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April 2012

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ffmpeg-devel.log.20120409
by burek 10 Apr '12

10 Apr '12

[00:19] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rc311afca0a 10ffmpeg/tests/ (codec-regression.sh ref/acodec/roqaudio): [00:19] <CIA-17> ffmpeg: Add roqaudio regression test. [00:19] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [05:39] <Compn> anyone feel like downloading a large sample file (1gb+) to see how our exr decoder handles it ? [05:39] <Compn> possibly report some bugs [05:39] <Compn> or create samples with blender and check for bugs [11:51] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * red68fd45d3 10ffmpeg/tests/ (codec-regression.sh ref/vsynth1/v408 ref/vsynth2/v408): [11:51] <CIA-17> ffmpeg: Add v408 codec regression test. [11:51] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [17:23] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r1d130328ed 10ffmpeg/libavcodec/aacdec.c: [17:23] <CIA-17> ffmpeg: aacdec: update debug code to always print the processed data. [17:23] <CIA-17> ffmpeg: When decoding LATM, this function will not process extradata [17:23] <CIA-17> ffmpeg: but a different buffer. [17:23] <CIA-17> ffmpeg: It seems this was forgotten to update when LATM support [17:23] <CIA-17> ffmpeg: was added. [17:23] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [18:01] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r438f3ef821 10ffmpeg/libavcodec/rv10.c: (log message trimmed) [18:01] <CIA-17> ffmpeg: RV20: try keeping aspect during resolution changes. [18:01] <CIA-17> ffmpeg: Resolution changes are usually only used to scale with [18:01] <CIA-17> ffmpeg: network bandwidth, the (full) resolution specified in the [18:01] <CIA-17> ffmpeg: RM header really is authoritative. [18:01] <CIA-17> ffmpeg: While I am not sure this is the best solution, it is a [18:01] <CIA-17> ffmpeg: conservative approach that still should fix the most [18:01] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rbbaf941e29 10ffmpeg/libavfilter/vf_scale.c: [18:01] <CIA-17> ffmpeg: scale: fix slice rendering with conversion between pal/non-pal. [18:01] <CIA-17> ffmpeg: We can't use whether the input format is paletted to decide that [18:01] <CIA-17> ffmpeg: the output format has a palette in data[1], too, that makes no sense. [18:01] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [18:01] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r24eac3cff5 10ffmpeg/ (3 files in 2 dirs): (log message trimmed) [18:01] <CIA-17> ffmpeg: Mark GRAY8 format as pseudo-paletted. [18:01] <CIA-17> ffmpeg: This fixes that the GIF encoder crashes with it because [18:01] <CIA-17> ffmpeg: it has no palette. [18:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.8 * r7ca2ed716d 10ffmpeg/libavcodec/x86/dsputilenc_mmx.c: [18:03] <CIA-17> ffmpeg: dsp: fix diff_bytes_mmx() with small width [18:03] <CIA-17> ffmpeg: Fixes Ticket1068 [18:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:03] <CIA-17> ffmpeg: (cherry picked from commit 73089eccd3e48539555349b36d8aabbf1cea416e) [18:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.8 * rc9d12a4692 10ffmpeg/libavcodec/pngenc.c: [18:03] <CIA-17> ffmpeg: pngenc: Fix incorrect mask used for interlaced mode. [18:03] <CIA-17> ffmpeg: Fixes Ticket1109 [18:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:03] <CIA-17> ffmpeg: (cherry picked from commit 15db6a959057b92245a384909ec7d413d5c16461) [18:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.8 * r3b18d820cc 10ffmpeg/Changelog: [18:46] <CIA-17> ffmpeg: 03Reinhard Tartler 07release/0.7 * r808686375d 10ffmpeg/Changelog: Update changelog for 0.7.5 release [18:46] <CIA-17> ffmpeg: 03ami_stuff 07release/0.7 * r10848d0862 10ffmpeg/libavcodec/x86/dsputil_yasm.asm: [18:46] <CIA-17> ffmpeg: Replace SSE2 instruction in scalarproduct_float_sse() by SSE equivalent. [18:46] <CIA-17> ffmpeg: Fixes an AAC decoding issue with the sample from ticket #213 on machines [18:46] <CIA-17> ffmpeg: with SSE but without SSE2. [18:46] <CIA-17> ffmpeg: Based on 89411a by Reimar. [18:46] <CIA-17> ffmpeg: (cherry picked from commit f6b78638086beae9bcab672d4c9de1790be5a928) [18:46] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * r7ca2ed716d 10ffmpeg/libavcodec/x86/dsputilenc_mmx.c: [18:46] <CIA-17> ffmpeg: dsp: fix diff_bytes_mmx() with small width [18:46] <CIA-17> ffmpeg: Fixes Ticket1068 [18:46] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:46] <CIA-17> ffmpeg: (cherry picked from commit 73089eccd3e48539555349b36d8aabbf1cea416e) [18:46] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:46] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * rc9d12a4692 10ffmpeg/libavcodec/pngenc.c: [18:46] <CIA-17> ffmpeg: pngenc: Fix incorrect mask used for interlaced mode. [18:46] <CIA-17> ffmpeg: Fixes Ticket1109 [18:46] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:46] <CIA-17> ffmpeg: (cherry picked from commit 15db6a959057b92245a384909ec7d413d5c16461) [18:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * r3b18d820cc 10ffmpeg/Changelog: [18:47] <CIA-17> ffmpeg: Changelog, delete, its too inaccurate, git log is better. [18:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * r9dd50da65b 10ffmpeg/: (log message trimmed) [18:47] <CIA-17> ffmpeg: Merge branch 'release/0.8' into release/0.7 [18:47] <CIA-17> ffmpeg: * release/0.8: [18:47] <CIA-17> ffmpeg: Changelog, delete, its too inaccurate, git log is better. [18:47] <CIA-17> ffmpeg: pngenc: Fix incorrect mask used for interlaced mode. [18:47] <CIA-17> ffmpeg: dsp: fix diff_bytes_mmx() with small width [18:47] <CIA-17> ffmpeg: Replace SSE2 instruction in scalarproduct_float_sse() by SSE equivalent. [18:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * r4f85e7b6ec 10ffmpeg/: [18:47] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 [18:47] <CIA-17> (3 lines omitted) [19:06] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.7 * re05fd37e02 10ffmpeg/ (Doxyfile RELEASE VERSION): [19:06] <CIA-17> ffmpeg: Update for 0.7.12 [19:06] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:06] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.8 * r4169912f39 10ffmpeg/ (Doxyfile RELEASE VERSION): [19:06] <CIA-17> ffmpeg: Update for 0.8.11 [19:06] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:52] <ubitux> https://ffmpeg.org/trac/ffmpeg/ticket/1180 [19:52] <ubitux> it looks like to me that the video is waiting for a keyframe [19:53] <ubitux> while the sound gets immediately copied [19:53] <ubitux> or something like this [21:15] <nevcairiel> for the record, that file plays fine for me using avformat/avcodec, so its not the libraries, more the implementations using them [21:16] <nevcairiel> seeking in avi should be "ok", so something must be odd in ffmpeg [21:21] <ubitux> the original file is fine; it plays fine with anything [21:23] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3532a87a25 10ffmpeg/libavcodec/exr.c: [21:23] <CIA-17> ffmpeg: exr: optimize exr_halflt2uint() [21:23] <CIA-17> ffmpeg: 30% faster [21:23] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:25] <nevcairiel> oh, i somehow misread that issue. yeah after running that command the video is oddly broken. It plays at around 11 fps for me, until the audio ends, then the remaining video plays at proper speed, very odd [21:37] <ubitux> yes behaviour differs quite a bit between players [21:37] <ubitux> but it's always broken :) [21:38] <ubitux> i expect this kind of issue pretty often when trying to extract samples& [21:38] <nevcairiel> i guess leavint out the -ss works, eh [21:39] <nevcairiel> leaving* [21:56] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r7f66912f1a 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/gif): [21:56] <CIA-17> ffmpeg: Fix gif regression test on big-endian. [21:56] <CIA-17> ffmpeg: Decode output must be converted to rgb24 to avoid CRC difference [21:56] <CIA-17> ffmpeg: due to palette being stored in machine endianness. [21:56] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [23:20] <CIA-17> ffmpeg: 03Nicolas George 07master * rb2db35995f 10ffmpeg/libavutil/ (audioconvert.c audioconvert.h avutil.h): [23:20] <CIA-17> ffmpeg: audioconvert: make av_get_channel_layout accept composite names. [23:20] <CIA-17> ffmpeg: Signed-off-by: Nicolas George <nicolas.george(a)normalesup.org> [23:20] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:20] <CIA-17> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [23:20] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r695ec04e6a 10ffmpeg/avconv.c: [23:20] <CIA-17> ffmpeg: avconv: only set SAR once on the decoded frame. [23:20] <CIA-17> ffmpeg: No point in repeating the assignment for each output stream. [23:20] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rb6c4518a87 10ffmpeg/avconv.c: [23:20] <CIA-17> ffmpeg: avconv: remove OutputStream.picref. [23:20] <CIA-17> ffmpeg: It's only used inside transcode_video() and there's no point in [23:20] <CIA-17> ffmpeg: preserving it between subsequent calls. So use a local variable instead. [23:20] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r18ed3788b0 10ffmpeg/libavutil/ (samplefmt.c samplefmt.h): avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc() [23:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r0109a09dc3 10ffmpeg/libavutil/ (avutil.h samplefmt.c samplefmt.h): avutil: use align == 0 for default alignment in audio sample buffer functions [23:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rc58846f3a8 10ffmpeg/libavcodec/ (avcodec.h utils.c version.h): [23:21] <CIA-17> ffmpeg: avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame() [23:21] <CIA-17> ffmpeg: Use default alignment in audio_get_buffer() [23:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r679a973e97 10ffmpeg/avconv.c: avconv: use default alignment for audio buffer [23:21] <CIA-17> ffmpeg: avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc() [23:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r1337de0c4b 10ffmpeg/libavcodec/utils.c: [23:21] <CIA-17> ffmpeg: avcodec: validate the channel layout vs. channel count for decoders [23:21] <CIA-17> ffmpeg: Set avctx->channel_layout to 0 if the channel count does not match [23:21] <CIA-17> ffmpeg: avctx->channels. [23:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r4d693b023c 10ffmpeg/ (4 files in 2 dirs): [23:21] <CIA-17> ffmpeg: avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt() [23:21] <CIA-17> ffmpeg: Based on a patch by Clément BSsch <ubitux(a)gmail.com> [00:00] --- Tue Apr 10 2012

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ffmpeg.log.20120409
by burek 10 Apr '12

10 Apr '12

[00:00] <relaxed> cnx: I don't think '-f mpeg' is correct, try '-f mpegts' [00:01] <cnx> [mpegts @ 0x2446eb0] Could not detect TS packet size, defaulting to non-FEC/DVHS [00:01] <cnx> [mpegts @ 0x2446eb0] max resync size reached, could not find sync byte [00:05] <cnx> anyone? [00:32] <darkstarbyte> How would I copy over certain audio streams and leave the others out, like 0.4 and etc [00:34] <relaxed> darkstarbyte: -map 0:a:4 [00:39] <darkstarbyte> oh [00:40] <cnnx> im having trouble getting ffmpeg to record from /dev/vide0 is there a generic parameter set I can try instead? [00:41] <darkstarbyte> I think you are supposed to have a special program for that [00:42] <cnnx> mencoder [00:42] <darkstarbyte> I was thinking something more like [00:42] <darkstarbyte> a dvr program [00:43] <cnnx> i wanted to write my own cli script [00:43] <cnnx> you cant? [00:43] <darkstarbyte> There is some junk data in most of those streams from the broadcaster [00:43] <cnnx> /dev/video0 [00:43] <cnnx> what broadcaster? [00:44] <darkstarbyte> all of the common ones [00:44] <cnnx> what are you talking about [00:44] <darkstarbyte> EIT data I believe it is called [00:44] <cnnx> /dev/video0 is an input source [00:44] <darkstarbyte> for one [00:44] <cnnx> from a video capture card [00:44] <darkstarbyte> over the air [00:44] <cnnx> i didnt say over the air [00:45] <darkstarbyte> sorry that is what most of those are used for [00:45] <cnnx> sorry [00:45] <darkstarbyte> perhaps you could mount it, and see the incoming data [00:45] <darkstarbyte> perhaps it would make files on its own. [00:46] <darkstarbyte> then all you would have to do is copy them. [00:51] <darkstarbyte> I have wondered if a hdmi cable could record things. [00:51] <darkstarbyte> port* [01:23] <barhom> If I just want to copy audio without doing anything at all using ffserver.conf, what should I add? (Im guessing something like; AVOptionAudio flags copy) [01:23] <barhom> Im transcoding the video, but I want to keep the audio as is [03:35] <darkstarbyte> What should I use to extract an individual audio track? [03:36] <darkstarbyte> the map option did not work [03:37] <relaxed> pastebin.com the command and all output of what didn't work. [03:42] <darkstarbyte> I got it to work after adding some stuff to what the last person said. [03:42] <darkstarbyte> sorry [10:33] <barhom> how do you add a filter from ffserver? [10:33] <barhom> AVOptionVideo filter yadif (something like that I guess) [10:34] <barhom> I can't find any documentation regarding it, or Im just blind and hopefully somebody can direct me [10:39] <Pacefist> Hello! [10:40] <Pacefist> I'm getting a digital TV signal package by package [10:40] <Pacefist> And I need to make frames from it [10:41] <Pacefist> an then convert it in openCV image [10:42] <Pacefist> It looks like typical problem, but I can't find any example, how to do it [10:42] <Pacefist> (sorry for bad english) [11:27] <burek> barhom, did you check the ffmpeg example config file [11:28] <barhom> yes but there is nothing about filters [11:28] <burek> Pacefist, do you need to do it using ffmpeg or libav? [11:28] <barhom> http://ffmpeg-users.933282.n4.nabble.com/ffserver-ignores-filters-td4310672… < I think I understand that filters is not supported in ffserver [11:28] <barhom> ffserver doesn't support filters, and yes that would be a nice feature < forum post 2012-01-20 [11:29] <burek> well [11:29] <burek> http://ffmpeg.org/pipermail/ffmpeg-user/2012-January/004386.html [11:30] <ubitux> we desperately need a ffserver maintainer actually :) [11:30] <burek> why don't you apply filter before you send it to ffserver [11:30] <barhom> cuz I didn't think about it? :P [11:30] <barhom> Ill try that [11:30] <burek> ok :) [11:58] <cnx> when I run ffmpeg -f mjpeg -s 640x480 -i /dev/video0 out.mpg I don't get any errors but it doesn't create my output file, any idea why? [12:05] <burek> well [12:05] <burek> cnx, first of all [12:06] <burek> what do you want ffmpeg to do? what is your goal [12:06] <cnx> burek: the statement says it [12:06] <burek> the statement is very confusing [12:06] <burek> so im not sure [12:06] <cnx> go to /dev/video0 using the mpeg codec and dump to a file [12:06] <burek> you even know [12:06] <burek> what statement did you write [12:06] <burek> thats why i ask [12:07] <cnx> scroll up [12:07] <burek> i see [12:07] <burek> you are missing -vcodec and -acodec [12:07] <burek> if you just want to dump [12:07] <burek> use -vcodec copy [12:08] <cnx> iused -vcodec, still doesnt create any output file [12:08] <cnx> and I dont care about acodec (audio) [12:09] <Pacefist> burek: using ffmpeg [12:10] <Pacefist> burek: what is libav? [12:10] <burek> cnx does the fps info change [12:10] <burek> or everything is just idle [12:10] <burek> Pacefist, how do you get "packages" [12:10] <cnx> burek: bttv: 0: Setting PLL: 28636363 => 35468950 (needs up to 100ms) [12:10] <cnx> bttv: PLL set ok [12:11] <Pacefist> cnx: I have a message with 188 bytes of information [12:11] <Pacefist> (from transport stream) [12:11] <Pacefist> burek: I have a message with 188 bytes of information [12:12] <Pacefist> it is one packet in the stream [12:12] <burek> Pacefist, [12:12] <burek> can you please use pastebin.com, to show your command line and its output? [12:12] <burek> cnx, too [12:12] <Pacefist> burek: I'm not using consloe [12:13] <Pacefist> burek: I'm using library [12:13] <cnx> kernel: [55714.402692] bttv: 0: SCERR @ cf823000,bits: VSYNC HSYNC OFLOW SCERR* [12:13] <cnx> this looks like an ERROR [12:13] <cnx> resolution [12:14] <burek> Pacefist, i dont understand [12:14] <Pacefist> burek: I should make a function [12:14] <burek> cnx, ffmpeg output [12:14] <cnx> http://pastebin.com/rs7bk3nn [12:14] <cnx> then I hit control-C [12:14] <cnx> and no file is generated in ls [12:14] <burek> Pacefist, what library [12:15] <Pacefist> openCV::Image getImage(char[188] ); [12:15] <Pacefist> ffmpeg [12:15] <Pacefist> sorry, not [12:16] <Pacefist> void getImage(char[188]); [12:16] <burek> Pacefist, then try to ask for help on OpenCV support page? [12:16] <burek> cnx, the entire output [12:16] <Pacefist> Oh, god, let me explain [12:16] <burek> not just half [12:16] <cnx> burek: that is [12:16] <burek> it's not [12:16] <cnx> on the cli it is [12:16] <burek> then your ffmpeg is fault [12:16] <burek> try to get the latest one [12:17] <burek> faulty* [12:28] <cnx> i compiled the newest version but now it doesnt recognize any of the arguments on the cli [12:28] <cnx> Unknown decoder 'codec' [12:28] <cnx> Option video_size not found. [12:28] <cnx> etc [12:28] <burek> can you please use pastebin.com, to show your command line and its output? [12:28] <cnx> that is the only output [12:28] <cnx> and the same command line as before [12:29] <burek> just copy/paste it [12:29] <cnx> http://pastebin.com/SQxNehFx [12:29] <burek> -vcodec codec ? [12:30] <burek> seriously? :) [12:30] <cnx> yeah, in this version its not copy [12:30] <cnx> look in the -h [12:30] <burek> :D [12:30] <burek> try reading _more_ careful [12:30] <cnx> then fine [12:30] <cnx> none of the args work anyhow [12:30] <burek> it never was "codec" and will never be.. it was always "copy" and shall stay that way [12:31] <cnx> my mistake [12:31] <Pacefist> burek: I have MPEG-TS transport stream. On each next packet getImage(char*) function is used. And there is one more function: Frame* getFrame(), that return present frame in the stream. And last, there is a function, getImage(), that should returns present image in openCV format. There is an example, how to convert an image from ffmpeg to openCV format, but there is no examples, how to get ffmpeg [12:31] <burek> I would say it's more like you are so impatient, cnx, and are making trivial mistakes [12:31] <burek> take it easy and it will work [12:32] <Pacefist> burek: ffmpeg picture [12:32] <cnx> http://pastebin.com/nr70c1dj [12:32] <burek> Pacefist, where did you read info for getImage() ? is there any link? [12:32] <burek> cnx, another mistake [12:32] <burek> the order of parameters does matter [12:33] <burek> move -vcodec copy after -i input [12:33] <Pacefist> burek: I can use some variables from in function [12:33] <burek> Pacefist, do you have any URL to the documentation for getImage function? [12:34] <Pacefist> shared variables: getImage and getFrame can use the same variables [12:34] <cnx> http://pastebin.com/tpDAbhNZ [12:34] <Pacefist> burek: no, no, no [12:34] <Pacefist> burek: wait a minute [12:35] <burek> cnx, try removing -r [12:36] <Pacefist> burek: well, getImage - isn't a good name [12:36] <cnx> http://pastebin.com/PVait5fR [12:36] <Pacefist> burek: I can make an example of using [12:36] <burek> cnx, remove -s [12:37] <burek> Pacefist, I'm trying to figure out what on earth are you doing [12:37] <burek> that's why I ask what is getImage [12:37] <cnx> http://pastebin.com/seWpL64f then i hit control C and there's no out.mpg [12:37] <burek> cnx, what did you connect to /dev/video0 [12:37] <cnx> video capture device and camera that owrks with tvtime [12:38] <burek> cnx, try this: ffmpeg -f v4l2 -i /dev/video0 out.mpg [12:38] <cnx> http://pastebin.com/3GbujFb2 [12:39] <burek> cnx, also -vcodec copy goes before output name (out.mpg) [12:39] <cnx> now it created the out.mpg but playing back the video is a bluescreen, can I select the right channel (multiplexoer) for composite1? [12:40] <burek> try installing v4l-utils [12:40] <burek> or v4l-tools [12:40] <burek> and use v4l2-ctl [12:40] <burek> to set what you need [12:41] <burek> Pacefist, can you copy/paste your code on pastebin so that we can figure out what are you dealing with [12:44] <Pacefist> http://pastebin.com/LPfPHk39 [12:44] <burek> ok, what libraries do you use [12:46] <Pacefist> And sometimes I will call functions getFrame and getImage [12:46] <Pacefist> Image* - pointer to picture in openCV library [12:47] <Pacefist> MPEG_devices is already done [12:47] <Pacefist> now I should realise nextPacket() function [12:48] <Pacefist> using some libraries, for example, ffmpeg [12:48] <burek> I see [12:48] <burek> ok [12:48] <burek> well [12:48] <burek> you are probably looking for libav libraries [12:48] <burek> used in ffmpeg too [12:48] <burek> so, try joining #libav channel [12:48] <burek> and ask your question there, to get more help :) [12:49] <Pacefist> burek: thanks [12:51] <burek> :beer: :) [12:56] <Pacefist> burek: silence on libav... [12:58] <burek> well, be patient :) [12:58] <burek> we are not all in the same time zone [12:58] <Pacefist> I know [12:58] <burek> try using google for libav in the meantime [12:58] <burek> you might find some examples or something [12:59] <Pacefist> but while there is no example for using ffmpeg? [13:00] <burek> ffmpeg is command line utility [13:00] <burek> it internally is using libav [13:00] <Pacefist> ok [13:01] <Pacefist> but ffmpeg-codek in VLC [13:01] <burek> what? [13:01] <Pacefist> let me think that ffmpeg is library] [13:02] <burek> "burek> ffmpeg is command line utility" [13:02] <burek> libav is library [13:02] <Pacefist> burek: shy [13:02] <burek> :beer: :) [13:45] <barhom> anyone who knows about using preroll/buffer in ffserver? [13:46] <barhom> Cant seem to get it working, it worked once, with the same settings, it seems to be random - but I think this function isn't used that often [13:53] <Tthread> Hi [13:55] <Tthread> Could you please give me advice& [13:56] <Tthread> I use ffmpeg for decode live video [13:56] <Tthread> by RTP [13:58] <Tthread> but I want to make faster this process [14:00] <Tthread> now i spend about 20 ms for each frame for decode (frame size is about 1,5 kbyte) [14:01] <Tthread> how can i tune this? [14:01] <Tthread> for decoding i use avcodec_decode_video2 [14:27] <Mavrik> Tthread: check lines around 2367 in ffmpeg.c to see how to set multi-thread decoding [14:27] <Mavrik> that could help [14:30] <Tthread> Mavrik: aha! ok! Thanks a lot! [15:19] <pahan> Hello. I am trying to grab video from X11. What is the best(fastest) lossles codec for it? [15:28] <pahan> How to make ffmpeg to do lossles grabbing of x11? [15:29] <JEEB> ffvhuff is probably one of the fastest lossless codecs in libavcodec [15:30] <pahan> JEEB, thanks [16:18] <Hacker-CB> Hi All! [16:18] <Hacker-CB> Is somebody knows, how to get current pts after 'av_seek_frame' without calling 'av_read_frame'? [16:34] <andrew__2> Hey, when I use the flag "-vpre fast", it says the "File for preset 'fast' not found" [16:35] <andrew__2> Any ideas as to why this is happening. Did I not install something correctly? [16:37] <sacarasc> andrew__2: Try -preset fast [16:37] <sacarasc> The x264 presets are built in, not in external files any more. [16:37] <andrew__2> oh ok [16:38] <andrew__2> now when I do that, I get errors about broken ffmpeg default settings again [16:38] <sacarasc> Which version of ffmpeg are you using? [16:39] <andrew__2> stable release 0.10.2 [16:39] <sacarasc> Can you paste your complete command line and the output to a pastebin? [16:41] <andrew__2> one moment [16:42] <andrew__2> http://pastebin.com/3XvnmBam [16:43] <sacarasc> Try moving the preset to after the -vcodec libx264. [16:45] <andrew__2> pretty much the same output [16:45] <sacarasc> -preset:v fast [16:46] <sacarasc> Maybe... I forget how it's meant to be with these new builds... [16:47] <andrew__2> yea, still getting the same thing [16:47] <sacarasc> I thought that the default wasn't broken any more, so I am totally stumped. [16:48] <andrew__2> it wouldn't have anything to do with my installation, would it? [16:48] Action: sacarasc shrugs. [16:50] <andrew__2> ok, just trying to think up possible sources of error [16:50] <andrew__2> im on Ubuntu, so im using http://ubuntuforums.org/showthread.php?t=786095 as my installation guide [16:51] <andrew__2> this just seems really odd for a stable build [16:52] <andrew__2> which makes me think I did something wrong [16:52] <ckb> What kind of specs should I look at for a strict encoding server? [16:53] <ckb> AKA: I have uploaded videos I'm encoding. Should I try for 16GB RAM / QUAD+ core? [17:02] <ckb> anyone? bueler? [17:02] <andrew__2> sacarasc: I was able to get it to work when I set the output as a local file, rather than a ffm file for ffserver [17:07] <andrew__2> brb [17:57] <burek> Hacker-CB, try asking the same question in #libav [17:57] <burek> andrew__2, your ffmpeg is too old, please update to the latest git [17:59] <sacarasc> 0.10.2 is too old? [17:59] <burek> oops my bad [17:59] <burek> :D [17:59] <burek> I wanted to say that -vpre is deprecated :) [18:56] <Diogo> what i need to do to add watermark in a movvie? http://pastebin.com/8AZuvd2b Appear: Could not find codec parameters(video : png) [18:56] <Diogo> thnaks.. [18:56] <burek> hm [18:58] <burek> try taking a look here http://ffmpeg.org/ffmpeg.html#movie [18:58] <burek> no wait [18:59] <burek> overlay [18:59] <burek> http://ffmpeg.org/ffmpeg.html#overlay-1 [19:00] <burek> also type ffmpeg -codecs | grep png [19:01] <burek> see if you compiled the png lib [19:13] <Diogo> http://pastebin.com/Kdcnt0Tm [19:13] <Diogo> i have this codecs... [19:18] <burek> ffmpeg -codecs [19:21] <Diogo> what is the lib to support png? [19:21] <Diogo> i'm using debian.. [19:22] <burek> let me check [19:24] <burek> libpng12-dev [19:25] <ubitux> ffmpeg doesn't use libpng [19:25] <ubitux> ffmpeg -codecs|&grep png [19:25] <ubitux> DEV D png PNG image [19:26] <ubitux> decode/encode/video, direct rendering [19:26] <burek> what do you mean it doesn't use it? [19:27] <Diogo> libpng12-dev is already the newest version. [19:27] <Diogo> when i do ffmpeg -codecs | grep png appear this http://pastebin.com/Kdcnt0Tm [19:28] <ubitux> i don't see png in your paste [19:28] <ubitux> burek: there is a native encoder and decoder [19:29] <ubitux> so no link with libpng; you can check with ldd [19:32] <ubitux> Diogo: can you test if ./ffplay -f lavfi 'testsrc[a]; movie=a.png[b]; [a][b]overlay' works for you? [19:32] <ubitux> (given a a.png file) [19:34] <burek> type ffmpeg -codecs | less [19:34] <burek> and see if there is png in the list [19:34] <burek> oh [19:34] <burek> it uses zlib's implementation? [19:34] <burek> Diogo, can you type ffmpeg -codecs and copy/paste to pastebin [19:36] <ubitux> burek: yes the zlib is used, but it's a subpart of the codec [19:38] <Diogo> burek i don0t have png in that list.. [19:39] <level09> can ffmpeg convert quick time references exported from Avid ? [19:40] <Diogo> burek: what i need to include in ./configure to compile with support for png images? [19:40] <burek> Diogo, apt-get install zlib1g-dev [19:41] <juanmabc> diretly libpng-dev or like, will get you any dep [19:49] <Diogo> thanks [20:32] <andrew__2> Hey, so im still having an error where ffmpeg thinks that default settings are broken [20:33] <andrew__2> Does anyone have an idea about that? [20:33] <andrew__2> Im converting to h264 [20:34] <andrew__2> And it only happens when I try to feed the data into ffserver [20:36] <roxlu> hey! [20:37] <roxlu> I'm looking for a way to create annimated gifs. If I'm correct libav can do this. I was wondering if libav uses a 3rd party lib for this? [21:53] <cnx> when i use v4l2-ctl -i 1 to set to composite1 it does but when I run ffmpeg it resets it to 0 and doesnt use 1, why? [21:59] <cnx> how do you make changes permanent ? [22:08] <cnx> how do you select composite1 in ffmpeg? [22:23] <cnx> when I type v4l2-ctl -i 1 it changes to composite1 but doesnt save the changes for when i use ffmpeg, how can I make it do this? [23:09] <cnx> alright, can one of the developpres tell me which variable in which .c or .h file sets the default input (i.e composite1,2,etc..) [23:09] <cnx> not the /dev/video0, but for /dev/video0 which input [23:11] <cnx> hello? [23:26] <meekohi> Is there anyway to share the r,g,b values in the expressions when using -vf luvrgb? For example if I want to enhance saturation in a video? [23:30] <meekohi> lutrgb I mean [00:00] --- Tue Apr 10 2012

1 0

ffmpeg-devel.log.20120408
by burek 09 Apr '12

09 Apr '12

[00:10] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * rf0bb1a599c 10ffmpeg/ffprobe.c: [00:10] <CIA-17> ffmpeg: ffprobe: use avbprint API [00:10] <CIA-17> ffmpeg: Simplify, increase robustness. [00:17] <Daemon404> michaelni, ping [00:17] <ubitux> saste: can't we use the bprint api to get rid of the fast_asprintf stuff too? [00:17] <Daemon404> or someone who knows how to use av_seek_frame properly [00:18] <saste> ubitux: possible [00:19] <nevcairiel> whats there to do wrong with av_seek_frame? [00:21] <Daemon404> nevcairiel, i cant figure out how to easily seek to teh last keyframe [00:21] <Daemon404> based on duration [00:21] <Daemon404> the format fo teh timestamp parameter confuses me [00:21] <floater> hey [00:22] <nevcairiel> seeking is generally rather volatile [00:22] <nevcairiel> most formats dont have accurate seeks or even key-frame seeks [00:22] <Daemon404> this is solely for mp4 files [00:23] <Daemon404> which i believe should [00:23] <nevcairiel> hm i dont think i ever read the seeking code for mp4, but at least its potentially possible to implement it properly because its fully indexed [00:24] <Daemon404> im a bit confused how to get to the end given duration [00:24] <Daemon404> tho [00:25] <ubitux> the end of the iso base media specs describe a seek scenario iirc [00:25] <nevcairiel> set flags to AVSEEK_FLAG_BACKWARD and just seek to the last known timestamp [00:25] <nevcairiel> it should end up on the last keyframe then [00:26] <Daemon404> yes but i dont know how to mangle stream->duratin [00:26] <Daemon404> into a format seek_Frame wont barf on [00:26] <nevcairiel> you shouldnt need to [00:26] <nevcairiel> specify a stream id to av_seek_frame [00:26] <Daemon404> yeah [00:27] <nevcairiel> if you specify an id, it means all times are in that streams time_base [00:27] <nevcairiel> so just pass it st->start_time+st->duration [00:27] <nevcairiel> or something like that [00:28] <Daemon404> nevcairiel, that sure doesnt work [00:28] <Daemon404> lol [00:28] <Daemon404> i get some random frame [00:30] <nevcairiel> are you sure st->duration is set properly? [00:30] <Daemon404> lemme me doubt check [00:30] <nevcairiel> maybe you need to use avfmtctx->duration [00:31] <nevcairiel> which you need to rescale into the streams timebase [00:31] <Daemon404> -1 [00:31] <Daemon404> for stream index [00:31] <nevcairiel> or that, but i always prefer specifying one. ;) [00:32] <Daemon404> AHA [00:32] <Daemon404> im a retard [00:32] <Daemon404> i forgot to change a hardcoded stream index of 1 [00:32] <Daemon404> ro i [00:33] <Daemon404> to* [00:33] <nevcairiel> heh [00:34] <Daemon404> of course in order to get the decoded frame, i had to ignore an error [00:34] <Daemon404> h264 spits out a buttload of nal errors, but decodes correctly after a few iterstions [00:35] <nevcairiel> it always complains a lot, not always does it mean something is broke [00:35] <Daemon404> ah [00:36] <Daemon404> not i just need to master encode2 [00:36] <Daemon404> s/not/now/ [00:36] <nevcairiel> if you're only decoding one frame, you should make sure frame threading is turned off, it makes live a bit easier [00:36] <Daemon404> oh [00:37] <Daemon404> thats a good idea [00:37] Action: Daemon404 reads doxy [00:55] <burek> err.. :)) can I get a little help please :) http://pastebin.com/32xCVFKr [00:56] <Daemon404> ... it is not clear what your problem is [00:56] <burek> Unrecognized option 'h' :) [01:51] <burek> +1 for -follow_mouse option :) [02:39] <burek> is it possible to tell ffmpeg to write directly to the soundcard [02:39] <burek> like to set the output to /dev/snd [02:39] <burek> or something? [02:39] <burek> the guy can't install ffplay for some reason, but can install ffmpeg [02:40] <burek> to play his audio [02:41] <Compn> uh [02:41] <Compn> well you can pipe pcm audio to /dev/dsp [02:41] <Compn> how you do that with ffmpeg ... dunno [02:42] <Compn> mkfifo out.wav /dev/dsp [02:42] <Compn> ffmpeg -i file.mp3 out.wav [02:42] <Compn> dunno [02:42] <burek> i see :) [02:42] <burek> cool, let us try that :) [02:43] <Compn> i'm not good with mkfifo so better read manual [02:44] <burek> I will :) thanks a lot :) [02:45] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rd506ef05a0 10ffmpeg/tests/lavf-regression.sh: [02:45] <CIA-17> ffmpeg: Fixes for third argument do_image_formats. [02:45] <CIA-17> ffmpeg: Remove it from two places where it is useless, do not apply [02:45] <CIA-17> ffmpeg: it to the encode command and make it apply to the output [02:45] <CIA-17> ffmpeg: instead of the input of the decode command. [02:45] <CIA-17> ffmpeg: Should fix the dpx test. [02:45] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [02:45] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * re9841505d2 10ffmpeg/libavcodec/options_table.h: [02:45] <CIA-17> ffmpeg: Allow setting CODEC_FLAG2_CHUNKS via the name "chunks". [02:45] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [02:45] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r58c25724b7 10ffmpeg/tests/ (5 files in 3 dirs): [02:45] <CIA-17> ffmpeg: Add nellymoser "fuzzy" FATE test. [02:45] <CIA-17> ffmpeg: Since we cannot specify decode parameters (and also because [02:51] <burek> one more thing, is it possible to stretch/shrink audio with ffmpeg? [02:51] <burek> like the stream that is 60 seconds long, to stretch to 70 seconds [02:53] <Compn> mplayer can do it with -af scaletempo and -speed , but i dont know if ffmpeg has such feature. you should ask in #ffmpeg [02:53] <burek> ok :) [02:53] <burek> I'm helping a guy there in #ffmpeg :) that's why I'm asking here :) [02:53] <Compn> ah [02:54] <Compn> i come from mplayer world, and ffmpeg is quickly catching up with filters, so i dont know whats in yet :) [02:54] <burek> I've seen once someone doing it something with PTS, but I forgot :/ [02:56] <burek> oh, finally I understood the difference between pts and dts :D maybe ffmpeg could include a few paragraphs from here in its documentation about pts/dts: http://dranger.com/ffmpeg/tutorial05.html [02:56] <burek> :) [03:02] <Compn> burek : want to make a small patch ? :) [03:02] <burek> :D [03:02] <burek> I would, but it's too complicated for me :S [03:03] <Compn> but i agree, pts/dts is confusing [03:03] <Compn> :) [03:04] <burek> well, I was always wondering about it and now I feel like an idiot like "well the very name suggested what it is" :D [12:55] <ubitux> !@# [13:05] <CIA-17> ffmpeg: 03Thierry Foucu 07master * rcb93946820 10ffmpeg/libavformat/mpegtsenc.c: [13:05] <CIA-17> ffmpeg: mpegtsenc: Add support to mux s302 stream [13:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:05] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r352a67e6f8 10ffmpeg/configure: [13:05] <CIA-17> ffmpeg: configure: fix - vs _ error [13:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:06] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r069cf86d32 10ffmpeg/libavcodec/ (avcodec.h utils.c): [13:06] <CIA-17> ffmpeg: Fix side-data memleak also for audio. [13:06] <CIA-17> ffmpeg: This uses the same code as in decode_video also in decode_audio. [13:06] <CIA-17> ffmpeg: Should fix valgrind FATE failures for nellymoser encode test. [13:06] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [15:42] <thumbnailer> Hi. I want to know more about the way FFMPEG seeks to a perticular time of a video [16:35] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r5612359501 10ffmpeg/libavcodec/ffv1.c: [16:35] <CIA-17> ffmpeg: ffv1: add a minor version field beginning with ffv1.3 [16:35] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:35] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1b295a17d7 10ffmpeg/libavcodec/ffv1.c: [16:35] <CIA-17> ffmpeg: ffv1: fix version=2 chroma handling [16:35] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8dfbc1c5cf 10ffmpeg/libavcodec/ffv1.c: [17:19] <CIA-17> ffmpeg: ffv1: allow enabling of version 2 by using some of its features. [17:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1d80c8db27 10ffmpeg/libavcodec/ffv1.c: [17:19] <CIA-17> ffmpeg: ffv1: set slice geometry based on user specified slices. [17:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:01] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * ra7b1420ca2 10ffmpeg/tests/ref/fate/wmv8-x8intra: [20:01] <CIA-17> ffmpeg: Update reference for disabled fate-wmv8-x8intra test to new format. [20:01] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:26] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r03ef5047d1 10ffmpeg/libavcodec/nellymoserenc.c: [20:26] <CIA-17> ffmpeg: Fix nellymoser encoder crash with hardcoded tables. [20:26] <CIA-17> ffmpeg: Use the correct init function to avoid crashes due to writing [20:26] <CIA-17> ffmpeg: to a rodata location. [20:26] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:28] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * ra5efc8c9bf 10ffmpeg/tests/Makefile: [20:28] <CIA-17> ffmpeg: Allow manually running disabled FATE tests. [20:28] <CIA-17> ffmpeg: This makes it easy to allow people to run tests that are disabled [20:28] <CIA-17> ffmpeg: (e.g. because they are broken) without having to hack Makefiles [20:28] <CIA-17> ffmpeg: by adding the test name to FATE_TESTS-no. [20:28] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:28] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r4f043c8a11 10ffmpeg/tests/fate/microsoft.mak: [20:28] <CIA-17> ffmpeg: Allow running fate-wmv8-x8intra manually. [20:28] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:48] <ubitux> http://prog21.dadgum.com/108.html [20:49] Action: Daemon404 spots "Haskell" [20:49] Action: Daemon404 shudders [20:57] <ubitux> i like this kind of spirit; the whole blog is pretty interesting [21:08] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r62c3c8ca78 10ffmpeg/libavformat/rtsp.c: [21:08] <CIA-17> ffmpeg: rtsp: Don't use uninitialized data if there are no streams [21:08] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:08] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r9294f538e9 10ffmpeg/libavformat/rtspdec.c: [21:08] <CIA-17> ffmpeg: rtsp: Don't use av_malloc(0) if there are no streams [21:08] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:08] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r5d6a40bc74 10ffmpeg/: (log message trimmed) [21:08] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [21:08] <CIA-17> ffmpeg: * qatar/master: [21:08] <CIA-17> ffmpeg: rtsp: Don't use av_malloc(0) if there are no streams [21:08] <CIA-17> ffmpeg: rtsp: Don't use uninitialized data if there are no streams [21:08] <CIA-17> ffmpeg: vaapi: mpeg2: fix slice_vertical_position calculation. [21:08] <CIA-17> ffmpeg: hwaccel: mpeg2: decode first field, if requested. [21:08] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r2ce7f4d4e6 10ffmpeg/libavformat/rtpdec_xiph.c: [21:09] <CIA-17> ffmpeg: decoder will correctly decode to the right line computed from the [21:09] <CIA-17> ffmpeg: appropriate top_field_first and is_first_field flags. [21:09] <CIA-17> ffmpeg: This patch aligns with DXVA's definition, which is what most HW and [21:09] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r456001486e 10ffmpeg/libavformat/ (8 files): [21:09] <CIA-17> ffmpeg: rtsp: Don't expose the MS-RTSP RTX data stream to the caller [21:09] <CIA-17> ffmpeg: This avoids exposing a dummy AVStream which won't get any data [21:09] <CIA-17> ffmpeg: and which will make avformat_find_stream_info wait for info about [21:09] <CIA-17> ffmpeg: this stream. [21:09] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:19] <CIA-17> ffmpeg: 03Reinhard Tartler 07release/0.8 * r808686375d 10ffmpeg/Changelog: Update changelog for 0.7.5 release [21:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.8 * r4f85e7b6ec 10ffmpeg/: [21:19] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 [21:19] <CIA-17> ffmpeg: * qatar/release/0.7: [21:19] <CIA-17> ffmpeg: Update changelog for 0.7.5 release [21:19] <CIA-17> ffmpeg: Merged-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Mon Apr 9 2012

1 0

ffmpeg.log.20120408
by burek 09 Apr '12

09 Apr '12

[01:05] <dadada> hey [01:05] <dadada> I'm batch converting a bunch of wma [01:06] <burek> cool :) [01:06] <dadada> I saw some examples where people use aq 100 [01:06] <dadada> is that necessary, would the quality by default not be 100%? [01:06] <burek> I think each encoder has its own default settings [01:06] <dadada> converting to mp3 that is [01:07] <burek> so you are best off specifying what you want [01:07] <relaxed> -aq 0 would be highest quality with libmp3lame [01:07] <dadada> what is the default? [01:07] <relaxed> probably 128k cbr [01:08] <relaxed> why do you care what the default is? [01:08] <dadada> is ogg quality better then mp3 or anything else I should look into? [01:08] <burek> aac+ ! :) [01:08] <relaxed> you mean vorbis, yes [01:09] <dadada> just to know if I can trust the defaults [01:09] <relaxed> you can not [01:09] <dadada> k [01:10] <dadada> why is vorbis good versus mp3? [01:10] <burek> why is blue better than green? [01:11] <relaxed> It's much newer for one, google for the rest. [01:12] <relaxed> If you're so worried about quality use flac [01:13] <dadada> flac would probably inflate the size too much [01:14] <burek> try aac+ if you really want to keep the best size/quality ratio [01:14] <Kuukunen> burek: blue is better because in double blind tests conducted by the University of Wisconsin, it got 7 points better average than green [01:14] <burek> Kuukunen :) [01:14] <dadada> lol [01:23] <diegoviola> hi [01:23] <diegoviola> can i record a single window with ffmpeg? [01:23] <diegoviola> a specific region of my desktop [01:44] <burek> diegoviola, yes [01:46] <diegoviola> i see, thanks [01:47] <diegoviola> any pointers please? [01:48] <burek> what you need is x11grab [01:48] <diegoviola> ty [01:48] <burek> if you are on windows, then vfwcap [01:48] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=25&t=13 [01:48] <diegoviola> i'm on linux, so x11grab, thanks [01:48] <burek> oh ok [01:49] <diegoviola> thanks for the help, appreciated [01:49] <diegoviola> :) [01:49] <burek> :beer: :) [01:49] <diegoviola> are x11grab and vfwcap something like ffmpeg modules? [01:50] <diegoviola> can something like that be written for wayland? [01:50] <burek> hmh.. you can say something like that, yes :) [01:50] <diegoviola> or different windowing systems, etc [01:50] <diegoviola> cool [01:50] <diegoviola> thanks [01:50] <burek> yw :) [02:12] <teresko> hi everyone [02:13] <teresko> i have two questions regarding http://ffmpeg.zeranoe.com/builds/ [02:13] <dadada> why does it say that vorbis is experimental and I need strict experimental? [02:14] <teresko> 1. what is the difference between shared and static builds ( i have only encountered such distinction when dealing with opera on bsd with QT libs ) [02:14] <burek> dadada, because we want to make sure that you don't come back here complaining too much, because the codec is still experimental :D [02:15] <burek> meaning _very_ beta :) [02:15] <burek> teresko, shared = uses windows dlls, static = doesn't use any external dlls (has got everything it needs in the exe file) [02:15] <teresko> damn .. nevermind [02:15] <teresko> its written in the page [02:15] <teresko> =/ [02:15] <burek> :) [02:17] <dadada> burek: that's a bummer, then I can forget ogg and use mp3 [02:17] <burek> use aac+ :) [02:17] <burek> it rocks! [02:17] <teresko> and the other question was as stupid as first one : can i used the external codecs from ffmpeg's trunk with mplayer ? [02:17] <dadada> burek: I doubt my router has support for that format :-) [02:17] <burek> teresko, ffmpeg's "external codecs" are just simple independent libraries [02:18] <burek> written just for that, so the other projects can include them in their own code :) [02:18] <teresko> so the answer is "yes" [02:18] <llrcombs> relaxed: OK, I reencoded the AAC MKA as an AC3 MKA. No issues with the duration, but it still plays very choppily in VLC [02:18] <burek> so the answer is, obviously, "yes, you can!" :D [02:18] <teresko> good [02:18] <teresko> my mplayer has been outdated for years [02:18] <burek> dadada, google for your router's user manual and check it out :) [02:18] <teresko> ( and yes , i am using mplayer on windows ) [02:19] <burek> llrcombs, try converting your audio to wav and play it in the vlc, to see if the stream is ok? [02:19] <llrcombs> burek: it plays fine in any other player [02:19] <burek> llrcombs, what are the errors in vlc? [02:20] <llrcombs> buffer too early, down-sampling [02:21] <llrcombs> buffer way too early, clearing queue [02:21] <burek> try remuxing with -vsync or -async [02:22] <dadada> burek: hmm, the file I converted to vorbis plays fine on my desktop but very badly on the router... madplay works only for mpeg and works great on the router... I have to play ogg with sox on the router but that's horrible, maybe ogg needs more hardware resources, sox has bad perfomance, or encoding problem? [02:22] <llrcombs> Is there a way to do that in MMG? [02:23] <burek> dadada, did you try using aac? -f aac [02:23] <burek> -acodec aac (sorry) [02:23] <dadada> burek: I don't know that there's a single tool that play that on the router [02:23] <llrcombs> burek: was that for me? [02:23] <burek> llrcombs, what is mmg? [02:24] <dadada> burek: it's a very limited env, mips cpu, busybox, totally stripped down, 400mhz [02:24] <llrcombs> MkvMerge GUI [02:24] <burek> dadada, which tool do you use to play it on the router [02:24] <burek> llrcombs, sorry, I've never used it, I thougt you were talking about ffmpeg [02:25] <llrcombs> burek: I'm encoding in ffmpeg and muxing in MkvMerge [02:25] <burek> why [02:25] <llrcombs> burek: because there's a nice, easy GUI for mkvmerge [02:25] <burek> why not mux it in ffmpeg? [02:26] <dadada> burek: sox [02:26] <dadada> burek: and madplay for mp3 (but mp3 is about all madplay can..) [02:28] <llrcombs> burek: 1. because I've been told it's preferable to mux in MkvMerge (though that doesn't really matter), and 2. because the muxing isn't the issue. Playing the MKA alone is choppy in VLC [02:29] <burek> dadada, did you try converting audio to aac, just to see if it plays? [02:29] <burek> llrcombs, I'm asking because: 1. it is easier to use 1 tool instead of 2 [02:30] <burek> and 2. muxing can cause audio/video async [02:30] <burek> so you get buffer too early for audio stream for example :) [02:30] <dadada> burek: no, but each tool gives a list of supported codecs, madplay just supports mpeg, sox supports a bunch but AFAIK not aac [02:30] <llrcombs> burek: it can, yeah, and I understand that if it only happened after muxing, MMG could be to blame [02:30] <llrcombs> burek: but it happens even when I take an MKA file straight from ffmpeg, so mkvmerge is definitely not the problem [02:30] <burek> dadada, how do you install things on that router and why don't you install ffmpeg/ffplay? [02:31] <burek> mka is audio only file right? [02:31] <llrcombs> it's an MKV with an alternate extension indicating that there are no video streams [02:31] <dadada> burek: opkg install package [02:31] <burek> i see [02:32] <burek> can you remux it into mp4 and try if it plays better, llrcombs ? [02:32] <burek> dadada, then dpkg -i ffplay :) [02:32] <dadada> burek: the repository has ffmpeg but no ffplay [02:33] <burek> then install ffmpeg (it will install ffplay too) [02:34] <llrcombs> burek: muxing with ffmpeg into any other encapsulation format causes the audio to play smoothly in VLC, but the duration decreases (as if it had been squeezed inward) [02:34] <burek> llrcombs, that would indicate that vlc might not support mka properly [02:34] <burek> or your mka is damaged [02:35] <llrcombs> burek: the stream or the mux? [02:35] <burek> (remuxing into other container proved that your audio stream is ok) [02:35] <burek> mka as a format/mux [02:35] <llrcombs> mka is an encapsulation format, never a codec, FYI [02:36] <llrcombs> and that doesn't explain the duration decreasing whenever I use any other encapsulation format [02:36] <dadada> burek: I installed the ffmpeg package, but no ffplay there, as I said, it's stripped down [02:37] <burek> llrcombs, that's why I would decode audio stream into wav [02:37] <burek> and check out the real duration [02:37] <burek> to see if mka is erroneous or not [02:37] <burek> before I do any other remuxing and testing [02:37] <llrcombs> burek: alrighty, reencoding as a wav. What -acodec do I give it? Just "wav"? [02:38] <burek> just set the output to out.wav [02:38] <burek> ffmpeg will figure it out [02:38] <llrcombs> writing... [02:38] <burek> dadada, hmh, let me think [02:40] <llrcombs> when it's a .wav, it shows the same, smaller duration [02:40] <burek> that means that is the real duration [02:40] <llrcombs> so maybe I need to stretch it out somehow, because it needs to be 2h4m31s [02:41] <burek> now, base your conclusions on that [02:41] <llrcombs> (the shorter one is 1h59m25s) [02:41] <llrcombs> I need it longer so it matches with the video [02:41] <burek> maybe it's damaged somewhere, so it has been truncated [02:42] <llrcombs> my thought is that the actual stream is 1h59m25s long, there's an MKV header saying to stretch it out, and VLC is interpreting the header wrong [02:43] <burek> why would the header tell anyone to stretch the audio stream? [02:43] <llrcombs> and that ffmpeg can convert it to a different codec in an MKA while keeping the audio long, because MKAs have that header, but in any other encapsulation format, it jumps to the real duration [02:43] <llrcombs> burek: so it syncs with the video [02:43] <burek> no [02:43] <burek> vlc does that by itself [02:43] <burek> or, should i say [02:43] <burek> it doesn't exactly do that, out of the box [02:43] <llrcombs> it'd need to know how how long it's supposed to be, though [02:43] <burek> it syncs the time stamps of audio/video [02:44] <burek> like any other player [02:44] <llrcombs> MKV/MKA definitely has a timescale header option [02:44] <burek> but it doesn't just stretch the audio to fit video, because they are of the different size [02:44] <llrcombs> meant for syncing up a video and audio track that are different lengths, but should be the same [02:45] <burek> well, if mka is something new, maybe no proper support has been written for it, yet? [02:45] <llrcombs> MKV/MKA aren't new [02:45] <burek> why don't you just use audacity or some other audio editor [02:45] <burek> and stretch your audio manually [02:45] <burek> and finish the task [02:46] <llrcombs> do you know how to do that in audacity, or shall I google? [02:46] <burek> dadada, doo you know how to use pipes? mkfifo and stuff [02:47] <burek> llrcombs, I think it's not a rocket science :) [02:47] <llrcombs> I just don't use audacity much [02:47] <burek> should be some option like stretch [02:47] <dadada> burek: yeah, mkfifo not often, but pipes I like [02:47] <burek> or so [02:47] <llrcombs> looks like google's got me covered on that [02:47] <burek> llrcombs, use your favorite audio editor then :) [02:47] <llrcombs> "change tempo" [02:47] <burek> dadada, try this: [02:47] <llrcombs> burek: I don't edit audio much in general :P [02:47] <burek> mkfifo out.wav /dev/dsp [02:47] <llrcombs> but I've got it, thanks [02:47] <burek> ffmpeg -i file.mp3 out.wav [02:47] <burek> :) [02:48] <burek> llrcombs, ok :) :beer: :) [02:49] <burek> dadada, or ln -s /dev/dsp out.wav [02:50] <llrcombs> oh wow, Audacity 2.0 [02:50] <llrcombs> or ffplay [02:50] <llrcombs> oh, you don't have that [02:50] <llrcombs> nevarmind [02:51] <dadada> burek: hmm, ffmpeg is missing libavfilter ... [02:51] <dadada> damn, that package is probably not well maintained [02:51] <burek> dadada :/ [02:52] <burek> try checking out what console players are available for linux [02:52] <burek> and try them one by one.. [02:52] <burek> llrcombs, I think it might be possible to stretch/shrink streams with ffmpeg and -vf option [02:52] <burek> using PTS something [02:53] <burek> but I need to check it out [02:55] <dadada> burek: I got ffmpeg running [02:56] <burek> weee :) [02:59] <burek> llrcombs, try reading this: http://ffmpeg.org/ffmpeg.html#setpts [02:59] <burek> I know someone was talking about it [02:59] <burek> some time ago here [02:59] <burek> on this channel [02:59] <llrcombs> burek: reading... [02:59] <dadada> burek: if I direct a wav file to /dev/dsp its pure garbage sound [02:59] <burek> about using setpts filter to speed up/slow down streams [02:59] <dadada> like white snow tv [02:59] <burek> dadada, try typing ffmpeg -formats [02:59] <rogelio> hola [02:59] <dadada> /dev/dsp probablly expects something like pcm [02:59] <rogelio> hello [02:59] <burek> you are looking for pcm something [02:59] <burek> yes [03:00] <llrcombs> that's for video, though [03:00] <rogelio> does anyone know how to crop borders with ffmpeg? [03:00] <burek> dadada, so just add -f pcm_something out.wav [03:00] <rogelio> i tried using -croptop -cropleft [03:00] <rogelio> but it aint include in my windows build [03:00] <burek> rogelio, http://ffmpeg.org/ffmpeg.html#crop [03:00] <dadada> burek: I se no pcm there [03:00] <burek> cropleft is deprecated long time ago [03:01] <burek> dadada, you should see at least 10 of them [03:01] <dadada> burek: I see 0 [03:02] <dadada> burek: http://fpaste.org/71UX/ [03:02] <rogelio> burek i have already tried that one, but it said: "Failed to set value '100:100:250:250' for option 'crop') [03:02] <burek> rogelio, [03:02] <burek> can you please use pastebin.com, to show your command line and its output? [03:02] <rogelio> yes [03:03] <dadada> dadada: sorry for not using pastebin and using my distros version of the same thing :-) kidding [03:04] <burek> dadada, I think that build of ffmpeg is useless to you [03:04] <burek> :/ [03:04] <rogelio> burek, here http://pastebin.com/A04qiuJ0 [03:04] <dadada> burek: what about "alsa" format? [03:05] <burek> hm [03:05] <burek> you can try it [03:05] <burek> but I'm not sure if it will work [03:05] <dadada> yeah, I tried it with -acodec alsa , says not supported [03:05] <burek> anyway, sox, I think, has an option of creating a virtual audio device [03:05] <burek> so you can mix several outputs into an input device [03:06] <burek> and stuff like that [03:06] <burek> so you might be able maybe to exploit that? :) [03:06] <burek> rogelio, [03:06] <burek> crop is not an option for ffmpeg, its a video filter [03:06] <burek> so you need to do something like this [03:07] <burek> ffmpeg ... -vf 'crop=...' [03:07] <dadada> burek: sox is slow as hell.... or did you mean rogeleo? [03:07] <burek> rogelio, if you read http://ffmpeg.org/ffmpeg.html#crop you should see there are examples at the bottom [03:08] <dadada> also strange thing, I converted ogg to wav with ffmpeg but it plays slooow, whereis until now every wav I tried on the router played normal [03:08] <burek> dadada, sox was for you, but let me check are there any other console players for linux [03:08] <burek> that you can use [03:08] <burek> can you play that wav on your computer [03:08] <burek> to see if it's damaged [03:11] <rogelio> burek thanks it worked [03:12] <burek> dadada, http://moc.daper.net/ [03:12] <burek> rogelio, :beer: :) [03:13] <dadada> burek: http://downloads.openwrt.org/snapshots/trunk/ar71xx/packages/ the repository I can use is pretty short :-) [03:13] <dadada> maybe I could try gst, but maybe not simple to setup I don't know, has so many darn packages [03:13] <burek> dadada, I hope you are not chatting over telnet? :) [03:14] <burek> you're timing out frequently :) [03:14] <burek> http://ffmpeg.gusari.org/irclogs/ffmpeg.log.20120408 [03:14] <dadada> no, it's just my wifi that makes probs [03:15] <dadada> thanks, that logs doesn't show the timeouts though [03:15] <burek> it shows just chat text that google can index :) [03:16] <burek> and it's real time [03:16] <dadada> in other words, this is totally not private [03:16] <burek> for archive, you can check ffmpeg's mailing lists [03:16] <burek> yes, it's not :) [03:16] <dadada> heh [03:16] <dadada> no problem [03:16] <burek> it says in the topic too [03:17] <burek> my friend would use to say: if you want privacy, just stay at home :) [03:17] <burek> but we are already at our homes, so.. :) [03:17] <dadada> makes sense, I also log most of my irc stuff, after all it's a significant part of my life ... [03:17] <dadada> that says something about my life omg [03:17] <dadada> heh [03:18] <burek> :) [03:18] <burek> that's ok :) we still love you :))) [03:18] <dadada> :P [03:18] <burek> but you should practice /join #real_life from time to time ^^ [03:19] <dadada> I just checked, nobody in that channel [03:19] <burek> :O it's time to wake up then, Neo :) [03:20] <dadada> rather to sleep, I'm a night-owl the last couple of weeks, I should try to normalize my rhythm so that I could function in day-life, but it's already too late at 03:20 [03:21] <burek> same here :) [03:22] <burek> but I need to do some more things, so I'm not gonna sleep yet [03:23] <dadada> I don't need to do things, but I've anxities about the next day so I rather stay in "this day" and try to avoid tranfering into the next day-day or something like that... of course you can always find things to do on the command line [03:24] <burek> you should read more on astral projections :) that's a cool way to spend a night, while sleeping :) [03:24] <burek> you soon realize you have so much spare time :) [03:24] <dadada> do you do that? [03:25] <burek> :) [03:25] <dadada> seriously? [03:26] <burek> it's kind offtopic, but :) let me just say that it doesn't take any of your "real" time :) [03:26] <burek> you would spend it in sleep anyway :) [03:26] <burek> so, you can try if you like :) [07:55] <_SKiTZO> ive tried using -loop_input and -loop in combination with -t to have my audio input repeated, however i just get the input once and then silence. [07:55] <_SKiTZO> is it even possible to loop audio? it works great with video [08:13] <Khmar> Hello everyone, I'm trying to create a screencast using ffmpeg and I get messages like "[mpeg @ 0x7ca590]buffer underflow i=0 bufi=337262 size=341185". At the end, when I hit q, it seems to have recorded everything but there are lots of video frames skipped resulting in a choppy video. I tried passing -bufsize but then hitting `q` to exit the program doesn't work at all. [08:17] <Khmar> In case it's relevant, this is my command line - "ffmpeg -f x11grab -s 1280x800 -r 50 -i :0.0 -f alsa -i pulse -sameq ./emacsmovies.org/screencast.mpg" [08:39] <RPG-Master> I'm not sure if this is outside of y'all's expertice, but I keep getting this error from PS3 Media Server: [08:39] <RPG-Master> INFO 2012-04-08 00:33:02.299 [New I/O server worker #1-15] Starting transcode/remux of 02 - Breathe.flac [08:39] <RPG-Master> INFO 2012-04-08 00:33:02.832 [ffmpeg-20] Process ffmpeg has a return code of 1! Maybe an error occurred... check the log file [08:53] <brimestone_home> hey guys, how do i use ffmpeg to convert a progressive 59.94fps to 29.976fps [08:53] <brimestone_home> so technically, creating a slowmo effects [10:48] <burek> RPG-Master, [10:48] <burek> can you please use pastebin.com, to show your command line and its output? [10:48] <burek> or the entire log file [15:35] <Guest33361> hi. I want to create thumbnails from videos with ffmpeg, but at least for AVI files it seems that ffmpeg needs to read the whole file [15:36] <Guest33361> is there a way to generate thumbnails fast, without ffmpeg needing to read the whole file? [15:36] <Guest33361> I've seen that in ffplay, there is a "seek by bytes" option [15:37] <Guest33361> I couldn't understand how it works, but may be seeking by bytes instead of time could do the job [15:37] <Guest33361> is there a way to do this in ffmpeg? [15:39] <Guest33361> no way? [16:15] <zap0> thumbnailer, there are many methods, each file type is different. a video is an abstract concept. accept it. [16:15] <thumbnailer> .F [16:15] <thumbnailer> ok [16:16] <thumbnailer> but the fact that I don't need very exact seeking may help [16:16] <thumbnailer> I just want to create some thumbnails from a movie [16:16] <thumbnailer> and I don't need to seek to an exact position or key frame [16:17] <thumbnailer> theoretically, a rough seek in a file [16:17] <thumbnailer> and then searching nearby for keyframe should do the job [16:17] <thumbnailer> isn't it? [16:22] <burek> thumbnailer, [16:22] <burek> which command line did you use so far [16:22] <thumbnailer> I've tried ffmpegthumbnailer in the first place [16:23] <burek> did you try ffmpeg in the first place [16:23] <thumbnailer> yes [16:23] <thumbnailer> for example [16:23] <burek> can you please use pastebin.com, to show your command line and its output? [16:23] <thumbnailer> I didn't have any problems with extracting or thumbnailing [16:23] <thumbnailer> but it was too slow [16:24] <burek> I understand [16:24] <burek> can you show your command [16:24] <thumbnailer> because my files are on network location [16:24] <thumbnailer> yes [16:24] <thumbnailer> sure [16:24] <thumbnailer> just a moment [16:27] <thumbnailer> ffmpeg -ss 100 -vframes 1 -s 320x240 -i video.mp4 thumb.jpg [16:27] <thumbnailer> and I want to use it on a HTTP url [16:27] <thumbnailer> like [16:27] <thumbnailer> ffmpeg -ss 100 -vframes 1 -s 320x240 -i http://host.com/video.mp4 thumb.jpg [16:27] <burek> ok, so [16:28] <burek> if you try to do it on a format, that saves info about streams inside in the front of the file [16:28] <burek> then you are ok [16:28] <burek> otherwise it will be fun :) [16:28] <thumbnailer> but it seems that ffmpeg reads the file until it reaches the 100 second reaches [16:28] <burek> mp4 has the ability to save global headers [16:28] <thumbnailer> so, what that it means? [16:28] <burek> atoom structures [16:28] <burek> and similar at the front of the file [16:29] <burek> so that you retrieve that first [16:29] <relaxed> thumbnailer: That command is wrong. It should be ffmpeg -ss 100 -i http://host.com/video.mp4 -vframes 1 -s 320x240 thumb.jpg [16:29] <burek> that tells you which frames are keyframes [16:29] <thumbnailer> sorry, actually I've tested it with AVI [16:29] <burek> so you can seek it, etc, without downloading everything [16:30] <burek> and yes, relaxed is right, your command would read only 1 frame from the input [16:30] <burek> instead of generating 1 frame at the output [16:30] <burek> the order of options does matter [16:30] <thumbnailer> oh [16:30] <thumbnailer> i didn't knew this [16:30] <relaxed> and it defines the input as 320x240 [16:30] <relaxed> instead of scaling to 320x240 [16:32] <thumbnailer> I removed -s [16:33] <burek> http://ffmpeg.org/ffmpeg.html#image2-2 [16:34] <burek> try ffmpeg -i http://host.com/video.mp4 -ss 100 -vframes 1 -f image2 -s 320x240 thumb.jpg [16:36] <thumbnailer> you mean: ffmpeg -ss 100 -i http://host.com/video.mp4 -vframes 1 -f image2 -s 320x240 thumb.jpg [16:36] <burek> no [16:36] <burek> try what I offered [16:36] <thumbnailer> tried both [16:37] <burek> and? [16:37] <relaxed> "-ss 100" before the input seeks to the position; after it decodes to the position. [16:37] <thumbnailer> the one you said needs to get the whole file [16:37] <thumbnailer> it seems [16:38] <burek> if you set -ss before -i then ffmpeg needs to get meta data from the file, in order to know where are keyframes to be able to properly seek [16:38] <thumbnailer> when putting "-ss 100" before the input, it says: Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) [16:38] <burek> if you put it after -i then ffmpeg will decode stream as it downloads it and will use that info to seek to the desired position [16:38] <burek> can you please use pastebin.com, to show your command line and its output? [16:39] <thumbnailer> yes, but pastebin.com is not accessible here [16:39] <thumbnailer> this one is ok? [16:39] <thumbnailer> http://www.heypasteit.com/ [16:39] <burek> yes [16:39] <thumbnailer> but it's blocked here [16:39] <thumbnailer> you know? [16:39] <burek> ok, use that one [16:39] <thumbnailer> :-( [16:39] <burek> btw google indexes pastebin.com in the matter of seconds [16:40] <burek> it's the only site that I know of that has such a privilege [16:40] <thumbnailer> google even started to index what you see [16:40] <thumbnailer> :-) [16:40] <thumbnailer> with the latest internet-connected glasses [16:42] <thumbnailer> it doesn't exit [16:42] <thumbnailer> because the file is large [16:42] <thumbnailer> I have to press q to exit [16:42] <thumbnailer> but if I wait, it will do the job [16:42] <thumbnailer> I press q and send the result [16:43] <burek> then use -ss before -i [16:43] <burek> whatever does the work done [16:43] <burek> gets* [16:43] <thumbnailer> ok [16:44] <burek> anyway, if it works with -ss before -i does it do what you wanted? [16:45] <burek> brb [16:45] <thumbnailer> ok [16:48] <thumbnailer> This is the first (ss after input) [16:48] <thumbnailer> http://www.heypasteit.com/clip/0AA8 [16:48] <thumbnailer> and this is the second [16:48] <thumbnailer> (ss before input)) [16:48] <thumbnailer> http://www.heypasteit.com/clip/0AA9 [17:01] <thumbnailer> :-) [17:01] <thumbnailer> it seems that the problem is with the nerwork module [17:01] <thumbnailer> because it does the job great on local files [17:01] <thumbnailer> nearly instant [17:31] <thumbnailer> are u there? [17:37] <thumbnailer> what if i want to extract more frames? [17:37] <thumbnailer> -vframes 2 ? [18:32] <level09> why is my x264 video not running on html5 players ? [18:51] <thumbnailer> I think even with -ss before input [18:51] <thumbnailer> FFMPEG reads all the file [18:51] <thumbnailer> :-( [19:44] <cryptopsy> does anyone know how to keep the same bitrate on sox? sox is a tool that uses ffmpeg i think, that can be used to trim or split songs (at least that's what i'm using it for, amongst other features) [20:38] <juanmabc> i guess you are alone at "man sox" and searching for rate, bit-rate [21:56] <cryptopsy> juanmabc: it is mentioned only as a definition [21:57] <cryptopsy> are you telling me it's impossible? give me a flying fucking break [22:26] <relaxed> cryptopsy: If your sox has ffmpeg support then I would guess it could stream copy if all you're doing is cutting. [22:27] <relaxed> meaning you wouldn't lose any quality. [22:29] <relaxed> Keep in mind this isn't #sox and you have not really give any details about what format you're dealing with. [22:31] <cryptopsy> there is no #sox [22:31] <cryptopsy> ffmpeg and mplayer channels are like the video and sound editing communities on this network [22:32] <relaxed> Which is why I answered your question. [22:32] <cryptopsy> if sox cannot perform, then ffmpeg gains another user by figuring out how to trim audio [22:33] <relaxed> sox is much better at trimming audio [22:34] <relaxed> decode the audio to wav -> trim with sox and output wav -> compress wav to flac/alac via ffmpeg [22:35] <cryptopsy> mm actually i think sox uses lame [22:35] <relaxed> That method should be lossless. [22:36] <relaxed> if sox has ffmpeg support it can use pretty much anything. [22:38] <relaxed> So if your source is mp3, I assume trimming with sox would stream copy the section you want to the output. I'm not a 100% sure though. [22:39] <relaxed> sox does have a mailing list. [22:40] <cryptopsy> i'd rather use lame or ffmpeg if it's possible than email an abandoned software [22:42] <relaxed> It's not abandoned. Use whatever you like. [22:43] <cryptopsy> there is a fine line that is actually a really fat line which is drawn when determining if a software is abadoned. on myspace, girls will tilt their fat faces at an angle, to appear more desirable by the bourgeois. in software, mailing lists and websites are kept online, despite authors having no interest in fixing the issue. [22:46] <highzeth> how one can claim a project being abandoned, when the last commit to it was less than a day ago, well, that brings a new definition to the word for me [22:49] <relaxed> highzeth: Because there's no irc channel with people falling all over theirselves to help Mr. Piss Poor Attitude here. [22:50] <cryptopsy> highzeth: people like to participate, despite no long-term goals by the original authors [22:50] <cryptopsy> people write patches instead of starting their own forks under obscure names [22:51] <cryptopsy> relaxed: IRC channels are not made for helping single people. we live in a society, where people help each other and in exchange you get something done for you [22:51] <cryptopsy> that's what the smurfs was about, do you know the story? [22:51] <highzeth> relaxed: hehe [22:51] <cryptopsy> in the smurfs, doing a job meant getting 10 jobs done in return for you [22:51] <cryptopsy> that's what communities are about [22:52] <cryptopsy> irc channels are communities [22:55] Action: relaxed had Smurfs bedspread and pillowcases as well as all the cups from Hardee's when he was a kid [22:55] <relaxed> It was pretty smurfy. [22:58] <relaxed> I also had a cat named Azrael. [23:58] <cnx> i run ffmpeg -f mpeg -vc 1 -tvstd NTSC -i /dev/video0 -y test.mpg i get a few lines but when i hit control c no file is created, why? [00:00] --- Mon Apr 9 2012

1 0

ffmpeg-devel.log.20120407
by burek 08 Apr '12

08 Apr '12

[00:12] <ubitux> it seems threading failed again [00:12] <ubitux> (http://fate.ffmpeg.org/report.cgi?time=20120406212225&slot=x86_64-archlinux…) [02:03] <michaelni> ubitux, can you bisect this easily/automatically in case its a regression ? (some bug IIRC also pointed to a thread regression that only happens sometimes) [02:03] <michaelni> like maybe running the failing test 10 times and if all pass consider it a pass for git bisect [02:04] <michaelni> anyway, i need to sleep a bit, ill read read the irclog tomorrow [02:05] <ubitux> it will take quite a while to run this i guess [02:05] <ubitux> i don't know if it's a regression, i setup the threads instances recentrly [02:05] <ubitux> recently* [02:06] <iive> is there something that could be done to easy tracking threading problems? [02:06] <seriousman> valgrind's tools? [02:07] <iive> no... i mean, during normal operation. like lock holding blocking for a second or more... [02:07] <iive> something that would give useful debug information even when not specifically debugging. [02:07] <seriousman> looks like a good idea [02:08] <seriousman> divert pthread_mutex_lock() with a variant that calls usleep with random values? [02:09] <iive> i'm not familiar with pthread functions, isn't there some with a timeout? [02:10] <seriousman> not for locks [02:10] <seriousman> i meant to use dlopen tricks [02:11] <seriousman> e.g. pthread_mutex_lock(pthread_mutex_t *m) { real_pthread_mutex_lock(m); usleep(random()); } [02:14] <kierank> just write a macro to measure how long each lock takes [02:14] <kierank> and then printf something [02:18] <iive> from the man page PTHREAD_MUTEX_ERRORCHECK looks like nice thing to try [02:25] <iive> seriousman: i think i was more like using try_lock() and a sleep if it failed, so a check could be done after few iterations. [02:26] <iive> hum, there is timedlock() what does exactly what I want. [02:27] <iive> what/that [02:27] <iive> time to sleep. [02:27] <iive> have fun. [07:07] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rb717f35bfb 10ffmpeg/tests/lavf-regression.sh: [07:07] <CIA-17> ffmpeg: fate: try to fix >8bit dpx reg test [07:07] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:18] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rab553612e7 10ffmpeg/libavcodec/put_bits.h: [08:18] <CIA-17> ffmpeg: put_bits: switch assert to av_assert2() [08:18] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:18] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3fb945a935 10ffmpeg/libavformat/mpegenc.c: [08:18] <CIA-17> ffmpeg: mpegpsenc: fix user specified mux rate [08:18] <CIA-17> ffmpeg: Fixes Ticket1175 [08:18] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rfe5b7db07a 10ffmpeg/configure: [11:27] <CIA-17> ffmpeg: build system: support setting avassert() assert_level [11:27] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rd40132dab3 10ffmpeg/libavcodec/proresenc_kostya.c: [11:27] <CIA-17> ffmpeg: proresenc_kostya: fix encoded mantissa. [11:27] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:28] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * re3b7079ddf 10ffmpeg/ (libavcodec/adpcmenc.c tests/ref/acodec/adpcm_ima_qt): [11:28] <CIA-17> ffmpeg: adpcmenc: fix encoded s_0 value. [11:28] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:28] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r76e8b336cd 10ffmpeg/libavcodec/proresenc_kostya.c: [11:28] <CIA-17> ffmpeg: proresenc_kostya: Mention likely ancestry in the license header. [11:28] <CIA-17> ffmpeg: As the encoder contained the same bug and has similar structure [11:28] <CIA-17> ffmpeg: to anatoliys encoder, it is possibly based on it. [11:28] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:28] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rdd7d08ee89 10ffmpeg/libavcodec/proresenc_anatoliy.c: [11:28] <CIA-17> ffmpeg: proresenc_anatoly: fix encoded mantissa. [11:28] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:54] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * recd7455e96 10ffmpeg/libavcodec/ (aaccoder.c aacenc.c): [11:54] <CIA-17> ffmpeg: aacenc: Fix issues with huge values of bit_rate. [11:54] <CIA-17> ffmpeg: Do not pointlessly call ff_alloc_packet2 multiple times, [11:54] <CIA-17> ffmpeg: and fix an infinite loop by clamping the maximum [11:54] <CIA-17> ffmpeg: number of bits to target in the algorithm that does [11:54] <CIA-17> ffmpeg: not use lambda. [11:54] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [11:54] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r796097c696 10ffmpeg/tests/ (6 files in 2 dirs): (log message trimmed) [11:54] <CIA-17> ffmpeg: Add tests for audio encoders: aac, dca, ra144 and nellymoser. [11:54] <CIA-17> ffmpeg: The PSNR values are of varying usefulness, though at least [11:54] <CIA-17> ffmpeg: the DTS and AAC ones are useful with the right shift value. [11:54] <CIA-17> ffmpeg: Note: due to usage of floats some of these may fail on other [11:55] <CIA-17> ffmpeg: architectures. [11:55] <CIA-17> ffmpeg: In that case they should be converted into a CMD = stddev [12:16] <btsimonh> hi, trying to use ffmpeg to capture/forward webcam on openwrt, but not getting any output file. Anyone any experience of ffmpeg in small environments? [12:44] <michaelni> btsimonh, what are you doing exactly, i mean command line, error messages, ... [12:45] <btsimonh> i think it's something to to with the build, but all looks ok. [12:46] <btsimonh> let me dig out the simplest example.... [12:48] <btsimonh> e.g. ffmpeg -f video4linux2 -i /dev/video0 /tmp/video.avi - trouble now is i've just done a rebuild, and now i don't even get any stdout feedback :(, but also get no error and no output file? [12:54] <btsimonh> trying to get back to where i was.. just rebooting the router [13:04] <btsimonh> @michaelni: http://pastebin.com/4FJ0uc7t - output of ffmpeg on openwrt [13:05] <btsimonh> mjpg-streamer runs, and produces video no problem. when mjpg-streamer is running, ffmpeg can't get video, when it's not, it 'seems' to be working (but low cpu usage...), but after ctrl-c, no file produced. [13:09] <btsimonh> (i wanted to use ffpmeg to stream video and audio from this webcam to a distribution website; mjpg-streamer is great, but my upload speed won't support more than one user) [13:25] <michaelni> btsimonh, i suggest you try latest ffmpeg [13:26] <michaelni> and if it still fails try to run it under gdb and see where it hangs [13:26] <btsimonh> what's it on at the moment, verison-wise? [13:27] <michaelni> latest release in is topic :) [13:30] <btsimonh> 0.10.2 :) i'll see if i can update the source in my openwrt tree later, a bit scary going that many versions forward :) [13:31] <ubitux> releases are done more often than previously [13:31] <ubitux> it's not such a huge "jump" [13:47] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r10e3dd15a5 10ffmpeg/tests/ (codec-regression.sh ref/acodec/aac ref/acodec/nellymoser): [13:47] <CIA-17> ffmpeg: Remove AAC and nellymoser tests that do not work. [13:47] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [13:47] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r6d3166fcf0 10ffmpeg/tests/ (Makefile fate/aac.mak): [13:47] <CIA-17> ffmpeg: Add AAC FATE encode tests. [13:47] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [13:56] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r087b9c326f 10ffmpeg/tests/lavf-regression.sh: [13:56] <CIA-17> ffmpeg: fate: try to force le for the dpx decoder side too [13:56] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:59] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r2f83681c79 10ffmpeg/ (11 files in 3 dirs): [13:59] <CIA-17> ffmpeg: lavfi: port libmpcodecs remove-logo filter [13:59] <CIA-17> ffmpeg: The code is based on the remove-logo filter in MPlayer/libmpcodecs, by [13:59] <CIA-17> ffmpeg: Robert Edele, relicensed to LGPL with consent of the author. [13:59] <CIA-17> ffmpeg: Address trac issue #249. [13:59] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r6cb01c9dba 10ffmpeg/ (5 files in 3 dirs): [13:59] <CIA-17> ffmpeg: lavfi/mp: remove remove_logo wrapper [13:59] <CIA-17> ffmpeg: removelogo has been ported to libavfilter, and the wrapper was not [13:59] <CIA-17> ffmpeg: working. [13:59] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r32b1022f76 10ffmpeg/doc/filters.texi: doc/filters: remove reference to removed delogo wrapper [13:59] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * rfd00662d5a 10ffmpeg/doc/muxers.texi: [13:59] <CIA-17> ffmpeg: doc/muxers: update documentation for the framecrc muxer [13:59] <CIA-17> ffmpeg: Add mention to the pts and duration log fields, added in [13:59] <CIA-17> ffmpeg: d2afbd9a56e4e863bad01923006dd9b77f747035. [13:59] <CIA-17> ffmpeg: Fix trac ticket #1159. [15:13] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rcb25a186eb 10ffmpeg/tests/ (codec-regression.sh ref/acodec/dca regression-funcs.sh): [15:13] <CIA-17> ffmpeg: Fix DCA regression test to work across architectures. [15:13] <CIA-17> ffmpeg: Only the decoding step is not bit-exact, so avoid checksum for that. [15:13] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [15:16] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r82ca1f8004 10ffmpeg/tests/ (lavf-regression.sh ref/lavf-fate/mp3): [15:16] <CIA-17> ffmpeg: Test mp3 container muxing (and thus ID3 writing). [15:16] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [16:35] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r4395c058d2 10ffmpeg/ (libavcodec/dcaenc.c tests/codec-regression.sh): [16:35] <CIA-17> ffmpeg: dcaenc: Allow encoding without specifying a channel_layout. [16:35] <CIA-17> ffmpeg: It will print a warning, making the behaviour consistent [16:35] <CIA-17> ffmpeg: with the AC3 encoder. [16:35] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [16:35] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * re80ce4adea 10ffmpeg/tests/ (lavf-regression.sh ref/lavf-fate/mp3): [16:35] <CIA-17> ffmpeg: Do not decode the mp3 data in lavf test since that is not bit-exact. [16:36] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r192efcf768 10ffmpeg/libavcodec/h264_ps.c: [20:57] <CIA-17> ffmpeg: h264_ps: check log2_max_frame_num for validity. [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r6560fa390e 10ffmpeg/libavcodec/get_bits.h: [20:57] <CIA-17> ffmpeg: get_bits: check the number of bits parameter through av_assert2() [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rd3b0fb1048 10ffmpeg/libavcodec/put_bits.h: [20:57] <CIA-17> ffmpeg: put_bits: add av_assert2() to check out of array writes. [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r6c249392fa 10ffmpeg/libavcodec/mpeg4videodec.c: [20:57] <CIA-17> ffmpeg: mpeg4videodec: make sure f/b_code are not invalid values. [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r20676711fe 10ffmpeg/libavcodec/vc1.c: [20:57] <CIA-17> ffmpeg: vc1: fix incorrect show_bits() usage. [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r6c3d6a214c 10ffmpeg/libavcodec/alsdec.c: [20:57] <CIA-17> ffmpeg: alsdec: Check k used for rice decoder. [20:57] <CIA-17> ffmpeg: Values that fail this check will cause failure of decode_rice() [20:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:11] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r5012c07336 10ffmpeg/ (4 files in 2 dirs): (log message trimmed) [22:11] <CIA-17> ffmpeg: Support detecting and demuxing EIA-608 subtitles in mov. [22:11] <CIA-17> ffmpeg: The format is slightly proprietary. [22:11] <CIA-17> ffmpeg: DVDs use a format of [22:11] <CIA-17> ffmpeg: code byte (0x00, 0x01, 0xfe or 0xff), two data bytes [22:11] <CIA-17> ffmpeg: MOV uses instead [22:11] <CIA-17> ffmpeg: cdat/cdt2 atom, two data bytes [23:03] <CIA-17> ffmpeg: 03Joakim Plate 07master * rba24f12982 10ffmpeg/libavformat/utils.c: [23:03] <CIA-17> ffmpeg: libavformat: Only require first packet to be known for audio/video streams [23:03] <CIA-17> ffmpeg: It can take a long time before subtitles or data streams show up, [23:03] <CIA-17> ffmpeg: so we shouldn't wait for those before assuming we have all info [23:03] <CIA-17> ffmpeg: for streams. [23:03] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [23:03] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r4648fe6324 10ffmpeg/avconv.c: avconv: remove now pointless variables. [23:03] <CIA-17> ffmpeg: 03Anton Khirnov 07master * re06d2dfcde 10ffmpeg/ (Changelog avconv.c configure): [23:03] <CIA-17> ffmpeg: avconv: drop support for building without libavfilter. [23:03] <CIA-17> ffmpeg: Since the mandatory memcpy in vsrc_buffer has been eliminated, there [23:03] <CIA-17> ffmpeg: shouldn't be any significant reason to build without lavfi anymore. [23:03] <CIA-17> ffmpeg: This will make upcoming support for complex filtergraphs easier to do. [23:03] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r951e715ceb 10ffmpeg/avplay.c: [23:03] <CIA-17> ffmpeg: avplay: Don't try to scale timestamps if the tb isn't set [23:03] <CIA-17> ffmpeg: If get_filtered_video_frame failed above, tb might not be [23:03] <CIA-17> ffmpeg: initialized at all, so don't scale using it. [23:03] <CIA-17> ffmpeg: This fixes cases where avplay could crash if aborting [00:00] --- Sun Apr 8 2012

1 0

ffmpeg.log.20120407
by burek 08 Apr '12

08 Apr '12

[00:59] <CoasterMaster> what does --version-3 flag enable (i.e. what does enabling version 3 give me that I wouldn't have otherwise) [01:08] <JEEB> CoasterMaster, you usually have to enable it for certain libraries [01:08] <JEEB> for example libvo_aacenc needs v3 [01:08] <JEEB> since it is apache licensed [01:09] <JEEB> I don't remember if ffmpeg itself had anything v3-only [01:12] <CoasterMaster> JEEB: thanks...i thought i read somewhere that there was a second prores decoder that's used if --version-3 is enabled (and possibly something about optimizations), but I can't remember where I read that and I can't find it again so it might be just be me making things up [01:12] <JEEB> one of the prores implementations might have used it, true [01:12] <JEEB> anyways, mostly related to third-party libraries [01:12] <CoasterMaster> alrighty, thanks :) [01:48] <llrcombs> looks like the issue I mentioned was fixed in the latest verison, nevermind [04:29] <Sedici> hi [04:30] <Sedici> is there anyway to join to videos? [04:30] <Sedici> using ffmpeg? [04:50] <relaxed> Sedici: look at concat in the man page. [04:58] <Sedici> thanks relaxed, i saw a way to it, but it MUST be a mpeg file, if i encode mi mp4 file to a mpeg i will lose quality [04:58] <Sedici> i was looking for a way that dont lose quality [05:11] <relaxed> MP4Box can concat losslessly [05:11] <relaxed> it's part of gpac [05:16] <Sedici> thanks [05:16] <Sedici> ill try it [05:26] <llrcombs> OK, strangeness [05:26] <llrcombs> I take back what I said about that stuff being fixed [05:27] <llrcombs> when I try to reencode an MKA file with an AAC track in any format other than AAC, ffmpeg scales it down [05:28] <llrcombs> (e.g. the 2h6m-ish file is now 1h59m-ish) [05:57] <llrcombs> has anyone got this? [05:57] <Sedici> relaxed mp4box worked great, thanks! [05:59] <relaxed> you're welcome [06:42] <llrcombs> Has anyone ever seen ffmpeg change an audio file's duration when converting from one encapsulation and codec to another? [06:51] <relaxed> llrcombs: can you put up a sample of your source somwhere? [06:52] <llrcombs> relaxed: not sure if it'll still happen in a sample. I'm trying something real quick; if it doesn't work, I'll upload a sample in a sec [06:52] <relaxed> the whole source is fine [07:06] <llrcombs> relaxed: got a reccomended host for a 117MB file? Mediafire's complaining about a nonexistant permissions error [07:07] <relaxed> llrcombs: dropbox http://db.tt/cDWPLzP [07:12] <llrcombs> uploading... [07:12] <llrcombs> not the entire source audio/video, just the audio track extracted from the original MKV as an MKA [07:13] <llrcombs> (it also occurs in another AAC audio track from the same file, which I also have as an MKA, but just the one should demonstrate the issue) [07:28] <llrcombs> well, that took long enough [07:28] <llrcombs> relaxed: here http://dl.dropbox.com/u/46348978/Spirited%20Away%20720p%20HDTV%20(1).mka [07:37] <llrcombs> relaxed: ping [14:27] <raven> hi [14:30] <raven> ~2 years old wmv-files - several "ignoring overflows" and massively faulty frames [14:30] <raven> any ideas what's wrong there/changed anything? [16:03] <zap0> use a ~2 year old ffmpeg [16:03] <zap0> raven, boop! [16:04] <raven> useful hint [18:00] <orangey> hello all [18:01] <orangey> I'm going crazy with my videos all rendering / remaining out of sync. They're rendered by kdenlive, which depends on FFMPEG using this command: f=mp4 hq=1 acodec=aac ab=%audiobitrate+'k' ar=48000 pix_fmt=yuv420p vcodec=libx264 minrate=0 vb=%bitrate+'k' g=250 bf=3 b_strategy=1 subcmp=2 cmp=2 coder=1 flags=+loop flags2=dct8x8 qmax=51 subq=7 qmin=10 qcomp=0.6 qdiff=4 trellis=1 aspect=%dar pass=%passes . [18:01] <orangey> However, even when i try to restore sync, I just can't. [18:02] <orangey> the sync is off by 300ms, and the only way to even come close to restoring it is to use avidemux and re-encode as something other than x264 / aac [18:02] <juanmabc> -vsync or -async ? [18:03] <orangey> juanmabc: as in, add that option? [18:03] <orangey> which would you recommend? what's the difference? [18:03] <juanmabc> yep, it's sync from video or sync from audio, both should sync [18:04] <juanmabc> just various methods [18:07] <orangey> juanmabc: neither seem to do it [18:07] <orangey> juan can I do a copy with async / vsync? [18:08] <juanmabc> then wait for another answer [18:11] <orangey> juanmabc: does this command look right: ffmpeg -i In.mp4 -vsync 1 -acodec copy -vcodec copy -ss 00:00:02 -t 10 test.mp4 [18:11] <orangey> ? [18:13] <juanmabc> vsync does not take args, and i guess copy does nothing to the stream [18:15] <orangey> juanmabc: that was my fear. [18:15] <orangey> though I guess because async takes args I figured vsync must.. [18:15] <orangey> anyhow. [18:15] <orangey> thank you for your advice [18:42] <Ultimatespirit> How do I make ffmpeg find libx264? I am confident I compiled it correctly and installed it [18:42] <Ultimatespirit> it being libx264 [18:47] <juanmabc> --enable-libx264 [18:48] <Ultimatespirit> I mean when I did that it said ERROR: libx264 not found [18:49] <juanmabc> --help | grep -i x264 or --help | grep -i path [18:51] <Ultimatespirit> that returns " --enable-libx264 enable H.264 encoding via x264 [no]" [18:52] <juanmabc> and the path ? or lib search ? [18:53] <Ultimatespirit> assuming path meant the path to the libx264 binary it returned nothing [18:53] <Ultimatespirit> as just path it returned some general information [18:53] <juanmabc> nope, just a literal "path" [18:53] <Ultimatespirit> if it helps any i'm running mac osx [18:54] <Ultimatespirit> in which case it says " [18:54] <Ultimatespirit> wwops [18:55] <Ultimatespirit> http://pastebin.com/5FvKaswm [18:55] <Ultimatespirit> it wouldn't let me use ffmpeg.pastebin.com for some reason... [18:57] <juanmabc> then grep "dir" and/or "lib" [18:58] <Ultimatespirit> i did both: http://pastebin.com/6UPM7i2e [20:11] <llrcombs> relaxed: ping [20:36] <maujhsn> Can anybody give me the commands to use to do dvd or cd authoring with ffmpeg! [20:37] <maujhsn> I will working with data files! [21:44] <relaxed> llrcombs: Hey. Let me download it and take a look. [22:08] <relaxed> llrcombs: I encoded the file to mp3 in mka and the durations are pretty much the same. http://pastebin.com/kMKJZWAF [22:09] <llrcombs> relaxed: I've been encoding to .mp3s and such, would that make a difference? [22:10] <relaxed> Are you using a recent version of ffmpeg? [22:10] <relaxed> I'm encoding to .mp3 now [22:15] <DelphiWorld> hi all [22:15] <DelphiWorld> how can i list my actual audio/video codecs ? [22:15] <DelphiWorld> ubuntu server 10.04 [22:16] <juanmabc> ffmpeg -codecs [22:18] <DelphiWorld> LOL [22:19] <DelphiWorld> juanmabc: thank you but i'm unable to read them all :| [22:19] <juanmabc> ffmpeg -codecs | grep -i "xvid" [22:21] <briMestone_home> hey guys.. im using FFmpeg to dump a video to a raw codec and then from that to a specified codec. my question is how do i send the output of the first ffmpeg to the input of the second ffmpeg task to skip the intermediate file ( large file ) [22:21] <briMestone_home> so something like piping the output of the 1st one to the input of the second ffmpeg [22:22] <Mavrik> you can use pipes [22:22] <Mavrik> but the first question is - why would you want to do that? [22:22] <briMestone_home> well im trying to conform a 60p footage to playback at 23.98 [22:23] <briMestone_home> so from 59.94psf to 23.98psf [22:23] <briMestone_home> making the footage longer since it will play slower [22:24] <DelphiWorld> juanmabc: i'm trtying to multicast a satellite transponder through MumuDVB [22:25] <briMestone_home> right now, i write it this way. ffmpeg -i 00003.MTS -f rawvideo -pix_fmt yuv444p -vcodec rawvideo -s 1920x1080 -y raw.raw; ffmpeg -f rawvideo -pix_fmt yuv444p -sameq -r 23.98 -s 1920x1080 -i raw.raw -y 00003.MP4 [22:27] <briMestone_home> no one? [22:28] <DelphiWorld> briMestone_home: no, two :) [22:28] <briMestone_home> reallly? thats the easiest way to do it? [22:28] <relaxed> llrcombs: Hmm, I'm not sure why that's happening. [22:28] <Mavrik> briMestone_home: I still don't get why you're running ffmpeg twice instead of doing a straight conversion [22:28] <briMestone_home> how? [22:29] <Mavrik> ffmpeg -i 00003.MTS -f rawvideo -pix_fmt yuv444p -r 23.98 -s 1920x1080 00003.MP4 [22:29] <briMestone_home> if i do it in one instance, ffmpeg will throw the excess frames within a second [22:29] <Mavrik> is the equivalent of your command [22:29] <Mavrik> "-sameq" doesn't do what do you think it does so don't use it [22:30] <Mavrik> you'll want to pass h.264 encoding bitrate and parameters for that [22:30] <briMestone_home> what it does is 1 minute of footage creates 1 minute of output [22:30] <briMestone_home> i want to create a slow-mo effects by re-arranging the actual number of frames [22:31] <Mavrik> briMestone_home: er yea, put -r before -i [22:31] <Mavrik> same with -f and -pix_fmt [22:31] <Mavrik> and you'll get the same effect [22:31] <briMestone_home> let me test that.. [22:33] <briMestone_home> man.. i would kiss your toes if this work as plan [22:33] <briMestone_home> or what ever yer highness [22:33] <Mavrik> if not, you can try this: http://blog.grio.com/2012/01/fast-and-slow-motion-video-with-ffmpeg.html [22:45] <briMestone_home> that site is not what im trying to achive, they force it to slow via interpolation [22:45] <JEEB> if you want ffmpeg to read something at a lower frame rate, just set -r before the input (-i) [22:45] <JEEB> it overwrites the input's frame rate [22:45] <briMestone_home> ffmpeg -r 23.98 -i 00004.MTS -pix_fmt yuv444p -s 1920x1080 00004.MP4 [22:46] <briMestone_home> it still played at realtime speed [22:47] <JEEB> dunno, that's how it's supposed to go IIRC :V [22:47] <briMestone_home> redoing it like [22:47] <briMestone_home> ffmpeg -r 23.98 -i 00004.MTS -pix_fmt yuv444p -s 1920x1080 -r 23.98 00004_1.MP4 [22:52] <briMestone_home> any more idea? [22:59] <maujhsn> Mavrik can you give me the commands to use to do dvd or cd authoring with ffmpeg? [23:18] <kilo> HI IM in a pickle! Trying to make an ios5.1 arm7 wrapper of ffmpeg + lame getting this Undefined symbols for architecture armv7: "_ff_dot_product", referenced from: _estimate_pitch in libavcodec.a(g723_1.o)"_ff_acelp_weighted_vector_sum" [00:00] --- Sun Apr 8 2012

1 0

ffmpeg-devel.log.20120406
by burek 07 Apr '12

07 Apr '12

[00:17] <bcoudurier> what filter_complex proposal ? [00:19] <bcoudurier> ubitux, just implement the right way from the start [00:55] <CIA-17> ffmpeg: 03Carl Eugen Hoyos 07master * r76c79aa28f 10ffmpeg/ (libavcodec/raw.c libavformat/nut.c): [00:55] <CIA-17> ffmpeg: Support yuva444p rawvideo in nut. [00:55] <CIA-17> ffmpeg: Fixes ticket #1058. [01:25] <jb_far> ffv1 stores 10bits in 16bits chroma? [02:24] <FishB8> Just wanted to drop in here to say, that if anybody is interested in playing with a sample of the Sony SR codec, (from the F65 camera) a sample file is available from the Mango Project [02:24] <FishB8> http://mango.blender.org/production/format-woes/ [02:32] <kierank> "MXF (again, this would not be readable for FOSS)" [02:32] <kierank> why not [02:32] <kierank> what's wrong with FOSS reading mxf [02:33] <j-b> yeah [02:34] <j-b> EXR? [02:44] <FishB8> I think the statement about not being readable with FOSS was in reference to the video steam contained in it rather than demuxing MXF itself [02:48] <Compn> FishB8 : well we are interested, thanks for sample [02:49] <Compn> it'd be really nice if someone would actually take the time for a proper bugreport, just a simple ffmpeg -i sample.exr out.avi would suffice [02:49] <Compn> i'm guessing we'll hit resolution limit bug and some other things [02:49] <Compn> well resolution limit isnt a bug but ... yeah [02:50] <Compn> guess i can stick it in the ole ffmpeg bug tracker [02:50] <Compn> j-b : can vlc handle those samples ? :) [02:50] <j-b> not tested [02:50] <j-b> and I do not really care. :) [02:50] <FishB8> I don't think ffmpeg supports OpenEXR [02:55] <Compn> mm [02:55] <Compn> theres a patch [02:56] <Compn> michaelni : what do you say about taking over the openexr patch and doing a security review on it ? [02:56] <Compn> michaelni : http://ffmpeg.org/pipermail/ffmpeg-devel/2010-April/081747.html [03:00] <Compn> hmm wheres stefano [03:13] <Compn> FishB8 : i put exr bug ticket in , you can follow it here > https://ffmpeg.org/trac/ffmpeg/ticket/1172 [03:19] <FishB8> Nice. Would make the workflow of getting from EXR to DCP a little less crazy. [05:46] <CIA-17> ffmpeg: 03Baptiste Coudurier 07master * re636aa1a56 10ffmpeg/libavformat/ (isom.c mov.c): [05:46] <CIA-17> ffmpeg: movdec: handle 0x7fff langcode as macintosh per the specs [05:46] <CIA-17> ffmpeg: The correct point that seperates ISO and MAC language codes is 0x400 [05:46] <CIA-17> ffmpeg: according to the current QT spec. Old QT specs did not list where this [05:46] <CIA-17> ffmpeg: seperation is but apparently only defined the meaning of the first 137. [05:46] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:17] <michaelni> Compn, this patch is amongth the most buggy ive seen since a while but ill try to get it in better shape [06:17] <michaelni> av_log(avctx, AVERROR_NOFMT, ... [06:17] <michaelni> wtf [08:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r9e91a7d45b 10ffmpeg/tests/ref/lavfi/ (6 files): [08:47] <CIA-17> ffmpeg: fix fate tests after 76c79aa28fc8cf0bfe52915317a3cfc72383d0ef [08:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:47] <CIA-17> ffmpeg: 03Jimmy Christensen 07master * rb7ce3242f8 10ffmpeg/ (10 files in 4 dirs): [08:47] <CIA-17> ffmpeg: lavc: add a OpenEXR decoder [08:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r634c01bc18 10ffmpeg/libavcodec/exr.c: [08:47] <CIA-17> ffmpeg: exr: fix av_logs() [08:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rf148537c41 10ffmpeg/libavcodec/exr.c: [08:47] <CIA-17> ffmpeg: exr: various cleanup and security related fixes [08:47] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [10:41] <ubitux> bcoudurier: this one: https://lists.libav.org/pipermail/libav-devel/2012-April/025289.html [11:18] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3826d53f15 10ffmpeg/README: [11:18] <CIA-17> ffmpeg: README: be a tiny bit more verbose [11:18] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:18] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r25eae29ea1 10ffmpeg/libavfilter/vsrc_buffer.c: [11:18] <CIA-17> ffmpeg: vsrc_buffer: remove overwrite support. [11:18] <CIA-17> ffmpeg: the overwrite flag became unused when merging support for buffering [11:18] <CIA-17> ffmpeg: multiple frames. [11:18] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:18] <CIA-17> ffmpeg: 03Thierry Foucu 07master * re0f30a567f 10ffmpeg/libavcodec/s302m.c: [11:18] <CIA-17> ffmpeg: s302m: Add Channle Layout for 6 channels [11:18] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:00] <Compn> michaelni : wow. you work fast. :) [14:01] Action: Compn sends mail to jimmy christensen asking if any of his other patches were missed [14:02] <ubitux> should we make a "playground fate" system for ppl with slow machines? [14:03] <ubitux> (a repository where you push, and a hook call fate, report the results, and reset to its old state or sth like this) [14:06] <Compn> do you mean so multiple machines can submit to the same test? or ? [14:08] <ubitux> i'd say the report is made available for a while (or sent my email or sth) [14:08] <ubitux> but maybe it's not worth the effort [14:11] <CIA-17> ffmpeg: 03Clément BSsch 07master * re5df74dc56 10ffmpeg/ffmpeg.c: [14:11] <CIA-17> ffmpeg: ffmpeg: remove unused meta_data_maps. [14:11] <CIA-17> ffmpeg: This is unused since a7b5e841ffe4b0f8423288965b8d069bd2a7a792. [14:14] <ubitux> mmh and of course the typedef could be removed as well actually. [14:21] <ubitux> hey [14:21] <ubitux> threads=8 just failed with a h264 test [14:21] <ubitux> h264-conformance-cama2_vtc_b [14:21] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20120406121601&slot=x86_64-archlinux… [14:21] <ubitux> maybe i should add more thread instances [14:25] <CIA-17> ffmpeg: 03Clément BSsch 07master * r735fb8ce59 10ffmpeg/ffmpeg.c: [14:25] <CIA-17> ffmpeg: ffmpeg: remove unused MetadataMap. [14:25] <CIA-17> ffmpeg: This struct was forgotten after [14:25] <CIA-17> ffmpeg: e5df74dc56494068042f094d43a5d774c257ae69. [14:44] <michaelni> kierank, do you have some spec about s302m or can you explain how the channel maping is supposed to work ? [15:02] <kierank> there is no channel mapping [15:02] <kierank> it's that simple [15:04] <michaelni> how is s302m presented to the user if theres no channel mapping anywhere ? [15:07] <nevcairiel> isnt it essentially a studio/mastering codec? maybe the people using it are supposed to know wtf they are doing? :) [15:14] <kierank> yes the user decides how to map the channels [15:16] <michaelni> kierank, in this case the question still is what is wrong with the patch setting the most likely layout ? [15:16] <michaelni> i mean if the user has to override it anyway [15:17] <kierank> nothing i suppose as long as the user can override it easilyt [15:17] <michaelni> it doesnt make the patch or previous addition of layouts correct sure ... [15:17] <michaelni> but also doesnt really seem to worsen things [15:19] <ubitux> couldn't we just call av_get_default_channel_layout()? does it belong to the decoder? [15:20] <michaelni> i think the default for 8 channels is different [15:22] <ubitux> is it less appropriate than the arbritrary choosen one here? [15:23] <CIA-17> ffmpeg: 03Alexander Strasser 07master * rf363773042 10ffmpeg/doc/ffmpeg.texi: [15:23] <CIA-17> ffmpeg: doc: image2: update docs for glob input patterns [15:23] <CIA-17> ffmpeg: * adapt examples to new syntax [15:23] <CIA-17> ffmpeg: * mention that glob chars need to be enabled by a preceding % char [15:23] <CIA-17> ffmpeg: * note that globbing will be performed if both a printf and globbing [15:23] <CIA-17> ffmpeg: pattern would be possible judging from the input pattern [15:23] <CIA-17> ffmpeg: Signed-off-by: Alexander Strasser <eclipse7(a)gmx.net> [15:24] <michaelni> i dont know, i suspect what is in s302m.c would represent what is most likely to occur in real world s302m [15:24] <michaelni> or at least thats what should be in it [15:28] <ubitux> yeah right, ok [18:55] <ubitux> what's the common solution to pass a decode option to the generic decode functions? [18:56] <ubitux> add a field in the AVCodecContext? [18:56] <ubitux> mmh seems to be sth like this& [19:51] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * re8b1da007e 10ffmpeg/libavcodec/nellymoserenc.c: [19:51] <CIA-17> ffmpeg: nellymoserenc: fix crash due to memsetting the wrong area. [19:51] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [19:51] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r099a1875f0 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/dpx): [19:51] <CIA-17> ffmpeg: dpxenc: also test 10 and 16 bit formats. [19:51] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [19:51] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r50361e5144 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/mov): [19:51] <CIA-17> ffmpeg: Add regression test for mov rtp hinting. [19:51] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:15] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rcbb7183498 10ffmpeg/tests/tiny_psnr.c: [20:15] <CIA-17> ffmpeg: tiny_psnr: allow searching for optimal shift value. [20:15] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:32] <bcoudurier> s302m channel order should single channel [20:33] <bcoudurier> using anything else is most likely wrong [21:22] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rbd128e9bff 10ffmpeg/libavcodec/exr.c: [21:22] <CIA-17> ffmpeg: exr: fix mixed declaration and statements. [21:22] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:27] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r5cc51a5847 10ffmpeg/ (5 files in 3 dirs): [23:27] <CIA-17> ffmpeg: avutil: Add av_get_default_channel_layout() [23:27] <CIA-17> ffmpeg: Also, use the new function in the AC-3 encoder. [23:27] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rbb0618e68b 10ffmpeg/libavcodec/ (audioconvert.c audioconvert.h): [23:27] <CIA-17> ffmpeg: avcodec: remove avcodec_guess_channel_layout() [23:27] <CIA-17> ffmpeg: It is not public because the header is not installed, and its functionality [23:27] <CIA-17> ffmpeg: has been replaced by av_get_default_channel_layout(). [23:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2e0c360abd 10ffmpeg/: (log message trimmed) [23:27] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [23:27] <CIA-17> ffmpeg: * qatar/master: [23:27] <CIA-17> ffmpeg: cosmetics: Align muxer/demuxer declarations [23:27] <CIA-17> ffmpeg: mpeg12: Do not change frame_pred_frame_dct flag and demote error into a warning [23:27] <CIA-17> ffmpeg: avcodec: remove avcodec_guess_channel_layout() [23:27] <CIA-17> ffmpeg: avutil: Add av_get_default_channel_layout() [00:00] --- Sat Apr 7 2012

1 0

ffmpeg.log.20120406
by burek 07 Apr '12

07 Apr '12

[00:08] <N0hbdy> Hey all, is anybody knowedgeable about how ffmpeg calculates the FPS for a h264 video file? [00:14] <sacarasc> It's written in the header files, IIRC. [00:14] <sacarasc> *files header [00:14] <beastd> N0hbdy: Not sure. But chances that someone will help out are usually higher if you describe a more concrete situation/exaple. I won't help out because I will go to sleep now. Good luck. [00:14] <N0hbdy> beastd: thanks for the heads up, sleep well :) [00:15] <N0hbdy> sarcasasc: thats for the tip, thats what I thought as well [00:15] <JEEB> yes, if it's just a bit stream there can be a timestamp/whatever in the header. In most cases a bit stream is in a container tho, which means that you usually have a time stamp for every frame [00:16] <N0hbdy> use case is simply that I have a video that ffmpeg is saying is 13.11 fps, but when i step through frame for frame, I see 25 frames go past every second.... [00:16] <N0hbdy> so I'm trying to figure out why the 13.11 is being reported [00:16] <N0hbdy> but if its a header thing, then that helps a lot for narrowing down where to look for the issue [00:16] <JEEB> is it a bit stream or an actually muxed file? [00:16] <JEEB> s/file/stream/ [00:17] <N0hbdy> muxed file [00:18] <JEEB> I would guess that in real usage the H.264 stream's header is going to be ignored and the timestamps from the container will be used [00:19] <JEEB> also, there are many files where the frame rate is not constant, which might throw off any calculation of a "frame rate" for that file [00:20] <JEEB> checking the container's time stamps would be a good beginning [00:22] <N0hbdy> okay, thanks [00:50] <ckb> can I specify an absolute path to a preset? [00:54] <sacarasc> -fpre [01:11] <ckb> -vpre too? [01:12] <sacarasc> -vpre is for installed ones. -fpre is to give a file name/path. [01:57] <ckb> ah [01:57] <ckb> I can take the ffpresets off one machine and put it on the other? [02:31] <relaxed> Why do you need them? Current ffmpeg uses libx264's internal presets. [02:31] <relaxed> ckb: ^^ [02:33] <relaxed> What preset are you using? [02:33] <ckb> fast_firstpass [02:33] <ckb> and medium [02:34] <ckb> but I'm moving code from one machine to another [02:34] <sacarasc> The fast firstpass one is done automatically on the first pass now, I believe. [02:34] <sacarasc> And medium is ridiculously out of date. [02:34] <ckb> so what else should I use? [02:35] <ckb> I'm doing this on a web server currently [02:35] <ckb> I'll be looking to get a strict encoding server shortly [02:35] <ckb> so I'm not trying to zombify the machine [02:36] <relaxed> medium is now the default [02:36] <ckb> hm [02:36] <ckb> so I don't even need to specify any presets? [02:36] <sacarasc> I was thinking of normal, then... Or something. [02:37] <sacarasc> Sorry, ckb. I'm ill and barely coherent. [02:37] <relaxed> ckb: If you're using a recent version, no. [02:37] <ckb> 0.7.8? [02:37] Action: relaxed passes sacarasc a red potion [02:37] <sacarasc> \o/ [02:37] <ckb> sacarasc: I just came down with the flu ;( [02:37] <ckb> broke a 101 temp last night [02:37] <sacarasc> :( [02:37] <sacarasc> Nice. [02:38] <relaxed> maybe, 10.1 is the latest. [02:38] <ckb> doc gave me some syrup though :D [02:38] <relaxed> is it red? [02:39] <ckb> ;) [02:40] <ckb> so I definitely need to tell the hosting company to update ffmpeg [02:40] <ckb> ridic [02:40] <sacarasc> Are you using ffmpeg or libav? [02:40] <ckb> er? libav is a codec right? [02:40] <sacarasc> It's a fork of ffmpeg that uses the same binary names. [02:40] <ckb> -i gives me libavutil [02:41] <ckb> but it says ffmpeg version 0.7.8 [02:41] <sacarasc> By which dev team? [02:41] <ckb> ffmpeg devs [02:41] <sacarasc> Okay, that is ffmpeg. [02:42] <ckb> so.. original question: can I use the preset from one machine to the other? [02:42] <ckb> or should I just update ffmpeg? [02:42] <relaxed> you may not need it. [02:42] <sacarasc> You can, but updating is generally better anyway. [02:44] <ckb> damn [02:44] <ckb> my company uses managed hosting [02:46] <joe7817> Hi all, I can't seem to be able to successfully set up a live stream to ffserver. FFserver seems to disconnect and not accept any feeds from FFmpeg [02:47] <joe7817> I can see that ffmpeg is issuing a POST command to ffserver, but ffserver disconnects before accepting input. But if I point ffmpeg to a listening netcat, it's able to stream [02:52] <joe7817> Any idea why this might be happening? I'm trying to look through the source now, but I can't make heads or tails of it. [02:53] <funyun> hi. can someone help me with this error? http://pastebin.com/QZuHjcUv [02:58] <funyun> the error is "Application provided invalid, non monotonically increasing dts to muxer in stream 0: 67 >= 33" [02:58] <funyun> anyone know how i can fix this? [03:01] <funyun> anyone alive? [03:02] <funyun> i also get this error av_interleaved_write_frame(): Invalid argument [03:16] <funyun> is everyone on spring break or something? [03:28] <funyun> hi. can someone help me with this error? http://pastebin.com/QZuHjcUv [03:30] <funyun> sacarasc: you there? [03:34] <sacarasc> funyun: ? [03:35] <funyun> sacarasc: you usually help me when i have a problem. i was wondering if you could help me with this one? [03:35] <sacarasc> I was about to go to bed. I am ill and it's 02:35... [03:35] <funyun> okay, thanks anyway [03:41] <funyun> relaxed: any idea? [04:06] <relaxed> funyun: Try demuxing the streams, encode the audio and then remux. If all else fails use mkvextract and MP4Box. [05:03] <funyun> relaxed: i tried your last suggestion, it works but the audio is very out of sync. i don't understand your first suggestion "Try demuxing the streams, encode the audio and then remux"? [06:42] <relaxed> funyun: extract the h264 and audio, encode the audio, then copy them into the container of your choice [09:15] <kiloflame> trying to compile lame + ffmpeg + ios for mp3 export... can anyone help me with my config flags? [09:15] <kiloflame> END /var/folders/fk/mmxfjf157b73q4t04cbfttxr0000gn/T//ffconf.jNrjMfBc.c /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc --sysroot=/Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS5.1.sdk -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -DPIC -arch armv7 -mfpu=neon -miphoneos-version-min=5.1 -mcpu=cortex-a8 -std=c99 -fomit-frame-pointe [09:16] <kiloflame> error: lame/lame.h: No such file or directory ERROR: libmp3lame >= 3.90 not found [09:20] <grepper> kiloflame: you have libmp3lame development files installed ? [09:21] <kiloflame> thats with these build flags: yes they are in /usr/local/lib [09:21] <kiloflame> and lame compiles in ffmpeg fine until i add my ios5.1 flags [09:21] <kiloflame> so its like i can have one or the other - lame or ios, but not both [09:22] <kiloflame> ./configure --enable-static \ --disable-shared --enable-pic \ --sysroot="/Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS5.1.sdk" \ --extra-ldflags="-arch armv7 -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS5.1.sdk -miphoneos-version-min=5.1" \ --enable-debug \ --extra-cflags="-arch armv7 -mfpu=neon -miphoneos-version-min=5.1" \ [09:23] <kiloflame> --arch=arm --target-os=darwin \ --cc="/Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc" \ --cpu=cortex-a8 \ --enable-gpl \ --enable-debug \ --enable-small \ --disable-asm \ --enable-libmp3lame [09:24] <kiloflame> sorry http://pastebin.com/ES3HzTGv [13:44] <meco> Is ffmpeg command line only? I want to rotate a video 90? but I don't understand what to write to effect this. [13:44] <meco> Actually I want to rotate it 270?.. [13:47] <burek> meco, it is cmd line only [13:48] <burek> take a look at transpose filter [13:48] <meco> burek: Could you assist me and give me the command I need to this operation? [13:48] <burek> http://ffmpeg.org/ffmpeg.html#transpose [13:48] <meco> ok [13:50] <burek> ffmpeg -i input.avi -vf transpose=0 -vcodec libx264 out.mp4 [13:50] <burek> ffmpeg -i input.avi -vf 'transpose=0' -vcodec libx264 out.mp4 [13:51] <meco> Cool.. I'll try it! [13:54] <meco> "Unknown encoder 'libx264'". It's an mp4 file. [13:54] <JEEB> then you just don't have libx264 compiled (installed) into your ffmpeg [13:54] <JEEB> it's a separate library that provides H.264 encoding [13:55] <meco> Can I deal with that easily? [13:55] <JEEB> well, in most cases you would have to re-compile ffmpeg :) [13:56] <meco> ok [13:56] <JEEB> (if you're on windows finding another binary is also an alternative) [13:56] <burek> ffmpeg -i input.avi -vf 'transpose=0' out.avi [13:56] <burek> can you please use pastebin.com, to show your command line and its output? [13:56] <JEEB> note how you are not specifying a codec [13:56] <JEEB> it will select something most probably quite bad [13:57] <meco> I'm on Linux [13:57] <burek> meco, please provide pastebin [13:59] <meco> http://pastebin.com/PAEURjkS [13:59] <burek> Video: h264 [13:59] <burek> Audio: aac [14:00] <burek> well for audio you can use -acodec copy [14:00] <burek> and for video, you need libx264 if you want it to remain as h264 [14:00] <burek> are you using 64bit linux? [14:01] <meco> I am trying 'ffmpeg -i in.avi -vf "transpose=1" out.avi' [14:01] <burek> that command line will select some codec automatically, just like JEEB has said [14:02] <burek> in most cases it ends up being bad quality [14:02] <burek> but you can try your: ffmpeg -i input.mp4 -vf "transpose=1" out.mp4 [14:02] <meco> OK. I'll just check the result before proceeding. [14:02] <burek> and see what happens [14:02] <burek> anyway meco, are you using 64bit linixu [14:02] <burek> linux* [14:03] <meco> 32 [14:03] <burek> ok [14:03] <meco> Yes, I changed to mp4.. [14:03] <burek> can you please use pastebin.com, to show your command line and its output? [14:04] <JEEB> that just changes the output container to mp4 and still just picks some codec that happens to be possible to use in that container ^^; [14:04] <burek> input is also mp4 [14:06] <meco> http://pastebin.com/16WuUU39 - nice result [14:07] <burek> Stream #0.0(und): Video: mpeg4, yuv420p, 240x320, q=2-31, 200 kb/s, 15 tbn, 15 tbc [14:07] <burek> it reencoded video as mpeg4 video [14:07] <burek> if you want to influence the quality, you can play around with -b parameter [14:07] <burek> for example: ffmpeg -i input.mp4 -vf "transpose=1" -b 10000k out.mp4 [14:08] <burek> see http://ffmpeg.org/ffmpeg.html for more info [14:13] <meco> I appreciate the help. I'm jsut doing an emergency edit, so the result I've got as satisfactory. [14:14] <meco> is [15:26] <woozly> guys, how to make FF plays mp4 file ? without full download it? [15:26] <woozly> I mean, FF can't play .mp4 stream [15:26] <woozly> :/ [15:26] <woozly> it waits until .mp4 downloads [15:27] <Mavrik> does "FF" mean "Firefox"? [15:27] <sacarasc> Use qt_faststart on the mp4 to move the moov atom to the front of the file. [15:27] <woozly> Mavrik: yes.. [15:27] <woozly> sacarasc: oh... I'll try! [15:27] <Mavrik> yeah, Firefox doesn't support H.264/AAC [15:27] <woozly> oh :( [15:28] <Mavrik> so you're out of luck with that [15:56] <meekohi> Guys I don't understand this "Out of tree builds are impossible with config.h in source dir." Does that just mean I need to run ./configure after cd to the source directory? [16:07] <meekohi> staging /usr/local/src/ffmpeg-0.10.1$ yasm --version [16:07] <meekohi> yasm 1.2.0 [16:07] <meekohi> staging /usr/local/src/ffmpeg-0.10.1$ sudo ./configure [16:07] <meekohi> yasm not found, use --disable-yasm for a crippled build [16:07] <meekohi> whyyyy [16:09] <JEEB> o_O wat @ sudo [16:13] <meekohi> JEEB: Using puppet [16:14] <JEEB> I would check if you have the same path being used [16:14] <JEEB> also check config.log for more info [16:15] <meekohi> config.log is full of so much junk -- what do you look for? [16:16] <meekohi> yasm --version [16:16] <meekohi> ./configure: line 628: yasm: command not found [16:18] <meekohi> So when configure runs yasm it isn't found, but when I run it, it is. [16:18] <meekohi> Does configure use a different path? [16:18] <JEEB> not configure itself but how you run it, possibly [16:18] <JEEB> since you seem to run it in some specific way [16:19] <meekohi> Right now I'm just running it on the command line with sudo [16:19] <JEEB> sudo -s [16:19] <JEEB> check path [16:19] <JEEB> see the PATH difference basically [16:20] <meekohi> identical [16:20] <JEEB> I dunno, in all the cases I've run configure (I've never run it under sudo tho) the PATH has been used as-is [16:20] <meekohi> gotcha. [16:50] <meekohi> What do the "demuxers" do? How do I know if I need them when compiling? [17:14] <JEEB> meekohi, take a container and demux streams out of it [17:15] <JEEB> demux/splitting usually means taking various bit streams / information out of a container [17:15] <JEEB> also usually you don't need to disable things manually [17:15] <meekohi> How is that different from a parser? [17:16] <meekohi> In this case I do. I am getting weird errors when I include some things, but not others, and am trying to narrow down what causes the problem. [17:16] <JEEB> o_O [17:18] <meekohi> Consistent with all my previous experiences, ffmpeg is about the hardest piece of software to compile of all time. [17:18] <meekohi> I have never had it go smoothly on any machine. [17:18] <JEEB> funny enough my experiences have been mostly the opposite [17:19] <JEEB> except for msys/mingw where some extra hacks were needed [17:19] <meekohi> I'd be happy to hire you for a few hours ;D I'm trying to get it working on Amazon's Linux AMI. [17:20] <meekohi> I can get it to compile fine with the settings I used previously [17:20] <meekohi> Now I'm trying to add support for image2/mjpeg [17:20] <meekohi> And am not having much luck. [17:20] <JEEB> my configure options are usually rather minimalistic [17:20] <JEEB> I usually just disable networking and hwaccels on windows tho [17:21] <meekohi> It doesn't help that the compile time on ec2 is like 10 minutes :P [17:21] <meekohi> So everything I try takes forever. [17:22] <JEEB> on windows the configuration script and make distclean takes minutes too [17:22] <JEEB> because windows isn't exactly liking the fork() ing [17:22] <JEEB> har har [17:22] <meekohi> Alas ;) [17:22] <JEEB> I did find myself using some voodoo options I had picked up during the years some time ago [17:22] <JEEB> like still definining the define for static w32pthreads [17:23] <JEEB> even though I'm using w32threads [17:23] <JEEB> (as in native win32 threading) [17:25] <meekohi> Whatever is going wrong causes this error: [17:25] <meekohi> libavcodec/libx264.c:73: error: duplicate member 'stats' [17:25] <meekohi> seems like a bug in my opinion, but I am so confused about what's going on I hate to file a report. [17:25] <JEEB> which gcc btw? [17:27] <meekohi> 4.4.6 [17:27] <JEEB> hmm [17:28] <JEEB> http://permalink.gmane.org/gmane.comp.video.ffmpeg.cvs/48399 [17:28] <JEEB> I would guess your source code is older than this commit? [17:29] <meekohi> Nice find. I'm using 0.10.1 so I assume yeah [17:29] <meekohi> Do you think this is included in 0.10.2? [17:30] <JEEB> I have no idea, but the commit did reach the 0.10.x branch at least... I don't really keep up on the dates of various releases [17:30] <JEEB> if 0.10.2 was released after that commit, it should have it :) [17:30] <meekohi> Craziness. I'll try upgrading. [17:30] Action: JEEB just usually compiles git HEAD because there are usually no real reasons to use the releases [17:31] <meekohi> Same day lol [17:31] <meekohi> Yeah& the guy who set this up before me is all about picking specific releases and sticking with them. We have this system for automatically configuring new servers& :P [17:31] <meekohi> Hugely over-designed in my opinion, but whatever. [17:35] <meekohi> Ha. That patch was the ONLY change between 10.1 and 10.2 [17:35] <meekohi> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?p=ffmpeg.git;a=log;h=refs/ta… [17:35] <JEEB> fun [17:35] <meekohi> I wonder how many hours I just wasted on that... [20:57] <ffizz> hello. i'm trying to convert a video from mp4 to flv but the file size stays nearly the same. any tips on reducing the file size of the resulting flv? [20:58] <burek> can you please use pastebin.com, to show your command line and its output? [20:59] <ffizz> http://pastebin.com/gffiGwRt @burek [21:03] <burek> ffizz, that paste is missing the output [21:04] <ffizz> burek, wait have to convert again to capture the output [21:04] <burek> just press ctrl+c when it starts encoding [21:06] <ffizz> burek: http://pastebin.com/1aW1CZ0K [21:07] <burek> your input has [21:07] <burek> Stream #0:0: Video: mpeg4 [21:07] <burek> Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16, 128 kb/s [21:07] <burek> and what is your goal? why do you want to convert it to flv [21:08] <burek> because flv is a file format and an encoder, so its kinda ambiguous [21:08] <ffizz> burek: i have a website where i want to setup streaming video [21:08] <burek> and you use web player, like jw player or flow player? [21:08] <ffizz> burek: flowplayer [21:09] <burek> then you only need to remux into flv container [21:09] <burek> did you try ffmpeg -i input.avi -acodec copy -vcodec copy -f flv output.flv [21:09] <ffizz> burek: i also want to reduce the filesize to save on bandwith and storage costs [21:09] <ffizz> burek: trying it now [21:10] <burek> that will create a flv file containing the original streams (mpeg4 and mp3) [21:10] <burek> without any quality loss (it will just copy the data) [21:10] <burek> i see [21:10] <burek> then use something like this [21:12] <burek> ffmpeg -i input.avi -acodec libmp3lame -ab 64k -vcodec libx264 -preset slow -f flv output.flv [21:14] <ffizz> burek: trying it now [21:16] <relaxed> a ratecontrol is fine too [21:16] <burek> yes, -b 300k for example [21:18] <burek> ffmpeg -i input.avi -acodec libmp3lame -ab 64k -vcodec libx264 -preset slow -b 320k -f flv output.flv [21:18] <ffizz> without the -b 300k it still creates a large file. 5 MB for 30 seconds of video. trying it now with -b 300k [21:19] <ffizz> what i'm trying to do is get the same quality as the major adult tube sites [21:19] <relaxed> to restream or ? [21:20] <ffizz> relaxed: i have my own adult videos in various formats (mp4, avi, wmv) and try to stream them on my tube [21:21] <ffizz> just very new to ffmpeg and video encoding. [21:21] <burek> then you are probably looking for 2 pass encoding [21:21] <burek> which can target the file size [21:22] <ffizz> burek: how can i do that? [21:22] <relaxed> and will give you a better average bitrate [21:22] <ffizz> or where can i read about it? [21:23] <burek> actually that's an old stuff [21:23] <burek> http://ffmpeg.org/ffmpeg.html [21:23] <burek> search for -pass [23:55] <llrcombs> so, something weird's going on with a file I've got [23:56] <llrcombs> I'm trying to take an MKA with one AAC audio track and reencode it using faac as an M4A (not just a remux, an actual reencode [current encode plays choppily in VLC for some reason, but reencoded, it's fine]) [23:57] <llrcombs> when I reencode it, it goes from being 2h4m to 1h59m [23:57] <llrcombs> muxed with the video it goes with, it has a very bad desync [23:57] <llrcombs> variable desync, which can't be corrected with a delay [00:00] --- Sat Apr 7 2012

1 0

ffmpeg-devel.log.20120405
by burek 06 Apr '12

06 Apr '12

[01:48] <ramiro> michaelni: the thing is that the granularity is huge. it's 1s. it can be made smaller, but then the table grows proportionally. the index table doesn't allow setting the real timestamp inside that 1s. [01:48] <ramiro> for example in the regression tests the real timestamp is 960, but it gets indexed as 1900 [01:49] <ramiro> and the table is designed to work with backward seeks. so forward seeks shouldn't use it. [01:50] <ramiro> imo the index table should be used as an initial estimate. find the packet from the table, then look for the next and the previous keyframe and determine their real pts. from that, look further if necessary or just return the frame if it's found [01:52] <ramiro> or just drop the table altogether in asfdec and use normal binary search [02:16] <burek> is there a way to select certain/specific alsa audio output device with ffplay [03:57] <burek> is it too hard to make a feature, just like in vlc, that will allow ffmpeg to retrieve youtube videos directly? [04:10] <Compn> burek : no, either using a wrapper to youtube-dl.py or just copying what vlc does [04:10] <Compn> should only take probably 20 mins or less by any smart devel [04:11] <burek> http://www.vbforums.com/showpost.php?s=d7c80ec67591eccfca871a0e1c0a8d1e&p=3… [04:11] <burek> :) [04:13] <burek> yes, I noticed that a lot of people have this solved [04:13] <burek> like JW Player (web video flash player) [04:15] <burek> I see that youtube-dl has got support not only for yt but for a lot of content delivery websites too [04:15] <burek> cool thing :) [05:35] <CIA-17> ffmpeg: 03multiple authors 07master * r13cc3645b0 10ffmpeg/ (6 files in 4 dirs): [05:35] <CIA-17> ffmpeg: libavfilter: add colormatrix filter [05:35] <CIA-17> ffmpeg: Ported by: Baptiste Coudurier [05:35] <CIA-17> ffmpeg: cleanup+fate by ubitux [05:35] <CIA-17> ffmpeg: For detailed authorship of the original code please see avisynth [06:22] <CIA-17> ffmpeg: 03Ramiro Polla 07master * r97d36a1898 10ffmpeg/libavformat/asfenc.c: [06:22] <CIA-17> ffmpeg: asfenc: realloc index_ptr fewer times [06:22] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:22] <CIA-17> ffmpeg: 03Ramiro Polla 07master * r1ceff0859d 10ffmpeg/libavformat/asfenc.c: [06:22] <CIA-17> ffmpeg: asfenc: rename some variables [06:22] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:22] <CIA-17> ffmpeg: 03Ramiro Polla 07master * rae315a6a38 10ffmpeg/ (libavformat/asfenc.c tests/ref/lavf/asf): [06:22] <CIA-17> ffmpeg: asfenc: start at object 1 instead of 0 [06:22] <CIA-17> ffmpeg: This is how it is done in the official muxer. [06:22] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:22] <CIA-17> ffmpeg: 03Ramiro Polla 07master * rbd603494f9 10ffmpeg/ (3 files in 3 dirs): [06:22] <CIA-17> ffmpeg: asfenc: properly write index information [06:22] <CIA-17> ffmpeg: The index must take into account the pre-roll time and must seek backwards, [06:23] <CIA-17> ffmpeg: not forwards. [06:23] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:23] <CIA-17> ffmpeg: 03Ramiro Polla 07master * rf2fad251b8 10ffmpeg/libavformat/asfenc.c: [06:23] <CIA-17> ffmpeg: asfenc: reduce code duplication with new variable [07:02] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rca2b450cee 10ffmpeg/libavfilter/vf_idet.c: [07:02] <CIA-17> ffmpeg: vf_idet: use enum to represent the type. [07:02] <CIA-17> ffmpeg: This will simplify future code. [07:02] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:02] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r0477254dd2 10ffmpeg/libavfilter/vf_idet.c: [07:02] <CIA-17> ffmpeg: vf_idet: Print both the single frame based detection and the multi frame one. [07:02] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:02] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1a5c08eef6 10ffmpeg/libavfilter/vf_idet.c: [07:02] <CIA-17> ffmpeg: vf_idet: Add last_type variable to make code more readable. [07:02] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:33] <CIA-17> ffmpeg: 03Clément BSsch 07master * rbfd30fa8a3 10ffmpeg/tests/ref/lavfi/ (colormatrix1 colormatrix2): lavfi/colormatrix: add forgotten tests. [09:15] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r4d4df310f2 10ffmpeg/libavfilter/vf_idet.c: [09:15] <CIA-17> ffmpeg: vf_idet: print accumulated statistics for single and multi frame detection [09:15] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:15] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rc59e73d2c8 10ffmpeg/libavfilter/vf_idet.c: [09:15] <CIA-17> ffmpeg: vf_idet: change individual per frame statistics to debug log level. [09:15] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:15] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rfa1f92a4f1 10ffmpeg/libavfilter/vf_idet.c: [09:15] <CIA-17> ffmpeg: vf_idet: clean detected frame types based on a 4 frame history. [09:15] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:50] <ubitux> we don't have a placeholder for "ffmpeg regressions", right? [09:56] <michaelni> placeholder ? [09:56] <ubitux> i mean just like lavfi-regressions.sh [09:57] <ubitux> we don't have a "ffmpeg-regressions.sh" [09:57] <ubitux> (for cmd line parsing or generic options etc) [09:57] <michaelni> no, we dont have such a thing yet [09:57] <ubitux> ok [10:26] <spaam> ubitux: cant you use fate for it ? [10:28] <ubitux> you mean the fate/*.mak? [10:28] <ubitux> *-regression.sh scripts are part of fate [12:38] <CIA-17> ffmpeg: 03Thierry Foucu 07master * rc99bdd51b2 10ffmpeg/libavformat/utils.c: (log message trimmed) [12:38] <CIA-17> ffmpeg: lavf: Fix the last PTS to be generated using -fflags +genpts [12:38] <CIA-17> ffmpeg: To reproduce the problem, using ffprobe: [12:38] <CIA-17> ffmpeg: ./ffprobe -show_packets -print_format compact -fflags +genpts -i [12:38] <CIA-17> ffmpeg: fate_samples/mxf/C0023S01.mxf [12:38] <CIA-17> ffmpeg: You will notice that the last video frame does not have it's PTS being [12:38] <CIA-17> ffmpeg: set, even with using genpts. [15:56] <burek> is it possible to have ffprobe -show_streams with ffmpeg? [15:57] <Compn> its the same commands basically [15:57] <Compn> so should be [15:57] <Compn> ffmpeg -i file [15:58] <burek> i tried ffmpeg -show_streams but no [15:58] <burek> so I thought maybe it's ffprobe only [17:03] <burek> hm, im puzzled with this paste: http://pastebin.com/WJ7qaxA2 [17:03] <burek> is it possible that format and all the streams can contain metadatas each [17:04] <burek> metadata for format + meta for stream #0, meta for stream #1, etc.. ? [17:05] <nevcairiel> yes, there is stream based metadata [17:09] <burek> thanks :) [17:24] <spaam> ubitux: yes. if the output differ then something is wrong.. [17:25] <ubitux> heh? [17:33] <spaam> ubitux: yes. its like that [17:33] <CIA-17> ffmpeg: 03Clément BSsch 07master * r614716f583 10ffmpeg/libavformat/mov.c: (log message trimmed) [17:33] <CIA-17> ffmpeg: mov: do not print string len in handler_name. [17:33] <CIA-17> ffmpeg: The handler name is stored as a pascal string in the QT specs (first [17:33] <CIA-17> ffmpeg: byte is the length of the string), thus leading to an invalid metadata [17:33] <CIA-17> ffmpeg: string export. [17:33] <CIA-17> ffmpeg: Also add a second length check based on the first character to avoid [17:33] <CIA-17> ffmpeg: overwriting an already specified handler_name (it happens with Youtube [17:33] <ubitux> spaam: i don't understand what you are talking about :p [17:34] <spaam> ubitux: neither do i [17:35] <ubitux> :) [20:51] <bcoudu> how do we debug format timestamps in the latest version ? [20:52] <bcoudu> -fdebug ts doesn't work anymore [21:42] <michaelni> bcoudurier, -fdebug ts seems to need -v 99 [22:02] <bcoudurier> right [22:13] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3e4b5e68c1 10ffmpeg/cmdutils_common_opts.h: [22:13] <CIA-17> ffmpeg: cmdutils: fix -fdebug without -v xy [22:13] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:44] <CIA-17> ffmpeg: 03Anton Mitrofanov 07master * r14af74e9a0 10ffmpeg/libavcodec/h264.c: [22:44] <CIA-17> ffmpeg: h264: fix 4:2:2 PCM-macroblocks decoding [22:44] <CIA-17> ffmpeg: Fixes bug 239. [22:44] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [22:45] <CIA-17> ffmpeg: 03Diego Biurrun 07master * r1f068e4450 10ffmpeg/libavfilter/vf_unsharp.c: [22:45] <CIA-17> ffmpeg: vf_unsharp: Mark readonly variable as const. [22:45] <CIA-17> ffmpeg: This fixes the following warning: [22:45] <CIA-17> ffmpeg: libavfilter/vf_unsharp.c:106: warning: initialization discards qualifiers from pointer target type [22:45] <CIA-17> ffmpeg: 03Christophe GISQUET 07master * r6b81da2fd0 10ffmpeg/libavcodec/x86/ (dsputil_yasm.asm sbrdsp.asm): [22:45] <CIA-17> ffmpeg: dsputil x86: use SSE float instruction instead of SSE2 integer equivalent [22:45] <CIA-17> ffmpeg: All the more required since the users are pure SSE functions. [22:45] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [22:45] <CIA-17> ffmpeg: 03Christophe GISQUET 07master * rf9888520cc 10ffmpeg/libavcodec/x86/vp8dsp.asm: [22:45] <CIA-17> ffmpeg: vp8dsp x86: perform rounding shift with a single instruction [22:45] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [22:45] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rcfec77aaf8 10ffmpeg/configure: configure: add dl to frei0r extralibs. [22:45] <CIA-17> ffmpeg: 03Christophe GISQUET 07master * rcd88105f6f 10ffmpeg/libavcodec/x86/dsputil_mmx.c: [22:45] <CIA-17> ffmpeg: dsputil x86: remove deprecated parameter from scalarproduct_int16 prototype [22:45] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [22:45] <CIA-17> ffmpeg: 03Anton Khirnov 07master * ra03f8ea0a8 10ffmpeg/libavfilter/vsrc_buffer.c: vsrc_buffer: when no frame is available, return an error instead of segfaulting. [22:45] <CIA-17> ffmpeg: 03Diego Biurrun 07master * r0becb07842 10ffmpeg/libavcodec/ (h264.c h264.h h264_cabac.c h264_cavlc.c): h264: Factorize declaration of mb_sizes array. [22:45] <CIA-17> ffmpeg: 03Diego Biurrun 07master * re484265c97 10ffmpeg/libavcodec/aacenc.c: [22:45] <CIA-17> ffmpeg: aacenc: Mark deinterleave_input_samples argument as const. [22:45] <CIA-17> ffmpeg: This fixes the warning: [22:45] <CIA-17> ffmpeg: libavcodec/aacenc.c:524: warning: passing argument 2 of deinterleave_input_samples discards qualifiers from pointer target type [22:45] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2c5a2958e9 10ffmpeg/: (log message trimmed) [22:45] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [22:45] <CIA-17> ffmpeg: * qatar/master: [22:45] <CIA-17> ffmpeg: h264: Factorize declaration of mb_sizes array. [22:45] <CIA-17> ffmpeg: vsrc_buffer: when no frame is available, return an error instead of segfaulting. [22:45] <CIA-17> ffmpeg: configure: add dl to frei0r extralibs. [22:45] <CIA-17> ffmpeg: dsputil x86: use SSE float instruction instead of SSE2 integer equivalent [22:45] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r7756859a31 10ffmpeg/tests/ (15 files in 3 dirs): fate: add BMP tests. [22:45] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * rbc0bdda77b 10ffmpeg/libswscale/ (input.c output.c): [22:45] <CIA-17> ffmpeg: swscale: handle complete dimensions for monoblack/white. [22:45] <CIA-17> ffmpeg: Fixes bug 269. [23:38] <ubitux> bcoudurier: the -amerge code won't require much change when we will need to move to a better solution; it's likely we will just have to change the "amovie=foo.avi:si=3" into something like "[0:3]" as source (see -filter_complex patch proposition by Anton) [23:40] <ubitux> and anyway, the "hack" is pretty isolated [23:40] <ubitux> the visibility of the feature will also be restricted (i removed the changelog entry, and added a warn in the doc) [23:43] <ubitux> are you proposing yourself to help with the lavfi auto reconfiguration, or audio stretching? [23:54] <beastd> "-filter_complex" sounds pretty bad from a naming perspective. didn't read the proposal though. [23:56] <ubitux> it's still a RFC [23:56] <ubitux> from what i can tell :) [00:00] --- Fri Apr 6 2012

1 0

ffmpeg.log.20120405
by burek 06 Apr '12

06 Apr '12

[01:34] <lesshaste> ubitux, yes the first frames are bad! [01:34] <lesshaste> ubitux, why is that? [01:38] <mischanix> I'm trying to stream directshow inputs encoded with x264/lame to rtmp. However, rtmp seems to bottleneck above ~1Mb/s, even if I stream to a LAN rtmp server (connection is 100Mb). Streaming to file, no frames are dropped, but using rtmp, the buffer fills up and frames are dropped wildly above 1Mb. Using zeranoe's windows builds, have tried both x86_64 and x86 versions. http://pastebin.com/SNT0iiHJ [01:39] <burek> mischanix, try vlc [01:41] <mischanix> Does vlc's librtmp work yet? [01:42] <mischanix> I mean, I tried to make that work at first to no avail. [01:42] <burek> well, since you are trying windows builds, I'm pretty sure you're going to hit the wall sooner or later :S [01:42] <burek> btw, why do you need rtmp [01:42] <burek> any specific reason? [01:43] <mischanix> Trying to work around XSplit for justin/twitch/own3d so that a friend can stream without it if possible. [01:45] <burek> did you try raising -rtbufsize 10000000 param? [01:46] <mischanix> Yeah. I've had the buffer up to 4 GB in memory and the same thing happening [01:46] <mischanix> [Windows doesn't like that.] [01:46] <burek> also, you are using high resolution -s 1280x800 -r 24 [01:46] <burek> combined with -crf 20 [01:47] <burek> it's pretty cpu intensive [01:47] <burek> can you try, just for test to use -s cif [01:47] <burek> that's lower than vga (640x480) [01:47] <burek> just to see if it's a cpu issue [01:48] <burek> (also remove -s and -r before the output if it's the same as in input) [01:48] <mischanix> Yeah, cpu maxes at 65% or so using rtmp. [01:48] <mischanix> 100% flv. Using crf /40/ works better, but obviously the quality is meh. default crt, rtmp starts to bottleneck again [01:48] <burek> although, dshow shows those warnings, which means probably a bad driver or something, not sure.. [01:49] <burek> it doesn't even get to the encoder when the warning is raised [01:49] <burek> try using maybe -crf 23 (default) and also use -preset ultrafast [01:51] <mischanix> drops even more with cif. [01:52] <burek> maybe you didn't remove -s and -r before the output [01:52] <mischanix> Er, that was using vb 3500k though [01:52] <burek> so it tries to rescale from cif to that [01:54] <mischanix> Smooth as butter with crf 20 [01:56] <burek> now, try increasing -s [01:57] <burek> to see what is the pitch top for your machine [01:58] <mischanix> 1280x800 with crf 30 is smooth [01:58] <mischanix> But this is with about 34% going to another program [01:58] <burek> well, screen grabbing and encoding is really cpu intensive [01:59] <burek> so you should use some codecs optimized for screen grabbing (low cpu usage, but bad compression) [01:59] <burek> of course, if you need live stream [01:59] <burek> if not [01:59] <burek> then do it in steps [02:01] <mischanix> crf 28 1280x800 is about 55% [02:02] <burek> :) [02:09] <mischanix> so, using -maxrate 3500k -bufsize 3500k uses less CPU and drops more frames, but -maxrate 1500k -bufsize 3000k uses more CPU and drops less frames, even though the connection is 100Mb. [02:10] <burek> that doesn't have anything to do with the network connection :) [02:11] <burek> but with the desired output bitrate that you want [02:11] <burek> higher bitrate / less encoding / less cpu / less encoding time [02:11] <mischanix> In fact, with every setting I've tried, the dropping of frames has been proportional to the bitrate. Is that how it should be? [02:12] <jya> a very silly question, but I'm yet to find an answer& with ffplay how do you select which alsa audio device to use? I get no audio by default [02:12] <mischanix> i.e. lower bitrate -> fewer dropped frames. That doesn't make sense to me. [02:12] <burek> well, would you drop more plates if someone would throw them at you, like, 10 in a second or 1 in a second? :) [02:13] <burek> jya, did you read ffplay documentation? :) [02:13] <jya> I did... [02:13] <jya> if you find anything related to audio output device in there, you're good :) [02:14] <burek> mischanix, why doesn't it make sense [02:15] <mischanix> burek, CPU isn't my bottleneck. If I output the same settings to a local file, very few if any frames are dropped. However, when rtmp is introduced, bitrate becomes proportional to frame dropping. In the situation of outputting to a local file, the opposite is true until around 100MB/s [02:15] <jya> burek: I've been googling for a good 20 minutes on the matter, read all the manpages, and ffplay -h .. nothing in regards to audio output device [02:16] <burek> jya, just a sec [02:17] <burek> mischanix, I think the best chance would be to find the irc channel for librtmp developers and ask them directly [02:18] <burek> ffmpeg just uses its interface [02:18] <mischanix> Yeah, but if I were in their shoes, I'd just dismiss it as an issue with the cross-compiler toolchain [02:18] <burek> :) [02:20] <mischanix> But, yeah. I suppose I'll look around their parts of the internet now. Thanks for the insight :) [02:21] <burek> np :) [02:22] <burek> jya, so, you want to select a specific audio output, right? :) [02:22] <jya> well, I just want to play audio.. I get no audio by default. so I'm guessing it's to do with an issue with the default audio device. [02:23] <jya> ffplay file.mpeg give me no audio [02:23] <jya> so I was hoping that specifying the alsa audio device might help. similar to -ao in mplayer [02:24] <burek> did you check if file.mpeg has audio streams in it? :) [02:24] <jya> of course it has audio :) [02:24] <burek> ok :) well [02:24] <burek> you can always use alsamixer [02:25] <burek> and with f6 choose the default device [02:25] <jya> reason I'm trying ffplay is because I want to find when ffmpeg fixed a bug in AAC-latm decoding. [02:25] <burek> or f5 [02:25] <jya> alsamixer doesn't let you configure your default audio device [02:26] <burek> since when [02:26] <jya> since forever [02:26] <burek> o.O [02:26] <burek> did you even try it [02:27] <jya> it only let you define the mixer level& f6 let you define what audio device to use, for within alsamixer. [02:27] <burek> :) [02:27] <burek> ok [02:27] <jya> fwiw, i'm quite well aware of what alsa does and .. what it doesn't :) [02:28] <jya> hum&. activating pulse audio gives me audio in ffplay for some reason.. this will let me go by for the time being [02:28] <burek> :) [02:29] <jya> i find it amazing that ffplay doesn't have an option to set the audio output device [02:29] <jya> I certainly don't want to use pulse [02:43] <jya> git bisect isn't going to help me much& ffmpeg at the time of the last sync doesn't play that file anywhere like what mythtv does. myth has slight corruption every few seconds, ffplay play that file at extremely low speed [02:46] <jya> wrong window [03:45] <entreri> hello, is there a way to disable the need of an X server for ffplay ? I don't want to depand on X to play an mp3, and I do not need any graphic, any idea ? thanks. [03:47] <burek> did you ever try something like this: ffmpeg -i play.mp3 -acodec copy -f alsa hw:0 [03:52] <jya> entreri: use -nodisp [03:52] <jya> ffplay -nodisp file.mp3 [03:54] <entreri> jya: I tried but it seems I still need to specify a display, it seems to just disable the graphic animation in X [03:54] <entreri> "Unable to open mouse, Did you set the DISPLAY variable?" [03:54] <jya> entreri: I use this command on a headless server to play audio ... [03:54] <jya> what version of ffmpeg are you using? [03:55] <entreri> 0.10.2 [03:58] <entreri> I think I need to change the config, there are some switches initialised, one of them is --enable-x11grab [04:16] <66MAAA5MI> hello all [04:17] <66MAAA5MI> is there a ppa for ubuntu? [04:17] <66MAAA5MI> ah, found it [04:17] <66MAAA5MI> https://launchpad.net/~jon-severinsson/+archive/ffmpeg [04:23] <Shimmy> Hi everyone, is this a developer channel? [04:39] <Shimmy> Anyone in here? [06:21] <fling> hello! :p [06:22] <fling> can I capture from BlackMagic DeckLink SDI ? [06:22] <relaxed> fling: is there a v4l2 driver for it? [06:22] <relaxed> If so, yes. [06:36] <Shiron> Wow...lots of users... [06:36] <Shiron> Anyone active? [06:37] <Shiron> Hello...anyone here? [06:38] <Shiron> Knock knock... [06:40] <relaxed> do you have a real question? [06:42] <Shiron> Several. A lot. [06:43] <Shiron> Actually I have absolutely no experience with ffmpeg, but I really could use some help. [06:45] <Shiron> Right... [06:46] <relaxed> Help with what? Ask a specific question... [06:46] <Shiron> So, I've only ever used GUIs up until now. The most specific I can get is that I've just downloaded a windows guild of ffmpeg on its own and I have no idea how to run it. [06:46] <Shiron> *build [06:47] <relaxed> start -> run -> type "cmd.exe" [06:47] <relaxed> enter [06:47] <Shiron> Yes [06:48] <relaxed> navigate to the binary and run it. [06:48] <Shiron> Alright... [06:48] <Shiron> Is it already configured with all the libraries? [06:48] <Shiron> Zeranoe build [06:48] <relaxed> I think he offers static builds, so yes. [06:49] <relaxed> what is your goal? [06:50] <Shiron> Recompiling an SBS .avi into actual 3D output [06:50] <Shiron> I have the code [06:50] <Shiron> I just don't know how to run it [06:50] <Shiron> You put it in a .bat, right? [06:53] <relaxed> this script has ffmpeg commands in it? [06:53] <Shiron> -i -vol -s [06:53] <Shiron> Yes, I believe [06:54] <relaxed> then modify the script and use the full path to ffmpeg and the input I guess [06:55] <relaxed> I don't use windows, nor do I know how to script for it. [06:55] <Shiron> Yes, I've figured as much [06:56] <Shiron> But the internet cafe I'm in only allows for one IP, so I'm unable to run an internet browser to download ffmpeg in my Ubuntu VB [06:59] <Shiron> You know... [06:59] <Shiron> I should just start in a NAT... [06:59] <Shiron> Whoops. Be back soon? [07:06] <Shiron> Alright, what's the apt-get for ffmpeg? Still new to Linux. [07:11] <Shiron> know what, nevermind. [07:11] <Shiron> General unresponsiveness has really peeved me. [07:12] <Shiron> Thanks anyways, and good day. [07:55] <TACPILOT> what container handles RGB32 ?? [08:44] <orangey> hello all [08:45] <orangey> I'm having an absolutely infuriating time with ffmpeg. I'm making some videos with kdenlive in h264 or xvid or mpeg2, and they render with a/v sync issues [08:45] <orangey> I cannot for the life of me figure out *why*, nor correct the issues [08:46] <orangey> how do I even debug why the sync issues are arising? [09:39] <level09> I am converting my video to h264 file but for some reason I cant run it using html5 video player [09:39] <level09> what am I missing here ? [09:41] <level09> mix_cmd = 'ffmpeg -y -i input.f4v -vcodec libx264 -b 512K -threads 0 -preset fast -acodec libmp3lame -crf 20 out.mp4' [09:45] <thebastl> anyone in here using ffmpeg on android? [09:45] <thebastl> hi btw [09:46] <level09> hey [10:12] <cbreak-work> so... I'll try to link ffmpeg to my application on windows. Wish me luck. [10:13] <sacarasc> You can do it, cbreak-work! [10:33] <matt__> Hi, i' trying to decode Subtitles in the dvbsub format. I decode them using the avcodec_decode_subtitle without any problem. I need to reconstruct the image and save it. My problem is that i cannot understand how the image is stored (pix_fmt is -1) and i cannot find any documentation. Can you explain to me how the data is represented or tell me where i can find the info please? [10:36] <cbreak-work> sacarasc: thanks, it works! :) [10:36] <cbreak-work> was less trouble than I expected [10:36] <sacarasc> Of course it does, you have work in your nick! [10:37] <cbreak-work> linking a C library is so much nicer than C++ on windows [10:53] <burek> orangey, [10:53] <burek> can you please use pastebin.com, to show your command line and its output? [10:54] <burek> matt__, did you try asking your question in #libav ? [10:56] <matt__> no i didn't try on libav, i will give it a shot then [10:56] <matt__> Do you have an idea on how the data is organised? [10:58] <burek> no, sorry :) [10:58] <matt__> Arff, no problem. Finding appropriate documentation is quite hard! [10:59] <burek> :) at least you are not bored :) [11:00] <matt__> Yeah sure ^^ [11:00] <buhman> is it possible to multithread anything fancier than mpeg2? [11:01] <buhman> like say vp8 or h.264? [11:01] <burek> buhman, h264 :) [11:01] <burek> -threads [11:08] <buhman> hmm, my distribution's ffmpeg only seems to come with 7 ffpresets [11:10] <burek> buhman, [11:10] <burek> can you please use pastebin.com, to show your command line and its output? [11:10] <burek> just type ffmpeg [11:16] <buhman> burek: http://sprunge.us/ZXWV [11:16] <burek> that's ok [11:16] <burek> anyway [11:17] <burek> instead of vpreset [11:17] <burek> ffmpeg now uses native x264's -preset option [11:17] <burek> so type x264 --help [11:17] <buhman> O.o oh really [11:17] <burek> and read about -preset -profile and -tune [11:17] <buhman> when did that happen? [11:17] <burek> recently [11:17] <burek> several months ago :) [11:20] <buhman> O.O watching my new hyperthreaded 2 CPU quad core doing ffmpeg -threads 0 in htop [11:20] <buhman> that's pretty sexy [11:21] <burek> :))) [11:21] <burek> you could do live screencasting in hd :) [11:21] <buhman> particularly the part where I'm doing 65fps on the slow preset on 480p video [11:21] <buhman> right; that's what I was bout to say [11:22] <buhman> I could do not-slow and get away with 720p probably [11:22] <burek> you can try :) [11:23] <buhman> linpack told me 65 GFLOPs [11:23] <buhman> is that "good" in the spectrum of computers that are used to encode h.264 video? [11:24] <burek> there are some better things, like hardware encoders :) [11:24] <burek> and way cheaper [11:25] <buhman> O.o really? [11:28] <buhman> I mean, considering my last computer was only 0.8 GFLOPs (linpack), I was afraid I was going to make a mess in my pants when I compiled ffmpeg-git (just a few minutes ago) in 1 minute and 57 seconds (according to 'time') [11:31] <burek> cool :) [11:31] <burek> do you have ssd disk [11:54] <luc4> Hi! I'm using ffplay to play an mp3 on a http server but I noticed the AVERROR_EOF is never received from av_read_frame. For any file I place on that server. Anyone who experienced the same? [11:55] <burek> you can try asking your question in #libav too [11:55] <luc4> burek: thanks [12:22] <libv> libav sws_scale colour space question... [12:23] <libv> i am hacking together some tool to do better test kms planes (and it will also gain Xv support) [12:23] <libv> converting from RGB24 to 0RGB (as needed by the overlay) manually gives me nice blacks [12:24] <libv> using sws_scale with the defaults gives me blue. [12:25] <burek> you can try asking your question in #libav too [12:27] <libv> *sigh* forks are fun (said the guy who lived through part of the xfree86/X.org shit) [15:04] <kriegerod> is there, at iOS AVFoundation API, http://goo.gl/EWZ5J , among kCVPixelFormatType_*, some format that exactly matches libavcodec's PIX_FMT_YUV420P? Or which of formats from AVFoundation matches exactly to particular libavcodec pixel format, so i can convert it with swscale? [15:07] <burek> you can try asking your question in #libav too [15:09] <plm> Hi all [15:24] <kriegerod> burek, thanks captain, i've already done it ) [15:28] <burek> :beer: :) [15:29] <plm> how works (in low level) to sent video streaming via http? I think that sen streaming via http is hirrble, because http will back, from target, to source, the response for each POST/GET.. that are there the peace of video right? [15:30] <burek> plm, if you use each http request for each peace of video, well, yes, that will work slow :) [15:30] <burek> but why would you do that? :) [15:31] <burek> usually 1 http request/response is enough to start http streaming [15:31] <burek> piece* [15:31] <burek> lol peace :D [15:32] <plm> burek: sorry, just curous how taht works over http [15:32] <burek> well, you would use ffserver + ffmpeg [15:32] <plm> burek: so how really works, dont have repose for each post/get? or dont have post/get in that? [15:32] <burek> client does http request, server replies with http response + stream data [15:33] <burek> and the connection stays open indefinitely [15:33] <burek> (if it's a live stream for example) [15:34] <plm> burek: ok, and will have this request/response for each pieace of video stream? [15:35] <burek> each piece? [15:35] <burek> what is a piece? [15:36] <plm> burek: if is live stream.. that live stream has a prt of video.... [15:36] <burek> read my previous 5 lines [15:37] <burek> server sends stream data as it gets encoded [15:37] <burek> through the same tcp connection [15:37] <burek> it is all 1 http request/response [15:37] <plm> burek: I read that. After server replies with http response + stream data what happen? [15:37] <plm> burek: ohh [15:37] <plm> burek: just one request;respose, in start video only? [15:37] <burek> well, during time, client continues to receive stream data [15:38] <burek> connection stays open forever, and client receives stream through it [15:38] <plm> after is just tcp, without http right? [15:38] <burek> well, yes [15:38] <burek> it is data with undefined length [15:38] <burek> so the client keeps receiving it [15:39] <plm> burek: ok. I was thinking that as is over http, the request/respose will have everytime.. not just in start connectiion... [15:40] <plm> burek: and after receiving video, are the http still working, been used in this moment.. like as video still send via http protocol, or will use just in start connection? [15:41] <burek> server sends http response + stream data (forever) [15:42] <plm> burek: ok. But request is not send never more by client right? [15:42] <burek> correct [15:42] <plm> burek: thanks for clarify! [15:42] <burek> :beer: :) [15:43] <plm> burek: so that is not so heavy! [15:43] <burek> not at all :) [15:43] <plm> burek: just somes bytes more in response.. [16:05] <woozly> guys, how to copy metadata to file, on convert? [16:05] <woozly> metadata with duration (because video-js player html5/js waits until videofile downloaded) [16:07] <burek> try mediainfo [16:09] <woozly> can map_metadata help me? :) [16:09] <woozly> what is mediainfo? [16:10] <zimbatm> woozly: mediainfo is another tool that shows you media metadata [16:14] <woozly> zimbatm, oh.. good. thank you! [16:14] <woozly> burek, thanks! [16:55] <orangey> hello all [16:55] <orangey> I'm having some trouble with this command and a video with an AAC audio input: ffmpeg -i In.mp4 -strict experimental -vcodec copy -acodec copy -ss 00:00:00.1 -t 10 -an output.mp4 [16:55] <orangey> Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 353 kb/s [16:55] <orangey> but the output is sans audio [16:57] <burek> orangey, [16:57] <burek> can you please use pastebin.com, to show your command line and its output? [16:58] <orangey> http://pastebin.com/WJ7qaxA2 [16:58] <orangey> burek [16:59] <burek> ok [16:59] <burek> also, you only need -strict experimental if you are encoding audio with aac [16:59] <burek> if you use -acodec copy that's not needed [17:00] <burek> since no encoder is being used [17:00] <burek> the data is just copied [17:00] <orangey> burek: I figured as much, but since it wasn't getting copied and I know my incoming is aac, I figured I'd try it [17:01] <burek> remove -an [17:01] <burek> that means "audio none" [17:01] <Mavrik> yeah, "-an" disables audio on output [17:02] <burek> you can see in your output after the line: Output #0, mp4, to 'output.mp4': [17:02] <burek> look into all lines that begin with: Stream # [17:02] <burek> that'll show you streams in the output file, so you can see there is no audio [17:19] <orangey> ooh/ [17:19] <orangey> oops. thank you [18:04] <miknix> hello everybody [18:05] <miknix> does someone know if the libx264-normal preset was added after ffmpeg-0.7.8 or removed before that? [18:05] <miknix> I cant find that present in that version [18:08] <jermy> I think ffmpeg dropped x264 presets before then - we had to move to using the x264 preset names directly at some point [18:34] <meekohi> OSX: I have libx264 installed via macports and trying to ./configure --enable-libx264, but getting "ERROR: libx264 not found" [18:36] <meekohi> Any ideas how to get this working? libx264 doesn't compile for OSX seemingly. [18:44] <miknix> jermy, I see, so I can consider the normal preset as deprecated [19:52] <Radex> Hi, i getting Unrecognized option 'directpred' [19:53] <Radex> it's deprecetad option i new one ffmpeg version? [19:55] <Radex> s/i/in [20:09] <iive> Radex: direct_pred ? [20:10] <meekohi> I switched to homebrew. Had some problems but finally got it working. [20:10] <iive> looks like x264 option. [20:13] <Radex> iive: yeah, just found commit in git - thanks for help [20:55] <burek> meekohi, you need libx264-dev, not libx264 [20:55] <burek> miknix, just use -preset option in ffmpeg and type x264 --help to see more info on -preset -profile and -tune [20:56] <burek> those options are mapped directly to native x264 options [20:56] <burek> so you can use its help screen [20:58] <andrew_> Hey again, im getting a similiar option [20:59] <andrew_> not option, problem [20:59] <andrew_> sorry [21:00] <andrew_> I try normal syntax, "ffmpeg -i <input> <output> [21:00] <andrew_> I get errirs about broken ffmpeg default settings and asking me to use -vpre [21:00] <andrew_> but those don't work [21:01] <andrew_> any ideas? [21:33] <lanceio> PSP decent quality? [21:33] <lanceio> FAQ? [21:34] <JEEB> lanceio, http://x264.fushizen.eu/random/psp_encoding.txt [22:26] <Mista_D> where can I upload a sample file? mediafire seems to be down... [22:28] <Mista_D> FFmpeg 0.1.2 fails to decode MOV with LPCM 24bit audio 48 kHz, stereo... http://www.mediafire.com/?52cdkuqf8w4f431 Any advice on how to encode it to flash? [22:35] <juanmabc> when something fails i encode intermediarily to mp4 [22:40] <Aegwyn11> I have an transport stream with h.264 video that I'd like to decode to raw video in AVI. The source material is 29.97 1080i TFF, but it seems that the output raw video file is always progressive. How do I flag it do that the output AVI is interlaced? [23:01] <meekohi> Does anyone know how to debug the error: "%03d.jpg: could not find codec parameters" ? [23:02] <meekohi> I feel like I need to add something to ./configure, but don't know that. [23:03] <meekohi> That error comes from ffmpeg -i %03d.jpg out.webm by the way [23:06] <meekohi> --enable-decoder=mjpeg maybe? [23:07] <beastd> meekohi: no it should be enabled by default with a normal ffmpeg build. i try to reproduce your problem now. [23:07] <meekohi> beastd: I'm starting from --disable-everything in configure [23:07] <meekohi> beastd: I only want the bare minimum (have to compile when bringing new server online) [23:08] <iive> meekohi: i don't see framerate [23:08] <iive> hi beastd [23:09] <beastd> meekohi: than mjpeg is needed [23:09] <meekohi> Ah okay sorry I shortcutted: /usr/local/bin/ffmpeg -i "%03d.jpg" -g 3 -r 30 -b:v 2.0M -y out.mp4 2>&1 [23:09] <beastd> Oi iive! [23:09] <meekohi> Do I just need the decoder? Do I need "demuxed" too? [23:10] <beastd> you need img2 demuxer and mjpeg at least [23:10] <iive> meekohi: unless you have disabled things specifically, you should have everything you need for jpg decoding. [23:11] <beastd> iive: [23:07:32] <meekohi> beastd: I'm starting from --disable-everything in configure [23:11] <iive> oh... [23:11] <iive> bad idea :) [23:11] <beastd> :) [23:11] <meekohi> Yeaaah I know ::ashamed:: I'm trying to speed up the time to bring new servers up& might be shooting myself in the foot. [23:11] <iive> with cannonball [23:12] <iive> if you have multiple cores, build with `make -j4` for 4 cores. [23:13] <beastd> meekohi: img2 is named image2 [23:13] <meekohi> iive: only one core for now, doing this on EC2 [23:13] <iive> ec2? [23:13] <meekohi> okay I've added image2 mux/demux and mjpeg encode/decode [23:13] <meekohi> Amazon cloud [23:14] <meekohi> They don't package ffmpeg nicely for you unfortunately :( [23:14] <beastd> meekohi: you probably do not need encode/mux if I understood your usecase. though it won't hurt in any case. [23:14] <iive> what is the default encoder for webm? [23:15] <iive> vp8? [23:15] <meekohi> vpx I think? [23:15] <meekohi> I added that and h264 as well. [23:15] <iive> is it libvpx or ffmpeg have an encoder? [23:15] <iive> i know we have decoder, quite faster than the official. [23:15] <meekohi> No idea. [23:17] <meekohi> Okay I will get rid of --disable-everything and try working backward instead... [23:17] <beastd> meekohi: what was missing after enabling mjpeg and image2? [23:18] <meekohi> Says Unable to find suitable output format for .mp4, which is weird because this was working before I think. [23:21] <meekohi> What does the muxer/demuxer do exactly in the case of jpegs? [23:23] <beastd> The "demuxer" makes the bunch of jpegs into a video stream. [23:24] <beastd> The image2 "muxer" would do the opposite and create a series of jpeg image files from a video stream [23:26] <JEEB> <iive> is it libvpx or ffmpeg have an encoder? <- libavcodec only has a decoder, the encoder's libvpx [23:26] <JEEB> there's also a xvp8 under work [23:29] <beastd> meekohi: seems unlikely to me but you could try once more with disabled image2 muxer. if that doesn't work again you are probably missing the muxer for your desired output format [23:29] <iive> JEEB: i think at some point the x262, x264, x265 and xvp8 would merge, forming a new common encoder library. [23:30] <JEEB> lol [23:30] <iive> :P [23:30] <JEEB> x265 is the only one I haven't seen yet [23:30] <beastd> try to verify it is compiled in with ffmpeg -formats [23:30] <iive> I heard the standard is out of draft stage and is (almost) ready. [23:31] <JEEB> yeah [23:31] <JEEB> so in theory one could start watching out for changes in the JVT mailing list [23:31] <JEEB> and making some pre-work [23:35] <ckb> How should I decide what presets to use? [23:37] <JEEB> libx264? [23:37] <ckb> yes [23:37] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [23:37] <JEEB> basically select the slowest you can take [23:39] <ckb> okay, and running concurrent ffmpeg processes isn't a great idea, right? [23:40] <JEEB> in most cases [00:00] --- Fri Apr 6 2012

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