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June 2012

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ffmpeg-devel.log.20120629
by burek 30 Jun '12

30 Jun '12

[00:01] <CIA-41> ffmpeg: 03Alex Converse 07master * ra112822597 10ffmpeg/ (5 files in 2 dirs): movenc: Add channel layouts for PCM. [00:01] <CIA-41> ffmpeg: 03Luca Barbato 07master * r1cb34ea4fe 10ffmpeg/libavformat/flvenc.c: flvenc: K&R formatting cosmetics [00:01] <CIA-41> ffmpeg: 03Diego Biurrun 07master * r433492ac65 10ffmpeg/doc/git-howto.texi: doc: git: Add checklist with test steps to perform before pushing [00:01] <CIA-41> ffmpeg: 03Damien Fetis 07master * rb92c7ee662 10ffmpeg/libavformat/ (flv.h flvdec.c flvenc.c): [00:01] <CIA-41> ffmpeg: flv: add support for G.711 [00:01] <CIA-41> ffmpeg: Signed-off-by: Luca Barbato <lu_zero(a)gentoo.org> [00:01] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4453f6b861 10ffmpeg/: (log message trimmed) [00:01] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [00:01] <CIA-41> ffmpeg: * qatar/master: [00:01] <CIA-41> ffmpeg: flv: add support for G.711 [00:01] <CIA-41> ffmpeg: doc: git: Add checklist with test steps to perform before pushing [00:01] <CIA-41> ffmpeg: flvenc: K&R formatting cosmetics [00:01] <CIA-41> ffmpeg: movenc: Add channel layouts for PCM. [00:02] Action: Nickname123 is no longer away : Gone for 19 hours 26 minutes 20 seconds [01:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4e82bdea60 10ffmpeg/doc/git-howto.texi: [01:22] <CIA-41> ffmpeg: git-howto: partial rewrite of the push checklist to make it match sanity & reality. [01:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7c8b5d1d80 10ffmpeg/Makefile: [01:22] <CIA-41> ffmpeg: Makefile: remove checkheaders from the main check target [01:22] <CIA-41> ffmpeg: checkheaders doesnt pass and noone has even noticed since a very [01:22] <CIA-41> ffmpeg: long time. [01:22] <CIA-41> ffmpeg: checkheaders is also unmaintained (please add yourself to MAINTAINERS [01:22] <CIA-41> ffmpeg: if you want to maintain it) [01:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rd007f963c2 10ffmpeg/doc/git-howto.texi: [01:22] <CIA-41> ffmpeg: git-howto: cleanup, remove unreasonable recommendition. [01:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r8bff1d7cb0 10ffmpeg/doc/git-howto.texi: [01:22] <CIA-41> ffmpeg: git-howto: remove inconvenient and odd 24h limit on rsync [01:22] <CIA-41> ffmpeg: This was never true for FFmpeg in this form. [01:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc103dc0b48 10ffmpeg/doc/git-howto.texi: [01:22] <CIA-41> ffmpeg: git-howto: replace confusing and incorrect text about he testsuite by mostly a correct text. [01:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:24] <durandal_1707> "... especially these days where a lot of things are in dubious state due to the forks issues." [01:34] <durandal_1707> heh, did anyone tried to fix those valgrind fate errors?, it seems to be in lavfi... [01:39] <michaelni> i ignored them as i thought they would disappear after some patches from nicolas [01:40] <michaelni> ill mail nicolas and check if he has something that will fix it or is working on it before ill look [02:11] <Zeranoe> I'm wondering if anyone can recommend a good guide for creating git patches? The guide at http://www.mplayerhq.hu/DOCS/tech/patches.txt doesn't really cover using git branches and such. (I'm looking to submit a few patches for FFmpeg and I want to get the right formating). [02:13] <durandal_1707> Zeranoe: because that is for mplayer and mplayer do not use git but svn [02:13] <Zeranoe> durandal_1707: Right, but it is still recommend to read here: http://ffmpeg.org/contact.html [02:14] <Zeranoe> "have a look at the MPlayer patch guidelines most of what is written there applies to FFmpeg as well" [02:14] <durandal_1707> that is from days when FFmpeg was using svn [02:14] <llogan> the instrucitons for VLC are somewhat helpful: http://wiki.videolan.org/Git [02:14] <Zeranoe> Which is why I'm looking for another guide [02:15] <Zeranoe> llogan: Thank you [02:15] <j-b> http://wiki.videolan.org/Git#Submitting_patches_to_the_vlc-devel_or_x264-de… [02:15] <durandal_1707> http://ffmpeg.org/developer.html#Developers-Guide [02:16] <durandal_1707> doc/git-howto.txt [02:16] <llogan> Zeranoe: did you get my message a week or two back about libutvideo crashing with your build? [02:17] <llogan> i haven't tested it since [02:18] <Zeranoe> llogan: I don't believe I did... Did you file a bug report? [02:18] <llogan> no, because it worksforme in linux [02:19] <Zeranoe> What was the error? [02:19] Action: llogan fires up VM [02:20] <Zeranoe> I haven't personally used utvideo yet [02:28] <llogan> Zeranoe: no ffmpeg error. just a windows dialog pops up "ffmpeg.exe has stopped working". [02:28] <llogan> ffmpeg -i input -an -c:v libutvideo out.mkv [02:29] <llogan> using latest 64-bit static [02:30] <llogan> 10.2.4 is from january and i only tested with 11.1.0 on linux. [02:33] <Zeranoe> llogan: Thanks for the report, I'll dig into it soon. I'm working through a few things before I'll be able to get to it though. [02:33] <Zeranoe> utvideo seemed to compile fine [04:00] <CIA-41> ffmpeg: 03Paul B Mahol 07master * r8fbe11e623 10ffmpeg/ (9 files in 2 dirs): [04:00] <CIA-41> ffmpeg: Replace Libav with FFmpeg in license headers for files created by me [04:00] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [07:57] <rik316> elo [11:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r84d73e9d5d 10ffmpeg/libavcodec/dca.c: [11:37] <CIA-41> ffmpeg: dca: fix project reference in table name [11:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc496224374 10ffmpeg/libavutil/pixdesc.h: [11:37] <CIA-41> ffmpeg: pixdesc: fix project reference in comment [11:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rbe4ffb28b3 10ffmpeg/libavfilter/vf_ass.c: [11:37] <CIA-41> ffmpeg: vf_ass: fix table name to refer to correct lib. [11:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc83ed18d45 10ffmpeg/libavfilter/formats.h: [11:37] <CIA-41> ffmpeg: formats.h: fix project reference in comment [11:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r491846e4be 10ffmpeg/libavutil/avutil.h: [11:37] <CIA-41> ffmpeg: avutil: fix project name reference in doxy section [11:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [12:52] <saste> how it is possible to distinguish packed/planar format in a PCM file? [12:52] <saste> assuming that the codec used is the same for both "modes" -> av_get_pcm_codec [12:53] <saste> this is causing a problem to lavfi+aevalsrc, aevalsrc issues planar output but the device thinks it is packed [12:53] <saste> and fails [12:56] <saste> or said in other words, why only a few PCM planar variants do exist? [12:56] <saste> michaelni: ^^ [13:00] <jonsen_> is any magic required to stream .mov/h264 via .asf to windows media player? [13:00] <jonsen_> can't get it working with ffserver [13:01] <jonsen_> opening with browser takes that long as the stream isn't finished (header issue?) [13:01] <jonsen_> opening directly via WMP opens but shows nothing [13:25] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r4674718203 10ffmpeg/ffplay.c: [13:25] <CIA-41> ffmpeg: ffplay: fix wrong reference to function in audio_decode_frame() comment [13:25] <CIA-41> ffmpeg: The comment now references swr_convert(), rather than audio_convert(), [13:25] <CIA-41> ffmpeg: which was deprecated and/or dropped. [13:25] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r4fd07b9366 10ffmpeg/ffplay.c: [13:25] <CIA-41> ffmpeg: ffplay: avoid useless NULL checks in swr_free() [13:25] <CIA-41> ffmpeg: swr_free() already checks for nullness, no need to add the check in [13:25] <CIA-41> ffmpeg: calling code. [13:32] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * re8e733adcc 10ffmpeg/libavcodec/dca.c: [13:32] <CIA-41> ffmpeg: dca: favor native over ffmpeg in table name [13:32] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:51] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r567eb9d344 10ffmpeg/doc/muxers.texi: [14:51] <CIA-41> ffmpeg: lavf/segment: fix command with missing -list option [14:51] <CIA-41> ffmpeg: Replace -list with the correct option -segment_list. [14:52] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rd9355a03f2 10ffmpeg/libavformat/hls.c: [14:52] <CIA-41> ffmpeg: lavf/applehttp: add log message in case of applehttp_read_header() failure [14:52] <CIA-41> ffmpeg: Improve error reporting. [14:52] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r0692d4c890 10ffmpeg/libavformat/segment.c: lavf/segment: add some debugging logs [14:57] <Compn> The search warrants used by police to raid the New Zealand home of Megaupload founder Kim Dotcom have been ruled illegal. In addition, the data that was sent to the FBI was ruled to be unlawfully obtained. [14:57] <CIA-41> ffmpeg: 03Lou Logan 07master * r6851130fd6 10ffmpeg/ (25 files in 2 dirs): [14:57] <CIA-41> ffmpeg: cosmetics: minor libavcodec spelling errors [14:57] <CIA-41> ffmpeg: Also update some common misspelled words in patcheck [14:57] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:57] <CIA-41> ffmpeg: 03Martin Sliwka 07master * rdf531b0e10 10ffmpeg/libavformat/file.c: (log message trimmed) [14:57] <CIA-41> ffmpeg: avformat: disable seeking on FIFOs/named pipes [14:57] <CIA-41> ffmpeg: Patch is addition to my previous patch [14:57] <CIA-41> ffmpeg: (https://lists.ffmpeg.org/pipermail/ffmpeg-cvslog/2012-June/051590.html) [14:57] <CIA-41> ffmpeg: and disables seeking on FIFOs/named pipes by setting [14:57] <CIA-41> ffmpeg: URLContext::is_streamed (same as pipe: protocol does for stdin/stdout pipes) [14:57] <CIA-41> ffmpeg: Fixes Ticket986 [15:52] <ubitux> saste: it seems your patch doesn't work if you don't do "-f stream_segment,ssegment" [15:53] <ubitux> (it looks like -f stream_segment or -f ssegment doesn't work) [15:54] <ubitux> also, why did you change the -c copy into -codec copy? :P [15:55] <ubitux> not that it matters much but well... [16:06] <saste> ubitux: order of application of patches matters [16:06] <saste> which patch doesn't work? [16:07] <saste> 1) lavf: allow multiple names in output devices selected by av_guess_format() [16:07] <saste> 2) lavf/segment: add stream_segment variant of the segment muxer [16:37] <jonsen_> i troubleshoot the whole day now just to get multicast streaming to work [16:38] <jonsen_> which container and codec is well suited for that? [16:39] <kierank> mpeg-ts and h264 [16:39] <jonsen_> and audio? [16:39] <ubitux> saste: oh, right ok [16:39] <jonsen_> kierank: it's really tricky to do all this right [16:39] <kierank> aac [16:39] <kierank> jonsen_: what are you trying to do [16:40] <ubitux> saste: you should send them as patchset :) [16:40] <kierank> i run a number of multicast streams [16:41] <saste> jonsen_: don't forget to update http://ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide when you're finished ;-) [16:48] <jonsen_> kierank: trying to get ffserver stable to work. Should stream via Multicast. Would love to see it working on the Windows-7 clients in my company. I tried RTP with many different options (maybe not the right ones), tried several containers and codecs, but stream was either not stable or it didn't work at all. Windows Media Client never worked with RTP/Multicast, just with Unicast/HTTP and that never stable. Also the SDP extension for the multicast add [16:48] <Compn> jonsen_ : unfortunately, no one really has worked on getting ffmpeg to work with wmp [16:49] <Compn> lu_zero is the ffserver maintainer, you may want to ask him... [16:49] <Compn> or wait for michaelni [16:49] <Compn> also i'm a multicast newbie [16:50] <Compn> i know all other stream stuff somewhat :) [16:50] <Compn> jonsen_ : did you try gstreamer ? [16:50] <jonsen_> from Transport Layer perspective it's the same as Streaming via RTP/UDP Unicast [16:50] <jonsen_> just unidirectional [16:51] <jonsen_> just opening URLs to get the Multicast address is something special [16:51] <jonsen_> the whole stuff is really painful [16:52] <jonsen_> Compn: no, never tried [16:52] <michaelni> Compn, lu_zero is not ffserver maintainer [16:52] <ubitux> he is just a avserver mentor for gsoc afaict [16:53] <jonsen_> Compn: is it worth to have a look? [16:53] <Compn> michaelni : oh, i am mistaken then :\ [16:54] <Compn> jonsen_ : well, what are you trying to accomplish end goal ? [16:54] <michaelni> Compn, np :) [16:54] <jonsen_> I could also just use VLC to stream to the network [16:54] <Compn> just streaming to wmp ? [16:54] <Compn> vlc is also good [16:54] <jonsen_> Compn: yup, but via Multicast [16:54] Action: Compn never learned multicast :\ [16:55] <jonsen_> the Multicast is no problem for me [16:55] <michaelni> btw, for the record lu_zero has 2 commits in ffserver.c out of 467 commits total in ffserver.c [16:55] <jonsen_> and for the software neither [16:56] <jonsen_> Transport Layer is important [16:56] <jonsen_> udp/rtp tcp/http and so on [16:56] <jonsen_> in the first place without having any knowledge about ffmpeg I thought that it would be right tool for professional streaming purposes [16:57] <Compn> it is. wmp is the non-professional tool ;) [16:57] <Compn> ehe [16:57] <jonsen_> unfortunately there is no way around wmp [16:58] <jonsen_> i want to stream to 20k users, 19900 of them have WMP :-( [16:59] <Compn> ah [17:11] <jonsen_> argh isnt working again with VLC [17:11] <jonsen_> ffmpeg -i Prototype1920.ts -f mpegts rtp://239.1.2.3:5000 [17:11] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * ra9a7e215e5 10ffmpeg/libavformat/ (allformats.c segment.c version.h): [17:11] <CIA-41> ffmpeg: lavf/segment: add stream_segment variant of the segment muxer [17:11] <CIA-41> ffmpeg: This simplifies usage for segment streaming formats with no global [17:11] <CIA-41> ffmpeg: headers, tipically MPEG 2 transport stream "ts" files. [17:11] <CIA-41> ffmpeg: The seg class duplication is required in order to avoid an infinite loop [17:11] <CIA-41> ffmpeg: in libavformat/utils.c:format_child_next_class(). [17:11] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r3cd4f9fd76 10ffmpeg/libavformat/ (utils.c version.h): [17:11] <CIA-41> ffmpeg: lavf: allow multiple names in output devices selected by av_guess_format() [17:11] <CIA-41> ffmpeg: Consistent with av_find_input_format(). [17:11] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rc1abfbc47c 10ffmpeg/libavformat/segment.c: [17:11] <CIA-41> ffmpeg: lavf/segment: rename segment private context from "c" to "seg" in segment_start() [17:11] <CIA-41> ffmpeg: Consistent with the rest of the file, less confusing. [17:11] <jonsen_> nor that ffmpeg -i Prototype1920.ts -sn -an -re -f rtp rtp://239.1.2.3:5000 [17:12] <jonsen_> uh, and the rtp muxer can only stream one stream [17:12] <jonsen_> so this one is showstopper anyway [17:14] <saste> jonsen_: then use multiple outputs [17:15] <jonsen_> that works? [17:15] <jonsen_> i mean how does a client differentiate those streams? [17:16] <jonsen_> those will come from different source ports [17:16] <jonsen_> will give it a try [17:16] <jonsen_> but at the moment just nothing works :D [17:18] <jonsen_> just for the sake that anyone cares: http://de.pastebin.ca/2165549 [17:19] <jonsen_> for today I give up now [17:19] <jonsen_> took me 10 hours [17:46] <saste> michaelni: any hint on the pcm planar -> codecid mapping problem i mentioned this morning? [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * ra19e9f2d5c 10ffmpeg/libavformat/microdvddec.c: lavf/microdvd: rewrite using subtitles queue API. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * rf926d91611 10ffmpeg/ffmpeg.c: ffmpeg: fix a memleak in subtitles decoding. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * r0e7782c08e 10ffmpeg/ (libavcodec/ass.c tests/ref/fate/sub-srt): [20:29] <CIA-41> ffmpeg: lavc/ass: honor Default style. [20:29] <CIA-41> ffmpeg: The "Default" style written in the header is ignored unless you explicit [20:29] <CIA-41> ffmpeg: it in the Dialogue events (it was valid, just ignored). This requires an [20:29] <CIA-41> ffmpeg: update of the SubRip test since the ASS output obviously changes. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * r8f4ce626f8 10ffmpeg/libavcodec/microdvddec.c: lavc/microdvddec: support "DEFAULT" properties. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * r04568f8d1a 10ffmpeg/tests/ (Makefile fate/subtitles.mak fate/video.mak): fate: introduce subtitles.mak and move SubRip test in it. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * re301f2f8c6 10ffmpeg/tests/ (3 files in 2 dirs): fate: add JacoSUB and MicroDVD subtitles tests. [20:29] <CIA-41> ffmpeg: 03Clément BSsch 07master * rd948893dbd 10ffmpeg/libavformat/ (subtitles.c subtitles.h): [20:29] <CIA-41> ffmpeg: lavf/subtitles: add some SMIL helpers. [20:30] <CIA-41> ffmpeg: This is needed for SAMI and RealText demuxers. [20:30] <CIA-41> ffmpeg: 03Clément BSsch 07master * r53640f42be 10ffmpeg/ (13 files in 6 dirs): SAMI demuxer and decoder. [20:30] <CIA-41> ffmpeg: 03Clément BSsch 07master * r439e32f9b8 10ffmpeg/ (13 files in 6 dirs): RealText demuxer and decoder. [20:30] <CIA-41> ffmpeg: 03Clément BSsch 07master * r0ef28e119e 10ffmpeg/libavformat/jacosubdec.c: lavf/jacosubdec: fix FPE in case timeres is badly set. [20:30] <CIA-41> ffmpeg: 03Clément BSsch 07master * r7c9f9685ae 10ffmpeg/libavformat/ (Makefile subtitles.c subtitles.h): lavf: add internal demuxer helpers for subtitles. [20:30] <CIA-41> ffmpeg: 03Clément BSsch 07master * r60715511db 10ffmpeg/libavformat/jacosubdec.c: lavf/jacosubdec: use subtitles queue API. [20:32] <ubitux> hey btw [20:32] <ubitux> http://irq6.net/haproxy1.5-dev.git/commit/?id=6e0644339f3bcd455693d45219ba8… [20:32] <ubitux> it could be nice to do that [20:33] <ubitux> at the moment we have valgrind with --malloc-fill=0x2a [20:33] <ubitux> but this is somehow limited [20:33] <ubitux> it could be nice to have some lrng doing this for all the fate instances [20:40] <michaelni> nice idea and it should be fairly easy to implement ... [20:42] <michaelni> .... if you arent planing to implement it probably a good idea to open a feature req so its not forgotten ... [20:44] <ubitux> i'm not :) [20:45] <ubitux> well, i think the effort of opening an issue is almost equal to the effort of writing the feature& :D [21:17] <michaelni> hi iive [21:17] <iive> hi michaelni [21:18] <michaelni> do you want to implement filling av_malloc() output by random data ? so use of such data can be detected as difference in the output ? [21:18] <michaelni> the idea was mentioned by ubitux [21:18] <michaelni> but we are all to lazy [21:19] <iive> how urgent it is? [21:22] <michaelni> isnt urgent [21:23] <iive> then if nobody else does it in few days I'll give it a try. [21:24] <iive> is this functionality to be enabled by cli option? [21:26] <michaelni> yes cli option is probably the best and thanks [21:33] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r12863db840 10ffmpeg/libavcodec/wma_common.h: [21:33] <CIA-41> ffmpeg: wma_common: Fix license header [21:33] <CIA-41> ffmpeg: common wma code existed long before Libav [21:33] <CIA-41> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [21:33] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:33] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rfb7688a83f 10ffmpeg/libavcodec/pngdsp.h: [21:33] <CIA-41> ffmpeg: pngdsp: Fix license header [21:33] <CIA-41> ffmpeg: Libav did not exist in 2008 thus this file cannot have originated from there [21:33] <CIA-41> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [21:33] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7bbb6b38fb 10ffmpeg/libavcodec/ (proresdata.h proresdsp.h): [21:33] <CIA-41> ffmpeg: prores: Fix license header [21:33] <CIA-41> ffmpeg: Libav did not exist in 2010 thus the file cannot originate from there [21:33] <CIA-41> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [21:57] <Compn> spend a lot of time on stupid libav/ffmpeg copyright headers [21:57] <Compn> could we make it 'this file is a part of FFmpeg/Libav ? [21:57] <Compn> ask libav to do it too [22:01] <durandal_1707> ^ pointless [22:02] <bcoudurier> :) [22:11] <ubitux> iive: the backlog if you're interested in working on the feature: http://pastie.org/private/iiwgwasarxjrk6bqhobg [22:11] Action: ubitux feels lazy [22:13] <iive> btw, wasn't avmalloc created to explicitly zero the allocate memory? [22:13] <ubitux> no, we have av_mallocz for this [22:14] <iive> hum, my bad. [22:29] <ubitux> durandal_1707: is your public xpm branch up-to-date? [22:29] <durandal_1707> what you mean? [22:30] <ubitux> the patch you submitted on the ml has the encoder & decoder squashed [22:30] <ubitux> i was wondering of testing it with your remote repo [22:30] <ubitux> where the enc & dec are separate [22:30] <ubitux> i was wondering if there was other changes [22:30] <ubitux> (and which one is the more up to date in that case) [22:31] <durandal_1707> ahh what i posted is what i have locally [22:32] <durandal_1707> i usually do not update remote branch after i send patch (unless I fix something....) [22:33] <ubitux> can i assume the two are mostly similar? [22:36] <durandal_1707> they are same, except non-xpm code [22:36] <ubitux> and patch-split :) [23:18] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r0d3ffde64f 10ffmpeg/ffplay.c: ffplay: split overly long line in audio_decode_frame() [23:18] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r8179660222 10ffmpeg/ffplay.c: [23:18] <CIA-41> ffmpeg: ffplay: vertially align complex if condition in audio_decode_frame() [23:18] <CIA-41> ffmpeg: Possibly improve readability. [23:18] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r347ecfdc63 10ffmpeg/ffplay.c: [23:18] <CIA-41> ffmpeg: ffplay: move assignment in else block in audio_decode_frame() [23:18] <CIA-41> ffmpeg: Avoid confusing and pointless double assignment of variable [23:18] <CIA-41> ffmpeg: resampled_data_size. [23:18] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r94a00ec8af 10ffmpeg/ffplay.c: [23:18] <CIA-41> ffmpeg: ffplay: group together and vertically align correlated parameters in log function [23:18] <CIA-41> ffmpeg: Possibly improve readability. [23:27] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * rdfb57fc596 10ffmpeg/libavformat/rtpdec.c: [23:27] <CIA-41> ffmpeg: rtpdec: Don't explicitly include unistd.h any longer [23:27] <CIA-41> ffmpeg: unistd.h used to be required for gethostname. On windows, gethostname [23:27] <CIA-41> ffmpeg: is provided by winsock2.h. Now network.h includes both unistd.h and [23:27] <CIA-41> ffmpeg: winsock2.h if they exist. [23:27] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [23:27] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * re312fcde6a 10ffmpeg/doc/general.texi: [23:27] <CIA-41> ffmpeg: doc: Indicate that RTMPT is natively implemented in libavformat [23:27] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [23:27] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * rf985113075 10ffmpeg/libavutil/random_seed.c: [23:27] <CIA-41> ffmpeg: random_seed: Only read /dev/*random if we have unistd.h [23:27] <CIA-41> ffmpeg: unistd.h is used for open/read/close, but if this header does not [23:27] <CIA-41> ffmpeg: exist, there's probably no use in trying to open /dev/*random [23:27] <CIA-41> ffmpeg: at all. [23:27] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [23:27] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r8a85660d3b 10ffmpeg/: (log message trimmed) [23:28] <CIA-41> ffmpeg: * qatar/master: [23:28] <CIA-41> ffmpeg: file: Only include unistd.h if it exists [23:28] <CIA-41> ffmpeg: random_seed: Only read /dev/*random if we have unistd.h [23:28] <CIA-41> ffmpeg: doc: Indicate that RTMPT is natively implemented in libavformat [23:28] <CIA-41> ffmpeg: rtpdec: Don't explicitly include unistd.h any longer [23:28] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r3b1ab197be 10ffmpeg/ (configure libavformat/file.c): (log message trimmed) [23:28] <CIA-41> ffmpeg: file: Only include unistd.h if it exists [23:28] <CIA-41> ffmpeg: It is included for the open/read/write/close functions. On [23:28] <CIA-41> ffmpeg: MSVC, where this header does not exist, the same functions [23:28] <CIA-41> ffmpeg: are provided by io.h, which is already included. [23:28] <CIA-41> ffmpeg: On windows, these functions are provided by io.h. Make sure [23:28] <CIA-41> ffmpeg: io.h is included if it exists, regardless of the setmode [23:29] <ubitux> michaelni: it looks like the sunos box is missing some samples rsync [23:34] <michaelni> ubitux, fixed [23:34] <michaelni> the problem is i didnt find a rsync client that works reliable for sunos so i decided to run it by hand [23:36] <michaelni> the samples dont change often enough for that to rmatter ,much ,.. [23:37] <ubitux> ok :) [23:37] <ubitux> thx [23:55] <durandal_1707> michaelni: what blocksize dsp support? [23:57] <durandal_1707> the codecs uses blocksize of 4 and there is way to copy pixels horizontally and vertically [00:00] --- Sat Jun 30 2012

1 0

ffmpeg.log.20120629
by burek 30 Jun '12

30 Jun '12

[08:01] <rik316> Am I correct in saying that avio_rl32 will read the next 4 bytes from the stream and return them as an unsigned int? [11:23] <mikunos> Hi guys I am trying to convert a mov file to an avi file but I get this error: [buffer @ 0x88b0180] Invalid pixel format string '-1' [11:23] <mikunos> Error opening filters! [11:23] <mikunos> any idea? [11:23] <mikunos> this is the command executed: ffmpeg -i Scrivania/matrimonio1.mov video1.avi [11:29] <saste> mikunos: try to update your ffmpeg, that was a problem which was fixed some time ago [11:41] <mikunos> ok saste [11:59] <natrixnatrix89> hi guys.. could you please recommend a lossless video codec for x11 grabbing ? [11:59] <mikunos> saste: http://pastie.org/4170577 [11:59] <mikunos> nothing to do [11:59] <mikunos> I have updated the ffmpeg to the latest version [12:00] <mikunos> compiling all the dependencies [12:00] <mikunos> but I get this result: http://pastie.org/4170577 [12:00] <JEEB> natrixnatrix89, ffvhuff or libx264. ffvhuff should be quite fast but not really compressible, libx264 depending on your hardware can be quite fast and can offer more compression [12:01] <saste> mikunos: don't edit the output, the other info are useful as well [12:01] <mikunos> I have followed this tutorial: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [12:01] <natrixnatrix89> how do I set libx264 to be lossless? because if I set -b or -sameq it will compress.. how do I disable compression then? [12:01] <JEEB> natrixnatrix89, sameq is not what you think it is [12:01] <JEEB> and the documentation has since been altered to make it more apparent [12:02] <JEEB> you use either -crf zero or set however you set quants to zero [12:02] <natrixnatrix89> ok. but the question is.. how do I use libx264 for lossless recording? [12:02] <mikunos> ok saste [12:02] <JEEB> (I don't know how you set quants to zero in ffmpeg, but crf zero should in most cases work) [12:02] <JEEB> also you most probably want to set a faster preset [12:02] <mikunos> saste http://pastie.org/4170583 [12:02] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [12:03] <natrixnatrix89> yeah. fast preset should help [12:04] <JEEB> but yeah, ffvhuff is then something that should really be fast and straightforward'ish [12:04] <JEEB> if libx264 even with the fastest presets is too slow [12:04] <iTux> Hi [12:05] <natrixnatrix89> also is ffv1 good? [12:05] <saste> mikunos: weird, can't find the pixel format in the stream, but it goes on anyway and fails when configuring filters [12:05] <JEEB> ffv1 is a good codec, but I'm not sure if it's fast enough [12:05] <saste> you can try to force the pixel format with -pix_fmt ... -i INPUT [12:05] <iTux> I need some help, I can't configure ffmpeg with libass [12:06] <saste> and you may open a ticket on trac [12:06] <natrixnatrix89> but what is -crf ? can't find in documentation [12:06] <mikunos> how have I to use th e -pix_fmt [12:06] <JEEB> natrixnatrix89, it's a setting defined by the libx264 encoder [12:07] <JEEB> it's libx264-specific [12:07] <iTux> ./configure says "ERROR: libass not found" [12:07] <JEEB> basically it's "constant rate factor", and tries to be something a la 'constant quality' in a way [12:07] <JEEB> zero with a 8bit libx264 results in lossless [12:07] <natrixnatrix89> oh.. I see. thanks.. ok. so can use ffmpeg -I {my screen} -crf 0 -pre fast out.mov right? [12:08] <saste> JEEB: what about sending a doc patch for -crf? [12:08] <JEEB> -vcodec/-c:v libx264 -crf 0 -preset veryfast or something [12:08] <natrixnatrix89> right.. [12:08] <natrixnatrix89> thanks [12:08] <JEEB> saste, I think it's already there tho, in video codec libx264's help? [12:09] <JEEB> also I really want to get a first version of this video encoder done this weekend [12:09] <saste> JEEB: yes, so it is a libx264 option, allright [12:10] <JEEB> http://git.libav.org/?p=libav.git;a=blob;f=libavcodec/libx264.c;hb=HEAD#l469 [12:10] <JEEB> argh, wrong repo [12:10] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/libx264.c;h=d56df… [12:10] <JEEB> there [12:12] <saste> JEEB: yes you're right, no point into documenting external library options [12:13] <Vardis> Hello [12:17] <iTux> ./configure make a log, I paste it here : http://pastebin.com/p3eCiHJq [12:17] <Vardis> iff I splitting from biger MTS to smaller chunks and encoding to webm how is the smartest way with audio? extract from big all lenght audio then compress to ogg or extract small portion of audio and then compress to ogg and then encode with thet small ogg? [12:24] <Vardis> And also is there better way to remove interlacing from video or only yadif filter [12:24] <microchip_> yadif is your "best" option with ffmpeg [12:25] <Vardis> Oh [12:27] <Vardis> but somtimes where is sharp contrast like white hand on dark gitar its aniway get jerky [12:27] <microchip_> iirc, ffmpeg has only 2 deinterlacers; yadif and -deinterlace [12:28] <Vardis> deinterlace is absolete [12:28] <Vardis> as I recall correctly [12:28] <microchip_> it's possible [12:28] <Vardis> or meybi I'm wrong [12:28] <JEEB> yes, it's obsolete and shouldn't be used [12:28] <JEEB> it's very bad [12:28] <JEEB> (as a deinterlacer) [12:29] Action: microchip_ lets JEEB take over :p [12:29] <JEEB> oi, I'm actually working now :P [12:29] <Vardis> does vpre is always so slow? [12:31] <microchip_> try a faster preset :p [12:33] <Vardis> microchip_: I have turion 64x2 1.8GHZ using 720p preset for best quality as possible and got max out 2.9 and min 1.2 frames per/s [12:33] <Vardis> faster presset will degrade more qulity :( [12:34] <pron> taa! [12:34] <microchip_> well, get a faster CPU. Turion 64 is pretty slow and old. [12:34] <pron> pidarasi spameri zajebalji [12:34] <Vardis> microchip_: whoos is admin here? [12:34] Action: pron noslauka asinjainaas rokas un veero kaa pamazam tiiraas 20k mail queue [12:34] <microchip_> i don't know [12:34] <microchip_> pron: GTFO [12:34] <pron> omm [12:35] <pron> wrong chan [12:35] <pron> xD [12:35] <pron> soz [12:35] <pron> :} [12:35] <Vardis> pron: nu skaties kur raksties [12:35] <pron> haha [12:36] <pron> microchip_: soz saw similar few names and missed chan :} [12:36] <microchip_> no problem :) [12:36] <Vardis> pron: bet 20k pastelja ir shausmas gan piekriitu [12:36] <pron> Vardis: yep [12:36] <Vardis> man apolleklii ~15k [12:36] <Vardis> un kriit bezgala iekshaa [12:36] <pron> Vardis: but this one is eng only chan [12:37] <pron> <- tvnet :P [12:38] <microchip_> what language is that? [12:38] <pron> latvian [12:38] <microchip_> ah thanks [12:38] <pron> no clue wtf is that?:D [12:40] <jonsen_> hm [12:40] <Vardis> pron: ok I got Ya :D [12:41] <microchip_> Vardis: as for the encoding speed, you have 3 options: 1) buy faster hardware, 2) use faster preset or 3) scale down the content [12:41] <jonsen_> when I want to stream a .mov via ffserver is there anything needed prior to streaming to make it streamable eg. get header and stuff early? [12:41] <jonsen_> I do have format .asf applied to ffserver, but when connecting to the server the client downloads the whole file and waits until its finished [12:42] <Vardis> microchip_: whell I have that I got and nobody will sponsor Me for better HW but wideo want at least hd [12:43] <Vardis> And I only encoding for youtube [12:44] <microchip_> Vardis: youtube *always* re-encodes the video. So even if you encode it with "perfect" quality, youtube will encode it once more and degrade its quality [12:44] <Vardis> even webm? [12:44] <microchip_> i think so [12:45] <microchip_> JEEB: does youtube *always* re-encode? [12:45] <JEEB> yes [12:45] <microchip_> Vardis: there ^^ [12:46] <microchip_> Vardis: so your "perfect" quality video will look "not-so-perfect" after youtube encodes it once more [12:49] <microchip_> he timed out lol [13:08] <natrixnatrix89> What is the file format if I want to save video with ffvhuff codec? [13:09] <JEEB> I would guess you can mux it into avi or matroska (mkv) [13:10] <natrixnatrix89> thanks [13:13] <natrixnatrix89> pron: btw which was the channel in latvian on freenode? [13:19] <pron> natrixnatrix89: i have no clue :} [13:50] <jonsen_> its kinda frustrating that there is no documentation about how to stream simple things [14:11] <saste> jonsen: http://ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide [14:11] <jonsen_> helps nothing as it only covers one very specific case [14:12] <jonsen_> and finding an answer for "stream .asf from codex .foo could be done that way, consider this and that, problems are..." [14:12] <jonsen_> i'am doing hard to get simple stream working [14:47] <natrixnatrix89> ffserver stream? [14:48] <jonsen_> cool segmentation fault when accessing sdp [14:51] <jonsen_> natrixnatrix89: ffserver, yes [15:07] <jzmer> does ffmpeg support any form of hardware-acceled audio decoding? [15:48] <jonsen_> ffserver seems not to be very ready [15:48] <jonsen_> it crashs a lot [20:43] <MarioMey> Hello. I just want to clarify about ffmpeg... because, I used to use -vcodec... then, I had to change to -c:v... now, it says that ffmpeg is depracated...? [20:43] <MarioMey> Is there any page where I can read about this? [20:44] <MarioMey> Once I understand how to use it... now, I have to learn again. [20:44] <MarioMey> I updated to Ubuntu 12.04 (I was on 11.04 with a ffmpeg from who-knows-who's PPA). [20:45] <sacarasc> You don't have ffmpeg installed, but libav. [20:46] <JEEB> yeah, and it doesn't say that ffmpeg as a project is deprecated, but that the ffmpeg tool (in libav) is [20:47] <JEEB> also, -c:v is newer syntax on both libav and ffmpeg [20:47] <JEEB> s/on/in/ [20:47] <JEEB> you can still use vcodec tho, methinks [20:48] <MarioMey> What do you suggest...? [20:48] <MarioMey> Install ffmpeg? [20:48] <JEEB> uhh, the command line syntax in both should be similar, if you use libav you use avconv and if you use ffmpeg you use the ffmpeg tool [20:50] <MarioMey> Well... I want to use what isn't depracated... [20:50] <MarioMey> The one that won't disappear. [20:50] <JEEB> if you are using libav, avconv is the up-to-date one; if you are using ffmpeg, it has integrated the libav avconv updates into ffmpeg [20:50] <JEEB> you are now using libav [20:51] <MarioMey> How to install it? [20:51] <JEEB> huh? [20:51] <MarioMey> avconv is the name? [20:51] <JEEB> yes [20:51] <MarioMey> sudo apt-get install avconv? [20:51] <JEEB> you should already have it tho [20:51] <MarioMey> ah... [20:51] <JEEB> unless they put it into a separate package [20:51] <MarioMey> Yes, I have it. [20:52] <MarioMey> And -c:v or -vcodec? [20:52] <JEEB> whichever you prefer, -c:v is the newer way of setting a codec, -vcodec is the older [20:52] <JEEB> I think both still work [20:52] <MarioMey> Right. [20:52] <MarioMey> Another question... [20:53] <MarioMey> I want to concatenate 2 videos with audio. [20:53] <MarioMey> Can I do "avconv -i one.mod -i two.mod out.mp4? [20:54] <JEEB> I think the only concateration thingy in both ffmpeg and libav was the filter that basically works as if you were just inputting both as a single file [20:54] <JEEB> although I'm not sure [20:54] <MarioMey> Sorry, but I didn't understand. [20:54] <relaxed> -i concat:1.blah\|2.blah -c copy out.blah [20:54] <JEEB> relaxed, but wouldn't that just do a simple concateration of input files? [20:55] <JEEB> as in, not properly decode and put one input after another? [20:55] <JEEB> so it wouldn't work with stuff that can't be "just concaterated" [20:55] <relaxed> oh, sorry, I just glanced at his question. [20:56] <relaxed> I assume you can do it with -map but I don't have time to hold someone's hand. [20:56] <JEEB> I'm not that sure about that and I don't have enough info on that unfortunately [20:57] <MarioMey> Thanks guys, I have a work meeting RIGHT NOW. [21:27] <raptor67782> hi how to play with ffplay without video (only sound)? [21:28] <raptor67782> ffplay -vo null "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf" is not working [21:29] <sacarasc> -vn [21:30] <raptor67782> wow cool thanks [21:30] <raptor67782> http://linuxers.org/tutorial/how-remove-audio-and-video-streams-media-file-… [21:31] <raptor67782> ffplay -vn "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf" [21:31] <raptor67782> well, not working [21:31] <raptor67782> ffplay -vn "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf" teells that the stream is not found [21:31] <raptor67782> however the stream is well there streamed to inteernet at this url [21:33] <raptor67782> ffplay -vn -ab 128 "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf" is not working either [22:46] <burek> raptor67782, did you ever read ffmpeg documentation? [22:47] <burek> -vn is an output option [22:47] <burek> so put it where it should be [22:47] <burek> also -ab 128 (bits?) is an encoding option, not playback option [22:47] <burek> and can't be used with ffplay [22:48] <burek> what exactly are you trying to do ? [22:58] <MarioMey> relaxed: concat:Sándalo 1º parte.MOD|Sándalo 2º parte.MOD: No such file or directory [22:58] <MarioMey> It didn't work. [23:00] <JEEB> see my comment towards relaxed, the concat filter/input thingy most probably isn't what you need. Also, if your file names have spaces and other things you have to quote/escape one way or another you'll have to deal with that. [23:01] <MarioMey> they have quotes/escape, I used TAB. [23:01] <MarioMey> What should I use? [23:04] <intracube> hi [23:04] <intracube> is it possible to get ffmpeg to compress expand the 0-255/16-235 range of a video? [23:05] <intracube> I've noticed some camcorders don't peak out at Y=235, but instead go all the way to Y=255 [23:05] <intracube> is there a 0-255->16-235 filter/command? [23:05] <intracube> the -color_range option got my hopes up: [23:06] <intracube> http://ffmpeg.org/pipermail/ffmpeg-user/2011-July/001579.html [23:06] <intracube> but it doesn't seem to work [23:11] <MarioMey> How to enconde to libx264? "Unknown encoder 'libx264'" [23:11] <juanmabc> "ffmpeg -codecs | grep 264" search for "E" in the tags [23:13] <MarioMey> juanmabc: no tengo codificadores... sólo "D" [23:13] <MarioMey> Yo se que no está por default, ¿cómo lo instalo? [23:13] <llogan> MarioMey: use a pastebin site to show your ffmpeg command and the complete console output [23:15] <MarioMey> Only these: [23:15] <MarioMey> D V D h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [23:15] <MarioMey> D V D h264_vdpau H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) [23:15] <JEEB> then that libavcodec library you are using was built without libx264 [23:15] <JEEB> are you still using a packaged version? [23:15] <llogan> i meant your encoding command. i want to see the detailed information about your ffmpeg so i don't have to ask as many questions [23:16] <JEEB> MarioMey, if you are still using a packaged version, try installing the packages x264 and libx264 [23:16] <MarioMey> llogan: http://pastebin.com/nvX7uSRw [23:17] <MarioMey> I'm installing x264 [23:17] <llogan> good. install libavcodec-extra-53 [23:17] <llogan> you don't need x264 [23:17] <JEEB> oh, they have a separate libavcodec package? [23:17] <MarioMey> Should I uninstall it...? [23:17] <JEEB> didn't know [23:17] <JEEB> MarioMey, it won't do anything bad [23:17] <llogan> no, you don't need to uninstall [23:17] <MarioMey> Ok. [23:17] <JEEB> just install the libavcodec-extra package as llogan said [23:18] <smjms> so if I want to encode 2-pass H264, do I need to only specify -pass 1 and -b? [23:18] <llogan> smjms: basically, yes. [23:18] <JEEB> smjms, set -pass 1 for first pass with -b and then just switch s/1/2/ for the second pass [23:18] <MarioMey> llogan: libavcodec53 will be uninstalled... ok? [23:18] <JEEB> yes [23:18] <MarioMey> libavutil51 too. [23:19] <JEEB> it will install newer libraries that have libx264 support [23:19] <MarioMey> Yes! [23:19] <llogan> MarioMey: also, you are not using FFmpeg. you are using a fork that is not supported here (but we often help anyway). [23:20] <JEEB> I already told him that some time ago :) [23:20] <llogan> because i'm only an asshole 30% of the time [23:20] <JEEB> and since his problems most probably have a common solution command-line wise I didn't tell him to go to #libav [23:20] <smjms> okay, so does it matter what output codec I choose on the first pass, or does ffmpeg use some generic stats format libx264 can read? [23:20] <JEEB> smjms, uhh I wouldn't change the vcodec around [23:21] <smjms> okay, so libx264 on both passes [23:21] <llogan> JEEB: ah. i came in late, and i think i've only referred one or two people to #libav [23:21] <JEEB> stats files and mbtree stats with libx264 especially are libx264-specific [23:21] <MarioMey> Encoding! [23:21] <MarioMey> Thanks, gentlemen. [23:21] <JEEB> smjms, anyways libx264 automatically selects faster settings in case it finds out that you are doing a "first pass" [23:21] <MarioMey> ffmpeg doesn't have a x264 encoding? [23:21] <JEEB> (pass 1 is set) [23:21] <JEEB> MarioMey, the format is H.264 and x264 is the best H.264 encoder around [23:21] <JEEB> there has been no need to reinvent the wheel [23:21] <smjms> so I should define -pass after I set the -preset? [23:22] <JEEB> smjms, it doesn't matter if it's before or after the -preset, it just has to be after -i (thus setting an encoder setting) [23:22] <MarioMey> Ok, I just want to understand why you said this is not a ffmpeg topic. [23:22] <smjms> ok [23:22] <JEEB> MarioMey, you are using libav, which is a fork of ffmpeg [23:22] <JEEB> (as I said a long time ago) [23:23] <MarioMey> Ok. [23:23] <JEEB> and well, in this case the answers to your questions would be the same with both avconv and ffmpeg [23:23] <JEEB> thus I didn't tell you to go to #libav :) [23:23] <smjms> is there a setting for "target filesize"? [23:23] <MarioMey> Ok. [23:23] <MarioMey> Thanks. [23:23] <llogan> do users get much help in that channel? [23:23] <juanmabc> i saw you even take back some libav patches [23:23] <juanmabc> cool [23:24] <JEEB> llogan, yes -- it's just that not many ask things there to begin with [23:24] <JEEB> smjms, bitrate is pretty much target file size [23:24] <llogan> i guess mru doesn't hang out there then [23:24] <JEEB> nah [23:24] <JEEB> he isn't even in libav-devel [23:24] <JEEB> because his ways weren't exactly liked [23:24] <smjms> JEEB: okay, thanks for help [23:24] <llogan> unsurprising [23:25] <smjms> !beer JEEB [23:25] <smjms> I guess that was for another channel [23:25] <JEEB> smjms, bitrate is basically kilobytes or kilobits per second, so you multiply by the length in seconds and you'll get the file size for that bit rate and length [23:26] <JEEB> also, what I said about the fast settings in libx264 can be better read @ http://mewiki.project357.com/wiki/X264_Settings#slow-firstpass [23:26] <juanmabc> he's... gone [23:26] <JEEB> argh [23:26] <llogan> jim [23:27] Action: llogan disposes of remaining star trek references [23:27] <JEEB> lol [23:28] <JEEB> llogan, thanks for providing me with the info on the libavcodec-extra package, didn't know it worked like that [23:28] <llogan> it enables other stuff too. not sure what else anymore. i haven't been keeping up with it after the libav adoption obviously [23:29] <JEEB> probably all of the non-freetard-compatible stuff :) [23:29] <llogan> and medibuntu also provides it which adds libfaac and probably something else [23:29] <JEEB> thankfully libfaac should become relatively unneeded soon'ish :) [23:30] <llogan> what's the news? [23:31] <JEEB> fraunhofer's AAC encoder was released with a GPL'ish license via the big G, and wbs has written a library of it and is writing the stuff so it can be used with libavcodec [23:31] <llogan> interesting. where in hell have i been? [23:31] <JEEB> not LGPL, but better than the nonfree faac and libaacplus :) [23:32] <intracube> x264 .ffpreset files are broken with recent versions of ffmpeg [23:32] <JEEB> since it seemingly provides both LC AAC and HE AAC (and v2 of that IIRC) [23:32] <intracube> does anyone know a fix? [23:32] <JEEB> intracube, I don't think they're used any more? [23:32] <JEEB> or come with ffmpeg [23:32] <llogan> replace -vpre with -preset [23:32] <llogan> and then use a preset listed in "x264 --fullhelp" [23:33] <JEEB> or just look at the listing @ http://mewiki.project357.com/wiki/X264_Settings#preset [23:33] <JEEB> :) [23:33] <llogan> who is wbs? i have a terrible memory. [23:33] <JEEB> https://github.com/mstorsjo/ [23:34] <intracube> llogan: JEEB: that works. thanks. [23:34] <JEEB> np [23:34] <llogan> JEEB: thanks [23:34] <intracube> it would be nice if ffmpeg threw up a more helpful error message though... ;) [23:35] <llogan> what was the message? [23:37] <intracube> hmm, I can't post the command... trying again [23:37] <intracube> I mean the error [23:37] <intracube> "Invalid option or argument: 'directpred=3', parsed as 'directpred' = '3'" [23:37] <JEEB> oh [23:37] <JEEB> I guess the ffpreset file contains settings that are in one way or another unusable now [23:38] <JEEB> as the x264's internal presets got usable [23:38] <intracube> something like "ffpreset files deprecated" would be helpful [23:38] <llogan> use a pastebin site to show your ffmpeg command and the complete console output [23:38] <JEEB> well, the system itself isn't [23:38] <JEEB> and ffpreset files are still usable for other encoders that don't have pre-made presets [23:38] <JEEB> (and for custom settings for x264) [23:39] <JEEB> but I guess since a newer ffmpeg compile won't copy new ffpreset files for x264 I guess it ended up reading an old ffpreset file and failing at parsing it :) [23:39] <intracube> JEEB: thanks for the explanation [23:40] <intracube> llogan: yes, I'll use pastebin in future [23:40] <intracube> the encoding is working fine now [00:00] --- Sat Jun 30 2012

1 0

ffmpeg-devel.log.20120628
by burek 29 Jun '12

29 Jun '12

[00:15] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rb1a17953ec 10ffmpeg/libavcodec/lpc.c: [00:15] <CIA-119> ffmpeg: lpc: use av_assert [00:15] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:15] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rd1c74ca2be 10ffmpeg/libavcodec/lsp.c: [00:15] <CIA-119> ffmpeg: lsp: use av_assert [00:15] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:39] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * ree7214c59a 10ffmpeg/libavutil/log.c: [00:39] <CIA-119> ffmpeg: log: change color for filters from blue to bright green [00:39] <CIA-119> ffmpeg: The blue is difficult to read on several peoples terminals with black background. [00:39] <CIA-119> ffmpeg: Idea-by: Paul B Mahol <onemda(a)gmail.com> [00:39] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:39] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r7803a04041 10ffmpeg/libavformat/http.c: [00:39] <CIA-119> ffmpeg: http: try to detect live akamai streams and dont enable seeking for them [00:39] <CIA-119> ffmpeg: Fixes ticket1320 [00:39] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:39] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r4ccf6e3971 10ffmpeg/libswresample/ (resample.c resample_template.c x86/resample_mmx.h): [00:39] <CIA-119> ffmpeg: swr: MMX2 & SSSE3 int16 resample core [00:39] <CIA-119> ffmpeg: about 4 times faster [00:39] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:47] <ubitux> nice push :) [00:51] <ubitux> michaelni: resample_mmx.h seems to be the only one using __asm [00:51] <ubitux> why not __asm__? [01:03] <michaelni> ubitux, no reason, i hadnt noticed, feel free to change it [01:05] <ubitux> i'm curious to see if it will make some fate instance red [01:05] <ubitux> so maybe later [01:09] <CIA-119> ffmpeg: 03Jordi Ortiz 07master * r58f3e09ee4 10ffmpeg/libavformat/tcp.c: [01:09] <CIA-119> ffmpeg: tcp: Set AI_PASSIVE when the socket will be used for listening [01:09] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:09] <CIA-119> ffmpeg: 03Jordi Ortiz 07master * ref882e464a 10ffmpeg/libavformat/tcp.c: [01:09] <CIA-119> ffmpeg: tcp: Pass NULL as hostname to getaddrinfo if the string is empty [01:09] <CIA-119> ffmpeg: This gives you the proper v4 or v6 version of the "any address", [01:09] <CIA-119> ffmpeg: allowing receiving connections on any address on the machine. [01:09] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:09] <CIA-119> ffmpeg: 03Martin Storsjö 07master * r75d339e044 10ffmpeg/ (configure doc/protocols.texi libavformat/udp.c): [01:09] <CIA-119> ffmpeg: udp: Support IGMPv3 source specific multicast and source blocking [01:09] <CIA-119> ffmpeg: Based on an original patch by Stephen D'Angelo <SDAngelo(a)evertz.com>. [01:09] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:09] <CIA-119> ffmpeg: 03Ronald S. Bultje 07master * rfa84506177 10ffmpeg/ (configure libavcodec/dxva2_internal.h): [01:09] <CIA-119> ffmpeg: dxva2: include dxva.h if found [01:09] <CIA-119> ffmpeg: Apparently, some build environments require dxva.h even for dxva2, [01:09] <CIA-119> ffmpeg: while others lack this header entirely. Including it conditionally [01:09] <CIA-119> ffmpeg: allows building in both cases. [01:09] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:09] <CIA-119> ffmpeg: 03Martin Storsjö 07master * r46df708b45 10ffmpeg/ (configure libavutil/libm.h): [01:10] <CIA-119> ffmpeg: the assumptions we do in the new code. [01:10] <CIA-119> ffmpeg: Tested on an IMC sample: 430c -> 370c. [01:10] <CIA-119> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [01:10] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r87df986dcf 10ffmpeg/: (log message trimmed) [01:10] <CIA-119> ffmpeg: Merge remote-tracking branch 'qatar/master' [01:10] <CIA-119> ffmpeg: * qatar/master: [01:10] <CIA-119> ffmpeg: mss1: validate number of changeable palette entries [01:10] <CIA-119> ffmpeg: mss1: report palette changed when some additional colours were decoded [01:10] <CIA-119> ffmpeg: x86: fft: replace call to memcpy by a loop [01:10] <CIA-119> ffmpeg: udp: Support IGMPv3 source specific multicast and source blocking [01:10] <CIA-119> ffmpeg: 03Kostya Shishkov 07master * r8f5d573a83 10ffmpeg/libavcodec/mss1.c: mss1: report palette changed when some additional colours were decoded [01:35] <durandal_1707> michaelni: xpm decoder needs adding more colors to color table in lavu/parseutils [01:43] <michaelni> adding more should not be a problem [02:48] <Nickname123> The other day I opened a feature request: https://ffmpeg.org/trac/ffmpeg/ticket/1483 [02:49] <Nickname123> I toyed around with implementing it: https://github.com/thenewguy/FFmpeg/commit/c92a070d376cff11af60119113258938… [02:49] <Nickname123> Are github pull requests an acceptable way to submit a patch? [02:50] <durandal_1707> i don't think so [02:51] <Nickname123> I have read http://git.videolan.org/?p=ffmpeg.git;a=blob_plain;f=doc/issue_tracker.txt;… [02:51] <durandal_1707> Nickname123: http://ffmpeg.org/developer.html#Contributing [02:52] <Nickname123> How do I go about getting my patch approved by another person [02:52] <Nickname123> okay Ill read that one [02:52] <Nickname123> Oh nice. Github is accepted =) [02:53] <durandal_1707> pull request may be ignored, patches to ml list less so [02:53] <Nickname123> Okay. I didn't see a guide to creating patches. Should they be from the base FFmpeg repo? [02:54] <Nickname123> *guidelines [02:54] <durandal_1707> Nickname123: http://ffmpeg.org/developer.html#patch-submission-checklist [02:55] <Nickname123> well dang. my bad [03:02] <Nickname123> thanks durandal_1707 btw [03:09] <Nickname123> Is it possible to sign off on a commit after it has been committed? [03:27] <durandal_1707> michaelni: already available colors in parseutils conflict with xpm/x11 definitions, like green color it is more for SVG compliance [04:36] Action: Nickname123[A] is now away - Reason : sleep [07:36] <ubitux> Nickname123[A]: please follow the surrounding coding style, use av_strtok() instead of strtok(), use strtol() instead of atoi() [07:36] <ubitux> Nickname123[A]: also, since you're working on the segmenter, you should see with Stefano (saste) when he is here [07:36] <ubitux> he's done some work on it [07:36] <ubitux> (which he hadn't submitted yet...) [07:37] <ubitux> see https://gitorious.org/~saste/ffmpeg/sastes-ffmpeg/commits/misc-segment-fixe… for instance [07:37] <ohsix> is av_strtok like strtok_r [07:37] <ubitux> ohsix: yes [07:37] <ohsix> ic [08:22] <jesk> good morning [08:24] <jesk> before me opens bug maybe it would make sense to get confirmation here in the first place [08:24] <jesk> i'am too unexperienced if that could be expected behavior [08:25] <jesk> i have pretty simple configuration, more or less stripped down the example configs to only have one stream and one feed [08:25] <jesk> when adding the multicast options to the stream configuration the server won't response anymore to http request [08:26] <jesk> s,to http request, to any http request, [08:26] <jesk> i'am using 0.11.1, osx build [08:27] <jesk> when removing the multicast options the server responses again [08:29] <jesk> i tried the debug flag, but tells nothing about this [10:08] <michaelni> jesk, for which protocol do you need multicast and for what do you need http [10:08] <CIA-41> ffmpeg: 03Michael Bradshaw 07master * rfc5999d027 10ffmpeg/libavformat/ (avformat.h options.c): [10:08] <CIA-41> ffmpeg: lavf: add proper enum type for fmt ctx duration esitmation method [10:08] <CIA-41> ffmpeg: Signed-off-by: Michael Bradshaw <mbradshaw(a)sorensonmedia.com> [10:08] <michaelni> i assume you arent trying to get multicast http to work :) [10:21] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rccbcf482ad 10ffmpeg/libavformat/udp.c: [10:21] <CIA-41> ffmpeg: udp: use av_freep() instead of av_free() [10:21] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:20] <mwkenna> Hi guys. I have a question with regards to FFMpeg and LibAv. What's the difference? Also - does the old FFMPEG API mailing list still run? [11:21] <mwkenna> (I am using avcodec for my projects also) [11:26] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r782763ed2b 10ffmpeg/libswresample/resample.c: [11:26] <CIA-41> ffmpeg: swr: fix compilation with ancient toolchain that doesnt support SSSE3 [11:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:08] <DEgITx> hellow guys, may i ask some questions about ffmpeg api? [13:10] <michaelni> yes [13:13] <michaelni> j-b, about the vscprintf issue, can you provide more details about the build environment (to nicolas) ? [13:14] <DEgITx> first one about av_read_frame()... is there any posible way to know return maximum packet size? [13:14] <michaelni> both mingw 32 & 64bit fate.ffmpeg clients are also green [13:16] <michaelni> DEgITx, the maximum is limited by the file format, by INT_MAX of the size field and in some cases by the maximunm packet size of the codec [13:16] <michaelni> its also limited by max_alloc_size indirectly [13:18] <michaelni> it also is limited by the filesize in many cases [13:26] <DEgITx> ok ... MAX_INT bad choise for limitation in my way), so is any posible simple ways to get codec/file limit (independently from codec and file format) ? [13:27] <michaelni> theres no lisit in ffmpeg about such limits no, btw why do you need to know it ? [13:27] <DEgITx> i mean AVCodecContext or something like this [13:30] <DEgITx> i'm trying to play file while it's downloading (such task), and care about av_read_frame() to have enough bytes for reading [13:32] <michaelni> you would have to add support to the demuxer to return EAGAIN in if theres not enough data [13:32] <michaelni> that is if you want it to work well [13:32] <michaelni> with low latency [13:33] <michaelni> max frame/packet size can be quite large compared to average with many file formats [13:34] <DEgITx> about reading again, what do you mean (any example?) [13:36] <michaelni> that a demuxer and av_read_frame would return EAGAIN if there is not enough data instead of blocking or retunrning EOF [13:36] <michaelni> and that they then could continue/be retried when theres more data [13:36] <michaelni> but as said you would need to add support for this in libavformat to the demxuer that you need [13:42] <DEgITx> and there is no problem with truncation of packets in such way? [13:44] <michaelni> no, the demuxer would have to seek back and restore any changes to its state before retunring EAGAIN [13:55] <DEgITx> ok, i understand that it's good variant to add support direct to the demuxer... but i'm trying don't touch sources of ffmpeg and making realization separately from it as well as caring about integrity of file [14:44] <j-b> michaelni: normal debian, i586-migw32msvc from mingw.org [14:47] <michaelni> j-b, thx, i forwarded the awnser to nicolas [15:01] <funman> mingw.org disappeared from sid btw [16:52] <nevcairiel> people are still using old legacy-mingw? [16:57] <j-b> yes. [16:59] <nevcairiel> they should stop :P [17:03] <j-b> when ming-w64 will be stable, indeed [17:06] <funman> if you have an issue with mingw-w64 just tell me [17:09] <j-b> lack of maturity, so far. [17:10] Action: funman slaps j-b [17:12] <j-b> la violence est le dernier recours de l'incompétent [17:17] <av500> didnt you agree to be nice to each other? [17:17] <av500> in #videolan [17:17] <av500> apply the same rules here [17:18] <j-b> av500: we love each other. [17:18] <av500> pics or it did not happen [17:22] <j-b> av500: http://people.videolan.org/~funman/pics/coding/IMG_20120619_170645.jpg [17:23] <av500> ok, seems mostly IRC-love :) [17:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * re377208d43 10ffmpeg/ffmpeg.c: [17:30] <CIA-41> ffmpeg: ffmpeg: dont copy creation_time as the destination file is not created at that time [17:30] <CIA-41> ffmpeg: Fixes Ticket1439 [17:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:42] <TimNich> Thanks Michael... [17:46] <cbsrobot> j-b: the picture seems to be photoshoped - i do not see any porn [19:18] <CIA-41> ffmpeg: 03Paul B Mahol 07master * r61123fb8f7 10ffmpeg/libavutil/parseutils.c: [19:18] <CIA-41> ffmpeg: parseutils: fix sorting of colors [19:18] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [19:19] <CIA-41> ffmpeg: 03Paul B Mahol 07master * r3f59bbf5b6 10ffmpeg/libavfilter/af_amerge.c: [19:19] <CIA-41> ffmpeg: lavfi/amerge: silence warnings [19:19] <CIA-41> ffmpeg: The warning silenced was: libavfilter/af_amerge.c:144:29: warning: conversion specifies type 'long long' but the argument has type 'int' [-Wformat] [19:19] <CIA-41> ffmpeg: The warning was introduced after FF_API_SAMPLERATE64 removal. [19:19] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [19:57] <durandal_1707> michaelni: could i get help with making ffmpeg outputs filters AVOptions? [21:00] <michaelni> durandal_1707, sure, what exactly do you mean ? [21:17] <CIA-41> ffmpeg: 03Nicolas George 07master * r05d6cc116e 10ffmpeg/libavfilter/buffersrc.c: buffersrc: warn when there are too many buffers. [21:17] <CIA-41> ffmpeg: 03Nicolas George 07master * rfcf8706ed9 10ffmpeg/libavfilter/sink_buffer.c: sink_buffer: warn when there are too many buffers. [21:35] <jesk> why does ffserver stops responding to any http request as soon as those few multicast options are configured? [21:48] <CIA-41> ffmpeg: 03Nicolas George 07master * r2c793b8501 10ffmpeg/ffmpeg.c: [21:48] <CIA-41> ffmpeg: ffmpeg: warn when -t is used for inputs. [21:48] <CIA-41> ffmpeg: Using -t on an input already have surprising results. [21:48] <CIA-41> ffmpeg: Furthermore, using it on an input or an output makes [21:48] <CIA-41> ffmpeg: a real difference if there are speed-altering filters. [21:48] <CIA-41> ffmpeg: Implementing -t for inputs will probably result in some [21:48] <CIA-41> ffmpeg: behavour changes. [21:49] <CIA-41> ffmpeg: 03Nicolas George 07master * r8069db8633 10ffmpeg/ffmpeg.c: ffmpeg: warn that -t does not work with -filter_complex. [22:07] <durandal_1707> why I get "Audio timestamp 85 < 284 invalid, cliping" ? [22:23] <durandal_1707> resolved: was because get_frame_defaults was not called.... [22:26] <CIA-41> ffmpeg: 03Jeff Downs 07master * r8d9fd58113 10ffmpeg/libavcodec/h264.c: (log message trimmed) [22:26] <CIA-41> ffmpeg: h264: Fix maximum reference count check for non-b frames [22:26] <CIA-41> ffmpeg: Below fixes the maximum reference count check for second reference list in [22:26] <CIA-41> ffmpeg: non-B frames. There is nothing to prohibit full (field sized) reference [22:26] <CIA-41> ffmpeg: list in this case as far as I can tell, and this fixes several syntax-test [22:26] <CIA-41> ffmpeg: files here (this is a regression caused when this check was made more [22:26] <CIA-41> ffmpeg: stringent by [22:26] <CIA-41> ffmpeg: 03Nick Brereton 07master * r16f6c16ac0 10ffmpeg/libavcodec/dca.c: [22:26] <CIA-41> ffmpeg: Parse & decode DTS XXCH frames [22:26] <CIA-41> ffmpeg: Reviewed-by: Benjamin Larsson <benjamin(a)southpole.se> [22:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:26] <CIA-41> ffmpeg: 03Nick Brereton 07master * rcd8bef969a 10ffmpeg/libavcodec/ (dca.c dcadata.h): [22:26] <CIA-41> ffmpeg: Generate channel layout, reordering for DTS-XXCH extension and, undo embedded downmixes [22:26] <CIA-41> ffmpeg: Reviewed-by: Benjamin Larsson <benjamin(a)southpole.se> [22:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:28] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rb0fdd3489a 10ffmpeg/ (4 files in 2 dirs): [22:28] <CIA-41> ffmpeg: lavfi: deprecate av_buffersrc_buffer() function [22:28] <CIA-41> ffmpeg: Favor av_buffersrc_add_ref() instead, which is more powerful. [22:28] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r7877b50d18 10ffmpeg/ffplay.c: ffplay: simplify code by using avfilter_unref_bufferp() [22:55] <ubitux> llogan: do you have write access to push your patch or a developer need to push it for you? [23:14] <j-b> So, this is different codecs, almost, but same CODEC_ID? [23:20] <llogan> ubitux: i believe i only have write access for ffmpeg-web, but right now i'd prefer if others push it anyway [23:21] <llogan> if isn't not too much a PITA [00:00] --- Fri Jun 29 2012

1 0

ffmpeg.log.20120628
by burek 29 Jun '12

29 Jun '12

[00:48] <cbreak> http://paste.the-color-black.net/277890 :( [00:48] <cbreak> this is on OS X [00:48] <cbreak> so no ELF [00:51] <lolfrenz> hey [00:51] <lolfrenz> I have a .flv file that I'd like to convert to mp3 with libmp3lame. I'm doing ffmpeg -i file.flv -qscale 8 -ainput libmp3lame -ab 192k file.mp3 [00:51] <lolfrenz> but while the flv sounds nice, the mp3 doesn't (especially bass on higher volume) [00:53] <lolfrenz> in fact, even -acodec copy which creates an aac makes horrible sound [00:55] <llogan> use a pastebin site to show your ffmpeg command and the complete console output [00:57] <ubitux> cbreak: i think that's now fixed [00:58] <ubitux> or maybe it will be in the next merge [00:58] <lolfrenz> one moment [01:04] <lolfrenz> llogan, http://pastie.org/4162658 [01:04] <lolfrenz> the .flv sounds good but the .mp3 sounds horrible [01:06] <llogan> lolfrenz: replace "-sameq -ab 192k" with "-aq 4" [01:07] <lolfrenz> one min [01:08] <lolfrenz> btw, where is this documented? ffmpeg says "codec-specific" for -aq [01:08] <lolfrenz> so I don't even know what 4 means [01:09] <llogan> lolfrenz: yeah, the value varies depending on the encoder. in this case, consider it the same as "lame -V". [01:10] <lolfrenz> so why 4 and not 0? [01:10] <llogan> http://wiki.hydrogenaudio.org/index.php?title=LAME#VBR_.28variable_bitrate.… [01:10] <lolfrenz> oh, nice [01:10] <llogan> also scroll up and read the stuff under "Recommended encoder settings" [01:11] <llogan> 4 is default [01:11] <llogan> for lame, IIRC [01:11] <llogan> 4 should be close to perceptual transparency [01:13] <llogan> so to directly answer your question, most people probably can't tell a difference between 0 and 4. [01:13] <lolfrenz> yeah, I've read that on the wiki [01:13] <lolfrenz> just tried with 4, still sounds horrible [01:14] <lolfrenz> it's an youtube-downloaded .flv, downloaded using youtube-dl [01:14] <lolfrenz> you can probably reproduce with ./youtube-dl http://www.youtube.com/watch?v=9w1PYLJEpxk&feature=related and then ffmpeg -i file.flv -acodec libmp3lame -aq 4 file.mp3 [01:15] <lolfrenz> hmm, maybe it's some volume stuff [01:15] <lolfrenz> if I lower the volume a bit it sounds pretty much like the video [01:15] <lolfrenz> brb, sorry [01:19] <llogan> maybe you need this: http://www.poormojo.org/pmjadaily/archives/009864.php [01:20] <cbreak> ubitux: yep, that problem got fixed :) [01:20] <cbreak> but a next one appeared: http://paste.the-color-black.net/277937 [01:32] <cbreak> http://paste.the-color-black.net/277940 [01:35] <Jan-> has anyone ever p/invoked avformat/avcodec/etc from inside a C# application? [01:35] <Jan-> I guess it's *possible* [01:35] <Jan-> but has it ever been done [01:46] <Nickname123> what is the recommended aac encoder for ffmpeg? [01:50] <llogan> Nickname123: libfaac, libvo_aacenc, or aac (descending order of "quality" / bitrate). or libaacplus for lower bitrates. [01:51] <llogan> Jan-: did you just read Phil's post on ffmpeg-user mailing list? [01:51] <Nickname123> thanks llogan. i thought I read libfaac wasn't recommended [01:51] <llogan> why not? [01:52] <llogan> lolfrenz: when you come back try the command with -vn. perhaps the png "video" in the mp3 is screwing up stuff in your player. [01:52] <Nickname123> i haven't looked at using it in a year or so I don't really remember what the reasons were. just wanted to double check what was recommended now [01:54] <llogan> lolfrenz: also the input and output sound the same to me (with and without -vn), but i'm in a noisy room. [01:55] <llogan> you could also pipe from ffmpeg to an external encoder such as faac or neroaacenc [01:56] <Nickname123> I am going to give libfaac a try. My hearing isn't great so I defer to others opinions about sound quality [01:58] <Nickname123> What would be the bitrate cutoff for "lower bitrates" that you would reocmmend libaacplus? [02:05] <llogan> Nickname123: i'm not sure. <128k? [02:05] <Nickname123> Okay thanks [02:06] <llogan> with libfaac use -aq 100 if you just care about quality [02:07] <llogan> (see "faac -q") [02:08] <Nickname123> nice [03:58] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax? [04:01] <PurpleSkinnyJean> anyone here use ffmpeg? [04:05] <PurpleSkinnyJean> is anyone even here? [04:05] <brocatz> what [04:05] <brocatz> are you new to irc [04:05] <PurpleSkinnyJean> yes. this is my first time [04:06] <PurpleSkinnyJean> popping my cherry [04:06] <brocatz> okay, basically just wait around [04:06] <brocatz> people are all in different timezones [04:06] <brocatz> ask your question again in a couple of hours [04:06] <PurpleSkinnyJean> how about you? [04:06] <brocatz> if you see poeple chatting, send them a funny cat picture [04:06] <brocatz> then ask your question [04:06] <PurpleSkinnyJean> where do i get those? [04:06] <brocatz> people don't usually answer unless they know [04:07] <brocatz> i don't know [04:07] <PurpleSkinnyJean> can you answer my ? [04:07] <PurpleSkinnyJean> i will send you a nude pic of me if you do [04:08] <brocatz> that sounds risky [04:08] <PurpleSkinnyJean> not at all [04:10] <PurpleSkinnyJean> still there bro bro? [04:10] <brocatz> yep [04:10] <brocatz> i've helped as much as i can [04:10] <PurpleSkinnyJean> is everyone in here new to ffmpeg like us? [04:12] <PurpleSkinnyJean> burek: you! are you online? [04:29] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax? [04:59] <garme> hi, folks. [04:59] <garme> I'm trying to convert some wmv files to avi. [04:59] <garme> I'm using ubuntu 12.04 64 bits. [05:00] <garme> I did: ffmpeg -i input.wmv output.avi... but i got the following error: 'error opening filters' [05:00] <garme> What's happening, guys? [05:05] <diegoviola> garme: what ffmpeg version? [05:10] <garme> diegoviola, 0.8.3 [05:20] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax? [06:01] <relaxed> PurpleSkinnyJean: deinterlace, crop, then scale [06:36] <rm-rf> burek: my command is as follows: ffmpeg -i rtsp://192.168.145.200:554/video.mp4 -vcodec copy -ar 44100 -ab 128k -t 300 /tmp/ffmpeg.mp4 [06:36] <rm-rf> burek: the error i get is: [mp4 @ 0x87fb800] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 245 >= 245 [06:37] <rm-rf> (not trying to cross-post, just posting in the relevant channel after accidentally posting in #videolan) [07:16] <Viking667> hey ho. Can ffmpeg keep up with recording two video streams in addition to two audio streams? [07:16] <Viking667> I have it working well enough to one video stream on a hard disk and another two audio streams to a separate disk. [07:16] <Viking667> ... I was interested in including a webcamera feed in addition. [08:58] <iUnix> Ffmpeg how can i test the H.265? [08:58] <iUnix> currently HEVC draft 7 [09:44] <hi117> iUnix: code it yourself? i would be surprised if theres an encoder for it yet [09:55] <burek> rm-rf, can you please use pastebin.com, to show your command line and its output? [09:57] <JEEB> hi117, the lolreference lolimplementation? [09:57] <JEEB> http://hevc.kw.bbc.co.uk/git/w/jctvc-hm.git [10:01] <iUnix> JEEB, is it the working module according to the Draft 7? [10:07] <gnarface> hey, does ffmpeg support VOB files with multiple camera angles yet? [10:20] <iUnix> http://www.vcodex.com/h265.html - where is those Feb 2012 committee draft? [10:20] <iUnix> July 2012 - Draft international standard [10:20] <iUnix> upcoming [11:31] <astroler> How to decode the smooth streaming without silverlight, have any good suggestions. [11:50] <burek> what is the "smooth streaming" [11:53] <astroler> burek: Smooth Streaming, an IIS Media Services extension, enables adaptive streaming of media to Silverlight and other clients over HTTP. [11:53] <burek> what is so special about it? [11:57] <astroler> seamless switching bit rate. [11:58] <burek> ok [12:59] <voltagex> hey, how do I verify/report a GPL violation? [13:01] <voltagex> http://www.4videosoft.com windows version contains ffmpeg dlls and no mention at all [16:07] <THEBLACKKING> how can i compile ffmpeg with sdl support [16:07] <THEBLACKKING> ? [16:07] <THEBLACKKING> i compiled sdl [16:07] <cbsrobot> then --enable-libsdl (if im not wrong) [16:08] <THEBLACKKING> and configured configured ffmpeg with: [16:08] <THEBLACKKING> ./configure --target-os=mingw32 --cross-prefix=i586-mingw32msvc- --arch=x86 --pkg-config=pkg-config [16:08] <THEBLACKKING> sorry, *cross-compile [16:09] <THEBLACKKING> Unknown option "--enable-libsdl". [16:09] <THEBLACKKING> See ./configure --help for available options. [16:09] <THEBLACKKING> i add --enable-outdev=sdl too [16:10] <THEBLACKKING> but the configure output for sdl support still "no" [16:13] <THEBLACKKING> what should i do ?? [16:32] <zambo> is it possibl to append audio to the audio track of a video using ffmpeg? [16:42] <cbsrobot> zambo: join audio and video together ? [16:43] <cbsrobot> see http://ffmpeg.org/ffmpeg.html#Advanced-options [16:45] <THEBLACKKING> <THEBLACKKING> what should i do ?? [17:03] <shahriman0> if sws_scale sees that input and output parameters are identical will it just pass through the data or does the user have to check that himself? [17:10] <zambo> re: the map command: it looks like this is only for mapping multiple streams in one files, no? What I want to do is append multiple audio files to one audio stream in a video file [17:12] <shahriman0> zambo: multiple audio file to one audio stream? you mean mix the multiple input to one stream and mux it in a video? [17:13] <virtus_> hello, I was wondering if someone could help me out? I'm trying to using the lutrgb filter to split an input FLV into two output MP4s, one for RGB channels, one for ALPHA.. any ideas on best settings? [17:17] <zambo> shahriman0: Yes exactly [17:17] <shahriman0> there's one amix filter afaik [17:18] <zambo> I'm also trying to concatenate the m4a files separately first to make one audio track, but not having much success with that. [17:19] <shahriman0> I am not sure if ffmpeg can concatanate [17:19] <shahriman0> oh it can [17:20] <shahriman0> wait there is a concat protocol [17:20] <shahriman0> zambo: try to use that, I have used it with video before [17:20] <shahriman0> should work for audio too [17:21] <zambo> I highly doubt it will work on Android [17:21] <zambo> but I will try, thank you [17:21] <shahriman0> you doubt without trying? [17:21] <shahriman0> any reason for your doubt? [17:21] <zambo> yeah, if it uses command line shell cat [17:22] <shahriman0> it does not use command line shell cat [17:22] <zambo> I will take a look, thanks again [17:34] <virtus_> suppose at first I only want the RGB channels, what is wrong with this comment: [17:34] <virtus_> ffmpeg -i input.flv -vf "lutrgb=r=val:g=val:b=val:a=0" -vcodec libx264 -vpre slow -vpre ipod640 -b 200k -r 50 output_rgb.mp4 [18:16] <iam8up> i'm trying to get one frame out of an rtsp, so 1) is it possible to do this with ffmpeg and 2) is this the right idea/what's wrong? ffmpeg -i rtsp://admin:passwd@10.10.10.192:554/live/ch00_0 -an clip.jpg and here's my problem/log: http://pastebin.com/XhdTbd8a [20:06] <dericed> could this be true? "you WILL get a better and more accurate result in up-conversion from SD to HD if the original analog video was quantized at 10 bit and you keep the 10 bit data" [as opposed to scaling with 8 bit] [20:12] <cbreak> analog? [20:12] <cbreak> like pal? [20:12] <cbreak> or ntsc? [20:12] <cbreak> those are terrible [20:16] <shahriman0> sales people will say anything [20:26] <zap0> what comes off cameras is analog. [21:04] <zambo> When I try to use the concat protocol with as a parameters for -i, I get a file/directory not found error. I know the files are there, is it an ffmpeg build/config issue? [21:37] <jesk> why does ffserver stops responding to any http request as soon as those few multicast options are configured? [22:20] <function1> is there some way i can tell ffmpeg to use the libavformat implementation of rtmp rather than librtmp? man page seems abiguous... there is a generic listing for rtmp and then one specifically for librtmp, either simply uses rtmp://<server>... [22:26] <saste> function1: if you compiled with --enable-librtmp, ffmpeg always favors librtmp implementation [22:26] <saste> if you want to use native implementation don't use --enable-librtmp [22:26] <saste> please suggest how to change docs in case it is not clear [22:38] <function1> actually... it is clear, i read it too quickly :) [22:38] <nunofgs> can anyone help me? I'm using libavcodec/libx264 to encode frames in h264 and sending them through RTMP. When I try to playback the rtmp stream I get errors like: [h264 @ 0x7fef50828400] Missing reference picture and decode_slice_header error and concealing 396 DC, 396 AC, 396 MV errors [22:39] <nunofgs> but in flash media server I can preview the stream just fine. it's just when I try to use ffmpeg/libav to grab the stream that that happens [22:41] <nunofgs> my encoding must be wrong somehow. I suspect it could be the PTS. does anyone know if I also have to set the DTS and Duration in this scenario? [23:49] <Freakshow> nunofgs: did you post a pastebin? [00:00] --- Fri Jun 29 2012

1 0

ffmpeg-devel.log.20120627
by burek 28 Jun '12

28 Jun '12

[00:02] <ubitux> durandal_1707: are you on cvslog? [00:02] <CIA-119> ffmpeg: 03Mans Rullgard 07master * r0595334892 10ffmpeg/libavcodec/x86/fft_mmx.asm: [00:02] <CIA-119> ffmpeg: x86: fft: elf64: fix PIC build [00:02] <CIA-119> ffmpeg: In a 64-bit PIC build, external functions must be called [00:02] <CIA-119> ffmpeg: through the PLT. [00:02] <CIA-119> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:02] <CIA-119> ffmpeg: 03Hendrik Leppkes 07master * rea1c5011b3 10ffmpeg/libavcodec/dxva2_h264.c: [00:02] <CIA-119> ffmpeg: dxva2_h264: fix signaling of mbaff frames [00:02] <CIA-119> ffmpeg: The MBAFF flag may only be signaled if we're actually dealing with [00:02] <CIA-119> ffmpeg: a full frame, and not singular fields, as it can happen in mixed content. [00:03] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [00:03] <CIA-119> ffmpeg: 03Carl Eugen Hoyos 07master * rfbcaceb1ff 10ffmpeg/libavformat/mov.c: [00:03] <CIA-119> ffmpeg: mov: do not try to read total disc/track number if data atom is too short. [00:03] <CIA-119> ffmpeg: Fixes bug 308. [00:03] <CIA-119> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:03] <CIA-119> ffmpeg: lavfi: remove 'opaque' parameter from AVFilter.init() [00:03] <CIA-119> ffmpeg: mov: do not try to read total disc/track number if data atom is too short. [00:03] <CIA-119> ffmpeg: avconv: fix -force_key_frames [00:03] <CIA-119> ffmpeg: dxva2_h264: fix signaling of mbaff frames [00:03] <CIA-119> ffmpeg: 03Anton Khirnov 07master * ra5e8c41c28 10ffmpeg/libavfilter/ (44 files): [00:03] <CIA-119> ffmpeg: lavfi: remove 'opaque' parameter from AVFilter.init() [00:03] <CIA-119> ffmpeg: It is not used in any filters currently and is inherently evil. If [00:03] <CIA-119> ffmpeg: passing binary data to filters is required in the future, it should be [00:03] <CIA-119> ffmpeg: done with some AVOptions-based system. [00:03] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r19ad567311 10ffmpeg/avconv.c: [00:03] <CIA-119> ffmpeg: avconv: fix -force_key_frames [00:04] <CIA-119> ffmpeg: parse_forced_keyframes() relies in encoder timebase being set, so call [00:04] <CIA-119> ffmpeg: it from transcode_init() after it is known. [00:04] <burek> er.. which of these "ffmpeg -codecs | grep -i jpeg" is a jpeg still image? [00:04] <burek> mjpeg only? [00:06] <ubitux> afaik mjpeg are just like cat *.jpg [00:06] <ubitux> durandal_1707: anyway, if you missed it, http://ffmpeg.org/pipermail/ffmpeg-cvslog/2012-June/051932.html [00:07] <burek> ok, thanks [00:44] <burek> ok, I started the page that will contain all the formats with their supported media types in it https://ffmpeg.org/trac/ffmpeg/wiki/SupportedMediaTypesInFormats [00:44] <burek> if you feel something is wrong, please correct it and feel free to add formats that you know for sure what do they support [00:45] <burek> maybe there is an array defined in ffmpeg for this exact thing? [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r8d900aa4d0 10ffmpeg/libavfilter/ (avfiltergraph.c avfiltergraph.h graphparser.c version.h): lavfi: remove disabled FF_API_GRAPH_AVCLASS cruft [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r6c1e065bd4 10ffmpeg/libavfilter/ (af_resample.c avfilter.h version.h): lavfi: remove disabled FF_API_SAMPLERATE64 cruft [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r0b3b958135 10ffmpeg/libavfilter/ (Makefile buffersrc.c version.h vsrc_buffer.h): lavfi: remove disabled FF_API_VSRC_BUFFER_ADD_FRAME cruft [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r5e88b96f37 10ffmpeg/libavfilter/ (avfilter.c avfilter.h version.h): lavfi: remove disabled FF_API_DEFAULT_CONFIG_OUTPUT_LINK cruft [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r1961e46c15 10ffmpeg/libavfilter/ (avfilter.c avfilter.h formats.c formats.h version.h video.c): lavfi: remove disabled FF_API_FILTERS_PUBLIC cruft [00:47] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r5916bc4658 10ffmpeg/: (log message trimmed) [00:47] <CIA-119> ffmpeg: Merge commit '1961e46c15c23a041f8d8614a25388a3ee9eff63' [00:47] <CIA-119> ffmpeg: * commit '1961e46c15c23a041f8d8614a25388a3ee9eff63': [00:47] <CIA-119> ffmpeg: lavfi: remove disabled FF_API_FILTERS_PUBLIC cruft [00:47] <CIA-119> ffmpeg: lavfi: remove disabled FF_API_DEFAULT_CONFIG_OUTPUT_LINK cruft [00:47] <CIA-119> ffmpeg: lavfi: use proper FF_API guards for different deprecated functions [00:47] <CIA-119> ffmpeg: lavfi: remove disabled FF_API_VSRC_BUFFER_ADD_FRAME cruft [00:47] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r205e90249a 10ffmpeg/libavfilter/avfilter.c: lavfi: use proper FF_API guards for different deprecated functions [01:14] <durandal_1707> ubitux: why would i be on cvs log? [01:14] <j-b> michaelni: 69bf775e9 is broken IMVHO [01:16] <durandal_1707> ubitux: http://lists.libav.org/pipermail/libav-devel/2012-June/029906.html [01:16] <Compn> j-b : how? [01:16] <Compn> if you could explains [01:16] Action: Compn finding a lot of devels giving a lot of bad bugreports [01:17] <j-b> r = _vscprintf(format, va); [01:17] <j-b> without doing a check on vscprintf check... [01:18] <j-b> and, of course, it breaks mingw32 [01:20] <durandal_1707> is it fine to remove all this stupid get_buffer() failed logs? [01:21] <michaelni> j-b, iam not sure i understand (id fix it if i do), but best mail nicrolas, he isnt on IRC [01:22] <j-b> michaelni: well, vscprintf is not defined in all Mingw [01:22] <michaelni> ahh [01:22] <j-b> so, it breaks many implemetations. [01:22] <michaelni> missing confogure check ? [01:23] <j-b> probably, indeed [01:23] <michaelni> ill forward this to nicolas, maybe he has a better idea because [01:23] <durandal_1707> if init of decoder fails is close still called? [01:23] <michaelni> a ciónfigure check would still mean the return value would then be wrong again [01:24] <michaelni> when its not available [01:27] <michaelni> durandal_1707, no i dont think its called [02:07] <CIA-119> ffmpeg: 03Anton Khirnov 07master * rcb81e29138 10ffmpeg/libavfilter/avfilter.h: [02:07] <CIA-119> ffmpeg: lavfi: reorder AVFilterBuffer fields. [02:07] <CIA-119> ffmpeg: Place related fields together, remove holes. [02:07] <CIA-119> ffmpeg: 03Anton Khirnov 07master * rf14e685609 10ffmpeg/libavfilter/avfilter.h: [02:07] <CIA-119> ffmpeg: lavfi: reorder AVFilterBufferRef fields. [02:07] <CIA-119> ffmpeg: Place related fields together, remove holes. [02:07] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r9618080512 10ffmpeg/libavfilter/avfilter.h: [02:07] <CIA-119> ffmpeg: lavfi: reorder AVFilter fields. [02:07] <CIA-119> ffmpeg: Place related fields together, remove holes, move private fields to the [02:07] <CIA-119> ffmpeg: end and mark them as private. [02:07] <CIA-119> ffmpeg: 03Anton Khirnov 07master * r83ba22392d 10ffmpeg/libavfilter/avfilter.h: [02:07] <CIA-119> ffmpeg: lavfi: reorder AVFilterLink fields. [02:07] <CIA-119> ffmpeg: Move private fields to the private section, remove holes. [02:07] <CIA-119> ffmpeg: 03Anton Khirnov 07master * rb8c632a720 10ffmpeg/avconv.c: avconv: don't include vsrc_buffer.h, which doesn't exist anymore [02:07] <CIA-119> ffmpeg: 03Martin Storsjö 07master * r39dba5aa1b 10ffmpeg/libavformat/ (network.h sctp.c tcp.c udp.c): (log message trimmed) [02:07] <CIA-119> ffmpeg: network: Include unistd.h from network.h [02:07] <CIA-119> ffmpeg: This heaader is required for close() for sockets in network [02:07] <CIA-119> ffmpeg: code. For winsock, the equivalent function is defined in the [02:07] <CIA-119> ffmpeg: winsock2.h header. [02:07] <CIA-119> ffmpeg: This avoids having the HAVE_UNISTD_H in all files dealing with [02:07] <CIA-119> ffmpeg: raw sockets. [02:07] <CIA-119> ffmpeg: 03Ronald S. Bultje 07master * re64bceeac0 10ffmpeg/ (configure libavformat/os_support.c): [02:07] <CIA-119> ffmpeg: configure: Check for sys/time.h [02:07] <CIA-119> ffmpeg: Apparently this include is needed on some systems for building the [02:07] <CIA-119> ffmpeg: poll fallback (for the timeval struct for select?), but it isn't [02:07] <CIA-119> ffmpeg: available on all systems. Thus only include it if it exists. [02:08] <CIA-119> ffmpeg: network: Don't redefine error codes if they already exist in errno.h [02:08] <CIA-119> ffmpeg: Since the errno.h values don't match the error codes that winsock [02:08] <CIA-119> ffmpeg: returns, map the winsock error codes to the errno ones, to make [02:08] <CIA-119> ffmpeg: sure explicit checks against AVERROR(x) match. [02:08] <CIA-119> (49 lines omitted) [10:53] <ubitux> http://dev.w3.org/html5/webvtt/ [10:53] <ubitux> seriously [10:53] <ubitux> fuck them all. [10:53] <ubitux> they're stupid or what... [10:54] <av500> ubitux: yes [10:54] <av500> ask j-b [10:54] <ubitux> mmh, @google again [10:54] <ohsix> what are they supposed to do? [10:54] <ubitux> i hate this company more every day [10:55] <ubitux> ohsix: re-use an existing one? [10:55] <ohsix> which one [10:55] <ubitux> they don't even support SRT properly on youtube [10:55] <ubitux> (they print <i> tags in the subtitles....) [10:56] <ubitux> ohsix: subrip is fine, ass as well, TED use some json based one, ... [10:56] <ubitux> i mean they are tons of them [10:57] <ohsix> would be nice if there was a requirements section [10:57] <ohsix> it is pretty dumb to not use an existing one, unless you can't [10:57] <ubitux> and again they are using smil based crap :( [10:57] <ohsix> smil is used on phones for everything [10:59] <ubitux> anyway, one more subtitles format to add on my todo list [10:59] <ubitux> hopefully it will never be used [11:00] <ohsix> looks like programmatic access is paramount [11:00] <ubitux> it seems to be used by video.js [11:01] <ohsix> at least in the context it'll be seen most, google will probably support it with code as well :] [11:01] <JEEB> I think nielsm talked with W3C on as6, but I don't think it went anywhere as the first alpha implementation of as6 was made just a month+ ago [11:02] <JEEB> (and the talks were had 1.5-2 years ago [11:02] <ubitux> do you know if as6 will be backward compatible with ass? [11:03] <JEEB> it will have some similarities, but it will also try to basically be better spec'd and somewhat of a different format. Not sure if nielsm released his working drafts yet. The only real docs on it I've seen on it was the readme file that came with the "crashes more than works" test implementation [11:04] <JEEB> http://www.animereactor.dk/aegisub/as6render-20120521_1000.rar [11:05] <ohsix> these things, very important [11:05] <JEEB> if you want more info on as6 I recommend poking nielsm on it [11:06] <j-b> ubitux: very [11:06] <ubitux> :( [11:06] <JEEB> I think he didn't want to release specs publically yet to try and stop what happened with the AS5 which was never really born as random people started implementing non-finished specs [11:07] <JEEB> (and AS5 was IIRC "stuff on top of ASS" instead of a new format, but I must confess I never even checked the as5-related stuff in aegisub's svn repo) [11:07] <ohsix> does it do css3, lul [11:07] <ubitux> :D [11:08] <ubitux> SAMI supports 100% of CSS2 [11:08] <ubitux> this makes SAMI a superior subtitles format [11:10] <ohsix> it's all about context tho [11:10] <JEEB> https://privatepaste.com/157ca0ccd0 <- the readme of as6render [11:10] <ohsix> that stuff never just lives in a typed resource on a webpage, it's always munged and shuffled about [11:11] <ohsix> people aren't using it to subtitle anime, they're using it for the hearing impaired [11:12] <j-b> SAMI is crap [11:12] <j-b> WebVTT is beyond crap [11:12] <ubitux> :) [11:12] <j-b> and Anime formats are bad. [11:12] <av500> :) [11:12] <ohsix> so there's one true format, why do they still need webvtt [11:13] <j-b> there is not one true format. [11:13] <ubitux> NIH [11:13] <ubitux> xkcd/927 [11:13] <ohsix> what about not fit for purpose [11:14] <JEEB> anyways, I hope something comes up of AS6 or there is a better alternative found -- so that people can move on from the mess that was created by Gabest [11:14] <JEEB> (never going to happen, I know) [11:14] <j-b> We need a new format, indeed. [11:14] <j-b> and WebVTT is not the solution. [11:14] <JEEB> indeed [11:14] <ubitux> a new format? oO [11:15] <ubitux> ASS is fine, AS6 will just help covering more stuff [11:15] <JEEB> ASS is not fine because the spec is vsfilter [11:15] <ohsix> their requirements are probably different from yours [11:15] <ubitux> JEEB: the specs could be written [11:15] <JEEB> and by now various vsfilter versions disagree with each other, too [11:15] <ohsix> specs don't mean compliance [11:15] <ubitux> also, there are already implementations [11:16] <JEEB> yes, which is why ASS is being used atm [11:16] <j-b> ubitux: no, ASS is not fine. [11:16] <ohsix> w4m [11:16] <ubitux> j-b: except the timing issue, i don't see why [11:16] <j-b> ubitux: it is a bad format, not complete, and not even specified [11:17] <ubitux> then we should just improve the documentation [11:17] <JEEB> also, hands up who here remembers CoreCodec's XML'ization of ASS? [11:17] <j-b> with a lot, and I mean a lot of unspecified behaviour [11:17] <ubitux> implementation are used and working for most of the stuff [11:17] <j-b> Moreover, it is non-streamable [11:17] <j-b> and near impossible to parse. [11:17] <ohsix> does it support css3 [11:17] <j-b> aka crap [11:18] <j-b> it is a fun toy, but not a serious format. [11:18] <ubitux> why isn't it streamable? we are able to make ass packets in mkv [11:18] <j-b> right. Cut a ASS file in 2. [11:18] <ubitux> you just need to sort the dialogues by pts and put the header stuff somewhere [11:18] <j-b> play both. [11:18] <ubitux> j-b: i don't understand; just dup the header? [11:20] <ohsix> it's all bad, give up [11:20] <j-b> it is bad. [11:21] <ohsix> for the purpose of discussion, since bad hasn't been defined, i'll define it as good [11:21] <ubitux> :) [11:22] <j-b> We need a strong format, that is streamable, correctly parseable by browsers and players and boxes, that can allow any cool style, and can be used for timed-text information [11:22] <ohsix> i don't, i need one that is readable [11:22] <j-b> why? [11:23] <av500> j-b: I dont need one, I can understand english and I dont watch anime :) [11:23] <j-b> av500: :) [11:23] <ohsix> vision and foreign language impaired [11:23] <ubitux> what is a "strong" format? [11:23] <ubitux> a streamable format, except formats like SAMI or microdvd supporting the "last-to-the-next-event" feature, most of them are [11:24] <ubitux> parseable by browsers? seriously no [11:24] <ohsix> i also don't care where it is on the frame or what it obscures, or how clever people can be with it; it's for the suboptimal case of me not being able to hear it at all, or to understand it [11:24] <ubitux> it means SMIL based crap [11:24] <ubitux> it's a hell [11:24] <ubitux> overhead, unreadable, stupid, etc [11:24] <ohsix> it's not bad if you _are_ a browser :] [11:24] <j-b> ubitux: you are dellusional. [11:24] <ubitux> "any cool style" no& see how much pain it is to implement this [11:25] <ubitux> look at the time it took to dev libass, and how much problems it raises.. [11:25] <j-b> it took a lot of time because ASS is crap [11:25] <j-b> doing the same for USF was very faster to do. [11:25] <ubitux> lol usf [11:25] <ubitux> :D [11:25] <JEEB> lol USF [11:26] <ubitux> doing a rendering engine for subtitles is hell [11:26] <ubitux> font issues, vector stuff, etc [11:26] <j-b> then, don't take any format. [11:26] <ubitux> no, we just need to have a common rendering system [11:26] <ubitux> we have one, and it's libass [11:26] <ohsix> yes, rendering them is crazy hard [11:26] <ubitux> we should use it for all the subtitles [11:27] <j-b> you are confusing the worlds need and yours. [11:27] <ubitux> actually, that's already the case [11:27] <ohsix> more like cairo and pango \m/ [11:27] <j-b> and because of answers like yours, we got WebVTT [11:27] <ohsix> what, heh [11:27] <j-b> and it will be a major format, that we will need to live with. [11:27] <ubitux> [ffmnpeg] % ./configure --enable-webkit --enable-cairo --enable-js [11:27] <ohsix> they wrote it for their requirements [11:27] <ubitux> yepee [11:28] <ubitux> j-b: i never requested webvtt [11:28] <j-b> you just did. [11:28] <j-b> 11:24 <@ubitux> parseable by browsers? seriously no [11:28] <ubitux> then use json [11:28] <ubitux> at least it's a sane base [11:28] <ohsix> eh [11:28] <j-b> of course, it needs to be json or xml based. Without header, or with repeatable header. [11:29] <ubitux> without header? [11:29] <ubitux> and with aa lot of styles? [11:29] <ubitux> i think you are dreaming [11:29] <ubitux> i don't see the point of dropping the header [11:29] <j-b> try to read my sentence. [11:29] <j-b> just try. [11:29] <ohsix> style can be separate, the wonders of html [11:30] <ubitux> j-b: isn't ASS header repeatable? [11:30] <j-b> maybe, but it does not fit, because of under-specification and because of parsable format. [11:31] <ohsix> if you have your own requirements it might be a good start to write them down :] nevermind specifications [11:31] <j-b> I do not have requirements. I just look a bit at the industry [11:31] <j-b> and I've been at EBU, IETF, W3C meetings [11:31] <ubitux> the industry doesn't care, they use bitmap subtitles anyway [11:31] <ohsix> then you don't need a new anything [11:32] <ohsix> aside from the size, what's wrong with bitmap subtitles? :D [11:32] <ubitux> no particular issues [11:32] <ohsix> aside from not being machine readable, or always legible, or really any standard [11:32] <ubitux> except that it's hard to convert them to text ;) [11:33] <ubitux> of course it has a lot of benefits [11:33] <ubitux> since it doesn't require your dvd player to use fontconfig [11:33] <j-b> of course, it does not fit the accessibility requirement. [11:33] <ubitux> sure, it's another issue [11:33] <ohsix> not being machine readable is a huge downside [11:34] <j-b> well, it just ruled-out bitmaps [11:34] <ubitux> maybe you can name the bitmaps with the text subtitles content for accessibility ;) [11:34] <ubitux> you should be able to do that with mxf [11:34] <ubitux> muxing named-png's for subtitles [11:34] <ohsix> you could just do regular vbi sub text too [11:34] <ubitux> well anyway [11:35] <ubitux> the industry will just invent N more subtitles formats [11:35] <ubitux> needing N more rendering engines [11:35] <ohsix> when [11:35] <ubitux> no one will ever implement [11:35] <ubitux> except by wrapping them around libass [11:35] <ubitux> problem solved by itself. [11:39] <ohsix> shrug, i dunno what they're supposed to do if nothing that currently exists meets their requirements, even if they may with negligable effort; you have to deal with the people too :p it probably works in their favor that people dislike it on its face [11:41] <ubitux> they should just help writing the ASS specifications, and define a way of muxing it in JSON, just like matroska defined a way of muxing it in MKV [11:42] <ubitux> re-use the existing, limit the insane from-scratch rewriting, and help the current situation instead of trashing it even more [11:42] <ohsix> that takes time [11:42] <ubitux> less than rewriting a specification [11:43] <ubitux> and wait for partial implementation [11:43] <ohsix> plus there's personalities involved that would cause problems [11:43] <ohsix> there's nto a lot the webkit(/apple/google) & mozilla guys do halfway with that sort of thing [11:44] <ubitux> ok let me help you: {"header": "[ASS+ Style v42]...", {"events": {"pts": "00:01:02.012", "duration: "00:00:10.000", "text": "..."}, {"dialogue": ... [11:44] <ubitux> here you have it [11:44] <ubitux> you just need a json parser and here we go. [11:45] <ohsix> do we cite the webpage or whatwg [11:45] <ohsix> s/webpage/irc log [11:45] <ubitux> is that much complicated than rewriting a spec from scratch with no idea what they are doing? [11:46] <ohsix> how do you know they have no idea what they're doing? at the very least they know what they need or want subtitles for, and how it has to work to facilitate that usage [11:46] <ubitux> look at the result [11:46] <sandsmark> ubitux: much less satisfying for the ego, though :p [11:46] <ubitux> do you think SAMI, RT or webttv are sane? [11:46] <ubitux> or even USF... [11:47] <ohsix> i have no opinion, other than i know the workflow & how things are typically employed in a browser [11:49] <ohsix> i don't think i'd propose it for some streamable or embeddable subtitle format for general use outside of a browser either [11:50] <ohsix> that would make it way more than what it currently is [11:50] <ubitux> well, xml & json are not that streamable [11:50] <ubitux> a plain text like subrip is though [11:50] <ubitux> (oh it's implemented everywhere already.) [11:50] <ubitux> ohsix: a parser is not that a problem [11:50] <ubitux> the problem is mostly about styles [11:51] <ubitux> or the decoder part if you prefer [11:51] <ohsix> i'm sure there's a discussion on the w3c/whatwg lists about this, i don't know what their requirements actually were and why they aren't able to use subrip, or anything else [11:51] <ubitux> you could define multiple way of muxing ASS [11:51] <ohsix> is ass plaintext? [11:52] <ubitux> we already know how to mux ASS in the MKV, we could define a "json format" to mux them (see above) or a plaintext one [11:52] <ubitux> sure [11:52] <ohsix> that seems silly when you can just come up with something that's html-alike, stylable, uses familiar elements [11:57] <ohsix> it's kind of disjointed to venture out of that sort of thing for browser use, unless it's a completely binary something you're not supposed to really touch; but tell the browser to do things with it [11:57] <ubitux> video & audio are already binary, i wonder what's the problem with muxing non-browser-compliant subtitles [11:58] <ohsix> timing and styling in the dom or with javascript would be inflexible and dumb with a binary format [11:58] <ohsix> if something is actually multiplexed with the video i don't see why it couldn't be subrip or anything else, but this isn't that [13:29] <burek> sweet :) http://en.wikipedia.org/wiki/Comparison_of_container_formats [14:17] <av500> asf does not support b-frames, interesting [14:17] <av500> vc1 with b-frames in ASF works fine.... [14:18] <JEEB> as far as I know ASF is just fine with b-frames [14:18] <JEEB> it's not AVI after all [14:18] <JEEB> tl;dr "lol wikipedia" most probably [14:19] <JEEB> http://wiki.multimedia.cx/index.php?title=Microsoft_Advanced_Streaming_Form… I lol'd at the "There are 2 versions of ASF" [14:20] <JEEB> > 1.0, the used and unpublished format > 2.0, the published and unused format [14:20] <Compn> lol [14:26] <JEEB> also, oh boy @ the ASF version 2's specs [14:26] <JEEB> lots o' pretty colors [14:27] <av500> the specs from M$ match more or less the files I have [14:27] <av500> dunno if that is 1 pr 2 [14:27] <av500> or [14:35] <jesk> ok, i got it to manage inserting free in front of mdat [14:35] <jesk> but it seems there is no application which wants to make use of free :D [14:35] <jesk> all i tried will rewrite anyway [14:35] <JEEB> unsurprising [14:35] <jesk> \o/ [14:37] <jesk> JEEB, could you explain me why this isn't suprising you? :-) [14:37] <JEEB> because no-one was adding arbitrary extra free space into their files [14:38] <JEEB> and thus no-one even implemented that possible use case [14:38] <jesk> but all will remove free :D [14:38] <jesk> brilliant [14:38] <JEEB> yes, because in most cases it is just useless cruft) [14:39] <JEEB> it's not mpeg-ts we're talking about which has a set mux rate (which will be filled with null packets to reach that goal) [14:39] <jesk> this useless cruft could help to reduce tagging time from hours to seconds, but instead all are ore interested in saving Kilobytes :D [14:40] <JEEB> I do understand your point, but most people either don't tag, or tag during their initial mux [14:41] <jesk> from code point of view this shouldn't be much effort to implement, right? [14:42] <jesk> i would guess that the implemented rewrite is 10 times more code [14:43] <jesk> but i'am just big mouth without clue [14:43] <jesk> oh I forgot: [14:44] <jesk> iTunes is honouring free [14:44] <jesk> one of ten I tried [14:45] <jesk> but iTunes won't fetch anything automatically [14:47] <JEEB> well, I'm surely not against people implementing that stuff, but you just have to understand that the use case has been rather small until now which hasn't really made anyone before implementing it [14:48] <JEEB> in most cases it's just mux with tags -> do an index move to the beginning [14:49] <jesk> as I understand MP4 is an very advanced container format with very poor support, although its there for yeeears [14:49] <jesk> but could be wrong [14:50] <av500> jesk: just adding free at the front does not help if the MOOV atom is at the end [14:50] <JEEB> no-one fully implements the ISO media container :) [14:50] <jesk> but chance is there anyway that I just missed the right tool which could this magic [14:51] <jesk> av500, true, but AtomicParsley seems to work fine with placing moov at the front [14:52] <jesk> /and/ place free behind it [14:52] <JEEB> jesk, my short opinion on the container can be condenced into http://forum.doom9.org/showpost.php?p=1580058&postcount=3 [14:54] <jesk> i had look at the specification from Apple [14:54] <jesk> its overloaded with details on all those atom types [14:54] <jesk> the whole idea about is is very hard to get out of the spec [14:55] <jesk> would be cool if there would be at least one OSS reference implementation [14:56] <JEEB> other than libavcodec, there are two or so still upkept mov/mp4/ISO media container implementations [14:56] <JEEB> GPAC and L-SMASH [14:56] <jesk> Loyal to Spec of Mpeg4 and Ad-hoc Simple Hackwork. [14:57] <jesk> never heard about L-SMASH [14:57] <JEEB> it's a newer thing, started in 2011 or so because GPAC was getting way too huge and derpy, as well as failing with various things [14:58] <jesk> so its worth a look [14:59] <JEEB> definitely [14:59] <jesk> hopefully it will compile on my mbp :) [14:59] <JEEB> it's somewhat raw from certain sides (like the raw stream muxer being separate from the remuxing app) [14:59] <JEEB> yeah, it should [14:59] <JEEB> it doesn't have many dependencies [15:00] <JEEB> I've used the boxdumper tool at times to troubleshoot certain files [15:02] <jesk> those mp4 gurus should unify and build ninja-mp4-toolsuite [15:03] <JEEB> I think GPAC is quite hardened on their position and all :) [15:06] <jesk> gpac is brilliant in deleting free, just adding is missing :) [15:07] <jesk> oh yah l-smash builds without gap [15:09] <jesk> boxdumper seems a lot like AtomicParsley [15:09] <jesk> just more verbose [15:11] <jesk> cool stuff [15:15] <jesk> impressive how man chunks there are [22:58] <ubitux> saste: do you mind commenting on '[PATCH 4/6] documentation: change "Libavfilter" link to "Filters".' ? [22:58] <ubitux> i'm going to push half the patchset, but i'm still unsure about this patch [23:02] <CIA-119> ffmpeg: 03Michael Bradshaw 07master * r54942c2383 10ffmpeg/libavcodec/avcodec.h: [23:02] <CIA-119> ffmpeg: lavc: clarify docs for avpkt->destruct [23:02] <CIA-119> ffmpeg: avcodec_encode_audio2 docs are ambiguous about avpkt->destruct and imply [23:02] <CIA-119> ffmpeg: it gets reset. [23:02] <CIA-119> ffmpeg: Signed-off-by: Michael Bradshaw <mbradshaw(a)sorensonmedia.com> [23:02] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:03] <CIA-119> ffmpeg: 03Eric Petit 07master * rf1136b2b10 10ffmpeg/libavformat/udp.c: [23:03] <CIA-119> ffmpeg: udp: fix occasional crash on shutdown [23:03] <CIA-119> ffmpeg: Wait until the thread is down before destroying the fifo [23:03] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:18] <jesk> is there any common reason why ffserver ignores any http request to it? [23:19] <jesk> port is open, but that's all. [23:34] <ubitux> saste: well anyway, i did a s/libavfilter/filtering/, i'll push soon [00:00] --- Thu Jun 28 2012

1 0

ffmpeg.log.20120627
by burek 28 Jun '12

28 Jun '12

[00:03] <burek> rm-rf, if one of your cams is already encoding the video in mp4, then no need to re-encode it again [00:05] <undercash> hello [00:05] <undercash> i use this command to stream on justin [00:05] <undercash> http://pastebin.com/Tgsqehac [00:05] <phoenixson> what video formats are trailers on imdb in? [00:06] <undercash> i m wondering if it s possible to add outputs so i could send that stream to ustream as well? [00:07] <undercash> one transcoding.. 2 broadcasts. [00:12] <burek> undercash it might be [00:13] <undercash> ;) [00:13] <burek> but you are using rtmp [00:13] <burek> does ustream also use rtmp? [00:14] <undercash> good question [00:14] <undercash> i guess? [00:14] <undercash> let me check.. [00:14] <burek> one way I can think of is: ffmpeg -i input -vcodec ... -acodec ... | ffmpeg -i -c copy output1 -c copy output2 [00:15] <burek> but it's a quest to find out which format to use for pipe :) [00:15] <burek> anyway, that way you would use 1 ffmpeg to encode all you need and another one to dup the output to 2 destinations [00:16] <burek> effectively using only 1 encoding process and 2 muxing processes [00:18] <undercash> i can't add like a second output in the bash [00:18] <undercash> like out_file2="-f flv rtmp://ustream.tv/app/streamkey" [00:26] <undercash> yea they provide rtmp link [00:26] <undercash> donno about this example you proposed [00:27] <ShinyObjects> Hey all, I'm reading through ISO 14496-12 to learn about AVC Composition Times. [00:28] <ShinyObjects> It's sort of dense reading and doesn't supply much context (for instance, what is a composition time and why do I need it?") [00:28] <ShinyObjects> Does anyone have a link to a slightly more human-oriented explanation? [00:32] <JEEB> ShinyObjects, http://mpeg.chiariglione.org/faq/mp4-sys/sys-faq-esm.htm#ESM-27 [00:32] <ShinyObjects> Appreciated, JEEB [00:32] <JEEB> I would think this is similar [00:34] <ShinyObjects> Interesting. Ok, that makes a little more sense. I'm not sure why it's so important yet, but at least that tells me what they are :) [00:34] <ShinyObjects> Thanks again JEEB: it would have taken me a long time to find that fairly simple thing out. [00:42] <burek> undercash, you can, but it will use 2 encoding processes [00:44] <undercash> really [00:44] <undercash> not cool [00:45] <undercash> anyway they provide rtmp url + stream name, don't know how to put this in a single output url [00:45] <undercash> unlike for justin [00:45] <burek> iirc it has got something to do with implementing sinks and sources and it might be implemented now [00:45] <burek> I didn't check it [00:46] <burek> so I might be wrong [00:46] <burek> the best would be to start the cmd line and see cpu usage [00:46] <burek> if it doubles, then im right :) [00:48] <undercash> yep [00:49] <undercash> i did a quick test, the rtmp url for ustream doesnt work.. [00:49] <undercash> so donno if it would duplicate the encoding [00:50] <malinens> hi guys! does sse4.1 CPU feature is big factor in encoding ffmpeg with x264? [00:51] <llogan> someone put a dirty diaper in my mail box... [00:52] <malinens> I have one little bit older cpu with sse4.1 and I get 250 frames per second but with "more powerful" CPU without sse4.1 I get only 15 frames per second [00:53] <burek> undercash, try to set the 2nd output to a simple file [00:53] <undercash> why not yes [00:53] <undercash> good idea [00:53] <malinens> sorry, not 15 but 150 [00:53] <burek> malinens, well use the old one then :) [00:54] <malinens> nope, I need more than one server for ffmpeg [00:54] <llogan> are your ffmpegs using the same version and configure? [00:54] <llogan> did you disable-yasm? [00:55] <undercash> it works [00:55] <undercash> 2 output, and one transcoding [00:56] <malinens> one is ubuntu with 0.10.4 and other I just compilled on centos using this guide: https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide [00:56] <undercash> just need the f*** ustream rtmp.. [00:56] <undercash> :) [00:57] <burek> :) [00:59] <malinens> also on ubuntu my ffmpeg compilation is little bit weird: http://pastebin.com/DzCNedat but it still works better than on centos [01:01] <malinens> on centos I don't have such creepy stuff: http://pastebin.com/tmQeLWwJ [01:08] <malinens> it seems CPU speed is a factor. on centos I have 4 cores E5320 @ 1.86GHz but on ubuntu I have 4 cores Intel(R) Core(TM)2 Quad CPU Q9650 @ 3.00GHz [01:08] <malinens> both are xeons [01:09] <malinens> and admin told me that centos has more powerful CPU... bad admin :D [02:06] <Freakshow> any examples of using ffmpeg to output hls? [02:21] <Freakshow> yeah, that 10 line blurb is less than helpful [02:21] <Freakshow> to the googles! [02:29] <Nickname123> Can anyone tell me why the following configure doesn't build ffplay? [02:29] <Nickname123> http://pastebin.com/icwvdxCi [02:30] <Nickname123> I am building on windows with msys [02:34] <Freakshow> can you supply any relevant bits from the failure? [02:34] <Nickname123> I didn't see any errors or failure. It just wasn't built [02:35] <Nickname123> I didn't know if I had incompatible flags or something [02:36] <Freakshow> hmm... how about setting a logfile to see what happened? [02:36] <Nickname123> okay [02:36] <Freakshow> ./configure --logfile=./config.log <options> [02:37] <Freakshow> this could only be partially correct as I'm pulling my details from building in osx and freebsd [02:37] <Freakshow> however, I can't imagine that 'wouldn't' work [02:39] <Nickname123> running config with the following script http://pastebin.com/ar1XXdZf [02:39] <Nickname123> painfully slow for some reason. building now [02:39] <Nickname123> *after i mean [02:40] <Nickname123> making now [02:41] <Nickname123> here was the config.log: http://pastebin.com/Ad9Wh9sx [02:54] <Nickname123> Forgot to capture make output. Redoing it [03:05] <Nickname123> That one messed up... http://pastebin.com/T9izQnCx but it didn't mention ffplay [03:14] <burek> Nickname123, ffplay_deps='avcodec avformat swscale swresample sdl' [03:14] <burek> check if there are any errors regarding those deps [03:14] <burek> c:/users/gordon/mingw/bin/../lib/gcc/mingw32/4.6.2/../../../../mingw32/bin/ld.exe: reopening ffmpeg_g.exe: Permission denied [03:14] <burek> :D [03:19] <Nickname123> hrm I got a permission error again running in an administrator command windows [03:19] <Nickname123> *window [03:22] <Nickname123> any idea what the permission denied is about or how to correct it? [03:23] <Nickname123> on windows 7 using msys and mingw [03:24] <Nickname123> didn't want to let it get the best of me but i am having the hardest time lol [03:51] <Robert_> Could somebody help me on the following code as to why it doesnt encode [03:51] <Robert_> ffmpeg -i infile.mpg -vcodec libx264 -b:v 1400k -vf scale=624:352 -preset fast -threads 0 -acodec libvo_aacenc -ab 192k outfile.mp4 [03:51] <Robert_> the infile audio is ac-3 and it wont let me encode [03:53] <Robert_> i get this error everytime [libvo_aacenc @ 037F6EE0] Unable to set encoding parameters [03:53] <Robert_> Output #0, mp4, to 'E:\xampp\htdocs\dvd\WWE.Falls.Count.Anywhere.2012.DVDRip\cd [03:53] <Robert_> .mp4': [03:53] <Robert_> Stream #0:0: Video: h264, yuv420p, 624x352 [SAR 352:351 DAR 16:9], q=-1--1, [03:53] <Robert_> 1400 kb/s, 90k tbn, 29.97 tbc [03:53] <Robert_> Stream #0:1: Audio: aac, 48000 Hz, 5.1(side), s16, 192 kb/s [03:53] <Robert_> Stream mapping: [03:53] <Robert_> Stream #0:0 -> #0:0 (mpeg2video -> libx264) [03:53] <Robert_> Stream #0:1 -> #0:1 (ac3 -> libvo_aacenc) [03:53] <Robert_> Error while opening encoder for output stream #0:1 - maybe incorrect parameters [03:53] <Robert_> such as bit_rate, rate, width or height [03:53] <Robert_> whoops sory didnt know about pastebin just read it ughh sorry guys [03:56] <Robert_> heres the paste sorry again for copying here http://pastebin.com/Uxpb1aNx [03:56] <sacarasc> Probably gotta make it only have stereo, not 6 channels. (The audio.) [03:57] <Robert_> hmm how do i do that? [11:52] <rainmaker1> Hello, anyone have experience with fragmented mp4 files? I am trying to save multicast input (h.264 and aac) to fragmented mp4 file so I can place atom on the begining of file, but when I try to play it with vlc it just play a few seconds of video and then stops [12:21] <xero-exez> rainmaker1: qt-faststart [filename] [12:33] <murali> hi, while compiling ffmpeg from source, based on the following link http://ffmpeg.gusari.org/viewtopic.php?f=25&t=38 i'm getting the following error. LD ffmpeg_g libavcodec/libavcodec.so: undefined reference to `x264_bit_depth' libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_125' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 Please provide some help. [12:36] <murali> burek. you there? i hope you remember me. yesterday we had some conversation based on this. [12:40] <burek> murali, type ldconfig [12:41] <murali> burek, it didn't work. [12:41] <burek> ok, make distclean [12:41] <murali> i compiled x264 and then typed ldconfig [12:41] <burek> ldconfig [12:41] <burek> ./configure... make... [12:41] <murali> ok. wait. [12:41] <burek> do you have any other x264 installed [12:41] <burek> (from your repository maybe) [12:41] <murali> you mean, in the x264 directory? [12:41] <burek> dpkg -l | grep 264 [12:41] <murali> ok. i'll check. [12:42] <burek> if you have, uninstall it, since you dont need it if you compile the latest x264 [12:43] <murali> ok. gimme 5 mins. i'll do all these steps. [12:57] <murali> burek, thanks a lot. the compilation is done. now when running the ffmpeg command it displays [root@SS59-MAA ffmpeg]# ./ffmpeg ./ffmpeg: error while loading shared libraries: libavfilter.so.3: cannot open shared object file: No such file or directory [12:57] <murali> but, i can see all the so and so.* files are present in all the respective directory. [12:58] <burek> ldconfig [13:02] <murali> oops. correct. [13:05] <murali> burek. thanks a lot for your support and patience. will use this knowledge in future too. this irc is awesome and a nice place to get real help. [13:05] <burek> :beer: :) [13:07] <murali> ha ha.. sure. i'm an indian. i'll support KingFisher only. :-) [13:08] <burek> the best one [13:08] <burek> is a cold one :D [13:08] <murali> oh really? you that? which country are you from? [13:08] <burek> serbia :) [13:09] <murali> oh, that's a wow. thanks a lot and will serve you a KF when you come to india. [13:09] <burek> it's a deal :) [13:09] <murali> yeah. sure. you can count on me. :) [13:13] <varaderoguy> morning ffmpegers [13:15] <microchip_> \o [13:29] <jesk> anyone knowing mp4 tagger with imdb or similar online-db support? [13:29] <jesk> osx support would be cool, but not a must [13:29] <jesk> cli prefered, but not a must [13:30] <jesk> but must have is ability to *not* rearrange atom structure [13:31] <jesk> all taggers i tried are too stupid to consider free-atom [13:31] <jesk> or are rewriting the whole file although free is there [13:32] <jesk> *free and moov in front of mdat [13:32] <burek> what does it have to do with ffmpeg? :) [13:33] <burek> you have better chances with google I guess [13:33] <jesk> mp4 metadata is one of the abilities of ffmpeg :) [13:34] <jesk> i believe chance is not bad to get answer here :) [13:36] <jesk> google is another time clueless about that [13:51] <AlRazi> I'm facing a number of issues converting mp3 to HE-AAC .. can anyone help out please ? [14:01] <burek> yes [14:01] <burek> AlRazi, can you please use pastebin.com, to show your command line and its output? [14:02] <AlRazi> of course just a sec, thanks [14:03] <AlRazi> http://pastebin.com/dx9FCtgf [14:03] <AlRazi> when i try to play the resulting m4a file on itunes, it gives 7000+ hours duration [14:03] <AlRazi> i tried qt-faststart to no avail [14:04] <AlRazi> the only case that it actually worked was when i produced wav from ffmpeg, and used faac to convert wav to m4a [14:04] <AlRazi> the issue here is that faac does not support HE-AAC [14:05] <burek> AlRazi, can you provide complete output [14:05] <burek> not just cmd line [14:06] <AlRazi> yes second [14:07] <AlRazi> http://pastebin.com/xfU9z3p0 [14:10] <AlRazi> this one actually works, let me try the problematic mp3, sorry for that : / [14:10] <burek> ok :) [14:11] <AlRazi> http://pastebin.com/JubJ5wj3 [14:12] <AlRazi> Time on itunes -> 789:57:13 [14:13] <burek> can you type ffmpeg -i out4.m4a [14:14] <AlRazi> http://pastebin.com/DV5sv1ma [14:15] <burek> can you try mediainfo on that file [14:16] <burek> also vlc player [14:16] <burek> and some other media player [14:16] <burek> just to check [14:16] <AlRazi> mplayerx plays it [14:16] <burek> because the output looks fine [14:17] <AlRazi> but if i try to stream the file, the player pauses till it downloads it all and then starts streaming [14:17] <burek> which player [14:18] <AlRazi> mplayer [14:18] <AlRazi> oh [14:18] <burek> use qt-faststart [14:18] <AlRazi> soundmanager2 and jplayer [14:18] <AlRazi> i tried the python script [14:18] <AlRazi> the output said that atom was moved [14:18] <AlRazi> but same results [14:18] <burek> does vlc play it fine? [14:19] <AlRazi> don't have it installed, but mplayer plays it [14:19] <AlRazi> should I try vlc per say ? [14:19] <burek> I'm not sure [14:19] <burek> but if qt-faststart didn't complain [14:19] <burek> then I guess the file is fine [14:20] <AlRazi> yeah the python wrapper [14:20] <burek> try several media player just to be sure there are no odd errors [14:20] <AlRazi> but the example i sent you earlier [14:20] <burek> just to make sure is it the file that's bad or the itunes player [14:20] <AlRazi> did work [14:20] <AlRazi> the m4a give correct duration [14:20] <burek> well I don't see any problems in ffmpeg's output [14:21] <AlRazi> the difference in the two outputs are max_analyze_duration warning [14:21] <AlRazi> the ( Estimating duration from bitrate, this may be inaccurate ) [14:21] <AlRazi> and the ( Multiple frames in a packet from stream ) [14:21] <AlRazi> steam 0* [14:22] <burek> well [14:22] <burek> you can increase analyze duration if you like [14:22] <burek> ffmpeg -analyzeduration 10000000 ... [14:23] <AlRazi> warning still there can i link you the command only ? [14:23] <burek> increase it to 20000000 [14:24] <AlRazi> ffmpeg -i abc.mp3 -acodec libaacplus -ab 64K -ar 44100 -vn -analyzeduration 100000000 -map_metadata g out4.m4a [14:24] <burek> no man.. [14:24] <burek> look what I wrote above [14:24] <burek> ffmpeg, followed by analyzeduration option and then all the other options [14:25] <AlRazi> doh [14:25] <burek> the order of options does matter [14:27] <AlRazi> so far when i increase the warning still pops up at the new higher value [14:27] <burek> then your input mp3 might be damaged or something [14:28] <Spideru> ffplay -i rtp://<ip>:<port> return: "At least one output file must be specified". How can i say to ffplay to use sound output on my windows XP? Thank you [14:29] <burek> Spideru, can you please use pastebin.com, to show your command line and its output? [14:29] <Spideru> ok [14:29] <burek> also, you don't need -i when using ffplay [14:30] <Spideru> oh gosh i'm a donkey [14:30] <Spideru> last command was not ffplay but ffmpeg [14:30] <burek> :) [14:30] <Spideru> tab completition fails [14:31] <AlRazi> burek, sorry for this but one last question .. can i eleminate ( Estimating duration from bitrate, this may be inaccurate) ? coz by mere observation, when this pops up the resulting m4a is problematic [14:32] <burek> AlRazi, when you get that message [14:32] <AlRazi> in the conversion output [14:32] <burek> it means that input file did not contain such info (in metadata or container) so ffmpeg tries to calculate it itself [14:32] <burek> it's just a warning, nothing to worry about [14:32] <gnarface> even if the bitrate it detects is "bitrate: -2147483 kb/s" ? [14:33] <burek> if you are concerned if the input is invalid or not, then unpack your mp3 to wav [14:33] <burek> ffmpeg -i bla.mp3 out.wav [14:33] <burek> and use wav as an input for aac+ conversion [14:33] <AlRazi> using ffmpeg again ? [14:33] <burek> gnarface, such bitrates are usually detected on live streams [14:33] <AlRazi> i'll try it now [14:34] <burek> or if your ffmpeg is old :) [14:35] <gnarface> burek: 0.8.3 too old? [14:35] <burek> well I can't tell, I never used it [14:35] <burek> I always use latest git [14:36] <burek> to make sure all the bug fixes are up to date [14:41] <latenite> Hi folks, I have a timelapse made out of tiff images. I created this file with mencoder. Now I want ffmpeg to convert it inti flv but all I I get is this error: https://gist.github.com/3003836 [14:41] <AlRazi> burek, it worked but the quality of the m4a is a bit weak .. it gave this message converting from wav ( Guessed Channel Layout for Input Stream #0.0 : stereo ) [14:41] <latenite> Hiw can I fix that? [14:41] <burek> AlRazi, if the quality of 64kbps aac+ is weak, that means your input is weakly encoded [14:42] <AlRazi> can we change that ( sorry if it sounds too stupid ) [14:42] <burek> latenite, what is *.timelapse? [14:42] <burek> AlRazi, what I wanted to say is that your original mp3 file is of a bad quality already [14:43] <burek> i.e. find another original with better encoding quality [14:43] <latenite> burek, its a secqemce of images I created like so : mencoder 'mf://@tifiles.txt' -mf fps=${fps} -ovc copy vcodec=mpeg -o origtif.timelapse [14:43] <AlRazi> oh, got that .. thanks burek [14:43] <burek> latenite, so, it's MJPEG sequence? [14:44] <burek> try ffprobe origtif.timelapse [14:44] <burek> or mediainfo origtif.timelapse [14:44] <latenite> burek, I dont know about MJPEG? I told it to be vcodec=mpeg . I that MJPEG? [14:46] <burek> latenite, well, simply put, the ffmpeg error tells you that ffmpeg doesn't recognize the input [14:47] <burek> btw, your ffmpeg looks like a distro binary [14:48] <burek> is it normal that --cpu=core2 has been configured for a distribution [14:48] <latenite> burek, yes its gentoos ffmpeg. Is that a bad thing? [14:48] <burek> no, I'm just wondering [14:48] <burek> why would distro maintainer choose what kind of cpu the end user will have [14:48] <latenite> burek, but what is wrong with my file? Did I create it in a bad way? [14:49] <burek> latenite, I don't know, the point is that ffmpeg doesn't support your input, whatever it is [14:49] <burek> frankly I also don't know what's a timelapse file [14:49] <latenite> burek, I think with gentoo it lokks up a config...and sees 2 cpus here. [14:49] <latenite> ..then complies it with this option [14:49] <burek> ok [14:50] <latenite> burek, its a secquence of images taken every minute for hours....then playback in 30 sec [14:50] <burek> why don't you save that sequence in series of files [14:50] <AlRazi> do you need to have --enable-faac with --enable-libaacplus ? [14:50] <burek> AlRazi, no [14:50] <latenite> burek, I did. They are all fiff files [14:50] <latenite> burek, but I want it to be a video [14:51] <burek> latenite, then why did you provide only 1 file to ffmpeg input: origtif.timelapse [14:51] <burek> ? [14:51] <latenite> burek, here is some info on this file https://gist.github.com/3003900 [14:51] <burek> if you have a sequence like img001.tiff, img002.tiff, ... [14:51] <burek> then you can do it with ffmpeg [14:52] <latenite> burek, because I merged all the source images like so: mencoder 'mf://@tifiles.txt' -mf fps=${fps} -ovc copy vcodec=mpeg -o origtif.timelapse [14:52] <burek> do you have the original sequence of files on the disk in some directory? [14:52] <latenite> yes [14:52] <latenite> I say I can skip the mencoder part? [14:52] <burek> what is the file name format, how does it look [14:52] <burek> yes you can, of course [14:53] <latenite> file look slike sohttps://gist.github.com/3003908 [14:53] <latenite> https://gist.github.com/3003908 [14:53] <burek> can you type ls in that dir [14:53] <latenite> yes [14:54] <latenite> its about 700 files [14:54] <burek> I'm just curious to see how they are named [14:54] <burek> is it 1.tiff, 2.tiff, ... [14:54] <burek> or 001.tiff, 002.tiff, ... [14:56] <latenite> burek, liek so: ./output1.tif [14:56] <latenite> ./output2.6.tif [14:56] <latenite> ./output4.2.tif [14:56] <latenite> ./output5.8.tif [14:56] <latenite> ./output7.4.tif [14:56] <latenite> ./output9.tif [14:56] <latenite> ./output10.6.tif [14:56] <latenite> ./output12.2.tif [14:56] <latenite> ./output13.8.tif [14:56] <latenite> ./output15.4.tif [14:56] <burek> hm [14:56] <latenite> ./output17.tif [14:56] <burek> where is the pattern in that.. [14:57] <latenite> the nest larger the numer is the next file [14:57] <latenite> 2. is bigger than 4.2 [14:57] <burek> can you use some tool to rename it to a normal sequence [14:57] <latenite> sure [14:57] <burek> like 1,2,3,4... [14:57] <burek> well do it [14:57] <burek> and then [14:57] <burek> you can use [14:57] <burek> ffmpeg -i output%d.tif ... [14:58] <burek> for the encoding part, how do you want to encode your video exactly [14:58] <latenite> I *need* to be able to specify the framerate thats most important [14:59] <burek> ok, you will [14:59] <latenite> like I did with mencode [14:59] <burek> I'm just asking what the end result should be [14:59] <latenite> anything else is not so important [14:59] <burek> any specific target or something? [14:59] <latenite> al flv is fine [14:59] <burek> ok [14:59] <burek> then ffmpeg -r 25 -i output%d.tif -vcodec libx264 out.flv [15:01] <latenite> by that I get an error : https://gist.github.com/3003936 [15:02] <burek> oh [15:02] <burek> your input images are of invalid size for a video [15:02] <burek> should be 1920x1280 [15:02] <burek> and not 1281 :) [15:02] <latenite> so that was the reason in the fist place? [15:02] <burek> ffmpeg -r 25 -i output%d.tif -s 1920x1280 -vcodec libx264 out.flv [15:03] <burek> that should fix it, but since you are doing a resize, expect some small quality loss [15:03] <burek> or we can crop it using filter [15:03] <burek> that way you wont loose any quality [15:03] <latenite> yes that would be cool :D [15:04] Action: pron bites burek [15:04] <pron> nom nom nom [15:04] <burek> pron, :beer: :) [15:04] <pron> damn i want cheburek now ;( [15:05] <burek> latenite, try: ffmpeg -r 25 -i output%d.tif -vf crop=1920:1280 -vcodec libx264 out.flv [15:05] <latenite> burek, it does not throw an error but there is no aout.flv file created https://gist.github.com/3003960 [15:07] <burek> latenite, try to take a better look [15:07] <burek> the aout.flv should be there :) [15:07] <burek> also, latenite, did you rename those files [15:07] <burek> to form a proper sequence? [15:08] <latenite> burek, sorry I found the file. :/ [15:08] <latenite> bit its small 117K 27. Jun 15:05 aout.flv [15:08] <Spice_Boy> hi. I can successfully use mplayer to view a webcam, but ffmpeg (feeding to ffserver) dies with a bus error. The same OS, version, everything works fine when using the same camera on the laptop. any ideas? [15:09] <burek> latenite, did you rename all files? [15:09] <latenite> burek, yes the files are numbered from 1 to 450 [15:09] <Spice_Boy> (ie, works on laptop, not desktop) [15:09] <burek> Spice_Boy, can you see the cam with ffplay [15:10] <burek> latenite, can you do ls in that dir [15:10] <burek> and paste it on pastebin [15:10] <Spice_Boy> burek: I wanted to try that, but didn't know the command [15:10] <burek> Spice_Boy, how do you access your webcam in mplayer? [15:11] <Spice_Boy> mplayer tv:// -tv width=1280:height=960:device=/dev/video5 [15:11] <Spice_Boy> or smaller resolutions too [15:11] <burek> do you have /dev/video0 [15:11] <Spice_Boy> generally 640x480 to test [15:11] <Spice_Boy> I have 4 video cards that it detects first, the cam is on 5 [15:11] <burek> i see [15:11] <Spice_Boy> sorry, 5 video cards [15:11] <burek> oh yes, its in that line above [15:11] <burek> didnt see it :) [15:12] <burek> ffplay -f v4l2 /dev/video5 [15:12] <Spice_Boy> Bus error [15:12] <burek> ? [15:12] <burek> Spice_Boy, can you please use pastebin.com, to show your command line and its output? [15:12] <Spice_Boy> that's all it says [15:12] <Spice_Boy> ok [15:12] <Spice_Boy> moment [15:14] <Spice_Boy> burek: http://pastebin.com/LW4XwD0S [15:14] <Spice_Boy> that's both commands [15:15] <burek> your ffmpeg is seriously crippled [15:15] <burek> can you recompile it from git? [15:15] <Spice_Boy> I just compiled it as simple as possible to get it going. Also though, it's strange, because it works on a different host with the same liveCD distro [15:15] <burek> well [15:15] <burek> it's not the same [15:15] <Spice_Boy> I'm happy to recompile if that will help [15:15] <burek> you didn't install the same libs [15:15] <burek> as there [15:15] <burek> so when making ffmpeg [15:15] <ubitux> Spice_Boy: can you give a backtrace? [15:15] <burek> it didn't recognize needed libs [15:16] <burek> and didn't build the support for them [15:16] <Spice_Boy> how not the same? [15:16] <Spice_Boy> was same usb stick [15:16] <ubitux> Spice_Boy: gdb --args ./ffmpeg_g -f v4l2 /dev/video5 [15:16] <ubitux> run, and the bt [15:16] <Spice_Boy> ok, learning here :) [15:16] <Spice_Boy> ubitux: I'll do that now.. [15:16] <ubitux> i mean with ffplay actually [15:17] <Spice_Boy> ubitux: would it be bad if I didn't have the gdb command? :( [15:17] <ubitux> mmh try this before: [15:17] <ubitux> ./ffmpeg -f v4l2 -i /dev/video5 -f null - [15:17] <Spice_Boy> oh [15:18] <Spice_Boy> Failed to set value 'null' for option 'f' [15:18] <ubitux> with *ffmpeg*? [15:18] <Spice_Boy> oh sorry [15:18] <Spice_Boy> I did ffplay... [15:18] <ubitux> yeah, i want you to try ffmpeg now [15:19] <ubitux> to see if it's not a sdl issue or something [15:19] <Spice_Boy> same, Buss error [15:19] <ubitux> ok [15:19] <Spice_Boy> can you see my confusion as to why it works on one and not the other? [15:19] <ubitux> then please give a valgrind and/or gdb backtrace on the binaries with symbols (_g ones) [15:19] <Spice_Boy> you just lost me there [15:20] <ubitux> valgrind ./ffmpeg -f v4l2 -i /dev/video5 -f null - [15:20] <ubitux> gdb --args ./ffmpeg -f v4l2 -i /dev/video5 -f null - [15:20] <ubitux> etc. [15:20] <ubitux> check http://ffmpeg.org/bugreports.html [15:20] <Spice_Boy> ok, seems valgrind is a debugger... [15:20] <ubitux> not really [15:21] <burek> apt-get install gdb [15:21] <burek> and when you finish, later, type apt-get --purge autoremove gdb [15:21] <burek> that will remove it completely [15:22] <Spice_Boy> 2 things, this doesn't have apt-get but I'll compile it from source. Also, I just have to reset and it will be gone (livecd, unless I specifically want something saved) [15:22] <burek> what os are you using [15:23] <Spice_Boy> porteus (slackware based) [15:23] <Spice_Boy> I'm compileing valgrind now, and downloading gdb [15:27] <Spice_Boy> ok, I now have valgrind [15:29] <ubitux> please use ffmpeg_g instead of ffmpeg btw [15:29] <ubitux> since it will give much more information [15:29] <Spice_Boy> hang on, all hell has broken loose here... [15:30] <Spice_Boy> is that a configure option? [15:30] <Spice_Boy> to make it? [15:30] <ubitux> should be made by default [15:30] <ubitux> don't you have it in your source directory? [15:31] <Spice_Boy> I grabbed the snapshot, not git... is that maybe why? [15:32] <Spice_Boy> wait, I have it [15:32] <Spice_Boy> I just tried valgrind ./ffmpeg -f v4l2 -i /dev/video5 -f null - [15:33] <ubitux> can you pastebin the output? [15:33] <Spice_Boy> just about to :) [15:33] <ubitux> valgrind ./ffmpeg_g ... then, if you have it [15:34] <Spice_Boy> http://pastebin.com/wqN7S3w2 [15:35] <ubitux> looks like your system is broken [15:35] <ubitux> according to what valgrind is saying [15:37] <Spice_Boy> I assume that means something to you [15:39] <ubitux> not much, except that you're on your own [15:41] <Spice_Boy> haha [15:42] <Spice_Boy> ok then [16:07] <ePirat> is there an altrnative for av_set_parameters ? [16:07] <ePirat> or isnt it needed anymore? [16:15] <latenite> burek, I managed to get it working. Thank you for helping me out. Now I have this timelapse :D [16:15] <latenite> Say, is there a way to put two videos of smae length aside of each other into one new video. So I can playback and compare? [16:22] <jesk> is the documenation about ffserver's ability correct that it doesn't support currently streaming/reading from files? [16:25] <burek> latenite, try overlay filter [16:26] <burek> jesk, afaik it does [16:26] <burek> just use File <file> inside <stream> element [16:26] <burek> omg.. [16:27] <burek> #<Stream file.rm> [16:27] <burek> #File "/usr/local/httpd/htdocs/tlive.rm" [16:27] <burek> #NoAudio [16:27] <burek> #</Stream> [16:29] <latenite> burek, I read http://ffmpeg.org/ffmpeg.html#overlay-1 but I can not make sense of it. I wish they had a small example. [16:33] <burek> ffmpeg -i input -vf "overlay=..." output.avi [16:36] <alyawn> Fabric Manager... nice, burek [16:39] <alyawn> ok... do I have to call avformat_find_stream_info() if I already know the stream formats? if not, then what all needs to be populated in my FormatContext? [16:39] <latenite> burek, I dont know how to define the input file. My way -> will overwrite the source files https://gist.github.com/3004467 [16:39] <alyawn> latenite, remove with.flv [16:40] <latenite> alyawn, but with.flv and without.flv are my two source files that I want to show in one video [16:40] <alyawn> then add "movie=with.flv," to the beginning of the filter string [16:41] <latenite> alyawn, that gives me another error: https://gist.github.com/3004489 [16:42] <alyawn> change the space to a comma... no spaces allowed in the filter string [16:43] <latenite> alyawn, still there is something missing https://gist.github.com/3004499 [16:45] <alyawn> label the input to overlay: s/overlay/[in]overlay/ [16:46] <alyawn> all of this info can be found here, btw: http://ffmpeg.org/ffmpeg.html [16:46] <latenite> alyawn, sorry but it still fails : https://gist.github.com/3004536 [16:47] <latenite> I cant see how one can take the documentation and get this to work. The syntax is very very complicated [16:48] <alyawn> you need to read and understand filtergraphs to grasp the consept [16:48] <alyawn> you're almost there [16:48] <alyawn> read the output of the error and see if you can change values around to get it to do what you want [16:49] <latenite> Well I stll need it to be side by side not only 1ßpx abpart [16:50] <latenite> and I dont get the error. Is ot telling me that I need to define the endresults size? like 3840x1280? [16:52] <alyawn> do a little investigating... the answer is in the error output [16:53] <latenite> I dont get the error. It sais my overlay area (result video) is not in the main area. But what is main area? [16:54] <alyawn> right... the line above that tells you what values overlay is given... check there [16:56] <latenite> alyawn, I tried this https://gist.github.com/3004620 but I can make sense of "overlay area amd main area". Which one is what? [16:59] <alyawn> look at the overlay filter output above the error to see how the values you changed effected the parameters [16:59] <alyawn> you have 2 video inputs one is called main the other is called overlay [17:00] <latenite> alyawn, I found this syntax on google. It works ffmpeg -i with.flv -filter:v "[in]setpts=PTS-STARTPTS, pad=iw*2:ih:iw:0,[left]overlay=0:0[out]; movie=without.flv,setpts=PTS-STARTPTS[left]" output.flv [17:01] <latenite> alyawn, I will get into this later. I realy am interested...but I need this done now...So I have to skip the learning part for today [17:02] <latenite> alyawn, thank you very much for showing me the way :D [17:02] <alyawn> latenite, no problem [17:11] <latenite> alyawn, :D I guess to realy get into ffmopeg I need some time. This cant be done in a day or to...I figured. [17:13] <latenite> alyawn, one last question. If I dont wanf flv but a videoforamt that has less/none loss. Wich one would I use? [17:15] <alyawn> depends on what the player you're using supports [17:42] <rm-rf> burek: thanks for your help yesterday [17:43] <rm-rf> i'm still running into an issue where ffmpeg just dies randomly [17:44] <rm-rf> i tried to set logging, but for some reason it isn't logging anything [17:44] <rm-rf> any suggestions? [18:04] <delicado> hi, do i have to put an avpicture_free() after each call to avcodec_decode_video2()?. [18:06] <delicado> hi i need help. [18:15] <teratorn> delicado: are you using valgrind? [18:15] <delicado> i get disconnected, did someone answered my question?. [18:16] <delicado> teratorn. no, why? [18:16] <teratorn> you need to be [18:16] <burek> rm-rf, can you please use pastebin.com, to show your command line and its output? [18:16] <burek> latenite, can you please use pastebin.com, to show your command line and its output? [18:16] <teratorn> otherwise worring about memory will be a waste of time :) [18:17] <delicado> teratorn, im in windows. i tried looking for valgrind before, but i did not find a windows port. sorry i have a bad english. [18:17] <delicado> teratorn: yeah i got memory leak. about 4kb per second. [18:17] <teratorn> delicado: only commercial products [18:17] <teratorn> if you allocate frames [18:17] <teratorn> you free them [18:17] <teratorn> otherwise you don't [18:18] <teratorn> the decoder will need references to those objects until its done with them [18:20] <delicado> okay i have mine an avpicture_free() after each call to avcodec_decode_video2. ill now remove them. and see if it helps. maybe i got heap corruption or something like that [18:26] <teratorn> that shouldn't help, but who knows [18:30] <delicado> yeah i still get the leak. ill see ffplay.c then. [18:33] <latenite> burek, which command do you meen? [18:39] <teratorn> burek: hes not talking to you :) [19:04] <iTux> Hi [19:07] <iTux> I need some help (juste a little) [19:10] <iTux> I would like to configure (before build) ffmpeg whith "--enable-labass" argument but the script say that is an unknow option. I have the lastest version from the svn and I use Mac OS X 10.6 (Snow Leopard) on my computer (please don't troll). [19:10] <iTux> *"libass" sorry [19:11] <sacarasc> iTux: ffmpeg hasn't used svn since January last year. [19:12] <iTux> Oh, sorry ... where i can download a recent version ? [19:12] <iTux> On the git ? [19:12] <sacarasc> http://ffmpeg.org/download.html has the git repos. [19:12] <iTux> Ok, thans :) [19:12] <iTux> +k [20:04] <delicado> hi what cause this swscaler warning."Warning: data is not aligned! This can lead to a speedloss". i use sws_scale and sws_getCachedContext. i specified 640 x 480 as the destination. maybe it is the cause why i have an slow playback. what should i do? [20:10] <Freakshow> anyone have any experience with segmenter and ffmpeg? [20:10] <Freakshow> looking at this article [20:10] <Freakshow> http://www.ioncannon.net/projects/http-live-video-stream-segmenter-and-dist… [20:11] <Freakshow> seems promising but is horribly ancient [20:13] <Freakshow> sorry, wrong article [20:28] <Mavrik> delicado, your input data isnt aligned in memory [20:28] <Mavrik> since swscale uses MMX/SSE instructions, that can lead to slow execution [20:39] <delicado> Mavrik: okay, so its the input file that is not aligned? i guess i cant do anything about it. [20:39] <Mavrik> the input buffer. [20:39] <Mavrik> the image data. [20:40] <delicado> ah. what should be done? [20:53] <teratorn> delicado: you can use a union to align some buffer, or there are preprocessor/compiler directives [21:02] <delicado> but 640 x 480 is aligned right? SSE is aligned in 16-bytes. thats what i know. is there a function i could use? [21:05] <killown> Encoder (codec none) not found for output stream #0:0 [21:05] <killown> what this means? [21:06] <Mavrik> delicado, check your compiler documentation and look for intrisics in ffmpeg source to see how they solve it [21:07] <Mavrik> killown, you haven't set the encoder to use and ffmpeg can't guess it [21:07] <killown> ERROR: libmp3lame >= 3.98.3 not found < the problem :/ [21:14] <Jan-> how d'you make ffmpeg use aac audio? [21:14] <Jan-> I tried "libfaac" and "faac" and it just says "Unknown encoder 'whatever'" [21:21] <llogan> Jan-: there are 4 available AAC encoders, but 3 are external libraries and your ffmpeg may not have been compiled to support them [21:21] <llogan> use a pastebin site to show your ffmpeg command and the complete console output [21:24] <zambo> hey, is it possible for me to use 3 separate, short, audio files as the audio track for one video file output? [21:25] <zambo> or do I have to merge the audio files first separately [21:28] <llogan> zambo: if it's a "cat" friendly format you can try the concat protocol: ffmpeg -i concat:"input1.wav|input2.wav|input3.wav" -i video.mp4 ... output [21:28] <zambo> it is not, but thank you anyways [21:33] <llogan> zambo: don't forget the -shortest option. might be useful for this sort of thing once you figure out the concatenation [21:35] <Jan-> llogan: don't worry, I'll just use mp3 [21:35] <Jan-> it's just a pain to tell people they have to use VLC to play my videos, as a lot of things don't support h.264 with mp3 [21:37] <llogan> Jan-: you could always use the native aac encoder or pipe the output to faac. [21:37] <llogan> ffmpeg -i input -c:a aac -strict experimental -b:a 192k [21:37] <Jan-> Sorry, I'm not really an expert [21:38] <Jan-> it's a bad idea to have three AAC encoders, surely [21:38] <Jan-> three times the work and no benefit [21:38] <llogan> i just gave you the command. you don't have to be an expert to copy and paste [21:38] <Jan-> s'OK, I'm halfway through doing it in MP3 now :) [21:38] <llogan> not all encoders are equal and choice is a good thing [21:38] <Jan-> well er [21:38] <Jan-> use the best one? [21:39] <llogan> It Depends" [21:48] <JEEB> Jan-, if you want LC AAC (lib)faac is your choice, if you want HE AAC (low bitrate-specific), you use (lib)aacplus (I think that was its name), if you want to be adventurous you can use the internal aac encoder that at least doesn't add audible random noise any more :) [21:49] <Jan-> um er [21:50] <Jan-> I tried "libfaac" [21:50] <Jan-> it didn't work [21:50] <JEEB> well, yes -- then you just didn't compile your ffmpeg with that library -- or your distribution didn't [21:50] <llogan> that's why i asked you to: use a pastebin site to show your ffmpeg command and the complete console output [21:50] <JEEB> also windows builds most probably don't have it because it's non-distributable as it contains non-GPL-compatible source code in it :) [21:50] <Jan-> oh well. [21:50] <Jan-> open source people always like to shit on windows. [21:50] <llogan> you ask for help and then don't want it [21:51] <JEEB> Jan-, not really [21:51] <JEEB> I don't see this having anything to do with "shitting on windows" :) [21:51] <JEEB> if you are grabbing recent builds of ffmpeg they most probably don't have libfaac for the same reason that latter ffmpeg builds for linux don't have libfaac [21:52] <llogan> you can enable it in some distros with a single package [21:52] <Jan-> aaaaand that's why open source is stupid. Anyway, I found an acceptable workaround. [21:52] <JEEB> because it was found out that faac was heavily based on a reference implementation that is not exactly "free software" [21:52] <JEEB> Jan-, eh [21:52] <llogan> i give up [21:52] <JEEB> blame the faac folk if you want to blame someone Jan- [21:52] <Jan-> I only need to use it in like one meeting anyway [21:52] <Jan-> if people ask for the files I'll be a bit screwed [21:53] <Jan-> but oh well [21:53] <llogan> is mp3 officially supported by the MP4 spec? [21:53] <JEEB> yes [21:53] <JEEB> afaik [21:53] <Jan-> windows media player doesn't like it I don't think [21:53] <Jan-> or quicktime, either or, can't remember [21:53] <Jan-> anyway it causes grief [21:53] <JEEB> then just use the internal encoder at a bitrate 192kbps or more :) [21:53] <JEEB> should be just fine [21:53] <llogan> i thought it wasn't "official". i recall DS mentioning that, but i never got the specs. [21:53] <Jan-> VLC will handle it [21:54] <Jan-> but then VLC would play the great wall of china if you found a way to stuff it into a hard disk drive. [21:54] <Jan-> :) [21:54] <JEEB> llogan, I'm pretty sure MPEG-1 Layer 3 is in the ISO Media Container's specs :) [21:54] <llogan> ok. thanks. [21:56] <Jan-> bear in mind it isn't a good idea to do it though. [21:56] <Jan-> it is disliked by Things, Various. [21:56] <JEEB> well, that always depends :) I was just asked whether or not it's in the specification [21:58] <Jan-> that said everyone should just use VLC. [21:58] <JEEB> Jan-, also you really just could've used the internal AAC encoder like llogan wrote up the way to use it [21:58] <Jan-> But whatever. [21:59] <Jan-> I'm not sure if my version of ffmpeg is up to date enough to do that anyway [21:59] <Jan-> FFmpeg version SVN-r23418, Copyright (c) 2000-2010 the FFmpeg developers [21:59] <Jan-> built on Jun 2 2010 04:12:01 with gcc 4.4.2 [21:59] <llogan> that's why i asked you to: use a pastebin site to show your ffmpeg command and the complete console output [21:59] <JEEB> oh [21:59] <JEEB> that's somewhat old :) [21:59] <JEEB> might have bugs in the internal one [21:59] <Jan-> bear in mind [21:59] <Jan-> "somewhat old" in ffmpeg means "yesterday" [21:59] <JEEB> depends [22:00] <JEEB> at times it's OK'ish to be a couple of months backwards [22:00] <Jan-> I have no idea what version that actially is anyway. [22:00] <JEEB> well, it says June 2010 [22:00] <JEEB> and most probably taken from the trunk then [22:00] <Jan-> I did figure out how to build it on windows [22:00] <Jan-> but the problem is if you do actually build it from source, it has almost no real features. [22:01] <Jan-> Unless you somehow manage to convince it to include lots of external stuff [22:01] <Jan-> which quickly becomes a nightmare [22:01] <llogan> recent builds are available: http://ffmpeg.zeranoe.com/builds/ [22:01] <JEEB> nah, it has most -- I mean 99% of all decoders and parsers are already in the default LGPL configuration [22:01] <JEEB> then you most probably need something like libx264 [22:01] <Jan-> Mine wouldn't even do -vcodec libx264 [22:01] <JEEB> and that's it if you use the internal AAC encoder [22:01] <Jan-> which is all I ever use [22:02] <JEEB> yes, and that's one library for encoding? [22:02] <Jan-> I uhoh [22:02] <JEEB> which has the dependency of pretty much just a compiler and yasm [22:02] <Jan-> as far as I'm concerned, "libx264" is what you put in the command line if you want an mp4 file [22:02] <JEEB> and yasm is needed for ffmpeg, too [22:04] <JEEB> I don't really get your herping and derping to be honest :) Also, the configure outputs all the enabled functions after it finishes its checks. And ./configure --help is available. Also, you were saing "include a lot of external stuff" yet the only thing in most cases is needed is just x264 to have the H.264 encoding library :3 [22:04] <JEEB> But yeah, if you don't know of any of this surely it may of course be a bit derpy [22:05] <Jan-> well that and the fact that even on linux, ./configure is a command that a) creates a lot of impenetrable error messages, and b) exits. [22:05] <Jan-> Compiling code is for coders. [22:05] <Jan-> Not people. [22:05] <JEEB> well, you brought it up yourself and there are windows binaries available, remember :P [22:06] <llogan> I Am Not A Coder [22:06] <JEEB> also ffmpeg's/x264's configures are mostly very easy-to-read unless you happen to have a very broken compilation set-up [22:06] <Jan-> yes, but whenever you ask a question of an ffmpeg person they always refuse to help unless you're using "Latest SVN", despite the fact that using "Latest SVN" is impossible usually. [22:06] <Jan-> and mingw is a very broken compilation setup according to most ffmpeg people [22:06] <JEEB> lol [22:06] <JEEB> you have something very negative in the back of your head right now [22:06] Action: llogan didn't see JEEB refusing to help [22:07] <llogan> what's with all of the hatin' lately? [22:07] <JEEB> do note that I've been compiling ffmpeg/x264 for mingw for a relatively long time [22:07] Action: Jan- indicates a big elephant standing in the corner of the room that is called "MSVC" which would make all this easy [22:07] Action: Jan- dons flameproof suit [22:07] <JEEB> Jan-, then you'll love what BBB is baking [22:08] <JEEB> he is basically making a thing that converts the code to C90-compatible stuff [22:08] <JEEB> and thus enables your delightful MSVC compilation [22:08] <JEEB> which can indeed be useful for debug purposes [22:08] <Jan-> that and the intel C compiler produces faster-executing code. [22:08] Action: Jan- dons second layer of flameproof underwear [22:09] <JEEB> But I don't really know if it's really easier compilation-wise... it will make linking stuff easier esp. with static linking, and it will make debugging easier -- but it really isn't that hard to set up a mingw setup nowadays [22:10] <JEEB> Anyways, just do note that people are actually working on official'ish MSVC support somewhere :P [22:10] <JEEB> although I'm not really sure if you're just trolling at this stage or having a normal discussion :) [22:12] <Jan-> bit of both [22:12] <Jan-> Discussing MSVC with opensource nerds is like having a fireworks display at an oil refinery. :) [22:13] <JEEB> sorry for disappointing [22:13] <Jan-> I just get frustrated [22:13] <Jan-> I'd love to get involved with open source [22:13] <Jan-> but the prerequisites are such a bitch [22:14] <JEEB> I began around x264 [22:14] <JEEB> cool people in that community [22:14] Action: Jan- mutters [22:14] <Jan-> Jason Garret Glaser is a bit of a dick [22:14] <Jan-> and now I'm not even kidding [22:19] <JEEB> He does have a bit of derpy points, but in general I see him being quite helpful and nice to people looking at the past X years over at #x264,#x264dev [22:20] <meekohi_> Hey does -vb have any effect on a first pass of encoding? [22:20] <JEEB> If you started off like you just started here I'm not surprised if someone didn't have the nerves to actually get to your problems when you pretty much look like you're ignoring pretty much all the help you're possibly given + you just randomly call things names 8) [22:21] <meekohi_> Or in other words: Can I do a first pass of encoding once, and then use the log for many different quality encodings? [22:22] <JEEB> yes, the bitrate set mostly has an effect even on the first pass of encoding [22:23] <JEEB> and the second thing is possible depending on the video encoder in usage + possibly on settings [22:23] <JEEB> I'm not exactly sure how well the mbtree logs fare [22:23] <JEEB> (with libx264) [22:23] <meekohi_> JEEB: Gotcha. I'm using libvpx currently. So the safe bet is to rerun the 1st pass for each quality setting? [22:24] <JEEB> I've no idea about libvpx unfortunately [22:24] <JEEB> you might want to ask the vpx encoder folk [22:24] <meekohi_> Sounds good. [22:39] <jesk> anyone an idea what the reason could be ffserver opening http port not responding to requests? [22:39] <jesk> stat page, index redirect, sdp, all that stuff not accessable [22:40] <jesk> first time i'am playing with it [22:42] <jesk> is the sdp file just in memory or is there directory which serves per default [22:43] <jesk> streaming to multicast works [22:43] <jesk> although I couldnt decode the stream yet :D [22:52] <drazmo> I am trying to compile latest git ffmpeg on ubuntu 8.04. I was able to compile x264 but now I am getting this error during compilation...LD ffmpeg_g libavcodec/libavcodec.a(libx264.o): In function `X264_init_static': /home/user/ffmpeginstall/ffmpeg/libavcodec/libx264.c:538: undefined reference to `x264_bit_depth' libavcodec/libavcodec.a(libx264.o): In function `X264_frame': /home/user/ffmpeginstall/ffmpeg/libavcodec/libx264.c:156: u [23:44] <llogan> drazmo: probably means you also have an old x264 somewhere [23:45] <llogan> drazmo: did you see this? https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuideHardy [00:00] --- Thu Jun 28 2012

1 0

ffmpeg-devel.log.20120626
by burek 27 Jun '12

27 Jun '12

[00:44] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r8a0cd58729 10ffmpeg/libavcodec/sonic.c: [00:44] <CIA-119> ffmpeg: sonic: fix FPE [00:44] <CIA-119> ffmpeg: Fixes Ticket1397 [00:44] <CIA-119> ffmpeg: Found-by: Piotr Bandurski <ami_stuff(a)o2.pl> [00:44] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:54] <llogan> saste: any ideas for a new closed state for art tickets? I'm thinking "completed art". [00:54] <saste> or just "completed"... [00:55] <saste> but give me a night and I think at it when walking the dog [00:56] <llogan> or "other" or "misc" [00:56] <llogan> just spamming ideas [00:57] <saste> i hate "other"... means all and nothing [00:58] <saste> "archived" may be another... [00:59] <llogan> i'm fine with "completed" [01:01] <saste> for example a logo which is not completed, but to which the artist don't want to work no longer [01:02] <saste> *doesn't want to work any longer [01:02] <saste> completed (art), archived (art) [01:05] <llogan> seems like we are overcomplicating it. i'd prefer just one closed state for art. [01:07] <durandal_1707> i prefer if fffmpeg.org dont have images at all [01:07] <CIA-119> ffmpeg: 03Paul B Mahol 07master * re3c2670539 10ffmpeg/libavcodec/mss1.c: [01:07] <CIA-119> ffmpeg: mss1: check number of free colours [01:07] <CIA-119> ffmpeg: Prevents out of array write. [01:07] <CIA-119> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [01:08] <burek> how about video :D [01:08] <durandal_1707> what codec? [01:09] <CIA-119> ffmpeg: 03yang 07master * r9b72041f80 10ffmpeg/libavutil/x86/intmath.h: [01:09] <CIA-119> ffmpeg: x86/intmath.h: Fix mull operand constraints [01:09] <CIA-119> ffmpeg: Fixes Ticket1466 [01:09] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:09] <Compn> ubitux : it was just a suggestion, if you think clone is better first, go for it :) [01:30] <CIA-119> ffmpeg: 03Mans Rullgard 07master * r8725da49a2 10ffmpeg/libavcodec/x86/fft_mmx.asm: x86: fft: win64: fix stack alignment for memcpy() call [01:30] <CIA-119> ffmpeg: 03Mans Rullgard 07master * r963cdf39b4 10ffmpeg/libavutil/x86/cpu.c: [01:30] <CIA-119> ffmpeg: x86: cpu: whitespace (mostly) cosmetics [01:30] <CIA-119> ffmpeg: This adds whitespace around operators, aligns line continuation [01:30] <CIA-119> ffmpeg: backslashes, and breaks long lines. Also fixes an ifdef halfway [01:30] <CIA-119> ffmpeg: through a statement. The one line of duplication this saved is [01:30] <CIA-119> ffmpeg: not worth the ugliness. [01:30] <CIA-119> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [01:30] <CIA-119> ffmpeg: 03Ronald S. Bultje 07master * r246154a9af 10ffmpeg/libavutil/log.c: [01:30] <CIA-119> ffmpeg: log: Include io.h on windows [01:30] <CIA-119> ffmpeg: This is required for isatty, which exists on MSVC and is found by [01:30] <CIA-119> ffmpeg: configure, but is provided by io.h instead of unistd.h. [01:30] <CIA-119> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:30] <CIA-119> ffmpeg: 03Justin Ruggles 07master * r14a34d90ad 10ffmpeg/libavresample/x86/audio_convert_init.c: lavr: x86: merge some branches [01:30] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r3b0ad040b3 10ffmpeg/: (log message trimmed) [01:30] <CIA-119> ffmpeg: * qatar/master: [01:30] <CIA-119> ffmpeg: log: Include io.h on windows [01:30] <CIA-119> ffmpeg: lavr: x86: merge some branches [01:30] <CIA-119> ffmpeg: x86: cpu: whitespace (mostly) cosmetics [01:30] <CIA-119> ffmpeg: x86: fft: win64: fix stack alignment for memcpy() call [04:03] <llogan> how many more stupid mistakes in this patch will i make today? [04:03] Action: llogan blames the heat. wool was a bad idea. [04:09] <Compn> wool is always a bad idea [04:09] <Compn> so itchy [05:51] <Zeranoe> I'm a little confused by this patch: http://git.videolan.org/?p=ffmpeg.git;a=commit;h=9ec5e956a2a4e359f21d48cc70… "ffmpeg: disable threading on mingw, it doesnt work due to dependance on internal code." is this only for win32thread? [09:31] <CIA-119> ffmpeg: 03Carl Eugen Hoyos 07master * rc77bcbbb32 10ffmpeg/libavcodec/mss1.c: [09:31] <CIA-119> ffmpeg: Signal MSS1 palette change. [09:31] <CIA-119> ffmpeg: Reviewed-by: Paul B Mahol [09:55] <CIA-119> ffmpeg: 03Carl Eugen Hoyos 07master * r6eff277284 10ffmpeg/libavcodec/sanm.c: [09:55] <CIA-119> ffmpeg: Make LucasArts Smush SANM palette opaque. [09:55] <CIA-119> ffmpeg: Reviewed-by: Paul B Mahol [11:45] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * rf054dbee6c 10ffmpeg/tools/graph2dot.c: tools/graph2dot: make dot graph representation a bit more compact [11:45] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r8dd0e87d7b 10ffmpeg/ (5 files in 3 dirs): [11:45] <CIA-119> ffmpeg: lavfi: remove old video sink API [11:45] <CIA-119> ffmpeg: It was deprecated since a long time and removed after the 2->3 major [11:45] <CIA-119> ffmpeg: bump. [11:45] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r720ec62012 10ffmpeg/libavfilter/ (avfilter.h buffersink.h defaults.c formats.c version.h): [11:45] <CIA-119> ffmpeg: lavfi: drop deprecated and disabled packing API [11:45] <CIA-119> ffmpeg: It was deprecated and removed after the recent 2->3 major bump. [11:45] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r79a7451d06 10ffmpeg/ffplay.c: [11:45] <CIA-119> ffmpeg: ffplay: add configure_filtergraph() helper [11:45] <CIA-119> ffmpeg: Will help factorization with the pending -af patch, and add some checks [11:45] <CIA-119> ffmpeg: missing in the original code. [11:53] <burek> is anyone here interested in improving the documentation by making changes to allow easy editing for both developers and regular users? [11:53] <burek> I've left the link yesterday to DocBook: http://en.wikipedia.org/wiki/DocBook [11:54] <burek> but I didn't see anyone bothered to reply to it.. [11:54] <ubitux> i thought we agreed with the wiki trac :) [11:54] <ubitux> burek: btw i sent a patchset to add the wiki and stuff [11:55] <ubitux> in a "community"-like section [11:55] <burek> ubitux, hi :) [11:55] <burek> well, what's the point of the trac wiki if the documentation is still left split? [11:55] <ubitux> the main documentation will stay in the repository [11:56] <burek> by that, do you mean "editable only by developers" or "tracked for version changes" ? [11:56] <saste> burek: technical documentation -> git, user level documentation (guides, walkthrough, overview) -> wiki [11:56] <ubitux> that is IMO the best way to get sure developers will update it when there is a new features in [11:56] <ubitux> exactly what saste said [11:57] <burek> isn't that putting all the work for tech docs to developers only? [11:57] <burek> i.e. it will stay messy like it is now, for that exact reason.. [11:57] <cbsrobot> ah please not xml [11:58] <ubitux> :D [11:58] <burek> cbsrobot, it's not xml [11:58] <cbsrobot> maybe http://johnmacfarlane.net/pandoc/ could be used [11:58] <burek> read more thoroughly [11:58] <ubitux> burek: it won't be better with a wiki [11:58] <ubitux> for technical doc, it can't [11:58] <burek> it's saved as xml, but it offers wide variety of presentations, even man pages format [11:58] Action: cbsrobot is meber of the noxml party [11:58] <burek> you can edit it even in html if you like [11:58] Action: ubitux is a member of the killxml parthy [11:58] <ubitux> -h [11:59] <ubitux> html is like xml [11:59] <cbsrobot> lol [11:59] <saste> burek: in order to contribute effectively to technical documentation, you need to be a developer anyway [11:59] <burek> saste, to correct syntax errors too? [11:59] <ubitux> burek: just raise them on irc, we will fix them [12:00] <saste> so you already knows how to use git, patches, etc, using a wiki won't help, but will make harder to maintain the thing (wiki<->docs synch, versions, etc.) [12:00] <burek> ubitux, it's not a problem to leave things as they are.. but isn't it obvious so far that it's a double work [12:00] <ubitux> burek: we have reviews to avoid most of the syntax errors [12:00] <burek> maybe I'm wrong about all this.. [12:00] <ubitux> i think you are [12:00] <cbsrobot> burek: I like the intention, but the implementation is the problem [12:00] <ubitux> i can explain why again [12:00] <ubitux> but you will need to read me this time [12:01] <cbsrobot> hehe [12:01] <cbsrobot> read more guess less [12:01] <burek> fine [12:01] <ubitux> burek: to me there is just no user contrib, because the trac has not much content, so there is no double work [12:01] <ubitux> and as saste said, user documentation is a different documentation from user [12:02] <ubitux> from dev* [12:03] <saste> an idea could be i heard is implemented maybe by libboost, an user can edit the web page, the change is converted to a patch and sent to the ML [12:03] <saste> this may work but i guess it's a lot of work to setup [12:03] <ubitux> that's a bit what you can do with github [12:03] <ubitux> since you can edit files directly from the web ui [12:03] <saste> and most patches created in this way will need to be reviewed more or less, so it's again moving the work on the shoulder of the developers [12:05] <ubitux> or you can just raise the typo on irc and we will fix them asap [12:21] <saste> ubitux: ideally we should be able to set metadata on filtered frames, and use select accordingly (or a selectmetadata variant) [12:21] <saste> rather than adding more and more features to select [12:21] <saste> indeed i don't like the lavc dependency [12:21] <saste> not that it is an issue, since now lavfi depends on lavc anyhow [12:41] <saste> michaelni: what's your plan with the opaque field in filters init? [12:41] <saste> I'd prefer to keep it, it is used in the sink and i can see some scenarios when it is actually useful [13:07] <CIA-119> ffmpeg: 03Nicolas George 07master * rf767658414 10ffmpeg/libavcodec/x86/fft_mmx.asm: [13:07] <CIA-119> ffmpeg: Revert "x86: fft: win64: fix stack alignment for memcpy() call" [13:07] <CIA-119> ffmpeg: This reverts commit 8725da49a2090de05b4b2d05e33727f45cb9d970. [13:07] <CIA-119> ffmpeg: Necerrary to revert 82992604706144910f4a2f875d48cfc66c1b70d7. [13:07] <CIA-119> ffmpeg: 03Nicolas George 07master * rfd91a3ec44 10ffmpeg/libavcodec/x86/ (Makefile fft_mmx.asm fft_sse.c): [13:07] <CIA-119> ffmpeg: Revert "x86: fft: convert sse inline asm to yasm" [13:07] <CIA-119> ffmpeg: This reverts commit 82992604706144910f4a2f875d48cfc66c1b70d7. [13:07] <CIA-119> ffmpeg: It breaks shared builds on x86_64. [13:15] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * re6674e46ec 10ffmpeg/ (7 files in 3 dirs): (log message trimmed) [13:15] <CIA-119> ffmpeg: lavu/imgutils: create misc functions for dealing with buffers [13:15] <CIA-119> ffmpeg: Move the lavc/imgconvert functions and rename them as follows: [13:15] <CIA-119> ffmpeg: avpicture_get_size -> av_image_get_buffer_size() [13:15] <CIA-119> ffmpeg: avpicture_fill -> av_image_fill_arrays() [13:15] <CIA-119> ffmpeg: avpicture_layout -> av_image_copy_to_buffer() [13:15] <CIA-119> ffmpeg: The new functions have an align parameter, which allows to define the [13:36] <CIA-119> ffmpeg: 03Nicolas George 07master * ra2bd8a9384 10ffmpeg/libavfilter/buffersink.h: buffersink: group libav API functions. [13:36] <CIA-119> ffmpeg: 03Nicolas George 07master * r784675ca91 10ffmpeg/libavfilter/sink_buffer.c: sink_buffer: make opaque argument optional. [14:01] <ubitux> saste: ah yeah maybe [14:01] <ubitux> saste: didn't you say you wanted to add ifdef? [14:01] <ubitux> (in the select/scene) [14:01] <saste> yes [14:02] <ubitux> i can do it, maybe tonigh, otherwise [14:02] <saste> but i don't want the filter to do things which is not required to [14:02] <ubitux> sure [14:02] <saste> and decrease performance in case a certain "feature" has not to be selected [14:03] <ubitux> it doesn't run scene detect if you don't use it [14:03] <saste> but the more general approach is to decouple selecting logic from feature detection [14:03] <ubitux> it could be nice to access the metadata when requesting a string indeed... [14:03] <saste> that is you detect the feature and put the data in frame metadata, and use a select filter matching a metadata pattern [14:04] <saste> with no need to further extend the select code more and more [14:21] <ubitux> feel free to design something :) [16:18] <CIA-119> ffmpeg: 03Mans Rullgard 07master * r37c3864ef7 10ffmpeg/libavcodec/x86/fft_mmx.asm: [16:18] <CIA-119> ffmpeg: x86: fft: elf64: fix PIC build [16:18] <CIA-119> ffmpeg: In a 64-bit PIC build, external functions must be called [16:18] <CIA-119> ffmpeg: through the PLT. [16:18] <CIA-119> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [16:18] <CIA-119> ffmpeg: 03Nicolas George 07master * r91765594dd 10ffmpeg/libavcodec/x86/ (Makefile fft_mmx.asm fft_sse.c): [16:18] <CIA-119> ffmpeg: Revert "Revert "x86: fft: convert sse inline asm to yasm"" [16:18] <CIA-119> ffmpeg: This reverts commit fd91a3ec44de38251b2c15e03e26d14e983c4e44. [16:18] <CIA-119> ffmpeg: The bug it introduced has been fixed. [16:19] <CIA-119> ffmpeg: 03Nicolas George 07master * rd4c45b8adf 10ffmpeg/libavcodec/x86/fft_mmx.asm: [16:19] <CIA-119> ffmpeg: Revert "Revert "x86: fft: win64: fix stack alignment for memcpy() call"" [16:19] <CIA-119> ffmpeg: This reverts commit f767658414fc85dea4006cb82969b6a925fdd380. [16:19] <CIA-119> ffmpeg: The bug it introduces has been fixed. [17:00] <jesk> what is a simple/cheap way to create free atom space in mp4 [17:00] <jesk> don't want to code that :-) [17:15] <ubitux> jesk: use a "free" or "wide" atom? [17:27] <jesk> ubitux: any tool for doing that? [17:28] <ubitux> no [17:28] <ubitux> you can do it with your shell [17:28] <ubitux> as long as you're appending it at the end [17:33] <jesk> ubitux: i wouldn't have any problem with an initial rewrite, have to do that anyway to get the moov to the beginning [17:36] <jesk> how would you insert it to the end, and how would you rewrite so that moov and/or free atom is in the the beginning? [17:36] <jesk> sorry if those questions seem to be stupid, but google isn't that helpful with that.. :-( [17:36] <ubitux> i don't know what you want to do [17:37] <ubitux> are you willing to add a faststart to the mov muxer? [17:37] <jesk> that's something i'am not quite sure yet, when writing with quicktime or mp4box it has always a faststart [17:38] <jesk> with ffmpeg only with qtfaststart [17:38] <jesk> i'am troubleshooting yet what the more accepted way with my devices is [17:39] <jesk> but anyway, without free atom there is no cheap way of modifying metadata when the data will increase a bit [17:40] <jesk> and that's unacceptable [17:40] <jesk> so I would like to try out what the better practice is [17:40] <ubitux> 17:36:50 <@ubitux> i don't know what you want to do [17:41] <jesk> would like to try with free atom in the beginning, either with procedure which focus *only* on placing free atom or with procedure with moving moov and freeing some atoms [17:42] <jesk> creating free at the end is something I would also like to test [17:42] <jesk> but yet I have no clue at all :D [17:42] <av500> what point would a free at the end have? [17:42] <jesk> *no clue how to create free atoms for placing metadata in both ways [17:42] <av500> you can add that any time [17:43] <jesk> av500, true [17:43] <jesk> very true [17:44] <jesk> so lets focus with free in beginning moov :-) [17:45] <av500> well, so change the muxer to add that [17:45] <jesk> what would you suggest, which muxer can do that natively? [17:45] <jesk> does ffmpeg has option for that? [17:47] <ubitux> ffmpeg doesn't support faststarting yet [17:48] <ubitux> but it shouldn't be that hard to do [17:48] <av500> change ffmpeg then [17:50] <jesk> ok, nothing easier than that ;-) [17:51] <av500> you are in #ffmpeg-devel after all [17:53] <jesk> take it as feedback from user and developer but ffmpeg noob [17:53] <jesk> :) [17:53] <jesk> usecase for many apple fanbox [17:53] <jesk> s,fanbox,fanboys, [17:53] <av500> but what is the use case? [17:53] <av500> what do you need free for? [17:54] <jesk> creating mp4, compatible with iDevice and ATV, and ability to modify/add iTunes/Movie metadata without the need to wait hours for rewrite [17:55] <ubitux> if the header is at the end, it doesn't matter, you can just add the udta/meta crap at the end (and resize parent atoms) [17:55] <ubitux> if it's on the top you need to adjust the offsets [17:56] <jesk> header at the end is more or less incompatible with Playstation and Apple stuff (imho) [17:56] <av500> then you need to faststart it after encoding [17:56] <ubitux> then look how qt-faststart does [17:56] <jesk> ffmpeg is different in this regard I believe [17:56] <jesk> qt-faststart can create free? [17:57] <jesk> not that I know of, but will have a look again [17:57] <ubitux> well adapt the code [17:57] <av500> why do you need to create free? [17:57] <ubitux> av500: to make space to add metadata later [17:57] <av500> ah, ok for metadata [17:57] <av500> well, then modify faststart to add a free atom [17:57] <av500> no big deal [17:57] <jesk> yet it is for me, but I will investigate [17:58] <ubitux> iirc qt-faststart is far from perfect [17:58] <ubitux> especially while looking for the offsets table [17:58] <ubitux> (just a memsearch) [17:58] <ubitux> i wonder if it doesn't trigger false-positive often [17:59] <ubitux> anyway, [17:59] Action: ubitux & [18:00] <jesk> ffmpeg/qtfaststart has a lot of problems creating mp4s working on Apples [18:01] <av500> and you reported them? [18:01] <jesk> not yet, still in the debugging/understanding phase [18:01] <jesk> working with mp4's since a week now [18:05] <av500> k [18:09] <cbsrobot> last time I saw a file some adobe enc added xml at the end of a mov and qtfaststart just failed with that ... so if you rewrite it keep that in mind [18:25] <Compn> jesk : we'd love patches :) [18:25] <Compn> when you are ready [18:26] <jesk> cbsrobot: not only Adobe, Apple too [18:27] <jesk> Compn: ok :-) [19:05] <CIA-119> ffmpeg: 03Gavin Kinsey 07master * r37b5959d96 10ffmpeg/doc/examples/ (filtering_audio.c filtering_video.c): [19:05] <CIA-119> ffmpeg: examples/filtering: fix packet memleak [19:05] <CIA-119> ffmpeg: Free packets unconditionally after demuxing, and not only when the [19:05] <CIA-119> ffmpeg: packets belong to a given stream. [19:05] <CIA-119> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [19:05] <CIA-119> ffmpeg: 03Gavin Kinsey 07master * r9ebed95db6 10ffmpeg/doc/examples/filtering_video.c: [19:05] <CIA-119> ffmpeg: examples/filtering_video: update to the new API [19:05] <CIA-119> ffmpeg: Update the video filtering example program based on the audio one. [19:05] <CIA-119> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [19:38] <ubitux> the re-align wasn't required [19:38] <ubitux> and now it's wrong :) [19:42] <TXH350> Provide me ffmpeg PGP public key ID for release, plz. I can't find it. [19:47] <ubitux> http://ffmpeg.org/download.html#releases ? [19:49] <TXH350> there are only signature files [19:58] <Compn> TXH350 : ask michaelni for it [19:58] Action: Compn afk [21:42] <ubitux> burek: you should update your forum favicon [21:44] <llogan> do you think it's too qataresque? [21:44] <ubitux> yes [21:44] <ubitux> and i think it might get us some trouble [21:46] <llogan> what happened to the forum.ffmpeg.org proposal? (or am i confusing myself?) [21:50] <burek> ubitux, is it ok now? [21:53] <llogan> looks fine to me [21:55] <burek> ok [22:20] <Compn> llogan : what was the proposal ? [22:20] <Compn> iirc someone came up with a list of ffmpeg forums [22:20] <Compn> so its kinda weird for us to chose one to link to [22:22] <ubitux> burek: looks like it's still in the cache for me :p [22:25] <burek> ubitux, open in your browser this page http://ffmpeg.gusari.org/favicon.png and press ctrl+F5 [22:25] <burek> and then try the home page again [22:26] <burek> Compn, the idea was to merge forums into one, but the process has taken too long (more than 3-4 months) and finally it was abandoned (at least by me) [22:26] <llogan> what's the other forum? [22:27] <burek> not to blame anyone, just I don't think it should take that much time, so I didn't want to participate any more.. [22:27] <burek> well, there is a list of forums at ffmpeg's contact page I think [22:27] <ubitux> burek: i did it, it worked, but it didn't update the one in cache when browsing the site :p [22:27] <ubitux> creepy firefox [22:27] <ubitux> but i believe you ;) [22:27] <ubitux> thx :) [22:28] <burek> but there were 3 of them iirc.. what happened with the 3rd one? [22:28] <ubitux> i added yours [22:28] <burek> ubitux, ok [22:28] <ubitux> it should stress a bit the process [22:32] <llogan> burek's is the one linked to in #ffmpeg /topic. i don't see why forum.ffmpeg.org shouldn't point to it if he's willing to admin it. [22:34] <burek> well I am, I just didn't want to diminish the value of other existing forums [22:34] <burek> so I suggested that we merge all of them into one official [22:45] <ubitux> reminds me the story of the wiki& [22:45] <ubitux> ;) [22:57] <burek> well, all I'm trying to do is to improve something a little bit and make it easy for development of the project to get more help from other people [22:57] <burek> either by reporting bugs (forums) or by improving docs (wiki), etc [22:58] <burek> it's not something egoistic, it's just a wish to make things better [22:59] <ubitux> i know :) [22:59] <ubitux> and thank you for your contributions [22:59] <burek> :beer: :) [22:59] <ubitux> :) [23:00] <burek> :juice: :D [23:00] <ubitux> yeah much better [23:08] <burek> is there somewhere a table, a list, something.. of supported video codecs inside existing formats.. [23:08] <burek> like for example, flv can contain following video codecs: h264, flv, bla bla [23:09] <burek> and following audio: aac, mp3, bla bla [23:09] <burek> I'd like to compile a table that can have it all in one place [23:11] <durandal_1707> i know only two tables riff one and isom one [23:11] <burek> is there any url for them, please? :) [23:11] <JEEB> someone tell burek not to ignore me please [23:11] <JEEB> http://pastebin.com/pFt55P3v [23:12] <burek> I'm not ignoring you :) [23:12] <burek> I just choose not to talk to you :) [23:12] <JEEB> great [23:12] <JEEB> flv/f4v specs are also available @ http://www.adobe.com/devnet/f4v.html [23:13] <burek> thanks for that :) [23:14] <JEEB> I have no idea why ffmpeg let that dude remux mpeg-4 part 2 into flv (technically it's possible naturally, the format is just not spec'd for it) [23:28] <ePirat> anyone has an idea why static linking of lav* libraries seems to fail while shared works? http://pastebin.com/zqtu17Xt maybe I am doing a noob mistake there& [23:30] <nevcairiel> it seems odd that you provide a path to avformat and avutil, but use -lavcodec [23:31] <ePirat> oh& huh [23:33] <ePirat> hm now i Undefined symbols for architecture http://pastebin.com/MBwK1DXT [23:33] <nevcairiel> sounds like its expecting some more libraries to be linked [23:34] <ePirat> why does it worked when using shared libraries then? [23:35] <nevcairiel> probably because its also loading the shared versions of all the other things? I dunno, i prefer shared anyway [23:35] <ePirat> hm but i need static& else i have to recompile on every machine& [23:40] <ePirat> seems maybe there is something wrong with my static libraries& I'll recompile everything and see if it works then& [00:00] --- Wed Jun 27 2012

1 0

ffmpeg.log.20120626
by burek 27 Jun '12

27 Jun '12

[00:49] <Freakshow> < saste: DelphiWorld: rtmp native support in ffmpeg is not very good at the present moment> in what way is rtmp support in ffmpeg 'not very good'? [00:50] <saste> Freakshow: in the sense that not all protocols are supported, and that people had some issue in the past, so they're usually better served by the librtmp implementation, which is more tested and generally more used [00:51] <saste> i don't know the current status of affairs though [00:51] <Freakshow> and holy shit... burek / ubitux I just read through this evenings discussion re: trac vs. mediawiki, I'm sorry... but I seriously lol'd [00:51] <saste> maybe it's improved in the meaningwhile... [00:51] <Freakshow> true that... but generally speaking, isn't librtmp more for handling encrypted output rather than standard output? [00:52] Action: llogan waits for IRC log [00:52] <Freakshow> rtmpt, etc.? [00:52] <Freakshow> yes llogan: grep for HerbertPumpkin [00:52] <Freakshow> :D [00:53] <llogan> nice nick. must be full o' gold. [00:54] <Freakshow> my favorite: [00:54] <Freakshow> [07:11am] ubitux: then just report typo issues by opening an issue in the trac [00:54] <Freakshow> [07:11am] ubitux: and complete the doc in the wiki [00:54] <Freakshow> [07:12am] HerbertPumpkin: what's a trac? [00:54] <Freakshow> [07:12am] ubitux: it's a facebook group [00:54] <Freakshow> sorry for that spam.... [00:59] <Freakshow> seriously though... I'm curious if anyone has had a chance to review: [00:59] <Freakshow> https://ffmpeg.org/trac/ffmpeg/ticket/1446 [01:00] <Freakshow> how would I go about trying to get some eyes on this issue? [01:00] <llogan> is that your ticket? [01:02] <burek> well, to be honest, if nobody is interested to make the docs better without raping the free time of developers, then we did waste our time today on that discussion about wiki and docs [01:05] <heimlich> hi guys.. if i have a quicktime movie with timecode... is there a way to extract specific frames from the quicktime movie starting with timecode xxxx and ending with timecode yyyy...when I try to use the -ss option, I always get invalid duration specification for ss [01:09] <Freakshow> llogan: yes, that's mine [01:10] <Freakshow> burek: to be fair... I wasn't saying it was a waste of time. I think that there were some valid points brought up in that conversation [01:12] <burek> heimlich, can you please use pastebin.com, to show your command line and its output? [01:12] <burek> Freakshow, well yes, but we seriously offtopiced this channel :D [01:12] <heimlich> sure.. one sec [01:13] Action: Freakshow nod [01:14] <Freakshow> no one else was talking anyway [01:14] <heimlich> burek: http://pastebin.com/94mSGWqQ [01:16] <heimlich> weird.. didnt seem to paste all [01:16] <burek> heimlich, try -ss 02:29:04.200 instead of -ss 02:29:04:20 [01:18] <heimlich> burek: same result: http://pastebin.com/2brseNf2 [01:18] <burek> heimlich, note the dot "." [01:18] <burek> instead of a column ":" [01:18] <heimlich> ungh.. sorry.. one sec [01:19] <heimlich> yeah.. that worked.. [01:22] <heimlich> now end timecode [01:28] <burek> * llogan waits for IRC log -> see live logs here (if you need): http://ffmpeg.gusari.org/irclogs/ [01:29] <heimlich> burek: http://pastebin.com/6LTKUGva [01:29] <heimlich> btw.. thanks [01:29] <burek> heimlich, can you please use pastebin.com, to show your command line and its output? [01:29] <burek> uncut, entire output [01:29] <Freakshow> llogan: do you have any thoughts on that ticket? I'm just trying to get some movement on it if possible [01:30] <heimlich> here yo go: http://pastebin.com/EwQH6YKi [01:31] <burek> heimlich, your input file's duration is: Duration: 00:00:01.20 [01:31] <burek> and you are seeking to -ss 02:29:04.200 (2 hours, 29 minutes, 4 seconds and 200 mseconds) [01:31] <burek> :) [01:32] <heimlich> well.. the starting timecode of the file is 02:29:04:02 my initial goal, was to extract frames from a long quicktime, using certain timecode segments [01:33] <burek> I see [01:35] <burek> well, in stream #0.1 there is "timecode" metadata [01:35] <burek> you might get it using ffprobe or ffmpeg (parse the output of ffmpeg -i inputfile) [01:35] <burek> and subtract from the creation time maybe [01:36] <burek> (which is also in metadata) [01:36] <burek> I didn't say it right.. what I meant is to subtract from the first timecode (of the first segment) you have [01:37] <burek> so that you get the actual time into the video [01:38] <burek> -ss can only handle time in seconds or in "hh:mm:ss[.xxx]" form [01:39] <Freakshow> brb [01:39] <llogan> Freakshow: sorry. i've never used rtsp output. [02:12] <alyawn> using libavformat, how can I specify a specific output codec rahter than taking the AVFormatContext default? [02:13] <alyawn> think mpegts format with an h264 stream, rather than the default mpeg2video [02:48] <Nickname123> Can anyone help me get ffmpeg compiling on my windows mingw setup? Ffmpeg configure gives me the following error: "ERROR: libx264 not found" See http://pastebin.com/qTwYas8t for directory layout and command line [02:49] <Nickname123> oh i think i made a copy and paste error [02:53] <Nickname123> http://pastebin.com/xaMLLDUa [02:57] <Nickname123> I tried to fix it for the / paths that msys uses... still no luck: http://pastebin.com/sEwjDwfA [03:01] <grepper> I think you need to compile and install x264 first, at least that is how it is done on linux afaik [03:02] <grepper> then if need be give the path the the installed libraries and headers [03:02] <Nickname123> I did but the msys / mingw search paths are throwing me off [03:02] <Nickname123> so I tried to explicitly list them [03:02] <Nickname123> oooh [03:02] <Nickname123> I think i got it [03:02] <grepper> I see. Sorry, I don't use windows so dunno [03:02] <Nickname123> started over on my configure line [03:03] <Nickname123> Found http://stackoverflow.com/questions/8812827/build-ffmpeg-with-x264-for-andro… [03:03] <grepper> did you do a make distclean or at least a make clean first ? [03:03] <Nickname123> and it made me think to use relative paths for some reason lol [03:03] <grepper> ah [03:03] <grepper> okay, as long as you solved it [03:04] <Nickname123> Thanks [03:04] <Nickname123> I think the -I and -L options were choking on C:/ [03:04] <Nickname123> but a relative path woked [03:04] <Nickname123> *worked [03:04] <llogan> Nickname123: there is this: https://ffmpeg.org/trac/ffmpeg/wiki/MingwCompilationGuide [03:05] <llogan> but it's not as comprehensive as i'd like [03:05] <Nickname123> I read through that first [03:05] <Nickname123> It helped me find all the extra libs. Ty [03:06] <llogan> i would prefer that it had actual step-by-step commands for users to copy and paste. [03:07] <llogan> (un)fortunately i don't use mingw [03:08] <Nickname123> This is my first time compiling on Windows instead of compiling on Linux for Windows [03:19] <alyawn> figured mine out... I needed to re-assign the codec, not just the codec_id after calling avformat_new_stream [03:27] <Nickname123> it built =) [06:01] <joecool> /usr/lib/gcc/x86_64-pc-linux-gnu/4.7.1/../../../../x86_64-pc-linux-gnu/bin/ld: libavcodec/x86/fft_mmx.o: relocation R_X86_64_PC32 against undefined symbol `memcpy@@GLIBC_2.14' can not be used when making a shared object; recompile with -fPIC [06:01] <joecool> /usr/lib/gcc/x86_64-pc-linux-gnu/4.7.1/../../../../x86_64-pc-linux-gnu/bin/ld: final link failed: Bad value [06:02] <joecool> i run a no-multilib system [06:02] <joecool> why is this happening? [07:32] <ubitux> Freakshow :) [07:53] <alyawn> am I supposed to re-init my AVPacket before I call avcodec_encode_video2()? [10:46] <sgfgdf> hello, guys! i'm trying to convert a video for an iPad. here is my attempt: my preset file -- http://cl.ly/2G1v0y3n0M292G2g053V , log -- http://cl.ly/0B1o1P300F0z372i1g2z . anyone can help, please? [10:46] <Vardan> hi all [10:46] <Vardan> people how can I get my ffmpeg library version number via code? [10:51] <Mavrik> Vardan: there's a "version.h" header defined with versions for each library [10:51] <Mavrik> also macros that give you better output [10:52] <Mavrik> sgfgdf: your preset sets invalid audio settings [10:54] <sgfgdf> Mavrik, which one is incorrect? ar, ab, ac? [10:55] <Mavrik> considering this line: Stream #0:1(eng): Audio: aac, 48000 Hz, stereo, s16, 1200 kb/s [10:55] <Mavrik> it's ar [10:56] <Mavrik> since ffmpeg thinks you're setting audio bitrate to 1200kbps :) [10:58] <sgfgdf> Mavrik, so i should change b= or ar= ? it's strange because i found the command from here -- http://www.ioncannon.net/meta/1040/how-to-create-ipad-formatted-videos-usin… and it appears to work probably for the author. is it my mistake or it is wrong from the source i get it? [10:59] <Mavrik> b is video bitrate [10:59] <Mavrik> nothing wrong with it [11:02] <ssuclinux> kg2 [11:07] <sgfgdf> Mavrik, i put b above ab, because it appears to override ab. now it is working. thank you! [11:10] <burek> sgfgdf, -ab controls audio bitrate and -ar controls the sampling frequency (44.1KHz) [11:10] <burek> or 48KHz in your case [11:10] <burek> so try -ab 96k [11:12] <burek> btw, your -ab gets overriden most probably because you use -b to set video bitrate [11:13] <burek> but the syntax changed a little bit in latest ffmpeg [11:13] <burek> -b:v is used for video bit rate and -b:a for audio bitrate [11:13] <burek> setting just -b affects both video and audio [11:14] <sgfgdf> burek, can i use b:v=1200k and b:a=128k in the ffpreset file rather than command arguments? [11:14] <burek> yes [11:15] <sgfgdf> burek, okay so better use them to eliminate confusions. thank you! [11:16] <burek> :beer: :) [11:19] Action: sgfgdf sends http://blogsensebybarb.files.wordpress.com/2012/04/beer.jpg to burek. hope i don;t have so much future questions :) [11:20] <burek> :excited: [11:23] <sgfgdf> burek, before drinking them you should count them and think again :) [11:25] <burek> one beer at a time :} [11:33] <varaderoguy> morning all [11:33] <varaderoguy> ffmpeg compiling fun.... [11:33] <varaderoguy> need some help and advice [11:35] <varaderoguy> So - I'm running the standard Centos 5.7 RPM version of FFMPEG 0.6.5 [11:35] <varaderoguy> I'm trying to build the night build with the same arguments..... [11:35] <varaderoguy> This is not as simple as it sounds..... [11:37] <varaderoguy> The good news is that just using the standard ./configure; make and make install on Centos, things work just fine [11:37] <varaderoguy> Its just that libraries are not being included [11:37] <JEEB> better not to do it with exactly the same arguments, distros tend to just EnableEverything while you usually have a more limited set of things that you need (given the fact that 99% of the decoders are inside libavcodec) [11:37] <JEEB> but look at it that you basically have everything LGPL enabled by default [11:37] <JEEB> and then you enable stuff that you need (libx264., libfaac, for example) [11:38] <JEEB> and yes, the libraries on an old centos are probably old [11:38] <JEEB> libfaac is one of those libraries that was pretty much left to rot so that might be up-to-date [11:38] <varaderoguy> JEEB: lovely - well that good news....I will need libx264 though (for transcoding to flv file format) [11:38] <varaderoguy> ...and the ac3 audio stack [11:38] <rainmaker1> Hi, is it possible to output the same stream to multiple destinations? [11:38] <JEEB> ac3 decoder and encoder are included in libavcodec [11:39] <JEEB> ok, you then most probably need a current yasm and current x264 [11:39] <varaderoguy> Varaderoguy HUGS JEEB [11:39] <varaderoguy> (with applogies to everybody) - ffmpeg -version give me..... [11:39] <varaderoguy> libavutil 51. 62.100 / 51. 62.100 libavcodec 54. 29.100 / 54. 29.100 libavformat 54. 11.100 / 54. 11.100 libavdevice 54. 0.100 / 54. 0.100 libavfilter 3. 0.100 / 3. 0.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 15.100 / 0. 15.100 [11:40] <JEEB> go to your home dir, do 'mkdir builds' and move into that folder [11:40] <JEEB> wget http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz && tar xvf yasm-1.2.0.tar.gz && cd yasm-1.2.0 [11:41] <JEEB> do you want to install to some specific place or /usr/local, which is often the default? [11:41] <burek> rainmaker1, yes: ffmpeg -i input (output options) output1 (output options) output2 (output options) output3 ... [11:42] Action: JEEB pokes varaderoguy [11:42] <varaderoguy> jeeb: sorry - I was just fetching yasm [11:42] <JEEB> k [11:43] <JEEB> then you basically ./configure and set --prefix=/where/you/want/to/install [11:43] <rainmaker1> burek: tnx :) [11:43] <varaderoguy> jeeb: /usr/local will be fine [11:43] <JEEB> k [11:43] <burek> :beer: :) [11:43] <JEEB> then you configure, make and make install -- and then you can check if yasm --version gives you 1.2.0 [11:44] <JEEB> I guess you have git already installed so the url for x264 would be git://git.videolan.org/x264.git [11:45] <varaderoguy> got git installed....just doing a make install on yasm [11:45] <JEEB> you clone it and configure it with --enable-static [11:45] <varaderoguy> lovely - yasm is 1.2.0 [11:45] <JEEB> and set prefix accordingly (but I think /usr/local is the default) [11:46] <varaderoguy> hummm.....got a slight problem in so much that I can do an https, but not git protocol [11:46] <JEEB> http://git.videolan.org/git/x264.git is on http://git.videolan.org/gitweb.cgi?p=x264.git;a=summary [11:46] <varaderoguy> jeeb: ** [11:48] <JEEB> basically --enable-static will build a static library. They're easier to deal with in this context so I recommend them unless you know your ways around ldconfig and friends [11:48] <JEEB> (x264 with a default config will only build the command line encoder tool) [11:48] <JEEB> also you should naturally check that asm is enabled in the configuration screen that comes up after you run the configure [11:48] <varaderoguy> okay....just having fun and games with git.... [11:49] <JEEB> I'm actually waiting for your compiler to fail at some point because I remember many centos 5 users stumbling somewhere along the line... [11:49] <varaderoguy> jeeb: stupid Q [11:51] <varaderoguy> jeeb: just doing make on x264 [11:52] Action: JEEB is left a bit irritated by a non-asked question [11:52] <varaderoguy> jeeb: humm...that passed off without issue [11:52] <varaderoguy> jeeb: okay - what didn't I ask? [11:53] <JEEB> <varaderoguy> jeeb: stupid Q <- [11:54] <varaderoguy> jeeb: sorry - I have a nasty habit of not finishing sentences....and then figuring the problem out for myself....my fault [11:55] <varaderoguy> okay - so I'm now ready to rock..... [11:55] <varaderoguy> It is a just a case of reconfiguring ffmpeg and letting it pick up the libraries? [11:56] <JEEB> ok, so I guess you got libx264 installed? and it had asm enabled? [11:57] <JEEB> then ./configure --enable-gpl --enable-libx264 and see if it spots the newer library in /usr/local or the older one from the package management [11:57] <varaderoguy> jeeb: I can confirm that asm is Enabled in x264 [11:57] <JEEB> also, if you hadn't gotten git before, how exactly did you get the source for ffmpeg? [11:58] <varaderoguy> no - it was my fault and the fact that I'm trying to type too quickly..... [11:58] <varaderoguy> I really apprechiate your help JEEB [12:02] <varaderoguy> jeeb: ./configure looks good; now just trying the make; make install [12:02] <JEEB> you could check the contents of config.log for which it actually picked up to be sure :) [12:08] <varaderoguy> ls [12:10] <varaderoguy> jeeb: That is fantastic....many thanks Jeeb [12:34] <varaderoguy> Chaps: another question: does the nightly builds come with a man file for ffmpeg? [12:37] <varaderoguy> ls [12:51] <varaderoguy> okay: the -metadata function is still broken in the latest build :-( [12:52] <varaderoguy> has anybody managed to get ffmpeg to write metadata to the file header for an FLV file? [12:52] <kcm1700_> does anyone know what's YUV440 format? I was searching for the information but I don't see any. [12:54] <varaderoguy> kcm1700: its a broadcast video format [12:55] <varaderoguy> kcm1700: Its to do with the number of bits of information in terms of lumance, and chromance [12:56] <kcm1700_> varaderoguy: thank you. Is there any web page(or search keyword) to learn the format? [12:56] <varaderoguy> kcm1700: try this for starters: http://en.wikipedia.org/wiki/YUV [12:57] <kcm1700_> ah.. [12:57] <kcm1700_> Luminance/chrominance systems in general <- this section has information, I missed it. [12:57] <kcm1700_> thank you [13:01] <kcm1700_> Ah, is YUV440 reordered format of YUV422? [13:03] <zap0> probably. [13:04] <varaderoguy> kcm_1700 - in essence, although you will find that there is no V sampling information, which for the purposes of colour correction can be an issue....but in terms of most data packets these days, in don't really need the V packets....its saves space [13:07] <kcm1700_> thanks for the helps, zap_0, varaderoguy! [13:12] <zap0> glad my huge contribution was recognized ;) [14:18] <AlRazi> I am trying to convert mp3 -> aac+, the resulting file has 7000+ hours duration on itunes, i tried running qt-faststart on it but to no avail [14:19] <AlRazi> I tried using ffmpeg to produce a wav file, and then convert the wav file to m4a using faac, it works fine, but faac doesn't support HE-AAC [14:47] <AlRazi> I am trying to convert mp3 -> aac+, the resulting file has 7000+ hours duration on itunes, i tried running qt-faststart on it but to no avail, I tried using ffmpeg to produce a wav file, and then convert the wav file to m4a using faac, it works fine, but faac doesn't support HE-AAC [14:56] <rainmaker1> Hi, is there any tool to help me find the atom size? [15:02] <rainmaker1> found :) [15:17] <AlRazi> I am trying to convert mp3 -> aac+, the resulting file has 7000+ hours duration on itunes, i tried running qt-faststart on it but to no avail, I tried using ffmpeg to produce a wav file, and then convert the wav file to m4a using faac, it works fine, but faac doesn't support HE-AAC [15:35] <Spideru> Hi. I would to connect ffplay to an rtp stream created with ffmpeg. On linux ffplay -f rtp rtp://<ip>:<port> works, on window Xp (with different prebuilds) nope. What i am missing? Thank you [16:49] <jesk> adding/modifying metadata in mp4 is a pita [16:49] <jesk> always rewrite from the scratch [16:50] <jesk> how can I modify a mp4 so that is has enough free atom space, something around 5MB would make sense [16:53] <Na_Klar> Can I affect the PNG pre-filter with ffmpeg? I want to decode a movie to a png image sequence but want to use NONE pre-filter as specified in the PNG defination. [16:58] <Diogo> hi one question please.. [16:58] <Diogo> i need to comunicate with ffser using ffmpeg [16:58] <Diogo> i'm using this command: /servers/ffmpeg/bin/ffmpeg -i HD.mp4 http://localhost:8090/test.asf [16:59] <Diogo> but it is to fast... [16:59] <Diogo> i'm doing something wrong? [17:28] <xero-exez> First things first: "Thanks again for FFMPEG!" [17:32] <xero-exez> Would someone be so kind to check my syntax. It transcodes all input files except ProRes. [17:32] <xero-exez> Error: Error setting profile baseline [17:32] <xero-exez> http://pastebin.com/Nw65Ck4u [17:36] <sacarasc> You have -profile and -vprofile... [17:36] <sacarasc> Also, if you had read... [17:36] <sacarasc> x264 [error]: baseline profile doesn't support 4:2:2 [17:38] <xero-exez> @sacarasc Thank you for your response! I tried with one of -profile and -vprofile there is no differance [17:38] <sacarasc> But you're trying to use 4:2:2 with a profile that doesn't support it. [17:40] <xero-exez> @sacarasc Allright I'm reading about YUV an downsampeling now....If i understand right the color input scheme cannot be downscaled? [17:43] <gnarface> hey guys i'm trying to transcode some video (a decoded VOB ripped from a commercial dvd i own) and though it successfully produces video with little complaint, the apparent playback of resulting video is unsteady [17:43] <gnarface> it speeds up and slows down [17:43] <gnarface> on an even interval [17:43] <gnarface> like it plays real fast for a half second then slows down to less than 1fps, then plays a bunch real fast, etc... nice even cycle [17:44] <gnarface> same behavior with single and multi-pass encoding [17:45] <gnarface> same behavior with multiple audio codecs [17:45] <gnarface> http://pastebin.com/hyeheLaf [17:45] <xero-exez> @sacarasc Thanks for putting me on the right track.... I added -pix_fmt yuv420p [17:45] <gnarface> here's an example run; i just tried adding -vsync 2 [17:46] <delicado> hi guys what function is available in ffmpeg library to convert the YUV format to RGB? i tried looking in avutils and avpicture for functions that start with "convert" but i did not find any. [17:46] <xero-exez> @delicado -pix_fmt yuv420p :) [17:49] <xero-exez> @gnarface try to lower your freames-per-keyframe -g 25 [17:49] <delicado> xero-exez: but im using the library. :( [17:49] <gnarface> xero-exez: same behavior if i don't specify any at all :( [17:50] <Mavrik> delicado: that's the task of libswscale [17:50] <Mavrik> delicado: look for sws_scale and other sws_* parameters [17:51] <Mavrik> er, functions, not parameters, sorry :) [17:51] <gnarface> xero-exez: trying it anyway [17:51] <gnarface> xero-exez: think this could be relevant? "[mpeg @ 0x20d4ec0] max_analyze_duration reached" [17:51] <delicado> thanks Mavrik. [17:52] <xero-exez> @gnarface Don't listen to me I'm just a n00b that had a question befor yours...and trying to do something back [17:52] <cen|3> could someone explain to me how to convert an asf to gif using ffmpeg/avconv??? I keep getting two errors - one stating that the gif only handles rgb24 pix form (use -pix_fmt rgb24 (but errors when I try to do this)) and another error stating that it could not write header for output file... [17:54] <Mavrik> cen|3: make sure you're using ffmpeg and not avconv and then paste the whole output with command line. [17:55] <cen|3> what I type is: avconv -i file.asf file.gif [17:55] <cen|3> k - hold on... [17:58] <cen|3> pastebin.com/CveXpEW0 [17:58] <cen|3> http://pastebin.com/CveXpEW0 [17:59] <cen|3> I'd like to make the gif resolution in the end much smaller as well (like 640xXXX) [18:06] <gnarface> does anyone happen to know if this implies a bug in ffmpeg? "Seems stream 0 codec frame rate differs from container frame rate: 59.94 (60000/1001) -> 59.94 (60000/1001)" [18:08] <cen|3> any idea as to what my problem is Mavrik? [18:10] <Mavrik> cen|3: hmm [18:11] <Mavrik> what do you get when you pass the pix_fmt parameter? [18:12] <cen|3> Option pixel_format not found. [18:13] <Mavrik> yes [18:13] <Mavrik> now read again the previous error message and check for typos. [18:18] <gnarface> wuldn't gif be rgb8 ? [18:19] <gnarface> nevermind i guess [18:19] <ePirat> hello [18:20] <ePirat> can ffmpeg segment a given file into segments of x segments length in specified format? cause I just saw the -segment_* options and wondered how to use them& [18:23] <jesk> anyone knowing how to create free? [18:23] <jesk> i mean a free atom for MP4 [18:25] <cen|3> sorry bout' that - I had to take a call [18:25] <cen|3> gnarface - I tried that also and it gave the same error [18:27] <cen|3> and if I try to use pix_fmt gif it says: Failed to set value 'gif' for option 'pix_fmt' [18:28] <cen|3> what is the difference between musing and demuxing support? gif format only has muxing support... would that matter? [18:28] <cen|3> musing=muxing [18:29] <sacarasc> cen|3: Muxing is putting the file together, demuxing is taking it apart. [18:29] <gnarface> cen|3: demuxing is the reverse of muxing... since the gif format only supports one video stream and no audio/subtitle/camera angle streams there's nothing from which to demux... [18:30] <cen|3> ah... ;) thx [18:36] <meekohi_> Are there any clever ways of reducing filesize of a video if you know it *loops*? I'm encoding to webm. [18:36] <meekohi_> I'm trying to explore the settings available, but it's a little overwhelming to guess what will and won't work. [18:53] <cen|3> got it - apparently gif support sucks for ffmpeg - so everyone extracts the images with ffmpeg and using another program like imagemagick to create the gif - thanks anyways.... [18:53] <cen|3> thanks anyways guys [18:54] <Mavrik> doh [18:54] <Mavrik> you were trying to create an animated gif [18:54] <Mavrik> yeah, that doesn't work [18:54] <Mavrik> sorry, thought you're just trying to get an image :) [18:55] <cen|3> yup - sorry - guess I should have said that... ;) [18:55] <cen|3> I assumed it was obvious considering I was pulling from a video file... haha - But this process works great.... [18:55] <cen|3> I really appreciate your efforts though [18:58] <Mavrik> cen|3: haven't used those in a while, completely forgot you can do that :P [18:58] <delicado> hi is the video format in each call to avcodec_decode_video2 always YUV420P? [19:00] <Mavrik> delicado: nope [19:00] <Mavrik> it's what the video is encoded in [19:00] <cen|3> Mavrik - Yeah I haven't had much need of them for a long time myself, but I'm trying to post an image on a forum and it won't accept video formats, but it does accept an animate gif (never used ffmpeg before though)... So... :) [19:01] <delicado> oh not YUV420P, when i read an AVFrame::format i always get 0. so its YUV. [19:05] <Mavrik> delicado: most videos are encoded to YUV420P, however you shouldn't rely on that [19:05] <Mavrik> H.264 can take YUV422P and other formats [19:05] <Mavrik> even RGB [19:14] <gnarface> question: is this bad? Duration: 00:00:04.98, start: 0.205433, bitrate: -2147483 kb/s [19:15] <Mavrik> gnarface: or a bug [19:15] <Mavrik> :) [19:15] <gnarface> Mavrik: well, is that normal output for a (decrypted) VOB ripped from a commercially-encoded, region 1 DVD? [19:16] <gnarface> actually i'm more interested in what the possible complications are that this could cause [19:17] <gnarface> namely whether it could cause the complication i'm having (where in the apparent video framerates wildly vary in the resulting transcoded video) [19:21] <Freakshow> Spideru: how are you outputing the rtp stream locally with ffmpeg? I'm just curious [19:24] <jesk> damn [19:24] <jesk> no way to add a free atom? [19:25] <jesk> anyone? [19:29] <varaderoguy> night all [19:43] <Rockj> How would I go about reporting an issue where ffmpeg exits with return code 0 on read error? [19:44] <Rockj> nor does it look like error messages are printed to STDERR, I guess its not supposed to be like this? [19:54] <AlRazi> I am trying to convert mp3 -> aac+, the resulting file has 7000+ hours duration on itunes, i tried running qt-faststart on it but to no avail, I tried using ffmpeg to produce a wav file, and then convert the wav file to m4a using faac, it works fine, but faac doesn't support HE-AAC [20:08] <delicado> how can i get a 'PixelFormat' from an AVFrame? so i can use it as parameter 3 for 'sws_getContext'. because it does not accept the AVFrame::format that i give to it. the compiler says i need an PixelFormat? [20:10] <Mavrik> format of the frame, -1 if unknown or unset Values correspond to enum PixelFormat for video frames, enum AVSampleFormat for audio) [20:10] <Mavrik> read the dcs [20:10] <Mavrik> it's int because it can be used for audio and video [20:11] <Mavrik> you'll have to cast it [20:11] <delicado> Mavrik: i tried putting 0 [20:11] <Mavrik> why 0? [20:11] <Mavrik> what would that accomplish? [20:11] <delicado> it says i need PixelFormat. [20:11] <delicado> yeah [20:12] <delicado> i mean it does not accept int [20:12] <Mavrik> yes. [20:12] <delicado> it needs a PixelFormat? how do i get it? [20:12] <Mavrik> --->> "int Values correspond to enum PixelFormat for video frames" <<--- [20:12] <Mavrik> do you even know C? [20:13] <Mavrik> cast the int to pixelformat [20:13] <delicado> i dont know C in depth. does '(PixelFormat)0' work? [20:13] <delicado> ah okay. ill try [20:14] <sente> i have an flv and a bunch of mp4s, how can I find the specific format/codec info of the flv, and reencode all the mp4s to be of that same type? [20:14] <sente> (this is so they can be streamed with amazon's streaming cloudfront abilities) [20:14] <delicado> it works. thanks [20:15] <Mavrik> delicado: why 0? [20:15] <Mavrik> if you want to convert from INPUT format to whichever OUTPUT format you want [20:15] <Mavrik> you're supposed to pass the INPUT format and the OUTPUT format to sws_scale [20:16] <Mavrik> sente: that's not a trivial problem [20:16] <delicado> Mavrik: no it was just a sample. because AVFrame::format is int. sorry im bad in english. [20:16] <Mavrik> delicado: ah, yea [20:16] <Mavrik> so basically [20:16] <Mavrik> (PixelFormat)AVFrame::format [20:16] <delicado> yes. [20:17] <Mavrik> sente: that's not an easy problem - run "ffprobe -i <file.flv>" to see it's data [20:17] <Mavrik> but it won't show exact encoding settings [20:18] <sente> i just need to make sure the new videos are "as streamable" as the flv is [20:18] <sente> if that makes sense [20:18] <sente> (i dont know much at all about video/codec stuff) [20:19] <Mavrik> hrm [20:19] <Mavrik> I guess you'll have to learn ;) [20:19] <ePirat> when trying to compile a program I've written using the libav libraries I get "Undefined symbols for architecture x86_64" (more: http://pastebin.com/yQFDL90j)& Would be great if someone could help me. thanks in advance& [20:19] <Mavrik> sente: what are you streaming them with? what's your target? who are you streaming them to? [20:20] <Mavrik> ePirat: that sounds like you either have 32-bit libs while compiling 64-bit target, or you forgot "-l" linking parameters, or your library path is broken [20:20] <Mavrik> (you're working on Linux?) [20:20] <sente> Mavrik: using flowplayer [20:20] <sente> ffmpeg -i MP4/1_Code.mp4 -sameq MP4/1_Code.flv [20:20] <Mavrik> ICK. Don't use sameq. [20:20] <Mavrik> :D [20:20] <sente> [flv @ 0x1ff27a0] flv does not support that sample rate, choose from (44100, 22050, 11025). [20:20] <Mavrik> sente: the question is [20:20] <Mavrik> why aren't you streaming mp4 files to flowplayer? [20:21] <Mavrik> it's the preferred way in new flash [20:21] <ePirat> Mavrik, oh right forgot -l *blush* (I am using Mac OS) thanks [20:22] <Mavrik> hrm [20:22] <Mavrik> can't help you with MacOS sadly [20:22] <sente> Mavrik: hrm, well the mp4's won't stream from amazon [20:22] <sente> but an flv will [20:22] <Mavrik> OS X has some wierd cocktail of 32/64bit arch with wierd compilers, don't have any at hand sadly :\ [20:22] <Mavrik> sente: I see [20:22] <Mavrik> sente: "sameq" doesn't do what you think it does, plus it reencodes whole video [20:22] <sente> Mavrik: it says the clip isn't found, when I try and stream the mp4 [20:23] <Mavrik> sente: I suggest you try just remuxing them to flv first [20:23] <sente> Mavrik: i see [20:23] <Mavrik> with ffmpeg -i <file.mp4> -codec copy <file.flv> [20:25] <ePirat> Mavrik, can i find a sample file using the av libraries in order to see if it's a problem with my code or another? [20:26] <Mavrik> ePirat: that looks like a linker problem [20:26] <Mavrik> ePirat: if you're compiling with gcc [20:26] <Mavrik> ePirat: I suggest you add "-L" path to the libav 64-bit libraries [20:32] <zap0> anyone do video editing for final cut pro ? [20:33] <ePirat> Mavrik, ok now i get a lot of implicit declaration errors http://pastebin.com/VPN8Cxd0 huh& [20:41] <sente> Mavrik: thanks [20:41] <sente> ill give that a shot [20:41] <sente> Mavrik: didn't work [20:41] <Mavrik> ePirat: that seems like your include files aren't being included correctly [20:42] <Mavrik> ePirat: use -I on compile stage [20:42] <Mavrik> to pass directory for include files [20:42] <sente> http://i.imgur.com/g2lJw.png [20:43] <Mavrik> sente: ah, your audio has wrong samplerate [20:43] <Mavrik> you'll have to reencode it [20:44] <sente> hmm, no idea how to do that [20:44] <Mavrik> ffmpeg -i <file.mp4> -codec:v copy -codec:a libmp3lame -b:a 128k -ar 44100 <file.flv> [20:44] Action: sente debates calling the customer and telling them to just fix it on their side [20:45] <sente> nice, that's atleast working [20:45] <sente> thank you [20:51] <dericed> i have a collection of .Y.U.V files and am having trouble reading them. I use "ffmpeg -i frame_%06d.Y" and get "frame_%06d.Y: No such file or directory". Am I referring to it wrong? [20:53] <zap0> dericed, zero based or 1 based? [20:55] <dericed> zap0: ? [20:55] <zap0> ? [20:55] <dericed> :) [20:55] <beandog> try quoting the filename? [20:56] <dericed> same [20:56] <beandog> bummer [20:56] <beandog> dunno [21:04] <dericed> found problem, my filenames started at the wrong number. fixed now [21:13] <llogan> dericed: there is also the -start_number option. allows you to choose the image to begin the sequence. [21:22] <dericed> llogan: I say that, glad it got added. I just wrote a loop to renumber the files because I have some gaps in the numbers as well. [21:55] <relaxed> dericed: also, if they're jpgs you can cat them to ffmpeg as input [22:14] <alyawn> does anyone have a good link to an example of properly setting PTS when encoding using libavcodec? [22:20] <alyawn> I was under the impression that having time_base set correctly (1/60) and setting PTS to the frame number was sufficient [22:29] <burek> alyawn, could you check the source code of the setpts filter? [22:30] <alyawn> burek, ok [22:32] <alyawn> burek, it seems that all setpts does is set the PTS to the evaluated value given to the filter [22:34] <burek> I see [22:35] <burek> well, can you check then how does a specific encoder does it [22:35] <burek> which video encoder are you currently trying to write [22:35] <burek> I mean, which video codec do you use for your code [22:36] <alyawn> I'm attempting to use H264 wrapped in a mpegts [22:38] <burek> hm then you should check how libx264 is doing it [22:38] <burek> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/libx264.c;h=d56df… [22:41] <alyawn> thanks, burek [22:44] <delicado> hi im leaking memory in this code http://codepad.org/oTIoxAKo. i have SDL functions in there before i isolated the code that leaks. why is it still leaking? other than av_free_packet(). what function is missing so i dont have the memory leak? sorry for the bad english. [22:45] <sente> Mavrik: the ffmpeg -i <file.mp4> -codec:v copy -codec:a libmp3lame -b:a 128k -ar 44100 <file.flv> example created flv's which had a vcodec error [22:45] <sente> any idea? [22:45] <Mavrik> what's a "vcodec error"? [22:45] <sente> when i play it in VLC it says there's an error [22:46] <Mavrik> *shrug* [22:46] <sacarasc> What is the error? [22:46] <Mavrik> flv is an old deprecated container which has problems with modern video codecs [22:48] <delicado> anyone? am i missing something? [22:48] <JEEB> it's not really deprecated and I'm not sure if it has problems with modern video codecs (compared to say, avi -- which has no support for b-frames at all) [22:48] <JEEB> it might just not have anything else to go for it but the fact that it's simple [22:48] <Mavrik> delicado: and where do you free the decoded frame? [22:49] <Mavrik> JEEB: there's some compatibility problems with H.264 which is the cause for Adobe to push mp4 for flash in newer versions [22:49] <JEEB> wut [22:49] <JEEB> H.264 goes just fine into FLV as far as I know [22:49] <Mavrik> lemme find the article [22:50] <JEEB> it better be a technical one [22:50] <JEEB> mp4 for apple is just natural because their mov is the base for mp4, and naturally not being limited by Adobe's specs is a good thing [22:50] <JEEB> it's also much more robust generally [22:50] <Mavrik> hrrm: Use of the H.264 and AAC compression formats in the FLV file format has some limitations and authors of Flash Player strongly encourage everyone to embrace the new standard F4V file format.[7] [22:50] <JEEB> ... [22:50] <Mavrik> from here: http://en.wikipedia.org/wiki/Flv#cite_note-kaourantin-6 [22:51] <Mavrik> however the linked article doesn't say anything concrete [22:51] <JEEB> yes [22:51] <ePirat> Can anyone take a look at this: http://pastebin.com/5pDPDPSp and maybe give me an hint what could be wrong? Really annoying sitting here for 3 hours trying to get it work& [22:51] <JEEB> also that's from 2007 and adobe is still using flv just fine for its streaming solutions (rtmp and friends) [22:52] <JEEB> and as far as I know technically the format fits H.264/AAC just fine [22:52] <Mavrik> JEEB: perhaps [22:52] <rm-rf> i'm running ffmpeg using daemontools to capture video from an IP cam, and every 180 seconds it cuts the stream and starts a new one so that the filesize doesn't get ginormous. in doing this, i'm finding that i gets a bunch of little files (200k-2MB) scattered among the actual 3 minute files, but can't figure out why [22:52] <rm-rf> any thoughts? [22:53] <sente> VLC does not support the audio or video format "undf". Unfortunately there is no way for you to fix this. [22:53] <JEEB> Mavrik, if you ever get any technical insight on the actual problems related to H.264/AAC audio in FLV other than "the timescale can't be !1000" [22:53] <sente> the audio is there [22:53] <juanmabc> ePirat: you would actually need to link "-lavcodec" or whatever [22:53] <JEEB> Mavrik, do feel to tell [22:54] <JEEB> anyone else is free too, of course [22:54] <Mavrik> JEEB: will do [22:55] <delicado> Mavrik: do i have to free the frame? what function can i use? is it avpicture_free? [22:55] <Mavrik> delicado: of course you have to [22:56] <Mavrik> at least at the end [22:56] <burek> rm-rf, how do you mean "using daemontools" [22:58] <delicado> yeah i added it. but it is still leaking. [22:59] <burek> sente, can you please use pastebin.com, to show your command line and its output? [23:00] <JEEB> Mavrik, looking at the FLV spec I can't really find any technical limitation [23:00] <Mavrik> JEEB: hmm, deleting that statement from Wiki would be a good move then [23:01] <JEEB> the move by Adobe from FLV to F4V seems just purely "we can build upon an openly specified format and we get to be able to stuff a whole lot of more stuff into this" [23:01] <rm-rf> burek: daemontools is a process that monitors other processes, runs certain processes based on criteria, etc. [23:01] <JEEB> also, man -- the flv spec is short [23:02] <JEEB> I bet people who liked AVI will like this [23:02] <alyawn> ok... I'm an idiot... I was reseting PTS to zero after each frame encode. thanks again, burek [23:03] <burek> alyawn, :beer: :) [23:03] <rm-rf> burek: one process i have, starts up ffmpeg to grab an rtsp stream and write it to a file. another process that accompanies it, monitors the runtime of the ffmpeg process, and kills it when it gets to 180 seconds. daemontools then sees that the process has "died" and starts a new one [23:03] <ePirat> juanmabc, ok seems to work with shared libs but static seems to fail& hm [23:03] <rm-rf> burek: does that makes sense? [23:03] <JEEB> (one of the main reasons why certain tool authors never moved away from AVI-based stuff is because AVI seriously was simple, and mp4 was a... clusterfuck, and matroska had problems of its own and/or wasn't known) [23:03] <burek> rm-rf, why do you that? [23:03] <burek> what is your overall goal? [23:03] <sente> burek: http://pastie.org/pastes/4156527/text [23:03] <rm-rf> to limit file sizes [23:04] <rm-rf> and the runtime of each file [23:04] <burek> why don't you use -t for example [23:04] <burek> to limit the time of the output [23:04] <rm-rf> would that end the process, or just cut the file off and start a new one? [23:04] <burek> end the process [23:05] <burek> sente, mpeg4 video and mp3 audio inside flv..? [23:05] <burek> why? [23:05] <sente> burek: i have no idea what i'm doing [23:06] <burek> :))) at least you are honest :) :beer: :) [23:06] <sente> I just need to convert the mp4 to flv so i can stream it with amazon's CDN [23:06] <sente> :) [23:06] <burek> I see [23:06] <sente> I know i should be able to stream mp4, but i can't get it to work [23:06] <JEEB> burek, he is doing video copy and at least that right [23:06] <sente> i can stream the flv just fine though [23:06] <JEEB> but the problem of course is that flv doesn't support that format [23:06] <JEEB> (MPEG-4 Part 2) [23:06] <burek> sente, do you see video when you stream it? [23:06] <rm-rf> burek: that might work well. is it '-t' measured in seconds? [23:07] <sente> burek: no, just black [23:07] <JEEB> wait... ffmpeg can actually copy mpeg-4 part 2 into flv...? [23:07] <burek> rm-rf yes, or you can set it in hour:min:Sec.msec format [23:07] Action: JEEB goes check [23:07] <JEEB> sente, yeah... I guess you're able to do it [23:07] <JEEB> but you should check the flv specification, page... 72 [23:07] <JEEB> mpeg-4 part 2 is not a supported video format for FLV [23:08] <burek> sente, is it viable for you that you re-encode your video too? [23:08] <sente> define viable? [23:08] <JEEB> http://pastebin.com/pFt55P3v [23:08] <JEEB> what the specs say [23:08] <sente> i dont care much about quality, i just need these things as FLV files, and preferably not 100x larger than the original mp4 files [23:09] <JEEB> use libx264 and crf [23:09] <JEEB> find the highest crf value that still looks good for you and the slowest preset that is fast enough for you [23:09] <burek> i see [23:09] <burek> try this then instead [23:09] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [23:09] <JEEB> and start from crf around 23 [23:09] <burek> ffmpeg -i 4_Energy.mp4 -codec:v libx264 -codec:a libmp3lame -b:a 128k -ar 44100 4_Energy2.flv [23:09] <JEEB> also why are you re-encoding the audio? [23:10] <JEEB> it seems like the input already is mp3 [23:10] <sente> JEEB: Mavrik suggested I had to [23:10] <JEEB> sente, nah [23:10] <JEEB> it should work [23:11] <sente> JEEB: okay so what's the command line I need, without the extraneous stuff? [23:11] <sente> I appreciate this [23:11] <JEEB> in the basics it's like what burek just now said [23:12] <sente> JEEB: okay trying that line [23:12] <sente> thanks [23:13] <JEEB> just switch -codec:a to copy [23:13] <JEEB> and remove the -b:a and -ar [23:15] <JEEB> ffmpeg's libx264 interface, if it uses the same defaults as libx264 should default to crf 23 and preset medium [23:15] <JEEB> so if it looks good, you can try a higher -crf [23:15] <JEEB> and if it looks bad, you can try lowering the -crf a bit [23:16] <JEEB> if it feels like it's going slow, you can tweak the -preset (it controls the internal speed vs more compression settings, while the crf is "how much do I compress this" [qp etc. wise, which basically makes it the 'quality' lever]) [23:16] <JEEB> and if it's fast enough you can try a slower preset [23:18] <rm-rf> i have two streams on this machine currently, one is an asf (foscam) and the other is mp4 (intellinet). is it possible to use the same command that i'm using to grab the mp4 stream with ffmpeg to grab the asf stream? cmd='ffmpeg -i <url> -vcodec copy -ar 44100 -t 300 /path/to/file' [23:19] <burek> hmh.. basically to grab the stream, it is enough just to use ffmpeg -i url [23:19] <burek> but what did you want to do with the rest of the command? [23:20] <rm-rf> i'm using the rest of the command to output the stream into a file. it fails if i don't have the '-ar' option, and i'm not sure if i completely need the '-vcodec' option or not [23:22] <rm-rf> burek: btw, thank you for the '-t' suggestion. that works so much better than killing the process and restarting it :) [23:26] <burek> rm-rf, well, if you want to put them all into a specific format/container (like flv or mp4) then you can't just blindly use -vcodec copy [23:27] <burek> because those formats might not support just any vcodec that you encounter [23:27] <burek> -ar just changes audio rate (sampling rate, usually 44.1 KHz) [23:27] <rm-rf> burek: i have a process that runs and converts them to mp4 [23:28] <burek> and it was maybe suggested to you for a specific case (for example mp3 in flv, which doesn't support 48 KHz mp3 audio, or something similar) [23:28] <rm-rf> i had to use the -ar, because i tried forcing the samplerate to 128k, then calling '-ar 128000', and it sounded like crap [23:28] <burek> well -ar 128k is just wrong [23:28] <rm-rf> but i'm all ears [23:28] <burek> -ab 128k might sound normal [23:28] <rm-rf> this is all new to me [23:28] <burek> but -ar no.. [23:28] <rm-rf> ok, noted [23:29] <burek> sample rate tells you at which rate you take audio samples (actually your audio card does that) [23:29] <burek> human ears can hear up to 21 KHz audio [23:29] <burek> so the double the rate is fair enough for catching all the frequencies human ear can hear [23:29] <rm-rf> i'm willing to look at anything suggested to me since i don't know much about the mplayer/vlc/ffmpeg world [23:30] <burek> 48 KHz is just being fancy :) [23:30] <burek> and -ab or the audio "bitrate" [23:30] <burek> is how much of the bandwidth would you use [23:30] <burek> for each second of your audio [23:30] <burek> i.e. 128kbps [23:31] <rm-rf> ah, not actually a samplerate [23:31] <burek> so, tell me first [23:31] <burek> what is your goal [23:31] <burek> what is your logical requirement [23:31] <burek> without much technical details [23:31] <rm-rf> basically surveillance video, in 5 minute increments, searchable on a php based webpage [23:32] <rm-rf> this is for a client, so i can't divulge too much more, and also, i'm not a paranoid freak [23:35] <burek> ok [23:35] <burek> so, you get your video from the camera [23:35] <burek> is it an ip cam [23:40] <rm-rf> burek: yes, one is a foscam IP camera, and the other is an intellinet IP camera [23:41] <rm-rf> the foscam stream is ASF formatted, the intellinet stream is MP4 formatted [23:47] <burek> so, ffmpeg -i http://url_to_cam -vcodec libx264 -acodec aac -strict experimental -ab 128k -ar 44100 -ac 2 out.flv [23:47] <burek> or you can use .mp4 instead of .flv [23:47] <burek> and then for your second cam you can just use -vcodec copy [23:48] <burek> to save your cpu/time without encoding the video [23:50] <alyawn> is this the correct way to specify x264opts using libavcodec? av_opt_set(c->priv_data, "x264opts", "no-ssim", 0) [23:51] <alyawn> guess I can look in ffmeg.c, nvmd [23:57] <rm-rf> burek: ah, so you are thinking that one cam will always encode video on the file, then the other cam would capture the raw stream and encode after the fact? [23:58] <rm-rf> burek: i have to run, but thank you for your help [00:00] --- Wed Jun 27 2012

1 0

ffmpeg-devel.log.20120625
by burek 26 Jun '12

26 Jun '12

[00:38] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r166f386446 10ffmpeg/tests/fate/vcodec.mak: [00:38] <CIA-119> ffmpeg: fate: speedup dnxhd tests and reduce their memory requirements [00:38] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:39] <durandal_1707> finally [02:04] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rf15803e957 10ffmpeg/tests/fate/acodec.mak: [02:04] <CIA-119> ffmpeg: fate: Try to fix source path for fate-acodec-dca [02:04] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:48] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r5fd3e6965e 10ffmpeg/ (libavformat/vocenc.c tests/ref/lavf/voc_s16): [03:48] <CIA-119> ffmpeg: vocenc: use new header from codec tag 4 [03:48] <CIA-119> ffmpeg: this matches sox and should fix ticket1119 [03:48] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:48] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * r0abfb0a9d8 10ffmpeg/ (libavformat/vocenc.c tests/lavf-regression.sh): [03:48] <CIA-119> ffmpeg: vocenc: change default codec to 16bit [03:48] <CIA-119> ffmpeg: Hardly anyone would want 8bit today, 16bit is a much more reasonable [03:48] <CIA-119> ffmpeg: default. [03:48] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:48] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rbacbbd2b03 10ffmpeg/ (4 files in 3 dirs): [03:48] <CIA-119> ffmpeg: vocenc: fix sample rate rounding direction [03:48] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:09] <brocatz> hey guys, i just compiled ffmpeg with the mingw package so that i could apply a patch and everything works fine, but now when i try to use avisynth it fails, it says permission denied, but i think that error is a side effect of another error, my other version of ffmpeg from 2010 (when i last used avisynth with it) works fine [13:10] <brocatz> wondering if there are any obvious things i should check [13:10] <brocatz> er w/w [13:24] <Compn> find out what permission is failing [13:24] <Compn> probably script in wrong dir [13:25] <brocatz> the exact command works on an older binary [13:25] <brocatz> i have 3 binaries, a recent nightly, my build, and an older one [13:25] <brocatz> swapping them out, so everything else is the same, the older one runs fine [13:25] <brocatz> [avs @ 00000000016cf320] AVIFileOpen failed with error -2147221164 [13:25] <brocatz> C:/temp/output.mp4.1.avs: Operation not permitted [13:25] <brocatz> that's the error btw [13:25] <Compn> oh [13:26] <Compn> well then something may have broken, or changed, since then [13:26] <Compn> update your script to new avisynth syntax ? [13:26] <brocatz> i'm using the Version avisynth script [13:26] <brocatz> which is very basic [13:29] <brocatz> hum the newest nightly i just got is fine [13:29] <brocatz> could i be inadvertently compiling my ffmpeg as 64bit [13:32] <brocatz> http://dpaste.com/763205/ that's my configure string [13:38] <brocatz> maybe someone could integrate our small patch so i can just use the main version of ffmpeg :) [13:45] <Compn> what patch ? [13:46] <brocatz> https://ffmpeg.org/trac/ffmpeg/ticket/1463 [13:48] <Compn> ah yest hat one [13:49] <michaelni> brocatz, if i understood reimar correctly that would break support for pre win2k ? [13:50] <michaelni> that is a binary compiled on win2k wouldnt work on previous versions ? [13:51] <michaelni> while one compiled on previous versions would have non working ctrl-break on later [13:51] <piman> The patch doesn't break support for anything. [13:52] <piman> It adds support for Ctrl+Break on Windows >= 2k. [13:52] <piman> (Hi, I'm the patch author.) [13:52] <michaelni> hi piman [13:52] <piman> Is pre-2k Windows a legitimate target for you guys still? [13:52] <michaelni> honestly i dont know :) [13:53] <michaelni> but if it works before your patch then i would prefer it still does afterwards [13:54] <michaelni> if it doesnt then the configure check shouldnt be needed [13:54] <piman> I'd think any problems it introduces are identical to whatever problems are introduced by the use of e.g. GetProcessMemoryInfo which is done in the same way my patch is. [13:55] <piman> (And has been there for years, and is XP+ only) [13:57] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r972cad77fa 10ffmpeg/libavfilter/ (11 files): (log message trimmed) [13:57] <CIA-119> ffmpeg: lavfi: remove unnecessary inclusion of libavcodec/avcodec.h in avfilter.h [13:57] <CIA-119> ffmpeg: libavfilter API was designed in order to be clarly distinguished from the [13:57] <CIA-119> ffmpeg: libavcodec API, including avcodec.h in avfilter.h is not going to help to [13:57] <CIA-119> ffmpeg: stick to this principle. [13:57] <CIA-119> ffmpeg: The inclusion of libavutil/audioconvert.h in many files was required [13:57] <CIA-119> ffmpeg: because avcodec.h includes audioconvert.h. [13:57] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * rfec512a52c 10ffmpeg/ffplay.c: ffplay: give more meaningful names to the buffersink instances [13:57] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r43583fb85c 10ffmpeg/ (5 files in 2 dirs): (log message trimmed) [13:57] <CIA-119> ffmpeg: lavfi/avcodec: deprecate avfilter_fill_frame_from_*_buffer_ref API [13:57] <CIA-119> ffmpeg: Deprecate functions: [13:57] <CIA-119> ffmpeg: avfilter_fill_frame_from_buffer_ref [13:57] <CIA-119> ffmpeg: avfilter_fill_frame_from_audio_buffer_ref [13:57] <CIA-119> ffmpeg: avfilter_fill_frame_from_video_buffer_ref [13:57] <piman> If you'd rather the configure check be gone I can change it to #ifdef _WIN32 but I don't really see the benefit in that. I usually prefer to test for the features I need rather than platforms. [13:57] <CIA-119> ffmpeg: and schedule to drop them at the next API major bump. [13:57] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * ra31ab50712 10ffmpeg/libavfilter/ (avfilter.h version.h): [13:57] <CIA-119> ffmpeg: lavfi: move some FF_API_ definitions from avfilter.h to version.h [13:57] <CIA-119> ffmpeg: version.h seems the right place for FF_API_ definitions. [13:58] <piman> (e.g. maybe Windows 9 RT stops defining _WIN32 finally, my check would still work.) [13:59] <michaelni> piman, are versions prior 2k still supported by MS ? i mean in terms of security updates ? [14:00] <piman> The usual policy for that kind of stuff for a company that size is "if you pay us enough, we'll do it" [14:00] <piman> But no, it's not generally supported at all anymore. [14:01] <brocatz> XP Support is about to end too, April 2014 [14:01] <michaelni> i guess then the patch might be ok, let me take another look at it [14:02] <burek> too bad.. it's the best OS they've ever had [14:02] <piman> What exactly is the concern? I guess you're thinking someone compiles on Win7, gets a binary, and it doesn't run on Windows 98? [14:03] <michaelni> yes [14:03] <piman> But that's going to happen anyway because of GetProcessMemoryInfo, and the fact every Windows compiler generates i686 instructions, etc. [14:04] <michaelni> i was unaware of the GetProcessMemoryInfo issue, this maybe should be changed somehow if its possible [14:04] <Tjoppen> I find it interesting old 3.1 programs still work in XP [14:04] <Tjoppen> or clock.exe did at least [14:05] <Compn> michaelni : no, win2k is not supported anymore [14:05] <Compn> winxp is due to expire soon (next few years iirc) [14:06] <Compn> oh brocatz mentioned this already :P [14:06] <brocatz> most of the people i know doing ffmpeg + windows work use virtual servers or real servers running 2008 server [14:06] <Compn> virtual really took off lately [14:07] <Compn> multiple cores did it i think [14:15] <michaelni> piman, #ifdef are wrong, these things get defined to 0 so compile fails with the patch [14:24] <michaelni> piman, if you resubmit the patch with #ifdef fixed ill apply it in a day or 2 if noone objects [14:25] <piman> OK. To Trac or the mailing list? [14:25] <michaelni> want to give reimar a chance to comment [14:25] <michaelni> ML is better [14:28] <piman> OK, thanks. [14:28] <michaelni> np [14:46] <Nedwada> hello guys [14:46] <Nedwada> 'm having a problem, using ffmpeg version 0.10.2.git. If i open an m3u8, and this is redirected inside its stream, i get a segfault error [14:47] <Nedwada> http://privatepaste.com/2374deb5f0 [14:53] <Nedwada> can someone help? [16:39] <burek> just to say this and wont prolong the topic discussed there on #ffmpeg :) please take a look at http://doc-book.sourceforge.net/homepage/ and if it's a viable solution I can convert all the texi into XML files [16:39] <burek> so we can resolve that issue once for good [16:49] <burek> DocBookWiki can also be used to edit a DocBook document online, from the web. Editing is done one section at a time, so the editor selects first the section that he wants to edit, and then edits it. He can edit it in several modes: text (like wiki), xml (the original format), html, latex, texi, etc., whichever is more suitable for him. The changes, however, are always saved in the XML(DocBook) format. Authentication of the editors (with username, pas [16:49] <burek> sword) can be enabled as well, if necessary. Also, different editing permissions can be assigned to editors. [16:50] <burek> so, devels edit xml files in text editor (offline), submit it through git (just like they use with texinfo files) [16:50] <burek> with the difference that those same xml files can be nicely edited from the web, by regular users (i.e. not developers) [17:24] <ubitux> https://lists.libav.org/pipermail/libav-devel/2012-June/029761.html [17:24] <ubitux> this might interest ffmpeg dev as well [17:26] <saste> http://sophia.estec.esa.int/socis2012/ [17:27] <saste> which reminds me that we still didn't integrated the patch from socis 2011 [17:27] <saste> *patches [17:48] <michaelni> ubitux, ffmpeg will keep r_frame_rate, its essential for supporting some containers [17:49] <ubitux> can you explain a bit? [17:49] <michaelni> avi [17:49] <ubitux> i don't see anything directly related in avi muxer [17:49] <ubitux> (or demuxer) [17:50] <ubitux> or maybe it's calling one of the lavf/utils [17:52] <michaelni> iam not sure what i should explain, avi needs r_frame_rate or equivalent [17:53] <michaelni> if eleril removes it in libav thats just another feature we have more :) [17:53] <av500> for avi average frame rate == frame rate [17:53] <michaelni> no [17:53] <av500> no? [17:54] <michaelni> avi supports skiped frames [17:54] <michaelni> so its vfr strictly speaking [17:54] <michaelni> just not efficiently [17:54] <av500> well, strictly speaking it does not I would say [17:54] <av500> people put 0 sized frames inside it [17:55] <av500> but anyway, no point to argue that [17:57] <CIA-119> ffmpeg: 03Ronald S. Bultje 07master * r8123e0901f 10ffmpeg/ (11 files in 3 dirs): [17:57] <CIA-119> ffmpeg: x86: place some inline asm under #if HAVE_INLINE_ASM [17:57] <CIA-119> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [17:57] <CIA-119> ffmpeg: 03Mans Rullgard 07master * r8299260470 10ffmpeg/libavcodec/x86/ (Makefile fft_mmx.asm fft_sse.c): x86: fft: convert sse inline asm to yasm [17:57] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * ra6ff8514a9 10ffmpeg/: (log message trimmed) [17:57] <CIA-119> ffmpeg: Merge remote-tracking branch 'qatar/master' [17:57] <CIA-119> ffmpeg: * qatar/master: [17:57] <CIA-119> ffmpeg: wtv: Check the return value from gmtime [17:57] <CIA-119> ffmpeg: x86: fft: convert sse inline asm to yasm [17:57] <CIA-119> ffmpeg: x86: place some inline asm under #if HAVE_INLINE_ASM [17:57] <CIA-119> ffmpeg: Conflicts: [17:57] <CIA-119> ffmpeg: 03Martin Storsjö 07master * rdc53858063 10ffmpeg/libavformat/wtv.c: [17:57] <CIA-119> ffmpeg: wtv: Check the return value from gmtime [17:57] <CIA-119> ffmpeg: On MSVC, gmtime returns NULL for values outside of their supported [17:58] <michaelni> av500, btw i got the old beagle board half working :) [17:58] <michaelni> needed a powered usb hub for the lan usb to work [17:58] <michaelni> and the timing of connecting the hub has to be right [17:59] <michaelni> connect it too early and nothing will work on the otg usb [17:59] <av500> ok [17:59] <av500> yes, that otg is a pain [17:59] <av500> fighting it here at work since years :( [18:00] <j-b> beagle is the one in omap3? [18:00] <av500> yes [18:01] <michaelni> about r_frame_rate, if someone has an idea on how to do avi vfr without it iam very interrested [18:01] <michaelni> OTOH if this is just removial of a features then iam not and it sounds like it is [19:21] <durandal_1707> ubitux: when you gonna remove SVN from download page? [19:21] <ubitux> gonna push tonight [19:21] <ubitux> i want to do something about Compn comment first [19:22] <ubitux> and i will likely update the documentation page as well (to add the wiki URL at least) [19:23] <saste> durandal_1707: are you going to update your smtpebars source? or do you want me to refactor it? [19:23] <durandal_1707> saste: not any time soon, i'm working on stupid undocumented video codec(s) [19:23] <ubitux> saste: what is blocking you from submitting your segmenter branch? :) [19:23] <ubitux> durandal_1707: smush ? [19:24] <durandal_1707> ubitux: san variant [19:24] <ubitux> ok [19:24] <ubitux> durandal_1707: btw you forgot the @tab X @tab X [19:24] <ubitux> in general.texi [19:25] <ubitux> (put the X where you need to, but the @tab are missing, and output is "broken") [19:25] <ubitux> (look for Smush on http://ffmpeg.org/general.html#File-Formats) [19:25] <durandal_1707> omg [19:38] <CIA-119> ffmpeg: 03Paul B Mahol 07master * rebfcd6049f 10ffmpeg/doc/general.texi: [19:38] <CIA-119> ffmpeg: doc/general: fix output [19:38] <CIA-119> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [19:48] <ubitux> durandal_1707: thx :) [21:16] <CIA-119> ffmpeg: 03Carl Eugen Hoyos 07master * rbec21ce7f4 10ffmpeg/libavcodec/mss1.c: Make MSS1 palette opaque. [22:04] <durandal_1707> is there filter which merges multiple audio streams into single stream? [22:16] <ubitux> amerge? [22:23] <ubitux> yay, some free time at least. [22:23] <ubitux> (last*?) [22:29] <ubitux> Compn: i think the git clone should be before the snapshot url... [22:49] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r8a3544149f 10ffmpeg/libavfilter/avfilter.h: [22:49] <CIA-119> ffmpeg: lavfi/avfilter.h: reorganize headers disposition [22:49] <CIA-119> ffmpeg: Group lavfi headers together, slightly improve readability. [22:49] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r6be8cfa034 10ffmpeg/libavfilter/buffersrc.c: [22:49] <CIA-119> ffmpeg: lavfi/abuffer: increase logging level of the log showing initial parameters [22:49] <CIA-119> ffmpeg: Also show them in a more parsable/consistent fashion. [22:49] <CIA-119> ffmpeg: 03Stefano Sabatini 07master * r9b41ec4b9e 10ffmpeg/libavfilter/vf_drawtext.c: [22:49] <CIA-119> ffmpeg: lavfi/drawtext: use av_opt_free() to free private context [22:49] <CIA-119> ffmpeg: Simplify code. [22:49] <RobertNagy> shouldn't the atempo filter change the time_base of the fitler graph? [22:50] <RobertNagy> as it is now time moves at the same rate as without atemp [22:50] <RobertNagy> which causes video to run at it's normal rate when using audio as master clock [22:54] <ubitux> saste: check if it passes make checkheaders [22:55] <ubitux> (about your stddef.h include) [22:56] <saste> ubitux: I moved it up, how can that break something? [22:56] <ubitux> i was refering to the original patch :) [23:00] <saste> RobertNagy: the *pts* is rescaled, not the time_base [23:01] <RobertNagy> the pts I get out is wrong then [23:01] <RobertNagy> and why scale the pts? isn't it more accurate to change the time_base? [23:02] <RobertNagy> I have to do "av_q2d(time_base) * tempo" to get the correct time [23:02] <RobertNagy> though maybe I'm the one that's wrong [23:05] <saste> RobertNagy: time_base is conveniently expressed as 1/sample_rate, why to change that? [23:06] <Daemon404> 1/sample_rate seems like the best idea... [23:06] <RobertNagy> well the problem I have is that, even if i set temp=0.5, time moves at the same rate as before [23:07] <RobertNagy> I would expect time to move 0.5 times slower [23:07] <RobertNagy> so that when I sync the video clock with the audio clock [23:07] <RobertNagy> the video will got at the same rate as the audio [23:07] <RobertNagy> though I'm probably missunderstanding how the atempo filter is ment to be used [23:08] <durandal_1707> hmm, audio and video are separate streams [23:08] <RobertNagy> yes? I still need to sync them [23:10] <durandal_1707> for video stream you would use another filter... [23:11] <RobertNagy> now you lost me, what does this have to do with the video filters? [23:15] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rcd6aa95caa 10ffmpeg/libavcodec/h264_loopfilter.c: [23:15] <CIA-119> ffmpeg: h264_loopfilter: use av_assert [23:15] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:15] <CIA-119> ffmpeg: 03Michael Niedermayer 07master * rdc30c27eb1 10ffmpeg/libavcodec/h264_parser.c: [23:15] <CIA-119> ffmpeg: h264_parser: use av_assert [23:15] <CIA-119> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:18] <durandal_1707> amerge can merge only 2 streams into one [23:18] <Daemon404> can you chain multiple amerges? [23:18] <durandal_1707> lol, description is ffmpeg output is wrong [23:18] <durandal_1707> doc/filters mentions n streams [23:22] <saste> RobertNagy: how are you synching the two streams (A+V)? [23:23] <RobertNagy> audio_time = audio_frame->pts * audio_filter->time_base [23:23] <saste> or in other words, are you doing it programmatically or through ff* tools? [23:23] <RobertNagy> programmatically [23:24] <RobertNagy> video_time = video_frame->pts * video_filter_->time_base [23:24] <RobertNagy> while(video_time < audio_time) next_frame(); [23:24] <RobertNagy> something like that [23:26] <RobertNagy> while(video_time < audio_time) next_video_frame(); [00:00] --- Tue Jun 26 2012

1 0

ffmpeg.log.20120625
by burek 26 Jun '12

26 Jun '12

[00:27] <tpd> hey, all im going through the source looking for the amr narrow band code [00:28] <tpd> there is wide band c source, but i cant find them [00:29] <tpd> i see a ffmpeg/libavcoded/amrnbdec.c [00:29] <burek> Megapixar, yes you can [00:30] <tpd> oh shit, im tarded [00:33] <burek> raptor67682, http://ffmpeg.gusari.org/viewtopic.php?f=25&t=39 [00:33] <juanmabc> .oO(MegaPixal would be a cool nick) [00:36] <burek> "using the built-in segmentation feature" :D I already thought of segmentation faults that were recently discovered and connected the statement with that and loled for a couple of seconds :D [00:37] <burek> drno_, why dont you simply name your output files with current date [00:37] <burek> like ffmpeg -i ... output-`date "+Hms"`.avi [00:38] <lsb> hey all, im looking for a amr narrow band encoder in libavcodec, im looking at the source, and all i see is a decoder, [00:39] <burek> that should have been: ffmpeg -i ... output-`date +"%H-%i-%s"`.avi [00:39] <burek> or just use %F (full date) [00:39] <juanmabc> then if you are lucky it's work in progress, lsb [00:40] <lsb> hah, well aparently ffmpeg can encode to amr [00:40] <durandal_1707> lsb: via libopencore-amr [00:44] <lsb> so its not apart libavcodec? [00:44] <lsb> a part of [00:44] <JEEBsv> it's a library that then can be used via libavcodec [00:44] <JEEBsv> just like libx264 for H.264 encoding [00:51] <lsb> well here is what i have gathered [00:51] <lsb> do you knw about audacity [02:12] <drno_> burek: Sorry for the delay. =) Thanks for your reply. I have tried that actually. With the built-in segmenter, the filename is interpreted ONCE, and only once. [03:49] <Sashmo> Can anyone help me out here. I am getting tons and tons of errors while decoding a multicast stream. If use a none DVB source, just an IP source from a professional encoder, it works amaizng, but if I use a DVB source from satalite, I get tons of errors ->, http://pastebin.com/nm8Zt3HR [04:24] <lolfrenz> when using libmp3lame to convert to mp3, how can I make it use the best quality it can? (i.e. supply no bitrate) [04:25] <lolfrenz> I'm converting a flv to an mp3 using ffmpeg -i file.flv -acodec libmp3lame file.mp3 [06:00] <brocatz> i need to capture from dshow to disk then i need to use avisynth to edit the file, then i need to output it to x264 [06:00] <brocatz> what format should i use for the initial capture [06:00] <brocatz> audio and video [06:00] <teratorn> anything wrong with x264, high quality? [06:01] <brocatz> i don't think i need to actually encode it first time through [06:01] <brocatz> i was thinking yuv or whatever [06:01] <teratorn> what [06:01] <teratorn> format and how long? [06:02] <brocatz> 1280x720 @ 25 for a few minutes, but i can put a 500gig ssd into the machine easy enough [06:02] <brocatz> it's recording people give interviews [06:02] <brocatz> then automatically editing them afterward [06:02] <teratorn> you want to trust RAM to that? [06:02] <teratorn> :) [06:02] <brocatz> third generation intel ssds should be suitable [06:02] <teratorn> i guess [06:03] <teratorn> what's this for, if you don't mind me asking? [06:03] <brocatz> letting people give their impressions during a trade show, like a photobooth but for video [06:04] <brocatz> friend just finished building the booth shell this week [06:04] <teratorn> commercial or just personal stuff? [06:04] <brocatz> i'd say prototype [06:04] <teratorn> its funny i work in that space [06:04] <brocatz> we're just a couple of freelancers [06:04] <teratorn> well not that funny [06:04] <teratorn> cool [06:04] <teratorn> http://catturavideo.com that is where i work [06:05] <brocatz> oh cool, yeah i see what you mean [06:06] <teratorn> yeah :) [06:06] <teratorn> it's interesting [06:07] <teratorn> have you been to any shows lately? [06:09] <brocatz> what kind of shows [06:09] <brocatz> i'm in new zealand [08:24] <brocatz> hey, i'm trying to use ffmpeg with avisynth, i've done that before but now i'm getting the same problem as this guy is having http://ffmpeg.org/trac/ffmpeg/ticket/1410 [08:28] <TACPILOT> I am using avisynth with ffmpeg. Is there a way to use a piped stream as the input in an avs file ?? [08:40] <brocatz> hum looks like the ffmpeg build [09:41] <o]> I need to change the mp3 audio codec from a video to any other that my tv supports. [09:41] <o]> how can I do that with ffmpeg? [10:06] <brocatz> so my ffmpeg from 2010 runs avisynth fine, but the newer versions i have, including the one i compiled a few days ago, don't want to run it [11:18] <phoenixson> I have general question on video formats. What are the safest formats to go for in terms on web video atm? [11:19] <zap0> teh web is a transport medium. video is a file. they are different things. [11:20] <phoenixson> well I was thinking in terms of compability its tricky btw webm, mp4, ogg, flv [11:20] <Tjoppen> phoenixson: at work we're using mp4/h264/mp3, ogg/theora/vorbis and webm/vpx/vorbis [11:21] <Tjoppen> for container/video/audio [11:21] <phoenixson> and thats overs IE, Mozilla, Chrome, Safari and Opera? [11:21] <phoenixson> covers* [11:21] <Tjoppen> dunno. works in firefox and chrome at least [11:22] <phoenixson> ha its something [11:22] <Tjoppen> should work in recent versions of ei too, since win7 does mp4 [11:22] <phoenixson> but then the whole Mac army is using Safari and Opera [11:22] <Tjoppen> I'd be surprised if mac didn't do mp4 [11:23] <phoenixson> well I am will try the ones you suggested. [11:23] <phoenixson> content is going to be delivered to me and then I will do testing [11:23] <Tjoppen> of course there's plenty of pitfalls with what profiles etc. you choose (esp. for h264) [11:23] <Tjoppen> just fuck around and search the web a bit - you'll figure it out [11:24] <phoenixson> thats true...thats for your help tho [12:04] <HerbertPumpkin> Will ffmpeg use DPX sequences as input these days? Could I get it to mux in a soundtrack at the same time? If so, how? I've googled, but DPX-related stuff seems a bit obscure. [12:05] <HerbertPumpkin> Best I can find is something like ffmpeg -f image2 -i "Movie_Frame_%04d.dpx" but I'm not sure if that's correct. [12:05] <Tjoppen> dpx works, as long as they're named in a way you can use the percent syntax for [12:05] <Tjoppen> like any other image sequence [12:05] <HerbertPumpkin> what's the whole "image2" thing [12:06] <Tjoppen> magic. I'm not sure if it's required in this case [12:06] <burek> o], ffmpeg -i input.avi -vcodec copy -acodec ... output.avi [12:07] <HerbertPumpkin> so, perhaps something like: [12:07] <gavlig> hi. i'm using libav(codec, format and utils) from my application to cut videos. i'm doing that by copying the frames i need from one stream to the new one. It works most of the time but not with videos which have parser in their streams. If they have parser, output video doesn't contain framerate, fps, tps and bitrate info. Any thoughts what could cause that? [12:07] <HerbertPumpkin> ffmpeg -f image2 -i "movie_%04.dpx" -r 24 -acodec libx264 -b 5000k [12:07] <HerbertPumpkin> ...maybe? [12:07] <Tjoppen> -r 24 before -i [12:08] <Tjoppen> -vcodec libx264 [12:08] <HerbertPumpkin> oh. [12:08] <HerbertPumpkin> of course, that was more a slip of the fingers :) [12:08] <Tjoppen> use CRF mode instead of specifying a bitrate, like -crf 24 [12:08] <HerbertPumpkin> I find it's very hard to make ffmpeg stick to any particular bitrate anyway [12:08] <HerbertPumpkin> tends to do what it wants [12:08] <HerbertPumpkin> god knows how you're supposed to actually tell it how many bits per second you want\ [12:09] <HerbertPumpkin> what would 24 give me? [12:09] <Tjoppen> you wouldn't - you tell it what quality you want [12:09] <Tjoppen> 24 is x264's (the CLI) default [12:09] <Tjoppen> lower is better [12:09] <burek> gavlig, you can always read ffmpeg's source code to figure out some things [12:09] <HerbertPumpkin> I want about 5mbps [12:10] <Tjoppen> just run it and see what you get [12:10] <HerbertPumpkin> That's ridiculous; can't I specify numerically? [12:10] <Tjoppen> aiming for some bitrate only makes sense if you're broadcasting or need to stick the file on some limited size media [12:10] <burek> HerbertPumpkin, yes you can [12:10] <HerbertPumpkin> It makes sense if that's what I've been asked for. [12:11] <Tjoppen> -b should work fine then [12:11] <HerbertPumpkin> Why is CRF recommended? [12:11] <Tjoppen> let me turn that around: why would you want CBR? [12:11] <gavlig> burek: i knew you'd say so :) but i can't find the solution of my problem there. I'm copying everything to the new context as it is done in ffmpeg.c but that doesn't help. I think there is something less obvious than copying info from one context to another, but i have no idea what is it [12:12] <HerbertPumpkin> Tjoppen: I have no idea. That's what I've been asked for. Mine is not to reason why. [12:12] <HerbertPumpkin> The documentation is pretty poor, by the way. I looked at it to figure out how to set frame rate, for instance. [12:12] <Tjoppen> well, there you go then. I care about apparent quality, not bitrate [12:12] <HerbertPumpkin> It tells you what valid options are (things like 24000/1001), but not what the actual option is called, "-r". [12:12] <HerbertPumpkin> Which is crazy. [12:13] <HerbertPumpkin> What if I wanted to mux in a soundtrack too? [12:13] <Tjoppen> an extra -i and maybe use -map. I don't recall the syntax [12:14] <burek> gavlig|food, http://ffmpeg.org/developer.html [12:14] <HerbertPumpkin> Is there a -crop as well as -s for scale? [12:15] <burek> it is possible to use VBR (-crf) with defined max bit rate [12:15] <burek> i.e. to tell FFmpeg to make VBR but not to go too far with bit rate [12:16] <HerbertPumpkin> I assume it won't read the framerate in DPX headers [12:16] <HerbertPumpkin> so I'll need to give it -r 24000/1001 etc [12:16] <burek> HerbertPumpkin, from which url did you read the docs? [12:16] <HerbertPumpkin> Several. [12:16] <burek> did you read official docs? [12:16] <HerbertPumpkin> Exactly what constitutes "official docs" tends to change depending on who you talk to. [12:17] <burek> http://ffmpeg.org/ffmpeg.html#Video-Options [12:17] <HerbertPumpkin> if I specify -s 1920x1080 it'll scale the input to fit, as opposed to scale and crop, right? [12:17] <burek> and it doesn't depend on anything but the official domain name :) [12:17] <burek> first read that link [12:17] <burek> and then tell me what do you see [12:18] <HerbertPumpkin> How would I go about cropping, say, 100 pixels top and bottom [12:18] <burek> do you see -r on that page [12:18] <burek> -r[:stream_specifier] fps (input/output,per-stream) [12:18] <burek> Set frame rate (Hz value, fraction or abbreviation). [12:18] <burek> etc [12:19] <HerbertPumpkin> Yes we've worked out the rate thing, burek [12:19] <HerbertPumpkin> I'm trying to figure out cropping [12:19] <burek> you can read that same document [12:19] <burek> for any option you need [12:19] <HerbertPumpkin> Personally I have no idea what "`r[:stream_specifier]" means [12:20] <burek> nobody does at the first read [12:20] <burek> that's normal [12:20] <burek> for such a complex and useful tool like ffmpeg is, it is normal that the documentation will be somehow harder to understand if you are new to all that [12:20] <HerbertPumpkin> I'm trying to figure out how I'd go about cropping it down. [12:20] <burek> HerbertPumpkin, read that document and you'll find out [12:21] <burek> for example, try CTRL+F 'crop' [12:21] <HerbertPumpkin> OK, that's very nice, but that document doesn't seem to contain any examples and would appear to be a reference for software engineers, not users. [12:21] <burek> would you just RTFM? [12:21] <HerbertPumpkin> I found http://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide but it doesn't include much more information [12:22] <HerbertPumpkin> it's far from obvious what "crop=width:height:x:y" means [13:10] <brocatz> hey guys, i just compiled ffmpeg with the mingw package so that i could apply a patch and everything works fine, but now when i try to use avisynth it fails, it says permission denied, but i think that error is a side effect of another error, my other version of ffmpeg from 2010 (when i last used avisynth with it) works fine [13:10] <brocatz> wondering if there are any obvious things i should check [13:12] <Mavrik> hmm [13:12] <Mavrik> brocatz: haven't seen that error on my mingw builds yet [13:13] <Mavrik> maybe the binary can't find .dll files [13:13] <brocatz> well actually [13:13] <brocatz> i got another prebuilt ffmpeg [13:13] <brocatz> and it wont run it either [13:13] <Mavrik> brocatz: try running this: http://technet.microsoft.com/en-us/sysinternals/bb896645 [13:13] <Mavrik> filtering on the ffmpeg.exe [13:13] <brocatz> [avs @ 00000000016cf320] AVIFileOpen failed with error -2147221164 [13:13] <brocatz> C:/temp/output.mp4.1.avs: Operation not permitted [13:13] <Mavrik> and see which file it's trying to open [13:14] <brocatz> oh yep i have that already [13:14] <brocatz> ok [13:15] <Mavrik> usually those errors can come from different sources :) [13:15] <brocatz> yep [13:15] <brocatz> that's what i figured [13:16] <brocatz> i was wondering if my ffmpeg binary is inadvertently 64bit [13:16] <Mavrik> you'd get a "binary not supported" error on start [13:16] <Mavrik> plus, you need to try really hard to build a 64bit binary with mingw -_- [13:17] <brocatz> binary not supported from avs? [13:17] <brocatz> i guess avs is already sending a meaningful error [13:17] <brocatz> so it's been entered [13:17] <Mavrik> oh, sorry, forgot you're running through avs [13:21] <brocatz> how do i track what it tries to open [13:21] <brocatz> it closes too fast [13:32] <murali> i'm trying to install ffmpeg with x264 support in my linux fedora 14 machine. but couldn't locate the libx264.so file even though it's installed in my /usr/lib directory. please help. [13:33] <brocatz> murali: are they both 32bit? [13:33] <murali> yes. my machine by itself is 32-bit. [13:34] <murali> [root@SS59-MAA ffmpeg-0.11.1]# ./configure --enable-gpl --enable-libmp3lame --enable-libx264 --enable-shared ERROR: libx264 not found [13:34] <murali> [root@SS59-MAA ffmpeg-0.11.1]# ls /usr/lib/libx264.so* /usr/lib/libx264.so /usr/lib/libx264.so.102 [13:35] <sacarasc> murali: What does the config.log say? [13:36] <murali> gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -std=c99 -fomit-frame-pointer -pthread -E -o /tmp/ffconf.gChSmshI.o /tmp/ffconf.vlAAMhUb.c /tmp/ffconf.vlAAMhUb.c:1:18: fatal error: x264.h: No such file or directory [13:36] <murali> compilation terminated. [13:36] <murali> ERROR: libx264 not found [13:36] <microchip_> murali: install libx264-dev or similar [13:36] <brocatz> yeah [13:36] <sacarasc> You might have to upgrade x264, too... [13:37] <brocatz> it looks pretty planely like it wants x264.h which will be in the dev package [13:37] <juanmabc> if you manually placed the /usr/lib, the problem is /usr/inclucde [13:37] <murali> ok. i'm installing now. gimme 5 mins. [13:37] <juanmabc> do not put lib there [13:37] <juanmabc> its for headers [13:38] <juanmabc> ah, -dev, ok [13:39] <murali> it worked i guess, but some version conflict . it requires ERROR: libx264 version must be >= 0.118 [13:40] <murali> i'll compile it from source. [13:40] <sacarasc> Uninstall the package one first. [13:41] <murali> ok. [13:53] <burek> murali, type ldconfig [13:56] <murali> i compiled it from source. but the x264.h like similar header files are not copied to /usr/include [13:56] <murali> so i copied some 4 .h files to /usr/include. [13:56] <burek> /usr/local/include [13:56] <burek> don't copy anything manually man [13:56] <murali> the libx264 enabled is yes after configure script is run. [13:56] <burek> compile x264 from source and after that type ldconfig [13:57] <burek> then configure your ffmpeg [13:57] <murali> but it shows a compilation error. [13:57] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=25&t=38 [13:57] <burek> just skip the libaacplus part [13:58] <burek> i.e. configure with: ./configure --enable-shared --enable-gpl --enable-libmp3lame --enable-libx264 [13:59] <murali> it's already skipped. [14:00] <burek> did you compile x264? [14:00] <murali> yeah. i compiled. make and make install. [14:00] <burek> and ldconfig [14:00] <murali> but make install didn't work as expected. [14:01] <murali> ./configure --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-libmp3lame --enable-libaacplus --enable-libx264 [14:01] <murali> [root@SS59-MAA x264-snapshot-20120624-2245]# make install install -d /usr/local/bin install x264 /usr/local/bin [14:02] <juanmabc> the x264 you build, uses/should use ./configure ... --prefix=/usr, or ffmpeg would need --with-libx264=/usr/local or whatever (ffmpeg part not checked ./configure --help | grep -i x264) [14:02] <murali> sorry, ingore my previous paste. [14:03] <murali> i've changed my configure script input as suggested by burek and it's compiling. [14:03] <sanderj> Can ffmpeg handle live video? [14:03] <burek> ok [14:03] <burek> sanderj, yes [14:04] <sanderj> burek, do you got an example of it? [14:04] <burek> sanderj, it's too wide topic [14:04] <murali> no use. it's same. libavcodec/libavcodec.so: undefined reference to `x264_bit_depth' libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_125' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 [14:04] <burek> can you give some more specifics on what's your live video [14:04] <burek> murali, type dpkg -l | grep x264 [14:04] <burek> or yum [14:05] <sanderj> burek, to make a web video confreance system? [14:05] <sanderj> with multiple people.. [14:05] <burek> well I guess it's possible [14:05] <burek> but I never tried it in such configuration [14:05] <burek> I always used sip server for that [14:06] <sanderj> But how do I recive a stream as input, and sends it as output? [14:06] <sanderj> in ffmpeg. [14:06] <burek> ffmpeg -i input -vcodec copy -acodec copy output [14:09] <sanderj> And then I have to make a wrapper which recives it on a port, and sends it to stdin? [14:09] <burek> no [14:09] <burek> ffmpeg -i http://blabla:port/stream.flv -vcodec copy -acodec copy udp://blabla:port/ [14:11] <murali> burek, it's a fedora. it's yum. i'll try anyway? but tell me what should i find? [14:11] <burek> x264-dev [14:11] <burek> and x264 [14:11] <burek> and uninstall those [14:11] <sanderj> burek, How do I get a webbrowser to send data to ffmpeg? [14:11] <burek> because you are compiling the latest anyway [14:11] <burek> sanderj, why on earth would you do that? [14:12] <sanderj> Because I want to capture it from the web camera. and send it to ffmpeg. [14:12] <burek> where does webbrowser appear in that scenario? [14:13] <murali> burek, will this installer script help me? [14:13] <murali> or atleast, will it help sanderj. he he.. :-) [14:13] <burek> murali, what installer script? [14:14] <sanderj> burek, Because it will be an web video conferance app.. Just like google+ hangout. [14:14] <murali> http://www.ffmpeginstaller.com/ [14:14] <burek> murali, no need for that [14:14] <burek> just compile things regularly [14:14] <sanderj> burek, I don't see how ffmpeg gets the input from another machine. [14:15] <burek> sanderj, you need a webbrowser plugin that will capture the video, encode it and send it over the web to your ffmpeg [14:15] <murali> ok. fine. i'll do that compilation by myself. [14:15] <burek> at least that is how googla talk does it [14:15] <burek> sanderj, ffmpeg -i http://anothermachine:port/bla.asf [14:16] <burek> or ffmpeg -i udp://localhost:port [14:16] <burek> if you send udp traffic to ffmpeg [14:16] <sanderj> burek, but with that ffmpeg -i command.. it require that the webbrowser plugin have an open port I belive..? [14:17] <sanderj> I want ffmpeg to listen to a port for incoming video. [14:19] <burek> ffmpeg -i udp://localhost:incomingport [14:19] <sanderj> Cool. [14:21] <sanderj> burek, do you know about any such video webbrowser plugin? [14:22] <sanderj> Wondring how google hangout makes it without a plugin. [14:22] <burek> it uses plugin [14:25] <murali> burek, i did a ./configure --enable-satic --enable-shared --system-libx264 and all the header files are copied into their apropriate directories i guess. now i'm compiling ffmpeg again with ./congfiure options you suggested. [14:25] <murali> but still, the same error libavcodec/libavcodec.so: undefined reference to `x264_bit_depth' libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_125' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 [14:26] <sacarasc> murali: You didn't uninstall the old x264, did you? [14:26] <burek> why --system-libx264 ? [14:26] <burek> I thought the post on that forum was really straight forward [14:26] <murali> i was just trying my luck. sorry. [14:27] <murali> ok. will read it again. and get back to u . [14:27] <burek> uninstall all x264* you have [14:27] <burek> in your package manager (google how to do that properly on your os) [14:27] <burek> and then compile libx264 and then ffmpeg [14:27] <burek> it's a 4-5 minute of work [14:28] <xero-exez> Hi All, First things first "thnxz for ffmpeg" [14:28] <AlRazi> ffmpeg aac+ output file duration is 789:57:13 on itunes [14:29] <xero-exez> I can not find the answer to the following question: [14:30] <sanderj> burek, do I have to have a certain version of ffmpeg to make streaming possible? [14:30] <xero-exez> "Why is (using H264) only the baseline profile suported/accepted as a parameter for -vprofile" [14:31] <burek> sanderj, I don't know, usually I use the latest :) [14:31] <AlRazi> burek, ffmpeg aac+ output file duration is 789:57:13 on itunes [14:32] <burek> xero-exez, where did you read that? [14:32] <burek> AlRazi, that's cool :) [14:32] <AlRazi> lool [14:32] <AlRazi> yeah but when trying to stream it from our web app, it downloads the whole file first then start playing it [14:33] <AlRazi> the duration tag doesn't add up to the player i reckon [14:33] <xero-exez> @burek I did not read that, ffmpeg doesnt accept it [14:33] <burek> xero-exez, why would you use -vprofile? [14:33] <burek> xero-exez, can you please use pastebin.com, to show your command line and its output? [14:33] <sacarasc> AlRazi: To get around that, run `qt-faststart` on the MP4 file. [14:34] <burek> either that, or if your app is ffmpeg, tell it to use -flags +global_header [14:34] <xero-exez> @burek since -profile isn't compatable any more...thanks I will paste just a minute [14:34] <burek> (for AlRazi) [14:34] <burek> xero-exez, most probably you are using an old ffmpeg.. you should stick to native x264's -profile (mapped option in ffmpeg) [14:35] <AlRazi> if my app is ffmpeg ? excuse my ignorance here .. what happens is that when the user uploads an mp3 a background job initiates an ffmpeg process for conversion and then upload the resulting file to an S3 bucket [14:35] <AlRazi> do i add -flags +global_header to the ffmpeg process command ? [14:36] <xero-exez> ffmpeg version git-2012-05-23-7a0d00d built on May 23 2012 16:47:13 with gcc 4.6.1 Thank you for the hint I will check it out [14:36] <xero-exez> @burek [14:37] <burek> xero-exez, can you please use pastebin.com, to show your command line and its output? [14:38] <burek> AlRazi can you show your ffmpeg encoding command [14:38] <burek> use pastebin [14:38] <AlRazi> http://pastebin.com/YutdQ98d [14:38] <xero-exez> @burek I'm on it. Just a sec...and thanks [14:39] <AlRazi> and here is the configurations [14:39] <AlRazi> http://pastebin.com/E1hUjnMD [14:40] <AlRazi> the m4apath variable ends with (.m4a) should I try it with (.acc) ? [14:41] <sanderj> burek, This one got an "Unable to find a suitable output format for 'udp://localhost:4444'": ffmpeg -i udp://localhost:4444 -vcodec copy -acodec copy test-output.flv& ffmpeg -i test-input.flv -vcodec copy -acodec copy udp://localhost:4444 [14:43] <sanderj> The format is Video: flv, yuv420p Audio: mp3, 44100 Hz [14:43] <sanderj> Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 29.92 (359/12) [14:43] <sanderj> HMM.. [14:44] <burek> oh I see.. [14:44] <burek> AlRazi, now, you would like to stream that mp4, right? [14:44] <AlRazi> yes sir [14:44] <burek> well, you'll either have to use qt-faststart as sacarasc suggested [14:44] <burek> or don't save as mp4, but instead use .aac or .flv [14:45] <AlRazi> i already download the python version [14:45] <AlRazi> hmm let me try changing to aac quick and i'll let you know [14:45] <burek> sanderj, can you please use pastebin.com, to show your command line and its output? [14:45] <AlRazi> but if you don't mind explaining why would .aac solve it ? [14:46] <burek> different format/container [14:46] <burek> that is more suitable for streaming [14:46] <burek> instead of mp4 which is more suitable as a file storage [14:46] <burek> (keeps some important info at the end of file, that's why you need to download a whole file before playing it) [14:46] <AlRazi> but what i use is m4a not mp4 [14:46] <AlRazi> if that makes any difference [14:46] <xero-exez> @burek You helped me out ;) Had a question earlier about profiles (baseline,main, high)Thnxz [14:46] <burek> qt-faststart fixes that by moving that data to the front of the file [14:46] <AlRazi> oh [14:47] <burek> xero-exez, :beer: :) [14:49] Action: xero-exez is handing over a :beer: to burek [14:51] <sanderj> burek, http://pastebin.com/HymxXXZt [14:52] <AlRazi> burek, aac requires flash player as opposed to my html5 player, and qtfaststart output is still giving those false readings on itunes : / [14:52] <burek> html5 player also requires things (like installed codecs on your OS) so it's pretty much the same trouble both ways [14:53] <burek> oh iOS :) [14:53] <burek> good luck with that :) [14:54] <burek> sanderj, your ffmpeg is ancient [14:54] <burek> you really need to update it [14:54] <Nedwada> 'm having a problem, using ffmpeg version 0.10.2.git. If i open an m3u8, and this is redirected inside its stream, i get a segfault error | http://privatepaste.com/2374deb5f0 | [14:55] <Nedwada> can someone help? [14:55] <burek> Nedwada, this is probably a bug, can you report it please: http://ffmpeg.org/bugreports.html [14:56] <Nedwada> i'm not very good at those things [14:56] <Nedwada> but i will try [14:57] <burek> thanks :) [15:00] <burek> just testing :) [15:02] <AlRazi> is the qtfaststart python version accredited ? [15:02] <AlRazi> or should i use the c library ? [15:03] <burek> ask google? [15:03] <AlRazi> ok thanks [15:08] <swkide> Greetings! Can someone please point me to documentation how to rewrite a program, which still uses AVFormatParameters? [15:09] <HerbertPumpkin> fsvo "documentation" [15:10] <swkide> Ok I did dig this source but without success - I wil try again - thnks [15:11] <swkide> Ok I did dig this source but without success - I will try again - thanks [15:11] <HerbertPumpkin> Someone should write a basic user's guide to ffmpeg, but it would be a bitch to keep up to date, and you'd never get the developers to endorse it. [15:13] <saste> swkide: check doc/examples [15:14] <saste> in case something is outdated you are welcome to send a patch! [15:14] <burek> HerbertPumpkin, yes, VLC has solved that pretty simple [15:15] <burek> they installed wiki software and allowed simple users to edit the docs and contribute in that way [15:15] <burek> not everyone knows how to make a patch :) [15:16] <swkide> @saste will do thanks [15:17] <HerbertPumpkin> burek: I was thinking from a "conversion" point of view, not just a viewing thing. [15:17] <HerbertPumpkin> I use VLC for viewing, but I often need to convert [15:19] <burek> HerbertPumpkin, well I started this: http://ffmpeg.gusari.org/viewforum.php?f=25 [15:19] <burek> based on issues from this channel [15:19] <HerbertPumpkin> Really it needs to be at ffmpeg.org [15:19] <burek> I agree, but there is always something that stands in the way :) [15:19] <burek> so it's best to leave things as they are now :) [15:20] <HerbertPumpkin> I mean, if you go there and hit the "general documentation" button, the very first thing you get is "ffmpeg can be hooked up with a number of external libraries... by passing appropriate flags to ./configire" [15:20] <HerbertPumpkin> that's not general documentation [15:20] <HerbertPumpkin> that's engineering information [15:20] <burek> :D [15:21] <burek> I think I know what answer you'll get most probably :) patch welcome :) [15:21] <HerbertPumpkin> I'd be perfectly happy to write it [15:21] <HerbertPumpkin> but they'd never want to endorse documentation [15:21] <burek> what do you mean [15:22] <HerbertPumpkin> I've found ffmpeg developers to be, well, openly hostile to non-coders. [15:22] <burek> it's a professional deformation I guess :) we all don't have patience sometimes, especially when we are always crowded with a ton of work.. [15:23] <HerbertPumpkin> You're not the only ones. [15:23] <ubitux> ? [15:24] <ubitux> HerbertPumpkin: i agree the documentation is not perfect, but we are working on it quite often [15:24] <HerbertPumpkin> I suspect the real problem is that ffmpeg changes its behaviour every three hours so keeping documentation up to date would be quite the task. [15:24] <burek> HerbertPumpkin, do you have any suggestion how to overcome it? [15:24] <AlRazi> okay this is strange .. outputting m4a directly won't play, but outputting wav, and then using faac to convert to m4a, it's playable [15:24] <ubitux> HerbertPumpkin: if you want to add some examples, feel free to add some in http://ffmpeg.org/ffmpeg.html#Examples-2 [15:24] <saste> HerbertPumpkin: i agree "general documentation" for that is a retarded name [15:25] <burek> AlRazi, maybe your ffmpeg is outdated? [15:25] <ubitux> saste: documentation page need to be reworked :( [15:25] <ubitux> the "index" is not really good (http://ffmpeg.org/documentation.html) [15:25] <AlRazi> Sun May 20 15:20:47 2012 [15:25] <AlRazi> my latest ffmpeg commit [15:26] <burek> AlRazi, then it might be a bug.. can you please report it? [15:26] <HerbertPumpkin> I can't even find where the basics are discussed - "ffmpeg -i [input file] [options] [output file]" [15:26] <burek> HerbertPumpkin, at ffmpeg docs [15:26] <burek> "FFmpeg Documentation" [15:27] <ubitux> http://ffmpeg.org/ffmpeg.html#Synopsis [15:27] <ubitux> first section [15:27] <HerbertPumpkin> Yes, but the problem is not that it isn't there (it IS there), it's just too hard to find. [15:28] <ubitux> yeah, the ffmpeg.org/documentation page need to be re-organized [15:28] <saste> the first part of ffmpeg.texi is really generic, not ffmpeg-specific [15:28] <burek> while we are at it, would it be better to move "General Documentation" section (with those 7 links) below the "Command Line Interface (CLI) and Related Usage Documentation" [15:28] <burek> because, after all, ffmpeg is the main thing in this project [15:28] <saste> that could be moved to a generic file [15:28] <HerbertPumpkin> I mean, the "synopsis" section is a little, er, bald. [15:28] <ubitux> i'd move the CLI documentation links above the general one [15:28] <burek> and docs for ffmpeg could be the first link in docs [15:29] <saste> burek: it's confusing and wrong to pretend that "ffmpeg" the tool is somehow special with the regards to the other tools [15:29] <ubitux> and rename "General documentation" into "Supported stuff" or sth like that [15:29] <burek> or just "other" :) [15:30] <saste> unfortunately many devs have this feeling that ffmpeg/avconv is special and implement all sort of ad-hoc hacks into it [15:30] <ubitux> :) [15:30] <HerbertPumpkin> Personally I have been extremely unimpressed with the way ffmpeg developers engage with non-developers. [15:30] <HerbertPumpkin> Chip on shoulder, etc. [15:30] <burek> saste, I do think it's special, otherwise an entire domain wouldn't be named ffmpeg.org :) [15:30] <HerbertPumpkin> This may explain why the documentation is so dire. [15:31] <burek> at least that's the impression I get, as an ordinary user [15:31] <saste> burek: i mean "ffmpeg" the tool and FFmpeg are two distinct thing [15:31] <saste> ffmpeg is just a facet of a more generic entity [15:31] <burek> I just say it would be logical (to me) that ffmpeg docs are the first link at docs link, that's all [15:31] <burek> or CLI section, for that matter [15:31] <ubitux> 15:28:43 <@ubitux> i'd move the CLI documentation links above the general one [15:31] <ubitux> 15:29:02 <@ubitux> and rename "General documentation" into "Supported stuff" or sth like that [15:32] <saste> indeed i hope it will be possible to transcode with ffplay or ffprobe some (not very remote) day [15:32] <ubitux> anyone wants to send a patch for this? :) [15:32] <burek> yes exactly ubitux :) [15:32] <saste> or in other words every tools just emphasize a specific aspect, but functionality should reside in the library [15:32] <saste> (and application should tend to be as lean as possible) [15:33] <burek> saste, that's not the issue here :) [15:33] <burek> HerbertPumpkin, has got a point :) the first group of links in docs is somehow wrongly positioned at the top [15:33] <burek> when in reality most people are really interested in CLI section of links [15:34] <HerbertPumpkin> YEs. [15:35] <burek> for the reference, I'm talking about this page: http://www.ffmpeg.org/documentation.html [15:35] <saste> ubitux: all for the supported stuff, it would be cool to write a script which parses the souce and detects which formats are supported [15:35] <saste> that page naturally tend to be outdated and require continuous babysitting [15:35] <saste> well actually some people love to do that kind of work... [15:36] <ubitux> :) [15:37] <ubitux> saste: i would actually print the support and all the options on the same page [15:37] <ubitux> ideally with some nasty js to show/hide them [15:38] <ubitux> but i don't want to do that [15:38] <ubitux> and i don't want to become the website maintainer :) [15:39] <HerbertPumpkin> Well, I think you're right [15:39] <HerbertPumpkin> the amount of work it would take to keep a comprehensive ffmpeg user guide up to date is enormous. [15:39] <HerbertPumpkin> That being the case, perhaps it would be nice if every answer in this channel wasn't "RTFM", eh? [15:39] <burek> :) [15:40] <saste> HerbertPumpkin: they are called links to documentation, and apparently many users don't know that there is a manual apparently [15:40] <HerbertPumpkin> If you can't find it, there effectively isn't. [15:40] <ubitux> HerbertPumpkin: it looks like you are willing to improve things, that's great! Will you join us and contribute by helping the users? :) [15:40] <saste> i wrote much part of it (e.g. filters documentation) and i don't feel like i want to rewrite it (again) on the ML [15:40] <HerbertPumpkin> No. ffmpeg developers tend to be asshats, and I'm not willing to work with those people. [15:41] <ubitux> HerbertPumpkin: well, that's not that hard, clic documentation, and then ffmpeg [15:41] <ubitux> HerbertPumpkin: ok :( [15:41] <HerbertPumpkin> Sorry, but they brought it on themselves. [15:41] <saste> HerbertPumpkin: give it a try, otherwise your claims are injustified (and unfair) [15:41] Action: HerbertPumpkin enjoys a slice of pie [15:42] <ubitux> i think a bad or incomplete documentation will just bring more user complains [15:42] <HerbertPumpkin> So do I. [15:42] <ubitux> i think we should drop completely the documentation [15:42] <burek> btw, regarding the documentation, why there is no interest in creating a wiki docs? [15:42] <burek> it's much easier to maintain it [15:42] <ubitux> because it would split the work [15:43] <ubitux> the wiki is nice for examples though [15:43] <saste> burek: we have it on trac [15:43] <HerbertPumpkin> Most of the time when I google I tend to find that I get information from forum posts and the like, not ffmpeg.org [15:43] <saste> but we need: 1. to publicize it better (no links from the website??) [15:43] <burek> saste, ok, but why don't you inforce it for docs? why .texi and all that stuff? [15:43] <saste> 2. to have more people working on it [15:43] <burek> ubitux, how would it split the work? [15:44] <ubitux> burek: we like file-versionned documentation [15:44] <saste> burek: problem is that code and documentation are not different things [15:44] <HerbertPumpkin> Mainly you would need to get the developers to at least mark parts dirty where they've made changes [15:44] <saste> when i extend a component i update the documentation as well [15:44] <HerbertPumpkin> and you'll never get them to do that [15:44] <burek> ubitux, wiki is all about versioning [15:44] <ubitux> it's not file oriented [15:44] <saste> and the documentation is specific of the version that you're using [15:44] <burek> at least mediawiki, I don't know about ffmpeg's trac wiki [15:44] <ubitux> burek: and you can't duplicate it easily [15:44] <burek> its not wikimeda [15:44] <burek> mediawiki* [15:44] <ubitux> burek: i like working offline with the doc [15:45] <ubitux> and know there are some backups everywhere [15:45] <saste> when we ship ffmpeg version x.xx we want to provide bundled documentation, rather than a link to a website [15:45] <ubitux> also, by having them in the git, we can follow the changes easily [15:45] <HerbertPumpkin> ffmpeg doesn't ship versions. [15:45] <ubitux> (mail notice, git log, etc.) [15:45] <HerbertPumpkin> you can only claim to be "shipping versions" if you're releasing binaries, which you aren't. [15:45] <ubitux> HerbertPumpkin: http://ffmpeg.org/download.html#releases [15:45] <saste> HerbertPumpkin: that's another prejudice, ffmpeg has versions since 3 years at least [15:45] <HerbertPumpkin> That's source code. [15:45] <ubitux> HerbertPumpkin: look for "builds" on the same page [15:46] <HerbertPumpkin> The behaviour of that source code can change wildly depending on how it's compiled. [15:46] <burek> can't follow all of your arguments, but "ubitux> burek: and you can't duplicate it easily" -> yes you can [15:46] <burek> ubitux, also offline work is possible too, just update the online version when you are done [15:46] <ubitux> i can't duplicate the whole wiki content and history [15:46] <ubitux> easily. [15:46] <saste> burek: what is makes sense, is keeping "official" and unofficial documentation split [15:46] <ubitux> also, i like the plain text doc :) [15:46] <burek> also, it is not difficult to make wiki for x.x version of ffmpeg [15:47] <burek> and update it accordingly [15:47] <saste> that is official documentation - shipped with ffmpeg, and wiki for user-guides [15:47] <burek> (several wikis) [15:47] <ubitux> really, feel free to update the trac burek :) [15:48] <burek> ubitux, well trac is not mediawiki [15:48] <ubitux> so what? :) [15:48] <burek> and doesn't have all the options to ease the work [15:48] <burek> so it takes more time to work on it [15:48] <ubitux> saste: didn't you say you would add an example to https://ffmpeg.org/trac/ffmpeg/wiki/FancyFilteringExamples ? [15:48] <burek> compare the features and you'll see [15:49] <saste> ubitux: no you did (the mirror effect) [15:49] <burek> the point of wiki is that a lot of people can contribute on it, fixing typos, adding additional usage examples, etc [15:49] <saste> and i have a ton of todo with highest priority right now [15:49] <saste> like chatting on irc [15:49] <ubitux> burek: we already have 2 official incomplete sources of documentation, i don't think adding a third will solve anything [15:49] <ubitux> saste: oh this one, ok [15:49] <burek> ubitux, not the 3rd, but the combined and one [15:50] <HerbertPumpkin> I'm with burek, for what it's worth [15:50] <HerbertPumpkin> not that anyone in charge of ffmpeg will ever take any notice [15:50] <ubitux> that's the problem [15:50] <HerbertPumpkin> this conversation is a waste of time; no ffmpeg developer will ever agree to a big change that wasn't his idea [15:50] <HerbertPumpkin> especially if it came from a non-dev [15:51] <burek> I'm just saying it from the perspective of an ffmpeg user (not developer) that it's a lot easier to contribute in this way then to learn all the git/patch/texi stuff and practically loose time learning it, which would push me away from wanting to help at all [15:51] <ubitux> i don't get what's complicated and so painful with the trac wiki [15:51] <HerbertPumpkin> burek: I assumed that's the idea. [15:51] <HerbertPumpkin> The last thing developers want is non-developers getting involved. [15:51] <HerbertPumpkin> They might have to listen to them! [15:52] <burek> ubitux, I think I already answered that.. check diffs for features of mediawiki and trac wiki [15:52] <ubitux> also, despite the "sync" issue, it will be yet another source of documentation, so users will likely never know where to look for [15:52] <HerbertPumpkin> ubitux, the thing to understand is that there really isn't any documentation as it is. [15:52] <HerbertPumpkin> Other than google. [15:52] <ubitux> burek: you have to show me it's an issue, because i don't believe so; so you need to raise some specific issues :p [15:53] <HerbertPumpkin> What's on ffmpeg.org is outdated, fragmentary, disorganised, and hard to find. [15:53] <burek> ubitux, again.. I don't know why are you bringing same issues again, when I argued them previously.. it won't be "yet another source of docs" it will be one and only [15:53] <burek> like in VLC [15:53] <ubitux> the page referencing the documentation isn't perfect and need just some small rework [15:53] <burek> they joined all the docs and put them to wiki [15:53] <burek> and look how big the docs are and still well organized [15:53] <ubitux> the overall documentation is heterogeneous because we focused on the documentation only recently [15:53] <burek> why not learn from others success? [15:54] <HerbertPumpkin> There's a focus on the documentation?! Well, you can't tell! [15:54] <ubitux> HerbertPumpkin: http://git.videolan.org/?p=ffmpeg.git;a=history;f=doc;h=3d84ebf5add3bac99c3… [15:54] <saste> HerbertPumpkin: i consider your attitude a bit arrogant, considering the amount of my *free time* i dedicated to it [15:54] <ubitux> all of this affect what's in doc/ directory [15:55] <HerbertPumpkin> saste: I'm being blunt, I admit. [15:55] <HerbertPumpkin> But frankly this isn't really about points for effort. [15:55] <HerbertPumpkin> A lot of effort in open source is wasted for this reason. [15:55] <ubitux> are you helping in any way? [15:55] <saste> HerbertPumpkin: it's not like we got a company, money or whatever else backing us [15:55] <HerbertPumpkin> It doesn't matter how good ffmpeg is; people will still use Compressor or the inbuilt exporter in their NLE because they can't figure out how to use ffmpeg. [15:55] <HerbertPumpkin> At which point you have to ask why ffmpeg exists. [15:56] <HerbertPumpkin> If it exists to be useful to people, it needs good documentation. [15:56] <HerbertPumpkin> If it's just a science project, an experiment, a toy... well, do as you wish, but I don't think that's what you want. [15:56] <ubitux> burek: again, i'm personally against moving the documentation out the git source tree, but i'm ok splitting them in two: one from the dev, and one from/for the users (and the current trac) [15:56] <saste> HerbertPumpkin: it is useful on its own, and don't forget that it is the engine of most multimedia apps/services out there [15:56] <HerbertPumpkin> wtf is a "git source tree" [15:56] <saste> so at least someone is able to deal with it [15:56] <ubitux> s/and the current trac/the current trac/ [15:57] <HerbertPumpkin> is that something I need to know in order to understand documentation changes? [15:57] <ubitux> it's the history of the changes in the project [15:57] <ubitux> affecting source code, and documentation [15:57] <burek> ubitux, why the docs need to be in git anyway if mediawiki supports its own versioning of all the content there? [15:57] <ubitux> i already gave a few arguments for that [15:58] <HerbertPumpkin> I think the wiki idea is sensible. [15:58] <HerbertPumpkin> It's easy to contribute, and would spread the workload. [15:58] <ubitux> we have a wiki [15:58] <ubitux> and you can contribute [15:59] <burek> ok, all in all, the idea of wiki had one good thing in mind, to split the need of regular user to know advanced stuff, like git/patch/texi/etc [15:59] <burek> they just need to log in, use the cool html editor and contribute [15:59] <HerbertPumpkin> Yes. [15:59] <ubitux> we should add an url to the wiki in the documentation page btw. [15:59] <burek> even a person who is not that familiar with computers can contribute.. isn't that valuable? [15:59] <HerbertPumpkin> One of the ways you make people contribute is to make it easy for people to contribute [15:59] <ubitux> burek: https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide?action=history [16:00] <HerbertPumpkin> I would probably have tried to write patches for ffmpeg itself (though obviously they would be rejected) if it could be compiled in MSVCC. [16:00] <ubitux> various ppl contributed here [16:00] <HerbertPumpkin> They're making it very, very hard to contribute (intentionally, of course). [16:00] <ubitux> ok i think i'm wasting my time here [16:00] <ubitux> feel free to keep complaining [16:00] Action: ubitux & [16:01] <burek> ubitux, that's cool :) but, let me just point out one really bad thing of that trac.. at least the one that I was embarrassed to realize.. [16:01] <burek> when you go to the home page of it https://ffmpeg.org/trac/ffmpeg/ [16:01] <burek> where is the register button? [16:01] <burek> to create a new account? [16:02] <ubitux> login register? [16:02] <saste> ubitux, burek: i don't like that as well [16:02] <burek> I know, but come on.. 2 clicks? [16:02] <burek> why? [16:02] <ubitux> because it sucks [16:02] <burek> simple things that irritate people.. :/ [16:02] <saste> don't know if it has to deal with trac defaults [16:02] <HerbertPumpkin> What's "trac" [16:02] <burek> :) [16:03] <HerbertPumpkin> Why someone would use anything other than mediawiki for collaborative docs I have no idea. [16:03] <HerbertPumpkin> No, wait, I do have an idea - so they can keep out everyone except their little in-group :/ [16:04] <ubitux> because mediawiki doesn't handle issues [16:04] <HerbertPumpkin> Be more specific? [16:04] <ubitux> bug tracker [16:04] <burek> anyway, let's forget the trac/wiki/patches :) the thing is there is a problem for non-advanced users to help contribute to ffmpeg.. that was especially noticeable when Google Summer of Code was actual [16:04] <burek> we didn't have a way to give tasks to people because everything needed some basic knowledge just in order to start contributing [16:05] <ubitux> HerbertPumpkin: we want to limit the number of different services (load, consistency, security, ...) [16:05] <saste> burek: how is that different from "real world"? [16:05] <HerbertPumpkin> I would attempt to write code for ffmpeg. [16:05] <HerbertPumpkin> But I can't even compile it. [16:05] <burek> so I had to convert all those contribs for the docs into patches (had to learn how to do that of course) [16:05] <ubitux> HerbertPumpkin: and since we're using a bug tracker that does wiki, we use that feature [16:05] <burek> instead of just giving them link to the wiki and tell them "go there, edit what you find erroneous" [16:06] <ubitux> burek: then I might have a solution [16:06] <ubitux> go to the github mirror, and if you have an account, you might be able to directly edit from the web interface [16:06] <ubitux> and then request a merge etc [16:06] <saste> ubitux: program an AI will assist users with using ffmpeg, or contributing to it [16:06] <HerbertPumpkin> Just use bloody mediawiki [16:06] <HerbertPumpkin> what possible problem is there with it [16:07] <ubitux> 16:05:01 <@ubitux> HerbertPumpkin: we want to limit the number of different services (load, consistency, security, ...) [16:07] <HerbertPumpkin> not techy enough for you? too easy for your little linux heads? [16:07] <ubitux> 16:05:11 <@ubitux> HerbertPumpkin: and since we're using a bug tracker that does wiki, we use that feature [16:07] <burek> ubitux, and how exactly is that easier in comparison of "just giving them link to the wiki and tell them "go there, edit what you find erroneous"" [16:07] <ubitux> also, we already have some content in the wiki [16:07] <ubitux> burek: then just give them the link to the wiki [16:08] <ubitux> anyway, i think it's ok to report typo or error on the devel channel [16:08] <saste> HerbertPumpkin: "your little linux heads?" - that's offensive, are you aware of it? [16:08] <burek> ubitux, there is nothing in the wiki! [16:08] <ubitux> then fill it [16:08] <burek> because you split it [16:08] <ubitux> adding another wiki won't fill it [16:08] <burek> into texi and wiki.. [16:09] <ubitux> it's the third time i explained that git versionning for the documentation is IMHO way better [16:09] <saste> burek: i agree with ubitux, official and non-official docs should be kept separate [16:09] <HerbertPumpkin> Yes, ubitux, but you have to be an expert software engineer for that. [16:09] <saste> no need to keep them together, because they appeal to different kind of users [16:10] <saste> and... who does the work decide how to do it, especially when he's volunteering his/her own time [16:10] <saste> there are technical and social reasons for the choice of the wiki and for the use of texi in ffmpeg [16:10] <ubitux> main doc on git because it is decentralized (we have copies of it), doesn't require stupid web shit (plain text read & edit), it's versionned with tools we can make use of, we can track the changes, etc. [16:11] <ubitux> user doc on trac (use cases, etc) [16:11] <HerbertPumpkin> It may be versioned with tools YOU can make use of, but you're a linux-based software engineer. [16:11] <HerbertPumpkin> It's hopeless. [16:11] <ubitux> exactly [16:11] <ubitux> then just report typo issues by opening an issue in the trac [16:11] <ubitux> and complete the doc in the wiki [16:12] <HerbertPumpkin> what's a trac? [16:12] <ubitux> it's a facebook group [16:12] <ubitux> join it [16:12] <ubitux> and shut up forever [16:12] <ubitux> thx [16:12] <HerbertPumpkin> ..and that is why open source software is so poorly documented. [16:12] <burek> ubitux, http://www.mediawiki.org/wiki/Git/Workflow [16:13] <burek> it's not the best wiki software for nothing [16:13] <ubitux> burek: that's interesting [16:13] <ubitux> can you generate the manpages out of it? [16:14] <ubitux> will it be able to handle the generation & installation of them? [16:14] <ubitux> also, how do you control the users are not trashing the whole documentation? [16:14] <burek> ubitux, did you know that texi2html can also generate wiki code? [16:15] <HerbertPumpkin> Same was wikipedia does. [16:15] <HerbertPumpkin> IE it doesn't really happen. [16:15] <ubitux> it happens, they have moderators for this [16:15] <ubitux> and we have reviews. [16:15] <HerbertPumpkin> Although I suspect that in open source world, writing simple, understandable instructions that are useful to the layperson would be considered "trashing the whole documentation". [16:15] <burek> I'm just saying there is a need for better documentation.. One way of solving that problem is to open the way for non-advanced users who have free time and will to contribute the text for docs [16:15] <ubitux> burek: possible, so? [16:16] <burek> otherwise, because of all the restrictions and requirements present now, the developers are the only one who can do that [16:16] <burek> and we all know they lack free time always [16:16] <burek> so.. what's the point of our conversation then.. [16:16] <saste> burek: let's try to avoid to go in circles, we have already a wiki at zero cost, what we lack is people working on it [16:17] <HerbertPumpkin> burek, in open source world, the reaction to documentation improvement is generally "possible, so?" [16:17] <saste> one problem is that people don't know how to reach it [16:17] <HerbertPumpkin> M [16:17] <HerbertPumpkin> Nobody cares. [16:17] <HerbertPumpkin> Open source developers don't care. [16:17] <ubitux> yep [16:17] <ubitux> OTOH, microsoft let you edit msdn documentation [16:17] <burek> <saste> one problem is that people don't know how to reach it [16:17] <burek> true [16:17] <burek> but also nobody used that before [16:18] <burek> and it doesn't have a cool html editor, etc etc.. [16:18] <saste> so... either you post a patch, or you open a ticket, or you write a mail on the ML or you just tell it [16:18] <burek> that way you developers loose free time [16:18] <saste> if i do it myself, i'll never get rid of it, it is important that users which care do it themselves [16:18] <burek> and docs suffer.. all users suffer [16:19] <saste> or they will always rely on other people (especially people which has already a huge workload) [16:19] <burek> exactly my point [16:19] <burek> it doesn't have to be wiki [16:19] <burek> mediawiki* [16:19] <saste> that said i'll note it on my todo list, and i'll get to do some real work right now... [16:19] <ubitux> same for me... [16:19] <burek> it just needs to be something simple that other non-advanced users can easily use.. that's it [16:19] <HerbertPumpkin> I cant imagine why you wouldn't use mediawiki. [16:19] <saste> burek: the principle is that we don't have mediawiki, we have trac and is already up and working [16:20] <burek> saste, ok, let's use ti [16:20] <burek> it* [16:20] <saste> it may be not the best, but it's already there and imho good enough [16:20] <HerbertPumpkin> saste, the documentation is crap; whatever "trac" is, it is NOT working [16:20] <burek> no, really, let's use it [16:20] <ubitux> thank you. [16:20] <burek> if it suits both devels and ordinary users [16:20] <HerbertPumpkin> It might be technically doing what it's supposed to do, but it is NOT creating good docs, so it is not working. [16:20] <burek> then its perfect [16:20] <saste> if there is some interest and someone wanting to do the work, that's welcome (but again it needs some discussion, there is the risk of splitting resources and work, we can't afford that) [16:20] <ubitux> burek: i'll rework the documentation page tonight [16:20] <HerbertPumpkin> and you can afford bad docs? [16:21] <ubitux> burek: i'll add a trac URL, and re-organize a bit the links [16:21] <burek> cool thanks :) [16:21] Action: HerbertPumpkin bounces his head off the wall [16:21] <HerbertPumpkin> Just. Use. Mediawiki. It's free! [16:21] <ubitux> burek: if you have a minute, please open a trac issue to add a "register" link on the main page [16:21] <burek> one more thing that needs to be done is that all texi thing is either moved to wiki or wiki is linked to it or something [16:21] <burek> what ever devels write through texi files, ordinary users need to be able to edit through the wiki [16:21] <ubitux> no, texi files will stay on git [16:22] <ubitux> for the reasons i mentionned [16:22] <burek> otherwise we have a split work, which is bad [16:22] <HerbertPumpkin> "texi files?" [16:22] <ubitux> wiki should document different thinbgs [16:22] <ubitux> HerbertPumpkin: the files used to generate the documentation, they are on git [16:23] <HerbertPumpkin> *what* documentation!? [16:23] <burek> did we ever bother to take a look how did guys from videolan solve all their issues using mediawiki [16:23] <burek> because they have great docs [16:23] <burek> how did they do that? [16:23] <burek> why not just copy the idea? [16:23] <ubitux> HerbertPumpkin: http://git.videolan.org/?p=ffmpeg.git;a=tree;f=doc;hb=HEAD these files are used to generate pages like http://ffmpeg.org/ffmpeg.html [16:23] <burek> if it's good of course [16:23] <HerbertPumpkin> You're trying to tell me that you want to autogenerate a man page that describes the use of ffmpeg in detail?! [16:23] <HerbertPumpkin> That's crazy! [16:23] <saste> burek: different kind of users, different *number* of users [16:23] <ubitux> HerbertPumpkin: that's what we do... [16:23] <ubitux> HerbertPumpkin: these texi are used to generate the manpages we deploy [16:24] <ubitux> as well as the official documentation on the website [16:24] <ubitux> but we also have a user-editable wiki: https://ffmpeg.org/trac/ffmpeg [16:24] <HerbertPumpkin> bear in mind on windows I don't even see a man page. [16:24] <ubitux> which need to be completed. [16:24] <HerbertPumpkin> Oh fer chrissake [16:24] <HerbertPumpkin> Once again, user documentation for an open source project obsesses over merely compiling it. [16:24] <HerbertPumpkin> That's not documentation! Tell me how to USE it! [16:25] <ubitux> ? [16:25] <ubitux> wth are you talking about? [16:26] <HerbertPumpkin> Compiling software is not something users should have to do, and it isn't usually something they CAN do. [16:26] <burek> http://doc-book.sourceforge.net/homepage/ [16:26] <burek> DocBookWiki can also be used to edit a DocBook document online, from the web. Editing is done one section at a time, so the editor selects first the section that he wants to edit, and then edits it. He can edit it in several modes: text (like wiki), xml (the original format), html, latex, texi, etc. [16:26] <HerbertPumpkin> burek: No. Just use mediawiki. [16:26] <HerbertPumpkin> Everyone knows it, everyone's comfortable with it. [16:26] <HerbertPumpkin> Ain't broke, don't fix, [16:26] <burek> well HerbertPumpkin we need to respect the need of developers too.. [16:26] <burek> if we don't there won't be any anymore.. [16:26] <burek> :/ [16:27] <ubitux> the problem is not with mediawiki or any other service [16:27] <ubitux> it's what saste and i already stated several times [16:29] <ubitux> and btw if the documentation gets outside the repository, be assured that when adding new stuff, the documentation won't be updated by developers anymore [16:29] <HerbertPumpkin> Developers can go hang [16:29] <HerbertPumpkin> They write code once [16:29] <HerbertPumpkin> We use it a million million times [16:29] <sacarasc> Because developers never use their own code. :( [16:29] <burek> I don't know.. I understand the need for texinfo but this docbook wiki uses xml file to store the content and produces html, texi, latex, anything.. and what's most important it allows online editing of the content which bridges two worlds of devels and regular users [16:29] <HerbertPumpkin> Coder ease and comfort is massively less important than user ease and comfort. [16:30] <HerbertPumpkin> God almighty just USE MEDIAWIKI. [16:30] <HerbertPumpkin> What possible problem is there [16:30] <ubitux> install it, do your shit with it, and please stop complaining [16:30] <ubitux> we have more serious problem do deal with [16:30] <HerbertPumpkin> I don't think you do. [16:31] <HerbertPumpkin> I think you fail to understand how important good docs are. [16:31] <ubitux> now please stop being insultant or you won't stay long here [16:31] <burek> HerbertPumpkin, you can't have an attitude like that [16:31] <saste> ubitux: don't get your day spoiled ;-) [16:31] <burek> why is developer's comfort and ease less important [16:31] <burek> they are not payed for what they do [16:31] <HerbertPumpkin> Because they have what's called a "single task involvement" [16:31] <burek> if they were that would be completely different story [16:31] <HerbertPumpkin> Well, there's a mechanism for software engineers to get paid for what they do, it's called "commercial software" [16:31] <HerbertPumpkin> if they want to do that fine. [16:31] <HerbertPumpkin> If they don't, then this is the deal. [16:32] <burek> they do this because they like it and have free time to help contribute it.. if we make it so they dont like it anymore, there wont be any more devels and project will die [16:32] <HerbertPumpkin> So what? I can't make it do anything useful anyway, because the documentation is so bad. [16:32] <HerbertPumpkin> (slight exaggeration but you see what I mean) [16:32] <burek> no, my point is that you would talk differently if you were one of developers [16:32] <burek> if you would realize problems observed from their perspective [16:33] <burek> if we can't join/merge these 2 worlds, then we didn't help [16:33] <ubitux> saste: yeah, i'm going away :) [16:33] <HerbertPumpkin> I appreciate the issues [16:33] <HerbertPumpkin> I just think that writing good docs is part of being a software engineer. [16:34] <HerbertPumpkin> If the developers don't agree then I would say they are missing the point of what they're doing. [16:34] <brocatz> it's not literally [16:34] <brocatz> that's why there is a career in technical writing [16:34] <brocatz> the matter of documentation is an endless argument amongst programmers [16:34] <HerbertPumpkin> I know, I make quite a lot of money out of technical writing. [16:34] <HerbertPumpkin> :) [16:34] <burek> it's just too bad docs haven't been written in XML or some other more general format.. that way developers would be able to edit them in text editors and there are a lot of html editors able to edit XML content [16:34] <brocatz> a lot of purists believe there should be very little as it discourages writing readable code [16:34] <burek> that would be ideal IMHO [16:34] <brocatz> but you're coming at it from an end users perspective [16:35] <brocatz> and i would say a technical writer is more the kind of person you want [16:35] <HerbertPumpkin> There are a lot more end users than there are developers. [16:35] <brocatz> not a software engineer [16:35] <HerbertPumpkin> This means the end users' concerns are much, much more important. [16:35] <HerbertPumpkin> Sorry if that sucks. [16:35] <brocatz> the kind of documentation programmers write for each other would be of very little value to you [16:35] <HerbertPumpkin> But it's life. [16:35] <HerbertPumpkin> Oh, I know. [16:35] <HerbertPumpkin> I write code too. [16:36] <brocatz> so anyway, you want a technical writer [16:36] <HerbertPumpkin> But if that's the case, what's wrong with just having a wiki? [16:36] <brocatz> don't make out like that's a software engineers obligation [16:36] <HerbertPumpkin> You can't tell me that the docs on the current website are used by the engineers. [16:36] <brocatz> my problem with the recent documentation change is that they canged the selection color and now find is basically worthless in chrome [16:36] <brocatz> and for such a dense document that's unhelpful [16:37] <burek> brocatz, how? [16:37] <burek> it works for me [16:37] <brocatz> i didn't say it didn't work [16:37] <brocatz> i said the selection color changed [16:37] <brocatz> looks like it's changed again, or chrome was being a jackass, it's bright orange now [16:38] <brocatz> which is good [16:38] <burek> ? [16:38] <burek> i dont understand your issue.. [16:38] <HerbertPumpkin> If certain text colours are set wrongly, the find hilight becomes invisible. [16:39] <brocatz> ^ [16:40] <burek> select color is blue [16:40] <HerbertPumpkin> You could argue that's really Chrome's bug, but either way if it's easily fixable (and it is), it should be fixed. [16:40] <burek> find color is something yellowish orange something [16:46] <HerbertPumpkin> burek: it is possible to cause collisions, though. [16:46] <HerbertPumpkin> Possibly it shouldn't be, but it is, and web authors need to bear that in mind or you do gimp the browser rather. [16:48] <burek> well I don't see collisions, that's why I ask :) [16:48] <HerbertPumpkin> link me, I use chrome [16:48] <burek> link me? :) [16:49] <HerbertPumpkin> "provide me with a URL to the offending page" [16:49] <burek> oh, well I don't have an offending page :) ask brocatz :)) [16:53] <HerbertPumpkin> OK, must dash, dinner with a beautiful woman [17:04] <swkide> After this very interesting discussion I would like to ask where I could find information about deprecated functions in avformat. the program segmenter.c does compile with ffmpeg 0.8.3 but no longer with ffmpeg 0.11.3, because of not longer exsisting functions. Looks to me, taht you have changed the media handling (open/close) completely. I simply don't find a place to really open this... [17:04] <swkide> ...Gordian knot - which it is to me ... [17:07] <ubitux> < burek> it's just too bad docs haven't been written in XML or some other more general format.. // XML sucks and is not editable [17:07] <ubitux> i mean it's a pain. [17:08] <ubitux> texi is generic enough, though i have a preference for markdown and restructuredtext [17:09] <brocatz> yeah xml is a terrible markup format [17:17] <saste> swkide: but again i suggest to rely on the internal segmenter instead [17:27] <swkide> @saste I got the internal segmenter up and running. We have tons of flv movies, which should be transcoded to apple format. Segmenting works, but all audios are clipping . [17:28] <saste> swkide: i know audio segmentation is borken [17:28] <swkide> when transcoded and segmented from a flv format - other formats work without any problem [17:28] <swkide> ok thanks [17:28] <saste> try this one: http://gitorious.org/~saste/ffmpeg/sastes-ffmpeg/commits/misc-segment-fixes… [17:28] <saste> yes i'm aware i should integrate it, postponing since months... [17:30] <swkide> ok will try this [17:30] <swkide> thanks [17:49] <varaderoguy> hello all [17:49] <varaderoguy> I wanted to ask some advice out metadata in flv files [17:50] <varaderoguy> I am trying to add some metadata to an existing FLV file using ffmpeg 0.6.5 (Centos 5.7) and the metadata is not being written [17:51] <varaderoguy> command I typed was: ffmpeg -i PM_PL_11_12_NOR_STO-00.flv -map_meta_data 0:0 -vcodec copy -acodec copy -metadata canSeekToEnd="true" PM_PL_11_12_NOR_STO-00-fixed2.fl [17:51] <varaderoguy> unfortunely, 'canSeekToEnd' is not written to the file: [17:53] <varaderoguy> Output is shown at: http://fpaste.org/j8nq/ [17:55] <varaderoguy> However, I know it is NOT working, because when you use ffplay: [17:56] <varaderoguy> http://fpaste.org/OcVN/ [17:56] <varaderoguy> Thoughts Chaps???? [18:02] <varaderoguy> [coughs].... [18:04] <sacarasc> That last paste didn't seem to work. [18:06] <varaderoguy> arh - hello there sacarasc....let me repaste... [18:07] <varaderoguy> http://fpaste.org/KLbG/ [18:08] <varaderoguy> I notice that the DAG repos has an v.old version of FFMPEG - only 0.6.5....when you guys are running 0.11.1 [18:09] <varaderoguy> I'm on Centos 5.7 [18:09] <varaderoguy> I suppose I ~could~ try and roll my own.... [18:10] <varaderoguy> :-( [18:10] <sacarasc> To do that, you'd probably have to update a bunch of dependencies too. [18:10] <varaderoguy> joy.... [18:11] <varaderoguy> I'll see whether the static ffmpeg might work.... [18:11] <varaderoguy> maybe see whether I can get DAG to update his repos [18:12] <swkide> perhaps http://yamdi.sourceforge.net/ is an option? [18:14] <varaderoguy> okay - this might be a goer.... [18:15] <swkide> MIght throw some problems compiling in CentOS [18:15] <varaderoguy> Is it worth me placing a bugzilla report in for this...or am I going to laughed out of court? [18:17] <Mavrik> rolling your own is pretty much the only choice if you're doing anything that relies on ffmpeg [18:17] <Mavrik> the repo versions are just too old [18:18] <varaderoguy> okay - I wonder whether I have to do this myself.... [18:19] <varaderoguy> IF....and this is a BIG IF....I can get the distro to roll its own and get a buildfile to work; would you be interested? [18:19] <varaderoguy> I work in the media industry, so I suppose it is in my interest anyway [18:22] <sweb> http://superuser.com/questions/441361/ffmpeg-strip-all-metadata-from-all-fo… [18:23] <varaderoguy> arh - good news - Yamdi has produced better results that b4 [18:25] <varaderoguy> sweb: Sorry, but I'm not stripping data; I'm adding it..... [18:26] <varaderoguy> sweb: oddly enough - yamdi can strip all metadata [18:26] <varaderoguy> using its -M flag [18:27] <varaderoguy> right - gotta be off now....I think I'll be back here....you guys are so lovely! [18:27] <varaderoguy> Thanks for the pointers.... [18:27] <varaderoguy> Chou [19:18] <sweb> i have an error [19:18] <sweb> ffmpeg -i audio.mp3 -map_meta_data -1 -c:v copy -c:a copy out.mp3 [19:18] <sweb> Unrecognized option 'c:v' [19:19] <sweb> FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers [19:19] <JEEB> it's just old [19:19] <JEEB> -vcodec for older versions [19:19] <sacarasc> SVN hasn't been used with ffmpeg since January of last year. [19:19] <JEEB> yeah [19:19] <sweb> so i must move to github for latest version ? [19:20] <sweb> i think it's a mirror [19:20] <JEEB> http://git.videolan.org/?p=ffmpeg.git [19:29] <sweb> during compile ... ERROR: vfw32 not found [19:29] <sweb> what package required ? [19:29] <sweb> debian 64 .... 6.0.5 [19:29] <JEEB> don't try to enable everything, that's a windows only feature [19:31] <sweb> whitch --enable ? [19:31] <sweb> http://pastie.org/private/6f78yksmah8wk8y62depbg [19:32] <sweb> JEEB: ^ [19:33] <JEEB> enable what you're using [19:33] <sweb> so i want to know vfw32 ? [19:35] <sweb> find out [19:35] <sweb> --enable-avisynth [20:00] <sweb> make: *** [libavdevice/libcdio.o] Error 1 [20:14] <sweb> ffmpeg: error while loading shared libraries: libavdevice.so.54: cannot open shared object file: No such file or directory [20:14] <sweb> after compile [20:15] <durandal_1707> you will need to be more verbose to get any useful help [20:16] <alyawn> I'm attempting to run the muxing.c example changing the format to "mpegts" but I get: Requested output format 'mpegts' is not a suitable output format. Is there an additional step to output a ts? [20:18] <sweb> durandal_1707: this is my configure : http://pastie.org/private/aiier6dfcvw5smxq3tmxnw [20:18] <sweb> debian 64 ... 6.0.4 fully upgrated [20:19] <durandal_1707> sweb: perhaps you enabled stuff that fails to compile [20:20] <sweb> durandal_1707: which one ... during make i have no error [20:22] <beastd> sweb: Where do you get that error? Is it in the build directory or after installation? [20:22] <sweb> beastd: after make install in /usr/local/bin/ffmpeg [20:22] <sweb> after run ffmpeg [20:24] <beastd> sweb: Did you check that the libraries were created in the build directory? [20:24] <sweb> beastd: how can i check it ? after make ? [20:25] <sweb> in making source dir ? [20:25] <beastd> Just check if the files are present in the source dir after you successfully executed make. [20:30] <sweb> beastd: i get latest from 0.11 of github [20:34] <beastd> sweb: does "ls libavcodec/libavcodec.*" in the build directory list the library file? [20:36] <sweb> beastd: http://pastebin.com/0b0hgq0k [20:40] <beastd> sweb: Seems like the libraries were not build. So you should have some errors at "make" and probably at "make install" too. [20:42] <sweb> i will paste all operion during configure make and make install w8 [20:51] <sweb> beastd: http://pastebin.com/nrdUD95C [20:56] <ShinyObjects> Hey everyone. I'm comparing the data in my non-working h264 stream to a working one. I've noticed that the NALU header for the working one is 67 while mine is 27. [20:56] <ShinyObjects> That would mean that the working one has "3" for nal_ref_idc, while my stream has "1" [20:56] <ShinyObjects> I can't seem to find what on earth nal_ref_idc is, however. [20:56] <ShinyObjects> Does anyone know what nal_ref_idc is? [20:57] <ShinyObjects> And of course, after searching for about 20 minutes I finally find my answer after I break down and ask here. [20:57] <ShinyObjects> http://lists.mpegif.org/pipermail/mp4-tech/2006-June/006615.html [20:58] <ShinyObjects> Looks like the only difference between 1 and 3 for a nal_ref_idc is that 3 gets higher priority [20:59] <Mavrik> ^^ [20:59] <Mavrik> The Murphy's law of software development. [20:59] <ShinyObjects> Yep, totally. [21:50] <beastd> sweb: May it be that "ldd /usr/local/bin/ffmpeg" shows you that it finds none of the libav* libraries? If so you need to adjust your runtime linkers library search path. [21:51] <beastd> est [22:03] <beastd> sweb: May it be that "ldd /usr/local/bin/ffmpeg" shows you that it finds none of the libav* libraries? If so you need to adjust your runtime linkers library search path to also find stuff in /usr/local/lib. [22:40] <pisto> hello. I think that mp3 is one of the formats that supports raw concatenation. but unfortunately my mp3s have id3 tags. Is there a way to tell ffmpeg to strip them before concatenating? [23:10] <relaxed> pisto: look at -map_metadata and -metadata in the man page. [23:10] <burek> pisto, can you type ffmpeg -i a.mp3 [23:10] <burek> and use pastebin.com to show the output [00:00] --- Tue Jun 26 2012

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