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November 2011

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ffmpeg-devel.log.20111129
by burek 30 Nov '11

30 Nov '11

[00:02] <michaelni> thx anyway :) [00:37] <CIA-41> ffmpeg: 03Alex Converse 07master * r999e7ebd23 10ffmpeg/libavcodec/dca.c: dca: Replace oversized unused get_bits() with skip_bits_long(). [00:37] <CIA-41> ffmpeg: 03Alex Converse 07master * r2b45222b6a 10ffmpeg/libavformat/md5proto.c: md5proto: Fix order of operations. [00:37] <CIA-41> ffmpeg: 03Diego Biurrun 07master * rc88ebdb42c 10ffmpeg/ (18 files in 2 dirs): Eliminate pointless 0/NULL initializers in AVCodec and similar declarations. [00:37] <CIA-41> ffmpeg: 03Mans Rullgard 07master * r60084a1723 10ffmpeg/libavutil/timer.h: [00:37] <CIA-41> ffmpeg: timer: fix misspelling of "decicycles" [00:37] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:37] <CIA-41> ffmpeg: 03Mans Rullgard 07master * rb94a3b288e 10ffmpeg/doc/APIchanges: [00:37] <CIA-41> ffmpeg: APIchanges: fill in some blanks [00:37] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:37] <CIA-41> ffmpeg: 03Alex Converse 07master * ra27805189b 10ffmpeg/libavcodec/txd.c: txd: Fix order of operations. [00:37] <CIA-41> ffmpeg: 03Alex Converse 07master * rac47e014bb 10ffmpeg/libavformat/ (adtsenc.c mpegtsenc.c): [00:37] <CIA-41> ffmpeg: adtsenc: Check frame size. [00:37] <CIA-41> ffmpeg: Inspired by work from: Michael Niedermayer <michaelni(a)gmx.at>. [00:37] <CIA-41> ffmpeg: Signed-off-by: Alex Converse <alex.converse(a)gmail.com> [00:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4dcd1a3145 10ffmpeg/: (log message trimmed) [00:37] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [00:37] <CIA-41> ffmpeg: * qatar/master: [00:37] <CIA-41> ffmpeg: adtsenc: Check frame size. [00:37] <CIA-41> ffmpeg: txd: Fix order of operations. [00:37] <CIA-41> ffmpeg: APIchanges: fill in some blanks [00:37] <CIA-41> ffmpeg: timer: fix misspelling of "decicycles" [00:37] <CIA-41> ffmpeg: 03Kostya Shishkov 07master * r1469f943ad 10ffmpeg/libavcodec/indeo3.c: [00:37] <CIA-41> ffmpeg: indeo3: cosmetics [00:37] <CIA-41> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [00:52] <kcm1700> thanks michaelni [01:19] <michaelni> kcm1700, by evaluating the center point and considering its value and the dx and dy it should be possible to skip a larger class of cases [01:24] <kcm1700> What would be the maximum value of 'shift' variable? [01:25] <kcm1700> if it's not that large, it'll be possible to use LUT. [02:02] <j-b> violet: ask michaelni or burek [02:02] <violet> thank you [05:05] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rfc09bf57a6 10ffmpeg/ (17 files in 5 dirs): [05:05] <CIA-41> ffmpeg: movenc: Write file with minimal number of chunks for the given interleaving. [05:05] <CIA-41> ffmpeg: Reviewed-by: Baptiste Coudurier <baptiste.coudurier(a)gmail.com> [05:05] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:02] <ubitux> why fill_char=0x80 when sample format is u8 and not 0 like all the others? [08:02] <ubitux> in generate_silence() [08:03] <Tjoppen> .. because it's u8? [08:05] <ubitux> i don't see the point [08:05] <ubitux> sorry if it is obvious :p [08:05] <Tjoppen> u8 = s8+128 [08:07] <ubitux> oh i'm stupid, i was thinking we had U16, U32 and such and was surprised of to have it only in U8 [08:07] <ubitux> thanks, makes sense obviously. [08:07] <Tjoppen> yeah, u8 is basically only old formats [08:08] <Tjoppen> if you mix up u8 and s8 when playing out on a sound card you end up with.. fun [08:08] <ubitux> i guess i'll have to fix the map channel silence then [09:16] <ubitux> huh, got a really strange bug; i have a vfilter which i called like this: -vf foobar=10,scale=300:200, and one field in the private struct of foobar gets updated with the height of the scale parameter [09:17] <ubitux> quite fun, i wonder what i'm doing wrong here :) [09:17] <ubitux> with the height parameter* of the scale filter* [10:27] <ubitux> btw, http://ffmpeg.org/doxygen/trunk/index.html why no lavfi? :( [12:59] <j-b> good morning [13:39] <burek> morning :) [13:40] <burek> i have a lot of these spell checkings done, so I'll try to setup git now and update those docs :) [14:58] <burek> michaelni or pasteeater, on what e-mail should I send patch for spelling errors? [15:02] <cbsrobot> burek: ffmpeg-devel [15:02] <burek> that's an email? :) [15:03] <cbsrobot> heres the form: https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/ [15:04] <cbsrobot> :) [15:04] <burek> oh ok [15:04] <burek> :) [15:04] <burek> but [15:04] <burek> wouldn't it go public and stuff? am i supposed to make it publicly visible ? :) [15:17] <iive> burek: are the spelling errors already publicly visible? [15:17] <burek> i've just sent the patch to the ml [16:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r6ffdc262b0 10ffmpeg/libavcodec/ac3enc_template.c: [16:10] <CIA-41> ffmpeg: ac3enc: clenaup project name in one comment [16:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r313d30c180 10ffmpeg/libavutil/ (avutil.h opt.h): [16:10] <CIA-41> ffmpeg: avutil: revert project name messing [16:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:14] <michaelni> burek, you should split the spelling fixes so that each students work is one commit and give credit to each student in each commit [16:14] <michaelni> and subscribe to ffmpeg-dev :) [16:15] <michaelni> the list is subscribers only as a spam prevention [16:19] <burek> oh man.. [16:19] <burek> they will already get payed for that :) [16:19] <burek> and changes are really small [16:19] <burek> so its more work for me to split all that and stuff [16:19] <burek> then it is their benefit of credits for the same thing [16:21] <michaelni> its less than 5min work [16:22] <michaelni> just a git reset HEAD^ and a few git commit thisfile.texi [16:23] <michaelni> about credit, you and them should get credit [16:24] <michaelni> about work, its true gci is alot of work [16:24] <michaelni> maybe you should ask the student to send a applyable patch to the texi next time ... [16:28] <burek> i never used git for comitting and editting and stuff [16:28] <burek> so its not 5 min for me, unfortunatelly :S [16:29] <michaelni> burek, we will see :) [16:29] <michaelni> does git log show "that" commit as last ? [16:30] <michaelni> last is the topmost [16:30] <burek> er.. can you ask me something about beer? :D [16:30] <burek> im more likely to know the answer :) [16:31] <michaelni> whats your favorite beer? ;) [16:31] <burek> the cold one! :D [16:31] <burek> that was easy ^^ [16:35] <michaelni> burek, now try "git log -1" typed in the terminal where your git stuff is and confirm thats the commit with all the spelling correction [16:35] <michaelni> "git log -1 -p" would show the actual diff [16:35] <burek> commit fc09bf57a60d4c4a6d339b204b3282337067c06d [16:35] <burek> Author: Michael Niedermayer <michaelni(a)gmx.at> [16:37] <michaelni> burek, did you commit it to another branch ? [16:37] <burek> burek i think [16:37] <michaelni> git checkout "that branch" [16:37] <burek> yup it's the last [16:37] <michaelni> good, now just to make sure do a "git diff" and make sure theres no output [16:38] <burek> there is no [16:38] <michaelni> good, now a "git reset HEAD^" that will kill the last commit but leave the files as tehy are [16:38] <burek> ok [16:39] <burek> yes, it showed me resetted files i guess [16:39] <burek> all texi [16:39] <burek> 12 of them actually [16:40] <michaelni> and now just a few "git commit -s doc/thisfile.texi" for each commit you want to make and mention in each the student upon whos submission its based, you will be credited as author of the commit [16:40] <michaelni> automatically [16:41] <burek> i see, but some documents are spanned across several texi files [16:42] <burek> which are included with @include or something in the ffmpeg.texi [16:42] <michaelni> you can specify multiple files for commit if its all the work of the same student [16:42] <burek> so i guess i need to check which one is whose [16:59] <CIA-41> ffmpeg: 03Geek.Song 07master * r9cdf048ad2 10ffmpeg/libavformat/ (movenc.c movenc.h): [16:59] <CIA-41> ffmpeg: movenc: Remove unneeded chunkSize field from MOVIentry [16:59] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:12] <vivienschilis> abou [17:14] <CIA-41> ffmpeg: 03Clément BSsch 07master * r81a65b82fb 10ffmpeg/libswresample/audioconvert.c: swr: handle correctly muted channel with u8 sample fmt. [17:34] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4b4a02b847 10ffmpeg/libavcodec/utils.c: [17:34] <CIA-41> ffmpeg: lavc: dont call set_dimensions() on h264 codec init. [17:34] <CIA-41> ffmpeg: This fixes ffprobe showing an incorrect width with [17:34] <CIA-41> ffmpeg: http://panda-test-harness-videos.s3.amazonaws.com/panda.mp4 [17:34] <CIA-41> ffmpeg: Idea-by: Joakim Plate <elupus(a)ecce.se> [17:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:57] <burek> michaelni, i've sent 4 patches (as attachments) to the ml :) [17:57] <burek> can you check if thats all ok? [18:07] <burek> also, how can i now continue working, but to be able to know where did i stop now, to distinguish new changes from these, already made [18:15] <ubitux> burek: you can git commit --amend --author=... [18:15] <ubitux> instead of writing the credits in the commit message [18:16] <burek> ok, but i already started this way, so i'll just finish like that, and next time use --author :) [18:16] <burek> thx :) [18:18] <ubitux> well, if you don't do it, michaelni might do it instead :p [18:19] <ubitux> maybe you could give a hand ;) [18:19] <michaelni> ill reword the commit messages a little as they should contain the "what/where" in the first line [18:20] <michaelni> iam way too lazy to swicth author/credit around, both burek and the student should be credited [18:20] <ubitux> hehe ok :) [18:20] <michaelni> the students submited changed html we needed a patch to texi ... [18:21] <burek> i dont need to be credited :) [18:21] <burek> just please lets get this done with :D [18:21] <michaelni> exactly my thought [18:21] <burek> it takes too much time for simple tasks [18:21] <burek> :) [18:21] <michaelni> ill do it [18:21] <burek> thanks :D [18:21] <burek> :beer: :) [18:22] <michaelni> and btw --author needs full emails so i cant use it anyway ... [18:23] <michaelni> next time i think its better to let the student do all work and then credit just them [18:29] <michaelni> i mean at least require a changed texi file, its easy to make a patch but hard to convert between formats [18:34] <CIA-41> ffmpeg: 03root 07master * r1c212a6465 10ffmpeg/doc/general.texi: [18:34] <CIA-41> ffmpeg: general.texi: fix spelling errors [18:34] <CIA-41> ffmpeg: credits to: Samuel M (from Google Code-in) [18:34] <CIA-41> ffmpeg: Signed-off-by: burek <burek021(a)gmail.com> [18:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:34] <CIA-41> ffmpeg: 03root 07master * raadbf9f74c 10ffmpeg/doc/developer.texi: [18:34] <CIA-41> ffmpeg: developer.texi: fix spelling errors [18:34] <CIA-41> ffmpeg: credits to: KayC (from Google Code-in) [18:34] <CIA-41> ffmpeg: Signed-off-by: burek <burek021(a)gmail.com> [18:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:34] <CIA-41> ffmpeg: 03root 07master * rb0a90c2004 10ffmpeg/doc/faq.texi: [18:34] <CIA-41> ffmpeg: faq.texi: fix spelling errors [18:34] <CIA-41> ffmpeg: credits to: Philip (from Google Code-in) [18:34] <CIA-41> ffmpeg: Signed-off-by: burek <burek021(a)gmail.com> [18:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:35] Last message repeated 1 time(s). [18:35] <Compn> root is an odd name for a committer :P [18:48] <burek> fck.. i checked out master and made changes and now i have to do it again in burek branch.. [18:50] <burek> i knew i would crew up something.. :) [18:51] <burek> so, first i created burek branch, made all changes and sent an email to the ml [18:51] <burek> after that, i was doing those git commands you told me, and in all those cmds there was git checkout master [18:52] <burek> after that i made new changes (which should have been made in burek branch), because texi files overlap [18:52] <burek> and some changes have alredy been fixed [18:52] <burek> so, now i ended up with diff from master instead from burek [18:52] <burek> fun never stops :D [19:56] <CIA-41> ffmpeg: 03Reimar Döffinger 07master * r7076967786 10ffmpeg/libavcodec/x86/imdct36_sse.asm: [19:56] <CIA-41> ffmpeg: Consistently use %ifdef ARCH_X86_64 [19:56] <CIA-41> ffmpeg: One out of 3 places used ifndef, which is needlessly confusing. [19:56] <CIA-41> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:08] <bcoudurier> michaelni, stop procrastinating about avfilter flush [22:09] <bcoudurier> I sent you patch ages ago to fix this [22:24] <pasteeater> what is "coverage support"? [22:25] <Compn> code coverage, check melansons blog [22:27] <gnafu> http://multimedia.cx/eggs/using-lcov-with-ffmpeg/ [22:27] <Compn> that [22:34] <pasteeater> ah, thanks. [22:38] <michaelni> bcoudurier, do you remember the subject of the mail with the patch or something so i can find it ? [23:15] <CIA-41> ffmpeg: 03Clément BSsch 07master * rb6ffe441cd 10ffmpeg/ffmpeg.c: ffmpeg: do not use a negative total_size (AVERROR) in bitrate estimation. [00:00] --- Wed Nov 30 2011

1 0

ffmpeg.log.20111129
by burek 30 Nov '11

30 Nov '11

[00:17] <w3gi> hallo... i have a flv-file and need to know the codec... how can i do this? [00:18] <w3gi> i mean the vcodec [00:18] <inashdeen> hi do i record sharp video from my webcam [00:18] <sacarasc> w3gi: ffmpeg -i foo.flv [00:18] <sacarasc> That should tell you. [00:19] <sacarasc> inashdeen: What do you mean by 'sharp'? [00:20] <inashdeen> the common ffmpeg -f oss -i /dev/dsp -f video4linux2 -s 320x240 -i /dev/video0 ~/Videos/recording.mpg doesnt give me a nice video [00:20] <w3gi> Stream #0.0: Video: flv, yuv420p, 720x410, 65 kb/s, 25 tbr, 1k tbn, 1k tbc [00:20] <w3gi> Stream #0.1: Audio: mp3, 44100 Hz, 2 channels, s16, 224 kb/s [00:21] <sacarasc> inashdeen: Because MPEG1/2 is pretty poo... Try -vcodec libx264 -profile fast -crf 20 ~/Videos/recording.mkv [00:21] <w3gi> 1 min. / ca. 7-8 MB but when i use kdenlive (which use ffmpeg) the same vid. have 22-26 MB [00:22] <w3gi> some idea wich vcodec i should set to get the video smaller? [00:23] <sacarasc> Just set a lower bitrate. [00:23] <inashdeen> sacaracs : actually this is my pob. the libx264 gimme Unknown decoder 'libx264'. but this only occur if i run it in bash, if i run it from terminal, the problem do not occur [00:23] <sacarasc> ffmpeg -f oss -i /dev/dsp -f video4linux2 -s 320x240 -i /dev/video0 -vcodec libx264 -profile fast -crf 20 ~/Videos/recording.mkv [00:24] <sacarasc> That;st the full command, to get your error, you're writing it wrong. [00:24] <w3gi> i was using PiTiVi before... with the same bitrate... just the app crash all 5 klicks :( [00:25] <w3gi> i just try to get per kdenlive the same result with size and quality [00:30] <inashdeen> i got this with that, and i didnt even change one thing http://pastebin.com/VvMXz2mB [00:31] <pasteeater> change "profile" to "preset" [00:31] <inashdeen> Unrecognized option 'preset' ffmpeg: failed to set value 'fast' for option 'preset' [00:32] <pasteeater> pastebin the complete, uncut, full, total console output [00:33] <inashdeen> http://pastebin.com/D6fjtktN [00:35] <pasteeater> your ffmpeg is too old to use "preset". use "vpre" instead. [00:37] <inashdeen> how is that? just change it to vpre? [00:37] <pasteeater> yes [00:37] <pasteeater> your version used an older syntax, IIRC [00:38] <inashdeen> now it says "Unknown decoder 'libx264' " again :( [00:39] <sacarasc> You're writing the command wrong. [00:39] <inashdeen> then how should i do it [00:40] <sacarasc> ffmpeg [input options] -i input [output options] output [00:40] <sacarasc> -vcodec libx264 is an output option. [00:41] <inashdeen> ffmpeg -f alsa -ac 2 -i pulse -y -f video4linux2 -s 320x240 -i /dev/video0 -vcodec libx264 -vpre fast -crf 20 /home/ihsan/test.avi [00:41] <pasteeater> or you may need to install libavcodec-extra-53 [00:41] <pasteeater> don't use avi. it doesn't like H.264 video with b-frames. [00:42] <pasteeater> try mp4 or mkv instead [00:43] <inashdeen> the problem is constant with mp4 and my ubuntu says my libavcodec-extra-53 is already the newest. [00:44] <pasteeater> your error said "decoder" and I at first read "encoder", so ignore the libavcodec-extra-53 suggestion. [00:44] <inashdeen> is there a way not to use libx264? [00:44] <pasteeater> "the problem is constant with mp4". what is the problem? [00:45] <inashdeen> it shows "Unknown decoder 'libx264' " [00:45] <pasteeater> use a pastebin site to show your ffmpeg command(s) and the complete console output(s) [00:46] <inashdeen> http://pastebin.com/z6cKzLwC [00:47] <pasteeater> your command is missing [00:47] <pasteeater> include your command and the output in the paste [00:48] <inashdeen> i cant put the command directly because it is kind of auto generated. you know, something like zenity. but i can write it manually cause i wrote the script [00:48] <inashdeen> is it ok? [00:49] <pasteeater> somehow you need to show whatever command you used so we can see what is wrong [00:50] <pasteeater> it's like if you ask a mechanic what is wrong with your car, but you don't bring the car [00:50] <pasteeater> or show the script so i can at least have something to guess with [00:51] <pasteeater> until then i'm agreeing with sacarasc that your -vcodec libx264 placement is incorrect [00:52] <inashdeen> http://pastebin.com/gdurfwHB [00:53] <inashdeen> i copy this directly from the code [00:54] <pasteeater> why not run the command directly and avoid the script until you know that the command works? [00:54] <sacarasc> inashdeen: Change it to ffmpeg $audio $video $location/$name$format [00:54] <sacarasc> And format really should be mkv or mp4. [00:56] <inashdeen> amazingly that works [00:56] <inashdeen> sacarasc: thanx amillion :) [00:57] <sacarasc> You had an -i after -vcodec [00:57] <inashdeen> what that suppose to mean? [00:57] <sacarasc> All input comes before output. [00:58] <relaxed> before tells ffmpeg to decode using that codec [00:58] <inashdeen> ok.huhuhu. thanx will learn ffmpeg more next time.my first time using it, seriously [00:59] <relaxed> we should ban troubleshooting scripts in here. [01:12] <ZacS123> hows does one use the stereo3d filter? I can see the code for it, but I get a no such filter error when trying to use it [01:20] <pasteeater> ZacS123: hard to guess without seeing your command [01:56] <ZacS123> pasteeater: all I tried was ffmpeg -i input.mov -vcodec libx264 -vf "stereo3d" output.mp4 [02:03] <MarioMey> Hello, everybody. [02:03] <MarioMey> I have to convert a image secuence from 0151.png to 0200.png... FFMPG doesn't accept to start in 0151... [02:03] <MarioMey> How to do it? [02:04] <MarioMey> trying: [02:04] <MarioMey> ffmpeg -f image2 -i %04d.png -crf 20 -aspect 16:9 -r 30 -b 6000k -s 1024x512 out.mp4 [02:04] <sacarasc> cat *.png | ffmpeg -f image2pipe - [blah] [02:05] <MarioMey> why image2pipe doesn't apear in doc? [02:05] <relaxed> I'm not sure you can cat pngs unless they've fixed that recently. [02:05] <pasteeater> ZacS123: try -vf mp=stereo3d [02:06] <relaxed> MarioMey: If that doesn't work --> http://ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_003f [02:10] <MarioMey> relaxed: thanks...! [02:10] <MarioMey> sacarasc: thanks you too. [02:20] <ZacS123> pasteeater: that seemed to work, just need to figure out the arguments now to the stereo3d filter and how to feed both eyes in [02:38] <ZacS123> pasteeater: looking at it further it doesn't look like filter is what im after, im looking to produce a side-by-side, or above-below stereo output [02:38] <pasteeater> do you have an input for each side? [02:44] <echelon> how do i prevent audio/video from going out of sync? [02:44] <pasteeater> ZacS123: might not be much different than mplayer's stereo3d other than the parameters must be integers: http://pastebin.com/7ahtYJLP [02:45] <pasteeater> echelon: check your timestamps and frame rates. [03:00] <ZacS123> pasteeater: Is there any filter that I can use to merge 2 frames into a single frame, i.e. to create my side-by-side or above-below stereo frame? [03:04] <relaxed> ZacS123: you can do it with the overlay filter [03:06] <relaxed> along with the pad filter to create the "canvas" where the overlay will go [05:30] <relaxed> ZacS123: Did you figure it out? [05:50] <ZacS123> relaxed: I'm still working through it, I am building an app with libavcodec [05:51] <ZacS123> im just restructuring my application now to accept a left and right input frame and use an overlay filter [05:54] <relaxed> I have some examples if you need them. [07:00] <grepper> making a 720x480 8000kps dvd spec mpg, what is the best filter to use to reduce pixelation in dark areas ? [07:01] <grepper> *kb/s [07:02] <[two]> ffmpeg is really good with single audio/video file but why does it have so much problem with splitted vob/aob files with IFO [07:04] <relaxed> you can concat the vobs but it can't read the dvd video format if that's what you mean. [07:06] <Xd7mT> hello [07:11] <Xd7mT> i am used ffmpeg version 0.6.2-4:0.6.2-3 and recorded video with my webcam used huffyuv codec. ~3 hours in result 27 GB. container - mkv. result - file unplayable with vlc, but after coding it with ffmpeg2theora with 'pro' parameters (see offsite) iv got 1,6 GB (27 GB to 1,6 GB), but quality is too worst. Previously im tried to record with mpeg4 (default) codec simalteneously with recording and got ~6 GB and better (much better) quality. Only one question: [07:13] <Xd7mT> hm. latest version seems to be a 0.8.6, but in my debian testing i have 0.6.2 [07:16] <Xd7mT> no. my version is 0.7.2 [07:20] <[two]> relaxed even if you join splitted vobs files, it still needs to understand chapter information from .ifo file [07:22] <relaxed> [two]: you have to dump the title you want to encode before using ffmpeg. You can do it with `mplayer -dumpstream -dumpfile movie.vob dvd://` [07:23] <[two]> relaxed if joined .vob files has 3 videos inside; and i want to just use video2 with ffmpeg ; what do i type [07:25] <relaxed> do you still have the dvd layout somewhere? [07:25] <[two]> dvdlayout information is located in .ifo [07:28] <relaxed> Figure out how to play what you want to encode with mplayer. Something like `mplayer -dvd-device /path/to/dvd/dir dvd://`, then once you have that add '-dumpstream -dumpfile movie.vob' to the command. You need to google or ask in #mplayer because it's offtopic here. [07:55] <[two]> is it possible to put closedcaptioning other than mpeg2? (closedcaptioning NOT subtitle) [07:58] <[two]> i want to play dvd-audio (audio_ts folder) what is best player that can do this [10:52] <w3gi> hallo... wich vcodec makes with 800k vid. br. the smallest flv-files? [10:53] <Mavrik> um [10:54] <Mavrik> if you set 800k video bitrate the flv file will always be the same :P [10:54] <Mavrik> the quality will differ [10:54] <Tjoppen> flv supports H.264, right? -> x264 [11:02] <w3gi> itry libx264 ... but i get a file with 1009KB wich i can not play in totem e.g. ... [11:03] <Tjoppen> you probably have to use the right profile [11:03] <Tjoppen> like main instead of hig [11:03] <Tjoppen> h [11:04] <w3gi> i use actually kdenlive as frontend... also to compare videos... [11:05] <w3gi> but -vcodec flv makes 26MB files... -vcodec libx264 ca. 1MB files wich seems to be broken... is there any other codec i can use? [11:05] <w3gi> sorry but i am a totally noob in video editing... [11:06] <w3gi> i just try to comp. 2 short clips for our homepage... [11:07] <w3gi> but 26 MB for 1 min. looks for me not really inet friendly [11:07] <Tjoppen> look for an x264 encoding guide [11:17] <w3gi> ok it seems to be problematic... Tjoppen can you tell me another codec i can try? [11:18] <w3gi> or can someone tell me wich vcodec's i can use for a valid flv-file? [11:19] <Tjoppen> try -vcodec flv [11:23] <w3gi> that is the standard wich makes MB [11:26] <w3gi> is there just flv and libx264 and no other codec? [11:27] <JEEB> if you are making a clip to be viewable on the internet, and with flash f.ex. you will want libx264 [11:27] <JEEB> it's just hands up the best encoder (format itself is H.264) for its format, and currently it's hands up the best format for video that you want to give out to people [11:28] <JEEB> also, totem and others probably just fail loading the file :P happens often, you might want to play your output flv/mp4 file in flash or whatever :P [11:28] <JEEB> also, do you care about a specific file size, or quality? [11:30] <JEEB> as in, "every video must fit 5/10MB/XMB specific" or "I will let the encoder use as much space as needed for the quality I set (more or less, since we still don't have thousands of chinese children picking our results inside the encoder)" [11:32] <w3gi> look... vcodec flv gives me 22,5MB / min.... that is to high... libx264 gives me 1 MB / min. wich is nice but i have also problems with the flowplayer... so is there a 3rd option i can try? [11:32] <JEEB> w3gi, you are just using stuff wrong [11:33] <JEEB> now just tell me which option you want and I will try to help you get further [11:33] <JEEB> also, just checking but how old is your ffmpeg? [11:34] <JEEB> certain things have become much easier to use lately + having newer libraries is never a bad thing [11:34] <w3gi> i use kdenlive... f=flv acodec=libmp3lame ab=128k ar=44100 vcodec=libx264 minrate=0 b=800k progressive=1 [11:34] <w3gi> wich render per ffmpeg [11:34] <JEEB> ok, sorry -- I have no idea what exactly kdenlive does [11:35] <JEEB> I only know how to use the command line interface of ffmpeg/avconv to a level [11:35] <w3gi> Fmpeg version 0.6-rpmfusion, Copyright (c) 2000-2010 the FFmpeg developers [11:35] <w3gi> built on Jul 27 2010 03:51:48 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) [11:35] <JEEB> ouch [11:35] <JEEB> that's megassa old [11:35] <JEEB> go update your ffmpeg and libx264 with it [11:36] <JEEB> and LAME / possible aac encoding library while you're at it [11:38] <JEEB> of course kdenlive might be retarded and need a specific ffmpeg version, so it might be that you might not be capable of doing anything with it. In that case I'd just build a newer ffmpeg and friends into my home folder -> output a lossless ffvhuff + PCM avi or something for the actual encoding [11:38] <w3gi> ffmpeg-0.6-3.fc13 is the newest version [11:38] <JEEB> for your distro version, could be [11:38] <JEEB> it's by far not the newest in the real world tho [11:39] <JEEB> welcome to "Distros never keep up", which is more problematic in multimedia stuff [11:39] <JEEB> anyways, whatever was that -acodec format for 16bit 48kHz audio >_> [11:40] <JEEB> w3gi, so yeah -- I recommend that you output your clip as a lossless AVI first [11:40] <JEEB> then it can be encoded into whatever lossy format for distribution with a newer build of ffmpeg [11:41] <JEEB> Because to get a good result from an older ffmpeg, not to mention that it's some weird API being used [11:41] <JEEB> is much much harder [11:41] <w3gi> JEEB, ok maybe... but before i start a stunt with x updetes and and and... maybe i can try another codec i just need 2 videos... that is all and i dont plan in the future more stuff like that [11:41] <JEEB> and sometimes not possible :P [11:41] <JEEB> w3gi, uhhh [11:41] <JEEB> have you been reading at all? [11:41] <JEEB> what I have been writing? [11:42] <w3gi> yes i read [11:42] <JEEB> I'm just telling you "Because you probably don't want to touch your distro too much, you might want to build a newer ffmpeg/libx264/LAME/faac|whatever into your home directory, and just output a lossless intermediate file from kdenlive" [11:43] <JEEB> because unfortunately I'm not really capable or wishing to take you through all the possible quirks of using a lolold version of ffmpeg :3 [11:43] <JEEB> (and yes, that settings part you showed me got quite lulzy after the vcodec part) [11:44] <JEEB> kdenlive -> lossless ffvhuff/PCM file -> encode final clip with newer tools [11:46] <JEEB> so uhm, does this sound alright or do you want to keep banging your head onto something? [11:47] <w3gi> ok... i will try... would be just easier when i would had generally an idea what i do.... :D [11:47] <w3gi> i never have any contact with video editing... :( [11:47] <JEEB> uhh [11:47] <JEEB> why do I "ok... I will try..." for a question "does this sound alright"? [11:48] <JEEB> *why do I get [11:48] <JEEB> I'd like a yes/no answer so I know if I can ignore this channel or not for a while :3 [11:49] <w3gi> ok last question aac is ok for audio in flv or should i use mp3? [11:50] <JEEB> yes, you clearly don't read what I write [11:50] <JEEB> and that depends on flash [11:50] <JEEB> you can put aac into flv [11:50] <JEEB> but I mostly see it in mp4 with flash [11:50] <JEEB> (although yes, lately in rtmp streams etc. there's aac-in-flv so I guess it supports it) [11:52] <w3gi> i read it but i mean what is better... that this sure run... can also depends on the player version i think... [11:52] <JEEB> I think pretty much everyone by now has a flash player version that can take H.264/AAC [11:53] <JEEB> <w3gi> i read it but i mean what is better... that this sure run... <- also what does this mean [11:54] <Mavrik> Adobe claims you should use H.264/AAC in flash anyway [11:54] <w3gi> ok... i was trying to export a avi how you told and use ma old ffmpeg co convert to flv and get now a 1,5MB file ... wich looks nice... [11:54] <JEEB> you're a trainwreck -.- [11:55] <w3gi> i go ahead and try it in flowplayer... maybe the prob. was just in kdenlive [11:55] <JEEB> Mavrik, also -- is it in your opinion so hard to grasp from my text that I might actually tell him how to do most stuff if he only told me what exactly he wanted, starting from the "do you care about specific size or quality" part? [11:56] <JEEB> s/tell/hold his hand/ [11:57] <Mavrik> well [11:57] <Mavrik> JEEB, nope, but if he doesn't read what you write then I can see how there could be a communication problem ;) [11:57] <JEEB> Indeed [11:58] <JEEB> I should probably do something productive [13:10] <w3gi> JEEB, i see now what i need... that is exact what i want: [13:10] <w3gi> video: ffmpeg Flash Video (FLV) Sorenson Spark / Sorenson H.263 / ffenc_flv -> 800K [13:10] <w3gi> audio: L.A.M.E. mp3 encoder - 196k 48000 [13:10] <w3gi> just how to set this... ?? [13:10] <cryptopsy> WHAT? [13:12] <w3gi> i get this settings from another programm... [13:12] <cryptopsy> OKAY, SO WHAT DO YOU WANT TO DO? [13:12] <w3gi> cryptopsy, i want to use this settings for a flv file... i hope someone know what this mean... [13:12] <vivienschilis> is libvpx git master branch stable? [13:13] <cryptopsy> CHECK THE DESCRIPTION, BRO [13:13] <cryptopsy> w3gi: ARE YOU CONVERTING BETWEEN TWO FORMATS? [13:13] <cryptopsy> I JOINED AFTER YOU ASKED THE ORIGINAL QUESTION, I THINK [13:13] <w3gi> yes... avi -> flv [13:14] <cryptopsy> OK YEA ITS PRETTY EASY, CHECK OUT THE EXAMPLES ON EHOW.COM [13:14] <w3gi> audio should be lame mp3 196 / 48000 [13:14] <cryptopsy> USE -ACODEC COPY [13:14] <w3gi> but wich -vcodec is Sorenson Spark / Sorenson H.263 ?? [13:15] <cryptopsy> SET IT ON YOUR INPUT FILE AND USE -ACODEC CPOY [13:15] <cryptopsy> -ACODEC COPY [13:15] <w3gi> the input is much bigger [13:15] <cryptopsy> YEA ... [13:16] <cryptopsy> -umv and -aic are the only ones that have 263 in the name, in the entire man page [13:46] <cryptopsy> w3gi: see you around [13:50] <burek> w3gi, type ffmpeg -codecs | grep -i h263 [14:03] <w3gi> burek, i get this: [14:03] <w3gi> DEVSDT h263 H.263 / H.263-1996 [14:03] <w3gi> D VSD h263i Intel H.263 [14:03] <w3gi> EV h263p H.263+ / H.263-1998 / H.263 version 2 [14:04] <burek> well you can try either h263 or h263p [14:04] <burek> since they are (E)ncoders [14:04] <burek> produce an output [14:05] <burek> and check if it has properties you need [14:05] <fj_> hi everyone [14:05] <Guest90795> everyone high [14:05] <fj_> I'm trying to upgrade ffmpeg library [14:06] <burek> ok, and :) [14:08] <burek> fj_, and? :) [14:09] <fj_> well in my old code I did memcpy(triPkg->encDataBuffer(), packet.data, packet.size); and it works [14:09] <fj_> but now avcodec_decode_video2 takes a whole packet instead of packet.data [14:09] <fj_> so when I do memcpy(triPkg->encDataBuffer(), &packet, packet.size); [14:09] <fj_> it sometimes works but sometimes gives a segmentation fault [14:10] <fj_> is packet.size the size of the whole packet? or just packet.data [14:11] <burek> i guess whole packet [14:11] <burek> wouldn't make sense to be just for data [14:12] <burek> http://ffmpeg.org/doxygen/trunk/files.html [14:12] <w3gi> crash with this settings: f=flv acodec=libmp3lame ab=192k ar=48000 vcodec=h263p minrate=0 b=800k progressive=1 [14:13] <burek> fj_ http://ffmpeg.org/doxygen/trunk/libavcodec_2utils_8c.html#eb2441bb9d76c881f… [14:14] <burek> w3gi [14:14] <burek> can you please use pastebin.com, to show your command line and its output? [14:18] <fj_> thanks but decoding works fine if it doesn't segfault at memcpy [14:20] <burek> i guess it works out of luck [14:20] <burek> since the params changed too [14:20] <burek> its not packet.size anymore [14:20] <burek> read more carefuly [14:21] <burek> anyway [14:21] <burek> did you read the docs what is triPkg->encDataBuffer() [14:21] <burek> did you try with just memcpy(triPkg->encDataBuffer(), &packet.data, packet.size); [14:21] <burek> packet.data.size [14:27] <fj_> packet.size didn't change [14:28] <fj_> also doesn't really matter what size I use [14:31] <fj_> but thanks anyway : ) [14:31] <fj_> guess i will stick with the old ffmpeg [14:53] <Xd7mT> im determined why my video quality was worst. im forgot to set bigger resolution [14:53] <Xd7mT> what resolution is recommended in capture? [14:54] <Xd7mT> hd420 is good? [14:54] <Xd7mT> 480 [14:56] <relaxed> Xd7mT: what was your command? [14:58] <Xd7mT> ffmpeg -f video4linux2 -i /dev/video0 -s hd480 -r 25 -vcodec huffyuv filename.mkv [14:58] <burek> move -s before -i [14:59] <burek> you want to set size of the capture and not to resize an already captured frames [14:59] <burek> the order of parameters is important [14:59] <burek> :) [15:00] <Xd7mT> its very strange but after my calculations video with length ~3 hours must weight 70 GB, but in fact result weight ~24 GB. Ist it cool? [15:00] <burek> Xd7mT what exactly are you trying to do? [15:01] <Xd7mT> im capture video from webcam [15:03] <burek> and? [15:05] <burek> Xd7mT and do what with it? [15:07] <Xd7mT> i am used latest ffmpeg2theora with -p pro option and got not good quality in time when ffmpeg -f video4linux2 ...... [no '-vcodec' option] gives better quality. default codec is mpeg4 [15:08] <Xd7mT> but i wont mpeg. its proprietary, maybe its not fine [15:08] <burek> i see [15:08] <burek> try this [15:09] <burek> ffmpeg -f video4linux2 -s vga -r 25 -i /dev/video0 -vcodec libx264 -crf 20 out.mp4 [15:09] <burek> or .mkv instead of .mp4 [15:09] <Xd7mT> sorry, but h264 propryetary isnt it? [15:11] <burek> libx264 [15:12] <burek> a free variant of h264 [15:12] <burek> http://www.videolan.org/developers/x264.html [15:12] <JEEB> lawl [15:12] <JEEB> H.264 is still H.264 [15:13] <JEEB> so if this person actually cares about not using MPEG formats, then nothing can be done about it [15:13] <JEEB> unfortunately both theora and vp8 are worse off than H.264 compression-wise, but that's how it rolls :P [15:13] <JEEB> oh, and there's dirac and snow as well, but those are... [15:13] <JEEB> "experimental", shall we say [15:13] <JEEB> (as well as snow being 'dead') [15:14] <burek> i mean its free as encoder [15:14] <burek> h264 is just a format [15:14] <burek> o gosh [15:14] <JEEB> yes, but he asked "proprietary" [15:14] <burek> just ignore me [15:14] <relaxed> libvpx is the pinko commie codec of choice [15:14] <burek> :) [15:34] <niro> Hi, is there a way to make ffmpeg output 2 files. And by output 2 files, i mean, save a local copy and also create a stream for ffserver? [15:35] <burek> not right now [15:35] <burek> but you can try to use [15:36] <burek> udp as an output [15:36] <burek> and duplicate it [15:36] <burek> send one stream to a file and another to a streaming server [15:36] <burek> or use vlc [15:39] <niro> so, if i wrote a little program and specified an output file for ffmpeg, and then piped the output too my program, which then duplicated it, would that work? [15:40] <niro> s/too/to/ [15:40] <burek> hmh [15:40] <burek> the main problem is [15:40] <burek> you'll have to make your stream packetized [15:41] <burek> i.e. if you just pipe the output to your program [15:41] <burek> it will have to wait until all of the stream data is piped/transfered to your program [15:41] <burek> in order for it to read the metadata [15:41] <burek> which usually comes at the end [15:41] <niro> it would be a live stream... so that wouldnt work [15:42] <burek> well [15:42] <niro> could i pipe a live stream back into ffmpeg? [15:42] <burek> i'm not sure, i couln't make it in several months without udp or vlc [15:43] <burek> so if you find the way, I'll be more than glad to see an example :) [15:43] <burek> you can pipe video back and to ffmpeg [15:43] <niro> when you say video, do you mean video exclusively, because this is 100 audio [15:43] <niro> 100% [15:43] <burek> i guess you need something like capture - split < file, stream [15:44] <burek> hmh [15:44] <burek> you know what you could do [15:45] <burek> capture audio with ffmpeg, feed that to a media streaming server (like shoutcast or http server like ffserver) [15:45] <burek> and just use another instance of ffmpeg to connect to media server and get/save the stream to a file on localhost [15:46] <niro> i think that could be a brilliant idea... [15:46] <niro> my own worry is... [15:46] <niro> *only [15:47] <niro> im developing for an embedded system, im already having to do live encoding, then serving from the system and then i'll have to reencode to save it as it'll be an flv [15:47] <niro> im just hoping there's enough cpu to go around [15:50] <relaxed> niro: you can have two outputs. did you try it? [15:51] <niro> no, how would i specify it? -o file -o file2? [15:53] <relaxed> ffmpeg -i input.wav -acodec libmp3lame -ab 128k output1.mp3 -acodec libmp3lame -ab 256k output2.mp3 [15:58] <niro> ffmpeg -f alsa -i hw:0,1 -acodec libspeex -f flv rtm:/some/stream/ -acodec libspeex file.spx [15:58] <niro> would that work? [15:59] <relaxed> give it a whirl [16:00] <niro> cool, thanks for your help :) [16:00] <relaxed> maybe the last output should be file.flv [16:01] <niro> the local copy needs to be a .spx file really too keep the size to a minimum [16:12] <burek> well, the 2 outputs example above actually uses double the cpu [16:12] <burek> i think he wanted to use 1 process for encoding and split the work to (1) save to a file and (2) stream it live [16:13] <burek> so, actually the muxing part needs to be done in 2 ways, not encoding [16:27] <relaxed> burek: you could use tee [16:29] <burek> hardly [16:29] <burek> try to make a simple example command line, meaning, grab the audio/video from a webcam, split it with tee and save it / stream it to, say udp or ffm [16:32] <relaxed> Would this not work? ffmpeg -i .... -f flv - 2>/dev/null | tee local.flv | ffmpeg -f -flv -i - -vcodec copy -acodec copy output [16:34] <burek> well, the 2nd ffmpeg would wait until 1st one finishes [16:34] <burek> and then would do vcodec copy and .. [16:34] <burek> try without tee to pipe it like that [16:35] <burek> heck, ill try it now :) [16:36] <relaxed> I don't believe there would be any waiting [16:50] <relaxed> burek: works fine here [17:07] <bf4648> no one is chatting is this room? [17:30] <clockwize> hey, is 30fps hdmi good enough to watch movies? [17:31] <relaxed> should be [17:32] <clockwize> for a media centre, that's good enough? [19:00] <francogrex> Hi, I am trying this but it fails: ffmpeg -i D:/VIDEO_TS/VTS_02_2.VOB -ss 2138 -qscale 7 -vcodec libxvid -s 640x360 -r 23.976 -aspect 16:9 -ab 128k -ar 48000 -async 48000 -ac 2 -acodec libmp3lame -f avi -g 300 -bf 2 output.avi [19:01] <francogrex> ffmpeg just quits without any output... anyone has an idea why? [19:10] <francogrex> ? [19:11] <relaxed> '-async 48000' needs to go [19:11] <relaxed> pastebin your command and all output if that doesn't help [19:12] <relaxed> I'm not sure if qscale woarks with libxvid [19:12] <relaxed> works* [19:14] <francogrex> relaxed: please see: http://paste.lisp.org/display/126163 [19:15] <relaxed> try with -ss 2138 before the input [19:20] <francogrex> gives me: Error while decoding stream #0.1 and outputs a 10k file with nothing [19:20] <francogrex> -ss used to work fine with outher than .vob input [19:23] <beandog> did you actually decrypt the CSS on the dvd? [19:23] <beandog> It looks to me like you're trying to access it directly [19:24] <francogrex> beandog: i don't think that's the problem because without -ss the same command above works very fine [19:35] <francogrex> actually -ss 30 (or any small number) also works fine; it's when it goes to large numbers that it bugs [19:37] <pasteeater> your original -ss number was longer than your input duration [19:37] <pasteeater> if you used a more recent ffmpeg it would have told you to check your -ss [19:38] <pasteeater> i don't see this happen often, but this is the third time in a week or so. [19:38] <francogrex> pasteeater: ! that's probably it... something weird with my VOB file starts not at 0 but at 25 min! ok sorry guys [19:39] <francogrex> it's solved [20:00] <burek> relaxed, just tried it, works with default vcodec (when i dont specify anything), but doesn't work with h264.. :S [20:05] <burek> ffmpeg -f video4linux2 -s qvga -i /dev/video0 -f alsa -ar 44100 -i hw:0 -f flv - 2>/dev/null | tee local.flv | ffmpeg -f flv -i - -vcodec copy -acodec copy -f mpegts udp://burek:10001 [20:05] <burek> thats the command i used [20:05] <burek> vlc can play only audio [20:05] <burek> no video :/ [20:10] <newl> good to see you got rid of that iive clown [21:56] <Mista-D> how do I checkout "ffmpeg version N-34031-ge403a97" from git? `git checkout ge403a97` didn't work. [21:57] <newl_> git clone git://git.videolan.org/ffmpeg --depth 1 [21:57] <newl_> vat i useth [21:57] <newl_> why not get the latest? [21:58] <Mista-D> newl_: I need that specific version since it works in one particular instance for me. [21:58] <Mista-D> newl_: its a win32 bin that works. I want to build in in RHEL now. [21:59] <newl_> what is the problem ,,, issue? maybe it should be fixed? [21:59] <Mista-D> newl_: I get vertical lines (barely visible) in Linux and no lines in Windows. (MPEG2 -> x264) [22:00] <JEEB> Mista-D, the revision string after the number is a part of the sha-1 hash of the commit used [22:00] <JEEB> oh [22:00] <JEEB> that checkout should in theory work... [22:01] <JEEB> oh wait [22:01] <JEEB> git reset -hard ge403a97 [22:01] <JEEB> this should do it [22:01] <JEEB> *--hard [22:02] <Mista-D> JEEB: "fatal: ambiguous argument 'ge403a97': unknown revision or path not in the working tree." [22:04] <Mista-D> JEEB: can I use version number? N-34031? [22:04] <JEEB> no [22:04] <JEEB> what's your git --version btw? [22:04] <Mista-D> 1 hour ago [22:04] <Mista-D> oh sorry.. [22:05] <Mista-D> its 1.7.4 [22:05] <JEEB> hmm [22:06] <Mista-D> I'll bump it up to 1.7.7, just a sec. [22:06] <sacarasc> ge403a97 isn't the full thing, though... [22:07] <JEEB> well, I remember using short hashes with some repo tho... [22:07] <JEEB> is there an easy way to grep the full hash? [22:07] <JEEB> git log I guess [22:08] <JEEB> git log|grep "ge403a97" [22:08] Action: newl_ wonders how he knows he won't have the problem in the 'linux' version [22:10] <clockwize> is 30fps good enough for smooth video playback? [22:11] <Mista-D> newl_: I assume they operate similarly if all lib versions and configure are the same.. ? [22:12] <sacarasc> clockwize: PAL is 24fps. [22:12] <newl_> lots of parts/things going into compiling ffmpeg - are you doing ALL the parts from same source ? using mingw? [22:13] <sacarasc> Or was it 25... [22:13] <sacarasc> NTSC is only 30000/1001! [22:13] <Mista-D> I'll use same FFmpeg and x264 versions, don't need the rest. [22:13] <clockwize> sacarasc: 29? [22:13] <clockwize> ok, that's cool then.. [22:13] <sacarasc> 29.97.... [22:14] <sacarasc> Though that is rounded. [22:14] <pasteeater> newl_: i don't think a shallow repository with --depth 1 can checkout a previous hash...maybe. [22:14] <clockwize> indeed [22:14] <clockwize> . [22:14] <newl_> pasteeater: i just use if for all new source each time [22:15] <pasteeater> yes. it's fine for that. [22:15] <pasteeater> if i understand you correctly [22:15] <newl_> :) [22:18] <pasteeater> Mista-D: how can we duplicate the vertical line issue? it would be nice to see which revision causes this if it's a regression. [22:22] <Mista-D> pasteeater: adding all info to a doom10 post. just a sec. [22:28] <newl_> vertical line ? sounds more like something scrapping some of the bits off as it goes through the projector [22:46] <dericed> I'm trying to copy selected tracks from an input mxf to an output mxf. The input has 1 video, 8 audio, and 1 data track. I need the output to contain 1 video, 2 audio and 1 data track, so I'm using the -map option to specify the tracks and -c copy to copy them. However I get an error everything I try to copy the data stream, "Could not write header for output file #0 (incorrect codec parameters ?)" http://pastebin.com/czA2t53A [22:46] <dericed> Is it possible to copy data stream? I saw this commit, http://git.videolan.org/?p=ffmpeg.git;a=commit;h=e3b540b42464395eadbc42b873…, but perhaps it doesn't help me. [22:49] <Mista-D> pasteeater: http://doom10.org/index.php?topic=2034.0 -- vertical lines noise. [23:18] <burek> dericed1, use -dcopy [23:18] <burek> wait [23:19] <burek> dericed1, your -map is wrong [23:20] <burek> instead of 0:0 use 0.0 [23:21] <burek> crap.. its the same [23:21] <burek> 0:0 is the same as 0.0 [23:37] <newl_> o^O [00:00] --- Wed Nov 30 2011

1 0

ffmpeg-devel.log.20111128
by burek 29 Nov '11

29 Nov '11

[00:41] <CIA-41> ffmpeg: 03Vitor Sessak 07master * rca55606a51 10ffmpeg/libavutil/x86/x86inc.asm: [00:41] <CIA-41> ffmpeg: x86inc: Flag shufps as an floating-point instruction for the AVX emulation code. [00:41] <CIA-41> ffmpeg: Without this, code like "shufps m0, m1, m2, 0xaa" would not work in CPUs [00:41] <CIA-41> ffmpeg: not supporting SSE2. [00:41] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:14] <CIA-41> ffmpeg: 03Kostya Shishkov 07master * r498605b4ad 10ffmpeg/libavcodec/vc1dec.c: [01:14] <CIA-41> ffmpeg: vc1: select interlaced scan table by FCM element [01:14] <CIA-41> ffmpeg: Interlaced videos can contain progressive frames too and now wrong scantable [01:14] <CIA-41> ffmpeg: is selected for them. [01:14] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:14] <CIA-41> ffmpeg: 03Victor Vasiliev 07master * r12bc20502a 10ffmpeg/libavformat/ (Makefile avi.c avi.h avidec.c avienc.c riff.c riff.h wav.c): [01:14] <CIA-41> ffmpeg: Generalize RIFF INFO tag support; support reading INFO tag in wav [01:14] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:14] <CIA-41> ffmpeg: 03Alex Converse 07master * rf11b0e9543 10ffmpeg/libavcodec/wmavoice.c: wmavoice: Make format string match variable type. [01:14] <CIA-41> ffmpeg: 03Mans Rullgard 07master * rd9ba767d61 10ffmpeg/libavformat/mpc.c: [01:14] <CIA-41> ffmpeg: musepack: fix signed shift overflow in mpc_read_packet() [01:14] <CIA-41> ffmpeg: Using an unsigned variable avoids problems with overflows. [01:14] <CIA-41> ffmpeg: There is further no need for a 64-bit intermediate here. [01:14] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [01:14] <CIA-41> ffmpeg: 03Aneesh Dogra 07master * r97980db487 10ffmpeg/libavcodec/indeo3.c: [01:14] <CIA-41> ffmpeg: indeo3: error out if no motion vector is set. [01:14] <CIA-41> ffmpeg: This fixes a crash on a corrupt bitstream (bugzilla #93). [01:14] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:14] <CIA-41> ffmpeg: 03Vitor Sessak 07master * r6b6ee58249 10ffmpeg/libavutil/x86/x86inc.asm: [01:14] <CIA-41> ffmpeg: x86inc: Flag shufps as an floating-point instruction for the AVX emulation code. [01:14] <CIA-41> ffmpeg: Without this, code like "shufps m0, m1, m2, 0xaa" would not work in CPUs [01:14] <CIA-41> ffmpeg: not supporting SSE2. [01:14] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:14] <CIA-41> ffmpeg: 03Justin Ruggles 07master * r0df5e869cb 10ffmpeg/libavcodec/ (mpegaudiodec.c mpegaudiodec_float.c): [01:15] <CIA-41> ffmpeg: * qatar/master: [01:15] <CIA-41> ffmpeg: vc1: use an enum for Frame Coding Mode [01:15] <CIA-41> ffmpeg: doc: cleanup filter section [01:15] <CIA-41> ffmpeg: indeo3: error out if no motion vector is set. [01:15] <CIA-41> ffmpeg: x86inc: Flag shufps as an floating-point instruction for the AVX emulation code. [01:15] <CIA-41> ffmpeg: 03Alex Converse 07master * r028a2375e2 10ffmpeg/libavformat/mov.c: mov: Make format string match variable type. [01:15] <CIA-41> ffmpeg: 03Cheng Sun 07master * r3f5aa7dfa6 10ffmpeg/libavcodec/pthread.c: [01:15] <CIA-41> ffmpeg: pthread: track thread existence in a separate variable. [01:15] <CIA-41> ffmpeg: This fixes a compile error on mingw32 when using p->thread [01:15] <CIA-41> ffmpeg: directly (as if it were a pointer) to track thread existence, [01:15] <CIA-41> ffmpeg: because the type is opaque and may be a non-pointer. [01:16] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:16] <CIA-41> ffmpeg: 03Luca Barbato 07master * r9270b8a3d1 10ffmpeg/doc/filters.texi: [01:16] <CIA-41> ffmpeg: doc: cleanup filter section [01:16] <CIA-41> ffmpeg: Use the @command{} tag when needed and cleanup the examples. [03:44] <shifter1> anyone alive? [03:45] <kcm1700> everyone is alive. [03:45] <shifter1> that's good to hear [03:45] <shifter1> I am having problems with ffserver, I think I know where they are coming from [03:46] <shifter1> am I in the right place? [03:52] <Compn> well bug reports should be reported in ffmpeg-user list or the ffmpeg bug tracker , unless you have a patch, then it can be discussed on ffmpeg-devel list or here [04:01] <shifter1> Compn: I don't have a .diff file but I believe libavformat/sdp.c:405 just below case CODEC_ID_AAC should be 'if (fmt && fmt->oformat && fmt->oformat->priv_class' [04:02] <shifter1> this is in function sdp_write_media_attributes [04:02] <Compn> well thats close enough :) [04:02] <Compn> yeah, if you stick around, someone should pick up on it [04:03] <shifter1> ffserver segfaults because fmt->oformat is NULL, and it tries to access fmt->oformat->priv_class [04:03] <Compn> michaelni is always good at reviewing code [04:04] Action: Compn goes afk [04:10] <michaelni> shifter1, does it work with your suggested change ? [04:10] <shifter1> yes, the audio works, but IM recompiling now since I had it running in -g3 and -O0 [04:11] <shifter1> it was pretty laggy, but it didn't crash [04:14] <shifter1> michaelni: audio works, video doesn't seem to, but that might be a problem with my ffserver config [04:14] <shifter1> but it no longer crashes [04:18] <michaelni> shifter1, the change looks harmless, ill apply it, what should i use as author ? [04:18] Action: shifter1 blushes [04:18] <shifter1> uh, cmastrange3(a)gatech.edu ? [04:19] <michaelni> ok [04:19] <shifter1> carl if you need a first name [04:19] <shifter1> "Carl" [04:19] <shifter1> :P [04:19] <shifter1> I've never been asked [04:20] <michaelni> iam always trying to credit the author of a change, though very rarely i forget ... [04:20] <shifter1> Well thanks, glad to have helped. [04:21] <Compn> shifter1 : be proud, you will soon have code in ffmpeg :P [04:30] <CIA-41> ffmpeg: 03Carl 07master * r2cf4bd7751 10ffmpeg/libavformat/sdp.c: [04:30] <CIA-41> ffmpeg: sdp: Fix null pointer dereference with aac and ffserver. [04:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rd3fc335bac 10ffmpeg/libavcodec/huffyuv.c: [04:30] <CIA-41> ffmpeg: huffyuvenc: support alphaless rgb32 [04:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rb14a2381f8 10ffmpeg/libavutil/ (pixdesc.c pixfmt.h): [04:30] <CIA-41> ffmpeg: pixfmt: Add 32bit rgb without alpha formats [04:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r0af7d7082f 10ffmpeg/libavcodec/huffyuv.c: [04:30] <CIA-41> ffmpeg: huffyuvenc: store alpha for bgr32 [04:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rb89ce2d6af 10ffmpeg/ (6 files in 2 dirs): [04:30] <CIA-41> ffmpeg: swscale: alpha less rgb32 support. [09:35] <Tjoppen> bcoudurier: http://code.google.com/p/ffmbc/issues/detail?id=49 look like decent opatom samples, and seem to work with the patches. thoughts? [09:51] <Tjoppen> audio ones work if I add their essence container ul. for some reason it thinks there's two audio streams though [09:53] <bcoudurier> ah yes, I should integrate this [09:54] <Tjoppen> video seems to demux without a problem though [09:54] <Tjoppen> going to investigate why two streams are created for the audio atoms [09:57] <Tjoppen> only A1 triggers it [09:59] <Tjoppen> ah, it's using the same sourpackageid for A1 and V1 [10:39] <Tjoppen> there. fixable by checking that data_definition_ul is consistent between packages [10:40] <bcoudurier> nice [10:41] <bcoudurier> Im going to bed, have a good day :) [10:41] <Tjoppen> ok. I'll post on the ml [10:42] <Tjoppen> ah, he left [14:06] <CIA-41> ffmpeg: 03Clément BSsch 07master * rb1ca5634fd 10ffmpeg/libavcodec/ (avcodec.h mpeg12.c mpeg12enc.c options.c version.h): mpeg12: raise timecode to codec context. [14:06] <CIA-41> ffmpeg: 03Clément BSsch 07master * rfbe6e29646 10ffmpeg/ffprobe.c: ffprobe: print codec timecode if available. [15:15] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rae5873f13b 10ffmpeg/libavcodec/huffyuv.c: [15:15] <CIA-41> ffmpeg: huffyuvenc: switch from alphaless rgb32 to rgb24 [15:15] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:23] <kcm1700> is it worth optimizing code around libavcodec/mpeg4videodec.c:468 ? [15:24] <kcm1700> it's get_amv function [15:34] <ubitux> kcm1700: according to the comment, yes :p [15:35] <ubitux> i guess a LUT could be used [15:38] <kcm1700> that'd be nice. thanks [16:53] <iive> i don't see how it could be done with look up table. [16:53] <iive> the values are quite big. [18:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4b03d96022 10ffmpeg/libavcodec/ffv1.c: [18:10] <CIA-41> ffmpeg: ffv1dec: use PIX_FMT_0RGB32 when there is no transparency plane [18:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rf7b160e829 10ffmpeg/libavcodec/ffv1.c: [18:10] <CIA-41> ffmpeg: ffv1enc: Store transparency plane. [18:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:11] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r2027d073ae 10ffmpeg/libavcodec/ffv1.c: [18:11] <CIA-41> ffmpeg: ffv1dec: transparency plane support. [18:11] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:11] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r492aab8589 10ffmpeg/libavcodec/ffv1.c: [18:11] <CIA-41> ffmpeg: ffv1enc: PIX_FMT_0RGB32 support [18:11] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:22] <michaelni> kcm1700, the naive way to optimize the 16x16 summing is to do it recursive [18:22] <michaelni> that is when 2 v>>shift values match than all points between them must match too [18:23] <michaelni> so if you test the 4 corners and they match then you are done already just take the value *256 [18:24] <michaelni> if they mismatch smaller squares could be used [18:24] <michaelni> its sad that mpeg didnt just use the analytical value of the integral, that would have been nicer [18:25] <michaelni> but maybe theres a trick iam missing by which it can be calculated in a more pretty way ... [18:32] <michaelni> and note, when trying the recursice thing there is surely a point where its faster to stop spliting [18:33] <michaelni> another obvious way would be instead of doing things horizontal & vertical rather along lines that are close to constant along the plane [20:03] <burek> should I send these spell-checked documents from Google Code In to some mailing list :) [20:03] <burek> or should we just ignore it or what? :) [20:58] <michaelni> burek, general.html ? [20:59] <michaelni> its generated from doc/general.texi, we need a patch against that [21:01] <michaelni> if you diff the file the student will give you against the html you should see the changes [21:01] <michaelni> dunno which is the easiest way to integrate in the texi [21:02] <michaelni> if its not much then doing it by hand would be just a few minutes work [21:02] <michaelni> the student should have worked on the texi :) ... i think he didnt but actually i didnt look at all [21:02] <michaelni> i just looked at the list of tasks and saw general.html spellcheck claimed [21:04] <burek> general, faq, ffmpeg, ffprobe, ffplay.. [21:04] <burek> its not much [21:04] <burek> its just a spell check :) [21:05] <burek> i can do that, just dont know how :) [21:05] <michaelni> you dont know texi ? [21:05] <michaelni> you just open it in a text editor of your choice [21:05] <michaelni> it should be obvious [21:05] <burek> ok :) [21:05] <burek> ill figure it out :) [21:06] <burek> is there any link to an original document or is it in the git? [21:06] <michaelni> git [21:06] <burek> ok, ill find it :) [21:07] <michaelni> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/general.texi [21:08] <burek> thanks :) [21:09] <burek> i just gave them tasks as easier as i could [21:09] <burek> so that complete it, without a lot of trouble [21:09] <burek> anyway, there are not much changes, so it will be ok :) [21:11] <michaelni> i would have been surprised if there was alot that was misspelled in there [21:12] <michaelni> though someone might probably reword alot in our docs [21:12] <michaelni> some of it could be worded more clearly [21:14] <gnafu> michaelni: http://4.bp.blogspot.com/_D_Z-D2tzi14/S8TRIo4br3I/AAAAAAAACv4/Zh7_GcMlRKo/s… ? [21:14] <gnafu> ;D [21:15] <michaelni> :) [21:18] <pasteeater> burek: a generalized method to make a simple patch: http://pastebin.com/jJbUh1tb [21:19] <burek> thanks pasteeater :) [21:21] <pasteeater> there is also: http://git.videolan.org/?p=ffmpeg.git;a=blob_plain;f=doc/git-howto.txt;hb=H… [21:21] <pasteeater> ...but it's quite verbose. [21:32] <burek> thanks again :) [23:33] <Compn> wait a minute [23:33] <Compn> when did they start selling .xxx domains ?? [23:33] <gnafu> Compn: A while ago (by Internet standards). Where have you been? ;-) [23:33] <ohsix> quite a while ago, they only started running ads on tv for them about a week ago though :> and they are pretty funny with the guy in the suit [23:34] <Compn> damn, i'm under a rock [23:35] <Compn> this month tho, so thats good [23:35] <Compn> not like an entire year or anything [23:57] <Anssi> michaelni: a bit of a late response, but no, don't really have any ideas... [00:00] --- Tue Nov 29 2011

1 0

ffmpeg.log.20111128
by burek 29 Nov '11

29 Nov '11

[00:02] <praedo_> hello [00:02] <praedo_> i need help desperately [00:03] Action: sacarasc hands praedo_ a gun. [00:03] <praedo_> i have an important recording that was interrupted and it seems corrupt [00:03] <darkstarbyte> sorry [00:03] <praedo_> when i open it with ffmpeg it says moov atom not found [00:03] <darkstarbyte> pasteeater, sorry [00:03] <praedo_> how could i try to fix it? [00:03] <sacarasc> praedo_: I know of no way to fix MP4s without moov atoms. [00:04] <praedo_> really? [00:04] <praedo_> and why does that happen? [00:04] <praedo_> in fact it's a m4v file [00:04] <praedo_> but it's h264 anyway [00:04] <sacarasc> Because you didn't finish downloading the file. [00:04] <praedo_> it's a live recording of an event [00:05] <praedo_> power went off and that's what it got recorded [00:05] <praedo_> 1.5gb of data [00:05] <praedo_> so it's not a broken download exactly [00:05] <JEEBsv> unfortunately the container in question needs an index [00:05] <JEEBsv> although... [00:05] <JEEBsv> live? [00:05] <praedo_> is the index writeen only when it ended? [00:05] <praedo_> yes, live stream recorded on disk [00:06] <JEEBsv> I think there was a feature called fragments in the format [00:06] <JEEBsv> but... [00:07] <praedo_> what is that? [00:07] <JEEBsv> can it be played back by mplayer2's current trunk for example? [00:07] <JEEBsv> praedo_: index-like points in the middle of the file to enable streaming [00:08] <JEEBsv> I'm not sure but I think that libavformat should support that feature as well, so it sounds weird if it doesn't work at all [00:08] <JEEBsv> (since mplayer(2)'s support comes from libavformat) [00:09] <JEEBsv> you have hope if it has fragments and can be read, you have to resort to quite manual labor or deem it impossible if it doesn't [00:10] <JEEBsv> since yes, indexes are usually put into either the beginning or the end of the stream/file [00:10] <JEEBsv> if you could play it, there had to be some kind of an index given out, though [00:10] <JEEBsv> if it was just data downloaded and not playable straight out when you start reading it, that'd mean that the index is in the end [00:11] <praedo_> the command i run is: ffmpeg -i file.m4v -acodec copy -vcodec copy out.mp4 [00:11] <praedo_> and it complains about the moov atom [00:11] <praedo_> why suggestion with a different command? [00:12] <JEEBsv> how new is your ffmpeg? [00:12] <praedo_> why=any [00:12] <praedo_> it's ffmpeg-git [00:12] <praedo_> from 1 month ago [00:12] <JEEBsv> how old? [00:12] <JEEBsv> ok [00:12] <JEEBsv> try current trunk [00:13] <praedo_> anything changed? [00:14] <JEEBsv> not sure, but worth a shot, no? :V [00:14] Action: sacarasc removes i from ive. [00:14] <praedo_> i try now [00:15] <iive> praedo_: have you checked if there are big areas full of zeroes in your file? [00:15] <JEEBsv> also, I guess l-smash's boxdumper could give insight into the file [00:15] <praedo_> it's 1 hour of soccer at 2500 kbps [00:15] <praedo_> i tried on windows media player and it can't be played at all [00:16] <JEEBsv> it can output the data of the composition as well as values of the boxes in the thing [00:16] <praedo_> what is boxdumper? [00:16] <JEEBsv> an app in l-smash [00:16] <JEEBsv> http://code.google.com/p/l-smash/ [00:17] <JEEBsv> I use it to see differences in mp4/mov/related files' structure [00:17] <praedo_> sounds hard to handle [00:17] <JEEBsv> not really [00:18] <JEEBsv> boxdumper filename [00:18] <JEEBsv> outputs the info into stdout [00:18] <JEEBsv> :3 [00:18] <iive> praedo_: i highly recommend you to check if your file is not full of zeroes, before going with further attempts to recover its content. [00:18] <JEEBsv> IIRC [00:18] <JEEBsv> and yeah [00:18] <JEEBsv> I agree [00:19] <iive> jurnal filesystem should guarantee the data consistency of everything before the last flish/close . And I'm not quite sure how often recoding programs call these. [00:20] <iive> (well, close is called at the end of the recoding) [00:20] <JEEBsv> so in the worst case scenario you have a big lump o' zeroes [00:20] <JEEBsv> I guess [00:21] <iive> it recognizes it as mov/mp4/qt file, so there is something written in it. [00:21] <iive> you should check if there is more than just header. [00:22] <praedo_> using the last version of ffmpeg: [00:22] <praedo_> ç[mov,mp4,m4a,3gp,3g2,mj2 @ 0x8836a80] moov atom not found [00:22] <praedo_> /floppy/UEF_MONTCADA_0.m4v: Operation not permitted [00:23] <praedo_> the recording program was runjning on windows 7 with ntfs partition [00:23] <praedo_> how can i check the zeroes you said? [00:23] <praedo_> i have the file on linux now [00:23] <iive> hex editor? [00:24] <praedo_> won't it take forever to open a 1.5gb file? [00:25] <praedo_> hexedit UEF_MONTCADA_0.m4v [00:25] <JEEBsv> only if it tries to read it whole at once [00:25] <praedo_> that was fast :) [00:25] <praedo_> not full of zeroes [00:25] <praedo_> it shows data [00:26] <iive> praedo_: check the end of the file too [00:26] <praedo_> true [00:26] <praedo_> > [00:26] <praedo_> zeroes at the end [00:27] <iive> check the middle [00:27] <iive> then 1/4 1/8 etc.. [00:30] <praedo_> data until the midle more or less [00:36] <praedo_> any idea, iive? [00:36] <praedo_> information is there [00:36] <iive> praedo_: well, you are getting at least half your data... try everything you can :) [00:37] <iive> it's probably just few MB at the end that are lost. [00:37] <iive> you can try at 3/4 5/8 or 7/8 if you want to know how much exactly. [00:38] <iive> oh, you can also try mplayer -demuxer mov file.mov in case it ignores this atom. [00:38] <praedo_> yes, just a few pages of zeroes at the end [00:39] <praedo_> why .mov? [00:40] <iive> mp4 is subset of mov/qt [00:40] <JEEBsv> well, isn't it both a subset and a superset? [00:40] <iive> that's why same demuxer is used... your file was .m4v right? [00:41] <iive> JEEBsv: kind of :) [00:41] <JEEBsv> I'd think that MPEG spec'd BIFS for example [00:41] <praedo_> mencoder /floppy/UEF_MONTCADA_0.m4v -ovc copy -oac copy -o ue_figueres.mp4 [00:42] <JEEBsv> > mencoder outputting mp4 [00:42] <iive> praedo_: just try the mplayer... [00:42] <JEEBsv> unless mencoder really got better [00:42] <praedo_> WARNING: OUTPUT FILE FORMAT IS _AVI_. See -of help. [00:42] <praedo_> success: format: 0 data: 0x0 - 0x49fe0030 [00:42] <praedo_> libavformat file format detected. [00:42] <praedo_> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8a7b660]moov atom not found [00:42] <JEEBsv> that'll be quite the <beep> [00:42] <iive> nah, you need at least -of lavf [00:42] <iive> and i don't see the "-demuxer mov" [00:43] <iive> without it, it would use libavformat from ffmpeg :E [00:43] <JEEBsv> oh, mencoder can now write stuff with lavf? [00:43] <JEEBsv> cool [00:43] <sacarasc> It has been able to for years. [00:43] <JEEBsv> fun [00:43] <praedo_> mplayer /floppy/UEF_MONTCADA_0.m4v -demuxer mov uef.mov [00:43] <praedo_> is this command correct? [00:43] <sacarasc> No. [00:44] <praedo_> File not found: 'uef.mov' [00:44] <praedo_> Failed to open uef.mov. [00:44] <JEEBsv> well doh [00:44] <JEEBsv> leave that out [00:44] <JEEBsv> it should try to play it [00:45] <praedo_> doesn't play [00:45] <JEEBsv> ok [00:45] <praedo_> what did you mean mencoder outputting? [00:45] <JEEBsv> the internal muxers of mencoder are <beep> [00:45] <JEEBsv> for everything else but avi [00:45] <JEEBsv> anyways, grab l-smash and build it [00:45] <JEEBsv> pastebin the output of boxdumper [00:45] <iive> in the above command, it would write .avi file with .mp4 extension... [00:46] <praedo_> iive, for some reason it doesn't work [00:46] <praedo_> File not found: 'uef.mov' [00:46] <praedo_> that's suppposed to be the output, no? [00:47] <JEEBsv> someone is quite out on the moon it seems [00:47] <iive> praedo_: sorry I confused you, uef.mov is just the name of your file. [00:47] <praedo_> could this be useful maybe to fix it? http://renaun.com/blog/code/qtindexswapper/ [00:47] <iive> mplayer can play many files one after the other, so when you give it 2 files, it plays the first and then the second. [00:47] <JEEBsv> praedo_: uhhh [00:48] <JEEBsv> considering that ffmpeg et al can't find the index :P [00:48] <JEEBsv> also there is an app to do that same thing for the index in ffmpeg's source tree as well [00:48] <iive> i'm not sure it is the index that is missing. [00:48] <iive> i'm not familiar with mp4 atoms. [00:49] <JEEBsv> well, boxdumper should tell us [00:49] <praedo_> what app? [00:49] <JEEBsv> ... [00:49] <praedo_> JEEBsv, i'll install that boxdumper and try then... what is l-smash exactly? [00:49] <iive> boxdumper from l-smash packet. [00:49] <JEEBsv> you don't need to install it [00:49] <JEEBsv> just compile [00:50] <JEEBsv> praedo_: a muxing/parsing library etc. for mp4/mov/similar [00:50] <iive> isn't it easier to try with apt-get install ? [00:50] <pasteeater> probably not in repo [00:50] <JEEBsv> yeah [00:50] Action: pasteeater makes l-smash AUR build script [00:51] <JEEBsv> also, praedo_ -- I think the site says it rather well [00:51] <JEEBsv> Loyal to Spec of Mpeg4 and Ad-hoc Simple Hackwork. Yet another opensource mp4 handler" [00:51] <iive> debian is starting to slack more than slackware... [00:51] <JEEBsv> Also the L-SMASH guys make fixes and other stuff for libavformat as well [00:51] <JEEBsv> :3 [00:52] <pasteeater> JEEBsv: did you know your name is in AUR? [00:52] <pasteeater> https://aur.archlinux.org/packages.php?ID=51606 [00:52] <praedo_> i use archlinux [00:52] <praedo_> i'll see if it's in aur [00:52] <pasteeater> it is [00:52] <pasteeater> now [00:52] <JEEBsv> pasteeater: lawl [00:53] <praedo_> pasteeater, awesome :) [00:53] <praedo_> JEEBsv, thanks for the great recommendation of boxdumper... i hope it will be able to fix it [00:53] <JEEBsv> it won't be able to fix it, but it should give you more hints [00:53] <JEEBsv> unless it crashes on the file hard [00:54] <JEEBsv> in which case bug reports are welcome :P [00:54] <praedo_> what kind of hints? [00:55] <JEEBsv> like a map of the file? [00:55] <JEEBsv> 01:45 < JEEBsv> anyways, grab l-smash and build it [00:55] <JEEBsv> 01:45 < JEEBsv> pastebin the output of boxdumper [00:55] <JEEBsv> :V [00:56] <praedo_> installing l-smash from aur now... [00:56] <praedo_> i'll need you to teach me using it [00:56] <JEEBsv> I think boxdumper --help or just calling boxdumper should tell you if you actually need to add any settings to it [00:58] <praedo_> compiling... [00:59] <JEEBsv> heh, nice -- chikuzen actually made a configure for it [01:00] <JEEBsv> with various options [01:00] <praedo_> installed [01:00] <praedo_> which one should i pastebin of these? [01:00] <praedo_> --box Dump box structure [01:00] <praedo_> --chapter Extract chapter list [01:00] <praedo_> --timestamp Dump media timestamps [01:01] <JEEBsv> --box [01:01] <praedo_> and I redirect output with > file ? [01:01] <JEEBsv> yeah, if you don't want it all on your face [01:01] <JEEBsv> s/face/terminal/ [01:02] <praedo_> http://pastebin.com/gCu9ZzAF [01:02] <praedo_> here [01:02] <praedo_> i don't understand a thing :) [01:03] <JEEBsv> :/ [01:03] <praedo_> that means no luck? [01:04] <JEEBsv> you can also try the remuxer in L-SMASH but looking at that output I'm not sure you'll be getting something out of it [01:04] <JEEBsv> (yes, it has an app called 'remuxer' in it) [01:04] <praedo_> what info gives that pastebin? [01:05] <praedo_> how much valid data is there? [01:05] <JEEBsv> you have an mdat box at zero, then a 'wide' box at 32, and then... mdat but with a lulzy size? [01:05] <praedo_> # remuxer -i /floppy/UEF_MONTCADA_0.m4v -o uef.mp4 [01:05] <praedo_> Error: failed to set iTunes metadata. [01:05] <praedo_> Error: failed to set up preparation for output. [01:06] <JEEBsv> yeah, couldn't parse it [01:06] <JEEBsv> at least it didn't segfault [01:06] <praedo_> size is correct because it's what it usually takes to record an hour of live stream at 2500 kbps [01:07] <JEEBsv> I guess 1694498855 is the correct size? [01:07] <JEEBsv> or something around that [01:07] <praedo_> yes [01:07] <praedo_> 1.5gb [01:07] <praedo_> any other test i could do? [01:07] <JEEBsv> anyways, the fact that that thing at 40 has a humongous size kind of welps it [01:08] <praedo_> at 40? [01:08] <JEEBsv> yes, the mdat at 40 [01:08] <JEEBsv> look at boxdumper's output :P [01:09] <praedo_> ah, yes [01:09] <praedo_> that must be the last bit [01:10] <praedo_> could i try with other tools? [01:11] <JEEBsv> feel free, but I don't think you'll get much out of it [01:13] <praedo_> so it's definately an unrecoverable corrupt file :( [01:13] <praedo_> no moov atom = dead [01:13] <praedo_> as a rule [01:14] <JEEBsv> pretty much [01:14] <JEEBsv> unless it uses fragments [01:15] <JEEBsv> in which case it should be playback'able with a relatively new mplayer or whatever [01:15] <JEEBsv> also, I'm not sure if ffmpeg would throw up errors on such a file either [01:17] <praedo_> and no fragments are on this file? [01:18] <JEEBsv> I would guess that boxdumper should find them in that case [01:18] <JEEBsv> and that ffmpeg wouldn't derp on the file either :P [01:19] <praedo_> derp? [01:27] <pasteeater> as in 'fail' or similar. short for 'herp derp' and variants. [01:28] <cryptopsy> toodles ~~ [01:28] <praedo_> i see [01:28] <praedo_> so no fragments [01:29] <praedo_> i heard that a media server could playback the file anyway as VOD... could this be true? [03:34] <shifter1> I am getting a segfault using ffserver [03:34] <shifter1> I have some debug output [03:34] <shifter1> where should I bring this info? [09:05] <Boon> hello, is vhook unsupported with latest ffmpeg? watermark feature [09:07] <Tjoppen> vhook has been removed for quite some time. there's libavfilter now [09:08] <Boon> so you suggest using libavfilter to do watermark? [09:09] <ubitux> yes, look at the overlay filter [09:10] <Boon> ok thank you [09:10] <ubitux> http://ffmpeg.org/libavfilter.html#overlay [09:13] <Boon> any example working command to test out? [09:14] <Boon> i found this on google [09:14] <Boon> ffmpeg -i 01.mpg -vf "movie=watermark.png [logo]; [in][logo] overlay=10:main_h-overlay_h-10 [out]" -f flv 03.flv [09:15] <Boon> what is [logo] [in] and [out] [09:15] <Boon> ? [09:15] <ubitux> it is the link names [09:16] <ubitux> http://ffmpeg.org/libavfilter.html#Tutorial [09:20] <dirtycookie> hello people. i have a ubuntu os runnung and iam using your great software along with the tool youtube-dl, and when i want only to extract the audio if a youtube clip everything starts fine, but when it comes to the conversion of the file I get the follwing " WARNING: error running ffmpeg" can someone poing me to the right direction to fix this? [09:24] <andrew_46> dirtycookie: You are using youtube-dl's --extract-audio option? [09:26] <dirtycookie> andrew_46: yes [09:28] <andrew_46> dirtycookie: I should mention here that I normally ask for help here, not give it :). Can you give the commandline + youtube url you are using? [09:31] <dirtycookie> andrew_46: one sec [09:31] <dirtycookie> youtube-dl --extract-audio --audio-format=mp3 [url of youtube] [09:32] <andrew_46> dirtycookie: Sorry I meant give an exact youtube url so I can test one that has failed for you [09:36] <andrew_46> dirtycookie: And if you are using Ubuntu you may be aware that the repository FFmpeg normally has mp3 encoding stripped from it? [09:50] <naxa> how would I find parameters for a video encoder? [09:50] <naxa> (i mean available parameters) [09:52] <naxa> for example there is the 'x264opts' parameter. are there any parameters for the other encoders? [09:53] <naxa> the docs only mentions vpx as far as i can see [09:53] <naxa> *mentions only [10:10] <andrew_46> dirtycookie: and the youtube-dl script calls libmp3lame for mp3 conversion [10:54] <dirtycookie> andrew_46: true but, the package is already installed [12:03] <burek> if we specify something like this: -f alsa -ar 44100 -ac 1 -i hw:0,0 [12:04] <burek> does it mean ffmpeg will try to read the input @44100 Hz Mono? [12:05] <burek> dirtycookie, you can try with ffmpeg -codecs | grep lame [12:05] <burek> naxa, each encoder can have its own specific params [12:05] <naxa> burek: it's okay but how do I input them? [12:06] <burek> your best chance is to google something like: ffmpeg vcodec <encoder> [12:06] <burek> or to go to the official web page of that encoder [12:06] <burek> and read the docs [12:06] <naxa> burek: yes. it's just there is a parameter named "x264opts" and I don't know if there is one like "mjpegopts" I doubt it [12:07] <burek> i see [12:08] <burek> for mjpeg, there is -qscale [12:09] <burek> -qscale 1 means the best quality [12:12] <dirtycookie> burek: well, i got a lot of output [12:12] <dirtycookie> burek: nothing specific [12:12] <burek> shifter1: https://ffmpeg.org/trac/ffmpeg [12:13] <burek> dirtycookie, that command should give you a clue if your ffmpeg was compiled with libmp3lame support or not [12:13] <burek> regardless if you have the library installed [12:14] <dirtycookie> burek: this suggests that i dont have it with lame compiled? [12:15] <burek> exactly [12:15] <burek> that's why you should always compile your ffmpeg yourself [12:16] <burek> it takes no more than 15 minutes [12:16] <burek> but you get what you need [12:16] <burek> and of course you get the latest (and greatest) version, with all the bugs fixed (lol) :) [12:19] <dirtycookie> burek: so could you help me here guiding me through this? [12:20] <dirtycookie> brb [12:22] <burek> sure [12:22] <burek> http://ffmpeg.org/download.html [12:22] <burek> first of all, what is your overall goal [12:22] <burek> what do you want to achieve with ffmpeg [12:42] <dirtycookie> burek: i just use ffmpeg to convert videos into different formats and resolutions. I also want to use it in oder extract audio from youtubeclips [12:42] <dirtycookie> in mp3 form [12:43] <burek> i see, so you'll need libmp3lame [12:44] <burek> apt-get install libmp3lame-dev [12:44] <burek> if you will just extract audio from youtube and convert it to mp3, you can also compile (or just install) libx264 [12:45] <burek> i suggest compile, because its less than 5min [12:45] <burek> also, I would recommend you to compile libaacplus, and use AAC+ audio encoder, rather than mp3 [12:45] <burek> it's much better quality [12:46] <dirtycookie> burek: ill do your recommendations [12:46] <dirtycookie> i installed libmp3lame-dev [12:46] <dirtycookie> what nex [12:46] <dirtycookie> t [12:46] <burek> http://www.videolan.org/developers/x264.html [12:46] <burek> aac+ http://tipok.org.ua/node/17 [12:46] <burek> (use libaacplus-2.0.2.tar.gz) [12:47] <burek> http://ffmpeg.org/download.html [12:47] <burek> and that's basically it [12:47] <burek> when you configure libx264, you could use ./configure --enable-shared --enable-static [12:47] <burek> you never know if you are going to compile vlc or something and if you'll need shared/static version [12:48] <burek> for aac+, just ./compile && make && make install [12:48] <burek> for ffmpeg, ./configure --enable-aacplus --enable-non-free --enable-libx264 --enable-libmp3lame [12:49] <burek> or this ./configure --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-postproc --enable-libx264 --enable-libaacplus --enable-libmp3lame [12:49] <dirtycookie> burek: 1 sec on the phone [12:49] <burek> ok [12:49] <vivienschilis> question [12:50] <vivienschilis> does webm supports CRF? [12:50] <burek> that's x264's option [12:50] <vivienschilis> ok just wondering cause ffmpeg accepts to pass a crf option even for an webm format [12:51] <burek> http://ffmpeg.org/ffmpeg.html [12:51] <burek> search for crf [12:51] <burek> I see it only in libvpx [12:52] <burek> http://www.webmproject.org/tools/encoder-parameters/#10_sample_command_lines [12:53] <relaxed> there are some libvpx presets in ffmpeg/ffpresets [12:55] <dirtycookie> burek: i entered "./configure --enable-shared --enable-static", do i enter the others right after it or how? [12:55] <vivienschilis> what's the different between the vpx and vp8 codec? [12:55] <vivienschilis> one is ffmepg's [12:55] <vivienschilis> ? [12:55] <burek> dirtycookie, what are you configuring right now [12:56] <vivienschilis> encoding api. [12:56] <vivienschilis> just wondering if i should expect cry to be just h264 specific [12:56] <vivienschilis> crf* [12:58] <dirtycookie> burek: i just entered "./configure --enable-shared --enable-static" i didnt hit enter yet [13:17] <burek> dirtycookie ok [13:17] <dirtycookie> burek: what about the other enables? [13:18] <burek> dirtycookie, what are you configuring right now [13:20] <dirtycookie> burek: --enable-shared --enable-static [13:20] <dirtycookie> burek: that is all [13:20] <burek> yes but what: aac, x264, ffmpeg...? [13:20] <dirtycookie> burek: no idea really [13:21] <burek> ? [13:21] <burek> you should stick to repos then [13:21] <burek> :) [13:21] <dirtycookie> burek: well, kinda late i sent it to make and make install [13:22] <burek> ok :) [13:22] <dirtycookie> burek: lets see what comes out [13:22] <dirtycookie> ;) [13:22] <burek> if you don't know what you are doing, the results can only be good, right :d [13:22] <dirtycookie> LOL, make was always 4 me voodoo magic [13:23] <burek> well, for libaacplus only ./configure && make && make install will be enough [13:24] <burek> x264 ./configure --enable-shared --enable-static [13:24] <burek> and ffmpeg, read above :) [13:25] <dirtycookie> burek: i enter "./configure --enable-aacplus --enable-non-free --enable-libx264 --enable-libmp3lame" then make && make install [13:27] <burek> http://pastebin.com/P2S3SLHn [13:28] <dirtycookie> burek: ok ill do that thx [13:29] <burek> http://pastebin.com/LhRkXhzZ [13:29] <vivienschilis> is ffmpeg's aac enc is better than libfaac now? [13:31] <burek> http://pastebin.com/7ngnxisS [13:31] <burek> there it is :) [13:31] <burek> vivienschilis, honestly, both are kinda the same (low quality) comparing to aac+ [13:34] <vivienschilis> fine ;) [13:46] <JEEB> note tho that HE-AAC shouldn't be used if you're trying to actually use semi-high bitrates (>64kbps) [13:46] <JEEB> HE-AAC(v2) takes out a lot of information and tries to replicate it with other kind of stuff [13:46] <JEEB> which makes it quite useful for <64kbps use cases [13:47] <JEEB> (this is stereo-wise of course) [13:47] <burek> yes, and this says it all: "Data from this testing also indicated that some individuals confused 48 kbit/s encoded material with an uncompressed original." :) [13:47] <JEEB> point being "some individuals" [13:47] <burek> yes, but still :) [13:48] <burek> mp3 never had such comments, nor aac-lc :) [13:48] <JEEB> well, d'oh [13:48] <JEEB> neither was made to sound good with low bitrates [13:48] <burek> well low bitrates are good thing, no? :) [13:49] <JEEB> yes, unless too much data is being taken out. My experience with HE-AAC(v2) usually tends to be that it sounds good, but if you don't have very strict bitrate limitations you'll want to use AAC-LC which takes out less information [13:49] <JEEB> for phones and such I would use HE-AACv2 [13:49] <JEEB> for video encodes for internets where the video bitrate is low as well, sure'ish [13:50] <JEEB> for video encodes where the video bitrate is higher, I might as well go for AAC-LC [13:50] <burek> well you can go with flac then :) [13:50] <burek> if bitrate is not a problem :) [13:50] <JEEB> wavpack if I want lossless [13:50] <JEEB> uhh [13:51] <JEEB> there's a difference between "less or around 192kbps" and "768kbps+" [13:51] <JEEB> I don't say that compression algorithms that make stuff sound good with low bitrates in general is bad, just that the ways that HE-AAC(v2) uses generally tends to remove stuff (while still keeping audible artifacts mostly away) [13:52] <burek> i don't know.. i've been listening to the classic music encoded with aac+ and it was of a superb quality.. so, I can freely say that it's a very good encoder [13:53] <JEEB> that removal might or might not be a bad thing with source X, but given the algorithms I generally prefer a good AAC-LC encoder (neroaacenc/qaac) and somewhat higher bitrates if I want to do a relatively good quality lossy audio encode [13:53] <JEEB> or vorbis with aotuv stuff [13:55] <JEEB> HE-AAC(v2) is very useful for what it's meant for tho [13:55] <JEEB> which is <64kbps encoding [13:55] <JEEB> I think I tested it down to 32kbps or so, maybe 16kbps :3 [13:56] <burek> 32kbs is the epicenter.. 48kbps is for high quality (cd like, classic music and stuff) [13:56] <JEEB> for HE-AAC it was already low'ish, for HE-AACv2 it could take some less bitrate [14:13] <dirtycookie> burek: hi again [14:14] <dirtycookie> i get an error message at compiling ffmpeg just as you told me to do, it tells me that libaacplus is not found [14:15] <burek> did you do this step: http://pastebin.com/P2S3SLHn [14:16] <dirtycookie> burek: hold let me check again [14:21] <dirtycookie> burek: did that again now trying my luch with ffmpeg [14:21] <dirtycookie> ./config of ffmpeg went without any errors... now making [14:21] <dirtycookie> looking good [14:22] <burek> ok :) [14:23] <dirtycookie> burek: in any case i thank you 4 ur patience that you gave me as a noob [14:23] <burek> :beer: :) [14:23] <dirtycookie> i dont meet many people in irc having the patience to this, other than answering rtfm [14:24] <dirtycookie> burek:where u from [14:24] <burek> well, you should rtfm though ^^ [14:24] <burek> serbia :) [14:29] <dirtycookie> burek:i would agree but is a little too technical 4 me and i just want to get on with my work [14:31] <burek> i know exactly what you mean... [14:31] <burek> the worst thing is, the more you know, the less people can help you.. [14:32] <burek> so it's a good practice to learn to read manuals.. :/ [14:35] <dirtycookie> burek: if i have the time 4 it [14:36] <dirtycookie> but thanks again :) [14:36] <burek> :) :beer: [15:03] <dirtycookie> burek: ok i finished compinling it, tried now the follwing command: "youtube-dl --extract-audio --extract-format=mp3 [target url]" and gave me "youtube-dl: error: noc such option --extract-format" [15:04] <dirtycookie> burek: i left that out and it started downloading it in a .webm file and gave me a warining that it couldn't obtain the file audio codec with ffprobe [15:06] <burek> try typing just ffmpeg [15:06] <burek> to see what ffmpeg has been started [15:07] <burek> from repo or compiled one [15:07] <dirtycookie> burek: dont tell me that it has to be like "/home/me/ffmpeg/ffmpeg"? [15:09] <dirtycookie> burek: when entering ffmpeg in my console i get the follwing [15:10] <dirtycookie> "ffmpeg: error while loading shared libraries: libpostproc.so.51: cannot open shared object file: Noch such file or directory [15:11] <burek> oh i see [15:11] <burek> dpkg -l | grep ffmpeg [15:11] <dirtycookie> burek: i get, [15:12] <dirtycookie> ii ffmpeg 4:0.7.2-1ubuntu1 Multimedia player, server, encoder and transcoder [15:12] <burek> ok [15:12] <burek> dpkg -r ffmpeg [15:12] <burek> or apt-get remove ffmpeg [15:13] <dirtycookie> ii gstreamer0.10-ffmpeg 0.10.12-1ubuntu ffmpeg plugin for GStreamer [15:13] <burek> that one is ok [15:13] <dirtycookie> burek: ok removing, now what [15:13] <burek> hash -r [15:13] <burek> ldconfig [15:13] <burek> and try typing ffmpeg again [15:16] <dirtycookie> burek: it gave me [15:16] <dirtycookie> ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference [15:17] <burek> ok [15:17] <burek> dpkg -l | grep libav [15:18] <dirtycookie> http://pastebin.com/vtkMU1Vc [15:19] <burek> ok [15:19] <dirtycookie> burek: u there? [15:20] <burek> apt-get remove libavutil51 libavformat53 libavfilter2 libavdevice53 libavcodec53 [15:20] <burek> i am [15:20] <burek> ldconfig [15:20] <burek> and try again with ffmpeg [15:24] <dirtycookie> burek: ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference [15:34] <burek> hash -r [15:34] <burek> ldconfig [15:34] <burek> ffmpeg [15:52] <dirtycookie> burek: ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference [15:55] <vivienschilis> burek, so ffmpeg support he-aac+ with libaacplus [15:55] <vivienschilis> but what about he-aac [15:55] <vivienschilis> ? [15:56] <burek> he-aac v2 [15:57] <burek> you can read more here [15:57] <burek> http://tipok.org.ua/node/17 [16:01] <dirtycookie> burek: ideas running out? [16:02] <burek> ? [16:02] <burek> what ideas? [16:02] <dirtycookie> ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference [16:02] <vivienschilis> oh ok, the lib have support for v1 and v2? [16:03] <burek> well, dirtycookie, uninstall libavutil with apt-get remove [16:03] <burek> do ldconfig [16:03] <burek> and recompile (configure, make..) [16:09] <dirtycookie> burek: ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference [16:09] <dirtycookie> didnt i removed libavutil? [16:10] <burek> check in dpkg -l | grep libavutil [16:12] <dirtycookie> dpkg -l |grep libavutil [16:12] <dirtycookie> ii libavutil-extra-51 4:0.7.2.1ubuntu1 Libav utility library [16:12] <dirtycookie> rc libavutil51 4:0.7.2-1ubuntu1 Libav utility library [16:14] <relaxed> dirtycookie: are you using 64bit linux? [16:14] <dirtycookie> relaxed: no it it is i386 ubuntu [16:14] <dirtycookie> xubuntu sry [16:16] <dirtycookie> burek: ok i think i made it, i got now a different output [16:17] <dirtycookie> http://pastebin.com/gNuM2d7u [16:25] <Pascal_1> hello [16:26] <Pascal_1> i'm trying to make a video from several picture (differents size and different orientation) with this command line : ffmpeg -r 2 -b 1800 -i %05d.jpg video.mp4 [16:26] <Pascal_1> but all the frame of the video is a malformed picture [16:27] <Pascal_1> is there a way to make a video with all picture and ffmpeg keep the format of each pictures ? [16:46] <Pascal_1> any idea ? [17:12] <burek> dirtycookie, thats ok [17:12] <burek> now you have it compiled [17:13] <burek> Pascal_1, can you make input pictures all the same format/size [17:42] <pasteeater> burek: there is an ubuntu specific ffmpeg compile guide in /topic. [17:44] <burek> yes there is :) [17:51] <burek> maybe it would be a good idea to have 1 static build for linux/win/mac updated weekly, with all the codecs/features enabled, just for those who want to "just install and go", because there a lot of lazy people, coming from m$ world [18:40] <giwe> How do I produce a "fastforward" version of a movie? I want a 20min movie to play in 1min. [19:00] <vivienschilis> is there any conflict between libvo-aacenc and libaacplus? [19:03] <pasteeater> giwe: ffmpeg -i input -vf setpts=0.05*PTS output [19:04] <pasteeater> frames will be dropped, however [19:06] <burek> pasteeater, does it reencode video? [19:11] <pasteeater> yeah [19:12] <pasteeater> and the 0.05 = 1 minute / 20 minutes [19:19] <Hyperi> PTS is per time second or what ? [19:21] <giwe> pasteeater: thanks will try [19:22] <giwe> pasteeater: is there a webpage that explains this for me too bookmark? [19:28] <Hyperi> There is this bible [19:28] <Hyperi> called google .) [19:28] <giwe> pasteeater: nvm it is in the ffmpeg man [19:34] <HrznDefeated> Hello all, how good is streaming support in ffmpeg? If I am planning to consume and decode rtsp/rtp and mpeg2-ts streams, should I be looking at ffmpeg directly, or should I be looking at gstreamer? [19:39] <burek> pasteeater would this do the same thing: ffmpeg -i input.avi -r 3 -vcodec ... -acodec ... output.avi [19:39] <burek> or the vf does something more peculiar [19:43] <vcs> hi, when i use "-vcodec copy" and also set the framerate from a raw h264 stream, the video is copied but it plays back at the framerate i didn't specifcy (the one it guesses the video is at) [19:44] <vcs> here is the command for refference: "ffmpeg -i 11.264 -vcodec copy -r 10 11.mp4" [19:44] <vcs> it seems no matter what I do, ffmpeg reads the raw stream at 30fps [19:44] <vcs> is there some way I can force ffmpeg to realize that the raw stream is only 10fps? [19:45] <JEEB> properly set it on the encoder's side? [19:45] <JEEB> I always kind of had the idea of -r being the setting for frame dropping/frame blending/copying [19:46] <JEEB> so I'd probably just use L-SMASH's muxer or mp4box to mux it with a specific frame rate defined :3 [19:47] <vcs> hmm ok, ill give it a shot thanks [19:59] <pasteeater> burek: you could probably so something similar with: ffmpeg -i input -an -f rawvideo -r 12 - | ffmpeg -s 1280x720 -r 24 -f rawvideo -i - out [20:00] <burek> yes, I figured :) I was thinking to pass the stream as-is through the decoder and to drop sufficial frames at the encoder side [20:00] <burek> i just wanted to see if this could be done without reencoding [20:01] <burek> to somehow "mark" the stream as faster/slower, but it seems it cant be done :) [20:01] <pasteeater> oh, i think -vf is ignored with -vcodec copy. [20:02] <burek> vcs [20:02] <burek> you could place -r before first -i [20:02] <burek> that would do what you want [20:03] <vcs> hmm ill give it a shot [20:03] <pasteeater> Hyperi: PTS = presentation timestamp in input [20:04] <vcs> burek: i tried it, got 30 FPS back :| [20:04] <vcs> after reading the output file with ffprobe [20:04] <burek> well [20:04] <burek> try this [20:04] <burek> ffmpeg -r 10 -i 11.264 -vcodec copy -r 10 11.mp4 [20:05] <vcs> still reads back 30 fps [20:05] <burek> hmh [20:05] <burek> wait wait [20:05] <burek> does raw h264 store fps value? [20:06] <vcs> i dont think so [20:06] <vcs> ill check [20:06] <burek> vcs, try this what pasteeater suggested [20:06] <JEEB> it can [20:06] <JEEB> timestamps [20:06] <JEEB> which some muxers read, others don't [20:06] <vcs> hm [20:06] <burek> ffmpeg -i 11.264 -f rawvideo -r 10 - | ffmpeg -s 1280x720 -r 10 -f rawvideo -i - 11.mp4 [20:06] <burek> just fix the -s [20:07] <burek> i think forcing the rawvideo will get rid of the timestamps [20:07] <burek> which will give you what you need [20:08] <burek> anyway, this seems to be a trivial problem, like rearranging timestamps [20:08] <burek> does ffmpeg have some options which handle that, so the people could easily speed up/slow down videos, without reencoding [20:11] <vcs> hmm.. that may have worked [20:11] <vcs> its reading back 10fps now [20:11] <vcs> the interesting thing is though the stream has the same duration [20:11] <vcs> as it did when it encoded at 30fps [20:11] <vcs> well, not encoded i mean muxed [20:12] <burek> you wanted to speed it up? [20:12] <burek> slow down? [20:12] <vcs> slow it down, it was playing back at 30fps but recorded at 10fps [20:13] <burek> i see, so it just dropped frames [20:13] <burek> and still plays faster [20:13] <vcs> right [20:13] <burek> well [20:13] <vcs> i may give mp4box a shot [20:14] <burek> try this [20:14] <burek> ffmpeg -i 11.264 -f rawvideo -r 30 - | ffmpeg -s 1280x720 -r 90 -f rawvideo -i - 11.mp4 [20:14] <burek> wait [20:16] <burek> i think we should use named pipes here [20:16] <burek> im not sure [20:16] <burek> but it seems to me like the input stream is "faster" than the output, so no buffering will happen, just frame dropping [20:17] <burek> that's why i think we could save it maybe to a file with -r 30 [20:17] <burek> to have all frames [20:17] <burek> and rawvideo, to loose timestamps [20:17] <burek> and after that to read the file with -r 10 [20:18] <burek> ffmpeg -i 11.264 -f rawvideo -r 30 tmp11 [20:18] <burek> ffmpeg -s 1280x720 -r 10 -f rawvideo -i tmp11 out11.mp4 [20:21] <vcs> alright ill give it a shot [20:26] <vcs> that seemed to do the trick [20:26] <vcs> thanks [20:27] <Hyperi> pasteeater: ty [20:27] <burek> :beer: :) [20:44] <giwe> pasteeater: the "fastforward" did work with your PTS suggestion. It did drop frames as you said. Do you recommend any other way in such a way that frames are not dropped? [20:45] <pasteeater> giwe: i'm not sure. [20:49] <burek> giwe, read above [20:49] <burek> vcs had exactly the same problem, i guess [20:59] <burek> general, faq, ffmpeg, ffprobe, ffplay.. [21:31] <vivienschilis> how good is vo-aacenc? [21:31] <vivienschilis> what does it bring compare to faac [21:31] <pasteeater> vivienschilis: probably on par with faac, or maybe a little worse. [21:32] <vivienschilis> why is android not relying on faac? I think it use this one right? [21:32] <vivienschilis> just curious [21:32] <pasteeater> and to answer your previous question, i've had to issues with both libvo-aacenc and libfaac enabled. [21:32] <relaxed> vivienschilis: faac can not be distributed because it's nonfree [21:33] <pasteeater> *no issues [21:34] <vivienschilis> to activate libaacplus I had to ditch libvo-aaenc [21:34] <vivienschilis> not a big deal I wasn't using it [21:34] <vivienschilis> that's why I asked :) [21:34] <pasteeater> oh, duh. i now see you said libaaacplus, not libfaac [21:51] <vivienschilis> pasteeater, related to libaacplus, does supports he-aac and he-aac v2? [21:51] <vivienschilis> the web page is quite confusing [21:51] <vivienschilis> does it* [21:51] <pasteeater> vivienschilis: i don't know. burek would know. [21:53] <vivienschilis> ok thx [21:53] <burek> vivienschilis, why would you want he-aac [21:53] <burek> withous ps [21:53] <vivienschilis> I though it was depending on the nitrate [21:53] <vivienschilis> for middle nitrate [21:54] <vivienschilis> bitrate* [21:54] <vivienschilis> he-aac was better [21:54] <vivienschilis> I might have read a bad article about it [21:55] <vivienschilis> might question was more [21:55] <vivienschilis> if I use -acodec aacplus [21:55] <vivienschilis> It would use he-aac v2 by default? [21:56] <JEEB> <vivienschilis> why is android not relying on faac? I think it use this one right? <- because faac is "officially" GPL, while vo-aacenc is payware, which was then licensed by Google under the BSD license. Google wants Android stuff to be BSD or similar [21:57] <JEEB> Also, faac was found to be using reference implementation sources [21:57] <JEEB> which is why it isn't even GPL compatible any more [21:57] <JEEB> (since the reference implementation source code is not under a GPL-compatible license) [21:58] <vivienschilis> hum thanks JEEB [21:58] <relaxed> doesn't libaacplus suffer from that too? (using reference implementation) [21:58] <JEEB> vo-aacenc seems to similarily contain the source code of the reference implementation, which is why IIRC it is considered GPL-incompatible... If I recall correctly [21:58] <JEEB> yes [21:58] <JEEB> so most of those encoders, if you enable them you can't give out your binaries [21:59] <JEEB> because they're not (L)GPL compatible any more [21:59] <Mista-D> where can one download a w32 build of FFserver please? [22:22] <inashdeen> hi, i ran into this : Incompatible sample format '(null)' for codec 'mp2', auto-selecting format 's16' . what might be the cause. thanks in advance :) [22:43] <raven> hi [22:43] <raven> possible to write mp3 files with 8 bit instead of 16? [22:44] <Tjoppen> mp3 doesn't have bitdepth per se. we had a similar discussion in hear yesterday [22:45] <Tjoppen> so no. it's all in the decoder [22:45] <raven> ok i thought so [22:45] <raven> tnx [23:09] <pasteeater> inashdeen: use a pastebin site to show your ffmpeg command(s) and the complete console output(s) [23:46] <Sorikan> Does anyone have or know of performance numbers when running ffmpeg on a dual core Atom system? Thanks..... [00:00] --- Tue Nov 29 2011

1 0

ffmpeg-devel.log.20111127
by burek 28 Nov '11

28 Nov '11

[00:09] <ubitux> the issue with audio visualizations is similar to the issue with subtitles [00:09] <ubitux> in the first case we want a filter that takes audio in input, and video in output [00:11] <ubitux> and in the second case we want a filter that takes a subtitle type in input and output video [00:39] <CIA-36> ffmpeg: 03Anton Khirnov 07master * r488eec1044 10ffmpeg/libavcodec/avcodec.h: lavc: update doxy to use nondeprecated API. [00:39] <CIA-36> ffmpeg: 03K.Y.H 07master * r51f316a997 10ffmpeg/libavcodec/cook.c: [00:39] <CIA-36> ffmpeg: cook: fix apparent typo in extradata parsing [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Carl Eugen Hoyos 07master * r1484b5dec5 10ffmpeg/libavcodec/flicvideo.c: [00:39] <CIA-36> ffmpeg: flicvideo: check extradata_size before accessing extradata. [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Reimar Döffinger 07master * r785baa738a 10ffmpeg/libavcodec/nuv.c: [00:39] <CIA-36> ffmpeg: nuv: use FFALIGN. [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Reimar Döffinger 07master * rf6afacdb3b 10ffmpeg/libavcodec/nuv.c: [00:39] <CIA-36> ffmpeg: nuv: check per-frame header for validity. [00:39] <CIA-36> ffmpeg: Since it contains dimensions parsing an invalid one has rather [00:39] <CIA-36> ffmpeg: annoying effects. [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Reimar Döffinger 07master * r7a62ddb689 10ffmpeg/libavcodec/rtjpeg.c: [00:39] <CIA-36> ffmpeg: rtjpeg: check get_block return value for error. [00:39] <CIA-36> ffmpeg: This avoids crashes due to reading out-of-bounds. [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Reimar Döffinger 07master * r7fb55e0b02 10ffmpeg/libavcodec/rtjpeg.c: [00:39] <CIA-36> ffmpeg: rtjpeg: simplify get_block() by using get_bits_left. [00:39] <CIA-36> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [00:39] <CIA-36> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [00:39] <CIA-36> ffmpeg: 03Mans Rullgard 07master * r3bd1162a52 10ffmpeg/libavcodec/gifdec.c: [00:39] <CIA-36> ffmpeg: gif: fix invalid signed shifts [00:39] <CIA-36> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:39] <CIA-36> ffmpeg: 03Mans Rullgard 07master * r93c286e54f 10ffmpeg/libavcodec/qtrle.c: [00:39] <CIA-36> ffmpeg: qtrle: simplify 32-bit decode using intreadwrite macros [00:39] <CIA-36> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:39] <CIA-36> ffmpeg: 03Mans Rullgard 07master * r644bff6c9b 10ffmpeg/libavcodec/apedec.c: [00:39] <CIA-36> ffmpeg: apedec: fix signed integer overflows [00:39] <CIA-36> (73 lines omitted) [00:39] <CIA-36> ffmpeg: 03Mans Rullgard 07master * rff6d9cc558 10ffmpeg/libavcodec/snow.c: [00:39] <CIA-36> ffmpeg: snow: fix signed overflow in byte to 32-bit replication [00:39] <CIA-36> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:39] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r5c15b78e4a 10ffmpeg/: (log message trimmed) [02:16] <CIA-36> ffmpeg: 03Kostya Shishkov 07master * r801393bc96 10ffmpeg/libavcodec/vc1dec.c: [02:16] <CIA-36> ffmpeg: vc1: select interlaced scan table by FCM element [02:16] <CIA-36> ffmpeg: Interlaced videos can contain progressive frames too and now wrong scantable [02:16] <CIA-36> ffmpeg: is selected for them. [02:16] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:27] <ubitux> buffersink filters are only for debug, right? [02:29] <iive> have somebody tooked a look of elenril's decode_audio4() stuff. Am I wrong if I think that they would be using data[] pinters to place each channel into separate plane, and thus the new api comes with build-in limitation of 8 channels max? [02:30] <ubitux> isn't this ruggles stuff? [02:31] <iive> humm... right, apologies to elenril . [02:31] <iive> BBB then? [02:32] <ubitux> Justin Ruggles (ruggles) ` Ronald S. Bultje (BBB) [02:33] <ubitux> anyway, it looks like indeed to be limited to 8 channels [02:33] <ubitux> old api: 4, new api: 8, but that sounds weird [02:33] <michaelni> 8 ch is already today too little :) [02:33] <michaelni> 7.1 + stereo downmix for example [02:33] <ubitux> there is "uint8_t **extended_data;" [02:34] <ubitux> i think it can be extended through this [02:34] <ubitux> but i didn't read the details [02:34] <kierank> afaik they'll just change the #define when they want more channels [02:34] <ubitux> there was some ABI discussion about this in the first patchset [02:34] <michaelni> they should start at 16 minimum not 8 [02:35] <iive> it doesn't matter with how many they start, they will never be enough. [02:35] <ubitux> it can be extended [02:36] <ubitux> i don't think having this value dynamically allocated is a good idea [02:36] <iive> and yeh, i've started forgetting name<>nicks [02:36] <michaelni> sure but one doesnt start at a value that is already at the time one puts it there too small [02:36] <ubitux> but 8 sounds too little somehow [02:36] <ubitux> i agree with the 16 value :p [02:36] <ubitux> anything <= 64 sounds reasonable to me :p [02:36] <ubitux> but > 8 [02:36] <ubitux> :) [02:36] <iive> how many speakers are in a theater? [02:38] <iive> oh, justin is the ac3 guy. [02:39] <iive> anyway, you can easily represent the channels as height, and the number of samples as width. [02:41] <iive> then depending on the format (planar vs packed) the linesize could be the size of one plane or size of all-channels-sample. [02:42] <iive> aka planar -> linesize=width*sizeof(sample) ; packed = channels*sizeof(sample). [03:24] <CIA-36> ffmpeg: 03Peter Ross 07master * r3d977edb04 10ffmpeg/libavcodec/iff.c: [03:24] <CIA-36> ffmpeg: HAM6/HAM8 support for IFF ACBM decoder [03:24] <CIA-36> ffmpeg: Based on patch by ami_stuff [03:24] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:24] <CIA-36> ffmpeg: 03Peter Ross 07master * rb488679510 10ffmpeg/libavcodec/iff.c: [03:24] <CIA-36> ffmpeg: iff: fix invalid reads (ticket 689) [03:24] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:16] <relaxed> sudo shutdown -h now [04:16] <relaxed> sorry [06:12] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r2bb79b23fe 10ffmpeg/libavcodec/pthread.c: [06:12] <CIA-36> ffmpeg: pthread: next try on freeing threads without crashing. [06:12] <CIA-36> ffmpeg: This should fix mingw [06:12] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:24] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r47044625ba 10ffmpeg/libavcodec/pthread.c: [06:24] <CIA-36> ffmpeg: pthread: check pthread_create() return value. [06:24] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:19] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r102a2463d3 10ffmpeg/libavformat/img2.c: [15:19] <CIA-36> ffmpeg: img2: Allow writing multiple files onto the same output file. [15:19] <CIA-36> ffmpeg: Fixes Ticket687 [15:19] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:56] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * rb7c7eae7d9 10ffmpeg/libavformat/tta.c: [15:56] <CIA-36> ffmpeg: tta: better check for totalframes. [15:56] <CIA-36> ffmpeg: Avoids crash, Fixes Ticket 690 [15:56] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:40] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * re99c4bbdf3 10ffmpeg/libavformat/img2.c: [16:40] <CIA-36> ffmpeg: img2: update first file only when -updatefirst is specified [16:40] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:37] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r6d13499be0 10ffmpeg/ (avconv.c cmdutils.c cmdutils.h ffmpeg.c ffplay.c): [21:37] <CIA-36> ffmpeg: cmdutils: pass AVCodec to filter_codec_opts() [21:37] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:37] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r1b1223872d 10ffmpeg/libavcodec/ (aaccoder.c aacenc.h): [21:37] <CIA-36> ffmpeg: aacenc: add AAC_CODER_NB [21:37] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:37] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * re64edeed3c 10ffmpeg/libavcodec/ (aacenc.c aacenc.h): [21:37] <CIA-36> ffmpeg: aacenc: make the aac coder user choosable. [21:37] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:50] <j-b> ANNOUNCE: if you want to add more GCI tasks, this is NOW, or never. [21:50] <iive> ? [21:54] <JEEB> Google closing the registration? [22:05] <michaelni> j-b, how much time is left ? [22:06] <j-b> tonight [22:08] <michaelni> :( [22:08] <michaelni> anyone in here wanting to help mentoring ? [22:09] <michaelni> ill add a few more tasks but i dont have so much time, its better if i maintain ffmpeg than mentor ... [22:10] <Compn> you could ask that other project about it :P [22:10] <Compn> starts with an L [22:11] <michaelni> Compn, i dont think they will mentor for ffmpeg [22:11] <Daemon404> they arecurrently mentoring a crapload of libav stuff for gci [22:13] <Compn> ah [22:13] <Compn> the gci website sucks [22:13] <Compn> only 18 projects listed, dont see ffmpeg or libav [22:13] <Compn> only videolan [22:13] <michaelni> Compn, if you want to register i can help you [22:13] <Compn> no no, too much work [22:13] <Compn> just trying to look at it [22:13] <michaelni> yes you need to join videolan [22:14] <Compn> ah [22:14] <michaelni> the only way to register is to manually type in some urls that you can find with google on non google pages :) [22:16] <michaelni> "Find a report a new bug for FFmpeg" is what Code Documentation Outreach User Interface Research Training Translation Quality Assurance ? [22:16] <Compn> research [22:23] <michaelni> how many should i create ? [22:23] <michaelni> ... each task is only good for one bug report [00:00] --- Mon Nov 28 2011

1 0

ffmpeg.log.20111127
by burek 28 Nov '11

28 Nov '11

[00:35] <pinkette> is it true that aacencoder by android is better than faac or aac-ffmpeg [00:36] <JEEBsv> IIRC both based more or less on the example implementation sources for an AAC encoder > faac, libvo-aacenc [00:36] <JEEBsv> ffaac's encoder is still very much a work-in-progress [00:37] <pinkette> jeebsv sorry can you repeat that? are you saying libvo-aacenc > faac ? [00:37] <JEEBsv> so I'd say that libvo-aacenc and faac are better than ffaac [00:37] <JEEBsv> ffmpeg's aac encoder = ffaac [00:37] <pinkette> i see [00:37] <JEEBsv> but those two then should be quite similar [00:37] <pinkette> okay what about libvo vs faac then [00:38] <pinkette> libvo and faac is pretty much same? [00:38] <JEEBsv> aye [00:39] <JEEBsv> I think since both base on the example implementation sources, neither of them is really usable for builds that you think about releasing publically [00:51] <pinkette> what other aac encoder does ffmpeg support [00:52] <JEEBsv> libaac+ IIRC, which also uses reference implementation code as far as I can remember :D [00:53] <pinkette> what do you mean by implementation sources/code [00:55] <JEEBsv> the stuff that's in the AAC's specifications [00:55] <JEEBsv> there are reference implementations [00:55] <JEEBsv> and those are not exactly (L)GPL-compatible as far as I know [00:59] <pinkette> isn't all aac encoder = have to have aac specs [01:02] <JEEBsv> yes, of course -- but copying code out of the reference implementation... [01:02] <JEEBsv> because the specs also contain a decoder and an encoder reference implementation [01:03] <JEEBsv> Just like you have JM with H.264 [01:59] <faab> hi! i'm converting my ogg music to mp3 so i can put it on my ipod.. [01:59] <faab> but the mp3s don't have tags after the coversion! [01:59] <faab> -map_metadata 0:0 doesn't seem to fix the problem... a little help? [02:04] <pasteeater> faab: use a pastebin site to show your ffmpeg command(s) and the complete console output(s) [02:09] <faab> pasteeater: okay.. doing so.. [02:11] <faab> pasteeater: here it is ... http://pastebin.com/EwWLdhkY [02:11] <faab> and there are no tags in the resultant file.. [02:12] <faab> since i'm planning to do this a large part of my library, i figured it'd be nice if i could get the tags to copy over.. [02:15] <pasteeater> ffmpeg doesn't show any metadata for test.mp3? ffmpeg -i test.mp3 [02:17] <faab> pasteeater: it would appear not: http://pastebin.com/U9L328TF [02:17] <faab> :( [02:21] <pasteeater> same for me...although i recall it working previously, and -map_metadata wasn't needed, but maybe i'm mistaken. [02:21] <pasteeater> and i'm using Git head [02:23] <faab> i'm using a cygwin build from the cygwin ports project.. [02:23] <faab> ah well.. [02:28] <notfaab> pasteeater: thanks for confirming at least ;) [02:29] <pasteeater> notfaab: i'm still monkeying with it... [02:30] <pasteeater> anyone know how to convert a shallow git repo to a "normal" one? [02:31] <notfaab> pasteeater: you don't have to if you don't want to.. but if you figure it out, thanks! [02:32] <pasteeater> it's a good distraction [02:33] <notfaab> pasteeater: i know that feeling [02:36] <monstaRtruck> pasteeater my sound still out of sync [02:40] <pasteeater> monstaRtruck: I don't remember your issue [02:42] <monstaRtruck> i made my movie smaller [02:43] <monstaRtruck> but its not due to my comp [02:43] <monstaRtruck> ffmpeg is jsut weird [02:43] <monstaRtruck> does urs sync if u record a desktop [02:43] <monstaRtruck> im trying to record a video game [02:44] <pasteeater> i don't have any video games [02:44] <pasteeater> except dungeon crawl [02:44] <pasteeater> do you have a sample of your output? [02:44] <monstaRtruck> yes [02:44] <monstaRtruck> ther is some commant that will pull the sound back [02:45] <pasteeater> huh? [02:45] <monstaRtruck> my sound is 3 seconds late [02:55] <pasteeater> notfaab: duh. not working for me. why didn't i notice this before? [02:56] <notfaab> pasteeater: notice what? [02:58] <pasteeater> the metadata not copying from in to out [03:02] <notfaab> you mean, it didn't work for you way-back-when you remembered it working? :P [03:05] <pasteeater> i don't remember [03:07] <monstaRtruck> pasteeater how come its out of sync [03:07] <monstaRtruck> so many ppl complaining theirs out of sync too [03:08] <pasteeater> i don't know. [03:08] <monstaRtruck> hav u ever tried recording desktop [03:26] <notfaab> pasteeater: i'll just use musicbrainz picard to retag all the albums.. [03:26] <notfaab> thanks though! have a nice night.. [07:47] <kcm1700> ls -altp [07:47] <kcm1700> oh, sorry. [10:06] <cryptopsy> can ffmpeg be used to make sounds deeper? [10:07] <cryptopsy> as in, have someone's voice sound deeper [10:08] <cryptopsy> also known as the pitch ? [10:10] <cryptopsy> apparently sox can do it [10:16] <cryptopsy> what's the operation where you remove scratchy background noise from recordings? [10:16] <o3u> hi, i would like to stream selected videos that are processed by ffmpeg, i'm thinking of doing it by creating a pipe (lets say pipe.mp4), and use jsvideo on a webpage to read that pipe, it would involve running ffmpeg continuously until visitor leaves page, does anyone know if this would work? [10:17] <cryptopsy> hiss reduction [10:17] <cryptopsy> that's it [10:17] <o3u> ffmpeg would output to the pipe, my player would read from it [10:18] <cryptopsy> you want to video the result as it's being processed? [10:18] <cryptopsy> view [10:19] <o3u> pretty much, it can be processed slightly ahead of time, [10:20] <cryptopsy> what you said didn't make sense [10:20] <cryptopsy> a <-stream-> b [10:20] <cryptopsy> which direction? [10:23] <o3u> i have a bunch of videos i'd like to put together, concatenate so to speak, and before that happens they need to individually be processed by ffmpeg, once processed and concatenated they're streamed to my video player, however the whole concatenation only happens once the visitor is on the webpage, it cant happen before, and it must be done live [10:23] <o3u> or "almost" live, i don't really understand your diagram there [10:47] <cryptopsy> you don't have to concatenate them into one file [10:47] <cryptopsy> you can have your website play one after the other [10:47] <cryptopsy> for example if you play music in mplayer this way you can't tell two songs apart, same with video [10:47] <cryptopsy> make sure the end of clip n lines up with the start of clip n+1 and its file [10:48] <cryptopsy> have your website cache the video so its there for playing when the previous is finished [15:19] <muffin_666> hey all, I have a strange problem with decoding mpeg4/aac: if I try to open the file by the standard-way (without custom IO) the file gets played nicely, however, if I try to use custom callbacks, av_read_frame() will never find packets from the audio stream, even if the audio stream is set correctly and the av_dump_format() prints out everything correct. The result is that my audioQueue is never filled. But the custom IO works fine with oth [15:19] <muffin_666> er files e.g. h264/aac. Does anybody have an idea what might be wrong? (here is my code: http://pastebin.com/9znPAXCp) [16:30] <masch> hi. [16:31] <masch> Log: http://pastebin.com/N9JUSZv4 | Config: http://pastebin.com/7m1q93uN can anyone tell me whats wrogn? [18:47] <CSMan> is there a way to specify mp4 output? [18:47] <CSMan> like you do -f flv to create flv files [18:47] <sacarasc> How about... -f mp4 [18:47] <sacarasc> :O [19:04] <parkeyparker> Is anyone here using ffmpeg to encode HTML5 compatible videos? If you are what bitrates are you aiming for for a good balance between quality and file size? [19:04] <blez> hello [19:04] <blez> how to extract a frame from flv/mp4 if I had only a partial content from it [19:04] <blez> without the beginning [19:14] <ticapix> Hi [19:16] <ticapix> using ffmpeg is there a way to change the fps property of a avi (mjpeg) ? without actual modifying the video (no frame being removed or duplicated), just the property of speed playback. [19:17] <CSMan> how can I get ffmpeg to use all my cpu power? [19:20] <ticapix> The way I do it now is to convert my video to y4m and edit manualy the file to change fps in the header. I'n sure there is a smarter way, no ? [20:07] <michaelni> blez, using h263 / h264 raw demuxer may work [20:08] <blez> how to do that [20:09] <michaelni> -f h263 or -f h264 but theres a high chance it will not work [20:10] <blez> is there a way to get the last frame in one command? [20:33] <kriegerod> does anybody know what bug in transcoding h264 application, that uses libavcodec & x264, can cause grey crap like this http://tinypic.com/r/24en86r/5 instead of normal pic. This appears after several hours of good transcoding work [21:18] <Foxhoundz> Is there any way to make ffmpeg guess the format of the input file? The file doesn't have any extension attached to it. [21:18] <Foxhoundz> And this is for the ffmpeg binary on windows [21:20] <parkeyparker> Foxhoundz: you might need to detect that yourself with a script before passing it to ffmpeg... [21:20] <parkeyparker> Unless there is something mentioned in the documentation about it? [21:21] <parkeyparker> As to what script you would use I don't know... However I'm sure there is some form of application/script that will look at the header section of the video and work out its extension for you automatically [21:25] <iive> Foxhoundz: ffmpeg is supposed to guess the file type by content and use the extension as hint. there is ffprobe too. [21:26] <parkeyparker> Ah ok, I always like to be corrected when it comes to features I didn't realise existed :P [21:26] <Foxhoundz> It's giving me an unknown format when I do it [21:26] <Foxhoundz> ffmpeg.exe -i "testFile" -ab 192k test.mp3 [21:27] <Foxhoundz> I think I need to add in the -f switch too [21:27] <Foxhoundz> nevermind [21:28] <Foxhoundz> -f only forces a specific output format [21:28] <iive> Foxhoundz: if it can't recognize it, then it is bad... [21:28] <parkeyparker> However forcing the output might be better than it trying to use the same output than input [21:28] <Foxhoundz> iive: It's an FLV file and it's not corrupted [21:29] <iive> Foxhoundz: are you sure you use recent ffmpeg ? [21:29] <Foxhoundz> It's a few months old [21:29] <iive> flv should be easy to guess. [21:29] <Foxhoundz> I'll try to get the latest one and give it a shot [21:30] <iive> rename it, and see if ffmpeg guesses it then. [21:31] <iive> btw, are the first 3 bytes of the file data content a "FLV" string? [21:32] <Foxhoundz> back [21:32] <Foxhoundz> ok I downloaded the latest version [21:32] <Foxhoundz> and it worked [21:32] <Foxhoundz> It seems that my version was a year or so old [22:27] <darkstarbyte> How would I find out the audio bit rate of something? [23:18] <pasteeater> kriegerod: does the gray occur in all players? [23:19] <kriegerod> i test it on ffplay only [23:19] <pasteeater> use a pastebin site to show your ffmpeg command(s) and the complete console output(s) [23:20] <pasteeater> darkstarbyte: bitrate = file size/duration, or see what 'ffmpeg -i input' says. [23:21] <darkstarbyte> I thought that does not work because h.264 can compress files really well [23:21] <darkstarbyte> and [23:21] <darkstarbyte> have the same quality of mpeg2 [23:46] <pasteeater> darkstarbyte: you asked about audio bitrate. what's H.264 have to do with that? i assumed you meant an audio file. [00:00] --- Mon Nov 28 2011

1 0

ffmpeg-devel.log.20111126
by burek 27 Nov '11

27 Nov '11

[00:07] <michaelni> ubitux, if it allows code reuse and simplifies things its surely possible [00:07] <ubitux> i'm going to try [00:07] <ubitux> but it might take some while, i'm not familiar with that stuff... yet. [00:08] <michaelni> where to put it i dont know, where it fits in naturally best i guess [00:08] <ubitux> :) [00:09] <ubitux> it seems the drawutils are a good place [00:58] <CIA-36> ffmpeg: 03Stefano Sabatini 07master * ra11eeb9215 10ffmpeg/configure: configure: sort entries [01:17] <CIA-36> ffmpeg: 03Mans Rullgard 07master * r6b34fbba9b 10ffmpeg/libavutil/common.h: [01:17] <CIA-36> ffmpeg: MK(BE)TAG: avoid undefined shifts [01:17] <CIA-36> ffmpeg: Casting the left-most byte to unsigned avoids an undefined [01:17] <CIA-36> ffmpeg: result of the shift by 24 if bit 7 is set. This affects [01:17] <CIA-36> ffmpeg: the rm demuxer. [01:17] <CIA-36> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [01:18] <CIA-36> ffmpeg: 03Janne Grunau 07master * rd14d4d982c 10ffmpeg/tests/fate/aac.mak: [01:18] <CIA-36> ffmpeg: aacdec: add more fate tests covering SBR and PS [01:18] <CIA-36> ffmpeg: Add all seven test bitstreams of Coding Technologies "aacPlus Decoder [01:18] <CIA-36> ffmpeg: Check Package". The streams cover different ways to signal SBR and PS [01:18] <CIA-36> ffmpeg: in different formats. [01:18] <CIA-36> ffmpeg: 03Janne Grunau 07master * r117e2a30f2 10ffmpeg/libavcodec/ (pthread.c thread.h): [01:18] <CIA-36> ffmpeg: frame-mt: return consumed packet size in ff_thread_decode_frame [01:18] <CIA-36> ffmpeg: This is required to fulfill avcodec_decode_video2() promise to return [01:18] <CIA-36> ffmpeg: the number of consumed bytes on success. [01:18] <CIA-36> ffmpeg: 03Rafaël Carré 07master * rbe1e872582 10ffmpeg/configure: [01:18] <CIA-36> ffmpeg: configure: Store vda lib flags in extralibs instead of ldflags [01:18] <CIA-36> ffmpeg: This way the needed linking flags end up in libavcodec.pc. [01:18] <CIA-36> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:18] <CIA-36> ffmpeg: 03Mans Rullgard 07master * r00a856e3f9 10ffmpeg/libavcodec/ (arm/dca.h dca.c): [01:18] <CIA-36> ffmpeg: dca: ARMv6 optimised decode_blockcode() [01:18] <CIA-36> ffmpeg: This is a hand-tuned version of the code with impossible parts of [01:18] <CIA-36> ffmpeg: the FASTDIV function ommitted. [01:18] <CIA-36> ffmpeg: 2-5% faster overall on Cortex-A8. [01:18] <CIA-36> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [01:18] <CIA-36> ffmpeg: 03Mans Rullgard 07master * rcc276c85d1 10ffmpeg/ (21 files in 4 dirs): [01:18] <CIA-36> ffmpeg: Make channel layout masks unsigned [02:18] <ubitux> mmh i realize we don't have any audio filter to mix multiple streams? [02:18] <ubitux> for instance to add a background music to a video [02:18] <ubitux> (which already has some sound) [02:33] <michaelni> ubitux, shouldnt be hard as long as you dont try to implement AA sync [02:34] <ubitux> AA sync? [02:35] <michaelni> audio audio sync ;) [02:35] <michaelni> like audio video [02:35] <ubitux> ah, ok :) [02:36] <ubitux> i was still on the overlay factoring, seems like i'm tired :p [02:51] <ubitux> michaelni: it might need some audio standardization (number of channels for instance) [02:51] <ubitux> so certainly some usage of swr [02:53] <ubitux> michaelni: btw, about the call for maintainers, maybe you should just add a news to the website [02:53] <ubitux> that might be relayed by some large news website etc [02:54] <ubitux> we lack manpower, so that would hardly make things worse [02:58] <michaelni> ubitux, _very_ good idea [03:36] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r31a1342e7a 10ffmpeg/libswscale/swscale.c: [03:36] <CIA-36> ffmpeg: swscale: remove duplicate code from yesterdays merge. [03:36] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [10:46] <CIA-36> ffmpeg: 03Stefano Sabatini 07master * r28338bc2a3 10ffmpeg/ (7 files in 3 dirs): lavfi: add libass based subtitles renderer [11:00] <ubitux> \o/ [11:00] <ubitux> thx saste :) [11:01] <ubitux> lifesrc now ;) [13:26] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r8f37c8f0f8 10ffmpeg/cmdutils.c: [13:26] <CIA-36> ffmpeg: opt_pix_fmts: try to fix segfault on open solaris [13:26] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:52] <CIA-36> ffmpeg: 03kaptnole 07master * rcb8db6423a 10ffmpeg/libavcodec/aacdec.c: [14:52] <CIA-36> ffmpeg: aacdec: Fix Sound fragments after seeking [14:52] <CIA-36> ffmpeg: Fixes Ticket420 [14:52] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:50] <michaelni> ubitux, Compn bcoudurier cbsrobot do you have ideas for gci tasks ? [15:51] <michaelni> that question is of course not limited to the people above :) [15:53] <michaelni> Hi Anssi_ do you have some idea for a google code in task ? [15:53] <burek> I would suggest setting up wiki and forming a better documentation [15:53] <burek> with examples and stuff [15:54] <burek> students can help with testing mostly, since they are weak as coders (mostly) [15:54] <michaelni> wiki sounds good [15:57] <michaelni> our trac bugtracker comes with a buildin wiki that can easily be used [16:03] <burek> good :) we can also setup some smaller tasks, to get them involved, like to refactor some code or something [16:04] <burek> i mean we can't actually expect them to fix a lot of real bugs, because i fear they have got not enough knowledge for that, i dont know [16:05] <michaelni> yes, bugfixing is unrealistic [16:05] <michaelni> we have fixed all easy bugs already :) [16:06] <burek> I would go for some generic tasks (plus documenting the work while they do it), like: setting up live stream from webcamera to the outter world [16:06] <michaelni> theres the option of fixing warnings but its possibly not worth it, i can probably do it quicker without a student than with one [16:06] <burek> and some similar interesting tasks like that [16:06] <michaelni> this sounds quite interresting [16:07] <burek> yes, I believe the help with coding would more take away developers time, than it would help [16:08] <burek> also, try to see where do you loose a lot of time, while you don't write the code, and see if that everyday stuff can be transfered to someone as a task [16:09] <burek> does ffmpeg have a facebook page? :) [16:09] <michaelni> not that i know [16:10] <burek> well kids are really into that stuff and they could help with it more than we know :) [16:10] <burek> it might promote ffmpeg more, so there would be more coders joining the project? [16:12] <ubitux> michaelni: i don't have easy task in mind, but a lot of possible audio filters :p [16:13] <michaelni> they could go over the various forums like doom9 and see if there are bugs mentiond that are reproduceable with ffmpeg and then put them up on trac [16:13] <ubitux> michaelni: maybe the matrix configuration in libswr? [16:13] <ubitux> the update of the hall of shame? [16:14] <michaelni> you mean finetuning the matrix so itz sounds best ? [16:14] <ubitux> no, allowing the api to configure the matrix [16:14] <michaelni> the hall of shame is dead ... [16:14] <ubitux> hall of shame is dead because no one wants to maitain it, no? :p [16:14] <michaelni> ubitux, these are 13-17 year olds [16:14] <ubitux> yeah it's not easy :( [16:14] <burek> they will be motivated with money prizes and will be monitoring several projects to find the easiest tasks, so we might adapt tasks for massiveness, not for brainiacs? :) [16:15] <ubitux> michaelni: what about adding fate tests? :p [16:15] <ubitux> "find some untested codec/format, and add reg tests" [16:16] <ubitux> might be hard too... [16:16] <michaelni> possible though this will need help from a mentor as the fate system is not trivial to work with [16:16] <burek> they could also create some logos, icons, brandings, etc. [16:17] <ubitux> add a RSS stream generation from the news on the website? [16:17] <michaelni> burek, right i forgot that, we could use modified logos for each day of the year like google is using [16:17] <ubitux> that might require some thinking to change the way news are stored [16:18] <ubitux> looking for typo all over the code [16:18] <ubitux> and doc, and website [16:18] <burek> was the ffmpeg's webpage built using cms or in notepad? :) [16:18] <michaelni> run a spellchecker over the docs (and report the segfaults ;)) [16:19] <ubitux> :D [16:19] <burek> :) [16:19] <ubitux> "fix some warnings:" [16:20] <michaelni> burek, website uses a simple Makefile + very simple pieces of html [16:20] <burek> well, adding rss support can be added as a task then, because it would be trivial thing :) [16:20] <ubitux> no [16:21] <ubitux> you need to change the way news are stored [16:21] <burek> does it use db and why no? :) [16:21] <michaelni> no db [16:21] <burek> i see, file based only [16:21] <michaelni> just files + header + footer IIRC [16:22] <burek> even no php? :) [16:22] <ubitux> news = [{'date': ..., 'content': '<p> ... </p>'}, {...}, ...] [16:22] <ubitux> then generate index + rss based on this is a possibility [16:22] <burek> i see [16:22] <ubitux> it will require python support (for instance, could be perl or sth) on server side [16:23] <ubitux> but no cgi, just for generating static content [16:23] <ubitux> not sure how realistic it is [16:24] <burek> well it's better to remain that way (static) because a lot of people visit it [16:24] <burek> so the server is not that much loaded [16:24] <burek> overloaded* [16:24] <ubitux> mmmh maybe update the old FFmpeg tutorial? [16:25] <burek> i would go with docs and easy stuff, to get a lot of them involved, waiting for easy points and a little bit later adding more serious tasks [16:25] <ubitux> http://www.inb.uni-luebeck.de/~boehme/using_libavcodec.html [16:25] <burek> do tasks have difficulty (num of points)? [16:25] <ubitux> this could be updated to the current API [16:25] <michaelni> theres a difficulty but i dont know if it affects points [16:26] <ubitux> and merged into the main documentation at some point? [16:26] <burek> ubitux, this would really be a valuable part of ffmpeg's webiste [16:26] <ubitux> mmh, i was thinking of this one actually: http://dranger.com/ffmpeg/ [16:26] <burek> a lot of people are coming to the #ffmpeg asking for such support [16:26] <ubitux> yes [16:27] <ubitux> and those tutorials are old, and not maintained [16:27] <ubitux> syncing them with upstream and adding them to the doc could be nice [16:27] <michaelni> i fear a ffmpeg developer would be more qualified to update these tutorials than a high scool student [16:28] <burek> it would be perfect if you would use an auto generator for docs, just by using comments in the source code, like java is practicing, but i guess that's too big of a change [16:28] <ubitux> michaelni: building the examples and fixing the deprecated warning could a good start, and that looks accessible to me [16:28] <michaelni> burek, you mean ? http://ffmpeg.org/doxygen/trunk/index.html [16:28] <burek> this all could go into wiki and is really needed [16:29] <ubitux> mmh about this: maybe integrate the doxy to the website better? [16:29] <ubitux> (merge css, ...) [16:29] <michaelni> fixing some deprecated warnings is easy others is not [16:29] <ubitux> (add main website header to it, etc) [16:29] <michaelni> i mean wrapers that have to contain the deprecated stuff ;) [16:29] <burek> michaelni yes :) that's what I'm talking about, just each function would have comment above itself, to explain what it does and why/how [16:30] <burek> and when you change it, you change the comments accordingly [16:30] <burek> and you have your docs updated [16:30] <michaelni> the doxy linked above should change when the codes comments changes [16:30] <burek> yes [16:30] <burek> that's the idea :) [16:31] <burek> also merging of such things can easily be done just using dns subdomains for ffmpeg.org [16:31] <burek> to avoid constantly updating main website [16:31] <burek> they could help with website a lot [16:32] <ubitux> btw, what's the state of the forum? :p [16:32] <ubitux> no work available on it? :) [16:32] <burek> well, I don't receive any more emails about it, so I don't know really [16:34] <burek> I've sent the backup of the forum at my server to Kyle, and haven't heard from him since then [16:34] <michaelni> last mail i got was 16 Nov 2011 [16:34] <burek> me too [16:37] <burek> michaelni, this is a good example of doxygen: http://xerces.apache.org/xerces-c/apiDocs-3/classDefaultHandler.html [16:37] <burek> for each function there is a brief description of usage and parameters [16:42] <michaelni> http://ffmpeg.org/doxygen/trunk/group__lavu__tree.html [16:42] <michaelni> i think the configuration may need work (its in main git) [16:43] <burek> this is good also [16:43] <burek> is it accessible directly from the main webpage? [16:43] <michaelni> but it should be shown here too: [16:43] <michaelni> http://ffmpeg.org/doxygen/trunk/tree_8c.html [16:43] <michaelni> theres a link on http://ffmpeg.org/documentation.html [16:43] <burek> if you are already using it, its great then :) a lot less work then :) [16:44] <burek> excellent :) [16:48] <burek> one more thing, is there a tutorial or something on how to setup the evnironment for a new developer, that wants to contribute to the project by writing patches and stuff [16:49] <michaelni> not that i know also its very different between platforms [16:50] <michaelni> it would be a good idea for a task [16:50] <burek> do you guys use eclipse or something else? [16:53] <kierank> no [17:01] <burek> i think the major problem why people don't join too often as developers is that they find it very difficult to just setup the evnironment to make even a slightest code change [17:01] <burek> so, tutorial on that would be highly appreciated [17:02] <michaelni> yes i agree [17:03] <burek> also one of the tasks could be that users show what all can they do using ffmpeg, and the most exotic examples could be awarded as a task done [17:04] <burek> just to later showcase the ffmpeg's features and usability [17:04] <michaelni> the idea is good but i dont know if it can be fit into the gci framework [17:05] <michaelni> i think (but am not sure) that each task is first claimed by a student and then work submited and the mentor must judge if it passes needs amendment or fails [17:05] <michaelni> so multiple and take best wont work ... [17:06] <burek> i see [17:06] <michaelni> i also think (but am not sure) that each task can only be finished once [17:07] <nevcairiel> you can have the same task more then once, though [17:07] <michaelni> but you have to create the taskl multiple times or am i misisng something ? [17:07] <nevcairiel> yes, you have to create it multiple times, but i think there was some bulk creation thing [17:08] <michaelni> yes, j-b posted a link but it didnt work for me [17:08] <michaelni> i also googled and found the same link but that still didnt work :( [17:16] <michaelni> btw, nevcairiel, do you want to help mentoring gci ? [18:33] <ubitux> about adding "smovie" in the src movie filter, it should be possible add a "AVSubtitle" source, but i think it would make sense to actually raise frames which could be blended already [18:34] <ubitux> the issue is, smovie must be aware of the final WxH [18:34] <Compn> michaelni : some documentation on creating 3d (side by side) videos using ffmpeg would be nice. someone gave me a nice command line but i havent had time to test it and create docs [18:35] <ubitux> would it make sense to raise AVSubtitle, and make filters like overlay the ability to render and blend it? [18:35] <Compn> let me dig up the mail [18:35] <ubitux> i'm not sure where to put the rendering [18:35] <Compn> michaelni : http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2011-November/116890.html [18:36] <Compn> america is tricking lots of people into buying 3d televisions, now they have to figure out which formats those tvs support, and how to make those formats :D [18:36] <ubitux> smovie filter raises AVSubtitle, then you would have a "burn" subtitle filter, a "text" subtitle filter (print on stdout? freetype rendering?), etc.? [18:37] <ubitux> mmh maybe i'll do that. [18:37] <ubitux> </monologue> [18:43] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r61c2cec957 10ffmpeg/libavcodec/pthread.c: [18:43] <CIA-36> ffmpeg: pthreads: fix segfault due to the thread beimg killed before it has been allocated [18:43] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:43] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r1b84e06244 10ffmpeg/tools/lavfi-showfiltfmts.c: [18:43] <CIA-36> ffmpeg: lavfi-showfiltfmts.c: fix handling of null names [18:43] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:57] <Compn> http://www.guardian.co.uk/commentisfree/cifamerica/2011/nov/11/us-justice-d… [19:30] <CIA-36> ffmpeg: 03Clément BSsch 07master * rc1ae524e2b 10ffmpeg/libavfilter/avfilter.h: avfilter: fix AVFilterPad video only comment. [19:32] <kierank> Compn: we discussed the 3d thing at vdd. there are a lot of issues to deal with [19:34] <Compn> vdd ? whuh? [19:34] <Compn> oh videolan days ? [19:35] <Compn> kierank ? [19:36] <ubitux> i think so [19:37] <kierank> yes [19:39] <Compn> what 3d things did you discuss and is vdd videolan dev days ? [19:40] <Compn> if you feel like sharing [19:40] <kierank> yes vdd = videolan dev days [19:40] <kierank> there are a bajillion different ways of implementing 3d [19:44] <Compn> ah [19:44] <Compn> yes [19:45] <Compn> luckily mplayer has stereo3d filter :P [19:50] <michaelni> and ffmpeg has libmpcodecs wraper that supports stereo3d :) [19:57] <kierank> well demuxing 3d blu-rays is not trivial [19:57] <kierank> since you have two streams which need to be fed to the same decoder [19:58] <kierank> it is not at all clear how you do that [19:58] <kierank> how one could do that i mean [19:59] <Compn> oh yeah, interlaced bd 3d support [20:01] <kierank> mvc support isn't that hard [20:01] <kierank> the demuxing is hard [20:01] <Compn> i wonder if its more or less harder for sending to a hardware decoder [20:02] <michaelni> the additional data could be feeded in by using side data of AVPackets maybe [20:20] <iive> kierank: so the 3d is done by separate streams? [20:26] <Compn> iirc, yes [20:26] <Compn> within the same file [20:27] <kierank> yes within the mpegts [20:27] <nevcairiel> mvc is two or more streams, one "base view" which is just default AVC, and one or more MVC streams for the other views (for 3D, just one) [20:28] <nevcairiel> for decoding, the decoder needs to get both streams [20:41] <michaelni> are there some interleaving requirements so that one can expect the ts demuxer to spit out packets from related streams in sequence ? [20:55] <iive> so you need special decoder too? it doesn't work with 2 separate instances of standard decoder? [20:59] <nevcairiel> thats right [21:00] <kierank> there are dts requirements that should be compliant [21:00] <kierank> but lavf's demuxer doesn't follow dts [21:01] <Compn> do we have some 3d bd samples for testing? :) [21:12] <ubitux> michaelni: i think the current about page is likely to belong to the documentation one [21:12] <ubitux> or maybe home page presenting the project, but that index is already quite filled up [21:13] <michaelni> feel free to move it if you find a fitting spot, we also could keep it below the new one [21:14] <ubitux> yes maybe below [21:15] <ubitux> i'd split the list btw [21:15] <Compn> michaelni : did you make a new task for google code in about 3d documentation? :) [21:15] <ubitux> "FFmpeg provides various tools: [tools list] and developers libraries: [lib list] [21:15] <ubitux> " [21:16] <michaelni> Compn, join as mentor and add a few tasks :) [21:16] <ubitux> it should fit well in the about page with your introduction [21:17] <michaelni> yes, ill see what i can come up with and post that then [21:17] Action: Compn busy planning for his trip to florida [21:26] <ubitux> michaelni: "Whereever" while you are at it :p [21:27] <ubitux> oh and "dependancies" "dependencies" [21:28] <Compn> forget that, make a post-commit hook to run spell check on it :P [21:28] Action: Compn runs [21:29] <michaelni> Compn, we dont have enough diskspace for the output [21:44] <ubitux> hey btw, what does "FFmpeg" means? Fast Forward MPEG? wouldn't that belong to the About page too? [21:47] <michaelni> sounds like that should be in a seperate patch [21:49] <Compn> hehe [21:50] <Compn> are we sure we can have 'mpeg' in the name ? [21:50] <Compn> motion pictures experts group might not like it :P [21:51] <Compn> what i mean is [21:51] <Compn> fast forward MPEG is still an acronym [21:51] <Compn> "fast forward motion picture experts group" ?? [21:52] <ubitux> any video? any media* [21:52] <Compn> not that i want to bikeshed this [21:53] <Compn> sure media works too :P [21:54] <ubitux> it's more correct :p [21:54] Action: Compn still waits for vivo support [21:55] Action: Compn will have to learn to reverse engineer or something [22:01] <Daemon404> i thought vivo was done [22:01] <Daemon404> or close to it [22:01] <Daemon404> according to wiki [22:06] <Compn> i think drv had some patches, forgot the current status [22:07] <Compn> every few months i bug everyone about it [22:08] <Compn> i am the most unhelpful person here :) [22:23] <drv> i have a patch that's almost ready, but it still needs some cleanup [22:23] <drv> i will get to it one of these decades [22:58] Action: ubitux wonders if we will have -af some day :( [23:00] Action: michaelni wants filters that can have mixed media in and out for audio vissualization and all that [23:02] <ubitux> i'm working on it right now :) [23:02] <ubitux> but it will be a hack at first :( [23:15] <michaelni> the only things that are perfect at first are the ones that are never finished :) [23:20] <michaelni> and perfect implies finished thus to fullfill the statement there can be nothing perfect. [23:24] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r4fd5e7639b 10ffmpeg/libavcodec/vc1.c: (log message trimmed) [23:24] <CIA-36> ffmpeg: vc1: Fix pic_header_flag=0 (SA10143.vc1) [23:24] <CIA-36> ffmpeg: Bug introduced in: [23:24] <CIA-36> ffmpeg: commit 4509be3d2f46a52ada8e2ecb476faed93e19abf3 [23:24] <CIA-36> ffmpeg: Author: Michael Niedermayer <michaelni(a)gmx.at> [23:24] <CIA-36> ffmpeg: Date: Tue Oct 11 11:56:42 2011 +0200 [23:24] <CIA-36> ffmpeg: vc1: reset interlaced variables, prevent another bunch of crashes. [23:24] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r22cb8e7b34 10ffmpeg/tests/fate/aac.mak: [23:24] <CIA-36> ffmpeg: fate: enable new sbr tests after our rsync server has them now. [23:24] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:50] <ubitux> michaelni: :) [00:00] --- Sun Nov 27 2011

1 0

ffmpeg.log.20111126
by burek 27 Nov '11

27 Nov '11

[00:00] <pythonirc1011> preset didnt work for me [00:00] <pythonirc1011> lemme show you my setting... [00:00] <burek> can you please use pastebin.com, to show your command line and its output? [00:01] <pythonirc1011> http://codepad.org/90RnTcyw [00:01] <pythonirc1011> for some reason, presets dont work well on windows, nor does libfaac [00:01] <burek> put d:\Videos\00019 [00:01] <burek> .MTS [00:02] <burek> into quotes [00:02] <pythonirc1011> it works, without quotes...I can change that for sure. [00:02] <burek> it confuses somehow things, i donno [00:02] <burek> all those errors seem to be because of that [00:03] <pythonirc1011> I'm not getting any errors [00:03] <pythonirc1011> sorry , please ignore that "output" on codepad [00:03] <pythonirc1011> thats not the output of my execution [00:03] <pythonirc1011> I'm running some batch conversions right now, which will take some time...so cant paste the output yet [00:04] <pythonirc1011> burek: do you see any obvious errors in the command that i should look into/fix/tune? [00:04] <burek> i would use -threads 0 [00:04] <burek> to let ffmpeg autodetect No. of threads [00:05] <pythonirc1011> well with threads 8 it takes too much time for me...on my 8 core machine [00:05] <pythonirc1011> ah ok [00:05] <burek> its not quite 8 [00:05] <pythonirc1011> will change that...didnt know 0 meant that [00:05] <burek> you have 1 decoding and 8 encoding threads [00:05] <burek> something like that [00:05] <burek> so you miss one core [00:05] <burek> -threads 0 is the safe way to do it [00:05] <pythonirc1011> k, will change that for sure then [00:06] <pythonirc1011> anything else? [00:06] <burek> you see -cmp +chroma ... etc [00:06] <burek> you should use that only if you are expert [00:06] <burek> and know what you're doing [00:06] <burek> for all the others, use -preset [00:06] <burek> its much better tuned [00:06] <burek> and -tune too [00:07] <pythonirc1011> I found that on the web by someone who had done it...for converting between avchd and mpeg4 [00:07] <burek> well ok [00:07] <pythonirc1011> I cant use presets on windows for some reason, so had to go with that [00:07] <burek> well, give me the output, and I'll tell you why you can't [00:07] <burek> and how to fix it [00:07] <pythonirc1011> is there a "-tune" that i should add? [00:07] <pythonirc1011> perfect...give me some time. I'll show you a pastebin [00:07] <burek> ok [00:09] <pythonirc1011> http://codepad.org/IAtIA2Yo [00:09] <pythonirc1011> For some reason adding -threads 0 started giving me errors [00:10] <burek> hm, why do you reencode h264 video to h264 video? [00:10] <pythonirc1011> burek: here are the errors with thread 0 : http://codepad.org/dRFqSeej [00:11] <pythonirc1011> burek: The MTS file does not play with windows media player...the output mpeg4 does -- have no clue whats the difference [00:11] <burek> did you try ffmpeg -i input.ts -vcodec copy -acodec copy output.mp4 [00:11] <pythonirc1011> yes i did [00:11] <burek> and? [00:11] <pythonirc1011> windows media player does not understand the output [00:12] <pythonirc1011> just a black screen with audio [00:12] <burek> wmp is just a direct show player [00:12] <burek> meaning [00:12] <burek> it doesnt understand anything [00:12] <burek> if you dont [00:12] <burek> have codec installed [00:12] <pythonirc1011> when i play an mts file -- no video -- only audio [00:12] <burek> can you download vlc for windows [00:12] <pythonirc1011> when i play the mpeg4 after encoding (with the long command) -- both audio / video play well [00:12] <relaxed> why are you setting those options and not using a preset? [00:12] <pythonirc1011> vlc plays MTS and output without problems [00:12] <burek> well use VLC then [00:13] <pythonirc1011> relaxed: because on windows compiles, preset doesnt work [00:13] <pythonirc1011> burek: If WMP cant play a video, other video applications have trouble rendering it too [00:13] <relaxed> '-preset veryslow' doesn't work? [00:13] <burek> do you understand how wmp plays videos? [00:13] <pythonirc1011> relaxed: nope [00:14] <burek> it uses installed codec (encoder/decoder) [00:14] <burek> if that installed version of codec is old/wrong/bad.. [00:14] <burek> wmp wont be able to play it [00:14] <burek> the point is, your wmp might not play that video [00:14] <burek> but mine will [00:14] <pythonirc1011> burek: where can i get a codec for MTS files? I looked on the web...this is what i found as a solution...use ffmpeg to transcode [00:14] <burek> so its not a good tool to measure the correctness of a video [00:15] <burek> well, lets go a more precise way [00:15] <pythonirc1011> burek: according to your theory, my MTS should play, since h264 codec clearly is on my system? [00:15] <burek> what do you want to accomplish in global [00:15] <burek> it is, but what version of it? [00:15] <pythonirc1011> Ultimately, I want to be able to use it in adobe premier [00:15] <pythonirc1011> and be able to change its size if need be [00:16] <burek> you can resize (rescale) your video with ffmpeg too :) [00:16] <pythonirc1011> indeed...hence I was looking into ffmpeg [00:16] <burek> so, once again, whats the ulimate goal [00:16] <pythonirc1011> I cooked up that command so that i can run it on any MTS that the camera captures [00:16] <pythonirc1011> be able to work with the video in adobe premier... [00:17] <burek> then just remux your video into flv container [00:17] <burek> ffmpeg -i ... -vcodec copy -acodec copy out.flv [00:17] <pythonirc1011> i want quality control as well -- my cam is 24mbps -- will flv retain quality? [00:18] <burek> isn't it better to configure the camera's output quality on the camera? [00:18] <pythonirc1011> burek: for some reason my machine doesnt render that output [00:18] <pythonirc1011> flv or mp4 -- copy [00:19] <burek> update your x264 [00:19] <pythonirc1011> how? [00:19] <pythonirc1011> i talked to canon tech support [00:19] <burek> i can bet adobe's site can tell you that :) [00:19] <pythonirc1011> they said their videos dont play on windoez [00:20] <burek> but you said, it plays in vlc? [00:20] <burek> so the support might be wrong [00:20] <pythonirc1011> absolutely -- vlc renders everything... [00:20] <pythonirc1011> its not using win 7 codecs [00:20] <burek> god bless for that :) [00:20] <relaxed> pythonirc1011: you get get recent ffmpeg builds here that support -preset. http://ffmpeg.zeranoe.com/builds/ [00:21] <pythonirc1011> I agree, but the rendering is flickery...24Mbps is too much for even VLC [00:21] <relaxed> s/get get/can get/ [00:21] <pythonirc1011> thats where i downloaded mine from [00:21] <pythonirc1011> btw, most of those builds crash! [00:21] <burek> well, you're on win 7.. [00:21] <burek> what did you expect :( [00:21] <pythonirc1011> yes [00:21] <relaxed> oh? I don't use them [00:22] <pythonirc1011> burek: All I care about is to be able to get my videos in a format that win 7 understands without loosing quality [00:22] <burek> now we are talking things :) [00:22] <wolfman2000> alright, I'm doing something wrong. I figured all I needed to do was copy the static libavcodec.a and libavformat.a to my project, but that isn't doing it. I then wondered if I had to include the associated header files, but now I'm given what seem to be silly errors such as 'avcodec_decode_video' is not a member of 'avcodec'. What is the proper way to use the static libraries in a project again? [00:22] <pythonirc1011> and i wont take a russian codec pack on my machine :) [00:22] <burek> so your final goal isnt to be able to load video into adobe premiere, but to be able to play it in win 7? [00:22] <pythonirc1011> both actually... :) [00:23] <pythonirc1011> burek: whats really going on -- so MTS is also h264 and ffmpeg output is also h264 -- its just a version difference because of which the MTS doesnt render? [00:23] <burek> wolfman2000, i think your problem exceeds ffmpeg [00:23] <burek> you should google more for static linking [00:24] <wolfman2000> burek: I know about --enable-static. What else do you think I'm missing? [00:24] <burek> pythonirc1011, im not sure how to compare two h264s [00:24] <burek> you can maybe use mediainfo [00:24] <burek> or some other tool [00:24] <burek> but most probably its a different profile or something [00:25] <pythonirc1011> burek: When a camera manufacturer claims that their videos are 24Mbps -- should i set -b 24000k? [00:25] <pythonirc1011> or is there something like a VBR for best quality? [00:25] <burek> wolfman2000, did you try google first? [00:25] <wolfman2000> yes. so far, not helping [00:25] <burek> pythonirc1011, yes.. -crf [00:26] <pythonirc1011> burek: whats wrong with the command that i used...it converts my h264 that win7 doesnt understand to h264 that it does understand? [00:26] <ghostbar_> burek, got it, I'm trying right now [00:26] <burek> pythonirc1011 i told you already, it probably changes profile of h264 video [00:26] <burek> and your h264 can read that profile, but not your camera's [00:26] <burek> you should in any case update your h264 win codec [00:27] <pythonirc1011> burek: if you know where i can find a good h264 codec for win 7, please let me know.. :) [00:27] <pythonirc1011> Also: why does -threads 0 give these errors: http://codepad.org/dRFqSeej [00:28] <burek> well, im not sure, but i would try google for those errors and figure out why :) [00:29] <burek> wolfman2000, i just tried: http://www.google.com/search?q=ffmpeg+static+linking+example [00:29] <pythonirc1011> burek: i would be ok with WMP not liking MTS videos, but what i dont like is premier not liking them...the preview pane is pain to work with with mts [00:29] <burek> and I've found several links which are promissing [00:29] <burek> http://soledadpenades.com/2009/11/24/linking-with-ffmpegs-libav/ [00:30] <burek> well [00:30] <burek> maybe you have problems with how is it called in windows.. [00:30] <burek> not splitter.. [00:30] <burek> hmh [00:30] <wolfman2000> I'm on Mac, now Windows [00:31] <burek> filter i guess [00:31] <burek> oh sorry, i was answering pythonirc1011 :) [00:31] <burek> anyway, demuxer, which gets the video stream out of the container (flv, mp4, avi) [00:32] <pythonirc1011> burek: vlc sucks with the 24Mbps videos on my machine....some frames render after a second! [00:32] <pythonirc1011> another reason i've to move these videos to another format [00:32] <burek> pythonirc1011, tools - messages (log level = 2) [00:32] <pythonirc1011> or get a codec that works [00:32] <burek> and see what does it complain about [00:33] <burek> well, ok, then loose -threads completely [00:33] <burek> and try like that [00:33] <burek> of course, use -preset and -tune [00:33] <burek> otherwise, don't complaint for bad quality [00:33] <pythonirc1011> burek: http://paste.pocoo.org/show/512988/ [00:34] <pythonirc1011> thats from vlc [00:34] <burek> your cpu can't manage to play it that fast [00:34] <burek> i would try to tune the camera [00:35] <pythonirc1011> burek: If an i7-2600K 4GHz CPU cant play a video...Its not the CPU...its VLC :) [00:35] <burek> its 24mbps.. [00:35] <pythonirc1011> yes...and i want to keep it 24Mbps 1080p [00:36] <pythonirc1011> sucks : adobe discontinued its media player! :( [00:37] <burek> fun never ends :) [00:37] <pythonirc1011> canon and microsoft fighting...consumer gets killed... [00:38] <pythonirc1011> burek: does -crf take an argument? [00:39] <pythonirc1011> whats the float? what value should i use? [00:41] <burek> try -crf 2- [00:41] <burek> try -crf 20 [00:41] <pythonirc1011> trying 25 :) [00:42] <pythonirc1011> looks like its rendering 3Mbits [00:43] <pythonirc1011> burek: is there a way to tell vlc to use hw acceleration? [00:43] <pythonirc1011> got it [00:43] <pythonirc1011> use gpu acceleration [00:43] <pythonirc1011> trying it now [00:44] <burek> try asking in #videolan [00:44] <pythonirc1011> plain flat green video with gpu acceleration on :) [00:45] <pythonirc1011> burek: what does the number "-crf 25" stand for? [00:47] <ghostbar_> burek: it didn't worked. It grows the number to the infinite, not just until 9 :-( [00:55] <pythonirc1011> burek: what does CRF really mean? At 15 it seems it encodes at 20Mbps+ levels. [00:55] <burek> pythonirc1011, try x264 --help and -fullhelp :) [00:56] <burek> ghostbar_ :( [01:23] <darkstarbyte> Are the -async and -vsync options important, and if so how should I use them? [01:37] <burek> http://ffmpeg.org/ffmpeg.html [01:46] <darkstarbyte> thanks [01:46] <darkstarbyte> Would anyone know where I would go to get help with mkisofs? [01:55] <sacarasc> :| [01:55] <ubitux> :| [01:55] Action: burek sets mode: +beer sacarasc [01:55] Action: burek sets mode: +beer ubitux [01:56] <burek> feel any better? :) [01:56] <ubitux> i don't drink alcohol [01:56] <ubitux> gimme apple juice [01:56] <burek> thats what we all say :))) [01:57] <ubitux> i really don't like alcohol actually :p [01:58] <teratorn> any example code around of transcoding with a frame-rate change? [02:01] <makario> Is it possible to take a certain video's audio track and mute specific moments in audio? [02:03] <burek> teratorn what exactly do you want to accomplish [02:03] <burek> makario, i think not [02:03] <burek> you can extract certain parts though [02:03] <makario> burek, darn. do you know of any other way to do that? basically i've been given the task of creating a quick 'swear word' muter [02:03] <teratorn> burek: I'm transcoding, and I want to go from source frame-rate to a fixed, lower frame-rate [02:04] <burek> skipping unwanted ones [02:04] <burek> makario, using voice recognition? [02:04] <ubitux> makario: extract the audio track, edit the audio with sth like audacity, and remux the stream [02:05] <makario> burek, ubitux: I need to do it programatically using the subtitles track [02:05] <burek> teratorn: ffmpeg -i input -r 25 ... [02:05] <ubitux> then you might want to create an audio filter [02:05] <ubitux> but this is not a simple task [02:05] <makario> ubitux: audio filter? [02:05] <teratorn> burek: yeah, it's just somewhat of a task tracing around in ffmpeg.c trying to figure out how it does its magic some times :) [02:05] <ubitux> makario: do you know how to write in C? [02:06] <makario> ubitux: yes, kind of. wrote a gba game, but have never done anything for a desktop (or using ffmpeg) [02:07] <makario> ubitux: point me in the right direction and i'll bang my head against it until i figure it out [02:07] <ubitux> that will require some time (let's say a few days) to make it i guess [02:07] <ubitux> look at the various libavfilter/af_*.c files [02:07] <makario> ubitux: okay, cool. [02:07] <ubitux> libavfilter/af_pan.c is the most recent one [02:08] <ubitux> it's a filter which allows you to "levelize" the channels [02:08] <makario> are all of these in the source? [02:08] <ubitux> yes [02:08] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavfilter/af_pan.c;hb=HEAD [02:09] <teratorn> ubitux: do you know what technique is used to lower the frame-rate? does it just drop frames here and there? [02:10] <ubitux> no i don't know [02:10] <makario> ubitux, yeesh [02:11] <makario> it'll be a few weeks, i think [02:11] <teratorn> ubitux: k, thanks, I'll just try to decipher ffmpeg.c later [02:11] <ubitux> makario: git show 1fbf7165d59907a0632f8b72664a31f97f218656 for the complete way of adding the filter [02:11] <burek> teratorn [02:11] <makario> thanks [02:11] <burek> you can look at the -vsync too [02:12] <teratorn> erm, I mean to address those question to burek :/ [02:12] <teratorn> burek: OK [02:13] <burek> oh you mean how does it resample the video [02:14] <burek> well isn't it obvious :) if the input rate is lower, it duplicates frames [02:14] <burek> if higher, it drops [02:15] <ubitux> makario: a simple way of doing it is to make a filter which takes ts ranges; something like silent=3-4.5,10.3-15.7,... [02:15] <ubitux> then when the pts is in this range, you just put 0 [02:15] <ubitux> otherwise you copy the data [02:16] <ubitux> that should do the trick [02:16] <ubitux> if you suceed in it, you can send it to ffmpeg-devel(a)ffmpeg.org for review [02:16] <makario> ubitux, yeah, i think that's the plan. [02:16] <ubitux> so it can be added [02:16] <makario> all right, cool [02:16] <ubitux> good luck :) [02:17] <makario> very much appreciated! [02:17] <ubitux> you can ask for help on #ffmpeg-devel for that matter btw [02:19] <makario> will probably do [02:19] <makario> now to read up! [02:26] <ronag> why does a filter graph with yadif choose to convert yuv422p10 to yuv420p instead of yuv420p? [02:26] <ronag> *instead of yuv422p [02:45] <burek> can you please use pastebin.com, to show your command line and its output? [02:50] <Nagy> not commandline, code: http://pastebin.com/AHjMM3Gm [02:53] <Nagy> this the log http://pastebin.com/FkYUabF1 [02:53] <Nagy> for some reason it does fmt:yuv422p10le -> w:720 h:486 fmt:yuv420p [02:54] <Nagy> instead of what I would want: fmt:yuv422p10le -> w:720 h:486 fmt:yuv422p [02:54] <Nagy> I'm not sure how libavfilter chooses formats, but that would be the logical conversion, wouldnt it? [04:15] <jimlkl> Pasteeater....I figured out the problem I was having with certain videos being put on my iPod which had the sides cut off of them. [04:17] <jimlkl> Well, if he's not here I'll tell you all- the iPod video setting of "fit to screen" had to be selected as OFF. [04:31] <pasteeater> e [04:32] <pasteeater> ...i know this isn't my terminal, damnit. [05:05] <o3u> hi, how can i stream multiple files in a row, and do processing on each, at the same time? Right now i'm able to stream a single file, when i try to do consecutive calls to ffmpeg it doesnt seem to work, is there a way to do this? [05:14] <johnnydude> hey guys, i'm trying to encode a dvd .VOB to webm, resize the video so its small and get the video to be about 5mb... [05:16] <johnnydude> for some reason the file keeps coming out massive [05:16] <sacarasc> Set -v:b X [05:16] <sacarasc> Where X is the right bitrate to make it 5MB. [05:17] <sacarasc> Depends on how long the video is, though. [05:17] <johnnydude> alright cool let me try... [05:18] <sacarasc> I think it's -v:b. These new changes confuse me sometimes. [05:18] <johnnydude> i used some other tool to make a mp4, and it came out around 5mb with perfect quality... video is only 5min [05:18] <johnnydude> hmm well, i was trying like -b and other stuff, but it didnt seem to apply [05:19] <johnnydude> or if it did, not how i was expecting, im pretty new to this [05:20] <johnnydude> ffmpeg.exe -i D:\VIDEO_TS\VTS_01_1.VOB -f webm -s 480x384 -v:b 120 output3.webm [05:20] <johnnydude> and it crashes when i start it [05:20] <sacarasc> What does it say? Use pastebin.com for a large paste. [05:21] <johnnydude> well im on windows atm, so i just get a windows error exception thing [05:22] <johnnydude> w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: [scale @ 032DE900] w:720 h:576 fmt:yuv420p -> w:480 h:384 fmt:yuv420p flags:0x4 [libvpx @ 032DDF20] err{or,}_recognition separate: 1; 1 [libvpx @ 032DDF20] err{or,}_recognition combined: 1; 65537 [libvpx @ 032DDF20] v0.9.7-p1 [05:23] <johnnydude> my god, i thought this was going to be easy [05:23] <johnnydude> spent like 12 hours just trying to convert a video loll [05:25] <johnnydude> ooh is it supposed to be "-vb" isntead of "-v:b" ? [05:26] <sacarasc> I think it depends on version. [05:27] <sacarasc> And it should be -b:v anyway. :D [05:27] Action: sacarasc blames the cold. [05:27] <johnnydude> im using some static git win32 version [05:29] <johnnydude> alright, i do -b:v 160 [05:29] <johnnydude> but then it encodes at around 120? [05:30] <johnnydude> do you know where the man page for latest ffmpeg is? since that switch isnt in what im reading at all.. [05:32] <sacarasc> http://ffmpeg.org/ffmpeg.html [05:33] <johnnydude> damn running out of time.. gonna get fired cos im too retarded to encode a video lol [05:33] <johnnydude> yeah thanks man :) [05:36] <johnnydude> yeah i dont think -b:v -v:b or -bv are doing anything [05:37] <johnnydude> ooh i see, i have to put a k on the end [05:40] <o3u> any help on the streaming question? anyway to do it straight from ffmpeg? [05:40] <o3u> imma try using a pipe, but it'd be nicer if i could do it with ffmpeg straight up [06:42] <pasteeater> o3u: stream to what? "when i try to do consecutive calls to ffmpeg it doesnt seem to work" can you show your command(s) and output(s) on a pastebin site? [06:56] <o3u> stream it to ffserver [06:57] <o3u> pasteeater: i cant seem to get pipes working with that either [08:38] <o3u> concatenation sort of worked by using a pipe, then using ffmpeg to read that pipe to the server, and several other instances consecutively writing to the pipe, however i get a 'rc buffer underrun 'now, [08:38] <o3u> underflow rather [08:42] <o3u> here are my terminals: http://pastebin.com/Qh8WbwQj [10:16] <EOF-sensei> how would one mux 6 channels of flac from separate files into an mkv? [10:17] <EOF-sensei> (as well as encode a sequence of png files to dirac) [10:58] <burek> EOF-sensei, use -map [10:58] <burek> and -f image2 [10:58] <burek> more info here: http://ffmpeg.org/ffmpeg.html [11:02] <burek> o3u: TCP connection to localhost:8090 failed: Connection refused [11:02] <burek> read your output more carefully.. [11:04] <EOF-sensei> hmm [11:04] <EOF-sensei> god [11:04] <EOF-sensei> schroedinger really needs optimization work [11:04] <EOF-sensei> and people need to stop wimpering and figure out a way to make wavelets compress well [11:05] <EOF-sensei> </minirant> [11:06] <burek> you can always help by sending such a patch :)) [11:07] <burek> or donating a box of beers or such ^^ [11:12] <EOF-sensei> indeed [11:13] <EOF-sensei> every signal around me telling me I should put on my developer cap and stretch my C/++ muscles again [11:13] <ubitux> and apple juice/ [11:13] <ubitux> oo/ [11:13] <EOF-sensei> I should probably work at finding workable motivators [11:14] <EOF-sensei> I should probably start preemptive patenting [11:15] <EOF-sensei> to avoid problems proving prior art to keep patents free [11:20] <shevy> ffmpeg -ss 10 -i input.mp3 -t 30 new.mp3 [11:20] <shevy> am I reading this right: it says, start at position at 10 seconds, and copy the next 30 seconds into the file new.mp3 [11:20] <shevy> Because somehow, that does not seem to be the case... the resulting mp3 is 40 seconds long [11:21] Action: shevy scratches his head. [11:21] <shevy> hmm it seems also not quite accurate :( [11:22] <EOF-sensei> shevy: re-encoding mp3s is a lossy process [11:22] <EOF-sensei> I suggest using an mp3 clipper [11:22] <EOF-sensei> as it will not further degrade the audio quality [11:22] <shevy> ah, the above command will re-encode? [11:23] <EOF-sensei> I would imagine [11:24] <EOF-sensei> use mp3split [11:24] <EOF-sensei> you won't have issues [11:27] <EOF-sensei> although I suppose mp3split isn't in all distro repositories [11:28] <EOF-sensei> (including my own) [11:28] <EOF-sensei> I wonder why [11:28] <EOF-sensei> patent issues I guess [11:41] <EOF-sensei> is theholyduck everywhere? [11:50] <burek> EOF-sensei no need for other tool, ffmpeg can also do it [11:50] <burek> shevy, you can add -acodec copy [11:51] <burek> and put -ss after -i (to seek after decoding frames) [11:51] <burek> like this: ffmpeg -i input.mp3 -ss 10 -acodec copy -t 30 new.mp3 [12:01] <shevy> hmm [12:01] <shevy> thx! [12:03] <EOF-sensei> I wasn't sure if -acodec copy would split without artifacts at the split points [12:03] <EOF-sensei> good to know there's an option for that [12:03] <EOF-sensei> although [12:03] <EOF-sensei> /ffmpeg getting less unix-y/ [12:05] <shevy> I think I know what's with that file [12:05] <shevy> it seems to not have a proper header [12:06] <shevy> that explains why ffmpeg was reporting wrong duration, I think :) [12:56] <TryDent1> what ffmpeg version fixes the 2 byte bug [12:58] <burek> TryDent1 what 2 byte bug [12:58] <TryDent1> ffmpeg -i music.file output.wav [12:58] <TryDent1> it creates 2 bytes bigger than it should [12:59] <JEEB> what format is the music file? and is it in a container or a raw stream? [12:59] <TryDent1> input or output [12:59] <JEEB> input [12:59] <TryDent1> flac [12:59] <JEEB> ok, not sure about flac [13:00] <TryDent1> if i use flac program it creates exact same file as original.wav [13:01] <ubitux> TryDent1: we already explained to you [13:01] <JEEB> aac and mp3/ac3 and similar lossy formats at least have priming samples which would get decoded by ffmpeg because there's no standard on how to specify how many priming samples were used by the encoder [13:01] <JEEB> (unless it's muxed into a container) [13:01] <TryDent1> jeeb i am only comparing with lossless formats [13:02] <TryDent1> ubitux yes i was told there was a bug [13:02] <ubitux> no there isn't [13:02] <ubitux> it was told to you to report it to ffmpeg & flac if you think it is [13:02] <ubitux> the original flac was certainly made by flac [13:02] <TryDent1> how is it not a bug: it's creating 2 bytes bigger [13:02] <ubitux> so it's no surprise it is the same [13:03] <ubitux> if you remux the flac with ffmpeg, convert it to wav, then convert it back to flac, you will obtain the same [13:03] <TryDent1> what if zip was doing that; creating 2 bytes bigger [13:03] <ubitux> TryDent1: the audio data is the same [13:03] <ubitux> the headers might be different [13:03] <ubitux> just like you can have different zip with the same content [13:04] <TryDent1> ubitux i encoded using ffmpeg -acodec flac and decoded using flac app; and it created exact file as original.wav [13:05] <ubitux> i can create you 2 zip with the exact same content, one with 2 more bytes than the other [13:05] <ubitux> and this is *not* a bug [13:05] <TryDent1> ubutux show me: prove it [13:05] <ubitux> are you really trying to make me waste my time? [13:06] <JEEB> I feel like he's not really understanding what you're trying to say :V [13:06] <TryDent1> because i don't believe zip/rar/etc would have that kind of serious bug [13:06] <ubitux> JEEB: i think he's just trolling so i'm going to do something else :p [13:06] <TryDent1> ubitux i am serious [13:06] <TryDent1> ubitux i want to know about 2 bytes bigger in zip scenario [13:06] <JEEB> ubitux, sometimes it's a good idea to take a pause :3 [13:07] <TryDent1> ubitux i am just perfectionist; i want everything to be exact [13:08] <JEEB> TryDent1, if you do ffmpeg -i input.flac -acodec copy output.flac and then read that, does the resulting output.flac give the same output as input.flac? [13:09] <TryDent1> jeeb okay; let's start with original.wav [13:09] <TryDent1> my goal is original.wav to match output.wav [13:10] <ubitux> TryDent1: http://ubitux.fr/pub/shots/_flacdiff.png [13:10] <ubitux> see? [13:10] <ubitux> only the header is affected [13:10] <ubitux> basically what encoder was used [13:10] <TryDent1> that picture means nothing to me [13:10] <JEEB> ah [13:10] <JEEB> yeah [13:10] <JEEB> you should only compare the audio data in the wav :) [13:10] <ubitux> TryDent1: it's a diff between original flac and remuxed ffmpeg file [13:11] <JEEB> uh, wait [13:11] <ubitux> (with a wav step between) [13:11] <ubitux> again, the data doesn't change [13:11] <ubitux> only the header (meta information and such) change [13:11] <TryDent1> why you comparing flac files? [13:11] <ubitux> that is *not* a bug [13:11] <TryDent1> why not compare original.wav and output.wav [13:11] <ubitux> you were talking about flac. [13:12] <TryDent1> i never said ffmpeg had flac encoding bug; i said ffmpeg has decoding bug [13:12] <ubitux> share you sample [13:12] <ubitux> and the exact procedure [13:13] <TryDent1> okay [13:15] <TryDent1> ubitux i think there is big misunderstanding i was complaining that md5sum or original.wav and output.wav didn't match; i wasn't complaining about flac files [13:16] <JEEB> if you bindiff the files, is the difference in the beginning or end? [13:16] <TryDent1> let me try [13:16] <ubitux> TryDent1: give the exact steps to reproduce, and the sample [13:16] <JEEB> since I don't think that ffmpeg gives a completely different result [13:17] <ubitux> yes [13:17] <ubitux> that's exactly the same issue as flac [13:17] <ubitux> wav has a header too [13:18] <TryDent1> where do i get bindiff [13:19] Action: JEEB doesn't know of any good (visual) binary differs [13:20] <TryDent1> then why are you recommending me to use it [13:21] <JEEB> hell, you can open the file in a hex editor of your choice [13:21] <JEEB> and check the beginning and the end of the file :P [13:22] <burek> if i understand correctly, you encoded 1.wav with flac tool, making 1.flac and now you are decoding 1.flac with ffmpeg to 2.wav, and 1.wav differs from 2.wav? [13:22] <JEEB> yes [13:22] <JEEB> that seems to be the case [13:22] <TryDent1> burek correct but i used ffmpeg to encode [13:22] <burek> and flac to decode? [13:22] <JEEB> and it also differs from 3.wav that comes out of 1.flac done with flac's reference decoder it seems [13:23] <TryDent1> burek if i use flac to decode it matches; if i use ffmpeg to decode it does not match [13:23] <burek> oh i see, so if you do it all with ffmpeg, then 1.wav differs from 2.wav [13:23] <TryDent1> correct [13:23] <burek> can you upload somewhere those 2 wavs? [13:23] <TryDent1> sure [13:24] <TryDent1> but let me use small samples [13:24] <burek> ok [13:24] <JEEB> it's fine [13:24] <JEEB> as long as it replicates [13:24] <TryDent1> burek but if you have a wav file you can do it yourself [13:25] <TryDent1> assuming yuou have both ffmpeg and flac app [13:25] <burek> i dont :( [13:25] <burek> i dont use flac generally [13:25] <burek> only aac+ :) [13:26] <TryDent1> what is aac+? [13:26] <burek> what is aac+ ??? [13:26] <burek> dear god, forgive him [13:26] <burek> :) [13:26] <TryDent1> i know aac but not aac+ [13:26] <burek> its the future :) [13:27] <burek> lets just say that at 32/48 kbs you get "near cd quality" [13:27] <burek> its HE-AAC v2 [13:27] <burek> http://en.wikipedia.org/wiki/High-Efficiency_Advanced_Audio_Coding [13:27] <burek> with sbr and pps [13:28] <TryDent1> HE-AAC is better than LC? [13:28] <burek> you should've heard of it, since youtube for example encodes all its audio with that [13:28] <burek> way better [13:28] <TryDent1> i see [13:28] <burek> take a look at the link i gave you [13:29] <burek> and see the image on the right [13:29] <TryDent1> i had a aac-LTP once and some many players had problem with it [13:29] <burek> where is AAC LC and where is HE-AAC [13:29] <TryDent1> does he-aac v2 have compatibility problem like he-ltp [13:29] <burek> i donno [13:30] <TryDent1> i mean aac-ltp [13:30] <burek> but i know that, when something is so good, the others will converge to it [13:30] <burek> meaning, the manufacturers of audio players will tend to incorporate aac+ into devices [13:30] <TryDent1> what aac encoder support he-aac v2? [13:31] <burek> Scientific testing by the European Broadcasting Union has indicated that HE-AAC at 48 kbit/s was ranked as "Excellent" quality using the MUSHRA scale.[8] MP3 in the same testing received a score less than half that of HE-AAC and was ranked "Poor" using the MUSHRA scale. Data from this testing also indicated that some individuals confused 48 kbit/s encoded material with an uncompressed original. [13:31] <burek> nuff said :) [13:32] <burek> libaacplus (for ffmpeg) [13:32] <burek> its already in the ffmpeg git [13:33] <TryDent1> what file extension does aac+ use [13:33] <TryDent1> Unknown encoder 'libaacplus' [13:34] <burek> you need to ./configure --enable-libaacplus [13:34] <TryDent1> huh [13:34] <JEEB> you have to use the library, configure it and IIRC the encoder did use example implementation's code so I'm not sure if the resulting binary will stay GPL [13:34] <burek> it will not [13:34] <JEEB> (the last point only matters to people who actually give the binaries out tho) [13:34] <burek> exactly [13:35] <TryDent1> does itunes support aac-plus [13:35] <burek> surely it does [13:36] <JEEB> doesn't QT's aac encoder support he-aac anyways? [13:36] <TryDent1> what about nero or faac? [13:36] <JEEB> so you could as well use it [13:36] <burek> JEEB which one is that? [13:36] <JEEB> it has an API, several command line apps use it [13:36] <burek> faac is AAC LC only [13:36] <JEEB> I don't think you can use it within ffmpeg :) [13:36] <burek> nero is good too [13:37] <JEEB> nero has HE-AAC, a bit worse than QTAAC tho IIRC [13:37] <burek> JEEB ok :) i thought you are talking about libvo_aacenc, which also doesnt support he-aac [13:38] <burek> TryDent1, http://tipok.org.ua/node/17 go to Download section, get the file, unpack, configure, make, install [13:38] <TryDent1> i don't remember itunes has settings of changing lc or he-aac [13:38] <burek> and then get ffmpeg from git, configure (with --enable-libaacplus), make, install and thats it [13:39] <burek> well, why would player had such an option? [13:39] <JEEB> you don't need a setting in a decoder [13:40] <JEEB> because the bitstream tells the decoder which type of AAC it is [13:41] <TryDent1> there is no option of LC or he-aac in itunes [13:41] <burek> :) [13:41] <burek> read above [13:41] <JEEB> and there shouldn't be [13:42] <JEEB> or wait [13:42] <JEEB> you mean the encoder? [13:42] <JEEB> not sure if itunes has the settings [13:42] <JEEB> it's meant to be 'dumb' [13:43] <JEEB> http://sites.google.com/site/qaacpage/news/qaacrelease107 [13:43] <JEEB> this uses the QuickTime API [13:44] <JEEB> (cabinet = downloads) [13:44] <TryDent1> wow [13:44] <TryDent1> why is it in google site not in apple site [13:44] <JEEB> because it's not an apple app? [13:44] <JEEB> sites.google is free hosting [13:44] <JEEB> the QuickTime API is free for everyone to use :P [13:45] <TryDent1> then why can't qaac be used with ffmpeg [13:45] <JEEB> because no-one has made a patch? [13:45] <JEEB> d'oh [13:47] <burek> wait what is the diff between libaacplus and that link above (if there was a patch) [13:48] <JEEB> different encoder used [13:49] <JEEB> also, depending on the understanding of how linking to Apple's API is considered GPL-wise (MPC-HC and friends link to it without caring), it might actually be possible to give out binaries that can link to it. [13:49] <burek> so it could be considered GPL? [13:50] <JEEB> LGPL or GPL, the Apple SDK is... [13:50] <JEEB> I think it's BSD or something [13:50] <JEEB> of course it only works on systems that have Apple's libs [13:50] <JEEB> which limits it to Win/Mac [13:50] <JEEB> unless you use wine, that is [13:52] <JEEB> yeah, MPC-HC has the whole SDK in their repo it seems https://github.com/jeeb/mpc-hc/tree/master/include/qt [14:01] <TryDent1> is there big difference between he-aacv1 and he-aacv2 [15:05] <mehmetali> Hi, how i can enable svq3 pixel format. [15:08] <BlackBishop> can I stream by any chance the output of /dev/video0 ( my webcam ) to the internet so it can be viewed by vlc/mplayer or other players ? :) [15:10] <BlackBishop> as in .. is there any easy way ? ( one line thing ) [15:13] <BlackBishop> I'm currently using : mplayer -tv device=/dev/video0:driver=v4l2:input=1:width=1920:height=1080 tv://1 -zoom -aspect 4:3 [15:13] <BlackBishop> but I want it to be able to stream [15:38] <BlackBishop> mhm, trying to use ffserver, bind(port 3343): Address family not supported by protocol [15:43] <BlackBishop> ffmpeg 0.7.7 if it matters :| [16:58] <burek> BlackBishop, it has got something to do with ipv6 [16:58] <burek> it's a bug :( [17:04] <BlackBishop> ow, fsck, yeah .. I have an ipv6 address .. :/ [17:05] <BlackBishop> I guess it might be fixed in newer versions but newer ones break my xbmc :/ [17:06] <burek> what version of ffmpeg are you using [17:07] <burek> is it the latest git? [17:08] <BlackBishop> nope, 0.7.7 right now ... [17:08] <BlackBishop> last time I used ~amd64 ( in gentoo's repo ) broke xbmc .. [17:08] <BlackBishop> so .. I decided to wait [17:10] <burek> well [17:10] <burek> you can do more than wait [17:11] <burek> you can try the latest git [17:11] <burek> since, multimedia tools, like ffmpeg, get updated every day [17:11] <burek> bug fixed every hour [17:11] <burek> so, it might be wise to use git, rather than "stable" releases :) [17:11] <burek> also, it compiles in like 10-15 mins [17:12] <BlackBishop> yeah, compile time isn't a problem .. BUT .. I have xbmc .. and I want that to work more than the ffserver thing .. [17:14] <burek> can you try vlc-nox [17:17] <BlackBishop> what for ? [17:17] <BlackBishop> streaming ? [17:35] <burek> BlackBishop yes [17:35] <burek> or use udp if possible [17:35] <burek> ffmpeg -i ... -f mpegts udp://target.subnet [17:40] <BlackBishop> neah, I'll wait 'till this gets sorted out .. [18:40] <khali> can anyone suggest a simple frontend to ffmpeg for Windows? [18:50] <burek> why frontend? [18:50] <burek> why not cli [18:51] <sacarasc> Which is a front end, really. [18:51] <khali> burek: I'm happy with the command line, but I suspect my sister-in-law won't be [18:51] <burek> just give her the link to the documentation :) [18:52] <khali> sacarasc: it certainly is; but I asked for a _simple_ frontend ;) [18:52] <sacarasc> It is simple. [19:06] <khali> sacarasc: thanks for your help [19:24] <pasteeater> khali: winff [19:28] <khali> pasteeater: will suggest that, thanks [22:40] <monstaRtruck> guys how do i sync up sound with my video [22:41] <monstaRtruck> no recording setting will make it sync [22:41] <monstaRtruck> i jus hav to do it afterwards [23:20] <burek> what is the synonim for vlc's ffmpeg-hurry-up in ffmpeg? [00:00] --- Sun Nov 27 2011

1 0

ffmpeg-devel.log.20111125
by burek 26 Nov '11

26 Nov '11

[04:43] <CIA-36> ffmpeg: 03Ronald S. Bultje 07master * rbd97b2e1ce 10ffmpeg/libavutil/ (pixdesc.c pixfmt.h): pixfmt: add planar RGB formats. [04:43] <CIA-36> ffmpeg: 03Ronald S. Bultje 07master * r6b0768e202 10ffmpeg/ (4 files in 2 dirs): Clean up swscale pixfmt macros using av_pix_fmt_descriptors[]. [04:43] <CIA-36> ffmpeg: 03Luca Barbato 07master * r7f1b427018 10ffmpeg/libavcodec/ (Makefile snow.c snow.h snowdata.h snowdec.c snowenc.c): [04:43] <CIA-36> ffmpeg: snow: split snow in snowdec and snowenc [04:43] <CIA-36> ffmpeg: The common non inlined code goes in snow.c, the common inlined code in [04:43] <CIA-36> ffmpeg: snow.h, tables move in snowdata.h (included only by snow.c) [04:43] <CIA-36> ffmpeg: 03Ronald S. Bultje 07master * rdb431f7efe 10ffmpeg/libavcodec/ (h264.c utils.c): h264: add support for decoding planar RGB images. [04:43] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r92afb43162 10ffmpeg/: (log message trimmed) [04:43] <CIA-36> ffmpeg: Merge remote-tracking branch 'qatar/master' [04:43] <CIA-36> ffmpeg: * qatar/master: [04:43] <CIA-36> ffmpeg: snow: split snow in snowdec and snowenc [04:43] <CIA-36> ffmpeg: tiffenc: deprecate using compression_level [04:43] <CIA-36> ffmpeg: swscale: fix failing fate tests. [04:43] <CIA-36> ffmpeg: swscale: add support for planar RGB input. [04:43] <CIA-36> ffmpeg: 03Ronald S. Bultje 07master * rf7f1835258 10ffmpeg/ (libavutil/pixdesc.c libswscale/swscale_internal.h): [04:43] <CIA-36> ffmpeg: swscale: fix failing fate tests. [04:43] <CIA-36> ffmpeg: isGray() is left as a FIXME for later. [04:43] <CIA-36> ffmpeg: 03Ronald S. Bultje 07master * r185655c601 10ffmpeg/libswscale/ (swscale.c swscale_internal.h swscale_unscaled.c utils.c): swscale: add support for planar RGB input. [04:43] <CIA-36> ffmpeg: 03Anton Khirnov 07master * r8b7412fe4e 10ffmpeg/libavcodec/ (tiffenc.c version.h): tiffenc: deprecate using compression_level [05:22] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r6d16a96a95 10ffmpeg/libavfilter/vf_boxblur.c: [05:22] <CIA-36> ffmpeg: vf_boxblur: fix memleak [05:22] <CIA-36> ffmpeg: As the filter uses the default start frame, the cleanup done by the [05:22] <CIA-36> ffmpeg: default end frame is needed. [05:22] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:03] <Compn> [01:06] <Compn> theres no such thing as enough memory if you have flash plugin running. [13:27] <CIA-36> ffmpeg: 03Stefano Sabatini 07master * r65f24858ed 10ffmpeg/ffprobe.c: [13:27] <CIA-36> ffmpeg: ffprobe: always print int values with print_val() [13:27] <CIA-36> ffmpeg: In particular, make the json writer write size values, fix regression [13:27] <CIA-36> ffmpeg: introduced with the addition of the print_val() macro. [14:38] <CIA-36> ffmpeg: 03Dominique Leuenberger 07master * r12d276531c 10ffmpeg/RELEASE: [14:38] <CIA-36> ffmpeg: RELEASE: We're now at 0.8.7.git [14:38] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:38] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * re1d48786d5 10ffmpeg/ffmpeg.c: [16:38] <CIA-36> ffmpeg: ffmpeg: Warn if output file is empty [16:38] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:24] <CIA-36> ffmpeg: 03Anatoly Nenashev 07master * ra7cfef2994 10ffmpeg/libavcodec/h264_ps.c: [17:24] <CIA-36> ffmpeg: H264: Check if more RBSP data in PPS provided by current profile due to Annex A. [17:24] <CIA-36> ffmpeg: This patch also fix issue https://ffmpeg.org/trac/ffmpeg/ticket/685. [17:24] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:13] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * re9e642cbfb 10ffmpeg/libavcodec/indeo3.c: [18:13] <CIA-36> ffmpeg: indeo3: Check remaining bits in parse_bintree() [18:13] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:13] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r1afe49b062 10ffmpeg/libavcodec/indeo3.c: [18:13] <CIA-36> ffmpeg: indeo3: out of array read checks for decode_plane() [18:13] <CIA-36> ffmpeg: Fixes: avi+indeo3+++1-dog.avi [18:13] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:03] <ubitux> saste: hi :) [19:08] <ubitux> saste: why is there a need for av_get_token() in the init func? [19:08] <ubitux> since the filter just has the filename as parameter [20:34] <ubitux> btw, git question: any idea how to solve a git am that doesn't apply? [20:35] <ubitux> i generally manually patch -p1 < ... [20:35] <ubitux> and fix it like this [20:35] <ubitux> which is a bit awkward [20:35] <nevcairiel> doesnt the failed am leave conflict markers in the file for you to resolve [20:36] <michaelni> you need git am -3 to get conflict markers and you need the blobs in your repo that the patch referes to [20:37] <ubitux> oh, -3, thx [20:37] <ubitux> nevcairiel: not by default, but it seems to put the tree in a strange state (that doesn't show with git status) [20:38] <ubitux> "Repository lacks necessary blobs to fall back on 3-way merge." :( [20:38] <ubitux> well anyway, ok [20:38] <michaelni> ubitux, try plain patch too [20:39] <ubitux> you mean patch -p1 < bla.patch? :P [20:42] <michaelni> yes [20:44] <ubitux> yes that's what i usually do :p [20:44] <ubitux> but it sucks a bit ;) [21:13] <michaelni> j-b, please approve burek becoming gci mentor for ffmpeg [21:28] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r7d52f46db7 10ffmpeg/libavcodec/vc1dec.c: [21:28] <CIA-36> ffmpeg: vc1dec: fix used ER flags in vc1_decode_b_blocks() [21:28] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:28] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * r9e794d103c 10ffmpeg/libavcodec/vc1dec.c: [21:28] <CIA-36> ffmpeg: vc1dec: drop damaged B frames [21:28] <CIA-36> ffmpeg: Fixes: vc1_error_spilt.avi of Ticket606 [21:28] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:18] <pasteeater> damn power outages... [22:30] <ubitux> mmh i'm unable to make the ass filter working saste [22:31] <ubitux> oh my bad, the text was very small :) [22:32] <ubitux> i'm not sure about the alpha value [22:33] <ubitux> i'm trying to play a ass file with this style: [22:33] <ubitux> Style: Default, DejaVu Sans, 40, &H00FFFFFF, &H00FFFF00, &H00000000, &H00000000, 0, 0, 0, 0, 100, 100, 0, 0.00, 1, 1, 0, 2, 30, 30, 10, 0 [22:34] <ubitux> and i need to set the first byte to FF instead of 00 [22:34] <ubitux> i think there is something wrong [22:34] <ubitux> since it renders well with other players (which use libass) [22:34] <ubitux> are you sure it's not 0xFF - x [22:34] <ubitux> ? [22:35] <ubitux> alpha is 0, so no transparency [22:35] <ubitux> erh [22:35] <ubitux> i mean value is 0, so no transparency [22:35] <ubitux> mmh indeed it seems to be that [22:36] <ubitux> setting to 'AA' makes it transparent [22:45] <ubitux> anyway, replied on the topic [23:08] <ubitux> i wonder if the code in the vf_overlay to blend the bitmap couldn't be exported [23:08] <ubitux> so we could use it in vf_overlay and vf_ass at least [23:13] <michaelni> if its usefull it surely can be exportet [23:43] <ubitux> drawtext is also a potential target for this [23:44] <ubitux> that will require some thinking not to slower things though [23:45] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * ra3b3562b47 10ffmpeg/libavcodec/vc1dec.c: [23:45] <CIA-36> ffmpeg: vc1dec: fix 10l typo [23:45] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:46] <CIA-36> ffmpeg: 03Michael Niedermayer 07master * rfc75e6f168 10ffmpeg/libavcodec/vc1dec.c: [23:46] <CIA-36> ffmpeg: vc1dec: fix scantable for advanced P frames [23:46] <CIA-36> ffmpeg: Fixes: vc1 file from Ticket606 [23:46] <CIA-36> ffmpeg: Fixes: vc1+vc1+++artifacts*.vc1 [23:46] <CIA-36> ffmpeg: Fixes: mpeg+vc1+++salxxos.evo [23:46] <CIA-36> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:55] <ubitux> michaelni: do you think such code belongs to lavfi/drawutils, lavu/imgutils or sth else? [00:00] --- Sat Nov 26 2011

1 0

ffmpeg.log.20111125
by burek 26 Nov '11

26 Nov '11

[00:10] <pythonirc101> Any suggestions for converting a mts file to high quality video that media player can read? [00:10] <pythonirc101> I tried -i x.mts output.mp4 -- and the file size became 5 times smaller [00:20] <pythonirc101> -sameq for same quality? [00:48] <pythonirc101> instead of saying -b 6000k -- how can i say , use the sam quality setting as the input? [00:59] <pythonirc101> how much difference does "-g" make on an encoding? Whats the optimum value of it for 1080p videos? [02:07] <jesselang|laptop> Hi. I'm using ffmpeg to convert an image to a DV file. It seems to change and flutter a bit (especially the text on the image). I've tried using ffmpeg directly, and tried ffenc_dvvideo from gstreamer. [02:08] <jesselang|laptop> The folks in #gstreamer said that the encoder must not encode identical frames deterministically. [02:10] <jesselang|laptop> Could anyone help me out with this? [05:12] <navatwo> Could someone help me out with a command to copy audio from a mp4 container into another mp4 container? [05:12] <navatwo> They *are* the same length [05:33] <burek> just use -acodec copy [05:35] <burek> pythonirc101m [05:35] <burek> can you please use pastebin.com, to show your command line and its output? [05:36] <burek> you can use ffmpeg -i input -vcodec libx264 -crf 20 -acodec copy out.mp4 [07:51] <sam1> hi guys [07:51] <sam1> i want to play a movie (*.avi + *.srt) with ffplay,what's command iuse? [08:16] <TryDent1> does ffmpeg support appleloss codec? [08:17] <Tjoppen> you mean alac? yes, for several years I'm sure [08:19] <TryDent1> ffmpeg -i R:\1.vob -acodec whatdoiputhereforalac R:\1.alac [08:19] <relaxed> alac [08:19] <Tjoppen> :) [08:20] <TryDent1> where do i see the acodec and vcodec list [08:20] <Tjoppen> there might be an option to spend more cpu time to improve the compression, not sure [08:20] <sacarasc> ffmpeg -codecs [08:21] <TryDent1> anybody know what is the command to convert any audio format to wav [08:22] <TryDent1> wow list is huge [08:22] <TryDent1> wow it even support vc1 [08:22] <TryDent1> amazing [08:24] <TryDent1> sacarasc is the "ffmpeg -codecs list" legit? [08:26] <TryDent1> anybody know? [08:32] <grepper> -acodec pcm_s16le -ar 48000 or somesuch [08:32] <TryDent1> why le not be [08:33] <TryDent1> and why 16 [08:34] <TryDent1> ffmpeg -i R:\a.wav -acodec alac R:\a.alac does not work [08:36] <relaxed> ffmpeg -i R:\a.wav -acodec alac R:\a.m4a [08:38] <TryDent1> relaxed why is ffmpeg picky with file extension [08:39] <TryDent1> why won't this work? ffmpeg -i R:\a.wav -acodec wmalossless R:\a.wma [08:42] <relaxed> what is your goal? what are you doing? [08:42] <TryDent1> wmalossless; that's my goal [08:43] <TryDent1> why won't it work [08:43] <relaxed> ffmpeg doesn't support it [08:43] <TryDent1> it's in the codec list [08:43] <relaxed> it's not in mine [08:43] <relaxed> who uses wmalossless anyway? [08:43] <TryDent1> it's in mine [08:44] <TryDent1> relaxed then it shouldn't be in the list [08:44] <relaxed> I have never run across one [08:46] <TryDent1> relaxed okay ; i am confused about converting to wav [08:46] <TryDent1> which one do i use [08:46] <relaxed> ffmpeg -i input output.wav [08:46] <TryDent1> what do i put for acodec [08:48] <relaxed> you don't need to -acodec [08:48] <TryDent1> then what is this? <grepper> -acodec pcm_s16le -ar 48000 [08:49] <relaxed> well, ffmpeg will decide the correct acodec/audio rate from the format. [08:50] <TryDent1> how come "ffmpeg -codecs" does not show that i have to use .m4a for acodec alac [08:50] <TryDent1> or am i missing something [08:51] <relaxed> m4a is a format; alac is a codec [08:51] <relaxed> ffmpeg -formats [08:52] <relaxed> by the way, the world decided on flac unless you need it for some iProduct. [08:53] <TryDent1> m4a is not listed in ffmpeg -formats [08:54] <relaxed> D mov,mp4,m4a,3gp,3g2,mj2 QuickTime/MPEG-4/Motion JPEG 2000 format [08:54] <TryDent1> basically how would i know if acodec alac has to use m4a if you didn't told me in the channel [08:55] <relaxed> but it's basically the mp4 format [08:55] <TryDent1> that does not answer myq esution [08:56] <relaxed> how about you google and read a little? http://en.wikipedia.org/wiki/Apple_Lossless [08:57] <TryDent1> that doesn't answer my question still [08:58] <TryDent1> and this is ffmpeg question [09:07] <TryDent1> i found a huge bug in ffmpeg i had original.wav and converted to alac and flac and converted back to wav; original.wav and decodedback.wav does not have same md5sum [09:10] <Tjoppen> not a bug (probably) [09:11] <Tjoppen> try remuxing original.wav first, then do the same test [09:11] <TryDent1> remuxing orignal.wav what do you mean [09:12] <Tjoppen> original.wav could have come from anywhere, with all sorts of metadata [09:12] <Tjoppen> or some strange structure [09:12] <Tjoppen> ffmpeg -i original.wav -acodec copy remuxed.wav [09:12] <Tjoppen> then compare remuxed.wav to decodedback.wav [09:13] <TryDent1> i shouldn't have to remux it [09:14] <TryDent1> what if somebody did the same thing with zip and md5sum didn't match [09:14] <Tjoppen> uh.. zip? [09:15] <Tjoppen> use vbindiff [09:15] <Tjoppen> and compare the data atoms [09:15] <TryDent1> no i trust md5sum [09:15] <Tjoppen> or hell, just look att both files using vbindiff and you can clearly see the different [09:16] <Tjoppen> or you know, compare the raw audio properly by muxing to -f s16le [09:16] <Tjoppen> (IIRC) [09:25] <TryDent1> okay let me convert to wav using flac program [09:27] <TryDent1> if i convert back to wav using flac program then md5sum matches [09:30] <TryDent1> problem with ffmpeg when convert back to wav [09:31] <Tjoppen> just look with vbindiff. the difference will probably be obvious [09:31] <Nagy> Is it possible to make saw scale [09:31] <Nagy> Nvm, accidental send. [09:36] <TryDent1> trydent why don't i just use filecompare app or md5sum; that's more accurate [09:38] <Tjoppen> they don't help with diagnostics [09:39] <Tjoppen> also, are the files the same size? [09:39] <TryDent1> no, it's 2 bytes samller [09:41] <Tjoppen> look at this: https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ then look in vbindiff where the difference is [09:41] <Tjoppen> specifically if the size field for the data chunk is the same [09:42] <Tjoppen> it sounds like one sample might have gotten losst [09:43] <TryDent1> problem is not with ffmpeg encoding [09:45] <TryDent1> why does md5sum match when i use flac decoder then [09:46] <TryDent1> it's a bug; please fix it [09:48] <Tjoppen> file a ticket then [09:49] <Tjoppen> http://ffmpeg.org/bugreports.html [09:51] <TryDent1> vbindiff.exe R:\a.wav R:\b.wav [09:52] <TryDent1> what am i suppose to be looking at [09:52] <Tjoppen> do you see anything highlighted in red? [09:52] <TryDent1> yes [09:52] <TryDent1> what about it [09:53] <Tjoppen> or rather, is the other file so that the whole thing is shifted two bytes? [09:53] <TryDent1> tjoppen if you don't believe me; why don't you just try encoding/decoding a wav file yourself [09:55] <Tjoppen> I don't doubt you. anyway, reproduced locally [09:55] <Tjoppen> went wav -> flac -> wav and the header got 2 B bigger [09:55] <TryDent1> was it 2 bytes off? [09:55] <Tjoppen> yah [09:55] <TryDent1> sorry you are right; bigger not smaller [09:55] <TryDent1> my bad [09:56] <TryDent1> so where is the bug? [09:57] <TryDent1> trydent1 try flac decoder [10:01] <Tjoppen> seems to be writing 2 B of extradata. not sure why [10:01] <TryDent1> does remuxing help? [10:01] <TryDent1> your first idea [10:06] <Tjoppen> yeah, the remuxed file is identical [10:06] <Tjoppen> I see there was a fix to the wav muxer quite recently [10:06] <Tjoppen> which might explain why the header is larger [10:06] <Tjoppen> anyway, the data is the same which is what's important [10:06] <TryDent1> you mean it took thing long for somebody find this 2 byte bug? [10:06] <TryDent1> this* [10:07] <TryDent1> when was ffmpeg invented? [10:07] <Tjoppen> it has to do with matroska [10:07] <TryDent1> huh? what do you mean [10:07] <TryDent1> that' mkv [10:07] <Tjoppen> yeah. it shares code with the wav mxuer [10:07] <Tjoppen> *shrugs* [10:08] <TryDent1> so this bug is fixed now? [10:08] <Tjoppen> I wouldn't call it a bug. the wav header is somewhat flexible - expecting two files to have the exact same header is pushing it [10:08] <TryDent1> tjoppen flac decoder decodes exactly [10:10] <Tjoppen> 2c4e08d89327595f7f4be57dda4b3775e1198d5e is the culprit [10:10] <Tjoppen> in fact: [10:10] <Tjoppen> Since fate uses wav files for the audio test a larger number of tests [10:10] <Tjoppen> has changed checksums or shifted positions due to the 2 byte longer [10:10] <Tjoppen> wave header. [10:14] <Tjoppen> considering this is appearently the correct way to write the header I'd say the bug is in flac [10:14] <Tjoppen> if you feel it's a big issue I'd file a ticket in both projects' trackers [10:15] <TryDent1> hmm i found another bug [10:15] <TryDent1> ffmpeg -i a.mlp -acodec flac a.flac a.flac is smaller than a.mlp [10:15] <TryDent1> much smaller [10:16] <Tjoppen> mlp? [10:16] <TryDent1> a.mlp is 24bit but a.flac became 16 bit [10:16] <TryDent1> meridiallosslesspacking [10:17] <Tjoppen> -sample_fmt s24le [10:17] <TryDent1> huh? [10:17] <Tjoppen> -acodec flac -sample_fmt s24le see if that works [10:18] <Tjoppen> *s32 even [10:18] <Tjoppen> or.. uh [10:18] <TryDent1> oh, why would i have to do that? [10:18] <TryDent1> it's 24bit not 32bit [10:18] <Tjoppen> I know [10:18] <TryDent1> so which one do i use [10:18] <Tjoppen> it's supposed to not by this dumb I think [10:18] <Tjoppen> *be [10:19] <Tjoppen> that sounds like an actual issue. but to try -sample_fmt s32 and see what happens [10:20] <TryDent1> still producing 16bit [10:20] <TryDent1> let me try flac encoder [10:20] <TryDent1> doesn't work since flac cannot do from mlp [10:22] <TryDent1> even ffmpeg -i a.mlp a.wav is not doing properly [10:24] <relaxed> TryDent1: your ffmpeg version? [10:24] <TryDent1> C:\>ffmpeg --version [10:24] <TryDent1> ffmpeg version N-34906-g4e7b3ef, Copyright (c) 2000-2011 the FFmpeg developers [10:24] <TryDent1> built on Nov 16 2011 12:35:07 with gcc 4.6.2 [10:27] <TryDent1> relaxed; i thought ffmpeg -i a.mlp a.wav is smart enough to use same as source [10:27] <TryDent1> it keeps using 16bit [10:30] <TryDent1> again, flac decoder decodes properly and uses 24bit if the source is 24bit [10:30] <TryDent1> but not ffmpeg [10:31] <TryDent1> Tjoppen you going to blame this on mkv too? [10:34] <Tjoppen> to wav you can use -acodec pcm_s24le [10:35] <Tjoppen> it might be that the mlp demuxer isn't setting some field in AVCodecContext correctly [10:36] <Tjoppen> try ffmpeg -i a.mlp -acodec pcm_s24le a.wav and then ffmpeg -i a.wav a.flac and see what happens [10:37] <TryDent1> what about decode back to wav from 24bitflac [10:38] <Tjoppen> just see what ffprobe says about the resulting flac first [10:40] <TryDent1> ffmpeg -i a.wav a.flac is producing 16bit flac [10:42] <Tjoppen> is a.wav 24-bit? [10:42] <TryDent1> yes [10:43] <Tjoppen> flacenc.c:221: if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) [10:43] <Tjoppen> return -1; [10:44] <TryDent1> huh [10:44] <Tjoppen> so only 16-bit is supported [10:44] <Tjoppen> it should probably nah or something [10:44] <Tjoppen> *nag [10:44] <TryDent1> flac app does 24bit fine [10:45] <Tjoppen> writing a general purpose encoder tool like ffmpeg is harder, so no big surprise that there are corner cases where it behaves in unexpected ways [10:46] <Tjoppen> doesn't it print a warning though? [10:46] <TryDent1> i have a question for lossy compression formats; how do you know if it's using 24bit or 16bit or even 8 bit [10:51] <TryDent1> any idea? [10:51] <Tjoppen> it depends. most don'thave bitdepths per se [10:51] <Tjoppen> in some cases you can tell the decoder to use float instead of 16-bit [10:52] <Tjoppen> you'd have to be more specific [10:52] <TryDent1> lossy compression shows sample rate such as 44.1khz [10:56] <relaxed> TryDent1: it's listed in the stream info. Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 252 kb/s [10:57] <relaxed> s16 = 16bit Look at `ffmpeg -sample_fmts` [11:02] <TryDent1> relaxed what is the command to see the stream info [11:04] <relaxed> ffmpeg -i input [11:05] <TryDent1> wow [11:06] <relaxed> there's also ffprobe [11:06] <TryDent1> okay i am going to test this with 24bit mp3 [11:06] <TryDent1> this better work [11:07] <tdr> for osx, is it "better" to just compile ffmpeg or use macports to get it? [11:07] <TryDent1> lol fail [11:07] <TryDent1> Stream #0:0: Audio: vorbis, 96000 Hz, stereo, s16, 0 kb/s [11:08] <TryDent1> wow so many bugs [11:10] <relaxed> You want libvorbis, not vorbis. Until you know what you're doing please stop with the fail/bug comments. [11:11] <TryDent1> i didn't use ffmpeg to convert 24bitaudio to vorbis-ogg [11:12] <relaxed> If you have a problem, go to pastebin and paste your command and all output. [11:13] <TryDent1> R:\>ffprobe 24bit.mp3 [11:13] <TryDent1> Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 224 kb/s [11:14] <TryDent1> let me test 8bit.mp3 [11:16] <Tjoppen> unless I'm mistaken mp3 doesn't have bitdepth [11:18] <TryDent1> not just mp3; all the lossy format; i cannot seem to find bitdepth [11:18] <Tjoppen> also, there's the mp3float decoder [11:19] <TryDent1> R:\>ffprobe 8bit.mp3 Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s [11:20] <TryDent1> relaxed failed again [11:22] <Tjoppen> it should be possible to use mp3float if you need more precision. I don't know how though [11:24] <TryDent1> am i just using the wrong/buggy version or what [11:26] <ubitux> 24 bit mp3? heh? [11:27] <Tjoppen> that's the sentiment I've been trying to get across [11:28] <TryDent1> ubittux bitdepth exist for lossless formats [11:29] <ubitux> can mp3 be lossless? :p [11:29] <TryDent1> Format/Info : Free Lossless Audio Codec [11:29] <TryDent1> Duration : 8s 169ms [11:29] <TryDent1> Bit rate mode : Variable [11:29] <TryDent1> Bit rate : 101 Kbps [11:29] <TryDent1> Channel(s) : 2 channels [11:29] <TryDent1> Sampling rate : 44.1 KHz [11:29] <TryDent1> Bit depth : 8 bits [11:29] <Tjoppen> again, there's mp3float. just figure out how to use it [11:33] <TryDent1> tjoppen link? [11:34] <Tjoppen> I don't know [11:35] <Tjoppen> I just know there's a float decoder and it should be possible to use it somehow [11:36] <TryDent1> tjoppen which version fixes but 2 byte bug [11:37] <Tjoppen> you could use 2c4e08d89327595f7f4be57dda4b3775e1198d5e~1 if you don't want the fix that increased the size of the header [11:38] <Tjoppen> which is 582f231142c62a0bd6391efbd5a2ac119d73bb40 [11:38] <TryDent1> huh [11:40] <relaxed> to decode with mp3float you would use `ffmpeg -acodec mp3float -i input.mp3 ...` [11:40] <TryDent1> and what exactly does that do [11:42] <ubitux> from my understanding, you decide how to decode the mp3: default is using 16 bits samples, but you could use 24 bits (i'm not sure it would make sense) or float; the above command says: use mp3float to decode the input mp3 [11:42] <ubitux> afaiu, but i may be wrong [11:43] <TryDent1> what if you have 8bit.mp3 [11:44] <spyworldxp> how to select all the files inside the directory and convert it? [11:44] <relaxed> what kind of files? [11:44] <spyworldxp> mov to avi [11:45] <ubitux> for f in dir/*.mov; do ffmpeg -i $f ${f%mov}.avi; done [11:45] <ubitux> ? [11:45] <spyworldxp> yes [11:46] <spyworldxp> why linux video converter faster than windows? [11:46] <ubitux> TryDent1: mp3 is lossy, so i don't think the bit depth actually matters [11:46] <ubitux> i mean i'm not sure the information is actually in it [11:47] <TryDent1> spyworldxp how much faster [11:47] <spyworldxp> Windows take 1hour 30mins. Linux just 20mins [11:47] <TryDent1> wow big difference; are you converting exactly same video [11:48] <ubitux> "Technically speaking, bit depth is only meaningful when applied to pure PCM devices. Non-PCM formats such as lossy compression systems like MP3, have bit depths that are not defined in the same sense as PCM." [11:48] <TryDent1> with exact same settings [11:48] <ubitux> (https://en.wikipedia.org/wiki/Audio_bit_depth) [11:55] <Tjoppen> spyworldxp: asm disabled perhaps? [11:56] <spyworldxp> no [11:56] <spyworldxp> i know the different [11:56] <spyworldxp> linux using mpeg4 and windows using h264 (high) [11:58] <spyworldxp> for f in dir/*.mov; do ffmpeg -i $f ${f%mov}.avi; done --> How to add h264? [12:23] <spyworldxp> hi [12:23] <spyworldxp> Unrecognized option c:a [12:23] <spyworldxp> how? [12:54] <torii> Hi. [12:55] <torii> I have a question, install of ffmpeg. [12:56] <torii> I installed libx264 that version is "0.118.x", "x264-snapshot-20111124-2245-stable". [12:57] <torii> But cannot install last-ffmpeg. [12:58] <torii> Console display, http://pastebin.com/mxzUvtQY. [12:58] <sacarasc> Do you have any other versions of x264 installed? And did you install the dev files? [12:59] <torii> yes, ihave installed another version of x264. [13:01] <torii> cannot override x264 install? [18:01] <njbair> is ffmpeg built to take advantage of 64 bits? [18:35] <Diogo> hi [18:36] <Diogo> hi i have installed ffmpeg and i want to stream a mp4 file to rtmp..(wowza server) [18:36] <Diogo> this is possible..? [18:40] <kakapo> yes [18:45] <Diogo> and using mjpeg to rtmp [18:45] <Diogo> ? [18:46] <Diogo> i want o get the video fom a ip camera.. [18:47] <spm-Draget> I have an interlaed mpeg2 video source and encode it to xvid using 'ffmpeg -i test.mpg -f avi -vcodec mpeg4 -vtag xvid -flags +ilme+ildct -g 50 -qmin 1 -qmax 2 -ab 256 test.avi'. Especially the flags I am confused about. I want the resulting xvid video to be flagged and encoded as interlaced, since I want to deinterlace it later. [18:48] <Diogo> i'm using this configuration but don't stream.. [18:48] <Diogo> built on Nov 25 2011 16:48:34 with gcc 4.6.1 [18:48] <Diogo> configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab [18:49] <Diogo> i'm using this configurations [18:55] <spm-Draget> Okay, different question: Can I extract only the videostream from a file without reencoding? [18:57] <mateo`> spm-Draget: ffmpeg -i source -vcodec copy -an dest ? [19:01] <spm-Draget> mateo`: Hmm, this makes fcchandlers mpeg2 plugin more happy, indeed. Thanks [19:05] <Diogo> anyone use ffmpeg to do a restreming? [20:33] <Takyoji> Just need to switch container formats. How do I preserve the video and audio without re-encoding? [20:33] <burek> -acodec copy -vcodec copy [21:54] <pythonirc101> is there a way to tell ffmpeg to output in standard form instead of using fancy output (curses etc) [22:35] <jimlkl> Here's the ffmpeg- i information from a video I downloaded. (pastebin will follow after this post) It [22:35] <jimlkl> transfers and plays on the iPod BUT the width of the video is partially [22:35] <jimlkl> cut off on the iPod screen. Any idea why? Please let me know how to fix it. [22:36] <jimlkl> Here's the pastebin [22:36] <jimlkl> http://pastebin.com/2TbLeVq9 [22:39] <pasteeater> jimlkl: does it look cut off with ffplay too? ffplay rp.mp4 [22:42] <jimlkl> hang on .....I'll give it a try. [22:43] <jimlkl> No, it doesn't. It plays perfectly. [22:46] <pasteeater> what does quicktime say about the video? there's a menu with "video information" or something like that [22:47] <jimlkl> I don't know.....I mostly run on Ubuntu 10.04 and would have to reboot to go to Windoz [22:48] <jimlkl> It plays fine in iTunes 10.5 but gets cut off on the iPod [22:48] <jimlkl> Wait.... [22:48] <jimlkl> geez.....I can't remember if it gets cut off in iTunes. [22:48] <jimlkl> My head is a sive. [22:49] <pasteeater> looks like ffmpeg is fine with it then. blame quicktime/itunes/iDevice/apple. [22:49] <jimlkl> 'Had a disk crash and had to reinstall Windows and restore all of my music and videos to stupid itunes. [22:50] <jimlkl> I download the videos from Youtube and try and put 'em on the iPod. [22:50] <jimlkl> Previous to THE CRASH of my disk drive I had no problem at all. [22:50] <jimlkl> Now....nothing but problems. [22:51] <burek> jimlkl [22:51] <burek> can you please use pastebin.com, to show your command line and its output? [22:52] <jimlkl> Oh boy.....that's a tough one......I don't even remember it nor could I recreate it. When it comes to FFMPEG I'm a moron and the best I can do is ask people for a code line to convert to stuff to ipod compatible format. [22:53] <pasteeater> did you try the command i gave you on ubuntuforums yet? [22:53] <jimlkl> Surely in this wicked world there has to be one simple method to ffmpeg a video file (mp4) to a ipod format. [22:53] <jimlkl> Are you FakeOutdoorsman? [22:54] <pasteeater> that bastard? [22:54] <pasteeater> yes [22:54] <pasteeater> http://ubuntuforums.org/showpost.php?p=11486441&postcount=6 [22:54] <jimlkl> YOU ARE THE MAN!!!!!! [22:54] <jimlkl> Hang on....let me get the code line... [22:56] <jimlkl> Is this it......ffmpeg -i cba.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf 24 -acodec libfaac -aq 100 FAKEOUTD.mp4 [22:57] <jimlkl> I've tried so many different code lines I don't remember if yours worked. I think it did....but not sure. [22:58] <jimlkl> Hang on....I'll check it again. [22:58] <jimlkl> By the way.....are you really in Alaska? [22:58] <pasteeater> yes [22:58] <jimlkl> Wow. [22:58] <pasteeater> if your input is too big for your ipod then add: -vf scale="640:trunc(ow/a/2)*2" [22:58] <pasteeater> it will auto scale to an acceptable size. [22:59] <jimlkl> Yes....the code line above worked perfectly. [23:00] <jimlkl> ....that is....the FAKEOUTD.mp4 line [23:00] <jimlkl> What do you mean by "if your input is too big? [23:02] <pasteeater> you ipod probably can't play back videos larger than 640x480 or sometihng like that [23:02] <jimlkl> Using the auto scale you mentioned just above is this how the line should look? ffmpeg -i cba.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf 24 -acodec libfaac -aq 100 -vf scale="640:trunc(ow/a/2)*2" FAKEOUTD.mp4 [23:03] <Diogo> hi anyone know how can i restream to a ip camera to wowza server.. [23:03] <Diogo> using ffmpeg? [23:05] <pasteeater> jimlkl: yes [23:05] <jimlkl> Alright.....I'll try it. [23:05] <jimlkl> Is there a way to speed up the conversion process as it takes a long time to convert a 50 meg file? [23:08] <pasteeater> use a faster preset. you can see a preset list with x264 --help [23:10] <jimlkl> Oh boy....that stuff is so confusing to me. [23:12] <pasteeater> presets in order of speed: ultrafast, superfast, veryfast, faster, fast, medium, slow, slower, veryslow [23:13] <jimlkl> Fake.....thanks so much for your help. You really are a great help here and over at ubuntuforums. [23:14] <jimlkl> Peace. [23:14] <jimlkl> Paste......would you re-send that last line you wrote to me? [23:17] <Hyperi> 00:12:53 < pasteeate> presets in order of speed: ultrafast, superfast, veryfast, faster, fast, medium, slow, slower, veryslow [23:17] <Hyperi> yw :P [23:18] <jimlkl> thanks [23:30] <wolfman2000> Afternoon. I am trying to compile ffmpeg n0.8.7, and I get failures on the linking process. My issue is in the pastebin. http://pastie.org/2921531 If it helps, I'm trying to build on Mac OS X 10.7 Lion. [23:32] <pasteeater> wolfman2000: i'm ignorant of osx, but your ./configure would probably be useful for others who are familiar with osx. [23:33] <wolfman2000> pasteeater: same paste link has been updated. If it will also help, I'm starting with a configure provided by an old build script for a project I help contribute to. [23:34] <burek> Diogo, what command did you try so far [23:37] <wolfman2000> pasteeater: what may help is that Lion is a 64 bit OS by nature, but can also build 32 bit [23:39] <burek> wolfman2000, did you try: https://www.google.com/search?q=%22Undefined+symbols+for+architecture+x86_6… [23:39] <wolfman2000> burek: yes. right now reading [23:40] <ghostbar> hey! I want to grab a rtsp stream a put it on x.jpg but the only way I found is to use -vframes 1 and re-launch ffmpeg. Is there a way to make it rewrite the jpeg without re-lunching ffmpeg? [23:40] <burek> yes [23:41] <burek> add -y [23:41] <burek> wait [23:41] <ghostbar> i used something like ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo.jpeg [23:41] <burek> can you please use pastebin.com, to show your command line and its output? [23:41] <burek> ok [23:41] <ghostbar> burek: it doesn't works [23:41] <burek> so [23:41] <burek> ffmpeg -loop 1 -i foo.avi -r 1 -s WxH -f image2 foo.jpeg [23:41] <ghostbar> let me tell you what gives me [23:41] <burek> that works [23:41] <burek> wait wait.. my bad [23:42] <burek> ffmpeg -re -i foo.avi -r 1 -s WxH -f image2 foo.jpeg [23:43] <burek> that should work [23:44] <wolfman2000> burek: going to try http://ffmpeg.org/trac/ffmpeg/ticket/469 and make it use clang [23:45] <ghostbar_> burek: it gives me [image2 @ 0x9c20820] Could not get frame filename number 2 from pattern 'what.jpg' [23:45] <ghostbar_> that's the same output with -y [23:45] <ghostbar_> and without -y [23:45] <ghostbar_> ffmpeg -re -y -i shell-20110908-1.webm -r 1 -f image2 -s qvga what.jpg [23:46] <ghostbar_> that's the line i'm trying with [23:46] <wolfman2000> burek, pasteeater: --cc=clang to ./configure did the job. At least I got no errors. [23:46] <burek> ghostbar, it seems that ffmpeg expects at least one % param (sprintf) [23:46] <burek> so we need to trick it somehow [23:53] <burek> ghostbar_ [23:53] <burek> you know what you could do [23:53] <pythonirc1011> can ffmpeg use nvidia CPUs? seems like for my AVCHD to mpeg4 conversion, its pretty slow on my quadcore [23:53] <burek> since ffmpeg doesn't allow you to just set the same filename [23:53] <burek> it requires you to put %d into your filename [23:54] <burek> you could set filename like 'output%d.jpg' [23:54] <burek> that would presumably go from 1-9 (or 0-9) [23:54] <sacarasc> pythonirc1011: CPUs or GPUs? [23:54] <burek> and then loop [23:54] <burek> so, you just set -r 10 (instead of 1) [23:54] <burek> and read one of those 10 files you want [23:54] <burek> for example output1.jpg [23:55] <burek> do you get my point? [23:55] <wolfman2000> ...sad part here is that I'm now stumped. I don't know which of the .a files I now need for my project. [23:55] <burek> wolfman2000 :beer: :) [23:55] <burek> well, at least you are never bored, right? :D [23:55] <wolfman2000> ...something like that [23:57] <burek> pythonirc1011, im not sure really, i think ffmpeg cannot (so far) use gpus [23:57] <burek> but, vlc has got an option for ffmpeg, namely --ffmpeg-hw [23:57] <burek> This allows hardware decoding when available. (default disabled) [23:57] <burek> so, I don't know what to answer :) [23:58] <pythonirc1011> i'm looking to convert a AVCHD to MPEG4 that is playable on win 7 [23:58] <pythonirc1011> its using all my quad cpus and still takes a lot of time because the bitrate is high [23:58] <burek> are you using libx264 [23:59] <pythonirc1011> yes [23:59] <pythonirc1011> my camera claims to output 24Mbps. I'm using 12Mbps [23:59] <burek> did you read x264 --help, especially for -preset and -tune [00:00] --- Sat Nov 26 2011

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