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October 2013

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ffmpeg-devel.log.20131010
by burek 11 Oct '13

11 Oct '13

[00:17] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:5ac6b6028f17: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [02:43] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:65c2fe7159f7: avcodec/h264: call flush_change() on init [03:39] <BBB> michaelni: yes fixed [05:26] <mathstuf> hi, im looking at adding support for mid-stream ogg comment changes to libavcodec to get support for printing changes in an mpd stream in mpv [05:26] <mathstuf> i see "av_log(avccontext, AV_LOG_DEBUG, "Ignoring comment header\n");" [05:26] <mathstuf> in vorbisdec.c [05:27] <mathstuf> the comment parsing code is in avformat/oggparsevorbis.c currently; would it make sense to move the comment parsing into vorbisdec.c and call it from avformat or is there a better way? [05:28] <mathstuf> hrm, seems flac's picture parsing would go as well [05:40] <mathstuf> nope...looking at the call pattern from mpv, it looks like decode() needs some way to signal that a metadata packet is incoming and that the header parsing function on the avformat for the codec needs to be called [06:18] <Compn> mathstuf : just want to warn you, many developers have tried to work on ogg, but gave up :) [06:18] <Compn> good luck man! [06:19] <mathstuf> well, i just spent the last week in matroska edition land, so who knows [06:19] <mathstuf> and the code is all there...it "just" needs to be hooked together [10:38] <cone-976> ffmpeg.git 03Luca Barbato 07master:1018a92219a3: jpeg2000: Check block length [10:38] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:e96c80154f7c: Merge commit '1018a92219a38a812cf97761c6b3a5e66a400f4b' [10:40] <durandal_1707> saste: that command you mentioned for interleave does not work [10:41] <durandal_1707> try with different inputs, one testsrc, other smptebars or similar [10:41] <durandal_1707> i can't get one input in left and another in right [10:45] <cone-976> ffmpeg.git 03Luca Barbato 07master:1cac9accbd1f: avi: properly fail if the dv demuxer is missing [10:45] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:c589c4c52d13: Merge commit '1cac9accbd1f9b8596122d0735e37b97a844c514' [11:29] <cone-976> ffmpeg.git 03Luca Barbato 07master:756547ce7f4d: avi: DV in AVI must be considered single stream [11:29] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:dd0429e1de54: Merge commit '756547ce7f4d67a18663503e2157aebed3531703' [11:37] <cone-976> ffmpeg.git 03Luca Barbato 07master:ceec6e792e4b: avi: directly resync on DV in AVI read failure [11:37] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:5fcd0a80a25d: Merge commit 'ceec6e792e4b5baaa23b220f4fd33417631f5288' [11:43] <cone-976> ffmpeg.git 03Martin Storsjö 07master:41a10f3ba149: vp6: Support cropping to AVCodecContext.width/height [11:43] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:70025c07a703: Merge commit '41a10f3ba149a2012de499d0b4ad4955d81f28d5' [11:51] <cone-976> ffmpeg.git 03Martin Storsjö 07master:4d7aeff9a108: flvenc: Don't pretend to support muxing "plain" VP6 [11:51] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:0efbd98e2331: Merge commit '4d7aeff9a1080e64125f3374d993b2c9c4c2751b' [12:04] <cone-976> ffmpeg.git 03Martin Storsjö 07master:be1e1373d267: flvenc: Support muxing VP6A as well [12:04] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:c2555bcbd147: Merge commit 'be1e1373d267bae2af8a62d79eef736736f24565' [12:13] <cone-976> ffmpeg.git 03Martin Storsjö 07master:e6ed8668597c: flvenc: Write proper cropping for VP6 even if there's no extradata [12:13] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:7acb464f0e8c: Merge commit 'e6ed8668597cfea25dfb350a9b4df7fb2efc1d90' [12:21] <cone-976> ffmpeg.git 03Luca Barbato 07master:44690dfa683f: prores: Add a codepath for decoding errors [12:21] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:5c99ca2288f5: Merge commit '44690dfa683f620c77e9f0e8e9bc5682608636b1' [12:29] <cone-976> ffmpeg.git 03Luca Barbato 07master:2df7f7714a12: prores: Error out only on surely incomplete ac_coeffs [12:29] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:3fa4c528a92f: Merge remote-tracking branch 'qatar/master' [13:18] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:0391f4617f1b: avcodec/vmnc/reset_buffers(): reset cur_hx/hy too for saftey [13:18] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:ef6b85e9b825: avcodec/proresdec2: propagate error codes [13:36] <cone-976> ffmpeg.git 03Ronald S. Bultje 07master:20d78a86064a: libavcodec/x86: Fix emulated_edge_mc SSE code to not contain SSE2 instructions on x86-32. [13:41] <durandal_1707> ubitux: that ts_unit approach does not generally work as it resets pts to 0, which is not nice for audio [13:41] <ubitux> ah? [13:41] <ubitux> why 0 is wrong? [13:42] <durandal_1707> video stream pts may start from !=0, while audio starts from 0 [13:42] <durandal_1707> for example in ts [13:44] <ubitux> ah, i see [13:45] <ubitux> i guess you could have a ts_start [13:45] <ubitux> from the first frame [13:45] <ubitux> to use as a basis [13:45] <ubitux> if you have ts jumps you're doomed though [13:46] <ubitux> but well, not worse than currently [13:47] <durandal_1707> i just stole code from yadif [13:49] <cone-976> ffmpeg.git 03Timothy Gu 07master:7ab8bceabc33: doc/encoder/aac: Remove -cutoff trick [13:51] <ubitux> durandal_1707: did you submit/commit anything? [13:51] <ubitux> ah, just now in w3fdid ok [13:52] <ubitux> dif* [14:36] <lhs> hi [16:04] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:cdc9657a0c7b: doc/codecs: Document field_order [17:12] <cone-976> ffmpeg.git 03Paul B Mahol 07master:d1d875d30138: avfilter/vf_w3fdif: do not reset pts to 0 [18:08] <cone-976> ffmpeg.git 03Michael Niedermayer 07master:a830915b78e2: swscale: Use full resolution chroma for yuv2rgb when the input has full chroma [18:41] <cone-976> ffmpeg.git 03Maxim Poliakovski 07master:4fb8de08e110: atrac3: Remove unused gain compensation tables [18:41] <cone-976> ffmpeg.git 03Maxim Poliakovski 07master:2d52834910a4: atrac3: Better name for imdct window initialization [19:53] <llogan> BBB: i suppose pre SSE is too old for your testing (I am ignorant of the topic and I didn't really follow thread) [20:30] <cone-976> ffmpeg.git 03Paul B Mahol 07master:996c73490c29: avcodec/exr: remove pointless initialization [21:44] <durandal11707> i get following spam: deprecated pixel format used, make sure you did set range correctly [22:13] <durandal_1707> michaelni: pts resets to 0 when input size changes when filtering [22:14] <durandal_1707> this is suboptimal [22:25] <rcombs> 0x1010101 feels like a very black-magic-y constant [22:25] <rcombs> anyone know what it means? (it's in the ff_draw functions) [22:26] <Compn> nsa backdoor [22:26] <rcombs> nah, it's something with color blending [22:26] <rcombs> but fuck if I know why [22:27] Action: rcombs is noticing some funny artifacting around subtitles drawn with ff_draw, and is trying to determine if that's coming from the draw functions or from x264 settings, and can't understand what any of this code's actually doing [22:29] <rcombs> like, blend_pixel appears to loop across the entire draw area [22:32] <durandal_1707> rcombs: its just for rounding so no floats are used.... [22:33] <iive> rcombs: can you point to the code in question? [22:33] <durandal_1707> blend_pixel/line [22:50] <rcombs> I'm looking at http://ffmpeg.org/doxygen/trunk/drawutils_8c_source.html#l00398 [22:56] <durandal_1707> just rewrite it [22:57] <rcombs> I think it's just H.264 artifacts anyway, but that doesn't stop me from being concerned about how confusing and undocumented that code is [22:57] <durandal_1707> with this unmodified code? [22:58] <rcombs> huh? [22:58] <durandal_1707> ff_draw you use is unmodified or? [22:58] <rcombs> oh, yeah [22:59] <durandal_1707> well you can always use lossless codecs to make sure its not that code fault [23:01] <rcombs> yeah, I'll be screwing around with this a bit more [00:00] --- Fri Oct 11 2013

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ffmpeg.log.20131010
by burek 11 Oct '13

11 Oct '13

[00:08] <relaxed> burek: fix my url please: http://johnvansickle.com/ffmpeg/ [00:09] <burek> im on it [00:09] <burek> is it still for 2.6.26+ [00:09] Action: relaxed passes burek a pbr [00:10] <relaxed> 2.6.32+ and 3.x.x [00:10] <burek> should I leave it as is or update it or what? :) [00:10] Action: burek sets mode: +beer relaxed [00:12] <relaxed> change it to 2.6.32+ so people aren't disappointed when it doesn't work with their ancient kernel. [00:13] <burek> :) [00:13] Action: relaxed glares at rhel [00:13] <burek> i forgot to update it even before.. :facepalm: [00:13] Action: relaxed takes his beer back [00:13] <burek> :)) [00:14] <burek> i believe its ok now :) im off to sleep :) [00:14] <burek> o/ :) [00:27] <beastd> relaxed: nit: you typoed version on your download page (verison) [00:31] <relaxed> beastd: thanks! It's probably been like that forever too. [00:32] <llogan> i prefer verison. [00:32] <Klaus_Dieter> hello world [00:34] <Klaus_Dieter> I have a mkv file that contains a tv recording. I would like to cut parts of it to remove commercials. since I do not want to re-encode the whole video, I use mkvmerge to cut the stream at keyframes and ffmpeg to only encode the parts between the keyframes where the cuts are. this works perfectly fine for video. However if I extract the audio with the given time from the original mkv and then mux the audio stream and all video snippets into [00:35] <Klaus_Dieter> at first I thoughtI might have an issue with my math, but then I realized that audio is encoded in frames as well and it might be that this is the cause of the problem. What could I to to keep video and audio in sync without having to reencode the whole movie when cutting it [00:35] <Klaus_Dieter> ? [00:37] <Klaus_Dieter> I tried adding arbitrary offsets for the audio times to compensate for the effect but this does not work well. This is the script I am using: http://paste.debian.net/54662/ [00:39] <relaxed> Klaus_Dieter: so you're cutting and concatenating with mkvmerge? [00:39] <relaxed> oh, why not use mkvmerge for both? [00:41] <Klaus_Dieter> relaxed: what do you mean by "both"? [00:41] <relaxed> then serve the concatenated file to ffmpeg if you need to encode [00:41] <Klaus_Dieter> I cut with mkvmerge at keyframes [00:41] <Klaus_Dieter> but then there are parts where I want to cut somewhere between two keyframes in the source material [00:42] <Klaus_Dieter> so I use ffmpeg to seek to the spot and re-encode only a few frames and that will become a video chunk as well [00:42] <Klaus_Dieter> maybe I should not separate video and audio at all and always process both at the same time. [00:42] <Klaus_Dieter> so whenever I use ffmpeg to re-encode some video frames I also re-encode the audio data as well [00:42] <relaxed> that would probably help [00:45] <Klaus_Dieter> that would probably work with divx movies [00:46] <Klaus_Dieter> but it would not work when I want to process x264 movies since I cannot use ffmpeg to reencode the video data as this has to happen using a specially patched version of x264... [00:47] <relaxed> build ffmpeg aginst the patched libx264? [00:47] <relaxed> against* [00:48] <Klaus_Dieter> hmm could be worth a try [00:48] <Klaus_Dieter> this is getting uglier than I thought... ;) [00:49] <relaxed> I prefer to let the internet remove the commercials for me. [00:50] <relaxed> you use both [ ] and [[ ]] in the script [00:50] <Klaus_Dieter> hmm. this could be a way too but I do not mean tobreak the rules [00:51] <Klaus_Dieter> relaxed: yeah it is still a mess. I am building upon a really old hacky script [00:51] <Klaus_Dieter> in the hope of improving it [01:16] <Zeranoe> Is there any way to speed up Theora encoding, or vpx? I need some chrome/firefox HTML5 formats. They are both so much slower than x264 [01:21] <ezekiel> "any way" is very vague - to potentially guide those more knowledgeable than myself, I'm going to go ahead and guess that you're not willing to buy better hardware to encode faster? [01:21] <ezekiel> and that you're not actually looking for "any way", but for possible command-line options to give up some little bits of quality in favor of faster encoding on whatever your current hardware is? [01:22] <Zeranoe> ezekiel: Command based ways, I'm just using b:v 1M right now, wondering what is fastest [01:26] <Zeranoe> vp8 is supposed to be really fast, but even with -quality reamtime its displaying at like 1fps [01:28] <ezekiel> if I were you, I'd start by testing variations in -crf, -qmin, and -qmax as described here: https://ffmpeg.org/trac/ffmpeg/wiki/vpxEncodingGuide [01:28] <ezekiel> but I've never encoded with vpx for either fun or profit, so... [01:30] <Zeranoe> vpx or theora, I'm not sure which is actually faster [01:32] <Klaus_Dieter> relaxed: for divx it worked to not separate audio and video stream. [01:32] <Klaus_Dieter> relaxed: now I have to build ffmpeg against my custom libx264 [01:57] <Klaus_Dieter> how can I give the divx encoder a hint on how many keyframes to use? [01:58] <Klaus_Dieter> if I encode a short video sequence it will try to put in kjeyframes aver couple of frames [01:58] <Klaus_Dieter> couple being in the range of 5 to 10 [02:05] <sacarasc> Klaus_Dieter: I think -g does it, but I am unsure. [02:07] <Klaus_Dieter> sacarasc: I have found that -g 0 is supposed to let the codec chose but that is garbage. -g 0 leads to a stream that has only iframes [02:07] <Klaus_Dieter> but having been toold that -g might be the right knob I can now use something like -g 300 and have a good starting point [02:07] <Klaus_Dieter> thank you sacarasc [02:08] <Marcin_PL> Hello. Could FFMpeg somehow do a lossless rotate 180 deg. a MJPEG stream? As rotating a JPEG image&? [02:10] <Klaus_Dieter> Marcin_PL: you defenitely can save it as mjpeg, rotate them losslessly using jpegtran and then create a stream again using ffmpeg [02:11] <Marcin_PL> Klaus_Dieter: You mean save as separately JPEGs? [02:13] <Marcin_PL> What i have: Avi container with mjpeg + PCM s16le Wave stream. Rotated due camera montage. :) [02:14] <Marcin_PL> Could You write me a command to do so? Maybe I could use pipelining? [02:14] <relaxed> Marcin_PL: man ffmpeg-bitstream-filters | less +/' mjpeg2jpeg' [02:15] <Marcin_PL> relaxed: Thanks, i really didn't know how to begin ald what to look for. [02:33] <Marcin_PL> Smart thing that frame %d. Thanks again Guys for fast help. [10:23] <Silasrsdrsd1> if following the centos compilation guide if I want to have the binaries in /usr/local/bin which path should I change? [10:24] <Silasrsdrsd1> without messing my whole server [10:31] <allengreen> ffmpeg -g means gop, if I just wanna 1 keyframe n second, and I don't know the fps, How can I achieve my goal? [10:32] <relaxed> allengreen: find out [11:00] <allengreen> of couse I can ffprobe to get tbr and multiply x seconds, then I can get gop. [11:00] <allengreen> It is just not my best choice. [14:05] <ShotokanZH> hi guys [14:05] <ShotokanZH> i've a problem with ffmpeg [14:05] <ShotokanZH> i'm trying to dump an RTSP stream lossless [14:06] <ShotokanZH> i tried that way: [14:06] <ShotokanZH> ffmpeg -i rtsp://admin:admin@192.168.1.108:554/cam/realmonitor?channel=1\&subtype=0\&tcp -an -t 30 -y out.mp4 [14:07] <JEEB> -c copy [14:07] <ShotokanZH> JEEB, should do the trick? [14:07] <JEEB> well as far as I can see you're not setting any output vcodec, so it most definitely is not doing what you'd want to do (copy lossless) [14:08] <ShotokanZH> JEEB, Unrecognized option 'c' [14:08] <ShotokanZH> Failed to set value 'copy' for option 'c' [14:08] <JEEB> ok, so you have a very old ffmpeg [14:09] <ShotokanZH> JEEB, ffmpeg version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2000-2013 the Libav developers [14:09] <ShotokanZH> built on Mar 30 2013 22:20:06 with gcc 4.7.2 [14:09] <ShotokanZH> seems not |: [14:09] <JEEB> yes, that's an old ffmpeg binary from the libav project [14:09] <JEEB> use avconv [14:09] <JEEB> if you want to use the thing that comes with your distro [14:09] <JEEB> otherwise you can also grab static builds of ffmpeg's ffmpeg [14:11] <JEEB> ShotokanZH, and just so you know -- the build date doesn't really mean anything [14:11] <JEEB> your libav sources are from when the 0.8 release was made, which is by now... [14:12] <ShotokanZH> ok :) [14:12] <JEEB> 20 months ago [14:13] <JEEB> anyways, if you are going to be using libav it's a separate project from ffmpeg, and you should be using the avconv command in that case. You can also build or grab a binary of ffmpeg's ffmpeg if you want to use that. Static binaries are available [14:14] <ShotokanZH> trying now :) [14:25] <ShotokanZH> JEEB, does work fine [14:25] <ShotokanZH> thank you dude [14:27] <jure> zap0: so - how do I now force a channel layout for an input file? [14:28] <zap0> jure, it involves small dolls and and some pins. possibly the sacrifice of livestock. [14:28] <jure> :) [14:29] <jure> "ffmpeg -af aformat=channel_layout=quad -i inputquad.wav -ac 2 out.wav" doesn't work [14:30] <zap0> i've not used sytax like that before [14:30] <jure> "Option af (set audio filters) cannot be applied to input file" [14:31] <jure> yet otherwise ffmpeg "guesses" the layout to be 4.0, which is wrong [14:32] <jure> "otherwise" = putting "-af ... quad" after input file [14:35] <jure> oh look, there's -guess_layout_max, let's try that out [14:40] <jure> that only gets ffmpeg to stop guessing the channel layout [16:11] <asding> hi guys I can't figure out how to make a proper conversion from ogv to mpeg. I need to burn a dvd for a dvd player. Any clue? running debian wheezy [16:12] <Mavrik> ffmpeg -i <file> -target pal-dvd out.mpg ? [16:40] <TauNeutrino> hey guys [16:40] <TauNeutrino> how can I download an audio file from youtube= [16:41] <TauNeutrino> man ffmpeg is useless because complicated as fuck and too lazy to read everything [16:42] Action: durandal_1707 too lazy to reply [16:42] <impy> TauNeutrino: thats not what you can use ffmpeg for [16:43] <impy> TauNeutrino: once you downloaded a video you can split off the audio with ffmpeg [16:43] <durandal_1707> you can use ffmpeg to grab audio from youtube [16:43] <impy> if you want to wait the entire duration [16:44] <TauNeutrino> maybe there is some simpler program you can suggest?? [16:45] <impy> maybe i can. [16:45] <TauNeutrino> I used clive too, bit it downloads the whole video [16:45] <TauNeutrino> :) [16:45] <TauNeutrino> sry but I just want to download audio files from youtube and I don't want to know every details of you program like in the man page [16:45] <TauNeutrino> your* [16:45] <durandal_1707> you cant only download audio [16:46] <TauNeutrino> I can [16:46] <durandal_1707> no you can not [16:46] <TauNeutrino> sure I can [16:46] <impy> Then why do you need our help, if you can? [16:46] <TauNeutrino> I want to download it with a program and not via webbrowser [16:46] <durandal_1707> you need to fetch video, and dump it afterwards, but you still download video.... [16:46] <TauNeutrino> like the analog of filsh [16:46] <sacarasc> TauNeutrino: youtube-dl piped into ffmpeg with -vn and -c copy would work well. [16:47] <durandal_1707> sacarasc: ffmpeg can do it on its own [16:47] <sacarasc> It can? [16:47] <durandal_1707> just compile it with libquvi [16:47] <durandal_1707> sacarasc: perhaps you are on wrong channel? [16:47] <sacarasc> No, I just don't keep up with things I don't need. [16:48] <TauNeutrino> with this I can download only the audio file: http://www.listentoyoutube.com/ [16:48] <TauNeutrino> I guess ffmpeg can do a similar thing? [16:49] <durandal_1707> stuff on youtoube have video too [16:50] <TauNeutrino> Isn't there something like : ffmepg -seghosego "youtubelink" ? [16:50] <durandal_1707> yes but you need to have right version [16:50] <durandal_1707> what version do you have? [16:50] <TauNeutrino> version of ffmpeg? [16:51] <TauNeutrino> I guess the newest, just download it via apt [16:51] <sacarasc> Ubuntu doesn't have ffmpeg at all. And Debian is never fully up to date. (I think testing doesn't use ffmpeg, like Ubuntu, too.) [16:52] <TauNeutrino> ah ehm 0.8.6-6 "This program is deprecated" -.- [16:52] <durandal_1707> you need 2.0 compiled with libquvi [16:52] <sacarasc> That's avconv from the libav project, which is a fork of ffmpeg, not ffmpeg itself. [16:53] <sacarasc> You can read that for some background to the forking/Ubuntu. [16:54] <TauNeutrino> -.- [16:54] <TauNeutrino> forget it, I just use this listentoyoutube thing [16:55] <impy> fair enough [16:56] <TauNeutrino> thx anyway :) [17:13] <jure> I had his answer, but I'm too late. [17:14] <durandal_1707> what is answer? [17:24] <jure> other people have suggested it. simplest way: youtube-dl -x <link to yt video> [18:00] <MichaelP> Is there anyway to get a cleaner video then this is doing ? ffmpeg -video_size 1366x768 -framerate 50 -f x11grab -i :0.0 output.mpg [18:13] <durandal_1707> MichaelP: i assume you encode into yuv420p [18:14] <durandal_1707> which can't be clear [18:14] <durandal_1707> or perhaps you are using default encoder parameters that sucks [18:14] <durandal_1707> there are myriad examples on web how to grab with ffmpeg [18:15] <durandal_1707> you could use libx264 encoder and preset fast [18:15] <durandal_1707> you can grab with ffmpeg lossless too... [18:15] <MichaelP> Guess im useing default then.... Here's how it looks... trying to grab my kde+compiz https://www.youtube.com/watch?v=CUFPouHGjWU [18:17] <MichaelP> It's being a bit faster then my desktop is [18:18] <durandal_1707> try add -qscale 0 [18:19] <durandal_1707> ffmpeg -video_size 1366x768 -framerate 50 -f x11grab -c:v libx264 -preset ultrafast -i :0.0 output.mkv [18:23] <MichaelP> Unknown decoder 'libx264' .... libx264-135 ... h264ec and x264 installed [18:25] <MichaelP> http://dpaste.com/1412274/ [18:25] <durandal_1707> encoder [18:25] <durandal_1707> you put it in wrong place [18:26] <MichaelP> i copyed and pasted what you put there [18:26] <durandal_1707> ffmpeg -video_size 1366x768 -framerate 50 -f x11grab -i :0.0 -c:v libx264 -preset ultrafast output.mkv [18:28] <MichaelP> That is perfect !! [18:29] <MichaelP> Thanks [18:30] <MichaelP> is terminal the onlyway you can start and stop it ? [18:30] <durandal_1707> there are guis that calls ffmpeg... [18:31] <MichaelP> I know recordmydesktop has one.. But that package don't want to work in suse [18:48] <paradisal> hello [18:49] <paradisal> I want to have a sample code about using speex decoder [18:50] <paradisal> thanks in advance [19:57] <denysonique> ffmpeg -f alsa -ac 2 -ab 128k -i pulse -async 1 -f x11grab -s 1280x800 -framerate 30 -r 30 -i :0 -vcodec libx264 -preset ultrafast -crf 0 -acodec pcm_s16le -threads auto -y /tmp/video.mkv [19:57] <denysonique> The audio and video are out of sync, the video runs faster than recorded [19:57] <denysonique> ffmpeg version 1.2.3 [20:04] <llogan> denysonique: what player? does it play as expected with ffplay? [20:04] <Guest14461> Hi, I want a file to print "Muxing mode : Packed bitstream" In mediainfo, which option would do that ? [20:04] <denysonique> llogan: mpv [20:05] <denysonique> llogan: same result with ffplay [20:06] <llogan> you mean the video appears to be playing faster than it should? [20:07] <denysonique> llogan: yes [20:07] <denysonique> it even occurs when I just record the video [20:07] <denysonique> avconv records the video fine, but it doesn't record sound with video [20:08] <denysonique> ffmpeg records sound fine but the video is fast... [20:09] <denysonique> ffmpeg -f x11grab -r 30 -s 1280x800 -i :0 -vcodec libx264 -preset ultrafast -crf 0 -threads 0 /tmp/video.mkv [20:11] <denysonique> llogan: https://gist.github.com/denysonique/c30effb4b38b399cbc2b [20:17] <llogan> denysonique: maybe ffmpeg from git head will work for you. [20:17] <llogan> i can't duplicate any fast playing issue [20:20] <Guest14461> Hi, I want a file to print "Muxing mode : Packed bitstream" In mediainfo, which option would do that ? [20:21] <fede> Hi all [20:23] <fede> I wanted to change the volume on an ogg file with: ffmpeg -i file.ogg -af volume 4.000 file2.ogg [20:23] <fede> but the output is a flac file. Any ideas? [20:26] <denysonique> fflogger: try maybe -acodec theora [20:27] <llogan> you probably meant libvorbis [20:27] <denysonique> sorry fflogger: -acodec vorbis [20:27] <llogan> flac is default for ogg container unless you specify otherwise [20:28] <llogan> vorbis (the native, experimental encoder) is shitty. use libvorbis. [20:29] <fede> thanks for the hint. I just tried -acodec copy seems to work. [20:31] <denysonique> llogan: huffyuv seems to improve it, video runs at better pace, audio still out of sync, though [20:32] <llogan> does it work if you try -video_size 320x240? [20:32] <llogan> instead of -s 1280x800 [20:35] <denysonique> llogan: yes, with libx264 it works at a lower res [20:35] <denysonique> i5 not enough for 1280x800? [20:36] <JEEB> it should, how old is your x264/ffmpeg? [20:36] <JEEB> although which model do you have and how much other stuff you're doing at the same time? [20:36] <fede> -acodec libvorbis gives a lot of error msgs: [ogg @ 0xc0dc40] Non-monotonous DTS in output stream 0:0 [20:36] <JEEB> also make sure you're gettin assembly optimizations with libx264 [20:36] <JEEB> *getting [20:37] <denysonique> JEEB: ffmpeg 1.2.3 built today, on Gentoo [20:37] <JEEB> are you by any chance on 32bit? :P [20:37] <JEEB> I remember gentoo defaulting to disable-asm on 32bit [20:37] <denysonique> avconv records the desktop fine, but with no sound. hence I switched back to ffmpeg for ruther testing [20:37] <denysonique> I'm on 64 [20:37] <JEEB> ok [20:38] <denysonique> ok, what is the fastest -vcodec? [20:38] <denysonique> anything lighter than huffyuv? [20:38] <JEEB> rawvideo? which is ExactlyWhatItSaysOnTheTin [20:38] <JEEB> (and yes, this usually gets HDD-limited) [20:39] <JEEB> also if it works with avconv then the problem is probably somewhere else than video? [20:39] <denysonique> hah, HDD can be the bottleneck in this case. I'll give it a try [20:40] <durandal_1707> denysonique: try lossless h264 [20:40] <llogan> he did, IIRC [20:40] <JEEB> he seems to be setting -c:v libx264 and -crf 0 and -preset ultrafast [20:40] <denysonique> [rawvideo @ 0x7fe240000ff0] Pixel format was not specified and cannot be detected [20:40] <denysonique> when trying to play rawvideo [20:41] <JEEB> yes, naturally [20:41] <JEEB> you have to set -pix_fmt before -i when reading it [20:41] <durandal_1707> denysonique: isn't command you use rescaling 640x480 to 1280x800? [20:42] <durandal_1707> and with yasm disabled everything (important) you do will be slow [20:42] <JEEB> yes, I asked him if he had the asm enabled [20:43] <durandal_1707> also -r 30 is wrong [20:43] <durandal_1707> he should use -framerate 30 [20:43] <durandal_1707> burek: you are responsible for broken wiki examples [20:48] <llogan> why him? it's a wiki. [20:48] <tempus_fol> I can't see -framerate referenced on ffmpeg --help, manual or site... where could I find docs on it? [20:49] <durandal_1707> ffmpeg -h demuxer=x11grab [20:49] <llogan> durandal_1707: then the docs examples need to be updated [20:49] <llogan> http://ffmpeg.org/ffmpeg-devices.html#x11grab [20:49] <tempus_fol> *scrib* thank you, noted [20:58] <durandal_1707> well -s and -r should work too, as there is hack in ffmpeg to translate -r to -framerate, but this may not work if -s/-r is on wrong position or whatever... [20:59] <denysonique> ffmpeg -f alsa -ac 2 -ab 128k -i pulse -async 1 -f x11grab -s 1280x800 -framerate 30 -r 30 -i :0 -vcodec huffyuv -crf 0 -acodec pcm_s16le -threads 0 -y /tmp/video.mkv [20:59] <denysonique> huffyuv seems to be much better than rawvideo in my casae [21:02] <llogan> why apply -ab 128k as input option? also, you don't need that for pcm_s16le. you also don't need -framerate 30 and -r 30. are you sure you need -async? [21:02] <denysonique> -ab 128k is a leftover when using libmp3lame [21:05] <denysonique> Downgrading to 1.0.7 before I try out HEAD [21:21] <Guest14461> Hi, I have this file http://pastie.org/8393141 (mediainfo output) and I want to create a smaller (in duration) version of it, I got this one http://pastie.org/8393142, but on my old DVD player can't play the last one (it 'jumps' trough it (00:00:12 -> 00:00:24, etc) with the screen remaining black), any ideas on how to do this ? [21:47] <fede> fflogger, sorry about my unformatted post, actually this is the first time I use IRC. I will check the sites you mentioned and post again. [22:04] <denysonique> here are the flags I have built ffmpeg with: https://gist.github.com/denysonique/2e90252923a73f5b0234 [22:04] <denysonique> maybe there is something that I could do to improve the performance of my x11grab [22:05] <denysonique> flag = present, -flag = disabled [22:12] <llogan> Guest14461: ffmpeg -i input -ss 30 -t 00:05:00 -c copy output [22:37] <denysonique> Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1 >= 1rop=1 [22:37] <denysonique> av_interleaved_write_frame(): Invalid argument [22:37] <denysonique> I get this with avconv-0.8.7 [22:37] <denysonique> command: avconv -f alsa -ac 2 -ab 128k -i pulse -f x11grab -s 1280x800 -framerate 30 -r 30 -i :0 -vcodec libx264 -acodec pcm_s16le -threads 0 -y /tmp/video.mkv [22:38] <JEEBsv> in general if you want support for libav's binaries I recommend #libav [22:40] <denysonique> both avconv and ffmpeg not working... and this is a Thinkpad. [22:41] <JEEBsv> well, basically if it's ffmpeg's ffmpeg then this is the right place, if it's libav's ffmpeg or avconv (you should generally use the latter with libav) then you poke #libav [22:41] <JEEBsv> not that I really care, but some people seem to. And there are some specifics + depending on the age of the binary there might be some specifics (in both ffmpeg and libav) [22:41] <denysonique> already asked [22:42] <denysonique> but ffmpeg provided by libav isn't ffmpeg anymore or it is? [22:42] <JEEBsv> it is ffmpeg, just from before elenril's rewrite of a lot of stuff in it, which has also since been merged into ffmpeg's ffmpeg [22:43] <JEEBsv> so basically libav's ffmpeg is an old version left just hanging around (it was removed after the 0.8 release IIRC) [22:43] <JEEBsv> at least if I understood your question correctly :P [22:43] <denysonique> yep [22:44] <denysonique> or [22:45] <llogan> simply put: libav stuff (including their so-called ffmpeg) go to #libav, FFmpeg stuff go here [22:45] <denysonique> Better asked would be: Is /usr/bin/ffmpeg provided by the package libav still ffmpeg? [22:46] <denysonique> as opposed to the /usr/bin/avconv provided by avconv [22:46] <denysonique> libav* [22:46] <durandal_1707> no [22:46] <denysonique> how different is /usr/bin/ffmpeg from /usr/bin/avconv provided by libav? [22:46] <llogan> that's a question for #libav [22:47] <JEEBsv> libav's ffmpeg is basically the only binary libav had after the forking [22:47] <JEEBsv> then elenril did the great rewrite (the fruits of which we use with current ffmpeg as well), and the result of that rewrite is avconv on that side [22:48] <JEEBsv> for whatever reason the ffmpeg binary was left unupdated until the next release, at which point it was removed [22:48] <denysonique> right, thanks [22:48] <llogan> if you don't know which one you're using, see the first line of the console output and you'll see "...FFmpeg developers" or libav developers (I think) [22:48] <JEEBsv> yes [22:48] <JEEBsv> that marks from which repository it came from [22:49] <denysonique> so why did he decide to do a fork? [22:50] <llogan> because the dish ran away with the spoon [22:50] <JEEBsv> It's a group of people and it's a long story so I won't start a bedtime story here. Let's just say that both sides are doing active development and ffmpeg merges most if not all of the changes done over at libav. [23:25] <denysonique> ok, so it seems that libav 0.8.7 can record my desktop fine [23:26] <denysonique> except for problems when recording the sound and video at the same time [23:26] <denysonique> just to let you know [00:00] --- Fri Oct 11 2013

1 0

ffmpeg-devel.log.20131009
by burek 10 Oct '13

10 Oct '13

[00:07] <llogan> ffmpeg still compiles fine on a p2 [00:54] <cone-824> ffmpeg.git 03Lukasz Marek 07master:60136345e680: lavd/pulse: move common code to separate file [00:54] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:885739f3b4ca: ffserver: strip odd chars from html error messages before sending them back [04:17] <cone-412> ffmpeg.git 03Timothy Gu 07master:df08fd5e1efa: doc/default.css + doc/t2h.init: New styles [04:17] <cone-412> ffmpeg.git 03Lou Logan 07master:f115e83c81bf: MAINTAINERS: add Communication section [09:43] <cone-239> ffmpeg.git 03Carl Eugen Hoyos 07master:f4e85afd99ed: Add pulse_audio_common.h to skipheaders. [09:43] <cone-239> ffmpeg.git 03Carl Eugen Hoyos 07master:3aabfaa37b24: Fix make checkheaders for vp9 header files. [11:18] <cone-239> ffmpeg.git 03Luca Barbato 07master:b2148faca9e9: shorten: Extend fixed_coeffs to properly support pred_order 0 [11:18] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:7f22df3a4945: Merge remote-tracking branch 'qatar/master' [11:43] <ubitux> oh no :( [11:52] <Daemon404> durandal_1707, ping [12:03] <durandal_1707> Daemon404, pong [12:03] <Daemon404> durandal_1707, which environments are having issues with the new strip patch [12:04] <Daemon404> i know OS X warns, but not much else [12:05] <durandal_1707> there is ticket on trac and mail on list [12:05] <durandal_1707> http://forum.doom9.org/showthread.php?t=168963 [12:05] <Daemon404> i feel like my brain cells were lost by reading that post [12:06] <Daemon404> i do not see any mail on -devel [12:06] <Daemon404> and i am not subscribe to the user list [12:07] <durandal_1707> strip: unrecognized option: -wN [12:07] <Daemon404> so yes a warning [12:07] <durandal_1707> Usage: strip [-AnuSXx] [-] [-d filename] [-s filename] [-R filename] [-o output] file [...] [12:07] <Daemon404> when binutils is too old [12:07] <Daemon404> i.e. binutils on bsd is ancient [12:07] <durandal_1707> not here [12:08] <durandal_1707> not my problem [12:08] <Daemon404> i dont know if it's work adding a configure check for binutils version just to get rid of some harmless warning [12:08] <Daemon404> y binutils for 2-3 years ago has -wN at least [12:08] <Daemon404> my* [12:08] <durandal_1707> https://lists.ffmpeg.org/pipermail/ffmpeg-user/2013-October/017849.html [12:09] <Daemon404> oh it errors [12:09] <Daemon404> but continues [12:10] <Daemon404> pengvado, feel free to chime in (Since it's your patch) [12:13] <pengvado> no comment [12:14] <Daemon404> lol ok. [12:14] <pengvado> afaict you've correctly diagnosed the problem [12:14] <pengvado> not that I've tried to reproduce it [12:15] <Daemon404> i guess ill look into 'solutions' [12:45] <superware> I'm getting "[rtp @ 01e708a0] Unable to receive RTP payload type 96 without an SDP file describing it", is there NO WAY to play h264 rtp over udp without a physical SDP file? even directly with the ffmpeg lib? [13:01] <superware> anyone? :| [13:05] <ubitux> h += y; [13:05] <ubitux> - y = 0; [13:05] <ubitux> + y = 0; [13:05] <ubitux> } [13:05] <ubitux> lol [13:05] <ubitux> come on.. [13:06] <superware> heh [13:07] <durandal_1707> ubitux: why you do not like that? [13:07] <ubitux> http://pastie.org/8389053 i will feel so much better after such diff, at last i can breath, what a relief [13:08] <superware> ubitux: do you happen to know whether it's possible to play a h264-rtp stream directly from url/mrl, without an SDP file? [13:08] <ubitux> durandal_1707: because it's kind of stupid [13:08] <ubitux> superware: no [13:09] <durandal_1707> ubitux: i suppose you copy pasted this from fork? [13:09] <ubitux> yet another random k&r patch yes [13:09] <ubitux> i don't get it honestly [13:10] <durandal_1707> but how to show your boss that you are doing something useful? [13:14] <ubitux> } [13:14] <ubitux> - *got_frame = 1; [13:14] <ubitux> + *got_frame = 1; [13:14] <ubitux> if ((ret = av_frame_ref(data, &c->pic)) < 0) [13:14] <ubitux> lol [13:14] <ubitux> sounds like a reliable script [13:16] <ubitux> my bad on that one ^ [13:22] <cone-239> ffmpeg.git 03Clément BSsch 07master:56db86a0b2bf: doc/filters: remove extra spacing in vid.stab section. [13:25] <ubitux> so [13:25] <ubitux> i'm going to fix the examples [13:25] <ubitux> what method should i prefer? [13:25] <ubitux> old school or ref counted? [13:25] <durandal_1707> "fix"? [13:25] <ubitux> yeah, they crash and leak [13:26] <durandal_1707> ah, i was thinking about k&r [13:27] <iive> the funny thing is that in the above y__=0; the extra space actually indented so the equal sign is under the equal sign on the previous line, aka vertical alignment. [13:27] <iive> oh, my bad.... [13:27] <iive> that was the intended change. [13:28] <iive> but yeh... k&r is the second name of diego. [13:29] <durandal_1707> ubitux: why you reconsider old school? [13:29] <ubitux> because that's the default [13:30] <ubitux> ref counted requires an explicit user request [13:30] <ubitux> aka av_dict_set "refcounted_frames" ... [13:30] <ubitux> also, in a lot of situation, you don't need ref counting [13:31] <ubitux> i'm tempted to rename my example submission from a while ago to something like refcounted_decode_video.c [13:36] <ubitux> btw [13:37] <ubitux> in lavfi/src_movie.c (and likely in other demuxing code), we have sth like ret = avcodec_decodec...(); pkt->data += ret; pkt->size -= ret; [13:37] <ubitux> is this necessary for audio only? [13:38] <ubitux> it seems to be done for anything !video actually [13:38] <ubitux> but that source only supports a/v anyway [13:40] <durandal_1707> that is needed for some audio codecs [13:41] <ubitux> is this documented in some doxy? [13:42] <ubitux> ah yeah, in the decode audio [13:43] <ubitux> i wonder if that couldn't be used for dvb subtitles btw [13:43] <ubitux> instead of the ugly nb hack in ffmpeg.c [13:48] <ubitux> so much changes required.. [13:52] <durandal_1707> its needed for shorten, which have ugly design [13:52] <durandal_1707> and some adpcm codecs iirc [13:52] <ubitux> ape maybe? [13:53] <durandal_1707> its because you get several frames in single packet [13:53] <durandal_1707> nope, ape just have extremly big frames [13:54] <durandal_1707> and because of that it used CODEC_CAP_SUBFRAMES [13:57] <iive> ubitux: i thought this is how all decoders are supposed to work, even video ones. [13:59] <ubitux> no [14:00] <ubitux> it seems the ret is "fixed" for video [14:00] <ubitux> like in case the decode video doesn't return the size of the packet [14:00] <ubitux> see libavfilter/src_movie.c [14:05] <iive> in what sense fixed? [14:05] <ubitux> if (!ret || st->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) [14:05] <ubitux> ret = pkt->size; [14:05] <ubitux> pkt->data += ret; [14:05] <ubitux> pkt->size -= ret; [14:06] <ubitux> aka "if video, eat all the packet" [14:06] <iive> but for what purpose?! [14:07] <iive> does ret hold any other meaning? [14:07] <ubitux> i guess making sure it works with decoders who don't return the appropriate size [14:07] <iive> i still can not get any sense out of this... [14:08] <iive> what are these decoders and why they don't return the appropriate size? [14:08] <ubitux> i don't know [14:08] <iive> i can imagine that some packets are stuffed or padded, but the decoder should be able to skip that anyway... [14:09] <ubitux> ee97982408c83766a3dcb235a673761da629cf99 [14:09] <ubitux> https://ffmpeg.org/trac/ffmpeg/ticket/2556 [14:09] <ubitux> oh it's a bug i opened [14:18] <cone-239> ffmpeg.git 03Derek Buitenhuis 07master:9e3946aadaef: build: Require yasm 0.8.0 or higher [14:18] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:4ce82e6fa94c: Merge commit '9e3946aadaef6f7d7ef15c31b5ff1e73d0ba79a1' [14:24] <durandal_1707> saste: what about interleave patch? and mergeplanes? [14:24] <durandal_1707> is anybody writing any filter? [14:36] <cone-239> ffmpeg.git 03Henrik Gramner 07master:ad7d7d4f6a13: x86inc: Remove .rodata kludges [14:36] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:31d0d355608f: Merge remote-tracking branch 'qatar/master' [14:55] <cone-239> ffmpeg.git 03Paul B Mahol 07master:19fda0a1bcb1: avcodec/vmnc: use av_freep() [15:34] <michaelni> BBB, ff_emu_edge_vfix21_sse() contains SSE2 instructions it seems (https://trac.ffmpeg.org/ticket/3017) [15:48] <cone-239> ffmpeg.git 03Paul B Mahol 07master:8c7bc7d3e50f: avcodec/exr: use avpriv_report_missing_feature() [15:56] <durandal_1707> wouldn't pts code in decimate be just like one in fps? [16:40] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:8780f7fbfaac: avfilter/ff_filter_frame_needs_framing: fix handling of AV_NOPTS_VALUE [16:40] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:0f322feffdf2: avfilter/buffersink: try to fix AV_NOPTS_VALUE [20:08] <cone-239> ffmpeg.git 03Carl Eugen Hoyos 07master:3aa576513b31: lavd/xv.c: Include XShm.h before Xvlib.h. [22:14] <llogan> mpeg-2 video long gop. what is the typical gop size? [22:40] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:83fc6c822b06: avformat/wavdec: Dont trust the fact chunk for PCM [00:00] --- Thu Oct 10 2013

1 0

ffmpeg.log.20131009
by burek 10 Oct '13

10 Oct '13

[00:00] <llogan> sprite`: thanks. i'll have to recompile to test since i don't use libfaac [00:00] <llogan> rmelo: can you provide the input file so i can attempt to duplicate the issue? [00:01] <rmelo> its huge [00:01] <rmelo> :| [00:01] <sprite`> this is the one thats all distorted with libfaac: http://stream05.trillhd.com/trill/4636_240_5cwq9v18zj.mp4?start=40 vs the one i just made with aac experimental: stream05.trillhd.com/trill/4636_240_cr4k4esph2.mp4?start=40 [00:02] <rmelo> Is there any trick i can do, such as passing the wole file through ffmpeg (with some encoding)? [00:05] <llogan> rmelo: sorry, i don't understand the question [00:09] <rmelo> maybe converting the file with a different coding will help? [00:10] <llogan> rmelo: i'm don't know why -ss appears to not be working for you. is the input file the only one giving you problems? [00:13] <llogan> sprite`: i'll blame your faac build [00:14] <llogan> consider using libfdk_aac instead since it appears you're compiling ffmpeg anyway [00:15] <llogan> my faac doesn't cause the distortions [00:15] <rmelo> I'll try with another [00:17] <llogan> although ffaac was surprising better than i expected [00:17] <sprite`> yeah it seems better [00:17] <sprite`> thanks for the help [00:17] <sprite`> :) [00:18] <llogan> why 428 wide? [00:18] <sprite`> it was 16:9 i was just trying to preserve the aspect ratio but it throws an error if i do it exact since it says it has to be divisible by 2 ? [00:19] <llogan> maybe i should have asked, "why 240 height?" [00:20] <sprite`> trying to make low quality small bandwidth versions of the videos because a lot of people who use the site are on terrible computers and setups and that's just the first thing i came up with [00:20] <llogan> consider adding "-movflags faststart" so it can begin playback by the viewer before the file is completely downloaded [00:22] <sprite`> ooh oakay [00:22] <sprite`> i've been using MP4Box for that [00:30] <llogan> rmelo: does -ss also get ignored if you re-encode? (remove -vcodec copy -acodec copy) [00:56] <Maverick|MSG> if I'm using: -filter:a "asetpts=PTS-0.25/TB" [00:56] <Maverick|MSG> and I want to copy the audio codec [00:56] <Maverick|MSG> where could I put -c:a copy [01:06] <klaxa> you can't copy the audio stream if you are changing it [01:07] <Maverick|MSG> what codec is used then? [01:07] <klaxa> whichever you specify [01:07] <Maverick|MSG> how do I specify to use the same codec that is already being used? [01:07] <Maverick|MSG> er, the one in use by the source [01:08] <klaxa> you... can't really, you can use a codec that will produce the same format [01:08] <klaxa> i.e. if the audio is aac, use libfdk-aac, if the audio is mp3, use libmp3lame [01:09] <Maverick|MSG> there isn't a way to tell ffmpeg to autodetect it? [01:14] <klaxa> i know of none [01:14] <Maverick|MSG> ok, well what's the highest quality audio codec I could use then? [01:15] <klaxa> highest quality will be lossless audio, in that case every lossless codec should be equivalently good [01:15] <klaxa> otherwise, aac is good i think [01:15] <klaxa> use libfdk-aac for encoding to aac [01:15] <Maverick|MSG> ok, so I could do: -c:a libfdk-aac [01:16] <klaxa> i suppose so [01:16] <Maverick|MSG> would that have to come before or after the filter:a "asetpts=PTS-0.25/TB" [01:16] <Maverick|MSG> or would it not matter? [01:16] <klaxa> might want to specify a bitrate too [01:16] <klaxa> hmm... i don't think that it really matters, it must be after the input and before the output though [01:17] <Maverick|MSG> I think the most common audio input I'll see will be pcm_s16le [01:18] <Maverick|MSG> @ 44100hz, ~1400kb/sec [01:18] <klaxa> will you need the produced file for further processing? [01:18] <Maverick|MSG> nope [01:18] <Maverick|MSG> it is for uploading to youtube; I just want to make sure the audio is as close to the original as possible [01:18] <klaxa> do you care about things being lossless for archiving? (i for one do) [01:19] <Maverick|MSG> not really; in this case it is an encoding from an original [01:19] <klaxa> ah, well... youtube will encode it to a bunch of other codecs, including vorbis and aac [01:19] <klaxa> do you care about upload time and disk space? [01:19] <Maverick|MSG> not overly [01:19] <klaxa> if not, go with flac [01:19] <klaxa> flac will produce lossless output [01:20] <Maverick|MSG> how much bigger is the file be? [01:20] <klaxa> at about half the size of uncompressed pcm most of the time [01:20] <Maverick|MSG> as compared to aac? [01:20] <klaxa> 5-6 times larger maybe? [01:20] <klaxa> depends on the bitrate you choose for aac [01:20] <klaxa> if you go with 160 kbps, flac will probably be about 600-700 kbps [01:20] <klaxa> but flac will be lossless [01:21] <klaxa> aac won't [01:21] <klaxa> if you choose like 256 kbps with aac it should be good enough for youtube i think [01:21] <Maverick|MSG> ok, let me see if that works [01:21] <klaxa> maybe upload a little sample and wait for them to process it completely, then check it out [01:27] <Maverick|MSG> hm, whenever I specify that ffmpeg bails out [01:28] <klaxa> can you pastebin the command with complete output? [01:30] <Maverick|MSG> actually, the complete output is going to be an issue since I'm running it remotely [01:31] <Maverick|MSG> I can pastebin the command though [01:33] <Maverick|MSG> out of time for now, I'll hit this a bit later. thank you for the help klaxa [01:33] <klaxa> kk [01:34] <Maverick|MSG> http://pastebin.com/WXyJYF1b [01:34] <Maverick|MSG> there's the command; bbl [01:35] <Maverick|MSG> adding the -c:a flac causes it to bail [01:37] <dk0r> will someone please tell me what's wrong here? https://gist.github.com/dk0r/6893699 [01:38] <dk0r> I'm receiving the error: "Could find no file with with path 'psi%d.csv.png' and index in the range 0-4 psi%d.csv.png: No such file or directory" [01:39] <dk0r> if I just use psi*.csv.png then all of my pngs get changed to "psi000.csv.png" [01:40] <klaxa> dk0r: needs to be psi%03d.csv.png [01:40] <klaxa> %03d is the posix format string for a zero padded 3 digit value [01:40] <dk0r> klaxa: where is documentation on that %03d syntax? couldn't find anything control-f'ing through the ffmpeg manualk [01:40] <klaxa> it's posix, you should be able to find it in printf's manpage or something [01:41] <dk0r> what's posix? [01:41] <dk0r> I'm still learning linux [01:41] <klaxa> http://en.wikipedia.org/wiki/POSIX [01:41] <dk0r> k. ty [01:54] <Memphizzzzzz> Can anyone please tell me the correct command to convert a mkv video to mp4 so the ps3 van play it without transcoding? [01:56] <llogan> Memphizzzzzz: depends on the contents of the input file [01:58] <Memphizzzzzz> llogan: How do i find out? [02:04] <Memphizzzzzz> llogan: ok i found mkvtoolnix incl mkvinfo. mkvinfo show me a lots of information, what should i do now? [02:06] <klaxa> Memphizzzzzz, pastebin the info [02:06] <Memphizzzzzz> sec [02:08] <Memphizzzzzz> http://pastebin.com/vpd9RyW1 [02:09] <klaxa> try: ffmpeg -i input.mkv -map 0 -c copy output.mp4 [02:09] <Memphizzzzzz> [NULL @ 0xd13be0] Unable to find a suitable output format for 'out.mp4' [02:10] <klaxa> oh... right it has er... chapters [02:10] <klaxa> oh and subtitles [02:10] <klaxa> does mp4 support S_HDMV/PGS for subtitles? [02:11] <Memphizzzzzz> in most cases i dont need subtitles if that helps [02:11] <klaxa> then try: ffmpeg -i input.mkv -map 0:a -map 0:v -c copy output.mp4 [02:11] <klaxa> that will map only audio and video [02:11] <klaxa> without chapters and subtitles [02:11] <Memphizzzzzz> and if wouldnt need chapters as if it plays natively on the ps3 i can fast forward [02:11] <Memphizzzzzz> ah ok, sec [02:12] <Memphizzzzzz> [NULL @ 0xd13dc0] Unable to find a suitable output format for 'output.mp4' [02:12] <klaxa> mmh can you pastebin the complete output of ffmpeg? with command [02:12] <Memphizzzzzz> sure [02:13] <Memphizzzzzz> hmm i cant pipe the output of ffmpeg [02:13] <Memphizzzzzz> do i need a flag? [02:13] <Memphizzzzzz> clear [02:14] <klaxa> i mean the ffmpeg log [02:14] <klaxa> can't you copy paste from your terminal? [02:14] <klaxa> it should be sent to stderr though [02:14] <Memphizzzzzz> the output is bigger than the terminal window and im in a tmux session [02:15] <klaxa> so: ffmpeg <whatever command> 2> error.log [02:16] <Memphizzzzzz> http://pastebin.com/MBWPGfaW [02:17] <klaxa> what was the exact command? [02:17] <Memphizzzzzz> ffmpeg -i The\ Iceman.mkv -map 0:a -map 0:v -c copy output.mp4 [02:19] <klaxa> still tries to sneak in the subtitle stream [02:19] <klaxa> add -map -0:s [02:19] <klaxa> so ffmpeg -i input.mkv -map 0:a -map 0:v -map -0:s -c copy output.mp4 [02:19] <sacarasc> Memphizzzzzz: -sn [02:19] <sacarasc> No subtitles. [02:20] <klaxa> ah or that [02:20] <klaxa> seems easier after all [02:20] <Memphizzzzzz> i tried both, still the same error "suitable output format" [02:21] <Memphizzzzzz> ffmpeg -i The\ Iceman.mkv -map 0:a -map 0:v -map -0:s -c copy output.mp4 [02:21] <Memphizzzzzz> ffmpeg -i The\ Iceman.mkv -map 0:a -map 0:v -sn -c copy output.mp4 [02:22] <klaxa> maybe mp4 doesn't like the audio? :x [02:23] <klaxa> hmm no it should, it's dts apparently [02:23] <klaxa> can you run it with -loglevel debug and pastebin the output? [02:24] <Memphizzzzzz> sure [02:24] <Memphizzzzzz> the sn version? [02:24] <klaxa> doesn't really matter, but yeah [02:24] <klaxa> since my assumption that it tries to put in the subtitle was wrong [02:24] <Memphizzzzzz> http://pastebin.com/wW3jeJuH [02:25] <klaxa> uh... i seriously don't see what is wrong [02:26] <Memphizzzzzz> output.mp4: Invalid argument, whats that supposed to mean? do i have to specify something like -o? [02:27] <klaxa> nope, >Opening an output file: output.mp4. [02:27] <klaxa> it's opening output.mp4 for writing alright [02:28] <klaxa> but the container doesn't seem to like the codecs used [02:28] <klaxa> although it should [02:28] <klaxa> maybe we can try with one audio track first? [02:28] <klaxa> change the -map 0:a to -map 0:a:0 [02:29] <Memphizzzzzz> nope [02:29] <klaxa> i don't even... [02:30] <klaxa> let's try video only then? this is obviously not what you want, but to confirm it's the audio messing up we can try it [02:30] <Memphizzzzzz> ok [02:30] <klaxa> ffmpeg -i input.mkv -map 0:v -an -sn -c copy output.mp4 [02:31] <llogan> Memphizzzzzz: you find out with "ffmpeg -i input" [02:31] <Memphizzzzzz> nope same problem [02:31] <klaxa> eh? mp4 doesn't like h264 anymore? [02:32] <klaxa> sorry, i'm out of ideas :I [02:32] <Memphizzzzzz> Could not find codec parameters for stream 5 (Subtitle: hdmv_pgs_subtitle): unspecified size [02:32] <Memphizzzzzz> this is the thing where it stops isnt it? [02:33] <Memphizzzzzz> or is it just the last stream displayed? [02:33] <Memphizzzzzz> oh yeah it is.. :( [02:34] <Memphizzzzzz> llogan: do you have any idea? [02:36] <llogan> Memphizzzzzz: i'm guessing your build has too much stuff disabled [02:36] <Memphizzzzzz> im on a netgear readynas box and this is the only build i got from apt-get [02:37] <Memphizzzzzz> how can i install one without using apt-get? [02:37] <llogan> you could try to cross-compile one [02:38] <Memphizzzzzz> that sounds complicated :D [02:38] <llogan> or compile on the device [02:39] <Memphizzzzzz> is there a guide on how to do that? [02:41] <llogan> get the source, ./configure --enable-gpl, make [02:41] <llogan> "apt-get build-essential" first i guess [02:42] <Memphizzzzzz> ok its installing :) [02:46] <llogan> or maybe a linux build will work on it. i don't know. [02:46] <Memphizzzzzz> im just clsoning get git repo [02:46] <Memphizzzzzz> how can i see my kernel version? [02:47] <llogan> uname -a [02:47] <Memphizzzzzz> 3.0.87.RNx86_64.2.1 #1 SMP Fri Sep 6 11:31:20 PDT 2013 x86_64 GNU/Linux [02:48] <Memphizzzzzz> Unknown option "--enable-gpl,". [02:50] <Memphizzzzzz> ah seems the , is the problem [02:50] <Memphizzzzzz> llogan: ok looks like i need yasm as well [02:52] <Memphizzzzzz> so if the ./configure --enable-gpl, make does not work i have to run ./configure --enable-gpl and then ./configure make right? [02:52] <klaxa> make is independent from ./configure [02:52] <Memphizzzzzz> ah ok ./configure just finsihed [02:52] <Memphizzzzzz> now im running make [02:53] <fahadash> Hi Y'all [02:53] <Memphizzzzzz> libavutils are part of ffmpeg right? [02:54] <Memphizzzzzz> and libavfilter, i just hope im not compiling something else than ffmpeg [02:54] <Memphizzzzzz> but i guess these are dependencies of ffmpeg, correct? [02:54] <relaxed> yes [02:54] <fahadash> Thank you so much for helping me with my video editing the other day... I added ffmpegdevs in Credits. At least 30 people are going to see it [02:55] <llogan> burek: can you update the link to relaxed's builds for "!dl static"? http://johnvansickle.com/ffmpeg/ [02:56] <llogan> Memphizzzzzz: yasm would be nice to have if you're going to be doing any encoding on it [02:56] <Memphizzzzzz> llogan: haha ok maybe only yasm was missing? :D [02:57] <llogan> you probably have to compile it if the repo does not supply a recent enough version [02:57] <Memphizzzzzz> once this make is done, where will the "executable" be? in the same folder as the source? [02:57] <llogan> http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [02:58] <llogan> i guess i should have given that to you first [02:58] <Memphizzzzzz> well i was able to get yasm through apt-get [03:03] <Memphizzzzzz> wow compiling takes ages with this little atom cpu :D [03:28] <Memphizzzzzz> llogan, klaxa: ok it finsihed compiling, do i need somethign else like this x264 encoder? [03:28] <klaxa> not if you are just muxing files [03:28] <Memphizzzzzz> lol [03:28] <Memphizzzzzz> same error [03:29] <Memphizzzzzz> no wait [03:29] <klaxa> did you run it with ./ffmpeg ? [03:29] <Memphizzzzzz> haha no that was the problem [03:29] <Memphizzzzzz> ok its converting, transcoding, however you call it :) [03:29] <klaxa> muxing :) [03:29] <Memphizzzzzz> can i see a progress bar? [03:29] <klaxa> there should be a counter telling at what time it is in the video [03:30] <Memphizzzzzz> yup but i dont know the lenght :D [03:30] <klaxa> i do though [03:30] <klaxa> | + Duration: 6332.492s (01:45:32.492) [03:30] <klaxa> how fast is it progressing? [03:31] <Memphizzzzzz> wow this is cool, my first compilation :) [03:31] <klaxa> nice [03:31] <Memphizzzzzz> fps=434 [03:31] <klaxa> ah then it's good [03:31] <Memphizzzzzz> thats fast? [03:31] <klaxa> it's probably slowed down by your harddrive :P [03:31] <Memphizzzzzz> well thats a raid0 in the NAS [03:32] <klaxa> if it were somewhere lower like 30 or something it might be doing actual encoding work instead of just muxing [03:32] <Memphizzzzzz> aha [03:32] <klaxa> weird though, i'd think 1.2.2 should be recent enough [03:32] <Memphizzzzzz> so for the future, how do i know the map stuff? [03:33] <klaxa> http://www.ffmpeg.org/ffmpeg.html#Stream-selection [03:33] <klaxa> it's actually rather simple [03:34] <klaxa> -map [-]<input-number>:[<stream-number>|<stream-identifyer>:<stream-number>] [03:34] <klaxa> all numbers start at 0 [03:35] <klaxa> input number is the input you want to map, you can specify multiple inputs by using -i more than once [03:35] <Memphizzzzzz> oh shit i used the command witthout audio :D [03:35] <klaxa> stream number is the... number of the stream in the input file [03:35] <klaxa> lol [03:35] <klaxa> anyway, the number of the stream in the input file can be of any type [03:35] <Memphizzzzzz> ok these i can see by ffmpeg -i file.mkv [03:36] <klaxa> stream identifier is something like "v" "a" and "s" for video, audio and subtitles respectively [03:36] <klaxa> the stream number then again is the number of that stream of that type [03:36] <klaxa> if you use a "-" in front of the input number that stream is not mapped [03:36] <klaxa> actually the format i posted is wrong, let me fix that [03:37] <klaxa> -map [-]<input-number>[<stream-number>|<stream-identifyer>[:<stream-number>]] [03:38] <Memphizzzzzz> http://pastebin.com/BEPfCqCS [03:39] <Memphizzzzzz> so in this case if i want video 0 and autdio 01 and 3, how would the command look like? [03:40] <klaxa> -map 0:0 -map 0:1 -map 0:3 will map Stream #0:0(eng): Video: h264 (High), Stream #0:1(deu): Audio: dts (DTS-HD MA) and Stream #0:3(eng): Audio: dts (DTS-HD MA) [03:40] <Memphizzzzzz> ah ok so i dont have to specify v a or s [03:41] <klaxa> -map 0:v:0 -map 0:a:0 -map 0:a:2 will map the same streams [03:41] <Memphizzzzzz> i see [03:41] <relaxed> man ffmpeg | less +/' -map' [03:42] <Memphizzzzzz> lol now with audio its the same fps :D [03:43] <Memphizzzzzz> well almost 401 [03:43] <klaxa> i always fail to find the relevant parts of the manpages :/ [03:43] <Memphizzzzzz> :D [03:43] <Memphizzzzzz> and for the -c parameter? [03:44] <relaxed> it's the same syntax [03:45] <Memphizzzzzz> thats copy instead of convert right? so if its not h264 i can convert it to it? [03:45] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-copy [03:46] <llogan> Memphizzzzzz: you can encode to H.264 video if you installed x264 and added --enable-libx264 to your ffmpeg configure [03:47] <Memphizzzzzz> aha and what would the command be then? i mean it wont be copy then, simply -c x264? [03:47] <llogan> -c:v libx264 [03:47] <Memphizzzzzz> nice :) [03:47] <llogan> https://trac.ffmpeg.org/wiki/x264EncodingGuide [03:47] <Memphizzzzzz> Well thank you very much guys [03:47] <Memphizzzzzz> looks like i have to recompile :D [03:48] <llogan> https://trac.ffmpeg.org/wiki/AACEncodingGuide [03:48] <llogan> useful too probably [03:48] <Memphizzzzzz> thats audio right? [03:49] <Memphizzzzzz> looks like aac is the default audio format on the ps3 [03:49] <Memphizzzzzz> im going to check now if the new mp4 works as its dts [04:11] <Memphizzzzzz> :( looks like for DTS the PS3 need transcoding [04:14] <agllengreen> I convert input.mp4 to ouput.ts(mpeg2 ts), but the output file is much smaller than the input file, (input: 37M, output: 10M), how to avoid this problem? [04:14] <agllengreen> ffmpeg -i input.mp4 -y -c:a libfdk_aac -vcodec h264 -f mpegts output.ts [04:14] <agllengreen> sound like it is a bitrate problem. [04:16] <sacarasc> Try setting some bitrates. [04:16] <sacarasc> Also, you should use -c:v libx264 rather than -vcodec h264. [04:18] <agllengreen> but if I don't wanna set specific bitrate value, is there a option the make the input, output same bitrate [04:19] <sacarasc> Not really. What is the specs of your input file? [04:20] <agllengreen> input file may varies. [04:21] <sacarasc> If it's always H264 with AAC audio, you could just do ffmpeg -i input.mp4 -c copy output.ts [04:21] <agllengreen> currently, input.mp4 is converted from a rmvb file. [04:22] <agllengreen> no, the input av codec may varies, some of there are not h264 and aac. [04:24] <Maverick|MSG> klaxa, are you still around? [04:24] <klaxa> yes [04:25] <Maverick|MSG> to get back to the issue i was having, using flac I get an error -1 [04:25] <Maverick|MSG> I apparently don't have libfdk_aac as you mentioned [04:26] <Maverick|MSG> but libvo_aacenc works [04:26] <klaxa> well there's two ways to go about this now [04:26] <klaxa> either you recompile ffmpeg with flac support [04:26] <klaxa> or you encode the audio with the flac cli tool itself [04:26] <Maverick|MSG> what's plan c? [04:27] <Maverick|MSG> I'm using the precompiled windows binaries [04:27] <klaxa> uh... upload in pcm [04:27] <klaxa> will take rather long though [04:27] <klaxa> or use a different audio codec [04:27] <klaxa> like... use mp3 with 320 kbps [04:28] <klaxa> should be good enough [04:28] <Maverick|MSG> so libmp3lame? [04:28] <klaxa> yep [04:29] <Maverick|MSG> actually, I think this build does have flac [04:29] <Memphizzzzzz> klaxa: on ./configure of x264 what does --prefix and --bindir mean? [04:29] <Maverick|MSG> Could not write header for output file #0 (incorrect codec parameters ?): Error [04:29] <Maverick|MSG> number -1 occurred [04:29] <Maverick|MSG> lemme pastebin the error [04:29] <klaxa> Memphizzzzzz: it specifies where make will put the files [04:29] <klaxa> default for --prefix is /usr/local/ [04:29] <Memphizzzzzz> if i leave them it puts them where the source is? [04:30] <klaxa> ah you run "make install" to actually "install" the binaries [04:30] <klaxa> where installing actually is putting the files in the right places in the filesystem [04:30] <Memphizzzzzz> aha [04:30] <Memphizzzzzz> and whats -enable-static? [04:31] <Maverick|MSG> http://pastebin.com/rYVne4tH [04:31] <Maverick|MSG> codec not supported in contained? [04:31] <Maverick|MSG> *container [04:31] <klaxa> ah... hmmm... just put it in .mkv i guess? [04:31] <klaxa> youtube knows how to deal with mkv [04:32] <klaxa> Memphizzzzzz: that will enable building static libraries i think [04:32] <klaxa> there's a different between dynamic/shared and static libraries [04:32] <Maverick|MSG> hm, is there another codec that'll give lossless in an mp4 format? [04:33] <klaxa> DEA.LS dts DCA (DTS Coherent Acoustics) (decoders: dca ) (encoders: dca ) [04:33] <klaxa> try dca then [04:33] <klaxa> mp4 should support dts [04:34] <klaxa> apparently ffmpeg can encode it [04:34] <klaxa> so give it a try [04:34] <Maverick|MSG> ffmpeg says it is experimental [04:35] <Maverick|MSG> y'know, just to be easy I'll just go with a high bitrate with mp3 [04:35] <klaxa> probably [04:35] <klaxa> easier [04:35] <klaxa> and i mean i can't hear the difference between 320 kbps mp3 and lossless [04:35] <klaxa> hell, i can't even tell the difference between 192 kbps vorbis and lossless [04:36] <Maverick|MSG> yeah, I figure after youtube manhandles it it'll probably be lower anyway [04:36] <klaxa> i keep everything in lossless if possible though for archiving purposes [04:37] <Maverick|MSG> the original format is pcm_s16le; not overly familiar with that [04:37] <klaxa> well it's just raw pcm [04:37] <klaxa> samples are stored uncompressed [04:38] <klaxa> the direct digital equivalent to the analog source [04:38] <klaxa> http://en.wikipedia.org/wiki/Pulse-code_modulation [04:39] <Maverick|MSG> looks like if I don't choose an audio codec ffmpeg uses libvo_aacenc @ 128kb/sec [04:39] <Maverick|MSG> what is s16 (and s16p)? [04:40] <klaxa> s16 means signed 16 bit integers [04:40] <Maverick|MSG> and the p? [04:40] <klaxa> dunno 'bout that p [04:40] <Maverick|MSG> k [04:41] <klaxa> there's an "le" at the end [04:41] <klaxa> that means little endian and describes the byte order [04:41] <Maverick|MSG> gotcha [04:41] <klaxa> i forget which one is which all the time [04:42] <Maverick|MSG> trying to decipher all the abbreviations in the ffmpeg output is mind bloggling [04:45] <Maverick|MSG> according to wikipedia aac is better than mp3? [04:46] <Maverick|MSG> would it be better to use that at 320kb instead of mp3 then? [04:46] <klaxa> i don't think it really makes a difference, especially given that libvo_aacenc is probably not as sophisticated as lame [04:47] <Maverick|MSG> is libvo_aacenc the best (or only) codec for ffmpeg? [04:47] <klaxa> libfdk-aac is considered the best aac encoder [04:47] <klaxa> https://trac.ffmpeg.org/wiki/AACEncodingGuide [04:48] <Maverick|MSG> hm, pretty sure I don't have that available in the windows binary [04:51] <Maverick|MSG> yup, nevermind on that [04:51] <Maverick|MSG> I need to stop perseverating on this and move on [04:54] <Maverick|MSG> thanks again klaxa, much appreciated [04:54] <klaxa> no problem :) [05:04] <Memphizzzzzz> klaxa: i compiled yasm 1.2.0 as its needed by x264, then i compiled x264 but now if i run ./configure --enable-gpl --enable-libx264 on ffmpeg it tells me libx264 not found, any idea why? [05:04] <klaxa> did you run make install in the x264 source directory? [05:04] <Memphizzzzzz> yes [05:05] <Memphizzzzzz> http://pastebin.com/EqkdUwPn [05:05] <Memphizzzzzz> this is what came out when i did x264 make install [05:06] <klaxa> can you run "ldconfig | grep x264" and see if that outputs libx264? [05:07] <Memphizzzzzz> it doesnt [05:07] <klaxa> run ldconfig without parameters once then [05:07] <klaxa> it should update its cache [05:07] <Memphizzzzzz> nothing [05:07] <klaxa> yeah it will not output anything [05:08] <klaxa> did you try to run configure again in ffmpeg's source directory? [05:08] <Memphizzzzzz> thats what im trying [05:08] <klaxa> i see, then maybe the directory /usr/lib is not in your library path [05:09] <Memphizzzzzz> i did not add --prefix="$HOME/ffmpeg_build" when runing configure on x264 do i need it? [05:09] <klaxa> you should have a file /etc/ld.so.conf [05:09] <klaxa> it should contain a line like "include /etc/ld.so.conf.d/*.conf" [05:09] <Memphizzzzzz> thats the only line in there [05:09] <klaxa> that's good [05:10] <klaxa> now you will create a new file in /etc/ld.so.conf.d/ called usr-local.conf or just about anything ending with .conf [05:10] <klaxa> that file will contain one line, this one: /usr/local/lib [05:11] <klaxa> no quotes, no trailing slash [05:11] <klaxa> then run ldconfig again and try building ffmpeg [05:11] <Memphizzzzzz> still the sam error :( [05:12] <klaxa> did you run ldconfig ? [05:12] <Memphizzzzzz> yes [05:12] <relaxed> Memphizzzzzz: --extra-cflags="-I/path/to/prefix/include" --extra-ldflags="-L/path/to/prefix/lib" [05:12] <Memphizzzzzz> whats path to prefix? [05:12] <relaxed> in your case, /usr/local/include and /usr/local/lib [05:14] <Memphizzzzzz> ./configure --enable-gpl --enable-libx264 --bindir="/bin" --extra-cflags="-I/usr/local/include" [05:14] <Memphizzzzzz> --extra-ldflags="-L/usr/local/lib" [05:14] <Memphizzzzzz> still the same error :( [05:14] <relaxed> "make distclean" and try again [05:15] <Memphizzzzzz> /usr/local/include does have a files x264_config.h and x264.h [05:15] <Memphizzzzzz> /usr/local/lib contains files: libx264.a [05:15] <relaxed> I know [05:16] <Memphizzzzzz> in which folder do i have to run make distclean? [05:16] <relaxed> top level of ffmpeg's source [05:16] <Memphizzzzzz> i ran it in ffmpeg but ./configure still gives the same error [05:17] <relaxed> you checked out x264 from git? [05:17] <Memphizzzzzz> i ran git clone --depth 1 git://git.videolan.org/x264.git in the ffmpeg folder [05:18] <Memphizzzzzz> then i had a x264 folder and did the ./configure --bindir="/bin" --enable-static [05:18] <Memphizzzzzz> and then i had to get yasm 1.2.0, configure, make, make install [05:19] <Memphizzzzzz> after that i was able to ./configure --bindir="/bin" --enable-static again which worked [05:19] <Memphizzzzzz> then i did make and make install [05:19] <Memphizzzzzz> and now im trying to configure ffmpeg [05:20] <relaxed> pastebin your config.log [05:20] <Memphizzzzzz> from which folder? [05:21] <relaxed> Memphizzzzzz: oh, never mind. [05:22] <relaxed> you need --extra-ldflags="-ldl -L/usr/local/lib" --extra-cflags="-I/usr/local/include" [05:23] <Memphizzzzzz> aha! [05:23] <Memphizzzzzz> ok now its make-ing again :) [05:24] <Memphizzzzzz> relaxed: what are these flags and why do i need them only if i want to include x264 [05:24] <relaxed> llogan: please make a note under the "x264" section of https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide that says you need --extra-ldflags="-ldl" [05:27] <relaxed> llogan: I know it's included in the ./configure there but for people using it as a reference it will save them/us headaches. [05:29] <relaxed> actually, the requirements for each external lib would be great. For example, Under "x264" have --extra-ldflags="-ldl" & --enable-gpl [05:30] <Memphizzzzzz> so would my ./configure have worked without the /usr/local/include stuff by only adding --extra-ldflags="-ldl"? [05:30] <relaxed> Depends. [05:30] <Memphizzzzzz> on wether or not /usr/local/lib and include are in my path? [05:31] <Memphizzzzzz> what does -ldl do? [05:36] <relaxed> it's needed for libx264 to open a dynamic library [05:37] <relaxed> for opencl [05:37] <relaxed> if you compile x264 with --disable-opencl then it's not needed. [05:40] <Memphizzzzzz> relaxed: i guess its faster with opencl :) [05:42] <Memphizzzzzz> klaxa, relaxed, llogan: Thanks for all your help! Im going to sleep now as its pretty late already, my NAS is currently compiling ffmpeg which is going to take some time, but i will try converting my mkv to mp4 and aac tomorrow again and will let you know how it goes :) Thanks again! Gn8 [06:41] <Zeranoe> Is tcp streaming broken on Windows? I'm getting an "Unknown error" with: "ffmpeg -i "vid.mp4" -f mpegts tcp://localhost:5596" [07:48] <BoomerBile> http://dranger.com/ffmpeg/ <-- i was trying to follow this tutorial, typed in all the code, read through it.... it's far more out of date than the guy claimed... so I decided to fix it up using what i could dig up out of ffmpeg header files... I've got it to compile, here's the output http://pastebin.com/sDQfhzPW [07:48] <BoomerBile> http://pastebin.com/pJGtX80d <-- here's the new source code i've come up with, but i'm doing something wrong [07:48] <BoomerBile> the program outputs ppm files and they are readable by any image editor, however they are all black. [07:49] <BoomerBile> anyone care to help a newb? [08:13] <BoomerBile> Ha! got it [08:13] <BoomerBile> thanks anyway [08:21] <SnowRaptor> Hey there! Is it possible to add hard subtitles but keep the original codec options? [08:48] <zap0> yes [08:54] <SnowRaptor> zap0, thanks, it worked! [08:54] <zap0> im glad my help was so informative. [08:55] <allengreen> where can I find the prebuild ffmpeg binaries for linux(with libx264, fdk_aac) [08:57] <SnowRaptor> zap0, I googled deeply while waiting and blindly tried somethings ;) [08:59] <mark4o> allengreen: the licenses of libx264 and fdk_aac are not compatible [09:00] <Katharsi1> hi [09:00] <mark4o> allengreen: you can build your own ffmpeg that includes both, but distributing it would violate the license [09:00] <Katharsi1> this is the line which i use for recoring my screencasts [09:00] <Katharsi1> http://wklej.to/4xQ3f/text [09:01] <allengreen> I just wanna test the binaries. [09:01] <Katharsi1> audio is out of sync, can you tell me why? [09:01] <Katharsi1> file out: .mp4 [09:02] <allengreen> mark4o, there are binaries include libx264, but without libfdk_aac. [09:02] <allengreen> I need both. [09:03] <mark4o> allengreen: right, libx264 is GPL, which is compatible with ffmpeg's LGPL license, but fdk_aac license is not compatible with GPL [09:03] <Katharsis> i'm recording on linux (debian 7+) [09:16] <Katharsis> anybody help me? [09:18] <relaxed> allengreen: build your own [09:20] <BoomerBile> Katharsis, i had the same problem, i just split the audio and video and adjusted it... my split method is way out of date, but here ya go... http://pastebin.com/Jbyi4HUc [09:21] <Katharsis> python code ;] [09:22] <BoomerBile> yeah i use it to help other people by making them movies of what to do... however it's a work in progress and very messy... sorry about that [09:22] <BoomerBile> runs great in gentoo, probably debian as well [09:23] <BoomerBile> it will produce an mkv and an mp4 file on it's own... i did that so i could have a high quality file and one to upload to my web server (size issue) [09:24] <Katharsis> BoomerBile: what's about -async arg? [09:25] <Katharsis> I don't want to split it out, doesn't have a time for merge [09:25] <BoomerBile> been a long time since i edited this, not sure why i put the async arg in there... probably to try and fix the sound issue, i'd have to look it up [09:26] <BoomerBile> it's commented out as you can see [09:28] <BoomerBile> i never use sound anyway that's why i commented out the sound split routine [09:29] <JoeGibken> Hi! I've been googling and searching for a while (around 30mins already LOL), but can't seem to find how to combine 3 videos into. Not concatenate, but join them using layers. Is that possible using ffmpeg? [09:31] <BoomerBile> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=15&t=1208 <-- JoeGibken [09:40] <BoomerBile> http://broadcasterproject.wordpress.com/2010/05/18/how-to-layerremix-videos… <-- JoeGibken [09:40] <BoomerBile> hope that helps [09:55] <JoeGibken> BoomerBile: Thank you! Lemme check that out :D [09:55] <BoomerBile> JoeGibken, go for the second link i posted, much nicer [09:56] <JoeGibken> BoomerBile: Will do. Thanks again ^o^/ [10:02] <allengreen> if I don't compile ffmpeg with enable-libx264 can ffmpeg do h.264 encoding? [10:06] <BoomerBile> allengreen, i don't believe so [10:07] <BoomerBile> allengreen, i run gentoo, that means i compile everything i run... if i don't use enable-libx264 i get no h.264 [10:09] <allengreen> get it. [10:24] <JoeGibken> BoomerBile: Hmm... Let me re-explain my problem at the moment - I have a video that is in 4:3 that is 720x480 from a DVD. But I have another footage of the same scenes that is recorded in 1280x 720. [10:24] <JoeGibken> I'm trying to make the 720x480 into 1280x720 so I upscaled it to 1080x720 - but there are black borders padded on the left and right the upscaled 1080x720 [10:25] <allengreen> I config ffmpeg with --disable-static --enable-shared , but when I run ffmpeg, It always says "error while loading shared libraries: libfdk-aac.so.0" [10:25] <JoeGibken> I don't want the black borders on the side [10:25] <BoomerBile> allengreen, you need to install the fdk-aac libraries [10:25] <JoeGibken> Instead, made blurred version of the upscaled 1080x720 with black borders on the left (so that's the 2nd video ) [10:25] <allengreen> --disable-shared --enable-static [10:26] <BoomerBile> allengreen, that would build ffmpegs static libraries for you to use to write your own programs and not build the shared ones [10:26] <allengreen> I don't wanna insatll fdk-aac, I wanna contain it into ffmpeg binary. just one executable. [10:26] <JoeGibken> And then made a blurred version of the upscaled 1080x720 with black padded borders on the right (so it's the 3rd video) [10:27] <BoomerBile> one sec JoeGibken reading your stuff here [10:27] <allengreen> I just wanna build all third party av codecs into one binary. [10:27] <JoeGibken> So I need the video 1 upscaled (1080x720) and the 2 upscaled blurred 1280x720 (sorry, not 1080x720) on the left and right - so merging them together [10:28] <BoomerBile> you want your compilers -static (gcc) option for linking [10:28] <JoeGibken> BoomerBile: No problem. I'm already thankful you're helping :D [10:28] <BoomerBile> allengreen, in order to build a static version of ffmpeg you will also need all of the supporting libraries built statically [10:30] <allengreen> if I build fdk_aac both statically and shared, how to choose the static one? [10:30] <BoomerBile> you link to the static library instead of the shared library [10:32] <allengreen> only one place I can choose is --enable-libfdk_aac, but how to specify the static one? [10:32] <BoomerBile> JoeGibken, still a bit confused as to your final goal... you want the videos overlaid one on top of another? or you want to resize your video without the black borders? [10:34] <BoomerBile> allengreen, you have to tell your linker to statically link it... you can usually point directly at the static library (usually called <something>.a) i've never been able to build any static programs so you're kind of asking the wrong guy... [10:34] <BoomerBile> allengreen, i have ffmpeg installed and after you asked the question i simply enabled fdk with a use flag (in gentoo) and rebuilt ffmpeg, now i have x.264 and fdk in my ffmpeg install [10:35] <BoomerBile> building a static binary is a tricky job [10:35] <allengreen> I'll try again. [10:35] <JoeGibken> BoomerBile: What I'm trying to achieve is that the the upscaled 4:3, originally 720x480 - which I upscaled to 1080x720, will be 1280x720 16:9, but not with black borders, but very very blurred borders of the original video [10:35] <BoomerBile> i've been trying for months to build a static binary of the game i wrote just using sdl for libraries... i still can't get it to link properly [10:35] <JoeGibken> Basically, the padding of the borders are not black, but the borders are the very very blurred version of the original file [10:36] <BoomerBile> JoeGibken, so you want to resize the video and add fuzzy (blurred) borders to it [10:36] <JoeGibken> BoomerBile: Exactly, yes :D [10:38] <BoomerBile> leme see what i can come up with [10:40] <geomyidae> how much of a file does ffprobe need? [10:40] <geomyidae> is there a set number of minimum bytes needed? [10:42] <zap0> it likely reads something like 4k [10:42] <geomyidae> ok, I have to grab 64K anyway [10:44] <JoeGibken> BoomerBile: Thank you :D [11:05] <JoeGibken> BoomerBile: Have you by chance found anything I can use? [11:06] <BoomerBile> working on the crop option [11:06] <JoeGibken> Or I'm thinking, BoomerBile, I'll just crop the left and right borders that are superblurred [11:06] <JoeGibken> Then patch them both to be the borders [11:06] <JoeGibken> But sadly, I'm not sure what commands I could use to do that [11:08] <BoomerBile> i'm working on resizing and changing the aspect ratio, along with cropping the black borders off, then i'll add a blur filter [11:13] <JoeGibken> BoomerBile: Wow! You can do that in only 1 line?! Amazing! *A* Thank you for being fantastically brilliant BoomerBile! I'm really grateful for the power of ffmpeg <3 [11:14] <BoomerBile> alright, i took a 720x540 video and resized it to 1920x1080 with aspect 16:9, no borders left... now to work on the blur filter [11:17] <JoeGibken> BoomerBile: Thanks! Looking forward <3 [11:27] <BoomerBile> some days i really hate nvidia [11:32] <nlight> BoomerBile, me too, on the other days I merely despise them :D [11:32] <surf> hi guys I am running debian wheezy and I wonder if I can convert an ogv file into an mpeg one without proprietary codecs... should I enable deb-multimedia and download from there the codecs? [11:32] <BoomerBile> nlight, freeking sli mosaic keeps causing segfaults in the driver [11:33] <BoomerBile> locks my computer and blanks the screens one at a time [11:41] <JoeGibken> BoomerBile & nlight - I know EXACTLY what you mean. No wonder Linux Torvalds said this - http://www.youtube.com/watch?v=O0r6Pr_mdio [11:42] <JoeGibken> *Linus Torvalds [11:43] <BoomerBile> is that the f* nvidia video? [11:44] <BoomerBile> if so yeah i was so proud [11:44] <allengreen> here is my config.log gcc -L/home/lei/coding/ff1/build/lib -static -Wl,--as-needed -o /tmp/ffconf.KwgCwrDj /tmp/ffconf.zr87JJ3f.o -lx264 -lvorbisenc -lvorbis -logg -ltheoraenc -ltheoradec -logg -lspeex -lfreetype -lm -pthread -lz -lrt [11:44] <allengreen> /home/lei/coding/ff1/build/lib/libx264.a(opencl.o): In function `x264_opencl_load_library': [11:45] <allengreen> opencl.c:(.text+0x46f): undefined reference to `dlopen' [11:45] <allengreen> then configure failed. [11:45] <BoomerBile> ffmpeg -i <video in> -vf crop=in_w:in_h*0.5,scale=1920:1080 <video out> <-- there is most of what you need JoeGibken, you will have to play wth the 0.5 a bit to get the desired crop (this example will crop the top and bottom) [11:46] <JoeGibken> BoomerBile: Yeah! Haha - I always rewatch that whenever I curse at my Nvidia haha [11:47] <JoeGibken> Okay - thanks BoomerBile! Lemme try that - thank you for being brilliant for helping me :D [11:47] <allengreen> may I should add -ldl to extra-ldflags [11:47] <BoomerBile> the crop option works like crop=<width you want from input>:<height you want from input>:<start from x positon of input (width)>:<start from y position of input (height)> [11:47] <JoeGibken> But BoomerBile, I don't think I need cropping of the original 4:3 video you see [11:47] <JoeGibken> I just did some upscaling, there are no borders on the 4:3 video you see [11:48] <BoomerBile> ahh [11:48] <BoomerBile> then all you should have to do is -vf scale=<desired width>:<desired height> [11:48] <BoomerBile> as for the faded edges, i'd recommend using a png overlay [11:48] <BoomerBile> and that overlay tutorial i showed you [11:48] <BoomerBile> just overlay a blurred edge png file over the video [11:49] <JoeGibken> Oh, but the borders on the left and right are png files? Not the video itself? [11:50] <JoeGibken> I'm actually trying to achieve that the borders on the left and right came from the original 4:3 just cut off, but very very blurred [11:50] <JoeGibken> What you mean win the png files are static images? I'm actually trying to achieve borders that are still from the original 4:3 video, just very very blurred [11:50] <BoomerBile> well, you just make a png file that is the size of the desired output video, blur effect the edges of the transparent png file, then overlay it on top of the video... everything in the middle stays the way it is, everything on the sides gets blurred [11:56] <JoeGibken> Thanks again BoomerBile! I'll try it and will get back to you if I achieved it. [12:15] <BoomerBile> http://imgur.com/bZnLx0p <-- JoeGibken something like that right? [12:16] <BoomerBile> only instead of a white border you want it blurred [12:24] <JoeGibken> BoomerBile: Ah no no. Not like that [12:24] <JoeGibken> Kindly wait [12:32] <superware> I'm getting "[rtp @ 01e708a0] Unable to receive RTP payload type 96 without an SDP file describing it", is there NO WAY to play h264 rtp over udp without a physical SDP file? even directly with the ffmpeg lib? [13:17] <curfont> Hi, whether i put --enable-static or --disable-shared or --enable-shared, libavcodec.a comes out the same size, is this normal? [14:02] <Mavrik> curfont, of course. [14:02] <Mavrik> libavcodec.a is a static library [14:03] <Mavrik> if you build another shared one, it won't have an effect :P [14:09] <curfont> Mavrik: when I try to link it against zoneminder, it gives me strange "undefined references" such as libva and others [14:09] <curfont> Shouldnt that not happen if its static? [14:10] <curfont> /ffmpeg_sources/ffmpeg/libavcodec/vaapi.c:61: undefined reference to `vaUnmapBuffer' [14:10] <curfont> for example [14:15] <Mavrik> nop [14:15] <Mavrik> if you say static [14:15] <Mavrik> you mean "make libav* libraries static" [14:15] <Mavrik> not "make ALL dependencies static as well" [14:16] <Mavrik> if you don't have static dependencies available, the .a files will link to dynamic libs anyway [14:16] <LLckfan> Does any1 know how to get a stale house smell out of a clean house without opening windows as fresh air is of no help? [14:16] <Mavrik> and obviously your libva is dynamic only [16:30] <MichaelP> i was told i could use ffmpeg as a desktop recorder [16:33] <sacarasc> You were told correctly. [16:34] <MichaelP> sacarasc: it's cammand line right or is there a gui ? [16:34] <sacarasc> Command line. [16:36] <MichaelP> Hows the commandline work for it [16:36] <microchip_> MichaelP: read the wiki [16:36] <microchip_> there are examples [16:37] <durandal_1707> it done with x11grab, which does not work on Windows [16:37] <durandal_1707> *it is [16:37] <sacarasc> There is a windows version, too. [16:37] <durandal_1707> really? [16:39] <sacarasc> I believe so. [16:39] <sacarasc> https://trac.ffmpeg.org/wiki/How%20to%20grab%20the%20desktop%20(screen)%20w… [16:39] <sacarasc> Seems to be there. [16:40] <durandal_1707> that depends on specific dshow device which may not always be present [16:42] <MichaelP> This is what it giving me http://dpaste.com/1410958/ [16:48] <MichaelP> Got it working with ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg .. But it not clean clear quality [16:52] <durandal_1707> you are resizing vga to cif [16:52] <durandal_1707> please read x11grab documentation [16:53] <durandal_1707> -video_size WxH sets capture resolution [16:53] <durandal_1707> default is vga, which is 640x480 [18:52] <Insayne> hello, i am trying to compile ffmpeg with libmp3lame (on FreeBSD), i enable it during make config, then i make install clean, yet it does not include libmp3lame ? [18:52] <Insayne> any idea why ? [18:55] <durandal_1707> make config and make install are not part of ffmpeg build process, they are part of ports [19:07] <captainastro> If you're working from git clone or a release tarball you need to call configure with "--enable-libmp3lame". Look at configure for a full list of parameters. [19:16] <Insayne> im working from ports [19:16] <MichaelP> can't get ffmpeg to grab the hole screen... http://imgur.com/PwSCsPx ... ffmpeg -f x11grab -r 50 -s 1024x768 -i :0.0 out.mpg [19:16] <Insayne> make configure is what i do [19:17] <Insayne> i then get a list of options, and choose it, but yeah... [19:19] <Insayne> how else am i supposed to do so ? [19:23] <Insayne> any ideas ? [19:23] <MichaelP> I think they fell asleep [19:24] <Insayne> well you get yours to compile at least :P heh [19:24] <MichaelP> no.. i used rpms from my distro [19:29] <durandal_1707> Insayne: you are asking for help using ffmpeg ports on wrong channel [19:29] <durandal_1707> nobody here use FreeBSD ports to install ffmpeg [19:33] <MichaelP> anyone have any idea how i get ffmpeg to grab full desktop insted of half ? [19:33] <Insayne> okay, from souce, it does not find libmp3lame [19:33] <durandal_1707> Insayne: it should if it is installed [19:34] <durandal_1707> MichaelP: you need to use correct command, required args are explained in documentation: hint: use: -video_size [19:35] <durandal_1707> but i already told you this [19:35] <durandal_1707> perhaps i need to write full command for you... [19:37] <MichaelP> durandal_1707: that wiki is not the best [19:37] <durandal_1707> yes i believe you now [19:39] <MichaelP> That would be nice [19:40] <durandal_1707> what would be nice? [19:41] <MichaelP> <durandal_1707> perhaps i need to write full command for you... [19:42] <durandal_1707> MichaelP: have you read wiki? [19:42] <MichaelP> yes [19:47] <durandal_1707> what link? [19:47] <MichaelP> http://ffmpeg.org/ffmpeg-devices.html#x11grab [19:49] <durandal_1707> there is -video_size mentioned there [19:49] <durandal_1707> use it [19:53] <Insayne> hrm, it doesnt find lame.h [20:06] <durandal_1707> Insayne: the ./configure or port system? [20:20] <Insayne> well now i get gcc: -l/usr/local/include: linker input file unusued because linking not done [20:20] <Insayne> and i fail the C test [20:21] <Insayne> any idea what i can do about that ? [20:23] <durandal_1707> Insayne: unlikely as nobody can see what commands you actually typed [20:24] <durandal_1707> anyway ports system should find lame [20:24] <durandal_1707> and i assume stuff they do is actually tested [20:25] <Insayne> its not [20:25] <Insayne> okay, i cant sadly paste [20:25] <Insayne> so i write it for you [20:25] <durandal_1707> but you need to set both extra-ldflags and extra-cflags... [20:25] <Insayne> i did [20:25] <Insayne> and then the c compiler fails [20:26] <Insayne> if i dont do it, it doesnt fail the gcc test [20:26] <Insayne> let me type you my command: [20:26] <durandal_1707> use pastebin [20:26] <Insayne> i cant paste... [20:26] <Insayne> i just told you so, heh [20:27] <Insayne> ./configure --enable-libmp3lame --disable-yasm --enable-static --disable-shared --extra-ldflags=-l/usb/local/lib --extra-cflags=-l/usr/local/include [20:27] <Insayne> for which it screws up... with gcc [20:28] <durandal_1707> obviously as your -extra stuff is done wrong [20:28] <Insayne> how so ? [20:28] <Insayne> whats wrong ? [20:28] <ubitux> -l -I for includes [20:29] <ubitux> at least.. [20:29] <ubitux> lol @ --disable-yasm [20:30] <durandal_1707> without yasm you make bunch of stuff extra slow [20:31] <Insayne> so... : ./configure --enable-libmp3lame --enable-static --disable-shared --extra-ldflags=-l/usb/local/lib --extra-cflags=-I/usr/local/include [20:31] <Insayne> is correct ? [20:31] <durandal_1707> nope [20:31] <Insayne> whats wrong there now then ? i thought that was what you said [20:31] <Insayne> heh [20:31] <Insayne> i am really really confused here [20:31] <durandal_1707> also better install yasm if it is not available [20:32] <durandal_1707> me too [20:32] <Insayne> how come? you said its wrong [20:32] <durandal_1707> s/l/L [20:33] <Insayne> oooh that did work [20:33] <Insayne> heh [20:33] <Insayne> thank you, both [20:35] <Insayne> and now I can make it ? [20:35] <Insayne> as in "make install" ? [20:40] <durandal_1707> if you are right directory, maybe..... [20:40] <mittens> you know it says /usb/local not /usr/local? [20:42] <durandal_1707> probably typo [20:58] <mittens> yeah probs [21:36] <Insayne> welp [21:36] <Insayne> that does not work [21:37] <Insayne> it complains about makefile requireing an operator, etc [21:38] <Insayne> what am i doing wrong ? [21:39] <durandal_1707> what you are doing? [21:39] <Insayne> after the ./configure that was successfull, i typed [21:39] <Insayne> make install [21:39] <durandal_1707> gmake install [21:39] <Insayne> oh [21:39] <Insayne> heh, thank you [21:40] <Insayne> is there a way, that the port system gets the --extra-ldflags=-l/usb/local/lib --extra-cflags=-I/usr/local/include passed on ? [21:41] <durandal_1707> that is offtopic [21:42] <Insayne> is there a channel you know of where i could ask this ? [21:42] <durandal_1707> maybe on #freebsd [21:42] <Insayne> ok [21:42] <Insayne> thank you very much [21:44] <Insayne> and lastly, where do the compiled files get put ? [21:45] <Insayne> nvm, where they belong :D [22:15] <SILASSARSD> Just installed ffmpeg on centos 6 and followed the compilation guide [22:16] <SILASSARSD> everything works fine when as user root, but as user apache i can run the command ffmpeg [22:16] <SILASSARSD> on antother server i can do it [22:16] <SILASSARSD> the permissions for the binary ffmpeg are exactly the same [22:16] <SILASSARSD> only the location of the file is different [22:17] <llogan> the centos guide makes a "local" install in the home of the user who compiled it if that makes any difference to your setup [22:17] <SILASSARSD> the one not working is in /root/bin/ffmpeg and the other is in /usr/local/bin [22:17] <SILASSARSD> how can i give apache access to it [22:18] <llogan> you said apache can run the command [22:19] <SILASSARSD> yes on the other server [22:19] <SILASSARSD> and there it is in /usr/local/bin [22:19] <SILASSARSD> which is a root directory [22:22] <SILASSARSD> http://pastebin.com/3PsvrFV2 on the working server i can execute ffmpeg as apache as you can see in: http://pastebin.com/3PsvrFV2 [22:24] <llogan> sounds like a permissions issue and less of a ffmpeg specific issue? try ##linux or #apache [22:26] <SILASSARSD> http://www.pastebin.ca/2464688 you can see the other server, all permissions are the same [22:27] <klaxa> i doubt apache has read/execute permissions for /root/ [22:27] <klaxa> or rather, it shouldn't have by default [22:35] <SILASSARSD> what is the best method [22:35] <SILASSARSD> to install if I wnat to run as user apache [22:35] <SILASSARSD> create a user ffmpeg [22:35] <SILASSARSD> and set the /home/ffmpeg to 755 [00:00] --- Thu Oct 10 2013

1 0

ffmpeg-devel.log.20131008
by burek 09 Oct '13

09 Oct '13

[00:09] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:b6b9c150f06a: ffmpeg: add max_error_rate to allow tuning the point decoding errors are considered a failure of the whole transcoding. [00:17] <BBB> michaelni: I don't think x264 supports nasm [00:19] <BBB> michaelni: it sounds in short like the rodata table is not exported correctly, but it's hard to say for sure [00:20] <BBB> oh it's functions [00:20] <BBB> odd [00:20] <BBB> ... [00:21] <michaelni> x264 maybe doesnt but ffmpeg did, i didnt investigate at all what goes wrong ... [00:21] <Daemon404> it should still [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:3bd0cd26774d: dsicin: Add some basic sanity checks for fields read from the file [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:057f655dc6a9: smacker: Don't return packets in unallocated streams [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:47bb4d888e1e: smacker: Avoid integer overflow when allocating packets [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:01a58b439d5f: smacker: Make sure we don't fill in huffman codes out of range [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:8484351749d4: shorten: Use a checked bytestream reader for the wave header [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:d411f0775189: shorten: Break out of loop looking for fmt chunk if none is found [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:eba112b4e331: ape: Don't allow the seektable to be omitted [00:46] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:9e505065b55f: zmbvdec: Check the buffer size for uncompressed data [00:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:f8a72f041c04: mpc8: Check the seek table size parsed from the bitstream [00:46] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:af1fb1d4670f: Merge commit 'f8a72f041c049e812dfa1f32156327e9778f5710' into release/1.1 [00:55] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:557df77eab7d: mpc8: Make sure the first stream exists before parsing the seek table [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:9eba02d5dd70: omadec: Properly check lengths before incrementing the position [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:153deed18bed: sierravmd: Do sanity checking of frame sizes [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:e47543eda585: vocdec: Don't update codec parameters mid-stream [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:9f883e75e66a: cavsdec: Make sure a sequence header has been decoded before decoding pictures [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:b4c479a82adb: vp3: Check the framerate for validity [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:aeca97a7b5c1: svq3: Check for any negative return value from ff_h264_check_intra_pred_mode [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:9272aa469848: mace: Make sure that the channel count is set to a valid value [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:9f7a8b8f8f6a: matroskadec: Verify realaudio codec parameters [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:20c8f176293e: rv34: Check the return value from ff_rv34_decode_init [00:56] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:163196562fe7: oggparseogm: Convert to use bytestream2 [00:56] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:af26185bdc35: Merge commit '163196562fe744149ef599d754c30c08a9898381' into release/1.1 [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:fe8bc7d1ee09: mov: Don't use a negative duration for setting other fields [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:38bd229af9c4: ivi_common: Make sure color planes have been initialized [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:c8c93795e4af: mpeg4videodec: Check the width/height in mpeg4_decode_sprite_trajectory [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:2e12af458761: eacmv: Make sure a reference frame exists before referencing it [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:eac1c3f384ea: truemotion2: Use av_freep properly in an error path [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:3706c2299201: ffv1: Make sure at least one slice context is initialized [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:00516b5491fb: xmv: Add more sanity checks for parameters read from the bitstream [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:0b0f1cd44ece: rv10: Validate the dimensions set from the container [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:de75bc01cda5: idroqdec: Make sure a video stream has been allocated before returning packets [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:5358f1b50753: asvdec: Verify the amount of extradata [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:bacf5db1962a: mpegaudiodec: Validate that the number of channels fits at the given offset [01:11] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:4d90550cf95e: qpeg: Add checks for running out of rows in qpeg_decode_inter [01:12] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:5379c5184b9f: segafilm: Validate the number of audio channels [01:12] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:78dc022f6f8a: wtv: Add more sanity checks for a length read from the file [01:12] <BBB> I can try to find a nasm version [01:12] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:ce1dacb43546: rl2: Avoid a division by zero [01:12] <BBB> but it's kind of a waste of time [01:12] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:55aedd679a24: Merge commit 'ce1dacb435460dda1f9d453eaaeac44bd502aca4' into release/1.1 [01:12] <BBB> it's like supporting gcc-2.50 [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:97b6916f93e6: dcadec: Validate the lfe parameter [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:0c8c6b4419e0: wnv1: Make sure the input packet is large enough [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:b62704891d23: vc1dec: Fix leaks in ff_vc1_decode_init_alloc_tables on errors [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:a6a8f66608cc: vc1dec: Undo mpegvideo initialization if unable to allocate tables [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:17e7edf75b45: lavf: Avoid setting avg_frame_rate if delta_dts is negative [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:8bd27a167b6e: electronicarts: Check packet sizes before reading [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:4d60ab62e05d: vqf: Make sure sample_rate is set to a valid value [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:60701469ab9f: vqf: Make sure the bitrate is in the valid range [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:2e4c649b3e62: avidec: Make sure a packet is large enough before reading its data [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:2ff935a06008: xwma: Avoid division by zero [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:78aa2ed62017: alac: Do bounds checking of lpc_order read from the bitstream [01:33] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:04d2f9ace3fb: mvi: Add sanity checking for the audio frame size [01:33] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:5da68aff96ff: Merge commit '04d2f9ace3fb6e880f3488770fc5a39de5b63cbb' into release/1.1 [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:607863acaec8: riffdec: Add sanity checks for the sample rate [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:a89868d71470: electronicarts: Add more sanity checking for the number of channels [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:ad1223d6bcc6: bfi: Avoid divisions by zero [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:10f384e4f5d0: bfi: Add some very basic sanity checks for input packet sizes [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:c10f3fed259c: mov: Make sure the read sample count is nonnegative [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:a63e83cd4b43: pngdec: Stop trying to decode once inflate returns Z_STREAM_END [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:812955a12b19: pcx: Consume the whole packet if giving up due to missing palette [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:145de32896b3: xan: Use bytestream2 to limit reading to within the buffer [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:09ace619d6cc: xan: Only read within the data that actually was initialized [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:3ac156b70793: xxan: Disallow odd width [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:f06e39fe6b27: rpza: Fix a buffer size check [01:46] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:7e350b7ddd19: pcx: Check the packet size before assuming it fits a palette [01:46] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:8d14e1d64ee1: Merge commit '7e350b7ddd19af856b55634233d609e29baab646' into release/1.1 [02:07] <BBB> michaelni: fixed [02:07] <cone-430> ffmpeg.git 03Anton Khirnov 07release/1.1:fd2ba1807e9a: mxfdec: set audio timebase to 1/samplerate [02:07] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:fc4d11ec9b4c: asfdec: Check the return value of asf_read_stream_properties [02:07] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:9b9aee27f4e4: twinvqdec: Check the ibps parameter separately [02:08] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:99f953b3fe4b: rmdec: Validate the fps value [02:08] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:3ee5458aa8b2: svq3: Avoid a division by zero [02:08] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:98e1f0a2db41: fraps: Make the input buffer size checks more strict [02:08] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:99fe4c577fc8: r3d: Add more input value validation [02:08] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:494f2d4f9e83: vc1dec: Make sure last_picture is initialized in vc1_decode_skip_blocks [02:08] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:829e43017fa7: Merge commit '494f2d4f9e834db1eaf1a7d0160d497f9802013d' into release/1.1 [02:17] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:9597807ab197: vc1dec: Don't decode slices when the latest slice header failed to decode [02:17] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:42c7ed49a9e4: mpegvideo: Initialize chroma_*_shift and codec_tag even if the size is 0 [02:17] <cone-430> ffmpeg.git 03Martin Storsjö 07release/1.1:23e428416097: h263dec: Remove a hack that can cause infinite loops [02:17] <cone-430> ffmpeg.git 03Reinhard Tartler 07release/1.1:a9fc88ea93b2: Prepare for 9.10 RELEASE [02:17] <cone-430> ffmpeg.git 03Michael Niedermayer 07release/1.1:653cdf42ee90: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [02:24] <michaelni> BBB, thanks [02:34] <cone-430> ffmpeg.git 03Ronald S. Bultje 07master:ba9c557b92c4: avcodec/x86/vp9dsp: Fix compilation with nasm. [06:40] <Compn> carl is quick :) [06:40] Action: Compn afk [10:34] <cone-824> ffmpeg.git 03Loren Merritt 07master:25cb0c1a1e66: x86inc: activate REP_RET automatically [10:34] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:c1488fab3dd4: Merge commit '25cb0c1a1e66edacc1667acf6818f524c0997f10' [10:52] <cone-824> ffmpeg.git 03Loren Merritt 07master:e52567c2954f: build: Strip spurious labels [10:52] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:3d5a995eac2a: Merge commit 'e52567c2954f627d420b30f75f71af2f2e4afe80' [11:01] <cone-824> ffmpeg.git 03Henrik Gramner 07master:ad76e6e7e193: x86inc: Set ELF hidden visibility for global constants [11:01] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:faafffaf82a2: Merge commit 'ad76e6e7e193b98e7335156422d35467816f9ef1' [11:06] <cone-824> ffmpeg.git 03Henrik Gramner 07master:63f0d623100b: x86inc: Use SSE instead of SSE2 for copying data [11:06] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:006c0fcfeaed: Merge commit '63f0d623100bdb0c6081456127f4b6713e83d3db' [11:14] <cone-824> ffmpeg.git 03Loren Merritt 07master:49ebe3f9fe02: x86inc: fix some corner cases of SWAP [11:14] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:3352fdb2920f: Merge commit '49ebe3f9fe02174ae7e14548001fd146ed375cc2' [11:22] <cone-824> ffmpeg.git 03Loren Merritt 07master:3fb78e99a04d: x86inc: create xm# and ym#, analagous to m# [11:22] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:17d9c7c20891: Merge commit '3fb78e99a04d0ed8db834d813d933eb86c37142a' [11:37] <cone-824> ffmpeg.git 03Henrik Gramner 07master:bbe4a6db44f0: x86inc: Utilize the shadow space on 64-bit Windows [11:37] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:1f17619fe480: Merge commit 'bbe4a6db44f0b55b424a5cc9d3e89cd88e250450' [11:56] <cone-824> ffmpeg.git 03Derek Buitenhuis 07master:47f9d7ce5493: x86inc: Check for __OUTPUT_FORMAT__ having a value of "x64" [11:56] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:3f965ab95d94: Merge commit '47f9d7ce5493e119e09d1227d017414feaaf8d97' [12:02] <cone-824> ffmpeg.git 03Jason Garrett-Glaser 07master:71155665414b: x86inc: various minor backports from x264 [12:02] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:31d9aa6b2e1a: Merge commit '71155665414b551ad350622d5abed20e58371fbf' [12:09] <cone-824> ffmpeg.git 03Henrik Gramner 07master:3e2fa991db7e: x86inc: remove misaligned cpu flag [12:09] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:19c389081910: Merge commit '3e2fa991db7ef172579422accd61624d52777e5a' [12:17] <cone-824> ffmpeg.git 03Diego Biurrun 07master:6405ca7d4a55: x86: h264_idct: Update comments to match 8/10-bit depth optimization split [12:17] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:deb5addcff82: Merge remote-tracking branch 'qatar/master' [12:43] <cone-824> ffmpeg.git 03Ronald S. Bultje 07master:ad75d2b590f0: x86: Fix compilation with nasm on PPC & OS/2 [12:45] <michaelni> was there anyone who cared about checkheaders passing ? [12:45] <michaelni> if so see: http://fate.ffmpeg.org/log.cgi?time=20131007224020&log=compile&slot=x86_64-… [13:41] <BBB> michaelni: not ppc :) mac [13:42] <michaelni> oops :/ [13:43] <BBB> not important [13:43] <BBB> just funny :) [13:43] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:2dce4abe506a: Changelog: add max_error_rate [14:39] <cone-824> ffmpeg.git 03Michael Niedermayer 07master:8c0f565ee3ba: avformat/movenc: factor out find_fps() [14:39] <cone-824> ffmpeg.git 03Lukasz Marek 07master:361b56c37962: lavd: pulse audio encoder [15:27] <ubitux> http://forum.doom9.org/showthread.php?t=168947 [16:02] <Compn> lol first post is a troll on patents :D [16:29] <durandal_1707> Compn: what? [16:39] <michaelni> ubitux, are your fate clients still submitting data ? it seems after i changed their owner from "fate" to "ubitux" there where no more submissions did something got stuck or did i mess up ? [16:40] <michaelni> your clients always submitted quite frequently ... [16:47] <ubitux> let me check [16:48] <ubitux> huh wth [16:49] <ubitux> michaelni: "Slot x86_64-archlinux-gcc-threads-misc owned by somebody else" [16:49] <ubitux> i get that kind of warnings [16:49] <ubitux> michaelni: the uid is "ux" btw [16:50] <michaelni> do you get these errors still or is this from half an hour ago ? [16:50] <ubitux> right now it's running the tsan instance so it will take a while before it's trying again [16:50] <ubitux> but i can force one [16:51] <ubitux> [Tue, 08 Oct 2013 13:40:20 +0200 :: fate :: configs/threads-misc.cfg] [16:51] <ubitux> Slot x86_64-archlinux-gcc-threads-misc owned by somebody else [16:51] <ubitux> [Tue, 08 Oct 2013 13:43:18 +0200 :: end] [16:51] <ubitux> this was the last one [16:51] <michaelni> when i switched fate->ubitux i forgot the owner check, but i disabled it after i realized [16:51] <michaelni> so it should be working in theory [16:52] <ubitux> do you want me to test right now? [16:52] <michaelni> also if you want something else then ubitux there, "ux" "clement" or something just say so [16:52] <ubitux> no it's fine [16:52] <ubitux> i was wondering if you were checking the local uid somehow [16:53] <michaelni> nope, there are just names associated with each ssh key [16:53] <michaelni> and they where all fate thanks to probably copy and pasting [16:53] <cone-824> ffmpeg.git 03Paul B Mahol 07master:3abbf20962f7: dpx: abgr support [16:53] <cone-824> ffmpeg.git 03Paul B Mahol 07master:85b56e4803c1: avcodec/xwddec: use avpriv_report_missing_feature [16:53] <cone-824> ffmpeg.git 03Paul B Mahol 07master:162e22e14421: avfilter/af_join: remove now unnecessary goto [16:53] <michaelni> also no hurry with testing, it should be working as other clients are working [16:54] <michaelni> and theres a check that checks if previous matches current (which i disabled for now) [16:55] <michaelni> ill reenable that once all clients updated and everyone is happy with their owner name [16:55] <ubitux> ok [16:56] <ubitux> are you working on improving fate server? [16:56] <michaelni> no [16:57] <michaelni> just realized the owner = fate issue and wanted to fix that [16:57] <ubitux> ok, cool :) [16:57] <ubitux> michaelni: do we have a public repo for our fate changes? [16:58] <ubitux> or it's just accumulated local changes? [17:00] <durandal_1707> if request_channels is deprecate how one downmix? [17:01] <wm4> durandal_1707: request_channel_layout [17:01] <michaelni> i dont think baptiste put the thing anywhere in a public git repo but i might be wrong [17:01] <michaelni> also its only a few lines diff [17:02] <michaelni> burek, btw any news on the new fate server stuff ? [17:03] <durandal_1707> wm4: then dcadec needs to be updated [17:03] <wm4> yeah I heard this shit is inconsistent as fuck [17:04] <wm4> and the deprecation status is bullshit [17:04] <durandal_1707> michaelni: do you remember anything about FIXME FIXME FIXME FIXME XXX wrong line in cabac? [17:06] <michaelni> the code is only used for the selftest, and probly meant that the termination didnt match the spec or wasnt double checed agast the spec [17:26] <ubitux> michaelni: seems ok [17:27] <ubitux> tsan instance just got updated [17:32] <michaelni> good [18:42] <teratorn> can anyone tell me how to get the audio from "clip1" in this command, ffmpeg -f lavfi -i color=size=1280x720 -vf 'movie=source1.mp4[clip1]; movie=source2.mp4[clip2]; [clip1] scale=854x480[clip1_s]; [clip2] scale=426:240 [clip2_s]; [in][clip1_s] overlay=426:120 [first]; [first][clip2_s] overlay=0:120 [out]' [18:43] <teratorn> i'm just overlaying two videos on to a black canvas... but I lose the audio from clip1 and don't know how to add it back in to the output? [18:44] <durandal_1707> teratorn: because you use movie [18:44] <durandal_1707> that filter only takes video from input [18:44] <teratorn> it says it takes video and/or audio :-( [18:45] <durandal_1707> nope, amovie takes audio [18:45] <teratorn> I tried adding another [audio] output pad to the movie filter [18:45] <durandal_1707> movie cant give you audio [18:45] <teratorn> ok! [18:45] <teratorn> I believe you [18:45] <teratorn> so I can do another amovie filter to grab out the audio? [18:45] <durandal_1707> if you want to use movie filters and not -filter_complex add amovie=.... [18:46] <teratorn> yeah wasn't tackling filter_complex just yet [18:48] <teratorn> durandal_1707: so I'm using amovie=source1.mp4[audio] [18:48] <teratorn> durandal_1707: how do I reference that in the filtergraph to use it? [18:50] <durandal_1707> like video [18:51] <durandal_1707> please continue this on #ffmpeg [18:53] <teratorn> sure [19:34] <durandal_1707> https://github.com/lu-zero/libav/tree/avscale [19:35] <ubitux> so development started? [19:37] <durandal_1707> right question is when it will end [19:38] <cone-824> ffmpeg.git 03Michael Niedermayer 07release/2.0:9b02aa25937f: avcodec/parser: reset indexes on realloc failure [19:39] <cone-824> ffmpeg.git 03Paul B Mahol 07release/2.0:b4ccdf5e68c0: avcodec/ffv1dec: fix format detection [19:39] <ubitux> durandal_1707: sounds like a vapourware to me [19:40] <ubitux> i also doubt it will be better than sws in any aspect when it will start to be competitive with it [19:40] <durandal_1707> well its currently vapourware, (everthing starts from that) [19:40] <ubitux> and i'm guessing that unless it's sponsored by ffmtech it will be an epic fail, probably similar to config.d, but well... who am i to predict the future :) [19:40] <durandal_1707> sws sucks, and nobody wants to improve it: general packed yuv supports and refactoring ... [19:41] <durandal_1707> and more SIMD... [19:46] <cone-824> ffmpeg.git 03Michael Niedermayer 07release/2.0:a53fd4b7584a: update for 2.0.2 [20:50] <Daemon404> o it passed 200 [20:50] <ubitux> 8798 remaining [20:50] <ubitux> before we can joke about it [20:51] <Daemon404> its about ffaac [20:51] <Daemon404> its a joke by definition [20:51] Action: Daemon404 runs [21:03] <cone-824> ffmpeg.git 03Stefano Sabatini 07release/2.0:9d0bb7fc3991: doc/codecs: fix dangling reference to codec-options chapter (cherry picked from commit b4bd21b7fe2ad8be59b16538448586814e5db65b) [21:27] <cone-824> ffmpeg.git 03Paul B Mahol 07fatal: ambiguous argument 'refs/tags/n2.0.2': unknown revision or path not in the working tree. [21:27] <cone-824> Use '--' to separate paths from revisions [21:27] <cone-824> refs/tags/n2.0.2:HEAD: avfilter/af_join: remove now unnecessary goto [21:27] <Daemon404> \o/ [21:35] <durandal11707> the cone thing is broken [21:35] <durandal11707> michaelni: the strip thing for asm files looks to not work in some environments [21:41] <llogan> alright. another typoless tweet. [21:42] <durandal11707> Daemon404: what is the status of the lavfi deadlock? [21:42] <llogan> i assume i'm going to find something stupid after i submit it no matter how many times i review [22:30] <burek> michaelni, still waiting for baptiste to get in touch with you about that [22:35] <michaelni> baptiste, ping (I think burek needs some kind of access to the fate server) [22:37] <michaelni> burek, if baptiste is busy, then maybe you could fetch the data via rsync and run your server software on your box or something like that, dunno [22:37] <burek> no problem [22:37] <burek> let's give him a day/two to respond :) [22:38] <burek> and if there's no answer from him, we'll see what can we do about it, ok? [22:38] <michaelni> sure, no hurry as far as iam concerned [22:56] <cone-824> ffmpeg.git 03Lukasz Marek 07master:1ab9f322ee53: lavd/pulse: rename pulse.c into pulse_audio_dec.c [00:00] --- Wed Oct 9 2013

1 0

ffmpeg.log.20131008
by burek 09 Oct '13

09 Oct '13

[04:09] <allengreen> what's the difference between libfdk_aac and faac? which one is better? [04:11] <relaxed> libfdk_aac [04:22] <allengreen> relaxed, sounds like faac is awesome too. [04:30] <relaxed> allengreen: libfdk_aac is supposed to be much better. [04:30] <llogan> allengreen: https://trac.ffmpeg.org/wiki/AACEncodingGuide [04:38] <allengreen> thanks, fellows. [08:09] <StamEhad> Hi! I have a muxed UDP MPEG-2 multi-program transport stream. How can I use ffmpeg to dump this stream to a .ts file without any transcoding? [08:20] <luoluoluo> hi, noob question, what does v20544 for ffmpeg mean? what version corresponds to v20544? [08:21] <sacarasc> luoluoluo: Where did you get that from? [08:21] <luoluoluo> a private project that uses ffmpeg code [08:22] <JEEB> I don't think I've seen ffmpeg itself prefix the revision with "v" [08:22] <sacarasc> No. [08:22] <JEEB> but if that is indeed a commit number, it is rather old [08:22] <JEEB> like, years and years old :P [08:22] <sacarasc> Yeah, it's up in the 50ks now, isn't it? [08:23] <JEEB> 56k on my sep 15 build I'm using atm [08:23] <sacarasc> Or is it even higher? [08:23] <luoluoluo> Thanks guys. So it's a commit number? [08:24] <JEEB> most likely at least [08:24] <luoluoluo> OK, I will try to search the git log for it. [08:24] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=1f20782c0446ed373d6f915a26… [08:25] <JEEB> "only" four years old [08:25] <luoluoluo> thanks, how do you get this? I searched for a while and cannot find it. you are so cool [08:26] <JEEB> I could have searched in the git log from the command line, but in this case I was lazy and just searched the commits for that number via that webui :P [08:27] <JEEB> anyways, not sure you'll gain anything from getting such an old revision [08:28] <luoluoluo> ok. Since this project use it, I have to use it... [08:28] <JEEB> ugh [08:28] <JEEB> good luck with that :P [08:29] <luoluoluo> So the most recent version from it's committing date should be the version corresponding to it, right? [08:29] <JEEB> what? [08:29] <JEEB> no [08:29] <JEEB> also if this svn revision was used then it was used, not any version :P [08:30] <luoluoluo> Actually I am trying to build it for android ndk, so I am trying to search like "ffmpeg x.x.x android.mk" [08:30] <luoluoluo> that's why I wanna know the public version corresponding to 20544 [08:30] <JEEB> good luck with that :P [08:31] <luoluoluo> OK... Thanks JEEB for the hint [08:32] <JEEB> it's not a version, it's a random point of time as far as I can see. So it's that exact commit. And you'll surely have "extra fun" trying to compile such an old version for NDK or anything else like that [08:33] <luoluoluo> yeah, "extra fun" indeed. Caused much head ache to me.... [08:35] <luoluoluo> do you search from here? http://git.videolan.org/?p=ffmpeg.git;a=summary [09:00] <StamEhad_> Is it possible to directly dump a media stream to a file without transcoding? [09:26] <relaxed> StamEhad_: ffmpeg -i input -map 0 -c copy output [09:33] <StamEhad_> @relaxed: thanks! I'll try that. [11:50] <saschagehlich> hey, whenever I try to compile a video using libvpx as the video codec, I get a floating point exception (Ubuntu 13.04, installed via apt-get): https://gist.github.com/saschagehlich/04f0c9e8814977c3a4ce [11:53] <relaxed> saschagehlich: you need to compile a more recent version of ffmpeg, or use http://johnvansickle.com/ffmpeg/ [11:58] <saschagehlich> thanks relaxed [13:47] <WhiteNight> Hi, how can I decode a single h264 frame in my c++ program? [14:14] <zap0> WhiteNight, are you a unix beardo, basement dweller into nerd BDSM? if so, try linking in all same libXXXXX that ffmpeg does. [14:14] <zap0> WhiteNight, if you shower daily.. and sunlight doesn't scare you.. i'd just invoke ffmpeg and let it do all the work. [14:32] <durandal_1707> zap0: what a nonsense [14:35] <Woopa> Hey all. I need some help with ffmpeg. I'm trying to use -filter_complex but all i get is "Unrecognized option 'filter_complex'". This is my ffmpeg output so you can see which version Im using: http://pastebin.com/0RhmUZzf [14:38] <durandal_1707> Woopa: you need something more recent than 0.10 [14:38] <Woopa> Ok. I have to compile then if I'm not mistaken? [14:39] <durandal_1707> not at all [14:39] <durandal_1707> !downloads [14:39] <Woopa> Ok? [14:42] <Woopa> durandal_1707: Hmm. Ok. [14:45] <Woopa> durandal_1707: Crazy that the ppa they recommend for Ubuntu is that outdated. [14:50] <Woopa> now i get "Unknown input format: 'x11grab'" when im running the downloaded binary. Damn. [14:54] <durandal_1707> it is easy to compile ffmpeg yourself [14:55] <Mavrik> hmm, yeah, the static builds don't have X11 support [14:55] <Woopa> oh ok [14:56] <Mavrik> so I'm afraid you'll have to build ffmpeg yourself if you need X11 support [14:57] <Woopa> Compile it is then. :) [15:02] <durandal_1707> what about this one: http://ffmpeg.gusari.org/static/? [15:03] <Woopa> I'l try [15:10] <Woopa> Can't even extract the tar. xD And some builds are 0B in size. [15:10] <Woopa> Compiling now, hoping that will work. [15:19] <Woopa> nope still the same version. Wierd. [15:21] <durandal_1707> Woopa: what does not work? [15:21] <durandal_1707> did you compiled it yourself? [15:22] <Woopa> I followed this guide: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [15:22] <durandal_1707> and you instealled it? [15:22] <Woopa> sudo make install? [15:23] <durandal_1707> are you sure you enabled x11grab? [15:23] <durandal_1707> --enable-x11grab --enable-gpl [15:23] <Woopa> yep [15:24] <Woopa> this "ffmpeg 2>&1 | head -n1" tells me that I still use the old version. [15:24] <mittens> if you didn't pass --prefix to ./configure, its probably installed in /usr/local [15:25] <Woopa> aha [15:27] <Woopa> It says "./configure --prefix="$HOME/ffmpeg_build"" in the guide [15:28] <Woopa> Is that wrong? [15:30] <Woopa> mittens :) [15:36] <Woopa> I got a folder called "ffmpeg_build". [15:36] <Woopa> in ~/ [15:39] <Mavrik> then your compiled ffmpeg is in ffmpeg_build/bin [15:39] <Woopa> oh ok [15:40] <Woopa> which step should i alter to install it more proper? In bin, i guess. [15:41] <Mavrik> first uninstall the old one [15:41] <Mavrik> and if you don't pass --prefix to configure, it'll install into /usr/local which is in Ubuntu default search path [15:42] <Woopa> So I'l just run ./configure? [16:04] <Woopa> sort of got it now. [16:04] <avih> hi, could i use ffmpeg to create a virtual encoded file with random access from an arbitrary source to then be served over http (e.g. to mplayer or a flash player in a browser or webm in a browser). essentially, trying to create encoded chunks of fixed size which could be concatenated into a playable stream (even if not fully valid) , and with random access without transoding the entire video prio to the access point (but some prior decoding is acceptable). [16:04] <avih> it doesn't have to support many clients, and assuming the cpu is enough to encode at least 5x real time at my selected parameters. could this be achieved? [16:04] <Woopa> But i need to compile it with pulse. "--enable-pulse"? [16:05] <avih> and i could use extra tools, e.g. to pad each chunk with 0 at the end, etc [16:06] <avih> the part with which i look for help is the chunks with random acess using ffmpeg [16:09] <avih> also, i don't mind if it's a theoretically non-seekable container as long as the player could still somehow seek it [16:20] <Mavrik> avih, it looks like you need MPEG2-TS stream :P [16:20] <Mavrik> doing broadcasting? :) [16:24] <durandal_1707> Woopa: it will not enable pulse unless you use that flag [16:38] <avih> Mavrik: yeah, already started playing with just that :) thanks [16:39] <avih> Mavrik: can i create fixed-size chunks which could be concatenated into a playable and seekable stream? (Assuming i create a virtual file on the server on the fly according to range requests by the client) [16:41] <avih> Mavrik: not broadcasting. more of a media server of my local videos over hard network (socks, limited bacdwidth) [16:42] <avih> bandwidth* [16:42] <avih> Mavrik: and just for fun ;) [16:42] <Mavrik> yes [16:42] <Mavrik> M2TS is built for broadcasting, so you can cut it [16:43] <Mavrik> of course you need to cut on 188B packet boundaries [16:43] <Mavrik> and it helps if you cut on I-frames with PSS/SPS packets for H.264 streams [16:44] <avih> i'm flexible with the formats etc, as long as i can play a transoded on-the-fly and seek over a network [16:44] <Mavrik> seeking& is a problem [16:45] <avih> yeah [16:45] <Mavrik> you'll need a proper streaming server for that [16:45] <Mavrik> and of course a client that knows how to do it [16:45] <Mavrik> I had very good experience with Wowza and RTMP protocol for that [16:45] <avih> which is why i need to create a virtual file, so the server can present it as if it knows all the data in advance, but encode the chinks on the fly [16:45] <Mavrik> playing static mp4 files [16:46] <avih> rtmp might be hard to play over socks [16:46] <avih> but i don't mind an off the shelf solution. but so far i couldn't find one [16:46] <avih> (tried ps3 media server, lix media, and others) [16:47] <Mavrik> RTMPT works over socks IIRC [16:47] <avih> vlc kinda works, but its ui is terrible, and seek can't be done from the client (but it's workable with a remove desktop) [16:47] <avih> remote* [16:48] <avih> and creating such server would be fun ;) [16:48] <bencoh> its a bit of work [16:48] <avih> it is. [16:49] <avih> but depending on possibilities , it might be easier for a subset of the overall functions of such media server. right now i'm focusing on a virtual served transcoded-on-the-fly video [16:49] <bencoh> nginx-rtmp supports seeking under certain conditions [16:49] <Mavrik> transocidng on the fly with seeking is not really trivial [16:49] <avih> right [16:49] <Mavrik> doable, but you'll have to learn alot :) [16:50] <avih> heh [16:51] <avih> i was thinking of, e.g. if using a 100K/sec overall, then encode arbitrary 5s chunks, then pad them with 0 to some bigger size which would hold all such 5s transcodes with tight cbr [16:52] <avih> so then if the client asks for bytes 1M-2M, these would be chunks 2,3 (0 based), which all have fixed size [16:53] <Mavrik> that works [16:53] <avih> but for this i'll need a format+container which i could pad and concatenate, and that it stays playable with whatever player (preferably mplayer or vlc, for instance) [16:54] <Mavrik> well, I already told you which one that is [16:54] <avih> yeah [16:54] <Mavrik> MPEG2-ts was made for your use case [16:54] <Mavrik> it supports padding as well [16:54] <Mavrik> (not padding by zeros, but there are padding packets in the standard that are listened to by most clients) [16:54] <Mavrik> of course, not all players support TS, but alot of them do [16:54] <avih> so how would i encode the 5s chunk which starts at 100s at the source with mpeg2-ts? [16:54] <avih> (and which codecs would fit into that container?) [16:55] <Mavrik> pretty much all MPEG codecs work [16:55] <Mavrik> DVB-T e.g. currently puts either H.264/AAC or MPEG2/MP2 into it for broadcasting [16:55] <Mavrik> I suggest you stick with H.264/AAC for compatibility and efficiency [16:55] <avih> right [16:55] <avih> yeah, sounds reasonable [16:56] <avih> and can i transcode an arbitrary location at the source with relatively fast seek using ffmpeg? (preferably better seek than O(N), like O(log N) would be nice) [16:57] <avih> (and yeah, padding a non-0 pattern would be fine) [16:57] <Mavrik> seeking in ffmpeg is pretty fast always if GOP size is reasonable [16:58] <Mavrik> you'll probably just have to pad chunks manually, not sure if ffmpeg supports that [16:58] <avih> well, i don't have control over the sources, as they're very varied [16:58] <avih> i can pad manually, np [16:58] <avih> well, with some code :) [16:59] <avih> Mavrik: so for seek, should i use -ss on input? or output? the docs say the output is more accurate but slower. i guess i need accurate if i don't want duplicate frames etc? [17:00] <Mavrik> avih, if you put it on output ffmpeg will decode whole file until -ss [17:00] <Mavrik> if you put it in input it'll jump to approximate position, backtrack to first I-frame it can find and decode up until the exact point [17:00] <Mavrik> if you have a reasonable GOP (which you should), there's little reason of using -ss in output [17:02] <avih> right. sounds reasonable [17:02] <avih> so it would be roughly O(log n) + O (M) where N is the seek position and M is GOP size, right? [17:02] <Mavrik> mhm [17:02] <Mavrik> for most files it's a very small value [17:03] <avih> so if the disk IO is relatively fast, can i expect seek time of under 2s for, say, 1G source file of 2 hours? [17:03] <Mavrik> mhm [17:03] <Mavrik> for non-broken files it's very fast [17:03] <avih> just to get a feel of the seek time numbers [17:04] <Mavrik> especially if they have constant framerate [17:04] <avih> yeah, i think most files would have constant frame rate... i _think_ [17:06] <avih> do i want mpegts or mpegtsraw? [17:15] <avih> Mavrik: so, this kinda works (plain concat): ffmpeg -ss 0 -i source.mp4 -t 5 -vcodec libx264 -preset faster -vf scale=480:320:-1 -acodec libmp3lame -ab 32k -b:v 268k -f mpegts chunk0.mp4 [17:16] <avih> i made 3 chunks for 0s, 5s, 10s, concatenated them (no padding), and it plays, but for some reason in double frame rate.. [17:16] <avih> does this seem like a reasonable start? [17:17] <avih> also, the player (mplayer) seem to get confused with the video length [17:17] <Mavrik> use .ts extension for starters [17:17] <Mavrik> everything else looks ok [17:18] <Mavrik> yes of course, video length won't be valid :) [17:18] <Mavrik> you probably have to set constant framerate with -r [17:18] <avih> hmm.. will try both. this is fun :p [17:19] <avih> do i want mpegts or mpegtsraw? [17:20] <Mavrik> mpegts, the other thing is for internal use [17:20] <Mavrik> also, use h264_mp4toannexb filter [17:20] <Mavrik> some mp4 files will require it [17:20] <Mavrik> (it's a bitstream filter) [17:20] <avih> for the source file? [17:21] <Mavrik> when encoding the TS [17:21] <avih> does it effect decoding in ffmpeg? or encoding? [17:32] <bencoh> bitstream encoding/muxing [17:36] <avih> hmm.. i'm getting some duplicate audio chunks there, and also the video gets stuck at some stage (trying concat of 6 chunks). [17:36] <avih> can i use -r to set the output to use the input frame rate? [17:43] <avih> tried another video. similar symptoms: faster than expected frame rate and audio is not in sync, and also with duplicate audio segments. [17:43] <avih> i'm using this command to encode a single chunk: [17:44] <avih> ffmpeg -ss 3000 -r 23.976 -i source.mp4 -t 5 -vcodec libx264 -preset superfast -vf scale=480:320:-1 -acodec libmp3lame -ab 32k -b:v 268k -r 23.976 -f mpegts -bsf:v h264_mp4toannexb chunk0.ts [17:45] <avih> (symptoms are when playing concatenated few chunks, each encoded as above) [17:46] <avih> Mavrik: with the above command, can i expect the audio to align decently? [17:46] <Mavrik> there's no reason for audio to be out of sync, neither for video to run too fast [17:46] <Mavrik> unless your player is broken [17:47] <avih> i can try another player, but usually mplayer is pretty dependable... [17:49] <avih> vlc doesn't play with fast framerate. but it looks as if the chunks are not fully accurate. [17:50] <avih> (just tried also with pcm audio) [17:51] <avih> i could use avisynth as source. it creates an uncompressed virtual file, so maybe it'll be easier for ffmpeg to cut the chunks correctly.. [18:55] <durandal_1707> teratorn: but you cant use amovie inside -vf .... [18:55] <teratorn> durandal_1707: oh dear [18:56] <teratorn> OK time to -filter_complex [18:56] <durandal_1707> teratorn: you can use -af [18:56] <durandal_1707> -af amovie .... [18:56] <teratorn> durandal_1707: ah [18:56] <durandal_1707> that should work [18:58] <teratorn> durandal_1707: I get no audio output stream at all... [19:00] <teratorn> ffmpeg -f lavfi -i color=size=1280x720 -vf '...' -af 'amovie=source1.mp4' [19:01] <teratorn> it runs but just doesn't produce an audio stream [19:05] <teratorn> durandal_1707: any clue? [19:17] <durandal_1707> teratorn: what version you use? [19:17] <teratorn> durandal_1707: latest [19:17] <teratorn> durandal_1707: I figured it out one way with just, -f lavfi -i amovie=source1.mp4 [19:18] <teratorn> I would prefer to just acodec copy the audio, but meh [19:18] <durandal_1707> -c:a copy [19:20] <durandal_1707> but, yes, i think you need for copy to have audio in -i INPUt [19:22] <durandal_1707> but using filter_complex should be much friendlier [19:23] <durandal_1707> you specify all inputs with '-i inputX' [19:24] <teratorn> yeah... [19:24] <teratorn> durandal_1707: anyway I got it sorted. thanks! [19:33] <l1x> hi [19:33] <l1x> http://pastebin.com/3itcBs4m [19:34] <l1x> can somebody tell me how to instruct ffmpeg to set the pix_fmt to the correct value instead of -1? [19:35] <teratorn> durandal_1707: OK, problem... ffmpeg doesn't stop after exhausting the first movie... it just keeps pressing the last frame(s) forever and ever... any clue ? :-( [19:38] <teratorn> durandal_1707: sure I will if adding -t <time> doesn't fix it :) [19:39] <durandal_1707> -shortest [19:41] <Hurons> Hey [20:20] <Hurons> Anyone on that knows a bit about ffmpeg dshow for decklink cards? (HDYC uyvu422 support)? [20:24] <newguy123> anyone able to help a new guy with a basic restream command? [20:27] <llogan> newguy123: what are you trying to do? [20:29] <newguy123> I am trying to transcode and it was locking up so I tried just to do a simple restream and it does the same thing ffmpeg -re -f rtsp -rtsp_transport tcp -i rtsp://media-us-2.soundreach.net/s lcn_sports.sdp -vcodec copy -acodec copy -f rtsp rtsp://127.0.0.1/stream.sdp [20:57] <Hurons> Trying to capture from a decklink card using fdshow under windows. But cant get passed this buffer full error. Anyone have a clue? See : http://pastebin.com/XCDKxTV0 [21:02] <plm> Hi all [21:04] <plm> ffmpeg does it? Or are there a server that split the stream into two or more streams and sends one over each link and in other side gets stream pieaces and join again? I would like hav ea box with many 3g cards and split video over each one and other side join it and show.. [21:05] <avih> guys, i'd appreciate some help in forcing input video frame rate. for some reason, ffplay plays it correctly, but when i encode it with ffmpeg, i get twice as fast framerate (but audio plays normally, so audio comes out twice as long as the video) [21:06] <avih> i tried -r for the input or output, and also tried -vf fps=fps=25 for output, and still the player (mplayer) plays with double frame rate [21:06] <newguy123> http://pastie.org/8387463 <--- commands I have tried for restream [21:12] <avih> llogan: my scenario is complex and involves more than one file. my specific question was about forcing framerate of the input and/or output video with ffmpeg. [21:14] <llogan> good luck [21:16] <Hurons> Llogan, is using fdshow and/or windows with ffmpeg a strange thing? Cuz I've been pulling my hairs out so far. And stackoverflow, and the ffmpeg-user mailing dont seem to have any answers. Feels like I'm going about this the wrong way or something. [21:17] <avih> llogan: https://pastebin.mozilla.org/3219172 [21:18] <avih> the source is an avisynth script, on windows. [21:20] <avih> apparently the source is correctly recognized as 25fps, but when played with mplayer, the info it gives is that it's 50fps, so the video plays twice as fast and stops when the audio (which plays normally) is still playing [21:25] <llogan> Hurons: you could ask on ffmpeg-user [21:26] <Hurons> If you mean the mailing list, I have done so a few days ago. I will be a bit more patient, thanks. [21:27] <llogan> oh, i see it. sorry, but i don't have an answer for you (i have no decklink experience) [21:28] <Hurons> No worries. It's just frustrating since I have no audio/video/ffmpeg experience. So it's like a needle in a haystack atm. [21:29] <llogan> avih: do other players also have trouble with it, or only mplayer? [21:31] <avih> llogan: i just tested with vlc and directshow based (mpc) and both played correctly. however, my actual scenario is one which only mplayer can handle (of those players) [21:32] <avih> so i need it to work with mplayer. also, appreciating mplayer as the mothre of all players, i want it to work properly with it regardless [21:32] Action: llogan blames mplayer [21:33] <avih> llogan: so, would you help me with the ffmpeg syntax to force input and/or output frame rate, please? [21:33] <llogan> if it helps you can refer to the -r option and/or the fps video filter [21:33] <llogan> http://ffmpeg.org/ffmpeg-filters.html#fps [21:34] <llogan> but i don't know why mplayer is behaving like that [21:34] <avih> as i said, i already tried -vf fps=fps=25 per the example from that section [21:34] <llogan> the answer for when i don't know, however, is "try a newer (mplayer) build" [21:34] <avih> but maybe i got the syntax wrong somehow [21:35] <llogan> oh, i missed that [21:35] <relaxed> does ffplay play it too fast? [21:35] <avih> checking [21:35] <llogan> you said "ffplay plays it correctly" so i assumed you meant the output [21:36] <relaxed> If so, try mplayer -vfm ffmpeg input [21:36] <avih> i meant for the input. but now i also tried the output, and it's also fine. [21:36] <avih> (ffplay plays the output fine) [21:37] <avih> but my ultimate usecase is ugly concatenation of ts files, which i expect to play decently. and neithre ffplay, not vls or mpc could play the concatenated file. mplayer can. but it has double frame rate (eith in a single chunk or with the concatenation) [21:38] <llogan> is that why you are using avisynth, to concat? [21:39] <avih> no, i use avisynth to get frame accurate and sample accurate chunks [21:39] <avih> is i use -ss 1000 -i source.mp4 -t 10 and then anothre chunk which starts at 1010, and then concat the outputs, i get overlapping audio [21:41] <avih> and the video is also messed up. the avisynth script (funnily, with ffmpegSource2), produces accurate chunks which after encoding concatenate to a decently continuous video and audio [21:42] <relaxed> -ss after the input will be more accurate [21:42] <relaxed> if all else fails try tsmuxer [21:42] <avih> yeah, but i need random access to chunks, and -ss for the output will decode whatever happens before that timetamp, which i cannot do [21:43] <avih> from performance point of view [21:43] <avih> the avisynth source indexes the file once and caches the iframes index, so seeking is both fast and frame accurate [21:44] <avih> (i'm trying to build a transcoding server which transcodes on the fly and on demmand, with seek support, by exposing a virtual transcoded file, whose chunks i generate on the fly) [21:45] <durandal11707> avih: what ffmpeg version you use? [21:45] <avih> from the past few days [21:45] <avih> ffmpeg version N-56930-gee3d03b [21:47] <avih> so i have two problems. only mplayer can play the concatenated mpeg2-ts chunks decently, and mplayer plays it with double frame rate whether it's concatenated or not. [21:48] <avih> (eventually the chunks will be padded to a fixed size such that the virtual transcode is consistent and predictable, but for now i just play with the concat) [21:55] <Hurons> I'm out, thanks for the conversation Llogan. [22:01] <relaxed> avih: did you try mplayer -vfm ffmpeg input? [22:01] <avih> relaxed: no. not sure what that means. is this an ffmpeg option? or an mplayer one? [22:02] <relaxed> mplayer [22:02] <avih> will try now [22:02] <relaxed> if that doesn't work try "mplayer -demuxer lavfpref input" [22:03] <relaxed> The former forces ffmpeg's decoders, the latter its demuxers [22:04] <relaxed> try both for shits and giggles [22:05] <avih> didn't work. same double frame rate [22:05] <avih> gonna try the demux now [22:06] <newguy123> does anyone know why this simple command doesn't work? ffmpeg -re -f rtsp -rtsp_transport tcp -i rtsp://media-us-2.soundreach.net/s lcn_sports.sdp -vcodec copy -acodec copy -f rtsp rtsp://127.0.0.1/stream.sdp [22:07] <relaxed> -simple [22:07] <avih> relaxed: the -demuxer lavpref option doesn't play the file. it just says "end of file" and that's it [22:07] <relaxed> newguy123: there's a space in the input name? [22:08] <relaxed> if it belongs there use quotes [22:08] <durandal11707> newguy123: what it does? and what it should instead do? [22:08] <relaxed> avih: can you stick a small sample of the video up somewhere? [22:08] <avih> relaxed: sure [22:10] <avih> relaxed: i use this command to encode it, and it plays fine in vlc, mpc and ffplay, but not in mplayer: ffmpeg -ss 1025 -i source.avs -t 30 -vcodec libx264 -preset superfast -vf scale=480:320:-1 -acodec libmp3lame -ab 96k -b:v 268k -f mpegts -bsf:v h264_mp4toannexb 6.ts [22:10] <avih> (uploading now) [22:11] <avih> relaxed: http://dropcanvas.com/jig0z and ignore the incorrect aspect ratio [22:13] <newguy123> for some reason ffmpeg seems to freeze after it displays the stream title audio and video codec [22:13] <avih> relaxed: ultimately i'm trying to create a playable file by concatenating few of those, but only mplayer plays the concatenation, so i need to either fix the concatenation, or make mplayer play it properly (regardless of concatenation) [22:14] <relaxed> avih: mplayer2 and mpv, both forks of mplayer, play it correctly [22:14] <avih> relaxed: you mean mpc? [22:15] <relaxed> the original mplayer may too...are you using the latest svn version? [22:15] <avih> (didn't try with mplayer2 yet) [22:15] <avih> quite recent. the one which comes with smplayer for windows from maybe a month or two ago [22:15] <relaxed> no, I mean http://mpv.io [22:15] <relaxed> I recommend mpv out of the three [22:17] <avih> iirc mplayer2 couldn't be built for windows for some months now, and i've never tryed mpv. are there pre-built windows mpv binaries? [22:18] <relaxed> http://mpv.srsfckn.biz/ [22:19] <hernantz> which could be the ffmpeg alternative to this command? avconv -i input.avi -c:v copy -c:a aac -strict experimental -b:a 13k output.mp4 [22:21] <relaxed> hernantz: they take the same options pretty much [22:23] <relaxed> avih: how does mplayer fit into this puzzle? [22:24] <avih> relaxed: mpv seems to work :) thanks! what do you mean about mplayer? [22:24] <avih> (mpv also plays the concatenation flawlessly! :) ) [22:25] <relaxed> more work here is done [22:25] <avih> relaxed: :) but what did you mean about mplayer? [22:26] <relaxed> never mind, you were just curious why it didn't work with mplayer. [22:26] <avih> i still am. do you know the answer? [22:27] <relaxed> most likely there's a bug in the version you're using. [22:27] <avih> though the concatenation of the mpeg2-ts encoded chunks doesn't seem to be a valid video file, which is suboptimal. [22:27] <avih> tried 2 mplayer versions, few months apart [22:27] <relaxed> are you using ffmpegconcat: [22:28] <relaxed> ffmpeg's concat: [22:28] <avih> mpv manages to work around it, but so far it's the only player capable of playing it [22:28] <avih> not ffmpeg concat. i'm trying to ceate individual encoded chunks to be concatenated on the fly without ffmpeg [22:29] <avih> but use ffmpeg to create those chunks [22:29] <BoomerBile> what file do i need to include to use av_open_input_file [22:31] <BoomerBile> ? [22:36] <BoomerBile> alright forget that question, this is a linking error.. hmm [22:44] <avih> relaxed: you got any idea how to encode chunks with ffmped which i could later concatenate into a valid file which most players could handle? [23:15] <sprite`> question: i'm having some audio distortion/skipping issues throughout videos when i use libfaac with h264...doesn't seem to happen with libmp3lame but if i use that the audio doesnt play in our flash player so i'm at kind of a loss (input/output here: http://i.imgur.com/28647TJ.png ) ... any ideas why this may be happening? [23:30] <sprite`> http://pastebin.com/dGQhPdga [23:31] <llogan> sprite`: does the input file play fine in your player? [23:32] <sprite`> yes [23:32] <sprite`> no audio skipping [23:33] <llogan> why do you not stream copy the audio instead of re-encoding? (-codec:a copy) [23:33] <llogan> why use -async 1? [23:34] <llogan> is there distortion with another AAC encoder? (-c:a aac -strict experimental) [23:34] <sprite`> i do use copy right now thats the only way ive been able to do it...but some of our raw audio is quite large and im trying to support lower bandwidths [23:34] <sprite`> async 1 i started doing because it was out of sync [23:35] <sprite`> i also a lot of times get inputs that dont support the containers [23:35] <sprite`> i havent tried that [23:37] <sprite`> trying -c:a aac -strict experimental [23:39] <sprite`> well the distortion seems to be gone with that haha [23:39] <sprite`> so is my libfaac messed up [23:43] <sprite`> plays in the player too [23:44] <sprite`> well thanks for the help...any reason i shouldnt use this as opposed to libfaac? [23:51] <rmelo> high guys [23:51] <rmelo> hi [23:51] <rmelo> lol [23:51] <rmelo> sorry [23:53] <rmelo> When i use the -ss xx:xx:xx -t xx:xx:xx the splitted video always starts at beggining of the movie [23:53] <rmelo> any idea why? [23:54] <llogan> sprite`: the quality isn't as good at the same bitrate. https://trac.ffmpeg.org/wiki/AACEncodingGuide [23:54] <llogan> what fplash player are you using? what browser? [23:55] <sprite`> it's custom made [23:55] <sprite`> firefox [23:55] <llogan> do other flash players have the same issue? [23:56] <llogan> you can quickly try jw player via their wizard [23:56] <sprite`> oh really [23:56] <llogan> http://www.longtailvideo.com/jw-player/wizard/ [23:56] <sprite`> theirs plays the mp3 ones [23:57] <llogan> i meant the aac stutter issue [23:57] <llogan> s/stutter/distortion [23:57] <sprite`> oh..yeah the stutter issue is the same even if i play it in vlc or in the browser itself lol [23:57] <llogan> can you provide the input file? [23:58] <rmelo> fflogger: sorry, here it comes [23:58] <rmelo> http://pastie.org/8387921 [23:58] <llogan> sprite`: is it distorted in ffplay too? [23:58] <sprite`> http://stream05.trillhd.com/trill/4636_480_90vl2ljf7t.mp4 [23:59] <llogan> rmelo: do you mean that ffmpeg seems to be ignoring your -ss? [23:59] <rmelo> exactly [00:00] --- Wed Oct 9 2013

1 0

ffmpeg-devel.log.20131007
by burek 08 Oct '13

08 Oct '13

[01:21] <kierank> do we have any samples with aac drc [01:53] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:630c005b879a: avcodec/flac_parser: export sample_rate also when PARSER_FLAG_COMPLETE_FRAMES is set [01:53] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:8f2386b5894d: avformat/oggparseflac: fix handling of old flac in ogg [01:53] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:38e13f55a5a3: ffmpeg: dont detect slight (0.1sec) backward moving dts as discontinuity [03:05] <Compn> kierank : Daemon404 probably does [09:47] <wm4> any explanation about that last commit? (ffmpeg: dont detect slight (0.1sec) backward moving dts as discontinuity) [10:14] <ubitux> michaelni: we seem to often use '|' as list separator in filters AVOptions [10:31] <cone-430> ffmpeg.git 03Paul B Mahol 07master:ce26a016cd34: avfilter/vf_phase: remove dead initialization [10:40] <michaelni> wm4, theres a sample that has slightly bad timestamps that needed this, (it was given to me with "private dont share" so i cant share it [10:40] <durandal_1707> you can share it in private [10:40] <michaelni> no [10:41] <durandal_1707> by share, he/she meant do not put on torrent [10:41] <cone-430> ffmpeg.git 03Paul B Mahol 07master:ad934bc353b2: avfilter/vf_drawtext: remove dead initialization [10:45] <ubitux> durandal_1707: i don't think so [10:46] <michaelni> he didnt litterally say "share" i think, i dont remember the exact wording. If someone wants to see the sample i can forward your request to who sent me the sample. [10:50] <durandal_1707> yes, that seems reasonable [11:13] <cone-430> ffmpeg.git 03Paul B Mahol 07master:62078f25ee9a: avcodec/dpx: return different error code for unsupported depths [11:16] <durandal_1707> michaelni: i think using gbrp for dpx was bad idea [11:16] <durandal_1707> it would be much simpler if 10 & 12 bits are done like 16 bit rgb [11:33] <durandal_1707> is there some dpx exporter that does not sucks? (imagemagick sucks) [11:37] <cone-430> ffmpeg.git 03Diego Biurrun 07master:9adbc3f3a177: bmv: Remove unused variable [11:37] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:57f0f86f72a1: Merge commit '9adbc3f3a1770fec9d24b8f5be3438a6c8e9e6a6' [11:42] <cone-430> ffmpeg.git 03Diego Biurrun 07master:ce1e8045e04e: x86: fdct: Employ more specific ifdefs [11:42] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:b67cb58520ff: Merge remote-tracking branch 'qatar/master' [11:51] <cone-430> ffmpeg.git 03Niv Sardi 07master:49ba6e56bd32: lavu/parseutils: add more resolutions [12:02] <cone-430> ffmpeg.git 03Lukasz Marek 07master:c21be5845fa6: doc/indevs: make pulse dev formatting consistent with other devices [12:03] Action: Daemon404 sudders at the sight of pulseauio [12:04] <ubitux> wm4, saste, problem with av_get_channel_layout2() is the clash with libav if they decide to use a different behaviour [12:04] <ubitux> name needs to be completely changed [12:05] <Daemon404> av_get_less_retarded_channel_layout() [12:05] <Daemon404> ? [12:05] <saste> ubitux, get_channel_layout() is already different from libav [12:05] Action: Daemon404 runs [12:06] <ubitux> saste: ok.. [12:07] <ubitux> i didn't follow what exactly changed in the behaviour [12:07] <wm4> once more I'm glad that I didn't use this API [12:07] <ubitux> (was it a documented behaviour anyway?) [12:09] <Daemon404> wm4, ;) [12:15] <saste> ubitux, yes it is documented in the doxy, not in the manual [12:15] <saste> i plan to update utils.texi once we finalize this [12:15] <saste> the change should be explained in the commit log, and is related to the opt channel layout utils [12:15] <ubitux> well i'd be more for a av_get_channel_layout2() then, but well [12:17] <saste> ubitux, this only affects people who specify number of channels with "N" rather than "Nc" [12:17] <saste> i'll add a warning in the code, so they will get a warning in that case [12:18] <ubitux> i'd like to see the patch first anyway :p [12:20] Action: durandal_1707 do not want to see single nit kind of review any more [12:21] <wm4> <rant>with proper APIs, this would be a non-issue</rant> [12:22] <durandal_1707> proper API is like holy grail [12:23] <saste> ubitux, i'll already sent a patch [12:23] <ubitux> i guess i'm blind [12:24] <saste> *I already ... [12:45] <saste> durandal_1707, can you provide a commandline to reproduce the interleave issue? [12:58] <durandal_1707> ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc -lavfi interleave -vcodec ffvhuff /tmp/x.nut [12:59] <durandal_1707> drops half of frames [13:04] <saste> durandal_1707, that's ffmpeg which is dropping frames in that case [13:05] <saste> since it doesn't like two frames with the same PTS [13:05] <durandal_1707> and how to fix that? [13:05] <saste> durandal_1707, there is nothing to fix, that's how ffmpeg is supposed to behave [13:05] <saste> what's your objective? [13:06] <durandal_1707> what I should do to let ffmpeg not drop frames? [13:06] <durandal_1707> what about this one: [13:06] <durandal_1707> ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc -lavfi interleave,stereo3d=al:sbsl -vcodec ffvhuff /tmp/x.nut [13:06] <wm4> frames with same PTS doesn't seem to make sense? [13:07] <durandal_1707> you switch between both very fast when rendering [13:08] <wm4> that doesn't make sense either [13:09] <durandal_1707> maybe for you... [13:09] <Daemon404> [12:07] <@durandal_1707> you switch between both very fast when rendering [13:10] <Daemon404> this violates the definition of PTS [13:10] <durandal_1707> citation needed [13:11] <wm4> one frame would be displayed for the duration of 0 seconds, i.e. not at all [13:11] <saste> ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc,setpts=PTS+10 -lavfi interleave,stereo3d=al:sbsl -vcodec ffvhuff -y /tmp/x.nut [13:11] <saste> but this is not yet working [13:11] <wm4> so why encode that frame (or why was that pts even generated?) [13:12] <saste> so the question is why ffmpeg is not happy with that? [13:27] <durandal_1707> doing gray8->gray16->gray8 changes brightness [13:28] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:b8866783c6a8: avfilter/lswsutils: dont override the default scaler [13:33] <durandal_1707> hmm brightness is actually same (i was using different vo) [13:36] <saste> ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc,settb=1/50,setpts=PTS+1 -lavfi interleave,stereo3d=al:sbsl,showinfo -vcodec ffvhuff -vsync passthrough -y /tmp/x.nut -loglevel debug [13:37] <saste> durandal_1707, looks like stereo3d is wiping the timestamps [13:46] <durandal_1707> why swscale doesn't support only red/green/blue/alpha formats? [13:47] <durandal_1707> saste: perhaps because frame_rate is not set? [13:50] <saste> durandal_1707, with vsync passthrough that should not be a problem [13:50] <saste> may be a stereo3d issue? [13:52] <saste> why pts = pts_time = 0 in the output? [13:58] <cone-430> ffmpeg.git 03Paul B Mahol 07master:e745dc2d6f3b: avcodec/dpx: refactor pixel format selection [13:58] <cone-430> ffmpeg.git 03Paul B Mahol 07master:23824b9698bb: avcodec/dpx: support for 8 and 16 bit luma only files [14:08] <durandal_1707> michaelni: did you backported a27227d401adf12 to relevant branches? [14:16] <michaelni> just cherry picked to 2.0 & 1.2 locally, i dont maintain older branches, feel free to backport to older or take over their maintaince [14:19] <durandal_1707> i barely maintain master [14:56] <durandal_1707> saste: its because fps is not set [14:57] <saste> how's that? [14:58] <saste> durandal_1707, anyway, how should it be fixed? [14:58] <saste> my guess is that there is something wrong in stereo3d, since it's not setting output PTSs [14:59] <durandal_1707> interleave set fps to 1/0 [15:03] <saste> durandal_1707, yes, that means "undefined" [15:06] <Daemon404> isnt that 0/1 [15:07] <Daemon404> not 1/0 [15:09] <durandal_1707> saste: because its 1/0 ts_unit is 0 [15:09] <durandal_1707> workaround is using fps filter after interleave, [15:12] <saste> Daemon404, 7b42036b [15:12] <saste> i wonder if that was a typo from Nicolas [15:13] <Daemon404> saste, allowing 1/0 is quite bad idea [15:13] <Daemon404> only bad things can come from /0 [15:14] <saste> durandal_1707, yes singularities are bad [15:14] <saste> durandal_1707, shouldn't you add an exception to handle 1/0? [15:14] <durandal_1707> to what? [15:14] <saste> stereo3d [15:15] <durandal_1707> but to what? [15:15] <Daemon404> [14:14] <@saste> durandal_1707, shouldn't you add an exception to handle 1/0? --- this is retarded [15:15] <Daemon404> the proper fix is not have /0 [15:15] <Daemon404> in the first place [15:15] <saste> 1/0 == undefined [15:15] <saste> the frame rate is not always defined [15:15] <Daemon404> [14:13] <@Daemon404> only bad things can come from /0 [15:16] <Daemon404> lets allow accidental division by 0! [15:16] <Daemon404> because of a poorly hosen special constant [15:16] <Daemon404> and litter the coebse with checks [15:16] <Daemon404> but we forget? UH OH [15:16] <wm4> why does the filter chain have the concept of a framerate? [15:16] <saste> wm4, was introduced by Anton [15:16] <Daemon404> well that too [15:16] <wm4> curse him [15:16] <saste> that's required for several reason [15:16] <saste> i think for introducing the fps filter in the first place [15:16] <durandal_1707> i wonder how to do it without f* with frame rate... [15:17] <Daemon404> libav has avg_frame_Rate only [15:17] <Daemon404> so it's always there [15:17] <Daemon404> afaik [15:19] <durandal_1707> for proper interleave operation you need same rates on all inputs anyway [15:19] <durandal_1707> and than output rate is just number of inputs * input rate [15:20] <durandal_1707> but even if they are different filter should not need to buffer frams [15:20] <durandal_1707> *frames [15:22] <Daemon404> so how do you define interleaving on vfr content [15:22] <Daemon404> since libav* have no concept of vfr/cfr at all [15:22] <Daemon404> seems liek a total crapshoot. [15:22] <durandal_1707> i think framesync could allow to ignore pts... [15:22] <Daemon404> cant filters set the fps? [15:22] <durandal_1707> Daemon404: the filters start counting from 0 [15:23] <Daemon404> you could ignore the timestamps and take an input fps directly to the plugin [15:23] <Daemon404> instead of screwing with the whole framework [15:30] <durandal_1707> yadif=1 doesn't use this approach, i wonder what solution is really correct [15:30] <wm4> did MPlayer yadif even have correct pts handling [15:31] <wm4> ret |= vf_next_put_image(vf, dmpi, pts /*FIXME*/); [15:31] <wm4> haha. [15:31] <wm4> pts += vf->priv->buffered_i * .02; // XXX not right [15:31] <wm4> looks like it didn't [15:32] <durandal_1707> i amazed by such XXX/FIXME that do not say why something needs fixing [15:32] <Daemon404> welcome to mplayer [15:38] <durandal_1707> libavcodec//cabac.c: flush_put_bits(&c->pb); //FIXME FIXME FIXME XXX wrong ---> the king [15:39] <Daemon404> rofl [16:21] <saste> the fixme is there since forever (first commit in 2003) [16:22] <saste> so basically nobody felt the need/urge to fix it, and never will [16:27] <JEEB> reminds me of VSFilter [16:28] <JEEB> that thing has had a fixme: b0rken message that pops up during compilation for like ten years, too [16:30] <Daemon404> [15:22] <@saste> so basically nobody felt the need/urge to fix it, and never will [16:30] <Daemon404> i think you mean [16:30] <Daemon404> nobody knows *what* to fix [16:30] <Daemon404> since it doenst give any clue [17:10] <teratorn> is there any api for telling how many data planes are expected by a pix format? [17:11] <durandal_1707> yes there is [17:11] <kierank> there's a struct [17:11] <teratorn> oh what file is that in? [17:12] <wm4> libavutil/pixdesc.h [17:12] <durandal_1707> av_pix_fmt_count_planes [17:12] <teratorn> very cool, ty :) [19:51] <michaelni> BBB, ubitux it seems nasm doesnt like the vp9 code (http://fate.ffmpeg.org/log.cgi?time=20131007121022&log=compile&slot=x86_64-…) [19:52] <durandal_1707> it could be nasm bug? [19:52] <Compn> its undefined reference [19:52] <Compn> so probably missing include ? [19:53] <Compn> weird it works in yasm then... [19:54] <ubitux> the undefined are linker errors Compn [19:54] <Compn> oh i'm in the wrong part [21:41] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:abf2d53d807c: avcodec/options_table: add field_order [21:41] <cone-430> ffmpeg.git 03Lenny Wang 07master:2779b7b30a0d: avfilter/deshake [21:44] <durandal_1707> michaelni: can you at least eddit commit logs of stuff you push? [21:45] <durandal_1707> michaelni: have you tested deshake patch? [22:03] <michaelni> durandal_1707, yes i tested it, the filter works as well as before [22:04] <michaelni> also the code was obviously buggy [22:18] <cone-430> ffmpeg.git 03Michael Niedermayer 07master:728bb910ec04: avfilter/vf_deshake: fix block_contrast() lower brightness value [00:00] --- Tue Oct 8 2013

1 0

ffmpeg.log.20131007
by burek 08 Oct '13

08 Oct '13

[00:36] <MedDev> Hello, I'm trying to record my screen and the output of my sound card at the same time. It works, but only for 20 seconds then the sound cuts out. ffmpeg is ran using this http://pastebin.com/YPcD1Mtm (includes output of ffmpeg). [00:49] <klaxa> MedDev, i guess your video will also be speed up by factor ~2? [00:49] <MedDev> klaxa, yes [00:50] <klaxa> your overall system performance is only able to provide 14 frames per second (just as the log says), either decrease the input framerate to something you know your system can provide, or don't specify it at all and take as many (variable frame rate) frames your system can output [00:51] <MedDev> ok :) [03:31] <norbert_> hi, I'm trying to convert a 1920x1040 video to 1920x1080 [03:31] <norbert_> I'm using this command: ffmpeg -r 23.976 -i 1920x1040.mkv -vf scale=1920:1080 -b 4000k -ab 160k -r 60 -acodec libvo_aacenc out.mp4 [03:32] <norbert_> ffmpeg version N-54234-gef90639 Copyright (c) 2000-2013 the FFmpeg developers built on Jun 28 2013 05:12:50 with gcc 4.6 (Debian 4.6.3-1) [03:32] <norbert_> and I get this error: Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [03:32] <norbert_> it also says: [libvo_aacenc @ 0x9d7ba20] Unable to set encoding parameters [03:32] <norbert_> any ideas/suggestions? [03:33] <norbert_> somewhat related, I'm also wondering why anyone would create a video that's 1920x1040, seems like really weird dimensions to me [03:49] <klaxa> norbert_ some people are weird, also if the mkv already has aac audio, why not copy that stream? apart from that the recommended aac encoder is libfdk-aac [03:53] <norbert_> this appears to work: ffmpeg -r 23.976 -i 1920x1040.mkv -vf scale=1920:1080 -b 4000k -r 60 -acodec copy out.mp4 [03:54] <klaxa> you also shouldn't have to define the input framerate [03:54] <norbert_> even with -ab 160k it works [03:54] <norbert_> so basically it didn't accept -acodec libvo_aacenc [03:54] <norbert_> framerate -> ok [03:54] <klaxa> mmmhh if you copy the codec it won't apply any bitrate stuff [03:55] <klaxa> given the framerate is it anime by any chance? [03:56] <norbert_> not an anime [03:56] <klaxa> because in that case i'd like to know what group releases in 1920x1040 so i can avoid them if that's not the original resolution [03:56] <klaxa> ok [03:56] <norbert_> do you know of any way to preview a .mp4 file before the whole reencoding is finished? [03:57] <norbert_> totem plays the audio not video, vlc and mplayer do nothing [03:58] <norbert_> I don't understand your "if you copy the codec it won't apply any bitrate stuff" [03:58] <norbert_> "-acodec copy" means I copy the audio codec, right? [03:58] <norbert_> and audio has nothing to do with bitrate stuff [03:59] <norbert_> I'm going to keep the input bitrate just in case [04:01] <norbert_> anyways, seems to work fine now [04:01] <norbert_> will take a couple of days though :P [04:03] <norbert_> "Aspect Ratio: 1.85 : 1" [04:03] <norbert_> according to IMDb [04:03] <norbert_> so apparently that's how it was released [04:06] <norbert_> "The most common aspect ratios used today in the presentation of films in movie theaters are 1.85:1 and 2.39:1." according to Wikipedia: http://en.wikipedia.org/wiki/Aspect_ratio_%28image%29 [04:07] <norbert_> US widescreen cinema standard [04:12] <klaxa> <norbert_> and audio has nothing to do with bitrate stuff yeah but if you use -b:a 160k it would be audio bitrate [04:12] <klaxa> and that doesn't get applied if you use -c:a copy [04:13] <norbert_> yeah, okay [10:15] <bouba> hi, someone can help me find a better method to capture regular jpeg thumbs on h264 videos ? "-r 1/20" seems to have behaviors [10:15] <bouba> pleease [10:17] <bouba> durandal_1707: no problem with the cli command...but the capture interval is not static [10:18] <bouba> i guess "-ss" is better to capture [10:18] <bouba> but i dont know the total lenght of the videos [10:19] <bouba> THUMBCMD="ffmpeg -y -i $INPUT -an -r 1/20 -f image2 $INPUT_%05d.jpeg" [10:20] <bouba> ffmpeg sends no error [10:20] <bouba> and creates the thumbs according to the regex [10:25] <durandal_1707> replace '-r 1/20' with '-framerate 1/20' [11:01] <bouba> durandal_1707: -r is not an alias of -framerate ? [11:01] <durandal_1707> nope [11:04] <bouba> thx durandal, the behaviour seems different and correct this time [11:52] <bouba> durandal_1707: "-vf fps=fps=1/20" seems to work better for regular interval capture [11:53] <bouba> i don't really understand why [12:00] <durandal_1707> bouba: fps filter drops and dupes random frame to make video cf [12:00] <durandal_1707> *cfr [12:01] <bouba> crf ? [12:01] <durandal_1707> cfr [12:01] <durandal_1707> constant frame rate [12:01] <bouba> so the capture won't be at regular interval... [12:02] <bouba> found here : http://ffmpeg.org/trac/ffmpeg/wiki/Create%20a%20thumbnail%20image%20every%2… [19:17] <jure> could someone advise me on how to properly downmix a 5.1 song to a 2ch according to a well-established standard a la dolby digital [19:17] <jure> with ffmpeg ofc [19:23] <llogan> do you want just the FL and FR channels, or do you want to combine all channels to their respective L or R channel? [19:24] <jure> all of them [19:27] <llogan> does simply using -ac 2 do what you want? (i'm not sure if it adheres to the standard stuff you mentioned) [19:29] <jure> ffmpeg will probably select only L and R. but I'll try. [19:32] <durandal_1707> perhaps you need -matrix_encoding? [19:32] <jure> yeah [19:33] <jure> so ffmpeg definitely uses more than the L and R channels with -ac 2, but there's also needless attenuation of the signal. [19:34] <durandal_1707> -matrix_encoding dolby/dplii [19:34] <jure> now I wish I knew which matrix ffmpeg uses by default. [19:36] <durandal_1707> also you can mix with different coefficients using audio filters.... [21:07] <StamEhad940> Hi! I have a muxed UDP MPEG-2 multi-program transport stream. How can I used ffmpeg to dump this stream to a .ts file without any transcoding? [21:16] <StamEhad> Hi! I have a muxed UDP MPEG-2 multi-program transport stream. How can I used ffmpeg to dump this stream to a .ts file without any transcoding? [21:20] <jure> durandal_1707: how do I tell ffmpeg to phase-shift? [21:25] <durandal_1707> jure: what you mean by phase-shift? [21:25] <jure> durandal_1707: see for example http://en.wikipedia.org/wiki/Matrix_decoder#Dolby_Pro_Logic_II_matrix_encod… [21:37] <codepython777> how can i fix this :[dshow @ 000000000030e040] Could not find video device. ? [21:37] <codepython777> am trying this on windows : ./bin/ffmpeg.exe -f dshow -i video="UScreenCapture" output.flv [22:07] <jure> use BandiCam? [22:08] <codepython777> ? [22:08] <codepython777> jure: does it use ffmeg? [22:09] <jure> no, doesn't need ffmpeg [22:09] <codepython777> jure: I need a 35 minute recording, and dont want to be monkeying around patching/stitching [22:09] <codepython777> was hoping to use ffmpeg [22:09] <jure> well that's great, bandicam will do that job [22:10] <codepython777> jure: only 10 minutes [22:10] <relaxed> codepython777: did you look at https://trac.ffmpeg.org/wiki/How%20to%20capture%20a%20webcam%20input [22:10] <relaxed> wait, that's not right :/ [22:11] <relaxed> https://trac.ffmpeg.org/wiki/How%20to%20grab%20the%20desktop%20%28screen%29… [22:11] <codepython777> relaxed: that uses the camera on my laptop, i need to capture the screen [22:11] <relaxed> yeah, look at the second link [22:12] <codepython777> relaxed: ffmpeg -f dshow -i video="UScreenCapture" output.flv - gives - how can i fix this :[dshow @ 000000000030e040] Could not find video device. ? [22:12] <codepython777> (3:37:43 PM) codepython777: am trying this on windows : ./bin/ffmpeg.exe -f d [22:12] <codepython777> thats where i started and came here [22:12] <relaxed> did you try listing the devices? [22:12] <codepython777> no i did not [22:12] <durandal_1707> than what you wait? [22:13] <relaxed> do so [22:13] <codepython777> with what command? [22:13] <relaxed> maybe it's not "UScreenCapture" for your system [22:13] <relaxed> read the wiki I just pasted [22:14] <codepython777> http://bpaste.net/show/U2MtCuW4L7H0Et8X7nM1/ - here is the list [22:14] <codepython777> how do i capture my screen and speakers to an flv/mp4? [22:15] <durandal_1707> perhaps you instally some capturer that export itself via dshow ... [22:16] <codepython777> I need to ? [22:17] <codepython777> no hope without that? [22:20] <fazias> http://obsproject.com/ (you can just save to file instead of stream) [22:20] <fazias> on windows* [22:21] <codepython777> fazias: have you used it on windows? mp4 outut? [22:22] <fazias> yes [22:22] <fazias> you can also fully customize the x264 cmd line parameters if you feel like it, just need to disable some stuff [22:25] <codepython777> fazias: could not initialize directX [22:25] <codepython777> error -> Close [22:26] <fazias> :D [22:26] <fazias> good luck then [22:26] <codepython777> lol [22:31] <GoaLitiuM> codepython777: are you trying to use 64-bit binaries of ffmpeg with UScreenCapture? [22:31] <GoaLitiuM> if the filter is 32-bit only, then you need to use 32-bit ffmpeg [22:34] <codepython777> GoaLitiuM: I downloaded the 64-bit binaries for windos [22:34] <codepython777> how do i figure out what filter bit i am using? [22:34] <GoaLitiuM> then try downloading 32-bit ffmpeg binaries [22:34] <GoaLitiuM> i'm fairly sure most directshow devices only have 32-bit version available [22:36] <codepython777> ic [22:36] <codepython777> http://ffmpeg.zeranoe.com/builds/ [22:39] <codepython777> GoaLitiuM: http://bpaste.net/show/c5q9AB6WsWHw4YzYhIlk/ [22:39] <GoaLitiuM> which one is that [22:40] <codepython777> output of this - > ffmpeg -list_devices true -f dshow -i dummy [22:40] <codepython777> downloaded the 32- bit [22:40] <codepython777> static one- top [22:40] <GoaLitiuM> have you installed uscreencapture? [22:41] <codepython777> nope [22:41] <GoaLitiuM> well there you have it [22:41] <GoaLitiuM> http://www.videohelp.com/tools/UScreenCapture [22:41] <codepython777> downloading now [22:42] <GoaLitiuM> OBS didn't work? [22:43] <codepython777> GoaLitiuM: Nope [22:43] <codepython777> GoaLitiuM: how do i capture the speakers to the mp4? [22:43] <codepython777> ./ffmpeg.exe -f dshow -i video="UScreenCapture":audio="Microphone" output.mp4 [22:43] <codepython777> video=UScreenCapture:audio=Microphone: Input/output error [22:44] <codepython777> seems like the screen will get captured with UScreenCapture [22:45] <GoaLitiuM> i recommend using fraps or dxtory if you want to capture both screen and audio too [22:45] <codepython777> GoaLitiuM: will any of them work for a 35 minute recording without paying? [22:46] <GoaLitiuM> nope :/ [22:46] <codepython777> so how do i do audio ith ffmpeg? [22:46] <GoaLitiuM> next best option would be OBS/FFsplit/free version of Xsplit, and only record locally [22:47] <codepython777> obs didnt work [22:47] <GoaLitiuM> how old hardware you have on your system? [22:47] <GoaLitiuM> OBS requires DX10 support [22:48] <codepython777> this is a 2 year old laptop [22:55] <relaxed> codepython777: are you using the most recent version of ffmpeg? http://ffmpeg.zeranoe.com/builds/ [22:55] <codepython777> yes [22:56] <codepython777> tried both 32 and 64 bit - GoaLitiuM's help = screen capture possible it seems [22:56] <codepython777> but i need to capture speakers as well [22:56] <GoaLitiuM> perhaps with this: http://sourceforge.net/projects/virtualaudiodev/ [00:00] --- Tue Oct 8 2013

1 0

ffmpeg-devel.log.20131006
by burek 07 Oct '13

07 Oct '13

[00:03] <JEEB> <JEEB> doom9 is compromised [00:03] <JEEB> <JEEB> disabling JS seems "OK" for now, but there is no way of knowing if they got access to an admin as well, and not only a moderator [00:03] <JEEB> <JEEB> if you had opened a (sub)forum and it took a while and had an announcement, and you had JS enabled, you can be relatively sure that your username/password combo got sent out to a 3rd party server [00:44] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:6853e40106ca: avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits [01:45] <Compn> put doom9 news on the site [01:45] <Compn> luckily i use throwaway passwords on most things :) [01:46] <Compn> now wheres lou so he can make a tweeet about it :P [03:33] <rcombs> who can I pay, and how much, to fix the dithering-causes-banding-and-discoloration issue? [03:52] Action: rcombs wonders if a Riemersma dither could be done fast enough [03:59] <Compn> rcombs : open a trac ticket with a sample and command line [03:59] <Compn> and slap a bounty on it [04:04] <rcombs> Compn: already got a ticket (someone else filed, I added some sample images) [04:04] <rcombs> not sure how best to stick a bounty on an issue [04:05] <Compn> "$500 to person who fixes" ? [04:15] <rcombs> ^ relevant [04:16] <Compn> sounds good [04:16] <Compn> i'm kind of not paying attention here :D [04:16] <Compn> ehe [04:19] <Compn> rcombs : btw can you do --disable-optimizations or possibly --disable-yasm and rerun your tests ? [04:19] <rcombs> Compn: sure, gimme a minute [04:19] <Compn> wonder if its asm related [04:21] Action: rcombs facepalms; forgot that was a configure option [04:22] <Compn> yeah one of those [04:22] <Compn> recompile [04:23] <rcombs> yup [04:23] <Compn> so uh... can you make a small sample file [04:23] <rcombs> oh joy, a compile error [04:23] <Compn> i dont see one ... [04:23] <Compn> oh there it is [04:23] <Compn> vtst0.mp4 [04:24] <rcombs> Compn: also, if you take the rgb48 PNG I snapshotted, it makes a good sample source [04:24] <Compn> ok the png source shows the same thing ? :) [04:25] <rcombs> the RGB48 PNG looks good; using ffmpeg to convert the RGB48 to RGB24 shows the issue [04:26] Action: rcombs ran out of registers during register allocation [04:30] <rcombs> apparently clang doesn't like something or other that shows up when optimizations are disabled [04:31] <rcombs> and& gcc apparently screws up its output [04:32] Action: rcombs checks out n2.0.1; still has issue [04:33] <rcombs> libavcodec/x86/dsputil_mmx.c:266:9: error: ran out of registers during register allocation <-- joy [04:33] <Compn> add -fomit-frame-pointer ? maybe ? [04:33] <Compn> i forgot clang [04:33] <Compn> its old error i mean, but i dont know if that works with clang [04:34] Action: rcombs brings himself back to master [04:35] <rcombs> nope [04:37] Action: rcombs tries some more stuff [04:37] <rcombs> there we go [04:38] <rcombs> for the record, my successful configure line was ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-openssl --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --prefix=/usr/local --enable-static --enable-shared --cc="clang -m32 -fomit [04:38] <rcombs> -frame-pointer" --disable-optimizations --disable-yasm --disable-asm [04:39] <Compn> cool [04:43] <rcombs> http://dl.dropboxusercontent.com/u/46348978/Madoka_Snapshot_02_24_noasm.png <-- no change in output [05:11] <Compn> ok thanks for checking [05:11] <Compn> was a dumb idea [05:12] <Compn> theres dithering in there ? [05:12] Action: Compn doesnt even know what dithering means [05:12] <Compn> looks like luma is off [05:12] <rcombs> it is, yup [05:13] <rcombs> dithering (in this case) means improving the quality of color quantization by& it's hard to explain; the Wikipedia article has some great images demonstrating this [10:53] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:ccf96f8c642d: swscale/options: switch default to bicubic [11:12] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:931a4b433dce: avfilter/vf_scale: change the default scaler to bicubic [11:24] <cone-291> ffmpeg.git 03Luca Barbato 07master:4272bb6ef153: doxy: Document avlog [11:24] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:90cecd3c9b6b: Merge commit '4272bb6ef1533846a788c259cc498562d0704444' [11:30] <cone-291> ffmpeg.git 03Diego Biurrun 07master:0b8b2ae5e93d: x86: xviddct: Employ more specific ifdefs [11:30] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:d0b27036763f: Merge commit '0b8b2ae5e93d616c2ece59f7175f483154cff918' [11:36] <cone-291> ffmpeg.git 03Diego Biurrun 07master:258414d07718: x86: fdct: Initialize optimized fdct implementations in the standard way [11:36] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:7fb123429e6c: Merge commit '258414d0771845d20f646ffe4d4e60f22fba217c' [11:49] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:6b3ff6f91a53: swscale: provide a default scaler if none is set [11:49] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:5710f55e490d: Merge commit '6b3ff6f91a535d6383f41ca7bdf760165dcb6015' [11:55] <cone-291> ffmpeg.git 03Diego Biurrun 07master:2ddb35b91131: x86: dsputil: Separate ff_add_hfyu_median_prediction_cmov from dsputil_mmx [11:55] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:c86955d24a43: Merge commit '2ddb35b91131115c094d90e04031451023441b4d' [11:59] <ubitux> BBB: in the itxfm_wrapper(), why is the input block reset to 0? [12:00] <ubitux> is it simpler for the caller then? [12:08] <ubitux> well it seems useful anyway. [12:15] <ubitux> there are not much additionnal int in ssse3, but some look very useful to me& [12:33] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:90e15e340d97: oggparsevorbis: return meaningful errors [12:33] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:4634871b513c: Merge remote-tracking branch 'qatar/master' [12:40] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:3ee796760a89: avfilter/vf_scale: dont duplicate default. [12:50] <BBB> ubitux: someone has to reset the block contents to zero [12:51] <BBB> ubitux: otherwise the block reader has to do it after reading the end-of-block [12:51] <ubitux> is it always necessary? [12:51] <BBB> only for the ones that are non-zero [12:51] <ubitux> yeah well, ok [12:51] <BBB> so you really only need to memset(block, 0, sizeof(*block)*eob); [12:51] <BBB> but I didn't do that :) [12:51] <BBB> feel free to change the C code that way [12:51] <ubitux> ah the eob thing [12:51] <ubitux> ok [14:52] <ubitux> SUMSUB_BA should be used with a temporary register when available right? [15:02] <durandal_1707> wow. doom9 hacked [15:30] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:4916088a9302: fate: fix dpx on big endian [15:33] <JEEB> durandal_1707, at the very least a moderator's account was compromised, in the worst case they also got access to the server [15:34] <JEEB> the most visible part of it was that JS script embedded in an announcement that was shown on every subforum [15:35] <durandal_1707> so it was because forum software was not updated? [15:36] <JEEB> well, the forum software letting anyone put a script tag into a title (or anything for that matter) is fucked up, but I have no idea if the updates would have fixed that [15:36] <JEEB> I think they were more or less up-to-date with the 3.x branch [15:36] <JEEB> which is, granted, quite old [16:48] <durandal_1707> why was lavc minor not bumped when vp9 was added? [16:49] <ubitux> likely a mistake [16:49] <durandal_1707> well not minor but micro [16:49] <durandal_1707> minor is bumped when new codec id is added [16:50] <durandal_1707> or whatever [16:52] <durandal_1707> the commit check script (which check for tabs iirc) could check for this too [16:56] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:efff7564d4d6: configure: filters should select the parts they need not use deps [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:eec833b10dda: avcodec/parser: reset indexes on realloc failure [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:510da4fe2b45: apichanges: fix 2 wrong hashes [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:ba7cd748c179: doc/APIchanges: List merge commit hashes and version numbers [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:ff29290e265d: update all trac links to use the trac subdomain [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:0dff3171ce10: avcodec/rpza: Perform pointer advance and checks before using the pointers [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:38ca79b04d9a: jpeg2000: check log2_cblk dimensions [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:1b05b0005bec: matroska_read_seek: Fix used streams for subtitle index compensation [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:e3d643cf7587: avcodec/pngdsp: fix (un)signed type in end comparission [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:90ee388b2883: avcodec/dsputil: fix signedness in sizeof() comparissions [17:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:bd66456866e9: avcodec/ffv1enc: update buffer check for 16bps [18:17] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.8:a248117f2618: update for 0.8.15 [18:21] <durandal11707> Compn: why you put that on ffmpeg news? [18:47] <cone-291> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.8.15': unknown revision or path not in the working tree. [18:47] <cone-291> Use '--' to separate paths from revisions [18:47] <cone-291> refs/tags/n0.8.15:HEAD: configure: filters should select the parts they need not use deps [19:06] <cone-291> ffmpeg.git 03Vicente Jimenez Aguilar 07release/0.7:c10582e703c5: doc: filters: Correct BNF FILTER description [19:06] <cone-291> ffmpeg.git 03Paul B Mahol 07release/0.7:537c17385350: smacker: fix off by one error [19:06] <cone-291> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:e2d529424f3f: Fix type of shared flac table ff_flac_blocksize_table[]. [19:06] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:b6783b882682: indeo5: update AVCodecContext width/height on size change [19:06] <cone-291> ffmpeg.git 03Luca Barbato 07release/0.7:ee6b868ac835: oggdec: make sure the private parse data is cleaned up [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:8489c0599f0e: update year to 2013 [19:06] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:f13f6f82c63f: wmadec: require block_align to be set. [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:7e6625a9afbe: xxan: fix invalid memory access in xan_decode_frame_type0() [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:c313f3160ae1: Update RELEASE file for 0.7.8 [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:7d4c38d58d01: h264: check for luma and chroma bit depth being equal [19:06] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:d86a5ce03f69: vorbisdec: Error on bark_map_size equal to 0. [19:06] <cone-291> ffmpeg.git 03Luca Barbato 07release/0.7:5bfa208e6565: matroskadec: request a read buffer for the wav header [19:06] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:5025dbc577c9: wmaprodec: return an error, not 0, when the input is too small. [19:06] <cone-291> ffmpeg.git 03Justin Ruggles 07release/0.7:d785f6940144: shorten: validate that the channel count in the header is not <= 0 [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:a694b2b15865: shorten: set invalid channels count to 0 [19:06] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:5ebb5a32bdd9: shorten: report meaningful errors [19:06] <cone-291> ffmpeg.git 03Luca Barbato 07release/0.7:6d4d186e9e9f: shorten: use the unsigned type where needed [19:07] <cone-291> ffmpeg.git 03Mans Rullgard 07release/0.7:c65763a2c677: lavf: fix arithmetic overflows in avformat_seek_file() [19:07] <cone-291> ffmpeg.git 03Xi Wang 07release/0.7:78889be3fb4b: atrac3: avoid oversized shifting in decode_bytes() [19:07] <cone-291> ffmpeg.git 03Xi Wang 07release/0.7:f8d3bb8961d2: flacdec: simplify bounds checking in flac_probe() [19:07] <cone-291> ffmpeg.git 03Mans Rullgard 07release/0.7:799000af702a: dsputil: fix invalid array indexing [19:07] <cone-291> ffmpeg.git 03Xi Wang 07release/0.7:530d10792de2: lzo: fix overflow checking in copy_backptr() [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:9c713f30e491: parser: fix large overreads [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:33c9e18b09f4: avfiltergraph: check for sws opts being non-NULL before using them. [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:96cf80609af6: dfa: check for invalid access in decode_wdlt(). [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:280998b13cb8: wmaprodec: require block_align to be set. [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:76c97f19634f: lavf: make sure stream probe data gets freed. [19:07] <cone-291> ffmpeg.git 03Kostya Shishkov 07release/0.7:f844cb9bced3: iff: validate CMAP palette size [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:9b052bfb8609: rv10: check that extradata is large enough [19:07] <cone-291> ffmpeg.git 03Anton Khirnov 07release/0.7:96481c5e1843: qdm2: check that the FFT size is a power of 2 [19:07] <cone-291> ffmpeg.git 03Luca Barbato 07release/0.7:053c19cd88b6: oggdec: fix faulty cleanup prototype [19:07] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:ade4f3e74635: aac: check the maximum number of channels [19:07] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/0.7:3197a9c4fa46: Update changelog for 0.7.8 release [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:d04dc7b5a78b: h264_cavlc: fix reading skip run [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:23d835f6113b: alacenc: Fix missing sign_extend() [19:07] <cone-291> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:2a1bebfc83db: Autodetect idcin only if audio properties allow decoding. [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:c997dcd38be7: mpegts: only reopen pmt_cb filter if its different from the previous. [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:f8c4dbe45ee2: mpeg12dec: avoid reinitialization on PS changes when possible. [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:afe09e490a30: Merge commit '5025dbc577c9a9e0109cb363ac630a9eeda6dc1d' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:0a41da3e9d20: Merge commit 'd785f6940144eb6ce4c24309ed034056b81395bc' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:44ebb2556d09: Merge commit '5ebb5a32bdd910a8afb316c51ed0b322f5600ae5' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:49d597f058a9: Merge commit '9c713f30e4913a28d93eb37ea5db7f62be4c0ef6' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:70a1182a4844: Merge commit 'f844cb9bced3148fca2db5bbb092929526108005' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:9b0736c08ac5: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:eec833b10dda: avcodec/parser: reset indexes on realloc failure [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:510da4fe2b45: apichanges: fix 2 wrong hashes [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:ba7cd748c179: doc/APIchanges: List merge commit hashes and version numbers [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:ff29290e265d: update all trac links to use the trac subdomain [19:07] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:0dff3171ce10: avcodec/rpza: Perform pointer advance and checks before using the pointers [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:38ca79b04d9a: jpeg2000: check log2_cblk dimensions [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:1b05b0005bec: matroska_read_seek: Fix used streams for subtitle index compensation [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:e3d643cf7587: avcodec/pngdsp: fix (un)signed type in end comparission [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:90ee388b2883: avcodec/dsputil: fix signedness in sizeof() comparissions [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:bd66456866e9: avcodec/ffv1enc: update buffer check for 16bps [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:a248117f2618: update for 0.8.15 [19:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:0ca3dbbfcef0: Merge tag 'n0.8.15' into release/0.7 [19:08] Action: durandal11707 gonna ban cone-291 [19:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/0.7:63d5db37cba3: update for 0.7.16 [19:10] <rcombs> 0.o [19:10] Action: rcombs didn't realize releases that old were still maintained at all [19:13] <michaelni> rcombs, they arent really "maintained", iam just making new releases as some stuff has accumulated in the branches [19:13] <rcombs> ah [19:16] <cone-291> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.7.16': unknown revision or path not in the working tree. [19:16] <cone-291> Use '--' to separate paths from revisions [19:16] <cone-291> refs/tags/n0.7.16:HEAD: configure: filters should select the parts they need not use deps [20:24] <ubitux> BBB: are we supposed to play with special instruction for the rounding 14-bit rounding, or you manually add/shift just like in C? [20:28] <BBB> ubitux: depends, most of the (a*x+b*y+round) >> shift should be done as such (pmaddwd for a*x+b*y, then paddd for round, and psrad+packuswb for shift -> word) [20:28] <BBB> ubitux: some of them (like ((a+b)*x+rnd)>>shift) can use pmulhrsw if a or b is zero [20:29] <BBB> ubitux: so e.g. for the dc only 4x4 case, you can use out=pmulhrsw(pmulhrsw(pmulhrsw(dc, 11585*2), 11585*2), 512) [20:30] <BBB> ubitux: which is the same as the *11585, rounded-shift 14, *11585, rounded-shift 14, rounded-shift 6 [20:30] <ubitux> ok i can use the builtin-16 inst just for the t0/t1 cases in idct func [20:31] <ubitux> thanks [20:31] <BBB> for the 4x4/8x8, you can also use the pmulhrsw for the case where a and b are both nonzero (in ((a+b)*x+rnd)>>shift) [20:31] <BBB> but that doesn't work for 16x16/32x32 [20:32] <ubitux> mmh [20:33] <BBB> 4x4: see line 890-891 [20:33] <BBB> 8x8: see line 924-925 and 942-943 [20:33] <BBB> in vp9dsp.c [20:33] <BBB> those can use pmulhrsw [20:34] <ubitux> yes the t0/t1 i was mentioning [20:34] <BBB> right [20:34] <BBB> anything that looks like that in the 4x4/8x8 [20:34] <ubitux> BBB: what's the range of the values block values btw? [20:34] <ubitux> they really are full 16-bits? [20:35] <BBB> 4x4: 14bits, 8x8: 15bits, 16x16/32x32: 16bit [20:35] <BBB> that's why it only works for the 4x4/8x8 [20:35] <BBB> for the 16x16/32x32, it can overflow in theory [20:35] <ubitux> ah i see [20:35] <BBB> (the add/sub) [20:35] <BBB> that's why it can only be done if a or b is zero [20:35] <BBB> not if both are nonzero [20:35] <BBB> because the sub/add is 17bits [20:35] <ubitux> ok [20:35] <BBB> for 4x4/8x8, that's not a concern [20:36] <BBB> (I think) [20:36] <ubitux> no risk of signed 32-bit overflow with the mult in 16/32 btw? [20:37] <BBB> shouldn't happen no [20:40] <durandal_1707> imagemagick is full of shit [20:50] <llogan> how do i show the changes in point releases in the online git browser? [20:51] <durandal_1707> changes as? [20:51] <llogan> as in the commits added for 0.8.15 from 0.8.14 for example [20:52] <llogan> s/from/since/ [21:06] <cptspiff> llogan: http://git.videolan.org/?p=ffmpeg.git;a=shortlog;hp=<startrev>;h=<endrev> [21:16] <llogan> cptspiff: thanks [21:16] <BBB> ubitux: oh where I said packuswb, I of course meant packssdw [21:16] <BBB> ubitux: silly me [21:16] <BBB> (it's dword->word, not word->byte) [22:22] <cone-291> ffmpeg.git 03Paul B Mahol 07master:8ec328668a7d: avcodec: add dpx parser [22:22] <cone-291> ffmpeg.git 03Paul B Mahol 07master:ee3d03bf4ec6: avcodec/dpx: make .long_name more descriptive [00:00] --- Mon Oct 7 2013

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ffmpeg.log.20131006
by burek 07 Oct '13

07 Oct '13

[02:37] <buhman> why do I get random (seemingly) unrelated segfaults in avformat_open_input() whenever I simply add any call to avcodec_alloc_frame() in my program? [02:37] <buhman> simplest testcase: http://ix.io/8m7/c [02:38] <buhman> if I comment line 19, everything works fine, I can av_dump_format(), everything is sunshine and rainbows [02:39] <buhman> if I don't, I get weirdness like: http://ix.io/8m8 [05:30] <Geha3> How to make x264 aware of sse cpu instructions on gentoo ? I tryed to configure use flags,but it did not work. [10:20] <hans_henrik> i want to remove all frames that are exactly equal to the last frame from a x264 mp4 file.. ideas? [11:12] <fahadash> Do you know at which positions those frames are that you want to delete ? [11:25] <hans_henrik> fahadash, nope i don't, btw seems the libavfilter "decimate" does what i want ^^ [11:35] <hans_henrik> 3.5M fooplain.wmv [11:35] <hans_henrik> 3.6M foo.wmv [11:35] <hans_henrik> the filter, supposed to remove duplicates, made the file bigger! o.0 [11:36] <hans_henrik> maybe the compression algorithm didnt like it [11:41] <fahadash> You can run another command to compress it [11:41] <fahadash> apply a different codec [11:47] <fahadash> hans_henrik_: https://trac.ffmpeg.org/ticket/1879 [13:38] <asido> I am attempting to use ffmpeg in iOS app, but I get linker errors wich doesn't make sense [13:38] <asido> http://ideone.com/C1vWy9 [13:40] <asido> nm output: http://ideone.com/AKedqP [13:40] <asido> why is so? [14:33] <Hurons> Hey [14:38] <Hurons> I am trying to do something that seems simple enough. But have not been able to wrap my head around it so far. If anyone could spare a moment or two, it would be much appreciated. Trying to capture audio and video from a Decklink card and transmit it on the network so I can pick it up and show (preview). And on demand pick it up a second time to record it. Using ffmpeg.exe on Windows to create [14:38] <Hurons> the network stream. Trying to pick up with vlc. [15:49] <asido> what's the status with ffmpeg on Xcode5 and iOS7 ? [18:13] <illusion> hi, cn ffmpeg stream video playlist to rmtp server ? if yes cn some one give me an example command line [22:54] <guideX> anyone know where I can get an already compiled ffmpeg binary for windows [22:54] <guideX> I got as far as installing mingw but I'm not sure what to do after [23:26] <guideX> ah ok cool thx [23:26] <guideX> perfect [00:00] --- Mon Oct 7 2013

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