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January 2012
- 1 participants
- 62 discussions
[00:07] <michaelni> btw, if someone has an idea about the openbsd/sh segfault ...
[00:07] <michaelni> it seems working if bash is used ...
[00:07] <michaelni> but we should get it working by default
[00:08] <michaelni> the commit that "caused" it was the addition of the yuv4 test
[00:08] <michaelni> reverting that on HEAD doesnt fix it though
[00:13] <durandal_1707> probably need reverting v308 test too
[00:13] <durandal_1707> wich points that we can not add more tests
[00:14] <iive> is it the sh that segfaults?
[00:18] <Daemon404> [18:07] <@michaelni> it seems working if bash is used ... <-- well considering it doesnt even have bash by default... :P
[00:21] <ohsix> it probably doesn't have any of the libraries & headers you need to build ffmpeg either
[00:21] <ohsix> by default
[00:23] <michaelni> iive its /bin/sh which seems provided by ksh
[00:23] <cbsrobot> ksh is the default shell on obsd
[00:24] <Daemon404> ohsix, and it also likely has an ancient gcc version in its base
[00:24] <ohsix> awesome
[00:25] <iive> it may be better to report the segfault to openbsd before than trying to workaround it.
[00:25] <ohsix> could be a security issue ;]
[00:25] <durandal_1707> you will be flamed if it is fixed in newer version
[00:26] <Daemon404> he will be flamed regardless
[00:27] <ohsix> it is openbsd afterall
[00:30] <durandal_1707> fourcc A64L is our rgba64le
[01:33] <durandal_1707> wtf r210 still encodes crap on valgrind
[01:34] <ubitux> keep in mind it takes ~6-7 hours to make a run
[01:35] <durandal_1707> ahh, you are right it still compiles old revision
[01:36] <durandal_1707> michaelni: one of openbsd recovered, did you changed something?
[01:40] <durandal_1707> hmm, openbsd machines are compiling unknown revision....
[01:41] <durandal_1707> that is michaelni playing
[01:42] <ubitux> durandal_1707: also, the box will certainly get down in the night so i'll have to restart the run manually tomorrow ;)
[01:49] <michaelni> durandal_1707, i changed the first to bash
[01:50] <durandal_1707> michaelni: but other have same revision and are still red
[01:59] <michaelni> the other 2 run with /bin/sh
[02:00] <michaelni> i dont really want to depend on bash
[02:47] <burek> :/
[02:47] <burek> I still can't get rid of this stupid error: http://pastebin.com/rex228L7
[02:48] <burek> it compiles everything just fine, but linking in the end fails..
[02:57] <funman> burek: hm
[02:58] <funman> why does it pull libfontconfig ?
[02:58] <funman> ah freetype
[02:58] <funman> pkg-config --libs freetype2
[02:59] <burek> # pkg-config --libs freetype2
[02:59] <burek> -L/usr/local/lib -lfreetype
[02:59] <burek> I've compiled both freetype and fontconfig with
[02:59] <burek> --enable-static=yes --enable-shared=no
[03:17] <funman> then your freetype2.pc should have -lfontconfig
[03:17] <funman> which version?
[03:18] <burek> version of fontconfig?
[03:18] <funman> freetype
[03:18] <burek> 2.4.8
[03:19] <funman> yeah it's buggy
[03:19] <funman> hm oh well
[03:19] <funman> fontconfig depends on freetype it seems, not the opposite
[03:20] <burek> fontconfig is from latest git
[03:20] <funman> where does the -lfontconfig come from ?
[03:21] <funman> any way pkg-config --libs fontconfig
[03:21] <burek> let me check
[03:22] <burek> maybe libass
[03:23] <funman> my libass.pc has Requires: fontconfig >= 2.2.0
[03:23] <burek> mine too
[03:24] <burek> and freetype2 >= 9.6.3
[03:26] <funman> hm weird
[03:28] <burek> since "XML_GetCurrentLineNumber" is part of libexpat, it seems like libexpat is missing somewhere..
[03:28] <burek> but I've compiled that too
[03:28] <funman> from fontconfig
[03:28] <funman> .pc
[03:29] <burek> yes
[03:29] <burek> hmh..
[05:29] <CIA-108> ffmpeg: 03Aneesh Dogra 07master * rfef3771778 10ffmpeg/libavcodec/sunrast.c:
[05:29] <CIA-108> ffmpeg: sunrast: Replace magic number by a macro.
[05:29] <CIA-108> ffmpeg: Signed-off-by: Aneesh Dogra <lionaneesh(a)gmail.com>
[05:29] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[05:29] <CIA-108> ffmpeg: 03Aneesh Dogra 07master * r4ffb8e2c14 10ffmpeg/libavcodec/sunrast.c:
[05:29] <CIA-108> ffmpeg: sunrast: Remove if (unsigned int < 0) check.
[05:29] <CIA-108> ffmpeg: Note: This fixes the following GCC warning :-
[05:29] <CIA-108> ffmpeg: libavcodec/sunrast.c:94: warning: comparison of unsigned expression < 0 is always false.
[05:29] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[05:29] <CIA-108> ffmpeg: 03Aneesh Dogra 07master * r415f358a1f 10ffmpeg/libavcodec/sunrast.c:
[05:29] <CIA-108> ffmpeg: sunrast: Cosmetics
[05:29] <CIA-108> ffmpeg: Signed-off-by: Aneesh Dogra <lionaneesh(a)gmail.com>
[05:29] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[05:29] <CIA-108> ffmpeg: 03Ronald S. Bultje 07master * re92003514d 10ffmpeg/libavcodec/ (9 files in 2 dirs): png: move DSP functions to their own DSP context.
[05:29] <CIA-108> ffmpeg: 03Ronald S. Bultje 07master * r331e7c4cb3 10ffmpeg/libavcodec/x86/imdct36_sse.asm:
[05:29] <CIA-108> ffmpeg: imdct36: mark SSE functions as using all 16 XMM registers.
[05:29] <CIA-108> ffmpeg: On x86-64, it indeed uses all 16 registers (and on x86-32, this gets
[05:30] <CIA-108> ffmpeg: Merge remote-tracking branch 'qatar/master'
[05:30] <CIA-108> ffmpeg: * qatar/master:
[05:30] <CIA-108> ffmpeg: png: add missing #if HAVE_SSSE3 around function pointer assignment.
[05:30] <CIA-108> ffmpeg: imdct36: mark SSE functions as using all 16 XMM registers.
[05:30] <CIA-108> ffmpeg: png: move DSP functions to their own DSP context.
[05:30] <CIA-108> ffmpeg: sunrast: Add a sample request for TIFF, IFF, and Experimental Rastfile formats.
[05:30] <CIA-108> ffmpeg: 03Aneesh Dogra 07master * rf9708e9a0e 10ffmpeg/libavcodec/sunrast.c:
[05:30] <CIA-108> ffmpeg: sunrast: Add a sample request for TIFF, IFF, and Experimental Rastfile formats.
[05:30] <CIA-108> ffmpeg: Signed-off-by: Aneesh Dogra <lionaneesh(a)gmail.com>
[05:30] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[11:37] <CIA-108> ffmpeg: 03Carl Eugen Hoyos 07master * r6838df0488 10ffmpeg/libavcodec/ (imgconvert.c version.h):
[11:37] <CIA-108> ffmpeg: Add pix_fmt_info for yuva444.
[11:37] <CIA-108> ffmpeg: Reviewed-by: Paul B Mahol
[11:37] <CIA-108> ffmpeg: 03Carl Eugen Hoyos 07master * r36436a4032 10ffmpeg/libavcodec/ (pgssubdec.c version.h): Add option forced_subs_only for Bluray subtitles.
[11:37] <CIA-108> ffmpeg: 03Martin Lambers 07master * r6fe12c7ff2 10ffmpeg/libavformat/img2.c: Detect jps and pns files as (3D) jpg and png images.
[11:37] <CIA-108> ffmpeg: 03Carl Eugen Hoyos 07master * r5f27374c1b 10ffmpeg/libavformat/isom.c: Cosmetics: Add bit per component information to v308 comment.
[12:46] <weekee> anyone here who could possibly do a little modification to the .MOV muxing in ffmpeg for a bounty?
[12:46] Action: av500 would prefers a Snickers
[12:47] <weekee> hehe :)
[12:47] <weekee> man now i want a bounty too
[14:40] <ubitux> broken video: http://dewy.fem.tu-ilmenau.de/CCC/24C3/mp4/24c3-2355-de-tracker_fahrn-COMPA…
[14:41] <ubitux> it looks like something is wrong with aac
[14:41] <ubitux> i'll open an issue today if i don't forget
[15:29] <ubitux> michaelni: could you comment on http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2012-January/120146.html ?
[16:49] <Compn> http://mpg123.org/beating_mp3lib_in_mplayer/
[16:49] <Compn> anyone with 3dnowex skills want to help the mpg123 dev? :P
[16:49] <Compn> ehe
[16:53] <av500> duron?
[16:54] <av500> k6-3?
[16:54] <teratorn> computers
[16:54] <nevcairiel> ...very old computers
[16:54] <av500> were they 32bit already?
[16:54] <teratorn> yep
[16:55] <Compn> 300mhz > 1ghz ?
[16:55] <Compn> roughly
[16:55] <nevcairiel> but not 686
[16:55] <nevcairiel> or wait, the k6-3 might've been
[16:55] <teratorn> they've been 32-bit since the 386 in 1985
[16:57] <teratorn> k6 was 686 i think
[16:57] <teratorn> k5 was last 586?
[16:57] <teratorn> hm, or not
[16:58] <nevcairiel> i think it was 686
[16:59] <nevcairiel> although wikipedia says its "P5 Pentium-class", which is 586
[16:59] <nevcairiel> oh well
[16:59] <nevcairiel> who cares!
[17:00] <teratorn> but then wiki always lists K5 and K6 as PentiumPro/6th-gen competitors - http://en.wikipedia.org/wiki/Pentium_Pro
[17:01] <Compn> why is amd stuff on pentium page ?
[17:01] <nevcairiel> shows them under competitors
[17:01] <teratorn> in any case, 586 and 686 are the same architecture
[17:01] <Compn> x86
[17:01] <teratorn> with the same extensions
[17:01] <nevcairiel> 686 must have something that 586 doesnt
[17:01] <Compn> more cpu opts
[17:02] Action: Compn doesnt have a clue about this stuff
[17:02] <teratorn> MMX
[17:03] <teratorn> only diff would be weather it was AMD with 3dNow or Intel with SSE
[17:03] <teratorn> I forget when it was that intel and amd adopted/licensed each others 3dn/sse extensons
[17:04] <teratorn> nevcairiel: it's a different, more advanced microarchitecture
[17:04] <teratorn> but from software it's the same thing
[17:05] <nevcairiel> intel never cared to license 3dnow, unless i missed that bit over the years
[17:06] <teratorn> hmm, I guess not
[17:07] <Compn> 3dnow is dead, i think
[17:07] <nevcairiel> i remember AMD saying they are removing it
[17:08] <teratorn> yeah
[17:08] <teratorn> according to wiki they are
[17:08] <nevcairiel> also found an official post about it
[17:17] <ubitux> michaelni: thx
[18:39] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r6166bf3cbe 10ffmpeg/libavcodec/huffman.c: (log message trimmed)
[18:39] <CIA-108> ffmpeg: Deobfuscate ff_huff_build_tree.
[18:39] <CIA-108> ffmpeg: I have no idea what the idea was behind the original code,
[18:39] <CIA-108> ffmpeg: but the new code is equivalent to it.
[18:39] <CIA-108> ffmpeg: In that loop that places the new node nodes[j] contains
[18:39] <CIA-108> ffmpeg: always the data of the new node (since the steps are always
[18:39] <CIA-108> ffmpeg: in order: FFSWAP copies node[j] to node[j-1], j is decremented).
[18:39] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r2ab5feafe1 10ffmpeg/libavcodec/huffman.c:
[18:39] <CIA-108> ffmpeg: huffman: use a simple assignment instead of FFSWAP.
[18:39] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[18:42] <sagitter> Hello !
[18:43] <durandal_1707> hello
[18:47] <sagitter> i'm trying to build ffmpeg in Fedora 16
[18:48] <sagitter> in make of ffmpeg, obtain "collect2: ld returned 1 exit status"
[18:48] <sagitter> at this point: libavcodec/libavcodec.a(libx264.o): In function `X264_init':
[18:48] <sagitter> can you help me please ?
[18:53] <relaxed> sagitter: that is a #ffmpeg question
[18:53] <sagitter> ok, thank you
[18:53] <sagitter> sorry
[20:47] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r9729f140ae 10ffmpeg/libavcodec/diracdec.c:
[20:47] <CIA-108> ffmpeg: diracdec: Fix integer overflow leading to out of global array read.
[20:47] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[20:47] <CIA-108> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com>
[20:47] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:48] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rd2a47c9473 10ffmpeg/tests/fate.sh:
[20:48] <CIA-108> ffmpeg: fate: add support for specifying the shell used to run configure
[20:48] <CIA-108> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com>
[20:48] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:13] <Daemon404> michaelni, poke (because i have a question about a file you wrote)
[22:13] <durandal_1707> what file?
[22:13] <Daemon404> libavutil/x86_cpu.h
[22:14] <Daemon404> it;s teh only file that is stopping libpostproc from being standalone
[22:14] <Daemon404> (i.e. it is not installed during make install of ffmpeg)
[22:14] <Daemon404> i mean it's just a bunch of defines... so...
[22:14] <Daemon404> i might just package it with postproc
[22:15] <Daemon404> yeah ill do that
[22:16] <Daemon404> and remove config.h
[22:21] <ubitux> michaelni: your proposition seems to work (http://pastie.org/3284781) but i'd like to understand the cur_dts check; if cur_dts is AV_NOPTS_VALUE or any other value`0, what does this mean?
[22:22] <ubitux> i have very hard time understanding what's going on in lavu/utils.c
[22:40] <durandal_1707> would it have sense to make emboss filter or this is already doable with current code?
[22:43] <michaelni> vf_geq could probably be used but a dedicated filter would be faster & easier to use
[22:43] <michaelni> ubitux, cur_dts!=0 pretty much means there was a packet with known timestamp or duration
[22:43] <michaelni> thats also why mp3 failed
[22:44] <michaelni> it needs to demux & parse a few packets to determine duration
[22:44] <ubitux> can't this known timestamp be 0?
[22:44] <michaelni> i can
[22:44] <michaelni> it can
[22:44] <michaelni> but the first packet with mp3 is often some header liek id3
[22:44] <ubitux> and cur_dts will be ` in that case? :/
[22:44] <michaelni> and their information like duration /sample rate can be random
[22:44] <ubitux> (`0 i mean)
[22:45] <ubitux> also, how come cur_dts==AV_NOPTS_VALUE means the timestamp or duration is known?
[22:45] <michaelni> cur_dts will be set to the timestamp if tehres one and will count upward if a duration can be found
[22:45] <durandal_1707> michaelni: i wonder why that neat filter is not yet ported
[22:47] <michaelni> ubitux, theres no code that sets cur_dts to AV_NOPTS_VALUE unless there was a known timestamp before
[22:47] <ubitux> is it reliable to assume this will always be true?
[22:49] <michaelni> durandal_1707, it would be great if someone ported it ...
[22:50] <michaelni> ubitux, you could exclude AV_NOPTS_VALUE from the check
[22:51] <ubitux> ok, last question: anything in particular i need to check when doing modification in that code?
[22:51] <ubitux> i'm not sure "passing fate" is enough :
[22:51] <ubitux> :p
[22:52] <michaelni> improve fate :)
[22:52] <ubitux> hehe sure, but what kind of checks should i add? :)
[22:57] <michaelni> dunno, but anything that failes and is fixed could get a fate test in theory
[22:58] <michaelni> that is a fate that for every ticket we fix from 1 to where we are now :)
[22:58] <michaelni> s/that/test/
[22:58] <ubitux> sounds like a good GSoC ;)
[22:59] <ubitux> a bit boring maybe :)
[23:32] <CIA-108> ffmpeg: 03Clément BSsch 07master * r7c0d30b57b 10ffmpeg/libavcodec/ass.c: ass: use av_asprintf() instead of a temporary stack buffer.
[23:32] <CIA-108> ffmpeg: 03Clément BSsch 07master * rcc7342f712 10ffmpeg/libavcodec/assdec.c: assdec: fix wrong alloc pointer check.
[23:32] <CIA-108> ffmpeg: 03Clément BSsch 07master * r6a3cc21242 10ffmpeg/libavformat/mp3dec.c:
[23:32] <CIA-108> ffmpeg: mp3dec: fix truncating packet error message while decoding a complete MP3.
[23:32] <CIA-108> ffmpeg: The MP3 demuxer split the data in packets of 1024B which are later split
[23:32] <CIA-108> ffmpeg: in MP3 frames by the MPEG audio parser. The last read is "truncated",
[23:32] <CIA-108> ffmpeg: but this should not raise any error.
[23:32] <CIA-108> ffmpeg: Solution-by: Michael Niedermayer
[23:54] <CIA-108> ffmpeg: 03Michael Bradshaw 07master * rf0d6828dbe 10ffmpeg/libavcodec/libopenjpegdec.c: Supoort decoding of gray16 images with libopenjpeg.
[00:00] --- Tue Jan 31 2012
1
0
[00:03] <Brance> rpm -i http://download.obs.j0ke.net/multimedia:/EL5/CentOS_5/i386/librtmp-devel-2.…
[00:03] <Brance> this is
[00:04] <Brance> the correct
[00:04] <burek> maybe the version is old
[00:04] <relaxed> perhaps the version is too old
[00:05] <Brance> what is the newest version of librtmp-devel?
[00:06] <burek> 2.3
[00:06] <Brance> can you please show me how to install it?
[00:06] <burek> http://rtmpdump.mplayerhq.hu/
[00:07] <burek> you'll need to compile it i guess
[00:15] <Brance> yes i did it
[00:16] <Brance> i was installing old version rtmpdump all time
[00:16] <Brance> thank you everyone for your time
[00:16] <JEEBsv> actually the current version of rtmpdump it shows is 2.4
[00:16] <JEEBsv> IIRC
[00:17] <JEEBsv> RTMPDump v2.4
[00:17] <JEEBsv> (c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL
[00:17] <JEEBsv> yup
[00:17] <JEEBsv> :3
[00:17] <JEEBsv> copyright is quite old tho
[00:17] <JEEBsv> lol
[00:32] <weekee> Hi folks, I need to mux a mov together so that the packet order is very specific, 8 audio packets followed by 15 video packets, etc. I'm trying to get a video that was produced externally to play on a panasonic camera. Is this possible? And if it's not, would someone code it for a bounty?
[00:33] <weekee> I originally though that "mov/mp4 chunking support (equivalent to mp4boxs -inter)" would do the trick, but it doesn't appear to
[00:33] <weekee> without using that, it outputs the packets as VAVAVAVA
[00:33] <weekee> using the chunking, it does it in what seems like a fairly random order
[00:33] <weekee> clusters of packets, but no discernible order
[00:34] <weekee> If someone can point me in the right direction, or would modify the MOV muxing code for a fee, I'd appreciate it
[00:35] <weekee> a fee or a donation to the project/freenode/whatever, just need it done :)
[01:13] <SIFTU> I'm trying to pillarbox a (640x480) 4:3 video to 16:9 (720p) can someone please help me understand the padding feature
[01:22] <weekee> you'd want to resize it and add padding at the left and right
[01:22] <weekee> resizing up to what, 960x720?
[01:23] <weekee> then -padleft 160 -padright 160 and i guess -padcolor 000000
[01:24] <burek> what do you want as your final resolution SIFTU ?
[01:25] <SIFTU> weekee: my version of ffmpeg does support -padleft -padright
[01:25] <SIFTU> burek: I'm converting to dvd
[01:25] <weekee> DVD? then it's not 720p
[01:25] <weekee> 720p is 720 vertical pixels
[01:26] <SIFTU> weekee: yeah sorry
[01:26] <SIFTU> figured that out just then :)
[01:26] <weekee> :)
[01:26] <burek> so SIFTU what do you want as your final resolution?
[01:27] <SIFTU> I have videos in 640x480 and 720p need to convert to DVD
[01:27] <weekee> ah, padx is deprecated
[01:27] <SIFTU> whats DVD 720x480?
[01:27] <burek> dvd is not a resolution
[01:27] <burek> its a format
[01:28] <weekee> you know what he means
[01:28] <SIFTU> well the standard resolution for a DVD
[01:28] <burek> no i dont know what does he mean that why im asking him for the precise resolution
[01:28] <weekee> that's the most common res for NTSC dvd yeah
[01:28] <burek> twice
[01:28] <SIFTU> burek: 720x480
[01:29] <burek> so you want to widen up your video or to pad it with black?
[01:29] <SIFTU> yeah the 640x480 ones
[01:29] <SIFTU> black bars left and right to make them 720x480
[01:29] <burek> i ask you which option (of those 2) you want, and you say: yes
[01:29] <burek> :)
[01:30] <burek> ok
[01:30] <weekee> he said pillarbox so he means the latter
[01:30] <burek> then you can use -vf
[01:30] <burek> just a sec, ill give you exact command line
[01:30] <SIFTU> burek: thank you
[01:33] <SIFTU> burek: I have to run but thank you in advance.. I'll read it in an hour or so
[01:34] <burek> ffmpeg -i input -vf "pad=720:480:40:0:black" -vcodec ... output
[01:34] <burek> http://ffmpeg.org/ffmpeg.html#pad
[01:50] <freezway> is there any completely uncompressed video format?
[01:50] <freezway> i know files will be huge
[01:50] <freezway> like not just lossless, but COMPLETELY uncompressed
[01:50] <maister> yuv4mpeg is one
[01:51] <freezway> coolk
[01:53] <burek> or yuyv
[01:53] <burek> ffmpeg -codecs | grep uncompress
[01:53] <burek> sorry, can't do like that :)
[01:54] <burek> +google ffmpeg uncompressed
[01:54] <burek> :)
[01:55] Action: burek slaps fflogger
[01:55] <burek> wakey wakey :)
[02:12] <MestreLion> hi there! Im not sure if this is normal, but... im in Linux Mint 10, and when i use ffmpeg in terminal, even with no arguments, it shows a huge dump of configuration options
[02:12] <relaxed> it's normal
[02:13] <MestreLion> can i change this weird level of verbosity? I mean... it takes like 2 terminal screens
[02:13] <burek> ?
[02:13] <burek> can you please use pastebin.com, to show your command line and its output?
[02:14] <MestreLion> http://pastebin.com/WYLcFVYC
[02:14] <burek> weird
[02:14] <MestreLion> (i swear i tried to use ffmpeg.pastebin.com .. but it seems to redirect to "normal" pastebin
[02:14] <relaxed> that's so stupid
[02:15] <MestreLion> ffmpeg was installed from repositories.. and I dont remember setting any config for verbosity
[02:15] <burek> just out of curiosity, did you type alias ?
[02:15] <relaxed> I think you should compile ffmpeg by hand because that's not the default behavior
[02:15] <burek> is there any weird ffmpeg alias.. ?
[02:15] <MestreLion> nope burek , its not aliased
[02:15] <burek> you have those options repeated 3-4 times
[02:16] <MestreLion> relaxed: that would be overkill... im a newbie, and id rather stick with repo versions so i dont screw my system, at least for now
[02:16] <burek> you could file a bug report to your distro
[02:16] <MestreLion> once for every lib, burek
[02:16] <burek> would you like to use a static binary? :D
[02:17] <MestreLion> burek: how would i know if i want that?
[02:17] <burek> well, its like windows exe file
[02:17] <burek> you download it and run
[02:17] <burek> and it works :D
[02:17] <burek> do you want it? :)
[02:17] <relaxed> MestreLion: look at loglevel in the man page
[02:18] <burek> https://sites.google.com/site/linuxencoding/builds
[02:18] <MestreLion> burek: i cannot choose if i want that or not, given that i have *no* idea of the implications of such decision
[02:18] <relaxed> MestreLion: you should file a bug report against the distro's ffmpeg package.
[02:18] <burek> MestreLion, you can safely download it and try
[02:19] <burek> no implication
[02:19] <burek> that's why it's called static.. it doesn't link with your system at all
[02:19] <MestreLion> thats the main reason I usually stick with the repos... or at the most an official PPA
[02:19] <relaxed> avconv doesn't even list the ./configure line anymore.
[02:19] <burek> it has all it needs to run (except glibc) :)
[02:21] <MestreLion> i guess i know what is the conflict
[02:21] <MestreLion> maybe its about that "WARNING: library configuration mismatch" ?
[02:22] <MestreLion> i mean... should options from all libs match the one in ffmpeg's ?
[02:22] <relaxed> probably
[02:27] <MestreLion> its a version mismatch... ffmpeg and friends were originally from medibuntu repo
[02:27] <MestreLion> and i guess a newer version popped up in ubuntu repo, which is more restrictive
[02:27] <MestreLion> but just for ffmpeg
[02:28] <MestreLion> what would be the best way to fix that?
[02:29] <burek> download a static binary
[02:30] <burek> or compile your ffmpeg
[02:30] <burek> since distro packages are mostly outdated anyway
[02:32] <relaxed> ubuntu has made it too easy to add external repos, which is never a good idea.
[02:34] <MestreLion> burek: but both compile or download would be outside of packaging system, correct? would i lose updates, fixes, etc?
[02:34] <burek> static
[02:35] <burek> is just like exe in windows
[02:35] <burek> download and run
[02:35] <burek> thats it
[02:35] <burek> no dependencies, no relation with your system at all
[02:35] <burek> just try it
[02:35] <MestreLion> ill give it a try, thank you
[02:36] <relaxed> I need to add a picture of Ron Popeil to the static build page with a catchy caption.
[02:36] <relaxed> Hmm, what would Ron say...
[02:41] <drv> but wait, there's more
[02:47] <MestreLion> wow, you're the author of those static builds, relaxed :)
[06:09] <ccherrett> http://pastebin.com/LfdJfrLN
[06:09] <ccherrett> can anyone tell me what this compile error might be
[06:09] <ccherrett> it is a slomo app that uses ffmpreg
[06:09] <ccherrett> ffmpeg
[06:09] <ccherrett> missing compile flag?
[08:48] <LexSfX> what's the PC range pixel format of yuv420p (since that one is TV range) to specify for libx264 in ffmpeg CLI?
[08:49] <LexSfX> is it yuvj420p?
[12:36] <undercash> hi
[12:36] <undercash> what s the command to merge several mp4 ?
[14:26] <mr_> quick question that can hopefully result in a quick answer: I'm encoding FRAPS files - mediainfo says the audio/video streams are of exact same length. After going through ffmpeg (or even x264 directly), both audio and video streams differ in length from the original by milliseconds no matter what codec is used -- upon concatenating the resulting mp4 containers, this results in a rather large desync drift - is there any way to remedy this? (hour long googling/doom9 rea
[17:00] <madsage> morning
[17:03] <madsage> i got ffpmeg and re-encoding working great with wowza using mpegts. nNow i need to figure out how to build a stream from multiple pieces of media with ffmpeg outputting them into the stream.
[17:03] <madsage> can anybody direct me to some examples or docs that may help?
[17:08] <madsage> looks like Tribler might be what i am looking for
[17:09] <madsage> or not
[17:12] <madsage> i want to queue content into the stream with ffmpeg somehow
[17:26] <mgeary> hey relaxed, i'm trying now to recompile ffmpeg from git (git pull), and i'm giving it this: ./configure --enable-libmp3lame --enable-gpl --enable-libx264 --enable-libvo-aacenc. It's telling me "libvo_aacenc is version3 and --enable-version3 is not specified"
[17:43] <chandoo> hi
[17:43] <chandoo> is it possible to convert animated.gif to jpeg images?
[17:45] <Tjoppen> try something like ffmpeg -i foo.gif -qscale 2 out%d.jpg
[17:48] <chandoo> Incompatible pixel format 'pal8' for codec 'mjpeg', auto-selecting format 'yuvj420p'
[17:49] <chandoo> tjholowaychuk:{
[17:49] <chandoo> Tjoppen:{
[17:52] <Tjoppen> you'll need to pastebin more output
[17:53] <chandoo> http://fpaste.org/0tRb/
[17:53] <chandoo> pls check that Tjoppen
[17:54] <Tjoppen> I don't see any error. seems like it output a single frame
[17:54] <Tjoppen> what does ffplay spiralAnimated.gif look like?
[17:55] <chandoo> yes just one image
[17:55] <Tjoppen> well there you go then
[17:55] <chandoo> i tried with rate 10 and 5 its same thing
[17:55] <chandoo> i am expecting few images since it is animation gif
[17:56] <chandoo> like spiral is winding inwards
[18:01] <Tjoppen> sounds like a gif decoder bug. file a ticket
[18:01] <chandoo> okay let me try with some other gif then
[18:05] <chandoo> same with other gif as well
[18:09] <Tjoppen> could you upload one somewhere?
[18:19] <SIFTU> Hi all, I'm trying to convert a videoo into mpeg2 but the audio isnt coming over Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt'
[18:24] <relaxed> SIFTU: it should be fine
[18:24] <SIFTU> relaxed: I dont get any audio on the output video
[18:25] <relaxed> command?
[18:25] <SIFTU> ffmpeg -y -i "9.AVI" -vcodec mpeg2video -vf "pad=720:480:40:0:black" -acodec ac3 -threads 0 -sameq 9.mpg
[18:26] <relaxed> remove -sameq and replace it with "-qscale 3 -f dvd 9.mp9"
[18:26] <relaxed> er, 9.mpg
[18:27] <SIFTU> relaxed: worked thank you!! now to change my bitrate to match the others
[18:28] <relaxed> "-qscale 3" replaces "-b $bitrate". Only set a bitrate if you need a specific size. And in that case use 2 pass encoding.
[18:29] <SIFTU> relaxed: ok, that does audio bitrate?
[18:30] <relaxed> no, for ac3 use -b:a $bitrate
[18:30] <SIFTU> I just set -ab 448k to match my other videos which seems to have worked
[18:30] <SIFTU> ok
[18:31] <SIFTU> relaxed: thanks for your help, i was fighting for hours and you solved it in 2 seconds
[18:31] <relaxed> and you didn't get audio playback earlier because you were muxing into a mpeg1 container.
[18:32] <relaxed> which "-f dvd" solved
[18:34] <SIFTU> relaxed: yeah I'm no video expert.. but where did my command suggest I was going to mpeg1? the extension?
[18:35] <relaxed> I believe so. You are one of many who have hit the same problem.
[19:01] <sagitter> hello !
[19:01] <sagitter> i'm trying to build ffmpeg helping myself with this page:
[19:02] <sagitter> http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide
[19:02] <sagitter> i use Fedora 16 64 bit
[19:04] <sagitter> but during "make" of ffmpeg, the compilation stops with error:
[19:06] <sagitter> collect2: ld returned 1 exit status
[19:07] <sagitter> at line: libavcodec/libavcodec.a(libx264.o): In function `X264_init':
[19:07] <sagitter> Can someone help me please ?
[19:07] <relaxed> does $(rpm -qa "*x264*") return anything?
[19:09] <sagitter> nothing
[19:10] <relaxed> you checked out the latest source for ffmpeg and libx264?
[19:12] <sagitter> Yes, i think; are the lastest from this page http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide ? Or no ?
[19:13] <sagitter> Is useful the x264-devel rpm, too ?
[19:13] <relaxed> pastebin the complete error
[19:14] <relaxed> what do you mean? is shouldn't be installed if that's what you're asking.
[19:14] <relaxed> /is/it/
[19:14] <sagitter> ok, the complete error is:
[19:14] <iive> not here
[19:15] <iive> put it on some pastebin site
[19:15] <sagitter> are only 4 lines :)
[19:18] <sagitter> complete: http://www.fpaste.org/0tGr/
[19:22] <relaxed> you followed that guide exactly?
[19:24] <sagitter> sure, i have only tried a second time adding "--prefix=/usr" to "./configure" but with the same result
[19:26] <relaxed> you shouldn't install software to the /usr prefix because that will conflict with your package manager.
[19:28] <sagitter> ok, the first time i followed exactly the guide in the webpage
[19:28] <relaxed> follow the "Reverting changes made by this guide" section and then pastebin the output of: find /usr -name "*x264*"
[19:29] <sagitter> for all or only for ffmpeg ?
[19:29] <relaxed> all
[19:30] <sagitter> wow !
[19:30] <sagitter> ok
[19:30] <relaxed> except "rm -rf ~/ffmpeg-source" and "yum erase gcc git make nasm pkgconfig wget"
[19:31] <sagitter> sure :)
[19:36] <sagitter> done
[19:36] <sagitter> but the find output is too long
[19:42] <lobster> Hi, I'm trying to figure out how to scale down videos of different resolutions while (obviously) maintaing aspect ratio. For example a max width of 1000 and a max width of 500. So a 2000x500 vid would be scaled to 1000x250 and a 500x1000 would be scaled to 250x500 while any video within the boundary wouldn't be modified. Is this possible? The -vf scale options are quite confusing :)
[19:44] <sagitter> http://www.fpaste.org/0wEv/
[19:44] <sagitter> i used find /usr | grep x264
[19:48] <cbreak> find supports parameters
[19:48] <cbreak> you can tell it what you are searching for :)
[19:50] <relaxed> sagitter: are you sure you don't have x264-devel installed?
[19:54] <sagitter> cbreak, i paste the right "find" output :http://www.fpaste.org/jCiv/
[19:55] <relaxed> answer my question
[19:55] <sagitter> relaxed, x264-devel is not installed
[19:56] <cbreak> your paste indicates that your system has x264
[19:56] <sagitter> must be it ?
[19:56] <sagitter> right, sorry
[19:57] <sagitter> x264-devel is installed
[19:57] <cbreak> the /usr level is usually reserved for the system
[19:57] <cbreak> the /usr/local level for user installed things
[19:57] Action: relaxed yum erases sagitter
[19:57] <sagitter> :)
[19:59] <sagitter> so ? :)
[20:01] <relaxed> did you uninstall it?
[20:07] <sagitter> no
[20:08] <relaxed> uninstall it, "make distclean" in all the source dirs and start over
[20:09] <sagitter> ok, try again, thank you
[20:09] <relaxed> the guide should have mentioned that
[20:10] <sagitter> in guide: "yum erase x264 ffmpeg", no x264-devel :)
[20:11] <relaxed> yes, I'll see that it's corrected
[20:24] <sagitter> done ! Compiled and installed
[20:32] <sagitter> thank you
[20:57] <pasteeater> i updated the guide
[21:36] <madsage> relaxed, when i uninstalled x264 it also uninstalled the devel libs. as they are no longer needed without x264. worked this way with CentOS 5.7 x64
[21:37] <madsage> i also had no problems, following your document. worked great. I went one step furtehr and also included libaccplus
[21:37] <madsage> ffmpeg version N-37346-g371946b
[21:42] <madsage> anybody tried OpenCaster?
[21:43] <madsage> i need something to help put content together for live streaming
[21:56] <pasteeater> madsage: also see http://www.ob-encoder.com/
[21:57] <pasteeater> as in "guide" i mean the centos on on trac...
[21:57] <madsage> oh very cool. thank you for that. think i will try that first
[22:57] <indigo> hi
[22:58] <indigo> finally I succeded in compiling ffmpeg for win32
[22:59] <indigo> I'm now using a Fedora since it has mingw32-runtime 3.18
[22:59] <indigo> all was fine and I move the compiled files to a windows machine and they work!
[23:00] <indigo> now I'm trying to re-compile with libopenjpeg
[23:00] <indigo> I also added --extra-cflags="-I/usr/i686-pc-mingw32/sys-root/mingw/include/"
[23:01] <indigo> and --extra-ldflags='-L/usr/i686-pc-mingw32/sys-root/mingw/lib/'
[23:01] <indigo> but STILL the message "ERROR: libopenjpeg not found" is showing up!
[23:02] <indigo> I previously tryed on Ubuntu, Debian, CentOS and Slacware with no luck
[23:02] <indigo> *Slackware
[23:03] <indigo> this is the first time I'm able to cross-compile ffmpeg
[23:03] <indigo> I'm using the daily snapshot
[23:04] <indigo> is there anyone that can help me?
[23:04] <relaxed> you installed libopenjpeg with --prefix=/usr/i686-pc-mingw32/sys-root/mingw ?
[23:04] <indigo> is there anyone that can help me to compile ffmpeg with libopenjpeg?
[23:04] <indigo> ?
[23:05] <indigo> I installed the package mngw32-libopenjpeg resident on the fedora repositories
[23:05] <indigo> and I can found it there
[23:05] <relaxed> look at config.log and see why it's not working
[23:06] <relaxed> maybe there's a mingw32-libopenjpeg-devel package?
[23:06] <indigo> I'll go and check
[23:07] <indigo> these are the last lines of the config.log:
[23:07] <indigo> i686-pc-mingw32-gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -U__STRICT_ANSI__ -I/usr/i686-pc-mingw32/sys-root/mingw/include/ -std=c99 -fno-common -fomit-frame-pointer -c -o /tmp/ffconf.TPzoafmz.o /tmp/ffconf.ojcBMRXQ.c
[23:07] <indigo> i686-pc-mingw32-gcc -L/usr/lib/ -Wl,--as-needed -o /tmp/ffconf.isd33QMV.exe /tmp/ffconf.TPzoafmz.o -lopenjpeg -lavifil32 -lm -lpthreadGC2 -lpsapi
[23:07] <indigo> /tmp/ffconf.TPzoafmz.o:ffconf.ojcBMRXQ.c:(.text+0xc): undefined reference to `_opj_version'
[23:07] <indigo> collect2: ld returned 1 exit status
[23:07] <indigo> ERROR: libopenjpeg not found
[23:07] <plut0> how do i copy all source streams in the output?
[23:07] <relaxed> indigo: use pastebin.com from now on
[23:08] <relaxed> plut0: -c copy
[23:08] <plut0> relaxed: if theres multiple streams it doesn't copy them all
[23:08] <indigo> sorry, aright
[23:10] <indigo> I found libopenjpeg-devel and installed
[23:11] <relaxed> plut0: -c:v copy -c:a copy -c:s copy -c:t copy
[23:11] <indigo> have I to compile for mingw32?
[23:12] <plut0> relaxed: unrecognized option c:a
[23:12] <relaxed> indigo: now run "make distclean" and run configure, make
[23:13] <relaxed> plut0: change all the -c: to -codec:
[23:13] <relaxed> which version are you using?
[23:13] <plut0> relaxed: 0.7.8
[23:14] <indigo> still having the same error message
[23:14] <indigo> maybe it's better I download the source and recompile libopenjpeg
[23:15] <indigo> which files have to be present? libopenjpeg.h or what?
[23:17] <relaxed> plut0: ffmpeg -h 2>&1| grep "^-[a-z]codec"
[23:17] <plut0> relaxed: what are you looking for?
[23:18] <plut0> relaxed: i've always done -scodec copy but that only copies one stream
[23:19] <relaxed> try, ffmpeg -i input -map 0 -c copy output
[23:20] <relaxed> that works for me with avconv
[23:21] <indigo> relaxed, if I download the source have I to compile the libs for mingw32 or for my distro?
[23:21] <relaxed> plut0: also works with recent ffmpeg
[23:22] <relaxed> indigo: I would assume mingw32. Look in the forums here for tips --> http://ffmpeg.zeranoe.com/builds/
[23:22] <indigo> thanks
[23:22] <indigo> I have to go
[23:22] <indigo> thanks a lot
[23:22] <indigo> bye
[23:23] <plut0> relaxed: i don't have a -c command
[23:23] <plut0> relaxed: theres no -c flag rather
[23:23] <relaxed> -codec copy
[23:24] <plut0> relaxed: do you mean -scodec ?
[23:24] <relaxed> If -codec copy doesn't work, then yes. look at $(ffmpeg -h 2>&1| grep "^-[a-z]codec")
[23:25] <plut0> number of stream maps must match number of output streams
[23:26] <relaxed> this would be real easy if you had a recent version of ffmpeg.
[23:26] <plut0> relaxed: i had to downgrade due to a few bugs
[23:27] <relaxed> psatebin the output of ffmpeg -i file
[23:28] <plut0> relaxed: http://pastebin.com/rL7pq9Ew
[23:30] <pasteeater> plut0: what bugs?
[23:30] <relaxed> ffmpeg -i input -map 0.0 -vcodec copy -map 0.1 -acodec copy -map 0.2 -scodec copy -map 0.3 -scodec copy -map 0.4 -scodec copy output
[23:34] <plut0> pasteeater: video was all f'ed up unless i disabled b-frames, and another one that escapes my mind at the moment
[23:35] <plut0> relaxed: still getting "number of stream maps must match number of output streams"
[23:36] <pasteeater> plut0: how can i duplicate your issue(s)?
[23:37] <plut0> pasteeater: i was on 0.9.1 encoding a flv as libx264
[23:39] <plut0> so frustrating
[23:39] <relaxed> plut0: last try:
[23:40] <relaxed> ffmpeg -i input -map 0.0 -vcodec copy -map 0.1 -acodec copy -map 0.2 -scodec copy OUTPUT-HERE -map 0.3 -scodec copy -map 0.4 -scodec copy
[23:41] <plut0> relaxed: nope
[23:42] <plut0> pasteeater: oh the other issue was, i did a -acodec copy for nellymoser audio and the output file had no audio
[23:43] <relaxed> I asked you to file a bug report on that. Being lazy isn't going to see that it's fixed :/
[23:52] <pasteeater> plut0: do you at least have command line examples that I can try?
[23:52] <plut0> relaxed: i got it working
[23:53] <plut0> pasteeater: ffmpeg -i source.flv -vcodec libx264 -crf 26 -preset veryfast -acodec copy -scodec copy -threads 0 output.flv
[23:53] <plut0> pasteeater: source audio was nellymoser
[23:53] <pasteeater> thanks. i'll take a look at it later
[23:54] <pasteeater> ...and a sample input would be nice
[23:54] Action: pasteeater leaves for a while
[00:00] --- Tue Jan 31 2012
1
0
[00:58] <CIA-108> ffmpeg: 03Mans Rullgard 07master * raac46e088d 10ffmpeg/libavcodec/ (Makefile aacsbr.c aacsbrdata.h sbr.h sbrdsp.c sbrdsp.h):
[00:58] <CIA-108> ffmpeg: aacsbr: move some simdable loops to function pointers
[00:58] <CIA-108> ffmpeg: This prepares for assembly optimisations by moving the most
[00:58] <CIA-108> ffmpeg: time-consuming loops to functions called through pointers
[00:58] <CIA-108> ffmpeg: in a new context.
[00:58] <CIA-108> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com>
[00:58] <CIA-108> ffmpeg: 03Mans Rullgard 07master * r8996ed2b73 10ffmpeg/libavcodec/ (aacsbr.c aacsbrdata.h sbr.h):
[00:58] <CIA-108> ffmpeg: aacsbr: align some arrays
[00:58] <CIA-108> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com>
[00:58] <CIA-108> ffmpeg: 03Alex Converse 07master * r7181c4edee 10ffmpeg/ (91 files in 6 dirs): cosmetics: Remove extra newlines at EOF
[00:58] <CIA-108> ffmpeg: 03Mans Rullgard 07master * rbe822d77b6 10ffmpeg/libavcodec/ (5 files in 2 dirs):
[00:58] <CIA-108> ffmpeg: aacsbr: ARM NEON optimised sbrdsp functions
[00:58] <CIA-108> ffmpeg: Overall speedup of HE-AAC decoding 2.3x on Cortex-A8, 1.2x on A9.
[00:58] <CIA-108> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com>
[00:58] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r2d9535ad31 10ffmpeg/libavcodec/utils.c:
[00:58] <CIA-108> ffmpeg: avcodec_align_dimensions2: set only 4 linesizes, not AV_NUM_DATA_POINTERS.
[00:58] <CIA-108> ffmpeg: This function is video-only, so there's no point in setting more
[00:58] <CIA-108> ffmpeg: linesizes.
[00:58] <CIA-108> ffmpeg: Fixes stack corruption in avplay.
[00:58] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r44911f2985 10ffmpeg/doc/APIchanges:
[00:59] <CIA-108> ffmpeg: There was no minor bump for making avcodec_alloc_context3() public and
[00:59] <CIA-108> ffmpeg: deprecating the other two, so I'm using the first next lavc bump.
[00:59] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r9bfe218299 10ffmpeg/libavcodec/avcodec.h: lavc: extend doxy for avcodec_alloc_context3().
[00:59] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r91b3bfba88 10ffmpeg/avplay.c:
[00:59] <CIA-108> ffmpeg: avplay: use the correct array size for stride.
[00:59] <CIA-108> ffmpeg: AV_NUM_DATA_POINTERS instead of 4.
[00:59] <CIA-108> ffmpeg: 03Nathan Caldwell 07master * r2e626dd513 10ffmpeg/libavcodec/aacenc.c:
[00:59] <CIA-108> ffmpeg: aacenc: Fix LONG_START windowing.
[00:59] <CIA-108> ffmpeg: Forgot to add the equivalent amount to the incoming sample pointer as the output pointer.
[00:59] <CIA-108> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net>
[01:00] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rc065255bba 10ffmpeg/: (log message trimmed)
[01:00] <CIA-108> ffmpeg: Merge remote-tracking branch 'qatar/master'
[01:00] <CIA-108> ffmpeg: * qatar/master:
[01:00] <CIA-108> ffmpeg: aacenc: Fix LONG_START windowing.
[01:00] <CIA-108> ffmpeg: aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
[01:00] <CIA-108> ffmpeg: avplay: use the correct array size for stride.
[01:00] <CIA-108> ffmpeg: lavc: extend doxy for avcodec_alloc_context3().
[01:00] <CIA-108> ffmpeg: 03Nathan Caldwell 07master * rdc7e7d4dd9 10ffmpeg/libavcodec/aacenc.c:
[01:00] <CIA-108> ffmpeg: aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
[01:00] <CIA-108> ffmpeg: 10l: Forgot to adjust deinterleave for new location of incoming samples in 7946a5a.
[01:00] <CIA-108> ffmpeg: This produced incorrect, but surprisingly listenable results.
[01:01] <CIA-108> (2 lines omitted)
[02:18] <durandal_1707> some headers still heave deprecated stuff which is pointless because required code is gone
[02:53] <CIA-108> ffmpeg: 03Paul B Mahol 07master * rcbf1dc4eb4 10ffmpeg/tests/ (codec-regression.sh ref/vsynth1/y41p ref/vsynth2/y41p):
[02:53] <CIA-108> ffmpeg: fate: add Y41P encoding/decoding test
[02:53] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[02:53] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[02:53] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r6bcc8275a1 10ffmpeg/libavcodec/r210enc.c:
[02:53] <CIA-108> ffmpeg: r210enc: fix encoding for unaligned widths
[02:53] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[02:53] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[02:53] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r931ec4a2f6 10ffmpeg/tests/ (codec-regression.sh ref/vsynth1/r210 ref/vsynth2/r210):
[02:53] <CIA-108> ffmpeg: fate: add R210 encoding/decoding test
[02:53] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[02:53] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[03:31] <durandal_1707> hmm r210 fails on bigendian
[03:32] <durandal_1707> probably because test needs explicit rgb48le?
[03:39] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r9719528e05 10ffmpeg/tests/codec-regression.sh:
[03:39] <CIA-108> ffmpeg: fate: fix r210 test on big endian
[03:39] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[03:39] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[07:20] <funman> FYI doc/APIchanges is broken http://git.videolan.org/?p=vlc.git;a=commitdiff;h=5b55c4dbab0ee2700300f9d8b…
[07:59] <michaelni> funman, never trust APIchanges
[08:00] <michaelni> noone is maintaining it sadly
[08:01] <michaelni> always use git log / git blame to find at which revission things where introduced
[08:04] <funman> i found e.g. that commit 526604545fb1cc0c11af356fbffd5cddf8cdc95f introduced it
[08:04] <funman> but git describe fails on it
[08:04] <funman> and git describe --tags says nothing useful
[08:10] <funman> in vlc or rockbox i see e.g. 1.3.0-git-895-g4ca6ef0
[08:10] <funman> last tag - number of commits since -gHASH
[08:11] <nevcairiel> the non-linear history because of all the merges terribly confuses git describe
[08:12] <funman> it confused me as well :/
[08:13] <funman> write to lkml and suggest to Linus that his tool sucks
[08:14] <funman> hop, fixed in 2 days =)
[08:39] <michaelni> funman, try:
[08:39] <michaelni> git describe --tags --match N `git rev-list 52660454..HEAD --ancestry-path --merges | tail -1`
[08:39] <michaelni> dunno if theres a easier way ...
[08:40] <michaelni> s/HEAD/master/ probably
[08:42] <funman> nope
[09:16] <michaelni> well the command i suggested should return the merge commit that brought the change in
[09:17] <michaelni> and looking into its version.h files should show the corresponding version
[09:17] <michaelni> but maybe i misunderstood
[09:18] Action: michaelni goes to bed
[10:41] <vtorri> when compiling on windows, there are several warnings about printf, conversion characters that are unknown
[10:41] <vtorri> i would like to define some FMT_XXX, which have some values on windows, other on linux, to remove these warnings
[10:42] <vtorri> is there a "private" header where i can define them, and which is included by all the source files ?
[10:43] <funman> you use mingw?
[10:43] <vtorri> another question: git diff is sufficient for you to create a patch, or does one use someting else
[10:43] <funman> you should use -D__USE_MINGW_ANSI_STDIO=1 to have mingw use posix/c99 printf format
[10:43] <vtorri> funman: it's not important, as the libc of microsoft i used
[10:43] <nevcairiel> git format-patch is usually the better choice
[10:44] <vtorri> funman: not with visual studio
[10:44] <vtorri> nevcairiel: ok
[10:44] <funman> why building with visual studio?
[10:44] <vtorri> funman: we already had that discussion :)
[10:44] <funman> ah right
[10:44] <funman> =)
[10:44] <nevcairiel> ffmpeg isnt designed to build with visual studio anyway
[10:44] <vtorri> ha ?
[10:45] <nevcairiel> all the C99 features used will just blow up in your face
[10:45] <ubitux> vtorri: for your git diff question: git commit and use git format-patch -1 to generate a correct patch
[10:45] <vtorri> well, then FATE should be configured with that preprocessor macro :)
[10:45] <funman> vtorri: USE_MINGW_ANSI_STDIO is mingw specific
[10:46] <funman> ah there's a mingw fate?
[10:46] <vtorri> fate is using mingw-w64
[10:46] <vtorri> well, ffmpeg one
[10:46] <funman> yeah i see it
[10:46] <funman> though it says x86_64 mingw32 :P
[10:46] <ubitux> removing -fPIC from mingw build would be nice too btw :)
[10:47] <ubitux> the build system is quite complex to deal with though
[10:47] <nevcairiel> whats wrong with PIC?
[10:47] <ubitux> nevcairiel: http://fate.ffmpeg.org/log.cgi?time=20120129065158&log=compile&slot=x86_64-…
[10:47] <nevcairiel> except on x64 builds, it crys that all code is PIC enabled. :p
[10:47] <nevcairiel> but not so on x86 builds
[10:49] <vtorri> wher should I put the patch (it only fixes some warnings) ?
[10:49] <nevcairiel> i suppose having configure smart enough to set that based on mingw and which arch would be good, but configure is meh to work on
[10:49] <vtorri> where*
[10:49] <ubitux> vtorri: ffmpeg-devel ml
[10:49] <vtorri> arg
[10:50] <vtorri> i didn't want to subscribe :/
[10:50] <ubitux> you don't need to
[10:50] <ubitux> but you will have to wait for moderation
[10:50] <vtorri> ho
[10:50] <funman> vtorri: you can subscribe
[10:50] <vtorri> good enough for me :)
[10:50] <ubitux> (and you'll need to specify in the mail you want to be CC)
[10:50] <funman> then after you get your pass go to mailman management
[10:50] <ubitux> http://ffmpeg.org/contact.html btw, read the first two paragraph
[10:50] <funman> and disable 'Mail delivery'
[10:56] <vtorri> sent
[10:57] <vtorri> funman: USE_MINGW_ANSI_STDIO is mingw or mingw-w64 specific ?
[11:01] <funman> mingw
[11:02] <vtorri> ok
[14:17] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r05741d70c7 10ffmpeg/ (22 files in 3 dirs): (log message trimmed)
[14:17] <CIA-108> ffmpeg: Fallback to input timestamps for non-delay encoders.
[14:17] <CIA-108> ffmpeg: Causes FFmpeg to pass through the correct pts values,
[14:17] <CIA-108> ffmpeg: instead of clobbering all to AV_NOPTS_VALUE (the av_init_packet
[14:17] <CIA-108> ffmpeg: default) to then make up new ones based on only fps when muxing.
[14:17] <CIA-108> ffmpeg: Included are also the related FATE ref changes, which all
[14:17] <CIA-108> ffmpeg: some reasonable on quick investigation.
[14:17] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r6a9b565e0a 10ffmpeg/ (3 files in 2 dirs):
[14:17] <CIA-108> ffmpeg: FRAPS: Do not needlessly use reget_buffer.
[14:17] <CIA-108> ffmpeg: Codec has only I- and skip-frames, so there is no
[14:17] <CIA-108> ffmpeg: need for reget_buffer, change it so it works with
[14:17] <CIA-108> ffmpeg: get_buffer.
[14:17] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[14:22] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r95e873bb14 10ffmpeg/libavcodec/fraps.c:
[14:22] <CIA-108> ffmpeg: fraps: fix indentation.
[14:22] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[14:42] <ubitux> avctx->subtitle_header = av_malloc(avctx->extradata_size);
[14:42] <ubitux> if (!avctx->extradata)
[14:42] <ubitux> return AVERROR(ENOMEM);
[14:42] <ubitux> huh?
[14:43] <nevcairiel> looks like someone copy pasted too much. :)
[14:48] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * rf4e8292eb7 10ffmpeg/libavcodec/fraps.c:
[14:48] <CIA-108> ffmpeg: fraps: Add release_buffer forgotten when reget_buffer was removed.
[14:48] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[14:50] <durandal_1707> ubitux: who is goint to post patch?
[14:51] <ubitux> i have a local patch
[14:51] <durandal_1707> then post it
[14:51] <ubitux> i'm checking some stuff before sending it
[14:51] <ubitux> yes don't worry, i'll post it :)
[14:53] <CIA-108> ffmpeg: 03Paul Kendall 07master * r7df9937fcc 10ffmpeg/libavcodec/dvbsubdec.c: Fix dvb subtitle decoding when display segment is missing.
[16:27] <ubitux> i'm a bit lost with st->codec->extradata; it seems to be allocated in various places (ffmpeg.c, codec, demuxer), but also free'd from various places (codec, demuxer, lavf utils in the format close, ...)
[16:27] <ubitux> how is it supposed to work?
[16:29] <nevcairiel> th demuxer sets and frees it, decoders just read it but never change it, encoders set and free it, muxers just read it and never change it
[16:30] <nevcairiel> that should be the essence of it
[16:30] <ubitux> libavcodec/libx264.c: av_freep(&avctx->extradata);
[16:30] <ubitux> what about this one?
[16:30] <nevcairiel> its an encoder, isnt it. :)
[16:31] <ubitux> oh sorry.
[16:31] <ubitux> then, why:
[16:31] <ubitux> libavcodec/utils.c- if (codec_is_encoder(avctx->codec))
[16:31] <ubitux> libavcodec/utils.c: av_freep(&avctx->extradata);
[16:31] <ubitux> ?
[16:31] <ubitux> handle the encoders which don't free it?
[16:31] <nevcairiel> so that if an encoder creates it but doesnt free it itself, it gets freed?
[16:32] <ubitux> if there is such thing, why not just drop all the av_freep from the encoders?
[16:32] <nevcairiel> who knows
[16:32] <nevcairiel> x264 can have varying extradata in the same stream, maybe its somehow ralted to that?
[16:32] <nevcairiel> related*
[16:33] <nevcairiel> not sure x264 generates such streams, but the format is capable. :)
[16:34] <ubitux> "demuxer sets and frees it [...] encoders set and free it"
[16:34] <nevcairiel> whoever creates the data is also responsible for freeing it again
[16:34] <ubitux> do you have examples of use case for the two situations?
[16:35] <ubitux> the demuxer sets it to transmit format data to the encoders, and sometimes there is a need of transmitting data from the decoder to the demuxer?
[16:36] <ubitux> and if so, can it be possible to do both without "conflicts"?
[16:36] <nevcairiel> the demuxer sets it to transmit it to the decoders, because there its needed
[16:36] <nevcairiel> a demuxer and a encoder will possibly never work on the same format context
[16:36] <ubitux> oh wait i mixed up things with decoders.
[16:37] <ubitux> i need to read more carefully again; sorry.
[16:37] <nevcairiel> you demux, decode, encode, mux, the first two can share a context, and the second two can
[16:37] <nevcairiel> everything else will possibly result in trouble. :)
[16:57] <ubitux> ok, it makes sense now, thank you nevcairiel
[17:35] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r7fabef1f01 10ffmpeg/libavcodec/fraps.c:
[17:35] <CIA-108> ffmpeg: fraps: optimize pseudo-YUV to RGB conversion.
[17:35] <CIA-108> ffmpeg: With gcc 4.6 this part of the code is ca. 4x faster, resulting
[17:35] <CIA-108> ffmpeg: in an overall speedup of around 5% for fate-fraps-v5 sample.
[17:35] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[17:54] <CIA-108> ffmpeg: 03Carl Eugen Hoyos 07master * ra915618a29 10ffmpeg/libavcodec/wavpack.c:
[17:54] <CIA-108> ffmpeg: Improve decoding quality for lossy wavpack.
[17:54] <CIA-108> ffmpeg: This reverts e6e7bfc1 and 365e1ec2.
[17:54] <CIA-108> ffmpeg: The code may be incorrect both before and after the revert, but we
[17:54] <CIA-108> ffmpeg: do not have any samples that were fixed by the original commits.
[17:54] <CIA-108> ffmpeg: Fixes ticket #871.
[18:03] <CIA-108> ffmpeg: 03Carl Eugen Hoyos 07release/0.10 * rb2f27d2926 10ffmpeg/libavcodec/wavpack.c:
[18:03] <CIA-108> ffmpeg: Improve decoding quality for lossy wavpack.
[18:03] <CIA-108> ffmpeg: This reverts e6e7bfc1 and 365e1ec2.
[18:03] <CIA-108> ffmpeg: The code may be incorrect both before and after the revert, but we
[18:03] <CIA-108> ffmpeg: do not have any samples that were fixed by the original commits.
[18:03] <CIA-108> ffmpeg: Fixes ticket #871.
[18:03] <CIA-108> ffmpeg: (cherry picked from commit a915618a29f3f4197832151a4ed03ccdd585f9cf)
[18:45] <CIA-108> ffmpeg: 03Peter Ross 07master * r15d838b745 10ffmpeg/libavformat/bintext.c:
[18:45] <CIA-108> ffmpeg: bintext: use private options now that AVFormatParameters has been removed
[18:45] <CIA-108> ffmpeg: Reviewed-by: Stefano Sabatini <stefasab(a)gmail.com>
[18:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[18:45] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r8e46e12222 10ffmpeg/tests/ (codec-regression.sh ref/vsynth1/yuv4 ref/vsynth2/yuv4):
[18:45] <CIA-108> ffmpeg: fate: add yuv4 encoding/decoding test
[18:45] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[18:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[18:45] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r668a0b152b 10ffmpeg/libavcodec/libvpxenc.c:
[18:45] <CIA-108> ffmpeg: libvpxenc: update after FF_API_X264_GLOBAL_OPTS removal
[18:45] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[18:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[18:45] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r3b93a524c2 10ffmpeg/tests/ (codec-regression.sh ref/vsynth1/v308 ref/vsynth2/v308):
[18:45] <CIA-108> ffmpeg: fate: add v308 encoding/decoding test
[18:45] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[18:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[19:17] <durandal_1707> why is openbsd's fate failing in configuration?
[19:34] <durandal_1707> r210 failing in valgrind case could mean only one thing: we need to write explicitly write zeroes when padding width
[19:35] <kierank> why
[19:36] <kierank> we don't do it in v210 iirc
[19:39] <durandal_1707> kierank: there is memset
[19:39] <kierank> oh
[21:08] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r371946bc27 10ffmpeg/ (4 files in 4 dirs):
[21:08] <CIA-108> ffmpeg: r210enc: don't write uninitialized data
[21:08] <CIA-108> ffmpeg: Also fix r210 fate decoding test.
[21:08] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[21:08] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:26] <burek> does anyone know what are these compilation errors related to: http://pastebin.com/a0HE7i1z
[21:30] <durandal_1707> i never got that rtsp warning when compiling static
[21:30] <durandal_1707> but i'm not using linux....
[21:30] <burek> :D
[21:31] <iive> libfontconfig.a
[21:31] <iive> it seems to use some other lib for xml parsing, and that one is missing.
[21:32] <burek> hmh.. I've tried apt-file search XML_GetCurrentLineNumber but it didn't give me any results
[21:32] <burek> I'll try google
[21:33] <iive> that's function not library
[21:33] <burek> yes, but I don't know the name of the lib
[21:33] <burek> seems like libexpat
[21:33] <iive> check `pkg-config --libs fontconfig`
[21:33] <durandal_1707> by build of shared fontconfig depends on : libfreetype, libz, libc and libexpat
[21:34] <durandal_1707> you have sttix libexpat?
[21:34] <durandal_1707> s/sttix/static
[21:34] <burek> # pkg-config --libs fontconfig
[21:34] <burek> -lfontconfig
[21:34] <iive> --cflags ?
[21:35] <burek> umm didn't modify cflags
[21:35] <burek> only ldflags
[21:35] <burek> durandal_1707, I'll download/compile libexpat
[21:35] <burek> the rest I have
[21:35] <iive> add --static to the --libs
[21:36] <burek> -lfontconfig -lexpat -lfreetype -lz
[21:37] <durandal_1707> michaelni: why is openbsd failing?
[21:40] <funman> as a project?
[21:40] <durandal_1707> funman: lol, noo
[21:50] <durandal_1707> maybe it is sh bug in openbsd...
[21:53] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r3469c88804 10ffmpeg/libavcodec/fraps.c:
[21:53] <CIA-108> ffmpeg: fraps: Minor simplification, use local variable.
[21:53] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[21:53] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r0efdb942a6 10ffmpeg/libavcodec/fraps.c:
[21:53] <CIA-108> ffmpeg: fraps: Deduplicate some code.
[21:53] <CIA-108> ffmpeg: Also moves it before the get_buffer call so that most error exits
[21:53] <CIA-108> ffmpeg: happen before it.
[21:53] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[21:53] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * rf9eb622944 10ffmpeg/libavcodec/fraps.c:
[21:53] <CIA-108> ffmpeg: Fix offset validity checks.
[21:53] <CIA-108> ffmpeg: Offsets are relative to the end of the header, not the
[21:53] <CIA-108> ffmpeg: start of the buffer, thus the buffer size needs to be subtracted.
[21:53] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[21:53] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * rcd3ced1bb9 10ffmpeg/libavcodec/fraps.c:
[21:53] <CIA-108> ffmpeg: fraps: frame threading support.
[21:53] <CIA-108> ffmpeg: Codec is too simple to gain much from it at lower resolutions,
[21:53] <CIA-108> ffmpeg: but should help at very high resolutions, particularly for
[21:53] <CIA-108> ffmpeg: v3 and v5 where a not too optimized pseudo-YUV to RGB
[21:53] <CIA-108> ffmpeg: is done in the codec.
[21:53] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[21:57] <nevcairiel> and just a few weeks ago someone complained to me that his fraps recordings were not decoding fast enough, that gotta shut them up
[21:57] <durandal_1707> nevcairiel: you mean doom9 thread?
[21:57] <nevcairiel> yes
[21:58] <nevcairiel> thats usually where all the complaints pile up =p
[22:14] <michaelni> someone (who regularly reads doom9) should open tickets for all complaints found there
[22:15] <durandal_1707> hmm AV_PIX_FMT_ABI_GIT_MASTER is not in history
[23:04] <CIA-108> ffmpeg: 03Alexander Strasser 07master * r90bf7c7b41 10ffmpeg/configure: build: configure: Restore alphabetical order for CMDLINE_SET
[23:11] <durandal_1707> why x262?
[23:12] <Daemon404> [15:50] < durandal_1707> maybe it is sh bug in openbsd... <-- dont tell theo
[23:12] <Daemon404> OpenBSD is Perfect (TM)
[23:14] <durandal_1707> 4.8 is pretty old, there is 4.9
[23:14] <Daemon404> i dont really know who even runs openbsd realistically
[23:14] <Daemon404> :/
[23:14] Action: cbsrobot does
[23:15] <ohsix> he doesn't know you bro
[23:15] <ohsix> better watch out, things can get sketchy!
[23:15] <Daemon404> does it have non-fail SMP yet?
[23:15] <cbsrobot> not sure - it's all a bit slower but at least theres not a zillion changes afterwards
[23:16] <Daemon404> and does it support amd64 yet? lol
[23:16] Action: Daemon404 recalls openbsd was a bit... slow on that front
[23:16] <cbsrobot> let me check my uptime
[23:16] <Daemon404> lol
[23:16] <durandal_1707> openbsd is not about performance
[23:16] <Daemon404> i know
[23:17] <cbsrobot> 68 days ... on amd64
[23:17] <Daemon404> openbsd is about code audition and security
[23:17] <Daemon404> it's produced great software
[23:17] <Daemon404> i just dont like its kernel :P
[23:18] <cbsrobot> politicas aside but they made some nice tools aswell
[23:18] <cbsrobot> openssh, pf, ntpd are quite nice
[23:19] <cbsrobot> durandal_1707: 4.9 is old - theres 5.0
[23:19] <Daemon404> cbsrobot, yes
[23:19] <cbsrobot> and durandal_1707 x262 = mpeg 2
[23:19] <cbsrobot> x264 = mpeg 4
[23:19] Action: Daemon404 sort of wonders why people still use mpeg-2
[23:20] <cbsrobot> DVD ?
[23:20] <JEEBsv> cbsrobot: for future reference, MPEG-4 has multiple video codecs .-.
[23:20] <ohsix> it might be all those broadcast standards and hardware
[23:20] <JEEBsv> so you might want to specify which you are referencing
[23:20] <ohsix> part 10
[23:20] <cbsrobot> JEEBsv: well - thats just how I make it up in my brain ...
[23:21] <Daemon404> [17:20] < ohsix> it might be all those broadcast standards and hardware <-- pretty much every in-use standard allows avc iirc
[23:21] <Daemon404> tho existign hardware and $$ are plausible reasons
[23:21] <ohsix> gj
[23:22] <ohsix> unless you need to worry about bandwidth (eg. a cable provider) you don't need to be very nimble with backhaul and stuff, unless it saves you more money than the equipment you'd buy to adapt it
[23:22] <Daemon404> i did just say money was a valid reason
[23:22] <ohsix> and for broadcast you have to not use a bunch of neat mpeg4 features
[23:23] <Daemon404> not sure how thats relevant...
[23:25] <kierank> existing hw is the reason
[23:26] <Daemon404> yea thats what i said/figured
[23:26] <ohsix> surely "mpeg4 is awesome" and "mpeg2 sucks i don't know why they use it" implies some use of the features the former offer over the latter
[23:26] <Daemon404> do you do anything besides troll?
[23:27] <ohsix> you said you don't know why people use mpeg2, i guess you can continue to elaborate; but i pretty much already knew what you meant when you said it
[23:27] <durandal_1707> back to the topic: what's wrong with ffmpeg mpeg 2 encoder?
[23:28] <cbsrobot> lol
[23:31] <cbsrobot> durandal_1707: ask kierank
[23:34] <kierank> durandal_1707: it's easier to add mpeg-2 support to x264 than to add all of x264's decent features to ffmpeg's mpeg-2 encoder
[23:34] <kierank> i.e time taken to reach a world class mpeg-2 encoder is drastically lower
[23:35] <durandal_1707> kierank: you mean from code standpoint?
[23:35] <kierank> yes and by extension time
[23:42] <cbsrobot> kierank: whats the status on x262 ?
[23:43] <kierank> well progressive works
[23:43] <cbsrobot> is there a commit you recommend ?
[23:43] <kierank> and things may be changing in the near future but i can't comment on that
[00:00] --- Mon Jan 30 2012
1
0
[00:02] Action: DelphiWorld compiling dss
[00:10] <niro> Hi, when you're creating a jpeg with ffmpeg, eg. ffmpeg -f video4linux2 -i /dev/video0 -o image2 image.jpeg
[00:10] <niro> how can you specify the jpeg quality?
[00:10] <Mavrik> -qscale I think
[00:12] <niro> let me give that a go
[00:13] <Mavrik> -qscale 1 should be "best" quality
[00:14] <niro> ok, i have take one at qscale 0.1
[00:14] <niro> now for 255
[00:14] <relaxed> 1 - 39 is the range
[00:15] <relaxed> er, 31
[00:16] <niro> relaxed, i tried setting it to 0 initially, and it came up and said qscale must be set between >0.0 or <=255
[00:16] <niro> and, i've ran it with 0.1 and 255, and the quality is noticeably different
[00:17] <niro> so thank you for the help :)
[00:18] <niro> now i just need to change that into a percentage
[00:18] <relaxed> any reason you want jpg over png?
[00:20] <niro> only because it's been requested in the specification i've been provided with
[02:08] <plut0> whats a good audio codec i can use to save some disk space for home videos. right now i'm using libmp3lame and it's pretty good, just wondering if theres something better
[04:00] <plut0> i'm using ffmpeg -acodec copy on a flv file, the output has no audio. what am i doing wrong?
[04:03] <relaxed> pastebin the command and output
[04:09] <plut0> http://pastebin.com/Cc8Rh4qr
[04:11] <relaxed> the output lists the audio stream. did you try playing it with ffplay?
[04:11] <plut0> i tried vlc and mplayer
[04:13] <relaxed> what are you doing with this output? putting it on the web?
[04:13] <plut0> nah just playing it locally
[04:16] <relaxed> then I would recommend -acodec flac output.mkv
[04:16] <plut0> looks like its a bug with version 0.9.1
[04:17] <plut0> 0.7.8 works fine with the same source file and cli options
[04:19] <relaxed> then a bug report is in order
[07:09] <fling> what is sample format?
[07:09] <fling> -f alsa -ac 2 -i hw:0,0
[07:09] <fling> [alsa @ 0x24c4780] cannot set sample format 0x10000 2 (Invalid argument)
[07:10] <fling> relaxed: hello :p
[07:39] <DeezGz> whats up guys
[07:59] <fling> i have alsa xrun if i use slower, veryslow x264 presets or if i use 40mbps+ bitrate and a file is stored locally
[08:00] <fling> no xrun if i use ultrafast lossless preset and the output file is on NFS :[
[08:00] <fling> should i increase buffers or something?
[08:05] <fling> if i use pcm_s16le + mkv container i have:
[08:05] <fling> [matroska @ 0x1a6b770] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 752 >= 752
[08:05] <fling> av_interleaved_write_frame(): Invalid argument
[08:06] <fling> if i use pcm_s16le + libx264 + avi container, i have desync
[08:06] <fling> libx264 + libvorbis are fine with any container...
[08:15] <fling> what is encoder name for flac?
[08:16] <fling> oh, it is flac :]
[08:19] <fling> noone is here?
[08:21] <fling> how to set flac compression level?
[09:07] <fling> no xruns with 0.7.8
[09:07] <fling> also second streaming output works with 0.7.8
[09:07] <fling> 0.9.1 and 1.0 are totally broken
[09:12] <fling> test script > http://paste.pocoo.org/show/542556/
[10:31] <shevy> hmmm
[10:31] <shevy> ffmpeg -i foo.mp4 -r 1 -pix_fmt rgb24 animated.gif
[10:31] <shevy> the quality kinda is very very bad. lots of pixels
[10:33] <Mavrik> "gif"?
[14:15] <knowtheory> Hey folks i'm looking for some guidance. I've been trying to convert a WAV i recorded into ogg vorbis and having essentially no success either through an interface like audacity, or through ffmpeg directly. The wav is 38 secs long, but ffmpeg keeps producing files that are anywhere form 27s long to 44s, depending on how i tweak the settings.
[14:15] <knowtheory> Are there other tools anyone has used to get better conversion, or are there things i should be particularly paying attention to in the conversion process that i might be missing
[14:16] <knowtheory> (aside from bitate and sampling)
[14:18] <Mavrik> if the lenght of audio file isn't correct, then you're passing wrong input sample rate
[14:19] <knowtheory> Mavrik: see i find that puzzling because it happens both w/ ffmpeg, and with audacity (and audacity certainly knows what the sample rate is). Audacity reports that the sample rate for the file is 44100
[14:19] <Mavrik> well, obviously it isn't if your length isn't correct :)
[14:19] <Mavrik> how did you record it?
[14:20] <knowtheory> Yep, fair enough, i recorded it using audacity off of the internal mic on my macbook pro
[14:21] <Mavrik> it could be that your stereo settings aren't right
[14:22] <knowtheory> ah, i hadn't thought of that. i'll dig around then
[14:22] <knowtheory> the original file is mono&
[14:22] <knowtheory> so i'll have to see how ffmpeg handles that.
[14:23] <Mavrik> just make sure you're passing those parameters before the "-i" parameters
[14:24] <knowtheory> Mavrik: ah, does ffmpeg infer settings off of the file specified w/ -i otherwise?
[14:24] <Mavrik> yeah
[14:24] <Mavrik> knowtheory, for example if you say "-ac 1 -i <input>.wav" that meant "interpret input file as mono"
[14:24] <knowtheory> cool, thanks for the heads up
[14:25] <Mavrik> knowtheory, but "-i <input>.wav -ac 1" will mean "convert file TO mono"
[14:25] <Mavrik> same for sample rate etc.
[14:25] <knowtheory> Ahh, okay, right that makes sense.
[14:26] <knowtheory> that also explains the error i just got :D
[14:26] <knowtheory> gotta move -acodec libvorbis after the -i file.wav
[14:28] <Mavrik> ^^
[14:28] <knowtheory> Well thanks Mavrik that gives me additional direction to tinker with
[14:28] <knowtheory> i'll see if i can get it sorted out
[14:46] <shevy> Mavrik yeah, I tried to convert a .mp4 video, 3 seconds clip, without audio, to an animated .gif file
[14:46] <shevy> it would have become my new avatar for forums!
[14:47] <shevy> but somehow, it was all blurry and way too large anyway
[14:47] <shevy> then I realized, I am a total noob when it comes to ffmpeg really, and did something else. like watch youtube video clips of freerunners, parcours and tricksters ... :P
[15:14] <burek> shouldn't ffmpeg figure out what the format of the audio is if you only run something like: ffmpeg -i audio.wav -acodec ... output...
[15:18] <sacarasc> It does...
[16:42] <maister> Having some issues with the new avcodec_encode_audio2(). I can't seem to fill the AVFrame correctly. Any hints on how to do it properly?
[19:00] <j-b> hello.
[19:01] <j-b> How am I supposed to set the quality for jpeg for encoding?
[19:03] <Mavrik> -qscale
[19:03] <Mavrik> 1 being the best
[19:03] <Mavrik> 39 (afaik) being the worse
[19:03] <j-b> by the API.
[19:04] <j-b> How do we do that through the API?
[19:05] <Mavrik> what API?
[19:05] <maister> libavcodec I guess? :P
[19:05] <peace_> where i have to go to speak with avconv dev
[19:05] <Mavrik> I see. Check AVCodecContext structure.
[19:06] <iive> #libav
[19:09] <littlebit> hello people, i have a 3d movie where the screen is devided into 2 sections, is there a way how to convert that into a normal one??
[19:12] <JEEB> you can crop off one of the views
[19:12] <JEEB> and resize to matching aspect ratio
[19:12] <JEEB> and that way convert it to a "normal" clip
[19:16] <littlebit> the movie that i have is side by side
[19:16] <littlebit> and would be great if you could tell me the command that i have put int
[19:16] <littlebit> o
[19:17] <Mavrik> It would be nice if someone told me how to be rich in 1 step too.
[19:19] <littlebit> lol k
[19:20] <littlebit> i agree
[19:21] <Mavrik> littlebit, basically, look for "crop" video filter
[19:21] <Mavrik> in ffmpeg libavfilter documentation
[19:22] <peace_> but .... to get the same quality of input is there an option ?
[19:28] <littlebit> Mavrik: thanks ill look at it
[19:28] <Mavrik> peace_, what?
[19:29] <peace_> Mavrik: i want to convert a video with the same bitrate of input
[19:29] <peace_> same resolution same audio bitrate
[19:29] <peace_> same frequency
[19:29] <peace_> just change the codec
[19:30] <Mavrik> ok.
[19:30] <Mavrik> now where seems to be the problem?
[19:31] <peace_> i don't know how to do that , i used before sameq but it seems i can't find anymore on ffmpeg doc
[19:32] <peace_> ffmpeg -i stuff stuffout does video with bad video bitrate
[19:32] <peace_> so...
[19:37] <Mavrik> peace_, -sameq doesn't do that
[19:37] <Mavrik> and you can't do that automatically
[19:37] <Mavrik> you'll have to calculate average bitrate from filesize
[19:37] <Mavrik> but I very much doubt exactly the same bitrate in another format is what you want
[19:38] <peace_> wtf this sucks
[19:39] <Mavrik> well, not having cars run on seawater sucks too
[23:26] <orbisvicis> im compiling ffmpeg, the opencore amr* options only apply to some uncommon formats, right?
[23:28] <orbisvicis> and x264 is only for encoding..
[23:46] <Brance> hello
[23:46] <Brance> anyone here
[23:46] <Brance> i have some issue
[23:46] <Brance> with compily ffmpeg
[23:46] <Brance> Centos5
[23:47] <Brance> if someone is willing to help
[23:47] <Brance> i try with librtmp
[23:47] <Brance> but no luck
[23:47] <Brance> it says it's not found
[23:48] <Brance> and log says
[23:48] <burek> can you please use pastebin.com, to show your command line and its output?
[23:48] <Brance> 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 -c -o /tmp/ffconf.HHT20165.o /tmp/ffconf.bDS20157.c
[23:48] <Brance> gcc -Wl,--as-needed -o /tmp/ffconf.dZD20161 /tmp/ffconf.HHT20165.o -lopencore-amrwb -lopencore-amrnb -lmp3lame -lfaac -lm -pthread -lz
[23:48] <Brance> check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket
[23:48] <Brance> check_func_headers librtmp/rtmp.h RTMP_Socket -I/usr/local/include -I/usr/kerberos/include -L/usr/local/lib -L/usr/kerberos/lib -lrtmp -lssl -lcrypto -ldl -lz
[23:48] <Brance> check_ld cc -I/usr/local/include -I/usr/kerberos/include -L/usr/local/lib -L/usr/kerberos/lib -lrtmp -lssl -lcrypto -ldl -lz
[23:48] <Brance> check_cc -I/usr/local/include -I/usr/kerberos/include -L/usr/local/lib -L/usr/kerberos/lib
[23:48] <Brance> BEGIN /tmp/ffconf.bDS20157.c
[23:48] <Brance> 1 #include <librtmp/rtmp.h>
[23:48] <Brance> 2 long check_RTMP_Socket(void) { return (long) RTMP_Socket; }
[23:48] <Brance> 3 int main(void) { return 0; }
[23:48] <Brance> END /tmp/ffconf.bDS20157.c
[23:48] <Brance> 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 -I/usr/local/include -I/usr/kerberos/include -L/usr/local/lib -L/usr/kerberos/lib -c -o /tmp/ffconf.HHT20165.o /tmp/ffconf.bDS20157.c
[23:48] <Brance> ERROR: librtmp not found
[23:48] <burek> omg...
[23:48] <Brance> sorry
[23:48] <burek> can you please use pastebin.com, to show your command line and its output?
[23:48] <Brance> :(
[23:49] <Brance> http://pastebin.com/uKkvgmyw
[23:49] <Brance> :D
[23:49] <Brance> so?
[23:51] <Brance> comeone
[23:51] <Brance> anyone?
[23:51] <burek> be patient
[23:52] <burek> are you using debian?
[23:52] <burek> or ubuntu
[23:52] <Brance> red hat
[23:52] <Brance> RHEL
[23:52] <Brance> Centos
[23:52] <burek> it seems you dont have librtmp-dev installed
[23:53] <Brance> locate rtmp.h
[23:53] <Brance> usr/local/lib/librtmp/rtmp.h
[23:53] <Brance> i have it there
[23:54] <burek> all these too? -lrtmp -lssl -lcrypto -ldl -lz
[23:54] <Brance> how to do that
[23:54] <Brance> ?
[23:54] <burek> i dont know, i dont use centos :)
[23:56] <burek> anyway, I think you are still missing development version of librtmp
[23:56] <burek> try to google for it
[23:56] <burek> how to obtain/compile it
[23:56] <Brance> but i have installed via rpm
[23:56] <burek> what cmd did you use to install it
[00:00] --- Mon Jan 30 2012
1
0
[01:51] <CIA-108> ffmpeg: 03Paul B Mahol 07master * rf9ca1ac796 10ffmpeg/ (9 files in 3 dirs):
[01:51] <CIA-108> ffmpeg: sws/pixfmt/pixdesc: add support for yuva444p
[01:51] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[01:51] <CIA-108> ffmpeg: Conflicts:
[01:51] <CIA-108> ffmpeg: libavutil/avutil.h
[01:51] <CIA-108> ffmpeg: libavutil/pixfmt.h
[01:51] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * reb3f81e4ef 10ffmpeg/libavcodec/targa.c:
[01:51] <CIA-108> ffmpeg: targadec: Simplify RLE out of packet check.
[01:51] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[01:51] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r49f2056289 10ffmpeg/libavcodec/pngdec.c:
[01:51] <CIA-108> ffmpeg: pngdec: Convert 2/4 bit formats to 8bit.
[01:51] <CIA-108> ffmpeg: This way 2 and 4 bit gray, rgb & rgba are supported.
[01:51] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[01:51] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rf21b6159cf 10ffmpeg/libavcodec/mpegvideo_enc.c:
[01:51] <CIA-108> ffmpeg: mpegvideoenc: Fail if a buffer size is specified without a max rate.
[01:51] <CIA-108> ffmpeg: This combination makes not much sense.
[03:22] <pasteeater> michaelni: can you check to see if my key was added for ffmpeg-web?
[03:23] <pasteeater> i sent it to -devel on 20-nov-11.
[03:24] <pasteeater> becaue i get "fatal: The remote end hung up unexpectedly" when i "git push --dry-run"
[03:41] <michaelni> pasteeater, you seem to be in there
[03:41] <michaelni> does reading / fetch work ?
[03:43] <pasteeater> git fetch works
[03:44] <durandal_1707> michaelni: swscale loses alpha when doing yuva444p -> rgba, some more generic check if yuv format have alpha is missing
[03:45] <durandal_1707> there is already isALPHA
[03:49] <michaelni> pasteeater, also make sure you use url = gil@ffmpeg.org:ffmpeg-web
[03:50] <michaelni> not the git://
[03:51] <durandal_1707> michaelni: will post patch if fate passes
[03:51] <michaelni> durandal_1707, thx
[03:52] <durandal_1707> well, if fate passes it would indicate that alpha is not tested at all
[03:58] <durandal_1707> michaelni: yuva444p conversion into rgba sets alpha to 128 instead of 255
[03:59] <pasteeater> michaelni: ah, progress. at first i thought gil was a typo for git.
[03:59] <durandal_1707> ffmpeg -i ~/git/fate-suite/dxtory/dxtory_mic.avi -vf format=yuva444p -pix_fmt rgba /tmp/test.xwd
[04:00] <durandal_1707> rgb24->rgba works fine
[04:01] <durandal_1707> yuva420p->rgba works fine too
[04:11] <durandal_1707> michaelni: fixed that to, patch sent to ml
[04:16] <durandal_1707> michaelni: for this last one you would need to regenerate lavfi ref files, or i can do that?
[04:16] <durandal_1707> so this is what caused libav fate failure, ffmpeg puts 0x80 and libav some random shit
[04:17] <Compn> lol
[04:17] <Compn> boy theres been an explosion of pixfmts lately
[04:17] <Compn> how many different ways can you pack colors!?
[04:18] <drv> back in my day, all we had was 0 and 1, and we were happy with them!
[04:18] <gnafu> Yeah!
[04:18] <Compn> crazyness with 9bit, 10bit, 16bit too
[04:18] <durandal_1707> ignore all that crap, now is time for 64 bit
[04:19] <Compn> now if only i could play vivo with ffmpeg... :P
[04:19] <Compn> its my white whale
[04:19] <ohsix> all that porn from 1996 you could be playing!
[04:20] <durandal_1707> Compn: you dont use mplayer vivo demuxer?
[04:20] <Compn> i do
[04:21] <Compn> but still... its such an old format, it deserves preservation!
[04:21] <pasteeater> where's my cinepak encoder?
[04:23] <durandal_1707> pasteeater: you wrote it?
[04:24] <pasteeater> no. shitty joke about old formats.
[04:24] <Compn> wasnt there a patch?....
[04:26] <durandal_1707> it should not be hard once i build some exp, until then use mencoder
[04:26] <Compn> drv has a patch iirc
[04:26] <Compn> for vivo
[04:26] <Compn> i keep pestering him
[04:26] <drv> yeah, it'll be done sometime this decade
[04:26] <Compn> 2 years down, 8 to go :)
[04:26] <drv> unless the mayans were right
[04:27] <Compn> stop putting it off due to various calendar bugs
[04:27] <durandal_1707> vivo < tivo
[04:27] <drv> it is actually mostly working, i should send a patch to the ML and guilt trip somebody else into finishing it ;)
[04:27] <Compn> i'm guessing no one has looked at getting h263 decoder to decode viv2 tho
[04:28] <Compn> or vivoaudio being g723 something
[04:28] <durandal_1707> it is siren..somthing
[04:28] <drv> yeah, only the older files decode
[04:29] <drv> i dug up a bunch of samples from archive.org and such, but they are almost all the newer codecs
[04:29] <durandal_1707> how could one write decoder if he would first need to write demuxer?
[04:30] <Compn> durandal_1707 : use mplayer demuxer
[04:30] <Compn> build mplayer with ffmpeg changes
[04:30] <durandal_1707> Compn: that is just calling for more problems....
[04:31] <Compn> mplayer must output correct video stream in order for binary codec to handle it
[04:31] <Compn> binary codec is very specific about what data it wants
[04:32] <durandal_1707> mplayer can play those new vivo files?
[04:32] <Compn> yep
[04:32] <Compn> viv1 / vivo = old, viv2 = new
[04:33] <Compn> 'new' as in, newer than vivo , still a 15 year old codec.
[04:33] <durandal_1707> so drv demuxer when using with mplayer can still play that files?
[04:33] <drv> i haven't tried actually
[04:33] <drv> but the older format is just h.263 and g.723.1, works with ffmpeg's decoders
[04:34] <Compn> you implement seeking in your demuxer ? :P
[04:35] <burek> one question about GPL, licenses, violations, blabla..
[04:36] <burek> would this kind of page be considered illegal: http://ffmpeg.test-lab.ch/uploads/ffmpeg-custom-build.html
[04:36] <drv> nah, the timestamps are busted too, one of the things I still need to fix
[04:36] <ohsix> that's what people pay lawyers for
[04:36] <burek> :)
[04:37] <durandal_1707> burek: libaacplus builds are nonfree
[04:37] <burek> yes i know, its just a dummy temp page
[04:37] <burek> to show the idea
[04:38] <Compn> burek : if you give a link to source somewhere its fine
[04:38] <durandal_1707> web page appears to be hosted from .ch?
[04:38] <Compn> non-free license aside...
[04:38] <burek> well ok, i would create it in such way that it shows warnings all over when you select a non-free package and stuff
[04:38] <burek> but is it illegal?
[04:38] <durandal_1707> burek: stop this yagadada, and distribute ffmpeg via torrent
[04:39] <burek> :)
[04:39] <Compn> its legal to distribute as long as you also include the source
[04:39] <Compn> in a link
[04:39] <durandal_1707> burek: depends in which country is server/person
[04:39] <Compn> or something
[04:39] <burek> im not into piracy and stuff, i just want to make it easy to the people to build/use ffmpeg, thats all
[04:40] <Compn> most ffmpeg devs only care if the people distributing ffmpeg have a link to sourcecode
[04:40] <Compn> at least i hope so
[04:40] <Compn> and changes / patches published are really nice too
[04:40] <Compn> but i cant speak for whole project.
[04:40] <burek> yes, the link would be there too
[04:40] <pasteeater> enable-bzlib is autodetect
[04:40] <burek> pasteeater, it would be a static lib
[04:40] <burek> -lib +binary
[04:41] <durandal_1707> burek: how is gonna pay for electricity and data transfer?
[04:41] <burek> easy :) I can auto-upload to sourceforge or some other file sharing hosting if the upload starts going crazy :)
[04:41] <durandal_1707> imho better add precompiled stuff
[04:43] <durandal_1707> and i hope you will add x264, lame and xvid support at least
[04:44] <burek> well, I already managed to add all gpl stuff
[04:44] <burek> just alsa is making me headache so far
[04:58] <burek> how to tell cmake to build static libs, like --enable-static=yes ?:)
[05:05] <durandal_1707> google did not help?
[05:05] <burek> no :/
[05:12] <NapHtaKeRoSene> hi
[05:12] <NapHtaKeRoSene> my random is benchmarked at 3GB/s filling my graphic screen, the enhanced version is 470MB/s, /dev/urandom is 4.4MB/s
[05:15] <durandal_1707> how you watch images with mplayer?
[05:23] <Compn> mplayer mf://*jpg
[05:24] <Compn> -mf fps=1
[05:24] <Compn> -fps .1
[05:33] <NapHtaKeRoSene> interesting slideshow option
[05:48] <durandal_1707> i get strange last line with x11 output with mplayer, more noticeable with -zoom, can anyone here reproduce it?
[07:28] <vtorri> hey
[07:28] <vtorri> in os_support.h, the definition of lseek looks strange to me
[07:28] <vtorri> in msdn:
[07:28] <vtorri> http://msdn.microsoft.com/en-us/library/1yee101t%28VS.80%29.aspx
[07:29] <vtorri> -lseeki64 is already declared in io.h
[07:29] <vtorri> _lseeki64, not -lseeki64 :)
[07:42] <vtorri> lots of warnings with mingw :)
[07:48] <vtorri> also
[07:48] <michaelni> vtorri, can you send a patch :)
[07:48] <vtorri> hehe
[07:48] <vtorri> also
[07:48] <michaelni> its 8 in the morning here so i dont want to change things i cant test :)
[07:48] <vtorri> problem when creatin .def and .lib
[07:49] <vtorri> same here
[07:49] <vtorri> our countries are neighbour :)
[07:49] <vtorri> (i think)
[07:49] <vtorri> SLIB_EXTRA_CMD is defined as -lib ***
[07:50] <vtorri> i'm cross compiling (i'm on linux, using mingw-w64)
[07:50] <vtorri> lib.exe is not found
[07:50] <vtorri> so no .lib and .def
[07:50] <vtorri> that's annoying
[07:51] <michaelni> you should talk with Zeranoe or ramiro
[07:51] <michaelni> iam not the windows guy :)
[07:51] <vtorri> hehe
[07:51] <vtorri> i know
[07:52] <vtorri> i'm just descirbing the problem, in case someone with better skills than me read them and solve the problem
[07:53] <vtorri> lib.exe seems to be a microsoft tool
[07:53] <nevcairiel> it is, but it works much better then the mingw variant, which is why its being used
[07:55] <vtorri> problem is, when cross compiling on linux, there are no microsoft tools :)
[07:56] <nevcairiel> so the configure needs to take that into account, as long as it keeps being used when compiling on windows for windows. :)
[07:56] <vtorri> configure should also check for lib.exe avaialbility
[07:57] <vtorri> (on windows)
[07:57] <nevcairiel> also, whats wrong with lseek? doesnt look wrong to me
[07:59] <vtorri> i have that warning:
[07:59] <vtorri> libavformat/os_support.h:34:0: warning: "lseek" redefined [enabled by default]
[07:59] <vtorri> /home/torri/local/opt/mingw-w64-x86_32/bin/../lib/gcc/i686-w64-mingw32/4.7.0/../../../../i686-w64-mingw32/include/io.h:336:0: note: this is the location of the previous definition
[08:00] <vtorri> io.h is certainly included somewhere
[08:00] <vtorri> so lseek is already defined
[08:00] <vtorri> then in os_support.h, you redefine lseek
[08:01] <nevcairiel> the problem is that mingws lseek is an alias for _lseek (which only exists in mingw), which only takes a long position, not a int64 position
[08:01] <nevcairiel> so it cannot be used
[08:01] <vtorri> so maybe a:
[08:01] <vtorri> #ifdef lseek
[08:01] <vtorri> # undef lseek
[08:01] <vtorri> #endif
[08:01] <vtorri> before defining your lseek in os_support.h
[08:02] <vtorri> to remove that warning
[08:03] <vtorri> other redefinitions:
[08:03] <nevcairiel> shouldnt hurt, i suppose
[08:04] <vtorri> EPROTONOSUPPORT, ETIMEDOUT, ECONNREFUSED, EINPROGRESS, gai_strerror
[08:04] <nevcairiel> in my version of mingw, lseek is a function, not a define
[08:04] <nevcairiel> so it doesnt complain
[08:04] <vtorri> in libavformat/network.h
[08:04] <vtorri> nevcairiel: ok. I use mingw-w64
[08:05] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r5eda82aac9 10ffmpeg/libavcodec/frwu.c:
[08:05] <CIA-108> ffmpeg: frwu: check if allocation was successful
[08:05] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[08:05] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[08:05] <vtorri> mingw is not improving quickly
[08:05] <CIA-108> ffmpeg: 03Diego Biurrun 07master * r299ab0fd17 10ffmpeg/libavutil/ (Makefile file.c): libavutil: Remove pointless file test program.
[08:05] <CIA-108> ffmpeg: 03Mashiat Sarker Shakkhar 07master * rd51d6ae9c4 10ffmpeg/libavcodec/vc1dec.c: (log message trimmed)
[08:05] <CIA-108> ffmpeg: vc1dec.c: Don't use uninitialized values of dmv_[x, y]
[08:05] <CIA-108> ffmpeg: MVDATA may or may not be transmitted. If it is not, both
[08:05] <CIA-108> ffmpeg: dmv_x and dmv_y is to be assumed zero.
[08:05] <CIA-108> ffmpeg: This may not trigger wrong picture in all systems, but
[08:05] <CIA-108> ffmpeg: it's a bug nevertheless. Fixes SA10116.vc1 on my 64-bit
[08:05] <CIA-108> ffmpeg: Windows 7.
[08:05] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r324e818093 10ffmpeg/libavcodec/8bps.c:
[08:05] <CIA-108> ffmpeg: 8bps: K&R formatting cosmetics
[08:05] <CIA-108> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de>
[08:05] <CIA-108> ffmpeg: 03Paul B Mahol 07master * rfc115c80b7 10ffmpeg/ (9 files in 3 dirs):
[08:05] <CIA-108> ffmpeg: sws/pixfmt/pixdesc: add support for yuva444p
[08:05] <CIA-108> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com>
[08:05] <CIA-108> ffmpeg: 03Ronald S. Bultje 07master * r3b15a6d742 10ffmpeg/ (27 files in 4 dirs):
[08:05] <CIA-108> ffmpeg: config.asm: change %ifdef directives to %if directives.
[08:05] <CIA-108> ffmpeg: This allows combining multiple conditionals in a single statement.
[08:05] <CIA-108> ffmpeg: 03Alex Converse 07master * r08628b6afb 10ffmpeg/ (9 files in 3 dirs):
[08:05] <CIA-108> ffmpeg: Revert "sws/pixfmt/pixdesc: add support for yuva444p"
[08:05] <CIA-108> ffmpeg: This reverts commit fc115c80b7bf6f1f5a937ee7f6e14630d99eb4dd.
[08:05] <CIA-108> ffmpeg: Tests are broken.
[08:05] <CIA-108> ffmpeg: 03Ronald S. Bultje 07master * rb5d08c27c3 10ffmpeg/libswscale/x86/ (input.asm swscale_mmx.c swscale_template.c):
[08:05] <CIA-108> ffmpeg: swscale: convert rgb/bgr24ToY/UV_mmx functions from inline asm to yasm.
[08:05] <CIA-108> ffmpeg: Also implement sse2/ssse3/avx versions.
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r7063b6eaee 10ffmpeg/ (doc/APIchanges libavcodec/version.h tests/ref/lavf/ffm):
[08:05] <CIA-108> ffmpeg: lavc: increase major version to 54.
[08:05] <CIA-108> ffmpeg: The lavf-ffm test results change because ffmenc writes
[08:05] <CIA-108> ffmpeg: AVCodecContext.flags/flags2 and the defaults for those change.
[08:05] <CIA-108> ffmpeg: 03Alexandra Khirnova 07master * r570f6aaab1 10ffmpeg/cmdutils.c:
[08:05] <CIA-108> ffmpeg: cmdutils: fix options starting with 'no'.
[08:05] <CIA-108> ffmpeg: E.g. non_linear_quant mpeg2video private option.
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r99ace37e68 10ffmpeg/libavcodec/ (avcodec.h mjpegdec.c options.c version.h): lavc: remove disabled FF_API_MJPEG_GLOBAL_OPTS cruft.
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * ra17479dfce 10ffmpeg/ (doc/APIchanges libavformat/version.h): lavf: increase major version from 53 to 54.
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r2439f2ca82 10ffmpeg/libavformat/ (avio.c avio.h aviobuf.c options.c url.h version.h): lavf: remove disabled FF_API_OLD_AVIO cruft
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r38233fc131 10ffmpeg/libavformat/ (avformat.h metadata.c metadata.h version.h): lavf: remove disabled FF_API_OLD_METADATA2 cruft
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * recd5f41a06 10ffmpeg/libavformat/ (avformat.h utils.c version.h): lavf: remove disabled FF_API_PARSE_DATE cruft
[08:05] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r64f6f6f836 10ffmpeg/libavformat/ (avformat.h img2.c version.h): lavf: remove disabled FF_API_GUESS_IMG2_CODEC cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r997420abf2 10ffmpeg/libavformat/ (avformat.h utils.c version.h): lavf: remove disabled FF_API_DUMP_FORMAT cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * rcc10ab79ec 10ffmpeg/libavformat/ (avformat.h utils.c version.h): lavf: remove disabled FF_API_FIND_INFO_TAG cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r21824e5b88 10ffmpeg/libavformat/ (avformat.h utils.c version.h): lavf: remove disabled FF_API_PKT_DUMP cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r2fb7501938 10ffmpeg/libavformat/ (avformat.h utils.c version.h):
[08:06] <CIA-108> ffmpeg: lavf: remove disabled FF_API_FORMAT_PARAMETERS cruft
[08:06] <CIA-108> ffmpeg: Also remove now unused AVFormatParameters struct and
[08:06] <CIA-108> ffmpeg: AVOutputFormat.set_parameters().
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r7232bfbd93 10ffmpeg/libavcodec/ (avcodec.h libx264.c libxavs.c options.c version.h): lavc: remove disabled FF_API_X264_GLOBAL_OPTS cruft.
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * rb32e30faa1 10ffmpeg/libavformat/ (avformat.h version.h): lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r2d70912cc2 10ffmpeg/libavformat/ (avformat.h movenc.c options.c version.h): lavf: remove disabled FF_API_FLAG_RTP_HINT cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r3bbfd3026c 10ffmpeg/libavformat/ (avformat.h gif.c version.h): lavf: remove disabled FF_API_LOOP_OUTPUT cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * rdf0bb26aca 10ffmpeg/libavformat/ (avformat.h dvenc.c gxfenc.c movenc.c mxfenc.c version.h): lavf: remove disabled FF_API_TIMESTAMP cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * r6dc345555f 10ffmpeg/libavformat/ (avformat.h img2.c version.h): lavf: remove disabled FF_API_LOOP_INPUT cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * rf0cb13958d 10ffmpeg/libavformat/ (avformat.h mpegenc.c mpegtsenc.c options.c version.h): lavf: remove disabled FF_API_MUXRATE cruft
[08:06] <CIA-108> ffmpeg: 03Anton Khirnov 07master * radad5b88f8 10ffmpeg/libavformat/ (rtsp.c version.h): lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft
[08:06] <CIA-108> (63 lines omitted)
[08:08] <vtorri> there are also some conversion characters not supported by the printf family on Windows
[08:08] <vtorri> t, T
[08:09] <vtorri> hoo
[08:09] <vtorri> ./libavcodec/w32pthreads.h:226:5: warning: return with a value, in function returning void [enabled by default]
[08:09] <vtorri> :)
[08:13] <vtorri> nevcairiel: do you want the complete output of my build ?
[09:01] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r72bbe768ba 10ffmpeg/libavdevice/alldevices.c:
[09:01] <CIA-108> ffmpeg: devices: Disable v4l to fix compilation until someone
[09:01] <CIA-108> ffmpeg: has time to update it to the new API.
[09:01] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[09:16] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r81ab42a334 10ffmpeg/libavcodec/x86/ (diracdsp_yasm.asm dwt_yasm.asm):
[09:16] <CIA-108> ffmpeg: dirac_yasm: fix linking failure due to %ifndef
[09:16] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[11:05] <CIA-108> ffmpeg: 03Clément BSsch 07master * r23b639c44f 10ffmpeg/ (6 files in 5 dirs): Fix "ouput" typo and add a check.
[12:41] <CIA-108> ffmpeg: 03Clément BSsch 07master * r3bc92e34a4 10ffmpeg/cmdutils.c: cmdutils: re-add return 0 accidentally remove from opt_protocols().
[15:28] <nevcairiel> I'm seeing some rather puzzling behaviour - i have the demuxer and the decoder running on completely different format contexts, and they only communicate via AVPackets. Now i am playing a paletted format which has the palette in side-data, however i never forwarded the side data from the demuxer to the decoder
[15:28] <nevcairiel> Still, it decodes fine
[15:28] <nevcairiel> how the hell do the two communicate? :d
[15:38] <saste> nevcariel: telepathy
[15:48] <nevcairiel> seriously, i dont understand why it works
[15:48] <nevcairiel> not that i mind, but i want to know why. :D
[15:51] <saste> nevcariel: yes be happy as long as it works, even if you can't understand why ;)
[15:53] <nevcairiel> oh i see, it puts the side data into the raw data stream
[15:54] <nevcairiel> with merge/split_side_data
[15:54] <nevcairiel> another mystery solved
[18:16] <ubitux> if i need common code between lavf & lavc, where is it prefered to put it? lavc?
[18:45] <Compn> lavu
[18:45] <Compn> :P
[18:46] Action: Compn has no clue
[18:46] <Compn> what does the code do ?
[18:46] <ubitux> it's basically a text format directives array, it doesn't really belongs to lavu
[18:46] <ubitux> it's codec/format specific
[18:56] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r762311ff94 10ffmpeg/ (6 files in 2 dirs):
[18:56] <CIA-108> ffmpeg: swscale: fix conversion into yuva444p from non-alpha formats
[18:56] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[18:56] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[18:56] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r7bab43f29c 10ffmpeg/libswscale/yuv2rgb.c:
[18:56] <CIA-108> ffmpeg: yuv2rgb: better check if source format have alpha
[18:56] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[18:56] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[19:19] <ubitux> it seems lavc has no dep on lavf, but lavf has deps on lavc
[19:20] <ubitux> mh.
[19:20] <ubitux> it seems lavc has no dep on lavf, but lavf has *no* depson lavc
[19:20] <ubitux> what am i saying&
[19:20] <ubitux> i'm totally braindead tonight
[19:21] <ubitux> lavc has no dep on lavf, lavf has deps on lavc
[19:21] <ubitux> so common code belongs to lavc.
[19:21] <ubitux> pfiu. that was hard.
[19:22] <nevcairiel> i would recommend less drinking
[19:22] <ubitux> i don't drink alcohol at all
[19:22] <ubitux> maybe i should actually.
[19:28] <durandal_1707> av_set_pts_info was not removed
[19:51] <nevcairiel> alot of those removals really seem to make it harder to write an external demuxer
[19:51] <nevcairiel> i'll have to use private symbols again!
[19:52] <durandal_1707> why external demuxer? why not internal one?
[19:53] <nevcairiel> because its way easier to debug when i keep it in the code thats compiled with MSVC instead of the GCC code. :)
[20:09] <durandal_1707> ffmpeg/libav code smells full of memleaks
[20:14] <Compn> lol
[20:15] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rafe9105f66 10ffmpeg/doc/examples/muxing.c:
[20:15] <CIA-108> ffmpeg: example: update muxing example to purple API
[20:15] <CIA-108> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com>
[20:15] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:15] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rcb73e35660 10ffmpeg/doc/examples/decoding_encoding.c:
[20:15] <CIA-108> ffmpeg: example: Update decoding/encoding example to the new API.
[20:15] <CIA-108> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com>
[20:15] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:33] <ubitux> seriously, the ppl who did jacosub were on LSD
[20:33] <ubitux> \D Insert current date into the title. The date will be of the format DD MMM YYYY, as in 2 Apr 1996.
[20:34] <ubitux> wtf is this related to subtitles& ? :D
[20:34] <nevcairiel> wth is jacosub
[20:34] <ubitux> old school ASS
[20:35] <ubitux> for amiga
[20:35] <ubitux> http://unicorn.us.com/jacosub/
[20:35] <durandal_1707> av_log_missing_feature()
[20:36] <ubitux> btw, does anyone know if there is a "reset to default style" tag in ASS?
[20:36] <durandal_1707> maybe for those artistic type of films...
[20:37] <ubitux> oh, \r
[20:37] <ubitux> great.
[20:37] <durandal_1707> i read commit log of mplayer and they ignore \r
[20:38] <durandal_1707> perhaps mplayer have own ass parser ....
[20:38] <ubitux> huh?
[20:40] <durandal_1707> https://github.com/richardpl/mplayer2/commit/1ef4fe59e2c25964c64bd54b198e56…
[20:40] <iive> durandal_1707: please name the forks correctly.
[20:40] <durandal_1707> iive: that is mplayer(1)
[20:40] <ubitux> durandal_1707: i meant "{\\r}"
[20:41] <ubitux> ass tag
[20:41] <iive> durandal_1707: why is it named mplayer2 ?
[20:41] <ramiro> hi
[20:42] <durandal_1707> iive: mplayer2 is my own fork of mplayer2, it have branch master,mplayer2,mplayer1 and several others
[20:42] <ramiro> michaelni: which commit exactly broke dshow? I'm somewhat confused with those git merges with lots of commits
[20:45] <durandal_1707> i dont have impression it is broken, michaelni just did not removed old part of code which is already removed in libav
[20:46] <Compn> durandal_1707 : lol @ mplayer2 fork being named mplayer2
[20:47] <durandal_1707> Compn: i could name it mplayer3 and it would certainly cause bit popularity from anime fans
[20:48] <Compn> lol
[20:48] <durandal_1707> Compn: just give me some remote server and i will do that
[20:49] <Compn> i dont need it
[20:49] <Compn> just thought it was funny :)
[20:51] <durandal_1707> Compn: you dont need ordered matroska chapters and gapless playback?
[20:52] <Compn> durandal_1707 : the only use i've seen of ordered chapters has been to watch intros before each anime episode
[20:52] <Compn> and i certainely dont need to watch the same intro 40 times if i'm watching 40 episodes
[20:52] <Compn> once is enough.
[20:53] <nevcairiel> the feature sure has potential (like, mixing theatrical and directors cut of a movie into one file), but its only used for anime intros, so .. meh!
[20:53] <Compn> and gapless i just dont care about
[20:53] <Compn> not saying other people, just me
[21:16] <durandal_1707> michaelni: should i stop spamming ml?
[21:20] <michaelni> ramiro, 2fb7501938b7103624c9bef740ca498258cacdab
[21:20] <michaelni> this removed AVFormatParameters
[21:21] <michaelni> codec->time_base = ap->time_base; <---- this used AVFormatParameters
[21:21] <michaelni> durandal_1707, please continue sending patches
[21:22] <michaelni> the more the better
[21:23] <ramiro> michaelni: can I just set codec->time_base to (AVRational){framerate.den, framerate.num}?
[21:23] <ramiro> does it make sense?
[21:24] <durandal_1707> michaelni: vorbisenc, libfaac, alacenc (probably forgot about others) are harder to fix because they do not check for allocations failures at all
[21:24] <michaelni> makes sense based on my limited knowledge of dshow.c
[21:25] <ramiro> libavdevice/v4l.c still has AVFormatParameters, shouldn't it be removed?
[21:26] <michaelni> yes
[21:26] <michaelni> configure didnt enable it here so i missed it
[21:26] <michaelni> and just quickly disabled in alldevices before going to bed to fix compile
[21:28] <durandal_1707> did 12/20
[21:49] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r7b21b0f155 10ffmpeg/libavcodec/adxenc.c:
[21:49] <CIA-108> ffmpeg: adxenc: check return value of avcodec_alloc_frame()
[21:49] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[21:49] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:49] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r88fb935c1c 10ffmpeg/libavcodec/ac3enc.c:
[21:49] <CIA-108> ffmpeg: ac3enc: check return value of avcodec_alloc_frame()
[21:49] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[21:49] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:49] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r82c252972b 10ffmpeg/libavcodec/adpcmenc.c:
[21:49] <CIA-108> ffmpeg: adpcmenc: check return value of avcodec_alloc_frame()
[21:49] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[21:49] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:49] <CIA-108> ffmpeg: 03Paul B Mahol 07master * rf870fc2fb9 10ffmpeg/libavcodec/adpcmenc.c:
[21:49] <CIA-108> ffmpeg: adpcmenc: refactor some code.
[22:01] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r0b378e8aa9 10ffmpeg/ (libavcodec/dfa.c tests/ref/fate/dfa4):
[22:01] <CIA-108> ffmpeg: DFA: fix "skip frame" TSW1 encoding.
[22:01] <CIA-108> ffmpeg: Previously the decoder would raise an error.
[22:01] <CIA-108> ffmpeg: The end result is the same, the time stamps only change
[22:01] <CIA-108> ffmpeg: because regression tests create time stamps incorrectly.
[22:01] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[22:01] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * ra0eb8de5e0 10ffmpeg/ (libavcodec/eacmv.c libavformat/electronicarts.c):
[22:01] <CIA-108> ffmpeg: Merge palette-only header with actual video frame.
[22:01] <CIA-108> ffmpeg: This fixes the video frame pts (off by one for each MVIh)
[22:01] <CIA-108> ffmpeg: and makes the "key frames" decode stand-alone (MVIh
[22:01] <CIA-108> ffmpeg: contains only palette, such a palette-only frame being
[22:01] <CIA-108> ffmpeg: marked as key frame is not really correct).
[22:01] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[22:01] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r333f5b52e3 10ffmpeg/tests/fate/ (8 files):
[22:01] <CIA-108> ffmpeg: Remove unnecessary -vsync 0.
[22:01] <CIA-108> ffmpeg: The tests work fine without it, and it will cause issues when the
[22:01] <CIA-108> ffmpeg: rawvideo decoder is changed to properly handle pts values.
[22:01] <CIA-108> ffmpeg: The H.264 conformance tests however are still broken, usually losing
[22:01] <CIA-108> ffmpeg: the first frames without it.
[22:01] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[22:01] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * rbb2d42c22f 10ffmpeg/libavformat/framecrcenc.c: (log message trimmed)
[22:02] <CIA-108> ffmpeg: Set AVFMT_TS_NONSTRICT for framecrc muxer.
[22:02] <CIA-108> ffmpeg: Since it is set for e.g. webm muxer we should make it possible
[22:02] <CIA-108> ffmpeg: to test such streams with framecrc, too.
[22:02] <CIA-108> ffmpeg: Though the primary reason is that this allows the H.264 tests
[22:02] <CIA-108> ffmpeg: to not run into this check when fixing raw video encode to
[22:02] <CIA-108> ffmpeg: pass pts values on.
[22:02] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r34751f8313 10ffmpeg/libavformat/mov.c: (log message trimmed)
[22:02] <CIA-108> ffmpeg: Demux all parts of concatenated H.264.
[22:02] <CIA-108> ffmpeg: We may or may not be able to play the latter parts
[22:02] <CIA-108> ffmpeg: but not demuxing at all seems like the worst possible behaviour.
[22:02] <CIA-108> (13 lines omitted)
[22:08] <saste> michaelni: what is "purple API"?
[22:08] <ubitux> bikeshed API?
[22:08] <saste> ah got it :)
[22:09] <ubitux> it seems my jacosub encoder is starting to work
[22:09] <michaelni> :)
[22:10] <durandal_1707> michaelni: why ff_get_unscaled_swscale have check for YUVA420P?
[22:13] <durandal_1707> i think that check is not needed at all
[22:13] <michaelni> which ?
[22:15] <durandal_1707> YUVA420p to YUV420p and reverse
[22:16] <durandal_1707> \ /* simple copy */
[22:17] <michaelni> hmm, agree seems redundant
[22:18] <durandal_1707> hmm, need to check yuva444p to and from nv21
[22:24] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r3ac99cdc41 10ffmpeg/libavcodec/libgsm.c:
[22:24] <CIA-108> ffmpeg: libgsm: check return value of gsm_create & avcodec_alloc_frame()
[22:24] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:24] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:24] <CIA-108> ffmpeg: 03Paul B Mahol 07master * ra136c2cf3e 10ffmpeg/libavcodec/libmp3lame.c:
[22:24] <CIA-108> ffmpeg: libmp3lame: remove dupe assigment
[22:24] <CIA-108> ffmpeg: It is already done by avcodec_alloc_context().
[22:24] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:24] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:24] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r5291cedd6c 10ffmpeg/libavcodec/libgsm.c:
[22:24] <CIA-108> ffmpeg: libgsm: remove dupe assigment.
[22:24] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:24] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:24] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r766e160e87 10ffmpeg/libavcodec/libopencore-amr.c:
[22:24] <CIA-108> ffmpeg: libopencore-amr: check return value of avcodec_alloc_frame()
[22:24] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:25] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:25] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r2de6a7253e 10ffmpeg/libavcodec/roqaudioenc.c:
[22:25] <CIA-108> ffmpeg: roqaudioenc: remove redundant assignment
[22:25] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:25] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:25] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r3ab90687fd 10ffmpeg/libavcodec/roqaudioenc.c:
[22:25] <CIA-108> ffmpeg: roqaudioenc: check return value of avcodec_alloc_frame()
[22:25] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[22:25] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:25] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r1fa24d08b2 10ffmpeg/libavcodec/pcm.c:
[22:25] <CIA-108> (19 lines omitted)
[22:47] <ubitux> mmh about forgotten patches
[22:48] <ubitux> hasn't Aurelien Jacobs submitted a MicroDVD decoder?
[23:01] <michaelni> google finds a demuxer & muxer but i see no decoder, maybe iam searching for the wrong terms
[23:01] <ubitux> ok, then if it's only the demuxer & muxer it's upstream
[23:02] <ubitux> too bad we can't do much without the decoder
[23:03] <ubitux> is it common practice to have a common context struct layout between demuxer and decoder?
[23:03] <ubitux> in case of subtitles, i need to parse the timing stuff two times
[23:03] <ubitux> at demuxer level, and then at decoder level
[23:05] <ubitux> in the decoder, i can fetch it from the demuxer packet since it only has the start time, and not the duration
[23:08] <ubitux> (and i need a common context because there is context time shift and scale)
[23:15] <michaelni> demuxer and decoder cant access each others private context
[23:15] <michaelni> if thats what you suggest
[23:15] <michaelni> they could run in different threads
[23:16] <michaelni> with fifo between them ...
[23:18] <ubitux> oh?
[23:18] <ubitux> how can i do that?
[23:18] <ubitux> (i mean access demuxer context in decoder scope)
[23:21] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * rdbc7c6121e 10ffmpeg/libavcodec/libgsm.c:
[23:21] <CIA-108> ffmpeg: libgsm: Fix compilation.
[23:21] <CIA-108> ffmpeg: Move libgsm_encode_close before its first use and call it
[23:21] <CIA-108> ffmpeg: with the correct number of arguments.
[23:21] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[23:23] <ubitux> michaelni: through extradata, or another mean?
[23:46] <michaelni> ubitux, extradata, sidedata, normal AVPacket data
[23:48] <CIA-108> ffmpeg: 03Reimar Döffinger 07master * r97dece333d 10ffmpeg/libavcodec/rawenc.c:
[23:48] <CIA-108> ffmpeg: rawenc: initialize coded_frame via avcodec_get_frame_defaults.
[23:48] <CIA-108> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de>
[23:51] <ubitux> mmh
[23:51] <ubitux> ok
[23:52] <ubitux> thank you.
[00:00] --- Sun Jan 29 2012
1
0
[00:05] <marteo> same thing. more accurately, i stopped after one second of input, and he gave me 1025 png's.
[00:05] <burek> with ffmpeg -re -i <input> -f image2 out%04d.png
[00:05] <burek> ?
[00:06] <marteo> yes
[00:06] <burek> with ffmpeg -re -i <input> -f image2 -r 25 out%04d.png
[00:07] <burek> no
[00:07] <burek> ffmpeg -re -r 25 -i <input> -f image2 out%04d.png
[00:07] <burek> and drop -re :)
[00:09] <marteo> no, you actually meant ffmpeg -i <input> -f image2 -r 25 out%04d.png
[00:09] <marteo> and it seems to work fine
[00:09] <burek> well it does, but it duplicates/drops frames
[00:09] <burek> when you apply it to the output
[00:09] <marteo> it drops most of them, yes
[00:09] <burek> when you apply -r to the input it reads the input more slowly (i guess)
[00:10] <marteo> my ffmpeg does not know -r option for the input
[00:10] <marteo> i'll take 50 .pngs per video second, and then i'll reassemble it proprely
[00:11] <burek> ok
[00:11] <burek> but then
[00:11] <burek> put -re back
[00:11] <burek> to slow down the input
[00:12] <marteo> -re does not do anything
[00:13] <marteo> because the input is already under realtime speed
[00:13] <burek> yes but when ffmpeg converts input file to output file, then it tries to do it as fast as possible
[00:13] <burek> not in real time
[00:13] <marteo> because the video has an absurd fps rate
[00:13] <marteo> ah
[00:13] <burek> real time means at the speed of watching the video
[00:14] <burek> so, it should create png images more slow, but I don't think it will influence output quality
[00:15] <marteo> right
[00:16] <marteo> well thanks, using png's was a great idea
[00:21] <burek> :beer: :)
[00:21] <marteo> come to paris and i'll get you ine
[00:21] <marteo> *one
[00:21] <burek> its a deal :)
[00:22] <burek> all for beer and beer for all :D
[00:22] <marteo> so long :-)
[01:01] <boo> Decoding dvb-t stream and passing it to rtmp server always crash ~ 5.2 sec, when I add -analyzeduration 1500000000 then the time can be extended to 10.3 seconds, any ideas how to fix this? :/
[01:06] <pasteeater> does it crash if you output to a file instead of rtmp?
[01:35] <Jan-> Is it practical - I don't mean possible, I mean practical - to compile a useful ffmpeg on windows, including all the essential stuff that's generally external libraries?
[01:35] <Jan-> I assume it's possible but not really very worthwhile.
[01:36] <Jan-> Possible in theory, I mean, but not worth actually doing.
[01:49] <pasteeater> Jan-: it's easier to use this: http://ffmpeg.zeranoe.com/builds/
[01:49] <Jan-> uh, yeah
[01:50] <Jan-> the plan was to try and use some of its code in another project (open source and all) and I thought a reasonable first step would be to compile it vanilla to prove it can be done
[01:50] <Jan-> I assume it could in theory be done but from what I see on google it isn't often done.
[01:56] <Jan-> is "WARNING: pkg-config not found, library detection may fail." bad?
[02:01] <Jan-> is "array subscript is below bounds" bad?
[02:09] <Jan-> Welp, it produced an executable.
[02:09] <Jan-> Doubtless a useless one as it doesn't include any of the required external libraries.
[02:09] <Jan-> But hey, it produced an executable, which is more than I was expecting!
[02:12] <Jan-> what's a good test to see if an ffmpeg compile is "basically working"
[02:12] <burek> :)
[02:13] <burek> ffmpeg -version
[02:13] <Jan-> ffmpeg version 0.9.1.git
[02:13] <Jan-> built on Jan 28 2012 01:04:40 with gcc 4.6.1
[02:13] <Jan-> configuration: --disable-yasm
[02:13] <burek> I'm trying to build the static binary on linux for ffmpeg
[02:13] <burek> but I keep getting stuck
[02:13] <Jan-> isn't the "configuration" line usually a lot (lot, lot) longer than that
[02:14] <burek> Jan-, it depends on what you want your ffmpeg to be able to do
[02:14] <Jan-> I thought this basic version wouldn't even include things like mpeg-4, but I just ran "ffmpeg -i in.avi out.avi" and got a very bad FMP4 AVI as a result.
[02:15] Action: Jan- is absolutely astonished at how well this seems to be going
[02:15] <Jan-> what else should I add? And how?
[02:16] <drv> you should definitely install yasm; that affects speed a lot
[02:16] <Jan-> how?
[02:17] <Jan-> anyway I'm not that worried about speed
[02:17] <Jan-> oh, it's h.264 it doesn't have by default, right
[02:17] <Jan-> and vorbis, and...
[02:17] <Jan-> and some crypto stuff, right?
[02:19] <Jan-> hm it won't read h.264 with aac audio
[02:32] <Zeranoe> burek: stuck on what?
[02:45] <burek> with alsa
[02:45] <burek> it seems alsa links dinamically against some functions in time.h
[02:46] <burek> http://pastebin.com/Ld1SZWPX
[02:46] <burek> and this is in config.log about alsa: http://pastebin.com/e6hW0B2E
[02:47] <burek> that was for Zeranoe :)
[02:51] <Zeranoe> burek: can it find time.h?
[02:52] <burek> how can i check that
[02:56] <Zeranoe> burek: I'm checking into it niow
[02:57] <burek> thanks :)
[02:57] <MrNaz> relaxed i am now... what's up?
[03:36] <gretelmk2> i am trying to build a small frame grabbing program using the ffmpeg (version 0.10) libs and i am following the closest tutorial I can find, but my frames come up looking like near garbage
[03:37] <Jan-> At least you can actually get the libraries to... well, work, compile, appear at all :)
[03:37] <gretelmk2> if i use ffmpeg -i output.ps what%d.jpg, each of the frames come out looking ok
[03:37] <gretelmk2> right
[03:37] <Jan-> I can just about get a crippled ffmpeg to compile with no external libraries.
[03:38] <gretelmk2> so, i am essentially having it properly guess the format for me and then i'm doing the avcodec_decode_video2
[03:38] <gretelmk2> oh
[03:38] <gretelmk2> i seem to be ahead of you ;p
[03:38] <Jan-> I'm not really a software engineer.
[03:38] <Jan-> Or rather I am, but I'm used to visual studio, which works a lot better.
[03:39] <gretelmk2> i use a half assed approach of installing the package from the repo then doing an ffmpeg and stealing the configure parameters and adding a few more on the end
[03:39] <gretelmk2> ah
[03:39] <gretelmk2> so then i uninstall the sucker and build my own
[03:40] <gretelmk2> that gets me to a fully functional ffmpeg if you want to try that
[03:40] <Jan-> I think I'll let it be for now
[03:40] <Jan-> they clearly don't want people compiling it under windows
[03:40] <gretelmk2> i thought there was a guide for it...?
[03:41] <gretelmk2> or is it a cross compile thing with mingw?
[03:41] <gretelmk2> i'm not really trying under windows ;p
[03:41] <gretelmk2> so I guess you won't be able to tell me why my frame looks like crap when I convert it to a jpeg? ;p
[03:42] <Jan-> no.
[03:42] <gretelmk2> it's like i'm looking at the same image kinda blended together 3 times or so
[03:42] <gretelmk2> it sucks
[03:42] <gretelmk2> everything looks right
[03:42] <Jan-> most windows builds are cross compiled
[03:42] <Jan-> I don't think it's ever built on windows really
[03:42] <gretelmk2> well, you could go the cygwin route... maybe.
[03:43] <gretelmk2> just give up on your windows ;p or better yet get yourself VirtualBox
[03:43] <gretelmk2> get the iso for the flavor of linux you know or can stand and use that
[03:44] <gretelmk2> i've been doing that even though I actually run linux... it just makes it easy for me to do across all machines if i can drag my VM around
[03:45] <gretelmk2> so... are you the only one talking around here? has there been any other activity?
[03:45] <Jan-> very little
[03:45] <Jan-> Given how painful this whole compilation deal has been I'm really not tempted to try other linuxy things :)
[03:46] <gretelmk2> eh... this and VLC are the two toughest I've run across for compiling
[03:46] <gretelmk2> anything else is pretty easy
[03:46] <Jan-> I don't really want to compile ffmpeg anyway, I want to use the libraries it uses for something
[03:46] <Jan-> but it seemed like a good first step
[03:47] <gretelmk2> it's really the best step since most distros carry a crippled version of the thing
[03:47] <Jan-> well the version I had described itself as "crippled"
[03:48] <Jan-> and it had no external libraries
[03:48] <gretelmk2> yeah, that sucks
[03:48] <Jan-> I'm not sure how you'd have added them
[03:48] <gretelmk2> my little process above covered it
[03:48] <burek> Jan-, you could ask Zeranoe for help with windows builds
[03:49] <gretelmk2> it's half-assed, but you can't ignore the results
[03:49] <Jan-> this whole thing really does put the lie to the idea that "anyone can compile stuff"
[03:49] <burek> it is possible and it does take some knowledge to do it
[03:49] <Jan-> everyone has to be a software engineer in linux
[03:49] <burek> well windows runs only on 1 type of machines
[03:49] <gretelmk2> pfffft
[03:49] <burek> that's why everything "always works"
[03:49] <gretelmk2> i install linux on any machine that someone asks me to fix ;p
[03:50] <burek> linux has a lot of varieties so it's normal to be a little bit harder to get used to all that, but it pays off in the end
[03:50] <gretelmk2> "did you fix it", "Oh, I fixed it alright"
[03:50] <burek> :D
[03:50] <Jan-> this computer gets used for work stuff though
[03:50] <gretelmk2> use the VM option then ;p
[03:50] <Jan-> linux is OK for a hobby I guess
[03:50] <gretelmk2> yo, burek, got any ideas for my issue? ;p
[03:51] <burek> windows is good for games, i guess :)
[03:51] <gretelmk2> ha
[03:51] <gretelmk2> i've got one windows machine, just for games ;p
[03:51] <burek> anything even remotely serious is scary to run on windows
[03:51] <burek> well gretelmk2 did you try libav* libs?
[03:51] <gretelmk2> of course
[03:51] <gretelmk2> those are the ones
[03:51] <burek> which tutorials did you use
[03:52] <gretelmk2> pretty much trying to follow the example those guys did in ZoneMinder and also in the doc/examples/decode_encode.c taking in to consideration the recently deprecated functions in ffmpeg-0.10
[03:52] <burek> you could start off from here: http://ffmpeg.org/documentation.html
[03:52] <gretelmk2> started off there plenty ;p
[03:52] <burek> ok :)
[03:53] <gretelmk2> that's how i'm as far as i am
[03:53] <Jan-> it's open source
[03:53] <Jan-> don't expect docs
[03:53] <gretelmk2> some of the best docs i've seen have been from open source
[03:53] <burek> Jan-, why don't you pay Adobe to give you decent libs for your project?
[03:53] <burek> or Microsoft for that matter
[03:53] <Jan-> they just use quicktime
[03:54] <gretelmk2> that's not true, they use wma too :)
[03:54] <Jan-> well I could do it in directshow
[03:54] <Jan-> but that's completely unportable
[03:54] <Jan-> and I want a mac version.
[03:54] <gretelmk2> ew, macs ;p
[03:54] <burek> gretelmk2, did you try #libav ? :)
[03:54] <gretelmk2> didn't know it existed
[03:54] <Jan-> No. They won't support anyone using windows, I bet.
[03:55] <Jan-> I didn't hold out much hope here to be honest.
[03:55] <burek> well, why dont you contact apple and ask them to offer you their products/libs
[03:55] <burek> so you can have nice docs too
[03:55] <Jan-> They already do. Quicktime has great docs, it's just a bit of a hairy piece of software.
[03:55] <Jan-> 32 bit and only single threaded.
[03:55] <burek> well why dont you just pay for it and cut your pain :)
[03:55] <gretelmk2> i'm reading "terrible"
[03:56] <Jan-> but at some point it is possible to use quicktime
[03:56] <Jan-> I mean I have all the code for the libav libraries right here, but there's no info on how to actually get it into a state where I can use it.
[03:56] <Jan-> I'm just used to adding references in visual studio and things just ping into existence.
[03:56] <burek> Jan-, did you read http://ffmpeg.org/documentation.html
[03:56] <burek> I mean, I know you did, right? :)
[03:56] <burek> I'm just checking :)
[03:57] <Jan-> I skimmed it, but it's almost completely unhelpful.
[03:57] <Jan-> it's open source you don't expect docs
[03:57] <burek> blablabla
[03:57] <burek> you didn't even read it
[03:57] <Jan-> there are no examples or anything
[03:57] <burek> well, write one :)
[03:58] <burek> or donate to someone to write it for you :)
[03:58] <Jan-> I looked at libav.org too.
[03:58] <Jan-> There's nothing like you'd get on MSDN explaining how to implement anything.
[04:00] <Plorkyeran> the fun part is that by the time you finish writing something using ffmpeg, half the functions you used will be deprecated
[04:01] <Jan-> I'm just amazed that the whole IDE for linux appears to be "here's a big bunch of text files, you're on your own."
[04:01] <Jan-> I knew linux was pretty primitive, but this is just hilariously bad.
[04:03] <burek> omg..
[04:03] <pasteeater> someone probably got paid to write the MSDN docs
[04:03] <burek> if windows and msdn were so great, why are you even here Jan- ?
[04:03] <Jan-> I don't really care HOW they got there, I just care they exist.
[04:04] <burek> you are not even willing to read the docs
[04:04] <burek> you rather spend your time complaining
[04:04] <Jan-> There aren't really any docs
[04:04] <Jan-> I'm looking through the doxygen stuff right now and it's almost blank
[04:04] <burek> im a full to have this conversation with you at all
[04:04] Action: burek away
[04:04] <Jan-> Told you they wouldn't support windows development...
[04:08] <gretelmk2> eh, you need to be looking in the right doxygen files
[04:08] <pasteeater> the docs always need improvement. we are volunteers, but anyone can contribute.
[04:08] <Jan-> I think what I need to do is to write my own quicktime movie parser, sadly
[04:09] <Jan-> most of the types of file I need to use will have uncompressed audio, which is at least a plus
[04:09] <Jan-> the thing is I'd like to support various types of MPEG-4 streams as well, and at that point, it becomes kind of a big job to support them all.
[04:11] <burek> "mom, do I have to eat the dinner MYSELF? :crybaby:"
[04:11] <Jan-> jesus, do YOU want to independently develop quicktime and MPEG-4 TS readers?
[04:11] <gretelmk2> dude, the VM you can totally do it... cross compile that and voila
[04:11] <Jan-> even if that's true (which I doubt, having tried it on linux before) that would just get me an ffmpeg executable
[04:12] <Jan-> what I want is to be able to write C code, in windows, using the libraries.
[04:12] <Jan-> I assume that's possible but there seems to be no easy way to do it, short of just #including every file in the damn thing.
[04:12] <gretelmk2> would it not provide each of the libraries as well in DLL form?
[04:12] <gretelmk2> i'm just taking a stab there
[04:12] <Jan-> I guess I'd end up writing makefiles and configure scripts which I have no idea how to do.
[04:13] <gretelmk2> that's the spirit!
[04:13] <Jan-> I did once play with gcc, notepad, and DLL files representing the libav libraries.
[04:13] <Jan-> The problem is that they never had debug symbols in them so if it breaks, and it will break, you're clueless.
[04:13] <gretelmk2> well, dump notepad and use eclipse with the CDT package
[04:13] <pythonirc101> if I have a video file (could be flv, mp4, etc) and I want to convert it to mp4, can I just do ffmpeg -I input.vtype output.mp4?
[04:14] <burek> :)))
[04:14] <Jan-> pythonirc101: basically yes.
[04:14] <Jan-> you may want to set some sort of better quality output as the default is very low quality.
[04:14] <Jan-> but I don't know how to do that.
[04:14] <burek> pythonirc101, try ffmpeg -codecs
[04:14] <Jan-> gretelmk2: I think I'd rather keep it simple.
[04:14] <pythonirc101> burek: there are too many
[04:14] <burek> and then ffmpeg -i input -vcodec <codec> output,mp4
[04:15] <burek> is mp4 an encoder or a format or both
[04:15] <Jan-> strictly speaking I could p/invoke the avcodec and avformat libraries from inside c# which would be by far the easiest way to go.
[04:15] <pasteeater> mp4 is a container format and can contain several video formats.
[04:15] <pasteeater> i assume you want H.264.
[04:15] <gretelmk2> eh... it's only really messy when you start changing code to get it to compile
[04:15] <pythonirc101> indeed I do
[04:15] <burek> and you just want to remux the input into mp4 format/container?
[04:15] <pythonirc101> sometimes my input is indeed mp4
[04:15] <Jan-> But doing that without information on what those libraries want by way of data types and what their calling conventions were, could get messy.
[04:16] <pasteeater> ffmpeg -i input -c:v libx264 -preset medium -crf 24 output.mp4
[04:16] <pythonirc101> I get variable types as input. I want to make sure the output is mp4+h264 with the same quality as the input
[04:16] <burek> pythonirc101, well do you want to reencode video or just to remux it
[04:16] <gretelmk2> that's all in doxygen
[04:16] <burek> i see
[04:16] <pasteeater> pythonirc101: 1. 17:49:54 @michaelni | pasteeater, also make sure you use url = gil@ffmpeg.org:ffmpeg-web
[04:16] <burek> pasteeater, gave you the answer then
[04:16] <pasteeater> oops. stupid paste.
[04:16] <burek> :)
[04:16] <pasteeater> i do that too often
[04:16] <gretelmk2> it's practically all in these:
[04:16] <gretelmk2> #include <libavcodec/avcodec.h>
[04:16] <gretelmk2> #include <libavformat/avformat.h>
[04:16] <gretelmk2> #include <libswscale/swscale.h>
[04:17] <gretelmk2> doxygen those and *bam*
[04:17] <codepython777> pasteeater: what if the input is indeed in mp4 format with 264...does that command just detect it and copy input to output?
[04:17] <pasteeater> you have to tell it: ffmpeg -i input -c copy output.mp4
[04:17] <Jan-> gretelmk2: oh if only it were that simple
[04:17] <codepython777> thanks
[04:18] <Jan-> I did try that
[04:18] <Jan-> gcc complained
[04:18] <gretelmk2> oops
[04:18] <pasteeater> pythonirc101: 1) use the highest crf value that still gives an acceptable quality
[04:18] <pasteeater> 2) use the slowest preset that you have patience for
[04:18] <Jan-> Christ on a bike
[04:18] <Jan-> avcodec has a HUGE number of functions in it
[04:18] <gretelmk2> yep
[04:18] <gretelmk2> shitload
[04:18] <gretelmk2> and like someone else mentioned already, they deprecate pretty fast ;p
[04:18] <Jan-> someone *has* done a p/invoke wrapper for it
[04:19] <Jan-> but it's somewhat old and I suspect wouldn't work with modern versions of the libraries.
[04:19] <gretelmk2> there is an ffmpeg-java project out there... if you put stock in that sort of thing
[04:19] <Jan-> The problem I would anticipate is that the people who write ffmpeg are likely to be openly hostile to that sort of thing and actively work to make it break.
[04:20] <Jan-> The same issue exists with using it in C#.
[04:20] <Jan-> (which is technically more or less the same thing)
[04:21] <Jan-> I mean look at this: http://www.koders.com/csharp/fid1F783E459768CCD8254F1DAF770A489D19327B0A.as…
[04:24] <gretelmk2> eh, even if you're behind some, does it matter than much?
[04:24] <gretelmk2> i take the latest and greatest because I can, not because it's necessary
[04:24] <gretelmk2> though that sense of accomplishment is there ;p
[04:24] <codepython777> ` b09aSD
[04:25] <Jan-> That just makes life difficult.
[04:25] <Jan-> It's hard enough even to get hold of older versions.
[04:25] <Jan-> ah, fugheddaboudit
[04:26] <Jan-> another project nixed by opensource stupidity
[04:26] <gretelmk2> geez
[06:47] <vtorri> hey
[06:47] <vtorri> i'm cross compiling
[06:48] <vtorri> i have to set arch and cpu
[06:48] <vtorri> are those values the same than gcc options ?
[07:06] <plut0> is it possible to pass in x264 options to ffmpeg? such as sync-lookahead ?
[07:37] <vtorri> hmm, it's target-os that must be set
[09:44] <\DSAFEW\> I'm getting some issues trying to stream to justin tv using ffmpeg, it says the rtmp://live.justin.tv/app: No such file or directory for any rtmp file, I compiled it with rtmp libs, going to try it with native support next
[09:44] <\DSAFEW\> yep, same problem with native rtmp
[09:46] <\DSAFEW\> running a the command I found on some FAQ, with my options as the variables there, ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0 -f alsa -ac 2 -i hw:0,0 -vol 4096 -vcodec libx264 -vpre "$QUAL" -s "$OUTRES" -acodec libmp3lame -ab 96k -threads 0 -f flv "rtmp://live.justin.tv/app"
[09:47] <\DSAFEW\> and I commented out the stream key and flashver option, simply because I was testing for syntax errors
[15:44] <\DSAFEW\> I'm getting some issues trying to stream to justin tv using ffmpeg, it says the rtmp://live.justin.tv/app: No such file or directory for any rtmp file, I compiled it with rtmp libs, going to try it with native support next
[15:44] <\DSAFEW\> same problem with native rtmp
[15:45] <\DSAFEW\> running a the command I found on some FAQ, with my options as the variables there, ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0 -f alsa -ac 2 -i hw:0,0 -vol 4096 -vcodec libx264 -vpre "$QUAL" -s "$OUTRES" -acodec libmp3lame -ab 96k -threads 0 -f flv "rtmp://live.justin.tv/app"
[15:45] <\DSAFEW\> and I commented out the stream key and flashver option, simply because I was testing for syntax errors
[15:47] <Mavrik> hm, since you're using "-vpre"... just how old is your rrmpeg?
[15:47] <Mavrik> IIRC RTMP support is quite new in ffmpeg
[15:47] <\DSAFEW\> I'm just copying a command
[15:47] <solemnwarning> Hi
[15:48] <solemnwarning> Can anyone help me with encoding a series of PNG images and a wav file into uncompressed AVI?
[15:48] <\DSAFEW\> I've tried using the command with librtmp and with native support
[15:48] <\DSAFEW\> built on stable gentoo amd64
[15:48] <solemnwarning> I'm getting the video ok if I force pix_fmt to bgr24, but there's no audio and VLC sees the file as being 11:01 long (Is 1min worth of audio and video)
[15:49] <\DSAFEW\> ffmpeg-0.7.8 btw
[15:51] <Mavrik> \DSAFEW\, hmm... can you show the actual command line being executed with the exact output?
[15:51] <Mavrik> put it in a pastebin
[15:51] <Mavrik> (my local testing shows rtmp output working perfectly on ffmpeg 0.10)
[15:51] <\DSAFEW\> yeah I can do that, let me edit out my key
[15:52] <\DSAFEW\> http://paste.pocoo.org/show/542109/
[15:53] <Mavrik> not that
[15:53] <Mavrik> the EXACT command line that's run with all parameters
[15:53] <Mavrik> the EXACT ffmpeg output with all warnings
[15:53] <\DSAFEW\> that is the exact script I run, no parameters
[15:54] <\DSAFEW\> http://paste.pocoo.org/show/542110/
[15:54] <\DSAFEW\> output^
[15:54] <Mavrik> well
[15:54] <Mavrik> tell your script to print out exactly what's being executes
[15:54] <Mavrik> *executed
[15:55] <Mavrik> since as you see your parameters are wrong.
[15:55] <\DSAFEW\> wait a sec
[15:55] <\DSAFEW\> http://paste.pocoo.org/show/542111/
[15:55] <\DSAFEW\> sorry for the -v
[15:55] <\DSAFEW\> that was a mistake from me trying to troubleshoot
[15:56] <Mavrik> there's obviously something wrong with your generated command line
[15:57] <\DSAFEW\> compiling in the librtmp wrappers now
[16:01] <\DSAFEW\> http://paste.pocoo.org/show/542113/
[16:01] <\DSAFEW\> not sure but I suspect that's the same as before
[16:02] <Mavrik> as I said, your problem isn't an ffmpeg problem
[16:02] <Mavrik> fix your script.
[16:03] <solemnwarning> This is the command/output I'm trying to create an uncompressed AVI with: http://slexy.org/view/s2ZT45mh2G
[16:03] <\DSAFEW\> what's the proper syntax to do an flv stream on an rmtp protocol?
[16:04] <\DSAFEW\> I did read the man page, and I googled it for a few hours
[16:04] <Mavrik> solemnwarning, that's wierd... can you try specifying rate (-r) also after -i?
[16:04] <Mavrik> \DSAFEW\, nothing wrong with the syntax
[16:04] <Mavrik> something is wrong with how you're building the command line string.
[16:06] <solemnwarning> Mavrik: No change
[16:06] <solemnwarning> There's no audio or video data after the 1:00 mark I think, VLC just leaves the last frame displayed if I skip ahead
[16:07] <Mavrik> that's hellova wierd
[16:07] <solemnwarning> I can create x264 videos from the same input too, no problems
[16:08] <Mavrik> but only when trying to save them uncompressed?
[16:08] <Mavrik> you get the problem?
[16:08] <solemnwarning> Yeah
[16:08] <Mavrik> solemnwarning, it could be a problem with the .avi muxer
[16:08] <solemnwarning> I haven't tried x264 and PCM/AVI yet actually, I'll try that
[16:08] <Mavrik> solemnwarning, can you try to store the content in .mkv or something to see if it's indeed .avi issue?
[16:09] <solemnwarning> Sure
[16:12] <solemnwarning> x264/PCM in AVI has no audio, VLC warns that it doesn't support codec "undf" when I try playing the raw/pcm in an mkv, but it plays the audio fine
[16:14] <Mavrik> figured, containers don't support it :\
[16:16] <solemnwarning> As in AVI doesn't support raw audio? o_O
[16:16] <Mavrik> solemnwarning, more like raw video
[16:16] <Mavrik> at least in case of mkv it seems
[16:16] <Mavrik> sadly I can't help you much with AVI, never really did anything with that container :\
[16:17] <solemnwarning> x264/AAC in an AVI also has no audio
[16:19] <Mavrik> AVI has some serious limitations with modern formats, so check if you're not violating them
[16:19] <solemnwarning> Am I using a modern format with rawvideo/pcm_a16le?
[16:19] <solemnwarning> *pcm_s16le
[16:20] <solemnwarning> Or the bgr24 pixel format?
[16:20] <Mavrik> rgb support could be problematic, it usually is
[16:23] <solemnwarning> The default format (pal8) results in screwed up colours and no sound
[16:59] <elkng> is there options to "Rebuilds index of files" similar to "-idx" in mplayer ?
[17:09] <grepper> elkng: maybe by just using -vcodec copy -acodec copy ?
[17:24] <fling> hello :]
[17:24] <fling> i want to record a video from webcam and sound from two soundcards same time
[17:24] <fling> stereo from first and mono from second
[17:25] <fling> how to do that?
[17:25] <fling> this is my old script with single mono recording http://paste.pocoo.org/show/542158/
[17:26] <fling> how to record one more stereo track?
[18:01] <elkng> grepper: for ffplay
[19:22] <plut0> i want to encode a video to x264. i need to use the x264 app because certain flags aren't available in ffmpeg. seeing how x264 only does video, how can i merge the audio and video back together?
[19:26] <Mavrik> plut0, I suggest you use "x264opts" parameter for ffmpeg to pass the "missing" parameters to x264 instead of doing what you do
[19:26] <Mavrik> but otherwise you just use "copy" codec for video from source
[19:30] <plut0> Mavrik: hmm i think that worked. i tried x264opts last night without luck but maybe i was just tired
[20:24] <fling> Unknown input format: 'video4linux2'
[20:25] <fling> -f video4linux2 -s 864x480 -r 15 -i /dev/video0
[20:25] <fling> worked in the past... what is new value of v4l?
[20:29] <fling> [alsa @ 0x2547780] cannot set sample format 0x10000 2 (Invalid argument)
[20:37] <fling> ^ -f alsa -ac 2 -i hw:0,0
[20:37] <fling> but can record with audacity
[20:38] <fling> what am i doing wrong?
[21:23] <WilliamWS> I want to make a sound graph using C language. Someone can tell me a good way/library to do that?
[21:24] <relaxed> WilliamWS: try ##c
[21:25] <WilliamWS> relaxed, It is a sound bargraph
[21:25] <fling> still can't record... -f alsa -ac 2 -ar 48000 -i hw:0,0
[21:25] <fling> [alsa @ 0x14177c0] cannot set sample format 0x10000 2 (Invalid argument)
[21:25] <fling> hw:0,0: Input/output error
[21:25] <WilliamWS> I do not know how to call it in english... maybe a RTA
[21:29] <saste> WilliamWS: what type of "sound graph"?
[21:29] <WilliamWS> about 8 bars, each one for an specific frequence range
[21:31] <fling> what is 'sample format' ? why can't ffmpeg set it?
[21:32] <WilliamWS> something like this http://www.scifun.ed.ac.uk/pages/about_us/shows/tables/Hi-freq-large.gif saste
[21:32] <relaxed> WilliamWS: oh, like this? --> http://www.desktoplinux.com/files/article101/MandrakeMusicChapter/images/xm…
[21:32] <WilliamWS> but with movement... in real time
[21:33] <WilliamWS> yes relaxed
[21:33] <fling> like projectM? :p
[21:33] <saste> WilliamWS: you can do that with an A->V filter
[21:33] <saste> check my thread on ffmpeg-devel about showaudio
[21:33] <relaxed> WilliamWS: ffplay does something like that when you play audio. have you seen it?
[21:34] <Mavrik> WilliamWS, you'll need to do FFT
[21:34] <Mavrik> WilliamWS, I think DSP books/resources will help you :)
[21:34] <WilliamWS> I can do it in real time using the microfone input or line in?
[21:34] <WilliamWS> fling what is Project AM?
[21:35] <fling> http://projectm.sourceforge.net/
[21:36] <WilliamWS> It is for a RTA.
[21:36] <WilliamWS> Real Time Audio Analyzer
[21:36] <WilliamWS> It theoreticaly will be used for audio tunning
[21:40] <WilliamWS> How can I get the audio from sound card and process it and display the image? Has any lib that do at least a bit of this work?
[21:51] <relaxed> WilliamWS: ffplay already does something like this.
[21:54] <WilliamWS> yeah
[21:54] <WilliamWS> hey can I get the signal from the line in with ffplay?
[21:55] <relaxed> I would say you should look at how ffplay does it in its source and go from there.
[21:55] <Mavrik> WilliamWS, you basically need a library that can grab audio from microphone (shouldn't be hard to Google), then use a DSP package to process that
[21:55] <WilliamWS> If ffplay do it I can use ffplay in the RTA.
[21:56] <Mavrik> tying to do that with ffmpeg will be... painful
[22:04] <saste> Williams: ffmpeg (and thus ffplay) supports reading from microphone through alsa/pulse/jack whatever
[22:05] <saste> best approach is an audio visualization filter
[23:14] <DelphiWorld> hello folks
[23:14] <DelphiWorld> what the best codec to use for low bandwith streaming ?
[23:15] <Mavrik> x264
[23:15] <Mavrik> with AAC
[23:16] <DelphiWorld> Mavrik: the audio is important, there's no video
[23:16] <WilliamWS> if I have a 9Kbp/s ?
[23:17] <DelphiWorld> Mavrik: what do you suggest for a streaming point, streaming server and client ?
[23:17] <DelphiWorld> WilliamWS: :)
[23:17] <DelphiWorld> seriously, any recomandation ?
[23:17] <DelphiWorld> i am setting a radio station
[23:17] <DelphiWorld> for blind users
[23:17] <Mavrik> DelphiWorld, for very low bitrates (64kbps or less) I've had great results with HE-AAC+
[23:18] <Mavrik> (also known as AAC+)
[23:18] <Mavrik> but that's not universally compatible
[23:18] <WilliamWS> DelphiWorld I used a 9kbp/s internet connection for about 6 months
[23:18] <Mavrik> sadly, i've mostly done video streaming not just audio
[23:18] <DelphiWorld> Mavrik: what should i use please
[23:19] <DelphiWorld> like i say is for a radio station for blind users community
[23:20] <mchinen> I am recording the deskop with ffmpeg via x11grab to x264 mp4 output (for youtube) and would like to simultaneously stream this justin.tv via rtmp x264 flv with ffmpeg - is there some more efficient way to do this than opening up two instances ffmpeg (which means two x11grabs and 2 x264 encoders)?
[23:21] <Mavrik> mchinen, just add another output parameter chain to the command line
[23:21] <mchinen> to make it a bit more complicated a new instance of the mp4 ffmpeg is started and stopped every 15 minutes (to fit the youtube length restrictions)
[23:21] <Mavrik> mchinen, right after the first output file
[23:21] <mchinen> Mavrik: Oh cool, I didn't know you could do that.
[23:22] <Mavrik> mchinen, so you do something like ffmpeg -i <something> -codec:v libx264 .... output.mp4 -codec:v libx264 ... -f flv rtmp://...
[23:22] <Mavrik> you just need enough CPU power :)
[23:22] <Mavrik> and be careful of parameter ordering ;)
[23:22] <DelphiWorld> so, i'm open for suggestion
[23:23] <mchinen> Mavrik: great, I'll try it, thanks! We'll see how my cpu handles.
[23:23] <Mavrik> DelphiWorld, I gave you a suggestion.
[23:23] <DelphiWorld> Mavrik: client? streaming server? player ?
[23:23] <WilliamWS> vlc?
[23:24] <DelphiWorld> WilliamWS: sadly is bot too easy for blinds :(
[23:24] <DelphiWorld> QT...QT...QT... the worst
[23:24] <WilliamWS> why?
[23:25] <WilliamWS> what is the hard part of lc for blinds?
[23:25] <DelphiWorld> WilliamWS: the qt is not well accessible for blind users
[23:25] <WilliamWS> vlc*
[23:25] <DelphiWorld> mostly all blind users don't use it
[23:26] <WilliamWS> vlc can run in terminal...
[23:26] <WilliamWS> but it would be the same as ffmpeg, nosens sorry
[23:27] <DelphiWorld> WilliamWS: yes, and you expect users to listen through it?
[23:27] <DelphiWorld> no one can do that :)
[23:27] <DelphiWorld> need to make it too simple for blind users, maybe through web
[23:27] <Mavrik> use flash player then, and make sure it has accessibility features
[23:28] <DelphiWorld> Mavrik: yes, i allready use one for mp3
[23:28] Action: DelphiWorld love Mavrik
[23:29] <DelphiWorld> Mavrik: what about the streaming poing
[23:29] <DelphiWorld> Mavrik: what about the streaming point
[23:37] <DelphiWorld> Mavrik ?
[23:38] <cbreak> you want to use blind users as streaming point?
[23:39] <DelphiWorld> cbreak: yes, everything blind :)
[23:39] <DelphiWorld> streams, and listeners
[23:39] <cbreak> ffmpeg should be usable by blind people, as long as they have some means to read the terminal output
[23:40] <WilliamWS> I am thinking from the point that he have noa ccess to those blinds.
[23:40] <WilliamWS> if he does then vlc is suitable
[23:41] <DelphiWorld> WilliamWS: he's a windows user and he never used console :(
[23:41] <DelphiWorld> cbreak: he's a windows user. he use only gui ;)
[23:41] <WilliamWS> Is only 1 person?!
[23:41] <cbreak> gui for a blind person?
[23:41] <cbreak> isn't that kind of stupid?
[23:42] <DelphiWorld> cbreak: no, gui is usable a lot in windows
[23:42] <DelphiWorld> cbreak: i use gui also without any issue at all
[23:42] <DelphiWorld> cbreak: i am blind too
[23:42] <WilliamWS> cbreak touchscreen bro ahahaha
[23:42] <cbreak> touch screens have no feedback
[23:42] <cbreak> although I have seen some proto types
[23:42] <DelphiWorld> WilliamWS, cbreak Jaws for windows is the mostly used one
[23:42] <cbreak> but they didn't feel that interactive
[23:43] <DelphiWorld> cbreak: i use touchscreen but in iOs
[23:43] <cbreak> it was mostly just texture feeling
[23:43] <cbreak> and how do you know what to touch?
[23:44] <WilliamWS> you are blind not stupid ><
[23:44] <WilliamWS> te screen is a square
[23:44] <DelphiWorld> cbreak: i touch anything on the screen and i get a audio fidback
[23:44] <WilliamWS> you can make a image projection on your mind
[23:44] <DelphiWorld> but is not yet processed
[23:44] <DelphiWorld> i'm moving my finger
[23:44] <cbreak> so you're scanning the screen
[23:44] <DelphiWorld> looking for cbreak on my contact list
[23:44] <WilliamWS> not an image but a squared tate ideia
[23:44] <DelphiWorld> if i found cbreak, i can put a double touch into it and i'm done :)
[23:44] <DelphiWorld> same for all elemants
[23:45] <cbreak> sounds inefficient if you could just as well use a keyboard and a command line shell
[23:45] <WilliamWS> DelphiWorld is only 1 person?
[23:45] <DelphiWorld> cbreak: yes, i know but iOs... :P
[23:45] <DelphiWorld> WilliamWS: mean ?
[23:45] <WilliamWS> only 1 blind user for your intention?
[23:46] <cbreak> I use mplayer on the command line just fine, and I am not even blind.
[23:46] <DelphiWorld> WilliamWS: no, me, and lot of folks starting this station
[23:46] <WilliamWS> You want a radio or you only want stream audio for a singles person?
[23:46] <cbreak> (for playback)
[23:46] <cbreak> for streaming I've used cvlc, which is also a command line program
[23:46] <DelphiWorld> WilliamWS: a radio station like i say :)
[23:46] <DelphiWorld> but web
[23:46] <cbreak> but I've only used it for HTTP Live Streaming, although it supports a lot more
[23:46] <DelphiWorld> so i want blind users to be listening simply and easyly
[23:47] <WilliamWS> aha, vlc can stream to windows media player as well
[23:47] <WilliamWS> ffmpeg does the same
[23:47] <cbreak> HTTP Live Streaming supports iThings
[23:47] <DelphiWorld> yes i know WilliamWS but like i say, vlc is shit QT
[23:47] <WilliamWS> but with some effort to configure
[23:47] <DelphiWorld> and QT is not a accessible gui for blind users
[23:47] <cbreak> cvlc is console
[23:47] <WilliamWS> but you will be in the server not?
[23:47] <cbreak> (command line shell)
[23:47] <DelphiWorld> yes yes
[23:47] <DelphiWorld> any other solution for windows Os ?
[23:47] <DelphiWorld> but the server
[23:48] <DelphiWorld> i can do vlc no problem
[23:48] <WilliamWS> or the blinds will maintain the radio?
[23:48] <cbreak> you'd write a script that uses cvlc for streaming
[23:48] <DelphiWorld> but for the streaming point is too dificult
[23:48] <DelphiWorld> WilliamWS: server, no wory.
[23:48] <DelphiWorld> WilliamWS: server is me, but the radio is another streaming point that stream to the server like icecast
[23:48] <WilliamWS> cbreak I would suggest it but I was thinking he wanted to use a stream from his house for some blinds
[23:48] <DelphiWorld> and clients connect to icecast or some server and listen
[23:49] <cbreak> WilliamWS: so?
[23:49] <cbreak> cvlc can serve as streaming server
[23:49] <DelphiWorld> but what about the streamer
[23:49] <cbreak> it is a streamer.
[23:49] <cbreak> it streams.
[23:49] <WilliamWS> vlc is a streamming system not?
[23:49] <DelphiWorld> i don't want cvlc to stream from file or sound card, but to accept stream from a streamer and deliver it
[23:49] <WilliamWS> vlc is the most complete player (wqith gui) you will find
[23:50] <cbreak> I use cvlc to stream from a different program
[23:50] <DelphiWorld> :P
[23:50] <WilliamWS> it does...
[23:50] <cbreak> which I wrote myself
[23:50] <DelphiWorld> i see
[23:50] <DelphiWorld> let's browse arround and see
[23:50] <DelphiWorld> maybe i will try darwine streaming server
[23:50] <cbreak> if it spits out data that vlc can play, then cvlc can stream it
[23:50] <WilliamWS> vlc can get media from everywhere and send to everywhere.
[23:50] <WilliamWS> in every format
[23:50] <cbreak> not every
[23:50] <cbreak> but many
[23:50] <DelphiWorld> WilliamWS: what do you think about darwine streaming server
[23:51] <DelphiWorld> haha cbreak good catch:P
[23:51] <cbreak> I've only used darwin streaming server once
[23:51] <cbreak> it worked for files, then I switched to vlc. No particular reason.
[23:51] <WilliamWS> I do not know darwine sorry
[23:51] <cbreak> darwin streaming server is apple's streaming server
[23:51] <DelphiWorld> dss
[23:52] <WilliamWS> I am from Brasil, here we make the entire server many times ahahaha
[00:00] --- Sun Jan 29 2012
1
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[00:24] <durandal_1707> hmm, why hflip is only tested for some pixfmts?
[00:27] <michaelni> it should be tested for all that it supports
[01:05] <durandal_1707> Compn: you have videc.dll?
[01:05] <Compn> uhhh
[01:09] <durandal_1707> it is for fourcc VDEC
[01:09] <Compn> i dont see it on my drive
[01:09] <Compn> i'll have to look what installer installs it
[01:09] <Compn> do you have vdec sample ?
[01:11] <durandal_1707> Compn: downloaded sample from usual location
[01:11] <durandal_1707> vdec.avi in V-codecs, from 1996 year
[01:11] <Compn> oh cool
[01:11] <Compn> wonder when that showed up
[01:12] <durandal_1707> if ever manage to RE something ...
[01:14] <Compn> give me a few mins durandal_1707 , i'll try to locate dll again
[01:15] <Compn> durandal_1707 : ftp://ftp.logitech.com/pub/techsupport/quickcam/color/
[01:20] <Compn> and heres the dll file
[01:45] <durandal_1707> Compn: sending files via irc actually works?
[01:47] <Compn> it does if you hit accept butto n
[01:47] <Compn> want me to send again ?
[01:48] <durandal_1707> no need
[02:26] <CIA-108> ffmpeg: 03Martin Storsjö 07master * r8801fac365 10ffmpeg/tools/ismindex.c:
[02:26] <CIA-108> ffmpeg: ismindex: Fix build on mingw
[02:26] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[02:26] <CIA-108> ffmpeg: 03Alex Converse 07master * rd2ee8c1779 10ffmpeg/libavformat/matroskadec.c:
[02:26] <CIA-108> ffmpeg: matroskadec: Pad AAC extradata.
[02:26] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[02:26] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[02:26] <CIA-108> ffmpeg: 03Janne Grunau 07master * rb3461c29c1 10ffmpeg/libavformat/utils.c:
[02:26] <CIA-108> ffmpeg: lavf: prevent infinite loops while flushing in avformat_find_stream_info
[02:26] <CIA-108> ffmpeg: If no data was seen for a stream decoder are returning 0 when fed with
[02:26] <CIA-108> ffmpeg: empty packets for flushing. We can stop flushing when the decoder does
[02:26] <CIA-108> ffmpeg: not return delayed delayed frames anymore. Changes try_decode_frame()
[02:26] <CIA-108> ffmpeg: return value to got_picture or negative error.
[02:26] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[02:26] <CIA-108> ffmpeg: 03Paul B Mahol 07master * r7de9af65c7 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/xwd):
[02:26] <CIA-108> ffmpeg: fate: add XWD image regression test
[02:26] <CIA-108> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de>
[02:27] <CIA-108> ffmpeg: 03Alex Converse 07master * rdac56d9ce0 10ffmpeg/libavcodec/qdm2.c:
[02:27] <CIA-108> ffmpeg: qdm2: Check data block size for bytes to bits overflow.
[02:27] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[02:27] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[02:27] <CIA-108> ffmpeg: 03Alex Converse 07master * r9adf25c1cf 10ffmpeg/libavcodec/smacker.c:
[02:27] <CIA-108> ffmpeg: smacker: Sanity check huffman tables found in the headers.
[02:27] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[02:27] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[02:27] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r01fcbdf9ce 10ffmpeg/: (log message trimmed)
[02:27] <CIA-108> ffmpeg: Merge remote-tracking branch 'qatar/master'
[02:27] <CIA-108> ffmpeg: * qatar/master:
[02:27] <CIA-108> (4 lines omitted)
[10:00] <Tjoppen> bcoudu: what do you think about ignoring MaterialPackage and just looking for LinkedPackageUID in EssenceContainerData and just use the linked SourcePackage?
[10:02] <bcoudu> Tjoppen, that could be possible
[10:02] <Tjoppen> I got a fun sample at work with improperly linked SourceTracks, which got me thinking along those lines
[10:02] <Tjoppen> though it looks like I can nag them
[10:02] <bcoudu> although material is supposed to be the reference
[10:02] <Tjoppen> yeah. akin to MOV EDLs
[10:03] <Tjoppen> though we don't support StartPosition etc. atm :)
[10:03] <bcoudu> do people mess with startposition ? :)
[10:03] <Tjoppen> I ran mxf2dot on the file -> 15 MaterialTracks point to the same audio SourceTrack, including the video MaterialTrack :\
[10:04] <Tjoppen> so I'm going send a little note to them along the lines of "fix your muxer"
[10:04] <Tjoppen> I dunno. no one really uses MXF for what it's supposed to be used for AFAICT
[10:06] <Tjoppen> I could see one use case though: cutting out clips with sub-GOP precision
[10:07] <Tjoppen> odd. the file seems to have been written by MXF4mac. I thought they knew this stuff well enough to not make mistakes like these
[10:22] <bcoudu> true
[10:24] <Tjoppen> analysis sent. hopefully whomever is responsible fixes their code
[10:25] <Tjoppen> -ignore_mp AVOption could be useful. I'm not terribly concerned yet though
[11:41] <kierank> Compn: "i wish i could predict x264 merging with ffmpeg. that would be fun. then maybe x262 and xvp8."
[11:41] <kierank> x262 won't be merging with x264 fwiw
[11:41] <kierank> it's too unwieldy to maintain already
[13:06] <Compn> kierank : i meant x262 + ffmpeg
[13:11] <burek> something is broken with libaacplus in new ffmpeg, it gives segfault.. I'll need to investigate further with gdb
[15:27] <burek> is this a bug in configure script or I'm doing something wrong? http://pastebin.com/7Un8X4vp
[17:45] <durandal_1707> burek: perhaps it is looking for files in /usr/lib?
[17:46] <burek> hm, im pretty sure i apt-get removed all the stuff I compiled myself, but thanks, I'll check again
[17:46] <durandal_1707> this may (not) help: --extra-ldflags=-L/usr/local/lib
[17:46] <burek> oh, cool :)
[17:47] <burek> that way it first looks into /usr/loca/lib?
[17:47] <durandal_1707> it looks also in /usr/local/lib
[17:48] <burek> ok, I'll try that too
[17:49] <burek> bingo
[17:49] <burek> you're the man :D
[17:49] <burek> thanks! :beer:
[20:59] <ubitux> michaelni: it seems the max duration analysis is triggered by the fact that first_dts == AV_NOPTS_VALUE
[21:00] <ubitux> it also seems that not only mp3 are affected, but flac as well (and maybe more)
[21:06] <ubitux> the pts/dts stuff is really obscure to me
[21:06] <ubitux> and the heuristic in that file looks very complex
[21:06] <ubitux> do you have a hint on what i should look for?
[21:07] <nevcairiel> hm, do raw audio files like flac even come with timestamps?
[21:08] <ubitux> is flac really a raw audio file?
[21:08] <ubitux> iirc it can contains metadata and such
[21:08] <nevcairiel> hm
[21:08] <nevcairiel> i guess it can
[21:08] <ubitux> what makes it a "raw audio file" then?
[21:08] <ubitux> only one stream, and flac?
[21:09] <nevcairiel> dunno. I think of all files that contain only audio that way, for playback they don't need timestamps as every audio frame has a well defined duration (unlike video)
[21:09] <ubitux> mmh ok
[21:10] <nevcairiel> easy enough to check the flac demuxer and see if it fills pts/dts
[21:10] <nevcairiel> nope, it does not
[21:10] <ubitux> yep seems not
[21:11] <ubitux> so maybe the audio check could be removed here?: if(st->first_dts == AV_NOPTS_VALUE && (st->codec->codec_type == AVMEDIA_TYPE_VIDEO || st->codec->codec_type == AVMEDIA_TYPE_AUDIO))
[21:11] <nevcairiel> for real container formats it would still make sense, i suppose
[21:11] <ubitux> i guess it would break a lot of things here&
[21:12] <nevcairiel> mp3 doesnt even output full frames, it just always gives you 1024 bytes, relying on the parser to figure it out
[21:14] <ubitux> there is a parser check just above
[21:14] <nevcairiel> maybe a format flag would be helpful to identify such formats where waiting for a timestamp is useless
[21:14] <ubitux> but it doesn't seem to be triggered with the mp3
[21:15] <nevcairiel> flag on the demuxer, i mean
[21:15] <ubitux> mmh
[21:15] <ubitux> i remember a NOTIMESTAMP flag or something
[21:15] <ubitux> AVFMT_NOTIMESTAMPS
[21:15] <nevcairiel> oh right, that exists already
[21:16] <ubitux> the only decoder using it seems to be the ico one
[21:17] <nevcairiel> sounds reasonable to use it to disable the dts check in the probe function
[21:18] <ubitux> ok
[21:23] <ubitux> http://pastie.org/3265443
[21:23] <ubitux> this seems to do the trick
[21:24] <ubitux> thank you nevcairiel
[21:28] <ubitux> mmh unfortunately this breaks some seeking
[21:29] <ubitux> (with mp2)
[21:32] <nevcairiel> hm, it does?
[21:35] <ubitux> yes http://pastie.org/3265511
[21:42] <ubitux> saste: oh you're here :)
[21:43] <ubitux> saste: as i said on ffmpeg-devel, i tried to copy_props in the filter_samples callback to fix the -t being ignored with ffmpeg
[21:43] <ubitux> as well as copying pts/dts in the samples ref
[21:44] <ubitux> but it didn't help
[21:44] <ubitux> i'm sure i'm just missing something to do in the that callback
[21:44] <ubitux> because it works before the rewriting with libswr
[21:44] <ubitux> any idea what could it be?
[22:58] <saste> ubitux: debugging debugging debugging
[23:01] <ubitux> ok :)
[23:44] <burek> can I get some more help with building ffmpeg as a static library
[23:44] <burek> I've managed to build the basic thing, without much external libraries
[23:44] <burek> and I noticed I don't have 'alsa' in indevs/outdevs
[23:45] <burek> so I checked config.log and these were the errors I've got: http://pastebin.com/e6hW0B2E
[23:45] <burek> this is how I configured all that: http://pastebin.com/Ld1SZWPX
[23:47] <burek> btw, I have libasound.a and libasound.la in /usr/lib
[00:00] --- Sat Jan 28 2012
1
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[00:02] <madsage> that might be an option
[00:04] <boo> madsage, dumb q, but would it be possible to create an flv stream which would be at least watchable ( the quality) and would be streamed with ~ 30-40kb/s? :) I think there should, no problems with streeming perfect webcam video over skype..
[00:29] <michaelni> Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | For bug reports, read: http://bit.ly/cqvkhs | Ubuntu compilation guide: http://bit.ly/3xSE5 | FFmpeg 0.10 is released | For x264 encoding use -preset and/or -tune and/or -profile | FFmpeg forum: http://ffmpeg.test-lab.ch/ | Prores lives! | This channel is publically logged
[00:30] <ubitux> better with the /topic :)
[00:41] <madsage> hey i have a problem. i am streaming a file via ffmpeg to wowza. it seems when ffmpeg finished encoding the wowza server stream stops. like ffmpeg is goign to fast
[01:04] <pasteeater> madsage: aq value is dependent on the encoder. for example the values will be different for libmp3lame and libfaac.
[01:05] <pasteeater> as for libmp3lame -aq is mapped to 'lame -V'
[01:06] <pasteeater> and for libfaac -aq is mapped to 'faac -q'
[01:06] <pasteeater> see http://wiki.hydrogenaudio.org/index.php?title=LAME#Detailed_explanation_.28…
[01:06] <pasteeater> lower value is higher quality for libmp3lame -aq
[01:07] <pasteeater> undercash: no, i know nothing of sdp and rtmp
[01:07] <undercash> oki
[01:07] <undercash> it worked partially for me
[01:07] <undercash> just think i have a problem with rtmp port
[02:31] <Tony_V> so i'm trying to make a qcif (176x144) 3gp video for mobile using a 608x344 source video. the output still isn't coming out in widescreen
[02:32] <Tony_V> this is the command i'm using:
[02:32] <Tony_V> ffmpeg -i input.mp4 -f 3gp -s qcif -aspect 16:9 -vf scale=176:144 -vcodec h263 -b 200k -ar 24000 -ac 2 -acodec libfaac -ab 64k output.3gp
[02:32] <Tony_V> can anybody help a brotha out?
[02:33] <Tony_V> i think 608x344 for widescreen would be 176x100
[03:20] <pasteeater> Tony_V: -f 3gp -vf scale=176:-1
[03:55] <Tony_V> thanks pasteeater but it seems like i am getting this now:
[03:55] <Tony_V> [h263 @ 0x51ee20] The specified picture size of 176x100 is not valid for the H.263 codec.
[03:55] <Tony_V> Valid sizes are 128x96, 176x144, 352x288, 704x576, and 1408x1152. Try H.263+.
[04:36] <arbin_> Tony_V, try padding it out
[04:43] <DeezGz> hello every body
[05:22] <DeezGz> anyone in here
[09:04] <morteza> Hi can anyone help me ?
[09:04] <morteza> http://pastebin.com/4HjFy4dw
[09:05] <morteza> I can not convert wmv stream to .ffm
[09:05] <morteza> I want feed ffserver
[09:21] <morteza> any help for this paste bin: http://pastebin.com/4HjFy4dw
[09:21] <DeezGz> pastbin not valid
[09:22] <morteza> it's expired I think.
[09:22] <DeezGz> linux or windows
[09:22] <morteza> linux
[09:22] <DeezGz> do u have all the codec compiled
[09:23] <morteza> http://pastebin.com/B44EjSWW
[09:24] <morteza> I tried other commands changing vprofile or size with no success
[09:25] <DeezGz> i see
[09:25] <morteza> I want to feed ffserver
[09:25] <morteza> ;)
[09:28] <morteza> DeezGz may I ask for your help?
[09:28] <DeezGz> i dont know if i can help
[09:28] <DeezGz> is this a never ending stream?
[09:28] <morteza> yes
[09:29] <DeezGz> brb
[09:29] <morteza> I can convert it if I add -f mpegts but ffserver does not accept it
[09:32] <DeezGz> is ffm mp4 format?
[09:33] <morteza> how should I know?:D
[09:34] <DeezGz> lets google it to see
[09:34] <morteza> I couldn't find what is ffm
[09:36] <DeezGz> wait u where trying this huh?
[09:36] <DeezGz> ffmpeg -i "mmsh://217.218.67.243/presstv" http://localhost:8090/imamhossein.ffm
[09:37] <morteza> yes
[09:37] <DeezGz> try this : ffmpeg -i "mmsh://217.218.67.243/presstv" imamhossein.ffm
[09:37] <DeezGz> ffmped does not have a http server, and i wouldnt trust it to up load
[09:37] <morteza> its working now
[09:38] <morteza> :)
[09:38] <morteza> thank you. can you help me on ffserver?
[09:38] <DeezGz> i can try never used it befor
[09:39] <morteza> You know, I want to restream to rtsp and http
[09:39] <morteza> let me try it to see if I have any problem. I tried before with no success.
[09:39] <DeezGz> ok let see, u using ubuntu or what linux
[09:40] <DeezGz> do u have vlc?
[09:41] <morteza> ubuntu
[09:41] <morteza> I tried VLC
[09:41] <morteza> but it can not decode wmv3
[09:41] <morteza> I compiled both ffmpeg and vlc
[09:41] <DeezGz> ok wait one sec
[09:41] <morteza> I could get ffmpeg decode wmv3 but vlc can not decode it.
[09:41] <morteza> ok.
[09:44] <DeezGz> ok still looking brb
[09:48] <DeezGz> you said ffserver
[09:48] <morteza> yes
[09:48] <DeezGz> ok let me look
[09:49] <morteza> I could get ffserver running but I can not get rtsp out of it!
[09:52] <armetiz> Hi there,
[09:52] <morteza> Hi
[09:52] <armetiz> I'm trying to script ffmpeg encoding inside a Sheel script.
[09:52] <DeezGz> ask hi see if he can help
[09:52] <morteza> he has a problem for himself;) :D
[09:53] <DeezGz> hello armetiz, do u know how to work ffserver to stream a rstp stream
[09:53] <armetiz> I transform some 4/3 to 16/9 adding some pad. It work fine with static value and with specific input size / output size.
[09:53] <armetiz> DeezGz, I donno
[09:53] <DeezGz> ok
[09:54] <DeezGz> thx
[09:54] <armetiz> And I want to know If ffmpeg generate "variable" that can be reused concerning widthXheight of video input.
[09:54] <armetiz> I say this, because I have see that -vf & pad can be used with some variables.
[09:55] <armetiz> like in_w or in_h..
[09:56] <armetiz> DeezGz, I'm using HTTP server with x264 code-shop module to stream videos.
[09:56] <DeezGz> ok
[09:56] <DeezGz> linux right
[09:57] <armetiz> yes
[09:57] <armetiz> unix based I guess
[09:58] <DeezGz> yea i dont know about the var, i think it is possible with ffmeg
[09:58] <DeezGz> its for the web right
[09:59] <armetiz> I'm looking on google.. But, all resultat that I got is about using ffmpeg information about video & shell extraction. But, it's not so easy as ffmpeg solution :p
[09:59] <DeezGz> sorry morteza i am look for ffserver for win, i dont feel like lunching my vm and compileing ffmpeg and ffserver right now
[10:00] <DeezGz> yea documentation is all over the place in et languge
[10:01] <morteza> DeezGz Its not a problem with win or linux
[10:01] <morteza> :)
[10:01] <morteza> anyway thank you for your help
[10:02] <morteza> why do you need to compile ffserver or ffmpeg?
[10:02] <DeezGz> i dont have a unix env to help you
[10:02] <morteza> aha.
[10:02] <DeezGz> my vlc cond decode the stream just fine
[10:03] <DeezGz> but yours has a codec that wasnt compile for you vlc
[10:03] <morteza> yes in my own desktop ubuntu I can use vlc to decode wmv
[10:03] <DeezGz> so u can
[10:03] <DeezGz> ???
[10:03] <morteza> but in server I can not. I don't know what is happening
[10:03] <morteza> ( In my own ubuntu desktop I can, in my server with ubuntu server I can not )
[10:04] <morteza> I tried many libraries, compiling from source, installing from repositories
[10:04] <morteza> but couldn't get vlc decoding wmv
[10:04] <DeezGz> yea got to compile your self with the right depen and the right switches
[10:05] <DeezGz> its a pain and the ass
[10:06] <morteza> I'm wondering why two ubuntu systems with same versions differ;)
[10:06] <DeezGz> what do u mean
[10:07] <DeezGz> like server vs desktop version
[10:07] <DeezGz> its the packages it comes with that all
[10:07] <morteza> the only difference is x11
[10:08] <morteza> I installed all other libraries related to audio and video
[10:08] <morteza> like gstreamer
[10:08] <DeezGz> yea a package its more then that too
[10:08] <morteza> maybe.
[10:08] <DeezGz> you have to have the right codecs from speacial repos to work or compile with the right switchs
[10:09] <DeezGz> very very time consuming, you have to right a compile script to get it right
[10:09] <DeezGz> do u have a windows machine anywhere
[10:10] <morteza> I have but it's not a server
[10:10] <morteza> dual boot with my laptop
[10:10] <DeezGz> i doesnt have to be
[10:10] <DeezGz> use it with vlc stream
[10:11] <DeezGz> it
[10:11] <DeezGz> it will work without the hassales
[10:11] <morteza> aha. I know.
[10:12] <morteza> I can do it from my desktop ubuntu as I said.
[10:17] <DeezGz> ok go to rtsp://108.72.28.88:21/imamhossein.ffm
[10:17] <DeezGz> im streaming your stream from windows
[10:17] <morteza> yeah it's working
[10:18] <morteza> :(
[10:18] <DeezGz> so either use you desk top or win for now
[10:18] <DeezGz> but the server has to compile the right codec for it to work
[10:18] <DeezGz> if u can do vlc, it can do it all
[10:20] <morteza> Thank you for your attention
[10:20] <morteza> it really helped.
[10:21] <DeezGz> thank you
[11:09] <X0n> hi, I have a flv file 1024x720 and I want to resize it to 512x, is it possible to do it without loose to much quality ? thanks
[11:10] <Mavrik> well... yeah
[11:10] <X0n> oups it's 1024x576
[11:10] <X0n> and just want to divide this video /2
[11:10] <Mavrik> X0n, you'll have to reencode the video though
[11:11] <Mavrik> X0n, check the bitrate of your current video
[11:11] <Mavrik> then do something like
[11:11] <Mavrik> ffmpeg -i <file>.flv -codec:v libx264 -preset slow -b:v <bitrate> -vf scale=iw/2:ih/2 -codec:a copy output.flv
[11:12] <X0n> Mavrik, wow ;) how can I know bitrate ? (I'm noob with videos)
[11:12] <Mavrik> um
[11:12] <Mavrik> X0n, approx. (filesize in KB * 8) / (seconds of length)
[11:13] <X0n> Mavrik, ok Mavrik I try this thanks a lot
[11:13] <Mavrik> X0n, sadly there's no better way of determining the bitrate
[11:13] <Mavrik> X0n, also, since you're cutting the video size in half you can lower bitrate... try using just 60% or so of the number so you get a smaller file :)
[11:14] <X0n> Mavrik, ok I'll try this
[11:14] <X0n> Mavrik, bitrate can be 33245 kb/s ?
[11:15] <Mavrik> yeah but that's really really high :D
[11:15] <Mavrik> for 512xsomethin take something like 700k
[11:15] <Mavrik> or even less :)
[11:16] <X0n> ok
[11:16] <X0n> I try
[11:16] <Mavrik> (don't forget the k)
[11:16] <X0n> ffmpeg: unrecognized option '-codec:v'
[11:16] <X0n> hum
[11:16] <Mavrik> er
[11:16] <Mavrik> just how old is your ffmpeg? :)
[11:17] <X0n> FFmpeg version SVN-r0.5.6-4:0.5.6-3 <== debian stable
[11:17] <X0n> I'll try on ubuntu
[11:17] <X0n> same
[11:18] <X0n> FFmpeg version 0.6.4-4:0.6.4-0ubuntu0.11.04.1
[11:18] <Mavrik> pre-packaged FFmpegs are old
[11:18] <Mavrik> X0n, use -vcodec instead of "codec:v", -acodec instead of "codec:a"
[11:18] <X0n> ok
[11:19] <X0n> ffmpeg: unrecognized option '-preset'
[11:19] <X0n> unlucky
[11:19] <Mavrik> ok, update your ffmpeg
[11:19] <Mavrik> if it's so old that it doesn't recognise -preset, your libx264 output will be garbage
[11:19] <Mavrik> check the topic for guide, current version is 0.10
[11:19] <X0n> I'll grab git one
[11:19] <Mavrik> grab release/0.10 branch, masters can be unstable
[11:20] <X0n> oki
[12:43] <haled> do I need to compile ffmpeg myself to enable -acodec libmp3lame (Ubuntu)?
[12:47] <X0n> thx Mavrik it's works like a charm
[12:48] <Mavrik> haled, I think you just need the "-extra" libav versions from restricted repos
[14:00] <morteza> hi
[14:00] <morteza> It seems I don't have default presets for ffmpeg to use with vpre
[14:01] <morteza> -vpre hq or any other one does not work
[14:02] <Mavrik> new versions use -preset
[14:02] <Mavrik> to use libx264 built-in presets
[14:02] <Mavrik> update ffmpeg and use taht
[14:05] <morteza> I use -vprofile baseline
[14:06] <morteza> but ffmpeg says default configs broken
[14:06] <morteza> use vpre
[14:07] <Mavrik> yes
[14:07] <Mavrik> because your FFMpeg is REALLY old
[14:08] <morteza> I just compiled it!
[14:08] <Mavrik> well
[14:08] <Mavrik> morteza, http://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/
[14:10] <morteza> Mavrik I want to convert a video from wmv3 to h264 with ffserver
[14:10] <morteza> may I pastebin the config and ask you to take a look?
[14:11] <mdsh> Can I use the movie filter to load image2 style pngs?
[14:11] <mdsh> I thought -vf movie=text%04d.png:f=image2 would work
[14:12] <mdsh> but it doesn't: [movie @ 0xb6a59a0] Failed to avformat_open_input 'text/text%04d.png'
[14:18] <morteza> http://pastebin.com/muxL8DRJ
[15:28] <burek> morteza, use newer ffmpeg
[15:29] <burek> also, try libaacplus instead of just aac
[15:29] <burek> you can use hiQ audio at bitrates as low as 32kbps
[15:35] <madsage> libaacplus huh is that a better codec than using libmp3lame? thats problably apples and oranges?
[15:35] <madsage> i went to libmp3lame from aac
[15:37] <madsage> i'm having another issue. seems my ffmpeg stream to my wowza server finishes before the live stream from wowza and causes the wowza server to end premature. how do i deal with that.
[15:38] <madsage> i'd appreciate any feedback if anybody has resolved or ran into this
[15:39] <madsage> i'm using rmtp:
[15:47] <burek> madsage, libaacplus is the future today
[15:47] <burek> youtube is using it
[15:47] <burek> aac+ is not same as aac
[15:47] <madsage> ok cool
[15:48] <madsage> i'll re-compile with it
[15:48] <madsage> any idea on my second question?
[15:49] <madsage> the ffmpeg stream finishing before the wowza live stream output?
[15:50] <burek> libaacplus: http://tipok.org.ua/ru/node/17
[15:51] <burek> madsage
[15:51] <burek> can you please use pastebin.com, to show your command line and its output?
[15:51] <madsage> yeah its just one line
[15:51] <madsage> ffmpeg -i /usr/local/WowzaMediaServer/content/Extremists.m4v -threads 0 -c:v libx264 -preset slow -crf 25 -c:a libmp3lame -aq 5 -ar 44100 -ac 2 -f flv rtmp://127.0.0.1/live/stream1
[15:51] <burek> try ffmpeg -re -i ..
[15:52] <madsage> then wowza picks up the stream and broadcasts it rtmp on the wan interface
[15:52] <madsage> is it a buffer issue with wowza?
[15:52] <madsage> ffmpeg seems to be doing its job
[15:53] <madsage> too good almost. heh
[15:53] <madsage> i prolly need to consult with wowza on this one
[15:53] <madsage> ok
[15:54] <madsage> trying -re
[15:54] <madsage> hope its that simple]
[15:55] <madsage> i'm a noob at this stuff so pardon my ignorance
[15:55] <madsage> were learning
[15:56] <madsage> ahh already i can tell its differnt. two connects are in sync, they were not prior
[15:56] <madsage> connects/connections
[15:57] <madsage> they are within milliseconds atleast
[15:58] <madsage> it stopped again :\
[15:58] <burek> wait
[15:58] <burek> it stopped because
[15:59] <burek> it reached the end of file
[15:59] <burek> what's wrong with that
[15:59] <madsage> yes
[15:59] <madsage> well the wowza media server detects no more input and didnt finish what was streamed to it
[15:59] <burek> do you want to loop the input
[15:59] <madsage> i will yes
[15:59] <burek> -loop 1
[16:00] <madsage> ok cool. i was thinking of doing it in a bash script
[16:00] <madsage> but thats a nicer option
[16:00] <burek> ffmpeg --help has a lot of nice options :D
[16:00] <madsage> hehe sorry
[16:00] <burek> also this http://ffmpeg.org/ffmpeg.html
[16:00] <madsage> i'll read more too
[16:01] <burek> ok :)
[16:02] <madsage> is there any other options for streams that wont be intended to loop? or thats goign to be a function of wowza past this point.
[16:02] <madsage> obviously ffmpeg is doing what it is supposed to
[16:04] <burek> well, I'm not sure if I understand
[16:04] <burek> but
[16:04] <burek> ffmpeg sends stream to wowza
[16:04] <burek> and when the connection closes
[16:04] <burek> wowza ends the stream too?
[16:04] <burek> right?
[16:04] <madsage> yes
[16:05] <burek> so, what is your expected behavior?
[16:05] <madsage> it seems wowza detects end of ffmpeg stream and decides the stream is done
[16:05] <burek> and it isn't?
[16:05] <madsage> well i'm thinking its buffered and should continue to end of buffer
[16:06] <madsage> but again, i understand this is goign to be a function of wowza more than likely
[16:06] <burek> how much of a stream is lost?
[16:06] <madsage> like in some cases 70%
[16:06] <madsage> my box is that fast it appears
[16:06] <burek> -re will make ffmpeg read its input more slow, like real-time
[16:06] <madsage> that fast with the encoding
[16:07] <burek> and not to read all and burst into the stream
[16:07] <madsage> yeah i added that i got further
[16:07] <madsage> but still finished at about 50%
[16:07] <burek> so, i'm not sure why do you lose 70% of the stream
[16:07] <burek> it would be a good idea to ask wowza support team :)
[16:08] <madsage> yeah i appreciate the help, thought maybe somebody had ran into this before
[16:08] <burek> ok :) :beer:
[16:08] <madsage> and might have an idea. ffmpeg is doing a fine job
[16:08] <madsage> ok man, enjoy that beer. wish i could buy you one for your help
[16:09] <burek> :)
[16:18] <morteza> Any opensource program like wowza?
[16:19] <morteza> what do you recommend?
[16:24] <burek> ffserver? :)
[16:24] <burek> vlc
[16:25] <morteza> I tried VLC it's buggy in linux
[16:25] <morteza> it can not decode wmv3 even when it has codecs
[16:26] <morteza> anyway, if ffserver or vlc can do the job why one pay $996 for wowza?
[16:34] <FunkyELF> I just upgraded ffmpeg versions on Fedora 16. I can no longer use -acodec libfaac .... but I noticed I can use -acodec aac if I use -strict experimental
[16:34] <FunkyELF> is this using the same encoder?... for -acodec did libfaac got renamed to aac?
[16:35] <burek> morteza, did you use git version of vlc?
[16:35] <burek> FunkyELF, if you can, compile your ffmpeg
[16:35] <burek> and use libaacplus
[16:35] <burek> its WAY better
[16:38] <morteza> yes
[16:39] <morteza> burek I used latest ffmpeg and vlc from git
[16:39] <morteza> to compile
[16:39] <morteza> I still get this http://pastebin.com/muxL8DRJ
[16:39] <burek> well, I don't know then.. If vlc can't play it, then it's skinny chance to find something that's open source to work
[16:40] <FunkyELF> burek, I compiled it
[16:40] <burek> morteza, that pastebin doesn't show anything
[16:40] <burek> it's just cmd line + ffserver config
[16:41] <FunkyELF> burek, I have a little cheat sheet that used to work and then I could use -acodec libfaac. Now it doesn't work, so I'm wondering if -acodec aac is the same as -acodec libfaac used to be
[16:41] <burek> i think libfaac support has been dropped
[16:41] <burek> you can read changelog
[16:41] <burek> ctrl+f "libfaac"
[16:43] <morteza> http://pastebin.com/UgGzCUzp
[16:43] <morteza> sorry it was a wrong link;)
[16:43] <morteza> I want to setup ffserver with this feed
[16:43] <FunkyELF> burek, from "version 0.4.9-pre1:" I see "- AAC encoding with libfaac"
[16:44] <FunkyELF> burek, this is how I build ffmpeg on Fedora http://pastebin.ca/2106802
[16:46] <burek> let me see
[16:47] <burek> morteza, drop vprofile, use profile
[16:47] <burek> also read x264 --help
[16:48] <burek> (to see profiles, presets and tunes
[16:48] <burek> FunkyELF, I'm not sure
[16:48] <burek> if that's the latest git version of ffmpeg
[16:49] <burek> however
[16:49] <burek> you can try typing ./configure --help | grep libaacplus
[16:49] <burek> wait, you want libfaac support?
[16:54] <FunkyELF> burek, I just rebuilt it without the "--with faac" flag and it still lets me use -acodec aac when I add -strict experimental
[16:55] <JEEB> -acodec aac is ffaac, which is libavcodec's internal aac encoder
[16:55] <JEEB> which also happens to not really great
[16:56] <FunkyELF> so libaacplus is what I want?
[16:56] <JEEB> if you want HE-AACv2
[16:56] <FunkyELF> I'm just not sure how that maps to the rpmbuild command
[16:56] <JEEB> not all devices can take HE-AAC to begin with
[16:56] <FunkyELF> I want something so that my android phone can play converted videos
[16:56] <JEEB> ok
[16:56] <JEEB> just use ffaac then
[16:56] <FunkyELF> I used to use libfaac
[16:56] <JEEB> it'll be OK as long as the bitrate's not too low
[16:57] <JEEB> you can still use it
[16:57] <FunkyELF> don't know what happened or why it stopped working
[16:57] <JEEB> --enable-nonfree in the configure
[16:57] <JEEB> and --enable-libfaac
[16:58] <FunkyELF> I see --enable-libfaac listed in my configuration (the thing it prints out every time you run ffmpeg) but then I get Unknown encoder 'libfaac'
[16:59] <JEEB> then it wasn't built with it
[16:59] <FunkyELF> I also see --enable-nonfree
[16:59] <FunkyELF> so it was built with it, but then it still says Unknown encoder 'libfaac'
[16:59] <JEEB> it would be easier to see if you had the config.log out of that build
[16:59] Action: FunkyELF runs sudo updatedb
[16:59] <JEEB> in any case, -acodec libfaac should work
[17:00] <JEEB> or wait
[17:00] <JEEB> how the hell can you have libfaac in a repository?
[17:00] <JEEB> it's non-free
[17:00] <JEEB> you can't possibly give out binaries with it enabled
[17:00] <JEEB> you'd have to build it yourself
[17:00] <FunkyELF> JEEB, I added the rpmfusion repo
[17:01] <FunkyELF> I don't see any files named config.log
[17:01] <JEEB> I have absolutely no idea what that is, but if they're still giving out binaries with libfaac they're doing it wrong
[17:01] <JEEB> and it might be that they just got wiser
[17:01] <JEEB> and disabled libfaac from the binaries
[17:01] <JEEB> :P
[17:02] <FunkyELF> I isntalled faac from rpmfusion-nonfree, then I built ffmpeg myself using rpmbuild --rebuild ffmpeg*.src.rpm --with faac
[17:02] <FunkyELF> pretty sure I got the binaries from rpmfusion-nonfree
[17:04] <JEEB> libfaac -- possibly. Anyways, you might want to list the available codecs in your compile
[17:04] <JEEB> there was a switch for that IIRC
[17:05] <FunkyELF> ffmpeg -codecs | fpaste
[17:05] <FunkyELF> http://fpaste.org/w1Sg/
[17:07] <morteza> burek, I drop -vprofile and used -profile baseline with no success. I don't know what's wrong with it
[17:13] <relaxed> -profile:v baseline
[17:15] <burek> FunkyELF, AAC is the ancient version of audio codec (v0), AAC-LC is its successor (v1) and AAC-HE(v2) is the aac+ which also youtube uses
[17:15] <burek> if you want to have brilliant audio quality, use aac+ (but its not GPL, if you need that)
[17:16] <burek> it's way better than mp3 too
[17:16] <JEEB> AAC-HE(v2) is the aac+ which also youtube uses <- as far as I know, it doesn't
[17:16] <burek> whatever
[17:16] <burek> every video I've downloaded, was using he-aac
[17:16] <JEEB> orly
[17:16] <burek> it's preety much the majority
[17:16] <JEEB> I think they used mostly LC-AAC
[17:16] <burek> well use VLC to see the audio codec info
[17:17] <burek> 90% is aac+
[17:17] <burek> if not 100%
[17:17] <burek> morteza
[17:17] <burek> can you please use pastebin.com, to show your command line and its output?
[17:19] <relaxed> all the stuff I have downloaded from youtube is LC
[17:19] <burek> ok let's see
[17:19] <burek> let me open the frontpage of youtube
[17:20] <burek> and see a random video info
[17:22] <burek> hm.. it appears AAC-LC
[17:22] <burek> i dont understand now..
[17:23] <JEEB> it might use HE-AAC for very low bitrate things where the recepient can always take in HE-AAC
[17:23] <JEEB> but I haven't seen such yet
[17:23] <morteza> burek let me tell you what I'm doing and I will use pastebin
[17:24] <burek> i really cant figure out what happened
[17:24] <morteza> I'm trying to feed ffserver with a live stream. its mmsh in wmv3 format
[17:24] <burek> i started using aac+ because of that
[17:24] <burek> and now it appears as if i was wrong all the time
[17:25] <morteza> and I want to get rtsp output to play it on android
[17:25] <burek> and i know what i saw
[17:25] <relaxed> burek: everything youtube streams is encoded on the fly. maybe they changed their policy and that's why you don't see it anymore.
[17:25] <JEEB> anyways, if HE-AAC sounds good enough for you and you like the low bitrates -- nothing bad with that
[17:25] <burek> well who wouldn't like low bitrates? :)
[17:26] <burek> i do have a good bandwidth, but not everyone has :)
[17:26] <JEEB> those people who want the actual sound -- not something that sounds "good" yet isn't the same thing
[17:26] <JEEB> and don't start giving me that bullshit any more thank you
[17:26] <morteza> burek here is my ffserver config
[17:26] <morteza> http://pastebin.com/ddR2x99L
[17:26] <JEEB> enjoy it yourself but don't try to push it down everyone's throat
[17:26] <burek> JEEB, I don't know your experience with aac+, but aac+ doesn't sound "good" it sounds extraordinary
[17:27] <burek> I tested it with classical music which is the hardest thing to efficiently encode
[17:27] <morteza> burek and this is the command I use to feed ffserver ;
[17:27] <morteza> ffmpeg -i "mmsh://85.17.29.186/ih" -vcodec libx264 -vprofile baseline -level 1 -b:v 64k -s 240x160 -strict experimental -acodec aac -ac 1 -ab 24k -f mpegts http://127.0.0.1:8090/imamhossein.ffm
[17:27] <burek> considering its spectral range
[17:27] <JEEB> also, it's very much possible what relaxed said, but to be honest I didn't see anything PC-related get HE-AAC
[17:27] <burek> it sounded identical to the original
[17:27] <JEEB> mobile stuff might be getting HE-AAC, _but_ I do remember getting the usual artifacts there with the lowest bitrate level
[17:28] <JEEB> whatever it is, it's not what you think it is
[17:28] <burek> and if you are actually comparing aac-lc and aac-he, I think he-aac will beat it like 5-6 time
[17:28] <burek> morteza, vlc can do that too, I know
[17:28] <Mavrik> <JEEB> whatever it is, it's not what you think it is
[17:28] <Mavrik> JEEB, would you care to explain that?
[17:29] <JEEB> it is not his super special HE-AAC
[17:29] <morteza> burek I know vlc can, I did it before. but this time I can not get vlc working, decoding wmv3
[17:29] <JEEB> as far as I could think of by the artifacts
[17:29] <morteza> I don't know what's wrong with it.
[17:29] <burek> what artifacts?
[17:29] <JEEB> Mavrik, burek is this guy who keeps recommending HE-AAC(v2) to everyone and their mom
[17:30] <burek> well I wouldn't do it if it wasn't this good
[17:30] <Mavrik> JEEB, yeah, I know...
[17:30] <burek> what's wrong in suggesting people the good stuff?
[17:30] <JEEB> it is good for what it's meant for, that's all I mean
[17:30] <Mavrik> JEEB, I'm actually using the AAC+ on my low-bitrates as well... I'm just wondering what you meant with "it isn't what you think it is"
[17:30] <JEEB> Mavrik, he thought youtube is all HE-AAC
[17:31] <Mavrik> getting some insights about AAC+ is kinda hard... I did notice that AAC+ tends to sound better on really low bitrates
[17:31] <Mavrik> ah, ok
[17:31] <JEEB> also, yes -- HE-AAC is my recommendation for low bitrates.
[17:32] <burek> wait
[17:32] <JEEB> The thing is, though, that it isn't an accurate'ish depiction. Anyone who can actually bother to read what the encoder does can tell you that. It's just good at getting a relatively similar sound to you with low bitrates, that is what it is for.
[17:32] <burek> what exactly do you mean by "low bitrates"
[17:32] <JEEB> anything under 96-64kbps
[17:32] <burek> yes but
[17:32] <burek> are you talking about input or output
[17:32] <JEEB> huh
[17:32] <JEEB> you really are there in the woods
[17:33] <JEEB> now go get some coffee and come sit back
[17:33] <burek> if I get a superb audio quality at 32kbps output, what can possibly be wrong in that?
[17:33] <Mavrik> mhm
[17:33] <burek> it's like when you say for 7zip "it is good for what it's meant for, that's all I mean"
[17:33] <Mavrik> JEEB, any insights what to use for higher bitrates? It seems to me LC-AAC is prevalent there
[17:33] <burek> you get a small file (which was your goal after all)
[17:34] <burek> aka low bitrates
[17:34] <JEEB> Mavrik, some good LC-AAC encoder (qt, nero) or Aotuv's vorbis
[17:34] <morteza> where can I find all available options for ffserver?
[17:34] <Mavrik> I was afraid you're going to say that ^^ :)
[17:34] <burek> morteza, http://ffmpeg.org/ffserver.html
[17:35] <burek> i just don't understand who would want "higher bitrates" when one can get 32kbps?
[17:35] <JEEB> You get an audio image that sounds close enough for you burek -- that's all. If that is all you're after and you are ready to sacrifice both decoder compatibility as well as actual closeness to the source (instead of just mimicing similar sounds -- although that's what they all do in a way), we have nothing to discuss here
[17:35] <burek> it's like you want a bigger file when testing compressions software
[17:36] <relaxed> morteza: the setting you need to use is "-profile:v baseline", not "-profile baseline".
[17:36] <JEEB> someone who knows the AAC specs please put it in more technical terms for burek please
[17:36] <burek> no need
[17:36] <burek> I really don't see the point of it
[17:37] <Mavrik> anyway, more importantly: anyone managed to test the new ffmpeg thumbnail filter? :)
[17:37] <JEEB> or tell me that I'm incorrect and that HE-AAC(v2) is overall optimized for various scenarios where closer replication is needed instead of just "something that sounds similar"
[17:37] <burek> :)
[17:38] <burek> whatever man :) what ever you find better for your own usage :)
[17:38] <Mavrik> JEEB, honestly, isn't that what pretty much all lossy compression algorithms do? :)
[17:38] <JEEB> Of course
[17:38] <JEEB> but HE-AAC, esp. the first spec -- seems to have really been just optimized to give out a similar'ish sound with too big artifacts on low bitrates
[17:39] <JEEB> I'm less knowledge'able about the second spec, but I don't think it differs much as their area is similar
[17:40] <burek> well that's what i ask you all the time
[17:40] <burek> what do you mean by "on low bitrates"
[17:40] <Mavrik> JEEB, the second spec just adds something similar to "joint stereo", where you store the audio as a mono track with deltas for other channels
[17:41] <Mavrik> so again it's really just made for low-bitrate usage with nothing to gain when you need actual replication of original sound
[17:42] <burek> would anyone care to explain what do you mean by "low-bitrate usage"
[17:43] <burek> to me it sounds just like flac, just the output is way more compressed
[17:44] <Mavrik> burek, use cases when achieving low bitrate with decent sound quality is more important than achieving maximum sound quality
[17:44] <burek> that's the problem
[17:44] <burek> I'm not talking about "decent"
[17:44] <burek> man, the quality is extraordirany
[17:44] <burek> I compressed classical music
[17:44] <Mavrik> the quality is OK :)
[17:45] <burek> which sounded the same as original
[17:45] <burek> is that "decent" enough?
[17:45] <JEEB> it's decent enough if you can't spare more bits
[17:45] <burek> why would you want to spare more bits?
[17:45] <burek> to create a 7zip compressor which produces bigger output files? :D
[17:45] <JEEB> you're comparing lossy and lossless compression here
[17:46] <burek> yeah, i was ironic
[17:46] <burek> in case you didn't notice
[17:46] <JEEB> ask someone who has dealt with the internals of that codec and see if that person would use it for quality purposes
[17:46] <JEEB> and preferably someone with his foot on the ground
[17:46] <Mavrik> burek, because sometimes you just need more quality than HE-AAC+ can provide
[17:46] <JEEB> or want to achieve it
[17:46] <Mavrik> yeah
[17:46] <JEEB> not to mention that in most cases the audio bitrate iis already small
[17:47] <Mavrik> either you have a better sound system, or other requirements
[17:47] <JEEB> it's when the audio bitrate gets to be 1/2 of video bitrate or so
[17:47] <Mavrik> or your stream is even more complex
[17:47] <burek> ok, never mind
[17:47] <JEEB> when you want to start thinking about compressing it more because it makes no sense if the video is the main part
[17:48] <burek> i respect your opinions, although i don't agree, but ok
[17:48] <burek> diversity makes world the better place anyway :)
[17:49] <JEEB> when your video bitrates are 1000kbps+ the difference between a 192kbps audio track and 32kbps audio track is no longer that big
[17:49] <JEEB> heck, for DVD backups where the AC3 track is 320kbps+ I'd prolly leave it too
[17:49] <Mavrik> btw, how does AC3 match up to MP3/LC-AAC?
[17:50] <JEEB> crap IIRC
[17:50] <JEEB> now, when we have a specific need of getting video + audio into, say, 300kbps
[17:50] <burek> well in that case, why not use flac?
[17:50] <JEEB> that's where HE-AAC shines
[17:50] <burek> why bother with aac-lc
[17:50] <Mavrik> burek, because you don't have a decoder available
[17:50] <JEEB> because FLAC breaks the camel's back
[17:50] <JEEB> because you'll end up with 700kbps+
[17:50] <burek> or any other lossless audio encoder
[17:50] <burek> for that matter
[17:50] <JEEB> at least in most of my audio tracks that are 2ch
[17:51] <JEEB> you end up doubling and almost getting to the video bitrate
[17:51] <burek> if you need brilliant audio quality, 100% of the original, use lossless audio encoder
[17:51] <burek> no biggie
[17:51] <JEEB> of course, but the audio is not the main thing here
[17:52] <JEEB> we can spare some extra bits, but there's an amount that just takes it beyond diminishing returns usually
[17:52] <JEEB> more than 320kbps-500kbps for 2ch f.ex.
[17:52] <JEEB> unless you are doing something where the audio has to be sample-exact
[17:53] <burek> ok, but generally speaking, if you don't care about bitrates, use lossless encoder, if you do care, you most definitely want aac+
[17:53] <burek> if you want to be picky, use something in between
[17:54] <burek> but that discards it as generally usable
[17:55] <JEEB> basically, "everything is relative" is the name of the case.
[17:55] <JEEB> there is no one magic pill for everything, unless you are ready to make some kind of sacrifice for it
[17:56] <JEEB> thus I just rather select my tools depending on the case
[17:56] <burek> ok :)
[18:08] <haled> how do I undefine the movie aspect in h.264 ?
[18:12] <madsage> there are switches and scaling ratios for that
[18:12] <madsage> google for some examples
[18:14] <madsage> i have a question about audio quality -aq <integer) which is better 1 or 10 and what are the limits. is it 1-10? the docs arnt clear about this. it just says to use an integer
[18:14] <burek> madsage, what -acodec are you using
[18:14] <madsage> my hearing sux. they both sound teh same
[18:14] <madsage> wll i'm currently using libmp3lame
[18:15] <burek> then use -ar 128k or -ar 256k
[18:15] <madsage> but i'm goign to switch to libaacplus by recomendation from you.
[18:15] <haled> madsage, there is a -scale switch to define a ratio but none to remove it
[18:15] <burek> libaacplus is not gpl, just so you know
[18:15] <burek> if you are going to distribute your binaries, you'll need to pay the license
[18:15] <burek> if you are just encoding
[18:15] <burek> you are good to go
[18:16] <madsage> ahh ok. so -aq doesnt do anything with libmp3lame
[18:16] <burek> :)
[18:16] <JEEB> it's worse than that burek
[18:16] <JEEB> it's not (L)GPL compatible
[18:16] <madsage> yeah just encoding
[18:16] <JEEB> you can't distribute it
[18:16] <burek> then you're ok
[18:16] <madsage> it will reside on my server. nowhere else
[18:16] <JEEB> not even if you "pay the license"
[18:16] <burek> no problems then :)
[18:16] <JEEB> unless you get a license that lets you re-license the code
[18:16] <JEEB> which is unlikely
[18:17] <JEEB> anyways, just being pedantic as usual
[18:17] <madsage> by the way i am using libmp3lame -aq 10 -ar 44100 -ac 2
[18:17] <burek> ok, so no distribution of binaries, just the media :)
[18:17] <madsage> so i can just drop the -aq 10
[18:17] <burek> yes
[18:18] <JEEB> and if you really want to be pedantic, unless there are some clauses in the AAC licensing that you can encode/distribute content encoded in the format for free, there'd be the licensing problem overall :P
[18:18] <madsage> is libmp3lame decent? or should i try the libaacplus
[18:18] <JEEB> which depends on your country, legislation and so on
[18:18] <madsage> i am looking for the best quality at the lowest bitrate
[18:18] <JEEB> same for mp3 of course
[18:18] <relaxed> madsage: -aq does work with libmp3lame
[18:18] <madsage> relaxed, oh it does? which is better 1 or 10, and what is the limits.
[18:19] <madsage> i prolly want to start in the middle
[18:19] <relaxed> -aq 0 == lame's V 0
[18:19] <madsage> oh.. volume?
[18:19] <relaxed> no, vbr
[18:19] <madsage> oh oh. doh
[18:20] <madsage> heh sorry, i'm really that green
[18:20] <madsage> so 0 is best and 10 would be least?
[18:21] <relaxed> man lame
[18:22] <madsage> i'll look it up. i dont want to wear out my welcome with stupid questions. i tried the ffmpeg docs it says to use an integer is all
[18:23] <relaxed> I believe it uses the same settings as lame's -V, so consult the man its page
[18:23] <relaxed> /the man its/its man/
[18:24] <paideia> hi, trying to extract the frames of movie 4 in http://www.pnas.org/content/early/2012/01/10/1115323109/suppl/DCSupplemental
[18:24] <madsage> aye, thank you relaxed
[18:25] <paideia> with ffmpeg, but it just exports the first frame repeatedly
[18:25] <madsage> ok that was easy
[18:25] <madsage> There are 10 compression levels defined,
[18:25] <madsage> ranging from 0=lowest compression to 9 highest compression
[18:25] <paideia> seemingly only vlc can play it nicely
[18:25] <madsage> -V5 resulting in files averaging 132 kbps, -V2 averaging 200 kbps
[18:25] <paideia> mplayer does not do it
[18:26] <paideia> and mencoder fails when attempting to convert it too
[18:26] <paideia> any suggestions?
[18:27] <morteza> can vlc stream on rtsp for android ?
[18:29] <JEEB> yes, as far as I know it can do rtsp streaming. I remember finding an example by googling
[18:30] <JEEB> might or might not contain android, but it was rtsp for sure
[18:30] <JEEB> (I think the example was for the N900)
[19:26] <mgeary> so , i recently compiled from svn, and now i'm getting "encoder 'aac' is experimental and might produce bad results" messages. How do i add that '-strict experimental' flag to override this?
[19:27] <mgeary> or does that need to be added at compile time?
[19:27] <mgeary> aha
[19:28] <mgeary> hrm. okay, well that's not working. What's a good fallback acodec?
[19:30] <relaxed> compiled from git?
[19:30] <mgeary> believe so
[19:30] <mgeary> that is, it was either from git or svn
[19:30] <mgeary> but yes, self-compiled
[19:31] <relaxed> pastebin the command and all output
[19:32] <mgeary> http://pastebin.com/YHUXN3JU
[19:33] <relaxed> ... -f mp4 out.m4v
[19:33] <mgeary> same response
[19:34] <relaxed> then use -b:v $videobitrate and -b:a $audiobitrate
[19:36] <relaxed> If you still get the "[ipod @ 0x10180fa00] malformated aac bitstream, use -absf aac_adtstoasc" error then update your source and rebuild. I believe that bug has been fixed.
[19:36] <mgeary> ah, ok. i actually just realized i have to go out to flv/mp3, so i'm dodging the issue at the moment
[19:36] <mgeary> but i'll try that in a bit
[19:37] <relaxed> omit -vpre medium because it is the default and libx264 profiles are now called with "-preset $profile"
[19:38] <relaxed> libx264 presets*
[19:38] <relaxed> I really screwed that up
[19:38] <relaxed> omit -vpre medium because it is the default and libx264 presets are now called with "-preset $preset"
[19:42] <mgeary> hahah
[19:42] <mgeary> thanks
[19:52] <relaxed> mgeary: next time you compile ffmpeg add libvo-aacenc support
[20:02] <DeezGz> hello ffmpegers
[20:02] <madsage> man i'm fustrated. i open a stream on wowza in stream manager. player connects.. i start ffmpeg its all good until its done. ahead of the the live stream from teh server and it somehow closes the stream. does it send some signal to close the stream? thsi isnt making sense
[20:03] <madsage> or am i just retarded
[20:07] <mgeary> that's a dangerous question to ask
[20:11] <madsage> heh
[20:12] <madsage> ffmpeg is workign great.
[20:12] <madsage> my streaming shit is not cooperating
[20:12] <relaxed> MrNaz: you around?
[21:12] <LiroXIV> Okay, I'm making a little VB.net app that takes in a input file, an output location, and will call ffmpeg (as a shell command) to encode the video
[21:12] <LiroXIV> I need a single FFmpeg command that will encode a video file into WebM with audio
[21:14] <Mavrik> LiroXIV, http://www.virag.si/2012/01/webm-web-video-encoding-tutorial-with-ffmpeg-0-…
[22:14] <LiroXIV> Okay, I tried this 64-bit Windows binary of ffmpeg with a certain string of commands. When it tries to start actually encoding, it crashes
[22:27] <r0b-> hello anyone here?
[22:31] <r0b-> trying to stream a webcam off a server using ffmpeg
[22:40] <r0b-> :(
[22:47] Action: r0b- cant make ffmpeg stream
[22:48] <ehsan_> Hi I need to put video width into a variable so I can crop the video by half in a bash script.what shall I do?
[22:53] <ehsan_> Hi isnt there crop in ffplay
[22:53] <ehsan_> ?
[23:36] <marteo> hello !
[23:37] <marteo> I read the man page and didn't find what i was looking for, so i assume it simply does not exist, but i'll ask anyway :
[23:38] <marteo> is there some kind of "slow, repeated, double-checked" mode while uncompressing videos (essentially poorly encoded webms with h264 video)
[23:40] <marteo> because some frames do not appear correctly (green squares, that kind of thing)
[23:40] <marteo> (i point out those videos were terribly encoded by other people in my local newscast association)
[23:49] <burek> so, what is your overall goal marteo ?
[23:50] <burek> to decompress the video as good as possible?
[23:50] <marteo> uncompressing these videos to DV, as accurately as possible, to work again on them (with Premiere, etc.)
[23:51] <burek> you can try ffmpeg -re -i <input> ...
[23:51] <burek> that will make ffmpeg read input in real-time mode
[23:51] <burek> i.e. not as fast as possible
[23:51] <burek> but I'm not sure if that will help a lot
[23:52] <marteo> yeah, because a simple ffmpeg -i input -target pal-dv output already works at 2 frames / sec.
[23:52] <marteo> let me try anyway
[23:53] <burek> you might wanna try then to output to png images (uncompressing the input effectively)
[23:53] <burek> ffmpeg -i ... -f image2 output%04d.png
[23:55] <marteo> what the hell
[23:55] <marteo> allright, i guess i get it
[23:56] <marteo> does the command output each frame exactly, independantly ?
[23:57] <marteo> because I get something like 200 files per second of input
[23:58] <marteo> which would explains some of the trouble i went through
[23:58] <burek> well, 25fps
[23:58] <burek> should be
[23:58] <burek> or 50fps
[23:58] <burek> but 200 fps...
[23:58] <burek> that's just too much
[23:58] <burek> try with -re
[00:00] --- Sat Jan 28 2012
1
0
[00:12] <CIA-108> ffmpeg: 03Paul B Mahol 07master * rdd453f197c 10ffmpeg/ (6 files in 3 dirs):
[00:12] <CIA-108> ffmpeg: r210, r10k and avrp encoder
[00:12] <CIA-108> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com>
[00:12] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[00:12] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r807a045ab7 10ffmpeg/libavcodec/kgv1dec.c:
[00:12] <CIA-108> ffmpeg: kgv1dec: Increase offsets array size so it is large enough.
[00:12] <CIA-108> ffmpeg: Fixes CVE-2011-3945
[00:12] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[00:12] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[00:12] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rdef678956a 10ffmpeg/ (Makefile avconv.c configure doc/avconv.texi):
[00:12] <CIA-108> ffmpeg: Remove avconv
[00:12] <CIA-108> ffmpeg: All features have been merged into ffmpeg.
[00:12] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[00:12] <CIA-108> ffmpeg: 03Lou Logan 07master * r935c659c03 10ffmpeg/Doxyfile:
[00:12] <CIA-108> ffmpeg: remove avconv from Doxyfile
[00:13] <CIA-108> ffmpeg: Fixes CVE-2011-3947
[00:13] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[00:13] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[01:15] <bcoudurier> I just subscribed to spotify unlimited, it works great on my ubuntu out of the box
[01:15] <bcoudurier> they did a good job
[01:56] <Compn> giving riaa money ?
[01:56] <Compn> :P
[02:13] <bcoudurier> Compn, yes, way less that if I bought cds or from itunes though
[02:18] <Compn> every penny you give to riaa goes to killing the internet
[02:19] <Compn> same with mpaa :)
[02:23] <iive> well, at least for now they are killing "only" the internet...
[02:25] <bcoudurier> I'm not gonna stop listening to music, you are proposing that I steal it ? :)
[02:32] Action: Daemon404 feras much of the music he lieks isnt on spotify
[02:32] <Daemon404> and does that site even -have- a north american version yet?
[02:33] <Daemon404> looks like it does... only wiyh a cruddy selection
[02:43] <hyc> I create music. and buy music directly from my fellow musicians...
[02:45] <Compn> start buying indie music
[02:45] <Compn> :P
[02:45] <hyc> exactly
[02:45] <Daemon404> while youre at it
[02:45] <Daemon404> grow moustache
[02:45] <Daemon404> and wear ironci tshirts
[02:45] <Daemon404> a*
[03:20] <CIA-108> ffmpeg: 03Ronald S. Bultje 07master * rc3af52fa8b 10ffmpeg/libavcodec/x86/dsputil_mmx.c:
[03:20] <CIA-108> ffmpeg: dsputil: use vertical component for drawing bottom edge.
[03:20] <CIA-108> ffmpeg: Current code only writes 8 pixels of vertical edge for YUV422, which
[03:20] <CIA-108> ffmpeg: causes MC artifacts when subsequent frames use data from that edge.
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * r83988d58ed 10ffmpeg/ (4 files in 2 dirs):
[03:20] <CIA-108> ffmpeg: movenc: Add support for writing fragmented mov files
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * rf1caf01d5e 10ffmpeg/libavformat/ (avformat.h utils.c version.h):
[03:20] <CIA-108> ffmpeg: libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * r6ca3856894 10ffmpeg/libavformat/ (movenc.c movenc.h):
[03:20] <CIA-108> ffmpeg: movenc: Add a separate start_pts
[03:20] <CIA-108> ffmpeg: This fixes calculation of trackDuration if the MOVIentry array
[03:20] <CIA-108> ffmpeg: is cleared. This is required by the fragmentation support in the
[03:20] <CIA-108> ffmpeg: next patch.
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * rb613ff5e93 10ffmpeg/libavformat/ (movenc.c movenc.h):
[03:20] <CIA-108> ffmpeg: movenc: Allow the caller to decide on fragmentation
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * r4ddd54dab4 10ffmpeg/ (5 files in 2 dirs):
[03:20] <CIA-108> ffmpeg: movenc: Add a separate ismv/isma (smooth streaming) muxer
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * r3b5d4428ac 10ffmpeg/libavformat/movenc.c:
[03:20] <CIA-108> ffmpeg: movenc: Write the sample rate instead of time scale in the stsd atom
[03:20] <CIA-108> ffmpeg: For ismv/isma, the time scale might not be the same as the sample
[03:20] <CIA-108> ffmpeg: rate.
[03:20] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[03:20] <CIA-108> ffmpeg: 03Martin Storsjö 07master * rbc7d05177f 10ffmpeg/libavformat/ (isom.h mov.c):
[03:20] <CIA-108> ffmpeg: movdec: Calculate an average bit rate for fragmented streams, too
[08:48] <CIA-108> ffmpeg: 03Thierry Foucu 07master * r10e9d1f76b 10ffmpeg/libavcodec/mpegaudiodec.c:
[08:48] <CIA-108> ffmpeg: Fix a heap-buffer-overflow
[08:48] <CIA-108> ffmpeg: In some case, what left to read from ptr is smaller than EXTRABYTES.
[08:48] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[08:48] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r46095f427e 10ffmpeg/libavcodec/mpegaudiodec.c:
[08:48] <CIA-108> ffmpeg: mp3dec: Check for memcpy size to be positive.
[08:48] <CIA-108> ffmpeg: No, ive no testcase.
[08:48] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[10:56] <cbsrobot> nelsontuxd is pulluting the wiki
[11:00] <cbsrobot> michaelni: ^
[13:07] <CIA-108> ffmpeg: 03Clément BSsch 07master * ree0cab7721 10ffmpeg/doc/ffprobe.texi:
[13:07] <CIA-108> ffmpeg: doc: remove trailing 's' to metadata.
[13:07] <CIA-108> ffmpeg: metadata is already plural.
[13:07] <CIA-108> ffmpeg: Found-by: Alexander Strasser
[13:26] <Tjoppen> bah, git-send-email
[13:29] <ubitux> ?
[13:29] <ubitux> i got pwned by git-send-email recently
[13:30] <Tjoppen> cherry-picked and sent some mxfdec patches to libav. everything seems to have gone mostly though, except for michaelni possibly getting some CC spam
[13:30] <ubitux> ah got that recently too
[13:30] <ubitux> when i tried to use --cc
[13:30] <ubitux> :D
[13:31] <ubitux> but more recently, i took 15 minutes to write a git send-email --annotate, failed to write my password properly, and thus lost my message
[13:31] <ubitux> it seems there is a backup in .git/.gitsendemailXXX, but it seemed overrided when i git send-email -1 again :(
[13:32] <Tjoppen> --compose seems useful
[13:32] <ubitux> it's useful for a patchset
[13:32] <ubitux> not for a single patch
[14:00] <michaelni> cbsrobot, nelsontuxd deleted
[15:47] <ubitux> michaelni: you think mp3 duration could be estimated faster?
[15:48] <ubitux> oh, maybe it's not the duration which make the probing looping so much?
[15:49] <ubitux> duration estimation*
[17:15] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * re2291ea153 10ffmpeg/libavcodec/diracdec.c:
[17:15] <CIA-108> ffmpeg: diracdec: Check dirac_unpack_idwt_params parameters before storing them.
[17:15] <CIA-108> ffmpeg: Fixes CVE-2011-3949
[17:15] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[17:15] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[17:15] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rddf0c1d86a 10ffmpeg/libavcodec/diracdec.c:
[17:15] <CIA-108> ffmpeg: diracdec: Check num_refs.
[17:15] <CIA-108> ffmpeg: Fixes: CVE-2011-3950
[17:16] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[17:16] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[17:16] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r92115bb685 10ffmpeg/libavcodec/dpcm.c:
[17:16] <CIA-108> ffmpeg: dpcm: Round output buffer size up.
[17:16] <CIA-108> ffmpeg: Fixes: CVE-2011-3951
[17:16] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[17:16] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[17:46] <CIA-108> ffmpeg: 03Alex Converse 07master * r5af569aa30 10ffmpeg/libavformat/matroskadec.c:
[17:46] <CIA-108> ffmpeg: matroskadec: Pad AAC extradata.
[17:46] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[17:46] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[17:46] <CIA-108> ffmpeg: (cherry picked from commit d2ee8c17793201ce969afd1f433ba1580c143cd2)
[17:46] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[17:46] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r1860c66c54 10ffmpeg/libavformat/matroskadec.c:
[17:46] <CIA-108> ffmpeg: matroskadec: increase padding on several more extradata allocations.
[17:46] <CIA-108> ffmpeg: Inspired by: 5af569aa30b93f56344ea540936eb671760f568c by alex
[17:46] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[17:46] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r70dba1e3c8 10ffmpeg/libavcodec/kmvc.c:
[17:46] <CIA-108> ffmpeg: kvmc: Check palsize.
[17:46] <CIA-108> ffmpeg: Fixes: CVE-2011-3952
[17:46] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[17:46] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[18:28] <funman> hello
[18:28] <funman> kierank: I'm working on making libavcodec extract (and insert) CC to/from mpeg2/h264
[20:09] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r7ed6b1a128 10ffmpeg/libavcodec/mpeg12enc.c:
[20:09] <CIA-108> ffmpeg: mpeg1videoenc: disable slice threads
[20:09] <CIA-108> ffmpeg: It doesnt work (and as far as i tested also didnt in the past)
[20:09] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:09] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * rfe0089a6ed 10ffmpeg/ffmpeg.c:
[20:09] <CIA-108> ffmpeg: ffmpeg: fix -qscale X breaking audio codecs
[20:09] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:09] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r22eef8d738 10ffmpeg/libavcodec/snowenc.c:
[20:09] <CIA-108> ffmpeg: snowenc: dont crash with gray but exit with an error msg.
[20:09] <CIA-108> ffmpeg: Fixes Ticket839
[20:09] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:09] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r9decfc17bb 10ffmpeg/libavcodec/h264_sei.c:
[20:09] <CIA-108> ffmpeg: h264_sei: Fix infinite loop.
[20:09] <CIA-108> ffmpeg: Fixes not yet fixed parts of CVE-2011-3946.
[20:09] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[20:09] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:54] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r8847561f93 10ffmpeg/libavformat/mov.c:
[20:54] <CIA-108> ffmpeg: mov: Fix seeking regression in fragemnted movs.
[20:54] <CIA-108> ffmpeg: Regression introduced in 550f7c43ece1af27604407647d10e74b52e6fedf
[20:54] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[20:54] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r88d84dd8ea 10ffmpeg/libavformat/dv.c:
[20:54] <CIA-108> ffmpeg: dv: Fix out of array read
[20:54] <CIA-108> ffmpeg: Fixes part of CVE-2011-3936
[20:54] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[20:54] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:36] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r32281d309a 10ffmpeg/Changelog:
[21:36] <CIA-108> ffmpeg: Changelog: update for 0.10
[21:36] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[21:39] <durandal_1707> michaelni: your changelog have duplicated entries
[22:07] <michaelni> durandal_1707, which ?
[22:08] <durandal_1707> avrp lines, there is AVID codec few lines above
[22:08] <durandal_1707> OMA muxer is same as Sony OpenMG few lines above
[22:09] <durandal_1707> APC adpcm decoder support was long ago there but just under different name, but feel free to leave it as is
[22:18] <michaelni> durandal_1707, thx, ill fix it
[22:21] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r5ce1b214eb 10ffmpeg/doc/RELEASE_NOTES:
[22:21] <CIA-108> ffmpeg: RELEASE_NOTES: update for 0.10
[22:21] <CIA-108> ffmpeg: remove minor things and things that we had in many previous releases
[22:21] <CIA-108> ffmpeg: already.
[22:21] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:21] <CIA-108> ffmpeg: 03Michael Niedermayer 07master * r33c21378a8 10ffmpeg/Changelog:
[22:21] <CIA-108> ffmpeg: Changelog: remove duplicate lines
[22:21] <CIA-108> ffmpeg: Found-by: durandal_1707
[22:21] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:40] <ubitux> when the demuxer need to export information with extradata, is it supposed to allocate itself a buffer and craft data in it? or is it supposed to export "raw" pointers to the data?
[22:41] <ubitux> i have a format where the "header" is split around the file
[22:42] <durandal_1707> afaik extradata is used to send same stuff to decoders
[22:43] <durandal_1707> ubitux: what format?
[22:44] <ubitux> jacosub
[22:44] <ubitux> see for instance http://unicorn.us.com/jacosub/jscripts.html#c_palette
[22:44] <ubitux> "#PALETTE may be specified anywhere in the file"
[22:44] <durandal_1707> that just sucks
[22:45] <ubitux> yes kind of :)
[22:45] <ubitux> i need to parse the whole file :p
[22:45] <ubitux> i mean, i need two pass
[22:45] <ubitux> :p
[22:45] <CIA-108> ffmpeg: 03Alex Converse 07master * ra8ae00b68c 10ffmpeg/libavcodec/qdm2.c:
[22:45] <CIA-108> ffmpeg: qdm2: Check data block size for bytes to bits overflow.
[22:45] <CIA-108> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
[22:45] <CIA-108> ffmpeg: CC: libav-stable(a)libav.org
[22:45] <CIA-108> ffmpeg: (cherry picked from commit dac56d9ce01eb9963f28f26b97a81db5cbd46c1c)
[22:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:45] <CIA-108> ffmpeg: 03Martin Storsjö 07master * rb9e79a3f4e 10ffmpeg/tools/ismindex.c:
[22:45] <CIA-108> ffmpeg: ismindex: Fix build on mingw
[22:45] <CIA-108> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st>
[22:45] <CIA-108> ffmpeg: (cherry picked from commit 8801fac365549a43a639e239faba409d8f91ef86)
[22:45] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:46] <durandal_1707> parser for subtitle?
[22:47] <ubitux> yes
[22:47] <ubitux> i started to work on this last about 1yr ago
[22:47] <ubitux> and never get done
[22:48] <ubitux> but now that i want to make fancy stuff with subtitles, i'm trying to understand a bit the whole subtitles chain
[22:48] <ubitux> so i'm working again on this crazy format
[22:48] <ubitux> but i start to understand why i never got it done :D
[22:48] <durandal_1707> i would put some default colors and would ignore palette
[22:49] <CIA-108> ffmpeg: 03Michael Niedermayer 07release/0.10 * r7e16636995 10ffmpeg/doc/ffmpeg-mt-authorship.txt:
[22:49] <CIA-108> ffmpeg: doc: remove doc/ffmpeg-mt-authorship.txt for release/0.10
[22:49] <CIA-108> ffmpeg: we dont carry the whole git history in releases so theres no
[22:49] <CIA-108> ffmpeg: point in having this in them either.
[22:49] <CIA-108> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[22:49] <ubitux> durandal_1707: it's not the only global command
[22:51] <durandal_1707> ubitux: you have samples?
[22:52] <ubitux> i grabbed these around the web: http://blackhole.ubitux.fr/jss/
[22:53] <ubitux> a simple support can be done with not much trouble
[22:53] <ubitux> i think i'll ignore the header to start with
[22:53] <durandal_1707> and they all have palette in random place?
[22:55] <ubitux> not, afaik
[22:56] <durandal_1707> just write what you can and leave other to somebody else :/
[22:58] <ubitux> :)
[00:00] --- Fri Jan 27 2012
1
0
[00:46] <indigo> hi
[00:47] <indigo> I'm trying to get a help to compile lastest ffmpeg in mingw32
[00:48] <indigo> it keeps giving an error saying I need mingw32-runtime >=3.15 but I cannot find any mingw32-runtime >=3.15 release!
[00:52] <newl> http://sourceforge.net/projects/mingw/files/MinGW/Base/mingw-rt/
[00:53] <indigo> thanks
[00:59] <boo> Is there anyone here? I think that I have found the problem of my ffmpeg crashing, but I don't know the solution
[01:01] <newl> strace it - maybe it is something easy to see and use verbose in your command line ... which is what btw?
[01:17] <boo> new1 are you talking to me?
[01:19] <boo> new1 Facts - if I save the dvb-t stream with 'cat' to a file and paralel decode it with ffmpeg then nothing crashes. BUT if I encode dirrectly from dvb-t stick, then it crashes ~5.2 seconds +/- 2 ms so one guy suggested adding -analyzeduration 1500000000
[01:20] <boo> new1 wuwhu, now it didn't crash for 10,3 seconds..
[01:20] <boo> new1 need to find a way to ignore that 'analyzeduration' ... because you cannot add infinite value
[02:03] <indigo> well
[02:04] <indigo> I installed mingwrt-3.15.2 and ffmpeg ./configure keep telling me I need >=3.15
[02:05] <indigo> I recompile the entire mingw32 package via the sh script
[02:06] <newl> what is the configure line you are using -
[02:07] <indigo> ./configure --enable-cross-compile --arch=x86 --target-os=mingw32 --disable-shared --enable-static --enable-version3
[02:07] <boo> new1 no Ideas about mine problem?
[02:09] <newl> boo nope
[02:09] <indigo> it should be fine,
[02:09] <boo> new1 don't know anything about analyzeduration ?
[02:10] <indigo> check_cpp_condition _mingw.h "defined (__MINGW64_VERSION_MAJOR) || (__MINGW32_MAJOR_VERSION > 3) \
[02:10] <indigo> || (__MINGW32_MAJOR_VERSION == 3 && __MINGW32_MINOR_VERSION >= 15)" ||
[02:10] <indigo> die "ERROR: MinGW runtime version must be >= 3.15."
[02:10] <indigo> it search in _mingw.h
[02:11] <indigo> but WHICH _mingw.h?
[02:12] <newl> add a --cross-prefix=i686-mingw32-
[02:12] <newl> ymmv
[02:13] <newl> boo nope
[02:15] <indigo> I found my mistake
[02:15] <indigo> I never removed the old version
[02:15] <newl> always helps
[02:15] <indigo> and instead installing the newer in the same di9rectoty
[02:15] <newl> godo
[02:15] <newl> doog
[02:15] <newl> good
[02:16] <indigo> I installed in /usr/minw232 instead /usr/mingw32
[02:16] <indigo> :D
[02:17] <newl> is it in the $PATH?
[02:17] <indigo> yes
[02:18] <indigo> PATH=$PATH:/usr/mingw32
[02:19] <newl> a good ffmpeg.exe is only good if it works with wine - forget about testing on a win machine :)
[02:19] <Zeranoe> If I don't modify the source code for FFmpeg at all, and build for LGPL, does the source code need to still be "linked" or mentioned? Does it need to be packaged with the build?
[02:19] <newl> you selling your product for beaucoup yen?
[02:21] <indigo> ok, I fixed the mingw installation path
[02:21] <indigo> now let's do it again
[02:22] <indigo> doesn't work
[02:22] <indigo> to be shure i
[02:22] <indigo> updatedb
[02:22] <indigo> then locate _mingw.h
[02:23] <indigo> then cat /path/to/_mingw.h
[02:23] <indigo> and I can see it's 3.15.2
[02:23] <indigo> but nope
[02:23] <indigo> it's in /usr/mingw32/include
[02:26] <Zeranoe> indigo: Are you cross compiling?
[02:26] <indigo> yes
[02:26] <indigo> I'm on a debian
[02:26] <Zeranoe> what toolchain are you using?
[02:26] <newl> use the extra --extra-cflags='-I/mingw/include' --extra-ldflags='-L/mingw/lib'
[02:27] <indigo> ok, I'll try
[02:28] <indigo> ./configure --enable-cross-compile --arch=x86 --target-os=mingw32 --disable-shared --enable-static --enable-version3 --extra-cflags='-I/root/mingw/include' --extra-ldflags='-L/root/mingw/lib'
[02:28] <indigo> ERROR: MinGW runtime version must be >= 3.15.
[02:28] <Zeranoe> indigo: Can you post your config.log on pastebin.com?
[02:29] <indigo> give me a minute
[02:30] <indigo> maybe I solved
[02:30] <indigo> oh yeeeeeah
[02:31] <indigo> it was NOT /root/mingw/include BUT /root/mingw32/include
[02:33] <indigo> I have different errors this time, but I can handle them
[02:33] <indigo> thanks a lot
[02:33] <indigo> goodbye
[02:43] <_klk_> ladies and gentlemen, can ffmpeg transcode from h264 in mkv container to m2ts for burning onto a bluray?
[04:30] <shevy> Can I disable audio somehow?
[04:30] <shevy> I have a .mp4 file and want to only keep the video data
[04:30] <shevy> and discard audio
[04:33] <shevy> found it
[04:33] <shevy> -an
[04:55] <burek> shevy -an
[04:55] <burek> yes :)
[06:10] <Ginks> is there a way that I can place a scrolling marquee in a movie?
[06:10] <Ginks> like: "This movie is copyright Ginks, 2012"
[07:03] <sosman> I have some motion JPEG AVI (straight from digicam). When I use ffmpeg -i foo.avi foo.mpg they are 4x smaller but noticeable pixelation. Any suggestions on command line for improving the quality (I don't mind sacrificing compression).
[07:48] <grepper> default bitrate is 200 kbps, so you should indicate the bitrate.
[08:15] <cbreak-work> if you want h.264, consider a crf encode
[08:17] <grepper> yeah, your line is making mpeg1 so your compression could probably not get worse :)
[08:19] <Ginks> is there a way that I can place a scrolling marquee in a movie?
[08:23] <cbreak-work> text?
[08:24] <cbreak-work> ffmpeg has the drawtext filter if you configured it with --enable-libfreetype
[10:08] <Ginks> yah, text
[10:09] <Ginks> I found that filter, seems pretty complicated for what I need to do
[10:09] <Ginks> I'll see if I can figure it out
[10:09] <cbreak-work> drawtext is power :)
[10:19] <Ginks> basically I just want to scroll "This movie is copyright Ginks, 2012" a couple times thoughout the video across the top
[10:19] <Ginks> is that doable cbreak-work?
[10:21] <cbreak-work> yeah. Think so. It has temporaly dependent x-y
[10:21] <cbreak-work> don't ask me how :)
[10:37] <sosman> cbreak-work: thanks, crf 22 did the trick.
[10:39] <sosman> if I want to replace the audio in an mp4 file with the tracks in a separate .aac file how can I specify copy vs transcode please?
[10:39] <sosman> I tried ffmpeg -i a.aac -i v.mp4 new.mp4
[10:39] <JEEB> -acodec copy
[10:40] <sosman> But it was compressed quite a bit.
[10:40] <Mavrik> -codec:a copy
[10:40] <Mavrik> or -acodec copy
[10:40] <Mavrik> depending on which version you have :)
[10:40] <Mavrik> Morning everyone by the way :)
[10:40] <sosman> evening
[10:40] <sosman> version 0.7.8
[10:41] <Mavrik> just do -acodec copy to copy audio stream
[10:41] <Mavrik> if there's an audio stream in the source mp4 file you'll probably have to use -map as well
[10:41] <sosman> yes there is
[10:44] <Mavrik> then if you want to replace the audio you'll probably have to specify -map mappings for streams you want to keep
[10:53] <mrAZ> hi, input.mov is dvvideo, yuv420p, bitrate: 31041 kb/s, i am trying to lower bitrate with -vcodec dvvideo -b 1000k, but output is again bitrate: 27972 kb/s. what is the problem?
[10:53] <sosman> So far I have:
[10:54] <sosman> ffmpeg -i foo2.mp4 -i bg.aac -vcodec copy -acodec copy -map 0:v -map 1:a bg3.mp4
[10:54] <Mavrik> mrAZ, you can't change bitrate of DV, it has a static compression
[10:54] <Mavrik> sosman, that should work :)
[10:54] <mrAZ> Mavrik: thx, i will try to use something i can put in .mov
[10:54] <sosman> but I get stuff like [aac @ 0x1b411a0] channel element 0.0 is not allocated
[10:54] <sosman> [h264 @ 0x1b3ed80] missing picture in access unit
[10:55] <Mavrik> mrAZ, that would be H.264 ^^
[10:55] <Mavrik> sosman, that's... interesting
[10:55] <sosman> indeed
[10:56] <Mavrik> especially since it shouldn't even be trying to decode the aac
[10:56] <Mavrik> -map 0:v:0 -map 1:a:0?
[10:57] <sosman> same
[10:57] <Mavrik> can you paste the whole ffmpeg output to some pastebin so we can see if anything else goes wrong? :)
[10:57] <sosman> ok
[10:58] <sosman> Any command I can run on the input files to check my assumptions?
[10:58] <Mavrik> ffprobe
[10:58] <Mavrik> to see what's in them
[10:58] <Mavrik> and ffplay to see if ffmpeg can actually decode and play them
[10:59] <sosman> I straight away get errors with ffprobe on the mp4 file so I'll hang fire on the paste till I sort that out.
[11:00] <sosman> vlc plays it fine though
[11:00] <Mavrik> sosman, try doing just ffmpeg -i foo2.mp4 -vcodec copy -an test.mp4
[11:00] <Mavrik> and see if the test.mp4 file is playable
[11:00] <mrAZ> Mavrik: how does it go in ffmpeg? Unknown encoder 'h264'. normaly i use mplayer and mp4box for that.
[11:00] <Mavrik> mrAZ, something like -vcodec libx264 -preset slow -b 1000k
[11:01] <sosman> works fine with ffplay
[11:01] <Mavrik> sosman, should be fine then, there seems to be an ffmpeg bug report because of that message
[11:05] <mrAZ> Unrecognized option 'preset', and if i skip that : use an encoding preset (vpre) ... parameters such as bit_rate ...
[11:05] <Mavrik> ugh
[11:05] <Mavrik> your ffmpeg is ancient :)
[11:05] <Mavrik> I suggest udating it before you continue.
[11:07] <mrAZ> FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2 ... is that already ancient?
[11:07] <Mavrik> libx264 didn't really work very well with old ffmpeg versions
[11:07] <Mavrik> mrAZ, current version is 0.9.1 :)
[11:16] <sacarasc> ackjewt: It's been ancient for a long time. ;p
[11:47] <IamTrying> I have wmv file which is converted to flv works but i can not do it to "WEBM" i get this error: https://gist.github.com/1682200
[11:52] <Mavrik> IamTrying, you're trying to encode audio to 11050Hz with 1 channel which libvorbis encoder doesn't support
[11:52] <Mavrik> IamTrying, also, you probably shouldn't be using -sameq, it doesn't do that you (probably) think it does
[11:54] <IamTrying> Mavrik, ok thank you when i used this its doing now something. But will this drop the quality? e.g: $ ffmpeg -sameq -i file.wmv -ar 22050 output.webm
[11:55] <Mavrik> for audio, probably not, since your input is 11025
[11:55] <Mavrik> for video, you're probably losing quality because you're not specifying any settings and ffmpeg is defaulting to built-in
[11:58] <IamTrying> Thank you, much clear now what was the issue.
[11:59] <Mavrik> IamTrying, http://www.virag.si/2012/01/webm-web-video-encoding-tutorial-with-ffmpeg-0-…
[12:00] <Mavrik> IamTrying, this may help you
[12:00] <IamTrying> OK
[12:17] <armetiz> hi there
[12:18] <armetiz> I'm using ffmpeg ince 2 days..So i'm a newbe on it.
[12:18] <armetiz> I'm trying to encode master video file to different bitrate, it's okay for size, etc etc..
[12:18] <armetiz> But I have a problem ( the last one ), about color...
[12:19] <armetiz> The encoded ffmpeg video is color less regarding the original file..
[12:19] <armetiz> Do you have any clue ?
[12:20] <armetiz> I have build ffmpeg & x264 from master SVN / GIT.
[12:20] <armetiz> I have compile x264 with default option
[13:04] <ackjewt> sacarasc: Huh?
[13:05] <sacarasc> Sorry, I failed at reading, and thus tab completing.
[13:09] <ackjewt> Ah. Im getting used to it
[13:34] <armetiz> I'm trying to use some video filter, like pad
[13:34] <armetiz> and I have no result..
[13:34] <armetiz> or "Input area XXXXX not within the padded area XXXXX or zero-sized"
[13:35] <armetiz> do you have any clue ?
[13:35] <Mavrik_> obviously you pass wrong parameter to the filter.
[13:35] <armetiz> I have try the ffmpeg example
[13:35] <armetiz> and the video output is the same as without pad..
[13:36] <armetiz> I want to add pad on left / right from a 4:3 to a 16:9
[13:36] <Mavrik_> well
[13:36] <Mavrik_> ffmpeg tells you clearly
[13:36] <Mavrik_> your input area for padding isn't valid
[13:36] <Mavrik_> you passed wrong parameters to the padding filter for your video.
[13:36] <Mavrik_> go read the documentation again and check what each parameter does.
[13:37] <armetiz> oki
[13:37] <armetiz> i'll read the documentation again
[14:05] <undercash> hello
[14:05] <undercash> i have updated my ffmpeg
[14:05] <undercash> http://pastebin.com/QSLigsum
[14:05] <undercash> but now it seems my command script won't work anymore..
[14:06] <undercash> ffmpeg -i "testcc.avi" -y -acodec libfaac -ar 44100 -ab 96k -ac 2 -s 640x360 -vcodec libx264 -preset medium -profile baseline -level 3.0 -crf 30 -threads 8 -f ipod -f ipod mobiletest.mp4
[14:06] <undercash> something changed since july?
[14:07] <Mavrik> maybe.
[14:07] <Mavrik> maybe you should post the actual error.
[14:07] <undercash> here is output http://pastebin.com/bPv73nFz
[14:08] <undercash> yes ;)
[14:08] <Mavrik> [NULL @ 0x181f680] Unable to parse option value "baseline"
[14:08] <Mavrik> ah, yes, known problem
[14:08] <Mavrik> undercash, parameter collision between libfaac and libx264
[14:08] <Mavrik> undercash, use -profile:v baseline
[14:09] <undercash> seems to works!
[14:09] <undercash> hope i dont meet any other stuff , thanx
[14:22] <undercash> hmm i have been installing this "Local" version of ffmpeg for vlc use
[14:22] <undercash> now i start a conversion via script and i get 16 threads.. like if ffmpeg was running double
[14:58] <armetiz> Hi there again
[14:59] <armetiz> I'm sorry, but I have read the documentation many times without any success
[14:59] <armetiz> I still don't understand how video filter "pad" work...
[17:18] <boo> When decoding and streaming video from dvb-t stick to rtmp ffmpeg always quits, at ~ 5.2 seconds, don't know why is this, the only thing that I found out is that by adding -analyzeduration 1500000000 I can extend the running time till 10.3 seconds.. Can anyone of you help me? I'm running ffmpeg witht this method - http://pastebin.com/ZFQcbp5H
[17:20] <boo> If I save the dvb-t stream to a file with 'cat' and paralel read it with ffmpeg, everything is ok
[18:04] <fluXx> Im on Debian using a compiled ffmpeg. If I download VLC from the Debian repos can I use my own ffmpeg binaries and just use VLC as a front end for playback. At times I do use ffplay.
[18:04] <fluXx> Don't mind compiling VLC if that is needed.
[18:06] <JEEB> I think you'll have to compile your own
[18:06] <JEEB> if you want to use the ffmpeg you compiled
[18:06] <JEEB> also, I recommend that if you compile by yourself that you use the 1.2 branch :)
[18:06] <JEEB> which is about to be released as far as I know, and probably supports newer libav* libraries
[18:07] <malinens> hi guys~! I have problems with ffmpeg live transcoding. input stream is x264 level 4.1 but output streams are level 3.0. I get about 5 seconds video freeze at the beginning (sound plays normally) ior green srtifacts at the beginning. what could be the problem?
[18:08] <fluXx> JEEB yeah don't really see a point of having cruft on my system. Hmm I am using ffmpeg version N-32328-g23934e5 not sure what you mean by 1.2
[18:08] <JEEB> VLC version 1.2
[18:08] <fluXx> ahh
[18:08] <JEEB> it seems to be in release candidate
[18:09] <JEEB> it also seems to have its own branch on the git repository
[18:09] <JEEB> thus, 1.2 branch
[18:10] <fluXx> Ill toss it into a VM and give it a spin. If I like the binaries I make I put automate (script) the entire bundle to be compile with a cron job.
[18:10] <JEEB> sounds good enough
[18:10] <fluXx> Thx for the feedback.
[18:21] <boo> anyone? :(
[18:25] <boo> anyone having info about analyzeduration?
[18:56] <elkng> I want to cut audio from 1:0 to the end, what the option I should use ?
[18:58] <snapper> hi
[19:03] <boo> Anyone? Help with analyze duratioN?
[19:17] <arbin> does ffplay usually have trouble selecting audio streams for playback?
[19:18] <arbin> I'm running ffplay -ss 00:01:49 -t 00:20:45 -ast 2 video_file.ext and it crashes, even though there are three audio streams
[19:18] <arbin> only seems to happen on 50% of the files
[19:34] <elkng> what the percentage of ffmpeg's use on piracy videos ?
[19:35] <snapper> doubt you could get an accurate answer for that elkng
[20:17] <Bombo> i encoded this: http://www.file-upload.net/download-4062198/video.mpg.html with "x264 --preset superfast --crf23 -o video.mp4 video.mpg" and result is this: http://www.file-upload.net/download-4062023/video.mp4.html but it wont mux (e.g. ffmpeg): "ffmpeg -i video.mp4 -vcodec copy muxed.mp4" error is: "[mp4 @ 025EC020] Application provided invalid, non monotonically increasing dts to muxer in stream 0: -7200 >= -7200 ...
[20:17] <Bombo> ... av_interleaved_write_frame(): Invalid argument" 1. why 2. what can i do to mux the video.mp4?
[20:19] <madsage> ahh sweet. ffmpeg channel
[20:20] <Bombo> tried muxing with l-smash "muxer -i video.mp4 -o video_muxed.mp4" -> "Error: failed to open input file."
[20:23] <madsage> i'm using ffmeg to rtmp stream to wowza, all my webapp players require a filename. any clue how i provide this? i want to use jwplayer but i cant figure out what to use for filename. it wants a filename with the rtmp stream url.
[20:23] <madsage> ffmeg/ffmpeg
[20:24] <madsage> i'm pretty green with this stuff. if that wasnt obvious. heh
[20:25] <madsage> my ffmpeg command like looks like this. ffmpeg -i /root/usenet/done/Up/Up.mkv -threads 8 -vcodec libx264 -vpre slow -vb 500000 -strict experimental -acodec aac -ab 128000 -ar 48000 -ac 2 -f flv rtmp://127.0.0.1/live/
[20:25] <madsage> command line^
[20:26] <madsage> anybody around have any clue what i'm trying to do?
[20:30] <Mavrik> you're trying to use archaic version of ffmpeg with broken AAC encoder to stream live video :P
[20:31] <Mavrik> the problem with "player requiring filename" is something you need to solve in Wowza documentation
[20:36] <madsage> archaic version of ffmpeg?
[20:36] <madsage> FFmpeg version 0.6.1
[20:39] <Mavrik> yeah, that's even older than what ships with Ubuntu and even that one is old ;)
[20:41] <madsage> ahh ok. i can build from svn. this is what was in the yum RHEL 5.7 repo
[20:43] <madsage> my second problem is studdering. maybe newer version will help. seems to play HD nice for about 20secs and the stream hickups. but i was goign to try a differnt player. when i ran into the filename issue.
[20:43] <madsage> my machine is dual quad 2.66ghz with 16gig ram. shouldnt have any problem
[20:44] <madsage> okay first things first. upgrade ffmeg then search for the filename issue
[20:44] <madsage> thanks Mavrik, i have a direction
[20:48] <madsage> i'll try ffmpeg-0.9.1
[20:49] <Mavrik> madsage, yeah, I'm streaming to Wowza with MPEG-TS intead of RTMP, seems more stable
[20:49] <Mavrik> as for filename usage... go check Wowza mpeg-ts tutorial, it shows clearly how to access the live stream
[20:49] <madsage> ok cool. i appreciate your help. thanks
[20:56] <undercash> mavrick you can stream using ffmpeg?
[20:57] <Mavrik> yes.
[20:57] <undercash> but you need ffmpeg-mt no?
[20:57] <undercash> i m curious how it could work to stream to justin tv
[20:58] <undercash> since i can make vlc to do the job
[20:58] <undercash> cannot*
[20:58] <Mavrik> undercash, no, afaik ffmpeg-mt doesn't exist anymore
[20:58] <undercash> so any recent ffmpeg built can stream
[20:58] <Mavrik> depends on which format and container you need
[20:58] <undercash> you could give me an exemple of a command
[20:58] <undercash> i need to mux on mpeg ts too
[20:59] <undercash> }'}:rtp{dst=127.0.0.1,port=1234,mux=ts,caching=4000,ttl=12,group=10,sdp
[20:59] <madsage> sweet, built nice and clean.. ffmpeg version 0.9.1
[20:59] <undercash> what i use for vlc..
[20:59] <undercash> no idea how this would translate with ffmpeg though
[20:59] <undercash> but it s pretty interesting
[21:00] <Mavrik> madsage, don't forget codecs you need :P
[21:01] <madsage> Mavrik, enable via configure? i was going to ask.. assumed maybe default configure file would be set for standard usage
[21:01] <Mavrik> madsage, well, if you don't have headers for libx264, libfaac or anything else you want to use it's not going to enable them
[21:02] <Mavrik> and IIRC it doesn't enable them even if you have them by default
[21:02] <madsage> yeah ok
[21:02] <Mavrik> (not sure about the last one)
[21:02] <madsage> i was goign to build and try
[21:02] <madsage> i believe i have all the dev headers installed
[21:02] <Mavrik> madsage, as I said, don't forget libx264 and libfaac
[21:03] <Mavrik> (or libvo_aacenc instead of libfaac)
[21:05] <madsage> oh yeah it helps to read
[21:05] <madsage> - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
[21:06] <Mavrik> madsage, there's a great ubuntu guide in the topic ;)
[21:06] <madsage> yeah i'm using cent but should help. thanks
[21:14] <pasteeater> undercash: ffmpeg-mt was merged into ffmpeg some time ago.
[21:14] <undercash> hi pasteeater
[21:14] <undercash> ok
[21:14] <pasteeater> and i've used ffmpeg with justin.tv before, but that was just a video of my boring desktop
[21:15] <undercash> guess i should read the book ;)
[21:15] <undercash> i see
[21:15] <undercash> u remember of a command line maybe?
[21:15] <pasteeater> madsage: https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide
[21:17] <madsage> ahh ok cool. thanks for that
[21:17] <pasteeater> undercash: not really, but i'll try it...if i remember my login
[21:17] <undercash> i should look for rtp ffmpeg streaming i guess
[21:19] <undercash> assuming i find that working command i ll still have the problem of the sdp file , because I use jtvlc to send the stream to justin
[21:19] <undercash> unless ffmpeg can create this file
[21:23] <Mavrik> undercash, ffmpeg creates the SDP file when you start streaming
[21:23] <Mavrik> however I suggest you check if you can to RTSP, MPEG-TS(UDP) or RTMP which is easier to configure
[21:23] <undercash> yes i m reading that
[21:23] <undercash> if it creates sdp , it should be working
[21:31] <pasteeater> where do you find the stream key in justin.tv now? they've changed things since i logged in last time
[21:32] <pasteeater> ah, it's in http://www.justin.tv/broadcast/adv_other
[21:36] <undercash> yes
[21:36] <pasteeater> undercash: this worked fine for me: ffmpeg -f x11grab -r 30 -s 1024x720 -i :0.0 -c:v libx264 -preset ultrafast -crf 28 -f flv rtmp://live.justin.tv/app/<stream key>
[21:36] <undercash> wjen u chose destock streaming u have the key there
[21:37] <undercash> where is your input on this command?
[21:37] <pasteeater> -i :0.0
[21:37] <undercash> i need to put ffplay to play a file right?
[21:43] <Viper550> Are there FFmpeg binaries for 64-bit Windows?
[21:44] <Mavrik> Viper550, yeah, Zeranoe builds his here: http://ffmpeg.zeranoe.com/builds/
[21:45] <pasteeater> undercash: maybe something like this ffmpeg -i IronMan.mkv -c:v libx264 -preset fast -crf 32 -vf scale=iw/2:-1 -c:a libmp3lame -aq 7 -ar 44100 -f flv rtmp://live.justin.tv/app/<stream key>
[21:45] <pasteeater> my net connection is shit, so you'll need to tweak it to something less mental
[21:45] <pasteeater> workedforme
[21:46] <pasteeater> or use a video that's already good to go instead of re-encoding on the fly like my example
[21:47] <madsage> ffmpeg version N-37192-g88d84dd
[21:47] <madsage> wth is this. from trunk or something?
[21:48] <madsage> i used git clone
[21:48] <madsage> must be really new. =]
[21:50] <pasteeater> yes. N-<revision number>-<git hash>
[21:51] <undercash> thanks pasteeater
[21:51] <undercash> no luck yet but
[21:52] <madsage> pasteeater, thanks
[21:52] <madsage> was just currios
[21:58] <madsage> pasteeater, your cent compile url was a HUGE help. saved me hours. thank you
[22:00] <pasteeater> madsage: good. it's a wiki page, so anyone should feel free to modify/update/improve it.
[22:04] <madsage> worked like a champ, and all latest library releases
[22:04] <madsage> which all mine were old
[22:09] <undercash> [rtmp @ 0x217b020] Cannot read RTMP handshake response
[22:09] <undercash> rtmp://live.justin.tv/app/xxxxxx: Input/output error
[22:09] <undercash> it doesn't write any error using; ffmpeg -i "input.mp4" -c:v libx264 -preset ultrafast -crf 28 -f flv rtmp://live.justin.tv/app/xxxx
[22:10] <undercash> but it doesnt do anything either
[22:21] <madsage> what does -crf32 do
[22:21] <madsage> err -crf 32
[22:22] <madsage> i was lookign at the example pasted in the channel a few minutes ago
[22:22] <madsage> anyways i'm getting an error: Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[22:23] <madsage> when i try to use an mkv
[22:23] <madsage> ffmpeg -i Up.mkv -c:v libx264 -preset slow -vb 500000 -crf 32 -c:a libmp3lame -aq 7 -ar 44100 -f flv rtmp://127.0.0.1/live/
[22:24] <madsage> works with a .mp4 file, very nicely
[22:24] <madsage> oh i see it. i think
[22:25] <madsage> [libmp3lame @ 0x186eef40] Invalid number of channels 6, must be <= 2
[22:25] <Mavrik> madsage, -crf sets quality
[22:25] <undercash> crf define a flexible bitrate
[22:25] <madsage> ahh ok
[22:25] <Mavrik> madsage, don't use it with -vb because they can't be used together
[22:25] <madsage> i fixed the 6 channel
[22:25] <madsage> -ac 2
[22:25] <Mavrik> madsage, either you set bitrate or constant quality (crf)
[22:26] <madsage> crf should work well?
[22:26] <madsage> i'll try it
[22:26] <Mavrik> madsage, people usually prefer CRF
[22:26] <madsage> and the value 32?
[22:26] <Mavrik> because it makes files of similar quality
[22:26] <Mavrik> it's awful
[22:26] <Mavrik> start with ~25
[22:26] <madsage> ok
[22:26] <Mavrik> if your increase it, the quality worsens, if you lower it the quality gets better
[22:27] <madsage> its working =]
[22:27] <madsage> frame= 323 fps= 23 q=30.0 size= 1043kB time=00:00:10.63 bitrate= 803.1kbits/s
[22:27] <undercash> paasteater maybe it s because my rtmp port seems to be closed
[22:27] <madsage> ahh it just studdered :\
[22:27] <boo> trying to emerge ffmpeg http://pastebin.com/GxmJPuKH
[22:28] <boo> Mavrik, finally, you are here! :) so, I realized that if I set -analyzeduration 1500000000 then the decoding stops not at 5.2 seconds, but at 10.3, so there's something related to that analyze thing
[22:29] <Mavrik> boo, that just means it pre-buffers a larger chunk of data to analyze it :)
[22:30] <boo> Mavrik, can you disable that analyzing? or doesn't it give somekind of a hint where's the problem
[22:38] <madsage> Mavrik, hey when you get a chance, you mentioned using MPEG-TS instead of RTMP and a wowza tutorial. i've searched and find that tutorial. i'll keep looking maybe its my keywords.
[22:39] <madsage> btw, i have the RTMP stearm lookign pretty nice now! 10x better just upgrading my tools
[22:39] <madsage> stearm/stream
[22:46] <madsage> man this looks so sweet. and it only studdered once. i'm doing the encoding and streaming from the same machine tho. oh and i forgot. i have assigned threads yet. i have a max of 8
[22:46] <madsage> have/havnt
[22:47] <madsage> one of our graphics guys just said it looks like blueray. i guess thats a compliment
[22:57] <jermy> boo/Mavrik: Remember there are two values related to analyzing - one for time, one for size
[23:00] <jermy> (size probably being probesize)
[23:28] <madsage> if i use -bufsize command and i buffering everything before output? so thsi should help with any studdering (inputt not keeping up with output)
[23:28] <madsage> and/am
[23:28] <madsage> the docs just say 'Set video buffer verifier buffer size (in bits).'
[23:30] <madsage> also one more question. -aq which intiger is higher quality 1 or 10
[23:38] <boo> madsage, where from and how do you stream it? jermy, mn any ideas how to fix my ffmpeg from failing?
[23:39] <madsage> i'm streaming from a file
[23:39] <madsage> to wowza media server via RTMP
[23:40] <madsage> ffmpeg -i Up.mkv -threads 8 -c:v libx264 -preset fast -crf 20 -c:a libmp3lame -aq 7 -ar 44100 -ac 2 -f flv -bufsize 524288 rtmp://127.0.0.1/live/
[23:41] <madsage> Up.mkv is my media
[23:41] <madsage> and rtmp://127.0.0.1/live/ is where wowza media server pickup the feed
[23:41] <boo> madsage, why decided to take wowza, not red5? Was it difficult to set it up?
[23:42] <boo> madsage, I understand, I'm streaming dvb-t stick data to my rtmp server, have trouble with ffmpeg stopping because of the fact that input is a stream..
[23:42] <madsage> no pretty easy, and according to research i've done. wowza is the best as far as stability and load handling. my project will be used in production with some heavy loads
[23:43] <madsage> ahh, i havnt used ffmpeg with external input yet
[23:43] <madsage> i've seen some examples in my readings
[23:43] <madsage> googling around
[23:44] <madsage> you use red5? how is that one
[23:44] <madsage> i didnt even try it.
[23:45] <madsage> that was goign to be my alternative if i couldnt secure the wowza license
[23:45] <madsage> company i work for forked out for it
[23:46] <boo> madsage, setting up was 3sec, but I havn't configured it yet. Still sometimes seems that it acts wierd, Isn't wowza free?
[23:47] <madsage> wowza is comercial
[23:47] <madsage> they do have a 30day demos
[23:47] <boo> oh
[23:48] <boo> madsage I hate that input stream.. http://pastebin.com/vRAsWZCs if you don't have that analyzeduration, then ffmpeg manages to decode 5.2 seconds. if I set the analyze, then it fails at 10.3 sec
[23:50] <madsage> so you using a cam or what kind of media
[23:50] <boo> usb dvb-t stick
[23:50] <madsage> ahh ok
[23:50] <madsage> linux?
[23:50] <boo> yes
[23:51] <madsage> yeah you are using the /dev/whatever in the input path i take it
[23:51] <boo> could be that ffmpeg doesn't recognize some of the settings, therefore by trying to find some info it fails..
[23:52] <madsage> or are you piping the dvb device to another device
[23:52] <madsage> like some of the media centers do, myth comes to mind
[23:53] <boo> madsage, directly from the dvb-t stick to ffmpeg and then to red5
[23:53] <undercash> pasteeater do you know where ffmpeg create the sdp file?
[23:54] <madsage> boo, wish i could help with that. maybe see how mytv handles those devices. might be worth a look.
[23:55] <madsage> mythtv^
[23:55] <madsage> or maybe i'm complicating it too.
[23:55] <boo> madsage, will try to look around, this is driving me crazy for the 3rd day
[23:56] <madsage> i believe mythTV pipes the signal to a special device so as to seperate some of the otehr controls. like channel changing etc
[23:56] <madsage> if i'm not mistaken
[23:56] <madsage> ffmplayer maybe seeing that stuff i dont know
[23:56] <madsage> er ffmpeg
[23:58] <boo> should find a way to store for example ~ 10mb of that stream from dvb-t stick on hdd and then ffmpeg would take info from that
[00:00] --- Fri Jan 27 2012
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