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January 2015

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ffmpeg-devel.log.20150130
by burek 31 Jan '15

31 Jan '15

[00:05] <cone-588> ffmpeg.git 03Anshul Maheshwari 07master:93fac23b80e4: avcodec/ccaption_dec: Adding color and fonts for future use in screen [00:55] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:5008605bf25c: avcodec/hevc: Replace more mallocs by av_malloc_array() [00:55] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:2b215b7f5af0: avcodec/hevc: Check for av_malloc failure [01:10] <rcombs> is there a reason why segment.c keeps seg->list_pb open all the time when writing an HLS manifest or using a max list size? [02:55] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:305cbe76d327: avcodec/wma: remove unneeded #includes, there are no assert() only av_assert* [02:55] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:a73c4118092d: avcodec/aacdec: remove unneeded #include, theres no assert() in aacdec [10:03] <cone-864> ffmpeg.git 03Paul B Mahol 07master:b5004f32133f: avformat/nut: add GIF[0] [10:34] <cone-864> ffmpeg.git 03Paul B Mahol 07master:64815d1f0c78: avformat/nut: support WavPack [11:19] <durandal_1707> mplayer broken with yuv444p? [11:24] <wm4> durandal_1707: in what way? unless it broke, it should probably work [11:25] <JoshX> Good morning :) [11:53] <cone-864> ffmpeg.git 03Stefano Sabatini 07master:40b026b1d0a3: doc/muxers/segment: document strftime option [11:54] <JoshX> ah nice [11:54] <JoshX> i did trigger something ;) [12:01] <ubitux> http://b.pkh.me/out-dither-bayer-scale-0.gif http://b.pkh.me/out-dither-bayer-scale-1.gif http://b.pkh.me/out-dither-bayer-scale-2.gif http://b.pkh.me/out-dither-bayer-scale-3.gif scale=1 looks best, right? [12:02] <ubitux> http://pastie.org/pastes/9873593/text i'm using this [12:03] <ubitux> erm scaled twice, i'm an idiot [12:03] <ubitux> let me fix that.. [12:04] <ubitux> file updated, scale=2 looks the best to me [12:04] <ubitux> code now being http://pastie.org/pastes/9873596/text [12:05] <ubitux> i need that minus to avoid too much luma [12:05] <ubitux> but i'm also uncomfortable about scaling down the dither value directly, i should probably upscale the source, add the dither, and then scale down the whole thing i suppose? [12:10] <nevcairiel> ubitux: 2 and 3 have banding, 3 more so than 2, but 2 still shows it [12:11] <nevcairiel> dunno if a little less pattern is worth the banding :d [12:24] <ubitux> nevcairiel: should i make it configurable? [12:24] <ubitux> so you would prefer scale=1? [12:25] <ubitux> the pattern looks more ugly than the banding to me [12:26] <durandal_1707> imho why not allow all of them? [12:26] <ubitux> yeah i guess i'll add an option [12:26] <ubitux> i'm just wondering if i'm really implementing bayer here [12:26] <ubitux> since apparently the scale=0 is kind of the "reference" [12:28] <wm4> huh, stream_codec_tag got deprecated? [12:29] <wm4> and no decoders use it anymore... [12:30] <durandal_1707> you need it? [12:31] <nevcairiel> nah, he is just confused that its useless now [12:45] <saste> JoshX, you the guy of the %t patch, right? [12:58] <Diego___> Hello! [13:59] <JoshX> oh yes i am but saste is gone :-/ [14:04] <michaelni> kierank, welcome back, and sorry for myself being grumpy&unfriendly yesterday [14:09] <iive> durandal_1707: are you still looking for soft telecine samples? [14:28] <durandal_1707> iive: i already got one from michaelni [14:29] <iive> ok. sorry i could help you at the time. btw, i guess you've figured out why mplayer says the filter is functional only with mencoder. (because mplayer changes fps when it detect soft telecine) [14:58] <cone-864> ffmpeg.git 03Christophe Gisquet 07master:1fa637d2ef00: ffmpeg: free_input_threads requires HAVE_PTHREADS [16:38] <cone-864> ffmpeg.git 03Paul B Mahol 07master:80cbf1372559: avformat/rtpdec_qdm2: change assert to av_assert0() [16:38] <cone-864> ffmpeg.git 03Paul B Mahol 07master:5274860fe275: ffmpeg: remove unused header [16:38] <cone-864> ffmpeg.git 03Paul B Mahol 07master:3db9960e0aa9: avcodec/qdm2: remove unneeded #include, there are no assert() [16:38] <cone-864> ffmpeg.git 03Paul B Mahol 07master:be128c1e3426: avcodec/videodsp_template: remove unneeded #include, there are no assert() only av_assert* [16:44] <cone-864> ffmpeg.git 03Michael Niedermayer 07master:3531594017bd: ppc/mpegvideo_altivec: Change assert to av_assert2() [16:45] <cone-864> ffmpeg.git 03Michael Niedermayer 07master:d0f315ab6c7d: avcodec/ppc/h264qpel_template: Change if DEBUG assert() to av_assert2() [17:12] <durandal_1707> michaelni, saste: is it ok if i remove libmpcodecs? [17:13] <saste> durandal_1707, no objections from me [17:14] <michaelni> durandal_1707, no objections from me [17:16] <cone-864> ffmpeg.git 03Paul B Mahol 07master:9fd925dee58d: lavfi: remove mp=softpulldown [17:17] <arwa> I have to write process_command to change which settings of eq filter? [17:18] <durandal_1707> arwa: all: gamma, contrast, brightness, saturation [17:19] <arwa> Okay. [17:30] <wm4> it would be better if instead of a process_command, we had a function to change the avoptions of a filter [17:57] <akira4> ubitux, Could you have a look at this diff? http://pastebin.com/fGGvLL8M. [17:57] <ubitux> akira4: sure :) [17:58] <ubitux> akira4: does it work? [17:58] <akira4> No. Its giving me a segfault when I try to access the ASS_style struct [17:59] <akira4> I just wanted to make sure I'm working in the right direction :-/ [18:01] <ubitux> let me check the api [18:02] <ubitux> akira4: try to allocate a new style [18:03] <ubitux> in the track [18:03] <akira4> hmm. Okay. I'll try that. [18:03] <ubitux> in which you will force a few settings [18:03] <akira4> Do I need to set the values for all the style parameters? [18:04] <ubitux> what the doc api says? [18:06] <akira4> Uhm there are many parameters. The doc just mentions what they do. [18:07] <ubitux> just alloc a new style in the track [18:07] <ubitux> and set style->FontName = something [18:07] <ubitux> (you might to strdup or something, not sure) [18:07] <ubitux> then call the force style thing [18:07] <akira4> Yes.Also, when I try to give more than one parameter in the commandline it just takes the first one. [18:08] <akira4> Am I doing something wrong in the taking input part? [18:08] <akira4> as in it doesnt take the string after a comma. [18:09] <ubitux> mmh wait, i see there is ass_set_style_overrides() too [18:09] <ubitux> akira4: what exactly are you trying to do? [18:09] <akira4> something like ./ffmpeg -i ~/test.avi -vf subtitles=~/testing.srt:force_style=FontName=Arial,Outline=2 ~/out.avi [18:11] <ubitux> yes, the comma is the filter separator for successive filters [18:11] <ubitux> try -vf "subtitles=$HOME/testing.srt:force_style='FontName=Arial,Outline=2'" [18:12] <ubitux> look at ass_set_style_overrides(), you probably want to use that [18:13] <akira4> right. Thanks. [18:18] <ubitux> akira4: you might just need a call to ass_set_style_overrides() actually [18:18] <ubitux> try this before trying someting more complex like allocating a new style etc [18:19] <akira4> Oh.Okay. I'll try working with that then. [18:24] <ramiro> is anyone from here going to attend FOSDEM? [18:24] <ramiro> I know kierank is going as a speaker [19:15] <cone-864> ffmpeg.git 03Carl Eugen Hoyos 07master:50144b91ea50: Respect horizontal differencing predictor for 16bit gray tiff images. [19:15] <cone-864> ffmpeg.git 03Michael Niedermayer 07master:dffc16d52314: Merge remote-tracking branch 'cehoyos/master' [20:16] <cone-864> ffmpeg.git 03Paul B Mahol 07master:2ff7e81b3c5e: avcodec/vp3: use init_get_bits8() [21:41] <kurosu_> jamrial, looking at the sao patches, I have several comments, one is a big conceptual one [21:41] <kurosu_> but I'll first look at benchmark values [21:42] <jamrial> sure [21:54] <jamrial> btw, alignment is not important for offset_val, but you just made me realize that (assuming it works) a single movq + SPLATW [0123] may be better for the sse2 version [21:54] <jamrial> one load from memory instead of four [21:59] <jamrial> kurosu_: works. will have to benchmark it and see if there's any gain [21:59] <jamrial> probably nothing substantial. it's outside of the loop after all [22:00] <kurosu_> yes, and at several thousands cycles, that's noise [22:22] <kurosu_> jamrial, well my time is over for tonight [22:22] <kurosu_> so just a summary of things [22:23] <kurosu_> your assembly is around 8-10% faster than the intrinsic one - too bad it's not that heavy in the decoding [22:23] <kurosu_> as for the implementation, it's going to be cumbersome, but here's an idea: [22:23] <kurosu_> you actually have only 4 values [22:24] <kurosu_> the issue is that they apply to just a range of pixels value [22:24] <kurosu_> it's probably possible to add/subtract an offset to the pixels values and shift [22:25] <kurosu_> or to shift the pixel values and offset the table [22:25] <kurosu_> but loading the 4 values into a xmm reg, and using pshufb looks possible for 8 bits [22:25] <kurosu_> (not sure if you extend to 16 bits and if it is needed) [22:26] <kurosu_> for 10+ bit cases, yes, it's troublesome, because pshufb works on bytes [23:28] <jamrial> kurosu_: i think see what you mean [23:29] <jamrial> and extending to 16 bits is probably necessary. there are no byte shift instructions outside of the amd xop one afaik [00:00] --- Sat Jan 31 2015

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ffmpeg.log.20150130
by burek 31 Jan '15

31 Jan '15

[00:42] <TwisteR> Greetings! I have a problem similar to one described here: http://stackoverflow.com/questions/25849875/ffmpeg-very-big-duration-and-ve… [00:42] <TwisteR> trying to find out what is the problem [00:47] <TwisteR> lol, problem solved :D forgot to set pkt.duration ;) [00:48] <TwisteR> like this (unsure, if this is correct, but it works for me) [00:48] <TwisteR> pkt.duration = av_rescale_q(pkt.duration, ifcx->streams[i_index]->codec->time_base, ofcx->streams[i_index]->codec->time_base); [01:42] <debianuser> Hello, how to recover an incomplete .mp4 file? I.e. a recording process was killed while writing .mp4 file. So I have 1+GB file that fails to play because "moov atom not found". [01:42] <debianuser> ("recording process" was actually ffmpeg, if that can help) [01:44] Action: debianuser thinks it's a rather common problem, maybe there's a common solution to it? [02:03] <c_14> Short answer, you can't. [02:03] <c_14> At least not easily, and not reproduceably. [02:07] <debianuser> :( It there a long answer? I know almost everything about that video (h264 1920x1080 yuv420p 30 fps video, aac 48kHz stereo audio), what information is missing to make that file playable again? [02:11] <c_14> Intricate knowledge of the internals of the encoder/muxer/mp4 format and a willingness to work with raw binary data/hex as well as knowledge of at least 1 programming language. [02:19] <c_14> I've found a couple of links to programs that might be able to recover something from the file. [02:20] <c_14> But if you still have the source it's probably less work/time to just reencode. [02:23] <ribasushi> c_14: I find your answer kinda odd - if the parameters of the streams are known, isn't the moov more or less redundant...? [02:23] <ribasushi> I am not arguing with you, I am wondering what am I missing [02:23] <c_14> The moov atom is an index of everything, until you write it you don't know where in the file the video is, where the audio is etc [02:24] <ribasushi> bleh [02:24] <c_14> http://forum.doom9.org/archive/index.php/t-169815.html [02:24] <c_14> read the 2nd post [02:24] Action: ribasushi makes a note to always record to matroska (with a copy-remux to mp4 if needed) [02:24] <ribasushi> c_14: thanks! [02:25] <debianuser> No, I don't have the source, it was a stream capture. (currently trying to solve some ffmpeg linking problems http://pastebin.com/kc1VYWFr and build https://github.com/ponchio/untrunc) [02:25] <ribasushi> c_14: huh... that's one bizarre mux format... [02:26] <ribasushi> any idea why anyone would design things this way...? I mean there had to be some reason [02:29] <c_14> It's relatively simple [02:29] <c_14> The muxer can just start writing data however it wants and then when it's done write out what's where. [02:30] <c_14> And nobody thought about what would happen if the process aborted... [02:32] <ribasushi> c_14: that'd be fine if they were working on a greenfield thing [02:32] <ribasushi> but both avi and matroska existed before mp4 was finalized (afaik) [02:33] <debianuser> (well `-Lffmpeg/libswresample -lswresample -lrt` did the trick, now ./untrunk prints lots of something to the screen, waiting and hoping...) [02:34] <c_14> debianuser: oh, and next time. When doing stream captures, (or any long-runnig encode especially if you don't have the source) don't mux to mp4. If you need it in mp4 either remux afterwards, or at very least also mux to another format like matroska or mpegts [02:34] <debianuser> :) [02:35] Action: c_14 wonders if he should stick a giant disclaimer somewhere [02:37] <debianuser> Technically it should not be impossible to recover even in case of mp4. (1) guess the codecs: read first 10MB and attempt to decode as much as possible with all the available codecs, cut that, repeat. (2) knowing the codec: read and decode as much as possible, switch to another codec, repeat. [02:38] <c_14> It's not _impossible_, but bruteforce is neither fast, nor pretty. [02:39] <c_14> And not really something you want to was^h^h^hspend your time doing [02:40] <debianuser> It was just a little unexpected that there's no known tool doing that, since it's really easy to get a broken mp4 because of a camera/mobile phone shut down during recording... [02:44] <c_14> You'd think so, but most of the time it just isn't worth the effort [02:49] <debianuser> Ok, it took ./untrunc 67 seconds to process 50MB of that file, and the result was indeed playable, video was very broken, however, the sound was not bad. [03:15] <k_sze[work]> ugh, transcoding ffv1 from 8-bit grayscale to 16-bit grayscale doubled in size. [03:16] <k_sze[work]> even though my video should have reaaaally low entropy. [03:18] <ribasushi> k_sze[work]: ffv1 does not do any meaningful compression [03:19] <ribasushi> k_sze[work]: I do a lot of 264 lossless (it truly is lossless, did a ton of tests with reassembled stream checksums etc) [03:20] <k_sze[work]> Can H.264 do 8-bit or 16-bit grayscale lossless? [03:21] <k_sze[work]> I forget how to list codec-specific help using the ffmpeg command line. [03:23] <k_sze[work]> Or was that only possible with libav's avconv command? I forget. [03:23] <c_14> ffmpeg -h codec= [03:24] <ribasushi> k_sze[work]: here are some timings I did just 3 days ago: http://paste.scsys.co.uk/460352 [03:24] <ribasushi> (clearly x265 is a work in progress :) [03:25] <k_sze[work]> yeah, x265 is clearly not doable in real-time with current cheap hardware. [03:25] <ribasushi> k_sze[work]: note - these are CPUtimes, not walltimes [03:26] <k_sze[work]> c_14: strange, it's not doing the right thing. [03:26] <k_sze[work]> I have ffmpeg 2.5.3 [03:26] <k_sze[work]> `ffmpeg -h codec=libx264` isn't giving me any info about the libx264 codec. [03:27] <k_sze[work]> I used to be able to see the supported pixel formats on another computer. [03:27] <c_14> eh, sorry [03:27] <c_14> -h encoder= [03:27] <k_sze[work]> ah, right [03:27] <k_sze[work]> encoder, not codec [03:28] <k_sze[work]> hmm, no grayscale pixel format for h264 [03:29] <k_sze[work]> What's the difference between the yuv* and yuvj* pixel formats? [03:29] <k_sze[work]> yuvj* use the full dynamic range of 0-255? [03:37] <k_sze[work]> And I get "Multple keyframes with same PTS" when remuxing 8-bit grayscale ffv1 avi to nut. [03:37] <k_sze[work]> Not sure if I should be worried about that. [03:38] <k_sze[work]> e.g. would that affect seeking? [03:40] <k_sze[work]> shit [03:40] <k_sze[work]> so it discarded the frame. [03:42] <debianuser> c_14 (and maybe ribasushi): SUCCESS! I commented out a few lines in untrunc/track.cpp (diff: http://pastebin.com/qHFVRnD8 ) and got first 50MB of .mp4 file recovered PERFECTLY (perfect enough for mplayer to play it with no video/audio distortions). Now waiting for it to process the whole file... [03:42] <k_sze[work]> The original AVI was encoded with ffmpeg. Looks like ffmpeg has a bug when remuxing to nut [03:43] <k_sze[work]> if I remux the avi -> nut -> another nut, ffmpeg still drops a frame on the second remux [04:01] <k_sze[work]> Actually, it maybe it didn't drop a frame the second time, but it still gave me a warning. Very weird. [04:05] <debianuser> c_14: Yes, it looks working! So if someone else asks how to repair an incomplete .mp4 file suggest "untrunc", optionally with http://pastebin.com/qHFVRnD8 patch. :) Worked for my h264+aac mp4. [04:05] <c_14> I'll keep it in mind. [04:21] <k_sze[work]> when exactly was ffmpeg 1.2 released? I can't seem to find that info on ffmpeg.org [04:21] <k_sze[work]> march 15, 2013, right? [04:21] <k_sze[work]> (I wish there's just a release history table somewhere) [04:24] <relaxed> k_sze[work]: http://git.videolan.org/?p=ffmpeg.git;a=tags http://git.videolan.org/?p=ffmpeg.git;a=tag;h=a700d997996e94944fed4c440ff8a… [04:25] <k_sze[work]> ok [07:53] <deasy22> hello all [07:53] <deasy22> i want to ask about ffmpeg problem [07:54] <deasy22> i want to make video like this [07:54] <deasy22> https://www.youtube.com/watch?v=1Jdf_aIFCsE [07:54] <deasy22> i have a png file name test.png and video file name video1.mp4 [07:55] <deasy22> about ffmpeg pip.. [07:55] <deasy22> what command in ffmpeg to create video pip ? [07:56] <deasy22> thanks for answer :) [08:09] <grepper> deasy22: have a look at the -loop_input option [08:10] <deasy22> greeper [08:10] <deasy22> i use this command [08:11] <deasy22> ffmpeg -i video1.mp4 -loop 1 -i test.png -filter_complex "[0:v]scale=-1:624[bgVideo]; [1:v]scale=-1[myFrame]; [1:v][bgVideo]overlay=10:(main_h/2)-(overlay_h/2):shortest=1[myBgOverlay]; [myBgOverlay][myFrame]overlay=main_w-overlay_w-10:(main_h/2)-(overlay_h/2)[video]" -map "[video]" -s:v 1280x720 -map 0:a -codec:a copy output.mkv -y [08:11] <deasy22> but still eror [08:12] <deasy22> [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000002ede80] Invalid stream specifier: '[v]'. Last message repeated 1 times Stream map ''[v]'' matches no streams. [08:18] <deasy22> i dont know how to resolved that error [08:21] <grepper> dunno, perhaps it is asking for the output stream number? [08:22] <k_sze[work]> Weird, I have an H.264 in FLV video. [08:22] <k_sze[work]> Somehow the first frame decoded does not even start at time 0 [08:22] <k_sze[work]> it starts at 0.067 second. [08:22] <k_sze[work]> It's a 30fps video. [08:24] <k_sze[work]> And then there another one where the first frame starts at 0.0 second, but then the next frames are at 0.04, 0.073, 0.107, 0.14, etc. [08:28] <k_sze[work]> This makes my life really miserable when I want to perform frame accurate seek in different video files. [08:29] <k_sze[work]> Because there is no guarantee at all about the way the frames are timestamped. [08:33] <grepper> deasy22: I have used filters to overlay a movie on an image background, but nothing near as complex, I used the 'movie' filter [08:34] <grepper> -vf movie=/tmp/todisc-work.0/quick_menu_bg.png [wm];[in][wm] overlay=0:0 [wm2]; [wm2] setdar=4:3 [out] [08:34] <grepper> I honestly doubt I could be of much help, much of it is voodo to me [08:35] <deasy22> do you have reference about PIP in ffmpeg? [08:35] <deasy22> picture in picture [08:35] <deasy22> link about pip ffmpeg i mean [08:36] <grepper> I just know of these, which you have probably seen: https://ffmpeg.org/ffmpeg-filters.html and https://trac.ffmpeg.org/wiki/FilteringGuide [08:37] <deasy22> thanks greeper [09:53] <fffan> why some versions of ffplay use packet_temp.size -= result after avcodec_decode_audio4, some versions of ffplay dosen't ? [11:39] <JoshX> Goodday :) I'm trying to figure out if the segment filter provides me with some kind of event when a segment is finished? [11:39] <JoshX> all i can find is someone suggesting it in a forum and some patches that show there _could_ be such a thing :) [11:39] <JoshX> now the question is how to use it then :) [12:46] <saste> JoshX, no, there is no such thing as an event callback in segment.c [12:57] <Diego___> Hello! [13:10] <Diego___> Hi, does anybody have any experience compiling ffmpeg on centos? Im following this guide http://trac.ffmpeg.org/wiki/CompilationGuide/Centos but when I try to configure libvpx I get this error nasm -Ox -f elf64 does not support section alignment (nasm <=2.08?) Ive tried updating nasm but the most recent version I can find is 2.07, is it possible to configure it using yasm instead? [13:11] <durandal_1707> if you have yasm installed, yes. [13:26] <Diego___> I do have yasm installed, how would I get it to use yasm instead (I have tried searching but couldnt find anything) [13:27] <durandal_1707> maybe with --yasmexe=EXE [13:43] <Diego___> thanks for your help durandal, I got it working using as=yasm, I also had to specify the path to yasm because it couldnt find it (PATH=$PATH:/usr/local/bin) [14:22] <posseeDeestroyah> Hello. [14:22] <posseeDeestroyah> I have one trouble. [14:22] <JoshX> I wish i had only 1 problem ;-) [14:22] <posseeDeestroyah> Me too, yep. That's not my only problem. [14:23] <posseeDeestroyah> FFmpeg is capturing sound with -f pulse -stream_name record_monitor -i `pactl list sources | awk '/Name:/ && /monitor/ {print $2}'` -f pulse -channels 1 -i default. [14:23] <posseeDeestroyah> So there is monitor of playback and micro. [14:23] <posseeDeestroyah> But it captures only playback in 2 streams. [14:23] <posseeDeestroyah> Is that pulse bug? [14:24] <BtbN> what? [14:25] <posseeDeestroyah> FFmpeg captures playback twice instead of capturing playback and micro. [14:25] <posseeDeestroyah> I dunno what causes it. [14:27] <BtbN> propably you told it to do so? Or how do you know what default is? [14:28] <posseeDeestroyah> Yes, default is micro. [14:28] <posseeDeestroyah> Wait. [14:28] <posseeDeestroyah> Without 'yes'. [14:28] <posseeDeestroyah> Just 'default is micro'. [14:28] <posseeDeestroyah> pactl info | grep 'Default So' [14:28] <posseeDeestroyah> Default Source: alsa_input.pci-0000_00_1b.0.analog-stereo [14:31] <posseeDeestroyah> Maybe pulse is too old. [14:32] <BtbN> Or the ffmpeg pulseaudio source isn't intended to be used more than once [16:56] <brontosaurusrex> JEEB: http://shrani.si/f/I/RA/2CbHtqm/virtualenginesspeed.png < vmware test ..... [18:02] <ramiro> brontosaurusrex: why are you trying to run video encoding in a virtual environment? [18:02] <ramiro> ffmpeg runs equally well under linux, mac, and windows [18:02] <brontosaurusrex> ramiro: vm is meant to be copy/pasted among macs/windows [18:03] <ramiro> yes, but why? =) [18:03] <brontosaurusrex> ffmpeg is just part of the story [18:03] <ramiro> the performance hit is huge, like you've seen already [18:05] <brontosaurusrex> i could live with it, if the performance were 5-15 percent slower i guess [18:05] <brontosaurusrex> but not 400 percent [18:07] <ramiro> does the rest of the system only run on linux? [18:08] <brontosaurusrex> ramiro: what do you mean, guest? [18:08] <ramiro> yes [18:08] <ramiro> you say ffmpeg is just part of the story. what's the rest like? [18:08] <brontosaurusrex> no, it could be made into separated windows and osx version [18:08] <ramiro> because you make it sound like there's no way out of using the vm [18:09] <brontosaurusrex> there is a way [18:10] <brontosaurusrex> still surprised by this crapy performnce, i guess waiting for someone to tell me that i need to clicky-clicky this and that and performance will magically improve ;) [18:11] <ramiro> brontosaurusrex: https://bgrins.github.io/videoconverter.js/ [18:12] <ramiro> (this is just a joke, don't actually try it =) [18:12] <foonix> brontosaurusrex: maybe you dont have all cpu features/flags or not all cpu/cores assigned to your vm? [18:13] <brontosaurusrex> foonix: its not a cpu issues, bottleneck is definately io speed [18:15] <brontosaurusrex> which will be less imortant when encoding to say x264 or even dv [18:15] <brontosaurusrex> my examples is huge input and huge output [18:16] <ramiro> are you sure about that? try to encode from a test source and into /dev/null [18:16] <ramiro> that way you get I/O out of the equation [18:16] <brontosaurusrex> ramiro: what would be the test.source ? [18:17] <ramiro> https://trac.ffmpeg.org/wiki/FilteringGuide#TestSource [18:17] <brontosaurusrex> let me try ... [18:23] <brontosaurusrex> 53 fps vs 37 fps (but the vm has only half cores), so that makes sense, right? [18:23] <brontosaurusrex> its IO! [18:24] <brontosaurusrex> I used: ffmpeg_static -f lavfi -i testsrc=duration=10:size=1280x720:rate=30 -vf yadif=0,scale=1920:1080 -aspect 16:9 -c:v prores -profile:v 3 -pix_fmt yuv422p10le -f mov -y /dev/null [18:26] <ramiro> try a simpler non-multi-threaded codec [18:29] <ramiro> or use -threads 1 before /dev/null [18:29] <brontosaurusrex> why? [18:30] <ramiro> to have a better comparison in regards to cpu usage [18:30] <ramiro> oh, you did set your HD to a fixed size in VMWare, right? dynamic disks will take a toll on disk I/O because they have to expand. [18:32] <brontosaurusrex> ramiro: its reading writing to host shared folder [18:36] <brontosaurusrex> ok, x264 is 2.25x faster [18:37] <brontosaurusrex> using ffmpeg -f lavfi -i testsrc=duration=10:size=1920x1080:rate=25 -an -vcodec libx264 -preset slow -tune film -crf 21 -f mp4 -y /dev/null [19:14] <brontosaurusrex> http://paste.debian.net/plain/143120 < noio tests [19:25] <nyuszika7h> hi, how can I remove the subtitle track from an mkv file and get it as a separate srt file? [19:25] <c_14> ffmpeg -i mkv -map 0:s out.srt [19:25] <nyuszika7h> thanks [19:26] <c_14> Note, depending on your definition of "remove" it won't actually remove the subtitle track since it'll still be in the mkv. [19:26] <c_14> If you want an mkv without the subtitle track, `ffmpeg -i mkv -map -0:s -c copy out.mkv' [19:27] <nyuszika7h> the latter gives "Stream map '0:s' matches no streams." [19:27] <c_14> You can also combine both commands `ffmpeg -i mkv -map -0:s -c copy out.mkv -map 0:s out.srt' [19:27] <c_14> But the first command worked? [19:28] <nyuszika7h> yes [19:28] <nyuszika7h> there seem to be two subtitle tracks actually (I want the first one, the other one seems to be broken) [19:28] <c_14> -map 0:s:0 for the ones producing the srt [19:29] <c_14> And replace the -map -0:s with -map 0:v -map 0:a [19:30] <nyuszika7h> ok, thanks [19:30] <nyuszika7h> seems to work [19:32] <nyuszika7h> yup. my TV is picky about embedded subtitles, it won't let me change the settings like subtitle size [19:32] <nyuszika7h> it works with a separate srt though [19:54] Action: c_14 loves TVs [19:54] <c_14> They're so [19:54] <c_14> stupid [20:03] <klaxa> especially "smart" TVs [20:03] <klaxa> i just plug in a computer and do whatever i want [20:17] <nyuszika7h> klaxa: yeah, my smart TV has some pretty stupid restrictions in some places [20:17] <nyuszika7h> like, I can't adjust the volume in the file manager [20:18] <nyuszika7h> and I can't change some things like the menu language when I'm in the file manager or playing something from an USB device [20:19] <relaxed> connect pc through hdmi, mpv, ???, profit [20:20] <nyuszika7h> I'd need a really long HDMI cable - I have a laptop but moving it around is not an option [20:21] <nyuszika7h> and I don't think I could control the playback with my TV's remote in that case, anyway (pause, rewind, etc.) [20:22] <nyuszika7h> (also, someone may want to change s/experimental/sid/ in the topic) [20:22] <relaxed> push keys on the keyboard or use an ir remote [20:23] Action: relaxed [20:23] Action: relaxed 's muscle memory prefers the keyboard [20:23] <nyuszika7h> my laptop is too far from the TV, not an option [20:23] <nyuszika7h> also, my TV does have an universal remote feature but does that work with computers? [20:24] <relaxed> yes [20:25] <klaxa> i once wrote an android app and a simple rest-service in python to use my phone as a remote [20:26] <relaxed> nyuszika7h: go to monoprice and buy a super long hdmi cable [20:26] <nyuszika7h> it would be a PITA to walk past the cable every time I want to enter or exit the room though, so no thanks [20:29] <relaxed> yeah, that sound brutal [23:11] <anon-1235623463> Does ffmpeg support PixelCrop in MKV containers? http://matroska.org/technical/specs/index.html [23:12] <anon-1235623463> ^ The purpose being to take a 1920x1080 video with black bars on the top/bottom or left/right and losslessly crop it at the container level, without re-encoding. the alternative is a lossless-re-encode. I losslessly re-encoded an 11 minute HD video from 16:9 to 4:3 and it went from 2 GB to 11 GB. [23:13] <klaxa> lossless encodes tend to use up more storage [23:15] <klaxa> anon-1235623463: maybe try mkvtoolnix see https://www.bunkus.org/videotools/mkvtoolnix/doc/mkvmerge.html under 2.8. --cropping [23:15] <anon-1235623463> Thanks. [00:00] --- Sat Jan 31 2015

1 0

ffmpeg-devel.log.20150129
by burek 30 Jan '15

30 Jan '15

[02:16] <cone-51> ffmpeg.git 03Supraja Meedinti 07master:6ad749b53354: libavutil: Added twofish symmetric block cipher [03:18] <cone-51> ffmpeg.git 03Carl Eugen Hoyos 07master:3ea97767e49c: Add an ARES atom to extradata when encoding avui. [03:18] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:4155f2d7cc6a: Merge remote-tracking branch 'cehoyos/master' [04:59] <cone-51> ffmpeg.git 03Arwa Arif 07master:a21acd554a25: Fix frame-alignment in PP7 [07:13] <fffan> why av_frame_get_best_effort_timestamp does not work for wmv's audio stream? [07:16] <fffan> 4 j-b [10:23] <durandal_1707> so is variable frame rate possible with lavfi? [10:24] <nevcairiel> that entirely depends on the filters [10:24] <nevcairiel> it just passes timestamps with the AVFrames [10:24] <BtbN> stuff like scaling schouldn't care [10:28] Action: kierank doesn't set timestamps with lavfi and it works [10:29] <nevcairiel> many filters in lavfi wont care about timestamps or framerate, but your app might want to associate timestamps with frames, unless its strict cfr anyway and you just invent new ones later [10:30] <durandal_1707> but timestamps depends on timebase and timebase may change with vfr [10:31] <kierank> Time base isn't meant to change with vfr [10:31] <nevcairiel> timebase shouldnt change, if it does the format is stupid :p [10:31] <kierank> The frame interval changes, yes [11:02] <wm4> some filters actually want a framerate [11:03] <durandal_1707> but to what set pts when filter works in vfr [11:04] <wm4> to the timestamp [11:04] <kierank> durandal_1707: you have a bigger problem I realised [11:04] <kierank> because in mpeg2/h264 the PTS of a soft pulldown file is the PTS *after* pulldown [11:05] <kierank> the filter is broken by design i believe [11:05] <nevcairiel> yeah soft-telecined h264 streams have crazy pts, 24 fps that fit on timestamps of 30 fps content [11:06] <kierank> oh wait, it's libmpcodecs it's already broken by design [11:06] <wm4> kierank: nice [11:07] <kierank> durandal_1707: just delete the mpcodecs filter [11:07] <kierank> serious who cares if michaelni is obsessed with them even though they don't work [11:07] <kierank> seriously* [11:08] <durandal_1707> i can just set it to NOPTS value [11:09] <durandal_1707> the filter was designed to be used with mencoder [11:13] <cone-588> ffmpeg.git 03Stefano Sabatini 07master:d11fcf735f4a: doc/filters: apply some updates to the Filtergraph syntax section [11:13] <cone-588> ffmpeg.git 03Stefano Sabatini 07master:af7b89e08be8: lavfi: document assumptions about the input and output labels of a filter graph description [11:13] <cone-588> ffmpeg.git 03Stefano Sabatini 07master:d43c1ec684ce: examples/filtering: extend comments about setting the filter graph endpoints [11:17] <durandal_1707> michaelni: mp=softpulldown sets all frames pts value to NO_PTS [11:19] <durandal_1707> michaelni: what you prefer: removal of mp=softpulldown or bug to bug compatible port [11:22] <wm4> <durandal_1707> the filter was designed to be used with mencoder <- *chuckle* [11:22] <wm4> mencoder didn't use timestamps [11:23] <durandal_1707> so how it worked at all? [11:38] <wm4> durandal_1707: who knows whether it ever did? [11:39] <wm4> the filter was added in 2003 [11:40] <kierank> just delete the filter [11:42] <wm4> who is even against it [11:46] <kierank> nobody apart from the leader [12:48] <michaelni> durandal_1707, i prefer having softpuldown in libavfilter, we can fix bugs in it later [12:50] <durandal_1707> michaelni: so it's ok to push it bug for bug compatible version with unchanged pts [12:50] <kierank> wow that's a joke [12:51] <wm4> michaelni: why? [12:51] <wm4> also "fix bugs later" => project slowly turns into garbage [12:52] <michaelni> kierank, ? [12:52] <kierank> the filter is totally broken, doesn't do anything yet you want it in libavfilter [12:52] <kierank> it's inherently broken [12:52] <kierank> the only way to make the filter work is to apply the pulldown pattern separately [12:53] <kierank> I can craft files that have totally insane repeat patterns otherwise and the filter will try to "pulldown" them [12:53] <kierank> http://git.videolan.org/?p=x264.git;a=blob;f=x264.c;h=2dd819a30414780fd4c93… [12:54] <kierank> note how x264 only supports sane pulldown [12:54] <nevcairiel> wtf is euro [12:54] <kierank> it's for music videos [12:54] <kierank> where they don't want 4% speedup [12:55] <nevcairiel> 24->50 in pulldown? [12:55] <kierank> 24->25 [12:55] <nevcairiel> those pattern look crazy [12:55] <kierank> it's just repeating a field every 12 frames [13:02] <michaelni> kierank, if you want to write a filter as you describe, please do so. How is this related to the softpulldown filter? [13:03] <wm4> michaelni: I still want to know why a broken filter should be added, even though everyone points out how broken it is, and no argument was given so far how it is not broken or useful [13:04] <wm4> michaelni: or do we have to tolerate this bullshit because you're the leader and you say so? [13:04] <wm4> (watch by diplomatic skills) [13:05] <michaelni> wm4, no, we can drop the filter if its useless but the filter does basically what the mpeg2 spec requires IIUC [13:07] <durandal_1707> filter does not set pts and I dunno how pts can be set [13:07] <michaelni> durandal_1707, i can fix pts or rather try [13:08] <durandal_1707> also i'm not sure that default behavior is correct when no repeat_pict is set [13:08] <kierank> 12:05 PM <michaelni> wm4, no, we can drop the filter if its useless but the filter does basically what the mpeg2 spec requires IIUC --> the *creator* of the file sets up the correct PTSs in the container [13:08] <kierank> it's not the job of some program to try and guess what the PTSs are [13:09] <michaelni> kierank, why do you troll ? [13:09] <kierank> it's not a troll [13:09] <kierank> it's a fact [13:10] <kierank> I implemented it in x264 - I know how it works [13:11] <michaelni> its a troll because i want to fix the pts and you say the pts need tp be fixed to match the input isnt that what i wanted to do ? [13:12] <kierank> you can't fix the pts because you don't know how the timebase is meant to change because you can't predict the pattern [13:12] <kierank> unless you assume the timebase is post-pulldown which means the PTSs are right to begin with [13:14] <michaelni> kierank, what you call a timebase is if i take your vfr comments not a timebase but the framerate and iam afraid the more i argue the more you argue based on different terninology [13:14] <kierank> pulldown is inherently cfr yes [13:22] <durandal_1707> i tried softpulldown with x264 and it returns always repeat_pict = 0 [13:23] <wm4> I know some people who encode DVDs and take deinterlacing issues very seriously, and they'd piss their pants if you told them about the softpulldown filter [13:24] <michaelni> wm4, thats understandable the filter pretty much adds "interlacing" [13:26] <michaelni> iam also not sure "softpulldown" is the best name for the filter [13:27] <wm4> oh, wrong direction then? even more useless... [13:31] <michaelni> not sure what you mean by direction but possibly yes [13:46] <durandal_1707> so is it ok to drop mp=softpulldown? [13:53] <michaelni> durandal_1707, if noone objects i wont object either, i would slightly prefer to have the filter in libavfilter though but if people feel strongly that its better to completely drop so be it [13:58] <compn> michaelni : apparently people "feel strongly" about getting rid of mpfilters , might as well let them drop it [13:58] <compn> if someone comes along later they can pull a git revision of before the removal and fix softpulldown [13:59] <compn> wm4 / durandal_1707 (and kierank who left) : althought we just had a user in #mplayer who was using pullup in mencoder because it wasnt available in ffmpeg ? [13:59] <compn> isnt pullup available in ffmpeg? if its renamed , might want to say that in the docs [14:00] <durandal_1707> it is available and not renamed, same name - pullup [14:01] <wm4> if he uses mencoder, whack him [14:11] <Daemon404> wm4, i hope you mean 'whack' in the mob sense [14:12] <wm4> is there another sense? [14:14] <Daemon404> well. the literal sense. [14:14] <Daemon404> whack someone with a club [14:14] <wm4> anything is fine against mencoder users [14:15] <Daemon404> i am fairly certain youtube still uses mencoder in a VM [14:15] <Daemon404> for a small subset of codecs [14:15] <Daemon404> liek canopus hq(x) [14:16] <wm4> but that's google [14:16] <BtbN> Youtube at least messes up flv with 48kHz aac. [14:21] <cone-588> ffmpeg.git 03Martin Storsjö 07master:6996fd204a7f: libopenh264: Log debug messages to a non-null context [14:21] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:d39fa69dfc64: Merge commit '6996fd204a7f28b46a8c3c97bcf223998218c743' [14:32] <cone-588> ffmpeg.git 03Zhang Rui 07master:038f3a173f59: avformat/concatdec: avoid NULL dereference when failed to open file. [15:23] <compn> so uh, full html5 on youtube eh [15:23] <compn> no more 'dobe flash [15:24] <Daemon404> its default, not gone. [15:24] <BtbN> html5 was default for non-broken browsers for a while. [15:24] <compn> iirc dark_shikari was talking about optimizing our h264 decoder , dont remember if coreavc was still faster [15:24] <BtbN> So basicaly everything except IE and Firefox(Which is basicaly only Chrome) [15:25] <wm4> I wonder if this is the death of flash [15:25] <BtbN> no. [15:25] <compn> we can only hope. [15:25] <compn> but no. :( [15:25] <wm4> youtube is one of the most popular websites (or was), so "everyone" had to have flash installed [15:25] <BtbN> live streaming still doesn't work without flash. [15:25] <wm4> but now without this "killer argument" for flash it could fall into disuse [15:26] <ubitux> do we have color similarity code somewhere i could reuse? [15:27] <compn> color difference stuff? hmm [15:27] <compn> yeah , whats it called where you set the blacklevel [15:27] <compn> in encoders [15:27] <compn> or maybe cropdetect [15:28] <ubitux> i mean sum of squared (or absolute) diff of each r g and b component is kind of shit [15:28] <ubitux> so i was wondering if we had somethng hsv based or similar [16:19] <ubitux> i wonder if it will be overkill to rely on xyz+cielab and an advanced cie diff based [17:14] <durandal11707> michaelni: where i can find nut4cc.txt? [17:15] <michaelni> svn://svn.mplayerhq.hu/nut [17:16] <michaelni> theres also a git mirror of it somewhere [17:22] <durandal11707> michaelni: svn one seems obsolete, i found commits that are not in svn tree [17:28] <durandal11707> hmm doing git pull fetched rest of changes [17:53] <durandal11707> michaelni: why is GIF[0] in nut4cc but not in libavformat/nut.c [17:53] <durandal11707> ? [18:00] <michaelni> durandal11707, no idea, maybe it was forgotten [18:29] <ubitux> http://pastie.org/pastes/9871846/text https://en.wikipedia.org/wiki/Ordered_dithering [18:30] <ubitux> can i have some litterature on this? [18:30] <ubitux> i don't understand why this particular matrix, and the logic behind it [18:30] <ubitux> also, i'm surprised how the bayer in sws doesn't seem to use this matrix but still have the expected output [18:40] <michaelni> ubitux, the standard ordered dither matrix is trivial to generate, just start with a empty square and fill in one value at a time, always pick a spot fartherst away from all other already filled in spots [18:41] <michaelni> fartherst away in the sense if you would also consider wrap around [18:43] <ubitux> is there a paper on this? [18:44] <ubitux> also, i'm surprised that the ordered dithering is all about adding unsigned values; isn't this causing a luma bump? why isn't it made signed? [18:45] <michaelni> theres probably a paper on it and also probably many other ways to generate the matrix, if its generated by filling in points one could add a slight random component to it though [18:46] <michaelni> about signed, if one replaces (v + 32) >> 6 by (v + dither) >>6 then dither would have a 0..63 range [18:46] <michaelni> doesnt mean there are no luma offset bugs of course, if there are any please report [18:48] <ubitux> i'm not talking about ffmpeg in particular here (i'm actually quite surprised ffmpeg doesn't even use that bayer/ordered matrix (i don't get from where ff_dither_8x8_{32,73,220} comes from nor how they are supposed to be used) [18:50] <ubitux> OTOH the pp filters seems to have the exact bayer filter [18:50] <ubitux> (which they seem to scale somehow differently sometimes) [18:51] <wm4> gradfun has one too [18:51] <ubitux> here i have a lot better result if i just v + (dither>>3) (so... 0..63 scale to 0..7 scale) for a v in 0..255 range [18:51] <ubitux> but i'm probably just doing it wrong [18:51] <ubitux> wm4: not bayer [18:52] <ubitux> mmh, actually maybe it is with a x2 [18:53] <ubitux> yeah right, bayer * 2 in gradfun [20:36] <cone-588> ffmpeg.git 03Luca Barbato 07master:1279221cc4d6: lavu: Check av_dict_set allocations [20:36] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:bcadf5d940e7: Merge commit '1279221cc4d63bc4449df86ae7a98e633f8be425' [20:44] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:41e03e284ee7: DNxHD: More verbose error messages [20:44] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:e5b7e2224f62: Merge commit '41e03e284ee7b0d4caa3a5d28681ad46e3105d65' [21:00] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:598f7d046cbf: DNxHD: Simplify pixel format detection [21:00] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:64e7cf12532e: Merge commit '598f7d046cbf306706623210c5baafa3b7cd1df3' [21:34] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:1a07df31128d: DNxHD: Add support for id 1258 (DNx100 960x720@8) [21:34] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:56252a5e0979: Merge commit '1a07df31128da3a0020b66502399989b91770d44' [21:35] <michaelni> where can i find a DNx100 sample ? [21:42] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:302ca6b20ed0: mpegvideo_enc: initialize the encoding context [21:42] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:e18e5ae62ce2: Merge commit '302ca6b20ed01ac584f5b15d5bca3d3a92b7a77a' [22:01] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:08fa34bf7594: yuv4mpegdec: initialize field_order in yuv4_read_header() [22:01] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:c2ad22ff9934: Merge commit '08fa34bf75942f66796d770ff42a3721b2e3d2d4' [22:07] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:c01ccccbb13f: ituh263dec: use macro instead of #if [22:07] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:c16896f52551: Merge commit 'c01ccccbb13f464e74bb8498a8c573a66c430ca0' [22:41] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:70d246d5cc3d: flacenc: initialize sums matrix [22:41] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:89c35b113060: Merge commit '70d246d5cc3d214da11f711d997d8cbd8c3a23d1' [23:03] <cone-588> ffmpeg.git 03Vittorio Giovara 07master:22b985d59c00: hqdn3d: check memory allocations and propagate errors [23:03] <cone-588> ffmpeg.git 03Michael Niedermayer 07master:ccccfb59b14f: Merge commit '22b985d59c007c4422aefe3ef3fca0aa0daafa9f' [23:13] <cone-588> ffmpeg.git 03Lou Logan 07master:961f2e3aace0: doc/indevs: add some XCB info to x11grab [00:00] --- Fri Jan 30 2015

1 0

ffmpeg.log.20150129
by burek 30 Jan '15

30 Jan '15

[00:01] <Projectns> Hello i have a question , i wanna build a stream server on a debian server. I installed a rtmp server and now i need a program like obs to stream my video list (Video12,3,4,5,6 sequenz) to the rtmp port [00:01] <Projectns> is it posible with ffmpeg ? and how? [04:29] <Matador> hmmm [04:29] <Matador> Anyone know about the error ? --- Past duration 0.890617 too large -- repeating ? [05:01] <k_sze[work]> Using the libav* API, is possible to open a video file that is still being written and try to get frames until no more frame is available, and when I reach the last frame available in the video file, stay there in a consistent state so I can try getting a frame again? [05:03] <k_sze[work]> (effectively polling an unfinished video file for new frames) [05:25] <Matador> gahh [07:33] <stf> hey guys, please could anyone of you take a look on this, i do not find the error: http://pastebin.com/pLKwEWVY right now, the audio streams 0:1 will be transformed to libvorbis and 0:3 will be transformed to aac [07:35] <stf> 0:1 should be aac and 0:3 should be copy [07:59] <stf> http://pastebin.com/pLKwEWVY [07:59] <stf> I did? [07:59] <stf> oh sry [08:02] <stf> here the console output http://pastebin.com/mztqYDrW [08:07] <relaxed> stf: -c:a:0 aac -c:a:1 copy [08:08] <stf> thx [09:40] <k_sze[work]> ugh, is it actually possible to seek an unfinished file? [09:52] <k_sze[work]> I get the feeling that it should be possible because it looks like mplayer can do it. [09:53] <BtbN> depends on the container [09:53] <BtbN> mp4, nope. mpegts/flv should work [09:53] <k_sze[work]> mplayer could do it with an mkv file. [09:54] <k_sze[work]> (i.e. open an mkv file while video is still being recorded and added to the mkv file) [09:54] <k_sze[work]> And mplayer could at least seek within the timespan that was available at the moment mplayer opened the file. [09:56] <k_sze[work]> even with an oldish version of ffmpeg? [09:56] <k_sze[work]> 1.2 (from deb-multimedia wheezy-backports) [10:08] <k_sze[work]> BtbN: does it also depend on the codec (assuming codecs that are supported by the particular container format)? [10:09] <BtbN> no, it primarily depends on the container having a global header or not [10:09] <BtbN> mkv should also work [10:47] <k_sze[work]> ok, so it seems to work rather easily when I put H.264 in FLV [10:47] <k_sze[work]> but I also have other videos that *must* be encoded as FFV1 for losslessness. [10:47] <k_sze[work]> Hopefully FFV1 in MKV works. [10:49] <BtbN> h264 can also encode lossless. [10:50] <k_sze[work]> but h264 lossless is actually slower and bigger than ffv1 [10:50] <k_sze[work]> not to mention I need gray16le lossless. [10:52] <k_sze[work]> Hmm, could not find plane 1 [10:53] <k_sze[work]> Does that have to do with the FFV1 stream version? [10:56] <pzich> please create a paste with full command and output [10:58] <k_sze[work]> pzich: that's tricky [10:58] <k_sze[work]> because running the same file using the ffmpeg command line works. [10:59] <k_sze[work]> It's when I'm using the PyAV binding to the ffmpeg/libav* API that it fails. [11:03] <k_sze[work]> The error is coming from this anyhow: https://github.com/mikeboers/PyAV/blob/master/av/video/plane.pyx [11:03] <pzich> ah, so a PyAV problem [11:04] <k_sze[work]> yes, PyAV is so far the best Python wrapper to ffmpeg/libav* that I could find so far. [11:25] <k_sze[work]> How do I find out what container format can take FFV1? [11:30] <leni536> Hi, I have a question about rtmpdump, can I ask it here? [11:39] <leni536> Ok, nevermind, it was a broken rtmpe url [13:51] <Elirips> Hello. Is it possible to tell ffmpeg to save every 100 frames one frame to png-image while it is converting some live-stream? [13:52] <Elirips> Say, I currently have something like 'ffmpeg -i <rtsp-input> -f flv <rtmp-output>' and now I'm looking for some magic to tell ffmpeg 'and please save every 100th frame to /tmp/snapshot.png' [14:17] <King_Hual> hi there [14:18] <King_Hual> I'm trying to transcode an m4a audio file to an mp3 file [14:18] <King_Hual> doing this [14:18] <King_Hual> http://pastebin.com/NwPLZ6Lc [14:18] <King_Hual> (in subsonic) [14:18] <King_Hual> what am I doing wrong? [14:19] <BtbN> stream 0 is not the audio stream [14:19] <BtbN> try -map 0:a:0 instead [14:20] <King_Hual> that fixed it [14:20] <King_Hual> thanks a lot! [14:21] <King_Hual> where can i read more about the -map trigger if i may ask? [14:21] <King_Hual> i'm new to ffmpeg [14:37] <tetryds> hello [14:54] <relaxed> King_Hual: https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [14:55] <King_Hual> cheers [16:37] <jamie_> hello I've set up a stream an rtmp stream with ffmpeg and I would like a little help. [16:38] <justinX> jamie_: not that I know much about rtmp, but you should probably tell more about the problem... ? [16:40] <jamie_> When first starting streaming it lagged because I was transcoding the stream from an mpg to an flv. So before I before I streamed I converted the mpg to an flv with the desired encoding settings. Now however when I stream which works perfectly, The upload of my stream is faster then the playback. It creates a large buffer on the flash player. I would like to know how to keep it in tune with... [16:40] <jamie_> ...my playback with encoding. [16:41] <jamie_> without encoding [16:44] <jamie_> http://trac.ffmpeg.org/wiki/Limiting%20the%20output%20bitrate [16:44] <jamie_> i think this solves my problem [16:53] <jamie_> basically instead of the buffer building up on the client side I want to keep it on ffmpeg. If I'm streaming some video and the upload is faster than the playback, then the stream would of been completed before the video has finished. If someone wanted to watch my stream about 20 minutes into me streaming it then the an hour or so video will be nearly finished. [16:54] <jamie_> *then an hour or so video* [16:59] <relaxed> jamie_: did you see -re in the man page? [16:59] <jamie_> nope pls link [17:00] <relaxed> https://ffmpeg.org/ffmpeg.html [17:03] <jamie_> thanks are there any other values that go after -re? [17:03] <relaxed> ffmpeg -re -i input ... [17:04] <jamie_> ok thanks [17:23] <hansman> Hey, [17:24] <hansman> my system becomes unresponsive when I run some complex ffmpeg commands. the disk usage in the task manager shoots up to 100% and everything stops working. [17:25] <hansman> I am using windows 8 with an i3 and 4gb ram [17:25] <__jack__> (haha!) [17:25] <hansman> here is a link to the ffmpeg code which caused this [17:25] <hansman> http://pastebin.com/kwmHgzyx [17:25] <hansman> Please help [17:27] <hansman> guys, anyone? [17:32] <relaxed> hansman: encoding is a resource intensive task [17:32] <hansman> So will getting a VPS sort this problem? [17:32] <relaxed> try with -threads 1 [17:33] <hansman> Okay. I will learn more about threads then. [17:33] <hansman> If I get a VPS, will that be helpful for me? [17:33] <hansman> And if yes, what config should I look for? [17:34] <__jack__> hansman: it's an OS issue [17:34] <hansman> Hmm, how can I sort it __jack__ ? [17:35] <relaxed> -threads 1 will help, by default libx264 will use all of them [17:36] <__jack__> dunno, I'm not a windows expert; you can try to do not have intensive task, as relaxed said [17:36] <hansman> Okay. I get that. [17:36] <hansman> But again, will getting a VPS help me or should I spare that cost for now and try out threads? [17:37] <__jack__> try threads, it cost nothing [17:37] <__jack__> (to try) [17:37] <relaxed> put it after all the inputs [17:39] <hansman> Okay. thank you. :) [17:39] <hansman> this channel never disappoints [17:41] <relaxed> which cpu is you i3? [17:41] <relaxed> your* [17:41] <hansman> which cpu, as in? [17:42] <hansman> I am using it on localhost. [17:42] <relaxed> i3-<number> [17:42] <hansman> 2348m [17:43] <relaxed> ok, that cpu has 4 threads [17:43] <hansman> okay. [17:44] <relaxed> I would try -threads 3 and see how that performs [17:44] <hansman> Can I learn more about the logic behind it? Is there any reference online? An explanation for all this? [17:44] <relaxed> http://arstechnica.com/business/2011/04/ask-ars-what-is-a-cpu-thread/ [17:45] <hansman> Okay. [17:45] <hansman> thank you! [18:25] <brontosaurusrex> any speed tests i can do to remove the IO from equation? (mostly interested in prores and x264 encoding) [18:43] <JEEB> brontosaurusrex, `ffmpeg -i hurr.mov -f null -` for decoding only, and you can check encoding speed by using y4m or something from as fast medium as possible [18:44] <brontosaurusrex> JEEB I/O must be out of equation [18:45] <JEEB> well you can't remove it completely :P [18:45] <JEEB> if you are interested in decoding prores, then you test that first, see how fast it goes [18:45] <brontosaurusrex> why not? isnt there any signal/noise generators or something? [18:46] <JEEB> but your shit is not fucking noise [18:46] <JEEB> the best thing is to dump one of your sources as y4m or something [18:46] <brontosaurusrex> i/d just like to compare wm with a real hardware [18:46] <JEEB> and then load that up from a not-slow HDD or something [18:46] <brontosaurusrex> basically host vs guest [18:47] <brontosaurusrex> the "shared_folders" virtualbox thingy is slow as hell [18:47] <JEEB> yes, so you put it within the goddamn VM [18:47] <JEEB> I mean, my 7200rpm (I think?) normal WD disk is fast enough for 1080p 4:2:0 y4m to be loaded 200+ frames per second [18:48] <JEEB> or well, I haven't tested but x264 can go that fast with it with superfast or so [18:49] <JEEB> so I'd say y4m is a pretty good way of testing "encoding only" unless you know you can have a RAMdisk that doesn't degrade your performance otherwise or something [18:49] <JEEB> I'm pretty sure prores reading will be the bottleneck there, not IO [18:49] <JEEB> s/reading/decoding/ [18:50] <JEEB> but that way you can at least check if x264 will go faster than you can actually decode prores with that systme [18:50] <JEEB> *system [18:54] <JEEB> basically only if you are getting the same kind of figures with `ffmpeg -i hurr.y4m -f null -` as with `ffmpeg -i hurr.mov -f null -` (which is highly unlikely), at that point you can start poking around trying to find faster IO :P [18:54] <brontosaurusrex> Iam pretty sure its an IO, since encoding never gets more than say 20 fps [18:55] <brontosaurusrex> and the virtual cpus are not saturated [18:55] <JEEB> well that kind of test would make it sure :P [18:55] <brontosaurusrex> and the host is getting 4x speed [18:55] <JEEB> also virtualbox is fucking slow in various ways [18:55] <brontosaurusrex> so id like to compare raw ffmpeg vs ffmpeg [18:55] <JEEB> if that's what you're using [18:55] <brontosaurusrex> yes, its virtualbox [18:56] <JEEB> if you are trying to benchmark for prores decoding, then you test that. and if you are so fucking sure about the IO being the fucking bottleneck then actually test that. [18:56] <JEEB> then you have an actual FACT that defines that [18:57] <JEEB> also I hope you're not trying to read a file via network or something, virtualbox's network things are known to be slow [18:57] <JEEB> if you do something, do it from a virtual hd or something [18:58] <JEEB> and if you just want to compare dumb performance then you have to remember that you are testing with something that is not your actual use case [18:58] <JEEB> if you start testing with noise, the thing is not exactly the same, and if you are using a static pattern, that usually ends up being very fast for x264 to encode if it's not intra-only [18:59] <brontosaurusrex> ok [19:38] <brontosaurusrex> JEEB: you were right, smb share is even 10x slower ... [19:40] <JEEB> I used to run lunix under a windows system for :reasons: for a couple of years, and it didn't take too long to switch to vmware, as it was better regarding networking etc. although if I could have local access I would have just installed a goddamn hypervisor or something to minimize the overhead [19:42] <brontosaurusrex> vmware is payware? [19:42] <JEEB> vmware player is freeware for non-corporate use [19:43] <brontosaurusrex> what kind of hypervisor? [19:44] <JEEB> either vmware esxi or one of those lunix based things (xen/whatever) [19:44] <brontosaurusrex> and that would improve speed, how? [19:44] <JEEB> because you're not running it under a full operating system [19:44] <brontosaurusrex> ah right [19:45] <JEEB> or well, the linux kernel things are pretty much a full OS, but of course they're barebones for the non-kernel parts [19:46] <brontosaurusrex> my idea was to make a "portable" encoding station, should run on windows and osx hosts [19:47] <brontosaurusrex> but it is getting way to complicated [19:47] <JEEB> way better to make a bootable image with yocto or something with a very limited set of things, and boot from that via USB or something [19:48] <brontosaurusrex> cant, we are heavily connected to metasan, and no time for reboots [19:48] <JEEB> well, virtualization as you can see can be very much problematic :P [19:49] <brontosaurusrex> iam really dissapointed thought [19:50] <brontosaurusrex> on the other hand, the wm is incredibly cute and will be usefull as root for any lunix experiments [19:51] <JEEB> yes [19:51] <JEEB> virtual machines in general are very useful tools [19:53] <iive> are you talking about windows/linux usage at the same time [19:53] <iive> or it is all linux only? [19:55] <JEEB> his use case was to use virtual machines to provide lunix environments for windows and OS X users as far as I can see [19:55] <brontosaurusrex> iive: windows or osx host, linux guest [19:55] <iive> i see... [19:56] <iive> if you need another OS, you can't go without virtualization. [20:29] <gnm|> If I want to stream video where there are 1000+ names stored in a database, what would be the best practice? stream every file with ffmpeg or using ffserver? if using ffserver, how could I add the variableName.avi stream without having to add it to the conf file first? there are going to be many streams of many files at once.. :/ [20:30] <c_14> You want one output stream that's a concatenation of all the inputs? [20:31] <gnm|> several streams from several sources... each source can be sent to many at once.. [20:31] <c_14> hmm? [20:32] <gnm|> i.e. mediaserver [20:33] <c_14> A 'mediaserver' would probably spawn 1 ffmpeg per request [20:34] <c_14> But I'm not entirely sure what you're trying to do [20:36] <gnm|> each file on server can be sent as input to many users at once... and every file in theory could be streamed at once.. (in theory) [20:38] <c_14> If a user requests a file and then another user requests the same file while user1 is watching/listening to file1, will user2 start the file from the start or from where user1 currently is? [20:38] <gnm|> guess fork ffmpeg would be best way to to it... one fork for each stream.. :P [20:39] <gnm|> from start [20:39] <c_14> fork() an ffmpeg for each request [20:39] <c_14> Do you want to reencode the source or provide it as-is? [20:39] <kepstin-laptop> gnm|: depending what you're trying to do, it might make sense to create static dash manifests and just serve everything with a high-performance http server. [20:39] <c_14> ^ It does sound a lot like he wants to create a web-server [20:40] <gnm|> depends on client, so transcode is wanted... [20:40] <gnm|> kep, Il read up on it, thnx [20:40] <gnm|> c_14, thnx :) [20:40] <kepstin-laptop> live transcoding would be ridiculous, unless you're doing really low-volume streaming. Just pre-encode a couple of variants and serve them statically. [20:41] <kepstin-laptop> (if you're doing live video, that's a different story, of course, but you'd still only want to do the encoding once, then distribute the encoded stream) [20:42] <gnm|> its basicly a mix of both, guess I could use ffserver for the live part, and ffmpeg for everything stored [20:42] <kepstin-laptop> for the stuff that's not live, just encode to hls or dash, serve with nginx or something, put behind a cdn, done. [20:44] <kepstin-laptop> or even just static mp4/webm served in http would be fine for playing in desktop browsers [20:46] <gnm|> It has to work with android and ios devices as well, so I guess hls an/or dash would work fine? [20:50] <gnm|> you have been of great help! thanks! now over to the documentation ) [20:50] <gnm|> :) [21:04] <Bellspringsteen> Hello, would this be an appropriate place to ask a question on solving a problem encoding a MXF File with a Closed Caption vbi_vanc_smpte_436M stream? [21:07] <Bellspringsteen> The file looks like this [21:07] <Bellspringsteen> Input #0, mxf, from '/Users/XXX/Downloads/ExampleCopy.mxf': [21:07] <Bellspringsteen> Metadata: [21:07] <Bellspringsteen> uid : 138157ae-9bbe-1b41-871b-1bfc102cb188 [21:07] <Bellspringsteen> generation_uid : 8af54e43-25e4-184a-b85a-87bb9d8a4d8d [21:07] <Bellspringsteen> company_name : AVID [21:07] <Bellspringsteen> product_name : TRMG [21:07] <Bellspringsteen> product_version : 3.01 [21:07] <Bellspringsteen> product_uid : 00000000-0000-0000-0000-000000000000 [21:07] <Bellspringsteen> modification_date: 2014-12-31 19:29:53 [21:07] <Bellspringsteen> material_package_umid: 0x060A2B340101010501010D1313000000E785EB19DB75A340B703A96982AF2957 [21:07] <Bellspringsteen> timecode : 01:00:00;00 [21:07] <Bellspringsteen> Duration: 00:01:00.06, start: 0.000000, bitrate: 60846 kb/s [21:07] <Bellspringsteen> Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 50000 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn, 119.88 tbc [21:07] <Bellspringsteen> Metadata: [21:07] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:07] <Bellspringsteen> Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:07] <Bellspringsteen> Metadata: [21:07] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:07] <Bellspringsteen> Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:07] <Bellspringsteen> Metadata: [21:07] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:07] <Bellspringsteen> Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:07] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> Stream #0:9: Data: none [21:08] <Bellspringsteen> Metadata: [21:08] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [21:08] <Bellspringsteen> data_type : vbi_vanc_smpte_436M [21:08] <ezekiel> /ignored [21:09] <ezekiel> Bellspringsteen use a paste service like paste.debian.net [21:10] <Bellspringsteen> whoops, thanks, see here http://pastebin.com/VR6mqnqr [21:13] <Bellspringsteen> In the MXF File, I want to replace the first two audio streams in the MXF file with two separate audio streams. This works if I ignore the ninth stream, the CC data stream. But I need this CC stream, when I include this, it says the CODEC is not supported. [21:29] <Phlarp> random question that isn't necessarily FFMPEG related but maybe someone can give me some insight: [21:29] <Phlarp> I'm looking for a good video streaming service. Something I could drop into a webapp to stream one way video to a page? [21:31] <Phlarp> Essentially I'd need the functionality of chat roulette, but with IOS support [22:00] <sfan5> Bellspringsteen: did you use -c copy? [22:00] <Bellspringsteen> @sfan5 I am using -acodec copy -vcodec copy -dcodec copy [22:00] <Bellspringsteen> is -c copy different? [22:00] <sfan5> it does exactly the same [22:01] <Bellspringsteen> here is the total output [22:01] <Bellspringsteen> http://pastebin.com/VR6mqnqr [22:01] <Bellspringsteen> yes, then I am already doing that [22:03] <sfan5> Bellspringsteen: whats the error ffmpeg gives you? [22:04] <Bellspringsteen> [mxf @ 0x7fe0b2a42000] track 3: could not find essence container ul, codec not currently supported in container [22:04] <Bellspringsteen> for the data stream [22:05] <Bellspringsteen> Stream #0:3: Data: none [22:05] <Bellspringsteen> Metadata: [22:05] <Bellspringsteen> file_package_umid: 0x060A2B340101010501010D13130000006D9F377935593C4280BA79DD0090270A [22:05] <Bellspringsteen> data_type : vbi_vanc_smpte_436M [22:05] <sfan5> looks like the mxf muxer needs to know the format to embed the stream [22:40] <ramp> is the error "Non-monotonous DTS in output stream" a problem? [22:40] <ramp> or something to be concerned about? [22:43] <Bellspringsteen> @sfan5 is there a way to force the mxf muxer the format of the data stream? Or tell it the format? [22:47] <kepstin-laptop> ramp: it depends on the output format. In some formats, non-monotonous dts isn't allowed, in others it's ok. [22:50] <ramp> i'm outputting an mkv [22:50] <ramp> or rather, i'm taking a set of mkv's and removing some of the streams [22:55] <ramp> anyhow I just wanted to check if this was cause for concern [00:00] --- Fri Jan 30 2015

1 0

ffmpeg-devel.log.20150128
by burek 29 Jan '15

29 Jan '15

[00:01] <michaelni> llogan, added for ffmpeg-trac [01:18] <llogan> michaelni: thanks [01:27] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:840570d27b71: avfilter/vf_mcdeint: avoid uninitilaized fields in AVPacket [05:41] <cone-454> ffmpeg.git 03Carl Eugen Hoyos 07master:449b8cf04b7e: avformat/mxfdec: Fix cleanup in mxf_free_metadataset() [06:09] <cone-454> ffmpeg.git 03Mark Reid 07master:a181169868fe: tests/fate: added mxf tests for essencegroups and missing index [06:28] <cone-454> ffmpeg.git 03Timo Rothenpieler 07master:fb34c580bb24: avcodec/nvenc: Handle non-square pixel aspect ratios [10:03] <cone-110> ffmpeg.git 03Maneesh Gupta 07master:1600f85cbc45: doc/fftools-common-opts: update/extend documentation for -opencl_bench option [11:40] <durandal_1707> is anybody here with latest mplayer binary? [11:53] <wm4> durandal_1707: just build it [11:53] <durandal_1707> i tried and failed [11:53] <wm4> not many distros package an uptodate mplayer [11:53] <wm4> failed how? [11:54] <durandal_1707> in linkage, missing symbols [11:54] <wm4> just remember to check it out with svn [11:54] <wm4> if it's a build regression, try with an earlier ffmpeg, or poke michaelni or reimar [11:54] <durandal_1707> nope, this is some local nonsense [11:56] <durandal_1707> iive: do you have mpeg2 that is soft-telecined? [11:58] <kierank> iirc it's possible to make with x262 [11:58] <kierank> can't remember though if that was implemented [11:58] <kierank> tbh I don't even understand what the soft pulldown filter is even trying to achieve [11:59] <nevcairiel> i thought its meant to convert to hard-telecine [11:59] <nevcairiel> but i may be wrong [11:59] <kierank> so it's a pulldown filter [11:59] <durandal_1707> yes it changes soft-telecined material into hard-telecined [11:59] <kierank> soft pulldown means the pulldown is just flagged [12:00] <kierank> then the filter name is bizzare [12:00] <kierank> why not just call it pulldown [12:01] <kierank> and also pts needs to change and afaik pts doesn't [12:01] <wm4> mplayer2 had changes to correct the pts [12:02] <wm4> in vf_softpulldown [12:02] <kierank> the filter should therefore fail on anything not 24000/1001 [12:15] <nevcairiel> wm4: i wonder how this osx guy wants to build a multimedia api without ffmpeg behind it ... its not like he has any alternative multimedia frameworks to pick from :p [12:15] <durandal_1707> gstreamer [12:15] <j-b> VDA :) [12:15] <nevcairiel> without the ffmpeg plugins, gstreamers support is minimal [12:16] <j-b> webm+ogg! [12:19] <wm4> nevcairiel: didn't he say he wants to use ffmpeg, just with minimizing contact with it? [12:21] <rcombs> nevcairiel: QuickTime! [12:22] <durandal_1707> why there is no wrapper around x262 in lavc? [12:23] <BtbN> 262? [12:24] <av500> x261 ftw! [12:28] <durandal_1707> x262 have --pulldown flag [12:28] <ubitux> wtf @ #2078 [12:28] <ubitux> people though they are on github or what [12:31] <BtbN> It's... a dog oO [12:36] <durandal_1707> x262 --mpeg2 --pulldown 22 -o /tmp/o.mpeg ~/matrix.mpg: simply crashes [12:44] <wm4> ubitux: lol... it's always funny if some poor project gets bombed with retards in an issue because it got linked all over the internet [12:44] <wm4> social coding etc. [12:55] <kierank> wm4: lol [12:55] <kierank> durandal_1707: might not be implemented then [13:00] <compn> durandal_1707 : ask dalias if he has any samples for soft pulldown [13:00] <compn> he worked on those filters iirc [13:00] <kierank> why do you need samples [13:01] <kierank> unless you want to ivtc and then pulldown back and compare? [13:01] <durandal_1707> to test filter with actual file that have this flags [13:01] <compn> kierank : to see if durandal_1707 ported it correctly i guess :P [13:01] <durandal_1707> i believe that mp=softpulldown default behavior is wrong [13:02] <kierank> oh i see [13:04] <ubitux> wm4: thanks for the "social coding" words, that's what i was looking for [13:05] <BtbN> Well, it gets fun when someone complains that your code using master/slave terminology is offensive. [13:05] <kierank> FFmpeg-gate [13:05] <compn> oh man that master/slave debate, classic [13:06] <BtbN> There actualy is a new, politicaly correct, form of master/slave, but i forgot what it was. [13:06] <nevcairiel> it sounds stupid though, tech terms should need to be "politically correct" [13:07] <nevcairiel> should not* [13:08] <BtbN> It _is_ stupid. [13:09] <ubitux> slave mode of mplayer should be renamed apartheid mode to avoid any confusion [13:09] <compn> trigger warning! [13:09] <compn> :P [13:32] <iive> durandal_1707: i think i have, but right now i'm busy, ask me again in few hours [13:38] <JoshX> Hello, I made a little addiotion to libavformat/utils.c [13:38] <JoshX> so that the filename pattern could also use %t to add Ymd_HMSfff [13:39] <JoshX> useful for segmentation of files with start timestamp in the filename [13:39] <JoshX> now, i'm fairly new to github aqnd have never 'really' contributed or tried to get a simple add to a big opensource project [13:40] <JoshX> but my guess is, this could be quite useful for more people so it might be nice to merge it in a way [13:40] <JoshX> questing really is, 'how' :-) [13:46] <kierank> JoshX: use git format-patch and send a patch to ffmpeg-devel(a)ffmpeg.org [13:50] <JoshX> ok lets see if i can get that to work :) [13:53] <cone-110> ffmpeg.git 03Michael Niedermayer 07master:ccaa5dcb3135: avcodec/h264_parser: Rename close() [13:54] <cone-110> ffmpeg.git 03Michael Niedermayer 07master:0d37ca150c5e: avutil/aes: Rename crypt() [13:55] <compn> JoshX : or even a diff -u of the two files [13:55] <compn> if you know the command line diff program [13:57] <wm4> compn: no, shut the fuck up [13:58] <wm4> git format-patch is more convenient for everyone, don't go around and suggest lesser, ahrder methods [13:58] <wm4> *harder [13:59] <arwa_> Hey, When I am using fspp for a paricular image, I am getting a green patch in the bottom. [14:00] <arwa_> Can anyone tell me what can be the possible reason? [14:00] <durandal11707> what is size of image? [14:03] <JoshX> kierank, compn, no found the howto :-) branched it, changed it, committed it and created the patch :) [14:03] <JoshX> have to learn git anyway [14:04] <arwa_> It is 480X260 [14:04] <wm4> and once you know it, writing patches will be much easier [14:04] <JoshX> wm4: agreed, that's why i did it the 'right' way :) [14:05] <JoshX> ah it's held on the list since i'm not a member :) [14:07] <JoshX> i understand.. I'm trying to make a dvr-like sollution to save a h264 mp4 stream from an ip camera to 900 second chunks.. i don't really want to transcode anything since the format and bitrate of the camera are good for now.. so I try to use 'copy' as vcodec and -an (no audio) but i get a lot of warnings about non-monotonous dts frames.. [14:08] <compn> JoshX : accepted your mail from the mod queue [14:08] <compn> should be posted now. [14:08] <wm4> JoshX: subscribing would make things easier [14:08] <JoshX> cool thanks :) [14:08] <JoshX> yes.. just mail with subscribe as subject to the list?\ [14:09] <durandal11707> arwa_: can't reproduce with ffplay, does it depends on specific image? [14:09] <JoshX> ah found the subscribe page, and subscribed [14:09] <wm4> JoshX: don't know if this works; also see http://www.ffmpeg.org/mailman/listinfo/ffmpeg-devel [14:10] <JoshX> wm4: google beat you to it ;) [14:10] <arwa_> Yes, because, I tried it with different images, but for one specific image I was getting this. [14:10] <compn> wm4 : btw there is vf_overlay in mplayer fork i forgot about ;D [14:10] Action: compn runs and hides [14:10] <wm4> compn: which fork? [14:11] <compn> humm [14:12] <compn> no idea [14:12] <compn> http://permalink.gmane.org/gmane.comp.video.mplayer.devel/55122 [14:13] <compn> another vf patch , http://onebithq.com/root/mplayer/videomixer [14:13] <compn> shared mem filter :P [14:13] <wm4> that looks like a rejected patch [14:13] <wm4> and also useless [14:13] <compn> yep [14:14] <wm4> it mixes memalign and av_freep so it's buggy too [14:15] <wm4> compn: what was the conclusion of this thread? [14:18] <JoshX> oh my patch fails... :-(( going to fix it [14:23] <wm4> JoshX: it also has a buffer overflow and undefined behavior [14:23] <wm4> replying on the list... [14:25] <wm4> done [14:25] <ubitux> arwa_: can you share the image? [14:26] <arwa_> Okay. [14:28] <JoshX> wm4 ok.. i'm not a real c developer, but it worked for me.. now the build fails, trying to figure out what i've done different [14:28] <JoshX> can i pm you? [14:28] <arwa_> http://imgur.com/KwJhGny [14:29] <wm4> sure... [14:33] <ubitux> arwa_: asserts for me here [14:33] <ubitux> Assertion frame->height == link->h failed at libavfilter/avfilter.c:1170 [14:34] <arwa_> Okay, I am not getting any error! [14:34] <ubitux> that's normal, i have a higher assert level [14:34] <arwa_> And why is there an assertion failure? [14:34] <ubitux> basically the frame your sending doesn't match the configuration of your outlink [14:34] <arwa_> So, what should I do? [14:36] <ubitux> arwa_: you need to fix the output frame dimension [14:36] <ubitux> arwa_: fspp is requesting a width and height 8 aligned buffer [14:36] <ubitux> but that doesn't match the real dimension, is just to avoid overread/overwrite [14:37] <ubitux> so after getting the output frame, you need to reset them to the expected dimension [14:37] <ubitux> look how it's done in vf_pp [14:37] <ubitux> (see lines 129 to 136) [14:37] <arwa_> Okay, thanks :) [14:37] <ubitux> also make sure you haven't made the same mistake in the other pp based filters [14:37] <arwa_> So, I need to fix the code? [14:37] <ubitux> yes [14:38] <ubitux> pp7 has the same problem [14:38] <arwa_> Yes, it is giving the same kind of results for pp7 also. [14:38] <ubitux> please send a patch for both of them [14:38] <arwa_> I will write a patch correcting it. [14:38] <ubitux> (1 patch for each filter please) [14:38] <ubitux> thank you [14:39] <arwa_> Okay :) [14:40] <ubitux> btw, you might want to factorize that code between all the pp filters [14:40] <ubitux> same for the dither tables [14:41] <arwa_> by factorize, do you mean including it in internal.h? [14:41] <ubitux> the prototype maybe, but not the code [14:42] <ubitux> not sure where it could belongs, maybe a libavfilter/utils.c, dunno [14:42] <ubitux> or simply avfilter.c [14:42] <JoshX> ok building now with the fixes you suggesed in place, wm4 :) [14:42] <JoshX> *suggested [14:43] <ubitux> arwa_: you understand the issue about the aligned thing? (not how to fix it, but more the meaning of this) or you want more explanations? [14:43] <arwa_> I would like to know more [14:44] <arwa_> I didn't exactly understand. [14:44] <ubitux> ok so, most pp filters (maybe all?) code needs to work by block of 8x8 minimum [14:44] <ubitux> to avoid some boundary checks (they just process the whole 8x8 block [14:45] <ubitux> so when the frame doesn't match that "grid" (because of non multiple of 8 dimensions) there is no special code [14:45] <ubitux> but we still want to output a frame that is not a multiple of 8 [14:45] <arwa_> Oh, okay. [14:45] <ubitux> we just want to buffer to have a sufficient size to avoid writing or reading out of memory [14:46] <ubitux> so basically we request a frame in larger dimensions [14:46] <ubitux> and then actually fix the dimension to match the one frome the input, hiding these stashing areas [14:46] <arwa_> Oh, nice!! [14:46] <ubitux> and lavfi asserted here saying that the out link was configured to match the exact same dimension as the input [14:46] <ubitux> but you were sending frame that were larger [14:47] <ubitux> (because aligned to the grid) [14:47] <ubitux> anyway, hope that clears things up [14:47] <wm4> ubitux: does lavfi have any magic mechanism to request padded frames from preceding filters? [14:48] <ubitux> not sure what you mean :p [14:48] <wm4> well if a filter "requests" a larger frame than the input size, what exactly happens? [14:49] <arwa_> Thank you. It is giving me an overall idea of what is happening. [14:50] <ubitux> wm4: i wonder, maybe there is a pool somewhere so that frame is reused or something :) [14:50] <ubitux> but i don't know [14:51] <ubitux> /* TODO: set the buffer's priv member to a context structure for the whole [14:51] <ubitux> * filter chain. This will allow for a buffer pool instead of the constant [14:51] <ubitux> * alloc & free cycle currently implemented. */ [14:51] <ubitux> seems not :( [14:51] <wm4> heh [14:51] <ubitux> i thought we had some at one point [14:52] <wm4> there's one in lavc [14:52] <ubitux> it probably died as a casuality of the evil plan or something [14:53] <ubitux> yeah michael added one in lavfi 4 years ago [14:54] <ubitux> but it disappeared somehow, too bad [14:54] <cone-110> ffmpeg.git 03Paul B Mahol 07master:87577f5508ca: avfilter/vf_telecine: use the name 's' for the pointer to the private context [14:57] <ubitux> lol we still have that unused pool structure [14:58] <ubitux> would be nice to have a pool of 4 or 8 frames [14:59] <wm4> ideally there'd be a pool that can be used by both lavfi and lavc and API users [15:00] <ubitux> there is a lavu/AVBufferPool could be used [15:00] <ubitux> but the FramePool itself is local to avcodec afact [15:00] <wm4> yeah [15:00] <wm4> it's messed directly into utils.c [15:00] <wm4> (who needs code reuse) [15:01] <ubitux> could be an interesting work to factorize them [15:01] <ubitux> but not my priority right now, we might want to open a ticket or something [15:14] <JoshX> hmm could it be that what i am trying to do (and did against a source from the beginning of december) is already in the codebase now?? [15:15] <nevcairiel> anything is possible. what are you trying to do? [15:16] <JoshX> i'm trying to make filenames with the timestamp to the millisecond using the segement feature [15:16] <JoshX> and it only did %d before [15:16] <nevcairiel> wasnt strftime support added recently [15:16] <JoshX> but now there seems to be a strftime feature [15:16] <JoshX> ghe possibly :) [15:16] <JoshX> it would fix my problem ;) [15:17] <nevcairiel> i guess the answer is yes then [15:17] <JoshX> haha [15:17] <wm4> yeah, always check git master before trying to fix or add something [15:17] <JoshX> { "strftime", "set filename expansion with strftime at segment creation", OFFSET(use_strftime), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, E }, [15:17] <JoshX> just found this debugging my code ;) [15:17] <JoshX> well my idea wasn't half bad then ;) [15:17] <JoshX> now to see how to use it then ;) [15:18] <ubitux> it was added in 2013 [15:19] <wm4> that feature sucks a bit [15:20] <wm4> you can only have a strftime format, or ffmpeg's own one [15:20] <JoshX> its not in the examples, manpage, help, whatever?? [15:20] <Daemon404> i want a feature to disable filename expansion [15:20] <wm4> can't mix them [15:20] <Daemon404> i should send a patch [15:20] <Daemon404> because our filename expansion is applied to urls [15:20] <Daemon404> and complete breaks escaped urls [15:20] <wm4> lol [15:20] <wm4> maybe it shouldn't be applied to URLs [15:20] <Daemon404> im sure some person will claim to use it [15:21] <ubitux> ah i was actually mistaking it for the strftime img2 thing [15:21] <wm4> no, someone will claim that someone else might use it [15:21] <wm4> (ffmpeg business as usual) [15:21] <Daemon404> yep [15:21] <Daemon404> theoretical "users" [15:21] <wm4> every feature is sacred, every feature is good [15:22] <nevcairiel> speaking of segment, i've had a rather random failure with it, where the webserver serving the segments and playlist would return an empty playlist .. i think its a race caused by the code just re-writing the playlist in that instant ... [15:22] <nevcairiel> didnt I see a function in some other muxer to write to a tmp file and then replace the original in one operation after [15:22] <nevcairiel> i should copy that [15:24] <JoshX> ah.. my main problem now is that strftime does not provide milliseconds... [15:25] <Daemon404> microseconds is not enough? [15:26] <Daemon404> is you file like 9 billion fps? [15:26] <JoshX> haha [15:26] <JoshX> nope [15:26] <nevcairiel> micro is actually smaller than milli Daemon404 :p [15:26] <Daemon404> the sad part is that i grew up in a fully metric and SI country [15:26] <Daemon404> and forgot that [15:26] <JoshX> what would be the strftime pattern for YYYYmmdd_HHmmssfff? [15:26] <nevcairiel> shouldnt %f be milli [15:26] <JoshX> it it in a lot of languages [15:26] <JoshX> lets see [15:27] <Daemon404> yeah it could be MS [15:27] <Daemon404> %f is micro, so you can round [15:28] <JoshX> 20150128_152740%f.mp4 [15:28] <Daemon404> hmm [15:28] <JoshX> well.. [15:28] <Daemon404> pubs.opengroup.org/onlinepubs/009695399/functions/strftime.html [15:28] <Daemon404> POSIX strftime has no micro [15:28] <JoshX> indeed [15:28] <wm4> linux4lyfe http://linux.die.net/man/3/strftime [15:28] <JoshX> you need to use gettimeofday [15:28] <wm4> MORE FEATURES [15:28] <nevcairiel> are you really going to produce more than one segment per second so that milliseconds are required [15:29] <JoshX> i produce mp4 video which need to be synced to output of another system so the starttime must be ms-based [15:29] <JoshX> and then i count frames to get to the exact position i need to be [15:29] <nevcairiel> and you are syncing over the filename? that seems weird [15:30] <JoshX> can i get the timecode to the ms per frame in mp4 then? [15:31] <JoshX> it would be much better for me if i could put the exact timecode per frame.. [15:32] <JoshX> either in a seperate index file or embedded in the mp4 [15:32] <JoshX> i thought about abusing a subtitle field for that.. [15:33] <JoshX> nevcairiel: any idea? [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:231bf4d1c21c: avfilter/vf_colorbalance: use the name 's' for the pointer to the private context [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:abcf8610b8b3: avfitler/vf_il: use the name 's' for the pointer to the private context [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:f27cda48ec64: avfilter/vf_extractplanes: use the name 's' for the pointer to the private context [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:6121968a9e87: avfilter/af_biquads: use the name 's' for the pointer to the private context [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:67aaa7056d2b: avfilter/avf_avectorscope: use the name 's' for the pointer to the private context [15:43] <cone-110> ffmpeg.git 03Paul B Mahol 07master:fca50464dc7f: avfilter/vf_colorchannelmixer: use the name 's' for the pointer to the private context [15:44] <ubitux> haha [15:44] <ubitux> i can drop my most-boring-thing-ever-and-i-dont-want-to-finish-this-shit i guess now [15:44] <ubitux> +branch [15:45] <durandal_1707> there are still some filters left [15:45] <durandal_1707> i keep mistyping filter [15:46] <ubitux> if you're into a rename rampage in lavfi, you could rename the "picref" into frames [15:46] <saste> what about doing useful work instead? [15:46] <ubitux> :D [15:46] <ubitux> yeah, if you want to do useful stuff, you can add the frame pool we talk about today [15:47] <durandal_1707> that is too hard [15:47] <ubitux> really? [15:47] <ubitux> it needs a bit of work but it doesn't look that hard [15:48] <ubitux> i wonder how much the pool mecanism helps in comparison to modern malloc implementation [15:57] <wm4> ubitux: there's also some hidden cost, because these large allocations will go through mmap, which means the kernel has to zero pages in the background [15:57] <wm4> but all in all it probably doesn't matter that much [15:59] <durandal_1707> michaelni: i added av_frame_make_writable() right after cloning and it doesn't help [16:00] <wm4> no, you need to call it right before you write to a frame [16:01] <durandal_1707> huh, why? [16:03] <wm4> because that's how reference counting works [16:03] <durandal_1707> ahh it WORKS now [16:58] <JoshX> anyone who can help me with the timecode thing? i need a way to reproduce the time of the first frame to the ms when using the segment filter to be able to jump to a frame in a video file based on a timestamp to the ms [16:59] <ubitux> durandal_1707: .needs_writable? or it's not necessary all the time? [16:59] <JoshX> i'm saving a h264 stream from an ip camera in 15 minut chunks and i just need the timecode of the first frame (or the timecode of any frame) or an index file with frametimes/numbers [16:59] <JoshX> i can live with any of the 3 :) [17:13] <ubitux> hey, the void & cluster ordered ("blue noise") dithering is kind of interesting [17:14] <ubitux> i'm probably going to add this one instead of the ordered one [17:15] <anshul_mahe> ubitux: are u intrested to check that c608 patch, I was planning to move on at the patch by sam [17:17] <ubitux> sam? [17:17] <ubitux> you mean shan? [17:17] <anshul_> yes, I forgot his name [17:17] <wm4> ubitux: mpv does use something related to void & cluster (although it's static) [17:18] <ubitux> yes that's the point :) [17:18] <ubitux> error diffusion is a nightmare for gif encoding, or still frames in general [17:18] <ubitux> so i need to have a good static one [17:18] <ubitux> (assuming that's what you meant by static) [17:19] <ubitux> anshul_: i don't plan to do anything, but i'm happy to review some patches i guess [17:19] <ubitux> anshul_: also, do we have a FATE test for CC or not yet? [17:19] <anshul_> no should I give it priority over shan patch [17:20] <ubitux> because in addition to improve the coverage, it will also show how the markup evolve :) [17:20] <anshul_> ok, So I would make one fate test first [17:21] <ubitux> yeah, ideally [17:21] <anshul_> do we have fate test for other subtitles, I would take one for refrence fate test [17:21] <ubitux> yes, almost all of them [17:21] <ubitux> anshul_: look at 6dc99fdf0e4328ca2992b7bfcd139f7b1b636bf8 [17:22] <ubitux> it includes a test [17:24] <durandal_1707> ubitux: if i got i right it should not be necessary all the time, like when repeat_pict is used [17:24] <cone-51> ffmpeg.git 03Arwa Arif 07master:622936424c69: avfilter/vf_fspp: Fix frame-alignment in FSPP [17:24] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:375a0273cec4: avfilter/vf_fspp: check count before calling row_idct() [17:24] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:a6f9a5d0f635: avfilter/x86/vf_fspp: Fix loop condition for column_fidct() [17:24] <ubitux> durandal_1707: ok [17:25] <anshul_> ubitux: thanks [17:25] <ubitux> anshul_: make fate-foobar GEN=1 is your friend [17:25] <ubitux> make sure you select a sample with advanced "markup" [17:25] <ubitux> and make sure the dependencies for that sample are correct [19:06] <cone-51> ffmpeg.git 03Paul B Mahol 07master:c099783316cf: avformat/nsvdec: remove case which is no longer possible [20:35] <cone-51> ffmpeg.git 03Clay McClure 07master:6a808f5ae17f: libdc1394: Add support for MONO8 (gray) video mode [20:35] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:37984ca13334: Merge commit '6a808f5ae17f1fcdbcfb18055872c12aef70ffff' [21:03] <llogan> the xcb x11grab -y avoption seems to conflict with the -y global option. can someone confirm? [21:04] <llogan> ffmpeg -y -f x11grab -x 100 -y 50 -i :0.0 outptu.mp4 [21:04] <llogan> console "outptu" then squawks ":0.0: Protocol not found" [21:06] <llogan> does the same w/o -y flobal [21:06] <llogan> *global [21:16] <cone-51> ffmpeg.git 03Luca Barbato 07master:3c18a7b18807: avio: Do not consider the end-of-buffer position valid [21:16] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:0b4fc4bacd86: Merge commit '3c18a7b18807de81566381a1bcbe9f6103c0296b' [21:16] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:58ed1528574f: avformat/aviobuf: Allow seeking to the end of the buffer for read only buffers [21:33] <michaelni> llogan, you can pass x/y in the filename +%d,%d style [21:34] <llogan> i know, but the -x and -y are listed as AVoptions and that may confuse users who will attempt it that way [21:45] <michaelni> llogan, indeed, feel free to remove or document, whatever you think is better [21:52] <iive> :O [21:55] <llogan> michaelni: i'd prefer that the AVoption does not get confused with the global one (and keep it documented) but i don't know how to do that. [21:56] <llogan> for now i just didn't include -x and -y in indevs.texi [22:24] <ubitux> arwa: it might be interesting to add the reference image as well [22:25] <ubitux> (because sometimes we wonder what we would ideally get from that square garbage :D) [22:25] <arwa> Yes, I have added the reference image. [22:25] <ubitux> ah, cool :) [22:25] <arwa> I have one ques about pp filter. [22:26] <arwa> I am confused on what options are offered by the filter. [22:26] <arwa> I read the documentation, but I am not able to change the parameters. [22:26] <arwa> For example: [22:28] <arwa> pp=hb|40 is not working [22:28] <arwa> Is there some syntax error? [22:33] <michaelni> -vf pp=hb|40 would need some quoting like with '' or "" [22:35] <michaelni> you can use : instead of | IIRC [22:37] <arwa> Okay thanks :) [22:39] <michaelni> both -vf 'pp=hb|40' and -vf pp=hb:40 should work [22:44] <arwa> It i working. Thank you. [22:54] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:61928b68dc28: h264: Do not share rbsp_buffer across threads [22:54] <cone-51> ffmpeg.git 03Michael Niedermayer 07master:72db3c79dfdd: Merge commit '61928b68dc28e080b8c8191afe5541123c682bbd' [23:02] <arwa> I am getting error with this - 'pp=hb/vb/dr/fq=8' [23:02] <arwa> [Parsed_pp_0 @ 0xb8e1980] Option 'hb/vb/dr/fq' not found [23:16] <ubitux> heh, hannibal looks so sweet after deblocking :3 [23:17] <arwa> I know!! :P [00:00] --- Thu Jan 29 2015

1 0

ffmpeg.log.20150128
by burek 29 Jan '15

29 Jan '15

[00:02] <kc8hfi> i have a pcm audio file and a raw uncompressed video file. when I use ffmpeg to combine and encode these to mpeg/ac-3, the audio falls out of sync. [00:03] <kc8hfi> can I encode the video separately, then the audio separately, and then use ffmpeg to combine the two together without losing the audio sync? [00:16] <iive> kc8hfi: there is no problem to encode the video without audio and then try to mux them both at second stage. [00:16] <iive> however I doubt that you'd get a different result. [00:17] <iive> processing audio and video separately is risky thing, because you lose sync data. [00:17] <iive> e.g. is the pcm from a live source, capture or sound card? [00:19] <iive> these are known to use imprecise oscillators that result in drift in the audio frequency... [00:25] <kc8hfi> it came from a capture card [00:26] <kc8hfi> i captured the raw video and audio at the same time into a huge file. but the audio and video fell out of sync. so i split the audio and video into 2 pieces. [00:27] <kc8hfi> Then I used openshot to try and resync the audio and video. however, when playing back both tracks side by side, there were no audio/video sync problems. So I went ahead and let openshot encode them both in another file. the audio and video were in sync for awhile, but then came out of sync. [00:31] <kc8hfi> http://fpaste.org/176405/14224013/ is the command i'm using to encode just the video. [00:34] <kc8hfi> http://fpaste.org/176410/42240160/ is the command i'm gonna try to encode both the video and audio together into the output file. I haven't tested this yet to see if its syntactically correct yet. It should be close if it isn't [00:35] <kc8hfi> dang, something bad happened, http://fpaste.org/176413/40173814/ [01:07] <kc8hfi> any idea why i'm getting those buffer underflow errors like that? [01:59] <michael__> Hello [02:00] <michael__> I have some videos which have been sped up with an unknown program [02:00] <michael__> They are 119.9115782% normal speed [02:01] <michael__> The method of speeding up seems to have simply increased the framerate, and adjusted the audio accordingly [02:02] <michael__> I can get it to play at normal speed with the command ffmpeg -i input.mp4 -filter_complex "[0:v]setpts=1.199115782*PTS[v];[0:a]atempo=0.833947826[a]" -map "[v]" -map "[a]" -r 23.976 output.mp4 [02:03] <michael__> Unfortunately, although it is running slower, it is still high pitch. How can I modify the above command to reduce the pitch? [02:11] <michael__> I have some videos which have been sped up with an unknown program. They are 119.9115782% normal speed. The method of speeding up seems to have simply increased the framerate, and adjusted the audio accordingly. I can get it to play at normal speed with the command 'ffmpeg -i input.mp4 -filter_complex "[0:v]setpts=1.199115782*PTS[v];[0:a]atempo=0.833947826[a]" -map "[v]" -map "[a]" -r 23.976 output.mp4'. Unfortunately, although it is running [02:11] <michael__> slower, it is still high pitch. How can I modify the above command to reduce the pitch? [02:13] <michael__> I am now trying with asetpts instead [02:33] <michael__> Ok, that just results in flickering audio [02:34] <michael__> I'll try asetrate [03:12] <k_sze[work]> What's a good container format that would allow me to quickly get a frame count while the video is still being written? [03:13] <k_sze[work]> It appears that if I use raw h.264, ffmpeg needs to try and decode every frame just to get the stats and frame count. [03:14] <relaxed> frame count of what's been encoded so far? [03:14] <k_sze[work]> yes [03:15] <relaxed> ffmpeg's output should tell you [03:15] <relaxed> frame= <number> [03:16] <k_sze[work]> Problem is that the video is remotely encoded. [03:17] <relaxed> have it copy stderr to a file and parse that for the frame count [03:18] <k_sze[work]> the frame count in stderr is guaranteed to be flushed to disk? [03:20] <relaxed> hmm, good point. [03:21] <relaxed> are you monitoring progress or what? [03:22] <k_sze[work]> I have a remote machine that captures video and writes it as H.264 over samba, to this local machine. [03:23] <k_sze[work]> And I would like to periodically check the frame count on the local machine, programmatically. [03:24] <k_sze[work]> I need assurance that the frame count I get is the number of frames *flushed to disk* and immediately decodable. [03:24] <relaxed> the local machine is running Windows? [03:24] <k_sze[work]> no, Debian [03:25] <relaxed> I think matroska would be a good choice. [03:35] <michael__> If you use the sync command, it will be flushed to the disk [03:35] <k_sze[work]> michael__: you mean using the libav* API directly? [03:36] <michael__> k_sze[work]: No [03:36] <k_sze[work]> Then what sync command? [03:37] <michael__> On my system (Linux Mint), there is a command sync which flushes disk caches [03:37] <michael__> If you're using Mac, I assume it's there too [03:38] <michael__> Append the '&' symbol (no quotes) to the ffmpeg command and you can run sync in the same terminal [03:47] <k_sze[work]> And I'm having trouble remuxing a H.264 avi file to mkv [03:47] <k_sze[work]> I get Can't write packet with unknown timestamp av_interleaved_write_frame(): Invalid argument [03:48] <k_sze[work]> I've already tried adding -fflags genpts and it's still the same. [03:49] <k_sze[work]> oh, never mind [03:49] <k_sze[work]> need to put that *before* the -i [04:01] <coherence> welp, solved my issues - had to twiddle the start time bits using libavformat, and writing the playlist out myself at the same time. [04:01] <coherence> cheers! [04:02] <Godface> can anyone help me correct this error? "ffmpeg: error while loading shared libraries: libass.so.4: cannot open shared object file: No such file or directory" [04:02] <c_14> Did you compile ffmpeg yourself? [04:03] <Godface> yes [04:03] <c_14> What prefix? [04:04] <Godface> prefix? [04:04] <c_14> You did `make install', right? [04:04] <Godface> yes [04:04] <Godface> i used this https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [04:05] <c_14> You installed libass-dev ? [04:05] <Godface> yes [04:05] <c_14> does /usr/lib/libass.so exist? [04:08] <Godface> c_14: no [04:08] <c_14> Can you pastebin the output of `dpkg -L libass-dev' ? [04:09] <Godface> http://pastebin.com/uM1fcAuc [04:11] <c_14> How about `ldd /path/to/ffmpeg' ? [04:13] <Godface> how can i find the path? [04:13] <c_14> which ffmpeg [04:13] <Godface> ldd: ./ffmpeg: not regular file [04:14] <c_14> `which ffmpeg' returns ./ffmpeg ? [04:16] <Godface> c_14: http://pastebin.com/7aDNyCNn [04:17] <c_14> Ehhh, what does `/home/admin/bin/ffmpeg -version' output? [04:18] <Godface> ffmpeg version 2.5.git Copyright (c) 2000-2015 the FFmpeg developers. built on Jan 27 2015 10:26:29 with gcc 4.9.1 (Ubuntu 4.9.1-16ubuntu6) [04:19] <c_14> Now what command were you executing when you got the error abov? [04:19] <c_14> +e [04:20] <Godface> ffmpeg -i "$2" -vcodec copy -acodec libfdk_aac -ac 2 -b:a 384k "$1".mp4 [04:20] <c_14> Replace 'ffmpeg' with '/home/admin/bin/ffmpeg' [04:22] <Godface> that works. why won't just ffmpeg work? [04:22] <c_14> Because there's probably another 'ffmpeg' binary in your path somewher. [04:22] <c_14> +e [04:22] <c_14> export PATH="/home/admin/bin:$PATH" [04:24] <Godface> thanks for the help [04:24] <c_14> np [10:39] <k_sze[work]> I followed this build guide to build a local copy of ffmpeg: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [10:39] <k_sze[work]> And then I want to build the PyAV python binding against it. [10:41] <k_sze[work]> but I then get some weird error about lib/libavcodec.a(avpacket.o): relocation R_X86_64_32 against `.rodata.str1.1' can not be used when making a shared object; recompile with -fPIC [10:42] <k_sze[work]> What's that supposed to mean? *What* am I supposed to "recompile with -fPIC"? [10:42] <c_14> ffmpeg [10:45] <c_14> You can probably just add --enable-pic to your configure line. [10:58] <deepsystm> Hi All! [10:59] <deepsystm> anybody knows how to stream MP3, AAC, OGG to icecast with ffmpeg? [11:03] <deepsystm> I try: http://pastebin.com/MALq9w26 [11:04] <deepsystm> I try to search ffmpeg useage example to stream to icecast in google, but nothing found [11:19] <k_sze[work]> c_14: ok, --enable-pic got rid of that. [11:19] <k_sze[work]> And now I get ffmpeg/libavutil/mem.c:253: multiple definition of `av_calloc' [11:20] <k_sze[work]> Also /ffmpeg/libavutil/frame.c:30: multiple definition of `av_frame_get_best_effort_timestamp' [12:10] <deasy> hello all [12:10] <Guest21271> hello all [12:12] <Guest21271> i want to ask to everyone here [12:12] <Guest21271> how to make video like this http://www.imagebam.com/image/145b86385282890 with ffmpeg? [12:12] <Guest21271> video with overlay /frame .. [12:13] <Guest21271> thanks for answer :) [12:18] <c_14> Use the overlay filter? [12:18] <c_14> k_sze[work]: no clue [12:18] <c_14> How are you compiling your program? [12:21] <Guest21271> i have 1 video .. [12:21] <Guest21271> i want set frame to my video..with static image ..like in screenshot [12:22] <c_14> The overlay filter can take an image as input. [12:23] <Guest21271> how to use overlay filter command c_14? [12:23] <Guest21271> :) i'm new in ffmpeg [12:24] <Guest21271> https://www.youtube.com/watch?v=faHziFBwUeg <<this is sample video [12:25] <c_14> https://ffmpeg.org/ffmpeg-filters.html#overlay-1 https://ffmpeg.org/ffmpeg-filters.html#Examples-50 [12:26] <Guest21271> fast forward to 1:24 minutes [12:27] <c_14> You mean that surrounding picture frame thingy? [12:28] <Guest21271> yes [12:28] <Guest21271> something like that [12:28] <Guest21271> give my video picture frame .. [12:30] <Guest21271> or something like this https://www.youtube.com/watch?v=1Jdf_aIFCsE [12:31] <Guest21271> with overlay filter command? can we make video like that? [12:32] <c_14> Either use the picture frame as the backer with -loop 1 and overlay the video in the position you want, or pad your video to the total video+frame size, and overlay the image (the image needs to have a transparent portion or you won't see the video) [12:35] <Guest21271> can you give me example command @c_14 ? if i have 1 transparant picture ..test.jpg and i have 1 video ..upin.mp4 [12:39] <c_14> `ffmpeg -i upin.mp4 -i test.jpg -filter_complex '[0:v][1:v]overlay={x_offset}:{y_offset}[v]' -map '[v]' -map 0:a -c:a copy out.mp4' [12:39] <c_14> Where x_offset is the width of the 'frame' from the left and y_offset is the width of the 'frame' from the top [12:40] <c_14> eh, wait [12:40] <c_14> `ffmpeg -i upin.mp4 -i test.jpg -filter_complex '[0:v]pad={image_width}:{image_height}[iv];[iv][1:v]overlay={x_offset}:{y_offset}[v]' -map '[v]' -map 0:a -c:a copy out.mp4' [12:41] <c_14> Everything in {} is a variable that depends on your input, the {} should not be there in your final command line [12:43] <Guest21271> thanks c_14 [12:43] <Guest21271> i will try right now [15:26] <kc8hfi> i'm having a problem with vidoe and audio sync. When I encode a raw video and raw audio into a mpeg file, at the beginning of the file, the audio and video is in sync. around 48 minutes or so, the sound starts coming befor ethe video. from that time forward, the sync gets farther and farther apart [15:27] <kc8hfi> I've seen options where you can shift the audio eithe way to correct the sync. But I that just the shift on the whole thing [16:48] <oomkiller1> hi i'm getting an error while executing an ffmpeg command: http://pastebin.com/wdyrHK44 command is in first line, can someone help? [16:49] <oomkiller1> I don't understand what the exact problem is, I have an h264 decoder, all the other files with h264 are working [17:00] <justinX> oomkiller1: so you only get that error for that file? [17:01] <justinX> I mean, something with that file that don't work with "ffmpeg -i input.mkv -vcodec hevc -x265-params crf=28 -sn -acodec ac3 -map 0:0 -map 0:1 -map 0:2 output.mkv" ? [17:03] <oomkiller1> justinX: I'm not sure if I understand you, but the only difference from the other files I could see was that the Codec ID is avc1 instead of V_MPEG4/ISO/AVC [17:07] <justinX> hmm.. that is strange if that is only difference in the ffprobe output between a file that works and this that don't [17:17] <oomkiller1> justinX: heres the output of ffprobe: https://bpaste.net/show/f242286e6831 (I got my information from mediainfo) [17:19] <oomkiller1> justinX: could that be a problem? [mov,mp4,m4a,3gp,3g2,mj2 @ 0x23df250] stream 0, timescale not set [17:19] <justinX> sadly I'm no expert but maybe yes [17:20] <justinX> hopefully someone that knows more joins in here later in the evening [17:20] <oomkiller1> ok thx [18:06] <Vladimir_> Hi all [18:06] <Vladimir_> anybody not sleep? [18:07] <Vladimir_> how can help with stream audio to icecast with ffmpeg? [18:19] <relaxed> Vladimir_: man ffmpeg-protocols|less +/^' 'Icecast [18:22] Action: relaxed 's static build will have alsa indev/outdev and libvidstab support [19:08] <justinX> Vladimir_: I'm not sleeping. [19:39] <Vladimir_> No manual entry for ffmpeg-protocols [19:40] <Vladimir_> justinX how can give example how to broadcast to icecast [19:56] <c_14> https://ffmpeg.org/ffmpeg-protocols.html#Icecast [21:02] <llogan> man ffmpeg-protocols works for me [21:02] <llogan> Vladimir_: ^ [21:34] <ben_> hi. i have some videos with 5.1 audio that i want to add stereo audio. the command i'm using works great, but i was wondering if there's a way to set the new stereo track as the default? [21:34] <llogan> let me guess: QuickTime? [21:35] <ben_> no, for my roku [21:35] <ben_> can't handle the 5.1 audio [21:38] <JoshX> i'm saving a h264 stream from an ip camera in 15 minute chunks and I need the timecode tothe msec of the first frame (or the timecode of every frame) or an index file with frametimes/numbers [21:38] <JoshX> is there any way to do this using the segment filter and perhaps metadata? [21:38] <JoshX> or any other way? [21:41] <ben_> http://pastebin.com/bGZSCgDR [21:41] <ben_> thats the ffmpeg part from my script [21:42] <ben_> as i mentioned, it works fine but i'd like to mark the stereo track as the default one [21:44] <llogan> why did you not incluse the console output too? [21:46] <ben_> is it required? the output files are fine and work correctly [21:46] <llogan> it was requested. [21:47] <ben_> i just wanted to know if it was possible for ffmpeg to set the default audio track [21:47] <llogan> i ask because it seems at least half of people do not include the output [21:47] <llogan> and i'd like to know why [21:47] <llogan> so maybe i can re-word the request [21:48] <ben_> well i'd have to run another task to get console output [22:06] <ben_> http://pastebin.com/uZAuM45e [22:06] <ben_> there's the whole thing [22:07] <ben_> so, is it possible for ffmpeg to set the stereo track as default? [22:14] <llogan> ben_: do you have a similar file where it does have the correct default audio? [22:15] <ben_> no, the original files only have 5.1 audio so that is the default track [22:18] <ben_> i can tag the new stereo tracks as default with mkvpropedit but i'd like to see if i could get it all done with a single ffmpeg script [22:20] <DX099> hello all [22:21] <DX099> I have a problem regarding metadata, flac and m4a container [22:21] <DX099> http://paste.debian.net/142748/ [22:22] <DX099> when producing the file, ffmpeg acts like everything went fine, but when I use ffprobe or even try to import resulting m4a files within my Rhythmbox library, nothing shows up in place of the metadata [22:23] <Vladimir_> how was try to stream content from ffmpeg to icecast? [22:24] <BtbN> i don't think you want it to convert that cover png to h264 [22:24] <DX099> BtbN, should I specify it so that it stays png ? [22:24] <BtbN> no idea if m4a supports that [22:26] <DX099> BtbN, you were right ! removing the png and ffprobe is no longer confused, and certainly other software will be fine [22:26] <DX099> I'll try png [22:28] <DX099> ah "codec not currently supported in container" [22:29] <BtbN> m4a should support it. But ffmpeg currently doesn't. [22:32] <DX099> BtbN, I just saw that mjpeg is supported by m4a too but "Could not find tag for codec mjpeg in stream #0, codec not currently supported in containe" [22:32] <BtbN> yes, i don't think ffmpeg currently supports muxing cover art into m4a [22:32] <DX099> ok [22:33] <DX099> I don't know why I got the impression that it could somehow... maybe while coverting files with already embedded images [22:38] <DX099> BtbN, alright, I'll just be using external utilities thanks [00:00] --- Thu Jan 29 2015

1 0

ffmpeg-devel.log.20150127
by burek 28 Jan '15

28 Jan '15

[01:01] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:530bf8ece6f8: avfilter/vf_eq: Fix clipping code [01:34] <cone-701> ffmpeg.git 03Andreas Cadhalpun 07master:f8716d1e56d5: configure: use ar and ranlib in deterministic mode if available [01:55] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:f5b3257c506e: avfilter/vf_eq: mark src as const [02:57] <compn> <michaelni(a)gmx.at> (expanded from <webmaster(a)mplayerhq.hu>): host [02:57] <compn> mx01.emig.gmx.net[213.165.67.115] said: 550-Requested action not taken: [02:57] <compn> mailbox unavailable 550 invalid DNS A/AAAA resource record (in reply to [02:57] <compn> MAIL FROM command) [02:57] <compn> lol gmx.at [04:09] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:aa508a9c5f63: fate: Fix bitexactness for vsynth3-dnxhd-1080i-colr [05:12] <aetasx> awesome. new black sails [05:19] <wm4> so there's colr writing now, but not reading? [05:19] <wm4> including colorspace guessing directly in the format specific muxer, awesome [05:22] <aetasx> just realized I was in the completely wrong channel [11:23] <cone-693> ffmpeg.git 03Stefano Sabatini 07master:afa3c996fed4: lavfi/lut: apply minor compute_gammaval709() doxy fix [12:46] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:2af82a1ad9f7: hevc: store the escaped/raw bitstream in HEVCNAL [12:46] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:06894f1a04dd: hevc: store the short term rps flag and size in the context [12:46] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:b2e9b0f5d4dc: hevc: add hwaccel hooks [12:47] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:f8ecffa9b7e4: hevc: reindent after previous commit [12:47] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:36962ad2339d: Add DXVA2 HEVC HWAccel [12:47] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:5f2cdf9c3cc5: ffmpeg_dxva2: add hevc support [12:57] <ubitux> libavfilter/vf_fspp.c: { 0, 48, 12, 60, 3, 51, 15, 63, }, [12:57] <ubitux> libavfilter/vf_owdenoise.c: { 0, 48, 12, 60, 3, 51, 15, 63 }, [12:57] <ubitux> libavfilter/vf_pp7.c: { 0, 48, 12, 60, 3, 51, 15, 63, }, [12:57] <ubitux> libavfilter/vf_spp.c: { 0, 48, 12, 60, 3, 51, 15, 63 }, [12:57] <ubitux> we should probably factorized this ordered dithering table [13:02] <cone-693> ffmpeg.git 03Anton Khirnov 07master:f9f883af4fe6: h264: simplify code in flush_dpb() [13:02] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:ed8de1570db8: Merge commit 'f9f883af4fe615a832407a657752e248a96c6280' [13:09] <cone-693> ffmpeg.git 03Anton Khirnov 07master:1dd021929f81: hevc: clear unused refs on failure [13:09] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:74e5a5a849bc: Merge commit '1dd021929f8157b88529ce1e6ab6351dd2899e89' [13:11] <durandal_1707> ubitux: what about changing pallete in palletegen when scene changes? [13:21] <cone-693> ffmpeg.git 03Anton Khirnov 07master:443b71928b2f: hevc: unref the current frame if frame_start() fails [13:21] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:a7fa1b9aa104: Merge commit '443b71928b2f36362e805c037751e6c3c79ea4e8' [13:30] <cone-693> ffmpeg.git 03Andreas Unterweger 07master:749a89d1b8bb: examples/transcode_aac: properly select the output sample format [13:30] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:010311d67337: Merge commit '749a89d1b8bb73b4d4f14c48f33259a1300c1761' [13:36] <cone-693> ffmpeg.git 03Andreas Unterweger 07master:c9b19ac8928c: examples/transcode_aac: fix a typo [13:36] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:d078d57fb769: Merge commit 'c9b19ac8928c6c9b7f25c3988177204f110d5e0e' [13:54] <cone-693> ffmpeg.git 03Andreas Unterweger 07master:3a70c0c95fea: examples/transcode_aac: generate proper PTS and set the muxer timebase [13:55] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:e5c28d4f9a72: Merge commit '3a70c0c95feacb3844d05eebd579fc8189a77eee' [14:02] <ubitux> durandal_1707: just replied to you on the ml [14:03] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:b0593a4bca13: hevc: pass the full HEVCNAL struct to decode_nal_unit [14:03] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:36779a84051e: hevc: store the escaped/raw bitstream in HEVCNAL [14:03] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:183bffb3a3c4: Merge commit 'b0593a4bca138f1f026d8c21e8c3daa96800afe2' [14:03] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:f35c4f4a17a4: Merge commit '36779a84051eae6744cc936d91b1d428143665ba' [14:21] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:4b95e95dbae5: hevc: store the short term rps flag and size in the context [14:21] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:e72e8c5a1df6: hevc: add hwaccel hooks [14:21] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:2b5fa0e0be11: Merge commit '4b95e95dbae58c9b60891284bf8b5bbd83e5293a' [14:21] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:4e40e6128769: Merge commit 'e72e8c5a1df61447ac7af750531e96e8b62d02ba' [14:33] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:b82722df9b29: hevc: reindent after previous commit [14:34] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:7e850fa67e32: Add DXVA2 HEVC HWAccel [14:34] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:2d4b8af4f688: Merge commit 'b82722df9b2911bd41e0928db4804067b39e6528' [14:34] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:b6938c94d641: Merge commit '7e850fa67e32ebe98581c2e4ca4a4655dee7f116' [14:36] <saste> durandal_1707, what happened to the lut16 code in eq2? why did you remove it? [14:37] <saste> my assumption is that it was not used in mp=eq2, right? [14:37] <durandal_1707> it was only one line [14:37] <durandal_1707> it is used by mp=eq2 [14:38] <durandal_1707> but -benchmark showed mp=eq2 is slower so i did not cared to add lut16 [14:38] <saste> durandal_1707, so how is that mp=eq2 is bit-exact with eq? [14:38] <durandal_1707> so i just removed that single line in filter context because arwa didn't added more lut16 code [14:39] <durandal_1707> saste: i changed order of options to match mp=eq2 [14:39] <durandal_1707> end tested contrast, brightness and saturation at least [14:40] <arwa> But the options are still not matching the order in mp=eq2 code. [14:40] <saste> so the difference is related to gamma? [14:40] <durandal_1707> i did not tested gamma at all, but it should be bit-exact [14:40] <saste> arwa, that's OK, since we plan to remove gamma correction from eq [14:40] <arwa> Okay. [14:41] <saste> mp=eq2=GAMMA:CONTRAST:BRIGHTNESS:SATURATION [14:41] <arwa> I tested for gamma, it was working fine. [14:41] <saste> eq=CONTRAST:BRIGHTNESS:SATURATION [14:41] <durandal_1707> no lavfi is in alphabetical order [14:42] <durandal_1707> actually not, ignore me [14:42] <saste> so, again, mp=eq2 is using lut16 code, eq is not, and yet they are bit-exact [14:42] <saste> how can it be? sorry if I'm dumb and missing something [14:42] <durandal_1707> lut16 is just bigger lookup table [14:43] <arwa> Maybe it does the same thing, but its little faster. [14:44] <arwa> Because datatype for lut is uint8_t and for lut16 is uint16_t [14:44] <saste> arwa, yeah, I think so [14:44] <durandal_1707> lut16 comes from michaelni [14:53] <cone-693> ffmpeg.git 03Hendrik Leppkes 07master:a7e038049730: avconv_dxva2: add hevc support [14:53] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:a1cdd1f08e06: Merge commit 'a7e0380497306d9723dec8440a4c52e8bf0263cf' [15:06] <cone-693> ffmpeg.git 03Anton Khirnov 07master:cf1e0786ed64: error_resilience: move the MECmpContext initialization into ER code [15:06] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:3d0411707839: Merge commit 'cf1e0786ed64e69614760bfb4ecd7adbde8e6094' [15:16] <cone-693> ffmpeg.git 03Anton Khirnov 07master:9404a47a2d1d: h264: move parser-only variables to their own context [15:16] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:6e57d2da9058: Merge commit '9404a47a2d1df418946a338938eb6cdb3afed474' [15:22] <cone-693> ffmpeg.git 03Anton Khirnov 07master:ecab21ac47d0: h264: do not reset the ref lists in flush_change() [15:22] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:ad3412d0283b: Merge commit 'ecab21ac47d0d4ca604bebf494017ae5090853a8' [15:35] <kierank> ubitux: is there a way of getting a triangular wave with eval? [15:36] <ubitux> use if() to get a 1 -1 which you multiply by t%period (use mod() for %) [15:36] <ubitux> i guess? [15:37] <ubitux> you can probably avoid the if() being smart [15:49] <ubitux> kierank: ffplay -f lavfi -i "aevalsrc='abs(mod(t*100,4)-2)-1',showwaves" [15:49] <kierank> Thanks will try later [15:50] Action: kierank was going to Fourier decompose the wave [15:51] Action: av500 waves [15:52] Action: ubitux shows [15:52] Action: Daemon404 transforms [15:52] Action: saste decomposes [16:02] <cone-693> ffmpeg.git 03Anton Khirnov 07master:58ae8d595724: h264_parser: restore a comment lost in 0268a54 [16:02] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:f25f15b6e1c4: Merge commit '58ae8d595724150c407ca2c2df3aa4bd5580397c' [16:06] <reynaldo> ubitux: you around? [16:06] <ubitux> reynaldo: yep [16:07] <reynaldo> cool, willc continue privately -> [16:07] <ubitux> yep [16:14] <cone-693> ffmpeg.git 03Anton Khirnov 07master:167e004e1aca: h264: drop any pretense of support for data partitioning [16:14] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:392080cbe528: Merge commit '167e004e1aca7765686ed95d7cd8ea5064d4f6f6' [16:39] <durandal_1707> michaelni: will you comment on libmpcodecs removal? [16:43] <ubitux> durandal_1707: softpulldown doesn't work in mplayer? [16:43] <durandal_1707> it works differently in mpv [16:44] <durandal_1707> i have not compiled mplayer here [16:45] <ubitux> does it work with mpv? [16:46] <durandal_1707> in mpv it is most likely broken [16:47] <michaelni> durandal_1707, ill reply later, ping me in case i forget [16:49] <durandal_1707> also mp=softpulldown drops all frames by default [17:00] <durandal_1707> and for normal progressive videos it acts like interlacer [17:05] <cone-693> ffmpeg.git 03Anton Khirnov 07master:f771b3ab5d3c: avidec: do not export stream_codec_tag [17:05] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:be023405a7cd: Merge commit 'f771b3ab5d3c0b763ee356152be550f4121babd0' [17:05] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:a76440239428: avcodec/h263dec: Try to use codec_tag instead of the stream_codec_tag [17:28] <cone-693> ffmpeg.git 03Anton Khirnov 07master:e44b58924fe7: lavc: deprecate unused AVCodecContext.stream_codec_tag [17:28] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:3c831fba0869: Merge commit 'e44b58924fe7b180bf8b0c277c15d1a58210a0e9' [17:31] <arwa_> How do I generate an image with blocky artifacts for testing the output? [17:31] <nevcairiel> encode something wth a very low bitrate? :d [17:32] <Daemon404> http://needsmorejpeg.com/ [17:32] <nevcairiel> heh [17:35] <arwa_> But, I need to use some encoder from FFmpeg. [17:39] <kierank> encode with mpeg2 [17:39] <kierank> or something like that [17:39] <nevcairiel> default mpeg2 or mpeg4 settings are probably blocky as hell [17:43] <durandal_1707> michaelni: could you please take look at softpulldown port by me to find out why it differs with mp version? [17:58] <cone-693> ffmpeg.git 03Anton Khirnov 07master:80a11de7dca3: nutenc: do not use has_b_frames [17:58] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:b44456c90896: Merge commit '80a11de7dca315505bf203ce9c8c016e71724fd2' [18:16] <cone-693> ffmpeg.git 03Anton Khirnov 07master:728685f37ab3: Add a side data type for audio service type. [18:16] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:62a82c66cd3f: Merge commit '728685f37ab333ca35980bd01766c78d197f784a' [18:21] <kierank> durandal_1707: +1 about mpcodecs [18:24] <nevcairiel> Sometimes I wonder if some people are just too dense or if I'm just that smart, but i've never had big troubles understanding the ffmpeg API or the source code behind it, yet some people make it a holy mission to call it "cryptic" at best, or using strong swear words at worst [18:24] <nevcairiel> Certainly its not perfect and makes some things more complex than they need to be, but its not "hard" to use it, its just a bit time consuming to write boilerplate that may not be necessary [18:25] <JEEB> pretty much [18:25] <nevcairiel> also, i should unsub from libav-user [18:28] <wm4> too much trolling? [18:28] <nevcairiel> well its the thread you and that other guy are trolling each other about that cocoa api [18:28] <Daemon404> too much dumb [18:30] <kierank> the problem is I think people don't understand containers, timestamps etc [18:30] <nevcairiel> even a high-level API would need to touch that at some point, unless the toplevel API is "Play(file)" and a new window pops up :d [18:33] <cone-693> ffmpeg.git 03Anton Khirnov 07master:4227e4fe7443: lavf: add a convenience function for adding side data to a stream [18:33] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:0bdcc27d9598: Merge commit '4227e4fe7443733fb906f6fb6c265105e8269c74' [18:33] <wm4> nevcairiel: well, in most cases you'd get pretty good results by naively drawing a RGBA frame using some GUI widget API and assuming fixed frame rate [18:34] <wm4> the low level API requires understand packets and stuff [18:34] <wm4> +to [18:34] <wm4> so there are some orders of magnitudes of understanding between those [18:35] <nevcairiel> I dunno, a naive decoding loop that just gives you RGB in the end isnt that complex, you dont really need to understand more of the "packet" than to know that you get it from one API and shove it into another [18:36] <Daemon404> wm4, packets AND parsers [18:36] <Daemon404> and perhaps bsfs [18:36] <nevcairiel> it might help the whole issue if sws had a AVFrame API that was easy to use, instead of having to figure out how to pass stuff to sws [18:36] <Daemon404> sws API is easy as pie to use [18:36] <Daemon404> a few calls, done [18:36] <Daemon404> lavf and avc are super tedious [18:36] <Daemon404> and nonobvious [18:37] <Daemon404> esp. with the multiple-frames-per-packet, multiple-packets-per-frame, parsers, bsfs [18:37] <nevcairiel> assuming you just want decoding, you dont need to know this [18:37] <Daemon404> plus the need to flush [18:37] <Daemon404> you do [18:37] <Daemon404> because you need to make teh loop handle both [18:37] <nevcairiel> bsfs are irrelevant for decoding, as are parsers, those are handled in avformat internally [18:37] <Daemon404> well ok [18:37] <wm4> lavf would need a way to configure a buffer sink/src with an AVFrame, and it'd be relatively simple to use [18:38] <Daemon404> anyway i just use FFMS_GetFrame for that stuff [18:38] <Daemon404> and ignore all this [18:38] <nevcairiel> the only thing you need to handle is the initial setup, and then a loop with av_read_frame(), and avcodec_decode_video2(), and then sws [18:39] <nevcairiel> well plus branching of between different streams, i guess [18:39] <Daemon404> you need to decrement packet size if it is not consumed [18:39] <Daemon404> and pass it again [18:39] <nevcairiel> not for video [18:39] <nevcairiel> but for audio, yes [18:39] <nevcairiel> (technically the API allows that for video too, but noone does t hat there, and no decoder expects you to) [18:40] <Daemon404> what about packed bframe AVIs? [18:41] <nevcairiel> i think the parser splits those in avformat [18:41] <Daemon404> right [18:41] <Daemon404> i dont get that [18:42] <Daemon404> why have a parser do that, but also have the API allow multiple frames per packet [18:42] <nevcairiel> frame threading [18:42] <Daemon404> right... [18:42] <nevcairiel> i guess the API still allowing that is just old [18:42] <nevcairiel> but not even the ff* tools do this for video [18:43] <Daemon404> lol fun [18:43] <nevcairiel> and even on the audio side its very few codecs that require it [18:43] <Daemon404> but teh ff tools dont even follow the documented api [18:43] <Daemon404> so. [18:43] <Daemon404> yeah. [18:43] <wm4> <nevcairiel> and even on the audio side its very few codecs that require it <- I argued for removing this API artifact [18:43] <wm4> but nope, got to play super-obscure audio formats efficiently [18:44] <nevcairiel> well, there are a few audio codecs that need it because you can't really detect frame boundaries, but it could do this internally i suppose [18:45] <nevcairiel> should potentially remove the docs from decode_video2 that say this [18:45] <nevcairiel> not sure if it actually does a thing if you use chunked decoding [18:45] <nevcairiel> ...which never worked right for me in the first place [18:45] <wm4> the examples (lol) still do the partial packet video decoding [18:45] <wm4> but mplayer, the former libav* API reference, doesn't [18:47] <nevcairiel> not all examples [18:47] <nevcairiel> demuxing_decoding doesnt [18:48] <ubitux> michaelni: can you explain a bit more when it's incorrect? (re: nutenc + has_b_frames) [18:50] <cone-693> ffmpeg.git 03Anton Khirnov 07master:321257814874: mov: export audio service type as side data [18:50] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:6d34665c76ca: Merge commit '32125781487411ed3b1b28b32063d6cd4024d4fc' [18:50] <ubitux> 2.6 release note is going to be awesome :) [18:51] <nevcairiel> why [18:52] <ubitux> there are a lot of cool things in the Changelog and incoming [18:52] <nevcairiel> should probably add my hevc hwaccel in there [18:52] <ubitux> yes [18:53] <ubitux> if someone wants to review palettegen/paletteuse before it's too late, better do it now btw [18:53] <ubitux> also, same for MOV timelines... it's kind of ready since a long time [18:53] <ubitux> ah, and i need to finish the bc dep [18:53] <ubitux> i guess i'll get some stuff done by the end of the week [18:54] <michaelni> ubitux, plain violation of spec and totally breaks packet interleaving, aka ruins nut for any realtime use, as packets would be potentially several seconds mis-interleaved, it might not even play in some players [18:55] <michaelni> thats in worst case, it wont be that bad for every file [18:55] <ubitux> ok [19:11] <cone-693> ffmpeg.git 03Anton Khirnov 07master:a536a4e4bc52: lavc: support extracting audio service type from side data [19:11] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:4ce67c961c0f: Merge commit 'a536a4e4bc52d05f59869761337452fb1f1977f6' [20:01] <cone-693> ffmpeg.git 03Vittorio Giovara 07master:7c51d79ca7ba: nsvdec: validate channels and samplerate [20:01] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:b04cbbe25555: Merge commit '7c51d79ca7badfb370c410b8f44c9142b938e2e6' [20:19] <cone-693> ffmpeg.git 03Vittorio Giovara 07master:e71149a7a5b1: nuv: validate image size [20:19] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:f955abe33321: Merge commit 'e71149a7a5b10ed7baa5a06f47d0313c7bd8df52' [20:32] <cone-693> ffmpeg.git 03Vittorio Giovara 07master:607ad990d31e: dvbsubdec: check memory allocations and propagate errors [20:32] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:c5b6b711b291: Merge commit '607ad990d31e6be52980970e5ce8cd25ab3de812' [20:54] <cone-693> ffmpeg.git 03Vittorio Giovara 07master:8805589b803f: libopencv: Rework error handling in parse_iplconvkernel() [20:54] <cone-693> ffmpeg.git 03Michael Niedermayer 07master:d3d4d98764ea: Merge commit '8805589b803fab5f362008306319336ac79a3fa7' [22:31] <cone-693> ffmpeg.git 03Reimar Döffinger 07master:d96090e7b633: Support BMP files that do not properly align lines. [22:31] <cone-693> ffmpeg.git 03Reimar Döffinger 07master:6efd0ba977ae: swresample_internal.h: Move struct declaration before first use. [22:58] <nevcairiel> hm, mpeg2 simple profile with nothing but P frames ... does mpeg2 use some crazy schemes like intra-refresh ? o.o [23:26] <kierank> Nevcairiel: Yes I've seen some beforr [23:26] <nevcairiel> apparently its a sat feed [23:27] <nevcairiel> my code naively assumes there might be an I frame at some point, so to avoid showing too much corruption, it waits for one before showing anything [23:27] <nevcairiel> yeah well, that didnt work out quite well! [23:28] <kierank> There's a sample somewhere in the FFmpeg samples repo [23:33] <llogan> at least the spammer had to take 10 tries before he succeeded. [23:33] <llogan> usually they quit after 2 or 3 [23:34] <llogan> damned retarded though that BadContent shows them what blacklisted words they hit [23:37] <JEEB> lol [23:42] <llogan> oh, apparently that can be disabled somehow. [23:44] <compn> llogan : is there a way to send these possibly spams to a moderation queue instead of them being posted ? [23:44] <compn> much like the mailing list ? [23:44] <compn> then we just review the spams once a week or so [23:44] <llogan> michaelni: can you add "show_blacklisted=false" (possibly under [spam-filter]) in trac.ini? [23:45] <llogan> compn: i doubt that would be practical. we are currently getting up to 5 per minute. [23:46] <llogan> it's the humon spammers that are the problem though. i'd rather allow some spam in than have to wade through ham and make users wait [23:47] <llogan> parsing the ML is boring enough [00:00] --- Wed Jan 28 2015

1 0

ffmpeg.log.20150127
by burek 28 Jan '15

28 Jan '15

[01:09] <dvk> i get warning: first frame is no keyframe, how can i make sure that it is a keyframe? [01:09] <dvk> i'm using -ss time [01:41] <MaudeLebowski> Hey there. I am trying to create a video out of a set of jpgs that are named by unix timestamp. The pics arent one after another, i.e. 1.jog, 2.jpg etc so it breaks after just 1 image. Without renaming the files, how would you make a video starting with 1.jpg for example if the next image were 3.jpg, 7.jpg, and so on. [01:43] <MaudeLebowski> I have looked everywhere for help, and all I get is rename the images which I really dont wanna do for every video I make. [01:43] <MaudeLebowski> I have heard of piping images into it. [01:43] <MaudeLebowski> But cant seem to get the right list of images on the fly. [01:43] <c_14> look at the glob part of image2 [01:43] <MaudeLebowski> Like % [01:44] <MaudeLebowski> I did do that. [01:44] <c_14> https://ffmpeg.org/ffmpeg-formats.html#image2-1 [01:44] <c_14> -pattern_type glob [01:44] <c_14> *.jpg [01:44] <MaudeLebowski> But it stops after 1 image [01:44] <MaudeLebowski> if they arent sequential in number [01:44] <c_14> not glob [01:44] <MaudeLebowski> Ok [01:44] <MaudeLebowski> Will look into this [01:44] <c_14> do ls *jpg [01:44] <c_14> it does the same thnig [01:44] <c_14> *thing [01:45] <MaudeLebowski> Well *.jpg wouldnt be what I need unfortunately itd be like anything from 114.jpg to 489.jpg but it might not have 114.jpg or 489.jpg [01:45] <MaudeLebowski> I know globs should be able to do that just cant figure out what pattern [01:50] <MaudeLebowski> Globs is likely my solution though. Ill keep working with it. Thanks for the help. [02:21] <coherence> i may simply be confused, but is it possible (or even reasonable) to try to set PTS in mpegts audio streams? (there is a tiny bit of silence at the start of the files after conversion from AAC - maybe it's the AAC priming frames) [02:47] <pentanol> coherence may be -er flag? [02:47] <pentanol> no, -re flag [02:47] <coherence> pentanol: thanks, using it already [02:48] <coherence> i suppose a command line would be in order [02:50] <coherence> pentanol: are you suggesting that -er might be responsible for the artifacts? [02:50] <coherence> i mean -re [03:17] <ramp> hi hi [03:17] <ramp> can anyone help me with using ffmpeg to embed a subtitle file? [03:19] <relaxed> possibly, but we need more details than that. [03:19] <ramp> okay [03:19] <ramp> i have some mkv files, and some ac3 files and a .sub file [03:20] <ramp> and i'm using ffmpeg to output an mkv file with the relevant audio streams and subtitles embedded [03:20] <ramp> I'm just trying to figure out the format to embed the subtitles [03:20] <relaxed> you want the subs included in the container or burnt into the video? [03:20] <ramp> just included in the container [03:21] <relaxed> ffmpeg -i video -i audio -i sub -map 0 -c copy out [03:21] <ramp> there's only one -map command? [03:22] <ramp> right now the thing I'm using looks like this [03:22] <ramp> ffmpeg -i $input -i $a1 -i $a2 -map 0:v -map 1 -map 2 -metadata:s:a:0 language=eng -metadata:s:a:1 language=commentary -c copy -shortest $output [03:22] <ramp> $input, is the input file name, $a1 is the first audio track, $a2 is the second audio track and $output is the output filename [03:23] <ramp> sure, no problem [03:23] <relaxed> yeah, using -map 0 was wrong on my part [03:24] Action: relaxed just woke up [03:25] <ramp> http://pastebin.com/iw32Rdtr [03:25] <ramp> so i'm just wondering where to include $sub in that command [03:25] <ramp> and what modifications I need to make [03:26] <relaxed> why is -shortest in there? [03:26] <ramp> in case there are slight differences in file lengths [03:26] <ramp> it's just a precaution [03:27] <relaxed> read fflogger's message again [03:27] <ramp> what in particular would you like me to show you [03:28] <relaxed> the bot was pretty specific [03:30] <ramp> if you'd like to help me I'd be most appreciative, but if you want me to guess what information you require I might as well guess how to interpret the docs [03:33] <relaxed> apart from the missing -i $sub in the script? [03:34] <ramp> there also needs to be some accompanying -metadata I think? [03:39] <relaxed> I think you can set the language but it can be omitted too [03:40] <coherence> how would one extract just the audio stream from an AAC? [03:41] <relaxed> are there multiple audio streams? you want raw aac? [03:41] <coherence> relaxed: i *think* i want raw bitstream, just one stream in a file. i want to remove the priming and remainder bits. [03:41] <ramp> if I just include -i $sub the output file does not have subtitles [03:42] <relaxed> you need -map 3 [03:42] <ramp> cool thanks, i'll try that now [03:43] <relaxed> coherence: try, ffmpeg -i input -map 0:a -c copy output.aac [03:43] <coherence> relaxed: thank you, i'll try now. i'm clueless in audio codecs/formats, just learning and the ffmpeg docs are harder for me than man pages :) [03:43] <ramp> i'm getting an error now [03:43] <ramp> http://pastebin.com/0uhAqiTk [03:44] <relaxed> ramp: "Subtitle codec 1833195076 is not supported." [03:45] <ramp> what does that mean? the format my subtitles are in is not supported by ffmpeg? [03:47] <relaxed> ramp: try converting them to the ass sub format first. ffmpeg -i input.sub out.ass [03:48] <ramp> cheers [03:51] <relaxed> coherence: check out the wiki too if you haven't already. https://trac.ffmpeg.org/wiki [03:51] <coherence> thanks relaxed [03:55] <coherence> relaxed: my questions stem from using ffmpeg to push AAC ADTS files to nginx-rtmp to be served using HLS, and the resulting mpegts files *i think* have a tiny bit of dead air at the start [03:55] <coherence> i'm searching for the cause, and the priming bits in AAC files have my attention [03:57] <coherence> although i'm also curious about the MPEG2-TS PTS delay as specified here: http://www.bretl.com/mpeghtml/timemdl.HTM [03:58] <coherence> which is why i wondered if i could use ffmpeg to set the PTS in a given file [04:19] <pentanol> coherence did you get wrong PTS\DTS? [04:33] <coherence> pentanol: /bufbri [04:33] <coherence> whoops [04:35] <coherence> pentanol: the start_time shown for the audio stream when using ffprobe -show_streams is 0.7 [04:35] <coherence> but should be 0 [04:36] <coherence> so i *think* that means PTS is off [04:36] <coherence> sorry, i'm guessing [05:15] <{MiltonBerle}> Why would I get- image2pipe No such file or directory [05:15] <{MiltonBerle}> When I am definitely piping in files [05:15] <{MiltonBerle}> args=''; for x in `find . -type f -newer 1422284581.jpg ! -newer 1422284761.jpg`; do args="$args $x";done ; cat $args | ffmpeg -framerate 1/1 -i image2pipe -c:v libx264 -r 30 -pix_fmt yuv420p testing_movie.mp4 [05:15] <{MiltonBerle}> example [05:15] <{MiltonBerle}> The commands before cat by themselves show files [05:16] <{MiltonBerle}> If echoed [05:32] <coherence> pentanol: http://pastebin.com/4ucZ3s9t [05:32] <coherence> first packet from the corresponding audio stream in each file [05:33] <coherence> the MPEG2-TS file shows a PTS of 63000 wihile the AAC source file has a PTS of 0 [05:33] <coherence> again could be my ignorance [05:33] <coherence> but that seems wrong [05:49] <pentanol> coherence what happen if you reencode this file with ffmpeg -i ... ? [06:05] <coherence> pentanol: you mean the TS file back to AAC? [06:05] <coherence> (interesting question anyway) [06:09] <coherence> looks like the pts_times convert back to the original/like the source file [06:10] <coherence> size and pos is off though [06:25] <pentanol> coherence after convert with ffmpeg pts still 0.7 ? [06:31] <coherence> pentanol: converting AAC -> TS using ffmpeg, pts is 1.4. converting back TS -> AAC, pts is 0 [06:47] <pentanol> coherence .aac to .ts ? what's your command with ffmpeg? [06:52] <coherence> pentanol: http://pastebin.com/rEj2cbVP [06:53] <coherence> and some variations on that playing with -ar and -ab [06:53] <coherence> /that/those/ [07:03] <Darby_Crash> hi to all [07:04] <Darby_Crash> thanks for ffmpeg [07:05] <brdxufan> hello [07:05] <Darby_Crash> have someone a script for compiling ffmpeg/ffplay on nix? [07:06] <Darby_Crash> hi brdxufan [07:07] <Darby_Crash> i need a script for a static build of ffmpeg ffplay [07:10] <brdxufan> av_new_stream(), how to use? [07:11] <brdxufan> who can help me? [07:30] <pentanol> coherence I made the test out.ts with start_pts=126000 out.aac with start_pts=N/A [07:31] <pentanol> You don't have wrong pts\dts when you paying the file? [07:31] <pentanol> it not, that's okay [07:31] <pentanol> of* [07:31] <pentanol> if* [08:54] <k_sze[work]> How does ffmpeg deal with files that have been incompletely written? [08:55] <k_sze[work]> e.g. if I take an intact AVI file and chop off the last 9 kilobytes. [09:34] <storms> Hi. I'm trying to scale videos to a 720 height, keep width but also use square pixels. -vf "scale=-1:720,setsar=1" makes a video with wrong aspect ratio in regards to the width vs height on a video with 4:3 SAR. [09:34] <storms> A simple way to do this? [09:39] <Vladimir_> HI to all [09:40] <Vladimir_> can somebody help me.. [09:40] <Vladimir_> I need to mjpeg stream to mix in one [09:40] <Vladimir_> http:mjpeg stream... 1280x720 ----> to be 2560x720 [10:02] <ArunC> Using ffmpeg cmd line, I'm trying to split a video at the same time, preserve the timestamps. At present I can split the video but I am not able to preserve the timestamp. Any idea which option I should be using? [10:05] <ArunC> ie, if the video is 10 min long and if each video chunk is 5 min, the second chunk's starting timestamp should be 05:00 and not 00:00. [10:20] <BtbN> how should that work? 5 minutes of nothing in the beginning? [10:25] <ArunC> BtbN: Yeah, sort of. Just record the offset as the timestamp for the initial frame. [10:25] <BtbN> I don't think that's possible. Might be possible for audio, but depends a lot on the container. [10:27] <ArunC> BtbN: OK, primarily we deal with surveillance videos. So splitting is necessary to process the videos in parallel, without losing the frame timestamp. [10:28] <BtbN> Sounds like bad processing software, which isn't able to offset an entire file [10:28] <ArunC> BtbN: Actually, the processes reside on different boxes and not all can afford to download the entire video file (50+GB). :) [10:29] <BtbN> And you can't just tell it "this file is offset by X minutes"? [10:30] <ArunC> No, as that process is a 3rd party s/w and that the interfaces are fixed now. [10:31] <BtbN> Well, you can insert black frames at the beginning, but you will have to re-encode for that. [10:41] <Vladimir_> hi [10:41] <Vladimir_> I tray to mix to stream mjpeg in on... [10:41] <Vladimir_> what is best solution for that ? [10:42] <Vladimir_> using filter ? [10:45] <t4nk476> get_bits seeking functionality? [11:00] <ArunC> Found what I wanted. We can use -itsoffset to specify the offset in the output video chunk. [11:00] <ArunC> BtbN: ^ [11:02] <BtbN> Doesn't that add an offset to the entire input video? [11:03] <BtbN> As it's an input parameter [11:06] <ArunC> oh yeah, my bad.. [11:06] <ArunC> :( [11:07] <ArunC> its an input param [11:22] <ArunC> BtbN: Not sure if -start_at_zero will help. [11:44] <cousin_luigi> What's the difference between using opus and libcelt to decode CELT? [12:33] <D0L1k> hi all [12:35] <D0L1k> We have a problem with timeshifting. We need to get concrete time sequence of file. We are using ffmepg 1.12, we are producing mpegts, we can connect and watch that file (streaming is allright). We can also start it from defined time/date, but we dont have duration of that file.. So in VLC/etc. we can't see duration [12:36] <BtbN> I don't think there is a version 1.12 [12:37] <D0L1k> *1.2.11 [12:37] <D0L1k> sorry [12:38] <BtbN> I'd recommend updating. [12:39] <cousin_luigi> I'll rephrase my previous question: what codec overlap is there between libcelt and libopus and what happens when both are linked against? [12:39] <D0L1k> In new version (for ex. 2.5.3) we can't use codec copy on ffserver [12:40] <BtbN> With a recent enough ffmpeg, you can do ffmpeg -i someinput.mkv -ss start -t duration -c:v ... -c:a ... youroutput.mkv [12:40] <D0L1k> but also there is not possibility of using "start from time for duration" [12:40] <BtbN> You can't get a specific sequence from a file without re-encoding. [12:41] <BtbN> ffserver is not exactly maintained anymore [12:41] <BtbN> Depending on the codec you can start from an I frame, by putting -ss in front of the input [12:43] <D0L1k> and is there any other software, where we can use timeshifting? .. in ffserver timeshifting is ok, but we can't select endtime/enddate, just startdate.. [12:48] <D0L1k> or how can we select sequence from ffm file? for example this works: "http://localhost:8080/nova.ts?date=2015-01-27T09:00:00" .. but we want something like "http://localhost:8080/nova.ts?date=2015-01-27T09:00:00&enddate=2015-01-27T0…".. we can edit source code of ffserver, just need to know how set duration.. [16:55] <cousin_luigi> JEEB: Hello. Do you happen to be around? [21:53] <Projectns> Hello i have a short quesition, is it posible to stream a video list (more than one video ) to a rtmp server ? [21:53] <Projectns> *with ffmpeg [21:54] <kepstin-laptop> you could use any of the supported methods of concatenating videos in ffmpeg to generate a single stream, which could then be sent over rtmp [21:54] <Projectns> do u have some links to tutorials ? [21:54] <kepstin-laptop> http://www.ffmpeg.org/faq.html#How-can-I-concatenate-video-files_003f [21:55] <Projectns> or can i choose a directory with videos ? [21:55] <Projectns> who play all videos from there [21:57] <kepstin-laptop> no, but if you have a bunch of files that all have maching codecs and other parameters, then you might be able to put their names into a list file then use the concat demuxer/format to load them. [21:57] <Projectns> ah cool , that is exactly what i want [21:58] <kepstin-laptop> if they might have different codecs, it's more complicated since you have to decode them before concatenating (in this case, you'd use the concat filter) [21:59] <Projectns> ok , i try to convert the videos what i want and add them to the ffmpeg [21:59] <Projectns> is the file level with cat the command? [22:00] <kepstin-laptop> file level concatenation is equivalent to using the cat command, yes. [22:00] <Projectns> ok [22:00] <Projectns> its sound hard... :D [22:01] <Projectns> do u know some other broadcast tools for linux ? [22:01] <Projectns> or some scripts or pgoramm [22:01] <Projectns> progrma [22:02] <kepstin-laptop> I think most rtmp clients handle streams being stopped/started just fine, so one thing you might try is just running ffmpeg separately on each file. [22:02] <kepstin-laptop> e.g. using a shell script to loop over the files. [22:03] <Projectns> mhh ok [22:04] <Projectns> so my idea is to create a rtmp stream with some videos for the internet. And the server should running on a linux (Debian Server) without a gui...but the problem is to find a way to stream my video list to the rtmp server [22:07] <Projectns> i found something https://trac.ffmpeg.org/wiki/Concatenate [00:00] --- Wed Jan 28 2015

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ffmpeg-devel.log.20150126
by burek 27 Jan '15

27 Jan '15

[00:21] <cone-504> ffmpeg.git 03Arwa Arif 07master:4c38e960d0ca: avfilter: Port mp=eq/eq2 to lavfi [00:26] <jamrial> the license.md file should be updated with vf_eq.c [00:53] <michaelni> jamrial, fixed [00:53] <cone-504> ffmpeg.git 03Michael Niedermayer 07master:f994000dc5e4: LICENSE.md: add vf_eq [05:59] <cone-701> ffmpeg.git 03Carl Eugen Hoyos 07master:af39b8fec46b: Fix creation of ffprobe-test.nut on remote targets. [06:17] <cone-701> ffmpeg.git 03James Almer 07master:f4f061932754: fate: add Camellia test [10:40] <ubitux> i have issues with lavfi and palettegen [10:40] <ubitux> i'm currently pushing the last frame only on request_frame() [10:41] <ubitux> but this makes flushing not work in various cases [10:41] <ubitux> for instance with ./ffmpeg -f lavfi -i "movie=foo.mp4,palettegen" out.png [10:41] <ubitux> because there is no final explicit pulling [10:41] <ubitux> basically my understanding is that i need to push it from filter_frame() [10:42] <ubitux> but how am i supposed to know that's EOF? [10:50] <ubitux> i can also likely reproduce with the filtering example [10:51] <ubitux> but i have no idea what to do in the filter itself to fix that [10:51] <saste> ubitux, i remind nicolas musing on that [10:51] <saste> i mean about the problem related to EOF signallin [10:51] <ubitux> didn't he just fixed it in ffmpeg only? [10:51] <saste> signalling* [10:52] <saste> let me see the log, I remind that he pushed the patchset [10:52] <ubitux> i mean, even with ffplay we have that problem [10:53] <ubitux> so i probably need to do something at the filter level itself [10:53] <ubitux> https://github.com/ubitux/FFmpeg/commit/0103b517d4640dfd07806d8395c675f6876… [10:54] <ubitux> i have FF_LINK_FLAG_REQUEST_LOOP so basically the request_frame() could return 0 [10:54] <ubitux> (which is the case) [10:54] <ubitux> maybe i could write the loop form itself [10:54] <ubitux> to see if it helps [10:55] <cone-701> ffmpeg.git 03Paul B Mahol 07master:7ccd625a46c5: avfilter/vf_eq: fix leak of input frame [10:56] <ubitux> i can actually reproduce with ffplay foo.jpg -vf tile [10:56] <ubitux> typically [10:56] <ubitux> while it actually works with ffmpeg [10:57] <nevcairiel> isnt the user-code required to flush the filter graph at eof [10:57] <nevcairiel> i know my code does that [10:57] <ubitux> nevcairiel: ffplay doesn't, lavfi device code doesn't, our filtering code doesn't, ... [10:57] <ubitux> but yeah, probably [10:57] <ubitux> (filtering code example*) [10:59] <nevcairiel> in my usecase i just call av_buffersrc_write_frame with a NULL frame, and that somehow does it .. not sure how that works if your source is actually inside the filter graph [11:00] <ubitux> you mean av_buffersink_get_frame()? [11:00] <ubitux> ah, well [11:00] <nevcairiel> no, i mean what i write :d [11:00] <ubitux> mmh [11:00] <ubitux> so you write until get_frame() gives you a EOF? [11:00] <ubitux> instead of just pulling? [11:01] <nevcairiel> i write a NULL once and then query get_frame until it errors out [11:01] <nevcairiel> that seems to work fine [11:01] <ubitux> (ignoring EAGAIN?) [11:01] <ubitux> i wonder if the NULL write is necessary [11:01] <ubitux> but ok [11:02] <nevcairiel> EAGAIN means it needs more input, so ignoring that when i dont have more input seems kinda pointless [11:02] <ubitux> i see, ok [11:03] <nevcairiel> buffersrc has some checks to set eof flags when you feed it a NULL frame, I just figured from that, that I should be sending it one NULL [11:04] <nevcairiel> otherwise continous pulling with get_frame might not flush frames from the middle of the graph, ie. with yadif which would not process a frame because it waits for the next one to process [11:05] <nevcairiel> you would get EAGAIN and frames are stuck somewhere [11:05] <nevcairiel> somehow yadif needs to knwo that there is no more input, and it cant wait for the next frame to do temporal deinterlacing [11:08] <ubitux> ok [11:08] <ubitux> thank you :) [11:08] <nevcairiel> (lack of docs had me read the source of buffersrc and buffersink more than once when writing this) [11:09] <nevcairiel> you notice lost frames quite quickly if you try to deinterlace single frames, ie. from a dvd menu background image :d [11:11] <nevcairiel> I figure if you use movie src in a graph, it would figure out on its own when the file EOF's, and can let the downstreams in the graph know all by itself? [11:13] <ubitux> dunno [11:13] <ubitux> still have issues here [11:13] <nevcairiel> well if it could, it would work :D [11:13] <ubitux> so i get a bunch of EAGAIN, and after flushing i just get a EOF [11:13] <ubitux> and no frame [11:13] <ubitux> :( [11:15] <nevcairiel> softpulldown is the only mp filter left now? we're so close to ditching the whole thing now! [11:16] <saste> nevcairiel, yes, probably today [11:16] <saste> ubitux, from what I understand, there is no way to know a frame is the last one in filter_frame() [11:16] <nevcairiel> softpulldown was the thing that took soft telecine and made it hard telecine? [11:17] <ubitux> saste: yeah, ok [11:17] <ubitux> but then i wonder if my logic is correct [11:17] <ubitux> in the filter, and in the app side [11:18] <ubitux> like EAGAIN, EAGAIN, ..., EAGAIN, EOF [11:18] <ubitux> i just get no frame at all :p [11:18] <nevcairiel> yadif handles the eof logic in request_frame apparently [11:18] <nevcairiel> so shove it there! [11:19] <saste> ubitux, i don't think you can do nothing at the filter level [11:19] <saste> that is, I don't think the palettegen logic can be improved [11:20] <saste> it sounds more like a generic problem [11:20] <ubitux> ffmpeg is able to deal with it, so i guess yeah the filter logic might be able to deal with [11:20] <ubitux> but then following nevcairiel advice, it's not enough [11:21] <cone-701> ffmpeg.git 03Stefano Sabatini 07master:0ca5c4daded0: lavfi/mp: drop mp=eq and mp=eq2 [11:24] <ubitux> :) [11:25] <ubitux> too bad such a filter is gpl though [11:26] <nevcairiel> as all the legacy mplayer stuff, yea [11:26] <ubitux> i don't mind about the others [11:27] <ubitux> anyway.. why i don't get any frame... [11:27] <saste> maybe someone wants to bother and asks the original contributors? [11:27] <nevcairiel> you know how long those lists of people get with all these old mplayer things [11:28] <ubitux> maybe i need AV_BUFFERSRC_FLAG_PUSH... [11:37] <ubitux> doesn't help, wtf... [11:49] <ubitux> alright [11:49] <ubitux> it works if i change the filter like this: http://pastie.org/9861751 [11:50] <ubitux> but it's strange because ffmpeg works anyway without that [11:51] <ubitux> oh well. [12:07] <saste> ubitux, glad you fixed it :-) [12:30] <wm4> <ubitux> too bad such a filter is gpl though <- could have been trivially rewritten? [12:30] <wm4> I mean what the hell [13:02] <durandal_1707> saste: is arwa going to port softpulldown? [13:02] <saste> durandal_1707, I don't think so [13:02] <saste> I think the plan was to get rid of softpulldown [13:02] <saste> i'm not the telecine guru so someone should comment on that [13:06] <cone-701> ffmpeg.git 03Carl Eugen Hoyos 07master:fe47cba7e8da: Decode Prores 4444 XQ with the existing Prores decoder. [13:06] <saste> also carl was telling me that we lack an inverse telecine filter [13:07] <durandal_1707> what fieldmatch+decimate and pullup do then? [13:23] <durandal_1707> i'm going to port softpulldown and remove then mp [13:23] <cone-701> ffmpeg.git 03Paul B Mahol 07master:5a919ced0d3c: avfilter/libmpcodecs: remove unused headers [13:36] <BtbN> Good job Thunderbird. Failed to copy the mail to the Sent folder, so it sends the mail again. [13:36] <j-b> yes, lovely, right? [13:45] <wm4> why would it send the mail again [13:53] <nevcairiel> because its just not that smart [14:31] <cone-701> ffmpeg.git 03Paul B Mahol 07master:e44a4c1f5fcb: avfitler/vf_fieldmatch: fix typo [14:33] <ubitux> durandal11707: ah, i missed "avfitler", is it on purpose? [14:33] <ubitux> ;) [14:33] <durandal11707> nope [14:33] <durandal11707> i'm blind [14:36] <wm4> avhitler [14:36] Action: wm4 runs [15:00] <cone-701> ffmpeg.git 03rogerdpack 07master:c55fa2f09bda: dshow: add properties dialog for tv tuners [15:00] <cone-701> ffmpeg.git 03rogerdpack 07master:ce1bbb08f127: dshow: alert as to ramifications of switching crossbar routing [15:00] <cone-701> ffmpeg.git 03rogerdpack 07master:7c2e26270124: dshow: crossbar dialog was frequently being displayed twice, split up option so it can be just once [15:00] <cone-701> ffmpeg.git 03rogerdpack 07master:61974c7dcca8: dshow: tweak logging [15:00] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:fcb18ab8d03d: Merge remote-tracking branch 'rdp/dshow_tv_tuner' [15:03] <j-b> Seriously, how fucking hard is it to ask authors to have a correctly set git name? [15:03] <j-b> Roger Pack has been in the community for years... [15:06] Action: compn hides behind wm4 [15:07] <av500> j-b: I guess it passed review on the ml..... [15:40] <cone-701> ffmpeg.git 03Timo Rothenpieler 07master:914fd42b8ac5: avcodec/nvenc: Fix b-frame parameter handling [16:22] <ubitux> saste: would you mind documenting in doc/examples or somewhere why AVFilterInOut pointers seems swapped? [16:23] <ubitux> i mean the inputs being the "out"/sinks and the outputs being the "in"/sources [16:23] <saste> ubitux, ping me if I don't do it by tomorrow morning [16:23] <ubitux> it's very confusing and while you explained that to me one or two years ago, i can't figure it out again [16:23] <ubitux> thank you :) [16:27] <saste> ubitux, did I explain it? I remind you figured it out by yourself (note: I'm puzzled every time I read that piece of code after a while, so yes it needs some explanations in the comments) [16:27] <ubitux> i think you did, but maybe i was smarter a while ago [16:28] <wm4> is there a way to get a unique id per packet? I tried the file position, but it obviously doesn't work [16:28] <Daemon404> md5sum(packet) [16:28] Action: Daemon404 runs [16:29] <nevcairiel> i was going to suggest that, but i decided not to troll! [16:29] <wm4> Daemon404: doesn't work, different packets can have identical contents [16:29] <Daemon404> the least painful thing to do might be struct wm4packet { AVPacket; uid; }\ [16:30] <wm4> the hard part is making lavf return wm4packets [16:30] <Daemon404> why does lavf need to return them [16:41] <wm4> I'm trying to resume the demuxer from an arbitrary position [16:41] <wm4> basically, seek back, restart demuxing, and skip packets the decoder has already seen [16:47] <Daemon404> i see. [16:48] <Daemon404> i do that too, but only for the first N packets [16:48] <Daemon404> i used a packet queue system [16:51] <wm4> in this case, I ant to get packets I didn't get before (deselected streams) [16:51] <wm4> *want [16:51] <Daemon404> to what end? [16:52] <wm4> daster stream switching during playback [16:52] <wm4> *faster [16:53] <Daemon404> it sounds a bit funky... wouldnt you have to keep a record of every packet ever seen then? [16:55] <wm4> no, approximately seeking to the position where your packet queue "starts" works pretty well so far [16:56] <Daemon404> so if you already have a queue system, why is it so hard to add a uid to that queue system [16:56] <Daemon404> it seems like putting it in teh avpacket itself is a bad idea [16:56] <wm4> (I could also just decode everything again, but that'd be noticably slower) [16:56] <wm4> that's not the problem [16:57] <Daemon404> oh, cmp. [16:57] <wm4> I have my buffered packets, right? then, when I seek back, I get new packets from lavf [16:57] <wm4> and then I need to check whether a lavf packet is already in the queue, or if it's "new" [16:57] <Daemon404> if this is for stream switching, why cant you just check stream_index [16:57] <wm4> anyway, I guess this approach is just too fragile [16:57] <nevcairiel> also, why doesnt pos work? [16:58] <nevcairiel> shouldnt it be unique for every packet? [16:58] <wm4> nevcairiel: split packets from parsres [16:58] <Daemon404> nevcairiel, nope [16:58] <nevcairiel> well i guess the parser screws those up, but you could just ignore those [16:58] <wm4> Daemon404: on stream switching, I seek backwards to get packets for the "new" stream [16:59] <wm4> Daemon404: but I must also continbue to feed correct packets to the decoders associated with already selected streams [16:59] <Daemon404> sure... but you know which streams are selected [16:59] <Daemon404> and which were previously not [16:59] <Daemon404> why cant you check the index [17:00] <wm4> it's not enough... after seeking back with lavf, lavf will return old packets for streams which were already selected too [17:00] <wm4> and feeding those to the decoder would fuck up everything [17:01] <wm4> unless I actually reset the decoder and decode everything again [17:01] <Daemon404> checkign pos + stream index should mostly always work [17:01] <Daemon404> except if you changed streams exactly between some parsed packets [17:01] <Daemon404> which you should simpyl disallow [17:01] <wm4> that can and does happen [17:01] <wm4> adds a delay [17:01] <Daemon404> ONE WHOLE PACKET [17:01] <Daemon404> HOLY SHIT SLOW [17:02] <wm4> it's also kind of hard to do [17:02] <Daemon404> while (pos == prevpos) feed_packet(); ? [17:03] <wm4> ok, maybe... [17:04] <wm4> assuming lavf doesn't try to reinterleave packets coming from aprsers or some bullshit [17:04] <wm4> *parsers [17:05] <Daemon404> i dont think it does [17:05] <wm4> still, it's awfully fragile [17:05] <wm4> I wonder how other players solve this [17:05] <Daemon404> they dont rice like you [17:06] <wm4> how is it ricing [17:06] <wm4> let the user wait for a second after switching audio tracks? [17:06] <Daemon404> open a new decoder and decode future packets only [17:07] <Daemon404> kill the old one in the bg [17:07] <wm4> that would mean decoding from the start again [17:07] <wm4> for both audio and video [17:07] <wm4> probably not that bad [17:17] <durandal_1707> the softpulldown filter in libmpcodecs actually does not work correctly [17:19] <kierank> quelle surprise [17:19] <kierank> let's be honest libmpcodecs filters are there for show [17:25] <jamrial> time to finally rm -rf libmpcodecs then? [17:26] <durandal_1707> i'm trying to make my port not segfault [17:28] <wm4> kierank: freakshow? [17:37] <kierank> myra: would you be interested in working on aes-ni? [17:38] <cone-701> ffmpeg.git 03James Almer 07master:b8f3b0703c32: tools/crypto_bench: add Camellia support [17:42] <myra> kierank : I haven't done much programming in x86 language. But I would love to do it and learn if it's not urgent. [17:50] <durandal_1707> wm4: in mpv softpulldown dmpi is never returned only modified [17:59] <wm4> yeah, looks very broken [18:00] <wm4> I'll delete it later [18:01] <wm4> don't think anyone even tried to use it [18:01] <cone-701> ffmpeg.git 03Kevin Wheatley 07master:7b6f4191763a: avformat/movenc: Add simplistic 'colr' tag writing support to mov container [18:32] <cbsrobot_> jpeg2000 can have multiple layers with different compression rates - would it be accepted to pass the quality and no. of layers in command line like this: 0,10,20,30 ? [18:32] <cbsrobot_> or has anyone a better idea ? [19:44] <compn> cbsrobot_ : you just want to know what escape character to use ? or if it should be read from a file or something? [19:44] <compn> but: i dont know the answer anyways. :) [21:16] <llogan> http://ffmpeg.gusari.org/viewtopic.php?f=13&t=1902&p=5379#p5379 [21:16] <llogan> I should have him write news releases, tweets, etc. [21:20] <kierank> llogan: lol [21:22] <llogan> kierank: no reply from BL. [21:22] <kierank> :( [21:23] <llogan> i'll prod them again in a week or so [21:24] <llogan> kierank: can you find an existing user? [21:25] <kierank> nope [21:56] <cone-701> ffmpeg.git 03Andreas Cadhalpun 07master:7e857cd57108: configure: use ar and ranlib in deterministic mode [23:05] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:5a1e524ba7e5: Revert "configure: use ar and ranlib in deterministic mode" [00:00] --- Tue Jan 27 2015

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ffmpeg.log.20150126
by burek 27 Jan '15

27 Jan '15

[00:21] <c_14> framerate is specific to a couple of muxers/demuxers. -r is more general [00:21] <c_14> And the actual numbers differ [03:29] <lol123> Can ffmpeg extract video as delimited by title markers? [03:29] <lol123> Like if I just want to extract the theme song from something [03:30] <zumba_ad_> thanks c_14 [03:30] <zumba_ad_> i got disconnected, didn't notice [03:32] <zumba_ad_> need help on this, my sources are full 1920x1080 clip without side or top bars. When I execute this command to merge the 2 mp4 together, the resulting mp4 has huge black bars on side. My video is not squeeze to the middle - http://pastebin.com/k6gy6JDS [03:39] <lol123> -vf 'select=between(n\,211\,730)' [03:39] <lol123> That's the correct syntax, yes? [03:40] <lol123> For discarding everything except the video from frames 211-730? [03:41] <c_14> If the escaping works, ye. [03:42] <lol123> Oh no that recorded everything up to frame 211 [03:48] <lol123> The ffmpeg manual doesn't seem to document the select filter very thoroughly [03:55] <c_14> The expressions are documented in ffmpeg-utils [08:06] <k_sze[work]> I'm running into a weird seek bug. [08:07] <k_sze[work]> I try to exact the first three frames of a video, but ffmpeg returns exactly the same frame. [08:08] <k_sze[work]> This is the command I use: `ffmpeg -ss 0.000 -i my_video.avi -ss 0 -frames 1 -f rawvideo -pix_fmt yuvj444p frame0.yuvj` [08:08] <k_sze[work]> And then I repeat for -ss 0.034 and -ss 0.067 [08:10] <k_sze[work]> (It's a 30 fps H.264 video) [08:46] <relaxed> k_sze[work]: did you try demuxing the h264 first t0 see if that makes any difference [08:46] <relaxed> to, not t0 :) [08:53] <k_sze[work]> relaxed: you mean demuxing into a raw h264 file? [09:00] <relaxed> yes [10:13] <cousin_luigi> Greetings. [12:38] <hEx_Mv> hey guys, ive been using ffmpeg and tried to upgrade. After installing "deb-multimedia-keyring" my ffmpef throws out the following error: ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> ffmpeg: Symbol `av_pix_fmt_descriptors' has different size in shared object, consider re-linking [12:38] <hEx_Mv> Illegal instruction [12:38] <hEx_Mv> uhm well... only once for sure sorry for spam [12:39] <hEx_Mv> i have a raspberry pi with raspian [12:39] <hEx_Mv> already removed the dev multimeia keyring and also ffmpeg with apt-get --purge autoremove ffmpeg [12:39] <hEx_Mv> error still comes up [12:57] <BtbN> looks like you installed ffmpeg for the wrong architecture. [13:47] <opmrcrab> hi guys, hopeing some one can help - im using ffmpeg to capture a HLS stream (m3i8+ts files), my segment format ends in -%03d.ts, and always starts *-000.ts, is there an argument im missing to start it at a number of my choosing? [16:45] <c_14> opmrcrab: -segment_start_number [19:01] <Guest20637> what does [0:v] and [1:v] mean in filter complex? [19:02] <Guest20637> Is there any tutorial for this online? Or any place where I can learn about it in depth> [19:02] <c_14> https://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [19:02] <c_14> It's the same thing. [19:02] <c_14> Just looks a bit different. [19:03] <Guest20637> Basically 0 is the first stream? [19:04] <c_14> The first file [19:04] <c_14> [0:v] means all the video streams of the first file [19:04] <c_14> [0:0] would be the first stream of the first file [19:04] <c_14> [0:a] is all the audio streams of the first file [19:04] <c_14> [0:a:0] is the first audio stream of the first file [19:04] <c_14> etc [19:11] <Guest20637> If we are talking about an image, do we say [o:i] or [o:v] [19:12] <c_14> An image as in a picture is a video stream [19:12] <pzich> also, '0', not 'o' [19:20] <Guest20637> Okay, That clears it up. Thanks a ton! [19:34] <Wegg> I would like to use ffmpeg to record my screen as a sequence of images that I can compile together at a later date into a timelapse video. Right now I have this. . . http://www.pasteall.org/56324 [19:34] <Wegg> But the resolution is horrible. [19:34] <Wegg> Can someone help me figure this out a bit better? [19:35] <c_14> What do you mean, the "resolution is horrible" [19:35] <Wegg> quality of the jpeg it outputs. [19:35] <Wegg> clarity of the image. [19:35] <c_14> use -q:v 2 [19:35] <c_14> or 1 [19:35] <c_14> Just set it to something low. [19:37] <Wegg> Unrecognized option 'q:v' [19:38] <Wegg> I'd much rather use .png but the .png files it produces appear to have their alpha set and the images are transparent. [19:39] <c_14> Where did you put the -q:v ? before or after the input [19:39] <c_14> Also, I don't think the png should have any alpha set... [19:39] <c_14> Never tried it though [19:40] <Wegg> before the input. [19:41] <c_14> Has to be after. [19:41] <c_14> Right before the output filename. [19:41] <Wegg> -video_size 1920x1080 -framerate 1/10 -f x11grab -i :0.0+0,0 -q:v 2 ~weggingt/images/Screenshots/test%05d.jpg [19:42] <Wegg> still get the same error [19:43] <c_14> Works for me. What version are you running? [19:44] <Wegg> 6:0.816-1 [19:44] <Wegg> 6:0.8.16-1 [19:47] <JEEB> first of all, that is most probably a Libav binary (it probably says "Libav developers" and not "FFmpeg developers" in the copyright), and second of all in Libav the ffmpeg binary is something that didn't get updated ,so you will want to use the avconv binary [19:47] <JEEB> although I must say that in any case that is an old binary so I recommend you get a newer one, either Libav or FFmpeg. [19:50] <Wegg> ahh ok avconv works. [19:51] <JEEB> libav 0.8 is also the last version to contain the ffmpeg binary, so after that it will only have avconv, and FFmpeg only has ffmpeg [19:52] <Wegg> png output still is totally transparent when I open it in krita [19:56] <Wegg> avconv -video_size 1920x1080 -framerate 1/10 -f x11grab -i :0.0+0,0 -q:v 2 ~weggingt/images/Screenshots/test%05d.png produces blank images. [19:57] <howzah> why is -map used in ffmpeg? [19:57] <llogan> Wegg: avconv is not from the FFmpeg project. [19:57] <JEEB> also I'm not sure if -q:v does anything with the png encoder :P [19:58] <JEEB> and yes, it's a Libav binary, and it's old (0.8) [19:58] <Wegg> so avconv forked from FFmpeg? [19:58] <JEEB> as I said, I recommend you update your toolset. be it from FFmpeg or Libav [19:58] <JEEB> Libav is a fork of FFmpeg, yes [19:58] <JEEB> it's the reason why FFmpeg is better now than it was before the fork [19:59] <JEEB> howzah, because you need a way to select tracks? [20:00] <howzah> Can you explain it with an example, please? [20:00] <llogan> http://ffmpeg.org/ffmpeg.html#Advanced-options [20:00] <JEEB> ffmpeg -i hurr_durr.nut -map 0:a -c:a shitcodec loers.lara [20:00] <pzich> or https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [20:00] <JEEB> selects all audio tracks from the first input file :P [20:00] <JEEB> and yes, there's plenty of documentation [20:01] <howzah> OOh. Thank you kind people. :) [20:01] <llogan> also see http://ffmpeg.org/ffmpeg.html#Stream-selection [20:03] <Wegg> thank you for your help JEEB [20:03] <JEEB> np [21:04] <nodejs> is there a ffmpeg plugin video player for firefox ? anywhere [21:07] <QKO> hi [21:08] <QKO> how do I record from x11grab, my audio output and my microphone at the same time? [21:08] <nodejs> look into ffmpeg desktop command record on google [21:08] <QKO> I tried adding the microphone and -filter_complex amix=inputs=2 but it gives me [alsa @ 0x7f8d61a78c40] cannot set channel count to 2 (Invalid argument) [21:11] <llogan> https://trac.ffmpeg.org/wiki/Capture/ALSA [21:12] <llogan> http://ffmpeg.org/ffmpeg-devices.html#alsa [21:12] <QKO> I was a fool [21:12] <QKO> I was trying to use the hardware device rather than the dsnoop device [21:59] <hungrybird> my system hangs up whenever I run a complex ffmpeg command. [22:00] <hungrybird> what could be the problem? [22:01] <hungrybird> I read somewhere it is related to pipes. [22:01] <hungrybird> But I couldn't find a fix for it. [22:03] <hungrybird> please [22:21] <hungrybird> Here [22:21] <hungrybird> http://pastebin.com/6k4SPx2h [22:22] <llogan> hungrybird: where is the rest? [22:23] Action: llogan goes on lunch break [22:23] <hungrybird> just that. [22:23] <hungrybird> I am using it through CMD [22:23] <hungrybird> I am using Windows 8 [22:26] <hungrybird> My system's disk usage touches a 100% when I see in task manager [22:28] <hungrybird> fflogger, anything? [22:32] <hungrybird> guys, anything? [22:58] <itsme_> if I have a file like this http://paste.debian.net/plain/142426 , is there an automagic way to split by chapters? [23:21] <itsme_> if I have a file like this http://paste.debian.net/plain/142426 , is there an automagic way to split by chapters? < cough, anyone? (cant be done < is also an answer( [00:00] --- Tue Jan 27 2015

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