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April 2014
- 1 participants
- 60 discussions
[00:04] <ubitux> :)
[00:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:818aaa76363c: avcodec/libopencore-amr: use av_mallocz_array()
[00:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:8ff061095f15: avcodec/libzvbi-teletextdec: remove * 1
[00:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:2daf580284d0: avcodec/j2kenc: use av_malloc(z)_array()
[02:03] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:7b2b06eb43d2: swscale/utils: dont print message about filter size when the cause is a malloc failure
[02:03] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:1752b1459d09: swscale/utils: scale filter_size in warning so that it can be used as max for the compile time define
[02:25] <cone-291> ffmpeg.git 03Paul B Mahol 07master:718907cd881a: libtwolame MP2 encoding support
[02:25] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:6c019f05ed53: Merge commit '718907cd881a0b593264aed059c0e00da13f9e15'
[03:18] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:46439e156219: mp2: match twolame default options
[03:18] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:8b17243d1742: lavu: Add packed YVYU422 pixel format
[03:18] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:352e13e2a991: Merge commit '46439e156219d27f059cf687743ba5aacf238b87'
[03:18] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:0a1cc04e6c09: Merge commit '8b17243d1742279bcb2368f8d325ea71e66736c8'
[03:42] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:92b099daf4b8: swscale: support converting YVYU422 pixel format
[03:42] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:ed962414bd66: Merge commit '92b099daf4b8ef93513e38b43899cb8458a2fde3'
[04:13] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:b6a09ef70078: rawvideo: Support decoding YVYU422 FourCC
[04:14] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:233dbee3bc98: Merge commit 'b6a09ef700786b4e1e871dfa0f5c386ea43bbd0a'
[04:19] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:59388dac69f9: fate: update tests for YVYU422 pixel format
[04:20] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:e40d92b1e48d: Merge commit '59388dac69f9a79eddb003d1c6aad5ec89d4287b'
[04:33] <cone-291> ffmpeg.git 03Luca Barbato 07master:be59675add2a: doc: Change the multitable rendering in texi2pod
[04:33] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:8f23045b169e: Merge commit 'be59675add2a05a7400b1abf86201ab069d7b1dd'
[04:51] <cone-291> ffmpeg.git 03Luca Barbato 07master:0d671ade9905: doc: Use a 3 columns multitable
[04:51] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:43935842bd13: Merge commit '0d671ade99059fd47b155df85c10c35eb6d54878'
[05:13] <cone-291> ffmpeg.git 03Luca Barbato 07master:e6e8be542b8e: doc: Completely document the x264 options
[05:13] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:624327988f74: Merge remote-tracking branch 'qatar/master'
[07:37] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:d98688efb1a5: swscale/input: replace assert() by av_assert1()
[07:37] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:9d66aa2c8fa6: avcodec/mpegaudioenc_template: dont crash on bitrate=0
[07:37] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:44841528c197: avcodec/mpegaudioenc_template: reorder operations to prevent integer overflow
[09:39] <tmartiro> hi guys
[09:40] <tmartiro> I manage to compile ffmpeg.c into libffmpeg.so
[09:40] <tmartiro> thanks everybody
[10:07] <BtbN> tmartiro, well, that library does not sound too usefull
[10:24] <tmartiro> BtbN, maybe you are right. But for me it is
[10:24] <BtbN> compiling it into a library will not export anything usefull, what do you want to do with it?
[13:36] <cone-291> ffmpeg.git 03Thilo Borgmann 07master:c740e083bc46: configure: Fail on missing prototypes in check_oc.
[14:19] <cone-291> ffmpeg.git 03Vittorio Giovara 07master:60fd7d36c47d: fate: correctly set sample rate for mp2 tests
[14:19] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:60ef0c6f9097: Merge commit '60fd7d36c47d62d4c603bf16c213b1a924f5cfcf'
[14:31] <J_Darnley> 1987MwZLP4tsm29rrr6F
[14:31] <JEEB> my condolences
[14:32] <J_Darnley> WTF is that?!
[14:33] <J_Darnley> oh shit you idit password manager
[14:48] <iive> first openssl leaks private keys, now this...
[14:54] <J_Darnley> Yes, the openssl issue is insignicant to the error I just encountered!
[14:54] <J_Darnley> :)
[15:20] <Compn> J_Darnley : no special characters? for shame.
[15:20] <Compn> or non-alpha i mean
[15:21] <Compn> J_Darnley : btw this channel is logged onto the mailing list, so its there forever now
[15:30] <cone-291> ffmpeg.git 03wm4 07master:f6774f905fb3: mpegvideo: operate with pointers to AVFrames instead of whole structs
[15:30] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:59a53842d360: Merge remote-tracking branch 'qatar/master'
[16:23] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:2ca553ed9cbe: avcodec/snowenc: remove operations that have become redundant with AVFrame pointers
[16:23] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:acf112ba6e3d: avcodec/mpegvideo: remove AVFrame padding, as the hack isnt needed anymore
[16:47] <J_Darnley> Compn: No the stupid website only allows A-Z0-9
[16:47] <J_Darnley> Not even case sensitive
[16:48] <J_Darnley> I changed the password starigh away when I realised
[16:48] <J_Darnley> *straight
[17:23] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:dddacf780f1e: avcodec/sanm: use named consant instead of literal number
[17:23] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:7ac2f7e41305: avutil/log: factor "level != AV_LOG_INFO/8" checks out
[19:48] <cone-291> ffmpeg.git 03Anthoine Bourgeois 07master:d8ab7f31dd81: avcodec/dirac_arith: Fix build with PIC and stack-check options
[23:18] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:375013e3d64c: avcodec/mpeg12dec: drop confusing av_dlog()
[23:18] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:020600eed13f: avcodec/jpeglsdec: ask for sample for "oversized image"
[23:18] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:2773ab36cc64: avcodec/jpeglsdec/ff_jpegls_decode_lse: check len field validity
[00:00] --- Thu Apr 10 2014
1
0
[01:10] <synth_> is there a way to add a watermark to a file i'm generating? if so, how would I alter my current command http://pastebin.com/Bs8vUykw ?
[01:59] <klaxa> synth_: see the overlay filter and/or google for: ffmpeg watermark
[01:59] <klaxa> there are lots of guides
[02:18] <synth_> yeah i tried a couple but i couldn't figure out how to do a watermark + text
[02:22] <Moussekateer> hi guys. Can someone help me with this issue http://pastie.org/pastes/9009983/text?key=cn8uvmwuelt3laypinhpyq
[02:23] <Moussekateer> I see lots of google results for it but all the solutions seem to be of the form 'keep trying different encoding bitrates'
[02:25] <Moussekateer> the scripts I'm using have ffmpeg as a dependancy, so it would be nice to find a solution that doesn't require me to rewrite everything
[02:31] <c_14> Can you try adding a -loglevel debug?
[02:32] <synth_> is it possible to run multiple -vf commands?
[02:33] <c_14> Just separate them with commas, or use a -filter_complex
[02:33] <c_14> Or read the guide: https://trac.ffmpeg.org/wiki/FilteringGuide
[02:46] <Moussekateer> c_14: http://pastie.org/pastes/9011364/text?key=e4zt0p92kfcjpvu5inrlg
[02:48] <klaxa> >[mp4 @ 00000000057d87e0] pts (6016) < dts (6704) in stream 0
[02:48] <klaxa> your input is broken
[02:49] <synth_> c_14 http://pastebin.com/7z9xVQXf i'm getting a "[AVFilterGraph @ 0x2ed40e0] Too many inputs specified for the "movie" filter." error
[02:50] <Moussekateer> klaxa: Broken in what sense?
[02:51] <Moussekateer> I'm not sure exactly what that statement means
[02:51] <Moussekateer> the pts < dts one
[02:51] <klaxa> ah wait a sec...
[02:51] <klaxa> the... output is broken?
[02:52] <klaxa> can you remux to mkv? just to see if it's the container
[02:52] <klaxa> dts means decoding timestamp, pts means presentation timestamp
[02:52] <klaxa> in that case the presentation timestamp is smaller than the decoding timestamp
[02:52] <klaxa> meaning the frame would have to be displayed before it has been decoded which is impossible
[02:54] <c_14> synth_: I'm pretty sure your quoting is off somewhere.
[02:54] <Moussekateer> klaxa: Same issue http://pastie.org/pastes/9011775/text?key=uwyazobx1auf5rabrykyq
[02:55] <klaxa> can you play back the file without errors?
[02:57] <Moussekateer> klaxa: The original? Yes, it plays fine
[02:57] <synth_> i'm confused now
[02:57] <Moussekateer> seeking and all
[02:58] <klaxa> that is weird
[03:01] <synth_> http://ffmpeg.org/ffmpeg-filters.html#drawtext-1 for drawtext shows -vf drawtext="params here"; but if i wanted to also do movie params; other sites are showing me to do -vf "movie=params here" (http://www.idude.net/index.php/how-to-watermark-a-video-using-ffmpeg/) is that incorrect?
[03:01] <synth_> i'm using their code for placing the watermark on the bottom right
[03:02] <synth_> like: -vf drawtext="draw text params here","movie=watermark params here"
[03:02] <synth_> since you guys said i could separate the -vf commands with a comma
[03:03] <c_14> The problem is that your first "" pair is quoting 1 filter. Your second "" pair is quoting the movie filter _and_ some other stuff.
[03:07] <c_14> Try this: http://sprunge.us/SFCe
[03:07] <synth_> i got the movie filter from: http://www.idude.net/index.php/how-to-watermark-a-video-using-ffmpeg/ under the "Bottom right corner"; is his example incorrect? i basically just copied that and repathed to my watermark image
[03:07] <c_14> I don't see it doing anything with the video I tested it on, but at least it didn't error.
[03:08] <synth_> ok one second
[03:08] <c_14> synth_: You can either quote each filter individually or the whole filterblock at once, he quoted the whole block, while you quoted the first filter individually _and_ quoted the rest as a block.
[03:09] <c_14> You might need to get rid of either the first or the second [in], I'm not sure there are supposed to be two of them.
[03:10] <synth_> still getting an error using your example
[03:10] <synth_> i tried quoting the whole filterblock for the first as well
[03:11] <c_14> Try this one: http://sprunge.us/ibJL
[03:11] <c_14> And if that doesn't work, pastebin the result so I can see the error msg.
[03:15] <synth_> it worked!
[03:15] Action: synth_ flails
[03:18] <synth_> thank you so much
[03:18] <c_14> No problem.
[03:18] <synth_> i owe you a beer
[03:33] <synth_> is there a way to show the file frame number versus the frame number starting from 0? (i'm using: Frame: %{n})
[03:34] <synth_> sometimes we start our frame ranges at other numbers like 1001 instead of 0001
[03:40] <klaxa> synth_: see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
[03:40] <klaxa> >start_number The starting frame number for the n/frame_num variable. The default value is "0".
[03:40] <synth_> ahh okay
[03:41] <synth_> thank you!
[03:41] <klaxa> incidentally i used the drawtext filter today too :)
[03:42] <klaxa> it's easier to draw frame count/pts with drawtext than to write a script to generate a subtitle file
[03:48] <Moussekateer> klaxa: Is there any fix I can carry out then, other than resorting to other software?
[03:49] <klaxa> none that i know of
[03:56] <Moussekateer> mboo
[06:41] <orgy> hello?
[06:42] <orgy> i want to ask
[07:42] <llogan> synth_: you don't need movie filter for overlays
[07:49] <ScottSteiner_> klaxa: I tried using that guide to hardcode the subs, but it didnt work
[16:08] <quup> How do I resize a movie to fixed height and width by first scaling to fit the height and then cropping sides in one go?
[16:08] <quup> without any stretching that is
[16:09] <klaxa|work> quup: use the scale and then the crop video filter
[16:11] <klaxa|work> i.e.: -vf scale=-1:480,crop=w=640
[16:11] <quup> klaxa|work: so like -vf scale=-1:480,crop=w=480
[16:11] <quup> :D thanks!
[16:11] <klaxa|work> i think that should work, i have not tested it
[16:11] <quup> I'll give it a try
[16:15] <quup> klaxa|work: works great, thanks
[16:17] <klaxa|work> if you want to crop centered you will have to do some math though
[16:18] <klaxa|work> like (i_w-400)/2 for the left side and i_w-((i_w-400)/2) for the right side
[16:18] <klaxa|work> maybe
[17:10] <ronlut> Hey guys, I'm building an android application, and I need to use some ffmpeg commands. I've got a source video and I need to edit it: 1) Cut only the last 7 seconds. 2) Add a watermark(logo) on the top left side of the video.(00:00-00:07) 3) Add a video inside the source video on the bottom right side of the video.(00:00-00:07) 4) Add another video to the edited video from steps 1-3 (00:07-00:10) Can I do it in one ffmpeg command?
[17:10] <ronlut> The commands I used so far are: http://pastebin.com/nPuuuHh7 But I have a problem with it: I hear the sound of the video in step 3 instead of the main video. Thanks a lot!
[17:26] <deni> .close
[19:02] <synth_> is it possible when using the movie= -vf filter to add a semitransparent png overlay to a video? it looks like the transparent png i'm using is fully opaque and not antialiased around the edges
[19:24] <llogan> synth_: why are you using movie?
[19:35] <synth_> c_14 helped me set it up lsat night
[19:35] <synth_> llogan
[19:36] <synth_> yes i know i was just preparing it for you
[19:36] <synth_> everything works fine
[19:36] <synth_> llogan http://pastebin.com/kn4D2Y2s
[19:37] <synth_> no errors
[19:37] <llogan> you forgot the console output
[19:42] <synth_> llogan http://pastebin.com/dfUZ6EpF
[19:42] <llogan> thanks
[19:42] <llogan> can you provide your png file?
[19:43] <synth_> the watermark png file?
[19:43] <llogan> yes
[19:44] <llogan> you have two instances of "-s 1280x720" outside of your filtergraph. what is the purpose of the "-s 1280x720"?
[19:44] <synth_> http://mellow.me/tmp/bigblock_watermark.png
[19:44] <synth_> i thought the first -s was the input resolution and last -s was the output resolution
[19:46] <synth_> example of the output mov file it generated: http://mellow.me/tmp/Club_Fracture_V4_.mov
[19:47] <llogan> your main input is already 1280x720 and the output will inherit that same size
[19:47] <synth_> i still need to implement the start_number parameter to get the frame number part working
[19:47] <synth_> oh okay
[19:47] <synth_> i'll remove the second one then
[19:47] <llogan> you don't need either of them
[19:48] <synth_> so if the input was 1080, and someone wanted to render out a 720, could i implement them?
[19:48] <synth_> or would the source frames need to be 720
[19:49] <llogan> you can scale any input. i'll make an example once i finish something
[19:49] <synth_> ok
[19:49] <synth_> thanks man
[19:50] <llogan> also, you can use the glob pattern instead of -start_number. might be easier for you
[19:50] <llogan> example: http://ffmpeg.org/ffmpeg-formats.html#Examples-3
[19:51] <synth_> okay i didn't know about that
[19:51] <synth_> thanks
[19:55] <llogan> -vendor is ignored because it's a provate option for the prores_ks encoder only, AFAIK
[19:56] <llogan> *private
[20:04] <llogan> synth_: http://pastebin.com/XpPVjkT5
[20:04] <llogan> i'm not sure if you needed that probesize
[20:04] <llogan> the scale in the first filterchain shows that you can scale any input at any time in the filtergraph
[20:05] <llogan> the overlay filter needed its format option because by default it uses yuv420: http://ffmpeg.org/ffmpeg-filters.html#overlay
[20:06] <llogan> (it is not the same thing as the format filter used at the end)
[20:07] <llogan> but you won't need "scale=1280:-1,". it's just there as an example
[20:25] <synth_> i was reading somewhere that probesize is good for frame by frame scrubbing using arrow keys
[20:27] <llogan> it's used to "probe" the input to get stream information
[20:28] <llogan> what you're probably referring to is "-g", but i don't think it does anything with mjpeg
[20:30] <synth_> i'm not sure, it's on https://trac.ffmpeg.org/wiki/vfxEncodingGuide
[20:30] <synth_> for vfx related ffmpeg renders
[20:31] <llogan> that's using an encoder that uses "group of pictures", but mjpeg is just a series of images, and therefore does not need, nor can not use -g
[20:32] <llogan> i suppose i should read that verbose wiki page someday
[20:32] <llogan> so many words
[20:32] <synth_> yeah i've just been skimming over it
[20:35] <llogan> did my example fix the shitty looking png overlay?
[20:37] <synth_> let me see if i can restructure my code without breaking anything...
[20:37] <synth_> the pattern glob portion isn't going to be easy to fix
[20:37] <synth_> so i'll have to leave that part out
[20:38] <synth_> we have some renders that have very identical names but are different passes all dumped into one directory
[20:38] <synth_> s/very/almost
[20:40] <llogan> you can still use glob: -i "foo*.png"
[20:57] <brontosaurusrex> hi ~
[20:58] <synth_> llogan i think your example has an issue with quotes
[21:00] <synth_> llogan this is what i'm getting: http://pastebin.com/PzDfhK9j
[21:00] <synth_> i refactored your code to fit into what i already had and it looks like encapsulating the quotes within quotes for the -filter_complex for the draw_text parameter messed things up
[21:11] <llogan> synth_: s/drawtext="fontfile/drawtext=fontfile and s/10"[bg]/10[bg]
[21:16] <synth_> llogan new error: http://pastebin.com/sG7D14au
[21:22] <llogan> synth_: i forgot to add -map "[out]" in the example
[21:22] <llogan> as an output option
[21:22] <synth_> oh okay, does that come after the -filter_complex ?
[21:25] <synth_> wait nevermind
[21:25] <llogan> you can probably place it anywhere as long as it is an output option, but for humans it would make more sense to place it after the filtering
[21:26] <synth_> ok so just rendered out another test and the png doesn't look any better, same as before.
[21:27] <llogan> you could remove [out] from the filtergraph and then omit -map "[out]", but i prefer to explicitly llabel things so i know exactly what is going on instead of relying on defaults.
[21:27] <synth_> wait a second
[21:27] <synth_> it looks better as long as black isn't behind the png
[21:27] <synth_> let me upload an example...
[21:30] <synth_> llogan : http://mellow.me/tmp/3000_0010_lgt_scenestate1_v0001.mov so the logo now is jagged over black, but as soon as color starts coming in, it looks as expected
[21:31] <llogan> weird
[21:31] <llogan> i can't duplicate the issue, but i don't have similar inputs
[21:32] <synth_> i could upload the frames, and watermark if you'd like?
[21:32] <llogan> what is the file size for each typical frame?
[21:33] <synth_> can range from a 2-10mb to 200-300mb per frame
[21:34] <synth_> these frames max around 9-10mb
[21:35] <llogan> i guess the first ~48 will suffice
[21:35] <llogan> maybe even just ~38-48
[21:36] <synth_> ah crap sorry just uploaded all of the frames lol
[21:37] <synth_> ~340mb
[21:37] <synth_> want me to redo it?
[21:37] <synth_> er ~370mb
[21:38] <llogan> no, i'll download it while i'm taking my break
[21:40] <synth_> ok http://mellow.me/tmp/9999_testproject_3000_0010_lgt_cam003_v0001__frames.zip
[21:40] <synth_> do you work for ffmpeg?
[21:43] <llogan> FFmpeg is made up of volunteers. i work for a small media business where i use ffmpeg very often.
[21:44] <llogan> i'll be back in about an hour
[21:48] <synth_> okay
[22:40] <ronlut> Hey guys, I'm building an android application, and I need to use some ffmpeg commands. I've got a source video and I need to edit it: 1) Cut only the last 7 seconds. 2) Add a watermark(logo) on the top left side of the video.(00:00-00:07) 3) Add a video inside the source video on the bottom right side of the video.(00:00-00:07) 4) Add another video to the edited video from steps 1-3 (00:07-00:10) Can I do it in one ffmpeg command?
[22:40] <ronlut> The commands I used so far are: http://pastebin.com/nPuuuHh7 But I have a problem with it: I hear the sound of the video in step 3 instead of the main video. Thanks a lot!
[22:56] <groundnuty> hey, I'm having trouble with ffmpeg ffmpeg -pattern_type glob -i '*.png' $MOVIE
[22:56] <groundnuty> it takes files like ls would print them
[22:56] <groundnuty> and I would pref it would take them as ls -v would print thme
[22:56] <groundnuty> I would not like to reneme them all
[22:57] <groundnuty> i a concat option an answer here?
[22:57] <quup> groundnuty: what's the difference between ls and ls -v in this case?
[22:57] <quup> oh numerical sort
[22:59] <quup> groundnuty: if you replace "-pattern_type glob -i '*.png'" with just -i '%d.png' does that give correct order?
[22:59] <groundnuty> quup: no, the numbers are crazy
[23:00] <groundnuty> quup: tried play with %d
[23:00] <groundnuty> but they are like prefix.timestamp.soemthing
[23:00] <groundnuty> prefix.timestamp.soemthing.png ;p
[23:00] <groundnuty> I thought that glob worked till prfix was in range of 0-9
[23:01] <groundnuty> when 10 apeared it was taken after 1
[23:01] <quup> groundnuty: maybe there is an option to read input filenames from a file
[23:01] <quup> so you could just ls -v > input_files.txt and then ffmpeg --magic-thingy=input_files.txt
[23:01] <sacarasc> groundnuty learns the importance of leading zeros.
[23:02] <groundnuty> sacarasc: I wont comment on those filenames - not my program generates them :(
[23:03] <groundnuty> I dont see the option that would read input files from external file
[23:03] <groundnuty> sacarasc, quup am I missign something?
[23:04] <quup> groundnuty: don't think so, can't find any such option either
[23:05] <quup> I guess I shouldn't be too surprised ffmpeg doesn't have a -playlist option :D
[23:06] <quup> groundnuty: oh! it exists, -f concat -i thefilenamesinthisone.txt
[23:06] <quup> groundnuty: http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%20…
[23:06] <groundnuty> quup: I'm reading about it
[23:06] <groundnuty> I'm just not sure because its all about wav
[23:06] <groundnuty> in my case I have png and output mp4
[23:07] <quup> it's only getting the input files in order that matters
[23:07] <quup> rest is just throwing them at the right filters
[23:08] <quup> or maybe I missunderstand concat...
[23:08] <quup> oh well
[23:32] <llogan> synth_: try changing format in overlay from rgb to yuv422. also that maked the last format filter redundant.
[23:33] <llogan> *makes
[23:34] <llogan> ronlut: without the complete console output helping you will be challenging
[23:41] <synth_> llogan: cool! that worked!
[23:41] <synth_> thank you!
[00:00] --- Thu Apr 10 2014
1
0
[00:37] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:f9158b01d0f3: swresample/resample: Limit filter length
[00:38] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:a5290cb1ac04: swresample/dither: use av_malloc_array()
[00:38] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:5027f39712fd: swresample/resample: use av_malloc_array() where appropriate
[02:45] <cone-616> ffmpeg.git 03Carl Eugen Hoyos 07master:7fcb4f6c51f4: Fix compilation with --disable-everything --enable-decoder=rawvideo.
[02:45] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:e36aaeca3a30: avfilter/vf_colormatrix: fix memleak on error
[02:45] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:b1763205457f: Merge remote-tracking branch 'cehoyos/master'
[14:39] <ubitux> is st->codec always set after avformat_{open_input,find_stream_info}?
[14:39] <ubitux> like.. even for unknown streams?
[14:51] <wm4> ubitux: isn't there an unknown codec id value for such things
[14:52] <ubitux> probably :)
[14:52] <wm4> do audio encoders accept an arbitrary frame size?
[14:52] <wm4> the transcoding.c example seems to imply that
[14:52] <nevcairiel> those that dont should be using the audio frame queue to buffer stuff
[14:52] <wm4> I thought you'd need to write a frame according to the frame size of the codec
[14:53] <wm4> ok
[14:53] <nevcairiel> ubitux: st->codec is always set, since its just the AVCodecContext, st->codec->codec may be NULL however
[14:54] <ubitux> ok
[15:08] <ubitux> wm4: "The code was basically ported from mpv's loadfile.c@929793be7"
[15:09] <ubitux> that's why it's the same list as mpv, i guess :p
[15:15] <tmartiro> Hi everybody
[15:16] <tmartiro> how can I compile ffmpeg.c as a library
[15:16] <tmartiro> and export the function main
[15:16] <nevcairiel> its not designed for this, so you can't
[15:17] <wm4> ubitux: he didn't keep the comment about the possibly bogus mime type, though
[15:17] <nevcairiel> why not simply call the ffmpeg executable instead
[15:17] <ubitux> wm4: ah, ok; i didnt compare, feel free to raise sticks
[15:19] <tmartiro> thanks
[15:53] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:9b467d3cd593: examples/transcoding: use av_malloc_array()
[15:53] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:c8a1f10cf486: avcodec/ac3enc_float: use av_malloc_array()
[15:53] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:53487268e555: cmdutils: use av_mallocz_array()
[16:24] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:e4c180c05ac9: avcodec/a64multienc: use av_malloc(z)_array()
[16:24] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:33c9bb943b67: avcodec/avpacket: use av_malloc_array()
[16:24] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:e20e854ca056: vcodec/bitstream: use av_malloc_array()
[16:47] <Daemon404> sws: filterSize 208 is too large, try less extreme scaling or increase MAX_FILTER_SIZE and recompile
[16:47] <Daemon404> sws: initFilter failed
[16:47] <Daemon404> what the shit?
[16:47] <nevcairiel> what kind of resize did you try?
[16:47] <Daemon404> 5000x5000 image to 1920x1920
[16:48] <ubitux> i opened a ticket for that a while ago
[16:48] <Daemon404> thing is
[16:48] <Daemon404> ffms2 uses swscale
[16:48] <Daemon404> and doesnt do this
[16:48] <Daemon404> maybe it uses less
[16:48] <Daemon404> "extreme" params
[16:48] <nevcairiel> might depends on the scaling algorithm selected as well
[16:49] <Daemon404> swsctx = sws_getCachedContext(NULL, ctx->yuv->width, ctx->yuv->height,
[16:49] <Daemon404> ctx->avctx->pix_fmt, ctx->dest->width,
[16:49] <Daemon404> ctx->dest->height, ctx->dest->format,
[16:49] <Daemon404> SWS_BICUBIC | SWS_FULL_CHR_H_INT |
[16:49] <Daemon404> SWS_FULL_CHR_H_INP | SWS_ACCURATE_RND |
[16:49] <Plorkyeran> or any of the 20 undocumented magic flags
[16:49] <Daemon404> SWS_BITEXACT, NULL, NULL, NULL);
[16:49] <ubitux> http://trac.ffmpeg.org/ticket/3170
[16:49] <Daemon404> i should remove bitexact
[16:49] <Daemon404> coworker put that there
[16:49] <nevcairiel> its probably because of full chroma or something :)
[16:49] <Daemon404> nevcairiel, the "dont look like shit" option?
[16:49] <Daemon404> ;)
[16:49] <Daemon404> anyway ffms2 uses it too
[16:50] <Daemon404> i see
[16:50] <Daemon404> so basically im fucked
[16:50] <Daemon404> any other image sclaing libs that are ok?
[16:51] <nevcairiel> could do what the error tells you to do
[16:51] <Daemon404> nevcairiel, the other issue is that there is no way ro pragmatically detect this
[16:51] <Daemon404> its printed om stderr
[16:51] <nevcairiel> increase the define and recompile :p
[16:52] <nevcairiel> thats just error reporting in C, there is never a real good way to know what exactly is going wrong
[16:52] <Daemon404> swscale is especially bad
[16:52] <nevcairiel> unless you read textual error messages smewhere
[16:52] <Plorkyeran> just gotta parse av_log output obv
[16:52] <Daemon404> it has no error code
[16:52] <Daemon404> period
[16:52] <Daemon404> just returns NULL
[16:52] <Daemon404> and pritns to stderr
[16:52] <nevcairiel> like EINVAL would tell you more
[16:53] <Daemon404> heh
[16:53] <Daemon404> does fmtconv have an api?
[16:53] <Daemon404> or any good resizing libs?
[16:54] <nevcairiel> weird thing is, my MAX_FILTER_SIZE in sws is 256, how is your 208 too big
[16:54] <Plorkyeran> fmtconv doesn't even have a non-vcxproj build system
[16:54] <nevcairiel> wonder if it was changed recently
[16:54] <Daemon404> Plorkyeran, olol
[16:54] <Daemon404> i thought gimp had some resizing lib
[16:54] <Daemon404> or was that just for colorspace conv
[16:54] <Plorkyeran> imagemagick obv :v:
[16:55] <Daemon404> imagemagick is just about the worst option
[16:55] <Daemon404> im better off writing my own bicubic resize
[16:56] <nevcairiel> oh its accurate rounding, that makes it fail
[16:56] <Plorkyeran> I'm confused by fmtconv
[16:56] <Plorkyeran> he put a ton of work into writing a high-quality scaler library
[16:56] <Plorkyeran> but won't put in the effort to make it actually usable
[16:57] <Daemon404> Plorkyeran, i know right?
[16:57] <Daemon404> i mean
[16:57] <Daemon404> in theory i COULD use the C api
[16:57] <Daemon404> ofr vs
[16:57] <Daemon404> of*
[16:58] <Daemon404> or implement a dumb loader
[16:58] <Daemon404> but thats retarded.
[16:58] <nevcairiel> strip the accurate_rnd flag, use sws, go grab a few beers to forget about the evil you d id
[16:58] <Plorkyeran> I'd be more inclined to just fork and add an API
[16:59] <Plorkyeran> and actually package it so other people can use it
[16:59] <Daemon404> Plorkyeran, yeah but nobody can mainta8n it except him
[16:59] <Daemon404> with his templated intrinsics mess
[16:59] <nevcairiel> another non-programmer building plugins?
[17:00] <nevcairiel> (which would also explain why no build system or api)
[17:00] <Plorkyeran> as long as you don't touch the actual conversion stuff and he doesn't decide to redesign everything then updating it with new code dumps wouldn't be an issue
[17:01] <Plorkyeran> real goal is to get it to a state where someone who could write it but doesn't want to starts using it
[17:01] <Plorkyeran> and then gets tricked into taking over maintence
[17:08] <Daemon404> lu
[17:08] <Daemon404> lul
[17:15] <Compn> Daemon404 : whats gstreamer use to scale
[17:18] <Daemon404> Compn, looks like their own thing.
[17:19] <Compn> nevcairiel : but all those books and classes said anyone could write a wrapper and make libraries! :P
[17:20] <Compn> Daemon404 : can look at how mplayer uses swscale bwahaha
[17:20] <Daemon404> same limitations
[17:20] <Compn> whats the problem again ?
[17:20] <Compn> i mean, your problem
[17:21] <Compn> oh it fails
[17:21] <Daemon404> swscale cnat handle large images.
[17:21] <Daemon404> with a high scaling ratio
[17:21] <Compn> what about increase max_filter_size ?
[17:21] <Daemon404> that's a hack
[17:21] <Compn> its an arbitrary limit
[17:21] <Compn> which i've fought against
[17:21] <Daemon404> surely it must be there for a reason
[17:21] <Daemon404> michaelni, ^
[17:21] <Compn> because it may or may not slow things down . not even sure anyone did a benchmark
[17:21] <nevcairiel> memory concerns
[17:21] <Compn> Daemon404 : NO, THERE IS NO REASON
[17:21] <Compn> i can assure you of this
[17:21] <Compn> 200%
[17:22] <Daemon404> nevcairiel, so i dont go an allocate 900gb?
[17:22] <Compn> no
[17:22] <nevcairiel> all it does is define the size of some pre-defiined optimization tables
[17:22] <Compn> memory concerns on your p233 mmx cpu
[17:22] <nevcairiel> which are now at 4xMAX_FILTER_SIZE
[17:22] <Compn> theres literally no reason for any limits in sws
[17:22] <Compn> :)
[17:22] <nevcairiel> ie 1024 elements :p
[17:22] <Daemon404> nevcairiel, im not sure what these "memory concerns" are
[17:22] <Daemon404> they dotn seem veyr valid
[17:23] <nevcairiel> sws is from the stone age
[17:23] <nevcairiel> memory was a concern
[17:23] <nevcairiel> :D
[17:23] <Daemon404> ... lol
[17:23] <Compn> p233 days we needed our memory
[17:23] <Compn> and we didnt have 5000x5000 video :P
[17:23] <Compn> so its time to update the limits...
[17:23] <j-b> 8096 plz
[17:23] <Compn> 14k imo
[17:24] <Daemon404> Compn, why have limits?
[17:24] <Daemon404> thats my question?
[17:24] <Compn> Daemon404 : that too
[17:24] <Compn> just because no one wants to fuzz test it
[17:24] <Daemon404> why should the library say i cant
[17:24] <Daemon404> im the damn api user!
[17:24] <Compn> Daemon404 : why not fuzz test it for us
[17:24] <Compn> remove limit, fuzz it
[17:24] <Compn> benchmarks too
[17:24] <Daemon404> why would it need fuzz testing for memory limtis
[17:24] <Daemon404> [16:24] <@Compn> benchmarks too <-- working > notworking
[17:24] <Compn> dont want to OOM ?
[17:24] <Daemon404> benchmark done,.
[17:24] <Compn> lol
[17:25] <Daemon404> [16:24] <@Compn> dont want to OOM ? <-- this is not the responsibiltiy of a resizing lib
[17:25] <Daemon404> its the user
[17:25] <Compn> well you dont want to slow down 1080 playback .... :P
[17:25] <Daemon404> if i wanna pass it 9000gb of rgba data
[17:25] <Daemon404> i will
[17:25] <Compn> Daemon404 : i'm in agreement with ya
[17:25] <Daemon404> Compn, why would it slow down existing stuff
[17:25] <Daemon404> it's a limit
[17:25] <Daemon404> i.e. upper bounds
[17:25] <Compn> because magic
[17:26] <Daemon404> or is this for some macro's magic tables
[17:26] <Daemon404> that with grow N times in size
[17:26] <Compn> could be. i dunno
[17:26] <Compn> it allocates some stuff
[17:26] <Compn> iirc
[17:26] <Daemon404> i see
[17:26] <Daemon404> so it's yet another swscale "design" feature
[17:26] <Compn> thats why the limit
[17:26] <Daemon404> i love those.
[17:27] <Compn> well you see we get complaints when our sizes (1024k per frame or something) are too big
[17:27] <Compn> or 1024b , i try to ignore such things
[17:27] <Daemon404> certainly a problem on embedded devices
[17:27] <Compn> true
[17:27] <Daemon404> but its 2014
[17:28] <Daemon404> and 4k is a thing
[17:28] <Compn> we are crippled by embedded of course
[17:28] <Compn> which is why i suggested runtime limits
[17:28] <Compn> if arm > cripple
[17:28] <Compn> if cluster > unlimit
[17:28] <Daemon404> i think it might be kind of hard to modify sws's api to set it
[17:29] <nevcairiel> hm i cant figure out how the filter size is even calculated, none of the calculations make sense to reach such a high value of 208 for such a downscale
[17:29] <nevcairiel> basically its filterSize = 1+ factor * srcW / dstW
[17:29] <nevcairiel> factor is 4 for bicubic
[17:31] <Compn> nevcairiel : would it make sense to have a better error message ? or just fix the limit ?
[17:32] <Daemon404> also
[17:32] <Daemon404> i dont forsee it getting "better" with teh advent of 4k
[17:32] <Daemon404> only much, much worse
[17:32] <Compn> limits ?
[17:32] <Daemon404> yes
[17:33] <Compn> or you mean resolution stopping from getting larger ?
[17:33] <Daemon404> swscale limits
[17:33] <Compn> obviously, we are on exponential resolution now
[17:33] <Compn> bigger screens means we will get bigger res
[17:33] <Compn> also raw video processing
[17:33] <Compn> but those guys just recompile and stay quiet
[17:34] <Compn> are you recompiling yet ?
[17:34] <Daemon404> i would rather not have to patch swscale
[17:35] <Plorkyeran> should at least be a configure option or something
[17:35] <Compn> submit patch to -devel with rfc
[17:35] <Compn> should get fixed within this month
[17:35] <Daemon404> Compn, well see
[17:35] <Daemon404> i want to hear michaelni's opniion
[17:35] <Daemon404> as it is his code
[17:36] <Compn> wonder if fabrice started the limit code
[17:37] <Compn> oh nevermind, only michael has copyright there
[17:37] <Compn> i had no idea :)
[17:40] <Daemon404> Plorkyeran, any idea just how much slower fmtconv is than swscale?
[17:40] <Plorkyeran> no
[17:40] <Plorkyeran> other than "probably a lot"
[17:40] <Daemon404> yeah
[17:41] <Plorkyeran> iirc it's significantly slower than avs's scalers
[17:41] <Daemon404> latency matters here
[17:41] <Daemon404> unfortunately
[17:42] <Compn> Daemon404 : that MAX_FILTER_SIZE was at least in code from 2003... and has been unchanged since
[17:42] <Compn> the 256 limit i mean
[17:42] <nevcairiel> i'm still not sure why you hit the limit with your downscale
[17:43] <Compn> 208 is less than 256
[17:43] <nevcairiel> i would probably have to step through the code to find out
[17:43] <nevcairiel> if you use accurate_rnd, the limit is reduced to 170
[17:43] <nevcairiel> why, i dont know
[17:43] <Compn> are you outputting to the right colorspace bt701 or whatnot ?
[17:43] <nevcairiel> but 208 seems way too high
[17:44] <nevcairiel> 208 pixels input for a 5000->1920 downscale?
[17:44] <nevcairiel> that would make more sense for a 5000->192 downscale
[17:46] <Daemon404> Compn, this is also doing a yuv420p->rgba conversion
[17:46] <nevcairiel> usually those that failed tried to create tiny thumbnails
[17:46] <Daemon404> but thats irrelevant
[17:46] <nevcairiel> ie 1920 -> 16
[17:47] <Daemon404> i have no idea
[17:48] <nevcairiel> wonder if there isnt a bug somewhere
[17:49] <Daemon404> nevcairiel, aha
[17:49] <Daemon404> i am doing a 100x100 thumb too
[17:49] <Daemon404> that triggers it
[17:49] <nevcairiel> that makes more sense
[17:50] <nevcairiel> you could strip accurate_rnd from the thumb, that gives you a bit more headroom on the scaling
[17:50] <nevcairiel> no solution of course
[17:50] <Daemon404> yea
[17:50] <Plorkyeran> maybe make the thumb from the downscaled version rather than the original?
[17:50] <nevcairiel> or that
[17:51] <Daemon404> thats still no solution
[17:51] <nevcairiel> a "solution" is to rewrite all those parts that rely on a maximum filter size
[17:52] <wm4> you still can hope for avscale
[17:52] Action: wm4 hides
[17:52] <nevcairiel> as if thats ever going to get finished
[17:52] <Plorkyeran> s/finished/started/
[17:52] <nevcairiel> starting stuff is easy
[17:53] <nevcairiel> but implementing a scaling library that can do any<->any like sws
[17:53] <Daemon404> nevcairiel, i am seriously considering using vapoutsynth's C api to load fmtconv
[17:53] <Daemon404> this is sad.
[17:53] <wm4> (except sws actually can't)
[17:54] <nevcairiel> wm4: its good enough, it can take nearly everything as input, and all of the things you care about as output..
[17:54] <nevcairiel> that it butchers chroma in the process or is just ugly on the inside is of no importance :p
[17:54] <Daemon404> (dithering sucks too)
[17:54] <wm4> btw. is anyone of the opinion that libavcodec should automatically use libswscale/libavresample, if the library user doesn't support a specific format?
[17:55] <Daemon404> i think swscale could also be less of a mess if it didnt have 900 pack/unpack formats
[17:55] <nevcairiel> i have never really bothered to check how well it dithers
[17:55] <nevcairiel> most of the directshow formats I support dont have a av pixel format entry, so i have to mangle them myself anyway
[17:57] <Daemon404> michaelni, do you oppose a runtime settable option?
[17:57] <Daemon404> api call or otherwise
[17:58] <wm4> IMO swscale should just handle everything, as long as the image sizes are valid
[17:58] <wm4> and memory allocation doesn't fail
[17:58] <wm4> I think swscale used to crash on these cases instead of failing on init (?)
[17:59] <nevcairiel> probably
[18:00] <wm4> because I remember complaining about this too
[18:01] <michaelni> Daemon404, if theres no real world speedloss and it works then sure runtime settable is better
[18:01] <Daemon404> michaelni, it would be set in an API call
[18:02] <Daemon404> well id rather swscale handle it itself
[18:02] <michaelni> it should not be needed to be set at all
[18:02] <Daemon404> but... thats not so fun
[18:02] <Daemon404> yeah
[18:02] <Daemon404> im not very familiar with that code myself
[18:02] <Daemon404> or what its used for
[18:03] <Daemon404> hmm
[18:04] <michaelni> also i think there are some offsets in the asm that need some change if you change the max filter size
[18:04] <Daemon404> oh lovely...
[18:08] <Daemon404> so basically its a LOT of work to move it to runtime settable
[18:08] <michaelni> yes i think so
[18:08] <Daemon404> hrm.
[18:09] <Daemon404> configure-time would be pretty easy (Config.asm and pals)
[18:09] <Daemon404> but thats sitlll ugly and bad
[18:09] <nevcairiel> if you can find all the spots which need changing in asm, it doesnt seem that terrible
[18:09] <nevcairiel> instead of an array in the struct, its just a pointer and needs an extra deref
[18:09] <Daemon404> nevcairiel, runtime settable means the asm has to take an new param
[18:10] <Daemon404> or new struct member
[18:10] <Daemon404> my asm sucks :)
[18:10] <Daemon404> especially ppc asm
[18:28] <Zeranoe> Will 2.2.1 support the newer x265 API? http://ffmpeg.zeranoe.com/blog/?p=279
[18:28] <nevcairiel> I think it was decided to not try to backport the weekly API fixes to the release branch
[18:30] <nevcairiel> libx265 API is changing way too fast right now
[18:30] <Daemon404> i could have avoided some
[18:30] <Daemon404> like Rodeo pointed out
[18:30] <Daemon404> by using x265_param_parse for e.g. SAR
[18:31] <nevcairiel> no reason the semantic of that isnt changing at some point, and thats a silent breakage then
[18:31] <nevcairiel> but sure
[18:31] <nevcairiel> but not all could be avoided, the recent header repeat thing not for example
[18:32] <Zeranoe> So 2.2.1 will use the same API as 2.2?
[18:33] <JEEB> yes
[18:34] <JEEB> basically you could consider pushing libx265 into an ffmpeg release being somewhat too quick (since you only get support for a single API during that release)
[18:34] <Zeranoe> I'm really wondering if x265 is worth all the trouble
[18:35] <JEEB> I'd just use the cli encoder at this point to be honest. Having support for the current API in master is OK, but it's not gonna be much useful in the release due to the amount of changes in the API
[18:36] <wm4> <Zeranoe> So 2.2.1 will use the same API as 2.2? <- the same ffmpeg API... the code itself could still be fixed to deal with API breaks of dependencies
[18:37] <JEEB> but then you'd break building that release branch with an older version
[18:37] <JEEB> which used to work
[18:37] <JEEB> I think a release should be relatively stable regarding those things
[18:39] <nevcairiel> wm4: for the fast moving x265 target, it was kinda decided to not bother
[18:40] <nevcairiel> because they are typically not backwards compatible
[18:40] <nevcairiel> so its not only a "break", but requires a new major version
[18:41] <srcreigh> Can someone help me over in #ffmpeg? I'm using ffmpeg libs and avpicture_free is segfaulting. Sorry to be offtopic
[18:48] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:f6759d9ad4a8: swscale/x86/swscale_template: loose hardcoded dstw_offset
[19:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:955d7e26b644: swscale/swscale_internal: make the offset strings update themselfs when the MAX_FILTER_SIZE is changed
[20:28] -:#ffmpeg-devel- [freenode-info] channel trolls and no channel staff around to help? please check with freenode support: http://freenode.net/faq.shtml#gettinghelp
[20:56] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:2fcef4a04470: swscale/swscale_internal: remove obsolete warning
[20:56] <cone-291> ffmpeg.git 03Schenk, Michael 07master:358f9ae6390a: configure: AVI demux needs exif
[23:07] <cone-291> ffmpeg.git 03Anshul Maheshwari 07master:5b30f2759267: rtspcodes: don't check for >0 on error codes macro
[23:49] <wm4> ubitux: why is there an ass and a subtitles filter?
[23:51] <ubitux> ass was there in the first place, and it has the benefit of not using lavf
[23:51] <ubitux> the content of the file is directly sent to libass
[23:51] <ubitux> iirc.
[23:52] <ubitux> ...so i guess it's less error prone, and recommended when you're using external ass subs
[23:53] <llogan> subtitles file name must be specified. would be a nice feature to allow users to specify a stream
[23:56] <ubitux> don't want to add subtitles support in lavfi instead?
[23:56] <wm4> this would require overcoming some bullshit
[23:56] <wm4> like thinking you need a generic catch-all data structures to represent formatted subtitles
[00:00] --- Wed Apr 9 2014
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[00:12] <llogan> ac_slater: you can make a MJPEG from individual jpg images without re-encoding
[00:14] <Stalkr_> I am sorry, I have already pasted it now though. Do you want me to use pastebin anyway?
[00:15] <llogan> sure
[00:16] <llogan> ac_slater: and you can do the opposite: extract images from the movie without re-encdoing
[00:16] <llogan> https://www.ffmpeg.org/ffmpeg-bitstream-filters.html#mjpeg2jpeg
[00:16] <Stalkr_> Playing with FFmpeg and webm, but it won't burn the subtitles, any idea why? http://pastie.org/pastes/9001844/text -- I followed https://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20vid…, which looks easy enough. What am I missing?
[00:16] <llogan> where is the rest of the info?
[00:22] <Stalkr_> Oh, sorry... I forgot that. I was able to find out what's causing it, it is `-vf scale=-1:480`
[00:23] <Stalkr_> The subtitles are there without that, but they are gone when scaling
[00:25] <Stalkr_> llogan: This is without scale http://pastie.org/9001871, this is with scale http://pastie.org/9001873
[00:26] <llogan> Stalkr_: you should only need one filtergraph. do you want to scale first, and then place subtitles. or burn the subtitles and then scale everything?
[00:26] <Stalkr_> I think scale first, then subtitles would be prettiest
[00:27] <llogan> -filter_complex "scale=-1:480,subtitles=psych.srt"
[00:29] <Stalkr_> Looks like that's working, thank you very much. It does print `Neither PlayResX nor PlayResY defined. Assuming 384x288` but the output is 480p
[00:29] <llogan> does the output look ok? then you can probably ignore it.
[00:30] <Stalkr_> It's red, but that could just be my iTerm. It looks good
[03:48] <ScottSteiner-> How can I turn softsubs that are included as a stream in the video file into hardcoded subs? The stream is Stream #0.2(eng): Subtitle: [0][0][0][0] / 0x0000
[05:47] <klaxa> ScottSteiner-: http://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20video
[09:45] <stevenm> Hey do I go someplace else for avconv support?
[09:45] <sacarasc> #libav
[09:45] <stevenm> aha #libav - that'll be it
[09:46] <stevenm> lol
[12:43] <bainsja> Could anyone help with a question I have? I'm muxing an rtsp stream to an mp4 container using ffmpeg with the -t option to limit to a fixed duration. When the duration is reached ffmpeg doesn't exit, it just sits there until I kill it, is this expected behaviour? command line is ffmpeg -i "rtsp://x.x.x.x:554/" -y -t 60 -vcodec copy test2.mp4
[13:35] <eristisk> Why is ProRes 4444 not called ProRes 444?
[13:36] <eristisk> What does the 4th "4" refer to?
[13:49] <iive> probably alpha (transparency) channel.
[14:15] <jkli> hi all
[14:16] <jkli> i hope you are all well
[14:16] <jkli> I'm not sure if this is related to my encoding settings with ffmpeg or something else
[14:17] <jkli> When I try to stream a video via nginx-rtmp I get a green screen for the first frames
[14:17] <jkli> and then after a while sound and video appears
[14:17] <jkli> it varies from video to video but overall I get green screen pretty much for every video while streaming
[14:18] <jkli> any idea if this is related to encoding settings in ffmpeg?
[14:18] <jkli> I stream h264 mp4 files
[16:59] <nano-> Is it possible to generate a raw ALAC file from a flac file? I've tried "ffmpeg -i 09\ Lateralus.flac -f data -acodec alac lateralus.alac" but I get the error message "Output file #0 does not contain any stream". What I'm trying to do is to generate raw ALAC data that I can use to test my airtunes framework before doing the full ffmpeg integration. Easier to just read a file than to start interacting with a whole set of new apis.
[17:10] <c_14> nano-: Try adding -map 0
[17:51] <nano-> c_14: that produced an output file in the similar size range, hopefully correct :) thanks
[18:31] <srcreigh> Can I ask a question about using ffmpeg's av libraries here?
[18:32] <srcreigh> avpicture_free segfaults and I can't figure out what the cause of the problem is.
[19:01] <srcreigh> Sorry, I'm reading in the libav archives now. There seems to be some good discussion about this issue.
[19:35] <synth_> where can i find a build of ffmpeg that works on ubuntu that has the --enable-libfreetype feature?
[19:36] <synth_> i'd like to enable text overlays
[19:41] <synth_> i believe i actually compiled from the 3.x+ static build as of march 20th, is libfreetype included with it or is it something i have to compile seperately? i compiled using this guide: https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[19:43] <c_14> If you download the static build, it should just work. If you compiled it yourself, you will need to recompile with --enable-libfreetype and make sure that libfreetype is installed somewhere where you can include it.
[19:44] <synth_> hm okay
[20:29] <synth_> perfect, drawtext worked
[20:29] <synth_> just gotta tweak the settings
[20:31] <synth_> klaxa, i work in vfx so often we like to generate mov files from our frames for review by our clients and i've set up an intranet website that submits a bunch of information concerning the frames to a database which is read by a python script that runs ffmpeg and updates information on the database entry to where the final output video lives and i wanted to integrate information onto the overlay
[20:32] <synth_> that shows what project, sequence, shot, artist, version and any comments were reguarding that particular render.
[20:33] <synth_> klaxa this information would be helpful for our design department since they are rebranding NFL and have hundreds of submissions to send to the client so we need to track which helmet styles are in the renders or what teams, which version of the render, etc..
[20:33] <klaxa> hmm yeah i guess drawtext is suited for that
[20:33] <klaxa> i myself find it a lot easier to just fire up aegisub and compile a subtitle file and render that on top
[20:34] <synth_> i'm not familiar with aegisub
[20:34] <synth_> that would also be an option for us, but i'm not sure how to go about doing that
[20:35] <klaxa> aegisub is a rather sophisticated WYSIWYG subtitle editor
[20:35] <synth_> hm
[20:35] <klaxa> drawtext might be easier if it is for informational purpose only
[20:38] <llogan> drawtext can use a file as input too in case that is useful/easier for you.
[20:39] <llogan> or at least line breaks are easier that way
[20:39] <synth_> i think it might be easier for me to just compile the command in python for drawtext
[20:39] <synth_> is there a special character for line break or just \n ?
[20:40] <llogan> i'm not sure, but escaping might be annoying
[20:40] <llogan> http://ffmpeg.org/ffmpeg-utils.html#Quoting-and-escaping
[20:57] <jkli> hi all
[20:57] <jkli> hope you are doing well
[20:57] <jkli> can anybody tell me why i keep getting green screens when i stream via rtmp?
[20:57] <jkli> while http is just fine?
[20:58] <jkli> i read that green screens are a sign of frames being dropped or too low bandwidth
[21:03] <synth_> http://pastebin.com/XLCzfJFL any idea why i'm getting the error shown at the bottom? I've got the text= portion quoted
[21:08] <synth_> wait i think i got it now
[21:08] <DonGnom> Silver my problem
[21:08] <DonGnom> s/Silver/solved/
[22:31] <salohcin> is there a way to *indefinitely* pipe images to ffmpeg and encode and stream those images into video to be broadcast? I am generating images and would like to stream them on the fly
[22:31] <salohcin> right now I'm opening a pipe to ffmpeg in python and am trying to write raw rgb frames to the piep
[22:32] <salohcin> *pipes stdin
[22:37] <klaxa> you can use a image2pipe
[22:57] <voip> hello guys
[22:58] <voip> whats wrong, have error: av_interleaved_write_frame(): Invalid argument
[22:58] <voip> http://pastebin.com/YxwMTmBR
[23:14] <llogan> voip: didn't you encounter that issue a few days ago?
[23:14] <llogan> maybe, in this case, "-loglevel debug" may output something informative
[23:16] <voip> llogan, witth debug http://pastebin.com/PQbx0TDK
[23:22] <llogan> voip: does it occur with a local file output?
[23:27] <voip> llogan, with local file ok
[23:28] <llogan> i dont know. try the ffmpeg-user mailing list and mention that it works normally with a local file output
[23:30] <voip> ok, just confuzed av_interleaved
[23:31] <salohcin> what is the difference between *image2pipe* and simply using *-* as input
[23:48] <Sander_> Can someone help me with DVB-s2 input to transcode it to h264
[00:00] --- Wed Apr 9 2014
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[00:00] <BBB> I was about to say, smush sounds super-familiar
[00:00] <BBB> actually, so wait, libav is now back porting all of ffmpeg?
[00:00] <BBB> isn't that ... kind of a waste of time?
[00:01] <BBB> so you have now two identical projects, not just double-merging each other's patches, but active double-reviewing and sometimes even double-developing?
[00:01] <BBB> now to mention double-indenting
[00:01] <BBB> omg
[00:05] <iive> :)
[00:21] <cone-625> ffmpeg.git 03Carl Eugen Hoyos 07master:3d5c859fa6dc: Set Matroska private data when muxing Prores.
[00:21] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:01a5d7178914: Merge remote-tracking branch 'cehoyos/master'
[01:34] <Plorkyeran_> double-reviewing is probably useful
[01:34] <Plorkyeran_> more people looking over changes is rarely a bad thing
[02:15] <cone-625> ffmpeg.git 03Stephan Hilb 07master:5dae3cf35729: lavc/cpia: fix typo in log message
[02:15] <cone-625> ffmpeg.git 03Stephan Hilb 07master:541bebd41401: lavc/cpia: use avpriv_report_missing_feature()
[02:51] <cone-625> ffmpeg.git 03Janne Grunau 07master:8675bcb0addb: aarch64: add armv8 CPU flag
[02:51] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:4c57c6a7655f: Merge commit '8675bcb0addb1c7fb0b04682d1f3f95d5b8dae14'
[02:58] <cone-625> ffmpeg.git 03Janne Grunau 07master:d3789eeeed34: aarch64: implement videodsp.prefetch
[02:58] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:190106e884a2: Merge commit 'd3789eeeed3423bd1ca9dc40030a2f7a21ea5332'
[03:18] <cone-625> ffmpeg.git 03Timothy Gu 07master:5ce7ca68b868: libxvid: add working lumimasking and variance AQ
[03:18] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:072278c77798: Merge commit '5ce7ca68b86856ee8e9d6530dffdadc4eca4f8d1'
[03:27] <cone-625> ffmpeg.git 03Timothy Gu 07master:c389a8049430: libxvid: Add SSIM displaying through a libxvidcore plugin
[03:27] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:a7a82f2f22af: Merge commit 'c389a804943095ebf078daec6b64690d2c97069c'
[04:05] <cone-625> ffmpeg.git 03Luca Barbato 07master:e10fd08aa7fb: h264: Refactor decode_nal_units
[04:05] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:0ecb3075c1c2: Merge commit 'e10fd08aa7fbe8645545ad2e8721f0ed03c8e06a'
[04:16] <cone-625> ffmpeg.git 03Timothy Gu 07master:f73495686d10: adler32: Fix doxy group definition
[04:16] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:a6fa3a476afb: Merge commit 'f73495686d109ffffaa8c0387e790e7997326229'
[04:18] <cone-625> ffmpeg.git 03Timothy Gu 07master:486e3649cef5: xtea: Add Doxy @file and group
[04:18] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:13aa5757643e: Merge remote-tracking branch 'qatar/master'
[04:57] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:a75ba1e116a4: avcodec/h264/find_start_code: factorize addition out
[05:22] <cone-625> ffmpeg.git 03Thilo Borgmann 07master:fdaf8372c27e: configure: Check for generated output in check_header_oc.
[05:24] <cone-625> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:0c5a6ef886da: ffserver: nits & typos
[05:24] <cone-625> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:5267a5d78312: rtspcodes: add full list of RTSP status codes
[05:24] <cone-625> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:02497a5dc8c9: ffserver: don't hardcode RTSP status codes
[10:39] <ubitux> wm4: btw, about you utf16 question, maybe we should try to ask opensubtitles.org or a similar website
[10:39] <ubitux> they probably have tons of shit wandering
[10:41] <ubitux> > opensubtitles.org is supporting these subtitle formats: *.srt, *.sub, *.smi, *.txt, *.ssa, *.ass, *.mpl - please don't upload UTF 16/32 files
[10:41] <ubitux> heh.
[10:42] <ubitux> > utf32
[10:42] Action: ubitux shivers
[10:42] <wm4> how do you ask a website
[10:42] <wm4> anyway, *.mpl is actually used?
[10:43] <ubitux> dunno, we could contact the staff
[10:43] <ubitux> wm4: seems so
[10:43] <ubitux> maybe they provide some stats
[10:44] <ubitux> http://www.opensubtitles.org/en/statistics nothing really interesting here..
[10:57] <ubitux> http://halide-lang.org/?#overview opinions?
[12:10] <tmartiro_> hi folks!
[12:10] <tmartiro_> I need your help
[12:10] <tmartiro_> is there any example how to cut video by startime and endtime using C API ?
[12:10] <Compn> did you check the wiki ?
[12:10] <Compn> in any case, i dont know :)
[12:12] <tmartiro_> this is dev channlel , isn't this?
[12:13] <Compn> yes
[12:13] <Compn> stick around , maybe one of the devs knows the example
[12:13] <tmartiro_> ok, thanks
[12:14] <Compn> its a bit early in the morning for everyone (on the east coast of usa anyways)
[12:14] <tmartiro_> Compn, I will wait for the answer
[12:15] <wm4> <tmartiro_> is there any example how to cut video by startime and endtime using C API ? <- uh the API is so low level that this question doesn't make sense
[12:15] <wm4> libavformat can demux video and seek, libavcodec can decode video...
[12:15] <wm4> there's no such thing as "cutting" video on time boundaries
[12:16] <wm4> so the example you want would basically require a full-fledged transcoder
[12:16] <tmartiro_> working with command line, there is the option to copy video from input file using startime and duration into output file
[12:17] <tmartiro_> I need same functionality usin C api, to port it on android
[12:17] <tmartiro_> as library
[12:17] <Compn> ffmpeg runs on android, you dont want to just run it that way ? :P
[12:17] <ubitux> if stream copy, then it's using seek for initial time, and then stop when reaching a given ts
[12:18] <ubitux> that's all, rest is a simple classic packet copy loop which you can find in the examples
[12:19] <tmartiro_> thanks folks!
[12:48] <BBB> nice troll ubitux. and so true
[12:48] <ubitux> i'm trolling? :(
[12:49] <BBB> "instead of wasting our time, please keep wasting yours"
[12:49] <BBB> so true
[12:49] <BBB> that was you right?
[12:49] <ubitux> indeed
[12:49] <BBB> I think that's a troll, right?
[12:49] <BBB> very good one too
[12:50] <ubitux> :p
[12:50] <Compn> koda will think you are biting him on the mailing list
[12:51] <Compn> because of your mean words :P
[12:54] <ubitux> Compn: so what? it's not like he's trying to improve the situation, he just want ffmpeg to stop merging by taking a fake good samaritan posture
[12:57] <pross-au> "whats good for the goose, is good for the gander"
[13:09] <wm4> ubitux: these days, everything slightly provocative is called "trolling"
[13:13] <Compn> difference of opinion is trolling now :(
[13:14] <Compn> ubitux : yes i realized that and i apologize for bugging everyone about working with libav. i thought it was an honest attempt but nope
[13:14] <Compn> its the same 'do everying libav way , no negotiation, no change'
[13:15] <pross-au> granted vittorio is the nkob
[13:19] <wm4> ok I can't tell whether I'm joking or not in the last part of the mail I just sent
[13:20] <kierank> wm4: i was about to say
[13:20] <kierank> that's nuts
[13:21] <pross-au> wm4: isnt that whats happening now? its just really s-l-o-w.
[13:21] <wm4> yeah, and Libav is still pretty different from FFmpeg in some parts
[13:22] <pross-au> do you think the have the manpower to do it?
[13:24] <Compn> i dont think libav wants to merge from ffmpeg
[13:24] <Compn> or review+merge whatever
[13:25] <Compn> like ubitux said, its been 3 years, why start now ?
[13:25] <pross-au> they just want the patchez
[13:26] <wm4> Compn: recently they did start to cherry-pick more stuff
[13:26] <pross-au> yeah vittorio is doing it
[13:26] <wm4> probably panic due to debian picking up ffmpeg again
[13:26] <Compn> only koda (vittorio) , not the whole dev team
[13:26] <pross-au> what happened to the libav devs?
[13:26] <Compn> has started merging
[13:28] <pross-au> i recall there being many more.
[13:29] <Compn> pross-au : mans and jason (x264 guy) left
[13:30] <JEEB> the latter I think just mostly is being less involved in OSS in general, at least on these sides of things
[13:30] <Compn> they gained a few more people, it does seem less quiet
[13:31] <pross-au> fair enough
[13:31] <JEEB> mans was well... his usual self, and I'm actually rather happy that he no longer is around :P
[13:32] <Compn> developer numbers do fluctuate in both projects
[13:32] <JEEB> otherwise it's usual folk, elenril, kostya, lu, diego
[13:32] <JEEB> yeah
[13:36] <cone-285> ffmpeg.git 03Matt Oliver 07master:158a80cc0b51: Remove leal op to fix icl inline asm.
[13:38] <kurosu_> Justin Ruggles also did/was the last to do things nobody among them is really able to do (asm, audio stuff)
[13:42] <JEEB> talking about asm, janne's accomplishments on that field seem nice? he has mostly been playing around with ARM, though
[13:42] <JEEB> for audio there's still peloverde, mostly for review purposes as he's not overly active :)
[13:42] <JEEB> (he's one of the AAC maintainers)
[13:43] <kurosu_> I have only a narrow view on this topic
[13:43] <kurosu_> but currently, janne is the only one that makes some things possible
[13:43] <kurosu_> ie adapt arm code to change things in the C part
[13:44] <kurosu_> and yes, he has been very helpful to me - but that's, again, a small topic
[13:44] <kurosu_> peloverde contributed recently some missing stuff to the aac decoder - actually some unused dsp stuff is now tested
[13:45] <kurosu_> so probably but he looks inactive to me
[13:45] <JEEB> there's always plenty of lurkers
[13:45] <JEEB> didn't take long for me to get a review on a patch I was asked to poke around
[13:46] <nevcairiel> half of the active dev power goes into cosmetics though, so the actual-code-writing seems lower due to that
[13:46] <kurosu_> you mean, some lurker here reviewed it, you consider yourself as a lurker, or peloverde is actually lurking and reviewing?
[13:47] <nevcairiel> if you find a patch that directly hits the spot of some lurker, they might be motivated to review it at the very least, that includes peloverde and a bunch of others
[13:47] <JEEB> yeah
[13:48] <kurosu_> I personnally find this an inefficient use of several people (but who am I to judge?) - refactoring may be somewhat ok in the long term (difficult to judge)
[13:48] <kurosu_> but that's never a valid reason to cause regressions in things that worked in the first place
[13:48] <JEEB> if you're talking about stuff like TEP etc
[13:48] <JEEB> that was rather useful methinks
[13:49] <JEEB> same goes for some other API changes that come from libav
[13:49] <nevcairiel> there is a difference between API refactoring and just style
[13:49] <nevcairiel> some changes actually add useful functionality
[13:49] <nevcairiel> some just reshuffle things because someone thinks it has to change
[13:49] <kurosu_> I don't know what TEP is. What I meant is just that hypothertically K&R-only things have more drawbacks for me
[13:50] <nevcairiel> TEP was the refactor of the AVFrame based APIs to support reference counting
[13:50] <kurosu_> I can't recall everything - but there were (re)factoring and reducing the possibilities of bugs is ok to me
[13:51] <kurosu_> yeah, this one may have caused a lot of regressions initially (I don't know) but I believe it is better in the long term
[13:51] <nevcairiel> huge changes like that are rarely without a regression in this code base
[13:51] <kurosu_> anyway, I'm neither a ffmpeg/libav developper nor user
[13:53] <kierank> 12:49 PM <"nevcairiel> some just reshuffle things because someone thinks it has to change --> yeah like av_reverse
[13:53] <kierank> completely pointless deprecation
[13:56] <ubitux> wbs: you (not singular you) should really test the patches from koda before allowing him to push the commits
[13:56] <ubitux> it's not the first time he's pushing stuff he hasn't even compiled
[13:56] <ubitux> that happened with frei0r as well not long ago
[13:57] <nevcairiel> which was broken?
[13:57] <ubitux> https://lists.libav.org/pipermail/libav-devel/2014-April/058794.html
[13:57] <nevcairiel> oh
[13:58] <ubitux> ...and he's bragging review process on ffmpeg-devel after stuff like that
[13:58] <nevcairiel> i'm ignoring that thread, its so full of BS
[14:00] <Compn> nevcairiel : good idea
[14:15] <wm4> nevcairiel: taking the easy way out, huh
[14:16] <wm4> apropos, is this part of the topic still true: We need a volunteer Debian Package Maintainer.
[14:16] <wm4> wasn't there progress on this?
[14:18] <ubitux> wm4: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=729203
[14:27] <BBB> wbs: have you noticed the c89-to-c99 bugs? are we interested in maintaining it or have we stopped caring now that msvc14 supports ffmpeg natively?
[14:28] <BBB> wbs: I'm wondering what to do with the project
[14:28] <BBB> and anyone else that cares
[14:28] <nevcairiel> (its msvc12)
[14:28] <wbs> BBB: I've seen them yes... it would of course be nice to fix the bugs, although I dunno how much work it is to fix them
[14:29] <nevcairiel> didnt one of the reporters even post a patch to address his issues
[14:29] <wbs> BBB: I've got a few usecases where I still can't upgrade to msvc13 myself so there's still some need for life support, but libav at least doesn't trigger any of the issues
[14:30] <wbs> BBB: so basically, if easily fixable it would of course be nice, but other than that I wouldn't put too much effort into it
[14:30] <BBB> wbs: right, same for ffmpeg... I guess the question thus becomes do we care about non-ffmpeg/libav code compiling? or is it "send a patch"?
[14:30] <BBB> and yes one had a patch
[14:31] <BBB> maybe should merge
[14:31] <BBB> looked ok to me but I'm not the only person maintaining it - shall I just merge it anyway?
[14:31] <BBB> oh work, bbl tonight
[14:32] <wbs> BBB: I can have a look but unless I object sometimes soon, don't hesitate to merge it
[14:41] <Daemon404> curious, whats the usecase?
[14:45] <wbs> Daemon404: no idea, I guess you can ask in those posts
[14:53] <Daemon404> true
[15:09] <cone-285> ffmpeg.git 03Schenk, Michael 07master:b0a8521383e0: avformat/mov: reset drefs_count in close
[15:09] <cone-285> ffmpeg.git 03Schenk, Michael 07master:845414bbb122: avformat/oggdec: reset nstreams in close
[17:17] <wm4> getting an invalid free when seeking in some h264 files with ffplay: http://privatepaste.com/e733deecff
[17:17] <wm4> looks pretty scary
[17:19] <wm4> basically it's hard to check the sanity of the code because h264 uses AVFrame incorrectly
[17:20] <ubitux> ah nice gmx.de is refusing mails now
[17:20] <ubitux> gmx.* actually
[17:20] <wm4> ubitux: kind of bad for a mail provider
[17:21] <ubitux> :/
[17:21] <ubitux> wm4: regression i guess?
[17:21] <wm4> the seek issue? maybe
[17:21] <wm4> I only get a valgrind error, the original reporter had a crash
[17:26] <wm4> the command for the issue is: ffplay -rtmp_live recorded -rtmp_pageurl "http://www.canalplus.fr/c-divertissement/pid1784-c-les-guignols.html?vid=10…" -rtmp_swfverify http://player.canalplus.fr/site/flash/player.swf rtmp://vod-fms.canalplus.fr/ondemand/videos/1404/nip_NIP_9077_1500k.mp4
[17:26] <wm4> and then do a seek once it's started
[17:44] <wm4> updated to git HEAD, and I can't reproduce it anymore
[17:44] <wm4> but it's probably still in 2.2
[17:45] <wm4> 8710ee11d75eeb could be a candidate
[19:06] <cone-616> ffmpeg.git 03Carl Eugen Hoyos 07master:993a5afaada4: Read aspect ratio from tiff image files.
[19:06] <cone-616> ffmpeg.git 03Carl Eugen Hoyos 07master:836b60ce2b49: Fix standalone compilation of vp7 and vp8 decoder.
[20:54] <cone-616> ffmpeg.git 03Vittorio Giovara 07master:17a75a8c0902: libxvid: fix missing end of line character
[20:54] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:0d225863f2c9: Merge remote-tracking branch 'qatar/master'
[00:00] --- Tue Apr 8 2014
1
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[00:02] <DoomBoom> I downloaded the latest build of real ffmpeg and it works
[00:02] <DoomBoom> hurray
[00:14] <DoomBoom> btw is there anyway to speed up the video a bit so it stays below a certain duration?
[00:15] <klaxa> https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20…
[00:15] <DoomBoom> thanks!
[00:15] <klaxa> literally the first result on google, has 4chan not taught you?
[00:15] <DoomBoom> doh
[00:15] <DoomBoom> was looking in the ffmpeg documentation and couldn't find it
[00:16] <DoomBoom> I actually forgot to google
[00:16] <klaxa> remember it next time ;)
[00:16] <JEEB> yeah, the setpts filter I noted #elsewhere :)
[00:16] <JEEB> should speed up/slow down stuff nicely
[00:19] <yeraze> Hey.. Anyone here know some filter-fu to take a movie file & individually shift each of the 3 color channels by some static x/y offset on playback?
[06:16] <darius77> is there any way to rotate a video 90 degrees without transcoding?
[06:16] <darius77> without re-enconding
[06:17] <sacarasc> I don't think so. But you might find some weird option in some container that can do it...
[06:18] <darius77> ok
[06:18] <darius77> is there any way to re-encode with all the same codec and formatting?
[06:18] <darius77> some replicate options, with digging around to figure out exactly what the original was
[06:19] <darius77> using -vcodec copy prevents it from rotating video
[06:19] <sacarasc> http://ffmpeg.org/ffmpeg-filters.html#rotate
[06:36] <mohit_chauhan> hi, folks
[06:37] <mohit_chauhan> i am using ffmpeg in android
[06:38] <mohit_chauhan> I am getting error No such filter: 'null'
[06:38] <mohit_chauhan> http://pastebin.com/vYMrDG1L
[06:38] <mohit_chauhan> I am using this function for configure http://pastebin.com/7Z6t7Weq
[09:35] <mrskman> Hi, I'm trying to build ffmpeg with --enable-shared but there is a problem with binaries after doing that - libav* libs are not properly linked. Am i doing something wrong? Here is my configure line and ldd output. http://pastebin.com/7QnC5Sgx
[10:28] <relaxed> mrskman: run: LD_LIBRARY_PATH=/opt/ffmpeg-2.2.0-test/build/lib /opt/ffmpeg-2.2.0-test/build/bin/ffmpeg
[10:28] <relaxed> also, --bindir by default is $prefix/bin
[10:30] <mrskman> relaxed: thanks
[10:30] <relaxed> your system doesn't know where ffmpeg's libs are located
[10:31] <relaxed> mrskman: man ldconfig
[10:31] <mrskman> yes, I get it now... I thought it is linked with absolute path
[10:31] <relaxed> ok
[11:42] <tmartiro> hi folks
[11:42] <tmartiro> I need your help
[11:43] <tmartiro> is there any example how to cut video by startime and endtime using C API ?
[11:52] <username12565> repeating my question from yesterday: hey, i am not sure what happened. i was successfully making a video out of still frames and an audio track before but if i do it now, media players first play the audio and only later start the video. i used: ffmpeg -i frames/%04d.png -i music.mp3 -vcodec libx264 -crf 25 -preset veryfast -pix_fmt yuv420p -acodec copy out.mkv . i use ffmpeg version 2.2 on archlinux
[11:54] <relaxed> username12565: see if -async 1 after the last input corrects the issue
[11:55] <username12565> will do, thanks
[12:05] <username12565> relaxed: no luck, same behaviour: http://pastebin.com/H27WMwh5 (i stripped filenames)
[12:18] <relaxed> username12565: hmm, I don't know. try with my static build http://johnvansickle.com/ffmpeg/
[12:18] <relaxed> username12565: does you have the delay with ffplay?
[12:20] <relaxed> or try with mplayer -ao alsa input.mkv
[12:27] <username12565> relaxed: with ffplay video and audio started together. video plays too slow though. then some seconds in it quickly "warps" to a later point in time, audio stays playing properly. "-ao alsa" changes nothing, i use ossv4 and it is usually fine.
[12:27] <username12565> i should add that my mp3 is longer than the frames/video
[12:27] <username12565> previously mplayer just kept the last video frame and played the rest of the audio
[12:28] <relaxed> add -shortest if you want the audio to stop once the video stream is over
[12:30] <username12565> oh nice
[12:30] <username12565> i now downgraded to ffmpeg-1:2.1.4-1, lets see what happens
[12:30] <relaxed> upgrading is usually the desired direction
[12:33] <username12565> yeah but :\
[12:38] <username12565> i downgraded ffmpeg, mplayer and x264 to the last archlinux versions before march 24. now i get the desired outcome again.
[12:45] <relaxed> oh, 2.2 is the latest. you should file a bug report on that issue.
[12:58] <username12565> i dont have the time to properly investigate this :(
[14:10] <ubitux> username12565: found a regression?
[14:11] <ubitux> username12565: can you provide all the samples you used?
[14:11] <ubitux> i will report the issue if i can reproduce
[14:20] <username12565> i'll pm you
[15:02] <Bombo> Guest11536: yes.
[16:39] <john3voltas> greetings. i have a call recorder outputting g729a call recordings to a folder on a network share. i need to convert those recordings to wave file so that they can be played on a regular media player without g729 codec. i know g729 is proprietary. i'd like to try the conversion using ffmpeg. where can we get the ffmpeg decoder for g729?
[16:39] <john3voltas> oh, forgot to add that we're using windows 7 x64.
[16:41] <john3voltas> PS2: if it works fine and if we can create some scripts that can automate the conversion, we are interested in buying g729 codec.
[17:11] <ValdikSS> Hello. How does http output works?
[17:12] <ValdikSS> Is it designed only for streaming or I can use it for file uploading?
[17:12] <ValdikSS> ffmpeg -c:s ass -i http://localhost:8000/input.mkv -map 0 -c copy -c:v libx264 -preset veryfast -tune animation -crf 21 -chunked_post 1 http://localhost:8000/output.mkv
[17:12] <ValdikSS> This command uploads output file only after complete encoding process
[17:13] <ValdikSS> If the input file is big (more than ~400 frames), then encoding process just hangs.
[17:13] <ValdikSS> Oh, sorry, this could be my mistake in server. Just a minute.
[17:15] <ValdikSS> Sorry. Everything works as expected.
[17:33] <john3voltas> Hi ValdikSS . could I possibly PM you with an off-topic matter?
[17:53] <ValdikSS> john3voltas: sure
[21:28] <dvnl> Hello Everybody! Does anybody knows what to change in the config.h to disable time-related functions in ffmpeg? I'm trying to use the API on an embedded platform standalone - so I get a link problem "undefined reference to Žtimes' during the build process (without any specific info)
[21:29] <dvnl> I've tried disabling have_gettimeofday, have_unistd, have_windows_h and some others, which are included in libavutil/time.c
[22:01] <austinbv> Hey everyone, we are trying to make a distinction between audio and video files. The naive way we have approached it is checking if ffprobe has any streams with a codec_type of video.
[22:01] <austinbv> We realized that this is incomplete because mp3
[22:02] <austinbv> with album art have video streams of the art. Is there a better way to figure out if a file is a video or audio?
[22:49] <someday> if you want just audio/video grep for mvhd
[22:57] <someday> btw this is a good indication
[22:57] <someday> Stream #0:0(und): Video: h264 (Main)
[22:57] <someday> for eg
[22:58] <someday> it is doing what we are supposed to do.
[23:27] <austinbv> someday: mvhd in the ffprobe output?
[23:41] <ac_slater> hey guys, wikipedia doesnt have a good description on MJPEG. Is it just a a "codec" where each frame is a jpeg? Is there metadata? A preferred container?
[23:41] <ac_slater> thanks!
[23:44] <ac_slater> also, is there audio?
[23:45] <Stalkr_> Playing with FFmpeg and webm, but it won't hardcore subtitles. Any idea why? `ffmpeg -i psych.mkv -vf subtitles=psych.srt -ss 00:00:31.000 -to 00:00:43.000 -c:v libvpx -crf 4 -b:v 1800K -vf scale=-1:480 -an test.webm` -- I followed https://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20vid…, which looks easy enough. What am I missing?
[23:45] <Stalkr_> I tried with .avi too
[23:47] <sacarasc> ac_slater: Most of the MJPEG files I've seen have been in AVI from not very good cameras with MP3 audio.
[23:48] <ac_slater> sacarasc: interesting.
[00:00] --- Tue Apr 8 2014
1
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[01:41] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:1fec720afa49: avformat/cinedec: ask for sample when biBitCount isnt supported
[01:41] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:419800acc46a: examples/avio_reading: fix null dereference on error
[01:41] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:b978391ed5fd: avformat/oggparsevorbis: remove non functional null check
[03:07] <BBB> ubitux: I sort of expected that thread to devolve into that sort of rubbish :(
[03:40] <cone-788> ffmpeg.git 03Carl Eugen Hoyos 07master:59c1023ef544: Consider aspect ratio when writing tiff dpi.
[06:18] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:f87b3d552394: avcodec/pthread_frame: fix missing unlock on error
[06:18] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:754f84663e8b: avcodec/msrle: check return code for success before use
[13:34] <nevcairiel> is there an easy way to scan an assembled function if it uses instructions of a certain simd function set? ie. I want to verify a sse function doesn't accidentally use sse2 functions
[13:53] <J_Darnley> I don't know an existing solution but...
[13:54] <J_Darnley> you could grep a plain-text dump of the instructions for certain instructions
[16:02] <cone-625> ffmpeg.git 03Thilo Borgmann 07master:33462542cf47: configure: Add workaround for qtkit detection using ICC compiler.
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:edf4406c4dda: avformat/hls: add support for alternative renditions
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:cc1162d7751c: avformat/hls: always return EOF for unneeded playlist streams
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:da7759b3579d: avformat/hls: add support for byte-ranged segments
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:f7c7fc13698a: avformat/hls: add some logging messages
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:1cb8d986b8d4: avformat/hls: split read_from_url() out of read_data()
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:01b184e68d86: avformat/hls: parse ID3 timestamps for elementary audio streams
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:61e1a702783c: avformat/hls: track seeking on a per-playlist basis
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:4e85202bcc26: avformat/hls: improve segment selection when restarting list reception
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:d549b0910c34: avformat/hls: avoid unnecessary segment retrievals
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:8fd6875c83d5: avformat/hls: do not care about stream start timestamps
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:6b4b73e75da9: avformat/hls: do not use sequence numbers for packet ordering
[16:56] <cone-625> ffmpeg.git 03Anssi Hannula 07master:05ce529a596c: avformat/hls: factor identical playlist allocations out of parse_playlist
[16:57] <cone-625> ffmpeg.git 03Anssi Hannula 07master:ad701326b430: avformat/hls: open playlists immediately when AVDISCARD_ALL is dropped
[16:57] <cone-625> ffmpeg.git 03Anssi Hannula 07master:9aa0606e87a2: avformat/hls: properly take stream_index into account when seeking
[16:57] <cone-625> ffmpeg.git 03Anssi Hannula 07master:f29cb45bf2cf: avformat/hls: flush the subdemuxer when seeking
[16:57] <cone-625> ffmpeg.git 03Vladlen Y. Koshelev 07master:cdbd9e218d44: avformat/hls: support for seeking on event playlists
[17:14] <nevcairiel> so I finally found the offending instruction, and my disasm claims its a movq .. is movq sse2?
[17:14] <nevcairiel> 6A2F2851 movq mmword ptr [eax+ecx],xmm2
[17:14] <nevcairiel> specifically
[18:25] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:28f8114bd240: avcodec/vp8: fix checking vp7_feature_value_size
[18:25] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:c1b24ca762cb: avformat/oggdec: check for av_packet_new_side_data() failure
[18:25] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:4ccd7cb45b9a: avformat/oggdec/ogg_read_packet(): factorize failure code path
[18:28] <michaelni> nevcairiel, movq with xmm should be sse2
[18:28] <nevcairiel> it appears so
[18:28] <michaelni> 0x6a is a push though not a movq
[18:29] <nevcairiel> thats just the binary offset
[18:29] <nevcairiel> not the instruction
[18:29] <michaelni> ahh ok thought it was intended to be the intrsuctions bytes
[18:33] <nevcairiel> I sent a patch to the ML to correct it
[18:44] <cone-625> ffmpeg.git 03Hendrik Leppkes 07master:fc7e02f0ff34: dcadsp: fix SSE code to not use SSE2 instructions.
[19:29] <ubitux> why isn't movq defined on movh for avx2?
[19:30] <ubitux> (since movq xm0, blah seems to work)
[19:31] <ubitux> (it translates to vmovq)
[19:34] <ubitux> mmh i guess that's because movh means "half" and thus for ymm it would mean dq
[19:38] <ubitux> i don't get why movdq[au] (or mov[au]) would be 256 load with ymm though
[19:40] <cone-625> ffmpeg.git 03Reimar Döffinger 07master:57d10a28f41b: pulseaudio: move NULL check up before dereference.
[19:47] <ubitux> side note: today libav decides to NIH fontconfig support in drawtext, with a different configure option ofc
[19:49] Action: ubitux wonders if the answer will be "i'm not following all our downstreams"
[19:49] <nevcairiel> of course it will be
[19:49] <nevcairiel> its a handy excuse
[20:01] <kierank> ubitux: that was why i moved to ffmpeg
[20:01] <kierank> drawtext and friends
[20:30] <BBB> nevcairiel: I've used qemu for that in the past
[20:30] <BBB> nevcairiel: that actually worked
[20:30] <BBB> nevcairiel: but it's a little ... clunky
[20:32] <BBB> nevcairiel: also if move is all there is, see has an analog for that and I think x86inc.asm knows it?
[22:05] <cone-625> ffmpeg.git 03Reimar Döffinger 07master:a0568ed25440: rtspdec: check for strchr returning NULL.
[22:36] <cone-625> ffmpeg.git 03Paul B Mahol 07master:a420ccd4f2a0: LucasArts SMUSH SANM video decoder
[22:36] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:881f4e8d3440: Merge commit 'a420ccd4f2a011887451a7d5e1bebba4fd7c40e2'
[23:15] <cone-625> ffmpeg.git 03Paul B Mahol 07master:6f273093e54c: LucasArts SMUSH VIMA audio decoder
[23:15] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:787be6d07496: Merge commit '6f273093e54cba130f3ffde3d6433e74baa4ad89'
[23:37] <cone-625> ffmpeg.git 03Paul B Mahol 07master:eeadcdfd1a6f: LucasArts SMUSH demuxer
[23:37] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:5297003aa7a0: Merge commit 'eeadcdfd1a6f3089b6bf6e194d6ece8d3f113123'
[23:46] <J_D> Wow, someone went on a lucasarts bender.
[23:47] <nevcairiel> fwiw, those existed in ffmpeg for quire a while already
[23:56] <cone-625> ffmpeg.git 03Paul B Mahol 07master:a027d2a8731f: fate: add tests for SANM and VIMA
[23:56] <cone-625> ffmpeg.git 03Michael Niedermayer 07master:9e7f1474dd5f: Merge remote-tracking branch 'qatar/master'
[00:00] --- Mon Apr 7 2014
1
0
[00:29] <Daron> Can someone help me to transcode live input and to transcode it(h264) and forwarding it to rtmp. I have it already working but after a while it will give problems.
[01:16] <DASPRiD> greetings, just a short question. i have the following code which is meant to create a timelapse, prepend it with a splash screen and add audio from a separate file to it. everything basically works, except that the audio channel is not encoded into the video: https://gist.github.com/DASPRiD/fa5d8fac28b9710d7a39
[01:17] <DASPRiD> i supposed that it was because of the map option, but adding "-map 2:0" to it gives me an error "unknwon encoder '2'"
[01:21] <c_14> That would be because of the "-c:a 2" you have in the 4th to last line.
[01:22] <DASPRiD> oh
[01:22] <DASPRiD> guess that should be -ac
[01:23] <DASPRiD> thx, apparently i looked at the completely wrong place for the error, hehe
[01:23] <DASPRiD> mh, and now ffmpeg doesnt know libfdk_aac, hmpf
[01:24] <c_14> ffmpeg -codecs | egrep '^ .EA'
[01:26] <DASPRiD> there's aac and libvo_aacenc
[01:26] <DASPRiD> i guess using them will be fine?
[01:27] <c_14> It depends, iirc libfdk_aac provides better quality but you can always try it.
[01:28] <DASPRiD> yeah, ffmpeg wiki says the same
[01:28] <DASPRiD> c_14, okay, seems to work now, thanks. just one question. do you have any idea if the first step here could be skipped and done with a single ffmpeg call? https://gist.github.com/DASPRiD/b702749fda0858569613
[01:29] <smo_> hi i have loooot of erros when transcoding a live rtsp stream: what can do this ? http://pastebin.com/RBsA4yPN
[01:29] <smo_> tried many ffmpeg same thing...
[01:37] <c_14> DASPRiD: You might be able to, but I wouldn't know how.
[02:23] <Affy> Hallo, hallo. Anyone here?
[02:25] <iive> !hello
[02:25] <Affy> Hiya.
[02:26] <Affy> Got some...issues installing ffmpeg, actually. Silly, but...well, just won't go.^^
[02:26] <iive> don't ask to ask, just ask. If somebody knows the answer, they might answer.
[02:26] <iive> what issues.
[02:26] <iive> (i was going to leave...)
[02:27] <Affy> Sorry. Well, it's a Ubuntu 12.04 x64 server...I did basic ./ffmpeg, but that seems to do nothing. I tried some other stuff...and nowhere.
[02:27] <Affy> If you'd like to leave, go ahead. I don't want to waste your time.
[02:27] <c_14> Affy: Tried https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide ?
[02:27] <sacarasc> Affy: ./ffmpeg would try to execute the executable called 'ffmpeg' in the current directory, is that what you want to do?
[02:28] <Affy> Well, yeah...what I'm looking to do is install it. I have no version atm.
[02:28] <Affy> Pulled up guide...1 sec.
[02:28] <Affy> I downloaded the latest...that's what I was trying to install.
[02:28] <sacarasc> One of them?
[02:29] <Affy> Sickle.
[02:29] <Affy> The stable one he recommended below the latest.
[02:29] <iive> n8 ppl
[02:29] <Affy> Good night.
[02:29] <Affy> Too late. -_-
[02:30] <Affy> Going through compilation guide now.
[02:56] <Affy> Seems to have...worked.
[02:56] <Affy> ffmpeg version 2.2.git
[02:56] <Affy> built on Apr 6 2014 02:49:24 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
[02:56] <Affy> configuration: --prefix=/home/affy/ffmpeg_build --extra-cflags=-I/home/affy/ffmpeg_build/include --extra-ldflags=-L/home/affy/ffmpeg_build/lib --bindir=/home/affy/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
[02:56] <Affy> libavutil 52. 74.100 / 52. 74.100
[02:56] <Affy> libavcodec 55. 57.100 / 55. 57.100
[02:56] <Affy> libavformat 55. 36.101 / 55. 36.101
[02:56] <Affy> libavdevice 55. 11.100 / 55. 11.100
[02:56] <Affy> libavfilter 4. 3.100 / 4. 3.100
[02:56] <Affy> libswscale 2. 6.100 / 2. 6.100
[02:56] <Affy> libswresample 0. 18.100 / 0. 18.100
[02:56] <Affy> libpostproc 52. 3.100 / 52. 3.100
[02:56] <c_14> Waaaa, next time pastebin, please.
[02:56] <klaxa> use pastebin the nex-- yeah
[02:57] <Affy> Sorry. Every time I write anything it's an error. Maybe this reply is an error too.
[02:58] <c_14> Can you pastebin the full command and output of what you're trying to do.
[02:59] <Affy> http://pastebin.com/8Qw9SQhZ
[02:59] <Affy> Just checking version after an install.
[02:59] <Affy> Looks like everything's all right.^^
[02:59] <c_14> looks fine to me
[03:00] <Affy> I was just using the paste to show that I finished...'cause I had issues before.
[03:00] <Affy> Thanks for the help...*scrolls up*...errr...
[03:00] <Affy> You, actually, c_14...also sacarasc and @@ive.
[03:01] <c_14> np
[03:02] <Affy> Yeah, thanks.^^Thanks. Best of luck and an enjoyable end of the weekend to all.
[03:58] <SirCmpwn> until recently, I was able to use -vf "scale=trunc(in_w/2)*2:trunc(in_h/2)*2" to ensure that a file would have an even width and height
[03:58] <SirCmpwn> this does not appear to be working any more
[03:59] <SirCmpwn> in fact, -vf scale even with hardcoded parameters doesn't appear to help
[03:59] <klaxa> huh?
[03:59] <klaxa> pastebin please
[04:00] <SirCmpwn> https://gist.github.com/SirCmpwn/7aeabe5779a2ac44b606
[04:00] <klaxa> that doesn't seem to be related to the video filter
[04:00] <klaxa> same output with -loglevel debut please
[04:01] <SirCmpwn> https://gist.github.com/SirCmpwn/329f81831383d55c8e99
[04:01] <klaxa> also consider either putting both video filters in one command or use a filter complex (which i have come to love since yesterday)
[04:01] <SirCmpwn> ah, dropping yadif does help
[04:02] <klaxa> >[graph 0 input from stream 0:0 @ 0x1730e80] Setting 'video_size' to value '622x337'
[04:02] <klaxa> this looks wrong regardless
[04:03] <klaxa> oh wait, that's the input
[04:03] <klaxa> >Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 622x337, 1/100, q=-1--1, 90k tbn, 100 tbc
[04:03] <klaxa> but this can't work
[04:03] <SirCmpwn> it works if I merge yadif and scale into the same -vf
[04:03] <klaxa> why are you using yadif on a gif btw=
[04:03] <klaxa> ?
[04:04] <SirCmpwn> was decided on through an automated procedure
[04:04] <SirCmpwn> tried to detect interlacing and passed
[04:05] <klaxa> ah
[04:05] <klaxa> using yadif before scaling is probably a good idea too
[04:06] <SirCmpwn> alright
[04:06] <klaxa> to use the same thing as a filter complex use -filter_complex "[0:v:0]yadif[deint]; [deint]scale=trunc(in_w/2)*2:trunc(in_h/2)*2" instead of -vf
[04:07] <SirCmpwn> as much as I'd love to do that
[04:07] <SirCmpwn> the actual filters in use are determined programatically
[04:07] <SirCmpwn> and that is much harder to put together
[04:07] <klaxa> ah probably
[04:11] <SirCmpwn> https://github.com/MediaCrush/MediaCrush/commit/94e0d4b86ef880ee9e2a26033a3…
[04:37] <dhaval2712> So there's this.. the -to , -ss and the -t flags basically aren't working.
[04:37] <dhaval2712> http://pastebin.com/DJjppUML Oh sorry forgot the apste
[04:38] <klaxa> you are using avconv and not ffmpeg, see #libav
[04:38] <dhaval2712> c_14: So? Can you help me?
[04:39] <klaxa> not sure you should put quotes around the time either
[04:39] <dhaval2712> Hang on, klaxa, what does that mean?
[04:39] <klaxa> :I
[04:39] <dhaval2712> Ah.
[04:40] <dhaval2712> I tried it in ffmpeg in Fedora as well, that wasn't working either.
[04:41] <c_14> I'd get rid of the quotes around the time, also having -ss twice is rather redundant. Unless of course -ss is additive, in which case it would still be better to only have one -ss.
[04:41] <c_14> better as in easier to comprehend
[04:41] <dhaval2712> Tried both.
[04:41] <klaxa> ah, no i see what he's trying to do
[04:41] <klaxa> quick look-ahead before decoding
[04:41] <dhaval2712> There's a large flac file, I'm trying to extract the individual songs from it.
[04:42] <klaxa> *seek
[04:42] <dhaval2712> I tried to simply extract it, that didn't work either.
[04:42] <klaxa> see: https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg
[04:43] <dhaval2712> I tried to set the duration, didn't work.
[04:44] <c_14> Have you tried getting rid of the quotes around the time?
[04:45] <dhaval2712> I did.
[04:45] <dhaval2712> And I used -t. It worked!
[04:45] <dhaval2712> Thank you so very much!
[04:45] <dhaval2712> exit
[04:46] <c_14> I'm glad I helped?
[05:23] <absolutezeroff> Hello. Complete newbie here, I'd just like to know: what is a simple way to use ffmpeg to record video + audio from the desktop and write to a locally hosted stream using ffserver? I have everything installed and am running it on xubuntu.
[10:28] <FedoraUser> hi friends
[10:28] <FedoraUser> can I use ffmpeg to scan for mask marker and insert an image in its place?
[10:51] <FedoraUser> I don't know if it was clear what I wanted, so here is the example: http://i59.tinypic.com/mt5ls4.png
[10:52] <FedoraUser> I am not asking how exactly should I do it, just general pointers, what filters should I look into etc...
[11:07] <Mavrik> do we know of any OCR library for DVB subtitle recogniion?
[11:31] <EcoGiko> Hello, I have a problem with compiling ffmpeg with libvpx. These are the commands I am using and my path. https://dpaste.de/PWWZ
[11:33] <EcoGiko> https://dpaste.de/KcQn - forgot to mention the error itself...
[11:52] <EcoGiko> hello?
[12:06] <sacarasc> So, compiling libvpx does nothing at all?
[12:07] <fling> I need to join multiple videos together
[12:07] <fling> but how to take only few seconds with offset from each video with a single command?
[12:08] <fling> I don't want to split them to smaller ones first
[12:10] <fling> not possible?
[12:10] <EcoGiko> sacarasc: I think so
[12:10] <fling> all the files are encoded with the same huffyuv codec, same geometry, ar, framerate and no sound
[12:11] <fling> sacarasc: Hello. :>
[12:11] <EcoGiko> sacarasc: I am currently trying to compile it using this guide: http://wiki.webmproject.org/ffmpeg/building-with-libvpx
[12:14] <EcoGiko> sacarasc: but the problem persists...
[12:14] <sacarasc> EcoGiko: What is the output of ./configure for libvpx?
[12:15] <sacarasc> fling: I have no idea.
[12:15] <sacarasc> Which is why I said nothing. :D
[12:15] <EcoGiko> sacarasc: https://dpaste.de/uLWa
[12:16] <sacarasc> And then it says nothing at all really for make?
[12:17] <EcoGiko> sacarasc: then I make it and the last line says: make[1]: Nothing to be done for `all'.
[12:17] <EcoGiko> sacarasc: https://dpaste.de/P31b
[12:26] <EcoGiko> any help?
[12:39] <nell> hi
[12:40] <EcoGiko> hi
[13:03] <EcoGiko> Hello, I have a problem with compiling ffmpeg with libvpx. These are the commands I am using and my path, also the config.log - https://dpaste.de/uvwK
[14:52] <Sander_> Hi, i have some problems with mapping and to get my stream stable working. http://pastebin.com/cS5FNb2y
[15:20] <l_r> hello
[15:21] <l_r> i keep getting this warning when recording audio + video
[15:21] <l_r> [alsa @ 0x1d0ffb0] ALSA buffer xrun. 147kB time=00:00:03.04 bitrate= 396.2kbits/s
[15:21] <l_r> do you have any idea?
[15:21] <l_r> it happens every secon
[15:23] <l_r> do dd if=/dev/video0 bs=256k | ffmpeg -f alsa -ac 1 -ar 22050 -i hw:1,0 -loglevel verbose -analyzeduration 1 -f h264 -r 13 -i - -c:v copy -strict experimental -c:a pcm_s16le -f flv rtmp://myserver
[15:23] <l_r> that's the command
[15:42] <mr-foobar> hi all, trying out command line editing with ffmpeg. Here's what I want to achieve ... make a continuous video into a sequence of frames
[15:43] <mr-foobar> can I extract a frame and loop it a few seconds in a single line, with some filter magic ? I know I can do it in two lines
[18:10] <Affy> I successfully compiled and installed ffmpeg using https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide, but I can't seem to actually find it. rutorrent can't, and it's not in /usr/bin/ffmpeg. Any ideas where?
[18:11] <c_14> If you followed the wiki, in $HOME/bin
[18:12] <Affy> Aha. User bin. Reading the conclusion of the guide is sometimes useful...
[18:14] <Affy> -_- Thansk again.
[18:14] <c_14> np
[18:14] <Affy> Thanks* Gee. Can't even spell.
[18:15] <Affy> Thanks and Thansk...both.
[20:22] <absolutezeroff> Hello. I'm trying to do something with ffserver, but when I try it gives me an HTTP Error 404 http://pastie.org/8998712
[20:22] <absolutezeroff> This is my ffserver.conf http://pastie.org/8998715
[20:23] <absolutezeroff> http://pastie.org/8998716 and this is the command i use to try to output to ffserver
[20:43] <TakaradaService> Hello, I'm trying to make a slideshow out of pictures in a folder with the following line : "ffmpeg.exe -r 1/5 -i image%03d.png -vcodec libx264 out.mp4" but my output only has the four last files of folder
[20:43] <TakaradaService> Is there something obvious I'm missing ?
[20:45] <TakaradaService> Oh nevermind, found the problem
[20:45] <TakaradaService> One of the images isn't the same size
[21:10] <hannes3> hey, i am not sure what happened. i was successfully making a video out of still frames and an audio track before but if i do it now, media players first play the audio and only later start the video. i used: ffmpeg -i frames/%04d.png -i music.mp3 -vcodec libx264 -crf 25 -preset veryfast -pix_fmt yuv420p -acodec copy out.mkv
[21:10] <hannes3> did anything change recently in an update?
[21:10] <hannes3> i use ffmpeg version 2.2 on archlinux
[22:21] <t4nk311> Hello
[22:22] <t4nk311> I'm trying to use ffmpeg but I don't know where the file to convert goes, can anyone help?
[22:23] <hannes3> you need to be more verbose what exactly you are trying to do and how
[22:24] <t4nk311> I'm wanting to convert a file, so I use the command line, but it shows me this http://gyazo.com/be955d53802368435460ffff1c22e8c7 where do I put the original file to convert it
[22:26] <hannes3> you have to be in the same directory as the original file or supply the full/relative path to it
[22:26] <t4nk311> I see
[22:27] <t4nk311> Could you post me an example. It doesn't seem to be working.
[22:29] <hannes3> "ffmpeg -i ../source.mkv out.mkv" or "ffmpeg -i /home/me/somefile.mkv /mnt/somewhereelse/out.mkv"
[22:32] <EcoGiko> Hello, I have a problem with compiling ffmpeg with libvpx. These are the commands I am using and my path, also the config.log - https://dpaste.de/uvwK
[22:34] <klaxa> EcoGiko, your libvpx seems to be too old?
[22:35] <EcoGiko> klaxa: I just downloaded from source
[22:35] <klaxa> >/tmp/ffconf.p4xCZ1mb.c:1:29: fatal error: vpx/vpx_decoder.h: No such file or directory
[22:36] <klaxa> let's look at the configuration again... one sec
[22:37] <klaxa> hmm... are you sure libvpx is correctly compiled and installed?
[22:37] <klaxa> because it looks correct
[22:42] <klaxa> EcoGiko?
[22:42] <EcoGiko> klaxa: yeah?
[22:42] <klaxa> did you check the stuff?
[22:43] <klaxa> is there an include directory in your ffmpeg_build directory?
[22:43] <EcoGiko> klaxa: yes there is
[22:43] <klaxa> if so there *should* be vpx/vpx.h if it was correctly installed
[22:43] <EcoGiko> klaxa: there isn't a vpx.h
[22:44] <EcoGiko> encoders, decoders, images, integer, vp8 stuff but not vpx.h
[22:44] <klaxa> also, --extra-ldflags="-I$HOME/Programs/ffmpeg_build/lib" should be --extra-ldflags="-L$HOME/Programs/ffmpeg_build/lib"
[22:44] <klaxa> ah, vpx_decoder.h and such?
[22:45] <EcoGiko> yeah
[22:46] <klaxa> weird that ffmpeg reports that it can't find the files then
[22:46] <klaxa> can you run configure again and change the --extra-ldflags while at it?
[22:47] <EcoGiko> klaxa: oh wow, I replaced the flag -I with -L and it worked
[22:47] <klaxa> ah good
[22:48] <EcoGiko> now I will make it and install it
[22:49] <EcoGiko> klaxa: thanks a lot, I have been waiting for 13 hours for a response...
[22:49] <klaxa> woah
[22:49] <klaxa> well...
[22:49] <EcoGiko> all is well when it ends well
[22:50] <klaxa> compiler and linker flags are not really trivial (although these pretty much are)
[22:59] <EcoGiko> klaxa: what does the nonfree flag do?
[22:59] <klaxa> enable non-free encoders and decoders probably?
[23:00] <klaxa> > --enable-nonfree allow use of nonfree code, the resulting libs
[23:00] <klaxa> > and binaries will be unredistributable [no]
[23:10] <rjp421> is there some special way to pipe audio from a transcode in linux? Unable to find a suitable output format for 'pipe:'.... using cmd: 'avconv -i "urltoshoutcaststream" -acodec pcm_s16le -ar 22050 -ac 1 -' on debian7 and raspberry pi
[23:11] <klaxa> a) use -f to force a format, in your case either -f rawaudio or -f wav or something similar
[23:11] <klaxa> b) you are using avconv, see #libav
[23:12] <rjp421> ok ty, i use ffmpeg mainly it just warned to use avconv so i did
[23:33] <DoomBoom> hey, I'm trying to convert gifs to webm's, but the output always turns out to be a 1 frame-long webm when I test it in Firefox
[23:33] <DoomBoom> terminal output: http://pastebin.com/ymSPH8XV input file: https://dl.dropboxusercontent.com/u/26359447/Poland%20modern%20border%20his… output file: https://dl.dropboxusercontent.com/u/26359447/output.webm
[23:33] <DoomBoom> do I have to add some option so it takes the whole gif?
[23:35] <absolutezeroff> funny i found a guide for this on 4chan of all places
[23:35] <absolutezeroff> http://boards.4chan.org/g/res/41212767
[23:36] <DoomBoom> actually I tried that too, same result
[23:36] <DoomBoom> well, I tried that with avconv
[23:37] <DoomBoom> I also tried it with deprecated ffmpeg but that didn't work at all
[23:37] <klaxa> did you try with proper ffmpeg
[23:37] <klaxa> ?
[23:37] <DoomBoom> I noticed this: "Duration: 00:00:00.04" on the input file info, which makes me think avconv isn't getting the gif correctly
[23:37] <DoomBoom> I'll try ffmpeg again
[23:39] <EcoGiko> DoomBoom: try running ffmpeg directly from the bin folder ./ffmpeg, I got the same problem, all my stuff were 0.04 in duration
[23:39] <DoomBoom> it fails when I use the options that the 4chan thread recommends, and it does exactly the same as avconv when I supply no options except input and output files
[23:39] <klaxa> no problems here: http://dedi.klaxa.eu/public/Poland%20modern%20border%20history.webm
[23:40] <klaxa> looks really bad there for a second though
[23:40] <DoomBoom> something must be wrong on this end then
[23:40] <klaxa> although crf is set to 4
[23:40] <EcoGiko> is there a way for me to replace the avconv ffmpeg with the real ffmpeg?
[23:41] <klaxa> compile from source?
[23:41] <klaxa> afaik the libraries are not 100% compatible
[23:41] <DoomBoom> running ffmpeg from /usr/bin does the same as running it normally
[23:42] <klaxa> what?
[23:42] <klaxa> you are probably not using ffmpeg then
[23:42] <EcoGiko> good night, I am off
[23:44] <DoomBoom> oh
[23:44] <DoomBoom> I just installed ffmpeg from the debian repositories
[23:44] <klaxa> that is avconv
[23:44] <klaxa> see #libav in that case
[23:44] <DoomBoom> ah
[23:45] <DoomBoom> damn debian
[23:46] <DoomBoom> thanks anyway
[23:52] <absolutezeroff> ah
[00:00] --- Mon Apr 7 2014
1
0
[00:14] <cone-703> ffmpeg.git 03James Almer 07master:2025d8026fe5: x86/synth_filter: add synth_filter_sse
[00:14] <cone-703> ffmpeg.git 03James Almer 07master:81e02fae6ec1: x86/synth_filter: add synth_filter_avx
[00:14] <cone-703> ffmpeg.git 03James Almer 07master:c74b86699c86: x86/synth_filter: add synth_filter_fma3
[00:14] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:51fd962c0bbf: Merge commit 'c74b86699c86bdf62e8570f41d8a38be5710baa3'
[00:14] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:490d53e335dc: avcodec/x86/dcadsp_init: fix compilation failure without FMA3
[00:22] <cone-703> ffmpeg.git 03Alessandro Ghedini 07master:a49aa440c939: af_volume: implement replaygain pre-amplification
[00:22] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:37af48717067: Merge commit 'a49aa440c939e221194f8d95bf98673f8cf38a06'
[00:43] <cone-703> ffmpeg.git 03Anton Khirnov 07master:9c21c08fa7c5: af_ashowinfo: print a more descriptive name for AV_MATRIX_ENCODING_DOLBY
[00:43] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:b6850e132f36: Merge remote-tracking branch 'qatar/master'
[01:33] <BBB> michaelni: jason checked that a long time ago - the things we really wanted to inline are inlined, afaik, but I don't know details anymore
[01:33] <BBB> michaelni: if you want to inline more and think it gains, go for it
[01:34] <michaelni> no, just wanted to confirm that its ok as is
[01:42] <BBB> I think it is, I mean, stuff might have changed since then, but back then it was ok
[01:42] <BBB> it's a long time ago anyway
[02:04] <Skyler_> I think the inline limits are pretty high nowadays? I only noticed that happening on older gccs
[02:04] <Skyler_> especially since a lot of the inlining probably doesn't boost code size too much
[02:11] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:a2c6a99ba56a: configure: Fix testing of objective-c headers for qtkit
[02:38] <BBB> Compn: I'm not sure what the point is of asking them anyway
[02:39] <BBB> Compn: it's not exactly alike, but you're kinda asking a copyright violator in china to stop distributing ffmpeg... what's the point? there's no reason for them to stop doing what they're doing, it's perfectly fine... maybe morally rejectable, but who cares about that stuff anyway?
[02:39] <BBB> Compn: if they consider it a great good to cosmetically prettify that code, let them do it
[02:40] <BBB> Compn: if they claim that adds copyright claims, let them think so
[02:41] <Compn> koda asked me to work with the project, cant say i didnt try.
[02:41] <BBB> I prefer to make the end product better
[02:41] <BBB> that's why I work on both hevc as well as vp9, I mean politically that's the silliest thing to do
[02:42] <Compn> thought you were doing it for financial reasons
[02:43] <Compn> as in, you'd be the person to hire, no matter whos team was hiring. plus they could compete to hire you
[02:43] <BBB> I can honestly claim that I made $0 on ffvp9, and that I estimate my total income for ffhevc will not be any higher
[02:43] <BBB> actually that's a lie, someone gave me a dinner cheque for ffvp9
[02:43] <Compn> not on those, but on future projects like xevc or xvp9
[02:43] <BBB> but that was more bet-related that anything payment-wise
[02:43] <Compn> lol
[02:44] <Compn> not that i'm dispariging anyone for working for money . i think its good idea
[02:44] <BBB> I claimed I could make a ffvp9 decoder that'd be faster than ffh264 or ffvp8 for same-quality
[02:44] <BBB> I got laughed at - quite massively
[02:44] <Compn> and i know you've been working hard
[02:44] <BBB> I proved it could be done, and surprise, I got a dinner cheque for that bet
[02:44] <BBB> yay
[02:44] <BBB> anyway, no, I'm not being hired by any multimedia company
[02:45] <BBB> and I don't think I'm really in the market ATM
[02:45] <BBB> I'm ... how do you say that ... hard to get
[02:45] <Compn> i wish you luck! :)
[02:45] <Compn> with whatever you are doing :)
[02:46] <Compn> seems like libavcodec devels are in high demand, i dont see any problems for them
[02:49] <BBB> I don't work in that sector at all right now ;)
[02:50] <BBB> I work here: http://www.twosigma.com/
[02:50] <BBB> also, kids wanna play, brb
[03:31] <cone-703> ffmpeg.git 03Carl Eugen Hoyos 07master:87f29996415a: Force gif aspect ratio multiplication to 64bit.
[03:31] <cone-703> ffmpeg.git 03Carl Eugen Hoyos 07master:9f4ab4e5f9f0: Show stream number when warning about stream without start time.
[03:31] <cone-703> ffmpeg.git 03Carl Eugen Hoyos 07master:90d93eeb386f: Allow .y as extension for image2.
[03:31] <cone-703> ffmpeg.git 03Michael Niedermayer 07master:7a5537fd3fd0: Merge remote-tracking branch 'cehoyos/master'
[03:53] <cone-703> ffmpeg.git 03Thilo Borgmann 07master:8c161d7884f0: lavc/exif: Make EXIF IFD decoding part of private API/ABI.
[03:53] <cone-703> ffmpeg.git 03Thilo Borgmann 07master:2f08dad0286f: lavf/avidec: Read metadata EXIF tags from AVIF tag. Based on patch by Gregory Wolfe (Kodak Alaris) <gregory.wolfe(a)kodakalaris.com>.
[03:53] <cone-703> ffmpeg.git 03Thilo Borgmann 07master:ec8789ac6353: Reindent after last commit.
[05:47] <cone-703> ffmpeg.git 03James Almer 07master:7d2116dd09d1: x86/synth_filter: compile avx and fma3 functions unconditionally
[10:59] <J_Darnley> um...
[10:59] <J_Darnley> Has the layout of the fate page been deliberately changed?
[11:00] <J_Darnley> Or has something gone seriously wrong?
[11:00] <J_Darnley> Nevermind
[11:20] <superware> can I use the same packet from av_read_frame for two outputs? say avcodec_decode_video2 for display, and av_write_frame for file dumping?
[11:29] <superware> is the righ architecture to use when I have one input and two outputs is to reuse the same AVPacket (av_read_frame)?
[13:48] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:323c049c7e43: configure: more properly disable header when check_header_oc() fails
[13:49] <cone-668> ffmpeg.git 03Michael Niedermayer 07master:3c728ceec66d: configure: use TMPM in check_header_oc
[14:11] <cone-668> ffmpeg.git 03James Almer 07master:a1ac12bddd8d: x86/dcadsp: add ff_dca_lfe_fir0_fma3
[19:05] <cone-788> ffmpeg.git 03Diego Biurrun 07master:3dc6272bed78: Remove a number of unnecessary dsputil.h #includes
[19:05] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:5440151fa42d: Merge commit '3dc6272bed7890a49080e18eacf3c7a4a6594b0d'
[19:10] <cone-788> ffmpeg.git 03Diego Biurrun 07master:b42f49e42f8c: x86: dsputil: Eliminate some unnecessary dsputil_x86.h #includes
[19:10] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:874f27a8f7e3: Merge commit 'b42f49e42f8cde25a788b2d13d03e99ca2956647'
[19:16] <cone-788> ffmpeg.git 03Diego Biurrun 07master:b718b24bbe83: ppc: Drop a bunch of unnecessary dsputil_altivec.h #includes
[19:16] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:865c94bb57db: Merge commit 'b718b24bbe83d3e8787c1dc6449ccbffcd13a4e7'
[19:30] <cone-788> ffmpeg.git 03Diego Biurrun 07master:01c5779f56cf: x86: Drop some unnecessary YASM ifdefs
[19:31] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:ebb21887b853: Merge commit '01c5779f56cf708e6cb88b11cfdc248cae7e2ee8'
[19:36] <cone-788> ffmpeg.git 03Diego Biurrun 07master:c2c5be57494e: x86: h264_qpel: Simplify an #if conditional
[19:36] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:e3c3f277a9db: Merge commit 'c2c5be57494e6117086771bca34c8cd4c72c8e99'
[19:43] <cone-788> ffmpeg.git 03Diego Biurrun 07master:57b5b84e208a: x86: dsputil: Move ff_apply_window_int16_* bits to ac3dsp, where they belong
[19:43] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:e6f69b324e88: Merge commit '57b5b84e208ad61ffdd74ad849bed212deb92bc5'
[19:49] <cone-788> ffmpeg.git 03Diego Biurrun 07master:5f9220437025: dsputil: Move DV-specific ff_zigzag248_direct table to dvdata
[19:49] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:47fc82b5b346: Merge remote-tracking branch 'qatar/master'
[21:44] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:26e70fd5ba50: avcodec/vp8: fix name of copy_luma()
[21:44] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:d167faafe9df: avcodec/wma: use av_freep(), do not leave stale pointers in memory
[21:52] <lukaszmluki> hi, I have question about channel layouts in ffmpeg. For example AV_CH_LAYOUT_2_1, does ffmpeg always put samples per channel in the same order? In other words, how I know where left channel is placed in stream to not take right channel or LFE instead.
[21:56] <wm4> lukaszmluki: it puts them in the same static order
[21:56] <wm4> lukaszmluki: for each speaker, there's a bit set in the mask
[21:56] <wm4> and the bit index corresponds to the channel number
[21:56] <wm4> err
[21:56] <wm4> not really the bit index
[21:56] <wm4> more like: the Nth set bit has its audio in channel N
[21:58] <lukaszmluki> ok, i expected that, thank for confirm
[22:18] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:a84f9c75bc3f: avformat/omadec: fix probetest failure
[22:25] <Compn> lukaszmluki : theres a file that explains the channel order
[22:25] <Compn> source code
[22:25] <Compn> somewhere :D
[23:13] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:49579c963765: avformat/cinedec: fix sizeof()
[23:13] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:af2e5061bbca: avformat/mov: Fix width/height typo
[23:13] <cone-788> ffmpeg.git 03Michael Niedermayer 07master:47b03415f319: avformat/cinedec: remove redundant zero termination
[23:25] <ubitux> lol koda throwing trolls and asking for everyone to stop contradict him :')
[23:25] <ubitux> wth is wrong with that guy
[23:25] <iive> ?
[23:25] <ubitux> iive: "To be or not to be working with libav" thread
[23:25] <ubitux> Vittorio
[23:26] <ubitux> > But I had really hoped that we could use this thread for something more constructive than "you did this to me", "I don't like you because you smell" and other childish troll attempts.
[23:26] <iive> oh, i though that thread have died.
[23:26] <ubitux> even though he's basically doing that since the beginning
[23:26] <ubitux> :D
[00:00] --- Sun Apr 6 2014
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[03:32] <t4nk697> what's that?
[03:33] <t4nk697> has been ffmpeg developed in java?
[03:34] <sacarasc> No.
[11:19] <superware> can I use the same packet from av_read_frame for two outputs? say avcodec_decode_video2 for display, and av_write_frame for file dumping?
[11:32] <superware> is the right architecture to use when I have one input and two outputs is to reuse the same AVPacket (from av_read_frame)?
[11:46] <Mavrik> check documentation about ownership transfer when calling methods with that packet
[11:46] <Mavrik> if the callable function takes ownership you have to copy it
[11:54] <superware> Mavrik: well, I've checked avcodec_decode_video2 and av_write_frame documentation but couldn't find a reference to that..
[11:55] <superware> the avpkt parameter is defined as [in]
[11:56] <Mavrik> hmm
[11:56] <Mavrik> documentation fail then
[11:56] <Mavrik> it probably doesn't take ownership, but you SHOULD check that
[11:56] <superware> how? :|
[11:56] <Mavrik> you have full source
[11:57] <Mavrik> check the method you're calling and the decoder
[11:57] <Mavrik> it's not like there's alot of code.
[11:57] <superware> might be too complex
[11:57] <Mavrik> dude
[11:57] <Mavrik> it takes less time than you sat here waiting for answer ;)
[11:57] <superware> heh
[11:58] <superware> I look at the code, it's not that simple to be sure nothing is being changed
[12:00] <superware> http://www.ffmpeg.org/doxygen/trunk/libavcodec_2utils_8c_source.html
[15:05] <nell> im at a hackathon right now, we needa know what to embed within html5 and what the config looks like for that -- server config for html5 capable format
[15:05] <nell> we been tackling this problem already for many hours and we're drained >_<
[15:13] <Mavrik> nell, what's a "html5 capable format"
[15:13] <Mavrik> what are you trying to do exactly?
[15:14] <nell> We're trying to stream webcams to the browser
[15:31] <nell> anyone?
[15:31] <nell> ill pay someone doge or bitcoin
[15:32] <nell> for their help in solving this
[15:32] <klaxa> maybe you can use HLS
[15:34] <klaxa> i don't know in how far webm supports live streaming, but that might be worth looking into as well
[15:35] <klaxa> mp4 is somewhat live-streamable too
[15:35] <klaxa> ogv might be too
[15:36] <klaxa> i think that are all your options if you are going for html5 compatibility
[15:36] <klaxa> nell
[15:37] <BtbN> What was used before av_frame_free was introduced? Trying to add compatiblity for older ffmpeg versions to an existing application.
[15:38] <nell> thanks klaxa ill explore those
[17:26] <Moonlightning> How do I add chapter markers to an existing video file?
[17:27] <JEEB> you generally can't without a remux, for matroska you can create and mux chapters easily with, say, mkvmerge(gui) [part of mkvtoolnix]
[17:27] <JEEB> for mp4 you'll have to use either GPAC or L-SMASH's tools for it, but chapters aren't an official thing in mp4 so YMMV in which things will support those
[17:28] <Moonlightning> Wait, really?
[17:28] <Moonlightning> I swear I had an .mp4 video with chapters in it once
[17:29] <JEEB> yes, but both ways of putting those chapters there are not in the specification
[17:29] <JEEB> and one of them is now actually banned in the specs
[17:29] <JEEB> (the MOV way)
[17:29] <JEEB> the two ways are basically "the Nero way" and "the MOV way"
[17:30] <JEEB> aka something that Nero made up, and something that was used in MOV (which is what the MPEG-4 container mostly bases upon)
[17:40] <Moonlightning> &wait, does ffmpeg not do chapter marking at all?
[17:59] <alusion> how download ogv codec
[18:10] <Guest11536> I'm posting this again..I really need your help.. I don't think it's difficult for you to figure it out...
[18:10] <Guest11536> Hey guys,
[18:10] <Guest11536> I'm building an android application, and I need to use some ffmpeg commands.
[18:10] <Guest11536> I've got a source video and I need to edit it:
[18:10] <Guest11536> 1) Cut only the last 7 seconds.
[18:10] <Guest11536> 2) Add a watermark(logo) on the top left side of the video.(00:00-00:07)
[18:10] <Guest11536> 3) Add a video inside the source video on the bottom right side of the video.(00:00-00:07)
[18:10] <Guest11536> 4) Add another video to the edited video from steps 1-3 (00:07-00:10)\
[18:10] <Guest11536> Can I do it in one ffmpeg command line?
[18:10] <Guest11536> What do I need to consider when I'm editing a video for a cellphone (quality,sizing..)?
[18:10] <Guest11536> Thanks a lot!
[18:26] <Bombo> i'm trying to recode a 'Video: mpeg4 (Simple Profile), yuv420p' to theora with '-i foo.mp4 -c:v libtheora -b:v 1000k -c:a libvorbis -b:a 128k' but i get '[mp4 @ 00000000026dd9e0] track 0: could not find tag, codec not currently supported in container'
[18:28] <Bombo> oh
[18:28] <Bombo> theora in mp4.
[18:28] <Bombo> i'm dumb.
[18:28] <Bombo> sorry ;)
[18:30] <Bombo> wow 19fps
[18:33] <Bombo> are there different profiles like for x264 superfast? is encoding theora generally slow?
[18:52] <thebombzen> Bombo: x264 is particularly good about configuring how you want it to encode
[18:52] <thebombzen> most other encoders like libtheora only allow you to set the bitrate because the algorithm isn't advanced enough
[18:53] <thebombzen> to determine speed/quality at the same bitrate
[18:53] <thebombzen> Guest11536: you can do all that in several command lines
[18:54] <thebombzen> Guest11536: but you might need to probe it first to find out the duration so you know how far to skip
[19:34] <name_space> howdy guys. So I've configured ffserver for 4 feeds. When we execute ffmpeg -f mpeg /dev/video0 http://0.0.0.0:8090/feed1.ffm it runs until it crashes and I can't get a feed in the browser.
[19:35] <name_space> is there a format I can just embed into the browser? Am I configuring things right?
[19:38] <Guest11536> thebombzen: I'm new to FFMPEF technology. Would you write me the command so i could implement it?
[19:39] <name_space> ?
[19:54] <Bombo> thebombzen: hmmkay
[19:55] <Bombo> Guest11536: for watermarking i found http://stackoverflow.com/questions/10918907/how-to-add-transparent-watermar…
[19:56] <Bombo> Guest11536: not sure if that works with videos
[19:57] <alusion> hey
[19:57] <alusion> i need information on stream format & ffserver config file
[19:58] <Bombo> Guest11536: when you do ffmpeg -i inputfile you get time, you would get that and calculate the last seconds, then use -ss skip to timestamp, -t duration (or use ffprobe)
[20:06] <Bombo> Guest11536: also: you need a nick ;)
[20:13] <xenphor> hi im using the looping command found in the concatenating section of the help file except I want to alter it slightly. Instead of playing each file once and then looping, I would like to have it finish a complete loop of a file and then move on to the next one. This is the command I was using: http://pastebin.com/nTmAw9Hz
[20:19] <xenphor> actually sorry that's the exact command in the help I first altered it a little: http://pastebin.com/XMEEnKpH
[20:48] <llogan> xenphor: i don't understand. can you explain in more detail?
[20:48] <llogan> alusion: did you see https://trac.ffmpeg.org/wiki/Streaming%20media%20with%20ffserver
[20:51] <xenphor> ok maybe this will help: http://pastebin.com/czpezVZ8
[20:53] <llogan> xenphor: how many files are you working with?
[20:53] <xenphor> well probably eventually will be around a couple hundred or more
[20:54] <xenphor> they're all short 5 second clips
[20:54] <xenphor> so i want them to loop a couple times each
[20:54] <xenphor> before merging them together
[21:29] <Guest11536> Bombo: thanks a lot, I manage to do the commands one after another, but how i can combine them together? It is possible at all?
[22:50] <Dumpy> Hi
[00:00] --- Sun Apr 6 2014
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