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January 2012

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ffmpeg-devel.log.20111231
by burek 01 Jan '12

01 Jan '12

[00:18] <funman> Daemon404: when will we see a ffmpeg port for mmix ? [00:18] <Daemon404> people use mmix? [00:19] <Daemon404> actual hardware exists? [00:23] <funman> don't think so [01:28] <CIA-101> ffmpeg: 03Mike Melanson 07master * rae51b1a9ba 10ffmpeg/tests/ (fate/h264.mak ref/fate/h264-bsf-mp4toannexb): [01:28] <CIA-101> ffmpeg: FATE: Add test for H.264 MP4->annex.B bitstream filter. [01:28] <CIA-101> ffmpeg: This test uses an existing FATE sample (h264/interlaced_crop.mp4). [01:28] <CIA-101> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rf486fb338e 10ffmpeg/tests/ (4 files in 3 dirs): fate: drop pointless _audio and _video suffixes from xan tests [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rd4b63054d9 10ffmpeg/ (11 files in 3 dirs): cosmetics: Drop unnecessary parentheses around return values. [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rc7cb1730da 10ffmpeg/tests/ (Makefile fate.mak fate/demux.mak): fate: split off demuxer FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r51150cf070 10ffmpeg/tests/ (Makefile fate/voice.mak fate2.mak): fate: split off voice codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rab2d11f75b 10ffmpeg/tests/ (Makefile fate/audio.mak fate2.mak): fate: split off audio codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r394f0b18bf 10ffmpeg/tests/ (Makefile fate.mak fate/ea.mak): fate: split off Electronic Arts codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r9afc025bff 10ffmpeg/tests/fate/ (12 files): fate: whitespace cosmetics [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r628637f67b 10ffmpeg/tests/ (Makefile fate.mak fate/video.mak fate2.mak): fate: split off video codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r1a6a088f7c 10ffmpeg/: (log message trimmed) [01:28] <CIA-101> ffmpeg: Merge remote-tracking branch 'qatar/master' [01:28] <CIA-101> ffmpeg: * qatar/master: [01:28] <CIA-101> ffmpeg: fate: whitespace cosmetics [01:28] <CIA-101> ffmpeg: fate: split off video codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: fate: split off audio codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: fate: split off Electronic Arts codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rf6ba1f2641 10ffmpeg/tests/ (Makefile fate.mak fate/qt.mak fate2.mak): fate: split off QuickTime codec FATE tests into their own file [01:28] <CIA-101> ffmpeg: 03Diego Biurrun 07master * rcd8d84575b 10ffmpeg/tools/qt-faststart.c: qt-faststart: K&R reformatting; fix comment typos [01:55] <Matty> Is this right place to discuss, how ffmpeg puts ASS subtitles in Matroska files? [01:58] <ffmpeguser> Matty: this is the development channel, for ffmpeg usage questions go to this channel #ffmpeg [01:59] <ffmpeguser> developer* [02:01] <Matty> Ok, I think ffmpeg produces broken Matroska files containing subtitles that VLC of mplayer do not display. [02:02] <ffmpeguser> http://ffmpeg.org/trac/ffmpeg/ticket/858, my ticket was closed, what is the problem with the apache license? [02:02] <ffmpeguser> matty /j #ffmpeg [02:04] <nevcairiel> ffmpeguser: the apache license is not compatible with (L)GPLv2 [02:04] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * rb46fb615c4 10ffmpeg/libavcodec/tiff.c: tiff.c: Use switch / case instead of if / else where appropriate. [02:04] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * re767968c6e 10ffmpeg/libavcodec/tiff.c: Simplify tiff decoder by merging two functions. [02:04] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r7ee0649f95 10ffmpeg/libavcodec/tiff.c: [02:04] <CIA-101> ffmpeg: Fix decoding some 1bpp tiff files, they are palettized. [02:04] <CIA-101> ffmpeg: Fixes ticket #830. [02:04] <pasteeater> we support libvo_aacenc and it's APACHE, IIRC. [02:05] <pasteeater> but nothing is probably stopping anyone implementing apple's ALAC other than interest, time, or manpower [02:05] <ffmpeguser> nevcairiel: that means that the license is not a problem [02:06] <nevcairiel> the license is always a problem :) [02:08] <ffmpeguser> what is incompatible between the 2 licenses? [02:08] <nevcairiel> ask the FSF, not me [02:09] <nevcairiel> anyhow, replacing code is never a good option, but if someone wants to submit a wrapper that just uses an external copy of the apple alac library, i'm sure there are no objections [02:09] <nevcairiel> similar to how the libvo wrapper works [02:10] <ffmpeguser> in this case it would do good to replace it [02:10] <iive> isn't alac an apache license ? [02:10] <ffmpeguser> yes [02:10] <iive> i think it is gplv3 compatible. [02:10] <nevcairiel> and APL2.0 is not compatible with (L)GPLv2 [02:11] <nevcairiel> with 3 it is, but that would be a step backwards from a licensing standpoint [02:11] <ffmpeguser> how do you know? [02:12] <iive> there is gplv3 code in ffmpeg atm. this includes configure option to enable it. [02:12] <nevcairiel> indeed, such options would be used to also activate the wrapper around the apple code [02:12] <nevcairiel> however there still is no reason to replace the perfectly fine code that is already in ffmpeg [02:13] <iive> also, as I land in the middle of the conversation, I assume you are talking about encoder [02:14] <pasteeater> ffmpeguser: would you like me to modify the ticket title as a request to add wrapper support? [02:14] <ffmpeguser> i mainly want the decoder, i have no use for the encoder, but i'm sure that others will use it in the future if you add encoding support [02:14] <nevcairiel> maybe you should then name one thing that doesn't work in the decoder right now, and then that thing can be fixed [02:15] <nevcairiel> like you were asked to in the ticket. :) [02:15] <ffmpeguser> that would involve downloading more things instead of one, ffmpeg only [02:17] <iive> i thought ffmpeg had the decoder for ages... [02:17] <nevcairiel> it does [02:17] <ffmpeguser> yes, i would like that, i'll upload also a file that you can use for testing but on mediafire, because it is over the file size limit but below 10 MB [02:18] <nevcairiel> not the "official" code mind you, its own implementation, but apparently it works just fine [02:18] <ffmpeguser> it doesn't decode 6 channel alac streams [02:19] <ffmpeguser> but it does play it back the stream with -ac 1 [02:19] <ffmpeguser> -it [02:19] <ffmpeguser> yes [02:20] <ffmpeguser> in other words the decoding support is incomplete [02:21] <ffmpeguser> [alac @ 01090AE0] Unsupported channel count: 6 [02:22] <ffmpeguser> [caf @ 00A0B880] skipping CAF chunk: 75756964 (uuid), size 154 [caf @ 00A0B880] skipping CAF chunk: 75756964 (uuid), size 36 [caf @ 00A0B880] skipping CAF chunk: 6F767677 (ovvw), size 13176 [02:22] <ffmpeguser> these also [02:22] <nevcairiel> thats a demuxer, not related to the audio decoder [02:23] <nevcairiel> anyway, open issues for the missing features of the existing decoder, and that has a far higher chance of getting fixed [02:24] <ffmpeguser> the skippings could mean that there's a problem, that's why i mentioned those too [02:25] <ffmpeguser> alright, i'll open a new ticket [02:28] <CIA-101> ffmpeg: 03Jean First 07master * r9bfaeee5f3 10ffmpeg/libavcodec/8svx.c: [02:28] <CIA-101> ffmpeg: 8svx: remove unused variable in eightsvx_decode_frame [02:28] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [02:28] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:28] <CIA-101> ffmpeg: 03Jean First 07master * r3ceb4c907e 10ffmpeg/libavformat/cache.c: [02:28] <CIA-101> ffmpeg: cache: remove unused variable in cache_open [02:28] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [02:28] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:28] <CIA-101> ffmpeg: 03Jean First 07master * r576ada791a 10ffmpeg/libavformat/cache.c: [02:28] <CIA-101> ffmpeg: cache: use av_freep instead of av_free in cache_open [02:28] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [02:28] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:28] <CIA-101> ffmpeg: 03Jean First 07master * rb6ffceefb5 10ffmpeg/libavformat/mpegts.c: [02:28] <CIA-101> ffmpeg: mpegts: remove unused variable [02:28] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [02:28] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:28] <CIA-101> ffmpeg: 03Jean First 07master * r8aed73c355 10ffmpeg/libavformat/movenc.c: [02:30] <ffmpeguser> done http://ffmpeg.org/trac/ffmpeg/ticket/859 [02:40] <pasteeater> ffmpeguser: please show an example command of ffmpeg attempting to decode 6 channel alac and include the complete console output [02:40] <ffmpeguser> ok [02:53] <ffmpeguser> pasteeater: done [02:55] <pasteeater> ffmpeguser: thanks. if you edit your post and wrap the command and the output with the code block it will be easier to read. [02:58] <ffmpeguser> i don't understand [03:00] <ffmpeguser> text wrapping? [03:00] <pasteeater> http://ffmpeg.org/trac/ffmpeg/wiki/WikiFormatting#PreformattedText [03:01] <pasteeater> or the 5th botton from the left above the comment box [03:01] <pasteeater> button [03:01] <ffmpeguser> i'll check right now [03:09] <ffmpeguser> pasteeater: done [05:14] <buzz_> hmm 27d323577c19af218f8a5ac33364f213b623a023 is causing a problem in xbmc4xbox [05:15] <buzz_> causes lots of errors like "Truncating packet of size xxxx to 1" . without the patch playback is fine. dont know how to produce this on ffplay, since in this case it is a dvd being played via libdvdnav [05:20] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r5500e65342 10ffmpeg/libavformat/flvdec.c: [05:20] <CIA-101> ffmpeg: flvdec: Check avpriv_mpeg4audio_get_config() return [05:20] <CIA-101> ffmpeg: value before using its output. [05:20] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:20] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r4392b3d11e 10ffmpeg/libavcodec/tiff.c: [05:20] <CIA-101> ffmpeg: tiffdec: hotfix for fate failure. [05:20] <CIA-101> ffmpeg: Only use pal8 as output if the file has a palette or more than 2 colors. [05:20] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:56] <ffmpeguser> buzz_: the vob files in the DVD [05:56] <buzz_> unable to reproduce that way. [05:56] <ffmpeguser> hm [05:57] <ffmpeguser> you use windows? [05:57] <buzz_> linux and windows here [05:59] <buzz_> the error is av_log(0, AV_LOG_ERROR, "Truncating packet of size %d to %"PRId64"\n", size, remaining+1); - and size is say 2345 and remaining is equal to 0 [06:01] <ffmpeguser> you authored the dvd? [06:01] <buzz_> no [06:01] <ffmpeguser> who? [06:01] <buzz_> everything is fine if i revert that ffmpeg commit, but I dont know the ffmpeg code to know what could be going wrong [06:01] <buzz_> warner bros [06:02] <ffmpeguser> i'll see the commit [06:03] <buzz_> basically remaining is always 0 so if i add the logic "remaining>=0" to the if, it should work again. perhaps its due the way xbmc4xbox code uses ffmpeg specifically - perhaps ffmpeg doesnt know how much is remaining in this case ? [06:04] <buzz_> oh wait. brain glitch. >0 would still match hmm [06:05] <buzz_> i dont know what the variables are anyway. i just bisected it to find the commit that broke playback [06:05] <ffmpeguser> it doesn't play back? [06:06] <buzz_> no. [06:08] <ffmpeguser> open a ticket in the bug tracker http://ffmpeg.org/trac/ffmpeg/ [06:08] <buzz_> i didnt want to open a ticket until i understand the problem. im assuming remaining must be negative in the case where this works before the commit [06:09] <ffmpeguser> are you sure that the vob files don't play back? [06:10] <ffmpeguser> ffplay vobfile [06:12] <buzz_> they do play back [06:12] <buzz_> they play fine. [06:13] <ffmpeguser> you're on windows right now? [06:13] <buzz_> no [06:14] <buzz_> well. i have both running. why ? [06:14] <buzz_> are you an ffmpeg developer, or understand the ffmpeg code etc ? [06:15] <ffmpeguser> the error you saw, you saw that on your tv if i'm not mistaken, right? [06:15] <buzz_> i saw it in a log file. [06:15] <buzz_> why do you ask ? [06:17] <ffmpeguser> get the latest build for windows from http://ffmpeg.zeranoe.com/builds/ [06:19] <buzz_> im using the ffmpeg dlls in an application. i dont see how getting the latest build for windows will help ? unless im missing something (in which case please explain). the issue is with the latest build. reverting the mentioned commit solves it. im just trying to work out why i need to revert it now (adding some debugging in) [06:21] <ffmpeguser> xbmc4xbox is for the xbox, that means that the ffmpeg code runs on the xbox [06:21] <buzz_> yes [06:22] <buzz_> well seems that remaining is negative - 07:11:13 M: 27897856 ERROR: ffmpeg[184]: size = 2016, remaining = -137248 so ffmpeg doesnt know how much data is left, perhaps due to the way we are decoding. maybe i need an additional ffmpeg flag or something. need to trace this back [06:23] <ffmpeguser> we'll see if the same thing happens with a dec 26 build which is the latest from zeranoe [06:23] <ffmpeguser> that error has nothing to do with settings [06:23] <buzz_> it doesnt happy with commandline. its related to how we are using ffmpeg. [06:24] <buzz_> doesnt matter. i dont this you can help. [06:24] <buzz_> s/happy/happen [06:25] <ffmpeguser> if they have modified the ffmpeg code extensively then you should open a ticket in their bug tracker in they have one [06:25] <buzz_> well "remaining" variable starts at 0 and goes progressively negative with each call to ffio_limit [06:25] <buzz_> ffmpeguser, that would be opening a bug for myself then [06:26] <buzz_> i am developing xbmc4xbox [06:27] <buzz_> we havnt modified the ffmpeg code extensively.. slightly. but we are obviously doing something in a way this change doesnt like. ill continue debugging. thanks anyway [06:27] <ffmpeguser> maybe it is something obvious [06:40] <buzz_> hmm s->maxsize is always 0 in this case, so then it does int64_t remaining= s->maxsize - avio_tell(s); which sets remaining to a negative number (maxsize = 0, avio_tell = 1769495 for example) [06:56] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r91ab557b68 10ffmpeg/libavcodec/aacdec.c: [06:56] <CIA-101> ffmpeg: Revert "aacdec: Avoid unneeded reinit due to ADTS headers" [06:56] <CIA-101> ffmpeg: This has been fixed differently and its revert avoids error messages [06:56] <CIA-101> ffmpeg: when decoding xx.flv. This also reduces the difference to qatar. [06:56] <CIA-101> ffmpeg: This reverts commit 5a2b3f3a52070232a68582a502fb60a9226ca86a. [06:56] <CIA-101> ffmpeg: Reviewed-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [06:56] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:21] <buzz_> michaelni / any other devs, would appreciate any feedback regarding http://www.ffmpeg.org/trac/ffmpeg/ticket/861 (middle of night in eurozone i realise now). [07:26] <michaelni> buzz_, ill add aworkaround tomorrow but why does your avio_size() return 0 ? [07:26] <michaelni> that would be filesize = 0 [07:27] <buzz_> i assume the way the data is passed to ffmpeg. via libdvdnav. im not all that familiar with the code in this part of xbmc4xbox [07:30] <michaelni> workaround that ill probably add tomorrow: s->maxsize= newsize - !newsize; [07:31] <michaelni> that is if it still makes sense after sleeping [07:32] <buzz_> wish i understood some of this code better - this is another patch i had to revert for the dvd playback to work - http://sourceforge.net/apps/trac/xbmc4xbox/browser/trunk/xbmc/cores/dvdplay… again i assume its related to the way the data is sent to ffmpeg, so eof doesnt work right ? [07:33] <buzz_> xbmc are on an older ffmpeg, so they will bump into these issues when they upgrade, and no doubt understand it better since they wrote this code. im just maintaining the old xbox version of the player [07:33] <buzz_> but we have a newer ffmpeg. [07:33] <buzz_> anyway. have a nice sleep :) [07:48] <buzz_> yeh i guess we wont know the file size, since we are processing dvd video blocks. the seek call doesnt do anything but say seeking is possible, and seeking is handled via dvdnav [07:57] Action: buzz_ gets confused/lost by the code [11:46] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r549b53e8cb 10ffmpeg/libavfilter/avfilter.h: [11:46] <CIA-101> ffmpeg: lavfi: remove redundant definition of avfilter_copy_frame_props() in avfilter.h [11:46] <CIA-101> ffmpeg: Rationale: avfilter_copy_frame_props() was already defined in [11:46] <CIA-101> ffmpeg: libavfilter/avcodec.h, and keeping the lavc/lavfi API glue localized in a [11:46] <CIA-101> ffmpeg: specific file should ease maintainance and help the ones which use [11:46] <CIA-101> ffmpeg: libavfilter without depending on libavcodec. [11:46] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r8fb03b4d70 10ffmpeg/ (7 files in 3 dirs): (log message trimmed) [11:46] <CIA-101> ffmpeg: lavfi: port tinterlace filter from MPlayer [11:46] <CIA-101> ffmpeg: Port MPlayer tinterlace filter from MPlayer, with some ideas taken [11:46] <CIA-101> ffmpeg: from the FFmbc/libavfilter port, with the following main differences: [11:46] <CIA-101> ffmpeg: * added support for full-scale YUVJ formats [11:46] <CIA-101> ffmpeg: * added support for YUVA420P [11:46] <CIA-101> ffmpeg: * request_frame() on the filter is forced to return a frame [11:53] <CIA-101> ffmpeg: 03Jean First 07master * r81980bf740 10ffmpeg/libavdevice/lavfi.c: [11:53] <CIA-101> ffmpeg: lavd/lavfi: fix compiler warning for uninitialized variables [11:53] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [11:53] <CIA-101> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [15:58] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * re6362f3a3c 10ffmpeg/libavformat/utils.c: [15:58] <CIA-101> ffmpeg: lavf: ffio_limit(), treat avio_size()==0 as error. [15:58] <CIA-101> ffmpeg: This works around issues arising from inputs that claim to have a [15:58] <CIA-101> ffmpeg: filesize of 0. [15:58] <CIA-101> ffmpeg: Reported-by: buzz_ [15:58] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:26] <CIA-101> ffmpeg: 03Nicolas George 07master * re90a69e955 10ffmpeg/ (5 files in 3 dirs): lavfi: add astreamsync audio filter. [16:27] <CIA-101> ffmpeg: 03Nicolas George 07master * r4962edf889 10ffmpeg/ (6 files in 3 dirs): lavfi: add amerge audio filter. [16:31] <CIA-101> ffmpeg: 03Reimar Döffinger 07master * r42a1f1d7a8 10ffmpeg/ (libavcodec/lcldec.c libavutil/lzo.h): [16:31] <CIA-101> ffmpeg: Avoid uninitialized data in lcldec when ofs is 0 in MSZH. [16:31] <CIA-101> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [16:34] <buzz_> michaelni, i assume the problem in this case, is the size of the input is not known ? [16:35] <michaelni> avIo_size should return an error and not 0 in that case [16:36] <buzz_> hmm ok. ill take a look at our code, and see if elupus can help (he wrote it) [16:40] <michaelni> maybe an issue with AVSEEK_SIZE or SEEK_END handling [16:41] <buzz_> i emailed Joakim Plate (elupus) in the hope he can shed some light/explain the issue [17:11] <CIA-101> ffmpeg: 03Nicolas George 07master * r0098e79f8a 10ffmpeg/libavcodec/libvorbis.c: libvorbis: proper error logging and return. [17:15] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * rf69a766d11 10ffmpeg/libavcodec/pamenc.c: Fix mono* pam encoding. [17:17] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r810a14dbcc 10ffmpeg/libavcodec/aacdec.c: [17:17] <CIA-101> ffmpeg: aacdec: Evaluate further PCE elements. [17:17] <CIA-101> ffmpeg: Fixes Music\ Station\ Super\ Live\ -\ 2011.12.23.mkv [17:17] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:17] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rbd553d5ea9 10ffmpeg/libavformat/mxg.c: [17:17] <CIA-101> ffmpeg: mxg: fix compiler warning for uninitialized variables [17:17] <CIA-101> ffmpeg: Based on work by: Jean First <jeanfirst(a)gmail.com> [17:17] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:17] <CIA-101> ffmpeg: 03Jean First 07master * racc88f07d4 10ffmpeg/libavformat/id3v2.c: [17:17] <CIA-101> ffmpeg: id3v2: fix compiler warning for uninitialized variables [17:17] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [17:17] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:35] <durandal_1707> does anyone can actually get mplayer to display bt20 videos? [19:24] <durandal_1707> is it fine to do flipping at decoder level? [20:40] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r9d9316ae34 10ffmpeg/libavcodec/h264_refs.c: [20:40] <CIA-101> ffmpeg: h264: Fix seeking in 00017.MTS [20:40] <CIA-101> ffmpeg: Fixes Ticket852 [20:40] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:47] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r714d2f9796 10ffmpeg/libavcodec/tiff.c: Only expand raw and packed 1bpp tiff horizontally if it is palettised. [20:53] <buzz_> happy new year michaelni and all other ffmpeggers. have a good one! [22:37] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r2cba62bd84 10ffmpeg/libavformat/asf.c: [22:37] <CIA-101> ffmpeg: Do not map asf metadata key WM/Track to track. [22:37] <CIA-101> ffmpeg: WM/Track is 0-based (and deprecated in favour of WM/TrackNumber), [22:37] <CIA-101> ffmpeg: track is 1-based. [22:37] <CIA-101> ffmpeg: Fixes ticket #825. [23:40] <Granjow> Why do I get a «Could not open file» with av_write_frame() here on line 113? http://codepad.org/GL3N1Lhr [23:41] <funman> Granjow: you should ask on #ffmpeg (see topic) [00:00] --- Sun Jan 1 2012

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ffmpeg.log.20111231
by burek 01 Jan '12

01 Jan '12

[00:49] <emerica> Did some more tests with the ac3 sync on hardware decoder problem, some results here. http://pastebin.ca/2097568 [02:13] <Matty> In have a problem with packing ASS subtitles into Matroska files. I run "ffmpeg -i video.avi" -i subs.srt video.mkv". Resulting file contains ASS subtitle track. But VLC and mplayer do not show subtitles, while playing video.mkv . Does somebody know what am I doing wrong? [02:14] <pasteeater> Matty: I am very subtitle ignorant, but you can use a pastebin site to show your ffmpeg command and the complete console output [02:27] <sam1> ffmpeg could burn ass subtitle,you could convert *.srt to *.ass [02:28] <Matty> Here is console output of "ffmpeg -i sample.avi -i subs.srt -y sample.mkv" producinf MKV file with subtitles that do not show in all players I tested. http://pastebin.com/PUJdb2Uv [02:30] <Matty> I could burn the subtitiles, but I want to include more subtitles for later switch. Moreover I want to play the file on Android device and the MX VIdeo player displays only ASS subtitles. [02:32] <sam1> hi , Matty , what's format subtitle do you burn? [02:32] <fortress> how do i get ffmpeg to disregard the subtitle stream ? [02:33] <sam1> -sn [02:33] <fortress> thank you! [02:34] <sam1> you are welcome [02:35] <Matty> I do not want to burn them in. I want to pack them in matroska file. Packing srt works fine, but ass produces broken subtitles that other players do not display. [02:36] <sam1> Matty: I could burn the subtitiles, ?? you said you could do it . [02:37] <sam1> i want to konw how do you do that [02:37] <emerica> http://code.google.com/p/ass2srt/ as workaround maybe if srt works for you [02:38] <Matty> I can burn them in video like this: "ffmpeg -i sample.avi -vf ass=subs.ass -y sample.mkv" [02:40] <Matty> srt works fine. Unfortunetly I want to play the file where only ass is supported. [02:40] <sam1> dou you burn other formats? ,e.g *.sub *.srt ? [02:40] <emerica> ahh ok, so the text itself is packed in the mkv, more used to dvbsub and userdata captions here [02:41] <Matty> I do not know how to burn sub or srt using ffmpeg. Mplayer can to that. [02:42] <sam1> i konw mplayer could do it. i only burn *.ass [02:43] <Matty> By the way ffmpeg can convert srt to ass "ffmpeg -i subs.srt subs.ass". [02:44] <sam1> thanks matty [02:45] <Matty> I think the problem with ASS in MKV is that ffmpeg fails to add some fields that seems to be requred in mkv. Resulting in subtitles with empty text in VLC, mplayer and others. [02:50] <emerica> i dont see them in ffplay either, not sure it it's supposed to [02:51] <Matty> I have never seen subtitles in ffplay. [02:54] <emerica> i've seen captions i *think* [02:54] <emerica> or dvbsub [02:55] <emerica> probly dvb sub bitmap subtitles [02:56] <emerica> if i convert my srt to ass with ffmpeg, then do it, vlc see em [03:00] <Matty> I see the too. Problem is when ASS subtitles are part of mkv file. Not burn in the video, but as separate subtitle track. [03:01] <Matty> I use command "ffmpeg -i video.avi -i subs.srt video.mkv". [03:04] <Matty> It results in matroska file fith 3 tracks: video, audio and subtitles. [03:04] <emerica> try ffmpeg -i bad.ass good.ass ; ffmpeg -i sample.avi -i good.ass out.mkv is what im saying [03:04] <pasteeater> did you know it's converting your srt -> ass? did you want to keep it as srt or change to ass? [03:04] <emerica> thats what i found ya [03:04] <emerica> so seemed like it should do ass/ssa either way [03:05] <Matty> I want them to be included as ASS. I know ffmpeg is converting them. [03:05] <Matty> Using ASS as input do not help. [03:06] <emerica> the sameple here http://matroska.org/technical/specs/subtitles/ssa.html muced into my test ts to mkv and subs show in vlc [03:06] <emerica> yesterday git clone [03:06] <emerica> i had to change the timecode a bit to be in my sample [03:17] <Matty> Wow, sample ASS works. [03:19] <emerica> a step forward ;) [03:19] <Matty> Those converted from srt by ffmpeg do not. [03:19] <Matty> I think I have found the problem. [03:19] <emerica> mine did, an example from wikipedia [03:20] <emerica> it was just two lines though, no formatting if you can in srt [03:22] <Matty> Sample subtitles contain all possible fields. Those by ffmpeg only: Layer, Start, End and Text. [03:24] <Matty> I run "ffmpeg -i subs.srt sample.ff.mkv" "ffmpeg -i subs.srt subs.ass" and "mkvmerge -o sample.mm.mkv subs.ass". [03:25] <Matty> The result are two almost identical samples one by ffmpeg, second by mkvmerge. [03:29] <Matty> The one by ffmpeg miss align fields in subtitles so mplayer and vlc do not show subtitle text(it seems empty to them). FFmpeg takes fileds from source file and do not recpect those fileds are mandatory in ASS packed in MKV. [03:38] <Matty> Now I think this is a bug in the way how ffmpeg integrates ASS into Matroska files. I will fill a bug report about it, maybe I can even fix the code. [03:39] <Matty> Than you very mych for help. [03:51] <JohnP789> I'm having trouble with audio not being encoded with libfaac on some input file. [03:51] <JohnP789> http://pastebin.com/MnN4vW02 [03:51] <JohnP789> There's an audio stream shown in the stream mapping during encode, but the output file has no audio stream in it. [03:51] <JohnP789> Any ideas on what I'm doing wrong? [07:50] <defaultro> is -r 60000/1001 enough to change frame rate? [09:54] <Granjow> Good morning :) What is the best way if I want to use ffmpeg for reading and seeking in videos -- start with the code from the decoding/encoding example or something else? [10:01] <metvik> reading and seeking in videos to what end? [10:06] <Granjow> metvik: I first want to extract all frames, but scaled (thumbnail size), and later on individual ones in full size [10:13] <metvik> for #1, perhaps try: ffmpeg -i <input-video> -s <new-width>x<new-height> -f image2 "output-%05d.png" [10:14] <metvik> for #2, there is ffmpeg -i <input-video> -f image2 -ss <hh:mm:ss> -vframes 1 "output-%05d.png" [10:16] <Granjow> metvik: Could I also do this with the library directly? [10:16] <Granjow> (programmatically) [10:19] <metvik> i wouldn't be quite certain how, but i'm also interested to know how if anyone else has a solution [15:57] <hivemind> uh [15:57] <hivemind> i see the guide for ubuntu, but aren't see the guide for Debian [16:57] <defaultro> am I right folks that I can concatentate multiple mp4 files? [17:05] <knobo> Am I suppose to be able to send a stream like this: ffmpeg -i videofile.flv -re -acodec copy -vcodec copy -f flv rtmp://localhost/live/test [17:05] <knobo> and play it by doing ffplay rtmp://localhost/live/test? [17:38] <metvik> highzeth, I'm trying to convert LATM AAC audio to ADTS AAC audio (copy) but it results in a silent audio file. [17:39] <metvik> s/highzeth/Hi [17:40] <metvik> Googling found "[Change] LATM/LOAS AAC: Output to MP4 with FFmpeg muxer was not converting audio to ADTS resulting in silent audio." in VideoReDo's changelog. [17:41] <metvik> input: ffmpeg -i input.aac -acodec copy -f adts output.aac [17:41] <metvik> output: http://pastebin.com/0kKy3yRv [17:49] <Granjow> Good afternoon :) I'm trying to read a video and save the frames as PNG, but I get a segfault in avcodec_decode_video2 here on line 74: http://codepad.org/tGV14pgu [17:50] <Granjow> Why is this? [17:50] <Granjow> It happens with the first frame already. [18:02] <Granjow> Problem found -- did not avcodec_open2 the codec. [18:18] <Granjow> Now it segfaults on sws_scale on line 84 and I'm really clueless. Still http://codepad.org/tGV14pgu [18:25] <Granjow> ... which I could solve by not following the tuorial and use avpicture_fill but avpicture_alloc instead [20:00] <LRN> Happy New Year! [22:28] <alesan> hi [22:33] <alesan> so I need to convert this 1920x1080 60fps video in a smaller, say DVD resolution, 30fps video [22:33] <alesan> so that I can actually play it on my computer [22:33] <alesan> ideas? can ffmpeg do it? [22:33] <alesan> I mean, the fps conversion [22:37] <pasteeater> alesan: it can drop frames to make a 30 fps output [22:39] <alesan> ok, I am not sure what the best thing would be... simply dropping or mess up some frame-merging algorithm... [22:40] <pasteeater> is it 1080p or 1080i? [22:40] <alesan> p [22:40] <pasteeater> you could just try resizing and the computer will probably be able to handle 60 fps decode [22:41] <pasteeater> is there a particular output format you want? [22:41] <alesan> H.264 would be good so I can transfer it on the internet [22:41] <alesan> or you mean container? [22:41] <pasteeater> both [22:41] <alesan> I am open to any container that is good for this job. No particular preferenmces [22:41] <alesan> preferences [22:42] <alesan> it has 5.1 sound that I would not mind being converted to simple stereo [22:42] <alesan> (it's from a Panasonic TM900 btw) [22:43] <pasteeater> ffmpeg -i input -c:v libx264 -preset medium -crf 24 -vf scale=640:-1 -an -t 60 output.mkv [22:43] <alesan> oh, .mkv :) I like that [22:44] <pasteeater> this will ignore audio so you can focus on a video size that works well [22:44] <alesan> I'll check the man page to understand your options [22:44] <pasteeater> and make a 60 second output just as a test [22:44] <pasteeater> so you don't have to encode the whole thing [22:44] <alesan> the test video I am working on is 13s only [22:44] <pasteeater> ok. omit -t 60 [22:44] <alesan> later I have a 5 minutes one that is my real "target" [22:45] <pasteeater> adjust crf value for quality. lower is higher quality. [22:45] <pasteeater> use the highest crf value that still gives an acceptable quality [22:45] <alesan> sure [22:45] <alesan> hm [22:45] <alesan> it complains... about subtitles? [22:46] <pasteeater> you can keep or remove subtitles [22:46] <alesan> http://pastebin.com/QFZzb55r [22:46] <alesan> subtitles... I shoot this video myself with a camcorder... what subtitles? [22:46] <alesan> interesting... [22:47] <pasteeater> do you want the subtitles? [22:47] <alesan> no, I do not even know what subtitles there are [22:48] <pasteeater> then add -sn i think [22:49] <alesan> ok it's encoding... [22:50] <alesan> this still is a 60fps video [22:50] <alesan> I would like to try to convert it to a 25fps [22:51] <alesan> or 30 [22:51] <alesan> trying with -r 30 [22:52] <alesan> seems OK [22:52] <alesan> now I would like to put audio back in [22:53] <alesan> removing "-na"... [22:53] <alesan> -an [22:54] <pasteeater> you can copy the audio or re-encode it. [22:57] <alesan> how was to get the list of available presets? [22:57] <pasteeater> x264 --help [22:57] <pasteeater> generally you use the slowest one you have patience for [22:57] <Shimmy> https://ffmpeg.org/trac/ffmpeg/ticket/844 please change status to completed, and thanks for completing it [23:00] <Shimmy> Sorry for my last line, there is still a problem with the RSS, it only shows the title header, no content [23:00] <pasteeater> Shimmy: i see you modified it. did you want to keep it open? [23:00] <pasteeater> ah, i missed your last line in this channel [23:01] <Shimmy> Yes, should be kept open [23:01] <Shimmy> maybe u or i can do it, just need access to the server [23:01] <Shimmy> and make the rss include the content of each article [23:03] <pasteeater> that will be a little harder to implement [23:09] <Granjow> Why do I get a «Could not open file» with av_write_frame() here on line 113? http://codepad.org/GL3N1Lhr [23:10] <qxt> Is there a cmd in ffmpeg that I can define the size of a video with the audio. Or do I have to take the time and bit rates and calculate manually? [23:12] <qxt> Need the highest quality from a BluRay 48GB video to fit on a 700MB disk. [23:13] <qxt> As of now my 2ed pass looks something like this -pass 2 -vcodec libx264 -preset fast -b 512k -acodec libfaac -ab 128k -ac 2 output.mp4 [23:13] <qxt> just not the same bit rates [23:29] <pasteeater> qxt: manually i think [23:30] <pasteeater> and use -preset veryslow if you want the highest quality for your desired bitrate, although, it is very slow. [23:30] <pasteeater> what's the duration of the input? [23:32] <qxt> pasteeater looks that way. Been Google'n for a while. The movie is Drive its 1:40:47 long [23:32] <qxt> pasteeater kick ass movie btw [23:32] <pasteeater> nice [23:32] <qxt> :) [23:35] <pasteeater> that would end up with around -b:v 786k -b:a 128k [23:35] <pasteeater> for 675MB output for "safety" [23:36] <pasteeater> or (700 MB * 8192 [converts MB to kilobits]) / 6047 seconds = ~948.3 kilobits/s [23:36] <pasteeater> i think [23:37] <metvik> is there an accurate way to account for overhead? [23:37] <pasteeater> i don't know [23:38] <metvik> would you happen to know anything about encoding AAC audio via ffmpeg? [23:38] <pasteeater> depends on the question [23:39] <metvik> I'm trying to convert LATM AAC audio to ADTS AAC audio (copy) but it results in a silent audio file. [23:39] <metvik> <metvik> Googling found "[Change] LATM/LOAS AAC: Output to MP4 with FFmpeg muxer was not converting audio to ADTS resulting in silent audio." in VideoReDo's changelog. [23:39] <metvik> <metvik> input: ffmpeg -i input.aac -acodec copy -f adts output.aac [23:39] <metvik> <metvik> output: http://pastebin.com/0kKy3yRv [23:40] <qxt> pasteeater Wow talk about spoon feeding. I just am running it as -codec libfaac -ab 128k and the video at 786k. Thx I'll let you know how it works out =) [23:44] <qxt> pasteeater was wondering if I should use veryslow during the first pass also. My computer can deal with the time. [23:45] <pasteeater> you should use the same preset for both passes [23:45] <pasteeater> or encoding will fail [23:46] <qxt> pasteeater puuu, that was a close one. Had fast on the first pass and veryslow on the 2ed one. [23:53] <qxt> BTW Happy NEW YEAR in 7 min!!! =D [23:54] <pasteeater> not for another 10 hours and 7 mins here [23:55] <JEEB> the internets is a funny place [23:56] <JEEB> there is no certain time zone for the internet [23:56] <JEEB> although someone tried to bring up something called 'the internet time' [23:57] <qxt> Got to have a skål on my balcony. Fireworks like crazy. Back in 5 min or so! [00:00] --- Sun Jan 1 2012

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