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November 2013
- 1 participants
- 60 discussions
[00:20] <cone-424> ffmpeg.git 03Anton Khirnov 07master:d42db44cfe58: avienc: drop the vfr flag.
[00:20] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:9c52ab6f4f32: ffmpeg: treat avi as VFR in framerate conversion code
[00:20] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:c79c7ffc9da7: Merge remote-tracking branch 'qatar/master'
[01:27] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:891f33cc7052: avformat/h261dec: optimize probe
[01:27] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:557571ff7db5: avformat/h261dec: dont accept invalid gobs in probe as valid
[02:02] <ubitux> wm4: any news on the dvd subtitles issue you had?
[02:02] <ubitux> duration or sth
[02:02] <wm4> ubitux: just that the API fails to provide a way to indicate that the duration is unset
[02:03] <ubitux> ok
[02:03] <wm4> it uses 0 for that, so you can't distinguish to subtitles with duration 0 (which can exists too, but maybe only with text subs not image subs)
[02:03] <ubitux> should we add some flag of some sort? or maybe negative duration?
[02:03] <ubitux> (negative duration might break some stuff in users apps)
[02:03] <wm4> a negative value would be best I guess, bit it would be an API change
[02:04] <wm4> also there's a strange additional hunk of code in mplayer's dvd sub parser, but I don't know about this
[02:08] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:64f48169159b: avformat/mp3dec: 4x faster probing
[04:29] <BBB> durandal_1707: if you're interested, I can help you get that going
[04:30] <BBB> durandal_1707: it's slightly more involved, but still, not overly difficult
[04:30] <BBB> durandal_1707: I tried to structure it in such a way that that becomes really easy to add later on
[05:51] <Compn> http://lists.alioth.debian.org/pipermail/pkg-multimedia-maintainers/2012-Ma…
[05:51] <Compn> lol
[11:03] <ubitux> can anyone css-add an horizontal scrollbar on fate when unfolding failing tests?
[11:04] <ubitux> (see for instance what happens when unfolding dragonegg-3.4 instance)
[11:14] <michaelni> ubitux, i can apply a patch, i dont know what to add where ...
[11:15] <ubitux> ok let me have a look
[11:19] <ubitux> meh fuck css, overflow:auto doesn't want to work
[11:23] <ubitux> oh well, i'll see later
[12:14] <cone-654> ffmpeg.git 03James Zern 07master:eb7f7b797f0a: riff: add VP9 fourcc
[12:39] <cone-654> ffmpeg.git 03Clément BSsch 07master:dbb41f93c16c: build: avoid stdin stall with GNU AS probing.
[15:35] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:76a47d66c414: avformat/dv: about 3 times faster probe
[16:22] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:9825b99eca0f: avformat/mpegts: do not hardcode type used in memset()
[16:22] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:e624df5dc33f: mpegts: only calculate stat index when its needed during probe
[16:28] <wm4> what's the deal with optimizing probing?
[16:32] <Compn> wm4 : takes a long itme
[17:44] <kierank> need to get rid of these spammers
[18:11] <ods15> michaelni, odd, I never unsubscribed from nut-devel, I did not get those mails
[18:12] <ods15> oops, I know whats wrong, my email address is dead, I am no longer ods15(a)ods15.dyndns.org
[18:13] <michaelni> kierank, trac supports several spam filter services, like akismet, typepad, defensio, http:bl. But all these need API keys AFAIK. If someone provides such keys i can enter them in trac, that may (or may not) improve spam filtering
[18:13] <ods15> I can't really say that I am still a maintainer for libnut, but I don't mind approoving these trivial patches, if you don't mind replying in my name in nut-devel
[18:13] <michaelni> ods15, sure, ok, will do
[18:14] <ods15> just - please add parenthesis over entire macro in ftello(a) - should be: #define ftello(a) ((off_t)(_ftelli64(a)))
[18:14] <ods15> patches OK otherwise
[18:28] <ubitux> ._.
[18:33] <michaelni> ods15, patches applied
[18:51] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:ef8f29347285: avformat/dv: consider a header match at byte 0 to count as detection of dv in probe
[18:51] <saste> ubitux: do you know if it is possible to specify the metadata text encoding in a container?
[18:51] <saste> or do we alwasy assume UTF-8?
[18:52] <saste> why there are so many files around with the wrong encoding?
[18:53] <ubitux> demuxers should be able to access sub_charenc option
[18:53] <ubitux> (to be confirmed)
[18:53] <ubitux> we assume utf-8 but user can set it with -sub_charenc
[18:53] <saste> sub_charenc only regards subtitles, right?
[18:53] <ubitux> ah sorry, metadata
[18:53] <ubitux> yes i think we assume utf-8
[18:54] <saste> should we extend it to metadata as well?
[18:54] <ubitux> feel free to do it
[18:54] <ubitux> since we added the iconv thing, it should be doable
[18:54] <saste> also, should we validate string option values?
[18:54] <ubitux> i'm not much into philosophy sorry
[18:55] <saste> right now we can still inject non valid UTF-8 through -metadata and other options
[18:56] <saste> i'm asking because i'm validating the value of -validate_string_policy, but doing it just for an option seems stupid
[18:57] <saste> philosophy? no it's more about opinions
[18:57] <ubitux> it was a joke, i don't have any opinion here :)
[18:57] <ubitux> not enough insight on the subject
[18:58] <saste> i think you thought hard about such stuff, since you're our subtitles guru
[18:58] <ubitux> yes but for charset i'm honestly not the best bet
[18:59] <ubitux> you should probably troll about that with wm4 and nicolas
[18:59] <wm4> <ubitux> demuxers should be able to access sub_charenc option
[18:59] <wm4> <ubitux> (to be confirmed)
[18:59] <wm4> <ubitux> we assume utf-8 but user can set it with -sub_charenc
[18:59] <wm4> I'd fucking hate this
[18:59] <ubitux> wm4: some formats include specify charset
[18:59] <ubitux> at least microdvd does
[19:00] <ubitux> (we don't support this though)
[19:00] <wm4> yes, you want to be able to know what charset it is if it's guaranteed
[19:00] <ubitux> SAMI might as well support something like that
[19:00] <wm4> e.g. webvtt subs are _always_ utf-8, no matter what
[19:00] <wm4> while srt can be anything
[19:00] <ubitux> so what would you hate?
[19:00] <wm4> and it would be nice to know that (without having to consult a whitelist)
[19:00] <wm4> demuxers recoding charset
[19:00] <wm4> like nicolas plans it
[19:01] <ubitux> i'm sure you'll find webvtt not utf-8
[19:01] <wm4> the day it is committed, I'll NIH _all_ subtitle parsers
[19:02] <wm4> ubitux: at least that's a case where you should try not to misdetect subtitles as some other charset
[19:03] <ubitux> i honestly didn't really follow the subtitles charset troll
[19:05] <wm4> it's not a troll, it's a flame
[19:05] <wm4> a very angry one
[19:05] <ubitux> grrr
[19:07] <ubitux> saste: opinion on the fade color patch?
[19:07] <saste> ubitux, i'll review it in a few minutes
[19:08] <cone-654> ffmpeg.git 03Stefano Sabatini 07master:0f8bdfe3e807: ffmpeg: store value of -filter and -filter_script per-stream option in OutputStream struct
[19:08] <cone-654> ffmpeg.git 03Stefano Sabatini 07master:f591ba60ff4b: ffmpeg: generate more specific error message in case of inconsistent filtering options
[20:08] <saste> nobody will comment on my utf8 patch?
[20:53] <cone-654> ffmpeg.git 03Erik Olofsson 07master:f5bffd3a1a1a: libavf/avienc: store language metadata for XSUB streams
[21:00] <BBB> ubitux: can you review+commit that vp9 patch I put up?
[21:08] <ubitux> BBB: don't want to answer reimar question(s)?
[21:09] <ubitux> i don't mind committing but i can't really review blindly this
[21:25] <BBB> ubitux: it's a question for google, I can't answer it, right?
[21:26] <ubitux> maybe Reimar isn't aware you're not working at google anymore
[21:26] <ubitux> ...or that you have no contact with libvpx (anymore?)
[21:26] <BBB> hm, I see
[21:28] <BBB> well that should clear things up then for him
[21:35] <ubitux> BBB: applied
[21:35] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:dd3d0f46bdb0: vp9: fix mvref finding to adhere to bug in libvpx.
[00:00] --- Sun Nov 10 2013
1
0
[01:00] <barneygale> Hi. I'm trying to record my screen while playing a game, but the lack of power on this machine (i assume) is causing the resulting video to run at about 0.5fps. Is there anything that can be changed in the command I'm running to reduce CPU usage? I don't mind losing quality. Command: ffmpeg -f x11grab -s 1920x1080 -r 15 -i :0.0 ~/recordings/`date --iso-8601=s`.mkv
[01:43] <relaxed> barneygale: Play the game in windowed mode with a lower resolution.
[01:44] <relaxed> add -framerate 15 before ht input
[01:44] <relaxed> the, not ht
[01:45] <relaxed> or scale the framesize down to 720p
[01:46] <relaxed> -filter:v scale=w=1280:h=720
[02:08] <barneygale> Thanks relaxed
[19:42] <Razkin> Hey guys, I was converting a video from flv to pal-vob yesterday, but I stopped the process at 1:08:02, is there any way I could start converting again right from where I stopped? from 1:08:02?
[19:42] <Razkin> I used avconv.
[20:20] <l_r> hello
[20:20] <l_r> is avconv a new name for ffmpeg?
[20:23] <l_r> i so what's in debian
[20:24] <saste> libav/avconv
[20:24] <l_r> do the command have the same interfaces?
[20:24] <saste> more or less
[20:24] <l_r> what a pity
[20:24] <l_r> damn forkings
[20:26] <l_r> do you now how to preserve the image format in the output video? avconv -f video4linux2 -r 25 -input_format mjpeg -s 640x480 -i /dev/video0 video.mpeg
[20:26] <l_r> it looks like avconv converts mjpeg to yuv420p
[20:27] <l_r> but i d like to have mjpeg in the video ,without conversions
[20:51] <ubitux> l_r: ask #libav, or use ffmpeg
[20:51] <ubitux> we can't support the fork here
[21:06] <l_r> ok
[21:07] <l_r> i have ffmpeg installed
[21:07] <l_r> what i want is to stream the cam to a feed which has "format mjpeg" defined, now the cam supports mpeg too and i don't want any conversion
[21:08] <l_r> ./ffmpeg/bin/ffmpeg -f v4l2 -s 640x480 -r 25 -input_format mjpeg -i /dev/video0 http://localhost:8080/webcam.ffm
[21:08] <l_r> it looks is does convert to yuv420p
[21:52] <grim21> quick query.. is there a way I can transcode a video and supply an option to keep the same bitrates? I'm trying to move from mp4 to webm. I don't want to have to manually ffprobe each file to determine their bitrates.
[22:04] <jonno11> Hi guys. Trying to stream a file to an RTMP wowza server. It's streaming fine, but I'd like to loop the file. It dies with Failed to update header with correct duration.ate=3579.1kbits/s"
[22:04] <jonno11> I'm running it with "ffmpeg -re -i content/sample.mp4 -loop 1 -c copy -f flv rtmp://myserver.co.uk/live/test"
[23:11] <dgbaley27> Hey. I have a list of chapter start times (HH:MM:SS.sss), and no idea how I can add chapters to my video file.
[00:00] --- Sun Nov 10 2013
1
0
[00:00] <burek> llogan, it detects rss document change
[00:00] <burek> probably something was changed at that moment, although it shows only last 5 items
[00:01] <burek> so it appears as if it is repeating the same thing
[00:01] <burek> i'll investigate in logs
[00:01] <burek> and improve the behavior if possible
[00:02] <llogan> burek: thanks. now i owe you the beer.
[00:02] <burek> +1 :)
[00:02] <cone-764> ffmpeg.git 03Diego Biurrun 07master:86f910806b2c: Remove #undefs for formerly forbidden system functions
[00:02] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:175e689cc5de: avfilter/graphparser: zero filter_ctx in case of deallocation in create_filter()
[00:02] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:48f37664a8b5: Merge remote-tracking branch 'qatar/master'
[00:44] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:b7fea3daabc7: libavcodec/dvdsubdec: remove exit() call in debug code
[05:57] <anshul_> i was trying to run fate and got an following error
[05:57] <anshul_> tests/fate.sh doc/fate_config.sh
[05:58] <anshul_> fatal: premature end of pack file, 300 bytes missing
[05:58] <anshul_> fatal: index-pack failed
[05:58] <anshul_> Error fetching source
[06:23] <anshul_> I deleted the fate_work directory and tried to run again, and got again an error
[06:24] <anshul_> [anshul@daku_daddy ffmpeg]$ tests/fate.sh doc/fate_config.sh
[06:24] <anshul_> fatal: pack has 9 unresolved deltas
[06:24] <anshul_> fatal: index-pack failed
[06:24] <anshul_> Error fetching source
[06:24] <anshul_> rm: cannot remove fate_work/fate.lock: No such file or directory
[06:25] <anshul_> done git pull for fatal: pack has 9 unresolved deltas
[06:33] <anshul_> I am unable to run fate, can anybody help me
[06:33] <anshul_> [anshul@daku_daddy ffmpeg]$ tests/fate.sh doc/fate_config.sh
[06:33] <anshul_> tests/fate.sh: line 107: fate_work/src/version.sh: No such file or directory
[06:33] <anshul_> rm: cannot remove fate_work/fate.lock: No such file or directory
[06:34] <anshul_> version.sh is present at ./fate_work/fate_work/src/version.sh
[06:36] <anshul_> I think relative paths confuse the fate, trying with absolute path
[07:17] <anshul_> Still unable to run fate
[07:19] <anshul_> cat: /home/anshul/Software/ffmpeg/fate_work/build/tests/data/fate/*.rep: No such file or directory
[07:22] <anshul_> actually /tests/data/fate is at /home/anshul/Software/ffmpeg/
[07:22] <anshul_> no idea what to try next
[07:22] <anshul_> can any one share there fate_config file to cross verify
[07:24] <anshul_> if I remove all of the work directory , then it try to makedir with empty name
[07:27] <anshul_> I dont think cpoying tests/data/fate/ in fate_work is an good idea
[07:28] <ubitux> did you look at https://ffmpeg.org/fate.html ?
[07:28] <ubitux> did you check the config.log of your fate run?
[07:36] <anshul_> I was following the same link
[07:37] <anshul_> I dont know what to check in config.log
[07:39] <ubitux> if it fails or no
[07:40] <ubitux> http://lucy.pkh.me/fate-configs/
[07:40] <ubitux> my configs are here
[07:40] <ubitux> http://lucy.pkh.me/run.sh
[07:40] <ubitux> and this is my fate run script
[07:41] <ubitux> (which is located in ~/fate/run.sh, configs in ~/fate/configs, source in ~/fate/ffmpeg)
[08:08] <anshul_> hey ubitex in you configuration you have kept repo as file:/home/ux/fate/ffmpeg but in mine cfg i have kept git://source.ffmpeg.org/ffmpeg.git; does it make sense to make it git
[08:08] <anshul_> or i should also change the path to local git repository at my pc
[08:12] <anshul_> yeh command worked tests/fate.sh doc/fate_config.sh with echo$? = 0
[08:13] <anshul_> but when i check the https://ffmpeg.org/fate.html how to confirm that this fate is run y me
[08:14] <ubitux> it's the local path where the sources are pulled
[08:15] <ubitux> the script pulls the sources at regular interval in that directory
[08:16] <ubitux> or well, before each instance run actually
[08:17] <ubitux> on fate.ffmpeg.org, to check if it's yours, you click on the tests, you'll get all kind of info
[09:00] <anshul_> 1 slot=x86-openSuse-gcc-4.7-dakudaddy # some unique identifier
[09:00] <anshul_> 2 repo=/home/anshul/Software/ffmpeg/ # the source repository
[09:00] <anshul_> 3 samples=fate-suite/ # path to samples directory
[09:00] <anshul_> 4 workdir=/home/anshul/Software/ffmpeg/fate_work/ # directory in which to do all the work
[09:00] <anshul_> 5 #fate_recv="ssh -T fate(a)fate.ffmpeg.org" # command to submit report
[09:00] <anshul_> 6 comment=First_time_fate # optional description
[09:00] <anshul_> 7 build_only= # set to "yes" for a compile-only instance that skips tests
[09:00] <anshul_> 8
[09:00] <anshul_> 9 # the following are optional and map to configure options
[09:00] <anshul_> 10 arch=
[09:00] <anshul_> 11 cpu=
[09:00] <anshul_> 12 cross_prefix=
[09:00] <anshul_> 13 as=
[09:00] <anshul_> 14 cc=
[09:00] <anshul_> 15 ld=
[09:00] <anshul_> 16 target_os=
[09:00] <anshul_> 17 sysroot=
[09:00] <anshul_> 18 target_exec=
[09:00] <anshul_> 19 target_path=
[09:00] <anshul_> 20 target_samples=
[09:00] <anshul_> 21 extra_cflags=
[09:00] <anshul_> 22 extra_ldflags=
[09:00] <anshul_> 23 extra_libs=
[09:00] <anshul_> 24 extra_conf= # extra configure options not covered above
[09:00] <anshul_> 25
[09:00] <anshul_> 26 #make= # name of GNU make if not 'make'
[09:00] <anshul_> 27 makeopts= # extra options passed to 'make'
[09:00] <anshul_> 28 #tar= # command to create a tar archive from its arguments on stdout,
[09:00] <anshul_> 29 # defaults to 'tar c'
[09:00] <anshul_> This my config file
[09:00] <anshul_> but i dont get anything on http://fate.ffmpeg.org/?dsort=npass
[09:13] <ubitux> anshul_: please use pastebin/pastie/... when it goes up to 3 lines&
[09:14] <anshul_> ok
[09:17] <anshul_> in configure.log it shows that
[09:17] <anshul_> Out of tree builds are impossible with config.h in source dir.
[09:23] <ubitux> cleanup your source dir
[09:25] <anshul_> you mean make distclean
[09:26] <michaelni> " fatal: pack has 9 unresolved deltas" / "fatal: index-pack failed" <-- sounds like some problem with git
[09:26] <anshul_> I am trying to run your scripts now, with making enviorment similar to your
[09:28] <anshul_> @michaelni i pulled the git again, and that error was not seen again
[09:42] <ubitux> anshul_: yes, or git clean -fdx
[10:02] <plepere> hello. coding assembly for HEVC decoding (or at least documenting myself to). Should I do general x86 functions and add options for x64 support or can I just go with x64 from the start ?
[10:03] <ubitux> plepere: as i said, you can go for x64 from the start, and later add 32
[10:03] <plepere> ok thanks. I had to go last time so I didn't see the answer when I asked last time. :)
[10:04] <nevcairiel> It would be great to have 32-bit eventually
[10:04] <nevcairiel> its still in wide use
[10:04] <plepere> of course
[10:05] <plepere> same for SSE instruction use ? start with SSE4 first and support old MMX only processors after ?
[10:05] <iive> sse2 at least
[10:05] <nevcairiel> personally i wouldnt bother much with mmx
[10:05] <plepere> ok
[10:06] <nevcairiel> If you want to build SSSE3/SSE4 version first, and add a SSE2 compat later, thats also ok, if you even want to deal with mmx compat ... your choice :)
[10:07] <plepere> I think I'd rather try my hand on ARM optimization if I get the choice. :) so much to learn
[10:08] <plepere> it's what I love about this job : I'm always learning. Forever a student !
[11:15] <anshul_> 62622 Test filter-metadata-ebur128 failed. Look at tests/data/fate/filter-metadata-ebur128.err for details.
[11:15] <anshul_> 62623 make: *** [fate-filter-metadata-ebur128] Error 1
[11:15] <anshul_> 62624 make: Target `fate' not remade because of errors.
[11:15] <anshul_> Still unable to run fate
[11:15] <anshul_> these are last three line in test.log
[11:18] <ubitux> "Look at tests/data/fate/filter-metadata-ebur128.err for details."
[11:19] <anshul_> there is no file named filter-metadata-ebur128.err
[11:21] <anshul_> I there is no file named *.err even
[11:22] <anshul_> Is there any clue i can get from the report generated in work folder
[11:24] <anshul_> from first line of report fate:0:20131108100824:x86-openSuse-gcc-4.7-dakudaddy:N-57965-gb7fea3d:2:error testing:First_time_fate
[11:24] <nevcairiel> simply run that failing test manually to find out more
[11:24] <nevcairiel> make fate-filter-metadata-ebur128
[11:55] <anshul_> [anshul@daku_daddy ffmpeg]$ make fate-filter-metadata-ebur128
[11:55] <anshul_> Makefile:2: config.mak: No such file or directory
[11:55] <anshul_> Makefile:47: /common.mak: No such file or directory
[11:55] <anshul_> Makefile:90: /libavutil/Makefile: No such file or directory
[11:55] <anshul_> Makefile:90: /library.mak: No such file or directory
[11:55] <anshul_> Makefile:167: /doc/Makefile: No such file or directory
[11:55] <anshul_> Makefile:168: /tests/Makefile: No such file or directory
[11:55] <anshul_> make: *** No rule to make target `/tests/Makefile'. Stop.
[11:55] <anshul_> I dont why its kooking in /doc tarher it should look fordoc
[11:55] <ubitux> this is because the configure wasn't run from this directory
[12:11] <sspiff> can I add CCFLAGS to ffmpeg easily?
[12:11] <sspiff> I'm building for a platform that needs a bit of tweaking :)
[12:11] <sspiff> I can't find anything in ./configure --help, so I'm currently using CCFLAGS=foo make
[12:12] <anshul_> @ubitux from where should i run the make
[12:12] <JEEB> sspiff, --extra-cflags="-I/ikaruga/kou/include" as an example
[12:13] <anshul_> I am still not getting, from where fate.sh had run the configure.
[12:19] <anshul_> @nevcairiel i tried by configuring in ffmpeg directory and tried the same command but now its say make: *** No rule to make target `fate-filter-metadata-ebur128'. Stop.
[12:21] <cone-424> ffmpeg.git 03Hendrik Leppkes 07master:019247bdc326: avformat/mov: only force parsing for video tracks if stss is empty
[12:21] <ubitux> you need to run the tests from where you ran the configure and make.
[12:21] <iive> anshul_: debug with `bash -x fate.sh` , it would output every command as it is executed.
[12:29] <anshul_> @ubitux i didn't run any configure or make, they are done by fate.sh;
[12:30] <anshul_> @live that was good hack for scripts :)
[12:55] <BBB> durandal_1707: yeah, you want to do vp9-mt?
[13:03] <durandal_1707> BBB: it looks very complicated
[13:14] <BBB> it isn't
[13:14] <BBB> you've done mt before
[13:14] <BBB> it's exactly the same, with some cross-frame progress tracking, which is in whatever unit you like
[13:14] <BBB> it's just a few lines more
[13:16] <BBB> I can hep if you're interested, that is, you'll code and I can provide pointers of stuff to watch out for etc.
[13:16] <BBB> basically it involves one piece of code refactoring, and then the regular framt-mt code insertions of init_done, report_progress, wait_progress
[13:17] <anshul_> I cleaned up every thing and end up with command make filter-metadata-ebur128 -f ../ffmpeg/Makefile
[13:18] <ubitux> anshul_: what the hell are you trying to do?
[13:19] <ubitux> anshul_: create a build directory, run the configure and make from that directory
[13:19] <ubitux> then run the make fate-filter-metadata-ebur128
[13:19] <ubitux> (and make sure the SAMPLES variable is set)
[13:19] <anshul_> ok i will try again;
[13:20] <durandal_1707> BBB: refactoring?
[13:40] <BBB> durandal_1707: so vp9 currently decodes the bitstream in-line with the reconstruction
[13:40] <BBB> durandal_1707: i.e. do { decode block in bitstream - reconstruct it } while (!done)
[13:40] <durandal_1707> so basically i need to rewrite decoder?
[13:40] <BBB> no
[13:41] <BBB> they are separate functions
[13:41] <BBB> they're just called after each other
[13:41] <BBB> what you need to do is to extract that so it does 2 cycles
[13:41] <BBB> do { decode block - store it in a temp mem location } while (!done) thread_init_done; do { reconstruct } while (more blocks in temp mem location)
[13:42] <BBB> I can actually do that if you feel that's troublesome
[13:52] <anshul_> failed again with fate; tried below command
[13:52] <anshul_> ../ffmpeg/configure --samples=../fate-suite/
[13:52] <anshul_> make SAMPLES=../fate-suite/
[13:52] <anshul_> make filter-metadata-ebur128 SAMPLES=../fate-suite/
[13:52] <anshul_> make: *** No rule to make target `filter-metadata-ebur128'. Stop
[13:53] <nevcairiel> make fate-filter-metadata-ebur128
[13:53] <nevcairiel> missing a fate-
[13:54] <anshul_> same result :(
[13:54] <anshul_> [anshul@daku_daddy fate]$ make fate-filter-metadata-ebur128 SAMPLES=../fate-suite/
[13:54] <anshul_> make: *** No rule to make target `fate-filter-metadata-ebur128'. Stop.
[13:55] <nevcairiel> just give up on the out of tree build then :p
[13:55] <nevcairiel> you cna get a list of fate targets with make fate-list, i think
[13:57] <nevcairiel> oh, and ebur128 is gpl, so you need to configure with --enable-gpl, or it wont be available
[13:59] <anshul_> I am running fate second time; few months before also i gave up with some error, this time i am determined to make this work.
[14:00] <ubitux> anshul_: if that tests fail it's not really a problem
[14:00] <anshul_> at least it is compiling.
[14:00] <ubitux> it will be reported on fate
[14:00] <ubitux> and we will be able to see the problem
[14:00] <ubitux> what configuration are you setuping?
[14:01] <ubitux> if it's an msvc instance, this test is known to be broken because of a msvc bug
[14:01] <nevcairiel> which is fixed in msvc 2013, hooray
[14:02] <anshul_> i have not posted the ssh public key yet, so it would not be reported
[14:03] <anshul_> my configuration were just plane ../ffmpeg/configure --samples=../fate-suite/ --enable-gpl
[14:03] <ubitux> i mean what plateform?
[14:04] <anshul_> version-x86-openSuse-gcc-4.7
[14:10] <ubitux> special cpu?
[14:11] <anshul_> no ts normal intel corei3
[14:12] <anshul_> In my company 100 of employees use the same cpu
[14:14] <anshul_> @nevcairie --enable-gpl fixed the issue; i am trying to run fate again with passsing extra_conf=--enable-gpl and waiting for error report
[14:16] <anshul_> I think fate shoul not test those test files whose codecs are not enable in configuration. can this issue reported as bug
[14:21] <ubitux> anshul_: fate has --enable-gpl by default
[14:22] <anshul_> make was successfull at make filter-metadata-ebur128 SAMPLES=../fate-suite/
[14:22] <anshul_> but why it was failing in fate
[14:22] <anshul_> but why it was failing in fate report
[14:30] <anshul_> @ubitux i dont think fate has --enable-gpl by default
[14:31] <ubitux> then look at the file tests/fate.sh
[14:31] <ubitux> configure() function
[14:32] <anshul_> yaaa 8)
[14:49] <durandal_1707> BBB: what about tile threading?
[15:00] <anshul_> 1 ffprobe version N-57966-g019247b Copyright (c) 2007-2013 the FFmpeg developers
[15:00] <anshul_> 2 built on Nov 8 2013 19:17:56 with gcc 4.7 (SUSE Linux)
[15:00] <anshul_> 3 configuration: --prefix=/home/anshul/Software/fate/fate_work/install --samples=fate-suite --enable-gpl --enable-memory-poisoning --enable-avresample -- enable-gpl
[15:00] <anshul_> 4 libavutil 52. 52.100 / 52. 52.100
[15:00] <anshul_> 5 libavcodec 55. 41.100 / 55. 41.100
[15:00] <anshul_> 6 libavformat 55. 21.100 / 55. 21.100
[15:00] <anshul_> 7 libavdevice 55. 5.100 / 55. 5.100
[15:00] <anshul_> 8 libavfilter 3. 90.102 / 3. 90.102
[15:00] <anshul_> 9 libavresample 1. 1. 0 / 1. 1. 0
[15:00] <anshul_> 10 libswscale 2. 5.101 / 2. 5.101
[15:00] <anshul_> 11 libswresample 0. 17.104 / 0. 17.104
[15:00] <anshul_> 12 libpostproc 52. 3.100 / 52. 3.100
[15:00] <anshul_> 13 [Parsed_amovie_0 @ 0xa7b9a20] Failed to avformat_open_input 'fate-suite/filter/seq-3341-7_seq-3342-5-24bit.flac'
[15:00] <anshul_> 14 [lavfi @ 0xa7b8980] Error initializing filter 'amovie' with args 'fate-suite/filter/seq-3341-7_seq-3342-5-24bit.flac'
[15:00] <anshul_> 15 amovie=fate-suite/filter/seq-3341-7_seq-3342-5-24bit.flac,ebur128=metadata=1: No such file or directory
[15:00] <durandal_1707> can you learn to use pastebin or similar?
[15:00] <anshul_> sorry
[15:01] <durandal_1707> iirc, amovie filter needs full path
[15:01] <anshul_> found the error file
[15:03] <anshul_> actually i was not trying to use amovie, i was trying to run fate on my system successfully without any error
[15:04] <anshul_> so for this error do i need to file bug, because i think while passing file name some part has been added ",ebur128=metadata=1"
[15:05] <durandal_1707> nope, everything is explained in documentation
[15:05] <durandal_1707> and from error i do not see that extra stuff have been added to filename
[15:05] <anshul_> so do i need to change the fate test case for amovie
[15:05] <durandal_1707> also you are on wrong channel
[15:06] <durandal_1707> anshul_: what are you actually doing?
[15:07] <anshul_> i am analysing ffmpeg.
[15:09] <anshul_> last time one of my patch was rejected due to fate, so today i was just getting the basic understanding to use fate.
[15:10] <anshul_> I thought fate is ffmpeg_developer tools ffmpeg users might not know how to use it
[15:10] <durandal_1707> so what is patch about?
[15:12] <anshul_> actually it was about an memory leak, long time back, this time i am just solving 2077 ffplay crash
[15:12] <anshul_> This time i have not submitted the patch till it work properly at my place
[15:15] <anshul_> to test that my change work properly, i have to make the refrence system work, that was what i am trying to do.
[15:15] <anshul_> but fate took my all day.
[15:16] <durandal_1707> that is strange - it should not happen
[15:16] <durandal_1707> you sure you have fate-suite directory set?
[15:16] <durandal_1707> and all files fetched?
[15:16] <nevcairiel> you dont need to setup a full fate box just for testing, the important point is to simply run make fate and see if your changes broke anything :)
[15:17] <durandal_1707> iirc this is is also mentioned in documentation
[15:17] <durandal_1707> it looks like there are tests that fails ...
[15:20] <anshul_> yes there are some test failing, talk to you guys tommarow, already late night!
[15:26] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:4c439f6e3999: avformat/idcin: check the decompressed frame size during probing
[15:29] <ubitux> i wonder what the hell that guy is trying to do
[15:31] <durandal_1707> fix ffplay crash
[15:44] <michaelni> ods15, there are 2 patches for libnut on nut-devel
[15:47] <michaelni> ods15, do you want to review/apply them ? should i apply them ? do you still maintain libnut ? should we try to find a new maintainer? ...
[15:48] <michaelni> http://lists.mplayerhq.hu/pipermail/nut-devel/2013-October/date.html
[15:50] <durandal_1707> michaelni: isn't libnut separate project and have nothing to do with wrapper?
[15:55] <michaelni> durandal_1707, yes, the patches are for libnut not the wrapper, my note here is probably strictly speaking off topic for ffmpeg
[15:58] <durandal_1707> iirc i do not think ods15 is maintainer of nut code, just of wrapper, correct me if i'm wrong
[15:59] <durandal_1707> also have fourcc been synced with code in ffmpeg?
[16:09] <michaelni> durandal_1707, ods15 is the author of libnut, last comit to it seems from 2008, also dont mix up the nut spec, ffmpegs nut implementation, the libnut wraper in ffmpeg and libnut itself
[16:10] <durandal_1707> i do not mix stuff
[16:12] <wm4> why is there a libnut wrapper in ffmpeg?
[16:12] <durandal_1707> apparently ods15 is/was maintainer of libnut and libnut wrapper
[16:12] <durandal_1707> wm4: because
[16:13] <durandal_1707> how otherwise would you test it?
[16:14] <wm4> whatever else uses libnut?
[16:26] <av500> nothing
[17:12] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:f8a9cf77040e: avformat/lvfdec: check stream count during probing
[18:01] <cone-424> ffmpeg.git 03Diego Biurrun 07master:0338c396987c: dsputil: Split off H.263 bits into their own H263DSPContext
[18:01] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:a30f7918b52d: Merge commit '0338c396987c82b41d322630ea9712fe5f9561d6'
[18:09] <cone-424> ffmpeg.git 03Diego Biurrun 07master:cb56c3961b18: h263dsp: K&R formatting cosmetics
[18:09] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:e93523a06f27: Merge remote-tracking branch 'qatar/master'
[20:23] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:4f0e85517f5d: avformat/icodec: reduce score returned on probing
[20:24] <cone-424> ffmpeg.git 03Paul B Mahol 07master:2145e8f19353: avcodec/vmdav: use av_freep() and reset unpack_buffer_size
[20:24] <cone-424> ffmpeg.git 03Paul B Mahol 07master:b8f276e64574: avcodec/mimic: use av_freep() and reset swap_buf_size
[22:40] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:32584667ea51: tools/probetest: use named constant for the maximum number of formats
[22:40] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:6b4f58d448f4: tools/probetest: also print the time the probe functions needed
[22:41] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:a26e83da6c96: avformat/mxfdec: optimize probing
[23:42] <cone-424> ffmpeg.git 03Kostya Shishkov 07master:5462dde1947e: metasound: Fix error message argument
[23:43] <cone-424> ffmpeg.git 03Michael Niedermayer 07master:ac4447b6ec4e: Merge commit '5462dde1947e0ecfcb2db99ae29ce9dee8933b45'
[23:53] <ubitux> michaelni: ah sorry, just get it for AS
[23:53] <ubitux> will re-send a patch
[00:00] --- Sat Nov 9 2013
1
0
[00:39] <hailwood> Hey guys, I am using PHP-FFmpeg - should I ask here about issues with it or on #PHP? My specific issue is that one format is double running a command, but all the others are fine.
[00:42] <llogan> hailwood: we don't have anything to do with that
[00:42] <hailwood> Sweet as - to #php it is!
[00:43] <llogan> hailwood: this might be useful for you http://trac.ffmpeg.org/wiki/Using%20FFmpeg%20from%20PHP%20scripts
[00:43] <llogan> see first paragraph in "conclusion"
[00:47] <hailwood> llogan, Yeah; I'm not using ffmpeg-php instead I am using this: https://github.com/alchemy-fr/PHP-FFmpeg which is actively developed
[01:06] <paulfm> Hey everyone
[01:08] <paulfm> I'm having trouble with ffmpeg on osx - finding libmp3lame. I have the lib/headers installed and am using --enable-libmp3lame... and still get the following error:
[01:08] <paulfm> ERROR: libmp3lame >= 3.98.3 not found
[01:09] <paulfm> Any advice is much appreciated
[01:09] <paulfm> libs are in /usr/local/lib
[01:09] <paulfm> config.log says is couldn't find <lame/lame.h>
[01:10] <paulfm> This file is in /usr/local/lib/lame/lame.h
[01:10] <paulfm> Lastly... libmp3lame is 3.99.5
[01:13] <paulfm> correction: header is in /usr/local/lame/lame.h
[01:14] <hailwood> Hmm, what's the effect of running a command twice with the only difference being -pass 1 or -pass 2 ?
[01:14] <paulfm> ob wow... I mean file is in /usr/local/include/lame/lame.h
[01:15] <paulfm> Is there any way to tell configure the actual path of libmp3lame
[01:17] <triggerhapp> How well would ffmpeg deal internally with, say, a video source that can change dimensions? I dont suspect there's much support right?
[01:19] <paulfm> looks like --disable-yasm is what I needed...
[01:19] <paulfm> and leaving out --enable-libmp3lame...
[01:31] <llogan> paulfm: --disable-yasm is another way of saying "--enable-slowness"
[01:31] <paulfm> haha
[01:31] <llogan> did the tail of config.log have any useful info?
[01:31] <paulfm> llogan: well - it compiled.. with mp3lame... weirdly enough
[01:32] <paulfm> all it said was that it couldn't find the header
[01:32] <paulfm> As far as I can remember
[01:32] <llogan> looks like someone else had similiar issues (I guess) lately http://ffmpeg.org/pipermail/ffmpeg-user/2013-November/018364.html
[01:33] <paulfm> make check passed all tests
[01:33] <llogan> might be useful for you: https://trac.ffmpeg.org/wiki/MacOSXCompilationGuide
[01:34] <paulfm> Yeah... odd.. I even tried older version of the libs.. but still no luck. I find it odd that adding disable-yasm, and not even mentioning to enable libmp3lame... build ffmpeg with mp3 support
[01:35] <paulfm> Ahaa.. handy guide.
[01:35] <fundies> any way can I play a mp4 with a missing trex / header?
[01:35] <paulfm> will try it with those options... thanks llogan
[01:38] <paulfm> I get same build error with OS X build guid options
[01:38] <paulfm> --disable-yasm makes it all work
[01:39] <llogan> fundies: sure: ffplay -f rawvideo -s 320x240 -i input.mp4
[01:40] <paulfm> actually... something is wrong here... checking
[01:41] <fundies> llogan, ffplay -f rawvideo -s 640x480 ~/Downloads/CCLENG0001-480-1000K.mp4Seg1-Frag3
[01:41] <fundies> i got colored bits
[01:41] <llogan> that was a bad joke.
[01:42] <paulfm> hmmm... was too hasty.. mp3 decode is enabled(go figure).. no encode :-(
[01:42] <fundies> llogan, http://pastie.org/8464180
[01:43] <paulfm> llogan: do you know if there is a way to tell ffmpeg where exactly to look for libs/headers?
[01:43] <Danathus> hey
[01:44] <Danathus> I am trying to use ffmpeg (on windows, if it matters) to take a bunch of png files in order and put them in a video file, playing at 1 fps
[01:44] <llogan> fundies: i would guess not with ffmpeg, but i'm not sure. maybe JEEB does.
[01:45] <Danathus> I have it working except for the 1 fps part, and whenever I try to specify a frame rate I get videos with just one frame throughout
[01:45] <fundies> JEEB ?
[01:45] <Danathus> has anyone here had any success doing something like what I'm trying to do? Or possibly knows that it's not possible?
[01:46] <llogan> Danathus: http://ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_003f
[01:46] <Danathus> my command line looks like this: ffmpeg.exe -y -f image2 -i img%%03d.png video.mov (without attempting to set fps, works, but plays much faster than I want)
[01:46] <Danathus> thanks, I'll take a look
[01:47] <llogan> do you want it to show one frame per second, and/or do you want the output to actually be 1 fps?
[01:47] <Danathus> hmm that doesn't seem to cover setting a frame rate
[01:48] <Danathus> ideally the output to be 1 fps, but I'm not sure I'm entirely clear on the difference
[01:48] <llogan> for example, you can set input -r 1 and output -r 25. it will duplicate frames to reach -r 25 but the same image will be shown for one second
[01:48] <Danathus> I'd like it so that the images stored in the final output are just one image every second of video, so it's a smallish file
[01:48] <Danathus> ah I see
[01:48] <llogan> then add -r 1 as an output option
[01:48] <llogan> i mean input
[01:49] <llogan> note that some players may not work with -r 1
[01:49] <Danathus> -r 1 as input, and what for output?
[01:49] <llogan> nothing, since the output will inherit the input -r
[01:49] <Danathus> maybe that is my problem, the player not working -- testing with vlc
[01:49] <Danathus> okay
[01:50] <llogan> then add an output -r and increase the value, starting with 2, until it works if you are trying to limit output frame rate
[01:50] <Danathus> when I use -r 1 for input I get a video that is the correct length but when I play it it's just one still frame for the entire duration
[01:50] <Danathus> trying the -r 2 and up
[01:51] <llogan> paulfm: --extra-cflags --extra-ldflags
[01:51] <llogan> example http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[01:52] <paulfm> llogan: thanks - taking a look
[01:52] <Danathus> when I try with -r 24 on the output, it takes a lot longer to encode, I guess that's a good sign :/
[01:52] <Danathus> okay, output looks like it's correct then
[01:52] <Danathus> thans llogan
[01:52] <Danathus> *thanks
[01:53] <llogan> also see https://trac.ffmpeg.org/wiki/x264EncodingGuide
[01:53] <Danathus> seems well compressed still I think, I am satisfied
[01:53] <Danathus> hmm
[01:54] <Danathus> I think it will take me a bit to sift through that (:
[01:54] <llogan> if your output is not yuv420p then it won't work with dumb players like QuickTime
[01:54] <Danathus> I have a follow-up question
[01:54] <Danathus> alright
[01:54] <Danathus> works with vlc anyhow, that's what I care about now (ultimately to be read with custom software anyway)
[01:55] <llogan> ok. then don't worry about that
[01:55] <Danathus> my follow-up question is, really my input of png files are a temp intermedia format, and I am wondering if I might be able to have ffmpeg create the output from my source format
[01:55] <Danathus> I'll explain
[01:56] <Danathus> I am starting with a file which is a sequence of raw image data frames, one after another, no header information of any kind -- but I know the bit depth, dimensions, etc
[01:57] <Danathus> I am using other code to break this into a bunch of .raw files, one for each frame, and then I loop through making an ffmpeg call to convert each one to a png file, and then have my final ffmpeg command line call to put all the png files together in the mov file
[01:58] <paulfm> llogan: --extra-cflags="-I/usr/local/include/" worked!!.. thank you
[01:58] <Danathus> if I didn't have to convert from the .raw files to the png files that would be an improvement, and if I could get ffmpeg to read from the big fat sequential raw file, that would be even better
[01:59] <Danathus> but where img%03d.png seems to work, img%03d.raw doesn't, even if I am specifying -vcodec rawvideo -f rawvideo -pix_fmt rgb32 -s 480x640
[02:00] <Danathus> I *should* be able to get the .raw files to work, right?
[02:01] <llogan> i'd assume so, but i think the %03d syntax is specific to the image2 muxer
[02:02] <llogan> meaning i don't know if that would work with raw.
[02:02] <Danathus> hmm I see
[02:02] <llogan> you can ask on ffmpeg-user mailing list
[02:03] <llogan> did you try the big raw file as an input?
[02:06] <Danathus> I have
[02:06] <Danathus> it complains that it can't find the second frame
[02:07] <Danathus> okay
[02:08] <Danathus> for the big raw file attempt?
[02:09] <llogan> yes
[02:10] <Danathus> http://pastebin.com/7f7v8sSR
[02:10] <Danathus> sorry for the delay
[02:10] <Danathus> oh I'll include the error too
[02:10] <Danathus> my bad
[02:13] <Danathus> here http://pastebin.com/Ssm3fqwn
[02:15] <Danathus> updatefirst didn't seem to help either
[02:15] <llogan> -f image2 with mov container doesn't make sense to me
[02:16] <Danathus> okay, should I just omit the -f image2?
[02:16] <Danathus> oh wow it's working
[02:16] <llogan> isn't the ultimate goal to make a video out of the raw input? why even output to frames?
[02:17] <Danathus> only because that's how I was getting it to work
[02:17] <Danathus> but this is much better
[02:17] <Danathus> thanks a lot
[02:17] <Danathus> now I just need to combine this with the framerate bit...
[02:18] <llogan> by default it assumes the raw input is 25 frames/s
[02:18] <Danathus> alright
[02:19] <llogan> perhaps you can pipe the raw to ffmpeg from whatever makes the raw file
[02:19] <Danathus> the raw data isn't generated sequentially
[02:19] <Danathus> that's the only problem there
[02:20] <Danathus> okay I have a line that does exactly what I want now, and works swimmingly
[02:40] <fundies> noone can help me play this file?
[02:48] <Danathus> thanks again llogan
[02:48] <Danathus> I'm off
[02:48] <Danathus> good luck to you fundies
[02:48] <Danathus> ciao everybody
[03:28] <fundies> JEEB dead?
[05:06] <phunyguy> hello, I am trying to encode flac to mp3 in a script, which mostly works, but the -map_metadata option doesn't bring over the DATE tag.
[05:07] <phunyguy> it brings over everything else, including musicbrainz data, which makes this so strange.
[05:34] <phunyguy> le sigh. It was writing ID3v2.4
[05:39] <DANtheBEASTman> how does one convert m4a to mp3 v0?
[05:39] <DANtheBEASTman> looking through the wiki all I can find is video examples
[05:42] <DANtheBEASTman> nvm, figured it out! for f in *.m4a; do; avconv -i "$f" -qscale:a 0 "${f[@]/%m4a/mp3}"; done
[05:42] <DANtheBEASTman> thanks anyway!
[06:14] <kcm1700> I am creating mp4 file with some code. I'm feeding a packet to the AVFormat with av_write_frame. Which does AVPacket.pts use as a time base, AVStream.time_base or AVStream.codec->time_base ?
[06:21] <ChannelZ> Anyone familiar with the guts of h264/aac in an mp4?
[06:23] <ChannelZ> As in, I'm trying to help someone fix a corrupt mp4 (quicktime really) that crashed while recording... so no moov atom. But I've got the data in the mdat atom which should be the actual media streams, but am not sure what "format" it's in such that I could feed it to ffmpeg or anything to try to salvage what is there.
[09:36] <tiksa> hey guys, could you please tell what's wrong with my ffmpeg command? My intention is to just trim video so that only video+audio between seconds 2..4 are left. the command is:
[09:36] <tiksa> ffmpeg -i input.mp4 -filter_complex "[0:0]trim=start=2:end=4[video];[0:1]atrim=start=2:end=4[audio]" -map '[video]' -map '[audio]' output.mp4
[09:38] <tiksa> it does not throw any error but what happens, the first 2 seconds are silent and the frame at 0:02 is shown. then, the rest 2 secs are as it should, from 0:02 to 0:04
[09:42] <noobed> could some1 give me example source code of decoding an Vorbis audio stream and encoding it into MP3 one
[13:02] <jay_> why does a video player say the file(.mp4) contains no playable stream but .flv, .avi etc play when the file is not 100% complete?
[13:04] <ubitux> because the header is likely at the end
[13:04] <JEEB> flv has no index, and avi can be parsed without one :P
[13:04] <JEEB> and most people and apps create mp4 files that have a single index, at the end
[13:04] <JEEB> because the index has to contain everything in the file
[13:06] <jay_> hmm, thanks for the replies ubitux , JEEB. I'll have to read a little on that
[13:07] <JEEB> mp4 can have smaller indexes around the file, but not many things use that feature
[13:07] <JEEB> "movie fragments" the feature was called, IIRC
[13:08] <jay_> JEEB, to sum it up, mp4 requires an index unlike flv and avi and for some reason it is usually at the end of the file
[13:08] <JEEB> not for "some reason"
[13:08] <JEEB> but for the reason that you have to have the whole file created when you make the index
[13:08] <JEEB> usually it's put in the beginnig by first creating the file, and then rewriting it so that you write the index first :P
[13:09] <jay_> oh ok
[13:16] <jay_> is it possible to make a new index with part of a .mp4 file?
[13:17] <JEEB> no
[13:17] <JEEB> since you have no idea what's in the file if you didn't make it yourself
[13:17] <JEEB> and if you made it yourself you could just do that again and use movie fragments
[13:17] <JEEB> (or just put the index in the beginning if you want that)
[13:18] <jay_> interesting. Thanks JEEB . I think i have to go read about indexing mp4
[17:51] <exfalso> Hi, i'd like to write a small program that simply takes a single picture and audio file and creates a video out of it. what encoding is best for this? what libraries should i use? (i'm a total noob, not even sure i'm asking in right channel)
[17:52] <exfalso> or is there an existing way of doing this with ffmpeg?
[18:13] <exfalso> this was easier than i thought... now just need to figure out how to keep audio quality the same
[18:17] <exfalso> was easy too... ffmpeg rocks
[18:47] <uehuehue> What does ffmpeg return on error/success?
[18:48] <uehuehue> Is it OK to do "ffmpeg <blah> && rm file"?
[18:52] <ubitux> yes
[18:52] <ubitux> is file the output file?
[18:52] <uehuehue> file would be the input file
[18:53] <ubitux> k
[18:53] <uehuehue> As in, convert file to something and remove the input file
[18:53] <ubitux> i hope you always keep a copy of your original somewhere though :)
[18:54] <uehuehue> Heh, converting from mp3 to ogg. Not sure what could go wrong
[19:02] <ubitux> well it's lossy to lossy so quality will go down
[19:02] <ubitux> ah he left.
[19:28] <ac_slater> hey guys. Is there an x264 channel? If not, maybe someone knows how to cross compile x264 (for ARM7) AND enable asm. I can compile only with disable-asm. Do I need and arm yasm? I'm confused. Thanks!
[19:29] <JEEBsv> #x264 is the x264 channel
[19:29] <JEEBsv> and no, you don't need yasm for ARM compilation
[19:29] <JEEBsv> the asm is generally inline for that, the little there is of it
[19:29] <ac_slater> oh. Weird I joined it and it was empty, it isnt now. Thanks though.
[19:29] <cortexman> ffmpeg looks like it supports NIST Sphere files http://ffmpeg.mplayerhq.hu/doxygen/trunk/nistspheredec_8c_source.html
[19:29] <ac_slater> JEEBsv: So you dont think it'll be beneficial? I was hoping there was NEON support, etc
[19:29] <JEEBsv> let's move the discussion there
[19:32] <cortexman> can ffmpeg output raw audio?
[19:32] <JEEBsv> cortexman: yes
[19:32] <JEEBsv> see ffmpeg -codecs:a
[20:14] <cortexman> i have a NIST Sphere file with these headers: sample_coding -s4 ulaw, sample_rate -i 8000, sample_n_bytes -i 1, sample_sig_bits -i 8
[20:14] <cortexman> no matter what options i specify the output is 16k which makes the audio sound pitch shifted into a chipmunk cartoon
[20:25] <durandal_1707> cortexman: sample, link to it?
[20:27] <cortexman> durandal_1707: http://filebin.ca/11FdZnYT7BCH/sw02001.sph
[20:27] <cortexman> i'm trying to get the first channel rate converted to 16k without becoming chipmunky
[20:27] <cortexman> short ints
[20:28] <cortexman> raw format
[20:46] <cortexman> figured out a way to do it with sox!
[20:49] <durandal_1707> cortexman: it works fine here
[20:49] <durandal_1707> and it is ulaw
[20:50] <cortexman> durandal_1707 you were able to convert the file?
[20:50] <durandal_1707> with ffmpeg? sure
[20:50] <cortexman> i was not able to convert it to 16k and have it properly rate converted with ffmpeg
[20:51] <cortexman> https://gist.github.com/brianmingus/b8ca6659c541a0fb2e72/raw/7b991ecb053274…
[20:53] <durandal_1707> cortexman: you are not using nist demuxer but raw one
[20:53] <cortexman> the output should be raw
[20:54] <durandal_1707> and you are not giving full uncut console output but just worthless commands
[20:54] <cortexman> that's the only console output..
[20:56] <cortexman> i figured ffmpeg would choose nist automatically based on .sph
[20:56] <durandal_1707> cortexman: what you use to play raw file?
[20:56] <cortexman> `play`
[20:56] <cortexman> it's part of sox
[20:57] <durandal_1707> and do you set number of channels?
[20:57] <durandal_1707> what is 'play' command ?
[20:57] <cortexman> it plays an audio file
[20:57] <durandal_1707> yes but full command is?
[20:58] <cortexman> play file
[20:58] <durandal_1707> > play /tmp/o.raw
[20:58] <durandal_1707> play FAIL formats: bad input format for file tmp/o.raw': sampling rate was not specified
[20:58] <durandal_1707> raw can not work
[20:58] <cortexman> right.. there is play-sample-rate
[20:58] <cortexman> you can figure out what the sample rate of the file is by tweaking that
[20:58] <cortexman> until it sounds right
[20:58] <durandal_1707> but if you do not set channel number it will not be correct
[20:59] <cortexman> yeah i think that's the problem
[21:00] <durandal_1707> play -c 2 -r 8k -e signed -b 16 /tmp/o.raw
[21:00] <durandal_1707> this works fine
[21:02] <cortexman> ahhh! thanks!
[21:02] <cortexman> i used that on fmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le sw02001.raw
[21:02] <cortexman> i already know how to extract a channel with sox also
[21:04] <durandal_1707> you can do same with filters
[21:07] <durandal_1707> fmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le -af pan="mono|c0=FR" sw02001.raw
[21:08] <cortexman> oh man that would have taken me forever
[21:08] <cortexman> i had already tried like 10 methods:)
[21:09] <durandal_1707> like typing random stuff?
[21:10] <cortexman> interestingly you can still hear the second channel in the background. it's just very faint.
[21:12] <cortexman> ffmpeg must be doing something weird because the voices are encoded on separate channels
[21:13] <durandal_1707> what?
[21:13] <durandal_1707> you want to extract single channel or remix?
[21:15] <cortexman> ffmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le -map_channel 0.0.0 sw02001_1.raw -f s16le -acodec pcm_s16le -map_channel 0.0.1 sw02001_2.raw
[21:15] <cortexman> :)
[21:15] <cortexman> thanks for your help
[22:51] <maujhsn> I'm using an older version of ffmpeg version git-2012-01-21-9e53f62 Copyright 2012
[22:51] <maujhsn> i would like to know how to control the "rec" volume when I am recording mydesktop without using an external mic?
[22:52] <maujhsn> I thought I could do this in "pavucontrol" but its not increasing record volume!
[22:53] <share> do you recommend using ffmpeg to convert DTS track to AC3?
[22:54] <maujhsn> share Whom are you talking to?
[22:55] <share> to the users on this channel
[22:55] <maujhsn> Ok
[22:56] <DexterF> share: http://www.videohelp.com/tools/PopCorn-MKV-AudioConverter - off topic, but might be what you need. guess will run in wine if there's no native linux code
[22:57] <DexterF> that said I'd be interested, too, if ffmpeg can be used to convert and how it then handles the different dynamic ranges etc. usually dts means "fumble with the volume controls for two hours"
[22:59] <maujhsn> DexterF Was this message intended for me?
[22:59] <share> DTS to AC3 5.1
[22:59] <share> could be
[22:59] <DexterF> maujhsn: to the channel
[23:00] <maujhsn> Ooops!
[23:01] <share> EAC3toGUI ?
[23:01] <share> there is this....... https://github.com/JakeWharton/mkvdts2ac3/
[23:11] <share> Application provided invalid, non monotonically increasing dts to muxer in stream 0 ********* ?
[23:16] <maujhsn> I'm using an older version of ffmpeg version git-2012-01-21-9e53f62 Copyright 2012
[23:16] <maujhsn> i would like to know how to control the "rec" volume when I am recording mydesktop without using an external mic?
[23:16] <maujhsn> I thought I could do this in "pavucontrol" but its not increasing record volume!
[00:00] --- Sat Nov 9 2013
1
0
[09:07] <anshul_> help: i was trying to analyse bug 2093, need problem.rm video
[09:13] <sspiff> anshul_: isn't it linked at the top of the bug report?
[09:13] <sspiff> http://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket2077/problem.rm ?
[09:14] <sspiff> anshul_: downloads fine here
[09:14] <anshul_> thanks i never noticed that
[09:15] <sspiff> anshul_: you're welcome
[09:19] <anshul_> In that report michael and cehoyos were unable to reproduce the problem on linux, was it playable on linux
[09:21] <anshul_> at my place on linux, i am not able to play that video, but i get these warning
[09:21] <anshul_> [rm @ 0xb10025c0] Unsupported stream type 000003a
[09:21] <sspiff> anshul_: using ffplay?
[09:23] <sspiff> ffplay gives me the output seen in https://trac.ffmpeg.org/ticket/2093#comment:15
[09:23] <sspiff> mplayer plays the file fine
[09:25] <anshul_> ok
[09:40] <anshul_> sspiff isn't mplayer too use the libavcodec and component specified in bug report is avcodec
[10:02] <sspiff> anshul_: I think mplayer has its own decoders
[10:13] <sspiff> ubitux: My idea of how to solve the encoding issue was crap :) I don't see how it can be solved without having the encoder output packets directly (or keeping track of a queue of out-of-order packets that need to be sent later on each output stream)
[10:46] <plepere> hello
[10:48] <plepere> I'm currently working on the OpenHEVC decoder which works in ffmpeg, and I'm told that you don't really like SSE intrinsics, so I'm going to pass the low level functions (MC and all) to assembly.
[10:48] <plepere> is someone already on assembly HEVC decoding ?
[10:49] <nevcairiel> I havent heard of someone specifically working on assembly routines
[10:50] <ubitux> i think Skyler_ was talking about doing some at some point, but it's been a while now
[10:50] <nevcairiel> make sure to write them in YASM and not inline gcc asm thogh =)
[10:50] <plepere> nevcairiel, sure !
[10:50] <ubitux> you might want to ask smarter and mraulet
[10:51] <plepere> I just want to make sure that there won't be 2 people doing the same thing.
[10:51] <plepere> I work with mraulet. :)
[10:51] <ubitux> oh.
[10:54] <plepere> but thanks for the indication.
[10:57] <plepere> I also cross-compiled ffmpeg for android and managed to decode some HEVC on my phone, so it's quite solid. :)
[12:42] <cone-764> ffmpeg.git 03kowalsky 07master:4684539c9148: avformat/hls: fixed bug where custom http headers weren't kept for hls streams
[14:12] <sspiff> can anyone point me to where AVSubtitle.pts is ever set?
[14:12] <sspiff> I can't find it...
[15:03] <{V}> sspiff, https://github.com/FFmpeg/FFmpeg/blob/02a6ee51685eb74f7a878dd49553ecc1f8da9… ?
[15:04] <{V}> oh wait. that's AVPacket
[15:09] <ubitux> sspiff: avcodec_decode_subtitle2
[15:09] <ubitux> sub->pts = ...
[15:14] <sspiff> ubitux: thanks, I found it eventually :)
[15:24] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:cbe84b4ffae4: avformat/astdec: sanity check channels & sample rate
[16:12] <plepere> hmm
[16:13] <plepere> should I do x64 assembly or stick to 32bit code ?
[16:14] <ubitux> for vp9 8x8 idct i only added a x64 one
[16:14] <ubitux> because it was a pain for 32
[16:14] <ubitux> so i guess you can do x64 only for a start
[16:31] <durandal_1707> lol what review fail
[16:38] <ubitux> durandal_1707 ?
[16:49] <durandal_1707> ubitux: not your code
[16:51] <ubitux> durandal_1707: ok...
[16:52] <ubitux> durandal_1707: btw, are you going to add threading to vp9?
[16:52] <durandal_1707> why would I?
[16:52] <ubitux> because you love threading of course
[16:52] <durandal_1707> but i do not love vp9
[16:52] <ubitux> why? :(
[16:53] <ubitux> you prefer vf rotate?
[16:53] <durandal_1707> i prefer lossless
[16:56] <durandal_1707> ubitux: i think BBB is going to do it
[16:58] <ubitux> he's busy with 32x32 idct, and probably 16x16 later
[16:58] <ubitux> he asked me to do the threading because it was supposely simple (and would offer quite some optim)
[16:58] <ubitux> but i don't like it
[16:58] <ubitux> so i suggested you do it
[16:59] <durandal_1707> why you do not like it?
[17:01] <ubitux> i find debugging way too random
[17:01] <ubitux> not enough binary, too much black magic to me
[17:02] <durandal_1707> there is no magic at all
[17:02] <ubitux> speaking of random ^
[17:02] <durandal_1707> you just need to recognise memory that is going to be shared between threads
[17:03] <durandal_1707> for frame threaded decoding
[17:03] <durandal_1707> other one should be mush simpler, you just need to find out what can be threaded
[17:04] <ubitux> i just don't want to do it :p
[17:04] <ubitux> i'm busy with some other optims in vp9
[17:05] <cone-764> ffmpeg.git 03Paul B Mahol 07master:315e3cf047cf: avcodec/hnm4video: remove redundant log message
[17:06] <durandal_1707> michaelni: ^ that decoder can overread, i'm not motivated to fix such mess
[17:08] <ubitux> durandal_1707: there are 2 similar ones in libavcodec/libfdk-aacdec.c if you're interested :p
[17:08] <ubitux> (av log get buffer)
[17:15] <cone-764> ffmpeg.git 03Paul B Mahol 07master:9521236881cb: avcodec/libfdk-aacdec: remove redundant log messages
[18:07] <zidanne> I am getting error while trying to compile for Android NDK and x86. in apply_window_mp3 : http://pastebin.com/XQFX3AkH
[18:07] <zidanne> Any ideas?
[18:42] <michaelni> zidanne, exact compiler & compiler flags would be interresting
[18:44] <michaelni> zidanne, also what version of ffmpeg is that ?
[18:44] <zidanne> N-50101-g76415a9
[18:46] <michaelni> 9 month old
[18:46] <zidanne> You are right, I will update and try it again
[19:24] <zidanne> michaelni: I fetched the latest source from git. still the same error. config: http://pastebin.com/yiuXKNr7
[19:29] <wm4> af_aecho.c has a use-after-free problem
[19:30] <wm4> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavfilter/af_aecho.c;h=9c8…
[19:30] <michaelni> zidanne, thats a pretty long configure line, is that all needed for the problem to happen (also which line numbers does it fail at and how with git master) and its likely a compiler bug
[19:30] <wm4> it's pretty obvious: frame is possibly freed, and then accessed
[19:30] <wm4> durandal_1707: ?
[19:31] <michaelni> zidanne, also --disable-asm might fix it but will make the code slow
[19:31] <zidanne> disabling sse sse2 sse3 fixes the problem. But it's not cool :|
[19:33] <zidanne> configuring with arch x86 may be enough to get this error
[19:35] <ubitux> zidanne: unrelated, but you can --enable-demuxer=mpegts,mpegvideo,mpegps,ogg...
[19:35] <ubitux> same for the others
[19:36] <zidanne> thanks
[19:37] <zidanne> I was using Android-ndk and changing from x86-4.4.3 to x86-4.6 fixed the problem.
[20:03] <durandal_1707> wm4: fixed
[20:07] <cone-764> ffmpeg.git 03Paul B Mahol 07master:d9d752cfb55c: avfilter/af_aecho: free input frame after it is being used
[20:33] <cone-764> ffmpeg.git 03Clément BSsch 07master:f70db22999d7: avformat/image2: allow muxing gif files.
[20:33] <cone-764> ffmpeg.git 03Clément BSsch 07master:3c5e6e76356e: avformat/img2enc: reindent after previous commit.
[20:33] <cone-764> ffmpeg.git 03Clément BSsch 07master:c37b0daf336c: avformat/gif: fix final_delay unit in documentation.
[20:33] <cone-764> ffmpeg.git 03Clément BSsch 07master:3e036d407d18: doc/muxers: document animated GIF muxer.
[20:34] <cone-764> ffmpeg.git 03Clément BSsch 07master:3bcd5c1e31e2: avformat/img2enc: simplify split planes extension selection.
[20:39] <ubitux> durandal_1707: 'going to review displace
[20:39] <ubitux> durandal_1707: patch is the latest version or you have some recent changes in your git?
[20:44] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:3836af476534: avformat/mpegvideodec: check seq headers more completely during probing
[20:49] <durandal_1707> ubitux: yes and no
[20:49] <ubitux> ok
[20:49] <ubitux> pff
[20:49] <ubitux> /home/ubitux/src/ffmpeg/libavfilter/allfilters.c:126: undefined reference to `ff_vf_displace'
[20:50] <ubitux> well i can fix that
[20:52] <cone-764> ffmpeg.git 03Dave Yeo 07master:f9c6044a6f61: Fix compilation with os2threads
[22:11] <saste> how to ban trac users?
[22:12] <llogan> do trac ticket numbers get re-used after deletion?
[22:13] <saste> yes
[22:13] <saste> also who can delete tickets?
[22:13] <llogan> i can now. i deleted those others today
[22:16] <llogan> i don't have any user management tools though. i'm not sure why the spammer's name isn't linked like the rest
[22:16] <ubitux> is ffloger in infinite loop?
[22:16] <ubitux> it's been the third time it's reporting the same trac op
[22:16] <llogan> burek: ping
[22:18] <llogan> saste: i think carl and michael can also delete tickets. im not sure who else can.
[22:23] <cbsrobot> I think iive can too
[22:31] <Compn> what ticket needs to be deleted ?
[22:31] <Compn> im not sure if i have that power :P
[22:31] <llogan> Compn: i deleted them, but there will be more. like herpes.
[22:33] <llogan> Compn: if you view the ticket you may see "Delete" under "Description" on the right (next to reply)
[22:33] <iive> cbsrobot: i'm not sure I even have an account in the tracker...
[22:34] <cone-764> ffmpeg.git 03Lenny Wang 07master:7c02a77d250f: avfilter/unsharp: added optimized opencl kernels
[22:43] <Compn> i dont see delete, no
[22:57] <wm4> are there any known flv seeking bugs? here it seems to read the whole file until the seek target, instead of executing an avio seek
[23:12] <ubitux> wm4: i've expected a seeking issue with flv a while ago, so it wouldn't surprise me
[23:16] <nevcairiel> flv seeking is practically non functional
[23:38] <cone-764> ffmpeg.git 03Diego Biurrun 07master:cffecc0e7ebd: avfilter/graphparser: Directly free filter memory if initialization fails
[23:38] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:ec8e68c7633c: Merge commit 'cffecc0e7ebd642afaa1fb9f56fab1fcc283293c'
[00:00] --- Fri Nov 8 2013
1
0
[00:01] <noobed> compiling errors: /usr/local/lib/libavdevice.a(lavfi.o): In function `lavfi_read_packet': /home/vasko/experimental/ffmpeg/libavdevice/lavfi.c:334: undefined reference to `av_buffersink_get_frame_flags'
[00:01] <noobed> any help?
[01:07] <eristisk> What is the compression_level value that encodes to FLAC at the highest compression (smallest files)?
[01:08] <eristisk> ffmpeg -i blah.wav -acodec flac -compression_level 12 blah.flac
[01:08] <eristisk> ?
[01:21] <triggerhapp> I'm finding little documentation on the -g flag. I know so much that choosing 0 means it's all I-frames, but what do non-zero values denote?
[02:14] <Plorkyeran> maximum distance between keyframes
[04:06] <nrobbins> I am having trouble with the fieldmatch filter. When I use fieldmatch, decimate, my video plays twice as fast as it should. Can someone help me understand what I am doing wrong.
[04:23] Last message repeated 1 time(s).
[04:25] <roothorick> so clarify something for me. MP3 decoding is builtin and doesn't rely on an external library? And do I need a license upgrade for it?
[04:26] <roothorick> by MP3 I mean MPEG layer3 audio, not just the container
[04:26] <sacarasc> It's built in, I do not know about licensing ffmpeg.
[04:27] <therube_agone> http://www.ffmpeg.org/legal.html
[04:28] <roothorick> therube_agone: that does not answer the question of precisely what is available in GPL vs. LGPL
[04:29] <sacarasc> roothorick: Build without --enable-gpl and see what you get?
[04:29] <roothorick> I'm playing with configure right now
[04:30] <roothorick> what are "indevs"?
[04:34] <relaxed> input devices
[04:42] <nrobbins> No help on the fieldmatch filter?
[09:01] <noobed> some compiling erros .. could any1 help? /usr/local/lib/libavdevice.a(lavfi.o): In function `lavfi_read_packet':
[09:01] <noobed> /home/vasko/experimental/ffmpeg/libavdevice/lavfi.c:334: undefined reference to `av_buffersink_get_frame_flags' what library am I missing to point the linker to? I've tried -lavfi but no success
[09:02] <anshul_> i was trying to analyse bug 2093, need problem.rm video
[09:02] <anshul_> hey nobbed you did some change in source.
[09:03] <anshul_> whats ur platform, destro and all info
[09:04] <anshul_> whats ur(nobbed) platform, destro and other info
[10:56] <noobed> could any1 look @ cpp file and tell me where I'm decoding and writing raw audio wrong?
[11:03] Action: mrAZ is trying to find out what means "[3][0][0][0]" in : Stream #0:0[0x44]: Audio: mp2 ([3][0][0][0] / 0x0003), 44100 Hz, stereo, s16
[11:05] <viric> there are 4-byte identifiers per codec, I think. And I guess "[]" prints the byte value. 0 and 3 are non-printable characters, so hence they may be given as numbers in [].
[11:06] <viric> the FourCC, no?
[11:06] <mrAZ> viric: thx for hints
[11:07] <viric> my guess. no idea.
[11:08] <viric> the 0x0003 is a bit confusing though. It looks like two bytes, not four. :)
[11:25] <brontosaurusrex> ffmpeg -i video.mov (ignore audio if any) -i audio.wav (ignore video if any) -vcodec copy -acodec copy -cut to video length out.mov < how would the correct cli look like?
[11:27] <ubitux> -shortest maybe
[11:30] <brontosaurusrex> ubitux, no, since there is a big change that audio is shorter
[11:30] <brontosaurusrex> chance*
[12:15] <brontosaurusrex> uhmm, is there a way to generate silent audio if audio is shorter than video?
[12:28] <brontosaurusrex> so the solution might be if video longer audio then generate some sort of mute audio and mix it with audio added then mux ..
[12:28] <brontosaurusrex> dam*
[13:50] <brontosaurusrex> p.s. the solution was to use -t "$video" and -filter complex apad
[14:34] <viric> brontosaurusrex: thanks for reporting
[15:29] <noobed> am I writing a raw audio in the right way? - https://gist.github.com/noobed/7353451
[15:29] <noobed> why can't I open the myOutputTest file using avformat_open_input()
[18:06] <noobed> am I writing a raw audio in the right way? - https://gist.github.com/noobed/7353451
[18:06] <noobed> why can't I open the myOutputTest file using avformat_open_input() any ideas? any1?
[18:32] <ffmpegnoob> I am trying to stream a recorded file with ffserver via rtsp. VLC can play the stream perfectly. Quicktime, however, does not.
[18:33] <ffmpegnoob> Is there any information out there as to why quicktime outputs "RTSP Version not supported"
[18:34] <ffmpegnoob> yet vlc can stream it just fine? this is really frustrating :(
[18:36] <ffmpegnoob> Here is my ffserver configuration file: http://pastie.org/private/mjn4cq7c4gxcybo0hdkxa
[18:39] <ezekiel> ffmpegnoob: I don't know enough to say what the problem is - but as a matter of helping brainstorm,
[18:40] <ezekiel> it sounds like Quicktime doesn't support the version of RTSP used by ffserver
[18:40] <ezekiel> it sounds that way because Quicktime says exactly that in its error
[18:41] <ezekiel> assuming the error isn't lying, I would try two things - Upgrade Quicktime (I personally hate Quicktime, and would just dump it completely)
[18:41] <ezekiel> or try to convince ffserver to use a particular version of RTSP
[19:11] <joseph> ffmpegnoob: Quicktime can only play Quicktime files.
[19:11] <joseph> VLC can play anything.
[19:14] <joseph> But that's a quicktime file.
[19:14] <joseph> So ignore me.
[19:14] <joseph> :D
[20:00] <ffmpegnoob> joseph, ezekiel, thanks for the input
[20:01] <ffmpegnoob> ezekiel: is there any way to specify a version of rtsp?
[20:01] <ffmpegnoob> I can't upgrade quicktime any further
[20:02] <ffmpegnoob> ezekiel: ps, i fricking hate quicktime. I'd use VLC/mplayer any day over that P.O.S
[20:03] <ffmpegnoob> just so happens I am required to use quicktime for this specific use case
[20:26] <therube> ffmpegnoob: did you ? first two hits may give you some points to ponder: https://www.google.com/search?q=quicktime++%22RTSP+Version+not+supported%22…
[20:29] <ffmpegnoob> therube: i did look at those posts already. thanks, though
[20:29] <ffmpegnoob> I'll reponder them
[20:30] <ffmpegnoob> i've been scouring anything i can put my cursor on
[20:37] <therube> i don't have/use QT, but i believe one of the points is that /sometimes/ you can interchange http: for rtsp:, so you could try that? or maybe try a different source?
[20:37] <therube> rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov
[20:37] <therube> http://www.wowza.com/_h264/BigBuckBunny_115k.mov
[20:42] <ffmpegnoob> therube: yes, both of those succesffuly load, which makes me very sad
[20:42] <ffmpegnoob> *sniffle* *sniffle*
[22:19] <hailwood> Hey guys, on Fedora I have compiled FFmpeg manually, but it now resides in /home/MY_USER/bin/* which means that only my user can access it, Should I b OK to just cut/paste to /opt/local/bin ?
[22:19] <hailwood> *Should I be OK*
[22:24] <cbsrobot> hailwood: sure you can or you can add /home/MY_USER/bin/ to yout PATH environnement variable
[22:33] <hailwood> cbsrobot, It's actually for the apache user to access it! So hopefully moving it will just work!
[23:49] <tinyhippo> can ffmpeg download RTMP streams?
[23:55] <llogan> tinyhippo: ffmpeg supports RTMP as input
[23:55] <llogan> see ffmpeg -protocols
[23:56] <llogan> http://ffmpeg.org/ffmpeg-protocols.html#rtmp
[00:00] --- Fri Nov 8 2013
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[03:20] <cone-602> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.10.10': unknown revision or path not in the working tree.
[03:20] <cone-602> Use '--' to separate paths from revisions
[03:20] <cone-602> refs/tags/n0.10.10:HEAD: avutil/common: error out with clear message if __STDC_CONSTANT_MACROS is not defined with c++
[09:09] <plepere> hello
[09:16] <michaelni> hello
[11:10] <cone-150> ffmpeg.git 03Marton Balint 07master:105d4748cfc0: ffplay: avoid code duplication in AVFILTER enabled and disabled case
[11:10] <cone-150> ffmpeg.git 03Marton Balint 07master:61dd31977066: ffplay: add frame duration estimated from frame rate to VideoPicture
[11:10] <cone-150> ffmpeg.git 03Marton Balint 07master:5ecfcc7dff04: ffplay: add smarter method for determining video picture duration
[11:10] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:1b3d3a9c87f1: Merge remote-tracking branch 'cus/stable'
[11:58] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:a35e30b67222: avformat/libgme: decrease score by 1, fixes probetest failure
[11:58] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:50a3c4c5d263: avformat/rsd: Check channels and sample_rate fields
[11:58] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:7b7697646b9b: avformat/rsd: increase probe score, as misdetections are fixed
[13:04] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:ff1d81b08c06: avformat/swfdec: check version and size during probing
[13:04] <cone-150> ffmpeg.git 03Michael Niedermayer 07master:3e6b7bbce510: avformat/segafilm: check 2nd tag
[13:31] <sspiff> does anyone here have any knowledge of DVB subtitles?
[13:32] <sspiff> It looks to me like 1 packet contains a region update for a block/line of subtitles, and ffmpeg decodes multiple of these packets into one AVSubtitle
[13:32] <sspiff> but I'm not sure how it decides that a frame is done
[13:34] <nevcairiel> there is a "frame done" packet
[13:34] <nevcairiel> called a display segment
[13:37] <sspiff> nevcairiel: ah ok, thanks!
[13:37] <sspiff> nevcairiel: since you seem to understand this, can I ask you a few more questions?
[13:37] <nevcairiel> I dont really understand it too deeply, but feel free to try
[13:39] <sspiff> well, I'm having a hard time understanding what's going on ffmpeg.c:783 to 790, it's basically manipulating the AVPacket pts for DVB subtitles, but I don't know why and how
[13:40] <sspiff> I'm having a hard time figuring out the relation of the MPEG-TS pts'es to ffmpeg's internal pts'es, as well
[13:40] <nevcairiel> the mpeg-ts pts should just end up in the AVPacket.pts without much change
[13:41] <nevcairiel> not sure whats up with the dvb timestamps there, i only really used the libraries, not the ffmpeg application itself
[13:41] <sspiff> nevcairiel: alright, thanks.
[13:45] <sspiff> Looks to me like it's writing out two packets, and it's setting the PTS of the first to the start time of the subtitle, and the second to the end time (probably one to display and one to clear?)
[13:45] <sspiff> (I'm just thinking out loud, hoping someone will jump in and uncover the true problem :))
[13:47] <wm4> that code looks mysterious
[13:49] <sspiff> I'm getting the error (Application provided invalid, non monotonically increasing dts to muxer in stream 0: 551350 >= 309708) on the third AVPacket to go through do_subtitle_out, so I'm going to be guessing the third packet is the second subtitle being displayed, before the 1st has been cleared. (Possibly multiple display segments for one frame being issued)
[13:49] <sspiff> wm4: I've spent the past 2 days obsessing over it, hardly understanding anything
[13:49] <sspiff> but thanks to nevcairiel's comments, things are starting to make sense!
[13:51] <sspiff> Actually, there's a comment that says there's one packet to draw, one packet to clear. Doh!
[13:52] <kierank> it's obviously some hack of some sort
[13:52] <kierank> to fix some broken file
[13:54] <sspiff> what's the resolution of ffmpeg's PTSes?
[13:54] <kierank> comes from the container timebase
[13:55] <sspiff> Then what's "#define AV_TIME_BASE 1000000" for?
[13:55] <sspiff> because it seems like do_subtitle_out uses this (through AV_TIME_BASE_Q)
[13:58] <sspiff> but it seems like my statement is more or less correct
[13:58] <sspiff> it's not triggered by other subtitle outputs, because they don't have the blanking packet
[14:00] <nevcairiel> streams and codecs usually have their own timebase, but some global structures us the generic AV_TIME_BASE
[14:03] <sspiff> can I change dvbsubdec to hold on to subtitles until they're cleared, and outputting a new subtitle frame everytime something changed?
[14:04] <sspiff> that would make them bigger, but would negate the need for this kind of hackery
[14:12] <sspiff> anyone who has an oppinion on whether it would be acceptable to make the subtitles bigger for this case?
[14:13] <ubitux> there is a delay problem if you do that
[14:13] <ubitux> you're looking at removing the DVBSUB hack in ffmpeg.c?
[14:14] <sspiff> ubitux: yes
[14:14] <ubitux> i think marton was trying to achieve something similar with the teletext decoder
[14:15] <sspiff> ubitux: I wouldn't delay the first display, but I would just issue a clear & a new subtitle when the second comes along (possibly overlapping), with the new sub containing all the regions still visible
[14:15] <ubitux> maybe ask him directly, Marton Balint <cus -at- passwd.hu>
[14:15] <sspiff> ubitux: I will, thanks
[14:15] <sspiff> but can you elaborate what you mean by delay?
[14:15] <ubitux> sspiff: note that maybe i'm missing something, but when you call the decode function you feed it with a new packet everytime
[14:16] <ubitux> if half a time you return the decoded packet, and the other half an empty one (to end the previous)
[14:16] <ubitux> aren't you going to accumulate/skip half packets?
[14:16] <ubitux> there are some caps (check CAP_DELAY), look around this
[14:17] <ubitux> dunno if Marton got done with this
[14:17] <ubitux> problem is, doing this means API changes
[14:19] <sspiff> ubitux: a decode is called when the demuxer says "we have a complete packet", I suppose?
[14:20] <wm4> I find it ironic that the current API couldn't even implement a libass decoder
[14:20] <sspiff> and AVPacket needs to be fully constructed when it returns?
[14:20] <ubitux> sspiff: av_read_frame() returns a ready to decode packets
[14:20] <ubitux> with audio, you can (have to) call decode several time on that same packet sometimes
[14:20] <ubitux> (because the decode won't consume all the packet)
[14:21] <ubitux> this logic is not currently used for subtitles
[14:21] <ubitux> and we assume 1 packet = 1 decode
[14:21] <ubitux> sspiff: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/demuxing_decodi…
[14:21] <sspiff> nevcairiel: you seem to know more about DVB subtitles than us, does one packet contain all the segments to do one update?
[14:22] <wm4> having a subtitle interface that works differently could help with wrapping libass, and with teletext stuff
[14:22] <ubitux> this is the generic demux/decode logic, used for audio (for video, it seems 1 packet = 1 video frame all the time anyway)
[14:22] <nevcairiel> not necessarily no, it can have one segment per packet, so you need multiple packets to handle a full cycle, at least AFAIK
[14:23] <sspiff> nevcairiel: so then an decode doesn't necessarily result in a new frame being produced.
[14:23] <nevcairiel> right
[14:24] <sspiff> decoding a packet would result in the manipulation of the changes being staged for the next frame, but the frame wouldn't necessarily be done
[14:53] <sspiff> modifying ffmpeg.c to keep the clear packet in a big, dirty static AVPacket dvbsub_clear, and only sending it out as soon as a subtitle packet with a higher PTS occurs "fixes" the transcoding
[14:53] <sspiff> obviously, the last clear will be missing
[14:53] <sspiff> which is (probably? you tell me...) unacceptable
[14:53] <sspiff> ubitux: I'm guessing the last clear can't be omitted
[14:58] <ubitux> static is even more blocking since it allows processing only one stream
[14:59] <sspiff> There's an almost disturbing amount of DVB subtitle specific stuff in do_subtitle_out...
[14:59] <sspiff> ubitux: right
[14:59] <sspiff> I could put it in the context
[14:59] <ubitux> and yes, encoding subtitles api is heavily broken
[15:04] <ubitux> sspiff: feel free to try hacks and even submit them as rfc on ffmpeg-devel for better suggestion
[15:04] <ubitux> as long as it helps understanding what the issue is :)
[15:41] <sspiff> is an OutputStream unique to a single video/audio/subtitle stream?
[15:42] <ubitux> sspiff: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/ffmpeg.txt;hb=HEAD
[15:42] <ubitux> not sure this is still up-to-date
[15:42] <ubitux> but it might give you some hints
[15:42] <sspiff> thanks ubitux :)
[15:49] <sspiff> ubitux: am I reading that correctly, when I say a container file or transport stream or whatever else needs muxing is created from a bunch of OutputStreams, where every OutputStream is a single component (audio, video, subtitle, ...)
[15:51] Action: ubitux parse error around `needs muxing'
[15:52] <sspiff> an output file looks to me like it's something like a container etc, where you put in several streams, right?
[15:52] <ubitux> yes sure
[15:53] <ubitux> sspiff: formats is an alias for containers, handled by libavformat; demuxing/muxing
[15:53] <ubitux> sspiff: codecs is handled by libavcodec, decoding/encoding
[15:54] <sspiff> and a stream is something a codec works with, a format is something a muxing/demuxing plugin works with?
[15:54] <ubitux> each stream can be of type audio, video, subtitles (and a few others)
[15:54] <ubitux> some formats can contain several streams and type of streams
[15:55] <ubitux> some formats have limitations on the type of codecs, total number of streams, ...
[15:55] <ubitux> sspiff: http://ffmpeg.org/ffmpeg.html#Detailed-description
[15:56] <sspiff> ubitux: thanks
[16:21] <sspiff> ubitux: how would keeping track of which rectangles are still "active" in the decoder need an API change?
[16:23] <ubitux> i don't understand the question
[16:33] <sspiff> ubitux: I may have skipped a few steps in my mind :)
[16:35] <sspiff> so my proposal was to keep track of which areas are active in dvbsubdec.c, and whenever a packet arrives that calls for something to be displayed, to reconstruct the entire subtitle and produce an AVSubtitle with that
[16:36] <sspiff> and everytime a new AVSubtitle is created, the previous one's end_display_time can be determined and filled in.
[16:37] <sspiff> This would require you to check if any of the active rectangles have expired, and potentially create more than one AVSubtitle for one packet of data that arrives, and the use of CODEC_CAP_DELAY, as you said
[16:39] <sspiff> but I don't see why it would require the API to be changed (I think you said it would need changing for this earlier)
[16:41] <sspiff> ubitux: I've written up an email describing the problem, and asking for suggestions on how to solve it. Is it OK to send without a draft solution? I really can't find a fix that doesn't have many grave pitfalls...
[18:53] <ubitux> sspiff: ofc, feel free to ask any question about ffmpeg development on the ml
[19:49] <llogan> michaelni: more spam. 3122 3123
[19:51] <llogan> michaelni: if you give me the correct permissions i can delete them instead of having to ping you each time
[20:15] <llogan> ubitux: Tip Create a 15 second video sample for testing: ffmpeg -f lavfi -i testsrc=d=15 output.mkv
[20:15] <llogan> or maybe i should use -t 15
[20:16] <ubitux> both are fine
[20:16] <ubitux> =d=15 shows how to use filters options
[20:17] <llogan> downside is that they will not understand why their fave dumb player won't play it
[20:21] <llogan> unless i add format
[20:21] <llogan> maybe this is not such a good tip
[20:22] <ubitux> mmh?
[20:22] <ubitux> why won't it be played?
[20:23] <llogan> yuv444p
[20:23] <llogan> i guess dumb players won't grok mkv anyway?
[20:23] <nevcairiel> quite possibly :p
[20:23] <ubitux> s/mkv/avi/
[20:23] <nevcairiel> although i'm sure there are people that have mkv something support but no 444 h264 support
[20:23] <ubitux> pb fixed
[20:24] <nevcairiel> too bad the default settings of the mpeg4 encoder give such a horrible picture
[20:25] <llogan> i don't really like to encourage avi usage.
[20:26] <ubitux> s/mkv/gif
[20:26] <llogan> heh. i'll come up with something else.
[20:36] <ubitux> llogan: the mkv command displays a warning anyway
[20:37] <llogan> ubitux: good enough for me
[20:38] <llogan> talking about warnings, "deprecated pixel format used, make sure you did set range correctly" is confusing users
[21:08] <michaelni> llogan, both deleted, (it needed root but ive enabled support to allow admins to delete, carl can delete them too, and ive also tried to give you permission to delete tickets, tell me if it doesnt work)
[21:09] <llogan> michaelni: thanks
[23:56] <cone-874> ffmpeg.git 03Michael Niedermayer 07master:ea6331b2c602: avfilter/af_aresample: Calculate output buffer size more precisely.
[00:00] --- Thu Nov 7 2013
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[00:02] <ac_slater> llogan: right I needed a little than strings could give me thanks mate
[00:03] <ac_slater> (ie - nice output :p)
[00:31] <IFN10EBrain> Hello #ffmpeg. I'm trying to make a screencast using ffmpeg but the sound dies after a few seconds of recording. How can I fix this? The mic is a digital integrated one on a notebook.
[00:31] <triggerhapp> I'm trying to work out how to livestream with synchronized audio... currently no matter what I try the audio runs out of sync (specifically, behind) the video - is there something else I could be doing? http://pastebin.com/UPagVsP9
[00:31] <triggerhapp> sounds fun IFN10EBrain, what are you doing for a command atm ?
[00:33] <IFN10EBrain> Its this: "ffmpeg -f alsa -i pulse -f x11grab -r 25 -s 1024x600 -i :0.0 -acodec pcm_s16le -vcodec huffyuv -sameq Screencast.avi". The sound works for a few seconds and dies.
[00:34] <triggerhapp> why use alsa, couldn't you just '-f pulse'?
[00:35] <IFN10EBrain> triggerhapp: pulse errors: pa_simple_new failed: No such entity
[00:36] <IFN10EBrain> triggerhapp: it's an input/output error.
[00:36] <IFN10EBrain> That's when I use -f pulse.
[00:36] <IFN10EBrain> -f alsa kinda works. But the sound is just a few seconds.
[00:47] <IFN10EBrain> Oy oy... I found a solution. It works :)
[00:54] <triggerhapp> Ooh, what'd you try?
[00:54] <IFN10EBrain> triggerhapp: I just got sound to work :s
[00:54] <IFN10EBrain> I need to get video With sound.
[00:55] <IFN10EBrain> I'm trying a mischmash of different things I found on the net.
[00:55] <triggerhapp> I meant using what method :P
[00:55] <triggerhapp> but I can't say much, I'm doing the same trick to try to get my sound to sync :)
[00:56] <IFN10EBrain> triggerhapp: setting -i hw:0,0 worked for sound. But in combination with video...it doesn't work either.
[00:59] <IFN10EBrain> The worst thing about using an extremely powerful tool is not knowing how to use it.
[01:00] <triggerhapp> My issue seems worse. I'm using a powerful tool that seems to be butchered into unusable :)
[01:01] <triggerhapp> having to deal with avconv instead of ffmpeg
[01:07] <llogan> don't use -sameq. it does not mean same quality, makes no sense with a lossless encoder, and has been removed from ffmpeg anyway
[01:07] <triggerhapp> I... did use pastebin?
[01:07] <llogan> ah...i missed it
[01:07] <triggerhapp> :P
[01:08] <llogan> avconv is not from FFmpeg
[01:08] <llogan> if you want help here you'll have to use ffmpeg from FFmpeg
[01:12] <IFN10EBrain> I'll get back to it later. I need to go to sleep. Nite nite.
[01:26] <triggerhapp> llogan, using a static build and it doesn't recognise x11grab as an input format. Short of compiling (since this is intended to be a python app) what could I do?
[01:27] <llogan> triggerhapp: the static builds don't have x11grab support
[01:28] <triggerhapp> that one I could gather :P Short of compiling it myself how can I get a version for ubuntu which will support it?
[01:28] <triggerhapp> any advised PPA's at all?
[01:29] <llogan> you could try https://launchpad.net/~jon-severinsson/+archive/ffmpeg but it only offers 0.10 & 0.7 release branches
[01:30] <llogan> ...or compile http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[01:30] <triggerhapp> The issue with compiling is to then get a compatible version for every user of my script I either drop them a lot of binary libraries and exec, or make them compile too
[01:30] <triggerhapp> In theory I love compiling... just not for a python app
[01:34] <llogan> maybe it would be easiest for you to include a binary so you don't have to deal with various versions, forks, and option changes
[01:34] <llogan> ...and hopefully x11grab will work as expected for everyone.
[01:35] <triggerhapp> And people running on arm? :P
[01:37] <llogan> sure, if they capture 180x120
[01:39] <triggerhapp> The raspberry pi manages 1080p just fine, I'm targetting desktops and pi's right now
[01:41] <triggerhapp> next question... which version is it that filter_complex is in? seems to be missing from this 0.10
[01:44] <triggerhapp> ... I'm gunna sleep actually. Morning shift
[04:59] <hailwood> Hi Guys, is there a concensus on how to best web optimize a video with ffmpeg?
[09:09] <plepere> hello
[14:48] <sspiff> modifying ffmpeg.c to keep the clear packet in a big, dirty static AVPacket dvbsub_clear, and only sending it out as soon as a subtitle packet with a higher PTS occurs "fixes" the transcoding
[14:49] <sspiff> obviously, the last clear will be missing
[14:49] <sspiff> which is (probably? you tell me...) unacceptable
[14:53] <sspiff> oops, wrong channel again!
[16:44] <dharriso> hi, im using the ffmpeg API to transcode AAC video over HTTP in a proxy. this works well. I have a need to specify the AAC sample size up front
[16:44] <dharriso> in an index (moov box) thats sent to the client. However sometimes i need to pad the AAC sample up to the size I initaly specified in the index.
[16:44] <dharriso> In my tests I have padded frames with hex zeroes and it seems to play on some desktop player.
[16:44] <dharriso> Is padding with hex zeroes the best way to acheive this?
[16:49] <dharriso> should i post this question on ffmpeg-devel, though the ffmpeg.org website indicates this is the correct place
[17:16] <sspiff> dharriso: I think this is the right place. Also, people from #ffmpeg-devel are also in this channel, so they'll see it eventually, I hope :)
[19:51] <noobed> hello everybody - there're only 2 IRC channels and since I'm making software that uses FFMPEG I'm writing here.
[19:51] <noobed> Could I ask a couple of questions?
[19:51] <noobed> 1) I think I've extracted the audio stream from an .flv file and successfully wrote the raw audio - it's a 2 channel audio stream and i've used the resampling audio example
[19:51] <noobed> so is there any way I can test it's properly written on my hdd ?
[19:51] <noobed> 2) after extracting the raw audio I want to encode it into mp3 but I get
[19:51] <noobed> "Invalid data found when processing input "
[19:51] <noobed> when running
[19:51] <noobed> avformat_open_input(&avFormatContext, argv[1], NULL, NULL); // where avFormatContext is NULL at that point and argv[1] contains the raw audio file name]
[21:59] <memand> Hey guys, can I make ffplay stream to stdout instead of just streaming to everything on the soundcard?
[22:06] <relaxed> memand: isn't that a job for ffmpeg?
[22:07] <memand> might be, the thing I want to do is to be able to take a music file and output it into aplay so I can choose what sound port it comes out from
[22:10] <memand> relaxed: ^
[22:10] <memand> I tried this 'ffmpeg -i 3938033.mp3 - | aplay -f cd -D front'
[22:11] <memand> but it says:
[22:11] <memand> Unable to find a suitable output format for 'pipe:'
[22:11] <memand> pipe:: Invalid argument
[22:12] <sacarasc> memand: ffmpeg -i blah.mp3 -f pcm -
[22:15] <roothorick> so, curious, why does ffmpeg want yasm/nasm and doesn't just use GNU as?
[22:17] <ShadowJK> syntax
[22:18] <roothorick> what syntax specifically?
[22:19] <ShadowJK> it's less of a headache to write code for nasm syntax than gnu as
[22:19] <roothorick> is it a preprocessor thing?
[22:20] <roothorick> my curiosity was mostly spurred by a bizarre errorspam I've been getting while building 2.1 in mingw-w64, but only on one certain machine which just boggles my mind
[22:20] <relaxed> I don't think ffmpeg needs nasm, only yasm.
[22:22] <roothorick> it's either/or
[22:22] <roothorick> and I couldn't get yasm to compile under mingw so nasm it is
[22:27] <roothorick> the errorspam seemed to suggest that it was trying to assemble 64bit code in a 32bit mode
[22:27] <roothorick> but I'm having no issues on a 32bit mingw on a different 64bit machine
[22:31] <dbro> Can anyone help debug some video encoder output: https://s3.amazonaws.com/dbro/h264_madness/ffmpeg_1383772856149.ts
[22:31] <dbro> [h264 @ 0x7f9be2818400] corrupted macroblock 11 29 (total_coeff=-1)
[22:31] <dbro> [h264 @ 0x7f9be2818400] error while decoding MB 11 29
[22:31] <dbro> working on a FOSS Android library to mate Android's hardware encoder to ffmpeg: https://github.com/OnlyInAmerica/FFmpegTest
[22:36] <noobed> Could some1 help me - I'm getting a raw audio from an .flv file.... afterwards I try to open that raw audio with avformat_open_input() and I'm getting an error " Invalid data found when processing input " ... I've posted a thread on SO http://stackoverflow.com/questions/19823260/c-ffmpeg-decode-into-raw-audio-… ... this's been bugging me for days now :(
[22:38] <mikepack> Does anyone have experience converting .3gp files? I'm having trouble aggregating 3gp files into a compilation with other files types (mainly mp4).
[23:19] <noobed> several ffmpeg/libavdevice/sdl.c:64: undefined reference ..... any clues what library to add in my makefile?
[00:00] --- Thu Nov 7 2013
1
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[00:00] <iive> llogan: ?
[00:06] <llogan> iive: ffmpeg.org is having issues
[00:08] <cone-910> ffmpeg.git 03Stefano Sabatini 07master:988e2e7c1c18: lavfi/ladspa: cast return value of av_x_if_null to char*, fix warnings
[00:08] <cone-910> ffmpeg.git 03Stefano Sabatini 07master:246e323de8a1: lsws/utils: fix typo
[00:14] <ubitux> BBB: ping
[00:35] <Compn> ffmpeg.org still down ?
[00:46] <ubitux> Compn: seems so
[01:00] <Compn> maybe one should mail root@mphq
[01:02] <llogan> how would they get it if it is also the mail server (assuming more than httpd is down)?
[01:03] <michaelni> it seems more is down, i didnt had any luck with reaching other boxes on the subnet
[01:17] <Compn> llogan : i had that wonder too
[01:21] <Compnn> michaelni : any chance for fixing up the bayer rgb patch i pinged about ?
[01:21] <lukaszmluki> Hi, it is just me or ffmpeg.org/trac/ffmpeg/ is not working?
[01:23] <michaelni> Compnn, i guess i should ...
[01:28] <llogan> lukaszmluki: it is not just you
[01:39] <BBB> ubitux: pong sortof
[01:40] <BBB> (will be on and off in bits)
[01:43] <cone-910> ffmpeg.git 03Michael Niedermayer 07master:04047b6166b3: avformat/http: print http headers at AV_LOG_DEBUG level
[01:47] <michaelni> btw, until the server is online again (which i hope will be very soon, but i have no idea what is wrong), dont hesitate to CC patches to me, so i can push them if they are trivial & ok
[01:51] <Compnn> is anyone going to md5 the entire tree to make sure it wasnt backdoored when we werent looking? :P
[01:52] <llogan> i don't think the problem is with the server itself
[01:52] <Compnn> did they launch the university into orbit ?
[01:52] <Compnn> is hungary still a country?
[01:54] <wm4> Compnn: git is already sha-hashed by design, literally every ffmpeg developer would notice if someone tempered with the tree
[01:55] <wm4> the other stuff is a different question of course
[02:02] <llogan> Compnn: did pross make the initial bayer patch?
[02:28] <Compnn> uhm
[02:30] <Compnn> On Sun, 16 Dec 2012 12:48:21 +1100, Peter Ross wrote:
[02:30] <Compnn> >This is an rebase and cleanup of some earlier patches that make
[02:30] <Compnn> i dont know
[02:30] <Compnn> http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2008-March/056719.html
[02:31] <Compnn> [MPlayer-dev-eng] [PATCH] add 8 bit Bayer format and filter
[02:31] <Compnn> Guennadi Liakhovetski g.liakhovetski at pengutronix.de
[02:32] <Compnn> i guess it was written as a filter back in 2008
[02:32] <Compnn> for mplayer
[02:34] <llogan> oh. thanks.
[02:41] <Compnn> 8bit bayer should be listed in v4l docs
[02:41] <Compnn> i have no idea what r3d uses with bayer
[02:46] <Paranoialmaniac> nevcairiel: the isom and mov spec doesn't forbid that audio has stss atom. stss atom just marks random access points. well, more restrictions depend on derived specifications though.
[07:26] <burek> ffmpeg.org seems to be down
[07:27] <Compnn> yes.
[07:28] <ubitux> still down? meeh
[07:28] <ubitux> :(
[09:31] <ubitux> yay it's back
[09:35] <ubitux> smtpd still down?
[09:35] <ubitux> or maybe under heavy load :p
[10:31] <cone-557> ffmpeg.git 03Diego Biurrun 07master:e2b5b097898c: x86: rv40dsp: Use PAVGB instruction macro where appropriate
[10:31] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:934e489ee8f4: Merge commit 'e2b5b097898c9155f4bdff4d83cdc54d5eef6930'
[11:09] <ubitux> yay mail reup
[11:19] <michaelni> mailman had a stale lock ...
[11:20] <cone-557> ffmpeg.git 03Diego Biurrun 07master:70cbf33405f5: vdpau: Add missing #includes to fix standalone header compilation
[11:20] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:e1459f4b0bb2: Merge commit '70cbf33405f50dfaf77e85f382a188acf17dc71a'
[12:14] <cone-557> ffmpeg.git 03Diego Biurrun 07master:48715285f69d: graph2dot: Fix use of deprecated API
[12:14] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:de970d49fede: Merge commit '48715285f69d70879353737d7bed049f89f58096'
[12:17] <cone-557> ffmpeg.git 03Diego Biurrun 07master:82c86d0b1839: graph2dot: Add missing #include for av_get_channel_layout_string()
[12:17] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:6a7de2361b94: Merge commit '82c86d0b1839b01b3cd0dd848fb6416ff21506dd'
[12:25] <cone-557> ffmpeg.git 03Diego Biurrun 07master:7d1a68e95baf: avfilter/buffer: Suppress deprecation warning for avfilter_unref_buffer
[12:25] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:be74cb5159ef: Merge remote-tracking branch 'qatar/master'
[13:11] <sspiff> I'm trying to figure out how the demuxing/decoding of dvb subtitles works, but I can't seem to figure it out.
[13:13] <sspiff> I'm looking at libavformat/mpegts.c, which is used for TS demuxing, but the only subtitle specific bit I can find is in ff_parse_mpeg2_descriptor, and it doesn't seem to pass through the subtitle case for my test streams.
[13:14] <kierank> dvb subtitling is done through a descriptor
[13:15] <kierank> and there's a table somewhere in the demux
[13:15] <sspiff> kierank: StreamType DESC_types[] { ... } is what you're talking about?
[13:16] <kierank> probably
[13:16] <kierank> can't remember the name
[13:17] <sspiff> looks like it, so these entries contain an AVCodecID, which I presume is used to identify the codec to pass the packets to?
[13:18] <kierank> yes
[13:21] <ubitux> saste: static const in the ScaleAlgo patch maybe
[13:22] <sspiff> kierank: this is used in ff_parse_mpeg2_descriptor, which I was looking at (and only there according to cscope), and it doesn't enter the switch-case clause that handles the subtitles
[13:25] <cone-557> ffmpeg.git 03Paul B Mahol 07master:7722ae35eb63: avcodec/c93: stop using deprecated avcodec_set_dimensions
[13:28] <cone-557> ffmpeg.git 03Paul B Mahol 07master:1a244c22ea83: avcodec/brender_pix: stop using deprecated avcodec_set_dimensions
[13:39] <cone-557> ffmpeg.git 03Paul B Mahol 07master:22c00962c283: avcodec/flashsv: use init_get_bits8()
[13:39] <cone-557> ffmpeg.git 03Paul B Mahol 07master:6e094f702639: avcodec/flashsv: stop using deprecated avcodec_set_dimensions
[13:39] <cone-557> ffmpeg.git 03Paul B Mahol 07master:290be34a745f: avcodec/brender_pix: stop calling av_image_check_size() twice
[13:41] <sspiff> never mind, I was wrong - turns out #define DEBUG 1 didn't turn on the av_dlog prints
[13:41] <durandal_1707> where you put it ?
[13:43] <ubitux> durandal_1707: ...
[13:43] <ubitux> - init_get_bits(&gb, avpkt->data, buf_size * 8);
[13:43] <ubitux> + if ((ret = init_get_bits(&gb, avpkt->data, buf_size)) < 0)
[13:43] <ubitux> where is the 8?
[13:43] <durandal_1707> awwwwwww
[13:45] <cone-557> ffmpeg.git 03Paul B Mahol 07master:ad1dfe04a1bf: avcodec/tscc2: use init_get_bits8()
[13:45] <cone-557> ffmpeg.git 03Paul B Mahol 07master:c3311515d513: avcodec/flashsv: fix typo
[13:46] <ubitux> thx
[13:48] <sspiff> durandal_1707: just under the header comment, before the includes
[13:48] <durandal_1707> sspiff: and what file is that?
[13:49] <durandal_1707> looks like atrac+ decoder is lost because of Don
[13:51] <durandal_1707> by insisting in putting stuff that only one decoder uses in separate .c file
[14:03] <cone-557> ffmpeg.git 03Stefano Sabatini 07master:2a6c95d85f98: lsws/utils: introduce scale_algorithms array
[14:05] <Compnn> don moir ?
[14:07] <durandal_1707> Compnn: Don Corleone
[14:10] <sspiff> durandal_1707: mpegts.c
[14:11] <Compnn> durandal_1707 : i dont see him in the list?
[14:11] <Compnn> maybe i should go back further
[14:12] <durandal_1707> sspiff: and no av_dlog appear?
[14:12] <sspiff> durandal_1707: none whatsoever
[14:12] <sspiff> replacing them by av_log(..., 0, ...) makes it work
[14:13] <sspiff> so there's something strange going on with the DEBUG
[14:13] <durandal_1707> perhaps you do not use -v debug?
[14:13] <Compnn> durandal_1707: i dont see don careone name anywhere in ffmpeg or libav list ?
[14:13] <Compnn> r carleone
[14:14] <durandal_1707> Compnn: watch film and you will see it
[14:15] <sspiff> durandal_1707: I was unaware :(
[14:15] <Compnn> oh lol
[14:20] <durandal_1707> 'spaghetti factor'
[14:21] <cone-557> ffmpeg.git 03Paul B Mahol 07master:b79f543c5869: avcodec/mss1: use init_get_bits8()
[14:21] <cone-557> ffmpeg.git 03Paul B Mahol 07master:82f5c4d737ca: avcodec/flashsv: use av_freep() for tmpblock
[14:21] <cone-557> ffmpeg.git 03Paul B Mahol 07master:e3f384b42835: avcodec/mss2: use init_get_bits8()
[14:38] <cone-557> ffmpeg.git 03Lenny Wang 07master:668255479085: avutil/opencl: compile kernels separately
[14:38] <cone-557> ffmpeg.git 03Lenny Wang 07master:89a3be8921e2: avfilter/opencl: compile kernels separately
[15:55] <sspiff> ubitux: dvb-to-dvd subtitle transcoding was also broken, right?
[15:59] <ubitux> sspiff: problably, i admit i didn't try
[15:59] <sspiff> ubitux: it works here
[16:00] <sspiff> (using my system ffmpeg, not built from source)
[16:00] <ubitux> ah, surprising, ok
[16:00] <ubitux> did you find the root cause of the problem?
[16:00] <sspiff> transcoding to theora/dvdsub mkv's works for me
[16:00] <sspiff> no
[16:00] <sspiff> still looking
[16:01] <sspiff> but I've been looking in the decoding/demuxing, which is probably fine since dvdsub transcoding works?
[16:01] <ubitux> honestly i don't know
[16:01] <sspiff> neither do I :)
[16:01] <sspiff> I'll continue looking
[16:02] <ubitux> dvb -> dvb remux only or transcoding are broken
[16:02] <ubitux> dvd -> dvb works
[16:02] <sspiff> does ffmpeg decode/re-encode for dvb->dvb?
[16:02] <ubitux> yes
[16:02] <sspiff> and dvb -> dvd works
[16:02] <ubitux> you can can't convert without transcode
[16:03] <sspiff> well you don't really convert anything if you stick with dvb subtitles, unless you want to change the clocks etc?
[16:03] <kierank> you can lower the colour depth iirc
[16:03] <sspiff> ah I see
[16:46] <ubitux> arg multistream sami
[16:46] <ubitux> why the hell do ppl still use sami?
[17:39] <michaelni> mateo`_, as you are movenc maintainer you should have op here
[19:37] <cone-834> ffmpeg.git 03Clément BSsch 07master:87434cf37340: avcodec/vp9: add ff_vp9_idct_idct_{4x4,8x8}_ssse3().
[19:56] <ubitux> no one else wants to work on vp9?
[20:30] <llogan> anyone find out what the deal was with the server connection?
[20:53] <Compnn> llogan : why not ask arpi ?
[21:06] <michaelni> llogan, Compnn long power break
[22:06] <cone-834> ffmpeg.git 03Diego Biurrun 07master:92352a94c165: configure: Move -Wno-maybe-uninitialized check into gcc section
[22:06] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:6cfaf9cde875: Merge remote-tracking branch 'qatar/master'
[23:39] <BBB> durandal maybe?
[23:42] <BBB> ubitux: you don't like vp9 anymore? ;( :P
[23:57] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:66872161cbb7: avutil/common: error out with clear message if __STDC_CONSTANT_MACROS is not defined with c++
[00:00] --- Wed Nov 6 2013
1
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[00:01] <makar> would any of you recommend a simple command line tool to demux a ts, that is, to extract a single PID from a ts?
[00:06] <foucist> hey folks, i encoded some new f4v videos at 640x360, but when i don't have stretch on in jwplayer, it shows up considerably smaller. you can see the ffmpeg -i for the new & old videos here: https://gist.github.com/foucist/137ba7f5ea796c0cad72
[00:07] <lvella> hey guys, do you know if I buy one of these, I'll be able to play blyrays in linux?
[00:07] <lvella> http://www.aliexpress.com/item/Special-price-free-shipping-USB-2-0-External…
[00:08] <llogan> foucist: that's not our ffmpeg.
[00:08] <llogan> we don't serve their kind here
[00:16] <foucist> llogan: well, i'll try asking in #libav, but i'm just figurng that the issue is a lack of general encoding knowledge rather than differences between the two.. i.e. the fact that the old videos have "[PAR 1:1 DAR 16:9]" where the new don't for example? or something to do with framerate?
[00:18] <llogan> hard to say without your command or console output, but that's a moot point since you're not using ffmpeg
[07:55] <burek> :)
[07:56] <burek> i was replying to a post on the forum, and I was too lazy to type over and over again the same answer :)
[08:53] <tiksa> hi guys. I should do some video editing with ffmpeg and I'm asking if it's possible to make an ffmpeg one-liner with -filter_complex that includes video trimming, concatenating, overlay videos/images, text and so on? what do you think?
[08:53] <tiksa> is it realistic?
[09:06] <Ronald_Reagan> I've encoded a video with these settings: -c:v libx264 -preset slower -crf 14 -c:a copy video.mkv Adobe products cannot open this it seems. I would've thought they could. What sort of options would I use to make a file that I can open? Totally new to this video encoding stuff.
[09:12] <viric> tiksa: I want to do something similar. That's what mlt seems suited for...
[09:12] <viric> but it takes a lot of ram, and I couldn't find how to set the encoder parameters in it
[10:01] <tiksa> ok
[10:02] <tiksa> Next question.. How can I continue filter graph (?) after concat? I have something like "... concat=n=3:v=1:a=1 [a] [v] ' -map '[a]' -map '[v]'" Now I should add an overlay image, for example, to the output. Thanks for any help..
[10:03] <tiksa> I wasn't able to find anything from google so far
[10:46] <bouba> hi there, anyone's able to give me tips about this plz ? http://pastebin.com/wmmDPD72 ; i just don't understand which param is not accepted by x264 (most probably in output)
[10:47] <bouba> the only thing which is not common is the INPUT file framerate flags
[12:37] <Baldino> Hi, is there any way to detect/select the forced subtitles of a subtitle stream in ffmpeg ?
[13:07] <sspiff> I'm trying to figure out how the demuxing/decoding of dvb subtitles works, but I can't seem to figure it out.
[13:07] <sspiff> I'm looking at libavformat/mpegts.c, which is used for TS demuxing right?
[13:09] <bencoh> right
[13:09] <bencoh> have a look at libavcodec/dvbsubdec
[13:10] <JEEB> sspiff, if you want to talk internals of libavformat, I recommend switching to the -devel channel
[13:10] <JEEB> yeah, the libavcodec/dvbsubdec is the decoder
[13:10] <JEEB> libavformat/mpegts is the MPEG-TS demuxer
[13:11] <sspiff> JEEB: oops, wrong tab!
[13:11] <sspiff> bencoh, JEEB: thanks
[13:11] <dharriso> Hi, Im using hthe ffmpeg API to re-encode video over a network stream (HTTP). Is it possible to pad out AAC audio samples to a specific size? Is it a case of appending hex 0 to samples or is there a construct for AAC like AVC Filler NAL to tell the player to ignore this data. I predict frame sizes up front for MP4 files to stream this content via HTTP
[13:11] <dharriso> thanks
[15:38] <dharriso> hi my connection dropped
[15:39] <dharriso> was there any answer to my question earlier in realtion to padding AAC frames to a specific size. Is it possible?
[15:57] <dharriso> shold i try the ffmpeg-devel list
[16:15] <tiksa> dharriso: no, there wasn't any answer
[16:20] <dharriso> tiksa: have sent same question to mailing list, thanks
[18:10] <sspiff> Is there a way I can convince ffmpeg to preserve the original pts'es?
[18:18] <Mavrik> sspiff, probably with setpts filter?
[18:40] <sspiff> Mavrik: thanks
[20:46] <llogan> michaelni: more spam 3121 3122
[20:46] <llogan> that was meant for -devel...
[20:59] <maujhsn> Is this command correct: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -c:a pcm_s16le -c:v libx264 -preset ultrafast -threads 0 ~/Desktop/music.mkv?
[21:01] <llogan> maujhsn: does the output look ok for you?
[21:02] <llogan> change -r to -framerate and -s to -video_size
[21:02] <llogan> (for x11grab)
[21:21] <renihs> ffmpeg stops recording audio when i select higher resolutions, e.g ffmpeg -f x11grab -r 30 -threads 6 -i :0.0 -f alsa -i hw:0,0 -acodec flac -vcodec libx264 bla.mkv works fine
[21:21] <renihs> but once i add a resolution like -s 1680x1050 -> no audio captured, not sure if am doing something dumb or even right channel though :)
[21:22] <renihs> irc channel i mean :)
[21:25] <renihs> i only have "something distorted" for like 1/2 second on the beginning of the video
[21:32] <renihs> llogan, thanks
[21:32] <renihs> http://bpaste.net/show/147080/
[21:34] <llogan> renihs: does this help? ffmpeg -f x11grab -framerate 30 -video_size 1680x1050 -i :0.0 -f alsa -i hw:0,0 -acodec copy -vcodec libx264 -preset ultrafast -qp 0 bla.mkv
[21:36] <renihs> llogan, with your line, the "audio distortion" at th beginning is less stretched
[21:36] <renihs> and its definitive audio from what should be captured, just speed up and shortened to like 1/2s
[21:37] <renihs> http://bpaste.net/show/147083/ paste with sound
[21:37] <renihs> with lower resolution
[21:40] <llogan> renihs: using git master is preferred over (old) release branches
[21:41] <llogan> also see https://trac.ffmpeg.org/ticket/615
[21:41] <llogan> running two ffmpeg processes may help (one for audio, one for video). not really optimal solution though.
[21:43] <renihs> yup thats my case :)
[21:43] <renihs> hmmm
[21:43] <renihs> good idea
[21:45] <llogan> i wonder if arecord works normally when you're using ffmpeg to capture video (probably does, but i'm not sure)
[21:49] <renihs> i guess it does
[21:49] <renihs> i only set arecord -D front > receives stuff :p
[21:49] <renihs> when recording and playing audio
[21:49] <renihs> video recording
[21:50] <renihs> hmm well i guess i have to eithr reduce resolution or use 2 captures and merge them afterwards or like
[21:51] <llogan> if you choose to merge you can see if you can still increase framerate. screencasts look nice at higher frame rates (-framerate 60), IMO
[21:52] <llogan> and use the lossless example i gave you. then when you mux you can re-encode if you need to
[21:53] <renihs> actually, it libx264 related
[21:53] <renihs> if i capture for example with ffvhuff
[21:53] <renihs> it captures normally
[21:54] <llogan> i guess that's a solution then
[21:54] <renihs> will try with 60, problem is i am trying to capture sc2, diablo3 and stuff like that from wine to show off how greatly the csmt patches improve performance
[21:54] <renihs> not want to bump it too much :)
[21:55] <llogan> also experiment with huffyuv and ffv1. one of those might perform better than others.
[21:57] <renihs> llogan, thanks
[21:57] <renihs> i sadly know nothing about the codecs around atm
[21:57] <renihs> 60fps seem to be no problem though
[21:58] <llogan> using -r or -framerate?
[21:58] <renihs> grmbl, didnt build ffmpeg with vpx :p
[21:58] <renihs> -r
[21:58] <renihs> lemme read man
[21:59] <llogan> see https://trac.ffmpeg.org/ticket/3045
[21:59] <renihs> dont fine the -framerate option
[21:59] <renihs> s/fine/find/g
[22:00] <renihs> oh hmm
[22:00] <llogan> man ffmpeg-devices
[22:01] <renihs> oh hmm will be using -framerate
[22:02] <renihs> recompiling ffmpeg first :)
[22:02] <llogan> git master i hope
[22:02] <renihs> cant really view the raw video
[22:03] <llogan> with what player?
[22:03] <renihs> llogan, um, no i will do that in a seperate prefix
[22:03] <renihs> mplayer2, will need to rebuild that too ;p
[22:03] <renihs> among some other deps
[22:03] <renihs> i touch ffmpeg like a raw egg tbh :)
[22:04] <llogan> i thought people used mpv these days (i usually just use ffplay or mplayer)
[22:05] <renihs> oh, i guess i will fetch that :)
[22:05] <renihs> didnt even know about it
[22:06] <renihs> hmm fmpeg -threads 8 -y -i bla.mkv -c:a libvorbis -q:a 3 -c:v libvpx -b:v 2000k bla.webm
[22:06] <renihs> doesnt care about threads :p
[22:06] <renihs> pure single core getting beaten
[22:06] <renihs> ... poor
[22:06] <llogan> because you're appling threads to the input
[22:06] <renihs> omg :)
[22:06] <llogan> http://trac.ffmpeg.org/wiki/vpxEncodingGuide
[22:07] <renihs> so much fail :)
[22:07] <renihs> grmbl
[22:07] <llogan> but i know nothing of that encoder or how it deals with threads
[22:07] <renihs> in my defense i was chasing a "missing audio channel" for some time until i realised it doesnt get captured on that resolution
[22:07] <renihs> yeah i just like it because it will work in frickelfox and whatnot everwhere
[22:07] <llogan> it's a bug
[22:08] <renihs> well it performs slightly better now
[22:10] <renihs> grmbl, guess need todo more reading, now no sound anymore in output file
[22:10] <renihs> i have to admit, the builtin screen recorder in gnome3 was nifty :)
[22:58] <viric> I'm trying the mkvtoolnix Headers Editor (in mmg)... It only allows me to set the title
[22:58] <viric> While here I see dozens of more tags: http://wiki.webmproject.org/webm-metadata/global-metadata
[22:58] <viric> How is it?
[23:14] <sacarasc> viric: This channel isn't affiliated with mkvtoolnix. But you can use ffmpeg to add meta data.
[23:14] <viric> right, but that requires remuxing.
[23:15] <viric> I asked here, because maybe there are some restrictions on metadata tags that I don't know (but mmg knows :)
[23:15] <viric> and maybe someone knows too.
[23:25] <CarlosTrading1> hi
[23:26] <CarlosTrading1> i am having this error Codec AVOption g (set the group of picture (GOP) size) specified for input file #0 (http://xxxxxxxxxxxxxxxx:xxxx/) is not a decoding option.
[23:26] <CarlosTrading1> does anyone know ?
[23:30] <ac_slater> Hey guys is there a simple way to retrieve the encoded bitrate of a file? I know that sounds weird but I have some video files and I'm curious what settings the person used to encode then with x264. Maybe I should ask elsewhere
[23:32] <sacarasc> Use the programme MediaInfo, it will tell you all the options used by x264 to encode the file.
[23:34] <ac_slater> sacarasc: what about one for linux?
[23:34] <ac_slater> oh nevermind
[23:34] <ac_slater> it's available for linux too
[23:34] Action: sacarasc nods.
[23:35] <ac_slater> ;) thanks
[23:55] <llogan> CarlosTrading1: how do you get the error?
[23:56] <llogan> ac_slater: you don't need mediainfo. "strings input.mp4 | grep x264"
[00:00] --- Wed Nov 6 2013
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