocolateArmpits> restreaming video or something [20:53] <wintershade> uhh... okay. I feel completely stupid. [20:54] <wintershade> okay, so what are my choices? MPlayer? anything else? [20:54] <ChocolateArmpits> MPC-HC [20:54] <JEEB> <ChocolateArmpits> VLC doesn't render the video gamma right <- this depends on the video renderer [20:54] <JEEB> (note: there's a whole lot of them in VLC) [20:54] <ChocolateArmpits> interesting [20:54] <wintershade> ChocolateArmpits: Oh. Umm... I'm on Linux. 10 years of FLOSS :P [20:54] <Plorkyeran> vlc originally stood for "videolan client" since it was the client for a client-server video streaming thing [20:55] <JEEB> wintershade, mpv [20:55] <Plorkyeran> but then the server got merged into the client [20:55] <JEEB> #mpv-player @ freenode [20:55] <Plorkyeran> and the name got silly [20:55] <JEEB> just needs opengl 2.2 and some extensions [20:55] <wintershade> JEEB: that's a fork of mplayer :D but thanks again for info :) anything else? [20:55] <JEEB> wintershade, it's a fork of a fork yes [20:55] <wintershade> JEEB: I believe I have opengl 2.2... nvidia blob driver here :-/ [20:55] <wintershade> JEEB: ...we need to go deeper. [20:56] <JEEB> it also has the best opengl-based video renderer that does Things Right [20:56] <JEEB> so it definitely is a good one to try [20:56] <wintershade> does things right? sounds scary... :D [20:56] <wintershade> let's see if I have it in my portage tree... [20:57] <wintershade> well what do you know, it's here. at least I know what's the next thing I'll be compiling today :D [21:01] <wintershade> anyway, I'm off now. [21:01] <wintershade> thanks everyone for your help! [21:01] <wintershade> and thanks JEEB for getting my facts right ;) [21:01] <wintershade> ...and everything else. [21:02] <wintershade> sometimes I'm a moron, but hey, at least I admit it :D [21:30] <voxadam> Is it possible to losslessy remux an MKV (h264/AAC), including all the video and audio streams including subs, to an MP4 with the same codedcs? I'm new to ffmpeg and have found nothing but conflicting information online (shocking!). [21:31] <Suchiman> h264 and aac should be possible [21:32] <Suchiman> ffmpeg -i Input.mkv -c copy output.mp4 [21:32] <voxadam> Wow... easy. [21:32] <voxadam> I love unix commands. [21:32] <Suchiman> the same does work on windows :o [21:34] <llogan> voxadam: mp4 doesn't support much for subs, AFAIK. [21:34] <Suchiman> voxadam: oh wait, i missed one part, "including Subs" if the Subs are in a subtitle file and not rendered upon the Video it won't work [21:34] <llogan> "mov_text" [21:34] <voxadam> Good to know. [21:35] <llogan> and "-map 0" will include *all* input streams incase there are multiples of each type (see http://ffmpeg.org/ffmpeg.html#Stream-selection) [21:35] <voxadam> Maybe I should just find a way to stream MVK directly. RTMP, DASH, or if I have to, HLS. [21:35] <Suchiman> either omit the subtitles, render them on the video (lossy re-encoding) or keep it as mkv, that was probably the reason it has been a mkv in first place [21:36] <Suchiman> ffmpeg can also do Streaming iirc [21:36] <voxadam> Maybe I could break the subs out into an SRT and pry it suncs. [21:36] <voxadam> s/suncs/syncs. [21:37] <voxadam> I'Il look into ffmpeg's streaming. [21:37] <llogan> depending on the subtitle type, you may be able to just add -c:s mov_text to convert it. [21:37] <voxadam> Thanks. I'll check into that. [21:37] <llogan> i don't know for sure though</subignorant> [21:38] <voxadam> :) [21:43] <RyanV> is there a way to prevent the logging of the configuration flags whenever running ffmpeg/ffprobe? [21:43] <t4nk221> EvolE: tried OBS but the screen positioning is a pain in the ass [21:44] <EvolE> t4nk221: what do you mean?) [21:44] <t4nk221> its very hard to position the screen recording [21:45] <t4nk221> because everytime i record a game i need to adjust the screen position [21:45] <t4nk221> different game different position [21:45] <t4nk221> its not easy as in Action! [21:46] <EvolE> in OBS you can create so called scene for each game and then just switch between them in 1 mouse click [21:46] <EvolE> also you can automatically grab game's/browser's window for each scene [21:47] <EvolE> basically many windows for each scene [21:47] <EvolE> you can presetup this and then you can just turn it on by selecting the scene [21:48] <RyanV> nvm, got it using flag -hide_banner [21:48] <EvolE> so basically you need to do the setup once per game [21:49] <t4nk221> the main problem is the there is a settings window behind screen rectangle and its very hard to set it [21:50] <t4nk221> anyway OBS is a great software [21:50] <t4nk221> a GUI version of ffmpeg [21:51] <EvolE> t4nk221: i don't understand what do you mean "settings window behind screen rectangle" ) [21:52] <t4nk221> nvm, i dont know how to explain it.. [21:53] <jgh-> hello, I am reading an RTMP stream via av_open_input / av_read_frame and can decode / display the video packets fine (I am using the hardware decoders for AAC and H264) as the SPS and PPS are included in the rtmp stream .... however, I am trying to find the ASC packet or at least information on the sample rate and channel count of the AAC audio. Where might I be able to find that (i.e. I'm not seeing anything obvious in [21:53] <jgh-> AVPacket or AVFormatContext but maybe I'm missing something) [21:53] <EvolE> t4nk221: you can select basic positioning with right-click menu if you want to reset it or something. in you need precise positioning then you can move layer with mouse i suppose [21:56] <t4nk221> yeah i need precise positioning for each game [21:59] <EvolE> t4nk221: is your game in separate window (windowed or full screen) or in browser? [21:59] <t4nk221> in browser [22:00] <t4nk221> something like http://www.ebog.com/management-games/16096/dreamfields.html [22:00] <EvolE> if it's in browser, then yes. it can be tricky.. but anyway game window will be at the same position every time in broswer, so you can setup it once and it will work all the time [22:01] <t4nk221> nahh its not in the same position, thats the problem [22:01] <jgh-> ah i think i found it in codeccontext [22:06] <EvolE> t4nk221: so how you doint that with action! ? it detects region by color or something? [22:08] <EvolE> t4nk221: you just click that game and action captures it without need to manually setup size with mouse? [22:10] <t4nk221> you just click the rectangle button in Action! and the program minimizes, only a rectangle with borders appear [22:12] <t4nk221> plus i m not recording the whole game window, for example if there is a menu in the game i dont need to record it [22:12] <t4nk221> only the actual gameplay [22:13] <t4nk221> http://prntscr.com/4n6nin [22:20] <EvolE> t4nk221: yeah, that snap (magnet icon) button should be pretty useful if it works with flash content in web pages [22:21] <EvolE> t4nk221: also, did you try to change video format for MP4 in Action! insead of avi? Maybe it will write video in h264 instead of own format? [22:22] <t4nk221> yeah, i checked that [22:22] <t4nk221> there is avi, mp4 and mp4 for intel... [22:23] <EvolE> t4nk221: isn't it h264 in mp4 ? [22:24] <t4nk221> i m not sure [22:24] <EvolE> t4nk221: so you will not have that problem with lagging cursor i suppose, if it's h264 [22:24] <EvolE> you can try xD [22:25] <t4nk221> yeah but i doubt that its h264 [22:26] <t4nk221> i will try it now [22:26] <EvolE> ok, will wait for the result) [22:40] <t4nk221> EvolE: seems like its h264 [22:40] <t4nk221> "codec_name": "h264", "codec_long_name": "H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10", [22:40] <t4nk221> https://www.dropbox.com/s/b18vsx2e2v84cry/spiredmanmp4.mp4?dl=0 [22:41] <shevy> anyone seen this error before? [VD_FFMPEG] DRI failure. [22:41] <shevy> when playing videos via mplayer, I get audio, but a black video screen [22:42] <shevy> I am not sure why exactly though - I reinstalled here a few days ago, switched from 64 bit to 32 bit again [22:42] <shevy> ffmpeg version 2.4 built on Sep 16 2014 01:13:26 with gcc 4.7.1 (GCC) [22:42] <shevy> mplayer was svn checkout from yesterday [22:43] <EvolE> t4nk221: yeah, seems like it is. and the cursor isn't lagging ) [22:44] <t4nk221> EvolE: nope [22:45] <t4nk221> EvolE: but i will need to encode it again thru ffmpeg [22:45] <t4nk221> EvolE: coz of the settings i use in ffmpeg....like scale etc.. [22:54] <t4nk221> EvolE: got a reason from my friend why he choose avi over mp4 in Action and he says that the bitrate for a direct encoding in mp4 is not enough high [22:57] <EvolE> t4nk221: maybe he's right. isn't there an option to setup scaling in action! ? if there is an option to set quality then you should set it to maximum if you want reencode later [22:59] <t4nk221> EvolE: yep, its already set to high quality [23:01] <lipizzan> When trying to maximize record quality when encoding a DV stream to h.264, would there be useful advantage in running ffmpeg w/o X and it's gui overhead? [23:01] <worak> First of all, hello guys! I've been trying to convert a ts file written from my DVB, but Im getting some problems. I'm a newbie with ffmpeg, I already search the documentation, but I could not find the answers to my problem. Here's the command line: $ffmpeg -i 000.ts -c copy output.mkv [23:02] <worak> and the error is [mpegts @ 0x1e55ba0] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size [23:02] <worak> Consider increasing the value for the 'analyzeduration' and 'probesize' options [23:03] <worak> I already change a lot of parameters of analyzeduration and still got the same problems [23:06] <worak> and also probesize... I'll be glad if some gentle soul can help me... Thanks! [23:13] <llogan> lipizzan: no [23:20] <rickbol> llogan: I have adopted\adapted the command line you provided me with a couple of days ago, have some questions and observations. [23:21] <rickbol> llogan: btw, I am lipizzan also ;-) [23:23] <rickbol> when I simply encode a DV stream to h.264, with libfdk_aac audio, and write to file, I see a certain fps and bitrate. I'm trying to maximize quality given my machine's capabilities. [23:25] <rickbol> If I overlay a small, transparent png (a watermarking bug), the bit rate goes up by ~1000Kbps, and the fram count goes up too. [23:25] <llogan> the x related deps are for x11grab input device used for screen grabbing, and probably for xv output device. you can omit those if you like, but keeping them probably won't make a measurable difference [23:26] <llogan> and of course won't affect your quality in any way [23:26] <rickbol> However, I may not actually be getting even the 30 frames I need. The time counter may even tick off 1 second, for 1+seconds realtime. [23:26] <llogan> sorry, but i don't quite follow [23:27] <rickbol> llogan: I was referring to cpu overhead being consumed by X, whereas just running console linux, I get that cpu back for more video processing. [23:29] <rickbol> llogan: the x deps didn't compile, so I built ffmpeg without them. [23:29] <llogan> that would be pretty easy to test to see if you even see a difference [23:29] <rickbol> llogan: yes I'm gonna do that shortly. Is there any way to profile runtime performance? [23:30] <llogan> -benchmark option might be what you're looking for [23:30] <llogan> or "time" command [23:30] <rickbol> llogan: yes [23:31] <rickbol> llogan: what I was mentioning above was that the counters that run at the bottom of the screen don't seem to be reliable for me depending on what filters etc, are running in my ffmpeg. [23:33] <rickbol> llogan: like just DV->h.264->file = 1500Kbps, but DV->h.264+overlay->file = 2500Kbps [23:34] <rickbol> llogan: and for the same examples, frames/sec might be 28, and 40, respectively? [23:38] <relaxed> ubitux: ffmpeg's opengl is pretty slick [00:00] --- Wed Sep 17 2014

1 0

ffmpeg-devel.log.20140915
by burek 16 Sep '14

16 Sep '14

[00:00] <ubitux> ok [00:00] <ubitux> this all sounds overcomplicated for no good reason though [00:00] <rcombs> indeed! [00:03] <rcombs> and unfortunately, I think Pango is probably too high-level for libass [00:04] <J_Darnley> ubitux: there is still only the log2_chroma_[wh] values rather than a value per component. This means that you need to do "tricky" things for >3 planes or if you want a loop that includes the luma plane. [00:05] <ubitux> >3 planes? alpha channel? [00:05] <J_Darnley> yes. [00:06] <J_Darnley> I don't recall anything with a 5th plane yet [00:07] <ubitux> let's add a plane for subtitles [00:08] <J_Darnley> z-data, to support a 3d volume! [00:09] <rcombs> one plane for each cone type in a mantis shrimp's retina! [00:09] <rcombs> (we're going to need a bigger Bayer filter) [00:17] <J_Darnley> okay, that looks like it is working. [00:30] <thardin> I did those very low bitrate video tests btw [00:31] <thardin> x265 is looking very promising :) [00:32] <thardin> and version string does indeed mess up rate control at these low rates [00:43] <J_Darnley> ubitux: I'm going to send a WIP to the list so you, and everyone else, can see it. [00:43] <J_Darnley> It works but is missing a few things. [00:43] <J_Darnley> like some documentation. [01:27] <Timothy__Gu> J_Darnley: CMYKA has 5 planes [01:29] <Timothy__Gu> Interesting English: "Why to crash, if don't crash it's better? :)" in #3950 [01:46] <cone-583> ffmpeg.git 03Michael Niedermayer 07master:6beea6f017f5: avcodec/tiff: fix odd dimensioned yuv [03:13] <cone-583> ffmpeg.git 03James Almer 07master:91459bd320b4: avcodec: remove obsolete FF_API_DSPUTIL cruft [05:32] <molikto_> hi, i know this is offtopic... but can anyone help me configure a minimal build for some specific input/output/command??? I am helpless and the only thing in my mind is to try divide and conquer.... [05:35] <molikto_> anyone here? [06:20] <molikto_> anyone can help me?? [06:25] <jamrial> molikto_: this channel is for development of ffmpeg. for user support you should ask in #ffmpeg [06:26] <molikto_> the question is not exactly a user-level question... although it is also not dev-related... nevermind... i will figure out myself [07:21] <ubitux> ff [07:21] <ubitux> oups [08:30] <sorinv> Hey guys .. quick question: do you know where this kind of error comes from? I am struggling to build Blender with ffmepg support in it getting this link error [08:30] <sorinv> /opt/lib/ffmpeg/lib/libavcodec.a(libtheoraenc.o): In function `submit_stats': [08:30] <sorinv> /root/src/blender-deps/ffmpeg-2.1.5/libavcodec/libtheoraenc.c:142: undefined reference to `th_encode_ctl' [11:15] <wm4> ubitux: still working on that subtitle thing? [11:16] <ubitux> yes, but today i'm at work :p [11:16] <wm4> in mpv I have some kind of subtitle "filter chain" [11:16] <wm4> but it's not very good and has some design issues [11:16] <ubitux> wm4: WIP @ https://github.com/ubitux/FFmpeg/compare/subtitles-ng [11:17] <ubitux> WIP commit is basically a draft i started yesterday as an attempt to create a text sub structure for the decoded form, with ass markup, as muxed in mkv [11:18] <wm4> hm [11:19] <ubitux> tell me more about your subtitle filter chain [11:19] <wm4> it passes the mpv equivalent of AVPacket from filter to filter [11:19] <wm4> and filters at the end of the chain have a function for retrieving bitmaps [11:20] <wm4> subtitle converts (like lavc "decoders") are filters in the middle of the chain [11:20] <wm4> renderers are at the end [11:20] <wm4> one renderer is libass, and the other is lavc (for vobsub and pgs) [11:21] <wm4> but things like charset conversion or timing adjustments (avoiding small gaps/overlaps) are unfortunately not part of the chain [11:22] <wm4> but it would be nice if a new ffmpeg subtitle api could handle these things [11:27] <ubitux> hum [11:28] <ubitux> the charset convert model is so broken in ffmpeg you can't use it? [11:29] <ubitux> what's this timing adjustment? for post seeking? [11:29] <wm4> ffmpeg just wraps iconv a little, it's utterly unusable [11:30] <wm4> the timing adjustment is just to avoid little gaps of no subs, or subs overlapping for a moment [11:30] <wm4> because of timing inaccuracies [11:30] <wm4> it's not used for ASS [11:36] <ubitux> i'm not sure what to propose for charsets.. [11:36] <ubitux> grr we really need that timeline thing for mov/mp4 [11:37] <wm4> isn't that similar to mkv ordered chapters? [11:39] <ubitux> dunno [11:48] <ubitux> wm4: btw, hb can be built without glib in the upstream [11:49] <wm4> I know [11:49] <wm4> but distros use glib anyway [13:47] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:8c1dc1f6ed11: avformat/network: add union for avoiding strict aliassing violations with sockaddr* [13:47] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:e587a428d75d: avformat/rtpproto: fix strict aliasing violations with sockaddr [14:07] <ubitux> anyone has a link to the previously rejected mov edit list patch from google? [14:16] <ubitux> michaelni: n2.5-dev LGTM [14:17] <iive> 2.4 have been release? [14:17] <iive> d [14:18] <ubitux> yes [14:21] <iive> nice :) [14:30] <wm4> seriously how did this ever happen: vf_eq and vf_eq2 [14:30] <wm4> they use the same asm [14:30] <wm4> at least almost [14:31] <wm4> or actually it's exactly the same [14:31] <J_Darnley> You are probably right with your "politics" comment. [14:31] <ubitux> :D [14:34] <iive> wm4: there are eq and eq2 native ffmpeg filters? [14:34] <wm4> no, libmpcodecs [14:35] <iive> oh, ok. [14:43] <iive> wm4: what is the trouble with vdpau and preemption, and what kind of preemption is at play there? [14:45] <wm4> iive: vdpau preemption can lose the vdpau state at any time (typically VT switching) [14:47] <iive> so you have to restart decoding from scratch? [14:49] <wm4> basically; you have to reinitialize everything again [14:51] <iive> major headache... [14:51] <iive> does this apply to the mesa3d gallium vdpau driver too? [14:53] <wm4> no [14:53] <wm4> apparently not [14:53] <wm4> just nvidia [15:28] <nevcairiel> you know its fasctinating how just days after a change lands in ffmpeg, the same change lands on the libav ML, but they didnt bother to keep authorship. Granted, its just a trivial one line change, but still, and don't even try to claim coincendence [15:28] <nevcairiel> -n [15:29] <wm4> don't let them hear that [15:49] <ubitux> nevcairiel: what are you thinking about? [17:01] <J_Darnley> I'm now wondering why contrast seems to have a different meaning and default in eq2. The default has moved from 0 to 1.0 and it seems to lack an add when it is used (in the mmx function). [17:01] <J_Darnley> (I'll read any replies later, I need to make dinner) [17:10] <iive> J_Darnley: hum? brightness and contrast form liner function where output=brightness+contrast*input. [18:10] <cone-192> ffmpeg.git 03James Almer 07master:4ae6bcc025e7: RELEASE: update to 2.4.git [20:15] <J_Darnley> Dammit! Why are you giving me a grey image? [20:16] <nevcairiel> grey ios the new black? [20:16] <J_Darnley> :) [20:24] <J_Darnley> dammit gdb! why can't you give ffmpeg a filename with spaces? [20:24] <ubitux> gdb --args ? [20:25] <wm4> gdb --args ./ffmpeg "fi le.mkv" [20:25] <J_Darnley> Doesn't work. gdb still seems to split at the space despite there being a backslash [20:26] <ubitux> shell issue? [20:26] <wm4> are you on windows? [20:26] <J_Darnley> Cygwin and bash [20:26] <wm4> windows treats the command line as a single string, instead of a list of arguments [20:26] <wm4> so you'll get random fuckups there [20:28] <J_Darnley> gdb is probably doing something special here [20:28] <J_Darnley> rather than what every other program does [20:38] <wm4> well, gdb has to pass the arguments to the child process [21:01] <cone-192> ffmpeg.git 03James Almer 07master:af7d2606261e: avutil: remove obsolete FF_API_LLS1 cruft [21:01] <cone-192> ffmpeg.git 03James Almer 07master:95a064f53034: avutil: remove obsolete FF_API_OLD_OPENCL cruft [21:01] <cone-192> ffmpeg.git 03James Almer 07master:ad1dadac86d4: avfilter: remove obsolete FF_API_BUFFERSRC_BUFFER cruft [21:02] <J_Darnley> Is the AVOption system supposed to initialize unspecified options as the default values? [21:04] <nevcairiel> yes [21:04] <nevcairiel> if something called the set-defaults function on your context [21:05] <J_Darnley> maybe I'm missing that. [21:05] <nevcairiel> usually you wouldnt need to do that yourself, unless you invented some kind of new context [21:13] <J_Darnley> oh nevermind, i'm just an utter idiot [21:30] <J_Darnley> There we go. eq and eq2 are now in one file. [21:33] <wm4> yay [21:42] <ubitux> in vf_hue.c ? [22:01] <J_Darnley> no. [22:31] <cone-192> ffmpeg.git 03Mika Raento 07master:f685f7d7a838: hlsenc: single_file, support HLS ver 4 byteranges [22:40] <cone-192> ffmpeg.git 03Mika Raento 07master:00431bf874e1: ismindex: handle time discontinuities and nonzero start time [22:40] <cone-192> ffmpeg.git 03Michael Niedermayer 07master:21561610560f: Merge commit '00431bf874e1044b01e09a2266ef85d4ff8d44cc' [23:26] <iive> "[h264 @ 0xb5204440] non-existing SPS 32 referenced in buffering period" what does this message mean? [23:26] <nevcairiel> it tries to use a SPS which it doesn't have yet [23:26] <iive> it's spamming my console when i play dvb-t stream. [23:26] <nevcairiel> if the message occurs during opening of the file, it can usually be safely ignored [23:27] <nevcairiel> how does playback look? [23:27] <iive> perfect. [23:27] <nevcairiel> and it keeps spamming the message during playback? [23:27] <iive> yes [23:27] <nevcairiel> thats weird [23:28] <nevcairiel> but as long as playback works! [23:29] <iive> it's horrible [23:29] <wm4> oh, libav having a flame about their review policy [23:29] Action: wm4 gets popcorn [23:30] <nevcairiel> and not even one of us that started it [23:31] <iive> funny thing, when ffplay reaches the end of the playback, the messages start to get grouped 200-300 together. [23:32] <iive> wm4: irc or maillist? [23:32] <wm4> irc [23:32] <nevcairiel> irc [23:33] <ubitux> i thought i was the only one to notice an hour ago [23:34] <ubitux> seems i'm not the only evil stalker [23:34] <nevcairiel> i mostly ignored it [23:34] <nevcairiel> its nothing but all the same [23:36] <ubitux> wm4: u evil [23:40] <ubitux> haha he's trolling hard [23:40] <wm4> courmisch threatening to fork [23:40] <ubitux> for the third time [23:40] <iive> wm4: remi? the one with the vdpau patches i asked about? [23:40] <ubitux> yes [23:40] <iive> what a coincidence... [23:41] <baptiste> courmisch in libav ? [23:42] <j-b> yeah [23:42] <nevcairiel> its not like he would run a full fork, more like my fork, application specific patches on top of upstream [23:42] <nevcairiel> many downstreams have such things [23:43] <nevcairiel> just not those trying to be d istributed on linux [23:43] <ubitux> it's probably a way to get it merged in ffmpeg without being explicit sending it to ffmpeg-devel [23:44] <j-b> And not a bad tactics [23:45] <ubitux> i'm not exactly sure what would be the point of doing so, except :ego: [23:45] <ubitux> but sure, if it works, whatever :) [23:45] <baptiste> j-b, why not sending to both ? [23:45] <wm4> or just fork and relicense as GPL3 [23:46] <j-b> baptiste: no clue. [23:46] <j-b> baptiste: probably because it's eaiser to send to one? [23:48] <nevcairiel> well if you want to rely on some new API, you kinda need to send it to libav, or it wont be available on a bunch of systems, and since ffmpeg merges anyway.. [23:48] <baptiste> yeah I guess adding another cc is more complicated ;) [23:49] <wm4> btw. did the debian ffmpeg package make it yet? [23:49] <nevcairiel> the dude said in a mail earlier today it might get in soon [23:49] <wm4> soon? ok, I'll ask again in 6 months [23:49] <nevcairiel> baptiste: if the api needs changing d ue to feedback, its better not to have it merged in one project already, but wait until its actually finished [23:49] <ubitux> it can, because it's in the top 5 pending packages [23:50] <ubitux> (if you count the 4+ months in queue before version bump) [23:51] <iive> is the queue visible somewhere? [23:51] <ubitux> https://ftp-master.debian.org/new.html [23:51] <iive> i kind of get the feeling it is more like a blackbox... [23:51] <ubitux> 2.2.1 waited 4+ months, 2.3.1 waited 1+ month, 2.4 was submitted... an hour ago [23:51] <ubitux> actually, 3 [23:52] <baptiste> nevcairiel, in my experience, getting any feedback as early as possible is better, but it's true that I haven't sent a patch since quite some time :) [23:55] <cone-192> ffmpeg.git 03Clément BSsch 07master:864d124bb7aa: build: simplify libwebp check [23:58] <iive> baptiste: it is nice to see you around. [00:00] --- Tue Sep 16 2014

1 0

ffmpeg.log.20140915
by burek 16 Sep '14

16 Sep '14

[00:30] <TheFakeazneD525> hi, I want to batch convert some .webms I have to mp4 files, with the same video bitrate and audio bitrate [00:30] <TheFakeazneD525> how do I go about doing this [00:39] <c_14> for i in *.webm; do ffmpeg -i "$i" [options] "${i%webm}mp4}"; done [00:39] <c_14> It might be '#' instead of '%', I'm not sure. [00:44] <hachi> hello folks, I'm getting Non-monotonous DTS in output stream with -vsync cfr... and without as well. Does anyone know why -vsync cfr doesn't seem to help that situation? [00:46] <TheFakeazneD525> c_14: [NULL @ 0x2381e60] Unable to find a suitable output format for 'video.mp4}' [00:47] <TheFakeazneD525> wait [00:47] <c_14> ehhhh [00:47] <c_14> that second } shouldn't be there [00:48] <TheFakeazneD525> ah, now it's doing a thing [05:06] <molikto> hi, i know this is offtopic... but can anyone help me configure a minimal build for some specific input/output/command??? [05:08] <molikto> anyone here? [05:13] <LanDi> how to convert an .m4a format to .mp3 ? [05:16] <LanDi> ok, got it [05:16] <LanDi> :P [05:26] <molikto> anyone here? [05:28] <LanDi> nope [05:29] <Nosomy> lol [05:30] <Nosomy> molikto, going to dev's channel. [15:56] <anshul_mahe> ffprobe -show_entries frames=media_type input_file is not working [16:03] <anshul_mahe> ohh, I got it correct command would be frame=media_type not frames=mediatype [17:23] <tb3> I need help converting the audio track of an mp4 file from ac3 to mp4a. I'm very new to ffmpeg and tried reading the man page but not understanding it. [17:25] <sacarasc> If you mean AAC... https://trac.ffmpeg.org/wiki/Encode/AAC [17:25] <kaotiko> hi [17:28] <tb3> I think that's exactly what I need. Thanks. [17:29] <t4nk435> hi [17:30] <t4nk435> anyone here? [17:31] <__jack__> yup [17:32] <t4nk435> i need help about avi to mp4 video lag [17:33] <t4nk435> when i encode avi to mp4 seems like frames are dropped [17:34] <t4nk435> here is the command: ffmpeg -i input.avi -c:v libx264 -profile:v high -preset slow -crf 24 -an -movflags faststart -vf scale=640:480 -r 24 out.mp4 [17:35] <t4nk435> any suggestions? [17:36] <__jack__> t4nk435: don't use -r 24, you're forcing framerate, this might damage the output [17:36] <__jack__> (aka "lag") [17:40] <t4nk435> here is it: http://pastebin.com/3PFhbwm7 [17:41] <t4nk435> k jack, trying without -r 24 [17:42] <t4nk435> same without -r 24 .. [17:43] <t4nk435> i noticed the lag by the mouse cursor, its not sliding as in the avi file [17:46] <t4nk435> tried everything i found on net... [17:46] <EvolE> t4nk435: maybe you are using some smoothing function with that avi which is not working with converted file? [17:48] <t4nk435> the avi is recorded with Milliris Action! software [17:49] <t4nk435> in most of the encoded mp4 videos there isnt a lag [17:49] <EvolE> so maybe decoder automatically smooth the video when you play it? [17:50] <EvolE> is this happens with all videos you have recorded with Action! ? [17:51] <t4nk435> hmm i play the mp4 thru browser... chrome/mozilla etc.. [17:52] <t4nk435> no no [17:52] <t4nk435> for example 10 out of 100 videos are with lag [17:54] <EvolE> but the video which lags is always the same? [17:55] <t4nk435> what do you mean by same? [17:57] <EvolE> i mean it's always the same video which lags after converting, rihgt? can you upload the video? [17:58] <t4nk435> yeah just a sec [17:58] <EvolE> the source video, so I can try to convert it too [17:59] <t4nk435> https://www.dropbox.com/sh/68tn5a1jud98jmw/AADj8Ek8lcSPOrh28SsN54jPa?dl=0 [17:59] <t4nk435> the 3rd video is avi the 4th is mp4 with lag [18:00] <ChocolateArmpits> Can anyone help? Remuxing a TS file (h264 video + aac audio) into MP4 produces a file with "r_frame_rate" of 50, while the original has it at 25. Both files have "codec_time_base" at 1/50. This leads to playback problems [18:06] <EvolE> t4nk435: video which you see in browser has 25 fps [18:06] <EvolE> the avi one.. [18:06] <EvolE> let me check mp4 [18:06] <t4nk435> and? [18:07] <t4nk435> did you noticed the lag in the mp4? [18:14] <t4nk435> 25 fps for the avi? [18:15] <ac_slater_> sigh. I've added `rawvideo` to all of the appropriate build options (mux, demux, encoder, decoder, protocol) and yet I cant use `av_find_input_format("rawvideo");` .. ie - I can use it, but it never finds rawvideo. [18:15] <ac_slater_> I can use it as an output however [18:16] <ac_slater_> ie - `ffmpeg -i input -c:v rawvideo ... etc` [18:16] <t4nk435> ffprobe reports 24 for the avi [18:19] <haspor> hi, anyone around who could help with adpcm_swf decoding to pcm ? [18:20] <EvolE> t4nk435: yeah, and when i download the avi video you see in broswer with ffmpeg, it becomes with 50 fps... very strange xD [18:20] <EvolE> but while downloading ffmpeg says it's 25 fps xD [18:21] <t4nk435> hmm [18:21] <t4nk435> so any suggestions? [18:21] <EvolE> need the codec for that Action! program... will try to download it now [18:23] <t4nk435> k i need to go now. you gonna be here later? [18:24] <EvolE> sure [18:24] <t4nk435> great, thanks for the help [18:34] <valder> anybody here have suggestions on how I can fix an audio sync problem I am having using ffmpeg to build a video? It seems my video is correctly rendered however there is too much audio. It seems to prepend the first x seconds( x is some number depending on the video size) to the video causing the total video length to be increased by x seconds. Howver, the video stops at the correct spot. Does this have to do [18:34] <valder> with the DTS, PTS relationship? The source video is a single source. Here's my code: http://pastebin.com/ViaACkJX Thanks for your help [19:52] <rickbol> is there a way I can determine the required versions for ffmpeg dependencies? I run an "squeeze" vintage of linux (AVLinux), but want to determine if I can compile ffmpeg. [21:19] <valder> anybody work with ffmpeg before? I'm having an audio sync issue I don't understand how to fix. Would appreciate any help you can provide. My audio is longer than my video so the output has the last four seconds stuck on the last video frame while the audio continues. What is supposed to happen is the audio has to stop at the end of the video. Here's the code I have so far:http://pastebin.com/ViaACkJX [21:24] <brontosaurusrex> how can i control cli update speed? [21:26] <c_14> valder: use -shortest [21:32] <valder> c_14: -shortest? where? how? during avcodec_open? [21:47] <t4nk335> EvolE here? [21:47] <EvolE> yes) [21:48] <EvolE> t4nk335: [21:48] <t4nk335> any luck? [21:48] <EvolE> t4nk335: well, i noticed that there is no lag in video except mouse cursor)) [21:49] <EvolE> t4nk335: but you see there is no cursor in dropbox for video spidermanlovefixmytiles.avi [21:50] <EvolE> t4nk335: cursor is visible when you play file in mirillis player (or for me it's PotPlayes which display cursor like mirillis player) [21:50] <EvolE> t4nk335: after ffmpeg cursor is lagging indeed, but no the video itself [21:51] <EvolE> t4nk335: that's what i found xD [21:52] <t4nk335> hmm i can see a cursor in spidermanlovefixmytiles.avi [21:52] <EvolE> t4nk335: when you play the file locally? [21:52] <t4nk335> localy and online [21:53] <t4nk335> ah wait wait [21:53] <valder> c_14: passing in "shortest", "1" to avcodec_open2 as key/value for the dictionary didn't seem to do anything different. Am I passing it in wrong? [21:53] <t4nk335> if i play it with windows media player there is a cursor [21:54] <t4nk335> but now i saw it that if i play it thru dropbox there isnt... [21:54] <EvolE> t4nk335: but there is no cursor if you play it from https://www.dropbox.com/sh/68tn5a1jud98jmw/AADj8Ek8lcSPOrh28SsN54jPa?dl=0#l… [21:54] <c_14> valder: hmm, it's an output option in ffmpeg. Not sure what it would be in the api [21:54] <t4nk335> yeah yeah you are correct, but thats its because i guess dropbox encodes the file somehow so you can view it thru browser [21:55] <EvolE> t4nk335: yes you are right, dropbox just reencodes the video and seems like disabling the curson showing during that process [21:55] <valder> c_14: thanks [21:55] <valder> c_14: what is supposed to do? Can't find docs on it [21:56] <EvolE> t4nk335: and when you plays it locally mirillis decoder shows cursor for you, and in this case cursor is moving more shooth than after ffmpeg [21:56] <t4nk985> didnt try to play it thru milliris player.. [21:57] <t4nk985> probably its same like WMP [21:58] <t4nk985> but yes, the cursors moves more smooth than after ffmpeg encoding [22:03] <EvolE> t4nk985: but the content inself is not lagging in converted video [22:04] <t4nk201> you think only the mouse lags? [22:04] <t4nk201> how is that possible? [22:06] <EvolE> t4nk201: probably ffmpeg can't (or don't want to) display cursor correctly, maybe it's reading cursor position more rarely than it should [22:07] <EvolE> t4nk201: you know that mirillis format keeps cursor data separately from the video data (because in other case dropbox couldn't display video without cursor) [22:08] <t4nk201> seems logical [22:08] <t4nk201> the point is that i have tried to record 2-3 videos and the mouse keeps laging.. [22:08] <t4nk201> after recording [22:09] <t4nk201> i thought that the other parts are laging also [22:09] <t4nk201> but its hard to tell [22:13] <t4nk201> k man, thanks for the help [22:18] <EvolE> np [22:19] <EvolE> sad that i didn't find how to render cursor correctly, will look for it now, but not sure if there is an option for this in ffmpeg [22:23] <t4nk201> k, i'll be here [22:23] <t4nk201> i was looking also but no info on google about that [22:24] <Suchiman> can i add chapters to a mkv file using ffmpeg if i have the timestamps? [22:36] <edoceo> I've got these two TS files, I made one by generate a blank, the other was sliced out of an FLV [22:37] <rickbol> I'm transcoding a dv clip to h264 mp4. The black and white video segments (start and end segments) look fine, but color segments (in the middle) get all scrambled and are useless. [22:37] <edoceo> individutally they look fine, but when I use 'concat:' it comes out super grainy [22:37] <edoceo> ffmpeg -i 'concat:pad0.ts|vid0.ts' -y 'work.ts' [22:38] <Suchiman> edoceo: maybe you want -c copy to prevent re-encoding? [22:38] <edoceo> I'll try it [22:41] <edoceo> Suchiman: #money - thx! [22:43] <rickbol> http://pastie.org/9556755 [22:43] <EvolE> t4nk201: found something [22:47] <t4nk201> yeah? [22:47] <t4nk201> whats that? [22:47] <rickbol> llogan: just tried "-pix_fmt yuv420p" ad this seems to fix it. [22:49] <rickbol> just compiled following the guide: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [22:49] <rickbol> I think there are some typos in the instructions [22:49] <llogan> where at? [22:49] <llogan> also, if you want to deinterlace try the yadif filter [22:50] <llogan> i haven't proofed/tested the compile guide in a while. [22:50] <rickbol> throughtout there's the statement PATH="$PATH:$HOME/bin" I had to change it to PATH="$HOME/bin:$PATH" [22:51] <llogan> shouldn't either work? [22:51] <rickbol> otherwise I pick up binaries in /usr/bin, et al before my local binaries [22:51] <llogan> ah. [22:51] <llogan> like what binaries? [22:53] <t4nk201> EvolE: is it working? [22:53] <EvolE> t4nk201: mplayer can load mirillis decoder and show correct cursor [22:53] <rickbol> also, libvpx compile failed - http://pastie.org/9556792 [22:53] <llogan> nobody uses that anyway [22:54] <llogan> but thanks for the info. i'll try to take a look at it sometime this week, but since it is a wiki you can change stuff if you think it's wrong [22:54] <rickbol> finally, ffmpeg compile couldn't find opus even though libopus compiled ok [22:57] <rickbol> llogan: I bombed on the first binary, yasm, and I made the change, so I don't know what else might be a problem. [22:59] <llogan> that was the binary in mind, but i guess an existing system yasm wouldn't be a problem unless it was too old (1.2.0 or newer for x264, 0.8 for ffmpeg IIRC) [23:01] <llogan> the "package e 1.3.0" is a typo in the guide. should be 1.2.0 [23:01] <t4nk201> EvolE: i dont see how can i "connect" the decoder with ffmpeg.. [23:03] <rickbol> llogan: I did have an old version of yasm [23:05] <EvolE> t4nk201: yeah, im looking onto in now.. player can write a file and then ffmpeg can read it, but i want to find a way to directly write to ffmpeg.. are you on windows? [23:05] <t4nk201> currently on linux, but i can switch on Windows now [23:05] <t4nk201> i mean the videos are recorded on Windows [23:06] <t4nk201> and encoded on windows [23:07] <Nosomy> ffmpeg on windows is with the Zeranoe builds [23:07] <Nosomy> the cmdlines are almost the sames [23:24] <rickbol> edoceo: Wondered if you could share your concat command. I'm trying to combine a leader and content, and the concat command is new to me. I definitely won't need any re-encoding. [23:24] <edoceo> rickbol: yes, need a few minutes [23:24] <rickbol> edoceo: thx [23:25] <valder> Hi all. I'm having an audio sync issue I don't understand how to fix. Would appreciate any help you can provide. My audio is longer than my video so the output has the last four seconds stuck on the last video frame while the audio continues. What is supposed to happen is the audio has to stop at the end of the video. Here's the code I have so far:http://pastebin.com/ViaACkJX [23:30] <EvolE> t4nk201: seems like this is working: mencoder.exe input.avi -of rawvideo -ovc lavc -o -|ffmpeg ... [23:31] <Suchiman> rickbol: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [23:31] <t4nk201> trying it now [23:31] <EvolE> t4nk201: if it can't access ficvdec_x86.dll then copy in to the mencoder's folder. you can find it in mirillis action folder [23:32] <t4nk201> let me try first on linux [23:34] <t4nk201> so with mencoder you are encoding to raw video? [23:34] <t4nk201> and them ffmpeg encodes that to mp4.. [23:38] <EvolE> t4nk201: not sure if mencoder doessn't preencode it to something... let me look into it [23:39] <EvolE> t4nk201: but yeah, the idea is like you described [23:39] <t4nk201> throwing some errors on linux, switching to windows. brb [23:51] <t4nk221> EvolE: what was the dll file? [23:51] <t4nk221> Assertion v>0 && v<=(1 ? 32 : 16) failed at libavutil/mem.c:226 [23:52] <EvolE> t4nk221: ficvdec_x86.dll [23:52] <Mikflut> Hello! I'm looking for the best place to get help with CLI FFMPEG. If this is not the best place, please let me know. I am having a problem with audio and RTMP. When I use a UDP Multicast output, everything works fine but when I use RTMP output, there is no audio output. [23:52] <llogan> rickbol: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [23:53] <t4nk221> i have only ficvdec_x64.dll but i m on 32bit system [23:53] <EvolE> t4nk221: seems like mencoder produces mpeg4 already( [23:53] <llogan> Mikflut: official help channels for users include this channel and the ffmpeg-user mailing list (and libav-user ML if using API/libraries) [23:54] <EvolE> t4nk221: you can get it downloading installer from their site and opening it like archive [23:54] <EvolE> t4nk221: but it's strange you don't have it on 32bit system.. [23:55] <EvolE> t4nk221: would be understandable if you wouldn't have ficvdec_x64.dll [23:55] <t4nk221> yep [23:56] <t4nk221> also there isnt a 64bit version of mirillis.. [23:56] <Mikflut> Thanks llogan. I'm using 64bit ffmpeg-20140910. ffmpeg -f dshow -video_size 720x486 -framerate 29.97 -i video="Decklink Video Capture":audio="Decklink Audio Capture" -vf scale=640x480 -r 29.97 -vcodec libx264 -tune zerolatency -acodec libvo_aacenc -ar 44100 -af volume=128.0 -f flv rtmp://localhost/myapp/mystream [23:58] <t4nk221> i have something like: ficvdec_x86.reg [23:58] <t4nk221> but thats a registry file [00:00] --- Tue Sep 16 2014

1 0