March 2014 - Ffmpeg-devel-irc

ffmpeg-devel.log.20140330
by burek 31 Mar '14

31 Mar '14

[00:00] <ubitux> J_Darnley: i wanted to avoid building it at all [00:00] <ubitux> at least on the embedded device [00:01] <J_Darnley> Yeah. I don't know how or if it handles a separate "machine" [00:02] <BBB> I thought it did, that is, you'd basically assign the build host to be the fate instance [00:02] <BBB> and have a fate run prefix (similar to valgrind) to "export" the run to the actual device [00:04] <ubitux> ah so you'd run from the host [00:06] <BBB> right, kinda like adb [00:06] <BBB> (assuming you've used that) [00:07] <BBB> in fact I believe I've been able to "run" fate using adb [00:07] <BBB> it's not pleasant, I mean, it's really quite terrible because dab doesn't understand stderr/out are different [00:07] <BBB> but other than that it's quite ncie [00:07] <BBB> dab=adb [00:14] <ubitux> i guess i'll consider something along those lines next time then [03:02] <cone-356> ffmpeg.git 03Vittorio Giovara 07master:53c20f17c78d: vp8: K&R formatting cosmetics [03:02] <cone-356> ffmpeg.git 03Michael Niedermayer 07master:ae3313e15486: Merge commit '53c20f17c78d1d8a0fc2505868f201e69ff59cc5' [03:23] <cone-356> ffmpeg.git 03Martin Storsjö 07master:508a84e6726a: golomb: Fix the implementation of get_se_golomb_long [03:23] <cone-356> ffmpeg.git 03Michael Niedermayer 07master:30e159366e8b: Merge remote-tracking branch 'qatar/master' [03:30] <cone-356> ffmpeg.git 03Michael Niedermayer 07master:c01ddf845dcb: avformat/replaygain: remove unused variable [06:32] <cone-462> ffmpeg.git 03Peter Ross 07master:73a2d16bfab5: avformat/wtvdec: demux mpeg2 extradata [06:32] <cone-462> ffmpeg.git 03Peter Ross 07master:31ac3f306c45: avformat/wtvenc: pad judiciously when writing mpeg2 extradata [10:17] <ubitux> lol koda trolling on ffmpeg-devel [10:22] <ubitux> should i reply to the last one? [10:30] <wm4> I suggest fighting with lethal weapons [10:32] <ubitux> well i guess his point is just generate drama so i'll ignore unless directed to me [10:32] <wm4> haha ffmpeg as libav downstream [10:32] <wm4> ok that's pretty trollish [10:36] <ubitux> i want to reply :( [10:36] <ubitux> so much nonsense :( [10:37] <wm4> I'm replying [10:38] <wm4> but I won't hesitate to shift some blame on ffmpeg too [10:55] <ubitux> :) [10:56] <ubitux> i couldn't resist :( [11:24] <j-b> wm4: nice trolling though, but a bit inaccurate, if I may say [11:24] <j-b> I find Nicolas mail way more truthful [11:25] <wm4> j-b: what was inaccurate [11:26] <wm4> and is everything trolling now [11:27] <j-b> wm4: first, whether you like it or not, FFmpeg is a downstream of libav. What you are right about this, is that it does not matter, since with merges, downstream and upstream are quite complex notions. [11:27] <ubitux> > game of words [11:28] <wm4> it's a downstream only in a very technical sense of the word I can't really take that seriously [11:28] <j-b> wm4: then, it was not only that they did not like "MiNi", it's that he refused to apply to himself the rules that the rest of the project had to follow. [11:29] <j-b> wm4: sorry, but with the complete merges of qatar/master, this is exactly what is called a downstream. Not to mention the remark of Nicolas on ABI/API. [11:31] <j-b> wm4: what I don't get is why does being a downstream matter. [11:31] <ubitux> making a point [11:31] <wm4> obviously the downstream should follow the upstream is proper behavior [11:31] <ubitux> ...on which is the most important project, the source of all the truth [11:32] <j-b> on that, noone cares [11:32] <ubitux> exactly [11:32] <ubitux> :D [11:32] <j-b> Both projects are mismanaged and a piece of pain for everyone [11:32] <ubitux> also called a pop~ [11:32] <j-b> FFmpeg was a pain for years and everyone was complaining about it [11:32] <j-b> So, the right thing was to make it worse [11:33] <j-b> by having 2 forks [11:33] <wm4> lol [11:33] <wm4> so, was forking the better or worse thing that could happen? [11:34] <j-b> probably the worse [11:34] <wm4> isn't it fun [11:34] <j-b> not really, no. [11:34] <ubitux> a real fork would have been almost fine [11:34] <ubitux> main problem is that it was forked without renaming libs [11:35] <wm4> yes [11:35] <j-b> Well, sorry to say so, but with a proper fork, you wouldn't have lived. [11:35] <ubitux> that's one of the main problem for downstreams currently [11:36] <wm4> j-b: so what do we do? [11:36] <j-b> wm4: nothing :) [11:36] <j-b> wm4: keep going like you did [11:36] <wm4> telling both projects how retarded they are is probably not very helpful [11:36] <j-b> not really [11:37] <j-b> Someday, enough downstream will be pissed [11:37] <j-b> and a third fork will emerge [11:37] <wm4> and then history will repeat? [11:37] <j-b> no, and it will get out [11:37] <j-b> because it will be less insane [11:37] <ubitux> that fork will merge the merges of ffmpeg? [11:37] <wm4> but yes, a third fork hopefully absorbing both forks would be a solution [11:38] <wm4> it would need a project leader with enough skills in, er, project leading [11:38] <nevcairiel> like thats ever going to work [11:38] <j-b> ubitux: the libavcodec ones? probably. the libavfilter ones? probably not [11:38] <wm4> and making people play well together [11:38] <ubitux> j-b: vlc is not the only ffmpeg user you know& [11:38] <j-b> ubitux: I know [11:38] <j-b> ubitux: but I discuss with a lot of FFmpeg users [11:39] <ubitux> we have a lot of lavfi users [11:39] <j-b> yes [11:39] <j-b> and maybe one day, you will split correctly your repository [11:39] <ubitux> what for? [11:39] <j-b> instead of adding more mess in the same... [11:40] <j-b> I will stop arguging with you on that, because, it's impossible to discuss with you. [11:40] <j-b> and I don't want to be banned from here. [11:40] <wm4> when will vlc split its modules into separate repos? [11:40] <ubitux> i'm not arguying, nor i planed to ban anyone [11:41] <ubitux> i'm just wondering what issue you're trying to fix by splitting our libs in different repository [11:41] <j-b> wm4: totally different. the libvlccore API is not meant to be used outside. [11:41] <ubitux> ...while a lot of our users would actually have a single library [11:42] <j-b> wm4: yet, vlc, vlc/android, vlc/ios, vlc/winrt, vlc/webplugins and other are already split [11:42] <j-b> ubitux: a single library? what library? [11:42] <ubitux> libffmpeg or something [11:42] <j-b> sure, but it does not exist [11:42] <ubitux> a few people suggested that several times [11:42] <j-b> yep, I really see people using that [11:42] <ubitux> yeah but splitting libs wouldn't be a step in that direction [11:43] <j-b> how so? [11:43] <ubitux> because they will probably be more in desync [11:43] <wm4> because libavfilter would be separate from libffmpeg [11:43] <j-b> wm4: not at all. [11:43] <wm4> but yeah, maybe libavfilter is the one thing that could actually be split off [11:44] <wm4> OTOH, ffmpeg.c and ffplayer.c depend on it [11:44] <j-b> ubitux: and that might force you to ACTUALLY eat your own dog shit [11:44] <ubitux> :) [11:44] <j-b> ubitux: and care a BIT about versionning [11:44] <ubitux> we do [11:44] <j-b> Bullshit [11:44] <ubitux> prove it [11:45] <ubitux> (and not the vdpau thing, that's related to library names between the forks) [11:45] <j-b> http://upstream-tracker.org/versions/ffmpeg.html [11:46] <j-b> so, yes, you should split libavcodec+libavformat(+libavutil) from the rest. [11:46] <j-b> including tools [11:46] <ubitux> j-b: not sure how to interpret that; also, aren't most of those abi/api changes from libav? [11:46] <ubitux> if we bump, what's the problem? [11:46] <wm4> j-b: that seems to be an automatically generated report, and it doesn't know about the ABI rules put up by ffmpeg [11:47] <j-b> wm4: having rules about bumping does not make it less worse [11:47] <wm4> e.g. adding a field into a middle of the struct is allowed by these rules [11:48] <pross-au> sigh [11:48] <j-b> http://upstream-tracker.org/versions/cairo.html [11:48] <j-b> This is cairo [11:48] <j-b> used by everyone [11:49] <wm4> I see ABI problems in minor releases (hurrr) [11:49] <j-b> right [11:49] <j-b> This is over 8 years [11:50] <j-b> ubitux: the fact that it comes from libav does not change a thing for the users. [11:50] <ubitux> j-b: i know :( [11:50] <ubitux> j-b: but we can't do much about it [11:51] <wm4> anyway, I do think making structs opaque would be a good thing [11:51] <wm4> unfortunately libav disagrees [11:51] <nevcairiel> most of the reports for ffmpeg are false-positives based on the api/abi contract (ie. don't use fields after the marker point "internal"), not that I think this is a good concept and should probably just be moved into an Internal sub-object, but it still wouldn't break [11:51] <wm4> so go argue with them? [11:52] <j-b> wm4: I won't argue with them. I was just explaining why those projects are bad with versionning [11:52] <j-b> nevcairiel: there are even breakage of behaviours without ABI or API bumps. [11:52] <ubitux> that's our compatibility curse :( [11:53] <j-b> Well, if you used your libraries, maybe you would be less bad, sorry to say [11:53] <wm4> j-b: and to be fair, you can't really compare ffmpeg and cairo [11:54] <nevcairiel> that I can agree to, even from a api design standpoint, some of the api design ideas seem like noone ever used them before :p [11:54] <wm4> ffmpeg is a low level lib that deals with a constantly evolving subject (multimedia) [11:54] <wm4> cairo is a rendering API that was inspired by the (proven) post script rendering model (or something) [11:54] <j-b> ffmpeg is NOT a lib [11:55] <ubitux> we could probably take some ideas from the gst guys about that issue [11:55] <j-b> wm4: how is libavfilter or libavdevice evolving all the time? [11:55] <ubitux> but no idea how that would fit with libav merges [11:55] <wm4> j-b: well, recently they added opengl output support to libavdevice [11:56] <j-b> wm4: SOOO much NEW technology. Amazing. Wow [11:56] <pross-au> just wait for libavmesh [11:56] <wm4> whether that change is a good/useful thing is one question, but it certainly changes the requirements on the API [11:56] <wm4> ubitux: better look at ffms2 [11:56] <wm4> ubitux: it had a stable API and ABI for 10 years or so (with only a few minor breaks) [11:57] <ubitux> we can't avoid the break; the project is too large and has a too chaotic history [11:57] <ubitux> the idea is how to make it less painful for the users to deal with that issue [11:58] <ubitux> also, unless you can convince libav to change its behaviour concerning the abi/api evolution, that's harder for us to deal with that [11:59] <j-b> sure, but don't say you care about versions then [11:59] <ubitux> they're so proud of being the ones defining the api/abi so... you should see with them [11:59] <ubitux> j-b: well, we do care for changes coming from us [11:59] <j-b> You think I never complain about that to them? [11:59] <wm4> ubitux: you know, you could send patches to libav any time [11:59] <nevcairiel> caring and not being able to fix it are different things :D [11:59] <wm4> ubitux: you're not mini so you won't cause a jerk reflex [11:59] <ubitux> wm4: i have no solution to propose for the abi/api [12:00] <wm4> (because I'm pretty sure some libav devs do hate mini, despite what everyone says) [12:00] <ubitux> i could send a patch to libav-devel to rename the libs in libqatar* but they won't like it much [12:00] <j-b> but, yes, adding more libraries to the same, is not a good idea, since it encourages the libraries boundaries violations. [12:00] <wm4> ubitux: well, patches with 0% trolling content [12:00] <ubitux> wm4: i don't have any to suggest [12:00] <wm4> troll free, may contain traces of sarcasm [12:40] <BBB> such bad trolls... [12:40] <BBB> j-b: when are you moving to NY again? *hug* [15:21] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:d9a3501c33a1: avutil/opt: dont crash on av_opt_set_dict() with NULL [15:21] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:7aa3979b8c20: avformat/avio: also set generic URL context options [15:31] <superware> I have an H.264 stream over RTP, and I want to use FFmpeg avlib* to open and dump it to a mp4 file (no decoding/encoding), is there a tutorial that demonstrates something similar? I've tried doc/examples/muxing.c but couldn't understand how to achive my scenario.. [17:19] <cone-372> ffmpeg.git 03Peter Ross 07master:e61973db6c01: avformat/mpegtsenc: move startcode validity check to ff_check_h264_startcode [17:19] <cone-372> ffmpeg.git 03Peter Ross 07master:92d657b5f1a4: avformat/wtvenc: advise user when H264 startcode is not present [17:24] <wm4> ubitux: ok... so, ffmpeg uses AVFrame.metadata [17:25] <wm4> ubitux: but there is also AVFrameSideData.metadata [17:25] <wm4> ubitux: so, couldn't ffmpeg just use an empty side data type, and use that second metadata field? [17:27] <ubitux> so the current trending is to use AVFrameSideData instead of frame->metadata? [17:40] <wm4> ubitux: yes, see replaygain stuff [17:40] <wm4> ubitux: and using structs for side data [17:40] <wm4> ubitux: replaygain doesn't use the AVDictionary [17:41] <wm4> actually I wonder when AVFrameSideData.metadata is used at all (?) [17:41] <ubitux> moving to AVFrameSideData will require updates in a small dozen of filters [17:42] <ubitux> i'm also assuming they're properly raised up to ffprobe [17:42] <ubitux> and usable at this stage [20:47] <cone-372> ffmpeg.git 03Vadim Kalinsky 07master:234f0bcb0c73: lavd: Add QTKit input device. [21:08] <cone-372> ffmpeg.git 03Timothy Gu 07master:9e4e35b4d7c4: avconv_opt: fix avio_open2() return code check [21:08] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:ce0ec108cd34: Merge commit '9e4e35b4d7c43a908944183a58aa389a23116fd6' [21:17] <cone-372> ffmpeg.git 03Timothy Gu 07master:68e95ab81be1: dnxhdenc: return meaningful return codes [21:17] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:f22d9b1e0b63: Merge commit '68e95ab81be1aa3f47ab148dceb8711ef5f4212d' [21:20] <kierank> <devils advocate> [21:20] <kierank> possibly ffmpeg should just break API/ABI [21:20] <kierank> completely from libav [21:20] <kierank> and see what happens [21:20] <kierank> </devils advocate> [21:22] <ubitux> i don't think that would be a wise move until it's packaged in debian [21:28] <wm4> the ffmpeg build system is so frustrating [21:28] <cone-372> ffmpeg.git 03Timothy Gu 07master:3a5a965493fa: avconv: make the ASCII flow charts narrower to fit onto TTY [21:28] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:9c77e57393cc: Merge remote-tracking branch 'qatar/master' [21:28] <wm4> update -> try to rebuild -> rebuild for some minutes -> suddenly some weird error about "No rule to make target..." [21:29] <wm4> it's that issue that happens when removing/renaming files or so [21:29] <wm4> try to fix it -> yeah no -> make clean & rebuild everything... [21:36] <superware> I've been trying all day to get an h.264 over RTP into a local mp4 file (dump/copy, no encoding/decoding). I'm reading the stream packets using av_read_frame into an AVPacket. How should I setup the output context? how can I define that av_write_frame won't do any decoding/encoding? [21:38] <ubitux> wm4: it sounds like something Diego would want to fix since he's moving files all the times and master the build system [21:50] <superware> or.. how should I mux AVPackets (h264/RTP) into mp4 file? [21:50] <ubitux> BBB: any suggestion for mc? (what i should do next) [21:50] <ubitux> since put/avg are done [22:12] <BBB> fullpel or 8tap subpel? [22:12] <ubitux> i did fullpell [22:13] <BBB> subpel is always number 1 in all profiles [22:13] <BBB> so I'd do subpel [22:13] <BBB> I haven't done the merged 2d x86 sims yet [22:13] <ubitux> (nothing new from yesterday except that it works - so what's sent in the patch) [22:13] <BBB> but I'll look into it :-p [22:13] <ubitux> ah you want to write some arm simd? [22:13] <ubitux> or you're suggesting me the 8tab subpel? [22:14] <BBB> yes [22:14] <BBB> I'll do the merged 2d x86 simd [22:14] <BBB> you should do the arm 8tap subpel simd [22:14] <BBB> I'd start with just 1d [22:14] <BBB> since 2d is well, hard [22:15] <BBB> and then a c wrapper around 2 1ds to get the 2d effect [22:15] <BBB> like we do for x86 [22:15] <BBB> but maybe later merge them all in assembly [22:15] <ubitux> ok [22:16] <j-b> m [22:16] Action: BBB hugs j-b [22:17] Action: j-b hugs BBB [23:13] <cone-372> ffmpeg.git 03Thilo Borgmann 07master:6d9bdd9d8b6a: doc/indevs: Fix example for QTKit usage. [23:42] <cone-372> ffmpeg.git 03Matt Oliver 07master:0f2588d7e5d2: Use intel compliant CDQ instead of CLTD in inline asm. [00:00] --- Mon Mar 31 2014

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ffmpeg.log.20140330
by burek 31 Mar '14

31 Mar '14

[10:22] <blockh34d-sleep> is ffprobe available by itself somehow? [10:23] <blockh34d-sleep> i wrote an app that uses ffprobe for some stuff, seems like overkill to install ffmpeg just for ffprobe [13:32] <Mavrik> ffmpeg is a tiny binary? O.o [13:55] <superware> I have an H.264 stream over RTP, and I want to use FFmpeg avlib* to open and dump to a mp4 file, is there a tutorial that demonstrates something similar? [14:01] <JEEB> look at the demuxing/decoding example under doc/examples [14:02] <JEEB> although in your case unless you want to re-encode the H.264 stream, you should be able to just use libavformat to demux the RTP stream, and then you can use another libavformat instance to take those packets and mux them into mp4 [14:02] <JEEB> there's most probably a muxing example as well, so take a look at both [14:07] <superware> ok, thanks [14:21] <superware> JEEB: couldn't really understand what to use in order to mux into mp4... :| [14:23] <JEEB> same as for the demuxing of packets [14:26] <julienb> hello, i have a problem with my server which is rebooting always when i make conversion with ffmpeg [14:26] <julienb> i would like to know if it can be ffmpeg or other thing? [14:34] <superware> JEEB: I've successfully opened the network stream, and looping over incoming AVPackets, but how should I open and write to the output file? [14:36] <superware> I guess it should be quite simple considering there's no decoding envolved [14:38] <JEEB> I can see a muxing example. That example also encodes a synthetic audio and video stream, but it also contains the basics of muxing [14:38] <JEEB> doc/examples/muxing.c [14:51] <superware> JEEB: avformat_alloc_output_context2; add_stream; avcodec_open2; avcodec_alloc_frame; avio_open; avformat_write_header; av_interleaved_write_frame per frame? [15:13] <superware> JEEB: sorry for nagging, you're probably my only option to do this right :) [15:15] <CokaCola> bin/ffmpeg -i $1 -acodec pcm_s16le tmp/output.wav --unable to find a suitable output format for 'tmp/output.wav' [15:15] <CokaCola> any ideas? I even tried just going ffmpeg -i $1 output.wav(also tried .ogg), same problem [15:52] <superware> JEEB: ? [15:56] <CokaCola> Weird, it works fine if I run it directly in the console [15:56] <CokaCola> So somehow running it in a script doesn't work : [16:46] <dagerik> I used ffmpeg to extract a clip from longer video. However, subtitles seem to be off by default. [16:46] <dagerik> To enable mplayer I need to hit j to enable it [16:46] <dagerik> To enable subtitles in mplayer I need to hit j [17:08] <jnvsor> How do I keep sound in sync when streaming flv? [17:43] <PKvid> Does anyone know how to create a waveform file like jpg/png with ffmpeg? [18:07] <blockh34d-zzz> does anyone know of any ffmpeg packages that only include ffprobe and smaller utils like that? [18:08] <blockh34d> i wrote a little app that uses ffprobe to get some media information, so now people need to install ffmpeg to use it, and it seems like overkill a bit just to get at ffprobe [18:09] <PKvid> I've got ffprobe [18:09] <PKvid> what command mus I use? [18:10] <blockh34d> to what? [18:10] <PKvid> Does anyone know how to create a waveform file like jpg/png with ffmpeg? Sorry I thought you were talking to me [18:10] <blockh34d> oh, no i was justtrying to figure out how to reduce my apps install size [18:11] <blockh34d> i'm confused by waveform file jpg/png [18:11] <blockh34d> what does that mean? [18:11] <sacarasc> blockh34d: Why not use the ffmpeg libraries and implement it yourself? [18:11] <blockh34d> wouldnt users still have to install ffmpeg to install my app then? [18:11] <PKvid> It is posssible to generate a waveform representation of the audio file [18:11] <sacarasc> Not if you statically built it! [18:11] <PKvid> I wondered if it's possible with ffmpeg to save it as an image [18:12] <sacarasc> Also, packaging is a distro thing. [18:12] <blockh34d> sacarasc: oh i was wondering about that... am i allowed to redist my compiled app like that? i thought it might be a license problem [18:12] <blockh34d> am i allowed to redist my app with a static build of ffprobe? [18:13] <sacarasc> Probably. [18:13] <blockh34d> btw i dont know much of anything about any of this, in case thats not obvious, just stumbling through it all here [18:14] <blockh34d> pkvid i use ffmpeg (avconv) to strip audio out of a mpg and save it as a mp3 [18:14] <blockh34d> is that at all what you're trying to do? [18:14] <sacarasc> They're trying to get an image of the wave form. [18:14] <sacarasc> Not get the audio. [18:14] <PKvid> sacarasc Indeed [18:15] <blockh34d> oh [18:15] <blockh34d> well thats a little harder i guess, or at least, i dunno how to do that with ffmpeg [18:15] <sacarasc> Especially as you're not using ffmpeg. :D [18:16] <sacarasc> #libav is the avconv channel. [18:16] <sacarasc> Etc. [18:16] <blockh34d> oh [18:16] <blockh34d> well i'm pretty confused at this point [18:16] <blockh34d> i thought ffmpeg and avconv were basically the same project [18:16] <blockh34d> just new names [18:18] <blockh34d> oh thanks for that stackof link that explains a bit [18:19] <blockh34d> fyi raspbian (debian) still shows the weird error message [18:19] <blockh34d> i think i'd rather use ffmpeg than libav, avconv, etc... [18:19] <blockh34d> libav sound like... jackasses [19:10] <PKvid> Does anyone know how to create a waveform file like jpg/png with ffmpeg? [19:11] <sacarasc> Have you tried reading the manual? [19:15] <PKvid> Yes, could not find [19:15] <PKvid> anything [19:17] <blockh34d> so sacarasc you've got me thinking about making my own playback app and ffprobe-ish app now [19:17] <blockh34d> how realistic to you think it would be for me to make a mp3 player that once or twice per second, output FFT peak data to stdout [19:18] <blockh34d> cause i have a player GUI i made (for raspberry pi's omxplayer), and i'd like to be able to have a winamp style eq in it [19:18] <blockh34d> the gui is atcuatll text mode so tui i guess... all curses and asciiart [19:19] <JEEB> raspberry pi isn't fast enough to encode AAC with the libavcodec encoder [19:19] <blockh34d> is that what i'd need to do to get that data? [19:19] <blockh34d> cuase i think omxplayer does all decoding in hardware [19:20] <blockh34d> so i dunno that i'd ever have access to the raw decoded data [19:25] <xlinkz0> I'm trying to capture the screen on windows 8.1 [19:25] <xlinkz0> but "ffmpeg -list_devices true -f dshow -i dummy" doesn't show the screen-record [19:26] <blockh34d> maybe scrot would work? [19:26] <blockh34d> i use scrot for screen capture on rpi debian [19:28] <blockh34d> n/m i guess if you wanted to use scrot you would! lol i didnt even know ffmpeg could record the screen, thats pretty cool [19:30] <xlinkz0> scrot seems to be linux only :( [19:30] <blockh34d> ah oh well. surprising there isn't more available options for win 8 [19:30] <blockh34d> i know how to fix this [19:31] <blockh34d> step 1: remove windows 8.1 [19:31] <blockh34d> forever [19:31] <blockh34d> step 2: _______ [19:32] <blockh34d> step 3: PROFIT! [19:33] <blockh34d> well i'm gonna go offtopic rant somewhere else [19:33] <blockh34d> have a nice day everyone [19:46] <jnvsor> Guess I'll ask again... How do I ensure video and audio will be synced streaming to twitch from ffmpeg? [19:54] <klaxa> i don't think you can, what command line are you using? (exclude your stream key) [19:56] <klaxa> when i get back, i'll see what i can do, afk for now [20:37] <jnvsor> klaxa: http://pastebin.com/jNRsURhe [20:56] <xlinkz0> nsfw : http://i.imgur.com/PPSwzlb.gif [20:56] <xlinkz0> thank you ffmpeg ! [20:56] <xlinkz0> lol [21:02] <ubitux> is this a boob version of 2048? [21:34] <superware> I've been trying all day to get an h.264 over RTP into a local mp4 file (no decoding/encoding). I'm reading the stream packets using av_read_frame into an AVPacket. How should I setup the output context? how can I define that av_write_frame won't do any decoding/encoding? only dump.. [21:49] <superware> or.. how should I mux AVPackets (h264/RTP) into mp4 file? [21:55] <superware> JEEB: can you please help me? [22:16] <Foxhoundz> I need help [22:16] <Foxhoundz> ffmpeg won't convert a wave file to mp3 for some reason [22:16] <Foxhoundz> I've installed libmp3lame [22:17] <Foxhoundz> I get the following error when running it: Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height [22:17] <Foxhoundz> and my initial input /usr/bin/ffmpeg -b 192k -i /var/www/drupal-7.26/sites/default/files/drmapan.wav /var/www/drupal-7.26/sites/default/files/drmapan.mp3 [22:18] <Foxhoundz> removing the -b switch doesn't help it either [22:18] <Foxhoundz> any ideas? [22:18] <JEEB> you're setting a bit rate for input/decoder [22:18] <JEEB> -b:a 192k after -i [22:19] <Foxhoundz> but it didn't work even after removing that switch [22:19] <JEEB> then it didn't have any good rate control mode set? [22:19] <JEEB> maybe pastebin the full output and link it here? :P [22:19] <MaZderMind> Hi all. I have a problem getting stills from anamorphic videos. running `ffmpeg -i http://ftp5.gwdg.de/pub/misc/openstreetmap/FOSSGIS2014/716-h264-hq.mp4 -ss 20 -vframes 1 thumb.png` results in a png with the native pixel numbers, presumably because pngs don't support none-square pixels [22:20] <MaZderMind> I don't know if all videos are 16:9-anamorphic, some may be 4:3, so I'd like to avoid hard-coding pixel-sizes. all should be in the video-file, ffprobe reports propper DAR of 16:9 [22:20] <JEEB> MaZderMind, https://trac.ffmpeg.org/wiki/Scaling%20%28resizing%29%20with%20ffmpeg [22:20] <JEEB> see the examples here [22:20] <Foxhoundz> JEEB: here's the output http://laravel.io/bin/X2VO [22:21] <JEEB> Foxhoundz, first of all when using libav-based tools you use avconv, not ffmpeg. second of all, try setting that bit rate [22:22] <klaxa> jnvsor, try to use -r instead of -framerate and move it after -i :0.0 [22:23] <klaxa> maybe someone should update the docs for that, it's a common command and many people run into synchronization issues with it [22:23] <MaZderMind> JEEB: thank you. I'm not sure I see an example that wil work without knowing the source pixel size. [22:24] <MaZderMind> all I'm looking for was a capture of the video-fram as ffplay would show it [22:24] <JEEB> well, you DO know it (as in there's a symbol for the scale filter for both SAR and DAR) [22:24] <JEEB> and goddamnit [22:24] <JEEB> you most definitely did not read that page to the end [22:25] <MaZderMind> JEEB: kk, got me. going to read before coming back :) [22:25] <JEEB> more here http://ffmpeg.org/ffmpeg-all.html#Examples-10 [22:25] <JEEB> (the last three or so examples in the latter are useful for you as well) [22:26] <JEEB> and then you can scroll up for the most used parameters you can give the scale filter) [22:26] <MaZderMind> JEEB: thank you, I wasn't able to google that constant listing. That helps [22:28] <Foxhoundz> JEEB: The drupal convertor uses ffmpeg so I don't have much choice [22:28] <JEEB> well you just defined the command line [22:28] <Foxhoundz> as long as avconv is being aliased as ffmpeg [22:28] <Foxhoundz> then it shouldn't matter [22:28] <JEEB> well in 0.8 it isn't [22:28] <JEEB> it's IIRC the old pre-updated ffmpeg [22:29] <JEEB> also I do recommend you update your libav if you're going to use it to one of the ones available from the multimedia team's PPA. 0.8 is /old/ , but unfortunately debian didn't finish the migration in time for a newer release to get into 13.10 [22:30] <MaZderMind> thank you, works. have a nice night [22:30] <JEEB> https://launchpad.net/~motumedia if you're running a server you probably want one of the PPAs giving out the stable version 9 or 10 for your version, not daily builds [22:31] <JEEB> or you can of course grab FFmpeg ffmpeg from one of the static build places [22:31] <JEEB> whichever fits your fancy [22:31] <Foxhoundz> what was bit rate you suggested again? [22:31] <Foxhoundz> -ab 192k? [22:32] <JEEB> well, 192k was what you wanted to set :P [22:32] <JEEB> but yes -ab should work with that old old ffmpeg [22:32] <JEEB> avconv and newer FFmpeg ffmpeg take -b:a just fine [22:32] <Foxhoundz> still a no go [22:32] <Foxhoundz> :( [22:33] <Foxhoundz> So I guess I have to upgrade this ffmpeg [22:33] <Foxhoundz> do you know any PPAs that have the updated ffmpeg binaries? [22:33] <Foxhoundz> clear [22:34] <JEEB> no, but you should be able to grab a static binary or so from somewhere linked from ffmpeg.org [22:41] <Foxhoundz> :D [22:41] <Foxhoundz> I think it worked! [22:41] <Foxhoundz> let's see let's see! [22:42] <esrax> hi guys, using avconv to stream x264 video in mpegts and the receiving end is complaining that the pts isnt changing sometimes [22:44] <JEEB> avconv comes from the libav project, #libav is the correct channel for it :) [22:46] <esrax> when i go to use ffmpeg on debian and it tells me ffmpeg is deprecrated, what is that about? [22:46] <JEEB> it means it's deprecated within the libav project [22:46] <Foxhoundz> yay it worked! [22:46] <JEEB> ffmpeg's ffmpeg is separate [22:46] <sacarasc> It means you're using avconv from libav. [22:46] <esrax> geez so how do i get the real ffmpeg on debian? [22:47] <esrax> i'll give that a shot [22:47] <JEEB> basically when elenril rewrote parts of ffmpeg's option stuff, libav also decided to rename the binary to avconv and leave the ffmpeg binary to wither around until the next release :) [22:47] <JEEB> FFmpeg merges most of the libav's changes so it contains those, too [22:47] <esrax> can i expect to see different behaviour between the two? [22:48] <JEEB> possibly some [22:48] <esrax> worth a shot then [22:48] <JEEB> and usually the repository stuff is just old [22:48] <JEEB> debian didn't finish migration to libav 9 in time for 13.10 [22:48] <JEEB> so basically every ubuntu release from 12.04 to 13.10 contains a libav release from very early 2012 [22:48] <JEEB> which is like... ugh [22:49] <JEEB> at least the libav 10 migration is now taking less time, although it did miss the 14.04 timeframe [22:49] <JEEB> so the new LTS will have an already year-old release >_> [22:50] <esrax> yeah still getting the pts not changing stuff on both the video and audio streams [22:50] <sacarasc> That Libav decided to go from 0.8.9 to 9 is kinda funny. [22:50] <JEEB> the third number doesn't really matter, but yeah -- a jump from 0.x to x [22:51] <JEEB> (Ž [22:51] <esrax> yeah i went to setup this relay server on rhel 6.x first and then...well...i know rhel is behind the times n all but wow [22:52] <sacarasc> ffmpeg 2.3.0 will be 23.0... [22:52] <sacarasc> Getting into a version race. [22:52] <JEEB> chrome style [23:45] <jnvsor> klaxa: You told me to use -r instead of -framerate but this tends to make desync issues far worse - if the device can't keep up then with -r it won't duplicate frames and will desync [23:46] <klaxa> did you put it behind -i :0.0 like i told you too? [23:48] <active8> Trying to get my screen grab in mp4 now. I've pasted the 2 commands and then the output of each with a big space in between so it's easier to read: http://pastebin.com/SaMb3Vr8 [23:49] <jnvsor> klaxa: Haven't had time to record a long enough video to see the desync yet [00:00] --- Mon Mar 31 2014

1 0

ffmpeg-devel.log.20140329
by burek 30 Mar '14

30 Mar '14

[00:00] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:067a9cf81a78: avformat/img2dec: make image2dec capable to be used from seperate demuxers [00:00] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:13bcb4de33b3: avformat: add image2 alias pix demuxer [00:00] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:075d6c066bc9: avformat: add image2 brender pix demuxer [00:00] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:1c13e1ef368a: avformat/img2dec: Use avformat probing interface to identify format if it has not been otherwise identified [00:49] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:2cffdcbdd7f6: avformat/img2dec: try to read PROBE_BUF_MIN instead of just enough for .pix probing [00:49] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:657cee1aef72: avformat/img2_alias_pix: rewrite probe function [03:11] <LoH> I've run into a problem compiling FFMpeg from source with avxsynth support. It looks like configure is looking for dlopen/LoadLibrary. [03:12] <LoH> I think what's going on is that I'm trying to compile on FreeBSD10, and that system comes with dlopen support bakedi n [03:12] <LoH> *baked in [03:15] <LoH> Is this the right channel, or should I go back to #ffmpeg? [03:17] <LoH> What I have done is comment out the check for dlopen in configure. Hopefully it works out [03:59] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:067ada04d196: avcodec/xbmdec: redesign parser to handle more cases [04:02] <michaelni> LoH, if you get no reply, maybe write the author/maintainer of the avxsynth code in ffmpeg [04:02] <Compn> LoH : it probably wont, bug reports and user questions still go to #ffmpeg tho [04:04] <LoH> Compn, it appears to work, but I don't have anything else to check the output with -.-; [04:06] <LoH> I'll head back to #ffmpeg though [04:07] <Compn> i dont mind, but some of these other guys :P [04:07] <Compn> ehe [04:13] <LoH> There's a guy who managed a freebsd port of ffmpeg, but their version is slightly older than what's wanted by ffms2 [04:17] <Compn> why using avxsynth with ffms2 ? [04:17] <Compn> er, i'm thinking of a diff project, nevermind [04:40] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:46f72ea507af: avcodec/vp7: check buffer size [15:03] <cone-447> ffmpeg.git 03Luca Barbato 07master:85698be461c0: cmdutils: Mark exit_program as av_noreturn [15:03] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:d840266633c6: Merge commit '85698be461c07be10d873dd34348bcfe9ffc56e0' [15:22] <jnvsor> How do you pipe the ffmpeg output to sed? The framerate updates don't newline so you get no flush [15:43] <cone-447> ffmpeg.git 03John Stebbins 07master:6adf3bc42e36: movenc: Add dvd subtitle support [15:43] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:b8f5b0713e81: Merge remote-tracking branch 'qatar/master' [15:43] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:8a9d0a156147: avformat/movenc: fix if vs if else [16:10] <J_Darnley> jnvsor: ffmpeg printes messages to stderr [16:10] <J_Darnley> *prints [16:14] <jnvsor> J_Darnley: Yeah, but it doesn't flush the fps messages so if I redirect stderr to stdout and pipe the lot to sed it doesn't flush [16:27] <J_Darnley> Other tools usually only flush with \n [16:28] <J_Darnley> ffpeg only prints \r for progress reports [16:42] <jnvsor> J_Darnley: So any way to get sed to flush on \r? [16:44] <J_Darnley> no idea [16:44] <J_Darnley> change it to \n? [16:45] <jnvsor> J_Darnley: Huh, didn't think that might work [16:48] <jnvsor> J_Darnley: Nah, with `sed -e "s%\r%\n"` it doesn't flush, guessing the flushing is done before the replace [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:fd786bad6321: tools/uncoded_frame: fix audio codec generation [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:27256e69ab2d: lavd/pulse_audio_enc: implement write_uncoded_frame callback [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:cd50a44beb01: lavu/mem: add av_dynarray_add_nofree function [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:85ed32d2ed15: lavd/pulse_audio_common: add device detecting code [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:255cf03af81e: lavd/pulse_audio_dec: implement get_device_list callback [17:13] <cone-447> ffmpeg.git 03Lukasz Marek 07master:3937b40e87c9: lavd/pulse_audio_enc: implement get_device_list callback [17:13] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:bcd5fd5346be: Merge commit 'lukaszmluki/master^' [17:25] <ubitux> BBB: [17:25] <ubitux> * dimension 4: x subpel interpolation (0: none, 1: 8tap/bilin) [17:25] <ubitux> * dimension 5: y subpel interpolation (1: none, 1: 8tap/bilin) [17:25] <ubitux> second line, "0: none", right? [17:26] <nevcairiel> its probably safe to assume as much [17:26] <ubitux> just wanna confirm before pushing :) [18:06] <BBB> ubitux: yes [18:06] <BBB> ubitux: did i write that> [18:06] <ubitux> seems so [18:06] <BBB> :( [18:06] <BBB> ohwell [18:06] <ubitux> lavc/vp9dsp.h [18:07] <ubitux> BBB: can you have a quick look to the patch for vp9 namespace in mc? [18:07] <BBB> okie [18:08] <BBB> yeah looks good [18:08] <ubitux> ok thanks [18:13] <BBB> no other patches right? [18:13] <BBB> I'm looking to see if there's a mc hevc update,e but don't see anything yet [18:14] <ubitux> no other patch yeah, i didn't have time to work on it until today [18:14] <ubitux> i'm slowly starting to work on neon again [18:17] <BBB> cool [18:21] <cone-447> ffmpeg.git 03Clément BSsch 07master:c4148a6668b7: x86/vp9mc: add vp9 namespace. [18:21] <cone-447> ffmpeg.git 03Clément BSsch 07master:af3b6aed0da5: avcodec/vp9dsp: fix typo in mc doxy. [19:34] <ubitux> anything better than this: [19:34] <ubitux> sub r3, #16 [19:34] <ubitux> vld1.8 {q0}, [r2]! [19:34] <ubitux> vld1.8 {q1}, [r2], r3 [19:34] <ubitux> ? [19:34] <ubitux> i'd like to get rid of the sub (stride -= 16) [19:35] <ubitux> something like "vld1.8 {q0}, [r2], #16" + "vld1.8 {q1}, [r2,#16], r3" [19:44] <ubitux> BBB: btw, is it ok for put4/avg4 to read 8 instead of 4? [19:45] <ubitux> i'm still not sure how to deal with those in neon [19:45] <ubitux> assuming it makes any sense [19:46] <ubitux> i mean, there is no simd to write for those [20:01] <BBB> ubitux: I believe it's fine, the buffer is padded, but you can't write 8 [20:02] <BBB> ubitux: isn't there an instruction to read sets of 4 bytes and interleave? [20:02] <BBB> you could read 4 sets of 4 bytes separated by "stride" bytes and interleave, to do 4x4 pixels in a single go [20:02] <ubitux> yeah probably, dunno [20:03] <BBB> neither do I :-p [20:03] <ubitux> put8/16/32/64 done btw, i'm working on avg [20:03] <BBB> cool [20:03] <BBB> you'll have to help me decrypt your asm btw, my neon isn't as good as yours [20:03] <BBB> like, I still don't know 100% sure what ! means [20:04] <BBB> it's something like pre or post-increment or something [20:04] <ubitux> something like that [20:04] <ubitux> :D [20:04] <ubitux> afaict it's postinc [20:06] <BBB> post-inc of what? [20:06] <BBB> and by what? [20:06] <BBB> shouldn't post-inc be vld1.8 {q0},[r2]!, r3? [20:06] <BBB> or so [20:06] <BBB> (dunno) [20:07] <ubitux> ! will post inc of #16 [20:07] <ubitux> but you can't do vld1.8 {q0},[r2],#16, so you do vld1.8 {q0},[r2]! [20:07] <ubitux> i guess your line would do #16+r3? [20:07] <ubitux> but... no idea :D [20:08] <ubitux> the doc really sucks honestly [20:11] <BBB> neon is kinda weird [20:11] <BBB> well anyway it doesn't have to be 100% perfect every first try, plus feel free to copy mpeg/h264 optimizations from neon, they should be self-explanatory [20:11] <BBB> or vp8 [20:12] <ubitux> yup [20:23] <ubitux> https://github.com/ubitux/FFmpeg/commits/vp9-arm not much atm [20:23] <ubitux> put8/16/32/64 and avg8 [20:24] <ubitux> i'd really like to avoid these sub in put32/put64 [20:52] <BBB> why does put16 use vld1.8? [20:54] <ubitux> the value aren't 8-bit coded? [20:54] <BBB> oh it's bits? [20:54] <BBB> I didn't know, I thought it was bytes (load 8 bytes) [20:54] <BBB> lol [20:54] <BBB> functions look good then [20:54] <BBB> I'd commit it already [20:55] <BBB> does it help speed-wise? [20:55] <ubitux> maybe it means something else... :x [20:55] <ubitux> BBB: not much so far [20:56] <ubitux> BBB: i wouldn't push them immediately [20:56] <BBB> and so ! is pre-increment [20:56] <ubitux> there are a few things i want to check first [20:56] <BBB> whereas , X is post-increment [20:56] <ubitux> pre? no don't think so [20:56] <BBB> anyway I'd be interested in the real mc [20:56] <ubitux> i'd say post [20:56] <ubitux> , X is also post increment [20:56] <ubitux> but with a reg [20:56] <BBB> we should still merge 2d 8taps in x86 simd [20:57] <BBB> oh [20:57] <BBB> I guess you're right [20:57] <BBB> mis-read stuff [20:57] <ubitux> also, there are some stuff with alignment [20:57] <ubitux> i can probably add align hints in some places [20:57] <BBB> yeah we never did that in x86 either [20:57] <BBB> plus prefetch still big todo [20:57] <BBB> all these can help quite a lot [20:58] <BBB> anyway, we're already fast so who cares :-p [21:12] <ubitux> these functions doesn't make much difference on overall time for now [21:13] <ubitux> don't* [21:13] <BBB> it's possible that when compiled with neon support, gcc outputs that same assembly already through memcpy [21:13] <ubitux> it's like 0m15.864s 0m15.756s [21:13] <ubitux> yeah probably [21:14] <BBB> 0.1sec is still a good start though, always worth it [21:14] <BBB> how much fps do you get on HD material (1080p)? [21:14] <ubitux> about 3 [21:14] <BBB> ... [21:14] <ubitux> :) [21:15] <ubitux> these ~16s were 50 frames of etv5k [21:15] <BBB> omg [21:15] <ubitux> :D [21:15] <BBB> I thought it was like 1000 frames or so [21:15] <BBB> ok, carry on, ignore me :-p [21:16] <ubitux> yeah i don't do 60 fps on arm :)) [21:16] <ubitux> BBB: src and dst are always aligned? [21:18] <ubitux> mmh seems not [21:19] <ubitux> source isn't but dst is. ok [21:34] <BBB> right [22:38] <Ove_88> Hello everyone. I'm trying to seek to an exact frame number using AVSEEK_FLAG_FRAME in av_seek_frame and avformat_seek_file, but they always return -1. Do you know what I'm doing wrong? [22:39] <Ove_88> I've also looked at the source code for the demuxers in ffmpeg, and as far as I can tell, none of them work with AVSEEK_FLAG_FRAME, although I might be wrong. [22:40] <Mavrik> hmm [22:40] <Mavrik> checking out how ffmpeg.c does it would probably help you [22:44] <Ove_88> ffmpeg.c does not use AVSEEK_FLAG_FRAME [22:45] <nevcairiel> I doubt it works, and even if it would accept such parameters, its doubtful it would be frame accurate [22:45] <Ove_88> also, ffmpeg.c can't seek using frame numbers, it can only seek using hh:mm:ss.ms [22:46] <ubitux> mmh how is there any special tricks i should be aware of for running a fate instance with cross compiled tools? [22:46] <ubitux> -how [22:48] <Ove_88> ok, I thought it worked since I saw it in the docs [22:48] <Ove_88> I'll have to seek by timestamp instead [23:15] <ubitux> BBB: going to merge the x86 2d 8taps? [23:24] <J_Darnley> ubitux: I guess that will depend on your host and target [23:25] <J_Darnley> I tried using cygwin's cross compilers for native windows but the test script was doing strange things to file paths [23:50] <ubitux> :/ [23:50] <ubitux> i guess i'll just build ffmpeg on the board and copy the cross compiled one after each iteration [23:59] <J_Darnley> Well if you're staying with linux paths my particular issue won't bother you [23:59] <BBB> ubitux: sure I can try [00:00] --- Sun Mar 30 2014

1 0

ffmpeg.log.20140329
by burek 30 Mar '14

30 Mar '14

[02:42] <kyle__> Incase anyone searches the logs for questions and answers, it turns out I have to tell ffmpeg the subtitle codec is dvdsub, so it converts dvdsub to dvdsub, instead of copying it. [09:27] <zoidey> Hello. I'm trying to find out what compile time options are required for the following command: ffmpeg -y -i 1010677091.webm -vcodec mjpeg -an -q:v 8 -vf fade=in:0:30:color=black 1010677091.mov [09:27] <zoidey> It runs with one ffmpeg binary I have, but with another it fails with: [09:27] <zoidey> Option 'color' not found [09:27] <zoidey> Error initializing filter 'fade' with args 'in:0:30:color=black' [09:56] <ubitux> zoidey: probably too old [09:57] <ubitux> it was added "recently" [09:57] <ubitux> you'll find the feature in the latest stable, and maybe the one before, 'cant remember [10:05] <zoidey> ah, makes sense, I'm using different versions [10:05] <zoidey> thanks! [13:28] <jnvsor> I'm trying to remove a streaming key from ffmpeg output but if I do I don't get the framerate output unless I use a command and flush the output [15:38] <relaxed> jnvsor: do what? [15:54] <CokaCola> Was using audacity to convert MP3s to signed 16bit pcm wav files [15:54] <CokaCola> Can't seem to get it to work with ffmpeg [15:54] <CokaCola> my output file is larger(has a higher bitrate) but -b(or -b:a) option seems to have no effect on the output file [15:55] <CokaCola> Wait..nope I got it nevermind :p [15:55] <jnvsor> relaxed: I have a twitch key I'm streaming to, I can remove it with sed but then I don't get fps updates [16:04] <klaxa> what? [17:53] <ffmpeg_simple> Hi, I am interested to use the ffmpeg for a project in Matlab. [17:54] <ffmpeg_simple> I would like to be able to use audio encode/decode passing an array to ffmpeg, along with the codec to be used, and array size. [17:55] <ffmpeg_simple> I have managed to do the ffmpeg-matlab integration, and now I would like to understand how to easily pass the array to ffmpeg, instead of the default solution: via files on the disk [17:59] <ffmpeg_simple> being a rusty programmer, I got lost in the examples included in the documentation [17:59] <ffmpeg_simple> is there anyone who could guide calling the audio coding/decoding methods with array as arguments ? [22:11] <active8> Have I croosed into the realm of the impossible? I captured a screen (x11grab) while a youtube was playing and I noticed that as I was moving the pavucontrol slider, the sound and video wasnt in sync and the ends of video and audio didn't coincide, of course. So I had the audio bit rate set wrong or something. Is there any way to find out the bit rates and all that, is there. [22:12] <klaxa> maybe your command line is wrong, can you pastebin your commandline and the complete output? [22:25] <active8> klaxa. ok. one sec or three [22:27] <active8> klaxa: here ya go: http://pastebin.com/10xHN5an [22:28] <active8> you reminded me of those other errors I meant to ask about. THe DTS and out of order (IIRC) stuff [22:29] <klaxa> as expected, you put the -r 30 before -i :0.0 which means that ffmpeg will wait until x11grab has provided 30 frames and will put those into one second [22:29] <klaxa> put -r 30 after -i :0.0 and try again [22:34] <active8> I'm letting it run. My notes have that command highlighted as the one that worked a month ago on another video I just watched it. Either the sound was synced correctly or I suck. Heh, Now that I think of it, there wasn't much visual cue to determine that. [22:38] <active8> damn klaxa. The errors are gone and the volume tracks with me moving the sliders. This is one of the top 2 irc channels, IMO. [22:39] <klaxa> :) [22:40] <active8> The reason I was playing with the sliders was it sounded garbled at one point and the levels were pegging the "VU meter" at the time so I thought it might be clipping. The one from a month ago didn't do that. [22:40] <active8> it still garbles, BTW. [22:41] <klaxa> hm... it's not the audio source? maybe the resampling does it [22:43] <active8> no. sounds great. If you never saw Shadow of the Sword, I can't reccommed it because I fell asleep. I woke up just as this awesome gal started singing the chorus of the ending theme soneg: https://www.youtube.com/watch?v=pU75eBAKISU [22:44] <active8> I don't think I have any more processes running this time, compared to last. The only thing that beats this box to death is firefox with 20, 000 tabs open [22:45] <active8> I'm running it with the pavucontrol at 50% to see. [22:48] <active8> no joy. at least I got it sync'd with none of those errors. Those 2 of 6 cores running virtual XP shouldn't matter. [23:04] <jnvsor> How can I prevent desync in screen capture? The audio seems to desync randomly between 10 and 1 second every half hour [23:12] <active8> klaxa: I let that run without touching the slider and (3rd time with distractions) I'm not hearing that garbling. [23:12] <active8> I hope I didn't just lie [23:12] <klaxa> heh [23:16] <active8> Sounds smooth and clear as a bell. I'm ffmpegged out for the day, klaxa. Thank you very much for your help. [00:00] --- Sun Mar 30 2014

1 0

ffmpeg-devel.log.20140328
by burek 29 Mar '14

29 Mar '14

[00:01] <wm4> it's being a bit impractical here, but I did it in another way already [00:21] <wm4> ubitux: here's a patch for ASS utf16 support: http://sprunge.us/EHcY [00:21] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:64b79141bdfd: avcodec/libx264: move where x264opts is applied down so it isnt overridden by avctx & defaults [00:21] <wm4> ubitux: other subtitle readers could be extended similarly, although they have more dependency on random other stuff [00:29] <wm4> why do people do this https://github.com/chelyaev/ffmpeg-tutorial/blob/master/tutorial05.c#L529 [01:16] <kierank> wm4: you had to do that back in the days [01:16] <kierank> with reordered opaque [01:16] <wm4> even then you could do it in a cleaner way [01:17] <wm4> at least in the last few years [03:23] <cone-447> ffmpeg.git 03rogerdpack 07master:773eb74babe0: dshow: show device name when outputting buffer overflow log message [03:32] <cone-447> ffmpeg.git 03Vittorio Giovara 07master:d37c96213a2a: lavc: restore copy_block{4,16} functions [03:32] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:03e4c2d8333d: Merge commit 'd37c96213a2a9e1fd8669122d5405f4ce6a99ed8' [04:04] <Zeranoe> llogan: I think you have an admirer http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=1358#p6489 [04:40] <cone-447> ffmpeg.git 03Paul B Mahol 07master:70daeacd6ef8: PAF demuxer and decoder [04:40] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:16ddc58bd7a2: Merge commit '70daeacd6ef8b354dd7d2d77ad393831a5bbf033' [04:45] <cone-447> ffmpeg.git 03Paul B Mahol 07master:a7a5e3850ecd: fate: add PAF audio and video tests [04:45] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:8a8472dd49d7: Merge commit 'a7a5e3850ecd94e726ad2272295b9e6c91841cf8' [04:46] <cone-447> ffmpeg.git 03Vittorio Giovara 07master:792e4c21f212: xbm: use av_frame_free on close [04:46] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:6bd05ed8bb30: Merge commit '792e4c21f212979f0e29bcdf107cb6b4f51645a4' [05:56] <cone-447> ffmpeg.git 03Vittorio Giovara 07master:678082b409ac: X-Bitmap decoder [05:56] <cone-447> ffmpeg.git 03Vittorio Giovara 07master:991362fab49b: fate: add XBM tests [05:57] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:296e63efa56f: Merge commit '678082b409aca711f9cf991df6b0200116489322' [05:57] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:4618084a5cdd: avcodec/xbmdec: merge ptr increase into dereference [05:57] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:1da186676738: avcodec/xbmdec: support X10 format [05:57] <cone-447> ffmpeg.git 03Michael Niedermayer 07master:2b570c9569a2: Merge commit '991362fab49b60d34d89b7b7d6dd00fbec3ce022' [09:23] <ubitux> wm4: interesting [09:24] <ubitux> now it would probably be a good thing to make ass demuxer use the lavf/subtitles api, and make that one support utf16 [09:26] <ubitux> actually, make it use ff_get_line() [09:27] <ubitux> mmh well, with AVBPrint you have a virtually unlimited buffer though [09:30] <ubitux> wm4: anyway, we should probably have a ff_subtitles_read_{line,chunk}(), each of them taking that FFTextReader context [09:31] <ubitux> most subtitles demuxers are already using ff_subtitles_read_chunk() and ff_get_line() [09:33] <ubitux> that might require a bit more work thought, but would homogenize them [09:34] <ubitux> ah, and you'll have ff_smil_extract_next_chunk as well (for sami & shit) [09:41] <ubitux> that's pretty cool actually now that you have this "r8" read [11:17] <cone-453> ffmpeg.git 03Paul B Mahol 07master:fb5cf145b6bc: bmp: add a standalone parser [11:17] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:a696b0917d1d: Merge commit 'fb5cf145b6bcfa4f83af94398e5560c1132cc410' [12:05] <cone-453> ffmpeg.git 03Vittorio Giovara 07master:e8e560f2a20e: fate: add a bmpparser test [12:05] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:e1d10017039d: Merge remote-tracking branch 'qatar/master' [12:58] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:92005c26360a: fate/aliaspix: Use standard test sample [13:55] <J_Darnley> To whoever runs trac: ticket #2823 has a spam comment, see: https://trac.ffmpeg.org/ticket/2823#comment:10 [14:36] <j-b> good morning! [17:40] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:2b4543ff6968: cmdutils/filter_codec_opts: do not discard all options for CODEC_ID_NONE [17:40] <cone-453> ffmpeg.git 03Michael Niedermayer 07master:de96e20be959: avfilter/lavfutils: call av_find_stream_info() before reading various information about the stream [17:42] <ubitux> wm4: going to submit the patch? [17:42] <wm4> ubitux: probably [17:42] <wm4> just wondering what I should submit [17:42] <ubitux> your patch [17:42] <ubitux> obviously [17:42] <ubitux> :) [17:54] <wm4> ubitux: any modifications I should make before submitting? [17:54] <ubitux> well if you're motivated, what i suggested a few hours ago [17:55] <ubitux> otherwise if you feel like it's good enough for now then go for it [17:55] <ubitux> a test would be welcome btw [18:56] <jnvsor> Suppose you guys would know more - in filter evals is there any way to get key states? (Ie push to talk by using if() in volume filter with key states?) [18:57] <wm4> no, ffmpeg is not an arcade game emulator [19:00] <jnvsor> Hmm, don't suppose there's any way to do that without separately scripting a keybind or something huh? [19:03] <ubitux> jnvsor: command inject probably [19:03] <ubitux> with ømq [19:04] <ubitux> look at http://ffmpeg.org/ffmpeg-filters.html#zmq_002c-azmq [19:04] <ubitux> if the filter has the command you might be able to interface with it through that [19:04] <ubitux> otherwise, directly the api [19:04] <ubitux> for what it's worth [19:20] <jnvsor> ubitux: Sweet! Looks like it will do the job nicely once I get it working [19:22] <wm4> ubitux: int ff_get_line(AVIOContext *s, char *buf, int maxlen) [19:23] <wm4> ubitux: so that uses a static buffer, not AVBPrint [19:23] <wm4> I expect I'd see resistance if I replaced AVBPrint with a static buffer [19:23] <ubitux> yes [19:23] <ubitux> that's why you shoud do the other way around [19:24] <jnvsor> Don't suppose you know anything that reacts to keyup as well as keydown? [20:01] <jnvsor> What about asendcmd? Could that work? [00:00] --- Sat Mar 29 2014

1 0

ffmpeg.log.20140328
by burek 29 Mar '14

29 Mar '14

[00:01] <voip> llogan, actualy i grabbed this from M3U file http://pastebin.com/tMve5s7x [00:01] <jnvsor> llogan: What if I record noise only to an audio file - can I subtract it somehow? [00:13] <llogan> jnvsor: i don't think ffmpeg has that functionality. check the available filters. maybe i'm wrong. [00:18] <llogan> jnvsor: or maybe sox can do it [00:36] <voip> Guys, who know how to play streams like http://pastebin.com/tMve5s7x ? [00:43] <llogan> voip: how do you know it is valid or reachable? [00:45] <voip> i can check witch vlc [00:52] <jnvsor> llogan: can sox do it in a stream? [00:54] <klaxa> jnvsor: sox's noise removal is okayish, if you use pipes it can be done in a live-stream [00:54] <klaxa> i used it to improve my microphone's quality for VoIP calls [00:55] <klaxa> there was a delay of some 100-200 ms though [02:13] <mshadle> wondering if I could get some help, I want to normalize all my video files to a standard def bitrate, a lot are much higher, or varied, so basically a script to intelligently transcode, maintain aspect ratio and resize if necessary. so i can shrink the size my video collection uses. probably would use ffmpeg. would pay for this, and probably quite easy for someone adept at the tools. :) [03:08] <mshadle> anyone? :p [03:34] <jbermudes> mshadle: The way I'd approach it would be in your loop to first call ffmpeg with no output to get a listing of the stream data/information. (or mplayer, I forget which one gives more detail). Then use that info about the input to know what options to give to the output [05:14] <wh-hw> hi,all , what does this error mean : Unknown AudioCodec: libmp3lame ? [05:19] <jbermudes> wh-hw: What operating system are you using? [05:25] <wh-hw> ubuntu [05:25] <wh-hw> jbermudes,i using ubuntu 12.04 [05:30] <jbermudes> wh-hw: Do you have the package ubuntu-restricted-extras installed? [05:57] <wh-hw> jbermudes, are you still there [05:57] <wh-hw> jbermudes, i doesn't have it [05:58] <wh-hw> jbermudes, i installing it now [06:10] <jbermudes> wh-hw: Did that fix it? [06:18] <wh-hw> jbermudes, any other thing i can do to fix it ? [06:20] <jbermudes> wh-hw: Since you're on Ubuntu, are you actually using ffmpeg or are you using avconv? [06:49] <relaxed> jbermudes: install libavcodec-extra and try again [06:50] <relaxed> er, I meant wh-hw [07:05] <SirCmpwn> weirdest shit happening here [07:05] <SirCmpwn> https://mediacru.sh/wzeuT8SKhH6L [07:05] <SirCmpwn> view that link in chrome [07:06] <SirCmpwn> that video is encoded like so: https://github.com/MediaCrush/MediaCrush/blob/master/mediacrush/processing/… [07:06] <SirCmpwn> only affects webm, it would seem [07:07] <wh-hw> jbermudes, avconv [07:08] <wh-hw> relaxed, libavcodec-extra installed [07:08] <SirCmpwn> ubuntu (and debian in general) is famous for being shit at packaging ffmpeg [07:08] <SirCmpwn> compile it yourself imo [07:13] <wh-hw> what about this issue : Feed '/tmp/feed1.ffm' stream number does not match registered feed? [07:52] <Jack64> ok so I wrote this script that splits a video in parts and then overlays different images over those parts and then uses concat with copy to merge them again. When I split I search for the key frame and feed ffmpeg with something like 180.161195 which is the key_frame start position according to ffprobe so that the temporary split videos before the overlay are split correctly (so that it works with concat copy) but I'm getting mixed results. With some inpu [08:12] <jbermudes> Jack64: I think your text got cut off at "With some inpu..." [08:16] <Jack64> With some input videos it works with others I get 1 second or 2 seconds of duplicated frames. Any ideas why? [08:22] <Jack64> there's a bunch of ways to do what I wanna do and I may be trying to do it the hard way. I suppose I could create a video with the images and then overlay the video right? [08:25] <jbermudes> I would think so [08:26] <jbermudes> I have no idea how to help you, but I'm wondering if maybe the problem is concat gets confused trying to handle such a specific position and just ends up using that frame until the time marker gets back to the next integer [08:28] <Jack64> well I'm only doing that because I do all this stuff in RAM so splitting and merging is virtually instantaneous [08:29] <Jack64> if using copy [08:29] <Jack64> and the dup frames appear right after the split so it's the split that's doing it [08:30] <Jack64> maybe instead of breaking the input video apart I should make a video with the overlays I want and just overlay that [08:30] <Jack64> it'll probably be easier [09:12] <troulouliou_dev> hi i m trying to sample an hevc mkv file to 5 minutes with -map 0 -c copy but i get that error : [09:12] <troulouliou_dev> Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input [09:19] <Mavrik> troulouliou_dev, that's usually accompanied by a more detailed error [09:19] <Mavrik> fix that. [09:19] <Mavrik> also, HEVC is still rather new and possibly buggy [09:19] <troulouliou_dev> Mavrik, seems so checking atm [11:27] <taku> Hi there [11:29] <taku> Sorry to be rude, but Im looking for some hel concerning HLS muxing and Timed metadata insertion with libav, would there be some experts in the room ? :) [14:13] <RenatoCRON> hello guuys [14:14] <RenatoCRON> i have a ffmpeg running 13h:08s, the input is a mjpeg stream. it configured to segment [14:14] <RenatoCRON> but after sometime [14:14] <RenatoCRON> it doest not die, but stop writting to the disk [14:16] <RenatoCRON> ok, another process. uptime 195h 28 min [14:16] <RenatoCRON> last segment video: Mar 27 23:53 [14:16] <RenatoCRON> current date: Fri Mar 28 13:16:31 UTC 2014 [14:17] <RenatoCRON> -loglevel is warning, and there's nothing on error [14:19] <RenatoCRON> should I do a code for detect this and restart the process or restart the process each N hours ? [15:07] <abique_> Hi, what is the shortest way to get an AVFrame of an image, if I have a source url? Thanks! [15:09] <JEEB> look at the demux/decoding example? [15:09] <JEEB> under doc/examples [15:34] <abique_> JEEB, thanks [17:22] <luc4> Hello! I'm using ffmpeg in my closed source application (build as a dynamic lib). Do I need to add some indication of that in my application? [17:31] <jnvsor> http://ffmpeg.org/legal.html [17:31] <jnvsor> Looks like it has everything [18:35] <jnvsor> Is there a way to rig the volume filter to keypresses and get a sort of push-to-talk filter? [20:58] <kyle__> I'm trying to convert an mkv to an m4v container using ffmpeg. I've tried other tools, but considering all of them including ffmpeg give the same error, I'm assuming they're just wrappers using ffmpeg. [20:58] <kyle__> Application provided invalid, non monotonically increasing dts to muxer in stream 0: -83 >= -83 [20:59] <kyle__> av_interleaved_write_frame(): Invalid argument [20:59] <kyle__> That's the error, although the confusing thing is it plays flawlessly. [20:59] <kyle__> How do I get past this? Even if I have to re-encode the audio? [21:40] <maujhsn> Currently watching a vid that using h264 video codec & pcm_s16le audio codec want to convert to mp4 using same audio...Whats the ffmpeg command? [21:58] <maujhsn> maujhsn Thanks, BUT NO THANKS! [21:58] <maujhsn> ffmpeg! SUCKS! [21:59] <maujhsn> Check that its not ffmpeg that sucks YOU DO! [22:13] <sacarasc> What? [22:14] <jnvsor> Whut was that? XD [22:23] <kyle__> fflogger: here we go http://pastie.org/8976970 [22:38] <jnvsor> http://pastebin.com/xeyz1cfg - why does -b:v 4096k stabalize at ~3500k? [22:40] <kyle__> Humm. I thought the problem was audio decoding (based on the error) so I tried to re-encode the aduio as an mp3, remux it into the mkv, and try the command again, but it gave me the same error [00:00] --- Sat Mar 29 2014

1 0

ffmpeg-devel.log.20140327
by burek 28 Mar '14

28 Mar '14

[01:09] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:0d4a66ee7f48: avformat/aviobuf: ffio_ensure_seekback: only copy the initialized part of the buffer [01:09] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:61b5ef775413: libavformat/aviobuf: keep track of the original buffer-size and restore it after probe/ensure-seekback [01:27] <cone-517> ffmpeg.git 03Vittorio Giovara 07master:e50f5d3cf9ef: Alias PIX image encoder and decoder [01:27] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:12ab07be4472: Merge commit 'e50f5d3cf9ef9a16982a5cb4d8b1916cd963aa5b' [02:05] <cone-517> ffmpeg.git 03Vittorio Giovara 07master:9718c31ef60c: fate: add Alias PIX tests [02:05] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:62094e2fddf5: Merge remote-tracking branch 'qatar/master' [04:07] <cone-517> ffmpeg.git 03Andreas Cadhalpun 07master:cf3bfc970c7a: Fix texinfo error due to wrong @subsubsection [04:07] <cone-517> ffmpeg.git 03Andreas Cadhalpun 07master:d473f2d18aef: Fix spelling errors in texi files: more informations --> more information allows to --> allows one to [05:03] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:d5c9843cd2e7: configure: fix VP7 standalone build [05:03] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:57e939d96380: avcodec/vp7: Fix null pointer dereference in vp7_decode_frame_header() [09:07] <nevcairiel> JEEB: i knew this day would come, someone gave me a AAC file which uses SSR... [11:24] <Compn> nevcairiel : does that mean you have to support it now? :P [11:30] <nevcairiel> nah i just have to tell them to get lost [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:a80a7131d11a: swscale/swscale: fix integer overflow [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:adad1ba5d86b: avcodec/vorbisdec: use the stored previous window type only when the actual previous is not known [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:2c566744c41e: avcodec/vorbis: fix decoding of single element huffman trees [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:314f055c294b: dox/scaler:fix bicubiclin typo [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:b40ab81d1f54: avcodec/x86/mpegvideoenc_template: fix integer overflow [13:03] <cone-106> ffmpeg.git 03James Almer 07release/2.2:1103aec1df38: x86/cpu: check for OS support before enabling AVX2 [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:ef0c503d3781: avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets [13:03] <cone-106> ffmpeg.git 03wm4 07release/2.2:81cfe391135c: vf_pullup: simplify, fix double free error [13:03] <cone-106> ffmpeg.git 03Michael Niedermayer 07release/2.2:194485cfbaac: avfilter/vf_pullup: zero freed memory for saftey [13:03] <cone-106> ffmpeg.git 03Andreas Cadhalpun 07release/2.2:3cd1c8653b51: Fix texinfo error due to wrong @subsubsection [13:03] <cone-106> ffmpeg.git 03Andreas Cadhalpun 07release/2.2:31c21d2f6905: Fix spelling errors in texi files: more informations --> more information allows to --> allows one to [13:18] <BBB> was my email html-formatted? [13:18] <BBB> I hope not [14:23] <ubitux> BBB: seems not [16:53] <superware> is there a way to hint avformat_find_stream_info about a network stream codec? I want to save time while ensuring the right parameters are being detected. [16:57] <Compn> dunno superware . try asking on libav-user list for 3rd party app development / api questions ? or stick around here and #ffmpeg wait for answer :) [16:57] <Compn> sorry i cannot help [17:13] <superware> ok, thanks [18:18] <haasn> I'd be interested in adding BT.2020 support to swscale, possibly so I can transcode BT.709 content to BT.2020 for testing and comparison purposes [18:19] <haasn> I'd also be interested in what kind of improvement we could see from using BT.2020-CL to encode even BT.709 or BT.2020-NCL content; when reducing the bitrate [18:19] <haasn> So for that I'd need correct conversion in swscale as well [18:22] <wm4> "good luck" [18:26] <kierank> 709 to 2020 and vice versa iirc is not as simple as you think [18:26] <kierank> there are various papers on the topic [18:27] <haasn> kierank: I don't follow, why? [18:27] <haasn> 709 to 2020ncl in particular should just be a simple matrix multiplication [18:27] <haasn> 709 or 2020ncl to 2020cl would be more complicated, I haven't quite worked out the details in my head [18:28] <haasn> worst case scenario is go through RGB [18:29] <kierank> it's 2020 to 709 apparently that's nontrivial [18:30] <kierank> linear approximations according to the discussion I see [18:30] <kierank> i don't know why though [18:30] <haasn> I guess it depends on what exactly we're talking about [18:31] <kierank> anyway afaik swscale doesn't do yuv matrix conversions like that [18:31] <kierank> ffmbc ported theirs from avisynth [18:31] <haasn> adjusting between primaries is a mess [18:31] <nikitos_> Hello! I have asked my question in #ffmpeg and libav-users. Does libavcodec supports vaapi encoding (uses hardware accels)? In other places they answered no, ffmpeg doesn`t support it, But when I`m building ffmpeg with libva (vaapi) support it have h264_accel in hardware accels section. What is it? [18:31] <haasn> but the encoding techniques should be equivalent, they all cover the exact same range [18:32] <JEEBsv> nikitos_: it's the _decoder_ [18:32] <JEEBsv> as I said [18:32] <haasn> kierank: how does swscale do Y'CbCr conversions? [18:32] <haasn> say, from R'G'B' to BT.709 Y'CbCr [18:32] <kierank> hardcoded to 601 [18:32] <haasn> err.. [18:32] <kierank> yuv to rgb will work though [18:33] <nikitos_> Oh! I see! thanks. [18:33] <kierank> also you'd need to add a 12-bit pixel format [18:33] <haasn> is the implication of that that encoding from an RGB source (like a .png) into Y'CbCr that it will *always* use BT601 Y'CbCr values? [18:33] <kierank> I believe yes [18:34] <haasn> interesting [18:34] <haasn> x264 uses swscale under the hood right? so if I encode with x264 input.png --colormatrix BT.709 # this will produce a mistagged file? [18:35] <JEEBsv> yes [18:35] <haasn> okay [18:35] <JEEBsv> x264 uses lavf/lavc/swscale for input and colorspace conversion [18:35] <haasn> I guess the bottom line for that is make sure you're using the correct Y'CbCr source and just don't touch it [18:35] <JEEBsv> the only thing x264 does internally is bit depth conversion [18:35] <haasn> kierank: BT.2020 supports 10-bit too; but I guess 12 bit will be needed for full compatibility, yes [18:36] <haasn> 12 bit shouldn't be much harder than 10 bit, no? It's the same calculation, you just replace in 2^(12-8) instead of 2^(10-8) when deriving the constants [18:36] <kierank> won't be difficult but there might need to be some assembly changes [18:37] <haasn> okay [18:37] <kierank> haasn: if you don't mind me asking how come you are interested in 2020 [18:39] <haasn> kierank: I implemented it in mpv and I want more sources/evaluation/testing beyond the clips I generated. I'm interested in seeing if BT.2020-CL can produce a quality/filesize improvement even for BT.709 content like Blu-rays [18:39] <haasn> As for how I got the idea to implement it in the first place? I.. don't actually know [18:39] <haasn> I forgot [18:39] <JEEBsv> he learned about it and then specs were thrown at him [18:39] <haasn> kierank: half of it is being prepared and one-upping every other player ;) [18:40] <JEEBsv> and he still is pure regarding swscale [18:42] <wm4> kierank: he's a color nerd, he can't not do it [19:25] <superware> is there a way to hint avformat_find_stream_info about a network stream codec? I want to save time while ensuring the right parameters are being detected. [19:25] <JEEB> not really, but you can make the amount of data it uses for checking small [19:25] <JEEB> *smaller [19:35] <superware> JEEB: if the default is 5 seconds, why does it always take 5 seconds if it can also identify it in 500ms? [19:36] <wm4> superware: well there are various things which can make it slow [19:37] <wm4> probe buffer size, analyze duration, and overhead when opening a file [19:37] <superware> it's a network stream [19:37] <wm4> e.g. in one case I found out that reading the mp4 headers took so long because it was reading over the whole damn file [19:37] <superware> wow [19:38] <wm4> so IMO you'll have to special-tune it in the worst case, and according to your use case [19:38] <superware> it seemed logical that hinting the codec might reduce the probing time [19:39] <BtbN> well, mp4 is a little bit strange there. The header is at the end [19:39] <superware> "at the end"? of each frame? [19:40] <BtbN> of the file. [19:40] <superware> I'm opening a SDP file describing the codec [19:40] <superware> BtbN: in my case it's a RTP network stream [19:40] <BtbN> a stream can't be mp4 [19:40] <superware> h.264 [19:40] <BtbN> that's not a container [19:41] <superware> hmm [19:43] <superware> it's an IP camera, which container can it be? [19:43] <superware> H264 over RTSP? [19:44] <superware> I mean over RTP [19:50] <wm4> BtbN: for mp4 it should just seek to the end, but in my case it was skipping over the whole file in increments; aynway, that was just an example how opening a file can become extremely slow [19:51] <BtbN> wm4, that's because it is not neccesarily at the end [19:51] <BtbN> it can be litteraly anywhere in the file [19:51] <wm4> BtbN: I understand that, but in this case it had something to do with index creation or something [20:25] <cone-106> ffmpeg.git 03Diego Biurrun 07master:c3a0b3eb64be: arm: build: Maintain decoder objects separate from infrastructure objects [20:25] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:68014c6ed98b: Merge commit 'c3a0b3eb64be441ca897629e8ecd80d5b51fded7' [21:14] <Compn> something something free speech [21:19] <JEEB> something something the type of people who usually play that card :) [21:32] <cone-106> ffmpeg.git 03Aleksi Nurmi 07master:ae17878fb2ab: BRender PIX image decoder [21:32] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:f392949f1ac7: Merge commit 'ae17878fb2ab100264226c84c58f5b95a703312f' [21:43] <cone-106> ffmpeg.git 03Vittorio Giovara 07master:bb36b9aa7ef8: fate: add BRender PIX tests [21:43] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:09ebd87a34b9: Merge remote-tracking branch 'qatar/master' [22:52] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:850642331829: avcodec/brenderpix: remove unused variable [22:52] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:a4f27a3f579b: avcodec/brenderpix: propagate error codes [22:52] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:72bff8da479a: avcodec: Make ff_print_debug_info2() independant of Picture struct [23:40] <cone-106> ffmpeg.git 03Timothy Gu 07master:cb11b9e89e15: dnxhdenc: make get_pixel_8x4_sym accept ptrdiff_t as stride [23:40] <cone-106> ffmpeg.git 03Timothy Gu 07master:9d34dce05ba7: x86: convert DNxHDenc inline asm to yasm [23:43] <wm4> ubitux: what the hell is with this AVBPrint stuff [23:43] <wm4> ubitux: and wouldn't it be easier to use a static buffer [23:59] <ubitux> wm4: that's just like bstr [23:59] <ubitux> string helper, a lot of benefits [00:00] --- Fri Mar 28 2014

1 0

ffmpeg.log.20140327
by burek 28 Mar '14

28 Mar '14

[00:00] <bencc> what's the difference? [00:00] <bencc> I mean cropping the height not the duration [00:01] <bencc> I don't need the bottom of the video [00:02] <bencc> with the following command I can play the video in chrome but not in FF: [00:02] <bencc> ffmpeg.exe -i output.mkv -c:v copy -movflags +faststart -c:a aac -strict experimental -b:a 96k full.mp4 [00:04] <llogan> can you provide a short sample of the output that exhibits the same issue? [00:04] <llogan> you can't crop losslessly (at least with ffmpeg) because the video filters require re-encoding [00:05] <llogan> s/losslessly/while stream copying [00:05] <llogan> you can crop "losslessly" with mjpeg using jpegtran [00:05] <bencc> thanks. I'll search for mjpeg and jpegtran [00:06] <bencc> I'll try to generate a short sample [00:06] <llogan> but that won't work for you sicne you want it to play in the browser, and you'd have to re-encode to get crusty, old mjpeg [00:06] <bencc> ok [00:06] <bencc> this is how I generated the mkv video: [00:07] <bencc> ffmpeg -f pulse -ac 1 -i alsa_output.pci-0000_00_05.0.analog-stereo.monitor -f x11grab -r 30 -s 1280x960 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset:v ultrafast -crf 0 -threads 0 output.mkv [00:10] <llogan> lossless is huge and you want to play that in a browser? [00:12] <bencc> llogan: I want to play mp4 [00:12] <bencc> but remove the bottom without lossing quality if possible [00:13] <llogan> you're stream copying lossless H.264 into MP4. lossless is huge. [00:13] <llogan> and i'm not sure what browsers can handle that [00:13] <bencc> how can I decrease the size and still have good quality? [00:15] <llogan> ffmpeg -i input -c:v libx264 -preset medium -crf 23 -vf crop=iw:ih-10,format=yuv420p -c:a aac -strict experimental -b:a 96k -movflags +faststart output.mp4 [00:15] <llogan> http://ffmpeg.org/ffmpeg-filters.html#crop [00:15] <llogan> https://trac.ffmpeg.org/wiki/x264EncodingGuide [00:15] <llogan> adjust -preset and -crf and crop to your needs [00:15] <bencc> thanks. trying [00:26] <bencc> llogan: with your command it works in FF [00:27] <bencc> and it cropps to a good size [00:27] <bencc> thanks [01:16] <relaxed> llogan: daily [01:18] <llogan> a regular movement. good. [05:33] <jbermudes> I'm recording h264 video from a webcam into an mp4 file with ffmpeg. What would be the best way to watch what is being recorded while the video is being recorded? [05:33] <jbermudes> On Linux, the webcam driver won't seem to let me have another process like mplayer or vlc try to read from the camera while ffmpeg is recording [05:46] <relaxed> jbermudes: if you use a different output container you can have mplayer play it [05:47] <relaxed> I recommend matroska .mkv [06:06] <jbermudes> relaxed: thanks, I'll try that out [06:07] <Felix__> hello all [06:08] <Felix__> is anyone familiar with graphstudionext? [06:12] <Felix__> or dts-hd ma decoding? [08:34] <Aetas> Anyone happen to be around with some FLV/Sorenson format knowledge? [09:37] <termos> I have compiled with libfdk-aac but when trying to encode with AV_CODEC_ID_AAC I get an error saying aac is experimental. How do I set the encoder to use libfdk-aac? [09:38] <bart__> Hello, I'm having touble putting a multiply blend filter over my videos. Every time I add a filter, I get a green overlay. My code, screenshots and output can be found on this stackoverflow question : http://stackoverflow.com/questions/22631761/multiply-blend-mode-using-ffmpe… [09:38] <bart__> Somebody that can help me out? [09:40] <kcm1700> I'm using 'mjpeg' encoder by code. which field in AVCodecContext should be set to improve quality? [10:00] <stonie_> hiho! how many -threads would you prefer for encoding? [10:07] <DrSlony> vague question [10:07] <DrSlony> for vp8/9 the docs say "number of physical cores minus 1" [10:08] <DrSlony> of course on my i7 with 4 physical cores and 8 virtual ones, sticking to that rule means less than half my cpu is used, so i use number of virtual cores minus 1, or without the minus one [10:16] <stonie_> ok ty [12:43] <diroots> hi there [12:46] <diroots> i'm trying to set a metadata on a file,... -vcodec copy -acodec copy -metadata:s:v:0 codec_name="DVCPRO HD 1080i50" on a .mov file. when doing this, I see the codec_name information on the ffmpeg output during processing, but when reading once more the output file, the codec_name meta disappear on the video stream,... is the specific meta application on the video stream (metadata:s:v:0) incompatible with vcodec:copy? [13:17] <troulouliou_dev> hi what is the easy command to recursively parse a directory for mkv files and create a 5 minutes sample of those files with the exact same video and audio codecs ? [13:19] <DrSlony> troulouliou_dev in msdos? [13:22] <troulouliou_dev> DrSlony, ? :) [13:22] <troulouliou_dev> no ubuntu [13:26] <DrSlony> :) [13:27] <troulouliou_dev> DrSlony, i mean what are the ffmpeg utils and param to just sample the first 5 minutes of a files and output it in the same container firmat with same audio and codec properties [13:30] <relaxed> troulouliou_dev: ffmpeg -i input -t 00:05:00 -c copy output [13:30] <troulouliou_dev> relaxed, big thanks [13:31] <troulouliou_dev> relaxed, if input is dts_hd ma i ll still have dts_hd ma in output ? not only the core stream [13:31] <troulouliou_dev> relaxed, it will be a raw copy of the frames i guess [13:31] <relaxed> yes, make that: ffmpeg -i input -t 00:05:00 -map 0 -c copy output [13:32] <troulouliou_dev> relaxed, ok big thanks [13:38] <DrSlony> troulouliou_dev try this: find . -iname "*.mkv" | while read -r f; do ffmpeg -i "$f" -t 00:05:00 -map 0 -c copy "${f%.*}_5m.mkv"; done [13:40] <troulouliou_dev> DrSlony, cool nice one :) [15:10] <rsdrsdrsd> is yasm website down? [15:36] <spaam> rsdrsdrsd: no [15:41] <rsdrsdrsd> spaam: the following file can't be downloaded http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz [15:42] <rsdrsdrsd> website isn't down [15:42] <spaam> its really slooow [15:48] <nikitos> DrSlony Hi! How do you think, is now right time for my yesterday question? [16:51] <superware> is there a way to hint avformat_find_stream_info about a network stream codec? I want to save time while ensuring the right parameters are being detected. [16:58] <termos> when transcoding audio with fdk_aac my audio is just noise. I am using the bitstream filter aac_adtstoasc and encoding to flv. What could be the problem? [17:50] <nikitos> Hello! Are here somebody, who can help me with ffmpeg(liav) and vaapi (libva). I have asked this question in mail-list ffmpeg-libav, and one man answered that ffmpeg don`t have gpu encoding features, but ffmpeg supports vaapi and have it in list of hardware accels. What type of support of vaapi does ffmpeg have? [17:57] <Jack64> nikitos: that should probably be asked in #ffmpeg-devel [17:58] <JEEBsv> hwaccels are decoders that depend on you doing some of the prerequisite stuff [17:58] <JEEBsv> it's not as simple as with "normal" decoders [17:58] <nikitos> Jack64 thanks, I will try ffmpeg-devel [17:59] <JEEBsv> you should see if any other project uses the same hwaccel that you're wanting to use [17:59] <JEEBsv> and of course you would have to actually know about libva [17:59] <JEEBsv> so having documentation around for libva and looking at how another project uses the hwaccel are quite important [18:00] <nikitos> JEEBsv I didn`t find any examples of encoding, only decoding [18:00] <JEEBsv> oh, encoding [18:00] <JEEBsv> does lavc support that? [18:00] <nikitos> I will check lavc [18:00] <JEEBsv> lavc is libavcodec [18:01] <JEEBsv> one of the libraries in FFmpeg or Libav projects [18:01] <JEEBsv> I'm asking if it has support for libva _encoding_ at all? [18:01] <JEEBsv> as far as I can see it only has the scaffolding for _decoding_ [18:02] <JEEBsv> yeah [18:02] <JEEBsv> can't find encoders with vaapi [18:03] <JEEBsv> so I don't know what exactly you're looking for in lavf/lavc for yourself? [18:04] <JEEBsv> I mean you could use lavf/lavc for decoding for your input, and then encode with va-api, and then mux with lavf [18:04] <JEEBsv> but the encoding part would have nothing to do with lavc then [18:04] <JEEBsv> unless you are interested in writing encoder(s) for vaapi into lavc as hwaccels [18:04] <JEEBsv> so yeah, specify what you want to do first [18:10] <nikitos> I want "simple" thing, I want to use integrated in intel hd graphics 4600 h.264 hardware encoder via ffmpeg [18:11] <nikitos> so, I have raw yuv420p frames and I need to encode them to h.264 bitstream [18:11] <JEEBsv> what do you mean with 'via ffmpeg'? [18:11] <JEEBsv> do you mean ffmpeg cli or lavf/lavc libraries? [18:11] <nikitos> via ffmpeg, I mean c/c+ programm that uses liabcodec API [18:11] <nikitos> second libs [18:11] <JEEBsv> ok [18:12] <JEEBsv> do you want the actual _encoding_ be done via libavcodec? [18:12] <JEEBsv> or do you just want the things around it to be done via libavcodec? [18:20] <nikitos> what does mean things around it ? [18:20] <JEEBsv> :| [18:20] <JEEBsv> anyways, if you want to use a lavc hwaccel to encode, you would have to code such yourself [18:20] <JEEBsv> since there is none at the moment [18:21] <JEEBsv> if you think you are up to the task, goto #ffmpeg-devel and hope that hwaccels' design is capable of encoding :P [18:21] <troulouliou_dev> is -c copy still usable in ffmpeg tool ? [18:21] <JEEBsv> yes [18:21] <troulouliou_dev> Failed to set value 'copy' for option 'c' [18:22] <JEEBsv> then you have an old ffmpeg or the last ffmpeg binary from Libav [18:22] <JEEBsv> what project does the copyright mention? [18:22] <JEEBsv> if it's Libav, just use avconv [18:22] <JEEBsv> if it's FFmpeg, then you just update :P [18:22] <troulouliou_dev> JEEB, libav ok ... thanks :) [18:22] <JEEBsv> and libav has its own channel @ #libav [18:23] <troulouliou_dev> JEEB, ill stick to ffmpeg :) just forgot ./ before ffmpeg in script :) [18:24] <JEEBsv> ok [18:28] <nikitos> JEEBsv thank you! [18:33] <nikitos> JEEBsv, may I ask here question about h.264 two pass encoding? [18:33] <JEEBsv> no-one stops you from _asking_ [18:36] <nikitos> When I encodes video second pass via libx264 (h.264) does it converts input data back to raw frames (for example yuv420) or uses h.264 bitsream to generete output? [18:37] <JEEBsv> encoding always encodes raw pictures [18:37] <JEEBsv> an encoder always takes in raw pictures [18:37] <JEEBsv> if it takes in something else, it no longer is an encoder only [18:46] <nikitos> so, than, why 2pass (or 3pass) encoding works better? In my mean Better - better qulity at same bitrate. How does it work? [18:47] <JEEBsv> uhm [18:47] <JEEBsv> let's get something clear [18:48] <JEEBsv> with libx264, CRF and 2pass ABR rate control modes give you the same quality for the same bit rate (except for the rare cases where multipass rate control fails badly) [18:48] <JEEBsv> it all depends on what you want to specify [18:49] <JEEBsv> for a specific file size (ABR), you will want to use 2pass because that way the encoder knows more or less the whole complexity of the whole video, and it can make decisions with rate control accordingly [18:49] <JEEBsv> for a specific rate factor (CRF), which is the closest thing we have to something that is like "constant quality", you use CRF [18:50] <nikitos> ok, maybe idea of 2pass encdoding is not clear for me. So what is primary feature os Npass encoding? [18:50] <JEEBsv> first pass or the Nth pass write logs of the whole clip [18:50] <JEEBsv> and then the 2nd or final pass only encodes reading in that log [18:51] <nikitos> oh, I see [18:51] <nikitos> thank you ver much! [18:51] <JEEBsv> it is useful when what you want is not a certain quality level, but when you have an ABR (average bit rate) that you want to fit [18:51] <JEEBsv> but saying that 2pass ABR is the mode that gives you "better quality at the same bit rate" is complete bollocks [18:52] <JEEBsv> at least with libx264 [18:52] <JEEBsv> of course not all encoders have something like the CRF mode [18:52] <nikitos> :-) I see, thanks, I will learn more about encoding [18:52] <nikitos> thanks [19:25] <superware> is there a way to hint avformat_find_stream_info about a network stream codec? I want to save time while ensuring the right parameters are being detected. [19:36] <voip> Hello guys, i am tryng to copy and convert (PAL->NTSC) stream. But as result i have quality loss and very pixelized video. Where is my mistake ? http://pastebin.com/NdspuB8g [19:37] <llogan> voip: do you want to use the mpeg2video encoder as ffmpeg has selected by default? [19:37] <nikitos> voip it seems you need to specify output bitrate [19:38] <nikitos> Stream #0:0: Video: mpeg2video, yuv420p, 1280x720 [SAR 3:4 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 29.97 tbc as I understand it setes 200kbit as default video bitrate [19:41] <voip> llogan, i dont want use mpeg2, but when i putting h264 (instead of -f mpegts -f h264) as result i have black screen [19:41] <llogan> and why upscale from 720x576 to 1280x720? [19:41] <llogan> you don't want -f h264 [19:41] <voip> nikitos, shuld i manualy increase bitrate ? [19:42] <voip> llogan, just for test . whith 720x480 same result [19:43] <llogan> why not simply stream copy the input to the output instead of re-encode? [19:43] <nikitos> voip yes, try -b 1000k or -v:b 1000k [19:45] <voip> llogan, basicly i need stream copy, i need only change to ntsc resolution [19:45] <voip> nikitos, tnx [19:45] <llogan> why do you need ntsc? [19:45] <nikitos> voip, does it works for you? [19:46] <llogan> nikitos: that will still use mpeg2video [19:46] <nikitos> yes, we just solves pixilized video triuble [19:46] <nikitos> trOuble [19:46] <llogan> the trouble with triubles [19:47] <voip> :))) [19:47] <nikitos> sorry ((( [19:48] <voip> llogan, my boss want see in ntsc resolution :))) [19:48] <voip> for me doest mater :) [19:48] <llogan> your boss probably won't be able to tell the difference between 25 and 30000/1001 (ntsc video) since ffmpeg will simply duplicate frames to compensate [19:52] <llogan> anyway, to possibly get what you think you want: ffmpeg -i input -c:v libx264 -maxrate 1800k -bufsize 3600k -vf scale=1280:-1,fps=ntsc -g 2 -c:a copy -f mpegts output [19:53] <llogan> see http://trac.ffmpeg.org/wiki/EncodingForStreamingSites [19:53] <llogan> it's msotly about justin.tv, twitch, etc, but it should provide useful info for you [19:54] <voip> llogan, problem when we copy 720x576 stream , not ergonmcly shows on hdtvs [19:54] <llogan> that -g is wrong. i keep confusing units. -g 60 [19:54] <voip> llogan, thahnk you [20:18] <dhaval2712> I think I've found a bug in FFMPEG. I updated and now VLC doesn't work at all and neither could I extract audio from it. [20:30] <jnvsor> Can you set an x264 crf value and a maximum bitrate or will the max bitrate override the crf? [20:32] <JEEB> jnvsor, VBV can be used with CRF [20:32] <JEEB> also remember that you need to set both maxrate and bufsize [20:32] <JEEB> basically VBV will limit the CRF [20:32] <JEEB> so that it will not go over the VBV limitations at any point [20:33] <jnvsor> Righty, great [21:24] <jnvsor> How come if I use x264opts to set vbv it works but if I use -bufsize it's corrupted and stuff? [21:46] <jnvsor> Streaming to twitch, getting an error: "Connection failed: Application folder ([install-location]/applications/jnvsor) is missing." [22:02] <FrEaKmAn_> hi all [22:02] <FrEaKmAn_> is it possible with ffmpeg to merge 2 videos? so I put 1 small video on top of other? [22:03] <jnvsor> FrEaKmAn_: Yep, look for the overlay filter [22:03] <jnvsor> syntax is "[below][above]overlay={opts}[output]" [22:04] <FrEaKmAn_> great, thanks [22:29] <voip> Guys, what is mean: frame= 828 fps= 14 q=-1.0 Lsize= 6978kB time=00:00:33.18 bitrate=1722.7kbits/s ? [22:30] <voip> fps = 14 is it output frame rate ? [22:30] <voip> http://pastebin.com/MnmhFhaf [22:31] <jnvsor> Yep [22:31] <Bumble-Bee> current processing fps [22:31] <jnvsor> Right, sorry, I use it mainly for realtime recording where it is the output framerate xD [22:31] <Bumble-Bee> yeah [22:32] <Bumble-Bee> terms are pretty much interchangeable [22:32] <jnvsor> Speaking of - what's the best nosie removal audio filter? I saw something about compand but the example didn't do anything [22:32] <voip> why fps is too slow ? [22:33] <Bumble-Bee> voip you set threads to more than 1 ? [22:33] <voip> i didnt set threads [22:34] <voip> http://pastebin.com/MnmhFhaf [22:34] <jnvsor> Copy codec + slow network? [22:34] <voip> jnvsor, 1000mb network :) [22:35] <Bumble-Bee> just for a laugh [22:35] <Bumble-Bee> set threads to [22:35] <Bumble-Bee> cat /proc/cpuinfo | grep processor | wc -l [22:35] <Bumble-Bee> minus 1 [22:35] <voip> 8 cores [22:35] <Bumble-Bee> 8 cores of 8 threads ? [22:36] <Bumble-Bee> amd or intel ? [22:36] <voip> intel xeon [22:36] <voip> how te set threats for http://pastebin.com/MnmhFhaf ? [22:37] <Bumble-Bee> -threads 8 [22:38] <voip> Bumble-Bee, ffmpeg will use 8 threats ? [22:41] <Aetas> there any way to get an output of a stream's timecodes other than mkvtimestamp_v2 which requires timecodes to be in increasing order? [22:42] <voip> Bumble-Bee, why i need to use 8 for simple stram copy, also i have 0% cpu ussage [22:42] <jnvsor> voip: Try it to a file. If it works the problem's with your network [22:43] <voip> jnvsor, ok [23:22] <voip> Guys, can i do this: ffmpeg -i http://source_A udp://aaa.aaa.aaa.aaa:5000 [23:22] <voip> ffmpeg -i http://source_B udp://aaa.aaa.aaa.aaa:6000 [23:22] <voip> ffmpeg -i http://source_C udp://aaa.aaa.aaa.aaa:7000 [23:22] <voip> with one command ? [23:23] <voip> Or shuld run N instances ffmpegs? [23:28] <llogan> you don't need to set threads with libx264. it will automatically set the correct number for you [23:29] <llogan> oh, you're stream copying. how fast is it with a local file output? [23:29] <voip> llogan, ok [23:30] <voip> but if i want copy N streams form differen sources can i use 1 command ? [23:30] <voip> ex: [23:30] <voip> ffmpeg -i http://source_A udp://aaa.aaa.aaa.aaa:5000 [23:30] <voip> <voip> ffmpeg -i http://source_B udp://aaa.aaa.aaa.aaa:6000 [23:30] <voip> <voip> ffmpeg -i http://source_C udp://aaa.aaa.aaa.aaa:7000 [23:30] <voip> ? [23:31] <llogan> yes. with the -map option [23:32] <voip> ok, thanks [23:32] <voip> ona more question :) [23:33] <llogan> ffmpeg -i input0 -i input1 -i input2 -map 0:v -map 0:a output0 -map 1:v -map 1:a output1 -map 2:v -map 2:a output2 [23:34] <llogan> i don't know if it will be faster or slower than 3 separate processes. you should test. [23:34] <voip> perfect ! [23:34] <voip> :) [23:35] <llogan> you may want to use -re as an input option [23:36] <llogan> or you won't need that because your input is not a local file. i keep forgetting stuff. [23:36] <voip> no i will copy from network [23:36] <llogan> -re should not be used with live network streams [23:38] <llogan> i thought you wanted to re-encode to scale and change fps? [23:38] <voip> thank you, logan [23:51] <voip> llogan, sorry maybe i asking too much questions [23:51] <voip> i have one more question [23:52] <voip> how to copy from stream format like: http://178.167.5.37:7777/udp/234.5.2.1:1234 [23:52] <voip> ? [23:53] <jnvsor> I have a heavy background noise on a microphone - is there an ffmpeg filter I can use to remove it? [23:55] <llogan> voip: i'm not sure [23:56] <llogan> jnvsor: if it is within a certain frequency band you could try filters. lowpass, highpass, equalizer, bandreject, etc [00:00] --- Fri Mar 28 2014

1 0

ffmpeg-devel.log.20140326
by burek 27 Mar '14

27 Mar '14

[01:25] <llogan> kierank: thanks. i closed it. [01:28] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:4090d5baa894: avcodec/h261dec: fix motion vector vissualization [02:25] <Zeranoe> x265 recently dropped support for Windows XP, and since the Windows FFmpeg builds include x265 it broke FFmpeg too. I'm debating reverting x265 and holding back updates till support is added again, but that also effects all other Windows versions. I'm looking for some input on what is more important: cutting edge x265, or support for XP [02:49] <Compn> Zeranoe : well win xp is on .... many millions of computers [02:49] <Compn> and x265 is crap [02:49] <Compn> so i'd say stick with winxp [02:50] <Zeranoe> Compn: Even though XP is on the end of it's life? I thought x265 was the next big thing [02:50] <Compn> just my opinion of course, not speaking for the project(s) at all. [02:50] <Compn> Zeranoe : 'end of life' stuff is just because microsoft wants people to buy upgrades [02:50] <Compn> :P [02:51] <Zeranoe> No more updates related to security will be sent to XP [02:51] <Compn> you could make two binaries [02:51] <Compn> one for winxp, one for win vista+ [02:51] <Zeranoe> Trying to avoid that [02:51] <Compn> well [02:52] <Compn> 1. get many million computers upgraded 2. get microsoft to patch winxp further 3. drop winxp support [02:53] <Zeranoe> 4. revert and hold back x265 updates [02:54] <Compn> yep [03:00] <J_Darnley> Why on Earth does x265 require vista? [03:10] <drv> probably threading primitives, but just a guess [03:11] <drv> https://bitbucket.org/multicoreware/x265/issue/44/threadingh-breaks-mingw-b… [03:31] <J_Darnley> ugh [03:32] <Zeranoe> I'm probably going to revert x265 and hold it back. No need to drop XP support yet if everything else works. [04:24] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:6795dcfa659c: avcodec/hevc: Export picture type [04:24] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:c05065aac0d9: avcodec/h261: move b_stride/b_xy under the if() where they are used [07:09] <Zeranoe> Looks like XP users might be out of luck until x265 can add support back. After reverting FFmpeg complained about libx265 version must be >= 9. Options are now: 1. don't include x265 till it's fixed (no ETA at this time), 2. drop support for Windows XP and include x265 [07:31] <ubitux> xp is not supported anymore by MS, isn't it? [07:31] <Case> it is until 8th of April [07:36] <relaxed> therefore Zeranoe must also support it until then [07:41] <wm4> why does it not support xp? [07:42] <wm4> I mean what api is it using [07:43] <JEEB> it's using the threading stuff that's not on NT5 [07:44] <wm4> like what? [07:44] <JEEB> <muggs> mimicing a condition var on XP is non-trivial [07:45] <JEEB> so condition variables it seems [07:45] <wm4> eh [07:45] <wm4> interesting [07:47] <ubitux> BBB: tests done [07:47] <ubitux> ^test^encode [07:48] <ubitux> ssim summary updated etc [07:49] <ubitux> difference is a lot smaller, but vp9 is still below at high bitrate [07:50] <ubitux> but they're almost the same tbh [08:53] <nevcairiel> suprising how many people come out of the woodwork to complain about lacking XP support in x265, personally I would just say screw them and their ancient OS and move on :p [08:55] <JEEB> ayup [08:56] <JEEB> too bad MCW does have that one guy who seems to be like Lord [08:56] <JEEB> using VC6 and barely updated to VS2008's compiler lately or so >_> [08:56] <nevcairiel> well muggs pretty much said "patch or gtfo" but hey [08:56] <JEEB> yup [08:57] <JEEB> btw nevcairiel -- do you happen to have any BT.2020 test content :P [08:57] <nevcairiel> dont think so [08:57] <JEEB> k [08:58] <nevcairiel> not sure any of those 4k trailers are actually bt.2020 [08:59] <JEEB> nah [08:59] <JEEB> I haven't seen /any/ [08:59] <JEEB> it's this magical pixie dust in the specs but I haven't even seen the equivalent of the SMPTE bars in it yet [09:00] <JEEB> The Japanese are going to have it in the satellite 4K spec, but I bet the first content that's going to be aired on that thing whenever they're gonna start doing it is going to be BT.709 :P [09:00] <nevcairiel> lets hope they flag it at least [09:01] <JEEB> they've been rather dilligent on that side so I would guess it should get flagged correctly [10:38] <fjxx> hi [10:39] <ubitux> :/ [10:40] <mmo> I have a problem using the ffmpeg libraries in my own program that plays video streams from a h264 ip camera. Is there someone that can help me? [11:05] <cone-517> ffmpeg.git 03Carl Eugen Hoyos 07release/2.2:3ab63abbd471: Do not set swscale sizeFactor to -1. [11:29] <BBB> ubitux: what was the link again? [11:29] <BBB> ubitux: that's actually an interesting result [11:29] <ubitux> http://lucy.pkh.me/encodes/ http://lucy.pkh.me/encodeshd/ [11:29] <ubitux> BBB ^ [11:31] <BBB> encodeshd is the new one right? [11:35] <ubitux> BBB: yes [11:35] <ubitux> from the bluray [12:16] <kierank> i am not sure it is a good idea to print PSNR values [12:16] <kierank> mostly because people won't understand you've used ssim optimisation [12:26] <smarter> ubitux, BBB: I saw a commit that recently changed --aq-mode=1 to be less agressive (but also less effective on some clips at least) [12:27] <smarter> ubitux: do you still see under/overshoot? I've been investigating vp9's ratecontrol recently but haven't found any reason for the 2x difference that happen sometimes [12:35] <ubitux> smarter: the hd encode i did are with the exact same settings and version of libvpx [12:35] <ubitux> it was done just to check the diff with the source [12:35] <smarter> oh [12:35] <ubitux> so yeah, same rate issue [12:35] <smarter> interesting indeed [12:36] <ubitux> feel free to grab the y4m [12:36] <ubitux> and do the test :p [12:37] <smarter> etvhd5k.y4m is the original? [12:37] <ubitux> yes [12:42] <BBB> ubitux: ah yes indeed much better (https://people.gnome.org/~rbultje/vp9/etvbr.png) [12:42] <BBB> smarter: yes it still massively overshoots by 2x on this clip [12:42] <smarter> and Daemon404 has a clip where it undershoots by 2x :p [12:43] <BBB> vs. original https://people.gnome.org/~rbultje/vp9/etv_ssim.png [12:43] <smarter> I've found some weird stuff looking through the ratecontrol code but nothing that should make things this bad [12:44] <ubitux> http://code.google.com/p/webm/issues/detail?id=717 not sure if this is somehow related to rate control too [12:44] <ubitux> ...but it's very annoying [12:44] <smarter> I don't think I'm motivated enough to look at vp8 ratecontrol :p [12:45] <ubitux> :D [12:45] <BBB> it's vp9, not vp8 [12:46] <ubitux> BBB: #717 is vp8 [12:46] <BBB> o [13:04] <BBB> I'll do a slight update to the blog post to add this as an addendum then [13:04] <BBB> not today, probably over weekend, useful for people to know [13:05] <BBB> but still doesn't address fundamental flaw of vp9 that we mentioned in the blog post: encoder-is-too-slow [14:32] <cone-517> ffmpeg.git 03Diego Biurrun 07master:d0aabeab2375: x86: h264_qpel: Fix typo in CALL_2X_PIXELS macro invocation [14:32] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:8e8347b89291: Merge commit 'd0aabeab23755ee906440505ad2097c0f1493e80' [14:38] <cone-517> ffmpeg.git 03Diego Biurrun 07master:d3c3c1664a95: dsputil: Move hpel_template #include out of dsputil_template [14:39] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:152c8fd856e5: Merge commit 'd3c3c1664a958923f234283e66fbcbfe69a6927f' [14:51] <cone-517> ffmpeg.git 03Diego Biurrun 07master:e7373585f827: dsputil_template: Move bits that are used templatized into separate file [14:51] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:326206910111: Merge commit 'e7373585f827d4ec05d952daa3877e8decfe3c08' [14:59] <cone-517> ffmpeg.git 03Diego Biurrun 07master:aba70bb5387f: Add missing headers to make template files compile (more) standalone [14:59] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:0371eaebcd1b: Merge commit 'aba70bb5387f12dfa5e6cd8cb861c9c7e668151f' [15:07] <cone-517> ffmpeg.git 03Diego Biurrun 07master:2c01ad8b206d: dsputil_template: Detemplatize the code [15:07] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:e9c6b93bdaa6: Merge commit '2c01ad8b206d326700974438f7193f22be416eb1' [15:17] <ubitux> so, when are we bumping lavf? [15:18] <wm4> libav wants libav 11 to be API compatible to 10, so not yet [15:19] <ubitux> ok [15:19] <ubitux> so we're going to keep having mkv demuxer output ugly ass packets [15:20] <ubitux> anyway, about http://trac.ffmpeg.org/ticket/3207#comment:3 [15:20] <ubitux> what's the meaning for SSA and ASS? [15:20] <ubitux> ASS in mkv, SSA in .ass/.ssa ? [15:20] <ubitux> mmh well no i see timing [15:21] <ubitux> so not mkv related [15:21] <wm4> ? [15:21] <wm4> #3207 is about the header [15:21] <ubitux> ASS being the new SSA ok [15:22] <ubitux> this shit depresses me everytime i look at it [15:23] <j-b> Unknown decoder 'copy' [15:23] <j-b> grr [15:23] <ubitux> ? [15:24] <cone-517> ffmpeg.git 03Diego Biurrun 07master:8011ac911b3f: hpeldsp_template: Detemplatize the code [15:24] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:6967cf3c6c57: Merge commit '8011ac911b3f282b9fb64a0fc15404f8bfc7b7ed' [15:25] <ubitux> i wonder how well our ass demuxer works for both ass and ssa [15:26] <wm4> well your ass demuxer doesn't even accept all "valid" modern ass files [15:27] <ubitux> really? [15:28] <wm4> e.g. you expect that [ScriptInfo] is the first line [15:29] <wm4> also don't take libass as reference, it's wrong too [15:30] <ubitux> :( [15:36] <cone-517> ffmpeg.git 03Diego Biurrun 07master:da5be235250a: dsputil: Move RV40-specific bits into rv40dsp [15:36] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:b4f64c58fc80: Merge commit 'da5be235250a61d6994408b054e3e3acf2e0f90f' [15:41] <cone-517> ffmpeg.git 03Diego Biurrun 07master:92ba965103d3: dsputil: Move draw_edges and clear_block* out of dsputil_template [15:41] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:2d15e0c01dfd: Merge commit '92ba965103d3884609730ba9bf293772dc78a9ef' [15:52] <cone-517> ffmpeg.git 03Diego Biurrun 07master:09d4389de10b: hpeldsp_template: Drop av_unused attribute from *_no_rnd_pixels16_8_c functions [15:52] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:a49bdcdee549: Merge commit '09d4389de10b03ea65a84eaf3d6c4b7a7538ad75' [16:55] <cone-517> ffmpeg.git 03Diego Biurrun 07master:55d7f26e7bcf: hpeldsp_template: Move content to hpeldsp [16:55] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:b7f0d39d2629: Merge commit '55d7f26e7bcf1dfb69ee986aa9fc21c62e0b3ae6' [16:59] <wm4> ubitux: hm libavformat's architecture makes it pretty hard to support what I intended to do for utf16 support [16:59] <ubitux> :p [16:59] <funman> wm4: time to fork it? [17:00] <wm4> maybe I could always assume that UTF16 files have a BOM [17:06] <Plorkyeran_> strictly speaking UTF-16 files always have a BOM (in that if there's no BOM they're UTF-16LE or UTF-16BE) [17:06] <Plorkyeran_> and you're only supposed to use the BE and LE versions if the exact encoding is transmitted in some external fashion [17:07] <wm4> assuming that a BOM always exists would make some things near-trivial [17:08] <Plorkyeran_> in the context of reading a file, either the file format specifies the exact encoding, the file has a BOM, or the file is invalid [17:48] <cone-517> ffmpeg.git 03Diego Biurrun 07master:efc7290eb668: x86: hpeldsp: Keep all rnd_template instantiations in hpeldsp_init [17:48] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:4998a72b497f: Merge remote-tracking branch 'qatar/master' [18:20] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:289b149cecb3: avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets [18:54] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:54e2e9fbc15e: avutil/frame: undeprecate AVFrame.motion_val API [18:54] <cone-517> ffmpeg.git 03wm4 07master:5b0ce5d4e366: vf_pullup: simplify, fix double free error [18:54] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:a44409e692e8: avfilter/vf_pullup: zero freed memory for saftey [18:59] <wm4> michaelni: what kind of safety is that? [19:11] <llogan> how big is NTV in Russia? [19:12] <JEEB> one of the bigger ones [19:12] <iive> does N stand for National? [19:13] <JEEB> I didn't think it stood for anything [19:14] <JEEB> oh, it was "national channel 4" from 1984 to 1991 [19:14] <JEEB> but after that it was without any N [19:17] <llogan> they wanted interview footage but i can't do it now. [19:18] <JEEB> I've only been on the 1st channel [19:18] <JEEB> and almost said the F word in Russian :P [19:18] <JEEB> https://dl.dropboxusercontent.com/u/175558/photos/oldies/snapshot2007090519… [19:18] <JEEB> have a young JEEB [19:21] <llogan> what were they interviewing you about? [19:22] <JEEB> politics and stuff [19:23] <JEEB> I then later went on to do brown-paper-bag work for TV Kultura, live translation of an interview with famous Finnish writers [19:24] <JEEB> (whom I didn't even know by face which is kind of embarrassing) [19:55] <michaelni> wm4, saftey as in zeroing out pointers so any attempts to dereference will be clear and reproduceable and not odd rare crashes [19:56] <wm4> michaelni: this could go just the other way as well, and it might actually _reduce_ the likelihood that a bug is accidentally found, which is bad [19:57] <wm4> and also, everyone reading this code will be very confused [19:57] <wm4> because writing to memory before freeing it does by definition nothing [19:57] <michaelni> ill add a comment explaining it [19:57] <wm4> and if you think this is so effective, why not zero memory in av_free? etc. [19:58] <michaelni> av_freep() does clear the pointer [19:58] <wm4> the pointer, but not the memory, so it's entirely different [19:58] <ubitux> CONFIG_MEMORY_POISONING to the rescue [19:59] <michaelni> wm4 if you want i can replace the memset by clearing the pointers one by one [19:59] <michaelni> its the same but more code [19:59] <wm4> ubitux: yes this is the job of memory debugging tools [20:00] <michaelni> av_freep() alone isnt enough because the double linked ring [20:40] <cone-517> ffmpeg.git 03Ben Avison 07master:15a29c39d9ef: truehd: add hand-scheduled ARM asm version of mlp_filter_channel. [20:40] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:3017239d3a6f: avfilter/vf_pullup: add comment to explain memset(0) [20:40] <cone-517> ffmpeg.git 03Ben Avison 07master:87b128d5ef6a: truehd: add hand-scheduled ARM asm version of mlp_filter_channel. [20:40] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:f38af0143c18: Merge commit '15a29c39d9ef15b0783c04b3228e1c55f6701ee3' [20:48] <cone-517> ffmpeg.git 03Ben Avison 07master:4e5aa080bb8d: truehd: break out part of rematrix_channels into platform-specific callback. [20:48] <cone-517> ffmpeg.git 03Ben Avison 07master:3f4e73afe927: truehd: break out part of rematrix_channels into platform-specific callback. [20:48] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:80e67feda806: Merge commit '4e5aa080bb8d83cb6de1ffbdd7b37ec34bc6b30b' [20:55] <llogan> two license violation tickets closed in two days [20:56] <cone-517> ffmpeg.git 03Ben Avison 07master:483321fe7895: truehd: add hand-scheduled ARM asm version of ff_mlp_rematrix_channel. [20:56] <cone-517> ffmpeg.git 03Ben Avison 07master:89135716fd4c: truehd: add hand-scheduled ARM asm version of ff_mlp_rematrix_channel. [20:56] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:44dc373d4a7e: Merge commit '483321fe789566dcb27b6387c00ea16dd86bc587' [21:05] <cone-517> ffmpeg.git 03Ben Avison 07master:fcf5fc444522: truehd: tune VLC decoding for ARM. [21:05] <cone-517> ffmpeg.git 03Ben Avison 07master:b9eb03416d93: truehd: break out part of output_data into platform-specific callback. [21:05] <cone-517> ffmpeg.git 03Ben Avison 07master:b01a2562ae3f: truehd: break out part of output_data into platform-specific callback. [21:06] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:179cf1483262: Merge commit 'fcf5fc444522d24caa9907225802817ae788f511' [21:06] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:fc64e128f008: Merge commit 'b9eb03416d93a5c4ece27ffef5e6e11c81bec6fa' [21:28] <cone-517> ffmpeg.git 03Ben Avison 07master:3b5946bccef6: truehd: add hand-scheduled ARM asm version of ff_mlp_pack_output. [21:28] <cone-517> ffmpeg.git 03Michael Niedermayer 07master:50b68e323c41: Merge remote-tracking branch 'qatar/master' [00:00] --- Thu Mar 27 2014

1 0

ffmpeg.log.20140326
by burek 27 Mar '14

27 Mar '14

[00:22] <voip> I have 2 versions of ffmpeg, 100% same command i can run on old version but have problem when i tryng to run on new version http://pastebin.com/urqpABup [00:23] <voip> any advice ? [00:27] <sacarasc> -s 720x480 won't do anything with -vcodec copy. [00:34] <newhoa> I am trying to use mencoder, setting audio to lavc. Looks like "-oac lavc -lavcopts acodec=libvorbis" [00:34] <newhoa> Is this incorrect? It returnsoggvorbis_encode_init: init_encoder failed [00:34] <newhoa> Couldn't open codec libvorbis, br=224. [00:40] <voip> sacarasc, can't understand , why 100% same commands gaves differnet results ? [00:40] <sacarasc> What different results? [00:43] <voip> sacarasc, with old version of ffmpeg its works, with new version ffmpeg cant starts. look outputs on http://pastebin.com/urqpABup [00:45] <sacarasc> It looks like it exited cleanly, so maybe it didn't get any info from the net. [00:47] <voip> sacarasc, i tried 100 times version 0.6.5 works, last build NO [00:47] <sacarasc> Add a -v or something. [00:48] <voip> sacarasc, what mean -v , verbose ? [00:48] <sacarasc> Yes. [00:48] <voip> ok [01:02] <newhoa> Is there a way to check the version of a codec ffmpeg uses? [01:16] <newhoa> https://ffmpeg.org/ffmpeg.html says ffmpeg --help encoder=encoder_name [01:16] <newhoa> For encoders, but it seems not to work. [06:00] <mo1991> I am using debian wheezy with drupal 7 and it seems that the ffmpeg version that shipped with wheezy is too old to properly encode mp4 files with h264. what is the best way to get a newer version of ffmpeg for transcoding? [06:12] <rjp421> mo1991, try avconv [06:12] <relaxed> mo1991: you can build a recent version yourself or use a static build http://johnvansickle.com/ffmpeg/ [06:44] <RenatoCRON> what?! why set -timeout make ffmpeg get 1 frame of a MJPEG faster?! [06:45] <RenatoCRON> please, tests for yourself: [06:46] <RenatoCRON> (i'm getting the log.. please wait 200 secs or so) [06:46] <julienb> Hello [06:46] <julienb> I need to install the last version of ffmpeg but do you have a good tutorial for that [06:46] <julienb> on debian [06:47] <RenatoCRON> julienb, I gest the most correct is you compile it from git [06:47] <RenatoCRON> https://github.com/FFmpeg/FFmpeg/blob/master/INSTALL [06:48] <relaxed> julienb: https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [06:49] <julienb> i m a little lost , i need the version with libx264 and libfaac just to encode mp4 videos [06:49] <julienb> what the difference between get it from git and the link relaxed gave me? [06:50] <julienb> i ever checked this link relaxed, but is this the easyer manner to install it? [06:50] <RenatoCRON> http://pastebin.com/raw.php?i=kksX37dF > my log of "ffmpeg -i MJPEG-STREAM -vframes 1 " vs -timeout 10 time comparation [06:51] <relaxed> julienb: the wiki compile guide is recommended [06:52] <julienb> relaxed, the thing i don't understand , is do i need to install all the library before? [06:53] <julienb> there are first librarys then there are other things like yasm, libmp3lame [06:53] <relaxed> you need to read the guide, it will walk you through the whole thing. [06:53] <relaxed> Or you can use my static build: http://johnvansickle.com/ffmpeg/ [06:53] <julienb> yes i installed the static build yesterday but there is not the libfaac encoder [06:55] <relaxed> so? use ffmpeg's native encoder: ffmpeg -i input -c:a aac -strict experimental ... [06:56] <relaxed> Why do you think you need libfaac? [06:56] <julienb> it's the same as the libaac? [06:57] <relaxed> yes, it's an aac encoder [06:57] <julienb> i don't know i just need to encode a lot of mp4 using libx264 and libfaac [06:57] <julienb> and i really don't know what the best if i compile ffmpeg or using the static buildl [06:57] <relaxed> the static build shoulod be fine [06:58] <relaxed> should* [06:58] <julienb> yes? [06:59] <julienb> in the case i want to compile it i have to install all those libs before ? autoconf automake build-essential libass-dev libgpac-dev \ [06:59] <julienb> libsdl1.2-dev libtheora-dev libtool libva-dev libvdpau-dev libvorbis-dev libx11-dev \ [06:59] <julienb> libxext-dev libxfixes-dev pkg-config texi2html zlib1g-dev [06:59] <relaxed> this is all covered in the guide [07:00] <julienb> i read the guide but do i need all those libs to just using libx264 and libfaac [07:00] <julienb> that's i don't understand [07:00] <relaxed> also, if you're going to compile ffmpeg with a nonfree aac encoder, forget libfaac and use libfdk-aac instead [07:01] <julienb> ok but i have to install all the dependencies before? i can't omit some? [07:02] <relaxed> you can [07:03] <julienb> but how i know what dependendices i can omit? [07:04] <relaxed> you don't, so just follow the guide [07:05] <julienb> so i install all? [07:05] <relaxed> it's not going to kill you to have a few extra supported libs installed. [07:07] <julienb> for example tu use it with libx264 and libfdk-aac i can omit libmp3lame libopus libvpx ? [07:07] <relaxed> yes [07:08] <julienb> so if i understand well, the dependencies we install at first are just need to compil ffmeg? [07:08] <julienb> i can remove them after ffmpeg is installed? [07:17] <relaxed> no [07:18] <julienb> i can remove rm -rf ~/ffmpeg_build ~/ffmpeg_sources ? [07:18] <julienb> and let the dependencies installed? [07:19] <relaxed> NO, you need to read more. [07:19] <julienb> i read it all the day but i m confused [07:20] <relaxed> Follwoing the guide and actually doing it will net you some experience and understanding. [07:21] <julienb> ok [07:21] <relaxed> It tells you how to revert all changes, too. So stop with the pointless questions and get to it. [07:22] <julienb> sorry i never did that before [07:23] <relaxed> I'm just saying...you never learn anything unless you do it. [07:23] <julienb> that's 100% right [07:36] <julienb> ok i just install like the guide but ffmpeg is not in /bin [07:36] <julienb> i have only vsyasm x264 yasm ytasm [07:37] <relaxed> did you run "make install"? [07:37] <julienb> yes [07:39] <relaxed> I'm guessing you didn't follow the directions to the T? [07:39] <julienb> to the T ? [07:40] <julienb> what you mean by "to the T" ? [07:41] <relaxed> Why don't you following the directions exactly until you know what you're doing. [07:41] <julienb> i made copy past and all goes very fast in console [07:42] <julienb> ok so now i have to use the reverting changes? [07:43] <relaxed> no, run "make clean;make distclean" in the ffmpeg source directory [07:43] <relaxed> then run its ./configure again followed by the rest of the steps [07:44] <julienb> make: *** No rule to make target `clean'. Stop. [07:44] <relaxed> ignore that and proceed [07:44] <julienb> make: *** No rule to make target `distclean'. Stop. [07:45] <julienb> ok i try configure [07:45] <relaxed> are you in the ffmpeg directory? [07:45] <julienb> i was in the ffmpeg source to do the 2 commands [07:45] <julienb> you told me [07:46] <relaxed> "make clean" does not return "make: *** No rule to make target `clean'. Stop." [07:46] <julienb> make: *** No rule to make target `clean'. Stop. [07:47] <relaxed> run "pwd" [07:47] <julienb> make clean returns me make: *** No rule to make target `clean'. Stop. [07:47] <julienb> pwd returns /root/ffmpeg_sources [07:48] <relaxed> which is not the ffmpeg source dir, it's probably /root/ffmpeg_sources/ffmpeg [07:48] <julienb> ah ok i have to do it in ffmpeg [07:48] <relaxed> Yes, exactly as it states in the guide [07:48] <julienb> now i have other errors [07:49] <julienb> Makefile:2: config.mak: No such file or directory [07:49] <julienb> Makefile:53: /common.mak: No such file or directory [07:49] <julienb> Makefile:93: /libavutil/Makefile: No such file or directory [07:49] <julienb> Makefile:93: /library.mak: No such file or directory [07:49] <julienb> Makefile:95: /doc/Makefile: No such file or directory [07:49] <julienb> Makefile:178: /tests/Makefile: No such file or directory [07:49] <julienb> make: *** No rule to make target `/tests/Makefile'. Stop. [07:49] <relaxed> run the ./configure now [07:49] <julienb> ok with all the dependendies ? [07:49] <relaxed> did you follow all the previous steps? [07:49] <julienb> sure [07:50] <relaxed> then yes [07:51] <julienb> ERROR: libass not found [07:51] <relaxed> run it again and omit --enable-libass [07:52] <julienb> ERROR: libfdk_aac not found [07:52] <julienb> that's very strange [07:53] <julienb> i past the install of libfdk-aac before [07:53] <julienb> why it's not found [07:54] <relaxed> rm -rf ~/ffmpeg_sources ~/ffmpeg_build [07:54] <relaxed> start from the top of the guide and follow each step [07:54] <julienb> ok [07:55] <julienb> i just omit libsdl1.2-dev libva-dev libvdpau-dev libx11-dev libxext-dev libxfixes-dev because it's for server [07:55] <julienb> and i don't use x11 [07:56] <relaxed> ok [08:05] <julienb> relaxed you want me to do line per line or i can copy past the block? [08:06] <relaxed> line by line [08:06] <julienb> ok [08:07] <RenatoCRON> relaxed, do you now how -timeout option react with others? [08:08] <RenatoCRON> I'm haveing trouble with that now! [08:09] <RenatoCRON> relaxed, first case: http://pastebin.com/raw.php?i=V5HXZ0Cn [08:09] <RenatoCRON> a camera that without passing -timeout ,it takes 2min to response. [08:09] <RenatoCRON> passing -timeout 10, it takes 2 secs ! [08:10] <RenatoCRON> and then, another camera, that , with -timeout, it says very fast, 'invalid data/format', and without, after 4mins it saves the pic fine : http://pastebin.com/raw.php?i=DQeQFrJ2 [08:11] <RenatoCRON> I'm going to compile again, because ffmpeg-HEAD-e2742d6 is 2 month's old, but it's very strange, dont ? [08:12] <relaxed> RenatoCRON: you're login info is in the output. You should change that now [08:12] <RenatoCRON> relaxed, isn't mine [08:12] <relaxed> well, do they want you sharing that info? [08:13] <RenatoCRON> relaxed, I will change postogasolina.dyndns.org pass, but the anotherone is public [08:21] <molavy> hi [08:22] <relaxed> RenatoCRON: try ffmpeg -probesize 5M -analyzeduration 5M ... -i input .. [08:22] <molavy> i have problem on compile librtmp [08:22] <molavy> can someone help me": [08:22] <molavy> http://stackoverflow.com/questions/22653241/comple-librtmp-for-android-erro… [08:24] <RenatoCRON> relaxed, well, it takes 1/4 less [08:24] <RenatoCRON> with 1MB [08:24] <relaxed> molavy: do you really need librtmp? doesn't ffmpeg support it natively? [08:24] <RenatoCRON> still, if I press ctrl+c a few seconds afer "decode frame unused 67 bytes" image is save. [08:25] <RenatoCRON> i changed to 0.2M and it takes now 19 seconds (still much longer than it need to be, but is getting better) [08:25] <molavy> relaxed: i post issue about this problem on Vitamio https://github.com/yixia/VitamioBundle/issues/121 [08:26] <RenatoCRON> relaxed, what should be the side effect about setting both '32' ? i'm sure i'm sending urls with mjpeg streams. [08:26] <RenatoCRON> with 32, it takes 1.7 seconds (fine, because is away from me) [08:28] <relaxed> RenatoCRON: man ffmpeg-all|less +/probesize [08:34] <RenatoCRON> relaxed, well, I'll keep it low and see what I get. to my understanding at present, i don't need to "probe the input", because the first multipart have all information to make 1 frame =D [08:34] <RenatoCRON> at least in MJPG steams. on RTSP things maybe change [08:39] <molavy> any idea? [08:40] <julienb> relax it worked ! [08:41] <relaxed> yay! [08:41] <julienb> i saw unzip was not installed so it could not unzip libfdk-aac [08:43] <julienb> relaxed i have a question [08:43] <molavy> no idea? [08:43] <julienb> what about those 2 folders ffmpeg_build and ffmpeg_sources [08:43] <julienb> can i remove them or move them in other folder? [09:39] <julienb> relaxed? [09:48] <relaxed> I would keep both where they are. [09:48] <julienb> yes? [09:49] <julienb> can i move them to a folder /ffmpeg so it's cleaner in the folder? [09:49] <relaxed> especially ffmpeg_build, since it contains your libs [09:52] <relaxed> ffmpeg expects your libs to be in ~/ffmpeg_build/lib, so no [09:52] <julienb> ah ok [09:52] <julienb> can i move the /bin? [09:52] <relaxed> you can repeat the process and use --prefix=/ffmpeg for everything it that's what you want. [09:53] <relaxed> yes, you can move the binary [09:53] <julienb> ok [09:53] <julienb> i tested a mp4 conversion [09:54] <julienb> it worked but i have 2 stranges messages [09:54] <julienb> [libfdk_aac @ 0x1b59fc0] Queue input is backward in time [09:54] <julienb> [mp4 @ 0x1b59140] Non-monotonous DTS in output stream 0:1; previous: 1024, current: 637; changing to 1025. This may result in incorrect timestamps in the output file. [09:54] <relaxed> Don't paste in the channel [09:54] <julienb> sorry [09:56] <relaxed> read https://trac.ffmpeg.org/wiki/x264EncodingGuide if you haven't already [09:57] <julienb> here is the pastbin http://pastebin.com/PYDW66mg [09:58] <relaxed> julienb: -threads 4 goes after the input [09:59] <relaxed> I think that was just an informative message [09:59] <julienb> oh [10:00] <relaxed> add "-movflags faststart" after the input too [10:00] <julienb> relaxed, i just test to put it after the input but i have the same message [10:01] <relaxed> it moves the mp4 index to the beginning of the file for better streaming [10:01] <julienb> yes that's replace the old qt fastart [10:01] <relaxed> right [10:01] <julienb> but those message are strange? [10:01] <julienb> about libfdk [10:01] <relaxed> test the exit status with "echo $?" [10:02] <relaxed> if it returns zero then everything is fine and dandy [10:02] <julienb> in order to put my output file i put echo $? ? [10:03] <relaxed> No, no [10:03] <relaxed> That was not an error message [10:04] <julienb> i test ffmpeg echo $? [10:04] <julienb> ? [10:05] <relaxed> julienb: That message was just informative, it wasn't an error message. [10:05] <julienb> ah ok perfect [10:06] <relaxed> julienb: about "echo $?" --> http://mywiki.wooledge.org/BashGuide/TestsAndConditionals [10:06] <julienb> i have another question relaxed, why in the old version i was using libfaac and now we use libfdk_aac [10:07] <relaxed> because libfdk_aac is the better aac encoder [10:07] <julienb> ah [10:07] <julienb> perfect then [10:08] <julienb> can i remove the bin i don't use like ffserver,vsyasm,x264,ytasm? [10:10] <relaxed> sure [10:10] <julienb> great [10:11] <julienb> and i can move for example /bin/ffmpeg and /bin/ffprobe to /usr/sbin ? [10:12] <julienb> and then delete /bin folder in ~ [10:13] <julienb> i wanted to tell /usr/bin sorry [10:13] <relaxed> I would use /usr/local/bin [10:13] <julienb> perfect [10:13] <julienb> thanks a lot for your help relaxed [10:14] <relaxed> you're welcome [10:17] <julienb> relaxed just a question [10:17] <julienb> if i remove x264 there won't be any pb when i use ffmpeg with libx264 ? [10:17] <julienb> in /bin [10:20] <relaxed> correct, ffmpeg uses it's lib [10:20] <relaxed> its* [10:20] <julienb> so i can remove ffserver, vayasm x264 it won't interfer ? [10:21] <relaxed> yes [10:21] <julienb> perfect [10:24] <bencc> I'm using this script to build on ubuntu https://github.com/stvs/ffmpeg-static/blob/master/build.sh [10:24] <bencc> but when trying to record the desktop it says alsa is missing [10:24] <bencc> http://dpaste.com/1757153/ [10:24] <bencc> how can I add it? [10:27] <relaxed> bencc: install libasound2-dev [10:27] <bencc> relaxed: before running make? [10:27] <relaxed> then run ffmpeg's ./configure to verify alsa was enabled, then rerun that script. [10:29] <bencc> relaxed: do I need to build again? [10:30] <relaxed> hmm, the script had --disable-devices [10:31] <relaxed> see what that does [10:31] <bencc> ? [10:31] <bencc> I don't understand [10:31] <relaxed> that sh script had ffmpeg's ./configure running --disable-devices [10:32] <relaxed> which might disable alsa support [10:32] <bencc> ok [10:32] <bencc> I'll try removing it [10:33] <relaxed> do you need a static ffmpeg? [10:34] <bencc> relaxed: I need to record my desktop under ubuntu [10:34] <bencc> is there a better way than static build? [10:39] <DrSlony> Hello. What is the best way to automatically deshake videos nowadays? Please only reply if you've done this within the last few months, as I tested all known methods/programs more than a year ago :] [10:39] <DrSlony> has deshaking between ffmpeg 1.0.8 and 2.2-rc2 changed? [10:46] <nikitos> Hello! Could somebody help me with libav and vaapi? I have debian wheezy on haswell i7 (4770K) and want to encode h.264 video from raw frames (yuv420) via [10:46] <nikitos> HD graphics 4600 (intel integrated GPU) [11:02] <nikitos> Maybe someone could advice me, where to ask my question? [11:16] <DrSlony> nikitos here, at a differenttime of day [11:58] <bencc> when trying to record the screen under windows I"m with: ffmpeg -f dshow -i video="screen-capture-recorder" output.flv [11:59] <bencc> I'm getting: could not enumerate video device [11:59] <bencc> using recent static windows build [12:11] <nikitos> DrSlony, what time is the best for my question? [12:36] <DrSlony> nikitos id say europe-evening time [12:50] <nikitos> DrSlony thanks [13:35] <towolf> hullo, can i change the options used for the default 'showspectrum' visual filter used for audio? [13:35] <towolf> i found how to construct a filter that reads a file with amovie, but im recording form a sound card. [15:45] <termos> I'm getting the error message more samples than frame size (avcodec_encode_audio2) when encoding AAC audio. I tried creating a FIFO to hold the samples and encode the max number of samples I can. But the buffer fills up, so it only works for about 3 seconds. Any ideas? [15:51] <t4nk536> Hi, I'm currently working on a project which needs video editing. Here for I'm using ffmpeg, which is working pretty well, except for 1 filter. Every video should get an overlay with the Photoshop multiply blend mode. I'm using the ffmpeg blend mode, but unfortunately the video gets a green overlay [15:51] <Mavrik> termos, you're not draining the fifo fast enough then. [15:52] <t4nk536> Here is my script [15:52] <t4nk536> http://pastebin.com/25GjcvMe [15:52] <Mavrik> there will be cases where you'll have to encode several frames worth of audio data to keep up [15:52] <Mavrik> since your frame sizes don't match [15:53] <t4nk536> and my output http://pastebin.com/ESbCVT6g [15:53] <t4nk536> if you are interested, this is my screen output http://stackoverflow.com/questions/22631761/multiply-blend-mode-using-ffmpe… [15:53] <termos> Ahh, so several calls to encode_audio? I guess I need to buffer up the AVPackets from encode_audio as well then? [15:53] <dhaval2712> So I have this command and with it is the output I'm getting. I'm trying to extract a song out of a .cue file from a FLAC album. COuld someone help me? What am I doing wrong? [15:53] <dhaval2712> [NULL @ 0x15a2020] sample/frame number mismatch in adjacent frames - This is the only error I seem to get. [15:54] <dhaval2712> http://paste.fedoraproject.org/88794/95845575/ [15:54] <dhaval2712> klaxa|work: Sorry yeah forgot to post it. [15:55] <dhaval2712> What did I miss? [15:57] <dhaval2712> Hello? [15:59] <dhaval2712> Hello? Am I lagging out? [16:02] <sacarasc> Yes. [16:06] <t4nk536> Hello, can somebody help me out? I'm trying to add an overlay to my video using the blend mode filter. When I use (http://pastebin.com/25GjcvMe) this command, I'm getting a green overlay. (see this screenshots http://stackoverflow.com/questions/22631761/multiply-blend-mode-using-ffmpe…) The ooutput I get is the following: http://pastebin.com/ESbCVT6g [16:07] <Mavrik> termos, you need to read your FIFO as long as you have enough samples for a full frame in it [16:08] <Mavrik> termos, then pass all that to encode and get packets out of it [16:08] <Mavrik> it's... just basic maths [16:08] <Mavrik> if output frame is smaller than input frame then OF COURSE you'll have eto sometimes create 2 frames to keep up [16:26] <termos> okey I think I get it now, just need to try it out. Is this the best practice way of doing it? I tried setting output frame_size to the same as the input, but it's being reset (to 1024) when calling avcodec_open2() [16:27] <Mavrik> termos, if AAC demands certain frame size you can't just change it [16:51] <esrax> i'm trying to mux a webcam stream and an mp3 file to output to mpeg-ts, is there a way i can tell ffmpeg to not sync them? i just want the mp3 to play at regular speed [16:52] <termos> Mavrik: thanks for the help, I can't quite get it working yet but will try some more tomorrow. [17:04] <maujhsn> SASL: added audacity: [PLAIN] maujhsn * [17:08] <maujhsn> Does anybody know this guy? [17:08] <maujhsn> https://launchpad.net/~jon-severinsson/+archive/ffmpeg [17:31] <maujhsn> Excuse me does anybody KNOW JON? [17:31] <klaxa|work> why exactly is that important? [17:34] <maujhsn> Just want to know if the command is sudo apt-get install ffmpeg after the ppa is install? [17:35] <maujhsn> klaxa|work Just want to know if the command is sudo apt-get install ffmpeg after the ppa is install? [17:35] <klaxa|work> i don't use ubuntu, have you tried reading FAQs and help-pages? [17:36] <klaxa|work> this seems elaborate enough, maujhsn: http://askubuntu.com/questions/4983/what-are-ppas-and-how-do-i-use-them [17:38] <maujhsn> klaxa|work The page doesn't specificly give all the information about complete info about his branch of ffmpeg! [17:38] <maujhsn> https://launchpad.net/~jon-severinsson/+archive/ffmpeg [17:39] <klaxa|work> i... maybe #ubuntu is more apropriate, i'm sure people will know how to use his ppa properly [17:41] <maujhsn> klaxa|work I'll just have to wait! I sent him an email! [19:04] <llogan> towolf: https://ffmpeg.org/ffmpeg-filters.html#showspectrum [20:21] <esrax> :/ [20:29] <seriously_random> hello, I have problem with "ffmpeg -i video.mp4 -r 20 -ss 233 -t 4 output%03d.png" in ubuntu 12.04 [20:30] <llogan> seriously_random: what is the problem? are you using ffmepg from FFmpeg? Ubuntu uses a fork instead. [20:30] <seriously_random> "frame= 0 fps= 0 q=0.0 size= -0kB time=10000000000.00 bitrate= -0.0kbit" keeps repeating in console until it is killed [20:30] <seriously_random> sry, thought one line is ok [20:30] <llogan> you can, of course, trim the repeating lines to a sane number [20:30] <seriously_random> I guess I have ubuntu fork then [20:31] <llogan> then you'll either have to get help from the fork or use ffmpeg from FFmpeg [20:31] <llogan> the static builds are easy. just download, extract, and execute [20:33] <llogan> relaxed: how often are your builds...uhh..built? [20:35] <esrax> i'm trying to mux a live stream and an audio file into an mpeg-ts to udp multicast, i find that ffmpeg starts and seems to go through the entire hour long audio file, so i see the time increase all the way, then it goes back to zero and seems to try to play at the rate of the video. watching the stream it's full of audio pauses. how can i just get the audio to play at the video rate? [20:37] <seriously_random> wow, static build is much nicer and doesn't swap like crazy. Why does Ubuntu 12.04 have it so much worse? [20:38] <llogan> ask libav [20:39] <Plorkyeran_> 12.04's version is, unsurprisingly, over two years old [20:39] <Plorkyeran_> and it was somewhat out of date even when it was first released due to shitty release cycles [20:46] <JEEB> nah, 12.04 was not out of date really, but everything after that was :P [20:46] <JEEB> because I think 0.8 was released in the very beginning of 2012 or so [20:47] <JEEB> by now it's unfunny amounts of old of course, and the fact that debian/ubuntu couldn't finish migration for the next release in time for 13.10 meant that 14.04 is the first ubuntu having an actually newer version at all .-. [20:48] <Plorkyeran_> and it's still a version behind due to libav 10 missing the cutoff [20:48] <JEEB> (although even that is not the newest release because of both libav's release process as well as the debian/ubuntu package migration) [20:49] <JEEB> there's like four or five packages left on the migration to the new APIs >_> [21:36] <towolf> llogan, [re: your reply @18:04 UTC] now i figured out that ffplay uses an inline function to visualize audio via ffplay -showmode RDFT, which is not implemented by lavfi showspectrum, and there are no options. [21:46] <bencc> "ffmpeg.exe -i file.mkv -c copy file.ogg" gives me could not write header for output file #0 <incorrect codec parameters ?>: Error number -1 occurred [21:46] <bencc> without the -c copy it works but I'm loosing quality [22:39] <JonOomph> Hi! Does anyone know why the GIF encoder requires the rgb24 pixel format, but does not list that pixel format in it's supported pix_fmts array? [22:54] <bencc> llogan: http://dpaste.com/1758116/ [23:08] <dhaval2712> Hello? So I'm trying to extract audio. http://paste.fedoraproject.org/88794/95845575/ This is the command I'm using. Unfortunately it's extracting the entire file. I understand that -to requires HH:MM:SS format which this isn't but I tried that and it doesn't work. [23:33] <llogan> bencc: did you try a more recent build? [23:35] <llogan> bencc: oh, and PCM (or at least that variant) is not allowed in MP4 [23:36] <bencc> llogan: my build is from few months ago. I'll try a more recent build [23:37] <bencc> I thought ffmpeg will convert from PCM to what MP4 uses [23:37] <bencc> should I se aac for playing mp4 in browsers? [23:37] <llogan> that's generally the norm [23:37] <bencc> ok [23:38] <llogan> -c:a aac -strict experimental -b:a 96k [23:39] <llogan> also add "-movflags +faststart" [23:41] <bencc> thanks. trying [23:41] <bencc> I'm doing some more tests of other files [23:43] <llogan> it wouldn't convert because you told it to stream copy. [23:43] <llogan> (regarding your earlier command) [23:44] <bencc> and mp4 doesn't support this audio codec [23:44] <bencc> ok [23:48] <llogan> i hope you're not encoding the whole file to test since it's 5 hours long [23:51] <bencc> llogan: I got mp4. thanks [23:53] <bencc> cropping h.264 is lossless? [23:58] <llogan> bencc: cropping? as in using the crop video filter? or stream copying and using -ss and -t/-to? [00:00] --- Thu Mar 27 2014

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