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October 2013

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ffmpeg-devel.log.20131025
by burek 26 Oct '13

26 Oct '13

[00:20] <Compn> if someone wants to save me from that evansport thread [00:20] <Compn> i'd be happy [00:21] <cone-698> ffmpeg.git 03Michael Niedermayer 07master:da54ad9a7208: ffmpeg_filter: Pass command line -threads X to filtergraph [00:21] <cone-698> ffmpeg.git 03Michael Niedermayer 07master:29ffeef5e73b: avcodec/h264: do not trust last_pic_droppable when marking pictures as done [01:23] <cone-698> ffmpeg.git 03Michael Niedermayer 07master:4fb14f84926e: avcodec/wmalosslessdec: fix data flushing at the end [01:23] <llogan> RE: #3045, i wonder if an info message "Use -framerate instead of -r for x11grab" would be good. [01:45] <michaelni> llogan, dunno but if you think so patch is welcome [01:51] <BBB> ubitux: cool [01:51] <BBB> ubitux: don't forget review, I may have comments, or real experts like loren/jason [01:58] <llogan> michaelni: i can try but it might be laughable [01:59] Action: llogan sticks to user help/trolling/asking for console outputs [02:21] <cone-698> ffmpeg.git 03Michael Niedermayer 07master:a443b9732f75: ffmpeg: make stream default if its the only of its kind and no source can be unambigously assigned to it [07:19] <ubitux> <@BBB> ubitux: don't forget review, I may have comments, or real experts like loren/jason // ofc, i won't push this out of the blue :) [09:18] <cone-822> ffmpeg.git 03Paul B Mahol 07master:0f15ebe588a7: MAINTAINERS: update [09:30] <wcpan_> hi, does libswscale support yuv420p to rgb24 conversion on arm? [09:30] <nevcairiel> it can convert anything with C code, however that may not be very fast [09:30] <ubitux> there is no arm optim in sws... :( [09:31] <durandal_1707> s/anything/most things [09:31] <wcpan_> I got some "No accelerated colorspace conversion found from yuv420p to rgb24" [09:31] <durandal_1707> that is is ... [09:31] <nevcairiel> it will still convert [09:31] <nevcairiel> it'll just be slow [09:33] <wcpan_> ouch, I see. thank you [09:46] <durandal_1707> saste: my fork will not have 'ff' in its name [10:53] <durandal_1707> i would really like to push mergeplanes soon [10:58] <durandal_1707> should i add vf_dlopen from mpv wrapper? [10:59] <durandal_1707> which is kind of wrapper around libmpcodecs [11:42] <durandal_1707> michaelni: will you apply '[PATCH] build: remove pointless condition' [11:42] <durandal_1707> ? [11:43] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:3db7ae6a9a25: doc/issue_tracker: theres no substatus in trac [11:43] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:2a28bad82202: doc/issue_tracker: remove patches as we dont use trac to handle patches [11:43] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:cdb46f15fff4: doc/issue_tracker: update resolution names [11:46] <durandal_1707> hah, i can open trac issue with type:sponsoring_request and subject: pizza request! [11:47] <michaelni> durandal_1707, iam trying to apply but it conflicts with 94d707af3c8d2eb7dde7048929b056a451daabcb [11:47] <durandal_1707> ah it is already done.... [12:24] <Daemon404> michaelni, ping re: null deref patchset [12:36] <ubitux> huh [12:37] <ubitux> TRANSPOSE4x4W has a weird output layout [12:37] <ubitux> i have this input: [12:37] <ubitux> 27: $st0 = -nan(0xffe90011fff7000b) [12:37] <ubitux> 26: $st1 = -nan(0xffe90022ffd3001c) [12:37] <ubitux> 25: $st2 = -nan(0xffe9003cffa10036) [12:37] <ubitux> 24: $st3 = -nan(0xffe9004dff7d0047) [12:38] <ubitux> i'm doing a TRANSPOSE4x4W 0, 1, 2, 3, 4 [12:38] <ubitux> the output is correct but the out reg looks shuffled [12:38] <ubitux> basically 1 is a "disabled" output [12:39] <ubitux> 0, 2, 3, 4 contains the appropriate output, randomized a little [12:42] <BBB> ubitux: look at the SWAP macro [12:42] <BBB> ubitux: it does indeed shuffle them [12:42] <BBB> ubitux: so m0 is no longer mm0 [12:42] <BBB> or so [12:42] <BBB> maybe m0 stays m0, but m1 becomes mm4, etc. [12:43] <BBB> TRANSPOSE4x4W uses BUTTERFLY which uses SWAP [12:56] <ubitux> ok [12:56] <ubitux> so i'm not doing something wrong [12:56] <ubitux> i'll just play with those registers instead [13:02] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:52297945ecd6: doc/issue_tracker: remove more mentions of patches [13:02] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:6abb9eb52523: avformat/wavdec: Fix smv packet interleaving [13:03] <BBB> ubitux: if output is correct and make fate-vp9 passes, you're done, really ;) [13:03] <ubitux> yeah sure [13:15] <ubitux> it seems i forgot to reload my code as well, so what i said wasn't exact either, just some shuffling [13:47] <cone-822> ffmpeg.git 03Derek Buitenhuis 07master:530cd2893e02: tree-test: Don't return restricted exit codes [13:47] <cone-822> ffmpeg.git 03Derek Buitenhuis 07master:8dc1b7bd228c: base64-test: Remove posibility of returning restricted exit codes [13:47] <cone-822> ffmpeg.git 03Derek Buitenhuis 07master:5331d2b93fe2: dct-test: Remove possibility of returning a restricted exit code [13:47] <cone-822> ffmpeg.git 03Derek Buitenhuis 07master:7177df90a043: fft-test: Remove possibility of returning restricted exit code [13:47] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:d9ad30507233: Merge remote-tracking branch 'qatar/master' [13:56] <saste> who wants to review my channel_layout patch? [13:56] <saste> i'll push it soon [13:57] <durandal_1707> who wants to review my mergeplanes patch? [13:57] <durandal_1707> i'll push it soon [13:58] <wm4> is the libswscale filter thing pushed yet? [14:01] <durandal_1707> saste: i see nothing obviously wrong/incorrect/misleading/evil/... in channel_layout patch [14:01] <saste> wm4, not if no-one will review it [14:01] <wm4> well, not like I'm a fan of it, it just blocks my own patch [14:02] <saste> michaelni, waiting for your review ^^ [14:20] <ubitux> wth [14:22] <ubitux> TRANSPOSE4x4W is doing some magic like randomizing my registers access [14:22] Action: ubitux doesn't get it [14:23] <durandal_1707> can't be [14:24] <ubitux> so, i have this in my .asm: http://pastie.org/8429562 [14:24] <ubitux> generated output is correct [14:24] <ubitux> http://pastie.org/8429563 [14:25] <ubitux> now i add a TRANSPOSE4x4W 0, 1, 2, 3, 4 before those pxor [14:25] <ubitux> this is what happen in the generated asm: [14:25] <ubitux> http://pastie.org/8429566 [14:26] <ubitux> any idea wth is happening? [14:26] <ubitux> BBB? [14:30] <durandal_1707> ubitux: you expect it to be same? [14:30] <ubitux> yes? [14:31] <ubitux> the thing is, beginning of the function is matching with the code [14:31] <ubitux> it's only post transpose that the output gets somehow fucked up [14:32] <ubitux> i'm certainly missing something really stupid [14:32] <BBB> ubitux: SWAP [14:32] <BBB> ubitux: check the SWAP macro in x86inc.asm [14:33] <BBB> ubitux: specifically, check how TRANSPOSE4x4W calls BUTTERFLY (or one of its variants), and that calls SWAP [14:33] <ubitux> oh shit [14:33] <BBB> it's expected, and it's ok [14:33] <ubitux> yeah ok i see [14:33] <BBB> SWAP is totally awesome [14:33] <BBB> it's like the best thing about x86inc.asm [14:33] <ubitux> so the yasm code is still consistent [14:33] <BBB> perhaps only second to it allowing the same code for mmx, sse and avx [14:34] <ubitux> it doesn't swap, it just redefines the registers? [14:35] <BBB> right, it's a swap only in code-sense [14:35] <ubitux> that's completely insane haha [14:35] <BBB> it's a swap without having to emit an instruction [14:35] <ubitux> i see [14:35] <BBB> = insanely fast [14:35] <ubitux> haha sure yeah [14:36] <BBB> like "//" being a speed operator in c/c++ [14:36] <ubitux> i like this, but it doesn't help debugging with gdb :D [14:36] <BBB> put before any line of code you'd like to speed up [14:36] <BBB> swap is similarly cheaty [14:37] <BBB> gotta go to work, bbl [14:37] <BBB> send me email if you need immediate help [14:38] <ubitux> no it's ok now thank you [15:26] <saste> durandal_1707, i'm going to review mergeplanes [15:29] <durandal_1707> that would be 5th review or so... [15:45] <cone-822> ffmpeg.git 03Stefano Sabatini 07master:a3505136f966: doc/utils: document channel layout specification [15:45] <cone-822> ffmpeg.git 03Stefano Sabatini 07master:b4991292aa83: doc: add a few links to the Channel Layout syntax section [16:53] <durandal_1707> what a brain damage, libav disables crc checking by default [16:53] <nevcairiel> why would crc be on by default [16:53] <nevcairiel> it wastes cpu and users dont care [16:53] <durandal_1707> it changes defaults [16:54] <nevcairiel> isnt it off by default here too [16:54] Action: Daemon404 fails to see a reason to keep it enabled by default [16:58] <durandal_1707> nevcairiel: if it is off by default why would libav change default to off? [16:58] <nevcairiel> maybe it was still on in whatever branch they have of the decoder? [16:59] <durandal_1707> you are breaking stuff all the time [16:59] <Daemon404> how DARE they break their own WIP patch [16:59] <Daemon404> that you chose to merge early [16:59] <Daemon404> what assholes! [17:00] <durandal_1707> lol, no ffmpeg did not merged that crc thing.... [17:02] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:f5dae4894d10: configure: abort if threading without atomics is used [17:47] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:cbb7f56dfeb1: metasound & twinvqdec: return the number of read bytes from read_bitstream [17:47] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:2e7a1fdd6d51: avcodec/twinvq: return the actual number of consumed bytes [18:01] <Daemon404> [jpeg2000 @ 0x1b32400] unsupported marker 0xFF5F at pos 0x10B [18:01] <Daemon404> [jpeg2000 @ 0x1b32400] Support for 4 components is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [18:01] <Daemon404> [jpeg2000 @ 0x1b32400] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list. [18:01] <Daemon404> [jpeg2000 @ 0x1b32400] error during processing marker segment ff90 [18:02] Action: Daemon404 wonders why this dcp file has alpha [18:02] <Daemon404> :( [18:02] <j-b> Daemon404: you have a DCP player? [18:02] <Daemon404> yes it's called vlc [18:02] <cbsrobot> maybe it's used for projection mapping [18:02] Action: Daemon404 runs [18:02] <Daemon404> either way [18:02] <ubitux> BBB: we will probably be able to re-use some of the mul/sum/round macro [18:02] <Daemon404> no image is outout [18:02] <Daemon404> output* [18:03] <Daemon404> http://ftp.nluug.nl/pub/graphics/blender/demo/movies/ToS/ToS_4k_DCP.zip <-- playing with this [18:03] <ubitux> BBB: http://pastie.org/8430133 so this is incomplete but it works [18:03] <Daemon404> tears of steal [18:03] <ubitux> BBB: only 1d [18:03] <ubitux> i should be done "soon"© [18:03] <durandal_1707> Daemon404: use other decoder? [18:03] <Daemon404> "other" ? [18:04] <Daemon404> the openjpeg one? [18:04] <cbsrobot> Daemon404: fortunately it takes only 2 min to download [18:04] <Daemon404> lol [18:04] <durandal_1707> michaelni: that block_align change breaks metasount in asf [18:04] <durandal_1707> *metasound [18:05] <j-b> Daemon404: ok, sorry. [18:05] <Daemon404> j-b, btw my friend says he saw some cinemas in either iceland or norway using vlc [18:05] <durandal_1707> for 3D? [18:06] <Daemon404> no? [18:06] <durandal_1707> pro cinemas? [18:06] <j-b> Daemon404: yes, I'm not surprised. [18:07] <j-b> Daemon404: hence the work on DCP for VLC [18:07] <mateo`> Daemon404: vlc can decrypt/play DCP packages and decode jpeg2000 real time now ? :D [18:07] <j-b> we're not yet there [18:07] <j-b> mateo`: big machine? [18:07] Action: cbsrobot saw a 5d cinema in a little village in sardegna a few weeks ago ! [18:08] <cbsrobot> well, let's say aonly the ad said it was 5d [18:08] <Daemon404> what are the 4th and 5th dimensions? [18:08] <durandal_1707> 3D + 8ch audio? [18:10] <durandal_1707> but not anaglyph [18:10] <nevcairiel> Daemon404: i've seen movement and smell [18:11] <Daemon404> i was hoping "time" and "space" [18:11] <durandal_1707> smell? [18:11] <Daemon404> i.e. free LSD [18:13] <durandal_1707> sight + hearing + taste + smell + touch + 6th sense [18:13] <durandal_1707> pain + balance are extra [18:15] <saste> cbsrobot, then you was in calagonone [18:19] <durandal_1707> but jpeg2000dec.c code supports 4th component [18:26] <wm4> the alpha dimension? [18:29] <durandal_1707> whant to find out? download 13G zip file [18:29] <Daemon404> i have a 100 mb cut too [18:30] <Daemon404> http://chromashift.org/sample.mxf [18:42] <cbsrobot> saste: exactly ! [18:43] <saste> cbsrobot, it was easy :) [18:44] <cbsrobot> saste: I was intreaged to go there, but I ended up eating pizza at the local pizza delivery shop [18:45] <saste> cbsrobot, basically as everyone else does [18:45] <cbsrobot> I hope not (for the cinema owner) [18:46] <durandal_1707> Daemon404: looks like it is bacause of unsupported marker [18:46] <cbsrobot> but the logo looked like a bad copy of a 3d software [18:46] <cbsrobot> else I might have gone there & ! [18:46] <cbsrobot> *logo of a 3d software [18:47] <durandal_1707> didn't you said it was 5d? [18:47] <cbsrobot> well the software is called cinema 4d [18:47] <cbsrobot> so you see the marketing concept ! [18:47] <durandal_1707> iirc that is not just software [18:50] <durandal_1707> michaelni: block_align is required for demuxer that do not give proper packets to decoder [18:50] <cbsrobot> just compare: http://www.vnix.com.sg/graphisoftsg/products/archicad/addons/cinema_4d/inde… vs http://imgur.com/Reb9WwP [18:52] <durandal_1707> first have nothing in common with second [18:59] <michaelni> durandal_1707, where can i find a file that my block alig change breaks ? (so i can fix it) [19:02] <durandal_1707> and how would you fix that? [19:03] <durandal_1707> the block_align is set depending on bitrate, etc... [19:03] <durandal_1707> and in your commit that fix bug i see no explanation of problem it solves and why [19:09] <michaelni> the packet size doesnt match block align in that file, the packet size also isnt constant in it IIRC [19:09] <michaelni> the result where warnings/errors of packets being smaller than block align [19:22] <durandal_1707> than it probably should return smaller one of two [19:22] <durandal_1707> because if demuxer give big packet with multiple frames in it, decoder will most likely return only one [19:22] <durandal_1707> all this sucks [19:23] <durandal_1707> so either demuxer should be fixed, or decoder returns multiple frames.... [19:53] <ubitux> yay double idct working [19:53] <ubitux> last step and i'm done [19:55] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:d04ec6ffc51c: avformat/rmdec: english typos [19:55] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:ba8716df7fb5: avformat/mp3dec: perform seek resync in the correct direction [20:10] <cone-822> ffmpeg.git 03Michael Niedermayer 07master:08a89761964b: avcodec/h264: reduce noisiness of "mmco: unref short failure" [20:19] <ubitux> can someone make me a summary of DEFINE_ARGS and STORE_DIFFx2 macros? [20:19] <ubitux> BBB? Skyler_ maybe? [20:37] <durandal_1707> i can't belive my eyes: [FFmpeg-user] Is "-q:v" the same as "-qscale:v"? [21:25] <lukaszmluki> Hi, I have lame question, why there are 2 methods av_frame_alloc and avcodec_alloc_frame which seems to do the same? [21:27] <wm4> lukaszmluki: because the ffmpeg API is freaking terrible [21:28] <wm4> but use av_frame_alloc and ignore the other one [21:28] <lukaszmluki> ok, thx [21:36] <ubitux> lukaszmluki: avcodec_alloc_frame() if not ref counting frames [21:36] <ubitux> av_frame_alloc otherwise [21:37] <ubitux> lukaszmluki: see http://ffmpeg.org/pipermail/ffmpeg-devel/2013-September/147921.html [21:37] <ubitux> lukaszmluki: nice to see you on irc btw :) [21:38] <lukaszmluki> :) [21:38] <wm4> ubitux: how is there a difference? because I don't think there is [21:38] <lukaszmluki> I log in when need something :) [21:38] <Daemon404> ...wait [21:38] <wm4> and libavcodec is using libavutil incorrectly [21:38] <Daemon404> we have alloc_frame and frame_alloc? [21:38] Action: Daemon404 facedesks [21:39] <lukaszmluki> thx for answer, make sense [21:39] <ubitux> wm4: it was (and still is in the examples...) causing various memleak or invalid r/w [21:39] <wm4> nice [21:39] <ubitux> that's the reason i wrote that example [21:40] <wm4> + if (use_ref_counting) [21:40] <wm4> + frame = av_frame_alloc(); [21:40] <wm4> + else [21:40] <wm4> + frame = avcodec_alloc_frame(); [21:40] <wm4> Daemon404 is going to get a heart attack [21:40] <ubitux> :) [21:40] <wm4> and me too [21:40] <ubitux> feel free to propose better& [21:40] <wm4> because what the flying fuck [21:41] <ubitux> well, that's because of TEP [21:41] <ubitux> and this is not really documented so far afaict [21:42] <wm4> there doesn't seem to be much difference between these two functions [21:42] <wm4> the frame defaults could be slightly different [21:42] <Daemon404> ubitux, i luckily moved all my code over to TEP as soon as it was pushed [21:43] <Daemon404> with a lot of yelling at its author for "docs" [21:43] <Daemon404> <_< [21:59] <ubitux> omg, reimar adding arm intrinsics to mplayer [22:03] <wm4> so stupid, he could be using libavresample [22:03] <wm4> or ffmpeg's nih'ed copy of it [22:04] <ubitux> wm4: ... ? [22:04] <wm4> like float<->int conversions for audio [22:04] <wm4> pretty sure libavresample has asm for it, or would appreciate it [22:04] <ubitux> (lavr is the nih of lsws) [22:04] <ubitux> you're allowed to troll, but only when you are right [22:05] <wm4> lavr was in the works, when mini ninja-nih'ed it [22:05] <wm4> or something like this [22:05] <ubitux> nih 1 year earlier? from a code he already wrote? [22:05] <ubitux> come on... [22:05] <ubitux> i'm pretty sure it was done before it was even started on the other side [22:07] <wm4> *shrug* [22:07] <wm4> libswresample is still less useful, because it's available only on one of the 2 projects [22:08] <ubitux> i was just commenting on your nih comment [22:08] <ubitux> feel free to escape the topic [22:08] <ubitux> wm4: should we drop lavr from the project? [22:09] <wm4> that would be way more sincere than keeping a disabled copy of it [22:09] <wm4> but you won't do it because your chance you can force someone to use ffmpeg insteadof libav drops by 1% [22:10] <ubitux> "force someone to use ffmpeg" lol [22:10] <wm4> a patch removing libavresample from ffmpeg would make a good troll though [22:10] <ubitux> yup [22:11] <ubitux> let's add the macro wrapper instead [22:12] <ubitux> ...and keep that wrapper disabled by default ;) [22:12] <ubitux> - --enable-avresample enable libavresample build [no] [22:12] <ubitux> + --enable-avresample enable libavresample wrapper build [no] [22:54] <cone-822> ffmpeg.git 03lemonlinger(a)gmail.com 07master:2cf94485de9a: avformat/rtpproto: when local rtp port and local rtcp port are not specified, try to make two successive ports and retry 3 times when failed [23:00] <Daemon404> line breaks, how do they work? [23:12] <ubitux> BBB: i tried to stole vp8 code but it crashes for some reason (in STORE_DIFFx2); https://github.com/ubitux/FFmpeg/compare/vp9-asm [23:12] <ubitux> for now the double idct is properly working (i'm getting lost in my own shuffling though) [23:43] <ubitux> oh i think i get it [23:45] <ubitux> doesn't crash anymore, almost there. [00:00] --- Sat Oct 26 2013

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ffmpeg.log.20131025
by burek 26 Oct '13

26 Oct '13

[01:42] <pfloyd> I'm trying to combine the m2ts in a BDMV folder to a single m2ts and I'm seeing a ton of these: [mpegts @ 0x805078a00] Non-monotonous DTS in output stream 0:1; previous: 84819058, current: 84819056; changing to 84819059. This may result in incorrect timestamps in the output file. [01:42] <pfloyd> using command line: ffmpeg -i "concat:f1|f2|f3|..." -c copy combined.m2ts [01:42] <pfloyd> is there a better way to convert BDMV to a format my popcorn hour will like? [01:43] <pfloyd> or does anyone know if a popcorn hour will just play a folder with BDMV in it like it does a DVD rip? [03:45] <Ulfalizer> how does the resampling stuff in libavcodec/avcodec.h (av_audio_resample_init(), audio_resample(), etc.) relate to libswresample? [07:27] <xsacha> hi [07:43] <xsacha> i have a weird crash in ffmpeg and was wondering if anyone could help [07:53] <xsacha> my system is blackberry10. i upgraded from ffmpeg 1.2 to ffmpeg 1.4 (latest) and now i get crashes from avformat_find_stream_info [07:54] <xsacha> specifically, when i remove this line it no longer crashes (even though this line was also in 1.2): https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/utils.c#L2750 [07:54] <xsacha> i followed its codepath to avcodec_open2 [07:55] <xsacha> when i comment out this line it no longer crashes: https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/utils.c#L1181 [07:55] <xsacha> it doesn't seem related to &tmp though. because if i do an: AVDictionary *foo = NULL; av_dict_copy(&foo, *options, 0); it crashes again [09:07] <dreamhawk> hello, can anyone help me here with "yasm/nasm not found or too old. " on a centos 6.4 ? [09:08] <JEEB> compile your own yasm and put it somewhere that goes into PATH :) [09:08] <JEEB> http://yasm.tortall.net/Download.html [09:08] <dreamhawk> JEEB: http://www.pawprint.net/news/article/105/Compiling-FFMpeg-on-Centos-5/ <-- i followed this, simply put all in a shell-script and executed it :) [09:09] <JEEB> yeah, no [09:09] <JEEB> that's three years old and contains no traces of a newer yasm [09:09] <dreamhawk> https://trac.ffmpeg.org/wiki/CentosCompilationGuide this gave me same error yesterday. :/ [09:10] <JEEB> that tells you to build your own yasm, so you most probably did not have yasm in your PATH, or had it come in PATH after a possibly package installed one [09:11] <JEEB> that seems to install binaries under /bin in your $HOME [09:11] <JEEB> echo $PATH and see if it is there [09:26] <dreamhawk> JEEB: not, no Yasm there [09:30] <JEEB> it's not Yasm, it's yasm (at least the binary name). And you did understand what I meant, right? [09:31] <dreamhawk> JEEB: i think so, this is a bit new for me, i've mostly yum installed everything, now i need aac-support and need to compile myself. and i didnt find anything with "yasm" in echo $PATH [09:32] <JEEB> ugh [09:32] <JEEB> do you know what PATH is? [09:32] <JEEB> it is not linux-specific and it serves a certain purpose [09:32] <JEEB> (windows and friends have a PATH system variable as well) [09:33] <dreamhawk> Yes i know about it. [09:33] <JEEB> ok [09:33] <JEEB> so you did not see the spot that script would install the yasm binary to in your PATH? [09:33] <JEEB> that would explain why it didn't work then :P [09:33] <JEEB> uhh, not script [09:33] <JEEB> compilation guide [09:34] <JEEB> (see the --bindir it sets pretty much everywhere) [09:34] <JEEB> also you don't really need nasm methinks, since everything in there that uses asm uses yasm, not nasm [09:35] <JEEB> also uninstall any packaged yasm that you might have installed [09:35] <dreamhawk> i have ran yum remove yasm [09:36] <JEEB> anyways, if you want to follow that guide, make sure ${HOME}/bin is in your PATH [09:36] <JEEB> since that is what it is based upon [09:37] <JEEB> oh, LAME uses nasm o_O [09:37] <JEEB> never knew [09:37] <JEEB> but if you don't need mp3 encoding you don't need LAME anyways [09:37] <JEEB> most people only need libx264 and libfdk-aac [09:37] <dreamhawk> ooh [09:39] <dreamhawk> JEEB: so i change --bindir to like, /usr/ instead? [09:39] <JEEB> generally on linux you don't want to use /usr/ as-is since packages generally tend to go there [09:39] <JEEB> /usr/local is generally used for locally built things [09:39] <JEEB> (that you want to be in a root-only-writable area) [09:40] <dreamhawk> Ok :), so how could i add yasm path to PATH ? [09:40] <JEEB> which are you going with? [09:41] <dreamhawk> that centos guide, seems the best [09:41] <JEEB> export PATH=${HOME}/bin:${PATH} [09:41] <JEEB> : is the delimiter of PATH directories [09:41] <JEEB> the guide uses ${HOME}/bin so I used that [09:42] <JEEB> after that if you echo $PATH or ${PATH} you should see that your home directory's bin directory is now in PATH [09:42] <JEEB> there are various files where you can put this to automate this so that when you launch a shell you will get that done [09:43] <JEEB> such as dot-bashrc or dot-bash_profile [09:45] <dreamhawk> JEEB: alright! :) [10:04] <chandra_> Hi, I am trying to compile ffmpeg 1.2.4 on ubuntu 13.10 with gcc 4.7 [10:05] <chandra_> there are my flags for configure --disable-yasm --enable-shared --disable-static --extra-cflags=-fPIC [10:05] <chandra_> I got a linker error : libavutil/lls.o: relocation R_X86_64_PC32 against undefined symbol `memset@@GLIBC_2.2.5' can not be used when making a shared object; recompile with -fPIC [10:06] <chandra_> Please can any one help me out [10:08] <relaxed> chandra_: try adding --extra-ldflags="-static" and removing --extra-cflags=-fPIC [10:08] <JEEB> first of all 1.2.4 is old, second of all if you needed PIC there's --enable-pic, third of all you really don't want to compile without yasm, fourth of all that error seems to be coming from glibc, not ffmpeg, so compiling ffmpeg with PIC will not exactly help :P Also make sure you clean your source tree every time you change configuration [10:09] <JEEB> heck, yasm 1.2 comes with 13.10 [10:09] <JEEB> so you could just install the packaged one [10:09] <chandra_> thank you [10:09] <chandra_> i try it now [10:10] <JEEB> also the standard compiler on 13.10 is 4.8, any specific reason you're using 4.7? [10:11] <chandra_> 4.8 has some compilation problem with my source code [10:11] <JEEB> (although since you are not setting --cc I bet you are actually compiling with 4.8, unless ffmpeg's configure takes in the value of CC env. variable) [10:12] <chandra_> ok [10:14] <chandra_> Actually, /usr/bin/gcc -> gcc-4.7 :) [10:14] <chandra_> in my system [10:29] <matthias_arch> I'm recording my desktop with ffmpeg while playing Minecraft. I've set the FPS to 30. If i start the recording before starting the game the fps stays at 30, but if i start the recording if minecraft is already started up the fps is first around 67 and goes slowly down and is not such stable like in option1. any ideas, I'm recording no sound [10:44] <matthias_arch> durandal_1707: pastebin.com/tZ6jsDXv [10:45] <matthias_arch> durandal_1707: if i start the game after starting recording the fps stays everytime at 30 [10:53] <durandal_1707> matthias_arch: that is not ffmpeg from FFmpeg but from fork [10:55] <matthias_arch> durandal_1707: should i ask in #fork? [10:56] <durandal_1707> if you intend to use fork [10:56] <durandal_1707> lol, and not literal fork [10:59] <matthias_arch> would it work with the "original" FFmpeg? [10:59] <JEEB> depends on a lot of things [10:59] <matthias_arch> yes i got that from the ubuntu repo [10:59] <JEEB> anyways, basically with libav-based binaries you should ask from #libav and use the avconv binary when possible instead of ffmpeg. Because they left the ffmpeg binary more or less unattended after rewriting parts of it as avconv. [10:59] <JEEB> those rewrites were then of course merged into ffmpeg's ffmpeg [11:31] <diroots> hi there [11:34] <diroots> I have P2 mxf files : 1 video mxf and 4 audio mxf, in which format can I convert this to, with losless conversion, and keeping the 4 mono audio tracks separated? mkv container? is there another format/container allowing 4 distincts audio tracks? [11:35] <diroots> or can I merge these mxf to one single mxf containing video and audios? will this be editable by common video tools?? [11:38] <durandal_1707> mkv can have any number of distinct audio tracks [12:26] <plm> morning all =D [12:33] <brontosaurusrex> diroots, get mkvtoolnix and try to remux [14:16] <hexy> Hey people, question.. How would I strip the audio from video A and replace it with audio from video B without re-encoding the video -- I don't know if it's possible or not without re-encoding. [14:17] <hexy> But any help would be very appreciated =] [14:17] <sacarasc> -c copy and -map [14:21] <hexy> sacarasc: So it would be something like ffmpeg -i a.avi -c copy -map ...???? [14:22] <sacarasc> Yes, something like that. [14:22] <hexy> sacarasc: I was asking what comes after.. if you don't mind [14:22] <hexy> sacarasc: I apologize, it's my first time using ffmpeg [14:22] <sacarasc> I've never got map to work, so I can't help any further. [14:23] <hexy> Ugh, sacarasc, suggesting something that I would break my head over. Thanks lol [14:25] <sacarasc> It would be something like ffmpeg -i blah.avi -i -audio.mp3 -map 0.0:0.0 -map 1.0:0.1 -c copy output.avi [14:25] <sacarasc> That's about as much as I can help. :D [14:28] <hexy> sacarasc: I'm reading about the -map option, maybe I'll figure it out. Also, I need to take the audio from a video file, not mp3, so it's a bit more tricky I'm assuming. But thank you anyways. [14:28] <sacarasc> Not really. [15:08] <troulouliou_dev> hi i have a s-video/rca capture card the i would like to stream as sap/rtsp stream. can anybody help ? [17:02] <LithosLaptop> + [17:02] <LithosLaptop> oops [17:09] <plm> people, anyone can help to understand about realtime stream and HLS using ffmpeg? [17:10] <plm> I would like to streaming a rtsp camera over internet, but just peer to peer. [17:12] <plm> rstp_camera---(eth0)box1---internet---box2. I would like in box1 to do a hls from video of rstp and in box2 I will watch that video.. [17:13] <plm> I try ffmpeg hls to generate files, but files not stop to create while video input not finish. But video will never finish, becouse is rtsp from camera, how I do? [17:15] <plm> I tryed this in box1 ./ffmpeg -y -i http://187.71.63.23:8082 -pix_fmt yuv420p -vcodec libx264 -acodec libfaac -r 25 -profile:v baseline -b:v 1500k -maxrate 2000k -force_key_frames 50 -s 640x360 -map 0 -flags -global_header -f segment -segment_list /var/www/hls/index_1500.m3u8 -segment_time 10 -segment_format mpeg_ts -segment_list_type m3u8 /var/www/hls/segment%05d.ts [17:15] <plm> sorry, I change rtsp by http on camera, so input on ffmpeg in http [18:10] <plm> anyone? [19:37] <zap0> i have some video i'm trying to convert to MP4, but i'm getting "width not divisisble by 2" errors. width's are odd :( is there some magic to crop 1 line off the edge? [19:37] <zap0> or add a line [19:40] <JEEB> zap0, there are crop and pad video filters for that [19:41] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#crop [19:41] <JEEB> see the examples part and so forth for usage examples [19:43] <zap0> crop=in_w-1:out_h=in_h [19:59] <zap0> seperate topic: is there a filter for detecting scene change/cuts in a video? [20:08] <yairgo> Is the only way to get ffmpeg on centos 6 to compile it from source? [20:10] <sacarasc> What version kernel do you have? [20:10] <yairgo> 6.4 [20:11] <sacarasc> That's the distro version number... What does `uname -a` say if you do it on the command line? [20:12] <yairgo> sorry, Linux video-server 2.6.32-358.23.2.el6.x86_64 [20:12] <sacarasc> You should be able to use the latter of them. [20:13] <sacarasc> Download it, extract it and use it as ./ffmpeg. Or add it to your $PATH and use it as ffmpeg. [20:13] <yairgo> ok, I don't have an issue with trying to compile from source, I was more curious [20:13] <sacarasc> There's that way! [20:13] <sacarasc> It's up to date and easier. [20:14] <yairgo> I didn't know if my google ability was missing a source repo somehow [20:15] <yairgo> thanks sacarasc [20:47] <cheri> hi I am able to capture elementary streams from a multicast ts it has h264 video and ac3 audio codec. I want to transcode it to flv on the fly how can I do it. any idea [21:14] <hexy> Hey people, question.. How would I strip the audio from video A and replace it with audio from video B without re-encoding the video [21:20] <JEEB> hexy, ffmpeg -i video.A -i video.B -map 0:v -map 1:a -c copy video.out [21:20] <JEEB> something like this [21:20] <JEEB> two inputs, map to output the video from input 0 (first file), and audio from input 1 (second file) [21:20] <JEEB> and then -c copy to copy the data instead of transcoding [21:21] <hexy> JEEB Thank you! [21:21] <cheri> hi can anyone help me how to encode mpeg ts into flash [21:25] <cheri> I am trying this but doesn't work ffmpeg -i input.ts -c:v flv -b:v 1000k -c:a libmp2lame -f flv -y output.flv [21:25] <hexy> JEEB: I'm only getting audio as the output [21:26] <JEEB> hexy, show used command and the full terminal output in a pastebin [21:26] <hexy> Okay, one sec [21:27] <cheri> sorry the typo thats actally libmp3lame [21:29] <hexy> JEEB: http://pastebin.com/uZQCt9Mt [21:30] <JEEB> hexy, you can see by yourself that after output you have H.264 and AAC in there [21:31] <JEEB> and in stream mapping you can see that it picked the first stream from the first input file, and second from the second fil [21:31] <JEEB> *file [21:31] <JEEB> so that part seems to be working fine [21:31] <JEEB> and most definitely should not contain only audio [21:32] <hexy> JEEB: I just ran through the entire outputed file again just to make sure, and still nothing. [21:32] <hexy> Only audio that is.. [21:32] <JEEB> also you should have been able to use -map 0:v and -map 0:a to mark that you want video and audio from those, no need to explicitly put the number there [21:33] <JEEB> although your ffmpeg is old as hell [21:33] <JEEB> (by now) [21:33] <JEEB> hexy, dunno -- either how you are reading it fails or that old ffmpeg build fails [21:33] <JEEB> it seems to be most definitely doing what you want [21:33] <hexy> JEEB: I literally installed the new compiled version today [21:34] <JEEB> it could be a newly compiled version, but it's from the 0.10 branch [21:34] <hexy> Should I build from source? [21:34] <JEEB> 20 months ago n0.10 FFmpeg 0.10 release tag | commit | shortlog | log [21:34] <JEEB> that's basically where it was branched :P [21:34] <JEEB> 20 months ago [21:34] <JEEB> hexy, or you could grab a static build [21:34] <hexy> Where? [21:35] <hexy> I was looking for one [21:35] <JEEB> http://ffmpeg.gusari.org/static/ [21:35] <JEEB> I guess here [21:44] <hexy> JEEB: Sorry, it was just a bad player... Got it working! Thanks dude [21:52] <smjd> hey... is it possible to record my desktop, and have always some buffer size filled? so that I have maybe four last minutes recorded on my hard drive, no more or less [21:57] <smjd> could be useful catching bugs and other things [21:58] <smjd> +for [23:19] <teratorn> is there already some common data structure somewhere for recycling AVFrame* objects? [23:33] <hexy> I am unable to sync the audio, why is that?? This is the command I used: ffmpeg -i first.avi -itsoffset 6.0 -i second.avi -map 0:v -map 1:a -c copy out.avi [00:00] --- Sat Oct 26 2013

1 0

ffmpeg-devel.log.20131024
by burek 25 Oct '13

25 Oct '13

[00:11] <Compn> burek : http://www-staging-uk.webex.com/fileadmin/GLOBAL_PDF/seminars/recording/ [00:17] <Daemon404> webex is just h264 SVC [00:50] <michaelni> BBB, maybe http://fate.ffmpeg.org/history.cgi?slot=x86_32-debian-kfreebsd-gcc-4.4-cpuf… [00:59] <Zeranoe> Is there any way to disable usage of Thread local storage in a FFmpeg build? Or do all instances of __thread need to be removed or something? [01:14] <michaelni> "git grep __thread" has no matches [01:16] <Zeranoe> oh... so it must be in the compile then? [01:20] Action: michaelni has no idea where the thread local comes from [01:21] <michaelni> do you know some line number or something else that could help pinpoint where its from ? [01:23] <Zeranoe> This is everything I've been given: http://paste.debian.net/61013/ [01:26] <iive> tls... that trace doesn't look like ffmpeg thing. [01:27] <Zeranoe> I wonder if I need to recompile my toolchain with --disable-tls... [01:39] <BBB> michaelni: hm... ok, I guess the problem is that it's a box that can do more ;) [01:39] <BBB> anyway [01:39] <BBB> I tried to help the guy I think it's fixed now, I'll send a new patch if it works [01:39] <BBB> ubitux: pong [01:50] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:2f31b73a11d6: avcodec/mpegaudio_tablegen: fix build with hardcoded tables [02:25] <BBB> does anyone have a katmai (intel pentium III) lying around? [02:26] <llogan> maybe someone on the mailing list will have one [02:27] <BBB> can you ask for me? it'd be really helpful for me [02:27] <BBB> (and yes it's ffmpeg-related :) ) [02:27] <BBB> it's actually somewhat tangentially related to a bug report on ffmpeg-users [02:29] <llogan> i can if you're unable to atm. what are your requirements and how should they contact you? [02:33] <BBB> reply-to rsbultje(a)gmail.com, requirement is a intel pentium III (katmai) with some method of being able to put software on it (i.e. either a compiler and assembler, or a network connection to another computer that has one), and a terminal so they can a terminal application [02:35] <BBB> ideally they are capable of compiling ffmpeg on it, apply a patch and run a subset of fate to show it doesn't break [02:35] <BBB> but I can write a simpler test if that's not possible [02:36] <llogan> ...and just to be soure i was referring to ffmpeg-devel mailing list. [02:36] <llogan> *sure [02:38] <BBB> oh I can write that myself, I thought you meant ffmpeg-users [02:38] <BBB> (wider audience, no?) [02:39] <llogan> heh. i was wondering... 1347 (-devel) vs 1802. [02:39] <BBB> oh [02:39] <BBB> hm... [02:39] <BBB> ok [02:39] <BBB> then I'll ask on -devel [02:39] <llogan> write it on -devel and i'll spam -user with it [02:39] <BBB> ok ty [02:42] <llogan> and if we get nothing i can make a post on the web site [02:44] <llogan> libav-user is 659 and 194 for cvslog for the curious [03:02] <BBB> llogan: I'll send tomorrow, don't wait for it - waiting for some more results [03:18] <BBB> ubitux: michaelni: patch sent that I think properly fixes it. it's not ideal since it'll slow down a little more on sse-only cpus, but I don't think anyone cares at this point [03:19] <BBB> now back to simd again [03:19] <BBB> (vp9 simd) [07:27] <ubitux> BBB: yeah sorry, i had a question about simd but i solved it [07:27] <ubitux> i fixed the rounding, and i'm getting closer to the final result [08:54] <cone-897> ffmpeg.git 03Stefano Sabatini 07master:ff9d11f95deb: lsws: use av_pix_fmt_get_chroma_sub_sample() [08:54] <cone-897> ffmpeg.git 03Lukasz Marek 07master:759388aa717a: lavd: add fbdev output device [08:54] <cone-897> ffmpeg.git 03Lukasz Marek 07master:c720d0d9df8e: lavd: rename fbdev.c into fbdev_dec.c [08:54] <cone-897> ffmpeg.git 03Lukasz Marek 07master:a547c496e99e: lavd/fbdev: move common code to separate file [09:45] <cone-897> ffmpeg.git 03Paul B Mahol 07master:426abbb632b0: avfilter/libmpcodecs/vf_eq: remove useless storage class specifier in empty declaration [09:48] <durandal_1707> michaelni: flvdec have unused flv_data_packet() function, merge failure? [09:48] <ubitux> iirc it's wanted [09:48] <ubitux> wi [09:48] <ubitux> i've heard it's broken or untestable [10:03] <durandal_1707> why libavcodec/mpegaudio_tablegen.c overrride CONFIG_HARDCODED_TABLES? [10:29] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:049169535843: avdevice/fbdev_enc: avoid declaring variables in for() [10:42] <cone-897> ffmpeg.git 03Daniel Kang 07master:0e730494160d: avfilter: x86: Port gradfun filter optimizations to yasm [10:42] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:1ea28ffc4dbc: Merge commit '0e730494160d973400aed8d2addd1f58a0ec883e' [10:53] <cone-897> ffmpeg.git 03Vittorio Giovara 07master:fb13fe8342fa: golomb: reduce scope of a few variables [10:53] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:caac12bd4984: Merge remote-tracking branch 'qatar/master' [10:53] <ubitux> "In this section maybe denoise_row[ 8] and desnoise_row[ 9] could be [10:53] <ubitux> aligned at the = instead of the [] ?" [10:54] <ubitux> lol [10:54] <ubitux> sounds important [11:01] <michaelni> BBB, where did you sent that patch ? i dont see it on any list ? [11:07] <durandal_1707> ubitux: ? [11:09] <ubitux> durandal_1707: cosmetics discussion [11:11] <durandal_1707> ubitux: real programmers do that every day [11:22] <saste> durandal_1707, and finally they do nothing else [11:33] <cone-897> ffmpeg.git 03Timothy Gu 07master:0be74cf0f5b1: doc/encoders: add libvorbis doc [11:34] <cone-897> ffmpeg.git 03Timothy Gu 07master:6879fe426593: doc/codecs: update skip_alpha documentation [11:34] <cone-897> ffmpeg.git 03Timothy Gu 07master:683d9cb1f282: lavc/libmp3lame: fix alignment and capitalization in AVOptions [11:34] <cone-897> ffmpeg.git 03Timothy Gu 07master:07e7bc9cbda1: lavc/libmp3lame: add comments about CBR/VBR modes [11:50] <saste> durandal_1707, [Parsed_displace_2 @ 0x3416de0] [framesync @ 0x3416ec0] Buffer queue overflow, dropping. Last message repeated 877 times [11:51] <durandal_1707> so? [11:52] <durandal_1707> 'full uncut console output missing' [11:54] <cone-897> ffmpeg.git 03Ingo Brückl 07master:94d707af3c8d: build: remove pointless condition [11:54] <cone-897> ffmpeg.git 03Derek Buitenhuis 07master:5dfdffeb6fe9: tree-test: Don't return restricted exit codes [11:54] <cone-897> ffmpeg.git 03Derek Buitenhuis 07master:cbbd0ac764b4: base64-test: Remove posibility of returning restrcted exit codes [11:54] <cone-897> ffmpeg.git 03Derek Buitenhuis 07master:f80ca1cb002c: dct-test: Remove possibility of returning a restricted exit code [11:54] <cone-897> ffmpeg.git 03Derek Buitenhuis 07master:b2fdf3fc35b6: fft-test: Remove possibility of returning restricted exit code [11:55] <saste> ffmpeg -i matrixbench_mpeg2.mpg -f lavfi -i nullsrc=s=hd720,geq='r=128+50*sin(2*PI*X/800):g=128+50*sin(2*PI*X/800):b=128+50*sin(2*PI*X/800)' -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -lavfi '[0]scale=hd720,setsar=1[t0];[t0][2][1]displace' -y out.mkv -nostats [11:55] <saste> durandal_1707, ^^ [11:55] <saste> if I use a synthetic video it doesn't happen [11:57] <durandal_1707> and it happens for first/last/all frames? [11:58] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/0.10:8096691a681b: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/0.11:5df35d9a21d5: avformat/wavdec: Dont trust the fact chunk for PCM [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/0.11:87f719a03f23: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/0.11:43af9624fe11: avformat/mov: force parsing of headers if stts is absent [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/0.9:1b3803e8c11f: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.0:1cfcc9ac8ec6: avformat/mov: force parsing of headers if stts is absent [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.0:e3b2b884fc33: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.0:3c142c196d41: avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.0:89b5f5f494a2: avformat/wavdec: Dont trust the fact chunk for PCM [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.1:de32de679bcc: avformat/wavdec: Dont trust the fact chunk for PCM [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.1:8ffdcd04c481: avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.1:6636dd551fda: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.1:802c4f523186: avformat/mov: force parsing of headers if stts is absent [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.2:6559bb893f5f: avformat/mov: force parsing of headers if stts is absent [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.2:eff0bf7defdf: avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.2:970109deaf47: avformat/wavdec: Dont trust the fact chunk for PCM [11:59] <cone-897> ffmpeg.git 03mrlika 07release/1.2:311e58e478c1: lavd/v4l2: do not fail when VIDIOC_ENUMSTD returns EINVAL without a valid match [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/1.2:1fe734f4d342: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03mrlika 07release/2.0:f581e25a6905: lavd/v4l2: do not fail when VIDIOC_ENUMSTD returns EINVAL without a valid match [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/2.0:59431fc84187: avcodec/h264_refs: modify key frame detection heuristic to detect more cases [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/2.0:736851264b79: avformat/wavdec: Dont trust the fact chunk for PCM [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/2.0:6de6d9e2d3ed: avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/2.0:7da810e68b76: avformat/gifdec: make GIF_APP_EXT_LABEL parsing more robust [11:59] <cone-897> ffmpeg.git 03Michael Niedermayer 07release/2.0:fb1fb462e598: avformat/mov: force parsing of headers if stts is absent [12:08] <saste> BC@;$ä => best username ever [12:18] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:a826efb55ad8: avfilter/x86/vf_gradfun_init: fix const and related warnings [12:20] <durandal_1707> michaelni: but i pushed fix to ml [12:24] <cone-897> ffmpeg.git 03Carl Eugen Hoyos 07master:f60c3a7c1b1e: lavd/x11grab: Support AV_PIX_FMT_PAL8. [12:24] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:e7811e48be26: Merge remote-tracking branch 'cehoyos/master' [12:25] <durandal_1707> saste: that appears to be framesync bug [12:25] <michaelni> durandal_1707, oops didnt see your patch [12:26] <durandal_1707> you do not read mails at all [12:27] <durandal_1707> perhaps your todo list is empty? [12:27] <durandal_1707> ;/ [12:28] <michaelni> i had seen the warnings after the merge and fixed them before looking at the ML [12:29] <durandal_1707> yes, somehow we started doing same things (not first time), this did not happened before.... [12:34] <durandal_1707> saste: it does not happen if you generate maps once, and reuse them later [12:35] <durandal_1707> it is also faster [12:54] <BBB> michaelni: f.. typo in git send-email command [12:55] <BBB> michaelni: re-sent [13:07] <durandal_1707> saste: so you have any more comments for displace filter? [13:11] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:210afae0ba65: avformat/utils: Check size argument in ff_add_index_entry() [13:19] <durandal_1707> what about following options for mergeplanes: 4 ints which are flags and each bit maps to single plane [13:19] <durandal_1707> 4 ints for up to 4 inputs [13:20] <durandal_1707> i hate '1+2|0|1|1' syntax [13:24] <saste> durandal_1707, did you manage to fix framesync or what? [13:25] <saste> from my test displace is unusable if you have non synthetic video [13:25] <durandal_1707> saste: it works if you save geq stuff into file [13:26] <durandal_1707> anyway the example you use have static maps [13:26] <durandal_1707> and there is no point in recomputing same image over and over again [13:28] <saste> durandal_1707, the example I use comes from the manual page [13:30] <durandal_1707> so what? [13:31] <saste> durandal_1707, it should work out of the box [13:31] <saste> if not you should specify the conditions under which it works [13:32] <saste> ideally we should fix what it needs to be fixed/extended [13:46] <durandal_1707> saste: so it should be fixed now? [14:10] <durandal_1707> saste: it is because geq is slow [14:11] <durandal_1707> and lavfi sucks [14:11] <durandal_1707> instead of slowing down decoding of first input it just continues... [14:14] <durandal_1707> same apply if filter output multiple frames at once, lavfi will happily cache all of them and eat memory [14:15] <wm4> jit'ed geq? [14:16] <durandal_1707> geq slowness is not problem [14:39] <durandal_1707> yes, trolling success [14:40] <ubitux> :') [15:00] <cone-897> ffmpeg.git 03Ronald S. Bultje 07master:1b3a7e1f42c3: avcodec/x86/videodsp: Properly mark sse2 instructions in emulated_edge_mc x86 simd as such. [15:20] <ubitux> player ubitux enters the game [15:22] <Compn> the best ones are the '_______ sucks' ones :) [15:23] <Compn> nice one ubitux :) [15:23] <ubitux> oh come on wm4 [15:23] <mateo`> well i guess i can join the game and add more trolling, s/vapoursynth/gstreamer/ [15:23] <ubitux> :D [15:24] <wm4> hm I guess I should finish my vsynth demuxer and submit it [16:36] <durandal_1707> saste: if i use movie=matrixbench... i do not get queue overflow [16:37] <saste> everyone complains that lavfi sucks, and so? [16:38] <durandal_1707> but video pts starts from 0 [16:38] <durandal_1707> so i wonder why movie filter helps [16:38] <durandal_1707> perhaps its slowness match geq.... [16:40] <saste> first thing, try to improve ffsynch debugging [16:41] <saste> durandal_1707, what is happening with regards to pts? [16:41] <durandal_1707> will see, i can reproduce same issue with overlay filter, so i will troll little about it now.... [16:52] <saste> durandal_1707, ah ok, it is explained in the ticket [16:52] <saste> i'll rename the ticket... [16:52] <durandal_1707> what is explained? [16:53] <durandal_1707> if you rename ticket, thus not accepting truth, i will rm -rf libavfilter/libmpcodecs [16:56] <saste> bounty?? [16:57] <saste> who added the bounty? [16:59] <saste> how do you measure the quality of encoded files? [16:59] <Compn> psnr ? [16:59] <saste> is something better available? ssim or something? [17:00] <durandal_1707> drawtext='acceptable' [17:03] <durandal_1707> vf_compare from dark side have dssim [17:05] <durandal_1707> also there is blend=all_mode=phoenix [17:06] <Compn> ssim should be , yes [17:06] <Compn> but both are somewhat ... not great indicators [17:09] <saste> durandal_1707, dssim... was it a patch from lu_zero? [17:09] <saste> why wasn't it never integrated? [17:16] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:f7106e00f7f1: avformat/rmdec: fix return code of ff_rm_parse_packet() [17:16] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:aaf02f6ede55: avformat/rmdec: add some error messages [17:20] <durandal_1707> saste: how do I know? [17:21] <durandal_1707> saste: you interested in bounty? [17:22] <saste> durandal_1707, from other developers? [17:23] <saste> that's a sort of taboo [17:23] <saste> also from what i know bounties on trac never worked [17:28] <Compn> how is that a taboo > [17:33] <cone-897> ffmpeg.git 03Paul B Mahol 07master:ea9632b7470e: avcodec/flicvideo: use avpriv_request_sample() [17:43] <durandal_1707> michaelni: do not merge libav fieldorder changes as they are incorrect [19:55] <saste> burek, ping on docu patch [19:56] <durandal_1707> saste: you just need to specify what bounty you want, best bidder wins [19:58] <llogan> michaelni: can you grep for rob.schmidt86 in the mail logs? I would but I'm not a sudoer. [19:59] <llogan> (and I don't know my password...just the ssh key) [20:04] <durandal_1707> llogan: password for what? [20:05] <llogan> avserver. i wanted to see if i was in sudoers, but can't test without a passwd [20:05] <llogan> someone is having trouble subscribing to libav-user [20:05] <llogan> exciting. carry on. [20:06] <durandal_1707> avserver? [20:06] <llogan> ffmpeg.org [20:07] <durandal_1707> so you lost you password, and now ask for it via irc? [20:07] <llogan> not at all. [20:07] <llogan> i was only issed a login via ssh key and i'm not asking for the passwd via irc. [20:09] <durandal_1707> so how we should know you are really llogan ? [20:11] <llogan> i've just added the mod bit to your address in -devel [20:12] <durandal_1707> what? [20:12] <llogan> you wanted proof. [20:12] <llogan> ok...i removed it. [20:13] <durandal_1707> i didn't notice change [20:21] <michaelni> mraulet, there are some hevc bugs on trac (https://trac.ffmpeg.org/query?status=new&status=open&status=reopened&keywor…) [20:23] <mraulet> ok [20:25] <mraulet> could you cc me hevc related bugs? [20:39] <michaelni> not sure i understand but you can add yourself to the CC list of any bug or you could subscribe to https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac/ to get all bug changes (which is alot), or i could try to notify you when i see new hevc bugs if thats what you meant but i tend to forget doing such things, carl tends to be better at such things [20:43] <mraulet> I don't want to get all of them [20:43] <mraulet> :-) [20:47] <michaelni> saste, did you see https://trac.ffmpeg.org/ticket/3054 ? (doc uninstall regression since "build: drop unused ALLMANPAGES variable") [20:54] <saste> michaelni, i'll fix it [20:54] <michaelni> saste, thx [20:56] <durandal_1707> michaelni: i do not abuse trac, i want to make critical issues as such [21:02] <michaelni> see doc/issue_tracker.txt for a description of each priority [21:03] <durandal_1707> heh, than i will fork [21:06] <saste> durandal_1707, i found the perfect name for it [21:06] <saste> yaff [21:07] <saste> yet another ffmpeg fork [21:10] <llogan> i prefer ffffffmpeg [21:11] <saste> https://trac.ffmpeg.org/tags => cool [21:21] <cone-698> ffmpeg.git 03burek 07master:229042a52d5d: doc/protocols: add description for the RTP protocol [21:21] <cone-698> ffmpeg.git 03Stefano Sabatini 07master:af1c5388501e: doc/Makefile: fix man pages uninstall path [21:21] <cone-698> ffmpeg.git 03Ingo Brückl 07master:6c9c636c0022: lavd/fbdev_common: fix compilation with older kernels [21:29] <saste> ^ trac fight [21:31] <michaelni> mateo`, can you take a quick look at a 1 line mxf patch "1024 21:28 To FFmpeg devel (0.8K) [FFmpeg-devel] [PATCH] avformat/mxfdec: dont truncate packets" ? [21:31] <durandal_1707> saste: more like fatchick fight [21:58] <durandal_1707> query_formats is called before init? [22:57] <wm4> I'm seeing a ffmpeg muxed file that has no audio frame marked as keyframe [22:57] <wm4> err, it's a matroska file [22:59] <wm4> are there even audio codecs where a a matroska "block" isn't on key frame boundaries? [23:05] <Compn> audio has keyframes ? [23:06] <wm4> in matroska it does [23:06] <JEEB> that marker in matroska is a misnomer [23:06] <JEEB> it means "random access point" [23:07] <wm4> yeah [23:07] <wm4> anyway, mplayer2/mpv's demuxer can't handle it if no audio block is marked as keyframe [23:07] <Daemon404> the bigger question is [23:07] <wm4> so I'm pretty sure almost any mkv file I looked at had these more or less correctly marked [23:07] <Daemon404> who the hell muxes mkv with ffmpeg? [23:07] <Daemon404> and why? [23:07] <wm4> the file I'm looking at is named gst-hackfest-2013-360p.webm [23:08] <wm4> | + Muxing application: Lavf54.29.104 [23:08] <wm4> | + Writing application: Lavf54.29.104 [23:08] <Daemon404> gstreamer. got it. [23:08] <Daemon404> AIDS. [23:08] <wm4> :) [23:21] <ubitux> BBB: 4x4 should be done by the end of the week end [23:21] <ubitux> i'm almost there [00:00] --- Fri Oct 25 2013

1 0

ffmpeg.log.20131024
by burek 25 Oct '13

25 Oct '13

[00:17] <zennist> Hi does anyone knows how to get the current playing time from libavcodec [00:18] <zennist> like ftell for files? [03:05] <dbro> Anyone have any pointers re: constructing AVPackets given raw encoded access units? I'm performing encoding with a hardware encoder and using ffmpeg for processing / muxing [03:05] <dbro> Can't find a suitable looking convenience function to accomplish this, and manually setting AVPacket fields seems daunting and doomed to fail (as it currently is) [03:07] <dbro> (Android's) HW encoder only provides a ByteBuffer for each encoded frame, as well as pts, length, offset and flags. No dts as AVPacket seems to require& see http://developer.android.com/reference/android/media/MediaCodec.BufferInfo.… [03:08] <dbro> help me birth a FOSS HTTP HLS server for Android (and iOS too!) [04:04] <netsrot> does running killall ffmpeg or pressing ctrl+c to stop recording work just as good as pressing q? [04:18] <netsrot> brb 10h [05:07] <Ulfalizer> i'm seeing errors along the lines of http://pastebin.com/gBDWKv47 when encoding with CODEC_ID_MPEG1VIDEO, and the resulting output file generates "internal data stream error" errors in ubuntu's Videos video player as well. if i do mpeg4 instead everything seems to work fine. i also saw similar errors for mpeg1 with libavformat/output-example.c (which i based my code on), though no playback errors (guess it might be becaus [05:08] <Ulfalizer> could paste some code, but thought i'd check if it's a well-known problem first [05:41] <Ulfalizer> tried setting rc_buffer_size in the AVCodecContext to a few different values too. that got rid of the warning at the beginning, but not the rest. [13:11] <netsrot> Hi, any idea why this file only plays in mplayer and not vlc? "ffmpeg -f x11grab -r 3 -s 1920x1200 -i :0.0 -vcodec libx264 -vf scale=640:360 game1.mkv" [13:11] <durandal_1707> netsrot: perhaps vlc does not support yuv444p? [13:12] <zap0> cause VLC can't play it. [13:12] <durandal_1707> you could also use '-vcodec libx264rgb' for better quality [13:12] <netsrot> ok arg should I change so vlc can play it? [13:13] <durandal_1707> -vf scale=640:360,format=yuv420p... [13:14] <netsrot> ok thanks I will try that. [13:14] <zap0> why don't you first find out what VLC can play. [13:15] <netsrot> I prefer mplayer but other ppl use vlc and needs it to work... [13:21] <durandal_1707> netsrot: tell them to update VLC [13:21] <zap0> or get something that can play videos. [13:22] <netsrot> durandal_1707: thanks, will try that. [13:23] <durandal_1707> i do not have VLC, but i hope it is fixed in latest, if not than complain on #videolan [13:24] <netsrot> btw does ffmpeg exit gracefully when pressing ctrl+c or running killall ffmpeg? just like pressing q? or would it be better to make a fifo to pipe q to ffmpeg? [13:35] <netsrot> was it a stupid question or how do ppl exit ffmpeg? I don't want the terminal to show in the recording. [13:50] <durandal_1707> netsrot: you could try it... [14:01] <netsrot> piping q to fifo seems to work so I'll go with that as I'm affraid of getting a corrupted ending. [14:04] <durandal_1707> well with ctrl+c you should not get corrupted ending [14:14] <netsrot> ok, thanks a lot. [15:38] <bouba> hi all, about H264 vids, anyone's familiar with lagging problem on last adobe flashplayer, but no log on ffplay, vlc or other players ? [15:38] <bouba> the videos are read locally [15:38] <bouba> but no lag* on ffplay, vlc, ... [15:54] <netsrot> I tried changing to format=yuv420p but it still doesn't work in vlc 2.0.7 =( [15:55] <plm> Hi all [15:55] <netsrot> and using libx264rgb I only get green screen. [15:55] <netsrot> Hi plm [15:58] <netsrot> The problem seems to be that I use -r 3. When I change that to -r 25 it works... [15:58] <plm> The ffmpeg/ffserver support HTTP Dynamic Streaming? [16:06] <plm> netsrot: ^^ [16:11] <bouba> i just dont understand why flash is lagging and not ffplay (locally) [16:45] <durandal_1707> bouba: flash is slow [16:45] <tmm__> hey all, I'm trying to compile ffmpeg for the iOS simulator running iOS 7.0, with Xcode 5 command line tools and I get a /Applications/Xcode5.app/Contents/Developer/Platforms/iPhoneSimulator.platform/Developer/usr/bin/gcc is unable to create an executable file. error [16:45] <tmm__> console.log file is here http://pastebin.com/cHtPnYq3 [16:46] <bouba> durandal_1707 : but, once I reencode the vid into ffmpeg, it is tagged as "CFR" instead of "VFR"...and it works without lag [16:46] <durandal_1707> ask flash devs [16:46] <bouba> i use MediaInfo and FFprobe for infos [16:46] <bouba> ok [16:50] <plm> anyone know about ffmpeg/ffserver support for HTTP Dynamic Streaming? I would like a adatative bitrate streaming over box1 and box2. [16:56] <spaam> ffmpeg can make the files IIRC [16:57] <bouba> http://sourceforge.net/projects/opensvcdecoder/ [17:12] <plm> bouba: hmm... maybe SVC is the solution. I will to check. I think actually the adobe support HTTP Dynamic Streaming. I need just streaming, not encoding. box1 streaming a video to box2 and between box1 and box need to adjust on fly the bitrate.. [17:13] <bouba> plm: SVC is "adaptive", perhaps you need "progressive" [17:14] <bouba> ffserver supports http progressive [17:14] <plm> bouba: what is difference? [17:16] <bouba> plm: progressive cannot modify the stream quality "on-the-fly" [17:17] <bouba> plm: vlc-cli is able to stream HTTP too [17:18] <bouba> as ffserver [17:21] <plm> bouba: So I need the adaptive. I need t adjust bitrate on the fly becouse bewteen box1 and box2 I have unstable networking. box1 has as input a local network rtsp video, and box1 need to transmit to box2 with always best quality without stop/congeals. [17:21] <plm> bouba: vlc-cli? [17:22] <bouba> plm: sorry, vlc-nox [17:23] <bouba> but, imho ffserver as more features [17:23] <plm> bouba: I'm actually using VLC to streaming that: cvlc -vvv rtsp://192.168.0.40 --sout '#standard{access=http,mux=ogg,dst=187.71.63.24:8080}' --network-caching=5000 [17:24] <plm> bouba: yes, vlc does not that adjust bitrate on the fly [17:24] <plm> bouba: sorry, ffserver adjust on the fly bitrate? [17:24] <bouba> plm: im not a guru, but imho, non, they do not support it yet [17:28] <plm> bouba: I think just adobe is capable to adjust bitrate on the fly :-( [17:28] <plm> bouba: or that SVC project..(need to check) [17:30] <bouba> plm: in order to adjust on the fly, the media file needs to be segmented, check the HLS protocol by Apple too [17:30] <bouba> plm: http://stackoverflow.com/questions/1093667/http-live-streaming-ffmpeg-ffser… [17:39] <plm> bouba: ok [17:39] <plm> bouba: I think HLS is the solution.. [17:40] <plm> bouba: what I know about HLS is that it create many video streaming bitrates and client choice what video bitrate get right? [17:40] <plm> So the client can change between video with high bitrate to low bitrate and vice versa on the fly.. I hope that works =D [18:31] <zennist> Hi I'm playing around with libavcodec's AVIOContext. What I only do is to simulate the default io implementation by just reading / seeking a file handle. But avformat_open_input keeps complaining moov atom not found for mp4 files (which are perfectly playable with ffplay, or with the default io implementation without my custom avio context) Any ideas? [18:40] <namhub> oh hai [18:43] <namhub> zennist: 1) I've never used avio.h 2) if I'm not mistaken, you need to first open the container with libavformat [18:43] <namhub> 3) then you read a packet, which you use libavcodec to decode. [18:44] <zennist> namhub: yes I've already done that [18:44] <namhub> zennist: show me your code [18:45] <zennist> namhub: pastebin.com/6tyKFAz2 [18:46] <zennist> it can play mp3's no problem; just not for aac [18:46] <zennist> interestingly as I said if I'm not using my custom avio then everything's fine [18:48] <namhub> why avio, might I ask? [18:49] <zennist> namhub: I'm testing this for streaming music [18:50] <namhub> hmm, did you qt-faststart your mp4? [18:50] <zennist> so I actually needs the custom io functions to get the bytes from network; avformat declares it supports url but in reality I haven't got it to work [18:50] <zennist> no [18:50] <namhub> well mp4 moov atoms aren't at the beginning of the file, normally [18:50] <zennist> but the fact is that I'm now testing with my file on my disk; so it should be able to see the moov atom at the end [18:50] <namhub> well, did you actually read that part? [18:51] <zennist> and in fact it works when I'm not using my avio; so ffmpeg's own implementation does the trick [18:51] <zennist> yes I get your point [18:51] <namhub> i'm guessing the reference implementation does hackery to make this work right [18:51] <namhub> which you'll also need to magic up ;p [18:51] <zennist> but my point is that the file is complete, so it should still work [18:51] <namhub> dunno; perhaps ask the ML [18:52] <zennist> ML? [18:52] <namhub> mailing list [18:52] <zennist> OK. Thanks still. [18:52] <zennist> I'm reading the source code... Hopefully I can see the part I need... [18:56] <namhub> let me know if you find the magic ;p [18:57] <namhub> I've been piddling with libav* for awhile too; but I'm no expert obviously. [19:59] <Borys> Hello, I have an issue with using VP8 with FFMPEG. I am getting error 22 with avcodec_open2() when using avOutputFormat->video_codec = AV_CODEC_ID_VP8;. I cross-compiled ffmpeg with libvpx with --enable-libvpx option. AV_CODEC_ID_VP9 works. What should I take a look at? [20:00] <saste> Borys, what's the error message? [20:00] <Borys> Is there a way to read it? [20:03] <saste> Borys, just copy&paste stderr [20:03] <saste> it is printed there by default [20:03] <saste> 22 = EINVAL [20:03] <Borys> I am actually using the libraries in C [20:03] <Borys> not the command line tool [20:04] <saste> Borys, still the library will log to stderr [20:04] <Borys> Ok, one moment, I'll see how to read it. I should mention that this the libraries have been cross compiled for Android via the Android NDK [20:06] <saste> Borys, also only vp8 decoding is supported [20:06] <Borys> Ah, encoding is not? [20:06] <Borys> I think there's my problem then :) [20:06] <Borys> Is VP9 encoding supported? [20:07] <saste> Borys, if you read the log it will tell you, that's what it's meant for, the error code alone is not that useful [20:07] <saste> Borys, i think so, it was added a few weeks ago (only decoding) [20:07] <ska> I can't seem to record video using: ffmpeg -video_size 1024x768 -framerate 25 -f x11grab -i :0.ffmpeg -video_size 1024x768 -framerate 25 -f x11grab -i :0.0+100,200 -f alsa -ac 2 -i pulse output.flv 0+100,200 -f alsa -ac 2 -i pulse output.flv [20:07] <Borys> Here it mentions that both VP8/VP9 encoding is supported via libvpx: http://ffmpeg.org/general.html#Video-Codecs [20:08] <saste> indeed [20:08] <Borys> I have to use the experimental flag to enable it [20:08] <ska> I don't have pulseaudio.. Is there any way I can use pure alsa? [20:08] <ska> ok.. [20:10] <ska> http://pastebin.com/f43U26Bk [20:14] <Borys> Is there a specific function that I can use to read from stderr? My console output only has what I am logging http://pastebin.com/39vg2akC [20:15] <saste> Borys, to read from stderr, you need the log and your eyes (or the equivalent if you're blind) [20:15] <ska> I have a USB device, but I don't know how to address it for ffmpeg [20:16] <Borys> This is the only line in my log that is present when the line executes: E/ERROR(16257): CRASHING! Could not open codec: -22 (where "ERROR" is my ERROR_TAG from the pastebin) [20:16] <llogan> ska: that's a fake ffmpeg [20:17] <llogan> http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [20:17] <Borys> I think I found my logging issue, I need to manually redirect stdio in Android. I will go try it now. [20:17] <ska> llogan: should I be using avconv? [20:17] <llogan> ska: if you want to continue using crappy, fake ffmpeg then go to #libav [20:18] <llogan> if you want to use avconv then go to #libav. [20:18] <llogan> avcorn also isn't ours [20:18] <saste> avconf? [20:20] <Borys> saste: I will be unable to log stderr as Android redirects that output to /dev/null. The only way to read it is to have a rooted device, which I do not have access to :( [20:21] <Borys> Source: http://stackoverflow.com/questions/10531050/redirect-stdout-to-logcat-in-an… [20:21] <saste> Borys, ok, you can rediret stderr to a file [20:21] <saste> *redirect [20:21] <saste> see the file libavutil/log.h [20:22] <saste> basically you need to specify a log callback [20:22] <Borys> I'll check it out [20:32] <Borys> It is the av_log_set_callback function, right? [20:32] <ska> Ok, I installed the "real" ffmpeg: http://pastie.org/8427546 [20:46] <ska> Now my recording hangs on libpostproc : http://paste.debian.net/61323/ [20:48] <llogan> ska: never seen that one before... [20:49] <llogan> and the 0.10 branch is considered old [20:53] <llogan> are you doing this via x11 forwarding? [20:54] <ska> No, just on same system [20:55] <llogan> try -s 1920x1200 instead of the xwininfo stuff [20:56] <ska> llogan: Same result.. I thought of that too. [20:56] <ska> I have the strace of the place it hangs.. [20:56] <llogan> damn, because for me the xwininfo output is not a useable format [20:57] <llogan> the next step is to confirm that ffmpeg from git head does or does not work. the guide i linked to does not interfere with system [20:58] <llogan> and you can skip all external libraries except yasm and libx264 (and you probably want libfdk-aac) [21:00] <llogan> damn. ubuntu 13.10 is out already and now i have to test it for that guide... [21:01] <Borys> What's the best (fastest w/ moderate quality) non-gpl encoding library? I was going to use libx264, but the licensing fee is too much for me to be able to afford. [21:01] <Borys> sorry, codec, not library [21:07] <Borys> saste: I was able to get the logging working! [21:07] <Borys> http://pastebin.com/66ce9xEf [21:11] <plm> HLS can have as input a rtsp video? [21:12] <plm> actually I have this # ./ffmpeg -y -i sintel_trailer-480p.mp4 -pix_fmt yuv420p -vcodec libx264 -acodec libfaac -r 25 -profile:v baseline -b:v 1500k -maxrate 2000k -force_key_frames 50 -s 640x360 -map 0 -flags -global_header -f segment -segment_list /var/www/hls/index_1500.m3u8 -segment_time 10 -segment_format mpeg_ts -segment_list_type m3u8 /var/www/hls/segment%05d.ts [21:13] <plm> I would like to substitute the -i sintel_trailer-480p.mp4 by -i rtsp://192.168.0.2. That will works? I will have the segmentation on realtime video or with HLS need to be offline video? [21:23] <llogan> you can always just give it a try [21:31] <plm> llogan: I'm not have a real time video (rtsp) here. But I would like to know if are there logical HLS with real time video.. [21:37] <plm> ahaha I try here -i http://192.168.0.3:8082 that is a real time video from camera, and ffmpeg don't stop to create segments. Why? [21:37] <plm> So it are waiting final file in http to not create more files HLS? [21:38] <plm> So HLS not works with real time video? [21:47] <Shiba> can someone explain to me or point me to documentation regarding why "--enable-nonfree" makes the resulting binary non-redistributable? [21:49] <JEEBsv> that's what it is [21:49] <JEEBsv> you are using code that you cannot feasibly mix with LGPL or GPL [21:49] <JEEBsv> examples: fdk-aac needs you to license AAC to distro, faac and aac+ are just plain incompatible [21:53] <Shiba> a-ha. So its a licensing (and thus a legal) issue regarding how certain modules are linked together in the resulting binary? [21:55] <Shiba> presumably, though, content produced with the non-redistributabe binaries is itself redistributable [21:56] <Shiba> i.e. if I produce a video and encode it with such a build of ffmpeg, no restrictions are placed on the video itself? [21:59] <pzich> anyone familiar with getting an iOS-friendly h.264 video? [22:00] <llogan> pzich: which device(s)? [22:00] <pzich> iPad 2+ primarily, iPhone if possible [22:00] <llogan> which iPhone(s)? [22:00] <pzich> probably 4+ [22:01] <pzich> I finally got this working correctly with Quicktime on desktop using the yuv420p pixel format, but it's still failing to work on iPad [22:01] <pzich> also, this served via a <video> tag [22:02] <llogan> main profile, level 3.1 [22:02] <llogan> maybe some other restrictions [22:02] <llogan> as well [22:02] <pzich> I'll check it out, thanks [22:03] <llogan> consider adding "-movflags +faststart" [22:03] <pzich> does that do the same as the qt-faststart tool? [22:03] <llogan> yes [22:03] <pzich> ah, nice [22:04] <llogan> pastebin your command and complete output if you want suggestions, flames, etc [22:04] <pzich> yeah, I'll try fiddling with this and get back to you [22:06] <llogan> pzich: i added a chart here https://trac.ffmpeg.org/wiki/x264EncodingGuide#Compatibility [22:06] <llogan> s/chart/table [22:07] <llogan> that page is getting messy. i blame rogerdpack. [22:07] <pzich> ah excellent, I was just going to ask if there was a chart like that [22:07] <llogan> i stole it from apple [22:19] <pzich> llogan: cool, this is working, thanks [22:19] <pzich> looks like part of the problem was the annoying maximum video dimensions they set [22:19] <pzich> can't play a video 2048px wide on there [22:21] <llogan> pzich: see scale filter http://ffmpeg.org/ffmpeg-filters.html#scale [22:21] <llogan> specifically "-1" and also -force_original_aspect_ratio [22:24] <pzich> yeah, I figured out scaling, I was just hoping to use the same (larger) video we were using for desktop [22:31] <Borys> I figured out my issue with libvpx: I had to disable the realtime only flag to get rid of error 22 on avcodec_open2 [22:32] <Borys> when working with vp8* [23:44] <dbro> Trying to create AVPackets from a hardware encoder's output, getting a SIGFPE in av_interleaved_write_frame. See https://github.com/OpenWatch/FFmpegWrapper-Android/blob/master/FFmpegWrappe… [23:45] <dbro> using gdb w android's ndk is proving difficult, but I'm working on that as well. Thanks for the braincycles! [00:00] --- Fri Oct 25 2013

1 0

ffmpeg-devel.log.20131023
by burek 24 Oct '13

24 Oct '13

[00:00] --- Wed Oct 23 2013 [00:35] <burek> is there any interest to implement some kind of a remote control for ffplay [00:36] <burek> in a way, for example, to make it listen to a tcp connection and get the playback commands out of there [00:36] <burek> rather than from the keyboard [00:37] <llogan> burek: what's the use case? [00:37] <burek> i assume you didn't read #ffmpeg but for example [00:37] <burek> multiple players could be synced remotely [00:37] <burek> to play the content in sync [00:37] <burek> just by simple seeking [00:38] <burek> and btw, vlc has got that feature with a lot of different types of controllers (telnet, http, rc, etc) and its quite cool and useful [00:39] <llogan> ah. i didn't see #ffmpeg yet... [00:39] <burek> it's a long conv [00:39] <burek> in short, multiple players synced from 1 point to display the content in sync [02:38] <Ruler2112> Hello. I have a suggestion about the elimination of a restriction present in ffmpeg. [02:40] <Ruler2112> I wrote a script many years ago to standardize digital video into a format that my TV can reliably play. It uses several tools, ffmpeg and virtualdub among them. Driving VirtualDub via command line, I set the ISFT metadata to be the name of my script. (This way, I could tell right away which have been processed & which have not.) [02:42] <Ruler2112> I'm currently re-working the script and eliminating dependencies. I was hoping to do everything with ffmpeg, but the ISFT (encoder/encoded with) flag is locked to be Lavf####. I would like to suggest that this field default to the string denoting the version of Lib AV Format, but allow it to be overridden by setting the command line parameter at the invocation of ffmpeg. [02:47] <Ruler2112> I do have an alternate plan to change this metadata field after the file has been generated, but it would be much more streamlined to simply do it via ffmpeg. (Plus, though I've nearly eliminated VirtualDub as a dependency, I'll need to add another dependency to change the metadata field.) Having it locked to a set value when other tools allow it to be changed is just silly IMO. [02:52] <cone-487> ffmpeg.git 03Michael Niedermayer 07master:60ea0a5236ce: avcodec/sgi: Fix dereferencing uninitialized pointer [02:52] <Ruler2112> join #ffmpeg [03:42] <BBB> ubitux: I think that's the same as reported ~1 week ago, it's sse2 instructions in emulated_edge_mc [03:42] <BBB> I sent a patch for that but it wasn't applied yet [03:42] <BBB> I haven't been able to look at it since then [08:30] <wm4> so libswscale calls mprotect() to make mmap'ed memory executable for god knows what reasons [08:31] <wm4> but it fails to check the mprotect() return value, so it'll just crash on "hardened" systems which do not allow self-modifying code [08:41] <Zeranoe> Is there any way to compile FFmpeg without TLS (thread-local-storage)? [09:31] <michaelni_> wm4, sws its used for the fast bilinear scaler which generates at runtime some optimized code, ill look at the return checks unless someone else is quicker [09:32] <wm4> isn't fast bilinear said to be broken anyway? [09:33] <michaelni_> last i tried it worked [09:35] <michaelni_> Zeranoe, iam not sure atm where we use thread locals but --disable-pthreads or something like that would be the sledge hammer solution to avoid that [09:35] <michaelni_> what problem do these thread locals cause ? [09:37] <cone-490> ffmpeg.git 03Derek Buitenhuis 07master:3511d4fc9784: pthread: Avoid crashes/odd behavior caused by spurious wakeups [09:37] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:d0f0ac426bdf: Merge commit '3511d4fc9784d5fbb024dce68ca7a0d7fdd74663' [09:38] <Zeranoe> michaelni_: Just someone asking for a version without it, I'm sure everything is fine [09:54] <cone-490> ffmpeg.git 03Derek Buitenhuis 07master:b1fcdc08ceb5: nut: Fix unchecked allocations [09:54] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:078abccdd6dc: Merge commit 'b1fcdc08ceb5df69fac34aa0d57c56905d32b8b4' [10:00] <cone-490> ffmpeg.git 03Alex Converse 07master:f457edb69794: aacdec: Use avpriv_report_missing_feature() instead of custom logging. [10:00] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:e0f291822306: Merge commit 'f457edb69794c7e13b3630456d72f7942fff19f7' [10:11] <cone-490> ffmpeg.git 03Alex Converse 07master:b3be41ca8252: aac: Add support for Enhanced AAC Low Delay (ER AAC ELD). [10:11] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:fdb48225592c: Merge remote-tracking branch 'qatar/master' [10:15] <durandal_1707> michaelni_: why you reverted my change to sgidec? [10:18] <durandal_1707> actually you should revert my change instead.... [10:18] <durandal_1707> of adding another commit [10:19] <durandal_1707> and this case looks to not be properly covered at all [10:19] <durandal_1707> otherwise fate would be yellow [10:20] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:eb4205cc8957: swscale/utils: check mprotect() return code [10:22] <michaelni_> durandal_1707, i also realized that fate doesnt cover it but out encoder can generate files that trigger the issue [10:22] <michaelni_> needs -coder 2 IIRC [10:23] <durandal_1707> anyway that raw sgi should use planar rgb [10:32] <durandal_1707> michaelni_: what happened to crash with unused pad i reported? [10:40] <durandal_1707> michaelni_: something fisshy happened in aac merge [10:41] <durandal_1707> i have big warning here [10:41] <durandal_1707> lol it from libav [10:47] <michaelni_> durandal_1707, "unused pad" ? which ticket or mail was that or how do i reproduce ? [10:48] <durandal_1707> ffmpeg -f lavfi -i color -f lavfi -i color -filter_complex "[1]null[x],[0][1]overlay" -f null - [10:48] <durandal_1707> do you get crash? [10:51] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:72cf47b241c7: avcodec/aacdec: fix avpriv_report_missing_feature() arguments [10:51] <durandal_1707> AAAAAAAAAAAAAAAAAAWW [10:52] <durandal_1707> i had same commit [10:52] <durandal_1707> next time i will not share my precious findings to anyone [10:54] <michaelni> durandal_1707, yep crashes [10:55] <durandal_1707> michaelni: lol you left 3 unfixed [10:58] Action: durandal_1707 oh i shared it again [10:58] <durandal_1707> michaelni: no, i'm not going to fix it as i removed my version [11:09] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:435a730e2190: avcodec/aacdec: fix more avpriv_report_missing_feature() arguments [11:24] <j-b> 'morning [11:36] <EKRAM> Greetings to all people show makes and contributes such a wonderful piece of software that makes wonder [11:36] <EKRAM> i have question regarding the Distributed Encoding of a single Videos using ffmpeg [11:37] <EKRAM> so that i can use spread a big single file over different encoders and finally stitch them togather [11:37] <EKRAM> is it possible or would you give me a suggestions how can i achive it [11:43] <j-b> mraulet: ever heard of hevc in avi and flv? [11:43] <mraulet> no [11:43] <mraulet> I am a crap mpeg guy :-) [11:43] <j-b> mraulet: well, now, you do :) [11:44] <mraulet> where? [11:45] <j-b> uploading now [11:48] <durandal_1707> someone just hacked ffmpeg [11:52] <JEEB> note: the hevc-in-flv shit has NOTHING to do adobe [11:52] <JEEB> it's a random chinese corp hacking gabest's dshow muxer [11:52] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:b36257921e52: ffmpeg: Dont crash on unconnected output pads before an error is printed [11:53] <JEEB> *to do with adobe [11:54] <JEEB> and the avi samples are just random people taking the dshow flv demuxer's output and pushing it into some dshow avi muxer >_> [11:55] Action: durandal_1707 bans random people [11:56] Action: michaelni thinks 90% of avi is from random people pushing random things in it [11:56] <JEEB> also as I noted elsewhere, surprised j-b didn't get a sample of hevc-in-asf yet :P that crap is already flowing through doom9 because someone decided it was a good idea to remux avi via dshow to asf [11:56] <JEEB> thankfully, the flv/avi/asf crowd seems to be just a single person [11:56] <JEEB> or so [11:58] <durandal_1707> i cant get this: geq=r=X+128 + 30*sin(2*(PI/200)*((Y-50*T))):g=X + 128 + 30*sin(2*(PI/200)*((Y-50*T))):b=X + 128 + 30*sin(2*(PI/200)*((Y-50*T))) [11:58] <durandal_1707> to be continuous, eg not change from 0 -> 255 [12:09] <Daemon404> [10:56] * michaelni thinks 90% of avi is from random people pushing random things in it <-- has anyone managed to mux an avi stream withn an avi yet? [12:10] <Daemon404> and do we support ASS/SSA in avi? :D [12:10] <wm4> no [12:10] <Daemon404> aww [12:10] <Daemon404> because thats a real thing [12:10] <Daemon404> for ages [12:10] <wm4> I know, I found such a thing recently on the samples server [12:11] <wm4> anyway, ffmpeg's avi demuxer generally behaves somewhat worse for incomplete files compared to mplayer's unmaintained crap [12:11] <Daemon404> even with indexing? [12:12] <wm4> well, mplayer requires you to pass -idx to make it seekable, but yes [12:12] <michaelni> wm4 upload samples + open ticket [12:13] <cone-490> ffmpeg.git 03Clément BSsch 07master:e01fba556213: movenc: Add an F4V muxer [12:13] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:76424b2da850: Merge commit 'e01fba5562133ac56a4f44c6f214a80ee33cb388' [12:20] <cone-490> ffmpeg.git 03Diego Biurrun 07master:f6633c55a3c0: avfilter: Fix typo in Loren's email address [12:20] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:f4f8499c1973: Merge commit 'f6633c55a3c0e93a5b2bab6aa0692fb608f2a38d' [13:12] <cone-490> ffmpeg.git 03Luca Barbato 07master:72072bf9de32: mpegvideo: Drop a faulty assert [13:12] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:3d3a7f5034a2: Merge remote-tracking branch 'qatar/master' [14:17] <BBB> pff [14:17] <BBB> ubitux: I'll look again [14:18] <BBB> do we have a sse/32bit x86 fate station [14:18] <BBB> ? [14:43] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:867e7bb4f101: fix order of operations in comments [14:43] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:1e2ab9846076: avcodec/vc1dec: use logical operation instead of bitwise for twomv [15:20] <durandal_1707> yes!, here goes ultimate filter [15:46] <cone-490> ffmpeg.git 03Robert Krüger 07master:83922cccb207: avfilter/vf_scale: Correct vf_scale mappings from arguments to color range [15:59] <burek> was there any attempt to implement the decoding of ARF files (cisco webex video) in ffmpeg? [15:59] <durandal_1707> no [16:02] <burek> ok, thanks [16:20] <funman> michaelni: do you have a plan for removing CODEC_CAP_EXPERIMENTAL on hevc ? Just let a few weeks pass and see how it performs? [16:20] <durandal_1707> once it is no longer experimental [16:21] <funman> I'm asking how to help [16:21] <funman> not when it will be done [16:22] <durandal_1707> for help send patch [16:25] <durandal_1707> burek: do you have sample? [16:26] <michaelni> funman, i dont really have a plan, the removial time depends on the authors/developers. that is if you post a patch and noone objects i guess it can be removed. maybe ill wait till after 2.1 though with applying it in that case, not sure ... [16:28] <cone-490> ffmpeg.git 03James Almer 07master:c994b42830cb: lavf/adtsenc: Add support for APE tags [16:29] <j-b> APE tags on ADTS out? brrr [16:30] <durandal_1707> users reports such bugs, and carl forgets to close them as invalid/wontfix [16:31] <Daemon404> or users report valid bugs, and carl loses them as invalid/wontfix [16:31] <Daemon404> or "full uncut output" [16:32] <durandal_1707> admit you are lazy to provide "full uncut output" [16:44] <saste> michaelni, ping on "lavu: add av_get_color_string() and use it in -colors option" [16:45] <durandal_1707> you copy/pasted description from source code? [16:48] <mraulet> michaelni, funman, smarter, I think we can go without CODEC_CAP_EXPERIMENTAL [16:48] <mraulet> *go on [16:54] <saste> any reason to prefer "avcodec:" to "lavc:"? [16:54] <saste> (in commit messages) [16:55] <durandal_1707> i think i already answered it [16:56] <saste> durandal_1707, we do it because libav does it? [17:00] <Compn> burek : i found arf/wrf samples the other day... should i put them on samples repo ? [17:01] <michaelni> saste, patch should still be fine [17:07] <durandal_1707> saste: we do it because you do not ;) [17:08] <durandal_1707> so there are devs that do use lavf/lavc.. and others that use avformat/avcodec/... and they hate each other [17:10] <saste> durandal_1707, ok let's start another fork ;) [17:11] <ubitux> i switched to avformat/avcodec/..., i don't mind at all lavf/lavc/... [17:11] <ubitux> <@saste> any reason to prefer "avcodec:" to "lavc:"? // ease copy paste, according to michael [17:12] <ubitux> don't remember the other arguments, maybe the fork following that convention [17:13] <durandal_1707> saste: just pick name and i'm in [17:13] <saste> uh? honestly I don't believe in such silly micro copy&paste optimization [17:13] <saste> durandal_1707, ssmpeg? [17:13] <durandal_1707> next [17:13] <saste> ff2 [17:13] <saste> libmm [17:14] <saste> ffnext [17:14] <saste> fffork [17:15] <saste> durandal_1707, what's the point of an option which is not implemented?? [17:16] <durandal_1707> i would already implement it if I do not need to answer such questions [17:16] <saste> so please do, and i won't ask such answers [17:23] <durandal_1707> saste: what about forking libavcodec/libavformat/libavfilter/libavutil/... each into separate project? [17:23] <ubitux> start with libpostproc [17:23] <ubitux> oh my bad, already done [17:23] <ubitux> great success [17:26] <durandal_1707> http://yro.slashdot.org/story/13/10/22/153259/call-yourself-a-hacker-lose-y… [17:28] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:3ce7c62e1b47: avformat/framehash: add software version to framecrc/md5 [17:28] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:094c40ca3787: avcodec/utils: use a default lock manager that uses a pthread mutex [17:32] <Compn> wm4 : yes, i dont remember if vaapi got a vdpau module, but someone just gave up and now vdpau supports intel atom [17:58] <durandal_1707> michaelni: i wonder why fieldorder does not accept vertically subsampled chroma [18:01] <durandal_1707> i will just remove that limit [18:11] <michaelni> durandal_1707, ok [18:11] <durandal_1707> if you could test my patch on pcc/bigendian,etc it would be great [18:11] <durandal_1707> because i suspect test is doing conversion with swscale [18:12] Action: Daemon404 uses a cross-chain + qemu-ppc [18:20] <michaelni> durandal_1707, which patch exactly? [18:21] <durandal_1707> just sent [18:21] <durandal_1707> on ml [18:22] <durandal_1707> i hope you did not do something similar already.... [18:28] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:423ae31354a9: avutil/log: make default av_log callback thread safe [18:28] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:05eb0f125c70: avutil/log: document log callback thread saftey requirements [18:31] <durandal_1707> but i think for vertcially subsampled chroma filter should do nothing actually [18:41] <michaelni> durandal_1707, yuv420 chroma is interlaced too, and patch seems to pass on mips-qemu [18:44] <durandal_1707> well it appears it fails with disabled-asm [18:44] <durandal_1707> but i'm more concerned about correctness of patch [18:52] <durandal_1707> but similar is done with field filter... [19:00] <cone-490> ffmpeg.git 03Diego Biurrun 07master:0d6d4a9e4a62: avstring-test: Mark pointer passed to av_free() as non-const [19:00] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:23dec54c5dca: Merge commit '0d6d4a9e4a626be6ff55839eab4114aeac45ba88' [19:02] <durandal_1707> michaelni: so for fieldorder fate failure are to solutions: add bitexact swscale flag or apply that patch (if patch makes sense) [19:16] <Mista_D> Anyway to see "drop_frame_flag" via FFprobe? Need to distinct between drop and non-drop frame videos. [19:21] <Mista_D> in -show_packets "flags" don't show anything like "drop_frame_flag=0 / closed_gop=1 / broken_link=0"... [19:23] <mraulet> michaelni: you can remove CODEC_CAP_EXPERIMENTAL to HEVC [19:29] <smarter> michaelni: I'm okay with removing CODEC_CAP_EXPERIMENTAL now, but I worry about potential security issues [19:29] <smarter> I'm not sure how much fuzzing has been done [19:33] <michaelni> the same applies to every other decoder, i dont think things would be more secure if we simply kept every new decoder 6month under CODEC_CAP_EXPERIMENTAL [19:36] <cone-490> ffmpeg.git 03Paul B Mahol 07master:3a75c4e93b3b: fate: fix fieldorder test [19:42] <cone-490> ffmpeg.git 03Luca Barbato 07master:1ce3ec24b35d: h263: K&R formatting cosmetics [19:42] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:4c18c31438d0: Merge commit '1ce3ec24b35d9d549b12f17bd7fd391abaf3a6f7' [19:51] <cone-490> ffmpeg.git 03Vittorio Giovara 07master:2c993e8b5eca: golomb: K&R formatting cosmetics [19:51] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:02a31f6a7490: Merge remote-tracking branch 'qatar/master' [19:52] <durandal_1707> K&R contest [20:01] <michaelni> sounds like a C coding contest [20:03] <durandal_1707> coding contest is when you write something out of nothing [20:08] <michaelni> to quote wikipedia "The C Programming Language (sometimes referred to as K&R, after its authors' initials) is a well-known computer programming book written by Brian Kernighan and Dennis Ritchie ..." [20:10] <michaelni> "... the latter of whom originally designed and implemented the language, as well as co-designed the Unix operating system with which development of the language was closely intertwined. ..." [20:10] <Daemon404> coould also be a beard growing contest [20:12] Action: michaelni just thinks using their initials for a specific whitespace formatting is somehow lame and degrading ... [20:13] <wm4> "Libav developers degrade C inventors" [20:13] <wm4> nice headlines [20:14] <ubitux> poor ritchie [20:14] <ubitux> having his name associated with cosmetics regularly [20:15] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:92f649b411eb: avcodec/hevc: remove CODEC_CAP_EXPERIMENTAL [20:58] <Compn> michaelni : make a hook that changes 'k&r' in commit log to 'Brian Kernighan and Dennis Ritchie' [20:58] <Compn> ;D [20:59] <Compn> wm4 : learn to troll better [20:59] <wm4> Compn: ? [21:00] <Daemon404> ubitux, costmetics are more of an RMS thing: http://farm5.staticflickr.com/4023/4535097784_1b40890d9a_o.jpg [21:02] <Compn> wm4 : nm [21:07] <durandal_1707> Daemon404: more! [21:36] <llogan> Daemon404: heh. that could also be a boy-band/neckbeard album cover. [21:36] <Daemon404> the backhair boys [21:36] <llogan> RMS: Toe Cheese [22:01] <ubitux> BBB: ping [22:18] <cone-490> ffmpeg.git 03Michael Niedermayer 07master:fbdc98c758d5: avcodec/metasound_data: Fix inconsistency in tables [00:00] --- Thu Oct 24 2013

1 0

ffmpeg.log.20131023
by burek 24 Oct '13

24 Oct '13

[00:00] <Eduard_Munteanu> It's much easier to just sync to a synced wall clock for displaying loops. [00:02] <Eduard_Munteanu> (AFAIK there's no such project, everything seems to be using explicit sync network traffic) [00:05] <burek> NTP-synchronized playback ? [00:06] <Eduard_Munteanu> burek: you sync each player's clock via NTP, and have them agree on a time reference, e.g. Unix epoch. Then if you keep playback synchronized to the clock modulo loop length, you get synced playback without explicit network communication. [00:07] <burek> hmh [00:07] <burek> what is the real-world usecase for that? [00:07] <Eduard_Munteanu> IOW you assume the loop started at Unix epoch and repeated until today, it's easy to compute what you should be playing now. [00:07] <Eduard_Munteanu> burek: displaying ads [00:08] <burek> on a network stream? [00:08] <Eduard_Munteanu> burek: no, each player has files stored locally. They are distributed over a network, but not streaming. [00:09] <Eduard_Munteanu> Mainly because streaming is unreliable and traffic-hungry. [00:09] <burek> couldn't you just make a cronjob / scheduled job for that and sync the system clocks prior to that? [00:10] <Eduard_Munteanu> burek: no, they go out of sync rather quickly. It's quite annoying if you have a lot of TVs or other devices in the same room. [00:10] <burek> sync them more often [00:11] <burek> it's less intensive operation to sync your clocks more often than to create a synced playback the way you described it [00:12] <Eduard_Munteanu> burek: depends how often. NTP itself isn't very high-traffic, and it can use a local time source, it doesn't need to be accurate. [00:13] <Eduard_Munteanu> I got down to 3ms lag on the same machine, but a couple tens of ms is very reasonable for a LAN, even wireless. [00:14] <Eduard_Munteanu> Lag as measured vs wall clock. And this involves adjusting the playback speed as well for fine tuning and making it stable. [00:14] <Kuukunen> Eduard_Munteanu: btw, there's ffmpegsource that's meant for easier api... dunno where it's going right now tho [00:15] <Eduard_Munteanu> Kuukunen: hm, I'll have a look, I was implementing this in Haskell though. I still wonder if I can just pipe stuff to it :) [00:15] <burek> wait, you are playing offline files and the only thing that is not working correctly is that your ads don't display at the correct time or even the background video starts to go out of sync between displays? [00:15] <Eduard_Munteanu> burek: yes, it has to be synced between players [00:16] <burek> why dont you create a multicast stream then [00:16] <Eduard_Munteanu> burek: because it's traffic-hungry and it doesn't really work on usual wireless hardware. [00:16] <Kuukunen> Eduard_Munteanu: also, you could just run a video player with some sort of playback control api... I think vlc as one? [00:16] <burek> err... no it's not [00:17] <Eduard_Munteanu> Kuukunen: that's what I was doing with mplayer, controlling the playback speed through the slave socket. [00:17] <burek> you have only 1 stream going over your network and when the stream reaches your multicast point (usually your LAN switch/router) it gets multiplied and sent to each player/client that requested the stream [00:17] <Eduard_Munteanu> burek: I want it to work for 1080p, high quality H264. Besides, it's unreliable. [00:18] <burek> ok [00:18] <burek> so you want your TVs (or whatever) to only show ads? or is there other content too? is that other content also offline? [00:18] <Eduard_Munteanu> burek: that might work on wired LAN but it doesn't really work on commodity wireless APs. Many of them just freeze or fallback to a *very* low data rate. [00:18] <Eduard_Munteanu> burek: nah, only ads... all content stored locally. [00:20] <Eduard_Munteanu> Besides I find streaming wasteful, there's more than enough storage. :) [00:21] <burek> well in that case, you can try to sync your devices each minute and create cronjobs on each device to play your ads at a given schedule/program (even better, make your devices download the "program" from a server each morning).. that would make it all run at the same time, given that the hardware is the same (it takes the same amount of time to load/run ffmpeg) [00:22] <Eduard_Munteanu> burek: every minute is way too little, it becomes noticeable if they don't sync even for 5s. [00:22] <burek> then your hardware is maybe faulty? [00:22] <burek> if your clocks drift that much [00:22] <Eduard_Munteanu> Think about it, one or two frames ahead is acceptable, but not more. [00:23] <Eduard_Munteanu> That's already like 80ms. [00:23] <burek> btw, if you created your own player, why dont you just send an udp broadcast packet as a signal to all the players to play the content at that specific moment [00:23] <Eduard_Munteanu> burek: no, they just drift, especially hw where I don't choose drivers. [00:23] <burek> to avoid constant syncing [00:24] <Eduard_Munteanu> burek: because I'd essentially replicate NTP :) [00:24] <Eduard_Munteanu> And it's unreliable and it's a lot of traffic, I found that out when testing mplayer's UDP sync. [00:25] <burek> not quite.. this way you dont care about the clock syncing... you just listen to an udp port and trigger the playback when the packet arrives, that's all [00:25] <Eduard_Munteanu> burek: oh, just for control? Yeah, that'd work. [00:25] <Eduard_Munteanu> I still need to sync though. [00:25] <burek> but i hope you are not telling me that, when you press play on two devices at the same time, that the content gets out of sync until the end of the commercial? [00:26] <Eduard_Munteanu> burek: they do [00:26] <burek> how is that possible :) [00:27] <Eduard_Munteanu> I figure many players don't care about drifting as long as the average playback speed is correct. [00:27] <burek> well any decent player should respect the timestamp info in the stream and display frames accordingly [00:27] <burek> if your system clock drifts for such a short period of time that much... man.. [00:27] <burek> you're screwed :) [00:28] <Eduard_Munteanu> burek: AFAIK they do, but they use an *internal* clock source [00:28] <Eduard_Munteanu> They don't care about the wall clock. [00:28] <Eduard_Munteanu> e.g. if the video twitches, it resumes playback as soon as possible, it doesn't care about any reference clock. [00:29] <burek> so, your ads are so short, but even in that case, it is possible for 2 devices to go out of sync..? [00:30] <burek> you could send udp sync packet at the start of each ad so the players end the previous playback (if any) and immediately display the new ad... i guess.. [00:30] <Eduard_Munteanu> burek: yes. Each is usually 30s long. [00:30] <burek> but if they go out of sync even for the duration of a single ad.. well... [00:31] <Eduard_Munteanu> burek: I do what mplayer's syncing algo does, seek above a threshold, vary the playback speed under it. [00:31] <Eduard_Munteanu> burek: mind some of those devices might not have hw acceleration, or not optimally configured. [00:32] <Eduard_Munteanu> So if the video displays mostly correctly, even if the jitter is a tad bit high and it's lagging a bit, I still want to use that device. [00:33] <burek> that's just a weak device or wrongly configured... it might be better to resolve the cause of the problem rather than the consequence.. [00:33] <burek> but i get the idea [00:34] <burek> it's quite interesting scenario :) [00:35] <Eduard_Munteanu> burek: there's visible lag even simply because TVs have different processing lag [00:36] <Eduard_Munteanu> burek: HDMI splitters add some too. So in any case, I want to add as little as possible. [00:37] <burek> so, all you need is to write your own player that gets synced frequently and seeks/skips frames accordingly :) [00:37] <Eduard_Munteanu> Yep. I already have one working, but I don't really want to depend on mplayer's socket. [00:38] <Eduard_Munteanu> Also because I want more logic in playlist management. [00:40] <burek> well, since you cant get a lot of help from ntp [00:40] <burek> (because it also takes time to update/sync the clock) [00:40] <Eduard_Munteanu> I should also say NTP isn't great either, so that might add a delta of a few tens of ms theoretically. [00:41] <burek> maybe you could create a remote beacon source, which would send timestamps over the udp broadcast and sync the players [00:41] <Eduard_Munteanu> burek: not a lot, with the right settings it gets into a usable margin rather quickly [00:42] <Eduard_Munteanu> I don't expect an open loop system like that to work right. [00:42] <burek> why not [00:42] <Eduard_Munteanu> burek: wireless alone adds rather large and not very deterministic lag [00:43] <burek> yes, but when you send the broadcast packet, it gets 255.255 [00:43] <burek> so its a single packet [00:43] <burek> received at the same time by all devices [00:43] <Eduard_Munteanu> Think about it, even NTP isn't really great under 10ms or so. [00:44] <burek> and you dont need to send it often.. say, 5 seconds interval would be enough [00:44] <Eduard_Munteanu> Wireless hardware doesn't really work like that, it depends on a lot of settings. [00:45] <Eduard_Munteanu> burek: and mind broadcast/multicast modes aren't well supported on many APs. [00:45] <Eduard_Munteanu> Some freeze if you try to send a 1Mbit multicast stream. [00:45] <burek> so, if i get this right, you can't properly sync devices over your wifi [00:45] <burek> so, what's the use of it then [00:46] <Eduard_Munteanu> Yeah, wifi was a big reason for this. [00:46] <burek> wait, why would you have 1mbit broadcast/multicast? [00:46] <bencoh> well, 1mbps isn't much [00:46] <burek> i am talking about a single udp packet, containing the current timestamp [00:46] <bencoh> so, why wouldnt you ? [00:46] <Eduard_Munteanu> burek: because they suck at multicast :) [00:46] <burek> it's less than 32 bytes [00:47] <burek> it practically gets broadcasted instantly, even on wifi [00:47] <bencoh> oh, your controle stuff, not the actual stream [00:47] <Eduard_Munteanu> burek: I know, but that single packet depends on power management settings on the AP, sender and receivers, like the beacon interval, whether they go to sleep inbetween etc. [00:47] <burek> all the clients have all the streams/files so no need for broadcasting the stream [00:47] <burek> only syncing the playback time through timestamps broadcast [00:48] <Eduard_Munteanu> burek: I agree alone it doesn't seem to justify it [00:48] <burek> Eduard_Munteanu yes, but the same packet will be broadcast at 1 point in time and all devices will receive it at that point in time [00:48] <bencoh> Eduard_Munteanu: actually the fact that it is sent on beacon/DTIM isnt a big problem for sync [00:49] <Eduard_Munteanu> But it gets rather nasty once all those nondeterministic lag sources get together. [00:49] <bencoh> (the fact that most APs/clients suck is, though) [00:49] <burek> my point is [00:49] <burek> even if it gets some kind of a lag [00:49] <burek> all the players will get that packet again at the same time [00:49] <Eduard_Munteanu> bencoh: yeah, you're probably right. Maybe I've seen devices misbehaving. [00:50] <burek> they will all lag [00:50] <burek> thus, they will keep the sync [00:50] <bencoh> Eduard_Munteanu: most of them are :) [00:50] <bencoh> (misbehaving) [00:50] <Eduard_Munteanu> burek: indeed, but it's not deterministic. There's a lot of different hardware. [00:51] <burek> if 32 bytes udp broadcast packet is not effective enough for the syncing, then you can safely forget about your wifi connectivity in general [00:51] <Eduard_Munteanu> burek: besides, I don't feel like measuring the lag myself and putting in a magic number, in case the sender is also a player. [00:51] <burek> Eduard_Munteanu, that's the point [00:52] <burek> the sender is not playing it aloud [00:52] <burek> it can be muted [00:52] <burek> it is only used as a source for syncing [00:52] <Eduard_Munteanu> Sure, no audio. [00:52] <burek> all the other players will play in sync [00:52] <burek> even if in lag [00:53] <bencoh> burek: that's not true [00:53] <bencoh> we're talking milisecs here [00:53] <burek> why it isnt [00:53] <bencoh> you can have much more jitter than that on a wireless link [00:53] <burek> so? [00:53] <Eduard_Munteanu> If you try to ping over wireless you'll see rather non-deterministic delays. [00:54] <burek> i agree [00:54] <burek> why would it hurt players? [00:54] <Eduard_Munteanu> And even if you're right, a closed-loop system is probably more robust across different hardware. [00:54] <bencoh> so having a few 50ms lag on a single 32bits-udp-packet would be quite common on "working" wireless connections [00:54] <Eduard_Munteanu> burek: because not all players get the same roundtrip time / lag. [00:55] <bencoh> oh btw, is this all about just syncing an internal clock, or triggering remote play/pause ? [00:55] <Eduard_Munteanu> I also don't want to have all players the same "metric" away from the source. E.g. my source is playing videos too. [00:56] <burek> this is my ping to an ap that is under a mile away from me http://pastebin.com/H3e9dX9p [00:56] <Eduard_Munteanu> bencoh: syncing playback, not just on play/pause [00:56] <burek> and you are telling me about 50ms lag on a LAN wifi? [00:56] <bencoh> burek: quite common in a not-so-busy-but-suboptimal environment [00:57] <bencoh> crowded urban area for instance [00:57] <Eduard_Munteanu> burek: think about your local mall with lots of APs around and on the same channel etc. [00:57] <burek> why would you put them on the same channel [00:58] <bencoh> 7 packets transmitted, 7 packets received, 0.0% packet loss [00:58] <bencoh> round-trip min/avg/max/stddev = 6.780/12.861/20.081/4.908 ms [00:58] <Eduard_Munteanu> burek: I don't, but other channels aren't free either :D [00:58] <Eduard_Munteanu> The allowed ones at least. [00:58] <burek> ok, wait, we are loosing the point here [00:58] <bencoh> indeed :) [00:58] <burek> even if you have at one moment a lag of 100ms [00:58] <burek> it doesnt matter too much [00:58] <Eduard_Munteanu> burek: well, I'm just saying it has to be robust :) [00:59] <burek> since all the players will get that packet at the same time, since its a broadcast [00:59] <Eduard_Munteanu> burek: I wouldn't assume that [01:00] <Eduard_Munteanu> burek: think about it, NTP has trouble being very precise even on a wired LAN, and that's a closed-loop system. [01:01] <Eduard_Munteanu> On a wired LAN you can get around 5-10ms deltas IIRC, but if you want better, you reach for PTP. [01:02] <Eduard_Munteanu> Well, it might do better on a small LAN, but still. [01:03] <burek> well there is even less point to use ntp if you think like that [01:04] <burek> oh i got it [01:04] <Eduard_Munteanu> burek: I do have a budget of 1-2 frames, so 40-80ms lag *overall*. [01:04] <burek> you could send an ARP request [01:04] <burek> that is an ethernet level packet [01:04] <burek> and all devices should get it exactly as a single broadcast packet [01:05] <Eduard_Munteanu> UDP itself doesn't add much lag. Most of the lag is physical stuff, I think. [01:05] <Eduard_Munteanu> Like how APs and wireless cards react to incoming packets. [01:06] <burek> i believe that best bet is to get a single beacon (packet) from your AP, received by all your players (assuming they are on the same AP), which would make them play in sync [01:07] <burek> arp (or any other ethernet level) packet is one of such, because it generates just a single packet sent once (and received by all receivers) [01:07] <Eduard_Munteanu> burek: I'm not saying it won't work, in fact it probably will in quite a few cases, but I don't want headaches. Besides, I'm reusing a rather popular protocol for time sync, so it's even easier than doing UDP stuff myself. :) [01:08] <burek> btw, if you can't get such a precise beacon, there is no point in having wifi connectivity, right? :) [01:08] <burek> oh boy :) just take a look on the internet for a simple udp send/receive example [01:08] <burek> it's like.. 2 lines of code.. [01:09] <Eduard_Munteanu> NTP was designed to work over the Internet and give you a semi-reasonable synced clock. [01:09] <Eduard_Munteanu> Even in the presence of large, undeterministic lag on the order of hundreds of ms. [01:11] <burek> ntp over wifi will work even worse.. [01:11] <burek> round-trip will be averaged [01:12] <burek> which, in case of such a variable ping time over wifi, will make it so unreliable [01:12] <Eduard_Munteanu> burek: um, no, that's the point, NTP accounts for round-trip as well. It's not an open-loop system. [01:13] <burek> well yeah... but if it gets different "ping reply time" each time, whats the point of the average round trip time [01:13] <Eduard_Munteanu> And you get much more precise timers ticking. [01:13] <Eduard_Munteanu> Not just a beacon. [01:13] <Eduard_Munteanu> burek: it just takes longer to converge, actually [01:14] <cbsrobot> Eduard_Munteanu: looking for a sync player or do you want to implement it ? [01:15] <Eduard_Munteanu> cbsrobot: I already have one, I was considering reimplementing it on top of a codec library instead of controlling mplayer, though. [01:16] <Eduard_Munteanu> Assuming I get VDPAU H264 decoding if I feed ffmpeg a video. [01:18] <Eduard_Munteanu> Hm, libx264 seems to be an encoder only. I wonder what's the actual codec that does hw-accelerated decoding. [01:19] <burek> http://en.wikipedia.org/wiki/Reference_Broadcast_Time_Synchronization [01:20] <burek> so it wasnt enough to go to eth (level 2) :) it needs to be level 1 [01:21] <Eduard_Munteanu> burek: yeah, also it's more complex since they do take feedback from roundtrip time. [01:22] <Eduard_Munteanu> burek: btw, if I wasn't clear, I'm just using a local NTP server, I don't care much about the global clock. [01:22] <burek> it figures [01:24] <Eduard_Munteanu> There are a couple of options you can pass the client to make synchronization happen very quickly, in a matter of minutes. [01:24] <burek> lets get something straight [01:24] <burek> once you sync your system clocks [01:24] <burek> there is no need to sync them in the next hour or so (at least) [01:24] <burek> right? [01:25] <burek> otherwise something is very wrong with the hardware [01:25] <burek> or lets make it a minute [01:25] <Eduard_Munteanu> burek: yes, that's rather accurate [01:25] <Eduard_Munteanu> Assuming you're using a modern timer in the kernel. [01:25] <burek> so, once you sync your system clocks on the players, there is no need for syncing anymore, at least for the next minute [01:25] <Eduard_Munteanu> Yep. [01:26] <Eduard_Munteanu> burek: however you need to sync playback to the system clock too [01:26] <burek> yes [01:26] <burek> you can do that with ffmpeg [01:26] <Eduard_Munteanu> Oh? How do you mean? [01:26] <burek> just not with the executable, but if you are using ffmpeg libs [01:27] <burek> just feed the decoder based on the system clock, instead of an internal [01:27] <Eduard_Munteanu> I think some formats have a notion of accurate timing themselves, e.g. MPEG. But most players don't care about timing much I think. [01:27] <Eduard_Munteanu> burek: oh, I see, cool. [01:27] <burek> well with ffmpeg you control the decoding process, so you can decode a frame whenever you like [01:27] <burek> with ffmpeg libraries* [01:28] <Eduard_Munteanu> burek: should I do the decoding synced, or just displaying? I'm worried decoding speed may vary across machines. [01:28] <Eduard_Munteanu> Decoding lag, even. [01:28] <burek> DTS is what you care about i guess [01:29] <burek> no wait [01:29] <burek> PTS is for a display time right? [01:29] <Eduard_Munteanu> Those terms seem a bit familiar, but I don't know much about that. [01:29] <burek> just make sure the frame is displayed at the desired point in time (i guess its a PTS - presentation time stamp or so) [01:30] <Eduard_Munteanu> Ah. [01:30] <burek> and stall it (or drop it) if it drifts [01:30] <Dave92F1> Hi all! Been struggling with this for a while - how can I get ffmpeg to pipe its output to mplayer for display? I need it to read from /dev/video1 (input 1, norm NTSC). [01:30] <Eduard_Munteanu> Thanks, that seems like a viable approach. [01:30] <Eduard_Munteanu> Better than varying the playback speed externally, I mean. [01:30] <burek> definitely :) [01:31] <Eduard_Munteanu> One issue with my approach was I had to make many frames keyframes. [01:31] <burek> i was trying to solve the problem with the ffmpeg executable, thus the udp syncing of the start of the playback of each ad [01:31] <Eduard_Munteanu> This might alleviate that concern. [01:31] <Eduard_Munteanu> Ah, I see. [01:32] <burek> Dave92F1, just output it to stdout [01:32] <burek> ffmpeg -i INPUT ... -f mpegts - [01:32] <burek> "-" means stdout [01:32] <burek> or any other format you need instead of mpegts [01:33] <Eduard_Munteanu> I think I should do the syncing bits from C directly. [01:33] <Dave92F1> burek: OK, how do I tell ffmpeg to read from input 1? (I think I can do '-standard NTSC' to get NTSC). [01:33] <burek> i need to go to sleep :) anyway, good luck with that player Eduard_Munteanu :) i hope you'll showcase it somewhere :) [01:34] <Eduard_Munteanu> burek: I'm considering publishing it at some point, yeah. Thanks :) [01:34] <burek> Dave92F1: ffmpeg -f <FORMAT> -i - ... [01:34] <burek> "-" means stdin in this case, because it is at the position for the input parameter [01:34] <burek> Eduard_Munteanu +1 :) [01:35] <burek> oops, Dave92F1, use -f v4l2 -i /dev/video1 [01:35] <burek> http://trac.ffmpeg.org/wiki/How%20to%20capture%20a%20webcam%20input [01:36] <Dave92F1_> burek: "ffmpeg -f v4l2 -standard NTSC -i /dev/video1 -f mpegts - | mplayer -" gives me "Cannot seek backward in linear streams!" [01:36] <burek> try for the output: -f rawvideo - | ... [01:36] <Dave92F1_> burek: Also, ffmpeg seems to read from input 0 by default (input 0 is the S-video input on my capture device; I need to read from input 1 which is the composite video) [01:38] <Dave92F1_> burek: "ffmpeg -f v4l2 -standard NTSC -i /dev/video1 -f rawvideo - | mplayer -" give me the same seek error [01:38] <burek> http://www.linuxtv.org/wiki/index.php/V4L_capturing#Capture_from_composite_… [01:39] <Eduard_Munteanu> Is there an API for decoding video generically, instead of calling the demuxers, decoders and display stuff myself? [01:39] <burek> try: -f mpeg - [01:39] <burek> try: -f mpeg - | mplayer .. [01:40] <burek> Eduard_Munteanu, take a look at source (directory src/examples i think) [01:40] <burek> or docs/examples [01:40] <Eduard_Munteanu> Hm, looking at http://www.ffmpeg.org/doxygen/trunk/examples.html right now [01:40] <Dave92F1_> burek: -f mepg - | mplayer - also gives seek error. I looked at that link, but don't see how to select input 1 vs. input 0. [01:41] <bencoh> there is a way to play from stdin using mplayer, but not this exact one [01:41] <burek> Eduard_Munteanu http://git.videolan.org/?p=ffmpeg.git;a=tree;f=doc/examples;h=57fd81bfb3f98… [01:41] <burek> yeah, thats the same [01:41] <bencoh> hmm, nevermind [01:42] <burek> Dave92F1 http://mplayerhq.hu/pipermail/mplayer-users/2005-April/052844.html [01:42] <Eduard_Munteanu> Would gstreamer be more straightforward? I get a feeling this is a rather lowlevel API. [01:42] <burek> you can read /dev/video1 from mplayer [01:42] <burek> directly [01:43] <burek> Eduard_Munteanu, you can try... i dont know precisely, but i also got a feeling that some things are easier to do in gs... try and see [01:44] <burek> anyway, off to bed :) gn o/ [01:44] <Eduard_Munteanu> burek: g'night [01:44] <Dave92F1_> burek: Thanks; I know how to read it with mplayer (mplayer tv:// -tv device=/dev/video1:input=1:norm=NTSC) works fine. But I want to do it with ffmpeg so I can tee the output from ffmpeg to both mplayer (for display) and mencoder (for recording). [01:45] <bencoh> -f rawvideo pipe: [01:45] <bencoh> in case it's raw video [01:46] <bencoh> but you'd have to specify the video size/format to mplayer/mencoder [01:46] <bencoh> (and framerate) [01:46] <Dave92F1_> Gotta go now...back later. [02:56] <Ruler2112> Is there any way to override the ISFT metadata field? Using -metadata ISFT="whatever" does not work; seems to be locked to Lavf#### [03:00] <drv> i think you want to set "encoder", which will get translated to ISFT [03:02] <Ruler2112> I've tried that - output file still has it set to 'Lavf55.16.102' [03:06] <Ruler2112> VirtualDub can set the field to what I want it to be, but I'd like to eliminate it as a dependency of my script if possible. [03:14] <drv> looks like it always gets overwritten unless you specify bitexact [03:14] <drv> so you can try -flags bitexact, seems to work here [03:20] <Ruler2112> Dude, you're awesome! Works like a charm [03:20] <Ruler2112> :) [03:21] <Ruler2112> I do not see bitexact in the help; are there any other effects? [03:21] <drv> it's probably hidden under the help for 'flags', not sure what else it does [03:21] <drv> grep source for BITEXACT ;) [03:23] <Ruler2112> I found this online - EDVAS use only bitexact stuff (except (i)dct) - It's like defining a word using part of the same word. ;) [03:30] <Ruler2112> Thanks again for the help. [03:35] <ez> hi guys, how are you ? Can somebody help me ? [03:37] <iive> nobody can help you [03:37] <iive> if you don't say what the problem is [03:38] <iive> don't ask to ask, just ask. [03:38] Action: iive out [03:39] <ez> I am creating a bash script for a customer that converts multiple images into a video-slideshow. I need the imagens to be rendered into a specific order. I can generate the video using the pattern * but i am struggling using individual files, like this syntax. http://pastebin.com/NF8U9bzB Another question is: There is a way to have different times for each image, something like "1st image stays for 2 sec, 2nd for 5 sec, etc" [03:40] <eZ> sorry, I was busy writing my question lol english is not my native language :) [03:43] <eZ> it seems to get only the first image ..do I need to copy all images to a temp directory and render using the pattern * or something like that ? Can I control the order of the images and the individual times it will be exposed on video ? [03:46] <eZ> Could I pipe instead ? [04:06] <eZ> somebody here ? [05:09] <mark4o> eZ: ffmpeg -f image2 -pattern_type glob -i "*.png" out.mp4 [05:11] <mark4o> eZ: http://superuser.com/a/619843/2087 [07:03] <eZ> mark4o: I need without the pattern ... [07:04] <eZ> mark4o: I need to build a specific order, without recopy the files ... there other files on dir [07:05] <mark4o> eZ: you can use the concat demuxer, see http://superuser.com/a/619843/2087 [07:06] <mark4o> or if you want to keep a separate -i option for each one then use the concat filter [07:06] <mark4o> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [07:07] <mark4o> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [07:09] <mark4o> or for that matter you can use -f image2 -i "{file1,file2,file3}.png" [07:09] <mark4o> instead of *.png [07:57] <Ruler2112> Hello. I'm having a problem with a MKV clip when transcoding to XVID. Behavior is the same whether I use the built-in mpeg4 codec or libxvid. The original MKV is of a 4:3 source, but has black bars on the sides, though the resolution is 976:720. If I resize using -s 530:400 and -vf scale=530:400, the resulting video does not have the black bars. If I do NOT resize the original video and simply convert it to XVID, the resulting [07:58] <Ruler2112> Any ideas of how I could refrain from resizing (with the resulting loss of quality & introduction of digital artifacts) and eliminate the side black bars? [07:59] <Ruler2112> I've tried faking out ffmpeg by passing the original resolution to the resizing functions, setting both sar and dar using -vf, etc. After about 40 tries, I decided to ask. :) [08:11] <mark4o> Ruler2112: Did you try -vf crop=960:720 ? (or whatever the resolution is without the bars) [08:13] <Ruler2112> I had not, but just did. Had no effect. :( [08:16] <mark4o> Ruler2112: Didn't it make the video 960x720 instead of 976x720? [08:17] <mark4o> If you didn't notice then perhaps it needs more cropping [08:17] <mark4o> The cropdetect filter should be able to tell you exactly how much [08:20] <Ruler2112> Just checked the resolution of the output and it did cut it down; not noticeable when watching though. I'll try chopping it down some more, but I find it curious that simply resizing it (using the same aspect ratio) eliminates them. A real resize of the entire picture would preserve the bars, would it not? [08:23] <mark4o> Some codecs only support certain sizes, or sizes that are a multiple of 16 (whole macroblocks). [08:24] <Ruler2112> I just tried 800x720 and there's still no difference; bars exist and video *looks* to be the same exact size. [08:24] <mark4o> But yes, if any size is supported, then the bars should not be affected other than shrinking [08:25] <Ruler2112> I've got to get to bed. You've given me some ideas to play with tomorrow after work - thank you Mark. [08:26] <mark4o> ok np, may want to try a different player also [08:27] <Ruler2112> That's a good idea... I've been using VLC, but have MPC loaded as well. (Listening to a football podcast in it.) [08:29] <Ruler2112> Just tried MPC & no luck. :( Ah well - I'll get back at it tomorrow. Hope you have a great evening and thanks again. [09:03] <aquarat> does anyone know if there's any kind of raw-codec support in ffmpeg ? like DNG, REDCode, Cineform, etc. [09:05] <ebalsley> no red code or cineform support. I've heard old R3D files are supported from before RED started scrambling their files. But that's mostly irrelevant because all modern R3D files are scrambled [09:06] <ebalsley> you can vote up the issue https://trac.ffmpeg.org/ticket/2690 [09:06] <ebalsley> can someone help me understand the --shared and --static configure options and their pros and cons? [09:09] <aquarat> thanks ebalsley [10:41] <Apic> A wonderful fine Sweetmorn (UGT)! [11:17] <xlinkz0> can i make the log put a timestamp before every line? [11:40] <burek> xlinkz0, try this http://stackoverflow.com/questions/21564/is-there-a-unix-utility-to-prepend… [11:40] <burek> ive used ts but you can pick whichever one you like the most [11:41] <xlinkz0> burek: i'm interested when every log message was recorded [11:41] <burek> yeah i understand that [11:41] <xlinkz0> how can a utility know when each line was recorded? :) [11:44] <kenansulayman> hi! [11:44] <kenansulayman> What's the fastest way to decode ATRAC files? [11:45] <burek> xlinkz0, because it is piped to the output of ffmpeg [11:45] <burek> and it can detect whenever a new line of log is outputed [11:51] <xlinkz0> burek: but how do i do that with the -report option? [13:40] <maep> hi, is there a way to signal packet loss to a decoder? [13:42] <Apic> Good Question! [13:43] <maep> Is that a 'no'? [13:43] <Apic> No, that's a "I do not know either, but would really like to hear the Answer from someone competent in here." [13:44] <maep> ah :) [13:49] <maep> maybe setting AVPacket.flags to AV_PKT_FLAG_CORRUPT? [13:49] <maep> is that passed to the decoder? [18:55] <brontosaurusrex> is there a way to "measure" a complexity of a clip (this will be short clips) ? [18:55] <durandal_1707> what kind of complexity? [18:55] <brontosaurusrex> durandal_1707, visual complexity, say visualy rich shot or shot with lots of motion [18:56] <brontosaurusrex> the idea is "a. from a very long clip take N short random clips" "b. measure "complexity"" "c. keep N/2 of the more "complex" shots" [18:57] <brontosaurusrex> or N/3 , N/4 < whatever it will work [18:57] <durandal_1707> there is scene change detect filter... [18:58] <brontosaurusrex> hmm, yes i have to scene-detect this shots as well [18:58] <brontosaurusrex> basically it will be probably "If scene-cut was detected then delete" [18:59] <brontosaurusrex> but that will happen before "complexity" evaluation [19:00] <durandal_1707> i do not think there is such filter, which will give some number to output that maps to 'complexity' [19:01] <brontosaurusrex> the idea is to just downsize a bit and then encode with x264 crf and then read the average bitrate [19:01] <brontosaurusrex> but i doubt that will really work [19:01] <brontosaurusrex> so, open to some better ideas [19:07] <userper> Hi there [19:07] <userper> I wonder if anyone can help... trying to get the libfdk_aac library to work following an Ubuntu compilation from source [19:08] <brontosaurusrex> userper, and? [19:08] <userper> Ran the ./configure command as listed on the compilation guide [19:09] <userper> But I am receiving "Unknown encoder 'lbfdk_aac' when I try to make use of it [19:10] <userper> libfdk_aac was listed as an enable audio codec during the configure process. But 'ffmpeg -codecs' does not list it [19:10] <brontosaurusrex> userper, --enable-libfdk-aac --enable-nonfree ? [19:11] <userper> Yep, I used both of those @brontosaurusrex [19:12] <userper> Copy/pasted the commands from the compilation guide. Are the instructions you've linked on Crunchbang a better bet? [19:13] <brontosaurusrex> userper, no, they are pretty old now i suppose [19:13] <brontosaurusrex> so, last time i tryed a command like "ffmpeg -i in -c:a libfdk_aac -flags +qscale -global_quality 2 -afterburner 1 -vn out.m4a" should work [19:14] <brontosaurusrex> that should produce an LC AAC thingy [19:16] <userper> The current git clone doesn't seem to support "-c:a" so I've switched it to use :acodec [19:17] <brontosaurusrex> mkay, and ? [19:17] <userper> But that still gives me the "Unknown encoder 'libfdk_aac'" error [19:18] <userper> "ffmeg -codecs | grep aac" only lists aac, aac_latm and libvo_aacenc [19:18] <brontosaurusrex> ffmpeg -codecs | grep fdk , gives me [19:18] <brontosaurusrex> DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfdk_aac ) [19:19] <brontosaurusrex> so i guess you missed something during the compile time [19:19] <userper> That command doesn't give me any results, just ffmpeg the header [19:19] <brontosaurusrex> you missed something during the compile time [19:21] <durandal_1707> userper: actually to get fdk you need to recompile with fdk enabled flag [19:22] <userper> durandal_1707 - I think I have, that was my reason for moving away from a static build. I included the "--enable-libfdk-aac" switch in the configure command. [19:22] <adiulici> Hello. I have a problem with ffmpeg: it keeps getting closed and exits with return code "137". It doesn't always do this, but in about 66% of the cases. The command I'm using is: http://pastebin.com/dFQ884VB [19:24] <durandal_1707> userper: where it is installed? [19:26] <brontosaurusrex> userper, my latest debian static compile, switches http://paste.debian.net/plain/60915 [19:26] <brontosaurusrex> (note that i don't have a clue about compiling, so all i do is trial/error) [19:27] <userper> I have noticed the version line I have suggests that I'm not using my compiled version, so going to check: ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers [19:28] <brontosaurusrex> good catch :) [19:28] <durandal_1707> that is not FFmpeg [19:29] <userper> Yeah I know, I've seen mentions of the Libav "ripoff" [19:29] <userper> Can't believe I hadn't spotted it until now [19:38] <adiulici> Q: I have a problem with ffmpeg: it keeps getting closed and exits with return code "137". It doesn't always do this, but in about 66% of the cases. The command I'm using is: http://pastebin.com/dFQ884VB [19:40] <userper> Yep sussed it, I was using /usr/bin/ffmpeg (which I thought I'd removed) instead of the compiled version in ~/bin. Thanks, sorry for taking up your time. [19:42] <Mista_D> Anyway to see "drop_frame_flag" via FFprobe? Need to distinct between drop and non-drop frame videos. [19:57] <sacarasc> adiulici: Can you paste the full output somewhere, too. [19:58] <adiulici> Yes, I'll paste it in pastebin. Just a sec [20:05] <adiulici> sacarasc: this is the output I get from my cron job: http://pastebin.com/B82QqeGe [20:06] <adiulici> sacarasc: and this is a normal output (when everything goes fine) : http://pastebin.com/z2QYeWt9 [20:25] <llogan> adiulici: does your host kill some processes that last longer than x minute(s)? [20:27] <adiulici> llogan: I will ask my sys admin, but I'm not sure that's the problem because I measured how long the ffmpeg runs when it gets killed and very often when it gets killed it only runs 2 or 3 seconds. Sometimes it gets to run for 30 seconds also, but it also happened that it ran for 50-60 seconds and finished with no problem. [20:28] <llogan> maybe, if they do have something implemented, it's triggered by CPU instead [20:28] <llogan> have you tried on another machine? [20:30] <llogan> i've never seen mv4 and aic used with libx264 before. [20:32] <adiulici> I really am not an expert with ffmpeg :) But this command was what worked for me and did it's job well. I wrote the sys admin now a message and asked him if there is a program that's triggered by CPU [20:32] <adiulici> Thanks [20:33] <llogan> those flags are ignored [20:34] <adiulici> Ok, i didn't know. Thanks :) [20:35] <llogan> https://trac.ffmpeg.org/wiki/x264EncodingGuide [20:40] <v0lksman> llogan: I've been through that encoding guide and tried a million things with no luck. Any docs or tips on encoding user submitted video to mp4 that will work on an IOS device? [20:42] <llogan> v0lksman: which device(s)? they can vary. [20:44] <v0lksman> well I'm doing my testing from an iphone5...but I need the video to be versitile...say iphone4+ [20:45] <llogan> H.264 Main profile 3.1 [20:46] <llogan> according to: https://developer.apple.com/library/mac/documentation/NetworkingInternet/Co… [20:46] <llogan> (see Preparing Media...) [20:46] <llogan> -profile:v main -level 3.1 [20:47] <v0lksman> will give that a shot...tried a bunch of profile and level (even ref) settings yesterday with no luck...I'll try to paste some of the commands and see if anyone can poke holes in them [20:47] <v0lksman> thanks! [20:48] <llogan> i'm not sure what the max frame size dimensions are though [20:49] <v0lksman> I think that's listed on that apple page in the same section [20:50] <llogan> you can use scale filter to automatically scale and keep aspect: -vf scale=1280:-1 [20:50] <llogan> also see -force_original_aspect_ratio in scale: http://ffmpeg.org/ffmpeg-filters.html#scale [20:51] <llogan> or scale="1280:trunc(ow/a/2)*2" to (probably) avoid a "non-divisable by 2" value [20:53] <llogan> ...assuming you're downscaling (no need to up it). [21:01] <v0lksman> so trying to keep it super simple: https://dpaste.de/Lw0Z [21:01] <v0lksman> won't stream on iphone5 (IOS7) [21:03] <v0lksman> basically I'm placing the output file on a webserver and hitting it directly in Safari on the phone [21:03] <v0lksman> works fine for other videos that I didn't encode [21:22] <v0lksman> llogan: sorry here's the full output http://pastie.org/8424900 [21:26] <llogan> v0lksman: does it work if you use -profile:v baseline -level 3.0? [21:29] <v0lksman> http://pastie.org/8424912 - nadda...same behaviour [21:34] <llogan> iDunno [21:34] <llogan> out of ideas... [21:34] <llogan> omit the audio stream with -an [21:35] Action: llogan ignorantly shooting in the dark [21:36] <v0lksman> will try that! thanks for even the blanks...I've tried a TON of different settings..nothing seems to work. My buddy has a video he encoded in some windows app that works fine. Even tried handbrake no luck [21:37] <llogan> can you provide the link? [21:43] <Xx_pk_xX> hello guys. I have a ffmpeg releated question. Is there a way to find exact frame using ffmpeg? Or exact sound. What I need to do is an automated system to find exact time of commercials in h264 mp4 files recorded from TV. Is there a way to do something like that using ffmpeg? Let's say I have a frame of ending commercial, or a sound file, is there a way to find it in a video? [21:46] <llogan> Xx_pk_xX: an example finding the exact frame (only works if the frame and image are literally exactly the same) http://superuser.com/a/663947/110524 [21:46] <llogan> ...so that probably won't work for you now that i actually read the question [21:47] <Xx_pk_xX> yes, I don't think that would work, because frames maybe won't be exactly same [22:02] <v0lksman> llogan: hey...in moving the file it works on a different host. So it must be something with my dev server. Thanks for the help though. I should have tried that earlier [22:04] <llogan> weird [22:52] <pure> Hi, is there anything wrong with what I'm doing here? http://pastebin.com/anLpVVf3 [22:52] <pure> I'm getting a blank stream when I look on twitch. [23:36] <brontosaurusrex> pure, any errors if you redirect this to a file? [23:52] <llogan> pure: this script again... you probably need to add -pix_fmt yuv420p, but this is a guess since you did not provide the complete console output [23:53] <llogan> which is a requirement when asking questions here [00:00] --- Thu Oct 24 2013

1 0

ffmpeg-devel.log.20131022
by burek 23 Oct '13

23 Oct '13

[02:46] <Daemon404> g 44 [08:48] <cone-588> ffmpeg.git 03Mickaël Raulet 07master:1912842045f2: hevc: remove disable_au option(cherry picked from commit e90b3f6753d645fec076e951a0597a5dc2d2fe31) [08:48] <cone-588> ffmpeg.git 03Mickaël Raulet 07master:a8fafa89785b: hevc: cleaning disable field in deblocking filter(cherry picked from commit 7dd7a27ae850a51b3c9cd07046c422677398f6d5) [08:48] <cone-588> ffmpeg.git 03Mickaël Raulet 07master:1c8de4dd949c: hevc: pretty print(cherry picked from commit 6332b3afe298b9e1060e8549aea3aa2771b43f5d) [08:48] <cone-588> ffmpeg.git 03Mickaël Raulet 07master:c68faca68e21: hevc: fix transform_skip which is only valid for 4x4(cherry picked from commit 740e5a71e5121dbf3fabdc4fec97829c18ecc2d8) [11:19] <durandal_1707> kierank: what 'magical' constants? [11:19] <kierank> durandal_1707: in the olden days [11:19] <kierank> before TEP [11:19] <kierank> get_buffer used to be some magic numbers in ffmpeg.c [12:44] <cptspiff> gpl violators: http://xbmc.org they use ffmpeg under gpl in signed ios binaries. https://github.com/xbmc/xbmc/pull/3437 [12:46] <durandal_1707> and why is that violation? [12:46] <cptspiff> because you cannot modify it without resigning. [12:47] <durandal_1707> modify what? [12:47] <cptspiff> whatever i feel like modifying. [12:47] <BBB> do you have a legal record showing that a judge agreed with you in US juridsdiction? [12:47] <BBB> (i.e. precedent) [12:47] <BBB> jury is fine also [12:47] <cptspiff> no i don't. i only have my ethics and i left the project due to this. do what you want, i just wanted to inform. [12:48] <BBB> that was quick [12:48] <durandal_1707> you left FFmpeg? [12:48] <Daemon404> BBB, well also i think apple bans GPL [12:48] <Daemon404> which is not a legal issue [12:48] <Daemon404> but a store issue [12:48] <BBB> that's between them and apple, not us and them [12:48] <Daemon404> yes [12:48] <BBB> I agree it sucks :) [12:48] <JEEB> was it GPLv3 that doesn't let you drm the binaries? [12:48] <j-b> that is not true [12:48] <j-b> It is between you and them [12:48] <BBB> durandal_1707: I think he left xbmc [12:49] <durandal_1707> anyway proper way to report something is via bug tracker [12:49] <j-b> Apple considers itself like a 3rd party distributor [12:49] <durandal_1707> but he laft us, so he left FFmpeg too [12:49] <durandal_1707> *left this channel [12:50] <j-b> GPL is not possible on iOS, and it is the developers who must complain [12:50] <j-b> GPLv2 and GPLv3, for different reasons. [12:51] <j-b> GPLv3 because of the Tivo clause, notably [12:51] <j-b> GPLv2 because of the limitations of usage that Apple adds to it. [12:51] <JEEB> yup [12:52] <j-b> moreover, there are no reasons to not use LGPL version of FFmpeg [12:52] <Daemon404> for decoding, anyway. [12:52] <JEEB> yup [12:53] <kierank> Daemon404: tell that to blue lucy media [12:53] <kierank> :) [12:53] <JEEB> or all those people who grabbed their ffmpeg config from ffdshow tryouts or LAV [12:53] <JEEB> for whatever reason [12:53] <Daemon404> kierank, no no their old player is deprecated! [12:54] <kierank> Daemon404: it was funny when we were talking about their gpl violation and he walked through [12:54] <kierank> I need someone to request a trial of blue lucy [12:54] <kierank> to see if it still violates [12:55] <kierank> I trolled them by requesting a trial and saying my interesting in the program was for "checking for gpl violation" [12:55] <kierank> interest* [12:55] <JEEB> lol [12:55] <Daemon404> lul [12:56] <kierank> naturally i didn't get access to the trial [12:56] <durandal_1707> you need to donate big bucks [12:58] <durandal_1707> is fmod() always available? [13:48] <cone-373> ffmpeg.git 03Michael Niedermayer 07master:da30d0cec3eb: libavcodec/mpegaudio_tablegen: clip value before casting [14:23] <Daemon404> man im smart [14:23] <Daemon404> forgot to --amend before i send the v2 [15:26] <ubitux> BBB: did you see the emu_edge bug raised by ingo on ffmpeg-user? [15:27] <durandal11707> http://ffmpeg.org/pipermail/ffmpeg-user/2013-October/018101.html [15:29] <ubitux> last post give some little insight, carl replied on cvslog, dunno if you're registered [15:29] <durandal11707> i'm not because i do not intend to receive merge spam [15:30] <ubitux> it was for BBB obviously [15:31] <durandal11707> but on cvs ml you receive all commits [15:31] <ubitux> i like it personally [15:32] <durandal11707> i don't [16:01] <ubitux> http://voices.canonical.com/jussi.pakkanen/2012/09/25/the-code-documentatio… [16:02] <ubitux> (old, i know) [16:02] <ubitux> just came accross it, 'find it relevant to the project somehow [16:03] <ubitux> (and yes i know i tend to add similar noise at times) [16:43] <ubitux> seriously, the K&R cosmetics patch are getting more and more insane [16:44] <ubitux> http://pastie.org/8421508 :') [16:44] <ubitux> the 80-line break breaks almost the whole patch [16:45] <ubitux> http://pastie.org/8421514 wtf is this non-sense split? [16:46] <ubitux> also lol: http://pastie.org/8421515 [16:46] <Daemon404> looks like automated shit [16:46] <ubitux> and how the hell can this be considered an improvement: http://pastie.org/8421517 [16:47] <ubitux> Daemon404: as always [16:47] <Daemon404> also in case you missed [16:47] <ubitux> "i'm applying the uncrustify, please review the output for me" [16:47] <Daemon404> https://gist.github.com/dwbuiten/7101755 [16:47] <Daemon404> lots of segfaults in ffmpeg from unchecked mallocs [16:47] <ubitux> yeah i saw you mentioning it [16:48] <ubitux> that's pretty cool [16:48] <ubitux> can you add it in tools/ or something with a simple Makefile? [16:48] <Daemon404> it's extremely linux specific right now [16:48] <Daemon404> GCC specific even [16:48] <Daemon404> no idea if its at all portable [16:48] <ubitux> that's why i'm suggesting tools/ [16:49] <Daemon404> right [16:49] <Daemon404> i have a load of mem check patches to send later today too [16:49] <Daemon404> ill do that after [16:50] <ubitux> i can add a fate instance with that if you don't [16:50] <ubitux> but if the file is tracked, that's simpler for me [16:50] <Daemon404> not sure how useful hate instances are [16:50] <Daemon404> a bunch will fail obviously [16:50] <Daemon404> but we're only interested in some of the failures [16:50] <Daemon404> (i.e. 139 exit codes) [16:50] <ubitux> well if you add one i won't need to do it [16:51] <Daemon404> i think it'd need some postprocessing after [16:51] <Daemon404> like report generation [16:51] <Daemon404> otherwise itll be lost in noise of normal failures (i.e. exit code 1) [16:51] <ubitux> av_assert(0) [16:51] <ubitux> gdb backtrace on the core [16:51] <ubitux> too archaic i guess? [16:52] <Daemon404> ill think about how to set up an automated page [16:52] <Daemon404> probably just a script/cronjob on my home pc [16:53] <ubitux> if you do, please make sure that code is available in case we need to setup another box [16:53] <Daemon404> sure. [16:55] <ubitux> doesn't such tool already exists btw? [16:55] <Daemon404> probably [16:55] <Daemon404> i googled a bit but i didnt see anything [16:55] <Daemon404> not for posix_memalign anyway [16:55] <Daemon404> plus i was bored. [16:57] <nevcairiel> BBB: ubitux: i have a vp9 clip which produces a lot of "[vp9 @ 05982060] Invalid frame marker" messages and shows corruption, it seems to play OK in Chrome, interested? [16:58] <ubitux> sure [16:58] <ubitux> not sure i'll have a look at it soon though [16:59] <nevcairiel> http://files.1f0.de/samples/Bad%20Meets%20Evil%20-%20Lighters%20ft.%20Bruno… [17:35] <ubitux> nevcairiel: works fine here [17:35] <ubitux> how do you reproduce? [17:36] <nevcairiel> i play the video :D [17:36] <ubitux> ffplay? [17:36] <ubitux> at what ts it happens? [17:37] <ubitux> i have no problem playing with ffplay or decoding the file [17:37] <nevcairiel> my code was 48 hours old, but i dont suppose it was fixed? guess i'll have to dig deeper first [17:49] <ubitux> lol @ functionnal changes in K&R comits [17:50] <Daemon404> where? [17:50] <ubitux> at the end at least [17:51] <nevcairiel> ubitux: i found the reason, avcodec_decode_video2 doesn't properly return the number of consumed bytes for vp9, so my code tries to call the decode function again with the remaining buffer and cause problems .. the ff* tools tend to simply ignore the return and assume the decoder decodes the full buffer :p [17:52] <nevcairiel> if i also ignore the return, it works fine [17:52] <Daemon404> sounds like a vp9 bug [17:53] <nevcairiel> my guess is line 3496 in vp9.c should be "return avpkt->size;" instead [17:53] <ubitux> nevcairiel: that should be fixed; but i though each decoder was supposed to decode a whole packet every time [17:53] <ubitux> at least for video [17:53] <Daemon404> thats untrue [17:53] <Daemon404> you can have multiframe packets [17:54] <ubitux> in video? [17:54] <nevcairiel> yeah, some decoders support that [17:54] <ubitux> it seems to me that we are always assuming 1 packet = 1 frame in all the code [17:54] <nevcairiel> h264 can have a field in every packet, needing two for one frame [17:54] <ubitux> i need such sample [17:55] <Daemon404> basically ffmpeg ignores its own api [17:55] <Daemon404> :P [17:55] <nevcairiel> anyhow, i'll send a patch for this for review [17:59] <nevcairiel> but thats what i get for implementing the API properly, headaches that ffplay cannot reproduce! [17:59] <ubitux> yeah [17:59] <ubitux> so now the question is [18:00] <ubitux> do we really have even a single video decoder that relies on this behaviour? [18:00] <nevcairiel> you can set a flag on some decoders that allow it to be fed chunked input [18:01] <nevcairiel> ie. basically it does what the parsers do internally then [18:01] <nevcairiel> i believe mpeg4 has that [18:01] <nevcairiel> otherwise, i have no idea, since ffmpeg probably also ignores the return, someone would've noticed, no? [18:07] <cone-487> ffmpeg.git 03Derek Buitenhuis 07master:692b93090443: lavfi/pthread: Avoid crashes/odd behavior caused by spurious wakeups [18:08] <cone-487> ffmpeg.git 03Derek Buitenhuis 07master:55ae13e3deff: nut: Fix unchecked allocations [18:14] <cone-487> ffmpeg.git 03Michael Niedermayer 07master:77ef53881867: avcodec/vcr1: Fix bitstream input size check [18:14] <cone-487> ffmpeg.git 03Michael Niedermayer 07master:88c27193c7fd: avcodec/vcr1: print the actual size when its insufficient

1 0

ffmpeg.log.20131022
by burek 23 Oct '13

23 Oct '13

[00:29] <llogan> brontosaurusrex: ffmpeg -i input -c:v prores -map 0 -f segment -segment_time 10 out%03d.mov [00:30] <brontosaurusrex> llogan, thanks, but not working either, output files are borken [00:30] <brontosaurusrex> i guess they are not "streamable" enough [00:31] <brontosaurusrex> i mean mov is not [00:31] <llogan> worked for me [00:31] <llogan> ffmpeg -f lavfi -i testsrc=d=60 -c:v prores -map 0 -f segment -segment_time 10 out%03d.mov [00:31] <brontosaurusrex> hmm [00:31] <brontosaurusrex> should that also work with transiting? [00:32] <brontosaurusrex> transcoding* [00:33] <brontosaurusrex> let me retry [00:33] <llogan> re-encoding? I always thought transcoding as re-encoding while using some info from the source like motion vectors [00:33] <Hfuy> Hello; does anyone here have any experience encoding prores for use with quicktime-based applications? [00:34] <llogan> i thought you didn't like ffmpeg [00:34] <Hfuy> What gives you that idea? [00:35] <llogan> maybe i think you're someone else [00:35] <Hfuy> Either way, my ire is currently reserved almost entirely for Adobe Premiere Pro CS5 [00:35] <Hfuy> Which is crashing on import of files created by ffmpeg when I'm not really aware there's any reason it should. [00:36] <llogan> i can try to duplicate the issue in CS6 [00:36] <Hfuy> That would indeed be an interesting experiment. [00:36] <Hfuy> I could even conceivably send you a file. [00:36] <llogan> sure [00:37] <llogan> but the command and console output will be useful too [00:37] <brontosaurusrex> llogan, http://paste.debian.net/plain/60272 < Data stream encoding not supported yet (only streamcopy) [00:37] <Hfuy> The underlying issue, so you know the background, is that this stuff was shot to an Atomos Ninja recorder set to record 4ch audio, which Premiere <6 dislikes. [00:38] <Hfuy> So, I thought I'd http://pastebin.com/mkt8ZV6F it. [00:38] <Hfuy> And now I have this issue where Premiere CS 5.0.3 crashes on import. Sometimes. Only sometimes. [00:38] <Hfuy> I have reinstalled both Premiere and Quicktime. [00:42] <Hfuy> Very short example file: http://filebin.ca/zEGr54Rm2Jb/00001_2ch_apcn.MOV [00:43] <Hfuy> I would try DNxHD, which would require reencoding, but I could live with that. But I can't find a commandline that works. [00:49] <Hfuy> brontosaurusrex: I had that earlier. Not sure what the fix is. [00:51] <brontosaurusrex> well, at least I have a fresh ffmpeg/x264 compiles today :) (8bit and 10bit even) [00:51] <llogan> brontosaurusrex: it will probably work if you omit or stream copy the data stream [00:51] <Hfuy> well, sure, but does it solve the problem? [00:52] <brontosaurusrex> llogan, right, thanks [00:53] <Hfuy> I'm not sure if what I'm trying to do is reasonable. [00:54] <Hfuy> I'd be happy to try dropping the audio entirely if it'd help. [00:57] <llogan> is the example file the input or ffmpeg output? [00:57] <Hfuy> Output [00:57] <Hfuy> Your CS6 should read the input as is, since they fixed the multichannel audio bug. [00:58] <llogan> oh, duh...i didn't see the "encoder" info in the file [00:58] <Hfuy> Two characters are comically omitted from "Atomos" and "Ninja" [00:59] <llogan> does it crash if the file contains no audio? [00:59] <Hfuy> Not tried it. [01:00] <llogan> how many times do you need to import into CS5 for it to crash usually? [01:00] <Hfuy> It seem to crash reliably at ten or twenty. [01:00] <Hfuy> I have 94 takes to import in total. [01:00] <Hfuy> I have not been able to get through 20 in a go, I don't think. [01:01] <llogan> can you confirm that this particular short sample also causes issues? [01:01] <Hfuy> Wait one. [01:02] <brontosaurusrex> Hfuy, do you need the TC track? [01:02] <Hfuy> Not desperately. [01:03] <Hfuy> llogan: Just crashed CS5 importing 10 copies of that test file at once. [01:03] <Hfuy> Locks up somewhere during the "importing" progress bar. [01:03] <Hfuy> Often right at the start. [01:03] <brontosaurusrex> it does import into cs6 here Hfuy [01:04] <llogan> ok. i guess i'll blow the dust off of Windows and try it too. [01:04] <Hfuy> I suspect it may be a QT thing [01:04] <brontosaurusrex> mac version of cs6 that is [01:04] <Hfuy> although in my experience quicktime is pretty tolerant of lavc prores [01:04] <llogan> which quicktime version? [01:04] <Hfuy> downloaded today, whatever that is [01:05] <Hfuy> 7.7.4 apparently [01:05] <Hfuy> VLC and Quicktime play them happily, fwiw. [01:05] <Hfuy> Bridge CS5 too. [01:05] <Hfuy> And AE. [01:08] <Hfuy> Oh well, that was fun while it lasted! [01:16] <brontosaurusrex> Hfuy, i'd try to remove the TC track and see what happens [01:16] <Hfuy> Love to [01:16] <Hfuy> I was playing with -map earlier [01:17] <brontosaurusrex> llogan has probably booted into windows and will be back, just guessing [01:17] <Hfuy> I assume that if there's only one line under "stream mapping" there's only one track. [01:17] <Hfuy> er, stream. [01:17] <Hfuy> Whatever the term is this week. [01:19] <llogan> Hfuy: i can duplicate the issue in PP 6.0.3, QT 7.7.2, Win 7 Pro 64bit [01:20] <Hfuy> Well I guess that's both good and bad. [01:20] <llogan> on the same input file, and also on a version with no audio and no data stream [01:20] <Hfuy> Me too. [01:20] Action: Hfuy taps chin [01:20] <llogan> do non-ffmpeg muxed inputs cause the same problem? [01:20] <Hfuy> Ummmm. [01:21] <Hfuy> I can import stuff from a Ninja which had the audio setting to "2 channel" straight off the recorder, if that's what you mean. [01:21] <Hfuy> So it does in general read prores quicktimes, yes. [01:21] <Hfuy> The entire rest of the show was shot on that configuration. The problem is the pickups day. [01:22] <llogan> i did not test any ProRes not from ffmpeg [01:22] <Hfuy> not sure what you mean by non-ffmpeg muxed [01:23] <llogan> any other prores in mov files. if they also make PP crash then that would (probably) exclude ffmpeg [01:24] <Hfuy> Well as I say I have lots of prores in mov files, the whole show is made of them. [01:24] <Hfuy> The problem is ffmpeg specific. [01:25] <llogan> ok. consider submitting a bug report. [01:25] <Hfuy> I shall, but that doesn't really solve my immediate-term problem. [01:25] <Hfuy> I was pondering transcoding it all to dnx [01:25] <llogan> i'll try some more tests later and i'll provide input if i find anything useful [01:26] <Hfuy> I'm not sure anyone on the coding team will be that interested. As far as I know none of them are editors so they're unlikely to be able to recreate the issue due to lack of Premiere. [01:26] <llogan> i have premiere [01:26] <Hfuy> Also since it's Premiere's crash it will be very hard to detail and fix. [01:26] <Hfuy> And arguably, thus Premiere's bug. [01:26] <Hfuy> If it can't deal with it, it should say. [01:27] <llogan> if only outputs from ffmpeg are the problem then something is probably wrong with ffmpeg [01:27] <Hfuy> I wonder if something is being caused to explode in the quicktime-to-application bridge. [01:28] <Hfuy> That is often the case with weird undocumented no-apparent-cause lockups. [01:28] <Hfuy> All that aside, I really need to be cutting this stuff. Any suggestions? [01:29] <llogan> and i wish it wouldn't crash as much with debugmode frameserver too [01:29] <Hfuy> It's weird because it does the same thing with duped video as far as I can tell. [01:29] Action: Hfuy tries it [01:30] <llogan> i guess you can try re-encoding it if you're on a deadline. lately i've been using utvideo for intermediates [01:31] <Hfuy> utvideo? [01:31] <Hfuy> a codec? [01:31] <llogan> ffmpeg has native encoder and decoder (also supports libutvideo). [01:31] <Hfuy> Has to go into premiere [01:32] <llogan> http://umezawa.dyndns.info/archive/utvideo/?C=M;O=D [01:32] <llogan> works fine in premiere [01:32] <Hfuy> Hmm. Glitch is confirmed with ffmpeg -i infile -c:v copy -map 0:1 outfile [01:32] <Hfuy> so it isn't even the prores codec, it's the muxer [01:32] <Hfuy> What's very strange is that pr imported ten of them happily but choked on 20. [01:33] <llogan> that's what i assumed you did in the first place (I already forgot pasted details) [01:33] <Hfuy> well obviously I'm not desperate to reencode the pictures. [01:33] <llogan> i know, but that's why i suggested a lossless encoder if you have the drive space [01:33] <Hfuy> But if it's that or not cut them, well. [01:33] <Hfuy> I have a 4TB RAID10 on tap. [01:33] Action: Hfuy patpats drives [01:34] <llogan> how do you like the RAID10? [01:34] <Hfuy> A hell of a lot better than I've liked any RAID5. [01:34] <Hfuy> It's cheaper for the same capacity (except at ludicrously high spindle counts), faster in use, faster to rebuild. [01:35] <Hfuy> It retains its full performance even when degraded, and retains greater redundancy when degraded. [01:35] <llogan> i've been reading about it lately and decided for raid10 (the old raid1 system died) [01:35] <Hfuy> I'm not sure why it's even a consideration to anyone. [01:35] <Hfuy> Unless you're using a huge number of disks (in which case you'll need RAID6 anyway) the cost of the extra drives for a given capacity will be less than the cost of a high performance RAID5 controller. Which are very expensive. [01:36] <llogan> i noticed those prices... [01:36] <Hfuy> Anything other than the big Areca controllers suck horribly. [01:36] <Hfuy> And the big Areca controllers can be the cost of five or six disks. So it would need to be really bloody noticeably better than RAID10. An it really isn't. [01:38] <llogan> i wonder if anything else in mov also crashes Premiere [01:38] <llogan> and if it is a regression or not [01:41] <Hfuy> Not a clue. [01:41] <Hfuy> What's a "regression" [01:42] <llogan> if something used to work in an older version of ffmpeg [01:42] <Hfuy> Oh, I see. [01:42] <Hfuy> I never had this problem before. [01:42] <Hfuy> And isn't the mov encoder essentially the same as the mp4 encoder? [01:42] <llogan> so more testing is needed. maybe i'll try it later this week and make a bug report if you don't [01:43] <Hfuy> "VLC does not support the audio or video format "ULRA"." [01:44] <Hfuy> CS5 doesn't like utvideo, even after I installed the codecs. [01:44] <llogan> did you restart CS5? [01:45] <Hfuy> Yes. [01:45] <Hfuy> "File import failure... Codec missing or unavailable" [01:45] <llogan> worksforme (in CS6). [01:45] Action: Hfuy queues up a long stream of expletives and lets them all go at once [01:46] <Hfuy> This was easier back in the days of 35mm and scissors. [01:46] <llogan> i don't see why it wouldn't work in cs5 [01:46] <Hfuy> If it's a directshow codec presumably wmp would play it. And it won't. [01:47] <Hfuy> Something is bork3d. [01:47] <llogan> http://trac.ffmpeg.org/wiki/FFmpegPremierePro [01:47] <llogan> shows utvideo being used [01:47] <Hfuy> lemme see if it's in the output options [01:47] <Hfuy> since nothing seems happy to use utvideo, I suspect the installer didn't. [01:48] <Hfuy> Hn, it's in the output list. [01:49] <Hfuy> Heehee, did you write this guide? [01:49] <Hfuy> "to make sure Premiere did not screw around with the frame size, frame rate, etc, as it is apt to do." Why yes, yes it is! [01:50] <llogan> yeah, i did. [01:50] <Hfuy> Well, thanks for doing so. [01:51] <llogan> i was sick of using AME [01:51] <Hfuy> I've never been an avid guy. [01:51] <llogan> same. maybe i should have [01:51] <Hfuy> Every time I see Avid, I'm reminded why not. [01:51] <Hfuy> Can't see the joy in it. [01:52] <Hfuy> Avid always seemed like computer software for video editing people, whereas I was always more of a computer person looking for video editing software. [01:52] <Hfuy> If that makes sense. [01:53] <Hfuy> Oh. AVI wrapper, not quicktime? [01:54] <llogan> they both suck so i just chose the first that i tried that worked [01:54] <Hfuy> AVI works. MOV doesn't. [01:54] <Hfuy> I assume Pr sends the MOV through Quicktime, which (understandably) responds "WTF?" [01:55] <Hfuy> whereas it goes to directshow with the AVI. [01:55] <llogan> too bad Premiere is too retarded to use matroska. (hell, i remember when it woudln't accept raw DV). [01:55] <Hfuy> I occasionally work for the company that currently owns and runs Lightworks. [01:55] <Hfuy> They would dearly, dearly love to throw out their entire I/O layer and use avformat and avcodec [01:56] <Hfuy> Unfortunately it would make Lightworks impossible to distribute in the US, which is not doable. So. The suck continues. [01:56] <llogan> why don't you use that instead of Premiere? i've been meaning to try, but i don't edit much anymore [01:56] <Hfuy> I know Premiere. [01:56] <Hfuy> I have no reason to change. [01:56] <Hfuy> Although this Creative Cloud sh*t is making me reconsider. [01:57] <llogan> note that the frameserver output can be out of sync with audio, so if you try that it will work if you export the audio by itself from premiere and then re-encode and mux with ffmpeg [01:58] <llogan> (I should update that page) [01:58] <Hfuy> Frameservers are cool. [01:58] <Hfuy> I once built a very, very rickety pile of software which used both ffmpeg and avisynth to produce low-res proxies with BITC on a digital cinema recorder. [01:59] <llogan> nice [01:59] <Hfuy> Well, no, it wasn't nice, but it worked :) [01:59] <Hfuy> I'm sure I'm going to comp sci hell as a result. [02:00] <Hfuy> avisynth must hook the filesystem to do that, surely [02:02] <Hfuy> Ha. Pr supports 4ch audio in AVI, after all that. Puts it on a 5.1 track. [02:08] <Hfuy> I must depart sleepwards. Thanks for the heads-up on utvideo. [02:09] <llogan> Hfuy: ok. ill let you know if i get to test more [02:10] <Hfuy> Could be a bit of a nit to find I fear. [02:38] <SSlater> Could anyone point me to a tutorial or similar on basics of video formats and containers. From security camera point-of-view, I need to learn more about what is best suited for viewing detail immediately vs needs for archiving and disk space. Then turn these different requirements int ffmpeg options. [03:28] <littlebat> Comparison of container formats http://en.wikipedia.org/wiki/Comparison_of_container_formats [03:28] <littlebat> Comparison of video codecs http://en.wikipedia.org/wiki/Comparison_of_video_codecs [03:29] <littlebat> Video Converting Configure http://www.learndiary.com/mobilemate/video_configure [03:32] <SSlater> littlebat: Thanks for that. The links in the last one look very informative. [03:34] <SSlater> I had come across the container format link, but didn't know enough tomake much sense of it. Will go back to study these. [09:51] <damir__> hello [09:51] <damir__> can ffmpeg be used to get information about video frame type for each frame? [10:44] <xingchao_> does anyone know background of this bug? https://trac.ffmpeg.org/ticket/1035 [10:45] <xingchao_> the patch can also fix my mosaic issue, but i really donot know why [10:47] <xingchao_> my question is here: http://ffmpeg-users.933282.n4.nabble.com/Mosaic-issue-when-playing-RMVB-wit… [10:49] <ubitux> this was commited beginning of last year [10:49] <xingchao_> yes, that's right [10:51] <xingchao_> it fixed the two audio tracks issue while switching to another one because the queue cached a lot of packets [10:52] <xingchao_> but i donot think it has any relationship with the mosaic [10:52] <xingchao_> for mosaic issue, i thought the "seeking" didnot find the keyframe [14:13] <renevolution> Hi there, can anyone guide me how or if i can display data created by astats filter using ffplay/ffmpeg? [14:27] <renevolution> Ok, i see it works if i reencode the file and use -filter_complex & is there a possibiliy to output this data live ? [14:28] <renevolution> i mean, continuously ? [15:22] <Hfuy> Do we know why VLC doesn't support the utvideo codec? I was under the impression that they used the same core libraries as ffmpeg. [15:24] <durandal11707> yes they do use lavf/lavc, but you should ask them [17:43] <stein_> hi all [17:43] <stein_> how may i convert P2 structure to friendlier single file format? [17:49] <steinchen> re [17:50] <steinchen> how may i convert P2 structure to friendlier single file format? just cat all together? [17:52] <xlinkz0> i want to write to /dev/null to test a rtsp server using the ffmpeg client [17:53] <xlinkz0> how do i set mp4 format? [17:53] <xlinkz0> because i get [NULL @ 0x1d552c0] Unable to find a suitable output format for '/dev/null' [17:54] <bencoh> -f fmt [18:03] <Hfuy> steinchen: Depends what sort of P2 it is. [18:03] <Hfuy> But in general, MXF is a nightmare. [23:54] <Eduard_Munteanu> Hi. [23:55] <Eduard_Munteanu> Would it be reasonable to use ffmpeg as an executable and pipe data from/to it to implement a media player? [23:55] <klaxa> i think using the C-API would be more advisable [23:56] <Eduard_Munteanu> I guess so, I'm weighing my options. [23:56] <JEEBsv> I don't think you could implement a player with just the ffmpeg executable [23:57] <JEEBsv> you would have to use the libraries to have any kind of proper control on it [23:57] <Eduard_Munteanu> Hm, I guess seeking might be a problem for one thing. [23:57] <JEEBsv> glad that you came up with it yourself :P [23:57] <Kuukunen> JEEBsv: why not? pipe stuff in, read uncompressed out? :p [23:58] <Eduard_Munteanu> However I don't really want to seek generally, just seeking forward (i.e. just dropping stuff). It's supposed to be an unattended media player. [23:58] <Kuukunen> ...what are you trying to do that you can't do with an existing player? [23:59] <Eduard_Munteanu> Kuukunen: NTP-synchronized playback. I did it with mplayer but it's a bit icky. [23:59] <Eduard_Munteanu> Kuukunen: because mplayer's synced playback is unreliable. [00:00] --- Wed Oct 23 2013

1 0

ffmpeg-devel.log.20131021
by burek 22 Oct '13

22 Oct '13

[00:26] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:712eff49789a: fate: add synthetic swresample resampling tests [01:22] <michaelni> mraulet, smarter, it seems icc doesnt like the hevc code, could like a compiler bug though (http://fate.ffmpeg.org/log.cgi?time=20131020212707&log=compile&slot=x86_32-…) [01:25] <mraulet> and with --disable-asm? [01:30] <michaelni> i dont know i dont have icc installed here, its thilos fate client [01:34] <michaelni> just sent thilo a mail, you are in CC [11:20] <cone-755> ffmpeg.git 03Mickaël Raulet 07master:09ba986cae14: avcodec/hevc: fix duplicate/wrong nal unit check and difference to openhevc_upstream [12:57] <durandal_1707> geq=p(X+sin(2*PI*Y/50+N)*5\,Y):p(X+sin(2*PI*Y/50+N)*5\,Y):p(X+sin(2*PI*Y/50+N)*5\,Y) [13:01] <durandal_1707> 2/win3 [13:10] <saste> michaelni, can't see ffprobe coverity issues, did you mean tee? [13:10] <Daemon404> i saw it earlier [13:10] <Daemon404> it listed it as ffprobe.c [13:12] <durandal_1707> there is no way to find out if expression matches previous one? [13:14] <saste> durandal_1707, st(0, expr0); st(1, expr1); if(eq(ld(0), ld(1), ...) [13:15] <saste> Daemon404, yep [13:15] <durandal_1707> saste: i mean inside code [13:16] <durandal_1707> idea is to not recalculate some values. which may be same all the time [13:16] Action: durandal_1707 thinking about making ripple filter, which would basicaly do above expression but much faster [13:17] <saste> durandal_1707, we discussed some way to mark constant expressions [13:17] <saste> in some cases it can be done, but we need some tweaks in the eval code [13:17] <durandal_1707> can i just to memcmp? [13:24] <saste> durandal_1707, AVExpr is a recursive struct IIRC [13:39] <cone-755> ffmpeg.git 03Stefano Sabatini 07master:ddaf33f5bba5: ffprobe: fix consistency checks in parse_read_intervals() [13:47] <Compn> or just optimize geq ? [13:47] <Compn> :P [13:47] Action: Compn is late [13:50] <cone-755> ffmpeg.git 03Carl Eugen Hoyos 07master:87eae03aacaf: Support FF_DISABLE_DEPRECATION_WARNINGS with icc on Linux. [14:34] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:80b8f78048ba: fate: enable hevc tests [15:46] <cone-755> ffmpeg.git 03Paul B Mahol 07master:f5ed83c3831f: avformat/mxfdec; parse random index pack to get last partition offset and use it when footer partition offset is missing [15:57] <cone-755> ffmpeg.git 03Paul B Mahol 07master:c65521693914: make it possible to compile ffmpeg without swscale [16:24] <mateo`> anyone at linuxcon ? [16:25] <durandal_1707> where is it? [16:30] <mateo`> edinburgh [16:51] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:ab2bfb85d49b: avfilter/ff_insert_pad: fix order of operations [17:29] <saste> wm4, what happened to your libswscale API patch? [17:39] <durandal_1707> picked invisible color [17:52] <wm4> saste: waiting for this filter stuff... [17:52] <saste> wm4, will send an updated patch now [18:05] <Daemon404> michaelni, ping [18:05] <michaelni> Daemon404, pong [18:05] <Daemon404> michaelni, what do you mean by do "i" have padidng input bytes? [18:06] <Daemon404> crash was produced with ffmpeg cli, not my own code [18:06] <michaelni> then theres a bug [18:07] <durandal_1707> in overread [18:08] <Daemon404> not sure what you mean [18:08] <Daemon404> @ michaelni [18:08] <michaelni> can you share the file / testcase that causes the crash ? [18:09] <Daemon404> let me see [18:12] <michaelni> if not some valgrind otput that shows where it was allocated might be interresting too [19:48] <cone-755> ffmpeg.git 03Diego Biurrun 07master:23e6d3d4e27c: fate: vpx: Remove one redundant $(eval) invocation [19:48] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:e45460aab0a9: Merge commit '23e6d3d4e27c83e06d00aa072c84d057bd1dad23' [19:55] <durandal_1707> http://tinypic.com/r/wcbw2a/5 [20:07] <cone-755> ffmpeg.git 03Ben Jackson 07master:5152196b2b47: pthread: Avoid spurious wakeups [20:07] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:2444fca523d7: Merge commit '5152196b2b47d446d9a509b9bfb318daa26a1f2b' [20:17] <cone-755> ffmpeg.git 03Luca Barbato 07master:ea71aafd6881: avio: Use AVERROR_PROTOCOL_NOT_FOUND [20:18] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:2f9422d951dc: Merge remote-tracking branch 'qatar/master' [20:20] <Daemon404> AVERROR_PROTOCOL_NOT_FOUND... thats a new and oddly specific code [20:21] <durandal_1707> should it be reverted? [20:22] <Daemon404> no [20:22] Action: durandal_1707 will push filter without documentation [20:27] <wm4> it's not really "new" [20:29] <wm4> though this use was added only in 2012 [20:30] <wm4> and the error code back in 2010 (odd...) [21:53] <ubitux> is daniel kang on irc? [21:57] <gnafu> If I recall correctly, that's Jumpyshoes. [21:57] <ubitux> indeed, just noticed that from grepping irc logs [21:58] <ubitux> Jumpyshoes: i would suggest to use a macro for the dup in mmx filter line [21:58] <ubitux> i didn't do that for the inline asm to avoid a big diff (and because it's not really as "free" as with yasm) [21:59] <ubitux> but i think it would be relevant here [21:59] <ubitux> also, the comments are actually useful, you should keep them (and puting this code in a macro will avoid the redundancy of them) [22:03] <ubitux> durandal11707: have fun fixing the deps [22:05] <durandal11707> ? [22:14] <durandal11707> ubitux: ??? [22:14] <ubitux> durandal11707: see fate failure [22:14] <ubitux> i'm assuming this is because of the assumed swscale dep in ffmpeg for the fate tests [22:15] <durandal11707> who added such flag? [22:17] <durandal11707> well previously ffmpeg would not get build at all, so nothing was tested at all [22:17] <durandal11707> i consider that a huge improvement [22:20] <cone-755> ffmpeg.git 03Paul B Mahol 07master:c9660999b9a7: avfilter: replaygain scanner [23:36] <durandal11707> michaelni: i get crash with this: ffmpeg -f lavfi -i smptebars -f lavfi -i smptebars -lavfi "[1]cop[x],[0][1]overlay" -f null - [23:37] <durandal11707> *ffmpeg -f lavfi -i smptebars -f lavfi -i smptebars -lavfi "[1]copy[x],[0][1]overlay" -f null - [23:40] <durandal11707> or this: ffmpeg -f lavfi -i color -f lavfi -i color -filter_complex "[1]null[x],[0][1]overlay" -f null - [23:41] <durandal11707> i have basic ripple filter, anybody wants it? [23:49] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:7e3e653618a5: avformat/bintext: protect against potential overflow of chars_per_frame [23:49] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:d47e14b53a39: h263dec: use init_get_bits8() and check its return code [23:58] <cone-755> ffmpeg.git 03Nicolas George 07master:59f809e9922a: lavf/tee: fix leak of strdup/strtoked buffer. [23:58] <cone-755> ffmpeg.git 03Nicolas George 07master:5a12142aca22: lavf/tee: fix leak of select option. [23:59] <cone-755> ffmpeg.git 03Nicolas George 07master:c4e6024adc18: lavf/tee: fix leak of bsfs array. [23:59] <cone-755> ffmpeg.git 03Michael Niedermayer 07master:23c06984995e: Merge remote-tracking branch 'cigaes/master' [00:00] --- Tue Oct 22 2013

1 0

ffmpeg.log.20131021
by burek 22 Oct '13

22 Oct '13

[00:41] <x__> Hey all. Is there a way I can convert my .mov 1.9 gb files to a smaller mp4 format? like around 700 mb for a two our video? [00:42] <x__> I tried to use ffmpeg -i input.avi -acodec libfaac -b:a 128k -vcodec mpeg4 -b:v 1200k -flags +aic+mv4 output.mp4 but it isn't any smaller [00:43] <zennist> JEER: I've now got the resampling function to work. But interestingly only one channel outputs correctly, and the other channel is still noise... [00:44] <x__> I tried to run it on an AVI file, but it come out bigger? [01:54] <SSlater> G'day all. [01:55] <SSlater> I have a question about resampling video. Particularly which options are best to use in ffmpeg. [01:58] <SSlater> I have security camera video recorded from rtsp stream to mpg4 files at 25fps. Each day's file is about 30G. The thought was to reduce the file size each day by reducing the framerate, then archiving when down to 1fps. [01:59] <SSlater> The command is: ffmpeg -i test0.mpg4 -r 20fps test0-20.mpg4 [02:01] <SSlater> I am getting messages about headers not complete; picture size invalid; invalid dts/pts combination and Unable to find a suitable output format for 'test0-20.mpg4'. [02:02] <SSlater> Is something wrong with the input file? It plays with audio fine. [02:56] <SSlater> First problem solved. Changed the command to: ffmpeg -i test0.mpg4 -r 20 test0-20.mp4. [03:16] <poiupoiu> good evening everyone. I was hoping someone more familiar in linux can help me with a problem... [03:16] <poiupoiu> I've compiled ffmpeg from the commands given at https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [03:17] <poiupoiu> a script runs perfectly fine as my username, but bombs when i run it as another user. [03:17] <poiupoiu> poke around for a bit and find that as myself, it is using the compiled version, as other users it is using the avconv "ffmpeg" [03:19] <poiupoiu> does anyone know commands or has a chance to give me a hand with solving this? [03:21] <poiupoiu> http://askubuntu.com/questions/313217/install-latest-ffmpeg-for-all-users-i… <-- I think i found my solution... [03:58] <poiupoiu> And that got it. Thanks for letting me drop in for a minute! [07:56] <hans_henrik> ./ffmpeg -i ../../www/writhing.wmv -c:v libx264 pipe:0 [07:56] <hans_henrik> Unable to find a suitable output format for 'pipe:0' [07:56] <hans_henrik> pipe:0: Invalid argument [07:57] <JEEB> hans_henrik, you need to specify a container format [07:57] <JEEB> -f container_of_your_choice [08:02] <hans_henrik> JEEB, yup that was it! thanks [08:04] <hans_henrik> can i tell ffmpeg to be otherwise, completely quiet? [08:04] <hans_henrik> except the output file, which is piped to stdout [08:04] <hans_henrik> .. or, tell it to pipe all the stuff it normally puts on stdout, to stderr? [08:04] <hans_henrik> or [08:05] <hans_henrik> um [08:05] <hans_henrik> this doesnt have to be anywhere near foolproof, i guess i can just pipe it to stderr [08:41] <xingchao_> why does av_seek_frame() return non-key frame even i donot set *ANY flag? [08:41] <xingchao_> i thought it should return the key frame anyway [08:41] <xingchao_> does anyone know that root cause? [09:56] <zennist> I'm using a custom AVIO with my AVInputFormat and I found that this causes avformat_open_input to hang on some audio streams i.e. m4a [09:56] <zennist> Anyone has any idea? [09:57] <zennist> FFMpeg's default io implementation doesn't cause any problems though [10:06] <shur> http://stackoverflow.com/questions/19488762 [11:27] <GeorgesLeYeti> Hi [11:29] <GeorgesLeYeti> I encounter a problem for converting files with ffmpeg. [11:29] <GeorgesLeYeti> When i try to convert my file on my server it returns me: moov atom not found [11:30] <sacarasc> That sounds like you have an incomplete MP4 file. [11:30] <GeorgesLeYeti> But the thing is that in my local machine (with exact same ffmpeg version and file) it works [11:31] <GeorgesLeYeti> sacarasc: I copy the file from my server to my computer [11:33] <GeorgesLeYeti> this mean if the file is corrupted on my server it still should be corrupted on my computer [11:58] <Trax|wrk> maybe the copy is a download or it's an open file, try comparing both files with md5 checksum then you'lre sure they are the same [12:49] <Macey> hi all, the windows builds of ffmpeg, how different is it to the Ubuntu 13.04 release? I'm trying to transcode a TS into FLV but on windows i get a "Invalid data found when processing input" whereas under Ubuntu she happily transcodes away [12:50] <JEEB> ubuntu uses libav, not ffmpeg first of all [12:50] <plm> morning all [12:50] <JEEB> and since ubuntu only uses release code it's probably somewhat old libav, too :) [12:51] <JEEB> so yes, pretty different from a build you might be able to catch these days from zeranoe [12:52] <Macey> ffmpeg version N-57235-g0bf8580 - Windows [12:52] <JEEB> also if you are going to use libav-based tools, I recommend switching to the avconv tool instead of the ffmpeg tool. Because basically elenril rewrote parts of the ffmpeg tool, and the rewrited/updated one was then called avconv in libav. ffmpeg's ffmpeg does have these changes as well, since ffmpeg merges most if not all that comes from libav :) [12:52] <Macey> ok cool, i tried avconv aswell [12:52] <Macey> same issue [12:53] <JEEB> so the same issue comes up with current ffmpeg and the avconv on ubuntu? [12:53] <Macey> yeah, ubuntu both ffmpeg and avconv both happily transcode [12:53] <Macey> though windows versions don't [12:53] <JEEB> uhh [12:53] <JEEB> you just said... oh wait, you meant that you grabbed a current libav avconv build and that failed too, while the older versions in ubuntu work fine? [12:54] <JEEB> anyways, sounds like a bug [12:54] <Macey> yeah so ubuntu are just apt-get installs [12:54] <Macey> windows is latest d/ls [12:54] <JEEB> if it fails with zeranoe's current ffmpeg builds, you can report it in ffmpeg's trac, and libav problems can be reported at their respective bugzilla tracker [12:54] <JEEB> :) [12:55] <Macey> i think that the windows versions aren't built with TS support [12:55] <JEEB> there's nothing special you need for that [12:55] <JEEB> mpeg-ts is libavformat, part of ffmpeg/libav [12:56] <JEEB> you would have to specifically disable things for it to be disabled, so no -- that's not it [12:56] <Macey> hum [12:56] <JEEB> it's most probably a bug in the current libavformat code [12:56] <Macey> "Invalid data found when processing input" [12:56] <JEEB> also you could try building a simple build of ffmpeg or libav on the ubuntu i guess [12:56] <JEEB> and try that too [12:57] <JEEB> if it's just reading the input that fails then `ffmpeg|avconv -i welp.ts -c:v rawvideo -f null -` should be all needed to replicate [12:57] <JEEB> and the build would only need ./configure without extra options set [13:01] <JEEB> and yeah, you'll need the packages git, build-essentials and yasm to build ffmpeg/libav :) [15:25] <plm> people, is possible to increase the buffer whe streaming via http? I'm streaming a video em box1 to box2. but between box1 and box2 I have a high delay networking and bandwidth unstable (is 3g). So I was thinking to have a bigger buffer in box2 (before show video) for don't have congeals the video. I know that increase the buffer I will have a bigger delay on video, but this is not problem for me. A 30s delay is just fine. [15:40] <EchoDev> Anyone here who can help me with compiling ffmpeg on windows? [15:41] <TheCycoTWO> EchoDev: Would these work for you instead http://ffmpeg.zeranoe.com/builds/ [15:41] <EchoDev> nope :( I need a modified one [15:42] <EchoDev> WARNING: Unknown C compiler gcc, unable to select optimal CFLAGS <- Im getting this error [15:42] <EchoDev> I think I'm making a mistake with mingw somewhere [15:46] <EchoDev> Hmm I have found a complete package for compiling ffmpeg [15:46] <EchoDev> It seems to be working fine >:) [15:46] <TheCycoTWO> spiffy [15:46] <EchoDev> except for the libopencore_amrnb which I can fix [15:48] <TheCycoTWO> anyone here up on the latest recommendations for libswresample vs libavresample? I read when it was brought in that libavresample would be unmaintained, but is that still the case? [15:48] <durandal_1707> libav* is maintained by Libav [15:48] <durandal_1707> *libavresample [15:49] <durandal_1707> and it had some serious bugs, dunno if it still have as I do not use it [15:50] <TheCycoTWO> We are currently using libswresample exclusively. Ubuntu users keep whining in the issue tracker, so I was wondering if the best course was to maintain two code paths or switch. [15:51] <durandal_1707> how much of your userbase are Ubuntu users? [15:51] <JEEB> TheCycoTWO, if you want to support both I would recommend you use libavresample, since that is also available in ffmpeg as well [15:51] <JEEB> although of course libavresample is only maintained by libav, and just merged into ffmpeg [15:52] <JEEB> that said, there are plenty of projects using it already [15:52] <TheCycoTWO> Mostly windows users. Thanks JEEB. [15:52] <durandal_1707> but it is full of bugs [15:52] <TheCycoTWO> ah [15:52] <JEEB> uhh, if it was that full of bugs [15:52] <JEEB> I'm pretty sure nev wouldn't use it :P [15:52] <durandal_1707> that is nev problem [15:53] <JEEB> yes, but I'm just saying that if it was fully borked it wouldn't get any usage [15:54] <durandal_1707> even worse stuff have bigger usage [15:54] <JEEB> yes [15:54] <TheCycoTWO> nice of you guys to track it anyway, not cause a complete rift [15:55] <JEEB> durandal_1707, just saying that I'm not aware of any completely broken parts in it [15:55] <durandal_1707> they are so similar that it should be trivial to support both [15:55] <JEEB> yes, not really surprising as such [15:55] <JEEB> and that's another alternative [15:55] <TheCycoTWO> ? [15:56] <TheCycoTWO> ah, nm [16:03] <JEEB> durandal_1707, btw not only have you not actually noted any actual bugs (which no doubt there exist, since there is no perfect software -- but "full of bugs" as far as I know is not what it is), but I must say that even some people who wholly recommend ffmpeg usage want it around http://ffmpeg.org/pipermail/ffmpeg-devel/2013-October/149693.html [16:04] <JEEB> and if core ffmpeg folk have a problem with it I don't really know... you wanted to make some effort of maintaining interoperability, but you don't want to take it to the end? [16:05] <JEEB> (full interoperability between libav and ffmpeg is probably impossible, but most other things are manage'able) [16:05] <durandal_1707> i see it other way around: Ubuntu not providing FFmpeg packages [16:05] <JEEB> that's a whole separate problem of course [16:06] <JEEB> in a perfect world we would not have forks of ffmpeg, and all software would be perfect, unfortunately that's not the case. [16:08] <hans_henrik> discussing libav again? [16:09] <TheCycoTWO> sorry [16:10] <durandal_1707> you do not need to apologize [16:10] <TheCycoTWO> no, I'm Canadian, it's an obligation ;) [16:11] <JEEB> Just mostly noting that he might not actually be looking at it calmly but more from the "we need to keep people our APIs" stance with some superlative-likes :) Which is understandable of course. That said, I can understand people not wanting to have two separate code paths for two different libraries, as well as since I see ffmpeg-using people using libavresample... [16:11] <JEEB> just had to comment on it :) [16:12] <durandal_1707> that is littly funny considering that libswresample is older [16:12] <JEEB> I have no idea about those things and to be honest I do not have much interest in it either :) [16:13] <JEEB> I mostly care about the fact that there is a visible line of being able to use the same code path with both projects [16:16] <JEEB> of course there are also people who have opted to support both since avresample is disabled by default in ffmpeg's configure, like the project whose main developer is asking for avresample to be enabled by default :) https://github.com/mpv-player/mpv/blob/master/audio/filter/af_lavrresample.c [18:56] <TSM> im confused about the SAR/DAR values and if they can be used to resize the input DV file i have out to a file where it does not need to be streached out as when people play on VLC etc it does not do it automaticly with the DV file [19:02] <thisislawatts> I am looking into overlaying an image onto footage using a multiply blend mode. Wide eyeed noob here, but I have stumbled upon the filter_complex which I believe should do the trick, but I am not sure I am understanding the syntax completely [19:11] <Hfuy> Is there anything wrong with this: [19:11] <Hfuy> ffmpeg -i 00000.MOV -c 2 -c:a pcm_s16le -c:v prores -ac 2 00000_2ch_apcn.MOV [19:11] <Hfuy> All I really want to do is drop two of the audio streams in the input. But I don't seem to be able to do that without reencoding the video. [19:19] <Hfuy> Hello? Anyone? [19:20] <durandal_1707> -c:a copy [19:20] <durandal_1707> or -c:v copy [19:21] <Hfuy> Instead of what? [19:22] <durandal_1707> when you do not want to reencode/transcode use copy codec [19:22] <Hfuy> I'm also not sure why I have -c 2 and -ac 2 [19:22] <Hfuy> but it doesn't seem to work otherwise. [19:22] <durandal_1707> -ac 2 means make audio 2 channels [19:23] <Hfuy> The input file has (oddly) four. [19:23] <Hfuy> Which is the problem. My NLE doesn't like four channel prores. For some reason. Which is beside the point. :) [19:23] <durandal_1707> -c 2 is strange way to set codec [19:23] <Hfuy> er, I did. ffmpeg -i 00000.MOV -c 2 -c:a pcm_s16le -c:v prores -ac 2 00000_2ch_apcn.MOV [19:24] <durandal_1707> remove '-c 2' [19:24] <Hfuy> Uhhuh. [19:25] <Hfuy> Hm. This is rapidly becoming more of an Adobe Premiere problem than an ffmpeg problem. [19:26] <Hfuy> It does not like ffmpeg's output in that case, to the extent of actually crashing premiere. [19:26] <durandal_1707> use '-c:v copy' if 00000.MOV is already prores video [19:26] <Hfuy> durandal_1707: I'm not completely stupid but it's obvious that I'm not understanding something you're saying. What are you asking me to do? [19:27] <durandal_1707> to copy your full ffmpeg output of ffmpeg command you use [19:27] <Hfuy> Well, it is prores, but I've a feeling it's prores 4444, or something. [19:27] <Hfuy> I'm not sure what the default prores profile used by ffmpeg is. [19:27] <durandal_1707> if you use copy it should not really matter [19:27] <durandal_1707> well you said you do not want to reencode vide, no? [19:28] <Hfuy> Well, I may have to, if Premiere doesn't like what ffmpeg is doing. [19:28] <Hfuy> Or rather doens't like the way it's reordering things/ [19:28] <Hfuy> http://pastebin.com/mkt8ZV6F [19:28] <Hfuy> Ah. The input is apch [19:29] <Hfuy> Which is 422 HQ. As expected. Wtf. [19:30] <Hfuy> The stream ordering is coming out differently, can I control that? [19:32] <Hfuy> On the input, 0:1 is the video and it's coming out on 0:0,vice versa for the audio. Any idea why ffmpeg is doing that? [19:32] <Hfuy> (and ideally it would be nice to copy the metadata streams) [19:41] <Dresk|Laptop> Hey guys, been using ffmpeg a long time to encode stuff (3+ years now), and I still consider myself a novice -> amateur. Anyway, right now I'm dealing with the -map option, which is new to me. I basically have an arbitrary input file (contains video and audio stream), where I extract the audio stream into PCM16bit, peak normalize it, and then in one command I want to re-encode the video stream from the original file into MP4 AND take [19:41] <Dresk|Laptop> another input file (the normalized wav file) and use that as the audio [19:41] <Dresk|Laptop> I'm a bit confused as to the order of precedence necessary on the command line to have ffmpeg ignore the audio stream in the original file and instead use this 2nd input file, which is the audio stream peak normalized [19:42] <Hfuy> Dresk|Laptop: If you find a decent -map reference please tell me. [19:42] <Dresk|Laptop> Hfuy: Well one of the things I considered is extracting the video as well, and just working with two raw input streams, which would be easier [19:42] <Dresk|Laptop> Hfuy: But I don't want to do that, I use 2 pass encoding and want to keep the command simple [19:43] <Hfuy> Not a clue. [19:43] <Hfuy> It is possible, as I understand it. But I have no idea how. [19:43] <Hfuy> Personally, I have a situation where a command is reordering the streams in the file when I don't want it to, which I suspect is causing my NLE to fritz out. But I don't know. [19:46] <Hfuy> If it's any consolation, I have found it very difficult to find up to date, authoritative documentation for this. [19:46] <Dresk|Laptop> The order in which you write your ffmpeg call does matter, correct? Or can you provide enough info on every command line option to make the order independent? I've always found it confusing to work "first with one stream" and then another, sometimes using the same parameter but for two different streams [19:47] <TSM> it it possable to detect using ffprobe if a DV video is shot as 16:9 instead of 4:3 [19:47] <Hfuy> There are some situations where option ordering on the command line matters. But I'm not sure when or how. Again, I can't find any authoritative docs. [19:47] <Hfuy> TSM: Probably, see if it outputs a value for pixel aspect ratio (PAR) or display aspect ratio (DAR). [19:48] <Hfuy> That information is probably duplicated inside the DV codec's payload in each frame, but I don't think ffmpeg can read that. [19:49] <TSM> Hfuy: no PAR any longer on new version of ffprobe, [SAR 8:9 DAR 4:3], [19:49] <Hfuy> I suspect S stands for Sample. Which may be the same thing as Pixel. But I'm not sure. [19:49] <Hfuy> But DAR seems to be what you need anyway. [19:50] <TSM> so the fact i can see it says 4:3 means that it should be shown as 16:9? [19:50] <Dresk|Laptop> Hfuy: Take a look at this apparently very straight-forward (but order dependent) command line for multiple stream mapping : ffmpeg -i a.avi -i b.avi -i c.avi -vcodec copy -acodec copy / [19:50] <Dresk|Laptop> -vcodec copy -acodec copy out.avi -map 2.0 -map 0.1 / [19:50] <Dresk|Laptop> -map 1.1 -map 2.1 -acodec copy -newaudio -acodec copy -newaudio [19:50] <Hfuy> TSM: Is it a 16:9 video? [19:51] <Dresk|Laptop> 3 input files, the video codec chosen is just to copy, it's only called twice, not sur ehow the third input file comes into play [19:51] <Hfuy> Dresk|Laptop: Sorry, that's advanced to me. [19:51] <TSM> Hfuy: yes it is, well its a dvvideo format [19:51] <TSM> direct from cam [19:51] <thisislawatts> Hm ok so I've made some more progress on my multiply overlay dilemma, but now getting what looks like just green channels. So I am using -filter_complex "blend=all_mode='multiply':all_opacity=1" full copy/paste --> http://pastie.org/8419218 [19:51] <Dresk|Laptop> Hfuy: I will solve this for us! Onward! [19:51] <Hfuy> I simply have a situation where it's mapping 0:1 to 0:0 and 0:0 to 0:1 [19:52] <Hfuy> It's also dropping 0:2 which is metadata I'd like to maintain. [19:52] <Hfuy> I can't find any uptodate information on what the behaviour is supposed to be, why it's that way, or how to control it. [19:56] <Hfuy> TSM: Sometimes I've known DV in AVI files where the DAR in the AVI doesn't get set properly. [19:56] <Hfuy> That way you can end up with the thing assuming PAR=1:1 [19:57] <Hfuy> or any damn thing inbetween [19:58] <TSM> grrr, how anoying [19:58] <Hfuy> so you're saying the DAR given by ffmpeg is 4:3 but it should be 16:9? [19:59] <TSM> essentialy yes, ive done a straight 'ffmpeg -i <in> -aspect 16:9 <out>' and the result is fine and plays without adjustment [19:59] <Hfuy> That'll be getting reencoded though I assume? [19:59] <TSM> but otherwise when i play it in VLC without any changes it is squished until i change the aspect ratio [19:59] <Hfuy> VLC didn't used to be known for obeying things anyway. [19:59] <TSM> let me do it with -vcodec copy [20:00] <Hfuy> I'm not sure if you can change DAR with -c:v copy [20:00] <Hfuy> But again, it's undocumented. [20:00] <Hfuy> Can anyone explain to me how I should tell ffmpeg to not reorder the input streams in the output? [20:01] <Hfuy> Or to put it another way, put input 0:0 into output 0:0, input 0:1 into output 0:1, and ideally maintain input 0:2 into output 0:2 unchanged (0:2 is a timecode track) [20:02] <TSM> does not allow me to do vcodec copy, hmmm [20:02] <TSM> ile be reencoding the videos anyway so not much of a problem [20:02] <Hfuy> what is it, quicktime, avi? [20:04] <TSM> dvvideo [20:08] <Hfuy> oh, shit. right. [20:08] <Hfuy> raw DV eh. [20:08] <Hfuy> Hmm. [20:08] <Hfuy> And this is captured from a camera, not encoded with software? [20:09] <Hfuy> If it's from a camera the information will almost certainly be in there. [20:10] <Hfuy> But I wouldn't be surprised if it didn't appear correctly in ffmpeg, which is probably more commonly used with containers such as quicktime. [20:10] <Hfuy> there are DV tools around which allow you to read DIF information from raw dv video files, which should reveal what's going on. [20:11] <Hfuy> http://www.avpreserve.com/avpsresources/tools/ look for DIF analyzer [20:13] <Hfuy> Anyone know what this means: "Data stream encoding not supported yet (only streamcopy)" [20:13] <Hfuy> It appears when I insert -map 0:2 into a command dealing with an input file that has quicktime timecode as stream 0:2. [20:13] <Hfuy> I'd be happy with a copy of the stream but I can't find out how to do that. [20:14] <TSM> oh the complexity my head is burning [20:15] <Hfuy> Complexity of what? [20:15] <TSM> working out video formats etc [20:15] <Hfuy> Heh. It can be a bit complicated. [20:15] <TSM> im just compiling up that DV analizer [20:16] <Hfuy> It doesn't help that the documentation for ffmpeg is... well, practically nonexistent. [20:16] <Dresk|Laptop> Hfuy: Well, it's definitely not a win, but I solved the issue by extracting the video stream and just using 2 input files, since they were raw streams, so no mapping was needed [20:16] <TSM> but on the flip side, who the hell still shoots real 4:3 DV video anyway, not in my industry they should not [20:16] <Hfuy> Dresk|Laptop: the way it seems to work is that you put several -map x:y args in the command. [20:16] <Hfuy> So, if I state: -map 0:0 -map 0:1 I get the same order as I had in the input. [20:17] <Hfuy> I'm not sure if you can simply state more than one -i and then use -map 0:1 -map 1:1 in the command to get the outputs you want. [20:17] <Hfuy> I'm guessing here. I don't know. I haven't tried it. [20:17] <Hfuy> TSM: I shoot quite a bit for news, believe it or not. [20:18] <Hfuy> Big companies may have 4K tvs to sell, but most of my work is still DVCAM. [20:18] <TSM> true i mean how much is in real 4:3, ie not anemorphic [20:19] <Hfuy> Not much here in London. [20:19] <Hfuy> In the US, where widescreen standard def was never so much of a thing, quite a bit I suspect.# [20:19] <TSM> not since we are now all 16:9 on freeview etc [20:19] <Dresk|Laptop> Does anyone know what parameters affect pass 1 on video streams? For example, I am making 4 video files from 1 source, and they all vary in resolution and bitrate - do I need to do a pass 1 for EVERY one because my chosen resolution and bitrate, or is the pass a general thing that can be used across all formats? [20:19] <TSM> yup ive seen quite a bit on US cheepo channels [20:20] <Hfuy> TSM I think they just expect you to go to HD if you want wide [20:20] <Hfuy> Does anyone know how you set prores profiles? [20:20] <x__> Hey does anyone know how I can compress my .mov files from 1.9 to about 700-800 mb? [20:20] <TSM> I have to consider all countries, work in UK but lots of the video content comes from US videographers [20:20] <Hfuy> TSM: are these professional news guys or what [20:21] <x__> I have one command that I used to get 1.9 to 1.2 but not show how to condense it a bit more: avconv -i 12\ Dates\ of\ Christmas\ \[2011\].mov -acodec libfaac -b:a 128k -vcodec mpeg4 -b:v 1200k -flags +aic+mv4 final.mp4 [20:21] <TSM> no way, just papz that get hold of a cam and hound celebs :p [20:21] <Hfuy> Oh puh-leease. [20:21] <Hfuy> x__: you might try reducing the 1200k. [20:22] <x__> Hfuy: To what? not sure what that actually means? [20:22] <x__> Hfuy: Should I put it to 800k ? [20:23] <Hfuy> I'm not sure, either, but I suspect "-b:v 1200k" may mean "bitrate for video should be 1200 kilobits per second." [20:23] <Hfuy> Try it. See what you get. [20:23] <TSM> b:v = video, b:a=audio [20:23] <Hfuy> Controlling the bitrate of h.264 is a bit approximate anyway. [20:24] <TSM> anyone tried the opencl stuff of ffmpeg 2 yet? [20:25] <x__> Hfuy: I am checking it out now [20:26] <Hfuy> How do I get it to drop a stream entirely? [20:28] <x__> Hfuy: I checked in libav's documentation and it said to set the video bitrate of a file to 64kbits/s you use: avconv -i input.avi -b 64k output.avi [20:29] <Hfuy> well there yo go [20:31] <x__> Hfuy: I will let you know how it goes. Thanks for the help. [20:31] <Hfuy> Good grief, I'm actually dispensing advice on #ffmpeg [20:31] Action: Hfuy ducks the flying pigs [20:32] <Hfuy> Does anyone know how to have ffmpeg drop an output stream entirely? [20:39] <Hfuy> Hn. Alternatively, is there some way I can get it to translate the video content from yuv422p10le to yuv422p10 so I can use an alternate codec? [22:23] <thisislawatts> Hey there, I am struggling with the -filter_complex "blend=all_mode='multiply'" outputting only green videos, would anyone like to point out the obvious error I am making? [22:27] <durandal11707> using yuv pixel format [22:30] <thisislawatts> durandal11707: what format would you recommend? [22:30] <durandal11707> gbrp [22:31] <durandal11707> format=gbrp for each input,blend... [22:33] <durandal11707> just: format=gbrp,blend=all_mode=multiply [22:38] <thisislawatts> Hm, now it's outputting just black. Are there any other formats you'd recommend? [22:38] <durandal11707> its black when one of images are mainly black [22:39] <durandal11707> it does A * B / 255 for each pixel in each image [22:40] <durandal11707> you can create own ones with expressions... there are some examples in documentation [22:40] <thisislawatts> ah, so that's how the expressions work, that's great I was a little perplexed by the documentation. [22:41] <durandal11707> you can use modes, but they are just limited - no fancy stuff like animations and dynamic opacity [22:42] <durandal11707> it all depends on what you really want to archieve [22:43] <thisislawatts> in truth what I am trying to achieve is a very basic, just a colour burn -> http://www.thefirefliestour.com/wp-content/themes/fireflies/img/riders-arch… [22:43] <thisislawatts> Well at least I am assuming it's very basic :) [22:44] <durandal11707> well there is burn mode [22:47] <thisislawatts> Yeah I have tried with that also and it's still outputting solid black [22:49] <durandal11707> what images you use? [22:52] <thisislawatts> These are them -> http://thisis.la/blendmode.png [22:54] <durandal11707> you mean you use same images for both inputs ? [22:55] <thisislawatts> At the moment yeah, just using that red box to test things out? [22:57] <durandal11707> i don't get it what is exact command you use? [22:57] <thisislawatts> ffmpeg -i sample.mov -i bw.mov -filter_complex "format=gbrp,blend=all_mode=burn" test$(date +"%s")\.mov [23:01] <durandal11707> and where is sample.mov and bw.mov? [23:01] <thisislawatts> in the same directory as where I am executing the command [23:03] <durandal11707> swap inputs [23:08] <burek> can "subtitles" filter ( http://www.ffmpeg.org/ffmpeg-all.html#subtitles ) take an input from the input stream (subtitle stream) or it strictly takes its input from files only? [23:08] <durandal11707> iirc from files only [23:09] <burek> ok, thanks :) [23:09] <brontosaurusrex> how would i use the segment thingy when encoding to prores (I'd just like multiple files as output say 1 minute long) ? [23:09] <brontosaurusrex> wanted container is mov, codec prores [23:10] <thisislawatts> I have tried swapping the inputs to no avail, I think I will have to take another look in the morning, I would prefer to generate this coloured overlay from within ffmpeg [23:11] <ubitux> burek: you can pass it a mkv [23:11] <ubitux> burek: it will pick the first subtitles stream iirc [23:11] <ubitux> but if it's ass you won't have fonts etc [23:12] <durandal11707> thisislawatts: pehaps you need linear burn and not color burn [23:12] <durandal11707> the burn mode is actually color burn [23:12] <burek> ubitux, the input is already mkv: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=1119 [23:12] <burek> the guy just wants [23:12] <burek> to do it all in one step [23:13] <ubitux> redirect him to the ticket requesting that feature [23:13] <burek> ok [23:13] <ubitux> i can find it for you [23:13] <burek> thank you :) [23:14] <ubitux> http://trac.ffmpeg.org/ticket/1305 [23:14] <burek> +1 [23:24] <brontosaurusrex> this is what i have so far, nothing; http://paste.debian.net/plain/60251 [23:27] <durandal11707> thisislawatts: linear burn is: blend=all_expr=A+B-255 [23:48] <thisislawatts> durandal11707: Thank you kindly, I will let you know how I get on :D [00:00] --- Tue Oct 22 2013

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