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July 2012

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ffmpeg-devel.log.20120730
by burek 31 Jul '12

31 Jul '12

[00:01] <Shimmy> Anyone please help: https://ffmpeg.org/trac/ffmpeg/ticket/747 [00:10] <durandal_1707> how to handle encoder that outputs stuff only every n-th frame? [00:26] <durandal_1707> i get floating exception from muxer because of this [00:27] <iive> ? [00:28] <durandal_1707> encoder outputs data after it gets enough input [00:29] <iive> sure, if you don't have output, don't send it to muxer. [00:29] <iive> codecs are supposed to work as zlib. [00:30] <durandal_1707> i set got_packet_ptr to 1 only if i get something [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rad2022ac83 10ffmpeg/libavfilter/vsrc_testsrc.c: lavfi/testsrc: add comment in options [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * ra74dcb7dc0 10ffmpeg/libavfilter/vsrc_color.c: [00:38] <CIA-41> ffmpeg: lavfi/color: cache and reuse colored picture in context [00:38] <CIA-41> ffmpeg: Avoid to fill the same picture again and again with the same content. [00:38] <CIA-41> ffmpeg: Optimize computation, and provides an example for the use of the [00:38] <CIA-41> ffmpeg: AV_PERM_REUSE permission flag. [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * re5d617aa6d 10ffmpeg/ (3 files in 2 dirs): lavfi/color: drop support for old deprecated syntax [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r47b6b7a2b9 10ffmpeg/libavfilter/vsrc_color.c: lavfi/color: reindent after the previous commit [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r9cf9214d0a 10ffmpeg/libavfilter/vsrc_color.c: [00:38] <CIA-41> ffmpeg: lavfi/color: free internal context options in uninit() [00:38] <CIA-41> ffmpeg: Keeping the set options in the context when the component is still used [00:38] <CIA-41> ffmpeg: can be useful for debugging purposes. [00:38] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r316f6df9cc 10ffmpeg/libavfilter/avcodec.c: [00:38] <CIA-41> ffmpeg: lavfi/avcodec: rename picref->samplesref variable in avfilter_get_audio_buffer_ref_from_frame() [00:39] <CIA-41> ffmpeg: The new name is more meaningful in that context. [00:42] <michaelni> durandal_1707, you need to set CODEC_CAP_DELAY for the encoder not sure if this is the issue though [00:44] <durandal_1707> that is for every frame? not just last one? [00:44] <durandal_1707> issue is i get floating point in mp3 muxer [00:44] <durandal_1707> nut works but output is unplayable [00:45] <durandal_1707> what i get segv with libmp3lame when muxing into nut [00:45] <Shimmy> Anyone please take a look: https://ffmpeg.org/trac/ffmpeg/ticket/747 [00:45] <durandal_1707> previously at least it would abort instead of crashing [01:01] <CIA-41> ffmpeg: 03Anton Khirnov 07master * rf66eeff1c8 10ffmpeg/libavformat/utils.c: lavf: round estimated average fps to a "standard" fps. [01:01] <CIA-41> ffmpeg: 03Anton Khirnov 07master * raba232cfa9 10ffmpeg/ (21 files in 3 dirs): (log message trimmed) [01:01] <CIA-41> ffmpeg: lavf: deprecate r_frame_rate. [01:01] <CIA-41> ffmpeg: According to its description, it is supposed to be the LCM of all the [01:01] <CIA-41> ffmpeg: frame durations. The usability of such a thing is vanishingly small, [01:01] <CIA-41> ffmpeg: especially since we cannot determine it with any amount of reliability. [01:01] <CIA-41> ffmpeg: Therefore get rid of it after the next bump. [01:01] <CIA-41> ffmpeg: Replace it with the average framerate where it makes sense. [01:01] <CIA-41> ffmpeg: 03Luca Barbato 07master * r41f43202cf 10ffmpeg/libavformat/flvdec.c: [01:01] <CIA-41> ffmpeg: flvdec: remove spurious use of stream id [01:01] <CIA-41> ffmpeg: We match streams by codec id now. [01:01] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7d1918affe 10ffmpeg/: (log message trimmed) [01:01] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [01:01] <CIA-41> ffmpeg: * qatar/master: [01:01] <CIA-41> ffmpeg: flvdec: remove spurious use of stream id [01:01] <CIA-41> ffmpeg: lavf: deprecate r_frame_rate. [01:01] <CIA-41> ffmpeg: lavf: round estimated average fps to a "standard" fps. [02:27] <durandal_1707> michaelni: i get segv when using lame encoder [02:32] <michaelni> durandal_1707, how can i reproduce it ? [02:33] <durandal_1707> try encode to mp3 container, can you reproduce it? [02:33] <durandal_1707> Daemon404: what happened with you work on nut? [02:33] <Daemon404> what work? [02:34] <Daemon404> im using nut at work [02:34] <Daemon404> trhough teh public api [02:34] <Daemon404> im not workong ON nut [02:34] <michaelni> ffmpeg -i matrixbench_mpeg2.mpg test.mp3 <--- works fine [02:43] <durandal_1707> michaelni: strange segv only happens if i compile with libshine support [04:00] <CIA-41> ffmpeg: 03jamal 07master * r938e4470ae 10ffmpeg/ (libavcodec/options.c libavformat/options.c): [04:00] <CIA-41> ffmpeg: options: Fix warning about incompatible pointer type [04:00] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:01] <CIA-41> ffmpeg: 03jamal 07master * r753a1b0988 10ffmpeg/libavutil/samplefmt.c: [04:01] <CIA-41> ffmpeg: samplefmt: Fix warning about discarded qualifier [04:01] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:13] <oneal> hello, for XVBA acceleration, Is there a implementation in ffmpeg source code. [04:14] <oneal> or Is there a plan to implement this acceleration [04:15] <durandal_1707> no and no [04:16] <oneal> Thanks [04:17] <oneal> I'm focusing on implement this, but I don't know whether community accpet this code or not [04:18] <durandal_1707> how something can be accepted if code is not reviewed? [04:21] <oneal> Okay ,I know your mean that I should submit my code at firstly, but I just want to know whether community is interesting this feature or not. [04:22] <ohsix> the community is interested in every feature! [04:22] <oneal> sounds good. Thanks [04:46] <CIA-41> ffmpeg: 03Paul B Mahol 07master * r8a6d673eff 10ffmpeg/libavformat/flacdec.c: [04:46] <CIA-41> ffmpeg: flacdec: fix memleaks [04:46] <CIA-41> ffmpeg: While here refactor code. [04:46] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:51] <Compn> oneal : yes, xvba and all hwaccel api will be accepted [04:51] <Compn> :) [05:09] <oneal> Thanks , @Compn [05:12] <Compn> oneal : its best to ask here or on the mailing list to make sure you design it so that it can be accepted [05:13] <Compn> i mean, which way to design it , which api to use, etc [05:24] <oneal> Okay [05:25] <oneal> I will give a detail design for dicussing befor implementing it. [05:26] <oneal> make sure we are the same page and it can be accepted by community. [11:43] <ubitux> oneal: i see you're talking about deshake on one of the libav channel; you know ffmpeg has one? [11:43] <ubitux> with an opened related ticket for improvment btw (https://ffmpeg.org/trac/ffmpeg/ticket/1430) [11:49] <maker> Hello. I have a few questions about code conventions I saw on your repo :) Following Accelerated C++, I got used to the habit of putting '!=' operator inside loops; while ffmpeg seems to use '<'. Apart from debugging reasons, have you considered that replacing the operand could make comparisons faster? At least, that's what I read there. [11:50] <maker> follows the second one :D [11:51] <maker> AVFilterLink->cur_buf->data has planes placed one immediately after the other? I mean, in https://github.com/mmaker/FFmpeg/commit/0b99e4e4d2ffeac7b833161ebc5562a2841… [11:51] <maker> I could replace that expression with just a sum, without reassignment? [11:52] <maker> s/I could/could I/ [12:02] <rjiejie> hi, i'm a newbie, someone knows about that which file is the "mp4" demuxer in ffmpeg source code ? thanks [12:03] <saste> maker: no, data is usually padded, that is linesize != pixel_step * width [12:03] <CIA-41> ffmpeg: 03Nicolas George 07master * r33449b1776 10ffmpeg/libavutil/ (audioconvert.c audioconvert.h): audioconvert: add av_get_channel_description(). [12:03] <CIA-41> ffmpeg: 03Nicolas George 07master * rf893904469 10ffmpeg/ (4 files in 2 dirs): [12:03] <CIA-41> ffmpeg: audioconvert: add av_get_standard_channel_layout(). [12:03] <CIA-41> ffmpeg: Also bump minor version and add APIchanges entry. [12:03] <CIA-41> ffmpeg: 03Nicolas George 07master * rcc55470472 10ffmpeg/libavutil/audioconvert.c: audioconvert: include the description of channels in the library. [12:03] <saste> maker: also consider that we have already a negate filter [12:04] <saste> regarding the != vs. < , i'm fine with != but I wonder how much difference it does [12:05] <saste> i also suppose the difference will be platform dependent [12:10] <saste> rjiejie: libavformat, grep for mp4 [12:14] <rjiejie> saste: thanks, i have found the file is related "isom.*" :) [13:35] <maker> re [13:58] <maker> saste: jsut testing. Uhm,concering the difference between '<' and '!=', I think that the compiler transforms '<' in a cmp instruction, which in practice is a subtraction; instead '!=' should correspond to a test instruction, alias logic and. But it's just a detail. [14:35] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rdfd8f2ff19 10ffmpeg/libavfilter/avfilter.h: [14:35] <CIA-41> ffmpeg: avfilter: fix minor typo [14:35] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:35] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rcbb2c95e47 10ffmpeg/ffmpeg.c: [14:35] <CIA-41> ffmpeg: ffmpeg: init simple filter graph before trying to extract framerate from its output. [14:35] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:35] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r3161958455 10ffmpeg/ (libavfilter/vf_yadif.c tests/ref/fate/filter-yadif-mode1): [14:35] <CIA-41> ffmpeg: yadif: update frame rate [14:35] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:39] <durandal_1707> michaelni: why pal8 -> pal8 requires that palette is always copied to frame->data[1]? pal8->rgb does not have such limitation [14:46] <michaelni> durandal_1707, swscale doesnt copy the palette autiomatically or do i miss-guess what you are trying to do ? [14:47] <durandal_1707> decoder does write palette into frame->data[1] if it does not change, only pal8->pal8 seems to be affected by this [14:48] <durandal_1707> *decoder does not [14:49] <michaelni> for reget_buffer() the decoder should not need to re-write, for get_buffer() the decoder always has to write as it could be a new buffer [14:57] <durandal_1707> so using reget_buffer should better than having pal buffer in decoder itself and doing memcpy? [15:43] <michaelni> reget_buffer() can be much slower than get_buffer()+memcpy of data[1] because reget will copy data[0] too in some cases [15:44] <michaelni> this depends on if the user application can handle reusing buffers (this is often not possible) [17:22] <Freakshow> in the event that I have some changes that may be beneficial upstream in trunk... how can I get these submitted for review? should I send an email out to ffmpeg-devel for review? [17:24] <nevcairiel> yeah, send a patch to the mailing list [17:24] <durandal_1707> Freakshow: use git format-patch, and send-email [17:25] <Freakshow> great... thanks guys. [17:25] <Freakshow> is there any particular subject keyword that should used? [17:26] <nevcairiel> just make it "[PATCH] <first line of commit message here>" [17:37] <Daemon404> git send-email does that on its own anyway. [17:55] <durandal_1707> michaelni: between performance and simplicity i choose later [17:55] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r64a2c51f7b 10ffmpeg/tests/ref/fate/zmbv-8bit: [17:55] <CIA-41> ffmpeg: fate: fix forgotten hunk for framerate order change in ffmpeg.c [17:55] <CIA-41> ffmpeg: broken by: cbb2c95e47cd83cb65b2ce77cbb0296e85fa506b [17:55] <CIA-41> ffmpeg: i had fixed that commit but pushed from an old branch [17:55] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:55] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r64f07754fe 10ffmpeg/ (common.mak configure): [17:55] <CIA-41> ffmpeg: buildsys: fix c++ compilation [17:55] <CIA-41> ffmpeg: Based on code by: carl [17:56] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:58] <michaelni> durandal_1707, its your decoder (PAF) its your decission if you use reget or get+data1 copy [17:58] <durandal_1707> the later case needs more lines in code [18:00] <michaelni> yes but it will be slower with most video players [18:00] <michaelni> thats because the have a que/fifo and cant reuse a buffer before it has been displayed [18:01] <michaelni> so they will copy data[0] + [1] [18:01] <durandal_1707> for game codecs with so small resolution that is really not noticeable [18:01] <michaelni> true [18:14] <michaelni> hi bcoudurier [18:14] <michaelni> any news about fate.ffmpeg.org ? [18:14] <bcoudurier> called them this morning [18:15] <michaelni> what did they say ? [18:17] <CIA-41> ffmpeg: 03Nicolas George 07master * ra85b4a5696 10ffmpeg/libavfilter/video.c: [18:17] <CIA-41> ffmpeg: lavfi: use correct PTS for link age. [18:17] <CIA-41> ffmpeg: When start_frame has returned, the buffer reference's PTS [18:17] <CIA-41> ffmpeg: may have been modified by another filter. [19:48] <Daemon404> j-b, ping. [23:32] <CIA-41> ffmpeg: 03Mans Rullgard 07master * rf3eb008343 10ffmpeg/libavcodec/ (7 files): (log message trimmed) [23:32] <CIA-41> ffmpeg: eamad/eatgq/eatqi: call special EA IDCT directly [23:32] <CIA-41> ffmpeg: These decoders use a special non-MPEG2 IDCT. Call it directly [23:32] <CIA-41> ffmpeg: instead of going through dsputil. There is never any reason [23:32] <CIA-41> ffmpeg: to use a regular IDCT with these decoders or to use the EA IDCT [23:32] <CIA-41> ffmpeg: with other codecs. [23:32] <CIA-41> ffmpeg: This also fixes the bizarre situation of eamad and eatqi decoding [23:32] <CIA-41> ffmpeg: 03Justin Ruggles 07master * rbe51e589cd 10ffmpeg/libavfilter/af_asyncts.c: af_asyncts: avoid overflow in out_size with large delta values [23:32] <CIA-41> ffmpeg: 03Mans Rullgard 07master * r87cf481aa8 10ffmpeg/libavcodec/ (mpegvideo.c mpegvideo_common.h): [23:32] <CIA-41> ffmpeg: mpegvideo: remove unnecessary inclusions of faandct.h [23:32] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [23:32] <CIA-41> ffmpeg: 03Mans Rullgard 07master * r591766a3a9 10ffmpeg/libavcodec/eamad.c: [23:32] <CIA-41> ffmpeg: eamad: remove use of MpegEncContext [23:32] <CIA-41> ffmpeg: There is no sense in pulling in this monster struct just for [23:32] <CIA-41> ffmpeg: a handful of fields. The code does not call any functions [23:32] <CIA-41> ffmpeg: expecting an MpegEncContext. [23:32] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [23:32] <CIA-41> ffmpeg: 03Mans Rullgard 07master * r70c8753d33 10ffmpeg/tests/fate/filter.mak: [23:32] <CIA-41> ffmpeg: fate: amix: specify f32 sample format for comparison [23:32] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [23:32] <CIA-41> ffmpeg: 03Justin Ruggles 07master * r70d71b5801 10ffmpeg/ (doc/filters.texi libavfilter/af_asyncts.c): [23:32] <CIA-41> ffmpeg: af_asyncts: add first_pts option [23:32] <CIA-41> ffmpeg: This allows for padding/trimming at the start of stream. By default, no [23:32] <CIA-41> ffmpeg: assumption is made about the first frame's expected pts, so no padding or [23:32] <CIA-41> ffmpeg: trimming is done. [23:33] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r29d70274ec 10ffmpeg/libavformat/wv.c: wv: K&R formatting cosmetics [00:00] --- Tue Jul 31 2012

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ffmpeg.log.20120730
by burek 31 Jul '12

31 Jul '12

[00:00] <Shimmy> Anyone please help: https://ffmpeg.org/trac/ffmpeg/ticket/747 [00:02] <daniel___> Hello [00:03] <daniel___> Can I please have some help? [00:08] <daniel___> I have a problem with my ffmpeg it is when I try to upload my videos to youtube they play fine on my computer but on youtube they are displayed as a grey box whats the reason for this? can it be fixed? thanks :) [00:11] Last message repeated 1 time(s). [00:11] <burek> ask youtube? [00:12] <daniel___> I thiink it maybe somthing to do with the encoding from ffmpeg I have googled and cant find a fix that why I asked you guys :) [00:13] <frogs> probably encoding it into a codec/container that youtube can't handle [00:15] <daniel___> This is the code im using tell me what I need to tweak to make it work please: :) ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 -threads 0 myvideo.mkv then after the recording i type this for the encoding: ffmpeg -i myvideo.mkv -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -preset slow -crf 22 -threads 0 our-final-product.mp4 [06:05] <Buglouse> I want to censor the -bleep- censorship, does ffmpeg provide a capability to pass a file through a filter removing the specific tones? [06:21] <relaxed> unbleep the bleeps! [06:22] <relaxed> I'm pretty sure that's not possible with ffmpeg. [06:24] <michaeljwjr> Is it possible to write a script that can play one file, then another file when streaming without losing the stream? I want to play an intro, then jump into the stream. Thank you in advance. [06:26] <Buglouse> wget/mplayer dumpfile/dumpstrem [06:26] <relaxed> michaeljwjr: it might be possible with -i concat: [06:27] <soulstitchmmo> ok i'll look that up later. Thank you relaxed. [08:34] <axtroz> Greetings. I'm using FFMPeg to create videos from images on windows. I need to create a blank audio stream with that because the initial video is concated to another video that has audio, but the audio stream is getting lost and the second video has no sound. I red that under *nix and linux one can use /dev/zero. Is there any alternative for windows? NUL and NULL doesn't work :( [11:16] <twid> How guys.....Whats the use of avfilter in ffmpeg...Thanks in advance... [12:01] <rjiejie> hi, i'm a newbie, someone knows about that which file is the "mp4" demuxer in ffmpeg source code ? thanks [12:34] <twid> rjiejie : its libavformat/mov.c for demuxing and libavformat/movenc.c fro muxing... I hope this is what you looking for.. [12:56] <burek> twid [13:11] <rjiejie> twid: thanks for your help :) [13:19] <twid> NO prob... [14:50] <liquidee> is it possible to stream hardware video encoder's MPEG-4 output to ethernet? how can i achieve this with ffmpeg? [15:21] <someone-noone> liquidee, sure. Use tcp://hostname:port or udp://hostname:port as output filename [15:26] <liquidee> avconv -i /dev/video4 -codec copy -b 64k -r 25 -f h264 rtp://192.168.0.116:5004 [15:26] <liquidee> is this correct? because im getting black screen on the receiving laptop [15:28] <liquidee> heres the output: [15:28] <liquidee> http://pastebin.com/Ty6bZ71S [15:29] <saste> liquidee: no you can't read from /dev/video4 that way, you need v4l2 [15:30] <saste> also you're using avconv, here we support ffmpeg [15:30] <liquidee> using ffmpeg says its deprecated and that i should use avconv instead [15:31] <liquidee> how can i read it with v4l2? [15:31] <microchip_> that's a lie [15:31] <saste> liquidee: ffmpeg is deprecated only according to debian [15:31] <liquidee> oh [15:32] <saste> liquidee: http://ffmpeg.org/ffmpeg.html#video4linux2 [15:32] <liquidee> is there any option to command ffmpeg to use v4l2? [15:37] <saste> liquidee: yes read the examples in the linked document [15:38] <liquidee> Requested output format 'video4linux2' is not a suitable output format [15:38] <liquidee> : ( [15:54] <liquidee> what about: [15:54] <liquidee> avconv -v debug -f video4linux2 -i /dev/video4 -codec copy -f h264 rtp://192.168.0.116:5004 [15:55] <bogie> is there any way to access the list of available devices(e.g. dshow devices) via the ffmpeg api directly? as in not printing them to the console, i know setting the list_devices option will print it, but i want to have the list rather than the output. cheers [16:37] <intgr> Hi! Are there any instructions for migrating from deprecated API calls in ffmpeg, such as the av_open_input_file -> avformat_open_input change? [16:39] <saste> intgr: usually doxy tells you the replacement [16:39] <saste> usually it's just a rename, or an added (optional) params [16:39] <saste> *param [16:40] <intgr> Hmm [16:42] <intgr> http://ffmpeg.org/doxygen/trunk/deprecated.html doesn't list av_open_input_file for example... Am I looking at the right place? [16:44] <intgr> Ok, av_open_input_file is already removed (not deprecated) in recent versions... [16:49] <twid> intgr: for that best thing to do is checkout sourcecode... [16:50] <intgr> Looks like it's easier to just freeload Chromium commits for conversion examples ;) [16:51] <twid> Source says about av_open_input_file: "deprecated use avformat_open_input instead." [16:52] <intgr> Ok thanks :) [17:11] <intgr> And I take it that it's not documented anywhere, what version macros I should use for these API calls? [17:29] <twid> intgr : version.h shows 0.10.2 [17:49] <w34> Hi, I am currently trying to port the simple field filter from libmpcodecs [17:50] <w34> and ran into an issue with some mplayer flag [17:51] <w34> Posted my code on pastebin: http://pastebin.com/KjRYXmyM [17:55] <saste> w34: please post development issues on ffmpeg-devel [17:56] <saste> anyway the mp fields are useless in ffmpeg [17:58] <saste> w34: if you want to support only planar formats, you should accept only planar formats [17:58] <saste> that is you need to declare in query_formats the accepted formats [17:59] <w34> alright [18:00] <w34> so query_format is pretty much the right way to implement the flag thingy [18:04] <bogie> is there any way to access the list of available devices(e.g. dshow devices) via the ffmpeg api directly? as in not printing them to the console, i know setting the list_devices option will print it, but i want to have the list rather than the output. cheers [18:07] <saste> bogie: no there is no such API for querying the devices [18:08] <bogie> hmm imho thats a major defect for my purpose, i do not want to deal with the winapi to get a list of devices, would be better if ffmpeg would list those, as it does in dshow_cycle_devices [18:09] <saste> bogie: designing a *generic* interface for that would prove to be rather difficult, since it should work with all devices, that is it should be generic enough [18:09] <saste> anyway patches are welcome [18:10] <bogie> i suspected that ;) [18:53] <gionnico> hi [18:54] <gionnico> can I concatenate mpeg2 files with ffmpeg ? [19:09] <twid> gionnico : yes..and Google gives "cat abc.mpg def.mpg | ffmpeg -f mpeg -i - -vcodec copy -acodec copy ghi.mpg" [19:12] <sacarasc> You can concatenate multiple mpeg2 files using cat a.mpg b.mpg > c.mpg [19:53] <relaxed> The correct way is -i concat:a.mpg\|b.mpg -c copy [21:05] <filo1234> Hi all, can someone exlplain me how to convert a video theora .ogg to 3gp mobile format? I have git version of ffmpeg compiled [21:08] <filo1234> well problem was zixe and audio bitrate [21:09] <filo1234> sorry :) and bye [21:27] <blight> hi guys [21:27] <blight> i have a movie with an mp3 audio stream which is strange [21:29] <blight> the time base of the audio stream is 1/19983 and the pts seems to slowly drift away from the real time [21:30] <blight> i count how many samples i get from the decoder and i calculate the sample position from the time base and pts, i think some rounding errors accumulate so the pts is a bit too slow/fast [21:30] <blight> i guess there is no way to fix this? [21:30] <blight> because after seeking i have to trust the pts [21:30] <blight> (to know where the audio stream is and to sync it with the video) [21:31] <blight> any ideas/input on this problem? [22:01] <fabiobik> hello guys. i want to stream my webcam via internet. [22:02] <fabiobik> i have a trust webcam lsusb gives me this: [22:02] <fabiobik> Bus 001 Device 005: ID 093a:2468 Pixart Imaging, Inc. SoC PC-Camera [22:02] <fabiobik> and my dmesg is [22:02] <fabiobik> just a second [22:03] <fabiobik> http://pastebin.com/1ghNMAVv [22:03] <fabiobik> can please someone help me streaming? [22:07] <llogan> fabiobik: http://ffmpeg.org/ffmpeg.html#video4linux2 [22:07] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20capture%20a%20webcam%20input [22:08] <fabiobik> llogan, thank you for your reply. Im trying using an raspberry to a webcam streamer [22:18] <fabiobik> llogan, i guess video4linux dont have support to my webcam [22:21] <llogan> fabiobik: sorry, i have little experience with webcams. [22:21] <fabiobik> llogan, dont worrie, thanks [22:22] <llogan> someone else might know if you provide more info [22:22] <fabiobik> llogan, im completly noob. what command i should use? [22:27] <llogan> fabiobik: do you know the device path? "/dev/videoN" probably /dev/video0 [22:28] <fabiobik> llogan, well i guess so. it appears on dmesg? [22:28] <fabiobik> http://pastebin.com/1ghNMAVv [22:28] <Element9> I'm trying to record the screen with sound and I can get a got video or good audio, but I record both, audio is too fast [22:29] <Element9> here are command I'm using: http://pastebin.com/KWD28n3Q [22:29] <Element9> am I doing something wrong? [22:32] <llogan> fabiobik: i don't know. try ffmpeg-user mailing list if you don't get an answer here today [22:33] <relaxed> Element9: try setting a framerate before the input [22:34] <relaxed> also, don't use avi as the output container. [22:37] <stampe> any experience here in reading the base media header for timecode data? [22:37] <Element9> relaxed: what do you mean by "before the input"? is the paramter order important? [22:37] <relaxed> yes it is, -r 25 -i $input [22:38] <Element9> relaxed: oh yeah, I do need mp4, not avi, but with mp4 the audio is chopy [22:38] <Element9> relaxed: where can I read about how the order of parameters afect the meaning of the parameter? [22:38] <Element9> relaxed: and thanks for helping me [22:39] <relaxed> no problem, the man page is good [22:39] <Element9> relaxed: it seems like it's not faster when audio plays but it's stutters [22:40] <Element9> relaxed: can ffmpeg.exe produce the same text as "man ffmpef"? I'm Windows [22:40] <Element9> well, there's surely man page for it on the net [22:44] <Element9> and i get tons of these messages "real-time buffer 142% full! frame dropped!" [22:44] <relaxed> Element9: http://dl.dropbox.com/u/24633983/ffmpeg.pdf [22:49] <Element9> is there a specific section that you recommend which could help me with sync problems? or should I just sit down and go through all of it? :) [22:49] <relaxed> Hmm, do you really need lossless? [22:50] <relaxed> Try lowering the input frame rate to 15 or 20. [22:51] <Element9> relaxed: no I don't, I just thought that would be more cpu friendly and less prone to choppiness [22:51] <Element9> let me try that [22:51] <relaxed> and -crf 18 [22:51] <fabiobik> annot find a proper format for codec_id [22:51] <fabiobik> *C [22:52] <relaxed> fabiobik: pastebin your command and all output [22:53] <fabiobik> relaxed, oh sorry [22:53] <fabiobik> one second [22:53] <llogan> generally lossless is used for the capture and then re-encode later to take advantage of lossless capture speed [22:53] <fabiobik> relaxed, http://pastebin.com/VvkdNDjH [22:54] <Element9> relaxed: same result, unfortunately [22:54] <relaxed> fabiobik: -f avconv ? [22:55] <fabiobik> relaxed, wot? [22:55] <llogan> Element9: remove -b 500k. it won't make a difference since it's probably being ignored, but it's mutually exclusive when used with -crf [22:55] <relaxed> llogan: Yeah, I was thinking lossy might be faster and less choppy. [22:56] <Element9> llogan: that's exactly what I'm trying to do. I would like to do realtime compression, but I'm not sure if it's doable with x264 on doable on 2.1ghz dual core cpu. is it? [22:56] <fabiobik> relaxed, sorry not understand the question [22:56] <relaxed> fabiobik: Are you using linux? [22:56] <fabiobik> relaxed, yes :) [22:56] <llogan> Element9: i know nothing of dshow, so i don't know. [22:57] <fabiobik> in my laptop its only linux [22:57] <Element9> llogan: I wouldn't expect dshow to be bottleneck there, but, of course, I'm not sure [23:00] <fabiobik> relaxed, do you know how do i solve this? [23:00] <Element9> relaxed: it looks like dshow uses -framerate instead of -r. or are those unrelated? [23:01] <relaxed> fabiobik: for the audio you probably want "-f alsa -i hw" [23:01] <fabiobik> if i use this avconv -t 10 -f video4linux2 -s 176x144 -r 30 -i /dev/video0 -f avconv -i /dev/dsp1 -f mpeg webcam.mpeg [23:01] <fabiobik> relaxed, i dont want audio [23:01] <fabiobik> just image [23:02] <relaxed> then omit "-f avconv -i /dev/dsp1" [23:02] <relaxed> I'm guessing you don't want mpeg1 either. [23:03] <fabiobik> i just want to test my webcam :) [23:03] <fabiobik> record something [23:04] <relaxed> fabiobik: avplay -f video4linux2 -i /dev/video0 [23:05] <fabiobik> relaxed, i dont have graphic display [23:05] <fabiobik> *graphical desktop [23:06] <relaxed> fabiobik: avconv -f video4linux2 -r 15 -s 640x480 -i /dev/video0 -q:v 3 -y out.avi [23:07] <fabiobik> video4linux2 @ 0x105b400] The V4L2 driver changed the video from 640x480 to 352x288 [23:07] <fabiobik> [video4linux2 @ 0x105b400] Cannot find a proper format for codec_id 0, pix_fmt -1. [23:07] <fabiobik> /dev/video0: Input/output error [23:07] <fabiobik> :S [23:07] <fabiobik> my problem is with codecs? [23:08] <fabiobik> this is a little weird [23:09] <relaxed> Add "-pix_fmt yuyv422" before the input, and stop pasting the error in the channel. Use pastebin.com [23:09] <fabiobik> sorry [23:10] <fabiobik> relaxed, before the input? [23:10] <relaxed> yes [23:10] <fabiobik> before webcam? [23:11] <relaxed> -i = input [23:11] <fabiobik> same error [23:13] <relaxed> change the resolution to -s 352x288 [23:14] <fabiobik> same... [23:15] <fabiobik> avconv -f video4linux2 -r 15 -s 352x288 -pix_fmt yuyv422 -i /dev/video0 -q:v 3 -y out.avi [23:15] <fabiobik> its what im trying [23:15] <relaxed> remove -pix_fmt yuyv422 [23:16] <fabiobik> :\ [23:16] <fabiobik> same error [23:16] <fabiobik> relaxed, thank alot for your help man but appears nothing solve this [23:18] <Element9> just tried: just tried http://pastebin.com/mynFTCYx with the same result as always. CPU is not maxing out and wasn't maxing out on any of the command I tried [23:18] <Element9> maybe there's something from with video and/or audio devices [23:18] <Element9> they do work good separately though [23:21] <Element9> does this work for anyone? with appropriate linux video/audio sources, of course. I guess everyone is on linux here [23:22] <Element9> I mean with settings like this [23:33] <vitrums> Hi there. I'm running ubuntu 12.04 and I recently made some changes in my compiz and unity settings. Then brought all the changes back to the fabric. The problem is that ffmpeg is encoding with 2x speed. What could be the reason of it? [23:34] <Element9> good night everyone [23:35] <piie> hi, I've got a problem with streams when remuxing a video: http://nopaste.info/71b9c12570.html [23:36] <piie> somehow streams are transposed [23:37] <piie> in principle it works, but mplayer is then not able to play the second audiostream [23:58] <llogan> vitrums: encoding twice as fast? isn't that a good thing? i don't see how compiz/unity will generally affect encoding speed. [23:58] <vitrums> llogan: hm... I'll describe my problem a little bit more accurate [00:00] --- Tue Jul 31 2012

1 0

ffmpeg-devel.log.20120729
by burek 30 Jul '12

30 Jul '12

[02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r76888c64b0 10ffmpeg/libavcodec/x86/rv34dsp.asm: rv34: port x86 SIMD to cpuflags. [02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r158744a4cd 10ffmpeg/libavcodec/x86/ (vp56dsp.asm vp56dsp_init.c): [02:26] <CIA-41> ffmpeg: vp56: only compile MMX SIMD on x86-32. [02:26] <CIA-41> ffmpeg: All x86-64 CPUs have SSE2, so the MMX version will never be used. This [02:26] <CIA-41> ffmpeg: leads to smaller binaries. [02:26] <CIA-41> ffmpeg: 03Martin Storsjö 07master * r41ecbbc7aa 10ffmpeg/libavformat/tls.c: (log message trimmed) [02:26] <CIA-41> ffmpeg: tls: Return AVERROR_EOF if the TLS_read/write functions return 0 [02:26] <CIA-41> ffmpeg: OpenSSL returns 0 when the peer has closed the connection. GnuTLS [02:26] <CIA-41> ffmpeg: doesn't return that though, but returns [02:26] <CIA-41> ffmpeg: GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed [02:26] <CIA-41> ffmpeg: without a clean close notify packet. [02:26] <CIA-41> ffmpeg: Tested-by: Antti Seppälä <a.seppala(a)gmail.com> [02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r2734ba787b 10ffmpeg/libavcodec/x86/vp56dsp.asm: vp56: port x86 simd to cpuflags. [02:26] <CIA-41> ffmpeg: 03Martin Storsjö 07master * r8ebacfb598 10ffmpeg/libavformat/hls.c: (log message trimmed) [02:26] <CIA-41> ffmpeg: hls: Proceed to the next segment at any error code [02:26] <CIA-41> ffmpeg: Previously, we returned any error code except AVERROR_EOF to the [02:26] <CIA-41> ffmpeg: caller - only if AVERROR_EOF or 0 was returned, we proceeded to [02:26] <CIA-41> ffmpeg: the next segment. [02:26] <CIA-41> ffmpeg: With some setups of web servers, using Connection: close in https [02:26] <CIA-41> ffmpeg: and GnuTLS, we don't get a clean error code at the end of segments. [02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * rd07ff3cd5a 10ffmpeg/libavcodec/x86/ (dsputil_mmx.c h264_chromamc_10bit.asm): h264_chromamc_10bit: port x86 simd to cpuflags. [02:26] <CIA-41> ffmpeg: 03Mans Rullgard 07master * r23565c2641 10ffmpeg/ (Makefile configure): [02:26] <CIA-41> ffmpeg: build: support non-standard replacements for -c flag [02:26] <CIA-41> ffmpeg: This allows non-standard replacements for the -c compiler flag. [02:26] <CIA-41> ffmpeg: Some compilers use other flags or no flag at all in place of [02:26] <CIA-41> ffmpeg: the usual one. [02:26] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * ra5bbb1242c 10ffmpeg/libavcodec/x86/ (h264_deblock.asm h264_deblock_10bit.asm h264dsp_mmx.c): h264_loopfilter: port x86 simd to cpuflags. [02:26] <CIA-41> ffmpeg: 03Anton Khirnov 07master * refd34918ba 10ffmpeg/libavformat/utils.c: lavf: remove commented out cruft in avformat_find_stream_info() [02:26] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r4d777eedfd 10ffmpeg/libavcodec/x86/ (vp3dsp.asm vp3dsp_init.c): [02:26] <CIA-41> ffmpeg: vp3: don't compile mmx IDCT functions on x86-64. [02:26] <CIA-41> ffmpeg: 64-bit CPUs always have SSE2, and a SSE2 version exists, thus the MMX [02:26] <CIA-41> ffmpeg: version will never be used. [02:26] <CIA-41> ffmpeg: 03Diego Biurrun 07master * rbfe9f48ad7 10ffmpeg/configure: configure: Move parts that should not be user-selectable to CONFIG_EXTRA [02:26] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r0aa907cfb1 10ffmpeg/libavcodec/vc1dec.c: [02:26] <CIA-41> ffmpeg: vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value [02:26] <CIA-41> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [02:27] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r61f8bb74f3 10ffmpeg/libavformat/mp3dec.c: mp3dec: remove commented out cruft. [04:11] <CIA-41> ffmpeg: 03Paul B Mahol 07master * ra3a0774be8 10ffmpeg/libavformat/ (Makefile allformats.c wvenc.c): WavPack muxer [04:11] <CIA-41> ffmpeg: 03Paul B Mahol 07master * rbd93f96540 10ffmpeg/libavformat/ (Makefile apetag.h apetagenc.c): APE tag writer [04:11] <CIA-41> ffmpeg: 03Paul B Mahol 07master * rc25dc1f9c7 10ffmpeg/libavformat/wvenc.c: [04:11] <CIA-41> ffmpeg: wvenc: support for ape tags [04:11] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:27] <CIA-41> ffmpeg: 03Paul B Mahol 07master * rc3c9c4d006 10ffmpeg/doc/general.texi: [04:27] <CIA-41> ffmpeg: doc/general: add missed WavPack muxing support information [04:27] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:27] <CIA-41> ffmpeg: 03Paul B Mahol 07master * r29ba3aacb1 10ffmpeg/libavformat/smacker.c: [04:27] <CIA-41> ffmpeg: lavf/smacker: remove bogus video from .long_name [04:27] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:32] <CIA-41> ffmpeg: 03Paul B Mahol 07master * re4f3a9693d 10ffmpeg/libavformat/ (apetag.c apetag.h apetagenc.c): [04:32] <CIA-41> ffmpeg: lavf/apetag: move common stuff between writer and reader to single file [04:32] <CIA-41> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [11:48] <ubitux> is it me or the convergence_duration is actually never used? [11:49] <ubitux> (since it's been removed from mkv mux & demuxer) [12:32] <nevcairiel> mkv demux still uses it [12:32] <nevcairiel> unless duration is changed to int64 some day, it also has to or the duration will just overflow [12:34] <nevcairiel> or you cant have subtitles longer then 2-something seconds in MKVs with nanosecond timebase [12:34] <ubitux> mmh what did i smoke again. [12:35] <ohsix> for when theres 40 million fps video [12:57] <saste> 40 million fps video: useful when filming the big bang [12:58] <ubitux> speaking of this, i just saw http://www.ted.com/talks/ramesh_raskar_a_camera_that_takes_one_trillion_fra… [13:10] <iive> wasn't that one that actually was taking one pixel at a time, so it relied on repeating the same event over and over again? [13:10] <iive> i think they filmed how the light is moving. [13:11] <ubitux> yeah afaiu they send the light several time with shift [13:13] <ubitux> lavfi question: assuming i have a filter with same input & output format/sizes, in case of NULL start_frame callback, the out_buf will be set to a new similar buffer; can I assume the linesizes are the same for src & dst? [13:13] <ubitux> iirc linesize includes the padding, may it differs between the two? [13:17] <ohsix> if you could assume that, it would be API, and for that it's a poopy thing to guarantee :p [13:20] <saste> ubitux: no [13:20] <ubitux> ok :( [13:20] <saste> ubitux: the default start_frame() will call get_buffer on the next frame [13:21] <saste> you could have a pad for example, which will allocate a buffer with a big linesize [13:21] <ubitux> mmh i see [13:21] <saste> so you can't do predictions on the linesizes [13:22] <ubitux> ok, thx [13:37] <maker> uhm, I'm taking some examples from libavfilter/vf_* for creating mine; but (excluding multimedia.cx docs, which as michaelni said, it's obsolete), I see from doxygen that draw_slice should be used for processing [13:38] <maker> even though most of the filters uses start_frame for processing the link->cur_buf->video->data. Am I missing something? [13:40] <saste> maker: draw_slice is useful when you can process frame slices independently [13:41] <saste> that's not true for all the filters, and sometimes it is simpler to do all the processing at once in end_frame() [13:41] <saste> the idea is that slicing processing should improve overall performance relying on cache locality principle [13:42] <saste> so it really depends on the filter logic, and if it slice processing is worth the extra complexity [13:43] <saste> check drawbox for a simple example [13:46] <maker> Ok, I was referring to crop an vflip, which are nearest to what I intend to do. [13:47] <maker> Also, what are planes? [13:55] <saste> maker: an images usually consists of several planes [13:55] <saste> a planes contains several related pixels, depending on the pixel layout [13:56] <saste> for example for RGB you have a single plane containing the sequence of RGBRGBRGB... pixels [13:56] <saste> with planar YUV you have a single plane for Y, U, and V [13:56] <saste> you can have planar RGB when you store each "channel" on a different plane [13:57] <saste> pixfmt.h should explain this, once you get the idea [13:57] <saste> layout information is stored in lavutil/pixdesc.h [13:57] <maker> Thanks! [13:58] <saste> you can choose the better layout which fits with your filtering logic, and let the explicit conversion to the framework [13:59] <saste> *leave the explicit... [14:00] <ubitux> http://ubitux.fr/pub/pics/edgedetect.png [14:00] <saste> ubitux: sobel? [14:01] <ubitux> yep [14:01] <ubitux> do we have that already? [14:01] <saste> ubitux: no, only in frei0r [14:02] <saste> but there are several edge detector operators, we could make it configurable [14:03] <ubitux> i was trying to make a canny edge detector [14:04] <ubitux> right now i'm just using boxblur filter + manual sobel [14:04] <saste> also a generic convolution filter may help [14:04] <ubitux> i also have a gaussian filter but i need to play with several buffers within the filter [14:04] <ubitux> i'm not sure how to do that yet [14:05] <saste> ubitux: i'm reading the buffer permission thread, i should reply to it today, then we shall improve documentation on that still somewhat obscure area [14:06] <ubitux> yeah would be nice :) [15:55] <saste> ubitux: would you mind posting your rdft patch as a WIP or an RFC, that may also help in the discussion related to buffer perms [15:55] <ubitux> i already did a month ago or two [15:55] <ubitux> iirc. [15:56] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2012-May/125037.html [15:56] <ubitux> not much up to date though [15:56] <ubitux> i'll rebase & resubmit in a few hours [15:57] <saste> ah ok, i completely forgot, thanks [15:58] <ubitux> if you want filter examples for the buffer perms stuff [15:58] <ubitux> color is a nice one [15:58] <ubitux> (re-use the same local buffer ref) [15:58] <ubitux> showspectrum is indeed a good candidate because there is the issue of updating a local bufferref [15:59] <saste> does color do it now? [15:59] <ubitux> nope iirc [15:59] <ubitux> Nicolas pointed that out when durandal_1707 was trying to submit the color bars filter [15:59] <saste> yes indeed [16:00] <ubitux> and another hypothetical candidate, in the exact same spirit of showspectrum, would be an histogram video filter which i'd like to write [16:00] <durandal_1707> hey please finish that color bars filter (i'm not going to back to it anytime soon) [16:01] <ubitux> saste: and btw that might also applies for showwaves; if given N samples you can't raise a complete frame, maybe it could raise a duplicate of the previous [16:02] <saste> ubitux: no i don't like that very much, at least for the default behavior [16:02] <ubitux> ok [16:02] <saste> durandal_1707: ok, i'll try to update your patch [16:07] <saste> i wonder if it would be a good time to drop the deprecated syntax of color, so I can merge it with testsrc [16:09] <durandal_1707> saste: drop it [16:21] <CIA-41> ffmpeg: 03Nicolas George 07master * r5c81a9ff55 10ffmpeg/libavcodec/libx264.c: [16:21] <CIA-41> ffmpeg: libx264: list possible profiles. [16:21] <CIA-41> ffmpeg: The values are listed if setting them fails. [16:21] <CIA-41> ffmpeg: Using "-profile help" or "-profile list" have that effect. [16:21] <CIA-41> ffmpeg: Similar to 3aba391. [16:21] <CIA-41> ffmpeg: Suggested by "rogerdpack" in trac ticket #1529. [16:32] <nyuhu> I've been getting a grey image when testing my filter port (hue) and Ive just realized that it also does that when I dont do any processing at all in the end_frame function (i.e. just making a call to ff_end_frame) ; any idea of what could be the cause ? this looks like a rookie mistake imo& [16:33] <CIA-41> ffmpeg: 03Nicolas George 07master * r981d97f697 10ffmpeg/doc/filters.texi: doc/filters: document TB variable for vf_setpts. [16:34] <ubitux> nyuhu: is there any auto inserted scale filter (look at the cli output)? [16:35] <ubitux> if so you might have requested a gray input or output somewhere [16:35] <ubitux> (check your query_formats) [16:36] <nyuhu> no I didnt put any gray format in my query_formats function [16:37] <nyuhu> and it doesnt seem like theres an auto inserted scale [16:37] <durandal_1707> nyuhu: where is output? [16:38] <nyuhu> what do you mean ? [16:40] <durandal_1707> for that problem you have [16:41] <ubitux> nyuhu: what's your command line? [16:42] <nyuhu> ./ffplay test-2.mp4 -vf hue=0:1 [16:44] <saste> nyuhu: post the code on pastebin [16:45] <nyuhu> http://pastebin.com/2hd71jTY [16:50] <saste> nyuhu: ok so you're using the default start_frame, a buffer is requested on the output link, and stored in outlink->out_buf [16:50] <saste> then you don't write nothing to this buffer, so it will in general contain just random data (or it will be zeroed, can't remember) [16:51] <nyuhu> oh I see [16:52] <saste> what happens when you actually *write* into out_buf? [16:52] <nyuhu> then I would have to put the input link buffer in the output link buffer if I didnt want to do anything right ? [16:53] <nyuhu> gray image too [16:53] <saste> you will have to *copy* the data from inlink->cur_buf to outlink->out_buf [16:54] <nyuhu> (well this is true when I test it with hue = 0 and saturation = 1, which would of course output the same image ; the other cases need some debugging) [16:54] <saste> out->data[0] = in->data[0]; [16:54] <saste> this is wrong [16:56] <saste> you're making out_buf pointing to the data contained in cur_buf [16:56] <nyuhu> yes [16:57] <saste> forget about the special cases for now, consider the input and output buffers as distinct entities [16:57] <nyuhu> got it [16:57] <saste> there are some subtleties involved with buffer management, anyway you're not supposed to change the pointers to data in a buffer [16:58] <nyuhu> thank you ! [17:33] <ubitux> saste: ok it seems i don't leak anymore actually [17:33] <saste> heisenbug? [17:33] <saste> maybe the anton changes helped in that regard [17:34] <ubitux> possible [17:35] <ubitux> i'm adding the color and will re-submit [17:58] <ubitux> saste: isn't insamples->audio->channel_layout always supposed to be set when calling filter samples? [17:59] <ubitux> seems it is assumed to be correctly set [18:00] <saste> ubitux: I think so, but that shouldn't be strictly required since most filters don't care about what is written there and assume the configured value in the link [18:00] <saste> that's the same as for picref->video->w/h [18:00] <saste> but yes it's better to set it anyway, since a filter may read from it [18:01] <ubitux> ./ffmpeg -f lavfi -i 'aevalsrc=sin(440*2*PI*t)' -t 5 -ac 2 -y test.wav [18:02] <ubitux> then try to showwaves it [18:02] <ubitux> ./ffplay -f lavfi "amovie=test.wav,asplit[out1],showwaves=s=640x480[out0]" [18:02] <ubitux> div by 0 since the invalid channel layout leads to nb_channels = 0 [18:10] <ubitux> seems like i just found another issue [18:11] <ubitux> wget 'lucy.pkh.me/samples/6ch.flac' && ./ffplay -f lavfi 'amovie=6ch.flac,aformat=channel_layouts\=stereo' [18:11] <ubitux> segfault [18:35] <saste> ubitux: why chlayout is not set in the samples buffer? [18:55] <ubitux> saste: why do you ask me? :)) [18:56] <saste> ubitux: don't know, maybe you already debugged it ;^) [18:56] <ubitux> nop i just love raising problems [18:58] <saste> the problem is that chlayout is not set in the AVFrame [18:58] <maker> lol [19:01] <saste> and it is not set in AVFrame because it was not set in the codec context [19:12] <saste> michaelni: what's the best way to deal with an unknown chlayout in the decoder? [19:13] <Daemon404> doesnt have have a predefie nd layour for 5ch [19:13] <Daemon404> er 6ch [19:13] <Daemon404> doesnt flac* [19:13] <saste> should be set in avcodec_open() to the chlayout guessed from the number of channels? or should we leave it to 0 in case it is not explicitely defined by the decoder? [19:15] <saste> Daemon404: the problem is not only with flac, i have the very same problem with pcm and 8svx streams [19:15] <michaelni> hmm, if the decoder spec specifies some default then that should be used [19:15] <michaelni> otherwise iam tending to suggest to leave it 0 [19:16] <saste> ok, maybe we should specify that in the docs (chlayout = 0 => unspecified/unknown) [19:16] <michaelni> yes [19:18] <saste> on the other hand at the filtering level we always need to set it in the samples buffer [19:19] <michaelni> why? [19:21] <ubitux> saste: i've updated my showspectrum branch; but i still have issues with cluttered playback in some cases [19:22] <ubitux> (i've added the colors and there is no leak though) [19:22] <ubitux> (i'm sure i've messed the permissions though) [19:22] <saste> michaelni: some filters read chlayout and use that value for getting the nb channels [19:23] <michaelni> so it will crash if i set 1 channel layout=stereo ? [19:24] <saste> michaelni: why, can you really have that situation? [19:24] <saste> anyway yes, the number of channels info is guessed from the chlayout [19:24] <saste> same is for links iirc [19:26] <saste> also in the frame we store the chlayout info, which is then read and passed to the lavfi samplesref [19:26] <saste> since there is no channels field in AVFrame, the only way to pass the nb_channels info is through the chlayout [19:28] <michaelni> a decoder surely could set the layout and channels to inconsistent values [19:31] <saste> i see several possible solutions: we always set the chlayout information in AVFrame, if not specified we use get_ch_layout_nb_channels() [19:32] <saste> at the decoding stage we're sure that chlayout is consistent with nb_channels, or the open function will complain and fail [19:32] <michaelni> it could be changed after open [19:33] <michaelni> i think a channels field shoudl be added to AVFrame [19:33] <michaelni> otherwise a unknown channel layout with X channels becomes tricky [19:34] <michaelni> also do we have a sane default channel layout for any number of channels ? [19:38] <saste> up to 8 channels [19:48] <michaelni> so 9 channels will fail if theres no channel layout stored [22:04] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r72dabdfc58 10ffmpeg/libavcodec/ (aacpsy.c psymodel.c psymodel.h): [22:04] <CIA-41> ffmpeg: aacenc: new default cutoff [22:04] <CIA-41> ffmpeg: Improves subjective quality [22:04] <CIA-41> ffmpeg: Formula and testing by: kamedo2 <fujisakihir90(a)yahoo.co.jp> [22:04] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:09] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * rdcb7ef5483 10ffmpeg/ (avconv.c avprobe.c): avprobe/avconv: fix tentative declaration compile errors on MSVS. [23:09] <CIA-41> ffmpeg: 03Loren Merritt 07master * r60b9785530 10ffmpeg/libavfilter/vf_hqdn3d.c: [23:09] <CIA-41> ffmpeg: vf_hqdn3d: cosmetics [23:09] <CIA-41> ffmpeg: Change code style to match the rest of libav. [23:09] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [23:09] <CIA-41> ffmpeg: 03Loren Merritt 07master * rfb44e7401f 10ffmpeg/libavfilter/ (vf_delogo.c vf_gradfun.c video.c video.h): [23:09] <CIA-41> ffmpeg: factor identical ff_inplace_start_frame out of two filters [23:09] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [23:09] <CIA-41> ffmpeg: 03Loren Merritt 07master * r1ad715dbf3 10ffmpeg/libavfilter/vf_hqdn3d.c: [23:09] <CIA-41> ffmpeg: vf_hqdn3d: support 9 and 10bit colordepth [23:09] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [23:09] <CIA-41> ffmpeg: 03Loren Merritt 07master * r85e228c71d 10ffmpeg/libavfilter/vf_hqdn3d.c: [23:09] <CIA-41> ffmpeg: vf_hqdn3d: simplify and optimize [23:09] <CIA-41> ffmpeg: 14% faster on penryn, 2% on sandybridge, 9% on bulldozer [23:09] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [23:09] <CIA-41> ffmpeg: 03Anton Khirnov 07master * rfe1c1198e6 10ffmpeg/ (3 files in 2 dirs): (log message trimmed) [23:09] <CIA-41> ffmpeg: lavf: use dts difference instead of AVPacket.duration in find_stream_info() [23:09] <CIA-41> ffmpeg: AVPacket.duration is mostly made up and thus completely useless, this is [23:09] <CIA-41> ffmpeg: especially true for video streams. [23:09] <CIA-41> ffmpeg: Therefore use dts difference for framerate estimation and [23:09] <CIA-41> ffmpeg: the max_analyze_duration check. [23:09] <CIA-41> ffmpeg: The asyncts test now needs -analyzeduration, because the default is 5 [23:09] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7c26761b81 10ffmpeg/: (log message trimmed) [23:09] <CIA-41> ffmpeg: Merge commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23' [23:09] <CIA-41> ffmpeg: * commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23': [23:09] <CIA-41> ffmpeg: lavf: use dts difference instead of AVPacket.duration in find_stream_info() [23:09] <CIA-41> ffmpeg: avf: introduce nobuffer option [23:09] <CIA-41> ffmpeg: fate: make yadif tests consistent across systems [23:09] <CIA-41> ffmpeg: 03Loren Merritt 07master * r0f583e6cc5 10ffmpeg/libavfilter/vf_hqdn3d.c: [23:09] <CIA-41> ffmpeg: vf_hqdn3d: reduce intermediate precision [23:09] <CIA-41> ffmpeg: 11% faster on penryn, 7% on sandybridge, 5% on bulldozer [23:09] <CIA-41> ffmpeg: Negligible change to output. [23:09] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [00:00] --- Mon Jul 30 2012

1 0

ffmpeg.log.20120729
by burek 30 Jul '12

30 Jul '12

[00:33] <soulstitchmmo> Good morning everyone. I've gotten my stream working on Twitch (yay) however in Windows I use XSplit and I'm able to change scenes, and run promos and do all manner of scene switching and such. What I would like to know if while a stream is open am I able to send other sources? Or at the very least when I start the stream stream 1 video, then the live stream? [00:49] <burek> judget [00:49] <judget> I am using that guide [00:50] <burek> what's the exact message you get [00:50] <judget> il have to go back and run the config aagin meanwhile it did build [00:50] <burek> ok [00:52] <judget> heres the message: [00:52] <judget> WARNING: Please upgrade to VA-API >= 0.32 if you would like full VA-API support. [00:52] <soulstitchmmo> burek! yay :D [00:53] <burek> judget dpkg -l | grep libva [00:53] <burek> and update it [00:53] <burek> also libva-dev [00:53] <burek> soulstitchmmo :beer: :) [00:54] <judget> how do I update it (DUH) [00:54] <burek> judget :) read the ubuntu manual :) [00:54] <burek> or ask in #ubuntu [00:54] <judget> OK tx [00:56] <soulstitchmmo> burek did you see my question from earlier? [01:52] <pzaljr> so I'm installing on a Ubutuntu 12 precise server [01:52] <pzaljr> getting the error: /usr/bin/ld: libavutil/intfloat_readwrite.o: relocation R_X86_64_PC32 against undefined symbol `ldexp@@GLIBC_2.2.5' can not be used when making a shared object; recompile with -fPIC [01:52] <pzaljr> on the make [01:59] <vitrums> I've got an issue with pulseaudio, streaming online via rtmp. FFmpeg starts repeatedly writing warnings: DTS 301435891, next:302266281 st:0 invalid dropping. This just happens at some point and there's no way back to have it working again until restart of the stream. Any idea? [02:04] <iive> pzaljr: i've heard that this happens if one of the libraries is not compiled as PIC, i hope it is not glibc in your case. gentoo? [02:10] <rud> hi, i am currently using HEAD snapshot of ffmpeg. my current case scenario is crtmpd > pull> ffmpeg (transcode) > push > crtmpd+auth & it works perfectly fine until i enable RTMP auth. my streaming app supports said RTMP auth and works fine, but i can't find the proper syntax to tell ffmpeg to push the stream using a given login&password & here is what i'm currently doing : http://pastebin.com/nGLGzw3S .. if someone has a clue it would be much appreciated. [02:17] <pzaljr> iive: ubuntu precise with a bitnami stack [02:30] <Macey_home> Hi All, Sorry is there a decent c# binding for ffmpeg, i've searched all day wasted hours on sharpffmpeg which is horribly out of date... Any idea if there is a updated bindings? [02:54] <vitrums> I've removed pulseaudio, but I'm still running at the same issue [03:17] <vitrums> I'm running Ubuntu 12.04 x64. Not so log ago I had a trouble with -fPIC g++ compiler option due to x64 arch, but I really hope it's not my case ^^ Though, talking about ffmpeg... it can be just anything :paka: [04:50] <soulstitchmmo> Why does FFMPEG use so much CPU? [04:50] <soulstitchmmo> I am watching my htop, and I can see when I start ffmpeg streaming either ffmpeg, or compiz jump to 100% and even out at about 60-80% [05:00] <relaxed> encoding is cpu intensive [05:07] <soulstitchmmo> Why? In windows using Xpslit, it can do it flawlessly with barely any perceptible difference. In Linux it basically eats all my CPU. [05:46] <relaxed> You can limit its cpu usage in linux too. [05:54] <soulstitchmmo> relaxed: How do I limit it's cpu usage? [05:57] <relaxed> "it's" is a contraction for "it is". Look at cpulimit. [09:27] <noisymouse> Hi, I have a grayscale source and I'm looking for a codec that only encodes the darkness and no color (and saves space by doing that) [09:28] <frogs> option 'gray' [09:28] <frogs> doesn't give much help as the majority of bits go to the lightness already [09:29] <noisymouse> ok -- I would have thought that color would take up 2 or 3x extra space [09:30] <noisymouse> also I noticed when I used just the -gray option that the video comes out lighter (even though I can tell using VLC's brightness filter that the information all still there) [09:31] <noisymouse> err I mean darker [09:31] <noisymouse> does it cut off black at the most black pixel and then reassign brightness values? (this is with x264) [09:32] <frogs> 'gray' it self does not, but the encoder can. [09:32] <noisymouse> I see [09:33] <frogs> try encoding to a lossless format and see if the video is processed in the way you want [09:33] <frogs> like ffvhuff [09:33] <frogs> then you can encode that lossless one and see where the loss of quality is occuring [09:36] <noisymouse> ok, thanks [09:59] <soulstitchmmo> can someone show me how to use the cpulimit command with my ffmpeg command? [10:00] <soulstitchmmo> I'm not quite understanding where to put the cpulimit command [10:03] <tdr> soulstitchmmo, if you mean "cpulimit" command, i found a guide that says to start your ffmpeg, then cpulimit -e ffmpeg -l percentage ... but I don't know how that would scale threaded across cores [10:05] <tdr> you could also just nice it so it doesn't get in the way of other apps [10:06] <soulstitchmmo> the nice command did not seem to work, it was still running at near 100% cpu usage. [10:10] <soulstitchmmo> tdr, I figured it out, but I can't run it all in one script? [10:10] <soulstitchmmo> I have to open two terminals? [10:10] <tdr> soulstitchmmo no idea, i've never used it [10:10] <soulstitchmmo> ahh ok [10:20] <soulstitchmmo> WOW [10:20] <soulstitchmmo> what a huge difference [10:27] <vitrums> What might cause unwanted behaviour of the audio, regardless if I use pulse or directly a card:device. It just goes crazy in several minutes since the RTMP stream is up, and every package of audiostream is being dropped. I've tried to stream up on justin.tv and own3d.tv. The result is the same. Therefore I'm literally force not to use sound at all :rage_why: ?! [10:28] <vitrums> forced* [11:06] <raptor67782> can ffmpeg for debian stable play bbc radio: ? ffplay "http://bbcmedia.ic.llnwd.net/stream/bbcmedia_intl_lc_radio1_p?s=1343545294&…" [11:33] <karatemyran> Hey iam having some problem to set up a brodcast using ffmpeg. Iam new 2 linux and dont know what 2 do [11:35] <soulstitchmmo> first do this: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [11:35] <soulstitchmmo> where are you trying to stream to? [11:35] <karatemyran> justin [11:36] <karatemyran> justin.tv [11:36] <karatemyran> i have instaled that [11:36] <soulstitchmmo> you went through that particular page? [11:37] <soulstitchmmo> karatemyran, you installed it according to that guide, or some other guide? [11:37] <karatemyran> that guide [11:38] <soulstitchmmo> ok do you know how to make a script in your .bashrc? [11:38] <karatemyran> not rly thats the problem [11:38] <karatemyran> iam trying [11:38] <soulstitchmmo> ok goto Home, and hit CTRL+H to show hidden files. [11:38] <soulstitchmmo> then find your .bashrc and open it (should open in a text editor [11:38] <karatemyran> yes [11:39] <soulstitchmmo> then copy paste: http://pastebin.com/MRagM2E3 at the bottom [11:39] <soulstitchmmo> and of course at the top for stream key, use your own stream key [11:39] <soulstitchmmo> so for STREAM_KEY="" put your stream key between the quotation marks [11:40] <karatemyran> yy i got that =) [11:40] <soulstitchmmo> then save it, open a terminal, type: source .bashrc [11:40] <soulstitchmmo> then type twitch and hit enter [11:40] <soulstitchmmo> (if you really want to you can change the twitch to justin if that makes you feel better) [11:41] <karatemyran> nothing happens =( [11:41] <soulstitchmmo> you type in twitch, and you don't even get an error? [11:42] <karatemyran> wait [11:42] <karatemyran> now it came [11:42] <karatemyran> somthing [11:42] <karatemyran> [x11grab @ 0x96046a0] device: :0.0+0,0 -> display: :0.0 x: 0 y: 0 width: 1920 height: 1080 [x11grab @ 0x96046a0] shared memory extension found X Error of failed request: BadMatch (invalid parameter attributes) Major opcode of failed request: 142 (MIT-SHM) Minor opcode of failed request: 4 (X_ShmGetImage) Serial number of failed request: 11 Current serial number in output stream: 11 [11:43] <soulstitchmmo> what's the resolution of your monitor? [11:43] <karatemyran> ohh [11:43] <karatemyran> i feel stupid now [11:43] <karatemyran> wait [11:43] <soulstitchmmo> lol :p [11:44] <soulstitchmmo> INRES = the resolution you want to record at, I have min set to my monitor. [11:45] <karatemyran> but iam streaming now [11:45] <karatemyran> but got black screen [11:46] <soulstitchmmo> are you streaming sound as well [11:46] <karatemyran> naa =( [11:46] <karatemyran> http://sv.twitch.tv/karatemyran [11:47] <soulstitchmmo> hrmmm. [11:47] <karatemyran> maybe i got to hight output [11:47] <karatemyran> ? [11:48] <soulstitchmmo> Well that's about as far as my expertise goes. I just went through this process myself. [11:48] <soulstitchmmo> You have it streaming, just play with the settings now till you get it workign right. [11:48] <soulstitchmmo> Hopefully someone more knowledgeable can help you. [11:48] <soulstitchmmo> You play LoL in Ubuntu? [11:48] <karatemyran> okej [11:49] <karatemyran> THX [11:49] <karatemyran> for all your help [11:49] <karatemyran> your awsume [11:49] <karatemyran> <3 [11:49] <karatemyran> btw how do i stop the stream [11:49] <karatemyran> just close terminal? [11:49] <soulstitchmmo> q [11:49] <soulstitchmmo> q should cancel it. [11:50] <karatemyran> y it did [11:50] <karatemyran> =) [11:50] <karatemyran> thx alot [11:50] <karatemyran> rly [11:50] <soulstitchmmo> ok that's it for me. bed time. [11:50] <soulstitchmmo> good luck [11:55] <AlRazi> I have a question, when i'm trying to download libaacplus their website states ( latest ffmpeg git tree already patched ) but when i try to configure ffmpeg with libaacplus support i get ( ERROR: libaacplus >= 2.0.0 not found) [11:55] <AlRazi> what am i missing ? [12:10] <vitrums> Hi. I've got a task to grab a screen with x11grab and also grab my webcam so that the output video will contain them both simultaneously and the webcam video region would take only a small region of the sceen. Is that possible with ffmpeg? [12:11] <vitrums> I actually mean muxing 2 video streams [12:15] <saste> vitrums: overlay filter [12:25] <karatemyran> can someone help me iam trying to bradcast on twitch but, i just get blackscreen [12:27] <vitrums> karatemyran -pix_fmt yuv420p [12:28] <vitrums> use this pixel format [12:28] <karatemyran> okej and what does that do [12:29] <kauwlna> i am too stupid to cut accurately, "-t 00:00:52.800" seems to cut on the nearest keyframe. is there anything cutting on a frame basis? [12:29] <vitrums> your output .flv would be supported by most services, including justin, twitch and so on [12:29] <vitrums> I killed the whole yesterday before I found the solution [12:30] <karatemyran> okej so i just put that in the bashrc? [12:30] <vitrums> put it next to your vcodec parameters [12:30] <karatemyran> shuld i replace -preset:v? [12:31] <vitrums> if you output locally you recordings as -f flv out.flv and ffprobe out.flv, you'll notice the new pixel format 444, which is still not widely supported [12:33] <karatemyran> http://sv.twitch.tv/karatemyran [12:33] <karatemyran> WOHO [12:33] <vitrums> preset must be chosen to ultrafast i guess [12:33] <karatemyran> haha it worked [12:33] <vitrums> you're lucky) [12:33] <karatemyran> yes iam [12:33] <karatemyran> can you help me with the sound 2 ? [12:33] <karatemyran> btw is the stream laggy or fine ? [12:34] <vitrums> I'm still fighting with it ( Can't get it work stable [12:34] <karatemyran> the sound? [12:34] <vitrums> run fullscreen flash video and we'll see [12:34] <karatemyran> now youu lost me [12:35] <vitrums> btw don't toggle mute sound, if you open your broadcast on stream [12:35] <vitrums> see? Drops( I've got the same issue [12:36] <karatemyran> y i see that =( [12:36] <Kuukunen> karatemyran: seems a bit laggy here [12:36] <vitrums> try playing youtube video [12:36] <karatemyran> ok [12:36] <Kuukunen> otoh, I'm on an Android phone :P [12:36] <vitrums> we'll test your grab capabilities [12:37] <karatemyran> get eny sound ? [12:37] <vitrums> we don't have sound, it's all dropped. Make your video fullscreen plz [12:38] <vitrums> no, I mean a real video... some motion picture, not a static photo) [12:38] <karatemyran> okej [12:39] <karatemyran> did it work? [12:39] <vitrums> you've got at least two issues. First: bandwidth. Try to cap it with b:v 1500k option (or what you can afford) [12:40] <karatemyran> ffmpeg -f x11grab -s $INRES -r $FPS -i :0.0+0,0 \ -f alsa -ac 2 -i pulse -vcodec libx264 -pix_fmt yuv420p -preset:v $QUAL -s $OUTRES \ -acodec libmp3lame -ab 96k -ar 44100 -threads 0 \ -f flv rtmp://live.justin.tv/app/$STREAM_KEY [12:40] <karatemyran> where ? [12:40] <vitrums> Second: glitching Unity + top menu bar. I've been fighting upon this issue for the last day. Still can't seem to find any solution [12:41] <karatemyran> where do i pu 1500 k option? [12:41] <vitrums> make QUAL=ultrafast and put after "-s $OUTERS" -b:v 1500k [12:41] <karatemyran> okej btw [12:41] <karatemyran> what upload speed do i need 2 have for that 1,5 mbit ? [12:42] <vitrums> I've got 2.2Mbit on speedtest.net. It seems like b:v 1650 is a limit [12:43] <vitrums> 1650k * [12:44] <karatemyran> ok i got 2.1 [12:45] <vitrums> ok. You really want to have 30+ fps, ultrafast preset and b:v limit. That's what will make your stream smooth [12:46] <karatemyran> okej [12:47] <karatemyran> but dont think my comp can handel thatt [12:47] <karatemyran> i have it on 35 [12:47] <vitrums> If you were apt to stream LoL, than afaik this settings would be more then necessary [12:49] <karatemyran> i have the setings you said now [12:50] <karatemyran> still no sound [12:50] <vitrums> have your lol running for a minute [12:51] <karatemyran> its droping alot =( [12:51] <karatemyran> PTS 119214130, next:120834790 invalid dropping st:0 DTS 119220661, next:120836123 st:0 invalid dropping PTS 119220661, next:120836123 invalid dropping st:0 DTS 119227084, next:120837456 st:0 invalid dropping PTS 119227084, next:120837456 invalid dropping st:0 DTS 119233402, next:120838789 st:0 invalid dropping PTS 119233402, next:120838789 invalid dropping st:0 DTS 119239616, next:120840122 st:0 invalid dropping PTS 119239616, n [12:52] <vitrums> the tutorial on that HoN forum is kinda legacy, since modern pulseaudio does have a different interface. I thought it's all about pulse, however the direct audio input isn't better. The same issue appears [12:52] <vitrums> please, use pastebin.com to post your logs [12:52] <karatemyran> okej [12:53] <vitrums> don't spam the IRC with your configs [12:53] <saschagehlich> hey, I'm trying to convert a .mp4 movie into single .gif images but everything i get is completely grey gif images. is there a setting I have to change to avoid that? [12:54] <saschagehlich> (os: mac osx mountain lion, got the same problem on osx lion though) [12:56] <vitrums> unfortunately ffmpeg is a way way more heavy than XSplit and you've got to be more careful with cpu setup. Actually i'm working on it right now. There must be a way to assign cpu units directly [12:57] <karatemyran> wow this is laggy [12:58] <karatemyran> i rly cant stream [12:59] <karatemyran> its taking to mutch of my cpu =( [13:01] <vitrums> karatemyran, what is your HW? [13:02] <karatemyran> HW? [13:03] <vitrums> afaik ffmpeg is the best amids other tools. There's one Java based application and VLC. Couldn't make them work [13:04] <vitrums> hardware [13:04] <karatemyran> aha wait [13:05] <karatemyran> Intel® Core" i3 CPU 530 @ 2.93GHz × 4 [13:05] <karatemyran> GeForce GTS 250/PCIe/SSE2 [13:05] <karatemyran> 4 g ram [13:06] <karatemyran> brb [13:20] <blight> hi guys [14:26] <vitrums> saste, <3!!! webcam works just perfect [14:37] <vitrums> So there're actually two major issues yet unresolved. First is that the audio starts lagging and freezes completely over the time. Autoremoving pulseaudio didn't solve it. [14:40] <vitrums> Second is about Unity (or perhaps just top menu bar), which appears to be glitching upon fullscreen movies (regardless if it were flash via browser, or locally played video with Totem) [19:20] <xastey> I'm running into some problems encoding flac->mp3 .. I have followed http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide but it still fails [19:20] <xastey> http://pastebin.com/ZcUvFDwy output with -format showing lamemp3 is installed [19:21] <relaxed> -c libmp3lame [19:21] <xastey> add that to my encoding command? [19:22] <relaxed> hmm, wait [19:22] <xastey> Invalid encoder type 'libmp3lame' [19:23] <relaxed> ffmpeg -codecs | grep libmp3 [19:23] <xastey> libmp3lame is installed into /usr/local/iib [19:23] <xastey> EA libmp3lame libmp3lame MP3 (MPEG audio layer 3) [19:25] <relaxed> ffmpeg -i input -vn -c:a libmp3lame -ar 44100 -ab 128k -t 90 output.mp3 [19:26] <xastey> PERFECT!! [19:27] <xastey> so whats the -vn and -c:a part for.. [19:27] <relaxed> -vn = ignore video | -c:a specifies the audio codec [19:28] <xastey> gotcha.. thanks a lot [19:28] <relaxed> the man page is good reading [19:28] <xastey> yeah I read it a bit.. just didn't know were to start [19:28] <xastey> on windows it just worked [19:28] <xastey> moved to unix and fail everywere [19:29] <relaxed> It has nothing to do with the move to unix [19:29] <xastey> so why did the same file work on windows.. with the same command? [19:29] <relaxed> You're probably using different version of ffmpeg. [19:30] <xastey> a few weeks yeah [19:30] <relaxed> The same version would behave the same on linux, windows, or OS X [19:31] <xastey> N-42915-g307a20c vsN-42347-g299387e [19:31] <xastey> o well.. its fixed.. thanks a bunch [20:17] <frogs> anybody can compile latest ffmpeg? [20:18] <frogs> i get a large quantity of undefined references when linking ffmpeg (to various ff_* functions) [20:27] <frogs> i last successfully built a few days ago [21:45] <delicado> hi, after a file has reached an EOF, i have used av_seek_frame() to return to the first frame. i thought the EOF will be gone the next call to av_read_frame(), but it is still returning EOF, why is that? all i want to do is simple. i just want to loop that video file. but if i reach an EOF, i cannot play it again, so i have no choice but to re-load the file again. any idea? [23:01] <someone-noone> ./ffmpeg -i test-hd.mp4 -vcodec libvpx -r 24 -quality realtime test.webm produces video frames not in real time. How should I encode for real real time ? :) [23:02] <someone-noone> I'm getting alway ~23-22 fps [23:02] <frogs> hdtv encode in realtime? [23:03] <someone-noone> yes [23:03] <someone-noone> 720p [23:03] <frogs> you have multi core? [23:03] <someone-noone> core 2 duo [23:03] <frogs> use -threads [23:04] <someone-noone> ./ffmpeg -i test-hd.mp4 -vcodec libvpx -r 24 -quality realtime -threads 2 test.webm produces 21 fps [23:06] <sacarasc> someone-noone: Try -threads 0 [00:00] --- Mon Jul 30 2012

1 0

ffmpeg-devel.log.20120728
by burek 29 Jul '12

29 Jul '12

[00:10] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r2357f60687 10ffmpeg/libavformat/rtmpproto.c: [00:10] <CIA-41> ffmpeg: rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets [00:10] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [00:10] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * re49e6b6451 10ffmpeg/libavformat/rtmpproto.c: [00:10] <CIA-41> ffmpeg: rtmp: Allow having more unknown data at the end of a chunk size packet without failing [00:10] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [00:10] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r8ea1459bc3 10ffmpeg/libavformat/rtmpproto.c: [00:10] <CIA-41> ffmpeg: rtmp: Check the buffer length of ping packets [00:10] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [00:10] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r30b45d9c38 10ffmpeg/ (6 files in 3 dirs): x86inc: automatically insert vzeroupper for YMM functions. [00:10] <CIA-41> ffmpeg: 03Diego Biurrun 07master * r8784959234 10ffmpeg/ (configure libavcodec/Makefile): build: Rename aandct component to more descriptive aandcttables [00:10] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r02ac28229a 10ffmpeg/libavutil/eval.c: [00:10] <CIA-41> ffmpeg: eval: fix printing of NaN in eval fate test. [00:10] <CIA-41> ffmpeg: This fixes "make fate-eval" on MSVC builds. Without this, the test outputs [00:10] <CIA-41> ffmpeg: "-1.#NaN" instead of "nan" on MSVS 2010. [00:10] <CIA-41> ffmpeg: 03Mans Rullgard 07master * rb9d3c37848 10ffmpeg/ (Makefile common.mak doc/Makefile tests/Makefile): [00:10] <CIA-41> ffmpeg: build: use COMPILE template for HOSTOBJS [00:10] <CIA-41> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:11] <CIA-41> ffmpeg: lavfi: better channel layout negotiation [00:11] <CIA-41> ffmpeg: Allow substitution of channel pairs in the input for nearby channel pairs in [00:11] <CIA-41> ffmpeg: the output in order to get a closer match. Also weigh LFE channel mismatch [00:11] <CIA-41> ffmpeg: differently to favor matching the same layout without LFE over one less [00:11] <CIA-41> ffmpeg: channel with LFE. [00:11] <CIA-41> ffmpeg: 03Justin Ruggles 07master * r81f548de57 10ffmpeg/libavcodec/alac.c: [00:11] <CIA-41> ffmpeg: alac: check for truncated packets [00:11] <CIA-41> ffmpeg: This will give a clearer error message when the error is caused by a [00:11] <CIA-41> ffmpeg: truncated packet. [00:11] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r5361e10a5e 10ffmpeg/libavcodec/x86/proresdsp.asm: proresdsp: port x86 assembly to cpuflags. [00:11] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc6963a220d 10ffmpeg/: (log message trimmed) [00:12] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [00:12] <CIA-41> (10 lines omitted) [01:47] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r0fa352623d 10ffmpeg/libavcodec/dvdsubenc.c: [01:47] <CIA-41> ffmpeg: dvdsubenc: switch to av_assert [01:47] <CIA-41> ffmpeg: we use av_assert0() because the previous code unconditionally [01:47] <CIA-41> ffmpeg: enabled the assert. [01:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:47] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r55cdd45446 10ffmpeg/libavformat/avio.h: [01:47] <CIA-41> ffmpeg: avio: Document that write_packet() is not allowed to change its input [01:47] <CIA-41> ffmpeg: Found-by: Reimar [01:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:52] <cptspiff> hmm, ^^ gut reaction, wouldn't const'ing the input tell the same story and have the compiler enforce as a bonus? [01:55] <michaelni> cptspiff, yes but its public API making it const might (but of course doesnt have to) cause problems [01:55] <michaelni> the docs change was the easy way [01:56] <michaelni> if someone wants to change it and double check that it has no sideeffects thats welcome of course [03:11] <cptspiff> the extent of tests i can do right now is compile. on a 2g link, not grabbing fate over it ;) it compiles, diff old.log new.log is blank so no new warnings emitted or such. obviously being public, i cannot guarantee user code, but handwaving i'd say going from a writable to a non-writable ptr should not cause much havoc there, other than some warnings about incompat ptrs (and possibly errors in c++). don't have a tool to compare binary files here [03:13] <cptspiff> (by incompat ptrs i refer to the function ptrs) [03:26] <michaelni> cptspiff, it could error with the right compiler flags, and as said i dont mind at all if its changed i just didnt want to do it as a quick warning fix without thought [05:39] <CIA-41> ffmpeg: 03jamal 07master * r64c53fa16a 10ffmpeg/libavcodec/ (dxva2_h264.c dxva2_mpeg2.c dxva2_vc1.c): [05:39] <CIA-41> ffmpeg: dxva2: Fix some warnings about incompatible pointer type [05:39] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:39] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * ra2f91e7bcd 10ffmpeg/libavcodec/mpegaudiodec.c: [05:39] <CIA-41> ffmpeg: mpegaudiodec: switch to av_assert [05:39] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:40] <ubitux> Daemon404: in your canopus patch, i think michael wanted you to explicit the codec id with a MKBETAG [10:47] <maker> I was wondering, since the task of porting vlc filters to libavfilter has been in the todolist for contributors for a few years, has anybody ported anything since now? [10:50] <ubitux> vf_blackframe.c: * Ported from MPlayer libmpcodecs/vf_blackframe.c. [10:50] <ubitux> vf_boxblur.c: * Ported from MPlayer libmpcodecs/vf_boxblur.c. [10:50] <ubitux> vf_cropdetect.c: * Ported from MPlayer libmpcodecs/vf_cropdetect.c. [10:50] <ubitux> vf_delogo.c: * Ported from MPlayer libmpcodecs/vf_delogo.c. [10:50] <ubitux> and a few other ones [10:52] <maker> Puzzle seems not to have been ported \o/ [10:52] <saste> ubitux: from *vlc* [10:53] <maker> Well, I did git grep "Ported from" , the pattern is the same [10:53] <saste> "Turns the video in a jigsaw puzzle game" [10:54] <saste> we have a tile filter but seems pretty a different thing [10:54] <ubitux> saste: argh sorry misread. [10:55] <maker> :D [10:56] <ubitux> maker: according to Daemon404 they have a good ivtc filter that would be welcome in ffmpeg [10:56] <ubitux> but that's not trivial [10:57] <maker> ubitux: I don't think you are going to jugde students for having accomplished something, what if I create a new repo somewhere and develop on ivtc as qualification task? [10:58] <ubitux> i'm not going to judge anyone ;) [10:58] <saste> maker: that's ok of course, but for review it's easier to send a patch anyway [10:59] <ubitux> maker: you're still looking for a filter to do? [10:59] <maker> as god" said [10:59] <ubitux> (maker: are you the epita guy?) [10:59] <maker> ubitux: yay. [10:59] <maker> epita? Probably not [10:59] <ubitux> oh ok, my bad then [10:59] <maker> saste: yep, just in case there won't be enough time for submitting a patch [11:00] <ubitux> i'm wondering if porting the vf geq wouldn't be a good exercise [11:00] <ubitux> i wanted to do it for a while but didn't have the time [11:00] <ubitux> (this is from mplayer again, not vlc) [11:00] <maker> geq? uhm [11:00] <saste> ubitux: i seems to remember having started them, but there were some design issues [11:01] <saste> the way it dealt with RGB/YUV was borken iirc [11:01] <ubitux> saste: "them"? [11:01] <ubitux> vlc ones? [11:01] <saste> ubitux: them => geq [11:01] <ubitux> ok [11:02] <ubitux> iirc the main issue was how to deal with reading the values [11:03] <ubitux> but a vf geq without the read luma & chroma feature could be a first [11:04] <ubitux> well, just a suggestion :) [11:05] <maker> mulitmedia.cx is down? [11:05] <ubitux> i just saw pretty interesting geq expression a while ago, and thought that was a nice filter [11:05] <maker> :o [11:05] <ubitux> looks like so [11:05] <ubitux> fate as well :( [11:16] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rfd2b6c1e4b 10ffmpeg/doc/filters.texi: doc/filters: itemize movie examples [11:17] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r66a5201819 10ffmpeg/ (7 files in 3 dirs): lavfi: add flite audio source [11:17] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r7cf9aaddcd 10ffmpeg/libavutil/bprint.h: [11:17] <CIA-41> ffmpeg: lavu/bprint: extend/clarify documentation for av_bprint_init() size_max value [11:17] <CIA-41> ffmpeg: Also define AV_BPRINT_SIZE_* macros before av_bprint_init() declaration, [11:17] <CIA-41> ffmpeg: and mention them in the av_bprint_init() doxy. [11:17] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * r32390d6767 10ffmpeg/ (4 files in 2 dirs): [11:17] <CIA-41> ffmpeg: lavfi: move movie and amovie to a "multimedia sources" section [11:17] <CIA-41> ffmpeg: Since the recent changes, movie and amovie are able to deal with more [11:17] <CIA-41> ffmpeg: than one type of stream, so they should be categorized as "multimedia [11:17] <CIA-41> ffmpeg: sources" rather than audio/video sources. [11:49] <CIA-41> ffmpeg: 03Nicolas George 07master * r9a757cca34 10ffmpeg/libavfilter/avf_concat.c: avf_concat: fix min value for v option. [12:31] <CIA-41> ffmpeg: 03Stefano Sabatini 07master * rfb0688b079 10ffmpeg/doc/general.texi: doc/general: add missing device support information [13:32] <vitrums> Hi there. I'm looking for some to solve my problem of screencasting via rtmp (justin.tv, own3d), when I always get black screen as an output. However local recordings work just perfect. Any ideas? [13:33] <vitrums> for some help* [15:30] <CIA-41> ffmpeg: 03Nicolas George 07master * r4ce87ecf2a 10ffmpeg/ (doc/filters.texi libavfilter/asrc_flite.c): (log message trimmed) [15:30] <CIA-41> ffmpeg: asrc_flite: do not crash on multiple instances. [15:30] <CIA-41> ffmpeg: The voice register functions return the same voice structure [15:30] <CIA-41> ffmpeg: upon multiple registration. It causes us two problems: [15:30] <CIA-41> ffmpeg: If we delete a voice without deregistering it, it leaves [15:30] <CIA-41> ffmpeg: a dangling pointer inside the library. [15:30] <CIA-41> ffmpeg: If we delete or unregister a voice at uninit, it may still [15:31] <CIA-41> ffmpeg: 03Nicolas George 07master * refcfcf8793 10ffmpeg/libavfilter/avf_concat.c: [15:31] <CIA-41> ffmpeg: avf_concat: fix silence duration. [15:31] <CIA-41> ffmpeg: concat_in.pts starts at 0 for each segment, so the duration [15:31] <CIA-41> ffmpeg: of the silence was too long, except at the first stitch. [15:31] <saste> michaelni: is it OK to extend lavu/integer in order to support bigger numbers, or to make the max value configurable at init? [15:32] <saste> i need it for xface, so I don't need to duplicate big integer code [15:33] <michaelni> saste, sure [15:37] <cptspiff> michaelni: right, then i misunderstood what you meant (and you can see i struggled to figure out what you meant double check it doesnt break anything ;)) [16:34] <durandal_1707> why is fate down? [17:06] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r05a6c625f4 10ffmpeg/tests/Makefile: [17:06] <CIA-41> ffmpeg: fate: switch rsync to fate-suite.ffmpeg.org [17:06] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:06] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r157f956973 10ffmpeg/ffmpeg.c: [17:06] <CIA-41> ffmpeg: ffmpeg: add code to perform ts wrap correction [17:06] <CIA-41> ffmpeg: Fixes ticket1385 [17:06] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:52] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r3adc9c9996 10ffmpeg/ffmpeg.c: [17:53] <CIA-41> ffmpeg: ffmpeg: allow using the r_frame_rate as fundamental timebase for stream copy in avi [17:53] <CIA-41> ffmpeg: Fixes Ticket1434 [17:53] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:05] <cptspiff> hmm i think i'm starting to see why http://trac.ffmpeg.org/trac/ffmpeg/ticket/604 sticks around [18:06] <nevcairiel> my guess is mostly because it doesnt affect the ffmpeg apps, only third party apps [18:06] <cptspiff> do i read this correctly that i's pretty much impossible to solve without a the painful reopen cycle? [18:06] <cptspiff> well it's an arch issue [18:06] <cptspiff> and it really hurts in these evil 10-bit animeidiot days ;) [18:07] <nevcairiel> thinking about it, i guess the decoder really would have a hard time changing its own threads value [18:08] <cptspiff> yeah, plus the hwaccel isnt really setup before the entire chain is setup (decode_slice_header for h264 afaict) so you cannot detect it up front and hop to 1 thread. [18:08] <nevcairiel> i just disable threading when dxva is requested, and since my dxva codepath is completely seperate from the software decoding, i do a full codec re-open anyway [18:08] <cptspiff> yeah we do the same [18:08] <nevcairiel> but i guess if you want to keep it simpler, you might not have that luxury [18:08] <cptspiff> but the full codec re-open hurts on streams. [18:09] <cptspiff> (in particular volatile urls and such crap) [18:11] <nevcairiel> when i have the h264 sequence headers present i manually scan those for hw compat to try to avoid that, but in streaming situations sadly those arent always available, so there is that [18:13] <cptspiff> yup. it's the one size fits all solution that is hard. [18:14] <nevcairiel> although you need the SPS before you can start decoding, and once you have it, you know if its hw compatible [18:24] <CIA-41> ffmpeg: 03Nicolas George 07master * r0f8f924847 10ffmpeg/doc/faq.texi: faq: suggest the concat filter. [18:24] <CIA-41> ffmpeg: 03Nicolas George 07master * r2d6f442ca1 10ffmpeg/doc/faq.texi: faq: disambiguate "join". [20:04] <maker> Uhm, is the documentation present in multimedia.cx concerning libavfilter up to date? [20:05] <maker> Becouse I see some variation on the constraints, but maybe apis and logis is stil the same [20:05] <maker> *because [20:05] <maker> *logics are [20:05] <maker> Sorry for that. :| [20:08] <michaelni> maker, best look at the source code in ffmpeg git (various headers and some simple filters) as documentation [20:09] <michaelni> i dunno how upto date multimedia.cx is [20:10] <maker> There isn't any doxygen somewhere? [20:20] <michaelni> maker, http://ffmpeg.org/doxygen/trunk/index.html [20:22] <maker> ty [00:00] --- Sun Jul 29 2012

1 0

ffmpeg.log.20120728
by burek 29 Jul '12

29 Jul '12

[00:10] <soulstitchmmo> Q won't stop my streaming in FFMPEG [00:10] <soulstitchmmo> never mind, caps lock was on, that's kinda sensitive. [00:33] <insano> I'm trying to converto to RealMedia using the following command: ffmpeg -i video.mp4 -vcodec rv20 -ac 1 -ar 44100 -acodec real_144 -f rm video.rm [00:33] <insano> But the audio is horrible, how I can fix it? [01:02] <insano> I'm trying to converto to RealMedia using the following command: ffmpeg -i video.mp4 -vcodec rv20 -ac 1 -ar 44100 -acodec real_144 -f rm video.rm [01:02] <insano> But the audio is horrible, how I can fix it? [01:12] <burek> change the audio codec [01:12] <burek> or use -ab [01:54] <Jesin> well [01:54] <Jesin> ffmpeg has broken some backwards compatability, hasn't it [01:54] <Jesin> http://sprunge.us/fLhO [01:55] <Jesin> sed -e 's|libavcodec/opt.h|libavutil/opt.h|g' -i $_svnmod/src/common/ffmpeg.cpp [01:55] <Jesin> was easy enough [01:56] <Jesin> and it looks like I'll have to add -e 's|av_write_header|avformat_write_header|g' [01:56] <Jesin> (I am aware that what I am doing to this source file is horrid, yes... <_< ) [01:57] <Jesin> I haven't been able to find a similar fix for av_set_parameters though. [01:59] <Jesin> .... [01:59] Action: Jesin discovers ffmpeg-compat [01:59] <Jesin> :D [02:09] <Jesin> ...now I just have to get my pkgbuild to find this somehow [02:34] <ShadowJK> Are you testing for the version define? [03:00] <Jesin> nope [03:00] <Jesin> err, well [03:00] <Jesin> I'm trying to compile https://aur.archlinux.org/packages.php?ID=16851 [03:58] <dericed> can ffmpeg encode to AVC-Intra 100? [06:51] <Spice_Boy> is it possible to pipe the output of rtmpdump to ffserver, so that I can restream the music on the LAN in mp3? [06:51] <Spice_Boy> Currntly, this is how I play the stream rtmpdump -r "rtmp://wms2.scahw.com.au:1935/live/" -a "live/" -W "http://www.triplem.com.au/brisbane/nowplaying/mmm_FM_Streaming_Player_20120…" -p "http://www.triplem.com.au" -y "4mmm.stream" | mplayer - [09:11] <anacrolix> Hi, I'm trying to encode an SRT subtitle file into an MPEGTS stream. What is a suitable subtitle codec for this? [09:48] <cbreak> into the mpeg stream or into the video? [10:47] <jonthn> hello, I'm trying to make a video out of an audio file, and so I thought I would try to use the 'showwaves' filter but I don't understand how it should work. Can someone help me ? [10:49] <ubitux> ffmpeg -f lavfi -i amovie=input.mp3[out0],showwaves[out1] output.mp4 or something like that [10:57] <jonthn> thanks it seems to work when using -f lavfi -i "amovie=audio.ogg,asplit[out0],showwaves[out1]" [11:02] <jonthn> ubitux: do you know if it would be possible to add(merge) an image behind the wave ? maybe use a watermark filter to approximate the result ? [11:04] <ubitux> you can most likely use the overlay filter [11:04] <ubitux> (oups yes sorry forgot to split) [11:05] <jonthn> ok thanks again :) [11:08] <vitrums> hello there [11:10] <vitrums> I use Ubuntu 12.04 x64 (core i5, NVidia GeForce 560) and I've completed the FFmpeg installation procedure according to this article http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide. [11:11] <vitrums> But when I run ffmpeg, the picture is totally black. Has anyone run into this problem? [11:12] <vitrums> I've already tried streaming to own3d and justin.tv. FFmpeg is working, however there's no picture but a black screen =. [11:37] <vitrums> is anyone online? [11:46] <microchip_> no [12:17] <vitrums> I've run the best quality config to store my video locally and it worked just perfect: "ffmpeg -f x11grab -y -s 1920x1080 -r 30 -i :0.0 -ab 96k -f alsa -ac 2 -i pulse -vcodec libx264 -crf 30 -preset ultrafast -s 1920x1080 -acodec libmp3lame -ar 44100 -threads 0 ./out.avi [12:19] <vitrums> However I still get no picture when I stream online (put -f flv rtmp://MyJustinTV instead of ./out.avi). Can't figure out where exactly the problem might be [12:20] Action: sacarasc DDoS's microchip_ offline. [12:21] <microchip_> evil! [13:27] <vitrums> guys, so far I've found but only two threads with my problem to screencast via rtmp. E.g. http://ubuntuforums.org/showthread.php?t=1953527 .Though, no answer ;-( [13:50] <vitrums> Hello, Michael Niedermayer. Can I ask u about ffmpeg? I've got a rare issue with screencasting via rtmp. I really need some help, cause googling just didn't help ( [13:52] <saste> vitrums: consider filing a bug request [13:54] <saste> also: complete commandline and output on pastebin [13:54] <saste> consider simplifying your commandline, removing all the possibly unrelated options [13:54] <saste> and you may ask directly to the justin.tv crew [13:55] <vitrums> actually I've been reducing my command line to locate the exact reason of my troubles and it's just about sending an flv flow via RTMP, nothing else :/ [13:56] <vitrums> but thanks for your tips, I'll do my best [14:00] <sgfgdf> hello, guys! i hava a movie with this audio -- "Stream #0:1(eng): Audio: dts (DTS), 48000 Hz, 5.1(side), s16, 1536 kb/s (default)" which is 5.1. but it doesn't play the audio from the program i want to play it (iTunes). AFAIK i should encode it to AAC. does AAC support 5.1 and how should i re-encode the audio with ffmpeg command? [14:13] <burek> Spice_Boy why don't you get the rtmp stream directly with ffmpeg/ffserver [14:13] <Spice_Boy> burek: I'd like to if I knew how [14:13] <Spice_Boy> I tried from someone's example somewhere, but didn't seem to work [14:14] <burek> something like: ffmpeg -i rtmp://... should work [14:14] <Spice_Boy> I'll have another go now and see what happens [14:15] <Spice_Boy> hang on, ffmpeg?.... then just feed that to ffserver?... [14:15] <burek> http://ffmpeg.org/ffmpeg.html#rtmp [14:15] <Spice_Boy> can ffmpeg play the rtmp stream by itself? (to take one step at a time) [14:15] <burek> yes [14:15] <burek> ffplay can, ffmpeg no [14:15] <Spice_Boy> I'm sure I tried this [14:15] <Spice_Boy> ahhh [14:17] <Spice_Boy> trying to piece which bit of the link goes where [14:18] <burek> ubitux, why asplit is needed? [14:19] <vitrums> I'm about to post a bug report. But since there's some activity here, perhaps there might be any idea why I get the blank screen, when I stream to RTMP regardless what rtmp service I actually use... [14:20] <burek> vitrums can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [14:20] <vitrums> of cause, gladly [14:21] <ubitux> burek: you have 1 input, and you want 2 outputs (one with the sound, and the other with the waves) [14:22] <burek> oh :) makes sense :) [14:23] <burek> sgfgdf, try: ffmpeg -i input -acodec ... (audio options) output [14:24] <burek> ffmpeg -i input -acodec aac -strict experimental -ar 44100 -ac 2 -ab 128k output.aac [14:25] <Spice_Boy> http://pastebin.com/9Zcgdcss [14:26] <vitrums> I've posted the command and output at: http://pastebin.com/LbSzWD3y [14:26] <vitrums> hope it might help [14:27] <burek> did you read http://ffmpeg.org/ffmpeg.html#rtmp [14:27] <burek> Spice_Boy [14:27] <Spice_Boy> yeah [14:27] <Spice_Boy> well, tried to [14:27] <burek> no you didn't [14:27] <burek> read again [14:27] <Spice_Boy> it's not the easiest thing to read [14:28] <vitrums> The issue doesn't have anything to do with grabbing my screen or anything of that kind. I've tested streaming a video file with -re flag, and it also works, but the video is blank [14:28] <burek> it says: rtmp_live [14:28] <burek> Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is any, which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are live and recorded. [14:28] <burek> ffplay rtmp://wms2.scahw.com.au:1935/live?rtmp_live=1 [14:30] <burek> vitrums, change -b to -b:v [14:30] <burek> with -b you are setting both audio/video bitrate [14:31] <Spice_Boy> if I put the /live in as that 'manual' page says for app, it complains that argument /live is provided as input filename [14:31] <burek> vitrums, your output also tells you that "Please use -b:a or -b:v, -b is ambiguous" [14:31] <burek> and you can see there is no audio in your output list of streams, so that's the issue [14:32] <vitrums> ah, I don't think these options like -b -bitrate are necessary. I've just tried -b:v and it gives the same result. It's better to remove these redundant options at all [14:32] <Spice_Boy> burek: if I do actually figure this out for ffplay, would that then be the same command to use in ffmpeg to feed to ffserver? [14:32] <burek> use -b:a and -b:v [14:32] <burek> Spice_Boy yes [14:32] <burek> ffplay input could be used as ffmpeg -i input [14:33] <Spice_Boy> cool, I'll see if I can find someone's example somewhere [14:33] <burek> ok [14:33] <vitrums> I intentionally removed any audio to avoid any confusion with audio and so on [14:34] <Spice_Boy> actually, instead of that, can I feed the working output of rtmpdump to ffserver? [14:34] <Spice_Boy> as I currently feed it to mplayer [14:35] <burek> vitrums, well your command line provided uses audio [14:35] <burek> so, I can't read your mind [14:35] <burek> remove all the audio options then and use -an [14:36] <burek> Spice_Boy, yes but only if rtmpdump supports ffm format [14:36] <burek> which is ffmpeg specific [14:36] <Spice_Boy> hmmm [14:36] <vitrums> -b:v doesn't resolve the case. I'll post you my original command, where I also use audio, so you won't be confused with it [14:36] Action: Spice_Boy is going 'round in circles [14:36] <burek> so, I'm pretty sure no [14:37] <burek> Spice_Boy, try: rtmpdump ... | ffmpeg -f flv -i - -c copy http://localhost:8090/feed1.ffm [14:37] <burek> (start ffserver before that command) [14:37] <Spice_Boy> ah, I'll give that a go :) [14:38] <burek> it's much easier to try to run all with ffmpeg and just feed it to ffserver [14:38] <Spice_Boy> haha, that seemed to work :) [14:38] <burek> ok :) [14:38] <Spice_Boy> in my book, if it works, then it's good [14:38] <burek> :) [14:38] <Spice_Boy> it would be nice to keep it all in ffserver, but I just can't figure it out [14:40] <Spice_Boy> hmm... maybe didn't quite work... I could see the ffmpeg working away nicely though, so I figure that's a start [14:41] <burek> ffplay "rtmp://myserver/live/mystream live=1" [14:41] <burek> try like that [14:42] <Spice_Boy> umm... my ffserver.conf has mp3 as the codec [14:42] <Spice_Boy> which I thought would convert it [14:43] <vitrums> here is my updated command and logs. I've tried to remove any redundant parameters this time: http://pastebin.com/vMAUNr1H [14:45] <burek> vitrums, the output seems ok [14:48] <Spice_Boy> burek: when trying to play that with mplayer, the cache starts filling, but stops at 1.16%... then says cache empty [14:48] <burek> I don't use mplayer :( [14:48] <Spice_Boy> vlc dies too [14:48] <burek> pastebin the log? (-vvv) [14:49] <Spice_Boy> log of... which program? [14:49] <burek> vlc? :) [14:49] <Spice_Boy> vlc is on windows though :( [14:50] <vitrums> burek, might it be some firewall issue or anything of that kind? I can see the stream online, when I have FFmpeg running, but the video itself is blank :=/ [14:51] <burek> Spice_Boy, in vlc, Tools - Messages, select verbosity to 2 and then try to play [14:52] <Spice_Boy> ah, ok [14:52] <burek> vitrums, did you try replacing output "-f flv rtmp://..." with "out.flv" [14:53] <Spice_Boy> burek: http://pastebin.com/QC5Zq4Q7 [14:53] <vitrums> yeap, I'm telling you, it's just some RTMP issue, nothing else. I've got a perfect video, if I store it locally in some file [14:54] <burek> vitrums, try now: ffmpeg -i out.flv -c copy -f flv rtmp://... [14:54] <burek> or better [14:54] <burek> ffmpeg -re -i out.flv -c copy -f flv rtmp://... [14:54] <Spice_Boy> burek: if I could get the right command for ffmpeg (or ffplay) to play the rtmp stream, that would probably be the best step forward [14:56] <vitrums> burek, it gives me the same result: lifecasting with blank image [14:58] <Spice_Boy> burek: I have these details for the media stream. Can you tell me how that's constructed as an ffserver, ffmpeg, or ffplay command? [14:58] <Spice_Boy> http://pastebin.com/K8XBDfft [15:00] <vitrums> burek, here's a topic about the same issue http://ubuntuforums.org/showthread.php?t=1703385. Though, I couldn't find any other topic, reporting this trouble across the internet neither any solutions. [15:01] <burek> vitrums, try changing rtmp hosting providers [15:01] <burek> use justin or something else [15:01] <burek> just to see if that's an issue [15:02] <burek> if it's the same with all providers, then I guess it's a bug in ffmpeg's rtmp support [15:02] <vitrums> burek, I did. You can actually see it in my very first http://pastebin.com/LbSzWD3y post [15:02] <burek> well, then file a bug report, if you want [15:02] <vitrums> guess I will [15:03] <sgfgdf> burek, thanks for your answer. (sorry for the delay, i was away). i did not understand if AAC support more than 2 channels though. [15:03] <burek> Spice_Boy I don't personally use rtmp, so no, I don't know :) [15:03] <vitrums> thank you for passionate collaboration, dude [15:03] <burek> vitrums :beer: :) [15:03] <burek> sgfgdf, you can try with -ac 6 [15:03] <burek> and if it works, even better [15:03] <Spice_Boy> I just discovered piping the otuput of rtmpdump | ffmplay -i - plays it [15:04] <burek> Spice_Boy google for "ffmpeg rtmp live" [15:04] <burek> you should find plenty of examples [15:04] <Spice_Boy> yeah, been searching similar [15:04] <Spice_Boy> but if piping it to ffplay works, then to ffmpeg might too [15:05] <sgfgdf> burek, but if i don't mention -vcodec copy will it stay as it is, or probably better to explicit mention it? [15:05] <burek> well, ok [15:05] <Spice_Boy> just going to try [15:05] <burek> sgfgdf, if you have a video stream which you would like to preserve, then -vcodec copy is a must [15:06] <burek> otherwise it's going to be re-encoded in something like mpeg or something [15:08] <sgfgdf> burek, thanks for your help! -- http://1adt.com/wp-content/uploads/2012/02/creative-corona-extra-print-ads-… (i don't know if you like such beer) :) [15:09] <burek> mmmm :> [15:09] <burek> there is a saying in our language "for a good pig, there are no bad swills" :) [15:10] <burek> meaning, every cold beer is what I like :) [15:11] <Spice_Boy> :( [15:12] <Spice_Boy> if I pipe the output to ffmpeg http://127.0.0.1:8090/audio.ffm it says it's missing an audio stream [15:12] <Spice_Boy> which I 'think' is because at the instant of startup, it hasn't got it feeding to it yet [16:30] <Macey_Home> Hi All, trying to encorperate libavformat/avcodec into an app, Could anyone point me in the right direction of doing an audio codec conversion from a stream documentation please? [19:29] <vitrums> Good day everybody. I'm running into an issue, when I stream via rtmp to justin.tv two different flv videos: both have audio, but only one has video - the second one just a black sceeen. Any ideas? [19:44] <vitrums> seems like it's in a wrong video format. I can download the broadcasted video and playback it locally without problems. However this file is shown without video, when I play it remotely from justin.tv archive [19:48] <juanmabc> it is awesome, how mplayer at fullscreen consumes 13-15% counting pulseaudio [19:48] <juanmabc> i guess mmx,sse,... are underused on a lot of projects, not in that [19:56] <iive> we recommend to avoid pulse audio unless you really need it. [19:57] <juanmabc> it's default on fedora, but yep, it adds a 3% [19:58] <juanmabc> pulse seems very softw intensinve instead of hardw intensive [20:34] <Peace-> hey guys i have compiled ffmpeg but i have a strange problem [20:34] <Peace-> my script works with ffmpeg provided by libav stuffers [20:35] <Peace-> instead with ffmpeg it exits in a loop [20:36] <Mavrik> um [20:36] <Mavrik> I don't understand... what for loop? [20:37] <Peace-> while read line ; do [20:37] <Peace-> [20:37] <Peace-> /home/shared/ffmpeg-04-04-12/ffmpeg -i "$line" -y "${line%.*}.$outformat" [20:37] <Peace-> # v [20:37] <Peace-> done< "$listoffile" [20:40] <vitrums> I just don't get it... What's so wrong with my .flv screen recordings, that when they are played via justin.tv it gives a blank picture, though with audio. But if I play them locally with Totem Movie Player, the video is fine. [20:41] <Mavrik> vitrums: well, do a ffprobe on them and pastebin the output [20:43] <vitrums> Mavrik, here you go: http://pastebin.com/CZL8t4iU. Hope it would reveal the mystery =/ [20:43] <Mavrik> vitrums: my first guess would be [20:44] <Mavrik> just.tvs transcoder (or player) can't handle YUV444 [20:44] <Mavrik> which isn't really a widely supported pixel format [20:45] <vitrums> is it managable via ffmpeg options? [20:45] <Mavrik> hmm, it should be yuv420p by default [20:45] <vitrums> so you're telling there's something wrong with my default settings? [20:46] <Mavrik> nah, just probably your input device grabs video in yuv444 format [20:46] <Mavrik> vitrums: try -pix_fmt yuv420p [20:46] <Mavrik> as a video encode parameter [20:47] <vitrums> sure, already doing it. Hope you were right [20:48] <Mavrik> also the resolution is kinda high, does justin.tv support 1080p? [20:49] <vitrums> Mavrik, just tell me your paypal... that's not funny at all ( I've killed my weekend day actually to solve it. So, notwithstanding the fact I'm happy to finally get it work, this literally hurts me ;-) [20:50] <Mavrik> ^^ [20:50] <Mavrik> glad to help :D [20:52] <Peace-> solved :D [20:57] <vitrums> may one close the bug branch: https://ffmpeg.org/trac/ffmpeg/ticket/1581 ? [22:58] <AlRazi_> i'm trying to configure ffmpeg with libaacplus .. i'm getting -> ERROR: libaacplus >= 2.0.0 not found [23:00] <AlRazi_> i'm on mac 10.7 [23:01] <AlRazi_> ./configure --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-libmp3lame --enable-libaacplus [23:07] <AlRazi_> do i need to install it seperately ? or is it shipped with the ffmpeg source ? [23:55] <judget> hey duh Im building a new ffmpeg on natty and during config i get a message about upgrading VA-API >- 0.32 [23:55] <judget> where can I get those directions please? [23:55] <iive> AlRazi_: yes it is separate library. [23:56] <iive> same for mp3lame. [00:00] --- Sun Jul 29 2012

1 0

ffmpeg-devel.log.20120727
by burek 28 Jul '12

28 Jul '12

[00:06] <burek> we need some beer factory to be hour donator/sponsor or something :) [00:06] <Daemon404> bikeshed beer [00:09] <burek> hour... omg... [00:09] <burek> our* [00:10] <burek> i think its time to sleep :D [00:21] <orbisvicis> is there an example of using av_audio_convert? particularly, how to create the output buffer [00:21] <orbisvicis> (without AVCODEC_MAX_AUDIO_FRAME_SIZE) [00:22] <burek> did you check ffmpeg's source code [00:29] <orbisvicis> burek: what in particular? [00:29] <burek> grep av_audio_convert |? [00:30] <orbisvicis> i mean there is the api documentation (not helpful for me in this case), but nothing in the examples [00:31] <burek> the best api docs is the source code itself [00:31] <burek> see how does ffmpeg use it [00:31] <burek> and copy/paste the logic [00:31] <orbisvicis> ok, ill take a look [00:36] <saste> orbisvicis: av_audio_convert is deprecated, you're supposed to use the new libswresample [00:37] <saste> orbisvicis: and yes some resampling/converting examples in doc/examples may help [01:26] <maker> saste: ! [01:26] <saste> maker: hi! [01:27] <saste> maker: just see your application [01:27] <maker> Considering that submissions for the summer of code in spece are going to end today, am I still in time for developing the application requirements? [01:27] <saste> note: the task will *not* require to finish all the listed items [01:27] <saste> that's only a list of ideas [01:28] <saste> maker: there is time for evaluation until july 31 [01:28] <saste> so until the next tuesday [01:29] <saste> so yes i suppose there is still time for the qualification task [01:29] <maker> I am confident that once understood libavfilter's descriptors logic most of the time will be spent on algorithmics. [01:29] <saste> qualification task doesn't need to be *completed*, it's just to give evaluation elements [01:29] <saste> of course the better you do, the more chances you have to be selected [01:30] <saste> maker: yes, keep in mind that libavfilter is quite complex, so it's quite difficult to get familiar with its API [01:30] <saste> luckily we have lots of examples [01:31] <maker> Is socis popular? I don't think is a private data, so I was wondering, as ffmpeg organization, did you receive many applications? [01:31] <saste> maker: you will know tomorrow ;-) [01:32] <maker> Which is today ;D [01:32] <saste> we have received three applications so far [01:33] <saste> maker: do you have an idea for a qualification task? [01:34] <maker> Nope, I was just googling for ideas/inspiration [01:35] <saste> maker: did you have some experience with VLC? maybe you could port some filter from there [01:41] <maker> saste: my experience with vlc steps just through the player's api, not filters. [01:58] <saste> great, James Ashton just replied to me [01:58] <saste> regarding a library he wrote more than twenty years ago o_O [02:56] <highgod> Hi,all.I want to ask a question.I read ffmpeg code,but I didn't find any CODA code to optimize ffmpeg.I want to ask why CUDA is unsuccessful in ffmpeg?Thanks [02:56] <highgod> Sorry,CUDA [03:02] <Skyler> What precisely are you looking to "optimize"? ffmpeg is a large application. [03:05] <highgod> video filter,such as scale [03:07] <Skyler> There's been some effort towards this in the Handbrake project, which AMD/Multicoreware contributed some GPU-accelerated code to [03:14] <highgod> Yes,they use opencl. [03:18] <highgod> So I want to ask why ffmpeg don't use it to optimize? [03:18] <Skyler> Because nobody has submitted a patch to do it [03:30] <highgod> Hehe.Thanks.our team want to add opencl code to optimize ffmpeg.will you accept it? [03:32] <Skyler> Michael Niedermayer would be the one to "accept" something, but nobody can "accept" something before you even write the code. A patch gets accepted if it passes code reviews and developers believe it should be included. [03:32] <Skyler> If you want to discuss the idea of adding it, start an RFC on the mailing list. [03:34] <Skyler> Going and developing a large patch independently and dumping it on the mailing list without discussion almost invariably leads to abandoned code. [03:41] <highgod> OK,Thanks.We have write some code.Witch developers should we send the code to? [03:49] <Skyler> You need to participate on the ffmpeg-devel mailing list. [03:50] <oneal> there is one pipeline path(software pipeline) in currently ffmpeg(decoder(demuxer --> single or multi thread)-->filters processing-->encoder(single or multi threads) -->muxer), in our optimizing plan, there is another pipeline path(acclerate pipeline). it's easy for use to switch between software pipeline and accelerating pipelin. this can make more flexibility. [03:54] <oneal> Are there any other suggestions or ideas about this. [03:55] <oneal> there are some algorithm which are suitable for running on CPU, but there are a lot of algorithm which can be suitable for running on GPU. [04:00] <oneal> according our profiling about the currente video transcode procedure, there are some block in the transcode pipeline. so I think if the computation of some stages in transcode pipeline are moved to other computing device. this can speed up the whole pipeline. [06:46] <philipl> ubitux: I've updated the mov_text branch and posted a new diff. [06:47] <philipl> ubitux: Also, I looked at mplayer, and I know why you saw weird timing. it's handling of packet duration is messed up. [06:47] <philipl> It doesn't read duration properly, and then it doesn't set the AVPacket dts or duration properly (no timebase conversion, etc). [06:47] <philipl> The end result is that it falls back to its 3 second default duration. [06:48] <philipl> Most people don't go down this path as they're decoding srt or ass where the inline timing information is used by ffmpeg and the broken AVPacket values ignored. [06:53] <philipl> ubitux: Ok, got it. So there are two problems. 1) It only looks at convergence_duration for subtitle duration. *sigh* 2) If I try and switch it from using its own minimal mov_text decoder to my one, the mplayer code doesn't get the timebase conversion right, so all times and durations end up rounding to zero. [12:02] <durandal_1707> michaelni: wavpack (demuxer) spits bogus error after addition of picture support in ape tags [13:15] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * ra2cd13f04f 10ffmpeg/libavcodec/zmbv.c: [13:15] <CIA-41> ffmpeg: zmbv: Fix warning about discarded qualifier [13:15] <CIA-41> ffmpeg: Based on patch by: jamal <jamrial(a)gmail.com> [13:15] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:15] <CIA-41> ffmpeg: 03jamal 07master * rc49e0d2cdd 10ffmpeg/libavformat/aviobuf.c: [13:15] <CIA-41> ffmpeg: aviobuf: Fix warning about discarded qualifier [13:15] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:15] <CIA-41> ffmpeg: 03jamal 07master * rcb40d36074 10ffmpeg/libavcodec/ffv1.c: [13:15] <CIA-41> ffmpeg: ffv1: Fix warnings about incompatible pointer type and discarded qualifiers [13:15] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:47] <CIA-41> ffmpeg: 03Marton Balint 07master * r3be02afb56 10ffmpeg/libavformat/mxfdec.c: [13:47] <CIA-41> ffmpeg: mxfdec: fix off by one error in d10 aes3 decoding [13:47] <CIA-41> ffmpeg: Without this fix the last sample was missing from the packet. [13:47] <CIA-41> ffmpeg: Signed-off-by: Marton Balint <cus(a)passwd.hu> [13:47] <CIA-41> ffmpeg: Reviewed-by: Matthieu Bouron <matthieu.bouron(a)gmail.com> [13:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:47] <CIA-41> ffmpeg: 03Matthieu Bouron 07master * r789f8cb03a 10ffmpeg/ (4 files in 2 dirs): [13:47] <CIA-41> ffmpeg: avutil: support 50 and 60 frame rates in timecode api [13:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [14:00] <CIA-41> ffmpeg: 03Steven Robertson 07master * rb3841db133 10ffmpeg/libavfilter/vf_alphamerge.c: [14:00] <CIA-41> ffmpeg: vf_alphamerge: Fix reversed conditional [14:00] <CIA-41> ffmpeg: Reviewed-by: Nicolas George [14:34] <ubitux> wow. [14:34] <ubitux> another huuuuuge patch from mips [14:59] <nevcairiel> i wonder if the ac3 fixed point decoder couldnt be made generally useful, wonder what makes it depend on mips [15:00] <nevcairiel> its also probably plenty of code duplication [16:04] <iive> nevcairiel: afaik ac3 decoders are compiled from common template. Probably mips doesn't have fpu so float point emulation makes it horribly slow. [16:05] <nevcairiel> mips isnt the only system with slow floats, so the question remains why not make it a generic option [16:06] <iive> ^_^ [17:48] <Daemon404> hey nevcairiel, does lav have its own channel? [17:57] <nevcairiel> no [17:58] <nevcairiel> i'm not really interested in a place where users can bother me all day :P [18:01] <Daemon404> nevcairiel, o [18:01] <Daemon404> lulz [18:01] <Daemon404> also: holy crapsticks @ jamal's patches [18:02] <Daemon404> a savior has appeared [18:02] <Daemon404> at least, someone willing to do it. [19:50] <CIA-41> ffmpeg: 03jamal 07master * r52a62f9085 10ffmpeg/libavcodec/x86/dwt.c: [19:50] <CIA-41> ffmpeg: dwt: Fix several warnings about incompatible pointer type [19:50] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:43] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * r4f2549e3d0 10ffmpeg/libavformat/Makefile: Fix aiff demuxer dependencies. [21:51] <slackyman> Hi [21:51] <slackyman> one simple question [21:51] <slackyman> what does the "enable-sram" option means? [21:52] <Compn> good question [21:53] <slackyman> obviously, I guess that it allows ffmpeg for using sram [21:54] <Compn> sure [21:54] <slackyman> but what's the point? FFmpeg can be more fast? Can it use sram cpu cache? [21:55] <slackyman> or whatever? [21:55] <slackyman> does it increase performances? [21:56] <Compn> sram sounds like secure ram to me [21:56] <iive> seems to be blackfin related. [21:57] <slackyman> thanks iive [21:57] <iive> it defines L1CODE macro, most probably level 1 cache. static ram. [21:57] <slackyman> so if I want to compile ffmpeg for linux workstations and pcs or cross-compile for linux I don't need, right? [21:58] <iive> correct, no effect on x86 architectures. [21:58] <slackyman> thanks, it's what I needed to know [21:59] <iive> and even in bfin it seems to be used only for swscale. [22:00] <slackyman> so, maybe my point of view it's a little hard, why ffmpeg doesn't refuse to compile with that option enabled when compiling for x86 archs? [22:01] <slackyman> I've just tried and it configure both for linux and windows without a word about that option [22:02] <slackyman> maybe something like "warning: you're using --enable-sram for non compliant arch!" can be of some help [22:03] <slackyman> but never mind [22:03] <iive> well, at lest the help could mention what it is about. [22:04] <slackyman> the help say that it enable on-chip sram [22:04] <slackyman> but even pcs can have an on-chip sram (e.g. the cache sram) [22:05] <iive> I personally find the configure horribly obfuscated and full of too clever hacks. sometimes to the point of becoming too hard to do actual correct checks. [22:06] <slackyman> I don't except for some options [22:06] <slackyman> something is really obscure [22:06] <slackyman> but the most are clear [22:07] <slackyman> someone here never tried to cross compile Opencv? [22:07] <slackyman> for windows, I meant [22:08] <slackyman> I succeded but then get lots of errors including it in ffmpeg [22:08] <slackyman> no matter how I use LDFLAGS, CFLAGS and all the hell of options [22:09] <slackyman> Opencv is great but it's a pain to cross-compile [22:09] <slackyman> and they support only windows native compiling via cygwin and mingw [22:09] <slackyman> My project is to build a windows and Linux huge-monolithic-static build of ffmpeg.current [22:10] <slackyman> so that no *.so and *.dll are required [22:10] <slackyman> but I give up in using opencv [22:11] <slackyman> ok, I have to go [22:11] <slackyman> bye and thanks [22:11] <iive> you are welcome. [22:44] <CIA-41> ffmpeg: 03jamal 07master * r94c3e11a6f 10ffmpeg/libavutil/imgutils.c: [22:44] <CIA-41> ffmpeg: imgutils: Fix warnings about incompatible pointer type and discarded qualifiers [22:44] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Sat Jul 28 2012

1 0

ffmpeg.log.20120727
by burek 28 Jul '12

28 Jul '12

[01:02] <felixge> ffmpeg allows me to extract individual pngs from a video using: '%4d.png' as the output file name [01:02] <felixge> is there a way to get this data written to stdout, rather than into those files? [01:02] <felixge> that is: I want individual pngs as an output stream (jpeg would be ok too) [01:03] <sacarasc> Maybe "-f png -" as the end of your command. [01:04] <felixge> sacarasc: [NULL @ 0x7f92b1844600] Requested output format 'png' is not a suitable output format : / [01:07] <eduardosouza> Hey guys I am trying to convert my video files into something that I can play on my website. But it seems like something is wrong with the conversion. Is there a quick way of just converting files into lets say mp4? [01:13] <d4forze> mp4 is a container [01:16] <perrth> wish that ffmpeg picked by gstreamer :( why libav?,fromn user POV ffmpeg has more features.. confusing [01:18] <perrth> and I like the openal and modplug stuff here, missed in libav [01:35] <eduardosouza> ok I got it but my videos are coming out without auto [01:35] <eduardosouza> suggestions?? [01:35] <d4forze> without auto? [01:35] <eduardosouza> sorry [01:36] <eduardosouza> audio [04:44] <randomguy123> Hi FFMPEG, I use a video server and I want to make ffmpeg talk directly to it. I have api lib and I can connect and READ raw yuv frames or WRITE yuv frames to it. Can you recommend where to look for clues? [04:45] <randomguy123> At first, I wanted to look for some device in implementation in avdevice lib [04:46] <randomguy123> but I also think it's possible to add virtual container format that stores raw yuv frames, but since the server is streamable then my container can't know the size etc. So, what is recommneded path I should take? [10:58] <roxlu_> hi guys, the pts and dts are those properties of a container format? or codec? [11:23] <vhdirk> hi all [11:24] <vhdirk> I recorded a video with some camera, and by default it records with a resolution of 960x720, PAR=4:3, DAR=16:9 [11:25] <Mavrik> aaaand? :) [11:25] <vhdirk> now, I'm looking for a way to convert this to 1280x720, PAR=1:1, DAR 16:9 [11:25] <vhdirk> this basically means splitting al rectangular pixels in half [11:26] <Mavrik> vhdirk: usually I do this chain: [11:26] <Mavrik> scale=iw*sar:ih,setsar=1:1,scale=1280x720 [11:26] <Mavrik> to convert to square pixels [11:28] <Mavrik> (the last scale is obviously scale=1280:720) [11:29] <vhdirk> Mavrik, thanks, but that does not seem to work [11:29] <Mavrik> well. [11:30] <vhdirk> here you go :) http://pastebin.com/1Ci934eu [11:32] <Mavrik> first of all, you're not even using ffmpeg [11:32] <Mavrik> vhdirk: and of course this won't work [11:32] <Mavrik> since you're just copying the video stream [11:32] <Mavrik> not reencoding it [11:33] <vhdirk> aha, that makes sense :) [11:36] <Mavrik> vhdirk: to change PAR you need to reencode video :) [11:36] <Mavrik> so choose an output codec/format and reencode it [11:36] <Mavrik> and then the video filter will work as well :) [11:38] <vhdirk> I thought -vcodec copy would just copy the codec parameters. silly me, it's still early here [11:41] <Mavrik> ^^ [11:41] <vhdirk> Mavrik, thanks that actually works :) [11:42] <Mavrik> np ;) [12:09] <roxlu_> I'm reading up on how dts/pts work with codecs and muxers. I found a post which says: "a timebase in ffmpeg is rational. So if your timebase is 100/2997, you will have 0,1,2,3,4...etc, when it is 1/2997, you will have 0,100,200,300,etc..". Can someone explain what is meant with a rational time base? (Maybe it's a language barrier I'm facing here) [12:20] <Mavrik> roxlu_: it just means it's a rational number [12:20] <Mavrik> roxlu_: http://en.wikipedia.org/wiki/Rational_number :) [12:22] <roxlu_> Mavrik: hehe thanks, that's what I got :) but. I would say that: 1/1000 results in smaller numbers then 100/1000 [12:22] <Mavrik> but that wouldn't mean the same [12:22] <Mavrik> the numbers would still be (And would have to be) rational ;) [12:23] <roxlu_> hehe ok I'm feeling kind of dumb now :) [12:23] <roxlu_> what do you mean with "number would still be rational", what numbers are you referring too? [12:24] <Mavrik> the timebase numbers ;) [12:24] <roxlu_> are the 1,2,3,4 etc.. and 0,100,200, the timebase numbers? (or numerators) [12:25] <Mavrik> no, 1/1000, 100/2997 etc. are :) [12:27] <Mavrik> 1,2,3,4; 0,100,200 would be timestamps in your context [12:27] <roxlu_> ah.. I think i'm almost there to understand this :) [12:29] <roxlu_> so because the timebase is 1/2997, the timestamps need to be bigger (0,100,200). as ((1/2997) * 100) == ((100/2997) * 1) [12:30] <Mavrik> roxlu_: timebase just tells you how much time passes between two timestamps (between 1 and 2) for examplee [12:30] <Mavrik> *example [12:30] <Mavrik> 1/2997 means that from 1 to 2 exactly 1/2997 second passes [12:30] <Mavrik> 100/2997 means that from 1 to 2 exactly 100/2997 of a second passes [12:30] <Mavrik> everything else is high-school math :D [12:31] <roxlu_> Mavrik: these timestamp / timebase are related to ffmpeg/encoding/decoding. I'm tryig to understand how it all relates. [12:31] <Mavrik> yeah [12:32] <Mavrik> "timebase" is a basic unit of time [12:32] <Mavrik> and timestamps on decoded (and encoded) packets are expressed in the units of timebase (not seconds) [12:33] <roxlu_> Mavrik: so the timestamp value is just an integer which goes from 0,1,2,3,4,5.... [12:33] <roxlu_> and the meaning of this number relates to the timebase [12:33] <Mavrik> yeah [12:34] <roxlu_> so the actually time in seconds is: timestamp * timebase ? [12:34] <Mavrik> if your timebase would be equal to 1 (1/1) [12:34] <Mavrik> 0, 1, 2, 3, 4, 5 [12:34] <roxlu_> yep, cool! thanks a lot [12:34] <Mavrik> would mean that frames are shown at 00:00:00, 00:00:01, 00:00:02... [12:34] <roxlu_> awesome! [12:34] <Mavrik> if timebase is equal to 0.5 (1/2) [12:34] <Mavrik> those same frames with those same timestamps would be shown at 00:00:00, 00:00:00.5, 00:00:01... t [12:35] <roxlu_> ok. I understand now! [12:35] <roxlu_> And then there is also something related with timestamps for muxers and codecs [12:36] <roxlu_> as a dts/pts for a muxer relates to both video and audio, am I right? [12:37] <Mavrik> roxlu_: yeah [12:38] <Mavrik> muxer only sees streams [12:38] <roxlu_> Mavrik: those are the AVStream* ? [12:38] <Mavrik> mhm [12:48] <roxlu_> or Mavrik what did you mean with stream? (like a stream of bytes?) [12:50] <Mavrik> container has several streams (which can be audio, video, data, etc.) [12:50] <Mavrik> muxer muxes those streams and produces the container with all data [12:50] <roxlu_> yes, ok. [12:51] <roxlu_> I was thinking how these timestamps are used when i.e. demuxing [12:51] <roxlu_> because, when looking at the avcodec functions, there are separate functions for encoding video and audio. Though the timestamps should match some how [12:53] <Mavrik> not really [12:53] <Mavrik> :) [12:53] <Mavrik> you check timebase on video stream, you show frame when it tells you to [12:53] <Mavrik> you check timebase on audio stream, you play frame when it tells you to :) [12:54] <roxlu_> ah ok [12:56] <roxlu_> Ok so when I'm encoding in live video and audio, the I actually need to make sure to set the pts correctly. [12:56] <roxlu_> based on the current time and scaled to the time_base of my video and for the audio to the timebase of the audio [14:43] <Tuxist> hi [14:43] <bennu> Haha, hi. =D [14:44] <Tuxist> I try to convert pictures to vp8 video but i don't which api i need for this [14:45] <Tuxist> i have already creat a index with the pictueres that i need [14:50] <bennu> Sorry man, I probably don't know enough to help you. However, I can do my best. [14:50] <bennu> You installed ffmpeg with libvpx enabled, correct? [15:04] <Tuxist> i have problems to understand things in the c api [15:37] <Element9> I'm trying to capture a video of the screen with sound and I can't get a good result - frame rate isn't that good and sound is chopy. I'm trying to record in 1600x900 and I have dual core cpu on 2.1ghz. Is that not good enough or I can't tweek the parameters right? [16:04] <balachmar> I want to put my video into another container, without transcoding it. Somehow, my TV doesn't like the file through DLNA. Even though it is a straight digital tv dump. I am trying to use -map to select the streams, because ffmpeg doesn't like one of the subtitle tracks. [16:05] <balachmar> I have a command like this: ffmpeg -i inputFile.mkv -map 0:0 -vcodec copy -map 0:1 -acodec copy output.mp4 [16:05] <balachmar> But I get the following error: [16:05] <balachmar> [mp4 @ 0x283f120] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 6 >= 4 [16:05] <balachmar> av_interleaved_write_frame(): Invalid argument [16:07] <balachmar> file information is pasted here: http://pastebin.com/hHh4F1q3 [16:13] <balachmar> I just figured out that I don't need the mapping, I can just use -sn to get the same result (error) [18:30] <delicado> hi im using libavcodec, what function should i use to replay a video after i reached an EOF? i mean i just want to loop the video. [18:32] <saste> delicado: you need to seek back, check libavfilter/src_movie.c for an example [18:33] <delicado> ill look into it. thanks. [18:41] <burek> Element9, you have a tutorial on wiki [18:42] <burek> balachmar can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [19:20] <bwalker898> Hi, I'm using ffmpeg to stream a flv video via rtmp to red5. It works, but for some reason ffmpeg runs very quickly and then finishes, so I can only watch about 3 seconds of video on my website. Does anyone know why ffmpeg quits so quickly? [19:36] <bwalker898> figured it out. apparently i needed to use ffmpeg -re to stop ffmpeg from sending the data as fast as possible [19:43] <soulstitchmmo> Good Morning. I'm trying to stream to Twitch, and I keep getting the following error: Unrecognized option 'preset' [19:43] <soulstitchmmo> Failed to set value 'fast' for option 'preset' [19:45] <relaxed> soulstitchmmo: command? [19:45] <soulstitchmmo> fmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0+0,0 \ [19:45] <soulstitchmmo> -f alsa -ac 2 -i pulse -vcodec libx264 -preset "$QUAL" -s "$OUTRES" \ [19:45] <soulstitchmmo> -acodec libmp3lame -ab 128k -ar 44100 -threads 0 \ [19:45] <soulstitchmmo> -f flv "rtmp://live.justin.tv/app/$STREAM_KEY" [19:46] <relaxed> try -preset:v fast [19:47] <bwalker898> does anyone know what the ffmpeg command would be to stream from a webcam? [19:47] <soulstitchmmo> Unrecognized option 'preset:v' [19:47] <soulstitchmmo> Failed to set value 'fast' for option 'preset:v' [19:47] <relaxed> ffmpeg version? [19:48] <soulstitchmmo> no idea. How do I find that out? [19:48] <relaxed> run ffmpeg and look at the top [19:49] <soulstitchmmo> ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers [19:50] <johnbender> soulstitchmmo: you and I sir are doing the exact same thing :D [19:50] <soulstitchmmo> lol [19:51] <johnbender> soulstitchmmo: https://gist.github.com/3f75d86183cea0fd40d7 [19:51] <johnbender> superfast works for me [19:51] <johnbender> though I'm on oneiric not precise [19:51] <soulstitchmmo> what is that link? [19:51] <johnbender> soulstitchmmo: its a pastebin [19:51] <johnbender> github [19:52] <johnbender> can anyone help me get my pulse audio sync'd up with a video overlay? [19:52] <soulstitchmmo> ok so... what does it do? Sorry I'm trying out linux gaming and I'd like to stream it. [19:52] <johnbender> is that possible [19:52] <soulstitchmmo> oh that's just your function. [19:52] <johnbender> soulstitchmmo: that will stream to justin tv [19:53] <johnbender> soulstitchmmo: I have it in ~/.bin/ [19:53] <johnbender> which is in my path [19:53] <johnbender> so I use it like: bash ~/.bin/stream-jtv `cat ~/.jtv-key` [19:54] <johnbender> I put my justin.tv streaming key in ~/.jtv-key so I don't spew it all over the internet [19:54] <johnbender> and that command just drops it in as the first argument the snippet I gave you [19:54] <soulstitchmmo> well, I don't have any errors, but my stream isn't live :p [19:54] <johnbender> soulstitchmmo: you'll probably want to tweak the resultion of the desktop capture [19:55] <johnbender> and I'm still trying to get the webcam overlay sync'd with the audio [19:55] <johnbender> soulstitchmmo: to test it out, you can replace the last line "-f flv ..." with "~/foo.mp4" [19:55] <johnbender> then view it with `totem ~/foo.mp4` [19:56] <burek> soulstitchmmo you are not using ffmpeg [19:56] <burek> you're using a fork of ffmpeg [19:56] <burek> so you might go to their support channel and ask there probably? [19:57] <johnbender> burek: where is that, #libav? [19:57] <burek> dunno [19:57] <burek> use google? [19:57] <johnbender> :/ [19:57] <johnbender> thanks [19:57] <burek> soulstitchmmo can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [19:57] <burek> oh man [19:57] <burek> I need sleep again :D [19:58] <burek> johnbender I didn't tell that to you [19:58] <burek> <soulstitchmmo> ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers [19:58] <burek> I was refering to him :) [19:58] <johnbender> correct, but I think you were addressing me about using google when I asked where the IRC channel was [19:58] <burek> now, can you, johnbender, provide your ffmpeg command line and complete/uncut output? :) [19:59] <johnbender> burek: I'm on ubuntu too, though so I might needs head over there myself [19:59] <johnbender> let me get the version [19:59] <burek> well, ok :) [19:59] <johnbender> burek: ffmpeg version 0.7.6-4:0.7.6-0ubuntu0.11.10.1 [19:59] <burek> can you use a pastebin [20:00] <johnbender> for the version? [20:00] <soulstitchmmo> http://pastebin.com/d7tA4CE6 [20:00] <burek> for the complete output [20:00] <johnbender> I'll gladly use a pastebin for the output if I'm in the right irc channel [20:00] <johnbender> burek: do you care if I use a gist? [20:00] <burek> soulstitchmmo, you can either continue with that fork [20:00] <burek> or you can uninstall it and compile your own ffmpeg [20:00] <soulstitchmmo> Sorry burek, what's a fork? [20:00] <burek> I would advise later [20:01] <burek> it's kinda a copy of project headed in another direction [20:01] <burek> johnbender feel free to [20:01] <soulstitchmmo> I followed that wiki, but ubuntu told me that I didn't have it installed and to sudo apt-get install ffmpeg-libavtools or something like that. [20:02] <johnbender> burek: https://gist.github.com/238d9905bfc0e0c9d6e6 [20:02] <bwalker898> Hi, does anyone know the ffmpeg command to stream from a webcam on ubuntu via rtmp to red5 media server? I can stream a static video correctly, but I can't figure out how to send the webcam [20:02] <burek> one of ubuntu's maintainer (or more precise debian's) is a head of the libav project (a fork of ffmpeg), so I wouldn't be surprised he would put any kind of message, just to turn users to libav instead of ffmpeg [20:03] <burek> they made it really hard for users to trully make their own decision, but ok [20:03] <burek> in their house they can do whatever they want [20:03] <johnbender> I deal with enough oss politics, I'll stay out of this discussion :D [20:03] <burek> smart choice :) [20:04] <burek> soulstitchmmo, well, I compiled ffmpeg on any distribution I got my hands on, including ubuntu [20:04] <burek> so I'm pretty sure you were doing something wrong, or something.. [20:04] <soulstitchmmo> ok i'll copy paset through the thing again. [20:04] <burek> bwalker898: ffmpeg -i input -vcodec ... -f flv rtmp://ip.of.rtmp.server/ [20:06] <bwalker898> burek: thanks. i assume input should be /dev/video0, what is the "..." for? [20:06] <burek> no [20:06] <burek> what is your input actually [20:07] <burek> ("-vcodec libx264" for example) [20:07] <johnbender> burek: bwalker898: ffmpeg -f video4linux2 -i /dev/video0 ?? [20:07] <burek> yes, also use v4l2-ctl [20:07] <burek> to see what youre device supports [20:07] <burek> -e [20:07] <bwalker898> burek: i don't know? i have a webcam and i've seen video4linux2 used a lot on the internet [20:07] <burek> bwalker898 usb cam? [20:07] <bwalker898> burek: yes [20:08] <burek> up there, what johnbender wrote [20:08] <bwalker898> burek: k [20:08] <burek> also, the full version is something like [20:08] <burek> ffmpeg -f v4l2 -s 640x360 -r 25 -i /dev/vide0 -vcodec libx264 -f flv rtmp://ip [20:10] <bwalker898> burek: is there a different encoder I can use? ffmpeg said libx264 was unkown [20:11] <johnbender> burek: you'll have to install x264 explicitly I think [20:11] <burek> you really want x264 [20:11] <johnbender> err misdirect [20:12] <burek> it's like the best video codec today [20:12] <burek> or at least one of the best [20:12] <burek> see compiling section [20:12] <burek> its really quick [20:12] <bwalker898> burek: i just removed the -vcodec line hoping the default would work and it did!. Is there an advantage to the x264 codec? [20:12] <burek> huh... [20:13] <burek> is ferrari better than opel? :) [20:14] <bwalker898> burek: thanks. i guess i'll try to install it. Does it make video quality better? [20:14] <burek> it does [20:17] <bwalker898> burek: the video works. but the frame rate is only 12, even though i think the -r 25 comand you said is supposed to make it 25fps. Do you know why this would be? [20:18] <burek> bwalker898 can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [20:21] <soulstitchmmo> ok now I am getting 'File for preset 'superfast' not found: http://pastebin.com/d7tA4CE6 [20:22] <bwalker898> burek: http://pastebin.com/Ui0FyCps [20:22] <burek> soulstitchmmo, type x264 --help to see the list of possible presets [20:22] <burek> they are mapped directly [20:24] <burek> bwalker898 Stream #0:0: Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 320x240, q=2-31, 200 kb/s, 1k tbn, 30 tbc [20:24] <burek> it is 30 fps [20:24] <burek> check with vlc player or some other [20:24] <burek> to see if the fps is 30 [20:24] <burek> should be [20:24] <soulstitchmmo> --preset <string> Use a preset to select encoding settings [medium] [20:24] <soulstitchmmo> Overridden by user settings. [20:24] <soulstitchmmo> - ultrafast,superfast,veryfast,faster,fast [20:24] <soulstitchmmo> - medium,slow,slower,veryslow,placebo [20:24] <burek> but, it might be that your camera is capturing the image [20:25] <burek> at 12 fps so that's why you get the slow fps [20:25] <soulstitchmmo> it says that it's an option. [20:25] <burek> oh, that's because you've set presets for both audio and video [20:25] <burek> change -preset to -preset:v [20:26] <soulstitchmmo> it says vpre not preset [20:26] <soulstitchmmo> should it be preset:v instead of vpre? [20:26] <burek> bwalker898, can you type: v4l2 --list-formats-ext > log.txt [20:26] <burek> and paste the log.txt to pastbein [20:26] <burek> vpre is deprecated [20:27] <bwalker898> burek: yeah one sec [20:27] <burek> -vpre and -preset:v are two different things [20:27] <burek> vpre looks into the preset file [20:27] <burek> while -preset:v tells libx264 encoder to use its native preset [20:28] <burek> without any preset file [20:29] <burek> bwalker898, sorry [20:29] <burek> v4l2-ctl [20:29] <burek> it's: v4l2-ctl --list-formats-ext > log.txt [20:29] <burek> and it's contained in v4l2-utils package in debian/ubuntu [20:31] <soulstitchmmo> ok, so... I have progress. I'm streaming... and it's going somewhere... lol. This is the command I am using to get it to work: ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0+0,0 \ [20:31] <soulstitchmmo> -f alsa -ac 2 -i pulse -vcodec libx264 -preset:v "$QUAL" -s "$OUTRES" \ [20:31] <soulstitchmmo> -acodec libmp3lame -ab 128k -ar 44100 -threads 0 \ [20:31] <soulstitchmmo> -f flv "rtmp://live.justin.tv/app/$STREAM_KEY" [20:31] <burek> soulstitchmmo, there is a reason we ask people to use pastebin [20:31] <soulstitchmmo> I don't see anything on my stream, but I can see in terminal that frames are being made. [20:31] <burek> so please try to take that into consideration next time :) [20:31] <soulstitchmmo> ok one sec. [20:32] <soulstitchmmo> http://pastebin.com/d7tA4CE6 [20:32] <burek> well I don't see anything here [20:32] <soulstitchmmo> my terminal is still running, I just don't know where it's going :p [20:32] <burek> just your command [20:32] <burek> take a look at the Output: [20:32] <burek> if it is like this Output #0, flv, to 'rtmp://128.104.153.123:1935/live/livestream': [20:32] <burek> then it's obvious :) [20:33] <bwalker898> burek: http://pastebin.com/gtvY64rL [20:33] <burek> bwalker898, are you sure that's the right output (did you type the command correctly) [20:34] <bwalker898> burek: v4l2-ctl --list-formats-ext > log.txt [20:34] <soulstitchmmo> Ok how's this? http://pastebin.com/d7tA4CE6 [20:34] <burek> then your camera is one really simple camera :) [20:34] <burek> it only supports continuous jpeg images [20:34] <bwalker898> burek: lol okay, i have another nicer one here. it sounds like i should try that one... [20:35] <burek> soulstitchmmo, why are you constantly cut your output? [20:35] <burek> what's the point? [20:35] <soulstitchmmo> Huh? [20:35] <soulstitchmmo> I don't understand the question. I apologise. [20:36] <burek> you always cut some part you find interesting :) [20:36] <burek> so I have to instruct you to do it again :) [20:36] <burek> try to always provide a full uncut output [20:36] <burek> to save time for both :) [20:36] <soulstitchmmo> Well it's a continous stream of information [20:36] <soulstitchmmo> I don't know what you need to see. [20:36] <burek> full output? [20:36] <soulstitchmmo> It's constantly updating [20:36] <burek> well [20:37] <soulstitchmmo> my terminal is just updating frame= blah blah [20:37] <burek> run it again like this: ffmpeg -t 10 (all the other options) 2> log.txt [20:37] <burek> that will run it for 10 seconds [20:37] <burek> and provide you with log.txt that we need [20:37] <soulstitchmmo> ok [20:39] <soulstitchmmo> ok let's hope this is what you need: http://pastebin.com/d7tA4CE6 [20:40] <burek> there are errors [20:40] <burek> as you can see [20:41] <johnbender> soulstitchmmo: you need to set those variables if you haven't [20:42] <johnbender> I'm assuming that INRES isn't set? [20:42] Action: johnbender could be wrong [20:43] <johnbender> burek: did you get a chance to look at my gist? [20:43] <johnbender> I know you're helping stitch so I'm happy to wait :D [20:45] <soulstitchmmo> burek: http://pastebin.com/d7tA4CE6 [20:45] <soulstitchmmo> I just put the variables in their directly. [20:46] <burek> johnbender I see it, what exactly is the problem (you're using libav not ffmpeg) [20:47] <johnbender> burek: the v4l input is delayed [20:47] <johnbender> ideally I'd like to sync it with the pulse audio input but I'd settle for "close" [20:47] <johnbender> burek: I'll try #libav, hopefully they'll know what to do [20:47] <burek> soulstitchmmo: Output #0, flv, to 'rtmp://live.justin.tv/app/live_22303455_pUP5U3DxOWvHfmBAuEJs7FiGGL44hu' [20:48] <burek> it is going to justin.tv [20:48] <burek> johnbender you might try updating ffmpeg [20:48] <burek> to latest version [20:49] <burek> since a lot of bugs are fixed each day [20:49] <burek> and you are also using very old version of it [20:50] <johnbender> burek: I tried building from git following the ubuntu install guide but all I got was continuous dropped frames :( [20:50] <burek> try to build ffmpeg with libx264 and libvo_aacenc [20:50] <burek> that should give you a stable stream [20:51] <burek> also, read a little about -async -isync and -vsync [20:51] <burek> see if that helps [20:51] <johnbender> burek: I did, using the guide on the ffmpeg site [20:51] <johnbender> burek: will do [20:51] Action: johnbender goes off to read [20:53] <randomguy123> Hi FFMPEG, I use a video server and I want to make ffmpeg talk directly to it. I have api lib and I can connect and READ raw yuv frames or WRITE yuv frames to it. Can you recommend where to look for clues? [20:53] <randomguy123> At first, I wanted to look for some device in implementation in avdevice lib [20:53] <randomguy123> but I also think it's possible to add virtual container format that stores raw yuv frames, but since the server is streamable then my container can't know the size etc. So, what is recommneded path I should take? [20:54] <burek> randomguy123, take a look at ffmpeg's source code [20:54] <burek> see how it handles yuv [20:55] <burek> and copy/paste the logic [20:55] <soulstitchmmo> That's where I'm supposed to rtmp too. [20:56] <randomguy123> burek: my main concern is to try to write a new avdevice or a new "virtual" container handler for avformat. Where to find and copy code - I'll figure out.. [20:56] <burek> soulstitchmmo that's where you are currently outputing your result [20:57] <soulstitchmmo> I just tried a different RTMP server, still not seeing anything on my stream. [20:58] <burek> well if it sends the stream without problems, then it's not ffmpeg's issue any more, is it :) [20:58] <soulstitchmmo> theoretically :p [20:58] <soulstitchmmo> what do I replace if I want to send it to a local file? So I can just make sure it's doing something [20:59] <burek> replace rtmp://... with bla.flv [21:00] <soulstitchmmo> YAY! [21:01] <soulstitchmmo> so that all works :D [21:01] <burek> :) [21:01] <soulstitchmmo> just a matter of getting the streaming to twitch thing to work. Thanks Burek. [21:01] <burek> :beer: :) [21:07] <soulstitchmmo> Fan freaking tastic. Set up a seperate script for local recording, so that's awesome. Just have to get the streaming now :p [21:09] <pertplus> hi I just converted a mp4 to wmv and there are grey artifacts/spots on parts that are supposed to be black, especially when they move.. [21:10] <pertplus> how can i fix this ;) ? [21:11] <pertplus> command i used was ffmpeg -i infile.mp4 outifle.wmv [21:33] <burek> pertplus, well specify some encoding quality options [21:33] <burek> don't leave it at default [21:34] <pertplus> thats what i figured but i cant find the name of the options [21:34] <pertplus> how do i do it at best quality [21:34] <burek> did you read the documentation [21:35] <burek> try for example with -b [21:35] <burek> or something, I dunno [21:35] <pertplus> i ok [21:36] <burek> best quality you would get without re-encoding :) [21:36] <burek> something like: ffmpeg -i input -c copy output [21:36] <burek> but I don't know if wmv format would support the codec from input [21:36] <pertplus> but i have to conver t from mp4 to wmv [21:36] <burek> mp4 is a format/container [21:36] <pertplus> ill try -b [21:36] <pertplus> oh [21:36] <burek> you don't convert a container [21:36] <burek> you convert streams inside it [21:36] <pertplus> ok [21:37] <pertplus> then ill try to copy [21:37] <burek> anyway, can you use pastebin to show the output of ffmpeg -i infile.mp4 [21:40] <pertplus> bunch of message about clipping coefficients [21:42] <soulstitchmmo> ok, i recroded a gameplay session, and the sound is way out of sync. [21:46] <pertplus> how do i copy with highest possible quality [21:46] <pertplus> -sameq ? [22:41] <relaxed> pertplus: -c copy [00:00] --- Sat Jul 28 2012

1 0

ffmpeg-devel.log.20120726
by burek 27 Jul '12

27 Jul '12

[00:06] <Freakshow> does out->size; include packaging overhead? [00:37] <michaelni> crtmpserver, i dont think you are on the right track with debuging this [00:38] <michaelni> can you try to remove probesize = 0 before av_find_stream_info ? [00:39] <michaelni> i mean in hls.c [00:45] <CIA-41> ffmpeg: 03Nicolas George 07master * r6bf87785e8 10ffmpeg/ffmpeg.c: ffmpeg: warn about new streams after startup. [00:45] <CIA-41> ffmpeg: 03Nicolas George 07master * rd0d9272580 10ffmpeg/doc/faq.texi: faq: explain about new streams, probesize and analyzeduration. [00:47] <michaelni> crtmpserver, also if removing probesize=0 works, please try "v->ctx->ctx_flags &= ~AVFMTCTX_NOHEADER;" instead of probesize =0 [01:03] <Mista_D> A quick question - why MP4 atom moov/trak(X)/tkhd/flag = 15? Shouldn't it be 1 or 7 (ISO/IEC 14496-12:2004). [01:09] <MP4_maniac> Mista_D: dunno. flag 0x8 indicates that the track is used in the movie's poster but it is defined in Apple's MP4 or MOV. ISO doesn't define it [01:10] <Mista_D> Any way to specify Apple/ISO format during encoding? [01:10] <cptspiff> -f applemustdie [01:11] <MP4_maniac> -f ipod [01:11] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r9054f6b66b 10ffmpeg/libavformat/utils.c: [01:11] <CIA-41> ffmpeg: probe_codec: fix memory corruption [01:11] <CIA-41> ffmpeg: Found-by: Tanami Ohad [01:11] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:12] <Mista_D> MP4_maniac: Thank you. [01:14] <Mista_D> MP4_maniac: still 0F, I think ipod uses default MP4 which is "Apple's" format, not ISO. [01:27] <crtmpserver> michaelni, I'm all over it :) [01:40] <Mista_D> MP4_maniac: x0F is a hard coded value in libavformat/movenc.c line 1540. [01:41] <MP4_maniac> yup [02:08] <crtmpserver> michaelni, removing probesize = 0 fixed it! [02:08] <crtmpserver> amazing! [02:08] <crtmpserver> :) [02:09] <michaelni> crtmpserver, does it still work with v->ctx->ctx_flags &= ~AVFMTCTX_NOHEADER; instead ? [02:10] <crtmpserver> let me check [02:10] <michaelni> this would reduce the amount of delay / latency [02:10] <crtmpserver> so, should I leave probesize in there? [02:10] <crtmpserver> and add v->ctx->ctx_flags &= ~AVFMTCTX_NOHEADER [02:10] <crtmpserver> ? [02:12] <crtmpserver> having both is not working [02:12] <crtmpserver> I guess that is not an issue, because HLS is already high-latency streaming [02:12] <crtmpserver> talking about 10-15 seconds [02:13] <crtmpserver> because of the nature of it [02:14] <crtmpserver> removing probesize=0 and adding v->ctx->ctx_flags &= ~AVFMTCTX_NOHEADER also wroks [02:14] <crtmpserver> I think probesize=0 was the faulty piece [02:50] <michaelni> crtmpserver, thanks alot for the testing ill commit a fix soon [02:52] <crtmpserver> I thank you for it. You saved my ass :) [02:53] <crtmpserver> now I can do -vcodec copy in a transrating environment [02:54] <crtmpserver> before, I was forced to re-encode the original [02:54] <crtmpserver> wasting valuable CPU horsepower [03:10] <CIA-41> ffmpeg: 03Piotr Bandurski 07master * rc5f23df2dd 10ffmpeg/libavformat/movenc.c: [03:10] <CIA-41> ffmpeg: movenc: fix remuxing of yuv2 [03:10] <CIA-41> ffmpeg: Attached patch fixes remuxing of yuv2 in mov (and does not break remuxing of yuvs) [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4e7f9ebdc2 10ffmpeg/libavformat/wav.c: [03:10] <CIA-41> ffmpeg: wav: switch to AVSTREAM_PARSE_FULL_RAW [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r6f6182372c 10ffmpeg/libavformat/hls.c: [03:10] <CIA-41> ffmpeg: hls: replace probsize=0 by noheader flag removial. [03:10] <CIA-41> ffmpeg: This fixes h264 timestamps in hls [03:10] <CIA-41> ffmpeg: Fixes ticket1572 [03:10] <CIA-41> ffmpeg: Tested-by: crtmpserver [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r12d1ee6a5e 10ffmpeg/libavformat/utils.c: [03:10] <CIA-41> ffmpeg: ff_add_index_entry: reject AV_NOPTS_VALUE [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r5ea4ee7bcb 10ffmpeg/libavformat/flacdec.c: [03:10] <CIA-41> ffmpeg: flacdec: mark as AVSTREAM_PARSE_FULL_RAW [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r2880c31660 10ffmpeg/libavformat/utils.c: [03:10] <CIA-41> ffmpeg: parse_packet: use pkt.pos always instead of the avparser offset. [03:10] <CIA-41> ffmpeg: a later point. [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rfdc371114f 10ffmpeg/libavformat/rawdec.c: [03:10] <CIA-41> ffmpeg: ff_raw_audio_read_header: mark as AVSTREAM_PARSE_FULL_RAW [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rea8c12c90f 10ffmpeg/libavformat/mp3dec.c: [03:10] <CIA-41> ffmpeg: mp3dec: mark as AVSTREAM_PARSE_FULL_RAW [03:10] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:10] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r2cd491a47c 10ffmpeg/ (libavformat/utils.c tests/ref/seek/mp2_mp2): (log message trimmed) [03:10] <CIA-41> ffmpeg: lavf: move generic index generation code to a later point [03:10] <CIA-41> ffmpeg: By moving it to a later point relative and unknown timestamps [03:11] <CIA-41> ffmpeg: are more likely to have been corrected [03:11] <CIA-41> ffmpeg: similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [03:11] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:11] <CIA-41> ffmpeg: Conflicts: [03:34] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r9c83f3e659 10ffmpeg/libavformat/loasdec.c: [03:34] <CIA-41> ffmpeg: loasdec: switch to AVSTREAM_PARSE_FULL_RAW [03:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:34] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * re083a62e9d 10ffmpeg/libavformat/aacdec.c: [03:34] <CIA-41> ffmpeg: aacdec: switch to AVSTREAM_PARSE_FULL_RAW [03:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:34] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r97c0d5357d 10ffmpeg/libavformat/omadec.c: [03:34] <CIA-41> ffmpeg: omadec: switch to AVSTREAM_PARSE_FULL_RAW [03:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [08:18] <rohankasat> hi , can anybody tell me . how ffmpeg calls a decoder ( from the different registered decoders) , so that it can process further ? any particular function call or wat ? [10:14] <CIA-41> ffmpeg: 03William Yu 07master * rd5d5e3dabe 10ffmpeg/libavfilter/af_amerge.c: lavfi/af_amerge: avfilter_copy_buffer_ref_props will overwrite output pts [13:16] <dougle> does anyone know of a filter to reduce 4:4:4 chroma down to baseline h264 compatible? [13:17] <dougle> have not used filters before and am a little lost [13:33] <michaelni> dougle, you probably search for the -pix_fmt option [15:08] <dougle> michaelni i salute you sir! very helpful thanks [16:36] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7a4e30f3b6 10ffmpeg/libavcodec/h264_cabac.c: [16:36] <CIA-41> ffmpeg: h264_cabac: switch to av_assert [16:36] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r506ffa793f 10ffmpeg/libavcodec/h264_cavlc.c: [16:37] <CIA-41> ffmpeg: h264_cabac: switch to av_assert [16:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r55fa883b30 10ffmpeg/ffmpeg.c: [18:08] <CIA-41> ffmpeg: ffmpeg: fix units of next_?ts in debug printing routine. [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r5938b4d398 10ffmpeg/libavcodec/mpc7.c: [18:08] <CIA-41> ffmpeg: mpc7: Initialize AVFrame properly [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * re15e5328a7 10ffmpeg/libavcodec/mpc8.c: [18:08] <CIA-41> ffmpeg: mpc8: Initialize AVFrame properly [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rb616405c16 10ffmpeg/ffmpeg.c: [18:08] <CIA-41> ffmpeg: ffmpeg: reset decoded_frame->pts after rescaling. [18:08] <CIA-41> ffmpeg: This ensures its not used afterwards. [18:08] <CIA-41> ffmpeg: This commit should not change anything as it should not have been used afterwards. [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r9b1fdb1c5a 10ffmpeg/ffmpeg.c: [18:08] <CIA-41> ffmpeg: ffmpeg: fix assumed decoded_frame->pts timebase [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r1a7af514de 10ffmpeg/ffmpeg.c: [18:08] <CIA-41> ffmpeg: ffmpeg: move decoded_frame->pts computation together [18:08] <CIA-41> ffmpeg: This clarifies the code. [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:08] <CIA-41> ffmpeg: ffmpeg: Fix rounding errors in decoded_frame->pts computation [18:08] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:42] <maker> Writing here, as burek suggested. Hello, I was interested in the Summer of Code in Space program, in particoular to the libavfilter project. Is saste, or any other backup mentor going to join before the deadline? [18:50] <maker> bbl [19:06] <durandal_1707> was? [19:27] <ubitux> maker: saste might come back tonight here [19:28] <ubitux> so you might want to wait for him until them, or send him a mail [19:29] <maker> ubitux: I will probably try to join tonight, thanks. I saw saste sending mails at 2AM, I suppose he did not change his habits :D [19:30] <ubitux> well maybe you're not in the same timezone :) [19:31] <maker> ubitux: I am :p [19:31] <iive> deadline? I thought that deadlines for sign up are somewhere between February and April. The deadlines that are coming is for finished projects. [19:31] <iive> are... [19:32] <maker> iive: for socis? [19:32] <maker> http://sophia.estec.esa.int/socis2012/?q=timeline [19:32] <iive> oh, sorry, i missed the "in space" bit. [20:00] <michaelni> maker, if you want to apply for SOCIS make sure you submit whatever is needed (application i guess) before the deadline [20:02] <michaelni> maker, but also wait and talk with saste, he is mentor of the libavfilter stuff IIRC and you will have to do some kind of qualification task [20:05] <philipl> ubitux: Any chance you can look at the timed text encoder? [20:05] <ubitux> yep i was purging my mails [20:06] <ubitux> i didn't have time for me lately sorry [20:06] <ubitux> i'll have a look tonight [20:06] <ubitux> sorry for the delay :) [20:07] <orbisvicis> is there a list of api changes, ie av_close_input_file -> avformat_close_input [20:07] <orbisvicis> etc [20:08] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/APIchanges;hb=HEAD [20:08] <orbisvicis> thanks [20:20] <ubitux> it seems diego wants to drop gopher support because protocols don't belong to the project [20:20] <ubitux> it makes me willing to support samba just like mplayer or vlc [20:24] <philipl> ubitux: no probs. thanks. [20:33] <durandal_1707> michaelni: you devastated Changelog in one of your merges [20:33] Action: durandal_1707 took look at Changelog and got WTF moment [20:34] <durandal_1707> git show 2cb4d516549526b5e17e941f6d2375a2c501ade6 [20:34] <nevcairiel> HEAD changelog looks ok [20:35] <nevcairiel> oh some lines went missing there [20:37] <ubitux> isn't it missing a few things? [20:38] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r307a20cca2 10ffmpeg/Changelog: [20:38] <CIA-41> ffmpeg: Changelog: put entries lost in 2cb4d516549526b5e17e941f6d2375a2c501ade6 back [20:38] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:38] <ubitux> thx :) [20:55] <ubitux> philipl: is your mov_text branch in sync with your patchset? [20:56] <philipl> Sorry, no. I forgot to push to that, and I can't push it right now. [20:56] <philipl> I can do it tonight. [20:56] <ubitux> ok no worry, i'll deal with the patches [21:28] <orbisvicis> what is the version test for lavu 51.8.0 ? LIBAV..._VERSION_INT [21:33] <orbisvicis> oh libavutil [21:37] <orbisvicis> is this a typo: [21:37] <orbisvicis> Add av_get_bits_per_sample_fmt() to libavcore/samplefmt.h [21:37] <orbisvicis> shouldn't it be libavutil/samplefmt.h ? [21:38] <orbisvicis> http://ffmpeg.org/doxygen/trunk/samplefmt_8c.html#ed9c391585b4d4d7c49a0ef88… [21:38] <orbisvicis> ie version test against libavu vs libavc [21:38] <nevcairiel> libavcore has been removed a long time ago [21:47] <orbisvicis> ah ok [21:48] <orbisvicis> how long ago? is it safe to completely replace av_get_bits_per_sample_format? [21:50] <nevcairiel> all its functions are still present, just moved to libavutil [21:55] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r9b498148ca 10ffmpeg/libavformat/rtmpproto.c: [21:55] <CIA-41> ffmpeg: rtmp: Factorize the code by adding handle_server_bw [21:55] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:55] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r912ecc9a19 10ffmpeg/libavformat/rtmpproto.c: [21:55] <CIA-41> ffmpeg: rtmp: Factorize the code by adding handle_client_bw [21:55] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:55] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r7be2a7d8ff 10ffmpeg/libavformat/rtmpproto.c: [21:55] <CIA-41> ffmpeg: rtmp: Factorize the code by adding handle_chunk_size [21:55] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:55] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r6d1c9945dd 10ffmpeg/libavformat/rtmpproto.c: [21:55] <CIA-41> ffmpeg: rtmp: Factorize the code by adding handle_invoke [21:55] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:55] <CIA-41> ffmpeg: 03Yang Wang 07master * r845e92fd6a 10ffmpeg/libavcodec/x86/dsputil_mmx.c: (log message trimmed) [21:55] <CIA-41> ffmpeg: dsputil_mmx: fix incorrect assembly code [21:55] <CIA-41> ffmpeg: In ff_put_pixels_clamped_mmx(), there are two assembly code blocks. [21:55] <CIA-41> ffmpeg: In the first block (in the unrolled loop), the instructions [21:55] <CIA-41> ffmpeg: "movq 8%3, %%mm1 \n\t", and so forth, have problems. [21:55] <CIA-41> ffmpeg: From above instruction, it is clear what the programmer wants: a load from [21:55] <CIA-41> ffmpeg: p + 8. But this assembly code doesnt guarantee that. It only works if the [21:55] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r79195ce565 10ffmpeg/libavcodec/ (8 files in 2 dirs): [21:55] <CIA-41> ffmpeg: x86/dsputil: put inline asm under HAVE_INLINE_ASM. [21:55] <CIA-41> ffmpeg: This allows compiling with compilers that don't support gcc-style [21:55] <CIA-41> ffmpeg: inline assembly. [21:55] <CIA-41> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [21:55] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * r48098788c2 10ffmpeg/libavcodec/vp8.c: [21:55] <CIA-41> ffmpeg: vp8: Replace x*155/100 by x*101581>>16. [21:55] <CIA-41> ffmpeg: Idea stolen from webp (by Pascal Massimino) - because it's Cool. [21:55] <CIA-41> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [21:55] <CIA-41> ffmpeg: 03Ronald S. Bultje 07master * ra1878a88a1 10ffmpeg/libavcodec/x86/vp3dsp.asm: [21:55] <CIA-41> ffmpeg: vp3: don't use calls to inline asm in yasm code. [21:55] <CIA-41> ffmpeg: Mixing yasm and inline asm is a bad idea, since if either yasm or inline [21:55] <CIA-41> (44 lines omitted) [21:55] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r7333798c85 10ffmpeg/: (log message trimmed) [21:55] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [21:55] <CIA-41> ffmpeg: * qatar/master: [21:56] <CIA-41> ffmpeg: libopenjpeg: support YUV and deep RGB pixel formats [21:56] <CIA-41> ffmpeg: Fix typo in v410 decoder. [21:56] <CIA-41> ffmpeg: vf_yadif: unset cur_buf on the input link. [21:56] <CIA-41> ffmpeg: vf_overlay: ensure the overlay frame does not get leaked. [21:56] <CIA-41> ffmpeg: 03Samuel Pitoiset 07master * r0ffd5161c4 10ffmpeg/libavformat/rtmpproto.c: [21:56] <CIA-41> ffmpeg: rtmp: Factorize the code by adding handle_ping [21:56] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [21:59] <orbisvicis> nevcairiel: so which version of avutil is the last to have av_get_bits_per_sample_format? [22:06] <orbisvicis> i see -> digging in the git history [22:08] <ubitux> hey saste, a socis student was looking for you [22:08] <ubitux> 18:42:19 < maker> Writing here, as burek suggested. Hello, I was interested in the Summer of Code in Space program, in particoular to the libavfilter project. Is saste, or any other backup mentor going to join before the deadline? [22:08] <ubitux> philipl: i'll review the other patch when i'll be able to test it, i'd like to make some tests [22:08] <saste> ubitux: yes, I see he applied [22:08] <saste> i'll contact him later [22:09] <saste> application deadline is tomorrow, for the selection there is time until July 31 [22:09] <ubitux> philipl: feel free to nudge me as soon as you update your branch [22:09] <ubitux> saste: ok; he might get back on irc [22:09] <saste> i'm going to post a reminder to the ffmpeg-* socis thread [22:10] <ubitux> ok :) [22:18] <philipl> ubitux: will do. [22:19] <philipl> ubitux: btw: movtext.h is not used. You can just remove it to test. [22:19] <ubitux> ah? great ok [22:20] <philipl> That's from my initial styling work. [22:20] <ubitux> then maybe i'll end the review tonight [22:20] <ubitux> ok [22:21] <ubitux> philipl: btw, that's great to be able to mux subtitles in mp4 [22:21] <ubitux> thank you for working on this :) [22:24] <ubitux> fate is broken :( [22:25] <CIA-41> ffmpeg: 03Clément BSsch 07master * r2755abff3e 10ffmpeg/libavcodec/realtextdec.c: lavc/rt: small wording fix in a comment. [22:25] <CIA-41> ffmpeg: 03Clément BSsch 07master * r316f8db2c2 10ffmpeg/ (13 files in 6 dirs): SubViewer demuxer and decoder. [22:45] <Freakshow> burek... or anyone else for that matter. Is av_log designed to output data in 500mS samples? [22:46] <philipl> ubitux: You're welcome [22:46] <philipl> I broke fate? [22:47] <Freakshow> nevermind, I see it in print_report [22:47] <Freakshow> < 5000000 [22:47] <Freakshow> sorry... 500000 [22:49] <ubitux> philipl: no no, server is down [22:50] <ubitux> philipl: btw, is there any player actually showing these subtitles? [22:50] <ubitux> seems like mplayer & vlc don't [22:50] <philipl> ubitux: mplayer and vlc both can handle them. [22:50] <philipl> I test with mplayer and ipad. [22:51] <ubitux> i did it wrong then ok [22:51] <philipl> mplayer has very basic support but it works for unstyled subs. [22:51] <ubitux> btw, two things: [22:52] <ubitux> (which might not be related to your patch) [22:52] <ubitux> 1) can't we avoid the -c:s mov_text? (auto select codec) [22:52] <ubitux> 2) seems like subrip demuxer has a problem: ./ffmpeg -f lavfi -i testsrc=d=60 -i ~/fate-samples/sub/SubRip_capability_tester.srt -map 0:0 -map 1:0 -c:s mov_text -y out.mp4 [22:53] <philipl> 2) Yes. That's the duration vs convergence_duration problem [22:53] <philipl> 1) I'm sure we can but there's no logic to do auto-matching for subs today. We'd have to add it. [22:53] <ubitux> ok [22:55] <ubitux> srt doesn't seem to have anything with convergence_duration [22:57] <philipl> Sorry. It's actually that there's *no* duration [22:57] <philipl> You need this: https://github.com/philipl/FFmpeg/commit/bd701c6dab3e57381d6c8bb1c9e651f755… [22:58] <ubitux> ah, makes sense. [22:58] <ubitux> did you submit this? [22:58] <philipl> No. But I guess I should :-) [22:59] <ubitux> yup [22:59] <philipl> On a related note: What's our official story on convergence_duration then? Matroska doesn't write duration. Which piece should be responsible for making things line up. [22:59] <ubitux> no idea, i didn't follow the convergence_duration story [23:00] <ubitux> it's all about having a 64-bit ts vs 32? [23:00] <philipl> Yeah. After we established that there was value in having 64bits, the discussion died off. [23:04] <ubitux> well does it make any sense? [23:05] <ubitux> a subtitle event hardly last for more than 1 minute [23:05] <ubitux> even with an insane timebase i don't see how that could really be a problem [23:05] <ubitux> or maybe i'm missing the point? [23:07] <philipl> I don't think it makes any sense myself. [23:08] <philipl> The one thing I did think about was if the container is like mp4, then there's no stored duration, and the duration gets calculated to the next sample at demux time. In that case, an empty sample might have a very long duration. [23:08] <philipl> But the decoder discards those samples, so their duration doesn't effect anything. [23:08] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r236ecc3502 10ffmpeg/tests/Makefile: (log message trimmed) [23:08] <CIA-41> ffmpeg: fate: change fate-suite rsync server. [23:08] <CIA-41> ffmpeg: there are some technical problems with fate.ffmpeg.org [23:08] <CIA-41> ffmpeg: thus split the subdomain between fate-suite and fate [23:08] <CIA-41> ffmpeg: fate-suite is now (temporary) provided by our main server [23:08] <CIA-41> ffmpeg: until fate-suite.ffmpeg.org is setup to point somewhere [23:08] <CIA-41> ffmpeg: we use fate-suite.avcodec.org [23:09] <ubitux> philipl: don't we use duration = -1 for these case anyway? [23:09] <philipl> ubitux: No. I'm not sure what magic goes on inside the mov demuxer, but it manages to correctly recover subtitle durations. [23:14] <philipl> It looks like there's some kind of sample index that is provided so it can calculate durations ahead of time. [23:19] <ubitux> i have some timing issue here, but i need to investigate [23:19] <ubitux> i'll take some time this week end [00:00] --- Fri Jul 27 2012

1 0

ffmpeg.log.20120726
by burek 27 Jul '12

27 Jul '12

[00:00] <llogan> i guess. i've never used it. [00:00] <llogan> probesize default is sudo checkinstall --pkgname=libvpx --pkgversion="1:$(date +%Y%m%d%H%M)-git" --backup=no \ [00:00] <llogan> wrong paste obviously [00:00] <llogan> probeseize default ~5 Mo [00:01] <dmitrykharlamov> thanks for helping me again llogan! [00:01] <dmitrykharlamov> I'm trying to increase the probesize during probing, just a min [00:01] <llogan> i'm not very familiar with those options but maybe they'll work for you [00:03] <dmitrykharlamov> analyzeduration did not yield any results [00:03] <dmitrykharlamov> still the same error [00:04] <llogan> maybe your probe needs to be bigger (sounds painful). [00:05] <dmitrykharlamov> lol [00:05] <dmitrykharlamov> what do you mean? [00:06] <llogan> -probesize <integer> [00:06] <llogan> where did your input files come from? [00:08] <dmitrykharlamov> ah-ha. It actually just worked. And the files I've been trying to test it on do seem damaged. Which is good and bad news at the same time [00:08] <llogan> what probesize did you use? [00:08] <dmitrykharlamov> when I set the -probesize to say 100, it then says that /path/video.mp4 is an invalid argument [00:09] <dmitrykharlamov> without probesize it worked [00:09] <dmitrykharlamov> when I enter 'ffmpeg -probesize 100 -i /video.mp4' it says that /video.mp4 is an invalid argument [00:12] <dmitrykharlamov> k, thanks very much for your help! I have found a problem on my end now! [00:12] <llogan> dmitrykharlamov: glad you figured it out [00:50] <Freakshow> I'm trying to capture the total number of bytes going out on a per stream basis, currently we've figured out the raw video, audio and sub-title data... but the decoded bitrate on the consumer side appears to vary a little more than expected. Is there a way to capture the true output data packet size taking into consideration network packaging overhead? [00:50] <Mista_D> Any explanation on why MP4 atom moov/trak(X)/tkhd/flag = 15? Should be 1 or 7 (ISO/IEC 14496-12:2004). [01:19] <DelphiWorld> burek :-P [01:54] <burek> johnbender, did you try reading the docs? :)) [01:54] <burek> DelphiWorld hi :) [01:54] <DelphiWorld> burek: ; [01:56] <burek> Freakshow, install tcpdump or pktstat [01:56] <burek> Mista_D, ask in #ffmpeg-devel [01:56] <burek> oh you already did ^^ [01:59] <Freakshow> burek: doing this in ffmpeg.c [02:00] <Freakshow> since the output logging is basically broken on a non-seekable output [02:02] <burek> why would you measure the bit rate in the code? [02:02] <burek> on the consumer side you could use vlc [02:02] <burek> and go to tools - codec information [02:02] <burek> tab statistics [02:03] <Freakshow> right, need this to scrap locally as we're using ffmpeg in an encoder on the client side [02:03] <Freakshow> scrap+e [02:03] <Freakshow> for output statistics [02:05] <Freakshow> it looks like in ffmpeg.c there was a comment to fix seeking in live streams [02:05] <Freakshow> this is kind of in reference to my ticket against output logging with rtsp [02:05] <Freakshow> https://ffmpeg.org/trac/ffmpeg/ticket/1446 [02:06] <Freakshow> I haven't seen any movement on the ticket, so we're fixing it ourselves and if it's worthwhile, we'll submit it for review as the resultant output data is much more accurate [02:07] <Freakshow> isn't it really late (or early) for you burek ? [02:46] <DelphiWorld> how to bind mysql to a specific ip? [02:47] <DelphiWorld> oh [02:47] <DelphiWorld> sory [02:47] <DelphiWorld> not here;( [07:13] <bwalker8981> Hi I'm trying to install the xuggler extension to ffmpeg, but when I run ant, I get build failed and it reports "/buildhelper.xml:1127: exec returned: 1". Any suggestions? [07:13] <bwalker8981> Or does anyone know what that means? [07:23] <bwalker8981> Does anyone know how to install xuggle for ffmpeg? [08:19] <rohankasat> hi , can anybody tell me . how ffmpeg calls a decoder ( from the different registered decoders) , so that it can process further ? any particular function call or wat ? [08:21] <t4nk823> hi , can anybody tell me . how ffmpeg calls a decoder ( from the different registered decoders) , so that it can process further ? any particular function call or wat ? [09:22] <johnbender> burek: I did, I've been going over them for a few hours [09:22] <johnbender> I think I need more background on basic video configuration :( [09:33] <johnbender> burek: I should say, I _was_ going over them for a few hours before I posted my question [09:33] <johnbender> I also tried building ffmpeg from git [09:34] <johnbender> unfortunately I was getting "invalid dropping" [09:43] <johnbender> in the meantime [09:43] <johnbender> I'm back :(, still experiencing a 4 to 5 second lag (seems consistent) in my webcam overlay. Still not sure how to correct it https://gist.github.com/84f47d78cccbf0c632e0 [09:43] <johnbender> as mentioned I've spent a long time on this without getting anywhere [09:44] <johnbender> I have little knowledge of how video manipulation works [09:44] <johnbender> so that makes it harder to figure out [10:56] <astropirate> is it possible to pipe audio from gst-launch into ffmpeg capture? [11:03] <saste> astropirate: ffmpeg -i - ... [11:05] <astropirate> saste, forgive my stupid question.. but should I save it to a file via gst-launch and then read it with ffmpeg? [11:05] <saste> astropirate: don't know how to use gst-launch, but you can read a pipe from ffmpeg [11:05] <saste> check the pipe protocol, maybe there is a more clever way, dunno [11:06] <astropirate> ok, thanks for your help [11:35] <burek> rohankasat, check how does ffmpeg do it in the source code [11:36] <burek> johnbender http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20speed%20up%20/%20slow%20down%… [11:37] <burek> oh, johnbender, you don't need setpts in that case [11:37] <burek> try reading about -async -isync and -vsync [11:38] <burek> also this might help http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(scree… [11:41] <astropirate> I am trying to record my desktop but also merge audio form an audio file into the recording [11:41] <astropirate> I am doing this ffmpeg -f alsa -i output.mp3 -f x11grab -r 25 -s 1280x720 -i :0 -vn -acodec flac -vcodec libx264 -preset ultrafast -threads 0 video.mkv [11:41] <astropirate> but it gives me this error: [11:41] <astropirate> [alsa @ 0x110ce20] cannot open audio device output.mp3 (No such file or directory) [12:27] <dougle> i'm setting profile:v to baseline for converting one png to a 5 second segment but i keep getting the x264 error "baseline profile doesn't support 4:4:4" anyone know how i can get away from it? [12:29] <burek> astropirate, what's the point of this statement "ffmpeg -f alsa -i output.mp3" ? [12:30] <astropirate> burek, to read the file [12:30] <burek> remove -f alsa [12:31] <burek> can you please use pastebin.com, to show your command line and its output? [12:32] <burek> dougle* [12:32] <astropirate> burek, I am in your debt sir/ma'am [12:32] <burek> astropirate :beer: :) [12:33] <burek> dougle can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [12:39] <dougle> burek: https://gist.github.com/3181423 [12:39] <dougle> (i think i owe you a keg by now burek) [12:41] <astropirate> here is another dumb question: [12:42] <madsy> I have a game emulator I'd like to screen capture with ffmpeg and its x11grab interface. But when I run the emulator in fullscreen it takes a fast YUV path with XVideo, so due to the hardware overlay, ffmpeg has nothing to copy. Is there a workaround? [12:42] <astropirate> ffmpeg stps the x11 grab when the music is finished. how do i make it just use silence :) untill the x11 grab is finished? [12:42] <burek> dougle try -profile:v instead of -profile [12:43] <burek> astropirate, why don't you first grab what you need with x11grab and when you finish, just add the audio to it [12:44] <burek> madsy, most emulators have an option to take a screenshot [12:44] <burek> try reading its manual for it [12:44] <astropirate> burek, because.. i'm doing a hack. system audio grab with ffmpeg sucks... lots of static. so i am using gst-launch to capture the audio [12:44] <astropirate> it saves it to file and ffmpeg reads it [12:44] <madsy> burek: pcsx-r does not have anything like that. Besides, I want video not still frames. [12:45] <burek> astropirate, then don't use the mp3 file, use the pipe for audio [12:45] <burek> so it won't end [12:45] <astropirate> burek, YES!! :D how do I do that/ [12:45] <burek> btw, astropirate what command did you use when you tried "system audio grab" :) [12:45] <astropirate> i tried figuring that out but couldn't figure out how to do it [12:46] <astropirate> gst-launch -e pulsesrc device="alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ! audioconvert ! lamemp3enc target=1 bitrate=128 cbr=true ! filesink location=output.mp3 [12:46] <dougle> same result, just no deprecation message [12:46] <astropirate> OHHH [12:46] <astropirate> sorry i misunderstood burek sec let me find out [12:47] <burek> ffmpeg -y -loop 1 -threads 0 -i input_image.png -f pcm_s16le -i - ... [12:47] <burek> some_script_that_produces_wav_output | ffmpeg -y -loop 1 -threads 0 -i input_image.png -f pcm_s16le -i - ... [12:47] <burek> the point is in "-i -" [12:47] <burek> (and -f before it) [12:48] <astropirate> i dont know if gst-launch will output audio to stdout [12:48] <astropirate> :( [12:48] <astropirate> for me to pipe it to ffmpeg [12:49] <burek> madsy, it appears you can't do a screen capture in fullscreen, because the emulator is not using system video output, but draws directly using the display hardware [12:49] <astropirate> ohh i think it does let me try it [12:49] <burek> so, try capturing in non-fullscreen [12:50] <burek> madsy, did you read this: http://www.linuxquestions.org/questions/linux-software-2/how-to-record-pcsx… [12:50] <burek> dougle, let me check [12:51] <burek> dougle, x264 does not support that chroma format, so you might use filter to first change that and then to feed it to encoder [12:52] <dougle> can you give me an example of such a filter [12:53] <astropirate> burek, Unknown input format: 'pcm_s16le' [12:53] <burek> dougle, try asking in #ffmpeg-devel, it's kinda advanced usage :) [12:54] <burek> it should go something like ffmpeg -i input -vf "some_filter_for_chroma_conversion" -vcodec libx264 ... [12:54] <burek> astropirate, just a asec [12:55] <dougle> will do thanks burek, am currently reading the wiki [12:56] <burek> astropirate can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [12:56] <burek> dougle :beer: :) [12:58] <burek> oh pcm_s16le is a codec, not a format.. silly me :) [12:58] <astropirate> :) [12:59] <burek> astropirate, so, "-f u16le -acodec pcm_s16le -i -" [12:59] <astropirate> so, where will ffmpeg output? [12:59] <burek> or -f s16le [12:59] <burek> im not sure now [13:00] <burek> try this: some_script_that_produces_wav_output | ffmpeg -y -loop 1 -threads 0 -i input_image.png -f s16le -acodec pcm_s16le -i - ... video.mkv [13:01] <burek> where ... is all that -vcodec stuff and encoding options etc [13:02] <astropirate> http://pastie.org/4335828 [13:02] <astropirate> :S [13:04] <burek> you can't use -vcodec libx264 before -i [13:05] <burek> it is treated as an input option [13:05] <burek> so ffmpeg tries to read that -i as libx264 stream [13:05] <burek> the order of params does matter [13:05] <burek> ffmpeg <input options> -i input1 <input options> -i input2 ... <output options> output1 <output options> output2 .. [13:06] <astropirate> ahh [13:06] <burek> also, no point in using "-i -" if you dont send anything to ffmpeg through the pipe [13:06] <astropirate> i am going to do that [13:06] <burek> i.e. you need to have that gst before ffmpeg followed by a pipe [13:06] <astropirate> i was just tyring to get the command right [13:06] <burek> ok [13:08] <burek> gst ... | ffmpeg -y -f x11grab -r 25 -s 1280x720 -i :0 -f s16le -acodec pcm_s16le -i - -acodec flac -vcodec libx264 -preset:v ultrafast -threads 0 video.mkv [13:08] <burek> something like that [13:08] <burek> but why not using just: ffmpeg -f x11grab -r 25 -s 1280x720 -i :0 -f alsa -i hw:0 -acodec flac -vcodec libx264 -preset:v ultrafast -threads 0 video.mkv [13:09] <burek> like described here: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(scree… [13:09] <burek> and here http://ffmpeg.org/trac/ffmpeg/wiki/Capturing%20audio%20with%20FFmpeg%20and%… [13:12] <astropirate> burek, because there is alot of static with that recording [13:12] <astropirate> to get around it one has to go into alsamixer and set it to "record" [13:12] <astropirate> then it becomes clear [13:13] <astropirate> but i need to do everything programatically [13:20] <burek> well you can [13:20] <burek> there is amixer [13:20] <burek> cmd line tool [13:20] <burek> read its manual and see how to always first set your recording device [13:20] <burek> and then fire up ffmpeg [13:21] <burek> it's far easier than using gst and piping with ffmpeg and stuff [13:22] <astropirate> HAHAH this failed soo bad [13:24] <astropirate> (I'm working on an uber slick GUI front end for ffmpeg x11 capture and it looks so sexy you can lick it) [13:52] <astropirate> burek, do you know if it is possible to change the x11 screen grab area while it is running? [13:52] <astropirate> idk via sending messages? [13:52] <astropirate> IPC? :S [13:52] <astropirate> that would be killer [13:53] <burek> I think not [13:53] <burek> once the thread has been initialized and given those coordinates, it runs separately, so I guess not [13:53] <burek> you might ask for a feature if you like :) [13:53] <burek> or even better, provide a patch :) [13:54] <astropirate> hahah i'm not that leet :p [14:49] <sanfun> hi all. i'm having trouble compiling ffplay.exe and ffserver.exe on windows (mingw). is this expected? [15:00] <sanfun> after all i managed to compile them, but ffmpeg.exe crashes when it should HTTP-stream to ffserver_g.exe . [16:26] <fris> what would be the best variables for converting mp3 to ogg im currently just using -acodec libvorbis -ab 192k [16:55] <stu> I am working on a project that is using ffmpeg to encode mp4. If I understand correctly, ffmpeg compiles with vdpau support by default. I only see flags to disable it when configuring. I have a nvidia GT 520 and have the ubuntu nvidia-currrent package installed. However, I am not seeing any decreases in decoding times. Can anybody give me some hints? [16:57] <stu> *decreases in encoding times* [17:07] <microchip_> stu: ffmpeg doesn't do GPU encoding ;) [17:13] <microchip_> stu: vdpau is only used for decoding, eg ffh264vdpau decoder [17:14] <roxlu_> hi! [17:14] <microchip_> hi [17:14] <roxlu_> I've asked this question in #libav too, but didn't get info there.. I hope someone in here can help me a bit further with this... all info is very welcome [17:15] <roxlu_> I'm working on a project where I stream generated RGB24 images as h264/flv to a flash media server [17:15] <roxlu_> this works all fine! [17:15] <roxlu_> I want to add audio now [17:15] <roxlu_> I'm wondering what approach I need to take to add this audio to the stream? [17:22] <stu> microchip_: thanks [17:27] <rexbron> Hey, Does anyone have any advice for copying a mov timcode track during a transcode? [17:29] <burek> roxlu_ can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output? [17:30] <burek> rexbron -c copy? [17:30] <burek> or -map 0 [17:38] <roxlu_> burek: thanks, but I'm using the C library [17:38] <rexbron> burek: I can't get it to copy it. I can post the ffprobe if you would like [17:38] <roxlu_> burek: I'm happy to paste my code if you can help me with that [17:39] <rexbron> burek: http://pastebin.com/zGphuBxZ [17:39] <bwalker8981> Hi, I'm trying to install xuggler on top of ffmpeg, but the build keeps failing and saying "buildhelper.xml:1127: exec returned: 1". Does anyone know what I am doing wrong? Or does anyone have any experience with installing xuggler? [17:55] <rexbron> ok figured it out [17:55] <rexbron> needed to add a -dcodec copy and -map [18:06] <maker> Hello, I was interested in the Summer of Code in Space program, in particoular to the libavfilter project. Is saste, or any other backup mentor going to join before the deadline? [18:34] <burek> rexbron it seems its unsupported.. [18:34] <burek> roxlu_ did you check ffmpeg source code to see how it does that? [18:35] <burek> so you can copy/paste the logic? [18:35] <burek> maker, ask in #ffmpeg-devel [18:35] <burek> bwalker8981, ask in xuggler support channel? [18:39] <bwalker8981> burek: the #xuggler chat room has been empty for a very long time (I've never actually seen anyone in the chat), so I was hoping the ffmpeg group might have some advice [18:41] <maker> burek: ty [18:41] <burek> bwalker8981, ffmpeg has got nothing to do with xuggler [18:42] <burek> the error you are trying to solve is xuggler's error, not ffmpeg's [18:43] <burek> btw, why are you not using JNI to connect your java with ffmpeg, why using xuggler? [18:44] <bwalker8981> burek: Xuggler uses FFmpeg to compress and uncompress media, and is built on top of it. I agree though, this is a xuggler error. There is very little xuggler documentation out there, so I'm trying to find someone who can help. [18:45] <bwalker8981> burek: my goal is to get a rtmp stream sent to red5 media server, and I have only found that xuggler built on top of ffmpeg can do it. Do you know of another program that can send rtmp streams to red5? [18:45] <burek> why do you use xuggler? [18:45] <burek> why not ffmpeg alone? [18:46] <bwalker8981> burek: i did not think that ffmpeg had native rtmp support. Does it? [18:46] <burek> of course it does [18:46] <burek> at least through librtmp [18:46] <burek> or something [18:47] <burek> download latest ffmpeg source code and type ./configure --help [18:47] <burek> search for rtmp [18:47] <bwalker8981> burek: k ill try it now [18:47] <burek> --enable-librtmp enable RTMP[E] support via librtmp [18:47] <burek> it does [18:47] <burek> are you on linux/win? [18:49] <bwalker8981> burek: im on rhel [18:50] <burek> well i never used it, but on debian/ubuntu you would just apt-get install librtmp-dev [18:50] <burek> i dont know how do you do that on rhel [18:50] <burek> when you install librtmp-dev, then compile ffmpeg with --enable-rtmp and you have it all [18:52] <bwalker8981> burek: k, I already installed ffmpeg via yum, does that mean i need to uninstall it and then compile after installing lirtmp-dev? [18:53] <burek> well, check first [18:53] <burek> type ffmpeg -formats | grep rtmp [18:53] <bwalker8981> burek: k [18:53] <burek> no, ffmpeg -codecs rtmp [18:53] <burek> o man [18:53] <burek> ffmpeg -codecs | grep -i rtmp [18:53] <burek> that :) [18:54] <bwalker8981> burek: okay, ill try that, im wating for the install on librtmp [18:56] <bwalker8981> burek: here is the output of the ffmpeg codecs: http://pastebin.com/kFWubaAu, does this look correct? [19:01] <burek> bwalker8981 uninstall it [19:01] <bwalker8981> burek: k [19:01] <burek> and compile ffmpeg, after you install librtmp-dev [19:38] <bwalker8981> burek: sorry this is taking so long, im having trouble installing git on rhel, so i haven't gotten the latest ffmpeg version yet [19:44] <delicado> hi what functions can i use to make rewind a video decoding that reached an EOF. ? [19:44] <delicado> hi what functions can i use to rewind a video decoding that reached an EOF. ? [19:58] <delicado> how many here are chinese? [20:11] <bwalker8981> burek: I don't understand how to run the ./configure command with the correct options. It tells me the options are not correct. see http://pastebin.com/Cc7LzviW What am I doing wrong? [20:12] <burek> why on earth would you enable all that bwalker8981? [20:12] <microchip_> Unknown option "--enable-liba52" [20:13] <burek> and the error itself is pretty self-explanatory [20:13] <bwalker8981> burek: ha, well I guess I don't know what I'm supposed to enable, and I found instructions online saying to do that [20:13] <bwalker8981> burek: k ill try with just the rtmp enable [20:13] <burek> what exactly do you want to do with ffmpeg? [20:13] <burek> what is the use case [20:14] <bwalker8981> burek: create a live video stream from my computer to a red5 media server. [20:14] <burek> live video encoded with what codec? [20:14] <burek> and what codec to use for audio? [20:14] <bwalker8981> burek: rtmp [20:14] <bwalker8981> burek: flv [20:14] <burek> ok, what about audio? [20:16] <bwalker8981> burek: doesn't matter I guess? Whatever is easiest. The video is more important. we are trying to create an autonomous blimp that flies around our building, with a live video stream onboard. [20:16] <burek> ok [20:16] <burek> then you only need librtmp [20:16] <burek> so [20:16] <burek> ./configure --enable-librtmp [20:16] <burek> and that's all [20:17] <burek> after that, if everything goes ok, do: make [20:17] <burek> and then: make install [20:17] <bwalker8981> it says that it's an unknown option again... hmmm [20:17] <burek> librtmp? [20:17] <bwalker8981> oh sorry, i did rtmp [20:17] <burek> are you doing the configure in the ffmpeg source directory [20:18] <bwalker8981> now it didn't complain about the enable librtmp, but is said that yasm is not found [20:18] <burek> yasm is small [20:18] <burek> and powerful [20:18] <burek> so it would be wise [20:18] <burek> to download the source code [20:18] <burek> and just type: ./configure && make && make install [20:18] <burek> it will finish in less than a minute [20:19] <bwalker8981> k, do you know where to get it? [20:20] <bwalker8981> nvm found it [20:24] <bwalker8981> does yasm need to be installed within the ffmpeg directory? I installed it in my home directory, and adjusted $PATH, but the ./configure still can't find yasm for some reason [20:25] <burek> ? [20:25] <burek> you don't have to do that [20:25] <burek> I don't know how did you install it [20:26] <burek> but if you compile it, it works [20:26] <bwalker8981> http://pastebin.com/kjHnbBHa [20:26] <burek> also, sometimes you need ldconfig [20:26] <burek> are you root [20:26] <burek> to do make install [20:28] <bwalker8981> burek: yes, I'm root, and the make install didn't show any errors i don't think [20:29] <burek> then just ldconfig [20:29] <burek> and continue with ffmpeg [20:40] <bwalker8981> burek: the configure worked now, i'm waiting on make to finish. [20:43] <burek> ok [20:53] <bwalker8981> burek: thanks. I've got it installed now, and it saws it supports rtmp. now i just need to figure out how to send a stream to red5... [20:54] <burek> what is your input [20:54] <burek> where from do you take it [20:55] <bwalker8981> burek: im not really sure where to start now. i have a webcam, but i have no idea how to get ffmpeg to take the video from the webcam... [20:56] <burek> search for grabbing/capture [20:56] <bwalker8981> burek: fflogger: k ill take a look now [21:24] <mikmu> Hey there, I'm capturing a stream from an ip camera and I want to stream it out via http via ffserver or vlc. I've tried both ffserver and vlc and the problem I'm having is that the client player always waits for the complete file to finish downloading before starting playback, instead of progressive download streaming [21:25] <mikmu> I'm transcoding to h264, are there any sort of headers or keyframes, etc that I have to add? Both programs are complaining that the stream does not provide global headers but that my output container requires them. Is this the issue? [21:40] <bwalker8981> burek: so i've run into a problem, i realized that my webcam was connected to my ubuntu computer (oops) so now im trying to install librtmp-dev on ubuntu. However, apt-get install librtmp-dev fails because it can't find the package. Do you know what to do? [21:44] <relaxed> bwalker8981: apt-cache search librtmp [21:45] <bwalker8981> relaxed: that pulls up librtmp0, but not librtmp-dev. I am trying to get rtmp support for ffmpeg on ubuntu now, and i thought i needed librtmp-dev installed. Wil librtmp0 also work? [21:51] <burek> librtmp-dev - toolkit for RTMP streams (development files) [21:51] <burek> you're doing something wrong bwalker8981 :) [21:52] <burek> try apt-cache search rtmp | grep -i dev [21:52] <bwalker8981> burek: lol yeah i know. that command produced no results [21:53] <bwalker8981> burek: synaptic package manager also cannot find librtmp-dev [21:53] <burek> which version of ubuntu are you using [21:53] <bwalker8981> burek: 10.04 [21:53] <burek> hm, ok [21:53] <burek> then you need to compile it from source :D [21:54] <burek> or update your ubuntu to 11.x [21:55] <bwalker8981> burek: hmmm... compiling from source would probably be an easier option. I tried installing it from source, but I just kept running into a million dependency issues [22:00] <bwalker8981> burek: im trying again... [22:01] <burek> ok [23:02] <Snaggle> Is the fate server down? [23:05] <saste> Snaggle: so it seems... [23:17] <bwalker8981> burek: it looks like 10.04 does not have multiarch support (for librtmp0), so I need to upgrade ubuntu. Just curious, have you ever done this before? Should I be worried about losing programs if I upgrade to 12.04? I've never done it before [23:27] <d4forze> ffmpeg > libav [00:00] --- Fri Jul 27 2012

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