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May 2014

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ffmpeg-devel.log.20140520
by burek 21 May '14

21 May '14

[00:00] <michaelni> yes [00:01] <iive> on both intel and amd? [00:03] <michaelni> i dont have a working amd here [00:21] <cone-612> ffmpeg.git 03Oliver Fromme 07master:ab6228316a59: avcodec/dvdsubenc: accept forced flag [00:23] <llogan> i have one if you want me to test [00:34] <michaelni> llogan, as you prefer, i dont think its very important ... [00:35] <cone-612> ffmpeg.git 03Lukasz Marek 07master:7336e39f3cd2: lavu/fifo: add av_fifo_alloc_array function [00:35] <cone-612> ffmpeg.git 03Lukasz Marek 07master:51bad41a13c7: lavd/jack_audio: use av_fifo_alloc_array [00:35] <cone-612> ffmpeg.git 03Lukasz Marek 07master:4d686674b265: ffmpeg: use av_fifo_alloc_array [00:35] <cone-612> ffmpeg.git 03Lukasz Marek 07master:d15db8e413e9: lavfi/buffersink: use av_fifo_alloc_array [00:35] <cone-612> ffmpeg.git 03Lukasz Marek 07master:ce051ceefc70: lavfi/vf_fps: use av_fifo_alloc_array [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:bc4f362c9281: lavc/frame_thread_encoder: use av_fifo_alloc_array [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:8aa29880365c: lavc/flac_parser: use av_fifo_alloc_array [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:74f9c5947d69: lavf/dvenc: use av_fifo_alloc_array [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:c61cdefa4d79: lavf/audiointerleave: use av_fifo_alloc_array [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:2ed9e17ed179: lavf/audiointerleave: check for allocation failure [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:37fa4b9bb967: lavf/audiointerleave: return more meaningful error codes [00:36] <cone-612> ffmpeg.git 03Lukasz Marek 07master:b9419b58826e: lavf/ftp: favour EPSV over PASV command [00:36] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:6218831844fa: Merge remote-tracking branch 'lukaszmluki/master' [00:38] <llogan> michaelni: i guess i'll pass. i don't know how to get dezicycles anyway, but feel free to ping me (w/ instructions) if you do need me to test some other time [00:40] <michaelni> ok [00:56] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:b2a8bc3d0482: ffmpeg/set_encoder_id: dont allocate encoder string if one is already set [02:16] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:be2bbc5266d1: ffmpeg_opt: dont copy "encoder" metadata unless stream copy is used [02:47] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:0739614c9c48: avcodec: remove no longer needed ff_codec_close_recursive() [02:47] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:8408720453d7: avcodec/smvjpegdec: Dont ignore avcodec_close() return code [02:48] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:9893e883ea21: avformat/mux: Remove localized encoder strings [03:30] <cone-612> ffmpeg.git 03Janne Grunau 07master:132a5711a2aa: check: add libavcodec/vda_internal.h to SKIPHEADERS-$(CONFIG_VDA) [03:30] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:447b988b5319: Merge commit '132a5711a2aaaa6c7d139a0a30046a6ad41b64aa' [03:38] <cone-612> ffmpeg.git 03Janne Grunau 07master:afe176265480: build: export library dependencies in ${name}_FFLIBS [03:38] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:9e6769d2dc7b: Merge commit 'afe176265480880e1f702c96a8ba434b0d88728b' [03:51] <cone-612> ffmpeg.git 03Janne Grunau 07master:c23c96b638cb: lavf: add av_stream_get_side_data [03:51] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:7d25af1547f8: Merge commit 'c23c96b638cbf6c489fd301e6b3d5555632fba37' [04:34] <cone-612> ffmpeg.git 03Katerina Barone-Adesi 07master:e50ae60d4618: avcodec/fate-idct8x8: Defined behavior: eliminate negative left-shifts. [06:24] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:6310eb8010b7: avutil/cpu: force mmx on selection of higher x86 SIMD features [14:32] <ubitux> [opengl outdev @ 0x1aa8520] SDL driver: 'x11'. [14:32] <ubitux> [opengl outdev @ 0x1aa8520] Provided pixel format 'yuyv422' is not supported [14:32] <ubitux> :( [14:33] <ubitux> isn't it supposed to insert the correct pix fmt convert by itself? [14:35] <thardin> what does it take? yuv420p? [14:36] <ubitux> yuv420p seems to be accepted yes [14:38] <thardin> can you insert one manually? [14:38] <thardin> (shouldn't be necessary of course) [14:38] <ubitux> yep sure that's wjat i did [14:38] <ubitux> what* [15:36] <cone-79> ffmpeg.git 03Yu Xiaolei 07master:2491579d4715: configure: arm: do not enable fast-unaligned if it is explicitly disabled via commandline [16:21] <Keestu> hello, avcodec_decode_video2 (), does it have an impact on thread_count value on Codec Context ? [17:22] <kurosu> BBB / michaelni: was leaving some inline asm stuff in C files intended? [17:22] <kurosu> I recently ported the fpel part but seeing the hpeldsp part, this looks like it was intended [17:23] <kurosu> (only the mmx implementations being left) [21:26] <cone-711> ffmpeg.git 03Alexis Ballier 07master:c9ab8790118c: Fix build with --enable-ladspa --disable-frei0r. [21:38] <cone-711> ffmpeg.git 03Michael Niedermayer 07master:be656c362d43: avcodec/h264_picture: Fix build without error resilience [21:40] <Daemon404> i always see these sorts of fixed [21:40] <Daemon404> why not just iterate over all combinations once and be done wth it [22:24] <nevcairiel> Daemon404: i bet libav keeps breaking it [22:32] <Daemon404> diego runs such a script [22:34] <nevcairiel> but they dont support that buildmode, iirc [22:34] <Daemon404> no its some script sitting on a pc somewhere [22:34] <Daemon404> small wrapper [23:08] <cone-711> ffmpeg.git 03Dale Curtis 07master:ca2fd25988d3: Fix compilation errors when deprecated features are disabled. [23:20] <kurosu> what does HAVE_MMX_EXTERNAL mean? yasm supporting mmx ? [23:21] <Daemon404> yes [23:21] <Daemon404> im not sure why that exists... [23:21] <Daemon404> maybe im mistaken [23:27] <kurosu> if I'm not mistaken, yasm/nasm is for x86 [23:27] <kurosu> so yasm not supporting mmx (only from sse2 on?) is as weird as a cpu having only sse2 on [23:28] <kurosu> but ok [23:28] <kurosu> I don't thing I'm going anywhere with the code in relation, anyway [23:31] <J_Darnley> I think it must come from the idea that someone may use a gcc so old it doesn't support inline mmx instructions so there may be a yasm so old that it too doesn't support mmx instructions. [23:48] <jamrial> those macros are mainly for when ffmpeg is configured with --disable-{opt} [23:49] <kurosu> ok [23:49] <kurosu> no that I think of it, most of those functions are used for when block size is 8 [23:49] <kurosu> *now [23:49] <jamrial> the oldest yasm version we support is 0.8, which can assemble up to AVX and FMA3 [23:50] <kurosu> so, they may actually be ran even on newer cpus [23:50] <kurosu> (because 128b-wide regs don't provide a benefit) [00:00] --- Wed May 21 2014

1 0

ffmpeg.log.20140520
by burek 21 May '14

21 May '14

[00:36] <mjuszczak> I'm having trouble re-using a ./configure line I've used in the past. It's failing on --disable-ss. Is that no longer a valid configuration option? When was it removed? [00:38] <c_14> It's not listed in ./configure --help so I'm pretty sure it isn't valid. As to when it was removed, I have no idea. You might want to look at the git commit messages pertaining to the configure script. [00:39] <llogan> i don't remember a --disable-ss [00:39] <c_14> You might be thinking of --disable-sse ? [00:42] <mjuszczak> http://pastebin.com/yDxfRRX8 [00:42] <mjuszczak> A literal copy of the output of $(ffmpeg) [00:43] <mjuszczak> unless that's a truncated ./configure :( [00:43] <mjuszczak> I'm trying to replicate a compilation and I only have the binary [00:44] <klaxa> do you remember what --disable-ss is supposed to be? otherwise check out the old source-tree and check ./configure --help [00:44] <klaxa> when i doubt, omit it and don't worry about it [00:45] <c_14> Judging by the opts around it, I'm guessing it's either --disable-sse or the sse3,sse4 variation thereof. [00:46] <mjuszczak> c_14: OK. IS there a chance other things got cut off as well? [00:46] <klaxa> probably [00:47] <mjuszczak> I don't suppose there's a way to take a binary of ffmpeg and know what options were configured into it, is there? [00:47] <klaxa> well... everything is in the paste you posted, but the options changed [00:48] <mjuszczak> I'm trying to recompile that exact version [00:49] <klaxa> if you want to compile that exact version you have to get the exact same source-tree [00:49] <klaxa> then the exact same configure options should work [00:50] <mjuszczak> I have it, but --disable-ss is failing. [00:52] <mjuszczak> Well, I lied. That's 0.10.2 and I'm trying to compile a later version of 0.10.x, because Icouldn't find 0.10.2. [00:52] <mjuszczak> But I didn't think configure options would change in a minor release [00:52] <llogan> why are you trying to compile the 0.10 branch? [00:52] <mjuszczak> legacy application, moving to a different OS, trying to reduce variables [00:53] <llogan> there is no --disable-ss, and there never was. c_14 already mentioned what it is probably supposed to be, but I see few reasons to actually disable SSE [00:53] <llogan> are you really compiling on/for AMD barcelona? [00:53] <mjuszczak> no [00:54] <mjuszczak> Just trying to keep things exactly as they were on the old setup :( [00:54] <mjuszczak> I'm not sure why the previous "guy" chose the flags he did, but he did.... many years ago. [00:55] <llogan> i recommend not doing what he did and just use a simple, sane configuration. [00:56] <mjuszczak> OK. [00:56] <mjuszczak> Thank yu [00:56] <mjuszczak> you* [00:57] <llogan> like: --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libx264 [00:58] <llogan> and whatever other useful external libraries you may need. [01:16] <mithodin> Hi guys. Can someone tell me why this creates a video with no audio output: http://bpaste.net/show/288236/ but when I leave out line 6, I get video with audio output? [01:18] <c_14> If you leave line 6 in you have 2 implicit outputs and ffmpeg will only use one if you don't name them explicitly and map them. [01:18] <mithodin> oh [01:18] <c_14> try adding an [a] at the end of line 6, an [v] at the end of line 10 and a -map '[a]' -map '[v]' somewhere before the output filename [01:19] <mithodin> does it matter if before or after -t ? [01:19] <c_14> Doesn't matter. [01:25] <mithodin> so this should work, right: http://bpaste.net/show/288243/ ? [01:25] <mithodin> because it does not [01:30] <c_14> From what I can see, it should work. [01:30] <c_14> hmm [01:32] <mithodin> http://pastebin.com/PemDjqwM this is Part of the output, maybe that helps? [01:36] <c_14> Do you get audio when you put [audio] in the map instead of [audio1] ? [01:36] <c_14> If that's the case it might be the afade muting all the audio for some reason. [01:37] <mithodin> yes, that works [01:38] <c_14> Maybe try sequentially adding the afade filters, start out with just the first one and so on? [01:38] <mithodin> I'm currently rewriting it so I afade before concatting [01:40] <mithodin> okay, for some reason, this works: http://bpaste.net/show/288268/ [01:41] <c_14> I guess the afades were fighting with each other and muted everything? [01:41] <mithodin> possibly... [01:42] <mithodin> I have a lot to learn about ffmpeg ^^ [01:42] <mithodin> thanks though [01:42] <c_14> np [01:42] <c_14> Not that I did much. [01:42] <mithodin> good night (or whatever time of day it is for you) :-) [02:53] <GEEGEEGEE> Anyone know what woudl cause this error in ffserver? stream number does not match registered feed [03:02] <Shurtagul> How do you use ffmpeg to change the resolution of a video? [03:04] <llogan> Shurtagul: http://ffmpeg.org/ffmpeg-filters.html#scale [03:04] <llogan> ffmpeg -i input -vf scale=640:-2 output [03:04] <llogan> ffmpeg -i input -vf scale=1280:720 -acodec copy output [03:04] <llogan> etc [03:07] <Shurtagul> Thank you :) [03:07] <azk> GEEGEEGEE: Iirc I have gotten that error when trying to hardcode subtitles on the fly for ffserver. [03:07] <azk> Instead it sends two streams to ffserver and it doesn't recognize one of them. [03:07] <azk> GEEGEEGEE: Getting actual logs instead of a single line would make it easier for people to debug. [03:08] <GEEGEEGEE> i cant find where the logs are saved :/ [03:10] <azk> run your commands and pipe their output to a file? [03:21] <dumdidum> hello, i have a vob/mpeg-1/2 file with broken time index. it runs without problems just the time is wrong. vlc shows a length of around 35 minutes but is about 2 hours. when playing the current time goes up to until 1:30 hour than the current time gets zero and goes up again to 35 minutes. than the video is over. how can i fix this? [03:57] <ScottSteiner-> How can I make ffmpeg use a subtitle's timestamps when i seek before the input file is loaded? ie ffmpeg -ss 00:01:00 clip.mkv -vf "ass=subtitle.ass" output.mkv [03:59] <sacarasc> So, if you seek to 1 minute, and a sub is meant to be up from like 58 to 1:02, you want it to show? [04:00] <ScottSteiner-> yes. if i seek to 1 minute and do a 10 second clip, it tries to show the subtitles for 00:00:00 through 00:00:10 instead [04:00] <sacarasc> Oh, that's weird. [04:00] <sacarasc> I don't know, then. [04:00] <relaxed> you'll probably have to edit subtitle.ass [07:28] <sim590> I'm creating a video with sound by merging an audio file and a video file. when I start the process, it seems to take alot of time. Do I always have to wait the length of media to wait for it to complete or what ? It seems that if I have a file that is 1 minute long, I have to wait 1 minute for it to process through ffmpeg... ? [08:34] <pagioss> hi all, is there an executable that install ffmpeg on windows just like a normal software? [08:45] <pzich> what do you mean like normal software? you mean not using the prompt to run it? [09:14] <pagioss> pzich: yes [10:19] <sim590> why doesn't ffmpeg stop at t=73 seconds when I do this : ffmpeg -t 73 -i out.mkv -ss "00:27:33" -i ~/Musique/Écoute/gaming/The\ Legend\ of\ Zelda\ Ocarina\ of\ Time\ Soundtrack-ZzwvItK3JPI.mp3 -af "afade=t=out:st=69:d=3" out_son.mkv [10:30] <blippyp> maybe the track is shorter than a 1 minute and 40.33 seconds? [10:30] <blippyp> I normally put the -t near the end, not sure if that makes a difference..?? [10:31] <blippyp> what are you trying to do - you have two inputs and your -t and -ss are at the beginning? [12:45] <intracube> hello. is there any way to make the deshake filter more aggressive? [13:14] <TheMechanist> hi all [13:15] <TheMechanist> Does anyone of you know a way to compare to video files in a player? In essence I have two files that I want to play simultaneously and flip back and forth between their display to see the (subjective) quality difference between them.... [13:57] <Sairon> I'm trying encode a png sequence with libx265, when I run 'ffmpeg.exe -r 59.94 -f image2 -i Intro_2%04d.png -strict experimental -c:v libx265 -preset slow test.mp4' I get '[libx265 @ 04825940] 4:4:4 support is not fully defined for HEVC yet. Set -strict experimental to encode anyway.', if I do I get a blank output as result. 4:4:4 seems to indicate some kind of chroma sub sampling, from my [13:57] <Sairon> understanding the problem is on the output end ( output stream is stated as yuv444p ). If that's likely the case, how do I change the output pixel format passed to libx265? [14:11] <intracube> Sairon: -pix_fmt yuv420p [14:12] <intracube> to show all available pixel formats: ffmpeg -pix_fmts [14:12] <Sairon> awesome, that seems to do the trick :) [14:12] <Sairon> thanks a lot [14:13] <intracube> not sure if yuv420p is optimum for you but it's what I always default to [14:19] <iive> 4:4:4 means that there is no chroma sub sampling, that is you have 3 full components for each pixel in yuv mode. [14:20] <iive> 4:2:0 means that color differential chroma components (UV) are at half resolution (Vertically and Horizontally, so totally quarter the size) [14:22] <iive> the lower resolution means data loss, that's why when converting from rgb yuv444 is picked. However most broadcast quality video is at the tv format 420 [14:22] <Sairon> ah [14:30] <Berra> I built ffmpeg from git and lost libavresample. How do I go about getting it back? [14:31] <Berra> http://sprunge.us/TbGa Is any of these the same thing in practice? [14:34] <excalibr> Hi! When you do direct copy from a video, is it possible to copy multiple timecode ranges in one go? [14:34] <excalibr> eg 10-60sec, 120-300sec [14:52] <c_14> ye, if you want one output file you'll need a filter_complex and the trim/concat filters. If you want them split to separate files, just use -ss and -t/-to before each individual output file. [14:53] <c_14> Berra: recompile? [14:54] <Berra> I updated my mpv to mpv-git and did not need that lib anymore. [14:57] <c_14> I'm confused. How did you lose the libavresample? Did you delete it? [15:00] <Berra> I installed the git version instead. Well now everything works anyway [15:07] <c_14> Well, long as it works. [16:18] <Keestu> hello, avcodec_decode_video2 (), does it have an impact on thread_count value on Codec Context ? [16:24] <Neppy> hm for issues with ffmpegsource, should i ask here, in #avisynth or where else? [16:29] <DeadSix27> Neppy, the owner of the repo is in this channel, and i cant see him anywhere else [16:29] <DeadSix27> so i assume yes, here? [16:36] <Neppy> ah [16:39] <Neppy> well trying to open a mkv file (others i've tried in the past worked fine, just not this one) with ffvideosource/ffaudiosource; ffvideosource just complains there is no video track though :o when i extract the videotrack with mkvcutter and use the extracted .h264 file with ffvideosource it goes on without error but still errors at the ffaudiosource (which also tries to use the mkv file) which [16:39] <Neppy> suggests either avs or ffmpegsource for some reason has an issue reading this mkv file.. i tried using track=0 and track=00 on ffvideosource (since mkvcutter adds _Track00 to the extracted filename) but then it says videotrack has no frames.. [16:39] <Neppy> using 32-bit avs with 32-bit (x86) ffms2 2.18 rc1.7 (windows 7 64-bit if that matters) [16:39] <Neppy> well that got a bit long ^^" [16:42] <DeadSix27> Neppy, that wasnt meant in the evil way btw, im not sure either, never had to ask for support yet. [16:42] <DeadSix27> (for that piece of software i mean) [16:44] <JEEB> Neppy, can you cut a small sample of that file to see how borked it is? You can use dd or something simple like that in most cases to cut off first 10-50MB [16:44] <JEEB> also generally you should be using as new as possible ffms2 builds [16:44] <JEEB> argh, this was not #avisynth [16:45] <JEEB> goddamnit for you posting it on multiple channels :P [16:45] <Neppy> haha [16:45] <JEEB> libav* are used by ffms2, but on #ffmpeg in general it isn't on-topic. esp. since the matroska parser isn't lavf's [16:46] <Neppy> ah [16:46] <muken> optionally, you can use lavf's though [17:51] <gneekd> how could i create a stream for ios devices from a number of different video files? programatically... [20:56] <voip_> hello guys [20:56] <voip_> i am copy video stream [20:56] <voip_> whats mean [flv @ 0x30cea20] pts (-74) < dts (40) in stream 0 [20:56] <voip_> av_interleaved_write_frame(): Invalid argument ? [20:57] <voip_> ok [21:01] <voip_> http://pastebin.com/Pg7wnFLv [21:06] <voip_> fflogger, llogan http://pastebin.com/Pg7wnFLv [21:16] <llogan> voip_: fflogger is a bot [21:17] <llogan> -preset does nothing when you are re-encoding [21:17] <llogan> s/re-encoding/stream copying [21:30] <voip_> llogan, s/re-encoding/stream copying ? [21:32] <llogan> -preset does nothing when you are stream copying [21:32] <llogan> not that this info helps you [21:37] <voip_> ok, i am runing witout preset, some error [21:45] <voip_> llogan, ffmpeg -i rtmp://10.10.10.10/live/Ch7_1200 -acodec copy -vcodec copy -f flv rtmp://11.11.11.11:1935/live/live same error [21:59] <llogan> voip_: i don't know. try the ffmpeg-user mailing list. [00:00] --- Wed May 21 2014

1 0

ffmpeg-devel.log.20140519
by burek 20 May '14

20 May '14

[01:47] <cone-365> ffmpeg.git 03Carl Eugen Hoyos 07master:e705d0ce1cd3: Set dwSuggestBufferSize to largest chunk size for every stream in avi. [03:47] <cone-365> ffmpeg.git 03Olivier Langlois 07master:76191c08f88f: ffmpeg: Use av_gettime_relative() [06:08] <john2_> I'm considering trying to add support for displaying pre-existing subtitles after seeking to the MKV demuxer, and I had a couple questions about implementation. [06:09] <john2_> Specifically, should I just make the demuxer's read_packet function return the pre-existing subtitle packets before returning anything else after the seek? [06:10] <john2_> If so, will the fact that the timecodes of the pre-exisintg subtitles come before the current timecode cause any problems with codecs/consumers? [06:12] <john2_> (An example of pre-existing subtitles is if we have a video with a subtitle that is on-screen from timecode 00:05 to 00:10 and I seek to 00:7.) [06:25] <jamrial> can't help you much since this is not my field of expertise, but afaik the CueDuration element was introduced specifically for this [06:26] <jamrial> http://lists.matroska.org/pipermail/matroska-devel/2012-September/004249.ht… this is an old email proposing the element. it's been in the spec for a while and might have changed some [06:27] <jamrial> so at least in theory, devices/demuxers that support that element should be able to handle pre-existing subtitles just fine [06:29] <jamrial> ffmpeg has been muxing matroska files with that element for a while. the demuxer however ignores it [06:45] <cone-365> ffmpeg.git 03Michael Niedermayer 07master:4a30f08505a4: avcodec/diracdec: move mc buffer allocation to per frame [10:18] <kurosu> crash: make fate-h264-reinit-small_420_8-to-large_444_10 CPUFLAGS=sse2 [10:19] <kurosu> fault in ff_hscale8to15_X8_sse2 [10:19] <kurosu> pmaddwd (%r10,%rax,2),%xmm1 [10:19] <kurosu> p/x $rax [10:19] <kurosu> $3 = 0x1 [10:19] <kurosu> can't pmaddwd from an unaligned address [10:20] <kurosu> p/x $r10+2*$rax : 0x8ba8b42 btw [10:20] <kurosu> ($r10 is aligned) [10:21] <nevcairiel> at least the crash is not in h264 code ;) [10:22] <kurosu> well, if you intend to test a change that is evaluated by h264, h264 better be working without the change ;) [10:23] <nevcairiel> that doesnt sound like something that might occur randomly though, and that test passes in fate generally [10:24] <kurosu> CPUFLAGS=sse or not specifying CPUFLAGS passes [10:25] <kurosu> so I guess the issue occurs whenever the cpu isn't > SSE2 [10:25] <kurosu> CPUFLAGS=sse4.2 passes [10:26] <kurosu> CPUFLAGS=sse4 doesn't crash but is incorrect [10:26] <kurosu> CPUFLAGS=ssse3 crashes, CPUFLAGS=sse3 (does it exist?) passes [10:27] <kurosu> I guess a sse2-only device is really needed for fate :D [10:27] <kurosu> well =<sse2 [10:28] <kurosu> is that swscale or swresample or ? [10:28] <nevcairiel> swscale [10:29] <nevcairiel> libswscale\x86\scale.asm [10:29] <nevcairiel> the only line matching that would be 292 [10:29] <nevcairiel> i think [10:32] <kurosu> well, it was to fill a bugreport [10:32] <kurosu> I'm not going to dig into swscale [10:51] <kurosu> the issue is most probably in the calling code, as i'm wondering what's to scale when the filter size is 1 (haven't checked what it does actually) [14:02] <cone-612> ffmpeg.git 03Mickaël Raulet 07master:04db5794cd97: hevc: templatize pred_planar [14:02] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:93318983c3e5: Merge commit '04db5794cd97e4b33ec2f963ef7f967722a456eb' [14:18] <cone-612> ffmpeg.git 03Anton Khirnov 07master:a1c2b48018b0: hevc: templatize intra_pred [14:18] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:844765855000: Merge commit 'a1c2b48018b09d2613f075ec0748c95bd520ac00' [14:27] <Voicu> any ideas why I get the entire buffer in the extradata field instead of just the necessary amount? [14:28] <Voicu> i.e. I open a buffer and av_find_stream_info consumes the entire buffer and puts it in the extradata field [14:28] <Voicu> instead of just the 25bytes or so the h264 codec would need [14:28] <Voicu> I also get the whole thing as a single frame when I do av_read_frame afterwards [14:34] <cone-612> ffmpeg.git 03Mickaël Raulet 07master:4c390b1ba98e: hevc/intra_pred: optimize EXTEND_()* [14:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:454920187f15: Merge commit '4c390b1ba98e6aec787aa0e7ea57ed7cb38283b7' [14:54] <cone-612> ffmpeg.git 03Mickaël Raulet 07master:25bb7eaf9eb1: hevc/intra_pred: drop unnecessary conditions in loops [14:54] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:e98d3a7d51da: Merge commit '25bb7eaf9eb14e6e00264bf584aa4d380d3bc29f' [15:09] <cone-612> ffmpeg.git 03Anton Khirnov 07master:c9f8809ee4c5: hevc/intra_pred: simplify neighboring sample derivation [15:10] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:3791741cf97f: Merge commit 'c9f8809ee4c576d5833865039bc1c85754448f67' [15:36] <cone-612> ffmpeg.git 03Anton Khirnov 07master:dc40d88625d7: avconv: do not use poorly defined and undocumented AVStream.pts [15:36] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:2de2b991cac6: Merge commit 'dc40d88625d7e402d58ac3f3df69fbf27aa31ea0' [15:36] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:87f5ede6b52e: ffmpeg: add last_mux_dts_plus_duration [16:47] <kurosu> nevcairiel, are you "he(ndrik)leppkes" ? [16:47] <nevcairiel> yes [16:48] <kurosu> regarding the bug on which you commented, for some reason, the sse4 code doesn't crash [16:48] <kurosu> doesn't make sense, but running without CPUFLAGS makes it pass [16:48] <kurosu> and my CPU has sse4.x at best [16:49] <kurosu> but the ssex instances should be all affected [16:49] <kurosu> I suspect the real issue is to try and convolve with a filter of size 1 [16:50] <kurosu> maybe it is intended to be generic, but sounds like unnecessary complexity [16:56] <kurosu> nevcairiel, could you check if those tests fails for you (as you tested the above): [16:56] <kurosu> make fate-msmpeg4v1 CPUFLAGS=mmxext [16:56] <kurosu> make fate-msmpeg4v1 CPUFLAGS=sse2 [16:56] <kurosu> both fail for me [16:58] <kurosu> mmx+mmxext/sse2 does not fail [16:58] <kurosu> maybe I'm the one failing at understanding CPUFLAGS [17:22] <nevcairiel> my bet is there is some stupid check somewhere that checks for if flags & MMX or something to change behaviour, but doesnt account for SSE2 only being enabled [17:35] <Plorkyeran> 0eec06ed8747923faa6a98e474f224d922dc487d appears to have broken statically linking ffmpeg with gcc on linux for me [17:35] <Plorkyeran> there's a configure check for if -lrt is needed for clock_gettime [17:35] <Plorkyeran> but it only adds -lrt to lavc's libs in the pkg-config file [17:36] <Plorkyeran> so the new usage in libavutil fails to link [17:40] <kurosu> michaelni, is running a fate test with eg only CPUFLAGS=sse2 supported? or should I specify CPUFLAGS=mmx+mmxext+sse2? [17:42] <michaelni> Plorkyeran, mail the author [17:43] <michaelni> kurosu, only sse2 possibly is not as theres no such cpu, but either the code should print a warnng or work it shouldnt just fail [17:47] <kurosu> ok I guess it is ok to assume mmx+mmxext+sse2 in the code [17:47] <kurosu> I'm going to run CPUFLAGS=mmx+mmxext out of curiosity (one test failing it seems) [17:50] <michaelni> we have a fate client that tests mmx+mmxext "http://fate.ffmpeg.org/report.cgi?time=20140519042836&slot=x86_32-debian-kf…" [17:50] <michaelni> though its hw supports more [17:53] <kurosu> fate-msmpeg-v1 was failing here but I'll confirm later on [17:53] <kurosu> (can't remember the test name actually and can't verify atm) [17:53] <kurosu> win64 [18:31] <kurosu> bah, passes now... [18:33] <cone-612> ffmpeg.git 03Anton Khirnov 07master:ed7922faac4e: mux: drop one of the hacks comprising compute_pkt_fields2() [18:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:dad54e4a62ae: ffmpeg: set dts for subtitles [18:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:6552e23d560c: avformat/mpegtsenc: fix dts for chained muxing aac in mpegts [18:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:ebbc33a42d1d: avcodec/libmp3lame: do not attempt to flush lame if no data was input [18:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:f29cdfec961d: avcodec/libvorbisenc: do not attempt to flush if no data was input [18:34] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:e3c1d9a56c71: Merge commit 'ed7922faac4ea4c893efc3cdf8102ebc49fcc011' [18:49] <cone-612> ffmpeg.git 03Anton Khirnov 07master:a312f71090ee: lavf: deprecate now unused AVStream.pts [18:49] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:999a99c865cc: Merge commit 'a312f71090ee620ee252f2034aef6b13e2dafe9c' [18:50] <michaelni> ubitux, can you check if dad54e4a62ae5d81c33530f2748f57e40363921d is correct ? (not setting dts for subtitles doesnt mean it gets the correct value thus it really has to be set but i dont know if thats the correct value) [18:56] <cone-612> ffmpeg.git 03Anton Khirnov 07master:cdf58f0599c3: avpacket: fix copying side data in av_packet_copy_props() [18:56] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:4a2fd251f477: Merge commit 'cdf58f0599c39852ee3beafe5f64af7d57d4215b' [19:01] <ubitux> michaelni: well, in the subtitles demuxing api we do set dts=pts all the time [19:01] <ubitux> now i don't know if that is meaningful for muxing in ts & friends [19:01] <ubitux> so& i have no idea [19:01] <ubitux> is that fixing anything? [19:02] <ubitux> ah it's after the drop from AVStream.pts [19:02] <ubitux> michaelni: shouldn't it be set in lavf itself? [19:03] <ubitux> because if that's necessary, won't it break apps? [19:03] <michaelni> maybe, also i didnt merge that removial yet as it breaks other things [19:04] <michaelni> (as written in the commit message) [19:05] <michaelni> about fixing anything, dts is needed for many muxers to know packet ordering, i dont know if the value and ordering is what ts expects [19:07] <michaelni> also the dts = nopts support removial itself can also break apps [19:08] <michaelni> iam also not sure we should try to remove it or if its actually a usefull feature, ffmpeg does use it in some cases [19:08] <cone-612> ffmpeg.git 03Anton Khirnov 07master:9929b3564c0d: pthread_frame: simplify the code by using new AVPacket API [19:08] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:af72f62d7b2a: Merge commit '9929b3564c0dca42ed7baa6798ef15b6f0013c83' [19:20] <cone-612> ffmpeg.git 03Vittorio Giovara 07master:bddd8cbf6855: Add transformation matrix API. [19:20] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:ef1d4ee2f8b6: Merge commit 'bddd8cbf68551f6405b2bf77cc3e212af9fbe834' [19:21] <j-b> ah, interesting [19:28] <cone-612> ffmpeg.git 03Vittorio Giovara 07master:853cc025d63e: mov: store display matrix in a stream side data [19:28] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:07941c2cb28d: Merge commit '853cc025d63ee2539fc0460dab62c5b9a3fd2043' [19:35] <cone-612> ffmpeg.git 03Martin Storsjö 07master:82fc9f4b3824: display: Include mathematics.h for fallback definition of NAN [19:35] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:cd7ae5af16d7: Merge commit '82fc9f4b38244236a6ca7f946662ca653044a04c' [20:06] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:55fe1582cb28: avcodec/pthread_frame: remove unused variable [20:18] <compn> i had an idea for a filter. make non-black color pixels brighter. so keep black and near-black pixels dark, but brighten up the other pixels. is that possible ? :) [20:19] <nevcairiel> thats called gamma [20:19] <compn> adjusting gamma still adjusts the black [20:30] <llogan> compn: play around with curves [21:15] <J_Darnley> Who organised ffmpeg's migration from svn to git? I seem to recall talk of a script to parse true authors out of commit messages. Was that ever published anywhere? [21:15] <Daemon404> no [21:15] <Daemon404> and iirc it was janne [21:17] <beastd> N [21:18] <beastd> J_Darnley: I do not know what was done for ffmpeg migration, but I remember migrating svn repos to git at work [21:18] <J_Darnley> Are you janne? [21:18] <Daemon404> janne is janne [21:19] <Daemon404> janne is over at libav [21:19] <J_Darnley> That was my next question. [21:19] <beastd> no, i am not janne of course ;) [21:19] <Daemon404> on the internet, nobody knows youre a dog [21:20] <J_Darnley> :) [21:21] <J_Darnley> I guess I'll send a email then.] [21:21] <beastd> J_Darnley: What exactly is your goal? There is only committer information in svn. Do you also want to transfer author information? i think that is not fully possible [21:21] <J_Darnley> It isn't anything serious [21:21] <J_Darnley> git-svn would probably suffice [21:22] <J_Darnley> beastd: yeah that's want I was looking for [21:22] <Daemon404> the bets you can do is try and regex the commit message [21:22] <Daemon404> beastd, little known fact: svn can ave a separate author [21:22] <J_Darnley> We used to say "patch submitted by $me" [21:22] <Daemon404> but nobody does it. [21:24] <beastd> when i converted svn repos to git i could specify a mapping for committer to committer/author i am sure [21:25] <Daemon404> yeah [21:25] <Daemon404> but usually the info is in the message (if at all) [21:25] <Daemon404> you more or less need to manually build up a table of nicks/names and regex [21:25] <Daemon404> and then poke it more [21:25] <Daemon404> it is super tedious [21:25] <beastd> Daemon404: if no one used it, it doesn't help :) That is sth git got right [21:26] <Daemon404> http://esr.ibiblio.org/?p=5634 [21:26] <Daemon404> relevant [21:26] <Daemon404> (yeah its esr, but for once his post is kinda interesting) [21:28] <beastd> ah yes, i heard about that conversion. i do not envy esr for that task [21:28] <Daemon404> rcs *and* arch [21:28] <Daemon404> only at GNU. [21:29] <Daemon404> this is actually the onl project i know which used arch [21:29] <Daemon404> only* [21:43] <J_Darnley> Wow 29 years [21:43] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:833ec9177d17: ffmpeg/do_video_stats: fix used timebase [21:52] <beastd> J_Darnley: If it is not so important I would first try to just convert and just map the Subversion committers. Unfortunately I cannot recall what I used to do it at work, but it may just have been git svn with an additional authors file provided. should be in the git docs or maybe the book. [22:11] <cone-612> ffmpeg.git 03James Almer 07master:b82274487b23: lavc: define STRIDE_ALIGN as 32 when compiling with AVX support [22:11] <cone-612> ffmpeg.git 03Thierry FAUCK 07master:41b928c5fad7: avcodec/ppc/asm: fix build with ABI v2 [23:08] <michaelni> ubitux, ok to apply: "[FFmpeg-devel] af_silencedetect.c: silencedetect reporting format change to follow blackdetect reporting format" ? (it has one reply that supports the idea but iam not sure if theres any script breaking issue and also the log output would be delayed for silence start IIUC) [23:16] <ubitux> mmh not really [23:16] <ubitux> just a moment [23:17] <ubitux> michaelni: yes i'm concerned about the delayed information [23:17] <ubitux> but people should not rely on that output anyway, metadata are here for that [23:17] <ubitux> a micro bump (and an entry in "behaviour changes"? :)) would be welcome i guess [23:24] <cone-612> ffmpeg.git 03James Almer 07master:d43c303038e9: x86/hevc_deblock: use constants instead of generating values at runtime [23:25] <michaelni> ubitux, ill leave the patch to you to reject/comment/apply, have uneasy feeling about it :) [23:26] <ubitux> sure, i'm writing an answer [23:31] <iive> i'd say this patch definitely needs proper benchmark. [23:59] <michaelni> iive, 172248 ->171248 dezicycles for master:d43c303038e9 [23:59] <iive> so the patch is faster? [00:00] --- Tue May 20 2014

1 0

ffmpeg.log.20140519
by burek 20 May '14

20 May '14

[00:01] <c_14> You're using libav, not ffmpeg. Either see #libav for support or install/download ffmpeg. [00:15] <waressearcher2> JEEB: sup [00:19] <waressearcher2> blippyp: sup [01:46] <GEEGEEGEE> Hi im trying to switch from VLC streaming x264 ts to ffmpeg streaming vp80 webm but I cant find any guides on where to start [01:46] <GEEGEEGEE> i currently run VLC to transcode the stream, and also VLC on my servers [01:53] <socksandsandals> how can I create an AVFrame in YUV420P format that's all black? [01:53] <socksandsandals> should I set all the data bytes to 0? [01:54] <llogan> GEEGEEGEE: http://trac.ffmpeg.org/wiki/vpxEncodingGuide [01:55] <GEEGEEGEE> llogan, but what about the output? Can ffmpeg output onto a network port like VLC? or will i need to configure ffserver/icecast first? [01:55] <GEEGEEGEE> also my input is from a HTTP link [01:56] <llogan> see ffmpeg -protocols [01:56] <llogan> and "man ffmpeg-protocols" [01:56] <llogan> or http://ffmpeg.org/ffmpeg-protocols.html if using recent ffmpeg [01:58] <GEEGEEGEE> ok thanks [02:26] <derric> I am trying to compile ffmpeg with a number of external libraries, and even though I get the libraries sucessfully built, ffmpeg seems to lack the ability to find them at linking time. [02:27] <c_14> can you paste the error and your configure command/output [02:28] <derric> ./configure --prefix=$INSTALL_PATH --enable-gpl --enable-version3 --enable-nonfree \ [02:28] <derric> 20:28 --extra-ldflags="-L$INSTALL_PATH/lib" \ [02:28] <derric> 20:28 --extra-cflags="-I$INSTALL_PATH/include" \ [02:28] <derric> 20:28 --enable-libmp3lame --enable-libtwolame --enable-libvpx --enable-libx264 --enable-avisynth \ [02:28] <derric> 20:28 --enable-libvpx --enable-libilbc --enable-libopus --enable-frei0r --enable-libxvid \ [02:28] <derric> 20:28 --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libwavpack --enable-libxavs \ [02:28] <derric> 20:28 --enable-x11grab --enable-libtwolame --enable-fontconfig --enable-gnutls --enable-ladspa \ [02:28] <c_14> eh, please use a pastebin service so you don't spam the channel [02:29] <derric> 20:28 --enable-libfreetype --enable-opengl --enable-libpulse --enable-libzvbi --enable-libaacplus \ [02:29] <derric> 20:28 --enable-libbs2b [02:29] <derric> output is ERROR: libbs2b not found [02:30] <derric> The output is similar when I try a number of other libs that I built and installed. [02:31] <derric> OK. [02:33] <c_14> Is libbs2b installed correctly in $INSTALL_PATH/{lib,include} ? [02:34] <derric> http://paste.debian.net/100522/ [02:34] <iive> derric: there is config.log that holds the tests and the errors (if any). try to find the relevant (bs2b) test and put it in pastebin site. [02:34] <derric> c_14: I believe so. I built and installed all the rest of the enabled items the same way and they get detected OK. [02:35] <derric> They all appear to have the proper files in those directories. [02:35] <iive> c_14: it would be good to always use the phrase "pastebin site" saves a lot of headache. [02:35] <derric> Even the ones that fail to get detected. [02:35] <c_14> iive: I think I'll do that from now on. [02:36] <c_14> derric: Have you tried with --extra-libs="-ldl" [02:37] <derric> About 25% of the libs I built fail to get detected by ffmpeg at compile time. I could make a list of which one work and which ones fail. [02:37] <derric> c_14: What with just --extra-libs="-ldl" verbatim? [02:38] <c_14> Adding that to your configure options [02:38] <derric> OK. [02:41] <GEEGEEGEE> How come libvpx looks so much worse than x264 at the same bitrate? [02:45] <c_14> Some codecs look better at certain bitrates compared to other codecs, it also depends a lot on the source and your encoding options though. I tend to get relatively decent quality on the vpx encodes I use using the same or a slightly lower bitrate. [02:45] <c_14> Are you just setting bitrate or bitrate and crf? [02:46] <GEEGEEGEE> im using -maxrate [02:46] <derric> c_14: Adding that line gets rid of all the extra libs that do get detected. :| [02:46] <c_14> GEEGEEGEE: That'll only set the maximum rate, the actual rate you're getting is probably much lower. [02:46] <c_14> derric: So it breaks more stuff? [02:46] <derric> Yep. [02:46] <GEEGEEGEE> so I should use -minrate as well? [02:48] <c_14> GEEGEEGEE: I'd go with -crf and -b:v like the wiki says: https://trac.ffmpeg.org/wiki/vpxEncodingGuide It's what I've been doing and it's treating me pretty well so far. [02:48] <GEEGEEGEE> ok thanks [02:48] <c_14> derric: did you set and export PKG_CONFIG_PATH ? [02:50] <derric> c_14: What should that point to? [02:51] <c_14> In your case $INSTALL_PATH/lib/pkgconfig [02:51] <derric> c_14: I'll try that. [02:56] <derric> c_14: That seems to have done the trick for libbs2b. Now let me see if the rest of the troublesome libs get detected now... [03:14] <waressearcher2> yo [03:14] <derric> c_14: libopenjpeg is still refusing to get detected, although I think that is problem with its installtion script, which seems to be doing an incomplete job. All of the other previously non-detected libs I have checked so far seem to detected now. I think the PKG_CONFIG_PATH is what I was missing. Ty. [03:15] <c_14> glad it helped [03:25] <GEEGEEGEE> When configuring ffserver why does it need to know the framerate, video size, and other settings? I dont want to transcode it on the server, just to distribute it to my clients [03:25] <c_14> ffserver sends that information to the ffmpeg process that's pushing content [03:25] <c_14> ffserver doesn't actually handle the encoding [03:26] <GEEGEEGEE> oh good [03:50] <GEEGEEGEE> hmm Im getting "av_interleaved_write_frame(): Connection reset by peer [03:50] <GEEGEEGEE> " and then ffmpeg closes [03:50] <GEEGEEGEE> ffserver shows the POST request though [04:33] <waressearcher2> can I ask about sox here ? [04:33] <sacarasc> Try their mailing list. [04:35] <waressearcher2> no [04:35] <waressearcher2> too long [07:54] <aarobc> I'm using a crf of 19 and -preset slower, but some output files are actually larger than what was put in! [07:55] <pzich> so they were poorer quality than 19 [07:57] <aarobc> pzich: so I thought that was the ammount of detail from the source. is that something that determines a constant bit rate then? [07:57] <pzich> well it controls how closely it matches to the source, 1 is near lossless, but at quite the file size [07:58] <aarobc> but as long as the source is a less efficient codec, why wouldn't it always be a smaller file size? [07:59] <pzich> because it's matching that quality as closely as it can, it doesn't know what are artifacts and what's desired quality [07:59] <aarobc> Indeed... [08:00] <pzich> if it's not something you're batching I'd recommend generating a various qualities and picking the smallest one that you're still happy with the compression in [08:01] <aarobc> Well, there are about 500 files, so it's set up to run in a script, but I want to make sure it's as good as possible because I don't want to have to run this again [08:02] <pzich> then I'd probably go for somewhere between 10 and 20, if you have the space [08:02] <pzich> here's an example with some 4K footage I worked on recently: http://i.imgur.com/1lJVcz0.jpg [08:02] <pzich> it's also crunched quite a bit from imgur, but it gets the point across [08:03] <aarobc> Indeed, that is a good example [08:04] <pzich> with a 330 MB 4K ProRes 709 goes between 239 MB and 164 KB, depending on the quality [08:04] <pzich> (we ended up going with something between 15 and 20) [08:05] <aarobc> pzich: so if the source is shitty it will just balloon the output from setting it low? [08:06] <pzich> every time you re-encode, if you're not encoding losslessly, you're lowering the quality [08:06] <pzich> but sometimes the parts that are lost are unimportant, or worth it because of file size [08:07] <pzich> if you want the obtain the best quality and file size without going through each video one by one, I'd recommend bucketing them by file type and resolution, and then dial in the quality for each of those buckets [08:07] <pzich> or do that for just one video and run that for all of them [08:10] <aarobc> yeah, that's kinda what I'm doing. [08:11] <pzich> so CRF 19 is bigger than the input file sizes, I'd look over the quality and then try to go smaller, and if you're happy with that keep going until you're not, etc., etc. [08:11] <pzich> or you can go the other direction since lower quality encodes are faster [08:12] <aarobc> pzich: do you use multi-threaded encoding or does that hurt the quality too much? [08:12] <pzich> I have 8 cores on this machine so I usually go with 8 threads, I haven't heard anything about that affecting the quality, but maybe someone else in here knows more about that. [08:14] <aarobc> yeah, I have 6 cores on the server, but encoding is pretty slow, so I'm right now looking at setting something up to pull the file to my local machine, encode, and upload the output. [08:14] <aarobc> google fiber is yummy [08:15] <pzich> yeah if you're local machine is faster and the connection is good, that's not a bad way to go [08:15] <pzich> err, your* [08:16] <pzich> cbreak-work: Disney Research Zurich? [08:16] <cbreak-work> hmm... yes? :) [08:16] <pzich> I was just trying to figure out your hostname [08:17] <cbreak-work> neat, isn't it? I picked it myself :) [08:17] <cbreak-work> (We are encouraged to use disney chars as names of our computers) [08:17] <cbreak-work> unfortunately, hal was already taken by a coworker... [08:18] <pzich> aww, that's a good one [08:18] <aarobc> Is GPU accellerated encoding a thing yet? [08:20] <pzich> I'm pretty sure it is, otherwise I'm not sure why we just put in three new GPUs on this encoder box [08:22] <aarobc> pzich: I haven't found any documenation for gpu accelerated encoding... does that only work on windows? [08:25] <pzich> hmm, there's https://forums.adobe.com/message/5478209 and some devices had hardware GPU decoding, which ffmpeg might take advantage of (I don't know though) [08:27] <aarobc> for shame. I'm running amd [08:27] <pzich> This thing says they use AMD's OpenCL, but I don't know anything about it: http://www.mainconcept.com/products/sdks/gpu-acceleration/#c14667 [08:28] <aarobc> Yeah I'm looking at that right now, but I hope that doesn't mean I have to install their proprietary drivers. [08:39] <aarobc> pzich: so you actually use ffmpeg for your video transcoding, and it's gpu accellerated? Are there any flags to set? [08:44] <pzich> aarobc: I can't confirm if or how well ffmpeg is utilizing the GPU. [08:49] <pzich> it sure looks to be using it while I'm encoding though [08:51] <aarobc> Going to try it in windoze then [09:00] <pzich> if someone sees aarobc tell him it doesn't look like the GPU really is getting used that much more, this is during and after encode: http://i.imgur.com/bCWAwrM.png [09:24] <excalibr> Hi! Is there way to hide those enabled compile/build msg when you run ffmpeg? [09:25] <excalibr> by msg i mean options [09:28] <excalibr> Never mind got it. -hide_banner is what I want [09:57] <decci_> Hi [09:57] <decci_> I am getting this error http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [09:58] <decci_> http://paste.ubuntu.com/7487380/ [09:59] <decci_> Any idea why its accessible through shell and not through web [10:01] <spaam> check PATH [10:02] <decci_> spaam: echo $PATH /usr/local/jdk/bin:/usr/lib64/qt-3.3/bin:/bin:/usr/bin:/usr/local/bin:/sbin:/usr/sbin:/usr/local/sbin:/usr/local/bin:/usr/X11R6/bin:/root/bin [10:02] <spaam> the web have the same thing ? [10:05] <decci_> spaam: http://198.20.89.58/~upeoapp2/ffmpegtest/phpinfo.php [10:05] <decci_> spaam: _SERVER["PATH"] /bin:/usr/bin [10:05] <spaam> and where did you install ffmpeg + [10:05] <spaam> ? [10:06] <spaam> in /usr/local/bin? [10:06] <decci_> spaam: yes [10:06] <spaam> there you have the problem [10:06] <decci_> spaam: /usr/bin/ffmpeg FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) configuration: --pre [10:06] <decci_> spaam: Just few lines above pasted [10:06] <spaam> you need to add /usr/local/bin to the PATH that PHP is using [10:07] <spaam> what? [10:07] <decci_> Where it says it using /usr/local/bin [10:07] <spaam> you said /usr/local/bin now /usr/bin ? [10:07] <decci_> spaam: its /usr/bin/ffmpeg [10:07] <decci_> spaam: Sorry if that confused you [10:08] <spaam> have you tried start /usr/bin/ffmpeg from PHP instead of using whereis things ? [10:08] <decci_> spaam: Looks like somethig to do with error: Cannot execute using backquotes in Safe Mode in /home/upeoapp2/public_html/ffmpegtest/test.php on line 4 [10:08] <spaam> there you have your problem [10:08] <decci_> spaam: The first thing is no command is working http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [10:09] <decci_> spaam: Looks like web host compan has everything in chroot [10:09] <spaam> this is an issue for ##php and not #ffmpeg [10:09] <decci_> spaam: But I was expecting atleast ls should have work [10:09] <spaam> read the error message [10:10] <spaam> talk to ##php on how to fix the issue. [10:10] <decci_> spaam: I dint understand that error message...what it actually mean [10:10] <decci_> spaam: #php Cannot join channel (+i) - you must be invited [10:10] <decci_> spaam: My bad [10:10] <spaam> ##php [10:11] <spaam> it cant execute pwd beacuse you are running php in safe mode. [10:11] <spaam> talk to those php guys on how to fix it. [10:12] <spaam> when you can start programs from php and still have issues with decoding / encoding videos with ffmpeg. feel free to come back and tell us about that issue. [15:26] <waressearcher2> is it possible to mix 10 mp3 files ? I use "sox -m 1.mp3 2.mp3 10.mp3 out.mp3", but is it possuble with ffmpeg ? [15:27] <c_14> What do you mean with mix? [15:28] <waressearcher2> c_14: to mix them I have different music in each of them and I want one mp3 to contain all of them [15:28] <c_14> Du you want to concatenate the files or do you want to have one file with multiple audio streams? [15:29] <c_14> s/Du/Do [15:29] <waressearcher2> one stream [15:29] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [15:29] <c_14> It'll work with music files as well. [15:29] <waressearcher2> they are all 15 minutes audio, I want one 15 audio with all audio from all of them [15:30] <c_14> If they're all the same codec you can use the concat demuxer, if they're different you'd have to use the filter. [15:31] <waressearcher2> they are all 320Kbit mp3 files 44100Hz [15:32] <c_14> Then you can probably just use the concat demuxer. [15:32] <waressearcher2> c_14: dude [15:33] <waressearcher2> c_14: concat is when begining of the second file starts after first file ends right ? I don't want to out.mp3 be 150 minutes long, I want it to be 15 minutes, as all 10 files are [15:33] <waressearcher2> I want "to mix" [15:34] <c_14> Look at the amix filter. [15:34] <c_14> That should do what you want. [15:34] <waressearcher2> I know how to concatenate, I want to "mix" or "merge" or whatever you call it [15:34] <c_14> https://ffmpeg.org/ffmpeg-filters.html#amix [15:36] <waressearcher2> dawgthanks [15:37] <waressearcher2> after 10 minutes of explanation I got the info [15:37] <waressearcher2> how much simpler world was before computers [15:45] <waressearcher2> I have many ace files, but I need to use "-i" in front of each one like: "-i 2_2.ac3 -i 2_3_2.ac3 -i 2_5_2.ac3", can it be done in a easier way, like "-i *_2.ac3" ? [15:46] <waressearcher2> s/ace/ac3/ [15:47] <mrskman> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [15:49] <waressearcher2> mrskman: where did you get that link ? [15:49] <waressearcher2> it does smell a bit [15:50] <waressearcher2> or do you suggesting to use "-i list_all.txt" where list of all files is in list_all.txt ? [15:50] <klaxa> waressearcher2: afaik there is no easier way unless you use something like ffmpeg $(for i in *_2.ac3; do echo -n "-i $i") <something> out.ac3 [15:50] <klaxa> hmm or something like that [17:29] <DelphiWorld> hi guys [17:29] <DelphiWorld> do ffmpeg exist in debian sid? its not installing... [17:30] <azk> Doesn't look like it [17:30] <c_14> the debian repos use libav by default [17:30] <DelphiWorld> no, i think i need the contrib one? [17:30] <c_14> If you want to use ffmpeg, you'll have to compile from source or use the static builds. [17:30] <c_14> Hmm? [17:31] <DelphiWorld> deb http://ftp.uk.debian.org/debian/ sid contrib-nonfree [17:31] <DelphiWorld> i think [17:31] <DelphiWorld> but its bad url [17:35] <c_14> I'm pretty sure debian doesn't have ffmpeg in its repos. ffmpeg is in debianmultimedia though if you want to try that. [17:36] <DelphiWorld> c_14: repo url please [17:44] <c_14> http://www.deb-multimedia.org/ [17:45] <c_14> The list of repos is at the waaay bottom [22:20] <jpsharp> I'm getting the dreaded Alsa buffer XRUN when trying to encode audio/video from a capture card using X264/AAC into an FLV container streaming to ngninx. But only when using X264. [22:21] <jpsharp> Sorenson works but it thrashes the hell out of the CPU. Mpeg4 seems to encode nothing but a black screen for video. [22:24] <pzich> I know nothing about this particular issue, but according to the ticket you can try increasing ALSA_BUFFER_SIZE_MAX http://trac.ffmpeg.org/ticket/615#comment:14 [22:28] <jpsharp> I'm not really in the mood to recompile alsa on a dozen boxes. [22:34] <pzich> well, it sounds like your options are to reduce the quality or size so that the buffer size is ample, or increase the buffer size [00:00] --- Tue May 20 2014

1 0

ffmpeg-devel.log.20140518
by burek 19 May '14

19 May '14

[00:12] <kurosu> one reason could be the differing calling conventions, as I'm assuming the 5th arg is passed through the stack [00:13] <kurosu> jamrial, might not be the case for x86_64, so an extra reg is needed there [00:13] <kurosu> worth than that [00:13] <kurosu> *worse [00:14] <kurosu> "System V AMD64" passes the 6 first args in regs [00:14] <kurosu> so I'm effectively overwriting them, and you can't save that many args in that case [00:15] <kurosu> on the other hand, one does not have to handle win32/x86_32 so... [00:46] <BBB> on win64, first 4 args are regs [00:47] <BBB> on unix64 (including mac), it's 6 [00:47] <BBB> on 32bit, everything is stack [00:48] <jamrial> kurosu: well, you effectively brought the function down to less than 8 xmm/gp regs, so it can be used on x86 builds as well now [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:0323a55cd898: avformat/rtpenc: switch to AVFMT_FLAG_BITEXACT [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:d524d424ec98: avformat/nutenc: switch to AVFMT_FLAG_BITEXACT [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:fab50573bb92: avformat/mp3enc: switch to AVFMT_FLAG_BITEXACT [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:0421409c5216: avformat/matroskaenc: switch to AVFMT_FLAG_BITEXACT [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:9646ea63df78: avformat/mmf: switch to AVFMT_FLAG_BITEXACT [04:41] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:17c8a069167e: avformat/framehash: switch to AVFMT_FLAG_BITEXACT [04:42] Action: michaelni hugs cone-266 [04:58] <Zeranoe> Is this a known bug? http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=1981 [09:20] <mraulet> kurosu, jamrial, https://ffmpeg.org/pipermail/ffmpeg-devel/2014-May/157734.html works for me [09:30] <kurosu> jamrial, yes, but you'd have to disentangle (?) the assignment I think [09:30] <kurosu> not really worth it because this function is seldom used (fades when the proper tool is on) [09:31] <kurosu> mraulet, ok, couldn't verify, so glad to know [09:35] <kurosu> anyway, to make it 32bits, the test must be changes to %if WIN64 || ARCH_X86_32 [09:35] <kurosu> don't know if there's a shorthand nowadays [09:59] <kurosu> pmulhw m6, m0, m2 [09:59] <kurosu> pmullw m0, m2 [09:59] <kurosu> punpckhwd m1, m0, m6 [09:59] <kurosu> punpcklwd m0, m6 [09:59] <kurosu> paddd m0, m5 [09:59] <kurosu> paddd m1, m5 [10:00] <kurosu> isn't that just pmaddwd but with samples not interleaved ? [10:00] <kurosu> (ignore the paddd actually) [10:04] <Skyler_> is that trying to work around a lack of pmulld or something? [10:09] <kurosu> Skyler_, ah right [10:09] <kurosu> probably [10:09] <kurosu> the funny thing is that the function is sse4 [10:10] <kurosu> this code really feels like it has been made for simplest reusability over thoroughness [10:11] <kurosu> eg loading coeffs into specified regs, and too bad if some of the previous regs aren't used [10:13] <Skyler_> pmaddwd is probably what you want there though [10:13] <Skyler_> since it's 16x16->32, not 32x32->32 [10:14] <kurosu> well, I'm firstly trying to reduce the reg usage, removing the aforementioned holes [10:15] <kurosu> not sure I'll get all the way here, because there's quite some work to do this for every function [10:15] <kurosu> *all the way there [10:42] <kurosu> michaelni / ...: please hold on, I'm squashing the next weighted pred reg realloc patch into the current one [11:46] <j-b> 'morning [12:00] <kurosu> oh wow, my macro loads coeffs into m14,m15 - core loops uses m0-m4 and coeffs - let's declare using 16 regs! [12:03] <J_Darnley> You should use them if you have them! [12:06] <kurosu> I fear that plepere develops under linux and never sees the dark side of such a code :( [12:59] <BBB> don't you want pmaddubsw? [13:00] <BBB> I mean, for 8bit at least [13:00] <cone-397> ffmpeg.git 03Anton Khirnov 07master:e1b66778b6ee: lavc: remove the locking code in avcodec_close() [13:00] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:ff17d8b56ec8: Merge commit 'e1b66778b6ee82a192b5895e23c4e135f7269326' [13:09] <cone-397> ffmpeg.git 03Anton Khirnov 07master:fd056029f45a: lavc: add avcodec_free_context(). [13:09] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:68c05185e229: Merge commit 'fd056029f45a9f6d213d9fce8165632042511d4f' [13:15] <cone-397> ffmpeg.git 03Anton Khirnov 07master:d256ed78ffe2: 4xm: allocate extradata properly. [13:15] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:6a510909242a: Merge commit 'd256ed78ffe202a4dcc8d625becffc716bfa3977' [13:25] <cone-397> ffmpeg.git 03Anton Khirnov 07master:b136564909ad: a64: remove unneeded struct qualifier [13:25] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:c7c11058abb7: Merge commit 'b136564909adec2043be4c94ade08b67ad443124' [13:31] <cone-397> ffmpeg.git 03Anton Khirnov 07master:3ef6c5264b25: a64: check that extradata exists before reading from it [13:31] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:47a721348a5f: Merge commit '3ef6c5264b2590781b4ed556443ff49709dd45fb' [13:42] <cone-397> ffmpeg.git 03Anton Khirnov 07master:e4dc1000d7bb: yuv4mpeg: split the demuxer and muxer into separate files [13:42] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:56d3cd1455ea: Merge commit 'e4dc1000d7bbbcb5b45cf9849fc5315f19578e37' [13:54] <J_Darnley> I wonder what benefit that ^^ is supposed to bring. I didn't think y4m was so complicated that you needed to split [14:08] <BBB> J_Darnley: smaller binary for demuxed-only builds, likely [14:08] <cone-397> ffmpeg.git 03Anton Khirnov 07master:b513bf6f69e2: yuv4mpegdec: do not set coded_frame properties [14:08] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:213e82b55c28: Merge commit 'b513bf6f69e26e724de6d5dca642c3582dcd0517' [14:11] <cone-397> ffmpeg.git 03Anton Khirnov 07master:ab3fdaa7131e: yop: only set extradata size after it has been successfully allocated [14:11] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:4fac4447d153: Merge commit 'ab3fdaa7131e2fd8e33006daea25a282724c01f0' [14:19] <cone-397> ffmpeg.git 03Anton Khirnov 07master:c3311d472a75: avienc: sanitize variable naming in write_header() [14:19] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:4f32efdff007: Merge commit 'c3311d472a7528c67f76d0d061704ae70a99b32e' [14:28] <cone-397> ffmpeg.git 03Anton Khirnov 07master:7a5f4f685304: lavf: extend avg_frame_rate documentation. [14:28] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:8636b11bdc6e: Merge commit '7a5f4f6853047eea74a6443d73f4d74d52b2799f' [14:51] <cone-397> ffmpeg.git 03Anton Khirnov 07master:93afb6c98df8: avconv: set output avg_frame_rate when known [14:51] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:8334850a9985: Merge commit '93afb6c98df876b15e3d911a9450ad55f92080ce' [14:58] <cone-397> ffmpeg.git 03Anton Khirnov 07master:79f2c426fde6: dv: do not set codec timebase [14:58] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:b06741d26794: Merge commit '79f2c426fde6e71c40b29504112d0528b85be623' [16:05] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:bf718bdf12be: avcodec/crystalhd: allocate padding for extradata [16:05] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:947ed8f292f5: avcodec/ffv1enc: allocate padding for extradata [16:05] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:5441eb460cb4: avcodec/movtextenc: allocate padding for extradata [16:05] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:6e8fe448154e: avcodec/wmv2enc: allocate padding for extradata and check malloc failure [16:15] <cone-397> ffmpeg.git 03Olivier Langlois 07master:0a150670476b: lavd: Use av_gettime_relative() [16:30] <cone-397> ffmpeg.git 03Christophe Gisquet 07master:f1793fe9cdf1: x86: hevc_mc: specify coefficients registers [16:55] <kurosu> BBB: some intermediate computations are specified to be on 32 bits [17:37] <BBB> kurosu: but do they use the upper 16 bits? [17:38] <kurosu> well, the input is on 16bits, but I haven't checked what the specs says on that [17:38] <kurosu> plus I don't really intend to [17:39] <kurosu> plus, this hevc thing is a hobby, it's not like my regular job is to implement hevc ;) [17:40] <BBB> are you sure about that :-p [17:40] <BBB> oh, right, implement [17:40] <BBB> hm... [17:40] <BBB> ok [18:25] <ubitux> Daemon404: so, no comment on writing_filters.txt, i thought you were complaining about that a while ago [18:25] <ubitux> ? [19:08] <Daemon404> ubitux, i have not read it [19:08] <Daemon404> just got home from the beach [19:08] <Daemon404> making dinner, so later. [20:14] <kierank> vps_max_num_reorder_pics out of range: 7 [20:14] <kierank> for hevc [20:25] <nevcairiel> the check in the code appears correct as the spec describes it [20:25] <cone-397> ffmpeg.git 03Anshul 07master:73a60633143b: ffprobe: fix crash happening because of new streams occuring [21:03] <cone-397> ffmpeg.git 03Anton Khirnov 07master:33082af7560a: doc/APIchanges: fill in missing hashes and dates [21:03] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:6eefc578a5c4: Merge commit '33082af7560a18179b52734665ef66f812902d33' [21:05] <cone-397> ffmpeg.git 03Anton Khirnov 07master:968a62a9d9e3: doc/APIchanges: mark the release 10 branch point [21:05] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:a6ba5184a5e2: Merge commit '968a62a9d9e3a785ce1f79b80facfe09c6c14544' [21:11] <cone-397> ffmpeg.git 03Anton Khirnov 07master:d246231e4714: wavenc: use codec descriptors to get the codec name [21:11] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:c01bcce4959a: Merge commit 'd246231e4714119faac6c7acd881d3b687bb8b11' [22:21] <Daemon404> ^ because lags for ffmpeg had to include an fp emulator to be portable [22:22] <kierank> EBROKENBYDESIGN [22:23] <cone-397> ffmpeg.git 03Anton Khirnov 07master:6656370b8583: avconv: set the "encoder" tag when transcoding [22:23] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:1e49439f04f4: Merge commit '6656370b858329ca07a60a2de954d5e90daa0206' [22:23] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:bbee02ed9c18: ffmpeg: Skip writing the version when -flags bitexact is used [23:22] <cone-397> ffmpeg.git 03Anton Khirnov 07master:0ba5299a805e: movenc: use the "encoder" metadata tag to write stsd Compressorname [23:22] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:5771efde87c1: Merge commit '0ba5299a805e9ccaef1a757381fc2ada4d54b8a1' [23:22] <cone-397> ffmpeg.git 03Michael Niedermayer 07master:0c152fe916e3: ffmpeg: prefix encoder with "Lavc " in bitexact mode [23:48] <cone-397> ffmpeg.git 03James Almer 07master:057ebf122276: x86/hevc_deblock: remove some duplicated instructions [00:00] --- Mon May 19 2014

1 0

ffmpeg.log.20140518
by burek 19 May '14

19 May '14

[00:21] <zybi1> hi [00:21] <zybi1> is there a command to merge audio to a (soundless) video without? [00:22] <c_14> ffmpeg -i sound -i video -codec copy outfile [00:22] <zybi1> I got a mov-video and a wav-audio, converting the audio to any working format (i.e. aac I guess) is fast [00:23] <zybi1> So which audio codec should I use for the following video type: MPEG-4 QuickTime Apple QuickTime Format AVC [00:23] <zybi1> is wav working with that? [00:24] <zybi1> can ffmpeg deal with that format well? [00:25] <iive> avc is probably advanced video coding, aka h264 [00:25] <zybi1> yes [00:25] <zybi1> Advanced Video Codec [00:27] <iive> it should work with mp2/3, ac3, dca, aac [00:27] <c_14> alac too [00:27] <iive> probably... [00:27] <iive> do we have alac encoder? [00:28] <c_14> ye, think so [00:28] <iive> alac was lossless, afair. [00:28] <c_14> it's in ffmpeg -encoders anyway [00:28] <zybi1> Unrecognized option 'codec' [00:29] <c_14> try -c:v copy -c:a $audio-codec [00:29] <zybi1> is -codec = -h264enc [00:29] <zybi1> ? [00:29] <iive> there is no h264enc in ffmpeg, we use libx264 like everybody else. [00:35] <zybi1> thanks [00:36] <zybi1> sorry: Failed to set value 'copy' for option 'libx264' [00:36] <c_14> paste your commandline [00:36] <zybi1> ffmpeg -i SOUND -i VIDEO -libx264 copy OUTPUT [00:37] <c_14> eeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh, what are you trying to do? [00:37] <c_14> Do you want to convert the video or the audio or just copy them? [00:39] <zybi1> convert the video by adding the audio [00:40] <c_14> just try ffmpeg -i sound -i video -c:v copy -c:a aac output [00:41] <c_14> If I'm understanding you correctly, you just want to mux the two streams together into an mp4 without actually reencoding the video, but since mp4 doesn't support wav (afaik), you'll have to encode the audio stream [00:43] <c_14> Or since you're using wav input, if you want the audio stream to remain lossless, use alac [00:44] <c_14> assuming the wav is actually lossless [00:56] <zybi1> Failed to set value 'copy' for option 'c:v' [00:56] <zybi1> if wav or aac is now not too important to me [00:56] <zybi1> sorry c_14 [00:57] <zybi1> http://pastebin.com/nhZe9jKE [00:58] <c_14> you're using libav, not ffmpeg. can you download one of the static binaries and try it with that? [01:04] <zybi1> success! [01:04] <zybi1> http://pastebin.com/iPL2PAY8 [01:04] <zybi1> but alac produced no audio [01:05] <zybi1> only copy audio with an existing ac3-file worked perfectly [01:06] <zybi1> thanks a lot!! [01:06] <c_14> np [04:01] <wlritchi> Say I'm splitting a video file with ffmpeg -acodec copy -vcodec copy. I've noticed that this produces weird artifacts around the start and end. [04:01] <wlritchi> I'm assuming this is because of block sizes of the audio codec, and non-keyframes in the video [04:02] <wlritchi> Purely for reasons of "because I can", I want to preserve as much quality as possible (ie no transcoding to lossy formats when not strictly necessary) [04:03] <wlritchi> I'm okay with transcoding the audio to FLAC because it's still a small enough part of the video file, but is there any way to keep -vcodec copy without these artifacts at the start? [04:04] <wlritchi> (example solution: re-encode the first frame only) [04:10] <c_14> You shouldn't be getting weird artifacts, ffmpeg should seek to the nearest keyframe and cut from there when used with -codec copy [04:21] <wlritchi> c_14: The problem is that the audio doesn't get seeked to match [04:22] <wlritchi> So I end up with video that (in most players, including ffplay) plays audio alongside timescaled video for the first few seconds [04:22] <wlritchi> Not sure whether the timescaling is done in encoding or decoding [04:23] <wlritchi> In any case, if possible, I'd like the split to not be seeked forward [08:15] <hendry> http://video.stackexchange.com/questions/10772/converting-intervalometer-ph… [08:51] <waressearcher2> anyone ? [08:51] <waressearcher2> hi [08:55] <waressearcher2> I have 15 minutes video with 1500k bitrate, and its resolution is 600x337, and the size is 161615KB, I converted it to video with smaller resolution 500x281 and now the size is 161463KB, so its quite the same size, why ? I mean, size of the picture decreases 30% but size of the file decreases just 0.01%, why ? [09:09] <wlritchi> waressearcher2: What other encoding options did you specify? [09:10] <wlritchi> If one of those was 1500k bitrate, I found your problem... [09:11] <wlritchi> (file size is a function of bitrate) [09:38] <waressearcher2> wlritchi: for both files I used that commands: "ffmpeg -i 1.avi -vcodec mpeg4 -vb 1500k -s 600x337 -y -f avi 2.avi" and "ffmpeg -i 1.avi -vcodec mpeg4 -vb 1500k -s 500x281 -y -f avi 3.avi", now 2.avi is 161615KB in size and 3.avi is 161463KB in size, they practically same size but resolution second one is 30% less, why ? [09:39] <waressearcher2> shouldn't there be difference in size of about 50MB ? [09:43] <blippyp> it's just like he said, it's because you didn't change the video bitrate, you got a smaller resolution out of the second video, but it did a higher quality encoding because of it (less artifacts). [10:01] <waressearcher2> sort of understood [10:02] <waressearcher2> so the encoder just used more bytes of data to encode ? [10:02] <blippyp> basically... [10:02] <blippyp> it's like takinga blu-ray [10:02] <waressearcher2> even if resolution is smaller there is more bytes per pixel encoded ? [10:02] <blippyp> then encoding it at say 60% quality (same resolution) [10:02] <blippyp> it's going to degrade the video in order to encode that [10:03] <blippyp> now resize the video - say to 80% of the original size [10:03] <blippyp> and encode it with the same settings, it's still going to degrade the video, but it will do a much better job at it... [10:03] <blippyp> both files end in the same size [10:03] <blippyp> this is what happened to you - because you kept the video bitrate the same [10:04] <blippyp> it probably wants it higher to do a perfect encoding, so it's keeping the size the same [10:05] <blippyp> long story short, to get a smaller filesize you will need to lower the bitrate, but your video quality will suffer for it [10:59] <smalltock> Hello, am ffmpeg noob. No matter what I do, I can't get ffmpeg to capture x11grab with the libvpx codec at a high framerate. It usually sticks around 5-6 FPS. If I specify a higher rate, it still captures at 5-6 FPS and the video file plays through them fast. My latest attempt was with the following command& [10:59] <smalltock> ffmpeg -f x11grab -r 16 -s 1920x1080 -i :0.0+0,0 -q:v 1 -b:v 3M -c:v libvpx -an /tmp/nope.webm [11:00] <smalltock> Am I doing something wrong or expecting too much performance or what? I appreciate all help. [11:01] <waressearcher2> smalltock: try "ffmpeg -r 30 -s 1920x1080 -f video4linux2 -i /dev/video0 output.flv" [11:02] <smalltock> No /dev/video0 [11:02] <waressearcher2> then you f**ed [11:02] <waressearcher2> go create one, search google for how-to [11:02] <smalltock> :3 Alrighty thank you [11:14] <smalltock> waressearcher2: What is the video0 device supposed to do? According to what google search has brought up, it's supposed to be a camera. Is X supposed to be recognized as a camera? Apologies for the silly question [11:15] <Mavrik> waressearcher2, uh, what are you talking about [11:15] <Mavrik> you don't have video0 for screen grabbing [11:15] <Mavrik> smalltock, the performance depends on how fast your computer is and how you grab it [11:15] <Mavrik> smalltock, in your case you're telling ffmpeg to "think" that desktop input is recorded at 16fps [11:15] <Mavrik> if you want it smoother set it to 30 but you will need enough CPU horsepower to encode it [11:16] <waressearcher2> smalltock: you can always turn your camera on you desktop and record it, no pun intended [11:16] <Mavrik> smalltock, also, I suggest you try with a fast x264 preset for starters to see if it's encoding CPU that's the limiting factor for you (e.g. -c:v libx264 -crf 22 -preset ultrafast something.mp4) [11:17] <waressearcher2> yes, I mistakenly suggested wront solution, haven't noticed he want's to grab a desktop not webcamera [11:17] <smalltock> I can grab from x11grab at 30 FPS if I specify it, just not when encoding with libvpx. If I use libx264, it's perfect, but I want libvpx output and don't want to encode more than necessary. Is libvpx encoding just that much more intensive? [11:18] <Mavrik> smalltock, yes, libvpx is significantly less optimized than libx264 [11:18] <Mavrik> smalltock, make sure you have the latest update of it and ffmpeg [11:18] <Mavrik> smalltock, there's also "quality" and "cpu-used" parameter [11:19] <smalltock> is cpu-used limited much by default? I have a decent CPU, FX 8350 if I recall correctly. [11:19] <Mavrik> try something like [11:19] <Mavrik> -quality rt -cpu-used 2 [11:19] <Mavrik> and play with it [11:19] <Mavrik> see http://www.webmproject.org/docs/encoder-parameters/ [11:20] <Mavrik> and https://www.virag.si/2012/01/webm-web-video-encoding-tutorial-with-ffmpeg-0… [11:20] <blippyp> wow - good thing there was a life guard on duty... haha [11:20] <smalltock> Alrighty, I'll read through those. Thank you very much! [11:21] <Mavrik> smalltock, if that won't work... you'll just have to dump raw / lossless [11:21] <Mavrik> and encode it later [11:22] <Mavrik> which is usually a better approach anyway, since your system hiccups won't corrupt video [11:23] <blippyp> Ya, I highly recommend that as well - You need a pretty good setup to not go lossless or near lossless x264 will give you some fair encoding while barely slowing you down... It will at least save some space while recording... [11:24] <Mavrik> yeah, I used the ffmpeg lossless format or huffyuv [11:24] <Mavrik> because I noticed that when recording raw the HDD sometimes tends not to follow [11:24] <blippyp> yup, me too [11:24] <Mavrik> and you need at least some compression to keep the I/O bandwidth down [11:25] <blippyp> exactly - it's just a better approach in my opinion as well.... [11:25] <smalltock> That seems to have done it! Getting the specified 16FPS now. Managed 24 a few moments ago when rate was unspecified. Thank you guys for the help! [11:26] <blippyp> my systems are pretty old, and even I can get away with encoding with a crf 17 with a preset of ultrafast [11:36] <ReX> Hi Guys, I am using libmp3lame with ffmpeg to convert video to mp3 using this (ffmpeg -i input -vol %volume% -y -acodec libmp3lame -ab %quality%k output) Is there a way to add an album art to the output? [11:38] <ReX> I tried using the lames's --ti command after libmp3lame but it doesnt work. [11:41] <ReX> could anyone please help me with this? [11:43] <blippyp> ReX, I don't think you can do that with mp3's - but you can probably attach the album art with a different container if you're not picky about that. Something like mkv can have attachments I think, but I've never used it, I could be sending you off in the wrong direction, just trying to give you a bone to chew on... [11:43] <ReX> Thanks blippyp, I can do this by (ffmpeg -i input.mp3 -i cover.jpg -map_metadata 0 -map 0 -map 1 output.mp3) [11:44] <blippyp> cool [11:44] <ReX> but this is not I want. I want to conver video to mp3 and add an album art to the output with 1 command [11:44] <ReX> is it possible? with libmp3lame? [11:44] <blippyp> I doubt it - mp3 is an audio only format as far as I know (thought) [11:45] <blippyp> You'd have to google libmp3lame to find out for sure I guess [11:45] <ReX> yes mp3 is only audio .. but it can have meta tags like album artist, title, genre .. album art etc right? [11:46] <blippyp> Unless someone else who actually knows otherwise comes on - give it time, I don't know what time it is where you are, but it's really early in the morning where I am [11:46] <blippyp> I don't think you have have art - just tags [11:46] <ReX> I tried googling libmp3lame but there is not much info on its uses with ffmpeg, can you put me into right directions? [11:46] <ReX> but this is not I want. I want to conver video to mp3 and add an album art to the output with 1 command [11:46] <ReX> blippyp, how do I use tags then? [11:47] <blippyp> you did it already with the map_metadata command [11:47] <ReX> I mean how to change tages with libmp3lame? [11:47] <blippyp> with the map_metadata command as far as I know??? [11:47] <ReX> the lame options doesnt work :( [11:48] <blippyp> I honestly don't think you can do what you're trying - I'm pretty sure you would need to accept a video container to do this [11:49] <blippyp> but there might be some other audio containers that will do what you want??? [11:49] <ReX> how to use map_metadata after libmp3lame? please give me some documentation if you know of any. [11:49] <blippyp> man ffmpeg-filters [11:49] <blippyp> search for metadata [11:50] <blippyp> or http://www.ffmpeg.org/ffmpeg-filters.html (same as the man page) [11:51] <ReX> Thanks .. trying that [11:51] <blippyp> good luck [11:52] <ReX> No manual entry for ffmpeg-filters :( [11:55] <blippyp> I posted it [11:55] <blippyp> search for metadata [11:57] <mithodin> Hi. [11:57] <ReX> :( [11:58] <ReX> is it possible to do using any other encoder? [11:58] <mithodin> I'm trying to put multiple images over a video with ffmpeg. They each get faded in and then faded out, so effectively it's only one image at a time. Is there a way to concatenate these images and then overlaying only one stream on the main movie? [11:59] <ReX> for example, how do I use lame with ffmpeg? [11:59] <mithodin> currently, I'm doing this: http://bpaste.net/show/285161/ [12:00] <ReX> i am getting unknown encoder when I try to use lame with ffmpeg. I have compiled ffmpeg with libmp3lame. Also I have lame installem [12:00] <ReX> *installed [12:00] <blippyp> mithodin: lookup concat in the man pages [12:02] <mithodin> blippyp: Yes, I'm aware of concat. The question is, when I concat two of my overlay streams, is the timing preserved? [12:03] <blippyp> I don't think so - it's a feature ffmpeg is missing as far as I'm aware... you would probably need to make 'short' clips and then concatenate them seperately afterwards... :( [12:03] <blippyp> If I'm understanding you correctly.... [12:04] <blippyp> but at least it's fairly easy to script [12:04] <mithodin> blippy: So export the first 15 seconds with the first overlay, do the same with the other two and then concat these files? [12:04] <blippyp> that's what i would do - someone else might know of a better way - I'd definately like to hear about it myself... :) [12:05] <mithodin> Okay, will this be seamless? I absolutely need a seamless result [12:06] <blippyp> wait, I'm not thinking clearly - I don't see why you couldn't concatenate this....??? you have specififed fade-in and fade-out times - concat isn't working for you? [12:07] <mithodin> I have not tried it yet... let me test it... [12:07] <blippyp> yeah, I would try it - I think it will work for you [12:09] <mithodin> so the synax is [v0] [v1] concat=n=2 ? [12:09] <blippyp> you should be able to get away with just [0:v][1:v] concat [out1] ..... [12:11] <mithodin> while it's streaming... does ffmpeg make use of multiple cores by default? [12:12] <blippyp> I think it depends on the codec being used, but I could be wrong [12:13] <mithodin> okay, I'v tried it like this: http://bpaste.net/show/285192/ and it doesn't work [12:13] <mithodin> in that I can see the first overlay appearing and disappearing, but not the second [12:14] <blippyp> I'm trying it myself - just give me a couple of minutes [12:17] <mithodin> maybe with select: http://ffmpeg.org/ffmpeg-filters.html#select_002c-aselect [12:24] <Mavrik> mithodin, not setting encoding parameters for video is usually a very horrible idea [12:24] <mithodin> I'm testing right now [12:24] <Mavrik> but yes, ffmpeg will use multiple cores if en/decoder uses that [12:24] <Mavrik> filters usually aren't multithreaded [12:25] <Mavrik> also alot of decoders aren't. [12:25] <mithodin> I see [12:26] <mithodin> select does not seem to work. Enconding just stops at frame 500 [12:27] <mithodin> which is where the select of the first overlay stops [12:27] <blippyp> it's enoding for me now, so I should see it soon [12:28] <blippyp> I'm past frame 500 if that means anything to you though??? [12:29] <mithodin> can you paste your code? [12:29] <blippyp> wait till we actually see the results.... doesn't look like it's working for me - all I get is audio [12:30] <mithodin> ok [12:30] <blippyp> let me look at the filter - explain what you want exactly [12:30] <blippyp> it looks like you have a video [12:30] <blippyp> and you want to overlay images from time to time on it right? [12:30] <mithodin> yes, that is exactly what I'm trying to do [12:31] <blippyp> k, let me play with it again [12:32] <mithodin> the thing is that overlaying all the (faded) images over the stream one by one is not very efficient and gives me a low encoding fps [12:33] <blippyp> that's because you have to re-encode the whole thing, all the while applying filtering to it [12:33] <Mavrik> honestly I think you should use a software made for video production [12:33] <Mavrik> not try to retrofit a video transcoder for production use :) [12:33] <mithodin> the thing is that I have tried using openshot and others and got a/v desyncing every time [12:34] <mithodin> with ffmpeg it works [12:36] <blippyp> actually ffmpeg is great for filtering some times... :) [12:36] <blippyp> now you do have the alpha channels setup on the images right? [12:36] <blippyp> or are you guessing/playing around with that not really understanding what it's doing? [12:38] <mithodin> well, what I'm doing is fading in each image (so from zero visibility to 1) and after a time fade it back out again. Each image is a png with transparency that puts some text or an image over the video [12:38] <Mavrik> blippyp, yeah, great for filtering but terrible for composition [12:38] <mithodin> I'm not sure I get your question [12:39] <blippyp> actually I find it's great for composition if you KNOW what you want beforehand and are willing to setup the scripts for it - then your video editing is a one-click process... ;) [12:41] <blippyp> my experience with alpha channels is that you basically need to convert everything to qrtle (if I remember that correctly) first to get it to work properly.... It's what I've always had to do for alphamerging.... Same issue here... [12:41] <blippyp> the format of the video likely needs to be converted first [12:41] <blippyp> are you getting any video at all with your attempts? [12:42] <mithodin> yes, and it looks fine [12:42] <mithodin> I can give you an example I'm getting with the slow approach [12:42] <blippyp> sure [12:43] <mithodin> (it will take a little moment to upload somewhere) [12:43] <blippyp> no problem [12:48] <mithodin> files.treffenstaedt.de/test.mp4 [12:49] <mithodin> http://files.treffenstaedt.de/test.mp4 [12:52] <blippyp> I'm such a dumb ass - my video was popping up behind my browser window for some reason! [12:52] <mithodin> :-D [12:52] <blippyp> it was working for me too - but like you it dies after 30 seconds [12:53] <blippyp> Mavrik is right though, something like this is done much easier in a video editor... ;) [12:54] <mithodin> can you recommend one (except openshot, because that one creates a/v desyncing over the video)? [12:54] <mithodin> for linux, of course [12:55] <blippyp> Yeah, I hate the video editors in linux, they absolutely disgust me - It's actually why I used ffmpeg for nearly everything now myself. [12:55] <mithodin> and here we are again ;-) [12:55] <blippyp> At least with ffmpeg I have no one else to blame but myself for my stupidity in my scripting... haha :P [12:56] <blippyp> I don't know, your command looks good from what I can tell - you're sure your timing is right in it for the fade-in/outs? [12:57] <mithodin> yeah, pretty sure. [12:58] <blippyp> wait - you're aware that YOU cut off video at 30 seconds right? [12:58] <mithodin> yes. but not at 20 [12:59] <blippyp> the video you sent me is 31 seconds long? [13:00] <mithodin> yeah, that used the other script that exports 60s, I killed it because 31 was enough to show what I want [13:01] <blippyp> ok, so I'm confused then - what's the problem? [13:01] <mithodin> I tried this: http://pastebin.com/tpvkdJkX [13:02] <mithodin> and encoding just seems to freeze at frame 500 [13:02] <mithodin> which is exactly where the first overlay stops (see line 6) [13:04] <blippyp> okay, but the filter was working fine wihtout the select in there? why do you want to use that? [13:05] <mithodin> because without it, the concat does not work [13:06] <mithodin> in that the second overlay does not appear at 20s [13:07] <blippyp> the [so][start] overlay? [13:08] <mithodin> no, the second part of the [start] overlay [13:08] <mithodin> so [titel] appears, but not [ortzeit] [13:10] <blippyp> okay, so you're happy with the first three fades yes? [13:11] <blippyp> (ignoring before you started playing with the select filter) [13:11] <mithodin> yes [13:11] <dloo> hello [13:11] <blippyp> then you want to get rid of everything and just have the video and the logo.png overlay [13:12] <dloo> if i want to record : vlc v4l2:///dev/video0 :v4l2-standard=SECAM :input-slave=alsa:// :v4l2-chroma= :v4l2-input=1 :v4l2-audio-input=-1 :v4l2-width=768 :v4l2-height=576 :v4l2-aspect-ratio=4\:3 :v4l2-fps=0 :no-v4l2-use-libv4l2 :v4l2-tuner=0 :v4l2-tuner-frequency=-1 :v4l2-tuner-audio-mode=-1 :no-v4l2-controls-reset :v4l2-brightness=-1 :v4l2-brightness-auto=-1 :v4l2-contrast=-1 :v4l2-saturation=-1 :v4l2-hue=-1 :v4l2-hue-auto=-1 :v4l2-white- [13:12] <dloo> balance-temperature=-1 :v4l2-auto-white-balance=-1 :v4l2-red-balance=-1 :v4l2-blue-balance=-1 :v4l2-gamma=-1 :v4l2-autogain=-1 :v4l2-gain=-1 :v4l2-sharpness=-1 :v4l2-chroma-gain=-1 :v4l2-chroma-gain-auto=-1 :v4l2-power-line-frequency=-1 :v4l2-backlight-compensation=-1 :v4l2-band-stop-filter=-1 :no-v4l2-hflip :no-v4l2-vflip :v4l2-rotate=-1 :v4l2-color-killer=-1 :v4l2-color-effect=-1 :v4l2-audio-volume=-1 :v4l2-audio-balance=-1 :no-v4l2-audio- [13:12] <dloo> mute :v4l2-audio-bass=-1 :v4l2-audio-treble=-1 :no-v4l2-audio-loudness :v4l2-set-ctrls= :live-caching=300 --sout 'file/ps:testbfm.mpeg' --ts-es-id-pid --no-video --audio-track-id=1005 , can i do the same with ffmpeg? [13:12] <dloo> hmm sorry, too much lines [13:12] <dloo> it : :v4l2-standard= :input-slave=alsa:// :v4l2-input=1 :v4l2-audio-input=-1 :v4l2-width=768 :v4l2-height=576 :v4l2-standard=SECAM [13:13] <mithodin> no, logo.png should be there always [13:13] <blippyp> okay [13:13] <blippyp> so what do you want to happen after the 3 fades? [13:13] <mithodin> there are 7 more fades later, but I wanted to get it right with the first three [13:14] <mithodin> the only thing I'm trying to fix is that the fps get very low when I just use overlays on top of overlays all the time [13:14] <mithodin> for encoding [13:15] <blippyp> so this is just a 'takes too long to encode' issue? [13:15] <mithodin> and so I thought that since overlays 1 - 3 don't really show up at the same time but sequentially, I could maybe concatenate them and have only two overlays, one of logo.png and one of the other ones combined [13:15] <waressearcher2> dloo: thats a lot of text [13:16] <mithodin> blippyp: basically, yes. [13:17] <waressearcher2> to get 360 seconds of silence I use "ffmpeg -f lavfi -i aevalsrc=0:0:0:0:0:0::duration=360 -ac 2 -ar 44100 -ab 320k -y -f ac3 s_360.ac3" is it correct command to generate silence ? [13:17] <mithodin> I'm at ~5 fps for as it is now, and as I said, I need 7 more overlays [13:17] <blippyp> I don't think you can solve that - I can give you a 'sort of quick fix' - but you'll still likely want to encode it again after the face (without the filtering required though) [13:17] <blippyp> face=fact [13:18] <blippyp> like right now you haven't specified the codecs you want to use at all... [13:18] <mithodin> yes [13:18] <blippyp> if you changed the -c:a copy test.mp4 to: [13:18] <waressearcher2> how to mix 10 audio files ? so far I use command "sox -m 1.mp3 2.mp3 10.mp3" but is it possible to use ffmpeg instead of sox ? [13:18] <mithodin> In the end I need one output for DVD and one for Youtube [13:18] <blippyp> -c:v libx264 -crf -qp 0 -preset ultrafast -c:a ac3 test.mp4 [13:19] <blippyp> it would decrease the encoding time, but it will still need to go through the 'filtering' [13:19] <blippyp> and your file size would be large [13:19] <mithodin> large as in 80 GB or large as in 20? [13:20] <blippyp> you'd still need to re-encode the final video afterwards to drop the file size down (i normally change -qp 0 to -crf 17) for my final videos [13:20] <blippyp> depends on the length of the video [13:20] <blippyp> how long is your main video? [13:20] <waressearcher2> I have 26 minutes audio with voice that read text and 3-4 seconds in between pauses, is there a way to remove those silent pauses ? some filter maybe ? [13:20] <mithodin> about 40 minutes [13:20] <blippyp> that will take a while regardless... [13:21] <blippyp> you might as well use -c:v libx264 -crf 17 -c:a ac3 to begin with and be done with it... [13:21] <mithodin> can't I still use -c:a copy? [13:21] <iive> audacity is non-linear audio editor, it might be better than linux video editors. [13:21] <blippyp> setup your filter with shorter times (so that you're only playing with a couple of minutes) [13:22] <mithodin> what do you mean, shorter times? [13:22] <blippyp> and get it working the way you like and then when it's the way you want change your fade times to what they should be, double check, triple check, and encode....??? [13:23] <blippyp> or you can probably set your PTS - that should drastically increase things for 'testing' it until you get it right [13:24] <blippyp> basically you could encode the 40 minutes into a 5 minute video instead - leaving all your fades and everything the way you want it - it would SHOW the fades as they should appear, but they will come and go quickly [13:24] <mithodin> ah, I see what you mean [13:24] <blippyp> but the entire video would be there (kind of like fast-forwarding through a movie) [13:24] <blippyp> setpts i the filter [13:24] <blippyp> i=is [13:25] <mithodin> good idea, thanks [13:25] <blippyp> to be honest, I'd probably go the pts route... [13:25] <blippyp> no problem [13:25] <blippyp> always glad to help (it doesn't always work out that way) haha ;) [13:26] <blippyp> but to be honest, with a project like this you'd be much better off with an actual video editor [13:26] <mithodin> if there was one which actually worked, I would agree [13:27] <waressearcher2> no video editor for linux ? [13:27] <blippyp> ya, there's a few [13:27] <blippyp> mostly lame, but they should work for what he wants [13:27] <blippyp> which one did you say you were using? [13:27] <mithodin> openshot does what I need [13:27] <mithodin> but it breaks the a/v sync [13:28] <blippyp> maybe try kdenlive? [13:28] <blippyp> I don't like it, but lots of people apparently do [13:28] <waressearcher2> last one I checked when you open video file that editor first convert it to pictures jpg or png, after that I gave up on video editors for linu [13:28] <waressearcher2> x [13:29] <mithodin> I don't want to have kde programs installed on my computer. They always pull a ton of useless stuff I don't need [13:29] <blippyp> ya, that's my issue with them as well [13:30] <blippyp> I've used kdenlive, it was alright - but in the end ffmpeg is much faster for my general needs, so I pretty much stick with ffmpeg [13:30] <blippyp> I think I even still have it installed, not sure though [13:30] <blippyp> it was a little buggy, but it was alright for a basic editor [13:31] <mithodin> is there a video editor based on ffmpeg maybe? [13:31] <blippyp> most of them pretty much are I think.... :P [13:31] <blippyp> kdenlive has ffmpeg options [13:31] <blippyp> actually they were probably x264 options [13:33] <blippyp> cinelerra looks someone decent [13:34] <mithodin> I tried it and I could not even figure out how to import a clip [13:34] <blippyp> really? looked like a nice easy editor......??? [13:35] <blippyp> I would guess clicking on file or video would give you an import option in the menu? [13:35] <mithodin> yeah I found it eventually, but still... for me it was utterly confusing [13:35] <mithodin> then again, I do prefer terminal programs ;-) [13:36] <blippyp> I miss premiere [13:36] <mithodin> what's that? [13:36] <blippyp> adobe premiere? [13:36] <mithodin> ah, I see [13:37] <blippyp> there are basically three decent video editors that I'm aware of - premiere, vegas and (I can't remember the other one - it's REALLY popular though, many consider it to be the only real professional app of the three) [13:37] <blippyp> I think it starts with an f [13:38] <mithodin> final cut? [13:38] <blippyp> yeah [13:38] <blippyp> never used it myself though [13:38] <mithodin> a friend of mine has it [13:38] <blippyp> I honestly can't imagine it having much more options that premiere - premiere does everything [13:38] <mithodin> seems to like it a lot [13:39] <blippyp> yeah I find people are usually fans of one of those three and of course the final cut guys hate the premiere guys as much as the windows guys hate the linux guys... yadda yadda yaddda - whatever [13:39] <blippyp> they all basically do the same damn thing... ;) [13:39] <blippyp> the great part of premiere though is the integration with the other apps in the suite along with the database for managing your videos - it's pretty cool [13:40] <blippyp> I imagine they all have the ability to have basically the same filters, so it pretty much likely comes down to person preference on the user interface [13:40] <mithodin> of course, with Adobe CS you get great interation, it's one package [13:41] <blippyp> yeah, but I wouldn't touch an adobe app for anything these days, I hate the whole cloud nonsense - I just threw my hands up over the idea (not that I'm using it anymore anyways, but still) [13:44] <blippyp> well, it's really late for me - I'm gonna go catch a few hours of sleep before my kid wakes up - good luck with your video mithodin - hope the pts idea helps you out [13:44] <mithodin> thanks, see you [13:44] <blippyp> bye bye *gone* [13:45] Action: blippyp vanishes without a trace [13:48] <waressearcher2> blippyp: sup [14:15] <dloo> anyones know how to change the v4l2 input? [15:30] <iive> dloo_: try -f v4l2 -i /dev/video1 [16:26] <dloo_> iive: already tried it :( [18:21] <socksandsandals> hi all [18:22] <socksandsandals> I am receiving some VP8 frames from another source that I want to write to a file and then turn into a video file when its all over [18:22] <socksandsandals> sometimes there are gaps in the frames, however, when the source is not sending any frames [18:23] <socksandsandals> would I have to fill in those gaps with a YUV black frame in order to make it video file-able? [18:23] <socksandsandals> and if so, do I write just one black frame and just set the timestamp of the next real frame to the size of the gap in time? [18:25] <socksandsandals> I guess what I'm asking is, do I have to have a frame written in the video file for every frame I would have received had it not been interrupted? or can a video file have a frame that's "repeated" until the next one comes along? [18:42] <nyuszika7h> hi, I'm using a fairly recent build for Windows, but the -to option doesn't seem to work as expected [18:42] <nyuszika7h> it seems to work about the same as -t [18:43] <nyuszika7h> okay, will do that when I get to my laptop soon [19:27] <nyuszika7h> https://gist.github.com/anonymous/1cca6444a980928a3f9d [19:28] <nyuszika7h> the end doesn't even get stripped [19:30] <c_14> If you use -ss as an input option, the startpoint of the video file is adjusted by that time. So, with what you have, you will copy everything from (00 to 42:25) adjusted forward by 40 seconds. [19:31] <c_14> So in reality everything from 00:40 to 43:05 [19:31] <nyuszika7h> yeah, that's with -t, isn't -to supposed to solve that? [19:32] <c_14> Only if you use -ss as an output option [19:32] <nyuszika7h> ah, so put it after -i [19:32] <c_14> If you use -ss as an input option, ffmpeg seeks $time seconds into the file before applying any output options therefore -to believes that the start point of the video file is wherever the beginning seek point is. [19:33] <nyuszika7h> I see [19:35] <c_14> Whether or not that is a bug or a feature is up for debate, but that's how it currently works. [19:35] <Schrostfutz> hi, I want to make a screencap with ffmpeg, but it is too fast, what am I doing wrong?: "ffmpeg -f x11grab -r 30 -s 1916x1041 -i :0.0+0,24 -vcodec libx264 -threads 0 video.mkv" [19:36] <nyuszika7h> Windows Media Player likes to get the total time wrong when I cut a video with ffmpeg [19:36] <nyuszika7h> mpv gets it right [19:39] <c_14> nyuszika7h: If the timestamps in the file are correct, I'd probably blame wmp for being shit at decoding. [19:40] <c_14> Schrostfutz: try -r 25 [20:27] <ChocolateArmpits> Or -r 15 [20:46] <Zeranoe> Does FFmpeg have a command line option to determine if it supports multithread (-threads)? [20:48] <fajung> ned help recording my desktop with speakers(system) audio, ffmpeg just records the mic and I don't know how to fix this: http://s2.postimg.org/jv6bmdgh5/Captura_de_pantalla_de_2014_05_18_15_49_09.… [20:50] <klaxa> fajung, are you running pulseaudio? [20:50] <klaxa> i would think so, because you are running ubuntu [20:50] <klaxa> if my guess is correct, try using pulse instead, use -i pulse instead of -i default [20:50] <fajung> don't know, new in Linux, how can I check it? [20:51] <klaxa> then open pavucontrol and select the monitor device for the ffmpeg recording stream in the recording tab [21:05] <fajung> I think that I get it now, thanks. why when I start ffmpeg it echo "configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-x11grab --enable-libpulse --enable-libx264 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-libass --enable-avisynth --en [21:05] <fajung> able-libsoxr" ? [21:06] <c_14> It just does that. Consider it a feature. [21:08] <fajung> if I compile it in my ubuntu14.04, if I upgrade my kernel I'll have to re compile it again? [21:09] <c_14> nah, why should you? [21:09] <fajung> ok, thanks for the help [21:25] <troyp> Hello. I'm compressing a mp4 to H.264 and it keeps rotating the video. Is there anyway to tell it to "leave it alone" lol [21:33] <c_14> rotating the video? [21:34] <troyp> ok one sec [21:39] <troyp> c_14: http://pastebin.com/KErZGTNq [21:39] <c_14> That transpose part is rotating the video. [21:40] <troyp> right, so let me explain. Without that, when I film in "portrait" mode on iPhone. It's rotated to "landscape" mode. This fixes that, but it breaks it for the other formats [21:41] <troyp> so i guess I need to detect when the file is filmed like this, and only add this command [21:42] <c_14> pretty much [21:42] <troyp> ok [21:44] <troyp> what's the best way to get this information? [21:44] <troyp> ffmpeg -i? [21:45] <c_14> probably [21:45] <troyp> it has some very interesting metadata it says "rotate 90" [21:46] <troyp> so i guess if i could extract that [21:46] <troyp> then i'll know [21:49] <troyp> very strange. There's multiple Metadata sections listed under input #0. Why is that? [21:50] <c_14> Magic? [21:50] <c_14> Can you paste it? [21:50] <troyp> http://pastebin.com/KErZGTNq line 28 [21:51] <c_14> The first one is for stream 0:0 the audio stream, the second one is for stream 0:1 the video stream [21:51] <troyp> http://pastebin.com/UgHtsAmh [21:51] <troyp> so it looks like it's returning only the audio stream with this command. Anyway to specify video? [21:52] <troyp> or maybe it's not the audio, but the overall metadata [21:57] <c_14> Hm, it should dump all the metadata by default, but you could always just dump the ffprobe output to a file ffprobe $file 2>out [21:57] <troyp> so they're calling it global metadata [21:58] <troyp> let me try an updated version and see if it's an old bug, one sec [21:58] <troyp> nope, same results. weird [21:58] <troyp> what are you saying about ffprob? [22:00] <c_14> ffprobe $file 2>outfile should dump all the output from ffprobe to a file and that should have the info as well [22:01] <troyp> ahh yeah [22:01] <troyp> nice [22:01] <troyp> ffprobe -v quiet -print_format json -show_format -show_streams 07E87390DD3011E3931EDDA6A0A26CF9.mp4 [22:01] <c_14> that works too [22:01] <troyp> this works [22:01] <troyp> thanks [22:03] <troyp> c_14: am i mistaken or is the input file already H.264? [22:03] <troyp> based on what i'm reading here [22:03] <c_14> it is h264 [22:03] <troyp> the only thing i'm trying to do is turn HD into SD... [22:03] <troyp> doing this takes makes the file-size 80% smaller [22:04] <troyp> so i guess it's really the other stuff i'm doing like bitrate [22:04] <c_14> yep [22:04] <troyp> got it, thanks! [23:29] <waressearcher2> is anyone ready to answer my questions ? [23:29] <waressearcher2> questions, plural [23:29] <c_14> just ask and if someone can help you, they will [23:30] <waressearcher2> I allready asked 3 of them yeasterday, no one bothered [23:31] <waressearcher2> c_14: can you scroll back for about 16 hours ? [23:32] <waressearcher2> anyway, I have mp3 file, 26 minutes, with voice, that is read text, phrases, and after each phrase there is 3-4 seconds pause, how to remove pauses ? is there any filter or something ? [23:33] <c_14> You might want to use audacity for that. That's not exactly ffmpeg's territory. [23:34] <waressearcher2> c_14: you mean by hand ? I have dozens such mp3 files and there are hundreds of pauses I need to cut out, can audacity do that ? [23:34] <waressearcher2> there should be some filter [23:35] <c_14> http://audacity.238276.n2.nabble.com/remove-silences-td293043.html [23:35] <c_14> not sure if audacity has some scriptable interface, but it might [23:35] <waressearcher2> another one, to generate 360 seconds of audio silence I use that command: "ffmpeg -f lavfi -i aevalsrc=0:0:0:0:0:0::duration=360 -ac 2 -ar 44100 -ab 192k -y -f ac3 s_360.ac3", is it correct one or is there a better method to get silence ? [23:37] <c_14> I think you can just use aevalsrc=0:duration=360, but that should be fine [23:38] <c_14> waressearcher2: For your first problem, apparently you can use sox: https://stackoverflow.com/questions/11834297/how-can-i-remove-silence-from-… [23:48] <vdm> I have file with raw audio data in G.711 MULAW codec format. how i can play the file data using ffmpeg? [23:49] <c_14> try ffplay? [23:51] <vdm> c_14, ffplay? [23:51] <vdm> i'm using linux, ffmpeg is installed but i have no ffplay command [23:52] <c_14> You sure you have ffmpeg not libav? [23:55] <vdm> c_14, i tried use ffplay, here is the error output: http://pastebin.com/hp6Ltgsh [00:00] --- Mon May 19 2014

1 0

ffmpeg-devel.log.20140517
by burek 18 May '14

18 May '14

[00:33] <BBB> Voicu: java is pretty, but tries to solve a problem that we don't have, it's higher-level [01:20] <wm4> <Voicu> [15:55:50] whoever designed the java API must have been drunk and/or high [01:20] <wm4> <Voicu> [15:56:27] there is no way a sane person would unleash something like that on a fellow programmer [01:20] <wm4> Voicu: is it specific to the java binding, or the ffmpeg API (assuming you're talking about a java ffmpeg binding) [03:02] <jamrial> just sent some hevc patches to fix and improve some minor things [03:23] <jamrial> BBB: sorry for the AVX inline :P [03:23] <jamrial> i tried to refactor the function two times, and as michaelni mentioned the best i could get was %14 slower than inline because of the calling overhead [03:24] <jamrial> only solution is probably rewriting the entire loop in yasm, and i'm not really up for that [03:30] <compn> jamrial : hehe its ok, there was a large inline asm purge a few years ago [03:30] <compn> converted a lot of stuff to yasm/nasm [03:30] <jamrial> i don't like inline either. which is why i tried to port that stuff to yasm [03:32] <jamrial> but %14 is indeed a performance hit too big to swallow [03:32] <compn> it might be worth having the %14 hit just for msvc anyhow [03:33] <compn> for msvc only i mean [03:33] <compn> the code is still faster than C right ? [03:33] <compn> than the c-code i eman [03:33] <compn> if you can also provide it [03:35] <jamrial> yes, even with that hit it would still be faster than the C version [03:35] <compn> then we'll have both versions, one for if_msvc and one for if_!msvc [03:35] <compn> :) [03:38] <jamrial> the yasm port i wrote is on the ml in case someone wants to make it work alongside the c and the inline version [03:38] <compn> ok good [03:39] <compn> we have to bend over backwards to appease the windows users :P [03:40] <compn> but its good to be faster on any platform too [03:40] <compn> optimize everything! [03:40] <BBB> jamrial: well yeah obviously inlining makes a massive difference [03:41] <BBB> jamrial: I can look at that, at least try to suggest on how to do it, not sure, don't have tons of time lately :/ [03:41] <BBB> anyway [03:41] <BBB> inline avx is beyond omg [03:48] <jamrial> and gcc 4.9.0 added AVX-512 inline :P [04:02] <wm4> why not use intrinsics [04:20] <bloody123> hi all, there's a minor description bug in http://ffmpeg.org/ffmpeg-filters.html#concat - anyone wanna fix? it's about replacing one word... [04:25] <bloody123> ...in option 'a', it reads: "that is also the number of video streams in each segment." - should be "audio streams", not "video streams", right? [05:11] <michaelni> j-b, is cone ok? she is a bit silent lately ... [06:41] <jamrial> for the record, my recent patchset did not attempt to address the problems with plepere's third patch [06:41] <jamrial> it's mostly the sse2 stuff i mentioned in the ml, plus some misc stuff [06:42] <jamrial> i still have to send the patch that adds the sse2 version of the luma functions in question [10:32] <Voicu> wm4: I was referring to the java binding for ffmpeg [10:32] <Voicu> wm4, I can't say I'm 100% happy with the ffmpeg API (C++) but that on the whole is pretty ok [10:59] <kurosu> mraulet, do you know if Pierre-Edouard finished working on MC? [12:40] <cone-664> ffmpeg.git 03Carl Eugen Hoyos 07master:4c49d0824a10: Fix alaw and mulaw muxing in caf. [12:52] <wm4> Voicu: well, you could write a better one [13:17] <Voicu> wm4, I really did think about a solution like that [13:18] <Voicu> i.e. make an API for my C++ code and use the C++ code instead of porting it to java [13:20] <Voicu> wm4, are you familiar with the java wrappers ? [13:20] <wm4> no [13:20] <Voicu> I'm asking because I think there might be a bug nad if it's not a bug it's a really weird behavior that should be rectified [13:25] <wm4> I'm just saying... this java wrapper might be some of the many projects depending on ffmpeg that don't receive enough maintenance [13:25] <wm4> because ffmpeg's API is a relatively fast moving target [13:26] <Voicu> wm4: I don't have anything against the ffmpeg API [13:26] <Voicu> it's not perfect but then again no API is perfect [13:27] <Voicu> and ffmpeg itself is awesome, it can do a lot of things [13:27] <Voicu> I just got frustrated with this project :D [13:27] <Voicu> android and ffmpeg apparently don't mix that well (yet!) [13:29] <Voicu> I certainly will make a post/article about my findings and troubles I had [13:30] <Voicu> and maybe even try and improve some of them [13:30] <Voicu> so others won't have to feel the same pain :D [13:41] <mraulet> kurosu, more or less yes [13:41] <mraulet> we dont exactly have the same performance than intrinsics for some functions [13:47] <mraulet> kurosu what do you have in mind? [13:47] <wm4> Voicu: what kind of insane stuff does the java binding do? I'm just curious [13:48] <Voicu> wm4, well I had to use an AVIOContext with a custom read procedure - I need to feed data from Android's MediaCodec into a ffmpeg stream [13:49] <Voicu> in C++ I did it with a callback of the form int (*p)(void *ptr, uint8_t *buffer, int buffer_size) [13:50] <Voicu> on the java side there are 4 different classes, one of which (I assumed) I had to extend [13:50] <Voicu> but... having more than 1 class (i.e. 2 callback) is not possible because somehow, somewhere a pointer gets overwritten [13:50] <Voicu> so all my avio contexts call the same callback [13:51] <Voicu> (the last one initialized) [13:51] <Voicu> so I said OK, I'll use one callback and use the pointer to differentiate between different callback behaviors [13:51] <Voicu> but! [13:52] <Voicu> I don't get the exact pointer object I provide. Instead I get a copy [13:52] <Voicu> so I had to resort to a horrible hack - I use the .address field of the Pointer object (which is meant to refer to a C++ side object) [13:53] <Voicu> the .address field is apparently the only one that gets preserved from the object I provide to the one I get in the callback [13:54] <Voicu> and that's just part of it [13:55] <Voicu> in C++ I can do avformat_alloc_output_context2(context=NULL, ... ) [13:55] <BBB> wm4: intrinsics have the problem of not knowing what goes in registers or memory (stack) [13:56] <BBB> wm4: plus it depends on the compiler doing well [13:56] <Voicu> in java I have to do context=avformat_alloc_context() first [13:56] <BBB> wm4: if compilers were so good, wouldn't we all use java or python? [13:56] <Voicu> otherwise the whole thing crashes [13:56] <wm4> BBB: well, it seems totally standard to use intrinsics on MSVC/C++ [13:56] <BBB> wm4: compilers suck, they are created using automated algorithms that work a little bit sometimes, but they typically generate inefficient code [13:56] <BBB> wm4: sure, you can, it's better than c; but yams typically beats intrinsics [13:56] Action: Daemon404 smashes BBB's stack [13:57] <BBB> wm4: yams written by a knowledgeable person [13:57] <BBB> hm, stack smashing, yummy [13:57] <ubitux> http://b.pkh.me/writing_filters.txt [13:57] <nevcairiel> MSVC is pretty good at translating intrinsics to asm, as long as you are aware of how many registers you use [13:57] <ubitux> review welcome, maybe i'll submit it as a patch in doc/ soon [13:57] <nevcairiel> i usually browse through the generated asm for my intrinsics to check [13:58] <BBB> yams has a higher chance of doing the correct thing [13:58] <BBB> apple's gcc used to do horrible on inline asm [13:58] <nevcairiel> also takes like double the time to write it though [13:58] <BBB> basically inserting movs after and before each and every single inline asm block [13:58] <BBB> kinda like windows 32bit inline asm [13:58] <BBB> nevcairiel: I find it fun .. [13:59] <nevcairiel> intrinsics take all the C crazy and just handle it for you, so you can focus on the actual algorithm [13:59] <wm4> intrinsics also don't require you to track down weird crashes because the yasm code might violate the ABI or do something invalid [14:00] <nevcairiel> might end up slightly less optimized [14:00] <BBB> people use python, people use java, people use c, people use asm [14:00] <BBB> likewise, people use inline, yasm, intrinsics, even crazy win32 inline asm [14:00] <BBB> use whatever suits you [14:00] <BBB> in ffmpeg, we use yasm :-p [14:01] <nevcairiel> someone should invent a way to inline a yasm block [14:01] <Daemon404> perl mypreproc.pl [14:01] <Voicu> hehe, when I was in highschool I made a graphics library in Turbo Pascal with inline assembly (16 bit) :D [14:01] <BBB> oh I love perl - and bash [14:02] <BBB> bash is so sweet, like honey [14:02] <Voicu> it was probably pretty bad code but fun to make [14:02] <BBB> and then this happens: http://stilldrinking.org/programming-sucks [14:02] <nevcairiel> actual inline algorithms have their right to live in many parts of the code, sadly yasm can't help us there [14:03] <BBB> if it's actually inlined... sure [14:03] <BBB> but if it's just another dsp function [14:04] <BBB> and you may think that, oh, this is just 1% here or there [14:04] <BBB> but all these 1%s add up to ffvp9 considerably beating libvpx-vp9 [14:04] <BBB> we didn't implement another algorithm, it's exactly the same decoder algo [14:05] <BBB> it's just more efficient, so yes, 1%s matter [14:05] <BBB> a lot [14:05] <mraulet> BBB it is not yet the case for MC intrinsics beat yasm on my laptop [14:16] <cone-664> ffmpeg.git 03Clément BSsch 07master:11e490334e0d: avfilter/edgedetect: reuse already defined ctx. [14:21] <BBB> mraulet: so fix the assembly :-p [14:22] <mraulet> let see what plepere can do [14:22] <mraulet> :p too [14:40] <mraulet> openhevc/ffmpeg/rext_new_mc (5.8s) ffmpeg/master (9.5s) [14:41] <mraulet> still some ASM to do on IDCT and SAO [14:54] <cone-664> ffmpeg.git 03Marton Balint 07master:ae6fe159f299: ffplay: increase AV_SYNC_THRESHOLD_MIN to 0.04 [14:54] <cone-664> ffmpeg.git 03Marton Balint 07master:1fab67b6851f: ffplay: fix compilation with Visual Studio [14:54] <cone-664> ffmpeg.git 03John Peebles 07master:e11697759d06: cmdutils: replace usages of "#ifdef __MINGW32__" with "#ifdef _WIN32" because MSVC only defines _WIN32 [14:54] <cone-664> ffmpeg.git 03Marton Balint 07master:0c8d8c0c8067: ffplay: try multiple sample rates if audio open fails [14:54] <cone-664> ffmpeg.git 03Marton Balint 07master:affc41047e96: ffplay: fix typo in docs [14:54] <cone-664> ffmpeg.git 03Marton Balint 07master:a583e2bebe77: ffplay: add support for toggling between multiple video filters with the w key [14:54] <cone-664> ffmpeg.git 03Michael Niedermayer 07master:5b0c7052fb2c: Merge remote-tracking branch 'cus/stable' [14:55] <mraulet> BBB 5.9s with MC intrinsics compared to 6.3s MC asm on BasketBallDrive QP 27 RA with one core [14:58] <BBB> mraulet: you're a university student, right? [14:58] <mraulet> not really student ^^ [14:58] <BBB> whatever, you've got an academic background [14:58] <mraulet> Research engineer :-) at a Engineering school [14:58] <BBB> so apply your academic principles on this problem [14:59] <BBB> your end goal shouldn't be to convince me that intrinsics are better than asm [14:59] <BBB> a) they're not, and b) it's irrelevant, we already decided to not use intrinsics in ffmpeg [14:59] <BBB> your goal should be to learn _why_ your particular intrinsics implementation behaves better than your particular asm implementation [15:00] <BBB> then use what you just learned, improve the asm [15:00] <BBB> and voila, you've got better-performing asm [15:00] <BBB> and everyone is happy [15:02] <wm4> didn't they recently decide to use intrinsics for some arm code in libav? [15:03] <cone-664> ffmpeg.git 03Carl Eugen Hoyos 07master:ef2713747f9d: Fix compilation of libavcodec/x86/hevc_deblock.asm with nasm. [15:03] <cone-664> ffmpeg.git 03Michael Niedermayer 07master:a7320c1574d2: Merge remote-tracking branch 'cehoyos/master' [15:03] <nevcairiel> probably because there is no proper inline support [15:03] <nevcairiel> arm is all kinds of crazy [15:08] <BBB> I believe they also found it to be slower (on arm 32bit) than asm [15:14] <kurosu> mraulet, some of the assembly is kind of lazy - suboptimal argument loading and register allocation [15:16] <kurosu> I have an example with 8 preloaded gprs, 10 needed and 11 xmms [15:16] <kurosu> turns out, it can probably be done in 5/5/7 [15:17] <mraulet> kurosu, do you plan to work on MC? [15:17] <kurosu> not really, I was mostly looking out of curiosity [15:32] <nevcairiel> 64-bit makes people lazy [15:32] <nevcairiel> so many registers to waste [15:37] <kurosu> on the other hand, I might be picky [15:38] <cone-664> ffmpeg.git 03Olivier Langlois 07master:0eec06ed8747: lavu: add av_gettime_relative() [15:45] <kurosu> mraulet, what would be a good sample to test the uni_w functions ? [15:46] <mraulet> I am using sample from here [15:46] <mraulet> ftp://ftp.kw.bbc.co.uk/hevc/hm-11.0-anchors/bitstreams/ra_main/ [15:47] <mraulet> BasketballDrive_1920x1080_50_qp27.bin [15:53] <kurosu> I'm getting no output on stderr or stdout with this command-line: ./ffmpeg.exe -threads 1 -i <file> -loglevel debug -f null - [15:53] <kurosu> Anybody has an idea why that would happen with mingw ? [15:54] <BBB> -v debug? [15:54] <BBB> not sure though [15:55] <kurosu> nope [15:55] <mraulet> your command line works for me [15:55] <kurosu> libav's avconv is ok [15:55] <mraulet> macos [15:56] <kurosu> well, my changes pass fate, but I can't get a START_TIMER to work [15:56] <BBB> does -v error work? [15:56] <BBB> instead of -v debug [15:56] <BBB> maybe someone broke -v debug to mean "only debug", not "all until debug" [15:56] <BBB> (START/STOP_TIMER is error) [15:57] <kurosu> -v error -loglevel error doesn't work either [15:57] <mraulet> but the stderr output has changed recently (it does not look really nice now :) ) [15:57] <BBB> sorry no idea :( [15:57] <BBB> change STOP_TIMER to do printf :-p [15:57] <kurosu> actually, I'm having strictly no output [15:57] <cone-664> ffmpeg.git 03Olivier Langlois 07master:b052bccbe4b5: lavc: Use av_gettime_relative() [15:57] <BBB> maybe your terminal is broken [15:57] <cone-664> ffmpeg.git 03Olivier Langlois 07master:41120e6e40e3: tools: Use av_gettime_relative() [15:57] <kurosu> not even the program prolog [15:57] <BBB> why don't you kill the terminal and restart it [15:57] <kurosu> yeah, the pity avconv works [15:57] <BBB> oh right [15:57] <kurosu> *is avconv [15:58] <BBB> ... [15:58] <BBB> I have no idea [15:58] <kurosu> mingw's terminal [15:58] <BBB> does printf work? [15:58] <BBB> like, if you add a printf somewhere, does it print? [15:58] <michaelni> kurosu, does it work with older ffmpeg ? if so which commit broke it ? [15:59] <kurosu> I believe not: I'm not even getting a program output (like ffmpeg (c) something or compiled with as I think it used to) [16:00] <kurosu> michaelni, need to bisect I guess, but given slowness of configure + compile... what would be a minimal build to test just that ? [16:01] <michaelni> i think there were some tricks to make configure faster on win32, not sure which though [16:01] <nevcairiel> kurosu: that means your ffmpeg was linked with SDL, which caused it to be a GUI app and not have stdout/stderr [16:02] <kurosu> even with --disable-outdev=sdl,opengl ? [16:02] <michaelni> i think configure is faster with dash than bash [16:02] <kurosu> nevcairiel, yeah, funky -Dmain=SDLmain [16:02] <kurosu> gonna remove that from config.mak [16:02] <nevcairiel> the real problem is -mwindows somewhere [16:03] <nevcairiel> which SDL enforces [16:03] <kurosu> I thought they were redirecting stdout/err to files also ? [16:04] <michaelni> btw configure with bash 12sec, configre with dash 4sec (on linux) [16:04] <nevcairiel> should really only apply the SDL CFLAGS to ffplay and not to ffmpeg [16:05] <nevcairiel> stupid -mwindows [16:05] <nevcairiel> i just dont have SDL installed anymore :p [16:05] <kurosu> I used to have another project relying a lot on SDL [16:05] <kurosu> but no longer today, so that's probably the best option [16:06] <kurosu> but really, yes, it should only apply to ffplay (patch welcome :D) [16:06] <nevcairiel> you could also mess with its pkgconfig file and strip the -mwindows [16:07] <ubitux> kurosu: well, ffmpeg needs libavdevice, and libavdevice has a sdl device [16:07] <ubitux> kurosu: you can use sdl with ffmpeg [16:07] <ubitux> typically, you can follow an encode by doing a -f sdl - [16:07] <kurosu> ubitux, I'm running configure with --disable-outdev=sdl,opengl [16:08] <kurosu> and --disable-ffplay [16:08] <nevcairiel> i never managed to get rid of SDL cflags in the build [16:08] <ubitux> then it shouldn't link against sdl :p [16:08] <nevcairiel> as long as SDL is installed, it'll add them [16:08] <kurosu> detecting and using SDL shouldn't even happen [16:08] <nevcairiel> so.. byebye sdl [16:08] <ubitux> i guess that's a bug :( [16:08] <kurosu> the problem is sdlconfigure passes -Dmain=SDL_main and relevant LDFLAGS/CFLAGS [16:09] <kurosu> *sdl_config iirc [16:09] <kurosu> if it's not through some pkgconfig detection [16:09] <nevcairiel> i suppose both ways exist [16:09] <kurosu> anyway, it's recompiling but I'll get rid of sdl then [16:10] <nevcairiel> the weird thing is that configure adds the sdl cflags globally to the main cflags, but then additionally to the ffplay one as well, as if someone tried to fix it once, but then it broke again :D [16:20] <ubitux> i guess the problem is that sdl is auto detected [16:22] <cone-664> ffmpeg.git 03Clément BSsch 07master:2ca97c7802b6: configure: make sure pkg-config flags are populated in FT test. [16:22] <cone-664> ffmpeg.git 03Clément BSsch 07master:9986e50a6ec8: configure: make vp9 decoder select the parser. [16:25] <kurosu> I can't find a sample testing it [16:29] <kurosu> ok, WP_A_Toshiba_3.bit it was [16:29] <ubitux> the sdl stuff in configure is pretty sick& [17:06] <kurosu> I don't really intend to work more on the hevc code, but on closer inspection, this one struck me as odd [17:06] <kurosu> I don't expect everything to be doable so as to make it work on 32bits systems, but still [17:11] <kurosu> nevcairiel: thanks for the hint btw, last time I noticed that, I thought it was a fluke [20:23] <cone-612> ffmpeg.git 03Olivier Langlois 07master:f78bc96b7c1c: lavf: Use av_gettime_relative() [20:27] <cone-612> ffmpeg.git 03nu774 07master:584f88409062: riff: Pass block_align to estimate frame duration [20:27] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:43c18fec6e7c: Merge commit '584f88409062f7a134e7391887899e8e723ab6ff' [21:05] <Daemon404> 300 get [21:11] <nevcairiel> we should just take the patch and apply it, can't get any worse [21:12] <cone-612> ffmpeg.git 03Olivier Langlois 07master:0ca0b4c29cf7: ffplay: Use av_gettime_relative() [23:02] <jamrial> michaelni: kurosu's patch compiles for me on mingw64 with Yasm 1.2.0, 1.1.0, 0.8.0, and NASM 2.11.2 [23:02] <jamrial> what are you using? [23:07] <michaelni> jamrial, yasm 1.1.0.2352 on ubuntu [23:09] <jamrial> odd [23:11] <mraulet> jamrial, it does not compile for me too [23:11] <mraulet> on osx [23:11] <jamrial> i'm booting a linux vm to check there [23:12] <mraulet> yasm 1.2.0 [23:13] <mraulet> exactly the same messages than michael [23:14] <jamrial> yeah, same here on Gentoo x86_64 [23:15] <jamrial> weird that it compiles fine in win64 [23:15] <jamrial> kurosu also tested it there now that i read his email again [23:25] <jamrial> changing the two r4q to r4d seems to fix it, and doesn't break mingw64 [23:25] <jamrial> can't run fate to test it, though [23:30] <mraulet> compilation ok but fate fails [23:58] <cone-612> ffmpeg.git 03Michael Niedermayer 07master:46380e8d2619: avformat/aviobuf/ff_get_line: also accept \r as end of line character [00:00] --- Sun May 18 2014

1 0

ffmpeg.log.20140517
by burek 18 May '14

18 May '14

[03:41] <decci> I have ffmpeg configured under /usr/local/bin/fmpeg. My web application is unable to run this ffmpeg due to permission issue [03:41] <decci> As I installed ffmpeg through root [03:42] <decci> My web application is running through normal user and so unable to run this command through PHP script [03:42] <c_14> What are the file permissions of /usr/local/bin/ffmpeg ? [03:42] <decci> c_14: ls -la /usr/local/bin/ffmpeg -rwxr-xr-x 1 root upeoapp2 19581640 May 16 07:59 /usr/local/bin/ffmpeg* [03:43] <decci> I tried modifying it with chown command but still not accessible [03:43] <c_14> Is /usr/local/bin/ in $PATH ? [03:44] <decci> c_14: echo $PATH /usr/local/jdk/bin:/usr/lib64/qt-3.3/bin:/bin:/usr/bin:/usr/local/bin:/sbin:/usr/sbin:/usr/local/sbin:/usr/local/bin:/usr/X11R6/bin:/root/bin [03:44] <decci> c_14: Yes its there [03:45] <c_14> Can your user execute the binary? [03:45] <decci> c_14: How to ensure that [03:45] <c_14> ffmpeg -version [03:46] <decci> c_14: http://paste.ubuntu.com/7475966/ [03:47] <decci> c_14: I ran it as root [03:47] <c_14> I'm guessing the path php is using isn't correct, have you tried running /usr/local/bin/ffmpeg from the script instead? [03:48] <decci> c_14: When I run as upeoapp2 user http://paste.ubuntu.com/7475970/ [03:48] <decci> c_14: If I run it as root user, it works [03:48] <c_14> If you run the php script as root user? [03:49] <decci> c_14: http://198.20.89.58/~upeoapp2/ffmpegtest/upload.php [03:49] <decci> c_14: This is what I am trying to build [03:49] <c_14> The file permissions are correct, and the user you install it as shouldn't prevent you from running ffmpeg. What error does it give you? [03:49] <decci> c_14: Try uploading any media video file and try to change the format..you will see that it shows http://paste.ubuntu.com/7475973/ [03:50] <decci> c_14: See this -i /home/upeoapp2/public_html/ffmpegtest/video/upload/testr.AVI -vcodec libx264 -maxrate 1024k -bufsize 1024k -f mp4 -strict -2 -y /home/upeoapp2/public_html [03:50] <decci> c_14: Its not showing /usr/local/bin/ffmpeg here [03:51] <decci> c_14: If I run it manually through root it works http://paste.ubuntu.com/7475981/ [03:52] <c_14> Do you have/can I see the php script? [03:52] <decci> c_14: Wait..I can show u its permission issue..confirmed [03:53] <decci> c_14: When I am running the command manually through root user it works but through upeoapp2 user it shows permission issue http://paste.ubuntu.com/7475984/ [03:53] <decci> c_14: The command is /usr/local/bin/ffmpeg -i /home/upeoapp2/public_html/ffmpegtest/video/upload/testr.AVI -vcodec libx264 -maxrate 1024k -bufsize 1024k -f mp4 -strict -2 -y /home/upeoapp2/public_html/ffmpegtest/video/converted/testr.mp4 [03:54] <c_14> It seems the user doesn't have write permission to the output directory. [03:54] <decci> c_14: How to fix it [03:55] <decci> c_14: -rw-r--r-- 1 root root 134453 May 16 08:08 retrer.mp4 -rw-r--r-- 1 root root 134453 May 16 20:51 testr.mp4 [03:55] <c_14> The upeoapp2 user needs to have execute permissions to all the directories above converted and write permissions in converted. And the user will also need write permissions to the output file if it already exists. [03:55] <decci> c_14: This is what it holds as I ran it through root user earlier [03:56] <decci> c_14: But it actually has upeoapp2 permission drwxrwxr-x 2 upeoapp2 upeoapp2 4096 May 16 20:51 converted/ drwxrwxr-x 2 upeoapp2 upeoapp2 4096 May 16 20:50 upload/ [03:56] <c_14> The folder yes, but not to the file and since the file already exists, it can't overwrite it, probably. [03:57] <decci> c_14: Do i just need chown -R upeoapp2:upeoapp2 * under pwd /home/upeoapp2/public_html/ffmpegtest/video [03:57] <decci> c_14: TRied but not working still [03:58] <c_14> same error? [03:58] <decci> Yes [03:59] <c_14> Can that user create files in that folder? try touch /home/upeoapp2/public_html/ffmpegtest/video/converted/test as that user [04:00] <decci> c_14: Yes [04:00] <decci> c_14: touch /home/upeoapp2/public_html/ffmpegtest/video/converted/test [04:00] <decci> c_14: /video/converted]# ls ./ ../ retrer.mp4 test testr.mp4 [04:00] <decci> c_14: I su to upeoapp2 and ran that command [04:01] <decci> c_14: Ohh...I am able to run ffmpeg command manually [04:01] <decci> c_14: http://paste.ubuntu.com/7476019/ [04:02] <decci> c_14: But I wonder if web is using other user [04:02] <sacarasc> It's not upeoapp2 running the command, it's your httpd user. [04:02] <decci> sacarasc: Oops [04:02] <decci> sacarasc: How shall I enable httpd if thats the case [04:03] <sacarasc> Look on `ps aux` for your httpd, check the user, su to that user, check the path. [04:03] <sacarasc> Or try to run the command. [04:04] <decci> sacarasc: http://paste.ubuntu.com/7476025/ [04:04] <sacarasc> So: nobody, I guess. [04:04] <decci> sacarasc: :( [04:04] <sacarasc> The user is called "nobody". [04:05] <decci> sacarasc: ok [04:05] <decci> sacarasc: how to fix it [04:06] <c_14> either chown the dir to user nobody, or find a way to run the httpd/php script as the usomething user [04:06] <decci> sacarasc: Do we need to provide write access to user nobody [04:06] <c_14> that would work too [04:06] <decci> sacarasc: chown -R root:nobody * [04:07] <c_14> You'll have to make sure that the dir has group write permissions [04:08] <decci> sacarasc: drwxrwxr-x 2 upeoapp2 nobody 4096 May 16 20:59 converted/ drwxrwxr-x 2 upeoapp2 nobody 4096 May 16 21:07 upload/ [04:08] <decci> sacarasc: drwxrwxrwx 2 upeoapp2 nobody 4096 May 16 20:59 converted/ drwxrwxrwx 2 upeoapp2 nobody 4096 May 16 21:07 upload/ [04:09] <decci> sacarasc: Still not working [04:12] <decci> sacarasc: I dont see ffmpeg working here [04:12] <clever> you also need +x on every directory above the one your writting to [04:12] <clever> and possibly +r [04:13] <decci> clever: /home/upeoapp2/public_html/ffmpegtest/video/converted is the one i changed the permission [04:13] <decci> clever: Where do i need +x [04:14] <clever> /, /home, /home/upeoapp2, /home/upeoapp2/public_html, /home/upeoapp2/public_html/ffmpegtest, /home/upeoapp2/public_html/ffmpegtest/video [04:14] <clever> all of those [04:14] <clever> if you lack +x at any point, you cant cd into that directory, and it cant go any deeper [04:16] <clever> and if you just give it 777, you leave yourself wide open for various kinds of abuse, dont leave it like that [04:16] <decci> clever: I have made chmod -R +x * below public_html directory [04:17] <decci> clever: but still its not working [04:17] <clever> -R and * shouldnt be mixed [04:17] <clever> -R defeats the entire point of * [04:17] <clever> and * wont get everyhting [04:17] <clever> also, dont just blindly +x everything in sight, test each directory to verify where the issue is [04:18] <clever> as the target user, try to cd into the directory, one level at a time [04:18] <clever> and verify if you can get into converted [04:18] <decci> clever:http://paste.ubuntu.com/7476065/ [04:18] <clever> if you can, then +x is already in place [04:19] <clever> switch over to the upeoapp2 user, and just try to cd /home/upeoapp2/public_html/ffmpegtest/video [04:19] <clever> does that work? [04:20] <decci> clever: yes [04:20] <clever> then you have +x on the required directories already [04:20] <clever> what is the actual error? [04:20] <decci> clever: id uid=504(upeoapp2) gid=502(upeoapp2) groups=502(upeoapp2) upeoapp2(a)upeoapp.com [~/public_html/ffmpegtest/video]# cd /home/upeoapp2/public_html/ffmpegtest/video upeoapp2(a)upeoapp.com [~/public_html/ffmpegtest/video]# [04:21] <decci> clever: Go to http://198.20.89.58/~upeoapp2/ffmpegtest/upload.php [04:21] <clever> does the php script run as upeoapp2 though [04:21] <clever> modify the php script to run id, and paste that output [04:21] <clever> <?php passthru('id'); ?> [04:24] <decci> clever: I added the entry into test.php http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [04:24] <decci> clever: Its running as upeoapp2 user [04:25] <clever> ah, good [04:25] <clever> <?php passthru('stat /home/upeoapp2/public_html/ffmpegtest/video/converted'); ?> [04:25] <clever> try this one [04:26] <decci> clever: http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [04:26] <decci> clever: nothing it shows [04:27] <clever> it may wind up printing errors to the server error log [04:27] <clever> <?php passthru('stat /home/upeoapp2/public_html/ffmpegtest/video/converted 2>&1'); ?> [04:27] <clever> this should redirect errors to the webpage [04:28] <decci> clever: http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [04:28] <decci> clever: sh: stat: command not found [04:28] <clever> yep, may need an absolute path [04:28] <decci> clever: But while I run it manually on console it works [04:28] <clever> /usr/bin/stat [04:28] <clever> PATH wasnt setup fully for the php env [04:29] <decci> clever: whereis stat stat: /usr/bin/stat /usr/share/man/man1/stat.1.gz /usr/share/man/man3p/stat.3p.gz /usr/share/man/man2/stat.2.gz [04:29] <decci> clever: But this fails http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [04:29] <clever> any chance the server is setup with chroot stuff at play? [04:30] <decci> clever: How to ensure that [04:30] <clever> <?php passthru('ls /'); ?> [04:30] <decci> clever: http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [04:30] <decci> clever: Z bin chroot dev etc home lib lib64 mysubdir tmp usr var [04:31] <clever> and do you have the same files when you ls / under a shell? [04:31] <decci> clever: http://paste.ubuntu.com/7476109/ [04:32] <clever> that was not 'ls /' [04:32] <decci> clever: http://paste.ubuntu.com/7476111/ [04:32] <decci> clever: sorry [04:32] <decci> clever: http://paste.ubuntu.com/7476111/ [04:32] <clever> so the answer is no, you dont have Z or mysubdir [04:32] <decci> clever: okay [04:33] <clever> so / under the real system is not the same as / under the php script [04:33] <clever> figure out what / maps to, maybe search for that 'mysubdir' thing, and then repeat everything under that [04:36] <decci> clever: You mean path issue [04:36] <decci> clever: $ffmpeg = trim(shell_exec('type -P ffmpeg')); [04:36] <decci> clever: type -P ffmpeg /usr/local/bin/ffmpeg [04:38] <decci> clever: When I am running this command under upeoapp2 user its working [04:38] <decci> clever: http://paste.ubuntu.com/7476128/ [04:39] <decci> clever: Shall I install ffmpeg under /home/upeoapp2 [04:39] <clever> where is 'mysubdir' on your system? [04:39] <decci> clever: whats that [04:40] <clever> a directory in / under php [04:40] <clever> 2014-05-16 23:33:25 < clever> figure out what / maps to, maybe search for that 'mysubdir' thing, and then repeat everything under that [09:19] <decci> I am able to run which ffmpeg through upeoapp2 user in shell prompt but with PHP script it dint work http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [09:19] <decci> What could be the issue [09:20] <decci> http://paste.ubuntu.com/7476924/ [09:22] <pzich> what do you see when you `which ffmpeg` on your shell? [09:23] <decci> pzich: # id uid=504(upeoapp2) gid=502(upeoapp2) groups=502(upeoapp2) upeoapp2(a)upeoapp.com [/usr/local/bin]# which ffmpeg /usr/local/bin/ffmpeg upeoapp2(a)upeoapp.com [/usr/local/bin]# [09:24] <decci> pzich: Tell me one thing . Is directly logging into as upeoapp2 user same as logging into root > and then su upeoapp2 [09:24] <decci> Looks like permission issue [09:28] <decci> pzich: What u say? [09:29] <pzich> could be permissions, also /usr/local/bin/ isn't in your $_SERVER['PATH'] [09:29] <decci> pzich: echo $PATH /usr/local/jdk/bin:/usr/lib64/qt-3.3/bin:/bin:/usr/bin:/usr/local/bin:/sbin:/usr/sbin:/usr/local/sbin:/usr/local/bin:/usr/X11R6/bin:/root/bin [09:29] <decci> It is [09:29] <pzich> that's your shell [09:29] <pzich> open http://198.20.89.58/~upeoapp2/ffmpegtest/phpinfo.php and search for _SERVER["PATH"] [09:31] <decci> pzich: Oops..how to fix it [09:31] <pzich> well, the easiest thing would be to run ffmpeg by calling /usr/local/bin/ffmpeg directly [09:33] <decci> pzich: What if I want to use from script as shown: //Path To your FFMPEG //$ffmpeg = 'ffmpeg'; // $ffmpeg = trim(shell_exec('which ffmpeg')); // or better yet: $ffmpeg = trim(shell_exec('type -P ffmpeg')); //$ffmpeg = '/home/upeoapp2/bin/ffmpeg'; [09:33] <decci> pzich: I really need this to work [09:33] <decci> pzich: or do i need to install ffmpeg under /usr/bin [09:34] <pzich> that would also work [09:34] <pzich> third option is to add it to your $_SERVER['PATH'], I'm not just not sure if you can just append it in your code [09:34] <decci> pzich: How to add that [09:35] <decci> pzich: I will rather add to the PATH if you can help me how to do it [09:35] <pzich> well, you can try $_SERVER['PATH'] .= ':/usr/local/bin' at the top of your script [09:38] <decci> pzich: You mean like this : <?php 2 $_SERVER['PATH'] .= ':/usr/local/bin' 3 echo 'name => '.$_FILES["upload_file"]['name']." "; 4 echo 'type => ' .$_FILES["upload_file"]['type']." "; 5 echo 'tmp_name => '.$_FILES["upload_file"]['tmp_name']." "; 6 echo 'error => '.$_FILES["upload_file"]['error']." "; 7 echo 'size => '.$_FILES["upload_file"]['size']." "; 8 echo "<hr/>"; [09:39] <pzich> you need to use a paste website [09:39] <decci> pzich: http://paste.ubuntu.com/7476969/ [09:39] <decci> pzich: Do i need to put it within <?php or outside [09:39] <decci> pzich: I see some error after I put it at the top [09:39] <pzich> definitely within if you want it to actualyl work [09:40] <pzich> also maybe on the test.php instead? [09:40] <decci> Parse error: syntax error, unexpected T_ECHO in /home/upeoapp2/public_html/ffmpegtest/upload_processor.php on line 3 [09:40] <pzich> or in addition [09:40] <pzich> well, yeah, you need to add a semicolon... [09:40] <decci> pzich: Just saw that..my bad [09:40] <pzich> how familiar are you with PHP, and coding in general? [09:41] <decci> pzich: Just a tired body [09:41] <decci> pzich: Oops..dint work [09:41] <decci> pzich: http://paste.ubuntu.com/7476975/ [09:42] <decci> pzich: See the second last line...it is not taking /usr/local/bin/ffmpeg [09:42] <decci> pzich: -i /home/upeoapp2/public_html/ffmpegtest/video/upload/apha.AVI -vcodec libx264 -maxrate 1024k -bufsize 1024k -f mp4 -strict -2 -y /home [09:43] <decci> pzich: http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [09:43] <pzich> can you just put "/usr/local/bin/" before the "ffmpeg"? that should make it work, if your PHP user can execute ffmpeg and read and write those file paths [09:43] <decci> pzich: I did try it but not working [09:43] <decci> pzich: It takes it as a string [09:45] <pzich> it should still work, are you sure that user can execute ffmpeg? [09:45] <decci> pzich: Here it shows : id uid=504(upeoapp2) gid=502(upeoapp2) groups=502(upeoapp2) upeoapp2(a)upeoapp.com [~]# which ffmpeg /usr/local/bin/ffmpeg upeoapp2(a)upeoapp.com [~]# [09:46] <decci> pzich: ffmpeg ffmpeg version git-2014-05-16-bac08cf Copyright (c) 2000-2014 the FFmpeg developers built on May 16 2014 07:59:25 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4) configuration: --prefix=/usr/local//ffmpeg_build --extra-cflags=-I/usr/local//ffmpeg_build/include --extra-ldflags=-L/usr/local//ffmpeg_build/lib --bindir=/usr/local//bin --extra-libs=-ldl --enable-gpl [09:46] <decci> pzich: Logged in as upeoapp2 user I can run ffmpeg but thats not working when on PHP [09:49] <decci> pzich: What shall I need to make it run through /usr/bin [09:49] <decci> pzich: Do i need to install it from start or just point it out [09:49] <Reventlov> Anyone for https://trac.ffmpeg.org/ticket/3630 ? [09:53] <pzich> I'm not seeing ffmpeg in your /usr/local/bin [09:54] <decci> pzich: I tried ln -s /usr/local/bin/ffmpeg /usr/bin [09:54] <decci> pzich: But still the problem [10:01] <decci> pzich: Tried putting it under /usr/bin..dint work [10:01] <decci> pzich: http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [10:05] <pzich> do `ls /usr/local/bin/`, where is ffmpeg? [10:10] <decci> pzich: let me add that to test.php script [10:10] <decci> pzich: 1 sec [10:10] <pzich> wait, where is ffmpeg? it's not on that computer [10:11] <pzich> take a look at http://198.20.89.58/~upeoapp2/ffmpegtest/find.php [10:12] <pzich> if you search for "ffmpeg", it's not there [10:13] <decci> http://198.20.89.58/~upeoapp2/ffmpegtest/test.php [10:14] <decci> pzich: Yes [10:14] <decci> pzich: I think thats the issue [10:14] <pzich> no, it absolutely is [10:15] <decci> pzich: So what u say [10:15] <pzich> ffmpeg isn't there, therefore you can't run ffmpeg [10:15] <decci> pzich: But its displayed when I run in shell [10:15] <pzich> are you sure that shell is on the same computer? [10:15] <decci> pzich: ls /usr/local/bin/ ./ findrule* lwp-mirror* phpize* srs* ../ gpgmailtunl* lwp-request* pwhich* srsc* bdf2gdfont.pl* graph-easy* lwp-rget* sa-awl* srsd* chartex* imgsize* oscartest* sa-check_spamd* stty.pl* cpandb* ip2cc* pear* sa-compile* tpage* crc32* ipcount* peardev* sa-learn* ttree* d [10:15] <decci> pzich: Yes [10:16] <pzich> well, search for /usr/local/bin/ on http://198.20.89.58/~upeoapp2/ffmpegtest/find.php , you can see it's not there [10:17] <decci> pzich: How did youi get this ile [10:17] <decci> file [10:17] <pzich> it's just there, just like http://198.20.89.58/~upeoapp2/ffmpegtest/find-usr.php [10:18] <decci> pzich: hmm [10:18] <decci> pzich: Does it mean its not accessible by upeoapp2 user [10:18] <decci> pzich: Its strange [10:18] <pzich> well it should at least be able to see the folder [10:19] <decci> pzich: What does ffmpeg* mean [10:19] <decci> pzich: Linked? [10:19] <pzich> I think executable in this case [10:19] <decci> pzich: -rwxr-xr-x 1 upeoapp2 upeoapp2 19581640 May 16 07:59 ffmpeg* [10:19] <pzich> try chmod 777 /usr/local/bin/ffmpeg [10:19] <pzich> hmm, that should be ok [10:21] <decci> pzich: No not working still [10:21] <pzich> well PHP still can't see it [10:21] <decci> pzich: chmod 777 /usr/local/bin/ffmpeg root@serv01 [~]# chmod 777 /usr/bin/ffmpeg [10:21] <decci> pzich: I am frustrated [10:22] <decci> pzich: What to do [10:22] <decci> pzich: I ran ln -s /usr/local/bin/ffmpeg /usr/bin/ffmpeg.. [10:22] <decci> pzich: Will it affect [10:23] <decci> pzich: I ran it 10 min back [10:24] <decci> pzich: [/usr/bin]# ls -la ffmpeg lrwxrwxrwx 1 root root 21 May 17 02:52 ffmpeg -> /usr/local/bin/ffmpeg*c [10:25] <decci> pzich: Wil this affect [10:34] <pzich> here, look at this: http://198.20.89.58/~upeoapp2/ffmpegtest/test2.php [10:34] <pzich> it's not seeing it in /usr/local/bin, but it is seeing it in ../../bin [10:34] <pzich> and it shows as being executable for everyone [10:35] <pzich> but when you run it with -version it still doesn't wokr [10:36] <decci> pzich: So what could be the issue [10:36] <decci> pzich: Is it server issue [10:36] <pzich> well, PHP probably can't access the actual executable, probably because of chroot or something [10:37] <decci> pzich: Will RPM based installation will fix this [10:37] <pzich> no idea [10:37] <pzich> but maybe try cp /usr/local/bin/ffmpeg /home/upeoapp2/public_html/ [10:38] <decci> pzich: Lemme try [10:40] <decci> pzich: Not workig still [10:43] <decci> pzich: What to do [10:44] <pzich> hang on, I'm trying to fix it [10:45] <decci> pzich: k [10:47] <decci> pzich: How are you creating those files? [10:48] <pzich> which files? [10:48] <decci> pzich: test2.php [10:48] <pzich> that wasn't just there? [10:48] <decci> pzich: ok [10:50] <decci> pzich: carry on [10:50] <pzich> so, I think the problem is that PHP still isn't allowed to execute ffmpeg for some reason [10:50] <decci> pzich: It loook like so [10:51] <decci> pzich: I wonder what might be preventing it [10:52] <pzich> take a look at http://198.20.89.58/~upeoapp2/ffmpegtest/ls.php [12:42] <t4nk015> flame me back if necessary... but what the is the deal with removing drm on m4v's from the iTunes store... obviously some apps out in the world do it, costing a pretty penny, I just want to be able to play stuff on my tv and not my laptop, hell it's not like I couldn't find a copy on a torrent, but I really do like well encoded video and I don't mind paying for it, but not being able to watch it on my TV, or 30" Apple Cinema Display th [15:41] <bencc> what file can save vp8 packets? [16:03] <t4nk015> test [16:07] <diesel420> hi, i try to capture a dvb-t stream via mencoder so i receive a growing *ts file, i want to stream it live via network with ffmpeg. i had problesm so i tried to stream it "local" first. here is my ffmpeg command and output: [16:08] <diesel420> http://pastebin.com/skxiSizd [16:09] <diesel420> i trird to open int with mpv and vlc, but it seems it does not work, i got no video and no audio either [17:01] <c_14> diesel420: your command works just fine for me [18:25] <SuperNoob> hi is it possible to concatenate videos with different nitrates? [18:25] <SuperNoob> bitrates? [18:25] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [18:25] <SuperNoob> thanks c_14 I've read that but I am confuse [18:25] <c_14> What about? [18:27] <SuperNoob> so I have a video I made out of After Effects rendered and set FPS at 60, I then have a video I captured on my screen using Quicktime I am trying to join them but I always get an issue with the audio the recorded video when joined produces no sound, out of sync etc. [18:28] <SuperNoob> I have checked the codecs and they are the same [18:28] <SuperNoob> bitrate for audio's are the same [18:28] <SuperNoob> Is there any doc I can read about Bit Rates? both Audio and Video? [18:29] <SuperNoob> I will do that [18:29] <SuperNoob> give me a minute [18:29] <c_14> The bitrate is just how many bits of data you have every second, there's not a whole lot to know. [18:30] <SuperNoob> I will show what media info says about the video I am trying to concat [18:32] <SuperNoob> what is the significance of sample rate in both Audio and Video? [18:32] <SuperNoob> can there be issues with joining if the input has different sample rate? [18:33] <c_14> It might be, but you should be able to downsample one of the audio tracks so that both have the same sample rate. [18:37] <SuperNoob> I am joining mov files any clue what -bsf:a aac_adtstoasc is for? is that a codec or something? I can't find details about it on the web [18:39] <c_14> https://www.ffmpeg.org/ffmpeg-bitstream-filters.html#aac_005fadtstoasc [18:42] <SuperNoob> ? [18:42] <SuperNoob> hi again here's the video I am trying to join http://pastebin.com/rPvgDKvC [18:44] <c_14> Frog what I can see, you should be able to merge them, can I see the ffmpeg command you tried using? [18:46] <SuperNoob> ffmpeg -f concat -i mylist.txt -c copy output [18:47] <SuperNoob> ffmpeg -f concat -i mylist.txt -c copy final.mov [18:48] <c_14> you might want to try the concat filter instead of the concat demuxer [18:50] <SuperNoob> what's the significance of doing that? if the codec, bit rate etc. are the the same? [18:59] <c_14> It gives you more control of what's happening. [20:06] <diesel420> c_14, thanks, yeah i needed just to install some video codecs or such =) [23:30] <chealy> Hi all, is this an appropriate place to discuss usage of libavformat functionality for an application I am writing? [23:31] <chealy> I'm trying to stream transport stream data using av_init_packet and av_write_frame as RTP and I'm having some difficulties. [00:00] --- Sun May 18 2014

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ffmpeg-devel.log.20140516
by burek 17 May '14

17 May '14

[00:08] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:bebce653e560: avcodec/x86/dsputil_mmx: Fix build with clang-usan [00:44] <cone-725> ffmpeg.git 03Carl Eugen Hoyos 07master:30aeab29e468: Use av_restrict instead of restrict in libavcodec/opus_celt.c. [03:19] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:a071c0b515e5: avcodec/utils: Fix undefined behavior in avpriv_toupper4 [03:19] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:1fb46858c249: avcodec/cavs: prevent out of array read [05:23] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:7970959a9412: avcodec/opusdec: fix some const correctness [05:43] <cone-725> ffmpeg.git 03James Almer 07master:cdff1cc98f3b: configure: add check for AVX inline support [05:43] <cone-725> ffmpeg.git 03James Almer 07master:a9bf713d350a: swresample: add swri_resample_float_avx [06:58] <cone-725> ffmpeg.git 03Masa Utashiro 07master:491d602283cd: avfilter/vignette: fix input frame memory leak. [07:13] <cone-725> ffmpeg.git 03Clément BSsch 07master:8a0ba44e1ac1: vignette: reuse already declared ctx. [07:13] <cone-725> ffmpeg.git 03Clément BSsch 07master:ed7bc5471b63: avfilter/vignette: add direct rendering. [07:13] <cone-725> ffmpeg.git 03Clément BSsch 07master:bac08cf086bb: avfilter/vignette: reindent after previous commit. [13:44] <BBB> really, inline avx? [13:44] <BBB> ;( [13:44] <BBB> I mean [13:44] <BBB> I'm going to cry [13:44] <BBB> can we please not do that [13:45] <nevcairiel> its too late [13:49] <BBB> let's throw a hassle on the ML [13:50] <nevcairiel> for the sse case of the same function it was apparently slower to do it in yasm [14:02] <plepere> BBB, nevcairiel I've submitted the DBF patch [14:02] <BBB> I noticed, I didn't look at it yet but I saw the earlier and it mostly looked ok [14:03] <BBB> there's some odd stuff in there (like calling beta/tsc no_q and q or so) [14:03] <BBB> but these are minor documentation things rather than anything else [14:03] <BBB> plepere: one thing that would be nice is if you could make register usage more contrained [14:03] <BBB> you're very quickly going from 6 to 15 registers from chroma to luma [14:03] <BBB> I'm not sure you need that many [14:03] <BBB> (GPRs, not XMMs) [14:04] <plepere> it's not my code, but ok. :p [14:04] <BBB> plepere: also, using named registers would be very cool (global function_name, n_args, n_gprs, n_xmms, arg1, arg2, arg3, arg4 [14:04] <plepere> I was hoping to get on the idct if this patch was accepted. [14:04] <BBB> so you can do [arg1q] [14:04] <BBB> I'm ok with it for now, mark them as FIXMEs so we can get back to them later [14:04] <plepere> BBB : I'm using that already [14:05] <BBB> you're using a lot of r1d, r1, etc. [14:05] <plepere> ah only in luma maybe [14:05] <plepere> ok ok, I'll do that. [14:05] <BBB> 15 gprs is also a little much, I don't think you use all 15 in practice [14:05] <BBB> it's minor, mark as FIXME to fix later [14:05] <BBB> but do mark it as FIXME because it's kinda silly [14:06] <plepere> I'll do that now [14:06] <BBB> t [14:06] <BBB> oops [14:06] <BBB> ty [14:06] <plepere> by the way, is there an easy way to check if a function has only SSE2 or SSSE3 instructions ? [14:07] <plepere> except checking each instruction [14:07] <BBB> you can run under qemu [14:07] <BBB> but that's kinda a lot of work [14:07] <BBB> so I don't know of an easy way [14:07] <plepere> ok, thanks anyways [14:07] <BBB> I typically just mark it as what I think it is or what the goal was for it to be [14:08] <BBB> if it's wrong, people will file bugs [14:08] <plepere> ok, I'll leave as it is then. [14:09] <plepere> I'm doing the named registers, try to limit the number of register usage in luma and then I'll work on the idct [14:09] <BBB> yay [14:09] <BBB> ty [14:09] <plepere> and then I'll be able to work on AVX2. That's what I'm looking forward to. [14:12] <kurosu> some perl wizardry might be able to catch most cases by checking forbidden insn [14:12] <kurosu> though there's the issue of insn forbidden with particular operands (like movd) [14:13] <kurosu> might also need to operate on the postprocessed or disassembled code [14:14] <plepere> kurosu, I'm working on assembly, so I could have a list of blacklisted instructions. that's a nice idea [14:17] <kurosu> I don't have a list, but your luma deblock code looked like it was ok once you had macro'ed pabsw [14:17] <kurosu> http://en.wikipedia.org/wiki/SSSE3#New_Instructions <- integer insn introduced by SSSE3 [14:17] <plepere> kurosu, maybe, but since I'm not sure, I prefer leaving this as it is.: [14:19] <kurosu> http://ref.x86asm.net/coder64.html <- a lot of info, among which the required info [14:20] <kurosu> nevermind, it lacks some ssse3 insn [14:21] <kurosu> http://ref.x86asm.net/coder.html <- better [14:23] <plepere> great, bookmarked. [14:23] <plepere> :) [14:23] <plepere> thanks [14:27] <plepere> BBB : you were right : I could even squeeze the chroma from 6 registers to 5. :p [14:27] <kurosu> plepere, although I guess you do sometimes, my suggestion is to look at the disassembled code for code with lots of macros [14:28] <kurosu> it's easy to miss obvious things, like m6 and m8 being used but not m7 [14:29] <plepere> I tend to look at pre-processor code rather than the disassembled one [14:36] Action: Skyler_ peeks in and sees asm [15:52] <Voicu> whoever designed the java API must have been drunk and/or high [15:52] <Voicu> there is no way a sane person would unleash something like that on a fellow programmer [23:39] <michaelni> what happened with cone ? [00:00] --- Sat May 17 2014

1 0

ffmpeg.log.20140516
by burek 17 May '14

17 May '14

[02:11] <AaronB_> Hey, is there anyone around that can spend a few minutes with me to help me understand differences in formats and codecs, and how to compile ffmpeg with support for as many as possible? This is for a debian system. [02:12] <satanist> you can try this repo git://anonscm.debian.org/collab-maint/ffmpeg.git [02:14] <AaronB_> Is there a site that links to that repo with compile instructions? I'm not really familiar with compiling, im a web dev [02:14] <satanist> git-buildpackage -us -uc -S [02:15] <AaronB_> I have already followed this but it doesnt support everything https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [02:17] <satanist> you can see the config options in the debian/rules file [02:19] <AaronB_> Anything that needs changed in the one you linked? [02:20] <satanist> yes in the debian/*-ffmpeg-dev.install you have to rename the pc-files form *.pc to *-ffmpeg.pc [02:23] <AaronB_> Ok, im firing up a VM now to start playing with it, will you be around for awhile to ask questions if needed? [02:24] <satanist> no, I will go sleep now [02:24] <satanist> sorry [02:24] <AaronB_> Alright, later mate [02:24] <AaronB_> Thanks [05:52] <hendry> i want to make a video of a long time series I captured. Basically I want each frame to be used as a video. so 24 captures = 1 sec of video. How do I do that? All the examples I see have a duration of 5 secs per frame [06:22] <musashi1> Hello, I used ffmpeg to create a video which doesn't seem to want to play in a browser. Could anyone reccomend a course of action to determine why it isn't playing? [06:56] <decci> I am trying to follow https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [06:56] <decci> I encounter this error while running ./configure --prefix="$HOME/ffmpeg_build" --extra-cflags="-I$HOME/ffmpeg_build/include" --extra-ldflags="-L$HOME/ffmpeg_build/lib" --bindir="$HOME/bin" --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 [06:56] <decci> Unable to create and execute files in /tmp. Set the TMPDIR environment variable to another directory and make sure that it is not mounted noexec. Sanity test failed. If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by confi [06:57] <decci> Any idea what could be the issue [06:57] <decci> http://paste.ubuntu.com/7471297/ [06:58] <decci> http://paste.ubuntu.com/7471298/ [09:04] <bencc> is it reasonable to do real time video transcoding on a 20$ VPS? (linode) [09:05] <another> depends on the resolution and such [09:06] <bencc> 1000 X 560 [09:06] <bencc> VP8 to h264 [11:37] <supernoob> hi [11:38] <supernoob> How do I concatenate mov files with different bitrates? [11:39] <supernoob> I've tried http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20medi… but it is not working for me [11:43] <supernoob> here's my Media Info Video 1 [11:44] <supernoob> Format : MPEG-4 Format profile : QuickTime Codec ID : qt File size : 165 MiB Duration : 12mn 12s Overall bit rate : 1 892 Kbps Writing application : Lavf55.25.101 Video ID : 1 Format [12:03] <dloo> hello [12:04] <dloo> how can i, in the ffmpeg cli, indicate that i want the 2nd input of my video card to record? [12:05] <dloo> (i got a black screen when recording on v4l2 -i /dev/video0 -vc1 [12:06] <anshul> dloo, can you pastebin ls /dev/video* [12:31] <dloo> ls /dev/video0 [12:31] <dloo> /dev/video0 [12:32] <dloo> right card, but not right video input [12:34] <anshul> dloo, have you tried same with any other player like vlc [13:07] <dloo> anshul: yes, and it is working; with ffplay too [13:13] <mrskman> Hi, is it bug or feature that this command never stops encoding? http://pastebin.com/dJpuiZND I guess that even with -shortest option it cannot detect end of input stream when there is an overlay? [13:17] <decci> I need a help regarding ffmpeg [13:18] <decci> I have a PHP script which uploads a media file and convert it to different format through ffmpeg. I installed ffmpeg under /root/bin/ffmpeg. [13:18] <decci> http://198.20.89.58/~upeoapp2/ffmpegtest/upload.php [13:19] <decci> While you upload any media, it tries to find out ffmpeg as shown ame => MOV06754.AVI type => video/avi tmp_name => /chroot/tmp/phpAtmHIR error => 0 size => 830900 Temp File: /chroot/tmp/phpAtmHIR exists. Uploaded File: /home/upeoapp2/public_html/ffmpegtest/video/upload/retrer.AVI -i /home/upeoapp2/public_html/ffmpegtest/video/upload/retrer.AVI -vcodec libx264 -maxrate 1024k -bufsize 1024k -f mp4 -strict -2 -y /home/upeoapp2/public_ht [13:19] <decci> The problem is it doesnt detect ffmpeg [13:19] <decci> I used this tutorial https://trac.ffmpeg.org/wiki/CentosCompilationGuide [13:19] <sacarasc> Well, /root is usually root readable only. [13:20] <sacarasc> And HTTPds most often don't run as root. [13:21] <decci> sacarasc: can you elaborate [13:21] <decci> sacarasc: Where shall I install ffmpeg [13:21] <decci> sacarasc: to make it run [13:22] <decci> sacarasc: "type -P ffmpeg" doesn't return where ffmpeg is [4:41:01 PM] Suman Ghosh: $ffmpeg = trim(shell_exec('type -P ffmpeg')); $conversion_command = $ffmpeg; [13:22] <sacarasc> /usr/local is usually a good place for your prefix. But whereever you put it, make sure the httpd's user has it in their $PATH. [13:22] <decci> How to ensure about that httpd user has its access [13:23] <sacarasc> Or, you could edit the script to use the full path, instead. [13:25] <mrskman> $ffmpeg = '/root/bin/ffmpeg'; doesn't help? [13:27] <decci> mrskman: No ffmpeg is not available from under /home/upeoapp2/public_html/ffmpegtest/ [13:28] <decci> mrskman: My script is not able to find ffmepg [13:28] <decci> ffmpeg [13:31] <sacarasc> mrskman is saying to edit the script, change trim() to just '/root/bin/ffmpeg' [13:31] <sacarasc> And then realise that you shouldn't install things in /root. [13:38] <Hello71> ... if you have access to /root [13:38] <Hello71> you can write *anywhere on the system* [13:38] <Hello71> what the hell are you doing [13:45] <decci> Hello71: Hi [13:45] <decci> Hello71: What u suggest [13:45] <decci> Hello71: Moving it to /home directory or /usr/local [13:45] <Hello71> read fhs [13:46] <decci> Hello71: If I install a new ffmpeg under /usr/local , how its going to take from that location [13:46] <decci> Hello71: Do I need to specify it [13:51] <decci> ? [13:55] <sacarasc> Hello71: httpds don't run as /root, though, and that's the user which is doing it. [14:11] <danylevskyi> Hi guys! I am trying to record my screen with avconv. It's great! But my audio has some arbitrary gaps. When I'm recording only audio it records fine. Where can be the problem? [14:43] <c_14> danylevskyi: avconv is part of libav, see #libav for support or install ffmpeg [16:15] <hendry> everytime i try convert images to a mp4, ffmpeg gets killed... any ideas? http://ix.io/cmF [16:19] <c_14> Do you have enough free ram? Tried checking syslog at the time it breaks? [16:39] <danylevskyi> Folks, I'd like to thank you for great ffmpeg tool! [16:40] <danylevskyi> Installed ffmpeg and all problems with sound have gone. avconv is really buggy. [18:07] <mrskman> Hi, is it bug or feature that this command never stops encoding? http://pastebin.com/dJpuiZND I guess that even with -shortest option it cannot detect end of input stream when there is an overlay? [18:22] <t4nk761> anybody tried coding the librtmp + avcodec? [18:23] <abhilashswe> i am writing a rtmp publish app using avcodec + librtmp [18:23] <abhilashswe> i request some help to start the coding [18:26] <abhilashswe> if anybody is interested in freelancing we can hire [18:33] <haspor> who knows how to loop atrac3p correctly with ffmpeg ? [18:34] <JEEB> what do you mean with that exactly? [18:34] <haspor> i mean that, how do i know when to make it seek back to loop point and how to get the loop point [18:35] <JEEB> that sounds like stuff that isn't in the actual audio stream [18:35] <JEEB> but in the container if even there [18:35] <JEEB> I wouldn't be surprised if it was in an unrelated thing that the app playing it would know :P [18:37] <haspor> i have figured out the offset in the header where the looppoint is but since i cant find any related documents and ffmpeg is not able to do the looping itself, i wanted to ask you guys [18:37] <haspor> sorry for being a bit confusing [18:37] <haspor> i tried various hacks but there is always something that is not ok [18:41] <JEEB> haspor, well libav* will not do that for you (it only does A->B stuff unless you seek yourself), so you'll have to seek to the looping part's start and start decoding from there again [18:42] <haspor> yep, thats what i do [18:43] <c_14> mrskman: overlay=y=70:eof_action=endall seems to do the trick. [18:44] <haspor> but there is always a tiny hiccup before the loop starts again cos sometimes those at3 files have nearly empty data at the end, so im not sure should i skip it always or only for specific files [18:46] <JEEB> compare to a reference decoder I guess, unless the file contains info? or if it's leftovers in some buffer, you could try to poke to that stuff out :P [18:47] <abhilashswe> hi JEEB [18:49] <haspor> it would help if i could know what those bytes mean in every packet header [18:55] <mrskman> c_14: thanks :-) [19:05] <dloo> hello [19:05] <dloo> did someone can help me to open my pinnacle card in ffmpeg, using the second input? (composite) [19:19] <brixon> gm [19:20] <brixon> can i send query params when publishing using rtmp using ffmpeg? [19:20] <brixon> for instance rtmp://myserver:1935/MyApp/small?param=1&param=2 [19:20] <brixon> ? [19:20] <dloo> what is the param to change the video input in v4l2 /dev/video0? [19:20] <brixon> it doesn't seem to work [19:35] <JEEB> haspor, you can look at http://wiki.multimedia.cx/index.php?title=ATRAC3plus [19:36] <JEEB> if there's not enough info feel free to add what additional things you know (either in the format's page or the container's) [19:36] <haspor> jeeb, thanks but checked that already [19:36] <haspor> :( [19:37] <AaronB_> Hey, is anyone able to make custom build scripts for debian with support for all codecs and formats? I can pay for the script or your time helping me do it. [19:38] <JEEB> AaronB_, personally I'd say that enabling all the things isn't exactly that useful. It's better to just set up something that fits all of your use cases since generally it's the input side that needs to be wider, and 99% of all input things are in lavf/lavc itself (and thus don't need anything extra) [19:39] <AaronB_> Agreed, input is most important, i mean "all" losely as i realize a lot of outdated or not used [19:39] <dloo> ffmpeg -f v4l2 -vc 2 -tvstd SECAM -s 768x576 -i /dev/video0 -ac 2 -ar 48000 -f alsa -i hw:0,0 -acodec ac3 -qmax 1 -y test.mov -t 4000 -vcodec libx264 [19:40] <AaronB_> JEEB ive already followed this guide https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [19:40] <AaronB_> and added h265 [19:40] <dloo> why this does not record my composite input on /dev/video0? get the sound from alsa but a black screen :-/ [19:40] <JEEB> AaronB_, then I recommend you check what things you need that you need to actually enable [19:41] <AaronB_> JEEB, I don't actually know what people will want to give me to be honest, I need to support anything that comes my way without fail [19:41] <JEEB> if it's files then you've pretty much got it all :P [19:42] <AaronB_> Does that guide contain libavformat? I dont see it anywhere [19:42] <AaronB_> JEEB this is my supported formats, it looks lacking http://codep.in/6G [19:42] <AaronB_> I expect D & E for every single one [19:42] <JEEB> uhh, that won't happen [19:43] <JEEB> I mean, there's plenty of formats like game stuff, that will never get a muxer [19:43] <JEEB> that looks completely normal to be honest :P [19:43] <AaronB_> Ok maybe not every one, but it still feels lacking [19:43] <AaronB_> like 3gp is only E [19:44] <AaronB_> I should be able to decode it too right? [19:44] <JEEB> 3gp is a special case of mov [19:44] <AaronB_> So it is supported? [19:44] <JEEB> and yes, muxer only because the mov demuxer handles it [19:45] <JEEB> basically you will gain pretty much zero demuxers with any external libraries, and I'm not even sure if you gain any by even doing --enable-gpl [19:45] <JEEB> so yeah [19:46] <JEEB> unless you specifically disabled something, you should have it all [19:46] <AaronB_> hmm ok then, i did have some issues in my initial testing though [19:46] <AaronB_> actually let me setup my test again so i can tell you accurate information, will you be around for a bit JEEB? [19:53] <AaronB_> I get an error: [Error: format mkv not available] when supplying mkv as the imput format of my video, I realize the extension does not = format, but what is the best way to get the format of my file to correctly pass it to my command? [19:53] <JEEB> by not passing it to the command? [19:53] <JEEB> unless your input has no file name or descriptor of any sort [19:53] <JEEB> (there's also byte-based probing) [19:54] <JEEB> if all that fails, /then/ you set your input format [19:54] <AaronB_> So passing it is never needed generally? [19:54] <JEEB> yup [19:54] <AaronB_> What about output format? [19:54] <JEEB> do you have an output file name? [19:54] <JEEB> or descriptor with an extension? [19:54] <AaronB_> say test.mp4 [19:54] <JEEB> then no [19:55] <AaronB_> That could be h264 or h265 [19:55] <AaronB_> sorry those are different things, ignore that.. [19:55] <JEEB> well, yeah... "format" in the context of ffmpeg is generally a container [19:55] <JEEB> (or well, whatever a video/audio/multiple streams gets written into) [19:56] <AaronB_> so its safe to not pass either input or output format? [19:56] <JEEB> if you have a file name, yes. and even if you don't have a file name there's a high probability that you will have it work [19:57] <AaronB_> Ok, in that case, is there a way to show supported output extensions? [19:58] <JEEB> no easy way, no [19:59] <AaronB_> what about supplying a filename without an extension to output too and giving it an output format, will it set the extension for me? [19:59] <JEEB> no [19:59] <JEEB> output file name is the output file name [20:00] <JEEB> if you do -f nut i_hate_chicken you will have a NUT file with exactly that name [20:01] <AaronB_> hmm so there is no easy way to go about making sure something is outputted correctly [20:01] <JEEB> just name it correctly [20:01] <JEEB> if ffmpeg doesn't know the extension, it will error [20:01] <AaronB_> Im giving control to clients that dont know what their doing lol [20:06] <AaronB_> Is this a big enough list of expected formats to support: mp4, ogg, webm, avi, mov? [20:06] <JEEB> sounds good to me [20:06] <JEEB> just collect some commonly used formats in those and limit the users to sane things with them, too [20:07] <JEEB> i mean video/audio formats [20:07] <JEEB> not the ffmpeg formats >_> [20:08] <AaronB_> is wmv supported? [20:08] <JEEB> yes, the real name of that container is ASF [20:08] <JEEB> and extension can be wmv [20:08] <JEEB> but I shall warn you, the encoders for WMV/WMA are /bad/ [20:08] <JEEB> not libavcodec vorbis bad [20:09] <JEEB> but bad [20:09] <AaronB_> Yea i read that supplying bitrate helps at least [20:09] <AaronB_> I wouldnt use it but i gotta support the case of someone doing it >< [20:09] <JEEB> yeah, just make the bit rates crazy high for ti [20:09] <JEEB> *it [20:10] <AaronB_> .webm is the correct extension for a vp8 video right? [20:11] <JEEB> ye [20:11] <AaronB_> Alrighty, this logic is going to be ridiculous to make sure this stuff works >< haha [20:12] <AaronB_> thanks for the help, lets see how far I get with this.. [20:13] <AaronB_> Oo last question, audio. so far I'm supporting aac,ogg,mp3, any more I can expect? [20:14] <JEEB> that should fit most use cases [20:14] <JEEB> also with ogg you mean vorbis [20:14] <JEEB> because ogg is the container thingamabob [20:14] <AaronB_> is ogg the extension? [20:14] <AaronB_> and what about an aac file, whats the extension on taht? [20:14] <dloo> someone can help me with my input v4l2 problem? [20:15] <JEEB> if you want audio-only AAC that most things that support AAC can play, I'd say mp4, but because some players are stupid just set it to m4a [20:15] <JEEB> same mov-based thing, but apple's extension for "audio-only such files" [20:16] <AaronB_> Let me test that real quick, I remember that not working for me.. [20:19] <AaronB_> hmm I lied, maybe I was supplying the output format = extension when that was failing [20:20] <AaronB_> Ok well this helps me a lot, thanks JEEB [20:21] <dloo> what -vc option is supposed to do? [20:21] <AaronB_> dloo sets video codec I believe [20:21] <AaronB_> I think it should be -c:v though [20:22] <dloo> hmm [20:22] <AaronB_> either might work [20:22] <dloo> what isthe optionto change the video channel input? (composite/svideo/tv) [20:26] <AaronB_> this should help dloo https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [20:47] <AaronB_> What are the video profiles available to h265? h264 supported baseline, main, and high (AFAIK) [20:47] <AaronB_> And levels also? [20:49] <JEEB> main and main10 profiles are currently standardized, and you don't really get the latter unless you code 10bit H.265 :P [20:50] <JEEB> main10 does support 8bit HEVC, but no-one marks 8bit stuff as main 10 profile [20:50] <JEEB> for ObviousReasons [20:50] <JEEB> levels you can check from the spec :P [20:50] <AaronB_> is baseline supported? or just main? [20:50] <AaronB_> baseline is best for greater cross device support right? [20:51] <JEEB> > implying that there is a baseline profile in HEVC [20:51] <JEEB> HEVC is main, main 10, main still [20:51] <JEEB> and all kinds of 4:2:2/4:4:4 profiles and >10bit profiles coming when the range extensions get ratified [20:52] <AaronB_> il just stick with main :) [20:52] <JEEB> and usually when you need to set a level, you are adhering to some hardware thingamabob's limitations. And those thingamabobs aren't on the market yet :) [20:53] <AaronB_> gotcha, so no need to set that? [20:53] <JEEB> basically just keep the output 8bit and you will get main profile, and let libx265 set the level [20:53] <AaronB_> ok [20:54] <JEEB> many people seem to think that there is a "high profile" in HEVC, but you're the first one who thought about baseline [20:54] <JEEB> (high profile was a thing that was added two years down the line in order to improve higher resolution content compression with AVC) [20:55] <AaronB_> wiki seems to think a high does exist too [20:55] <JEEB> AVC, yes [20:55] <JEEB> HEVC nope [20:55] <JEEB> HEVC is just main, main 10 and main still profiles [20:55] <AaronB_> grr too many options lol [20:56] <AaronB_> I could spend years just writing validation for this haha [20:56] <AaronB_> Screw it, contact support if you break it :P [20:57] <dloo> -map is not the right thing :( [20:59] <AaronB_> hmm i think i have libvorbis and libtheora mixed up.. [20:59] <AaronB_> vorbis = ogg, theora is? [20:59] <Plorkyeran> ogg = container [20:59] <Plorkyeran> vorbis = audio codec [20:59] <Plorkyeran> theora = video codec [21:00] <JEEB> <JEEB> also with ogg you mean vorbis [21:00] <JEEB> <JEEB> because ogg is the container thingamabob [21:00] <JEEB> :D [21:00] <AaronB_> gotcha, is ogg the common container for those? [21:00] <JEEB> yes [21:00] <JEEB> xiph formats generally go into OGG [21:00] <JEEB> one way or another [21:01] <AaronB_> hmm ok, lets see if i have these correct, i feel like im mixing up codecs and formats too much [21:02] <AaronB_> these are my supported video codecs that ive listed so far: "h265","h264","vpx","xvid","mpeg4","theora" [21:02] <AaronB_> ive shortend them for ease of use for the user, i convert them later [21:02] <JEEB> the name for the first one is hevc in libavcoec [21:02] <JEEB> *libavcodec [21:02] <AaronB_> not libh265? [21:03] <AaronB_> i think im going to leave it as that for the user to enter [21:03] <JEEB> libx265 is x265's library's name [21:03] <JEEB> just like libx264 is x264's [21:03] <AaronB_> when i use -c:v, libxh265 if correct for that value? [21:03] <JEEB> yes, for the encoder [21:03] <AaronB_> ok [21:03] <JEEB> although probably hevc would work too? [21:04] <JEEB> because nowadays ffmpeg can map the video format's name into an encoder for it in many cases [21:04] <JEEB> just like doing -c:v h264 for an encoder nowadays might get you libx264 [21:04] <JEEB> it's a weird world nowadays [21:06] <Venti> there is a high tier in hevc, but that relates to profile level restrictions [21:06] <AaronB_> should i have the user enter theora? i dont commonly hear that word.. i feel like entering ogg for that format would be better [21:06] <JEEB> yes, tiers are tiers [21:06] <AaronB_> though incorrect i know [21:06] <JEEB> AaronB_, if someone wants theora they probably know that name [21:06] <AaronB_> true [21:06] <JEEB> because only freetards do want it at times [21:06] <AaronB_> hah [21:06] <JEEB> the less knowledge'able freetards use vp8 [21:07] <JEEB> (because it got publicity more lately) [21:07] <AaronB_> should i leave it as vpx or set it to vp8? [21:07] <JEEB> libvpx is the encoder, so for that it's correct [21:07] <AaronB_> ok, audio codecs i have so far: "aac","vorbis","mp3" [21:08] <AaronB_> i dont like that vorbis is longer than the rest lol [21:08] <JEEB> you want to use fdk-aac for aac, libvorbis (preferably aotuv's fork) for vorbis and libmp3lame for mp3 [21:08] <AaronB_> thats what i convert those too [21:08] <AaronB_> for the user are those ok? [21:09] <JEEB> although the actual decoder (and thus the actual audio format names) names might be mappable to the "best" encoder for that format when encoding [21:09] <JEEB> and yes, those are OK [21:11] <AaronB_> does webm have its own audio format or is it just aac? [21:11] <JEEB> webm is a limited subset of matroska [21:11] <JEEB> and it means vp8 and vp9 for video, and vorbis and opus for audio [21:11] <AaronB_> Sorry, vpx [21:12] <JEEB> latter ones aren't yet on the spec pages so YMMV (although they are used by some) [21:12] <AaronB_> what container is opus in generally? [21:12] <AaronB_> that isnt one ive added yet, though its in my build [21:12] <JEEB> ogg, matroska [21:13] <AaronB_> clear [21:13] <AaronB_> oops lol [21:13] <AaronB_> you arent my terminal [21:14] <AaronB_> crap, h265 didnt work D: [21:14] <JEEB> yes, h265 is called hevc in ffmpeg space :P [21:15] <JEEB> because that's what the devs of the decoder liked and thus it became the name of the format in lavc [21:15] <AaronB_> sorry i mean it didnt compile [21:15] <JEEB> and ITU-T can suck their unpronounce'able eich-two-six-five [21:15] <AaronB_> and my ffmpeg is broke ;( [21:15] <JEEB> oh, wait. eich-dot-two-six-five [21:15] <JEEB> sorry ITU-T [21:16] <AaronB_> ./ffmpeg: error while loading shared libraries: libx265.so.18: cannot open shared object file: No such file or directory [21:16] <JEEB> you didn't do the ldconfig dance [21:17] <JEEB> and/or you decided to use shared libraries with stuff that doesn't get installed into LD_LIBRARY_PATH [21:17] <JEEB> in which case you can blame yourself [21:17] <AaronB_> Well yea i expected user error was the issue :P [21:17] <AaronB_> hm [21:18] <AaronB_> I did this http://codep.in/6H [21:18] <AaronB_> it built and installed correctly [21:19] <JEEB> well, run ldconfig as root and hope you set a prefix so that LD_LIBRARY_PATH actually grabs them [21:20] <JEEB> otherwise just re-compile x265 with a static library only, and link against it [21:21] <JEEB> basically linux only "notices" a shared library after it has been "indexed", which is what ldconfig does. Except that only works when that shared library is in a prefix that gets automagically checked for [21:21] <AaronB_> after ldconfig do i rebuild ffmpeg or should it work now? [21:21] <JEEB> if you had installed it into a prefix that is auto-checked, it should work [21:22] <AaronB_> yay looks like that did it [21:50] <AaronB_> Are speed presets (like veryfast, fast, normal, etc) available for all codecs or just h264? [21:50] <AaronB_> and are those available in h265 as well? [21:51] <JEEB> speed presets are something that the encoder library provides, they are not in ffmpeg itself [21:51] <JEEB> and yes, libx264 tries to implement something similar [21:52] <AaronB_> How do i find out which encoders support it? [21:52] <AaronB_> only x264 guide on ffmpeg site has those options [21:53] <JEEB> -preset is used by two encoders so far [21:53] <JEEB> libx264 and libx265 [21:53] <AaronB_> gotcha [22:08] <sacarasc> JEEB: Only 2 *video* encoders. IIRC, one of the AAC ones uses it to? [22:08] <JEEB> it does? I've only seen audio encoders use -profile [22:09] <sacarasc> Wait, I misread because I am half dead. [22:09] <sacarasc> Sorry. [22:09] <sacarasc> Ignore me. [22:09] Action: sacarasc slinks off into the pit of unworn glasses. [22:48] <AaronB_> I get this error when encoding an mkv to an h264 mp4: Non-monotonous DTS in output stream 0:1; previous: 1633624, current: 1633344; changing to 1633625. This may result in incorrect timestamps in the output file. [22:48] <AaronB_> is that a problem with my command or the video file, and can i fix it? [22:49] <AaronB_> Audio is off like 4 seconds [00:00] --- Sat May 17 2014

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