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April 2014
- 1 participants
- 60 discussions
[02:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:4b8e7a141494: avcodec/utils: prettify remap_deprecated_codec_id()
[02:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:161ec197ac33: fate/libswresample: add tests with filter_size=1 phase_shift=0
[02:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:6c8ee74af2af: swresample/resample: Fix fractional part of index in the filter_size = 1 filters = 1 case
[02:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:5e379cd3ee87: swresample/resample: simplify index/consumed calculation for the filter = 1 case
[06:48] <anshul_> michael, make distclean with cpp example patch running ok at my pc, what configuration you were using while doing distclean
[06:50] <michaelni> plain ./configure && make distclean
[06:52] <anshul_> it works fine at my place, http://pastebin.com/treRxz4p
[06:55] <anshul_> I think problem might be make -C doc/examples_cpp/ clean, that make does not come back to original directory
[06:56] <anshul_> michaelni, Is it gmake that you are using
[06:56] <michaelni> no plain make
[06:59] <michaelni> "make -C doc/examples_cpp/ clean" is causing it
[07:00] <michaelni> same with "make -C doc/examples/ clean"
[07:00] <michaelni> but make distclean doesnt cause it before the patch but does afterwards
[07:02] <j-b> 'morning
[07:03] <michaelni> j-b, hmm indeed, ill better hurry and go to bed
[07:08] <j-b> michaelni: hehe ;0
[08:41] <anshul_> michaelni, is normal make from doc/examples directory working fine at your pc
[09:20] <kcynice> hello, im puzzled if i should ask somequestions about using ffmpeg library for my development here, or #ffmpeg?
[09:25] <ubitux> on $ffmpeg
[09:25] <ubitux> #ffmpeg*
[09:25] <av500> %ffmpeg
[09:27] <ubitux> :(
[09:28] <av500> @ffmpefg
[09:28] <av500> :)
[09:30] <ubitux> mornings are hard av500, aren't they?
[09:34] <kcynice> thanks
[14:26] <cone-528> ffmpeg.git 03Martin Storsjö 07master:6477139721f5: rtmpproto: Make sure to pass on the error code if read_connect failed
[14:26] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:8e5f84047268: Merge commit '6477139721f559b26eafd415e23e13ea2b0c27e1'
[14:41] <cone-528> ffmpeg.git 03Stephan Soller 07master:4d40e073dcde: rtmpproto: Handle RTMP chunk size packets before the connect packet
[14:41] <cone-528> ffmpeg.git 03Stephan Soller 07master:36b9c27dae45: avformat/rtmpproto: Added handling of an initial RTMP chunk size packet.
[14:41] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:b11265c9891e: Merge commit '4d40e073dcdef335215c124d340ea156bc4bdf13'
[14:48] <anshul_> I was trying to debug dvbsubdec.c, png files are saved with 0 bytes, any idea what I am missing. I have defined DEBUG at top and using loglevel debug
[14:49] <ubitux> i remember using it and it was working pretty well
[14:51] <anshul_> may I share ts file, there might be problem in that, but ffplay is showing those subtitle
[14:52] <anshul_> If you have that file which you were using can you share that.
[14:52] <ubitux> mmh i don't have time right now, but i can send you something
[14:53] <ubitux> see in pv
[14:55] <kierank> I've used that PNG thing before in dvbsub
[15:01] <cone-528> ffmpeg.git 03Niels Möller 07master:b3d905b702dc: dcadec: Delete redundant init_get_bits call
[15:01] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:9468e9e5a2cd: Merge commit 'b3d905b702dcdda9e3ae2bcb7b5580ca77a96ab7'
[15:03] <ubitux> anshul_: how did you try it btw?
[15:03] <ubitux> ffmpeg -i bla.mts -c:s dvbsub -map s:0 -f null - or something like that should do the trick
[15:04] <anshul_> I defined DEBUG in the dvbsubdec.c file compiled the code and run using ffplay
[15:05] <anshul_> it saved png file with 0 bytes
[15:09] <anshul_> no your command have same behabiour files with 0 byte
[15:10] <anshul_> anyway I will look at it why its not working with ffmpeg binary, and update my command or code
[16:32] <ubitux> heh 4.9 :)
[16:44] <cone-528> ffmpeg.git 03Carl Eugen Hoyos 07master:63c03ea98d90: Fix writing RGB 5:5:5 rawvideo into avi.
[16:44] <cone-528> ffmpeg.git 03Carl Eugen Hoyos 07master:ef48ac6523c7: Set the bitrate when decoding speex.
[18:28] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:46d5625f4418: avcodec/x86/idct_sse2_xvid: fix non C99 inline function
[18:40] <cone-528> ffmpeg.git 03Rainer Hochecker 07master:e1fcd3a00759: flac demuxer: improve seeking
[18:40] <cone-528> ffmpeg.git 03Rainer Hochecker 07master:d8b19ee67296: fate: update seeking reference for flac
[18:40] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:3df787a1ef16: Merge branch 'flac' of https://github.com/FernetMenta/FFmpeg
[21:47] <cone-528> ffmpeg.git 03Michael Niedermayer 07master:b465aaa59f17: avformat/img2dec: fix typo
[23:35] <llogan> what's the secret forula to estimate/calculate the size of a rawvideo output?
[23:35] <llogan> *formula
[23:35] <llogan> size, as in file size.
[23:36] <llogan> bgr0
[23:46] <nevcairiel> framesize * fps * seconds?
[00:00] --- Tue Apr 15 2014
1
0
[00:05] <dwirc> I've tried the "-shortest" option, but when placing before the input (where I believe it is supposed tobe), I get the following error:
[00:05] <dwirc> Option shortest (finish encoding within shortest input) cannot be applied to input file
[00:06] <dwirc> Moving the "-shortest" option AFTER the input, ffmpeg executes, however when the input stops FFMPEG hangs again.
[00:13] <Hello71> reduce it to the most basic test case
[00:24] <dwirc> roger that
[00:25] <dwirc> Here is the pastebin with the command line, and full output:
[00:25] <dwirc> http://pastebin.com/QAwbK3xK
[00:40] <xlys> Maybe I'm just not looking for it properly, but when I run ffmpeg -i file I get a lot of file info, but as far as I can see the container format is not outputted. If I have a video file with no extension, can ffmpeg tell me what the container format is (what the right extension would be)?
[00:42] <clever> xlys: 'file' may identify it
[00:43] <xlys> clever: The program file? I thought about that, but it only says Video 13-04-14: MPEG sequence, v2, program multiplex
[00:43] <xlys> ?
[00:43] <clever> ah
[00:44] <clever> not sure then, my only guess would maybe be mpegts, program stream
[00:49] <xlys> clever: Okay, thanks! :)
[01:33] <dwirc> Any thoughts on why this would be happening with my source? Or is there any better way to detect a lack of input and have FFMPEG stop?
[02:05] <SirCmpwn> trying to use -progress
[02:05] <SirCmpwn> but ffmpeg won't do anything at all if I do
[02:05] <SirCmpwn> it just exits immediately
[02:05] <SirCmpwn> ffmpeg -progress http://127.0.0.1:5000/ffmpeg -i Hatsune\ Miku\ Live\ Party\ in\ Kansai\ Part\ 1.mkv -to 00:25:51 output.mkv
[02:12] <llogan> SirCmpwn: i've never tried that option, but the author, Nicholas, may provide some information on ffmpeg-user mailing list
[02:12] <SirCmpwn> llogan: thanks
[03:41] <Ertain> Hello everyone. I have a webm file that I'm trying to split up. I'm using the command, "avconv -i file.webm -sameq -ss 00:03:20 -t 00:03:28 output.webm", but that doesn't work. It only moves up to the 00:03:20 part of the video, and then just puts the rest of the stream into "output.webm."
[03:43] <klaxa> for avconv support see #libav
[03:45] <Ertain> Thanks, klaxa.
[04:16] <Jinx> I have a MOV with corrupted audio which I have repaired into a WAV file. How can I mux my original video (which still includes audio) but give it the new audio from the WAV file?
[04:17] <Jinx> My efforts thus far have yielded a "Seems stream 0 codec frame rate differs from container frame rate"... which occurs regardless of whether the WAV is 44.1 or 48khz
[05:16] <riveter> I'm trying to convert an h264 stream to pngs and am getting unacceptable lag. Is there a command line option that would tell it to only process iframes?
[06:44] <pzich> is there any tmp file cleanup I should be doing for an aborted or failed encode?
[08:27] <AndrzejL> is winff official ffmpeg frontend? I am asking coz I am looking for a nice set of x264 / mkv presets and cannot find them and I am to lazy or would I say I dont have sufficient knowledge to create them myself ;D.
[08:28] <AndrzejL> I have the official presets from winff site - I am looking for something more then that
[08:38] <diesel420> AndrzejL, i am not sure if it is the right place to ask, if you use winff and want a nice x264 preset. i never used winff and maybe i just dont understand your question properly.
[08:38] <diesel420> but my favorite x264 preset would be: --preset=placebo --tune=film --profile=high --aq-mode=2 --me=umh --merange=32
[08:39] <diesel420> is off of topic and i apologize for it ;)
[08:52] <AndrzejL> diesel420: nah you're fine - if winff is not officialy supported by ffmpeg then I am looking in a wrong place - hence my question :) - will try figuring something out - I always do ;D
[09:15] <kcynice> hello, its my first usage of ffmpeg. I want to reduce the output file size of my arm-linux. I only decode frames from a mp4 video with mpeg4 encoding. any compiling configure idea?
[09:16] <kcynice> i use the example code, demuxing-decoding.c, the output file size is up to 11M, eventhough i stripped it
[09:23] <ubitux> kcynice: start with --disable-everything, and add what you need
[09:24] <ubitux> typically --enable-demuxer=mp4 --enable-decoder=h264 --enable-encoder=mpeg4 ... --enable-protocol=file,pipe
[09:25] <ubitux> ofc other codecs are possible with mp4, depends on what you want to support
[09:36] <kcynice> thans ubitux, i will paste my configure, and the test result
[09:43] <kcynice> ubitux, what's --enable-protocol meant, i only native file, no other streams, such as network,pipe, etc
[09:43] <ubitux> then just enable file
[09:48] <kcynice> there is no mp4 demuxer, i use mpegvideo, right?
[09:48] <kcynice> for my last try, i got a failure at av_format_open_input, i did not know which option affected on it.
[10:07] <kcynice> ubitux, i use the same configure options, but got the the same failure for arm, failed at avformat_open_input. any help?
[10:08] <ubitux> what's your configure ligne?
[10:08] <ubitux> line*
[10:11] <kcynice> --disable-everything --enable-demuxer=mpegvideo --enable-decoder=mpeg4 --enable-protocol=file
[10:12] <ubitux> you want "mov" as demuxer
[10:13] <ubitux> you sure you have mpeg4 and not h264 in your mp4(s)?
[10:18] <kcynice> Im sure not h264 within it, but, the configure works well for pc version
[10:19] <kcynice> use mplayer play the original video and got an output:
[10:20] <kcynice> Selected video codec: [ffodivx] vfm: ffmpeg (FFMpeg MPEG-4)
[10:21] <ubitux> can you pastebin a ffprobe of your input file?
[10:21] <kcynice> OK. hang on.
[10:23] <kcynice> done. at http://pastebin.com/Eyiny7MP
[10:24] <ubitux> and so with the mov demuxer it doesn't help?
[10:45] <`Fibz> i have a dumbphone, the Pantect Crux. im trying to convert video to watch on this device. in windows i need just use windows movie maker. i've been trying all day in linux and it just doesnt want to play anything i encode with openshot
[10:45] <`Fibz> any suggestions?
[10:46] <ubitux> where is the ffmpeg question?
[10:46] <`Fibz> how to use
[10:46] <ubitux> you want me to paste the whole manual on irc?
[10:46] <`Fibz> no just some suggestions on codec, container and settings
[10:47] <ubitux> well, what does your device supports?
[10:47] <`Fibz> supposedly mpeg, mp4 and wmv,
[10:48] <ubitux> what about the codecs?
[10:48] <`Fibz> do you have any suggestions about how i can go about finding this out?
[10:48] <ubitux> read the specs of your device
[10:49] <ubitux> you can try to ffprobe a file that works as well
[10:49] <`Fibz> it says what i said. mpeg, mp4 and wmv
[10:49] <ubitux> http://trac.ffmpeg.org/wiki/x264EncodingGuide#Compatibility here are some examples for other devices
[10:49] <`Fibz> thank you
[10:50] <kcynice> its returned value was 0xBEBBB1B7, but i do not know what's it meant
[10:50] <ubitux> return value of what?
[10:52] <kcynice> avformat_open_input
[10:56] <ubitux> av_err2str()
[11:03] <kcynice> i got it: Error:Invalid data found when processing input
[11:57] <anshul_> kcynice,look whether data is valid. ;)
[12:04] <kcynice> anshul_, i was puzzled by that, because it works find for pc version
[12:07] <anshul_> you mean fine on another pc's
[12:35] <kcynice> no, the operation failed on arm-linux, ok for pc-linux
[12:43] <ubitux> with the same options?
[12:43] <ubitux> or you mean a full featured x86 buld vs your partial arm build?
[13:04] <mr-foobar> how do you delete a particular region, -ss t, in the video using ffmpeg ?
[13:21] <kcynice> ubitux, i have to leave now, im not sure i have time to enter here later. thanks for your help, anshul_ , too
[13:21] <kcynice> have funs
[15:50] <dwirc> Is there any way to have ffmpeg exit if an RTSP source stops sending data, or drops the TCP connection entirely? (Yes TCP - forcing the connection to TCP instead of UDP).
[18:25] <wickwire> Hi guys, I've been asked to implement a server-client demo for RTP/RTCP and video (h264). Since I'm new to this kind of technology, would it be possible for anyone here to confirm if ffmpeg is a good starting point for this, and if so, where to begin reading...?
[18:25] <wickwire> I've already downloaded ffmpeg's source code
[18:26] <wickwire> and have been reading several forums and guides online, but without much progress
[18:26] <wickwire> I've also found out about oRTP, an open source library using with linphone - got it to work, but for audio only
[18:28] <wickwire> to be noted, I was thinking about using ffmpeg in the hopes of understanding how this can be done, and possibly not have to use the whole of ffmpeg to do it but just the essential bits
[18:49] <barnoux> hello
[18:49] <barnoux> i need to install the last version of ffmpeg on a gentoo distro, do you have a tutorial for that?
[18:55] <klaxa> https://trac.ffmpeg.org/wiki/CompilationGuide
[18:55] <quup> barnoux: can't you just unmask and emerge?
[18:57] <barnoux> quup, i will try but does this version include the libfdk_aac ??
[19:00] <quup> no idea
[23:49] <naggi> Hello! :)
[23:50] <naggi> Just wonder, is there any program you guys know where i can take an MP4 file and see how its encode'd so i just can copy those settings and encode myself with those?
[00:00] --- Tue Apr 15 2014
1
0
[00:11] <kierank> ubitux: for clarity i think
[00:12] <kierank> it's explaining in pseudocode what the instruction does
[00:12] <kierank> so it can do what it likes
[00:13] <ubitux> ok
[00:48] <cone-727> ffmpeg.git 03Katerina Barone-Adesi 07master:d4a8a86c6863: doc: Libavfilter English cleanup
[00:48] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:275155081f2e: Merge commit 'd4a8a86c6863ecea208fa2f774690b6df4b5bc71'
[01:19] <cone-727> ffmpeg.git 03Lukasz Marek 07master:c1c3c3611ef1: lavd: no default device unless explicitly set
[01:19] <cone-727> ffmpeg.git 03Lukasz Marek 07master:afc898e7ce00: lavd/fbdev_enc: implement get_device_list
[01:19] <cone-727> ffmpeg.git 03Lukasz Marek 07master:defb87eef4dd: lavd/qtkit: add device category
[01:19] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:d3284a62082b: Merge remote-tracking branch 'lukaszmluki/master'
[01:36] <cone-727> ffmpeg.git 03Justin Ruggles 07master:60c4660ba035: swscale: fix an implementation-defined unsigned-to-signed conversion
[01:36] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:28875c418886: Merge commit '60c4660ba035bbfbcc84ac34129ce40e037c70ad'
[01:41] <cone-727> ffmpeg.git 03Justin Ruggles 07master:20c38c9c18b7: swscale: fix some undefined signed left shifts
[01:41] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:57570052e244: Merge commit '20c38c9c18b75e62f6c489d9b95bea88efdf3915'
[01:46] <cone-727> ffmpeg.git 03Justin Ruggles 07master:d9a542ace1b9: swscale: Set alpha to opaque for internal palettes.
[01:46] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:69505a89f1ed: Merge commit 'd9a542ace1b9ca5c0ea6d758127d03d4124b7801'
[01:56] <cone-727> ffmpeg.git 03Jean First 07master:bf2064f046af: tiffdec: rename variables for consistency and fix variable shadowing
[01:56] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:6145b1d9c11b: Merge commit 'bf2064f046af64c59a416c814474a39b0a457569'
[02:05] <cone-727> ffmpeg.git 03Justin Ruggles 07master:58bc38a5f224: tiffdec: use correct data type for palette entries and set alpha to 0xFF
[02:05] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:b394b9013ee4: Merge commit '58bc38a5f224d29b79338200459792c765c25fd5'
[02:06] <cone-727> ffmpeg.git 03Paul B Mahol 07master:de9d2705f61e: bytestream: add functions for accessing size of buffer
[02:06] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:d06a7530f7ea: Merge commit 'de9d2705f61ef569487ec5f8974a9c7ce34ec783'
[02:13] <cone-727> ffmpeg.git 03Justin Ruggles 07master:5748faf291fe: bytestream: add bytestream2_copy_buffer() functions
[02:13] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:31876ea62006: Merge commit '5748faf291fec297ef25d81962b52b3438f54278'
[03:27] <cone-727> ffmpeg.git 03Justin Ruggles 07master:0a467a9b594d: tiffdec: use bytestream2 to simplify overread/overwrite protection
[03:27] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:94cf83ff91b2: Merge commit '0a467a9b594dd67aa96bad687d05f8845b009f18'
[03:36] <cone-727> ffmpeg.git 03Justin Ruggles 07master:fdbe18b7b054: tiffdec: use a single strip if RowsPerStrip is 0
[03:36] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:75b4e4013796: Merge commit 'fdbe18b7b0545f9b5923d25a5433e4f735719ecc'
[03:51] <cone-727> ffmpeg.git 03Justin Ruggles 07master:a9b046fb0ae5: tiff: use a better name and enum values for PhotometricInterpretation
[03:51] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:b9b2f9d218ba: Merge commit 'a9b046fb0ae5395c4d1d8a82cd0d3e354b5034ed'
[04:00] <cone-727> ffmpeg.git 03Justin Ruggles 07master:345a96c327e8: tiffdec: remove an unneeded variable
[04:00] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:a64d6d515218: Merge commit '345a96c327e8f2a8077189af9f5e8d2b3f4ad5fe'
[04:03] <cone-727> ffmpeg.git 03Jean First 07master:e75ef2b7f48b: tiffdec: support predictor type 2 with RGB48 pixel format
[04:03] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:d7644e0fdb5c: Merge commit 'e75ef2b7f48b96a9b6c8646058713899d5ea5731'
[04:13] <cone-727> ffmpeg.git 03Tim Walker 07master:c8cf461c19e8: dcadec: Do not decode the XCh extension when downmixing to stereo
[04:13] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:952614b8996e: Merge commit 'c8cf461c19e8e35df4b7364d9b90aa42f1ab4560'
[04:50] <cone-727> ffmpeg.git 03Tim Walker 07master:801c39e1e305: dcadec: Use correct channel count in stereo downmix check
[04:50] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:a676e48d394c: Merge commit '801c39e1e3058fc4ba822bfb5d8612d777111e32'
[04:50] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:1dc2d4a8efd1: avcodec/tiff: remove unused variables
[05:42] <cone-727> ffmpeg.git 03Lukasz Marek 07master:88c4e5a049aa: lavd/avdevice: fix buildbreak when included to C++ code
[11:51] <wm4> this is the display matrix API which collects feedbacks and requests from
[11:51] <wm4> our downstreams users (such as VLC, mpv and ffmpeg) and developers.
[11:51] <wm4> trolling 2k
[11:52] <ubitux> & :/
[11:58] <ubitux> wm4: any more comment on the subtitles fonts patch?
[12:01] <wm4> no
[12:03] <ubitux> ok thx
[12:22] <Compn> wm4 : except i havent seen anyone on ffmpeg being upset by the downstream troll. its ineffective
[12:22] <nevcairiel> they keep trying anyway
[12:23] <bencoh> :/
[12:23] <Compn> i think the qatar thing annoys a few people :d
[13:08] <cone-719> ffmpeg.git 03Facundo Gaich 07master:7e6b3ad6930c: lavfi/subtitles: load attached fonts to libass.
[13:23] <pross-au> channel topic suggestion. *2.2.1* has been released
[13:23] <ubitux> you are +o
[13:25] <pross-au> thanks!
[13:43] <ubitux> wm4: still looking forward utf16 support btw :)
[13:43] <wm4> oh yeah I should do that
[14:13] <cone-719> ffmpeg.git 03Anton Khirnov 07master:f7c5fd81516f: resample: implement flushing
[14:13] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:cbfa71d689a1: Merge commit 'f7c5fd81516fb7b186c64c66dc080e1139586226'
[14:29] <cone-719> ffmpeg.git 03Anton Khirnov 07master:13164e479f1c: FATE: add tests for more resampling modes
[14:29] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:c955525f724d: Merge commit '13164e479f1c87829096c19003d2cc658ccc1fd4'
[14:34] <cone-719> ffmpeg.git 03Anton Khirnov 07master:ac976ed91e32: lavr: allocate the resampling buffer with a positive size
[14:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:c11aa9d29aae: Merge commit 'ac976ed91e323754e9a84509873ebdb437372797'
[14:46] <cone-719> ffmpeg.git 03Alessandro Ghedini 07master:aaab192df24a: af_volume: implement replaygain clipping prevention
[14:46] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:878f8b0d26e9: Merge commit 'aaab192df24a90f4450285cfb73b395cf495b462'
[14:48] <cone-719> ffmpeg.git 03James Almer 07master:0f524b6c6909: x86/synth_filter: remove the fma3 version ifdefs
[14:48] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:b6c1e80bf4a9: Merge commit '0f524b6c690962d7ebbd59ded74288e1e9b0f009'
[14:54] <cone-719> ffmpeg.git 03Peter Ross 07master:d95c9679f727: avformat/iff: remove unused code
[14:54] <cone-719> ffmpeg.git 03Peter Ross 07master:e311b7446a8c: avformat/iff: indent
[15:34] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:ca6dd53a734d: ffprobe: do not lose non ascii characters in non utf8 strings
[16:56] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:a31547ce2e5c: avutil/avstring: do not lose ascii characters when decoding non utf-8 with av_utf8_decode()
[18:13] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:289f02f923ac: avformat/utils: Inject global side data into first packet.
[18:13] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:c0d62d999d0e: avformat/utils: factorize if()
[18:14] <wm4> michaelni: uh
[18:14] <wm4> michaelni: you just broke the api
[18:15] <michaelni> wm4, ?
[18:15] <michaelni> please explain
[18:15] <wm4> it was decided that if side data remains constant over the whole file, it should be added to AVStream only
[18:15] <wm4> and avplay AFAIK has something special to copy the AVStream side data to the first packet
[18:17] <michaelni> "it was decided" <-- dont remember the mail on ffmpeg-devel
[18:17] <wm4> this wasn't discussed on ffmpeg-devel at all
[18:18] <michaelni> ok then it hanst been decided
[18:18] <michaelni> ffmpeg API must be discussed on ffmpeg-devel
[18:18] <wm4> ah it wasn't added to avplay: https://git.libav.org/?p=libav.git;a=commit;h=59444c76e6d43529a12dbd80b6dd2…
[18:18] <wm4> just avconv
[18:19] <michaelni> yes and ffmpeg as i merged that
[18:19] <michaelni> and iam sure not fond of that copying
[18:19] <michaelni> but IMHO as every application that uses avfilter pretty much needs it
[18:19] <michaelni> it would be better in avformat
[18:19] <michaelni> instead of in every app
[18:20] <wm4> you assume they even pass through side data
[18:20] <michaelni> i guess i shouldnt assume that but some filters need it
[18:20] <michaelni> like af_colume
[18:20] <michaelni> af_Volume
[18:21] <michaelni> also iam happy to revert if someone has a better idea
[18:21] <wm4> the reason it wasn't added automatically to the first packet is because it's not really clear what the first packet is
[18:21] <michaelni> its also not clear in ffmpeg
[18:21] <wm4> the first packet that is demuxed? after each seek? what if the application internally disables and then enables the stream again?
[18:22] <michaelni> what if the output from ffmpeg is broacast and the user starts viewing in the middle
[18:22] <wm4> and application could throw away the lavfi filter graph when disabling the stream, and build it again when reenabing the stream
[18:22] <wm4> same on seek (even ffplay recreates the lavfi graph on seek)
[18:23] <wm4> so the only reasonable way to auto-add it is probably adding it to _every_ packet
[18:23] <wm4> but that seems wasteful
[18:30] <michaelni> i agree but i think the copying itself is the problem and its better to copy in avformat until some other solution exists than in every app
[18:30] <michaelni> ill fix the seek case in a moment
[18:32] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:82796aacab1b: avformat/utils: reinject global side data on seeks
[18:34] <michaelni> the copying i avformat also should be enough for many apps, so only complex apps (which for example enable streams in the middle) would need to copy
[18:37] <wm4> also, the idea was that some side data types happen on _change_
[18:37] <wm4> so your changes still break the api
[18:42] <michaelni> is that documented and if so where ?
[18:44] <michaelni> also i was just trying to make the life of application devels easier by factorizing code that apps would need into avformat
[18:44] <michaelni> if app devels dont want that i can revert
[18:46] <wm4> michaelni: it's documented in the AVStream.side_data doxygen
[18:47] <wm4> it explicitly says that side data listed in AVStream never appears in the AVPackets
[18:47] <wm4> I can see that automatic injection might be helpful
[18:47] <wm4> but then it should be an option that is off by default
[19:36] <michaelni> wm4, posted patch that implements that
[21:29] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:eccec203978e: avformat/mp3enc: drop redundant and uninitialized variable
[23:45] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:5ec81929497c: avcodec/fft-test: use av_malloc_array()
[23:45] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:d42a61922be7: avcodec/psymodel: use av_malloc(z)_array()
[23:45] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:2ebacb0fc444: avformat/network: use av_malloc_array()
[00:00] --- Mon Apr 14 2014
1
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[08:57] <simpleirc2> #ffmpeg-devel
[13:28] <g44> anyone have an idea how to extract motion vectors out of ffmpeg?
[14:14] <pyBlob> how can you specify the video rate when using ffplay, the options -r/-vr don't work there?
[14:39] <pyBlob> ... or is it possible to change the speed without reencoding?
[15:34] <klaxa> pyBlob: see the setpts filter, or even easier, here: https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20…
[15:34] <klaxa> i think you can copy the video codec
[15:49] <LDawg> I try to convert my images (captured at 30fps) to a webm video with the following command, but I cant get the framerate right. The video is always too fast: http://pastebin.com/rAjJ2r7W
[15:49] <LDawg> I also tried setting -r to 30 for the input file but this didnt help
[15:57] <LDawg> But that is my ffmpeg command
[15:58] <sacarasc> What about the complete console output?
[16:19] <BlackBishop> Is it possible to control ffmpeg once it's launched ? As in, it's doing some transcoding and streaming to a rtmp server and I want it to pause/resume ..
[16:20] <BlackBishop> Preferably via some socket file so I can do it via some web interface
[16:27] <LDawg> I try to convert my images (captured at 30fps) to a webm video with the following command, but I cant get the framerate right. The video is always too fast: http://pastebin.com/rAjJ2r7W
[16:27] <LDawg> I also tried setting -r to 30 for the input file but this didnt help
[16:27] <sacarasc> LDawg: What about the output?
[16:29] <LDawg> http://pastebin.com/ThSPnYA0
[16:30] <sacarasc> That says the input is 25, which is fine, what about the output for the output?
[16:34] <LDawg> http://pastebin.com/aHPsxHjB
[16:56] <LDawg> anything unusual?
[16:57] <sacarasc> Not that I can see, but I suck at this kind of thing. :D
[20:07] <BlackBishop> so .. no way to control ffmpeg after it started ?
[23:30] <dwirc> Heya
[23:30] <dwirc> Question... reagding live streaming and data detection
[23:32] <dwirc> So, I'm pulling an RTMP stream live from a surveillance camera on my front door, then doing various things with the stream (restream and storage, bsaically, nothing too fancy, but that's beside the point)
[23:33] <dwirc> Occasionally, something hicchups, and the ffmpeg pull stops, and hangs, despite the camera still being available.
[23:33] <dwirc> Once it stops, it stops... I've let it sit for 24 hrs before and it never restarts, nor does it die.
[23:33] <dwirc> Is there any simple way I can have ffmpeg detect a lack of data for ... X number of seconds, or something like that, and retry the connection?
[00:00] --- Mon Apr 14 2014
1
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[00:13] <ubitux> do we have an audio denoiser?
[00:14] <BBB> ubitux: there's some filters thereof in various speech decoders (e.g. wmavoice)
[00:14] <BBB> you could extract/duplicate that into its own filter
[00:14] <BBB> I don't know how good it is obviosuly
[00:14] <ubitux> it looks like something we need
[00:14] <ubitux> a lot of talk recordings are pretty bad/noisy
[00:15] <ubitux> i wonder how much that could be improved
[00:15] <BBB> sorry not an audio expert, that's more a question for someone who knows audio
[00:15] <BBB> I know some basics but not much...
[00:15] <Skyler_> audio denoising is rather difficult
[00:15] <iive> maybe take a look of karaoke filter, and do the opposite :)
[00:16] <Skyler_> one approach is to allow the user to explicitly specify an area to use as a noise profile (e.g. what audacity does)
[00:16] <Skyler_> beyond that, automated noise reduction filters seem like the domain of high-priced high-latency VST filters
[00:16] <Skyler_> because it's way harder to distinguish noise and content than in video
[00:16] <BBB> yeah that's basically what wmavoice does, you select the fft partitions of interest and the rest is dampened
[00:16] <iive> but voice is usually low frequency (<3kHz) so you can simply cut anything above.
[00:16] <Skyler_> noise is very often all frequencies though
[00:17] <BBB> didn't know you were that much an audio expert Skyler_
[00:17] <BBB> that could be useful
[00:17] <iive> that's called white noise.
[00:17] <Skyler_> I'm not, really
[00:17] <Skyler_> the only VST filter I was able to find for realtime noise reduction (not gating) was expensive and required 40,000 samples latency
[00:18] <iive> most noise is high frequency and power outlet frequency
[00:20] <Skyler_> http://i.imgur.com/Cx1F3aj.png this is an fft of noise from my laptop micrphone at work
[00:20] <Skyler_> most is probably from the fans.
[00:20] <Skyler_> I think it's pink noise, not white noise.
[00:21] <iive> white noise should have same amplitude on all frequencies. afair
[00:24] <iive> random with constant power...
[00:38] <cone-476> ffmpeg.git 03Carl Eugen Hoyos 07master:8b122937af9a: Warn if rawvideo and an unreadable pix_fmt are written.
[00:38] <cone-476> ffmpeg.git 03Carl Eugen Hoyos 07master:c43222f436e9: Improve amr bitrate calculation for VBR files.
[00:38] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:27763db928b5: Merge remote-tracking branch 'cehoyos/master'
[01:44] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:66e959682c49: avcodec/mpegvideo_enc: override bitrate tolerance when its value is too small
[02:18] <cone-476> ffmpeg.git 03Michael Schenk 07master:513a431e073a: avformat/hls: reduce memory usage by using dynamically allocated url/key
[04:41] <michaelni> Skyler_, is "[FFmpeg-devel] [PATCH] avcodec/libx264: Implement reference frame count limiting based on level" ok to be used under LGPL ? (its based on (GPL) x264 code)
[05:03] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:f683895dee05: avformat/img2dec: support nanosecond file times
[05:52] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:46375e9a97c6: avcodec/diracdec: use av_malloc(z)_array
[05:52] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:fb4ac46d7f51: avcodec/elbg: Use av_malloc(z)_array()
[13:04] <pross-au> Can software under "BSD 3-clause" license be contributed into FFmpeg libavcodec?
[13:06] <BBB> yes
[13:06] <BBB> (not a lawyer)
[13:06] <BBB> :-p
[13:06] <pross-au> What! You're everything else :)
[13:06] <pross-au> Thanks
[13:23] <Mavrik> 3-clause BSD is compatible with GPL :)
[13:23] <Mavrik> http://www.gnu.org/licenses/license-list.html#GPLCompatibleLicenses
[13:30] <BBB> pross-au: I know! I'm thinking I should set up my own law firm just because ;)
[15:37] <BBB> smarter: (or jeeb): is 32x32 transform puma-only in hevc?
[15:37] <BBB> luma*
[15:37] <BBB> stupid spell correct
[15:41] <Rodeo> BBB: not that stupid, that made me laugh
[15:41] <Skyler_> yes, only big cats are allowed to use it
[15:41] <Skyler_> ocelots aren't allowed.
[15:41] <smarter> in 4:2:0, yes
[15:42] <BBB> it's very hard to have conversations about subjects that are not trivial talk if that's the only thing your dictionary supports and it auto-corrects
[15:42] <BBB> it's like spell-correction in text messaging
[15:42] <BBB> it's infuriating
[15:42] <BBB> every time I type my son's name, it corrects it from 'Fre' to 'fee'
[15:42] <BBB> so I keep sending my wife msgs how's fee doing
[15:42] <BBB> and she's like "who's fee"
[15:42] <smarter> hah
[15:43] <smarter> add it to your phone's dictionary? :p
[15:43] <BBB> if it doesn't auto-add corrected words, it's infuriating
[15:43] <BBB> it should auto-add
[15:43] <smarter> yeah it should do that
[15:43] <JEEB> I had that problem years ago, so I actually never enabled auto-correction on any of my mobile handsets
[15:43] <BBB> nowadays it's enabled by-default
[15:43] Action: JEEB mish-mashes finglishrussian
[15:43] <BBB> smarter: thanks for confirming
[15:44] <BBB> did I mention the clef coding looks pretty complex?
[15:44] <smarter> I never know how to make auto-correct work,
[15:44] <BBB> coef*
[15:44] <smarter> :D
[15:44] <smarter> it is, but I think there's a paper about it if you look on IEEE Spectrum
[15:44] <BBB> link?
[15:44] <smarter> err, ieee explore
[15:45] <smarter> sure
[15:45] <smarter> one sec
[15:45] <BBB> ty!
[15:46] <smarter> BBB: https://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=06324418
[15:46] <BBB> \o/
[15:46] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:36053aeff651: vcodec/error_resilience: use av_malloc_array()
[15:47] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:713e62e91d23: avcodec/faxcompr: use av_malloc_array()
[15:47] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:e89f3d0ed212: avcodec/ffv1: use av_malloc_array()
[15:47] <BBB> it doesn't load :(
[15:47] <BBB> crap
[15:47] <JEEB> yeah, you need to login first
[15:47] <smarter> I'm not logged and I can access it
[15:47] <BBB> their site seems dead to me
[15:47] <BBB> the css of the front page is missing
[15:47] <smarter> BBB: try http://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=06324418
[15:48] <JEEB> yeah, CSS is missing :D
[15:48] <smarter> your browser may prevent the https version from loading because of mixed content
[15:48] <BBB> ah that works
[15:48] <BBB> ty
[15:48] <BBB> oh so it's chrome
[15:48] <BBB> nice
[15:48] <JEEB> firefox does the same :)
[15:48] <BBB> let me read this
[15:48] <BBB> bbl
[15:49] <smarter> well, it's probably my fault because I use HTTPS Everywhere and it https-ed this link when it shouldn't have
[18:24] <Compn> ubitux : https://github.com/kodabb/libav/commits/rotation
[18:26] <ubitux> sure whatever
[18:28] <Compn> was worth a shot anyhow
[18:28] <Compn> maybe not
[18:31] <Compn> like talking to brick walls
[18:45] <Spideru> Hi, I am running fate on ffprobe, after applied a small modify to code (trying to fix http://trac.ffmpeg.org/ticket/3523) I launch the fate test with "make fate-ffprobe_compact". After a while I obtain an error http://pastebin.com/MHTTe8yb . the content of err file is http://pastebin.com/0kYw4cZa . How can I understand what test expect and what obtain? Thank you
[18:48] <ubitux> -packet|codec_type=audio|stream_index=0|pts=0|pts_time=0.000000|dts=0|dts_time=0.000000|duration=1024|duration_time=0.023220|convergence_duration=N/A|convergence_duration_time=N/A|size=2048|pos=572|flags=K
[18:48] <ubitux> +packet|codec_type=audio|stream_index=0|pts=0|pts_time=0.000000|dts=0|dts_time=0.000000|duration=1024|duration_time=0.023220|convergence_duration=N/A|convergence_duration_time=N/A|size=2048.000000|pos=572|flags=K
[18:48] <ubitux> the size changed from int to float or something?
[18:50] <Spideru> yes! Thank you, was just under my nose :|
[20:19] <ubitux> in the intel man, vinserti128 (1594/3355), operation section, the pseudo code is presented with some kind of temp memory area; what the reason of not writing in DEST directly?
[20:19] <ubitux> i suppose this is to avoid writing twice in the DEST?
[20:20] <ubitux> but if so, how is that important for a register?
[20:25] <ubitux> (btw, i think there is a typo in that page "... into an the destination operand" - no i'm too lazy to report)
[20:36] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:4f698be8f65d: avutil/log: add av_log_get_flags()
[21:49] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:617a1a98a6be: mp3enc: Properly write bitrate value in XING header
[21:49] <cone-727> ffmpeg.git 03Michael Niedermayer 07master:571d2ffa4745: Merge commit '617a1a98a6be3e59db6fbfc21afab2fb9a049c03'
[00:00] --- Sun Apr 13 2014
1
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[01:52] <jkli> hi
[01:53] <jkli> i want to reduce the motion from my encodings
[01:53] <jkli> any idea?
[01:53] <jkli> should i set rframes to 1?
[03:21] <SurfMan> can I concatenate h.264 videos using demux so I don't have to re-encode? I'm trying to make a title frame on a video, and have encoded silent stereo audio but seemingly have some sort of stream issue.
[03:25] <SurfMan> http://pastebin.com/80B6eqZ5
[03:25] <SurfMan> that is the detail on what I am trying to do
[03:26] <SurfMan> the resulting mix plays fine for the title frame duration, then plays gibberish video for the duration of what I would expect the mix to be.
[05:37] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key can someone help me how to download from such streaming
[05:38] <Zeranoe> adi10289: Link to video?
[05:39] <adi10289> 1 min actually i m trying to download premium video but i will provide free one for you hold on
[05:39] <adi10289> https://spuul.com/movies/6077-soundtrack
[05:40] <adi10289> you need to make free account on spuul first
[05:47] <adi10289> did u checked it Zeranoe?
[05:54] <adi10289> any update :(
[11:46] <teegee> I thought, by the time I get to extracting the frame and looking at packet->stream_index all dereferencing is done
[11:46] <teegee> av_read_frame() and all previous methods don't return an error
[11:51] <teegee> hmm
[11:52] <teegee> when I compile that code as an executable it works
[11:52] <teegee> but when I move it to a dll it fails
[12:04] <teegee> ah.. it's the struct member alignment
[12:05] <teegee> I set it to 1 byte to avoid problems when calling the library from LabVIEW. Apparently, that screws with the alignment of the dependencies too
[12:33] <teegee> alright, cheers
[12:33] <teegee> cu Mavrik
[13:53] <cm13g09> OK, so I've got an h264 RTP stream that I want to pump into ffserver (for streaming over HTTP)
[13:53] <cm13g09> problem is, ffmpeg seems to think the "best" thing to do is decode it, then re-encode it using libx264
[13:53] <cm13g09> It's OK - I'm going mad :P
[13:55] <cm13g09> Sorted with the help of a -vcodec copy
[14:14] <alfoos> hi, i am trying to stitch 4 live rtsp streams (from ip camera) using ffmpeg. I am getting huge packet loss (tried tcp) when doing live stitching. However i am not getting packet loss when saving individual stream to local file. what would be the most ideal way to do the stitching without packet loss
[15:45] <jkli> hi all
[15:45] <jkli> can somebody tell me in layman terms what subq is good for?
[16:05] <jkli> hm
[16:06] <jkli> delogo x and y, do they start from bottom left corner?
[16:06] <jkli> if yes, what do i do with videos who have different resolutions?
[16:18] <jkli> im using filter_complex is there an equivalent for delogo?
[16:34] <mikotoimouto> I've been looking around and I can't seem to find out how to bypass the fact that ffmpeg froces me to maintain aspect ratio. Using the aspect ratio argument only allows me to crop and the scale filter simply ignores your input and outputs the input aspect ratio anyway
[16:36] <mikotoimouto> http://pastebin.com/7g709YFM
[16:36] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key can someone help me how to download from such streaming
[16:37] <klaxa> mikotoimouto: and what exactly is the problem? that should scale your video to 480x272 resulting in aspect ratio ~16:9
[16:37] <mikotoimouto> klaxa: It doesn't
[16:37] <mikotoimouto> that's the problem
[16:37] <mikotoimouto> It forces the aspect ratio to remain the same as the input
[16:38] <mikotoimouto> In the manpage it even says "The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.
[16:38] <mikotoimouto> And there doesn't seem to be any way to bypass this
[16:38] <klaxa> have you tried to set -ratio 16:9 ?
[16:38] <mikotoimouto> -ratio doesn't seem to exist
[16:38] <mikotoimouto> I tried -aspect 16:9
[16:38] <mikotoimouto> But that crops the video
[16:38] <klaxa> ah
[16:38] <klaxa> i meant -aspect
[16:39] <JEEB> so your input is not 16:9, and you want to have a 16:9 clip come out?
[16:39] <mikotoimouto> Yes
[16:39] <JEEB> ok
[16:40] <JEEB> you'll probably want to then resize to aspect ratio-wise correct size, and then pad. that way you keep the ratio, and have a 480x272 picture
[16:40] <JEEB> or you just set either the width or height, and let the other be automagic :P
[16:40] <JEEB> (and don't pad it to 16:9)
[16:40] <mikotoimouto> I need the output to be stretched to the ratio though
[16:40] <mikotoimouto> I'm intentionally trying to make the output wider
[16:40] <JEEB> ah
[16:40] <JEEB> if you are intentionally breaking it
[16:41] <JEEB> then just use the setsar filter :P
[16:41] <JEEB> after the resize
[16:41] <JEEB> first scale, then setsar to 1:1
[16:41] <JEEB> but yeah, you should /only/ be doing that when you want to break the aspect ratio of the picture
[16:42] <adi10289> when it comes to encoding i always trust megui with avisyth :D
[16:42] <Krausen_> I'm attempting to generate video effects with ffmpeg (for now, a simple fade in, out) but the problem I am having is the resulting h.264 video cannot be concatenated (using demux) with the video from my camera. Could it be differences in the pix_fmt creating this apparent incompatability?
[16:43] <Krausen_> I really am trying to avoid transcoding
[16:43] <JEEB> Krausen_, most definitely. Also there can be plenty of other things regarding the parameter sets that can cause derps
[16:43] <mikotoimouto> Thanks, that worked perfectly
[16:43] <jkli> does anybody know how to pass relative positions to ffmpeg?
[16:44] <jkli> i want delogo to start from the bottom right corner with a retangle 50x50
[16:44] <JEEB> mikotoimouto, just remember that it breaks the picture so you don't want to really use it that way generally :)
[16:44] <mikotoimouto> I know that
[16:44] <Krausen_> I'm thinking the best approach would be to generate title slides / effects that match the camera's h.264 parameters, if that is possible.
[16:44] <JEEB> have fun with that :P
[16:45] <JEEB> it might be simple, or it might be harder. also depends on whatever you're wanting to play that stuff on
[16:45] <Mavrik> Krausen_, honestly the best approach is reencoding the whole set
[16:45] <Mavrik> because that will create the least broken output stream
[16:45] <Krausen_> I lack the cpu horsepower :(
[16:45] <Krausen_> but this sort of thing fairly typical?
[16:46] <Mavrik> concating stuff where SPS/PSS, timebases, pixel formats and encoding parameters don't match will give you partially broken outputs that won't be handled well by all players.
[16:46] <Krausen_> so there really isn't any advantage to cameras doing native h.264 if one intends to do any bit of editing?
[16:47] <Mavrik> you never do editing on encoded streams :)
[16:47] <Krausen_> cheap camera
[16:47] <Krausen_> its only format is h.264 encoded vid
[16:47] <adi10289> can we pass cookies in ffmpeg?
[16:47] <Mavrik> or, if I say it another way, I haven't seen an editing process that wouldn't produce the finished product by reencoding everything
[16:47] <JEEB> ^ this
[16:48] <Mavrik> usually called "rendering" :)
[16:48] <JEEB> while it can be theoretically possible to do minimal amount of re-encoding, you are going to have to go through that road yourself :P
[16:50] <Krausen_> seems like I've stumbled upon a nasty reality
[16:50] <adi10289> lol someone help me too :(
[16:50] <Krausen_> got all excited when I read about ffmpeg's demux concat too :(
[17:05] <jkli> argh
[17:06] <jkli> i cant believe delogo didnt came with input height and input width variables -_-
[17:15] <tulcod> the latest ffmpeg release does not build against the latest x265 library: the fix for this is in commit 4a2226451, which came out right after 2.2.1
[17:16] <JEEB> the supported x265 version in a release won't be changed
[17:16] <JEEB> because it would break backwards compatibility, among other things
[17:16] <JEEB> releases are like that, and x265 is a moving-as-hell target
[17:16] <tulcod> okay, let me look up what the supported version is then :)
[17:17] <JEEB> whatever there was at the time of the branching of the 2.2 branch
[17:17] <JEEB> actually it might have actually been a better alternative to not support libx265 at all in the releases yet, but I digress :P
[17:19] <tulcod> well i needed it at some point to play some video, so was happy to have it
[17:19] <JEEB> uhh
[17:19] <JEEB> it's an /encoder/
[17:19] <tulcod> ugh, i hate the bitbucket website. when did fscking x265 0.9 come out?
[17:19] <JEEB> you don't need it to /play/ anything
[17:19] <tulcod> err. okay i can't quite recall what i needed to do
[17:19] <tulcod> but for some reason i needed it
[17:19] <JEEB> libx265 is purely the encoding side of things
[17:20] <JEEB> libavcodec has its own HEVC decoder, which works fine
[17:20] <tulcod> okay whatever. i can't really have an opinion about this
[17:20] <JEEB> (and libx265 doesn't have any more of a decoder than what is absolutely needed for encoding, and it does not provide an interface for it unsurprisingly)
[17:20] <JEEB> :)
[17:20] <JEEB> anyways, just saying that what you said was invalid as libx265 has no decoding capabilities
[17:23] <JEEB> but yeah, random mercurial revision from the time of the 2.2 branching should work, I guess
[17:25] <JEEB> my real recommendation would be to just use the up-to-date command line encoder and piping the raw data from ffmpeg
[17:25] <JEEB> and ignore the libx265 libavcodec wrapper
[17:26] <tulcod> just trying to gather information to report to my distro's (unofficial) packager
[20:14] <adi10289> aahh no one helping out this poor guy :(
[20:14] <Mavrik> O.o
[20:15] <Mavrik> which.
[20:15] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key but not able to need help regarding same
[20:17] <adi10289> Mavrik : can u help me pls regarding same
[20:17] <Mavrik> I don't understand your sentence.
[20:17] <Mavrik> "regarding same" what's that supposed to mean?
[20:18] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key but not able to need help regarding same
[20:20] <adi10289> Mavrik : can u help me regarding above issue
[20:21] <Mavrik> I don't think so.
[20:22] <adi10289> c no one help this poor guy (
[00:00] --- Sun Apr 13 2014
1
0
[01:19] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:268b1eae2276: avcodec/mpegaudioenc_template: default to 384k bitrate as default
[01:27] <cone-593> ffmpeg.git 03Luca Barbato 07master:a7448064c526: mp2: Do not force a samplerate
[01:27] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:1de8a4f73665: Merge remote-tracking branch 'qatar/master'
[06:37] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:d9fef740bd6a: avcodec/dnxhdenc: use av_malloc_array()
[10:00] <relaxed> 2.2.1 probably needs mentioned on the front page
[11:20] <wm4> "IMHO, the fact that we merge Libav code does mean that we cooperate with them, even if Libav doesn't want to."
[11:20] <wm4> that thread is going places
[11:21] <ubitux> well that's pretty accurate
[11:21] <ubitux> it means there are some recognition to that project
[11:21] <wm4> you can't cooperate with someone against his will
[14:13] <Compn> iive : just submit a patch to libav-devel and call the bluff
[14:13] <Compn> dont bother arguing in mail threadsd
[14:14] <Compn> ubitux : bluff time!
[14:14] <Compn> :)
[14:25] <cone-476> ffmpeg.git 03Vittorio Giovara 07master:89ef08c992c4: DNxHD: K&R formatting cosmetics
[14:25] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:bad101ab8a54: Merge commit '89ef08c992c484a46711b1a68a988303679c288e'
[14:32] <ubitux> michaelni gmx.at: Envelope expired
[14:32] <ubitux> :(
[14:32] <wm4> wut
[14:32] <michaelni> what ?
[14:32] <ubitux> i've sent a mail to reimar as well
[14:32] <ubitux> it has expired
[14:32] <ubitux> i mean it's been trying to send it for a few days now
[14:33] <ubitux> maybe that's a problem with my smtpd though... :)
[14:38] <cone-476> ffmpeg.git 03Timothy Gu 07master:676856204b84: DNxHD: make get_pixel_8x4_sym accept ptrdiff_t as stride
[14:38] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:ee2a43798c2a: Merge commit '676856204b84dfaffe329dce8502e834a0dbdb19'
[14:45] <ubitux> Compn: huh?
[14:53] <cone-476> ffmpeg.git 03Timothy Gu 07master:71c32ed5335a: DNxHD: convert inline asm to yasm
[14:53] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:4899ccd29572: Merge remote-tracking branch 'qatar/master'
[17:24] <koda> hi, im designing an api for exporting rotation information from containers like mov and other codecs
[17:24] <koda> is there any preference in what should be exported?
[17:24] <koda> like full matrix, or just rotation angle or more fields even
[17:25] <ubitux> just like it's done currently, "rotate" tag is enough
[17:25] <ubitux> (at stream level)
[17:26] <koda> ok so no change would be your preference
[17:26] <ubitux> this information is currently extracted from a stream box header (tkhd, track header)
[17:26] <ubitux> so it's information at stream level to start with
[17:26] <Compn> ubitux : you dont want it in metadata or stream data ,how will vlc read it ?
[17:26] <ubitux> if at some point they decide to add frame level rotation that will be stored differently
[17:26] <Compn> er s/streamdata /side data
[17:27] <koda> the api i had in mind would remain in stream level, and there are codecs that export rotation at codec level
[17:27] <ubitux> Compn: by looking at the stream metadata?
[17:27] <koda> also metadata is usually for textual information, not this essential information
[17:27] <Compn> koda : so you want to export it how ?
[17:28] <koda> avstream side data, with possible avframe side data translation
[17:28] <ubitux> the "text metadata" is not optimal but it has a lot of benefits
[17:29] <ubitux> typically, it is automatically exported to ffprobe, so you can do stuff like that:
[17:29] <ubitux> - ffprobe -v 0 -of flat -show_entries stream_tags=rotate ~/samples/iphone-rotate/r180.mov
[17:29] <ubitux> streams.stream.0.tags.rotate="180"
[17:29] <koda> ok lets work on the not optimal and provide the same number of benefits with this api
[17:30] <ubitux> you could do both
[17:30] <ubitux> not sure how
[17:30] <koda> for instance you can do the same with the showinfo filter
[17:30] <ubitux> it requires decode, so it sucks
[17:31] <koda> well you usually want to do something with rotation info, so might as well decode the frame
[17:31] <ubitux> no
[17:32] <koda> just knowing that a video is rotated doesnt say much
[17:32] <ubitux> because it will be done at probe stage
[17:32] <koda> and the ffprobe case is somewhat limited/lmiting
[17:32] <ubitux> before setting up your decode workflow
[17:32] <ubitux> no it's not
[17:33] <koda> why?
[17:34] <ubitux> typically you extract that information at probe stage because you want to insert automatically transpose or vflip/hflip filter, or rotate filter for special rotation in another command, where you would drop the metadata itself
[17:35] <ubitux> if you can do that all transparently with a single command then that's fine as well
[17:35] <koda> the problem is that this is not textual information (metadata) but additional information essential for correct displaying
[17:35] <koda> and moreover you can only represent rotation with the current api, i wanted to add hflip vflip and maybe translation like you mentioned
[17:35] <koda> and adding a metadata entry for each of them is ugly
[17:36] <ubitux> let me re-explain
[17:36] <ubitux> currently the common workflow is the following:
[17:36] <ubitux> you probe the input file to get the rotation (and generally a lot of other stuff)
[17:36] <ubitux> then you setup a filtergraph to hard-rotate the output video based on that value
[17:37] <ubitux> depending on the value you construct a different filtergraph
[17:37] <ubitux> using transpose, vflip/hflip, or rotate filter
[17:38] <ubitux> (and you also add -map_metadata -1 to drop the metadata itself so the it's not rotated twice in the output)
[17:38] <koda> well thats a (quite complex) usecase, but its nothing that could not be done automatically with the new api
[17:38] <ubitux> this is what most people do to handle players with no soft rotate, which happens a lot
[17:38] <koda> instead of having to probe first, youd let the api detect the operation to perfom
[17:39] <koda> and the filter would just use this value to perform it automatically
[17:39] <koda> its somewhat less work for them overall too
[17:39] <ubitux> well if you can do that logic in ffmpeg & ffplay sure&
[17:39] <ubitux> but you know you can also do that with string metadata as well
[17:40] <ubitux> strcmp("90" ...) and the 4 other cases
[17:40] <ubitux> one nice thing missing would be to support special rotation matrix
[17:40] <koda> ok, but rotation is not really metadata, but something needed for correct displaying
[17:40] <koda> or we could just add every information like pixel format as metada then :p
[17:41] <ubitux> it belongs perfectly in metadata
[17:41] <ubitux> rotation in exif are also called metadata for instance
[17:41] <koda> exif is textual information, mov thd atom is not
[17:42] <ubitux> i'm not sure exif rotation is stored as "text" and it doesn't change anything
[17:42] <ubitux> it's really not much a problem to deal with strings here
[17:42] <koda> afaik exif reports it as rotation , mov as rotate , i see a little problem there too
[17:42] <Compn> i think koda means he wants to rotate the video before , so all of the player applications dont have to implement the check, and implement the rotate filtering ?
[17:43] <ubitux> Compn: you can't do that at demux level, you have to use a filter
[17:43] <Compn> 'auto-rotate'
[17:43] <koda> you can, if the api is not just metadata
[17:43] <ubitux> it doesn't change the data
[17:43] <ubitux> demuxer has blob of data not decoded
[17:44] <ubitux> you're not going to rotate in the demuxer, that's what i'm saying
[17:44] <ubitux> anyway, you could have stream metadata in a binary representation if you prefer that way
[17:44] <Compn> ubitux : maybe he wants a way for the filter api to read the demuxer rotate flag ?
[17:44] <ubitux> but i'm not sure it will simplify much things
[17:44] <Compn> right
[17:44] <ubitux> Compn: i understand what he wants you know
[17:44] <Compn> ok then :P
[17:44] Action: Compn goes back under his rock
[17:45] <ubitux> as long as it's not at frame level that should be fine
[17:45] <koda> ubitux: well binary represenation is again annoying to parse, i was thinking something more simple to report
[17:45] <ubitux> well i mean something API-usable
[17:45] <koda> eg fields or maybe an enum with data
[17:46] <ubitux> you also have the disposition stuff
[17:46] <ubitux> you can add a rotation disposition bit, meaning that some fields are set and should be honored
[17:46] <ubitux> and then use that
[17:47] <ubitux> AV_DISPOSITION_ROTATED
[17:47] <ubitux> then set whatever field you'd like to have
[17:47] <koda> so youd prefer enum for simple transformation
[17:48] <ubitux> note that "rotate" tags is already in heavy use
[17:48] <koda> + other fields
[17:48] <ubitux> ?
[17:49] <ubitux> you add a disposition (there is already an api around that, ffprobe partially supports it)
[17:49] <ubitux> then in the AVStream you add a rotation_matrix field with whatever struct you fancy
[17:50] <koda> ah, im not familiar with it
[17:50] <ubitux> it's simple
[17:51] <ubitux> in lavf/mov.c you video_stream->disposition = AV_DISPOSITION_ROTATED; video_stream->rotation_matrix = ...
[17:51] <cone-476> ffmpeg.git 03Lukasz Marek 07master:6db42a2b6b22: lavd: add device capabilities API
[17:51] <cone-476> ffmpeg.git 03Lukasz Marek 07master:c02e3dca2d49: lavd: add control message enums
[17:51] <ubitux> then add some print stuff in ffprobe.c
[17:51] <ubitux> that's all, and that should do it
[17:52] <ubitux> ah, and update the muxer to honor it as well
[17:52] <ubitux> we can already auto-insert the filters in ffplay btw&
[17:52] <ubitux> maybe we should do that heh
[17:54] <koda> oh i see, its avformat internal api though
[17:54] <ubitux> yes, format level, like currently, just not stored as string
[17:54] <koda> as soon as codecs implement rotation too its going to be in the way
[17:54] <ubitux> do you have one in mind?
[17:54] <koda> h264 and hevc have a sei message
[17:55] <ubitux> what happens if both are set?
[17:55] <ubitux> (codec and stream)
[17:55] <Compn> same thing that happens when codec and container have different sar/par
[17:56] <ubitux> what happens then?
[17:56] <Compn> ffmpeg picks one (i forgot which)
[17:56] <Compn> or maybe mplayer does
[17:56] <Compn> oh
[17:56] <Compn> fun fact
[17:56] <Compn> different players pick different ones
[17:56] <Compn> so it will look correct in mplayer but not wmp
[17:56] <Compn> :)
[17:56] <ubitux> koda: i'd say that in the case of codec level, the codec should do the rotation itself
[17:57] <ubitux> (because it can, as opposite to the format)
[17:57] <Compn> aspect ratio can change mid stream in some formats too. fun!
[17:57] <Compn> so future proof your thing to handle that imo
[17:57] <Compn> thing/api i mean
[17:57] <koda> ubitux: how can it rotate itself?
[17:58] <ubitux> well the codec has visibility on the raw decoded data
[17:58] <ubitux> and the rotation information (if stored at codec level)
[17:58] <ubitux> so it can do the rotation itself
[17:58] <koda> ah i see, but wed be cross layers too much
[17:58] <koda> imho
[17:59] <ubitux> i don't think so
[17:59] <Compn> you mean like codec flag is_flip or so ?
[17:59] <ubitux> exactly
[17:59] <Compn> decode the raw data opposite order to flip it
[17:59] <Compn> ah
[17:59] <koda> Compn: no i mean, lavc should provide decoded data, then its another layer who should do operations on it, like a player or a filter
[18:00] <ubitux> koda: some codecs support a "flip" flag
[18:00] <ubitux> they decode it "flipped"
[18:00] <ubitux> the same could happen for the rotation information if they have it
[18:02] <koda> ah like vp3
[18:02] <ubitux> or mjpeg and others
[18:03] <koda> they seem rather codec specific hacks to me
[18:03] <cone-476> ffmpeg.git 03Thilo Borgmann 07master:d2417061ef50: Add AVFoundation input device.
[18:04] <michaelni> also v-flip is free, rotate would likely not be free for most codecs
[18:05] <michaelni> that is unless one is crazy and transposes the idct scantable and scans mbs in columns and a few other things ...
[18:06] <koda> yeah, no, lets not do that :P
[18:07] <michaelni> transpose + flip == 90° rotate btw if that helps anywhere
[18:08] <ubitux> well you could raise the information to codec as well
[18:08] <ubitux> codec context*
[18:08] <ubitux> not sure if there is a "disposition" equivalent in lavc
[18:09] <koda> there is not, and id pefer to avoid breaking api too much
[18:09] <ubitux> well for the format the disposition is fine
[18:09] <ubitux> you need to find something else for codec
[18:10] <ubitux> 90 is transpose=1, 180 is hflip,vflip and 270 is transpose=2
[18:10] <ubitux> now for the other values we have rotate filter ofc
[18:11] <koda> yeah
[18:12] <ubitux> anyway that's for ffplay/ffmpeg
[18:14] <ubitux> koda: something really simple would be to just add a AVRotationMatrix in AVStream and AVCodecContext
[18:14] <ubitux> set AVStream->disposition
[18:14] <ubitux> add a AVCodecContext->disposition or just AVCodecContext->ihasrotation
[18:15] <ubitux> and that should do it
[18:15] <koda> that would work but itd break api, while side data would not, and again difficult to expand to cover other kinds of transformation
[18:15] <ubitux> break api? how?
[18:16] <ubitux> it would just be a minor bump
[18:16] <ubitux> just like anything you would do with side data i'm pretty sure
[18:16] <ubitux> (do we have stream and codec side data?)
[18:18] <koda> oh avcodeccontext not avcodec, sorry i got confused
[18:19] <koda> however id still be difficult to export if it is not carried outside in avutil
[18:19] <ubitux> you will define AVRotationMatrix in lavu ofc
[18:20] <koda> so youd go for having the full matrix there too, right?
[18:23] <ubitux> sure
[18:23] <ubitux> it can be heap allocated too
[18:24] <ubitux> NULL no matrix btw, that could work, no need for bit flags
[18:24] <ubitux> (but still a good idea to set the disposition for AVStream)
[18:27] <koda> im looking at it but its used mostly for textual information
[18:27] <ubitux> it's just bool flags iirc
[18:28] <ubitux> you'd just do st->disposition |= AV_DISPOSITION_ROTATED and set your matrix
[18:29] <koda> well you have to check two places to get it though
[18:30] <ubitux> ?
[18:30] <ubitux> of course
[18:30] <ubitux> it's important to keep a distinction between both
[18:31] <koda> hm not sure, you could just check if the rotation side data is present and then do that
[18:31] <ubitux> maybe you could set one or another depending on one's value
[18:31] <ubitux> but that depends on the politic of which override which
[18:31] <ubitux> why the hell do you want to put side data in every AVPacket?
[18:31] <ubitux> that has nothing to do with it
[18:31] <koda> avstream
[18:32] <koda> and id put it only once
[18:32] <ubitux> i only know avpacket side data
[18:32] <ubitux> what's this stream side data you're talking about?
[18:33] <ubitux> ah there is a stream->side_data
[18:34] <ubitux> oh that was some recently added nastiness for that exact purpose
[18:34] <ubitux> :)
[18:34] <koda> i wouldnt call it nastiness, but yeah
[18:34] <ubitux> i really don't see the point of this
[18:35] <ubitux> i mean, what's the benefit of this over special field for each entry?
[18:35] <ubitux> here api users need will have to mess with a side data api
[18:35] <ubitux> instead of just accessing fields
[18:36] <ubitux> afaict side data in avpacket makes sense because avpacket can (could?) be on the stack so it needs to be small
[18:36] <ubitux> AVStream is heap allocated so it doesn't matter
[18:36] <ubitux> ...but maybe i'm wrong.
[18:36] <nevcairiel> they dont like cluttering the struct with format specific fields, is the only reason
[18:36] <ubitux> it's not format specific
[18:37] <ubitux> rotation could be in various format
[18:37] <koda> because you cant just add fields everywhere everytime something new is available
[18:37] <nevcairiel> its video specific, thats reason enough :D
[18:37] <ubitux> nevcairiel: ahah
[18:37] <ubitux> koda: how so?
[18:37] <ubitux> you just add them at the end
[18:37] <ubitux> it's not stack allocated as i said
[18:37] <koda> its messy
[18:37] <ubitux> you use an api to create those streams
[18:37] <ubitux> how so?
[18:38] <koda> not every stream has rotation information, so its useless to add fields that do not apply to the majority of cases
[18:38] <ubitux> this is probably true for 80% of the fields in AVStream
[18:38] <ubitux> and it's fine to have unused fields
[18:39] <koda> i beg to disagree, and if its true for other fields, its not the case to add even more
[18:39] <ubitux> it's simpler to use for user
[18:39] <ubitux> simpler to understand
[18:39] <ubitux> and less access overhead
[18:39] <koda> umh not really simpler and you get overhead when passing that structure around
[18:40] <ubitux> ??
[18:40] <ubitux> you pass a pointer&
[18:40] <koda> ah right nevermind
[18:41] <koda> but its not like calling a function in a slowpath will reduce performance
[18:41] <koda> and its tidier and simpler for users, imho
[18:41] <ubitux> it's not like there is any single benefit of doing like this
[18:41] <ubitux> no it's not
[18:41] <ubitux> stream->field is simpler than av_get_side_data(stream) ... and interpret the binary shit side data is
[18:41] <koda> ive interrogated many downstreams and they seem pretty ok with accessors rather than filling up contextes
[18:42] <koda> and its not binary again
[18:42] <nevcairiel> i was never asked
[18:42] <ubitux> that's not an accessor
[18:42] <ubitux> look at how you use side data
[18:42] <ubitux> that's a pain
[18:42] <ubitux> while i can understand the logic for replaygain tags (and even that... it should be in metadata) i could understand
[18:42] <ubitux> but rotation is somehow generic
[18:42] <ubitux> it doesn't belong in this side data insanity
[18:42] <ubitux> just add a matrix and be done with that
[18:43] <koda> nevcairiel: i asked for feedback before and the patch on the other side is waiting for feedback for months
[18:43] <ubitux> don't try to overthing a shitty api, it's not worth the effort
[18:43] <ubitux> anyway, gtg
[18:43] <koda> i dont agree sorry
[18:43] <koda> its not metadata
[18:43] <koda> nor its disposition
[18:43] <ubitux> i was talking about replaygain
[18:43] <ubitux> that's something else
[18:43] <ubitux> (when talking about metadata)
[18:43] <koda> well side data can be both because its generic
[18:44] <ubitux> it's a generic and really painfull way of accessing non generic fields
[18:44] <koda> exactly which right now only 1 format exposes
[18:44] <ubitux> i'd say you should just add a generic rotate field and be done with that
[18:45] <ubitux> well it's not hard to imagine rotation field in various other formats
[18:45] <koda> ok thanks for the feedback, i disagree because of the reasons exposed ago
[18:45] <koda> above*
[18:45] <ubitux> you have no good reason to disagree
[18:45] <koda> i have
[18:45] <koda> read above
[18:45] <ubitux> well you didn't give any good argument
[18:45] <ubitux> i read
[18:45] <ubitux> and it's wrong
[18:45] <ubitux> gtg
[18:45] Action: ubitux &
[18:45] <koda> ubitux: you not liking something doesnt make something wrong
[18:46] <koda> and i gave enough arguments, from expandibility to tidiness
[18:46] <nevcairiel> you liking it doesnt make it right
[18:46] <koda> nevcairiel: no, my writing it makes it right :p
[18:46] <nevcairiel> every of your arguments is pretty weak, and can be taken as-is and turned around against it instead :p
[18:47] <koda> nevcairiel: well what would you suggest?
[18:47] <nevcairiel> first and foremost, make it easy to use, noone likes complicated api for the sake of cleanlyness
[18:48] <nevcairiel> when designing API, that should always be your primary goal
[18:48] <koda> nevcairiel: ok checking for disposition and then rotationmatix doesnt seem that simple either
[18:48] <nevcairiel> if you have to go and ask downstreams if they would mind this slightly more complicated approach, you're already doing it wrong :)
[18:49] <koda> nevcairiel: im asking downstreams because users proposed 4 different ways (now 5) of doing it
[18:49] <nevcairiel> especially because those established downstreams are already familiar with the oddities of the API, and may accept more crazyness than a new developer might before going crazy :D
[18:50] <koda> nevcairiel: im just trying to find the one that dissatisfies the least
[18:53] <koda> anyway i think i gathered enough feedback
[18:53] <koda> cya
[19:25] <cone-476> ffmpeg.git 03Anton Khirnov 07master:b9dea23766f5: resample: fix avresample_get_delay() return value
[19:25] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:c04961fa0ff6: Merge commit 'b9dea23766f52b8e059e72d34980bb7b456efe8f'
[19:29] <ubitux> > gathered enough feedback
[19:29] <ubitux> completely ignored them actually
[19:35] <cone-476> ffmpeg.git 03Anton Khirnov 07master:f20892eb67a7: resample: split the nearest neighbour path into a separate function pointer
[19:35] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:671f9086571a: Merge commit 'f20892eb67a7b4b5a3c080388a35a3380516a805'
[19:43] <cone-476> ffmpeg.git 03Anton Khirnov 07master:3c84f6b5d20c: resample: reindent
[19:43] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:fb7d64df7e3b: Merge commit '3c84f6b5d20cd345fac706f8cfb70c55e541ffb5'
[20:24] <jnvsor> Can't ./configure on sid any more - says something about missing hogweed?
[20:24] <cone-476> ffmpeg.git 03Anton Khirnov 07master:eed752d61da3: resample: drop useless abs()
[20:24] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:6d6bd16fce08: Merge commit 'eed752d61da332fb13e9893a175a90fed7b1d7d3'
[20:39] <cone-476> ffmpeg.git 03Anton Khirnov 07master:be394968c810: resample: add initial padding explicitly
[20:39] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:65088b925a0d: Merge commit 'be394968c81019887ef996a78a526bdd85d1e216'
[20:41] <michaelni> what configure command line causes this ?
[21:07] <cone-476> ffmpeg.git 03Anton Khirnov 07master:254c95cdd1b6: resample: split linear into its own function
[21:07] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:662a8d882758: Merge commit '254c95cdd1b669a722204a4a65974fc2523c8f83'
[23:06] <cone-476> ffmpeg.git 03Derek Buitenhuis 07release/2.2:96e13c989750: libx265: Use x265_param_parse to set the SAR
[23:48] <cone-476> ffmpeg.git 03Michael Niedermayer 07master:88ceede1fd78: avcodec/jpegls: print coding parameters stored in LSE
[00:00] --- Sat Apr 12 2014
1
0
[00:21] <karl_v> Hi, I am looking for a hint on how to convert a 3840x2160 YUV 4:2:0 8-bit-per-pixel H.264 recording to a 1920x1080 YUV 4:4:4 10-bit-per-pixel H.264 file. I have some experience with using ffmpeg, so scaling/encoding is the easy part for me, but I have not yet found the right options to convert to the right colorspace before scaling such that the binning of the pixels will actually retain maximum possible quality (combining 4 8-bit pixels of the
[00:21] <karl_v> original to 1 10-bit pixel in the output). Can somebody hint to the right options for doing so?
[00:28] <klaxa> you need to compile libx264 with 10 bit color depth
[00:28] <klaxa> that has to be enabled at compile time of libx264
[00:37] <karl_v> @klaxa: Yes, I understand that, and I did compile libx264 with 10 bit color depth. But I am wondering about two things: a) do I need to tell "configure" of ffmpeg also whether to use 8 or 10 bit upon compiling ffmpeg? And b) how do I make sure that there is no unneccessary loss of information/quality when downscaling - as it occurs when the destination of the downscaling is also just yuv420?
[00:37] <klaxa> you don't have to tell ffmpeg to use 10-bit in libx264 if you compiled libx264 with 10 bit (as it can't encode to anything else anyway)
[00:38] <klaxa> as for the downscaling, you will obviously lose information and probably quality too, there is nothing you can do against that
[00:39] <klaxa> i don't know about colorspace conversion details
[00:40] <klaxa> i would just encode short clips and check the quality
[00:40] <klaxa> see the -t option for that
[00:43] <karl_v> Maybe I found what I was looking for: "-vf format=pix_fmts=yuv444p,scale=1920x1080" seems to do the trick. Will have to check for the resulting quality - I'll paint a test picture with thin lines of equal luminance but different colors.
[00:48] <karl_v> Hmmm... "-vf format=pix_fmts=yuv444p" does spare me losing color information, but I wonder whether ffmpeg actually computes with 10bits per color channel, internally, then. It kind of would make no sense to encode to a 10bit per channel output format if the working data within ffmpeg was only 8 bits per channel, though...
[01:38] <relevart> Hi, I'm trying to stream video from udp. But the function av_read_frame hangs every now and then, stalling the stream. please help!
[09:52] <termos> Is it better to use a filter graph for audio and video instead of using the audio_fifo directly?
[10:38] <Aiena> Can someone please tell me which PCM is best for cross compatibility ? E.g. ffmpeg supports so many PCM types and I don;t know which is the most common
[10:41] <sacarasc> Aiena: 16 bit little endian, I forget how it's phrased when picking it/.
[10:41] <Aiena> thank you sacarasc
[10:41] <Aiena> in that case the codec is pcm_s16le
[10:41] <Aiena> sacarasc which container is best for PCM in this case WAV ?
[10:41] <sacarasc> Yeah, that one.
[10:42] <Aiena> i mean file container e.g .mp4 etc
[10:42] <sacarasc> WAV works well, yeah. Should work on just about anything you wanna play it on from the last 20 years.
[10:42] <Aiena> Ok thaks
[10:42] <Aiena> thanks
[10:43] <Aiena> sacarasc id the file name has spaces ffmpeg accepts quotes for the last output parameter right ?
[10:43] <Aiena> *if
[10:43] <sacarasc> Yes.
[10:43] <Aiena> thanks
[10:44] <Aiena> ffmpeg is one of the finest pieces of software but with so much power comes more complexity.
[10:44] <Aiena> thanks for your help sacarasc
[12:59] <grrk-bzzt> Hello
[13:00] <grrk-bzzt> I have a mkv{h264 + flac} file which video stream is longuer than the audio stream.
[13:00] <grrk-bzzt> I would like to drop frames with -vsync and -map but ffmpeg always end up producing a mkv with only an audio stream in it
[13:01] <grrk-bzzt> I use the following command
[13:01] <grrk-bzzt> ffmpeg -i file.mkv -vsync -1 -c:a copy -map 0:a file_fixed.mkv
[13:02] <relaxed> grrk-bzzt: you just want them to end at the same time?
[13:02] <grrk-bzzt> I want the video stream to end at the same time of the audio, yeah
[13:03] <relaxed> to cut the video stream where the audio ends, ffmpeg -i input -map 0 -c copy -shortest output
[13:06] <grrk-bzzt> Hum
[13:08] <grrk-bzzt> relaxed, -map 0:1 or 0:a in this case
[13:09] <grrk-bzzt> -map 0 just reproduces the same file as the original
[13:09] <grrk-bzzt> Wait no
[13:09] <grrk-bzzt> It doesn't work either
[13:09] <grrk-bzzt> It still makes mkv with no video stream
[13:19] <pdgendt> hi, i don't know if this is related to ffmpeg or alsa but when I play an audio file to a USB DAC, I get a lot of crackle (especially in the lower tones), I don't have it with regular headphone jack, FYI I am running on an ARM omap4 board, anyone who has an idea or could point me in the right direction?
[13:32] <pdgendt> nevermind other decoders have same issue, probably alsa thing
[15:36] <Venti> can I use the blackframe filter to remove all black frames from a video? how about white or boring frames in general?
[15:48] <dannixon> I'm trying to add an OGG audio track to an MJPG video, I have tried 'ffmpeg -i UNWRAP_MJPEG.avi -i AUDIO.ogg -vcodec copy -acodec aac -strict experimental FINAL.avi' however the audio teack appears to be empty when I try to play it
[18:17] <devinheitmueller> Are there any write-ups/documentation which describes the theory of operation for ffmpegs parallel encoding functionality? Im trying to understand how much sharing of state there is, whether its spreading the load based on GOPs (or whether its splitting up encoding tasks within a particular frame) etc.
[18:19] <devinheitmueller> For example, if its taking groups of frames that are within the same GOP and handing them off to a single CPU, then I would know that the multi-threading doesnt really help with realtime encoding.
[18:28] <clever> devinheitmueller: from what i understand, slice encoding is the main way to parallel it
[18:28] <clever> and i think its just splitting the entire frame into 2 (or 4), and then encoding it as entirely isolated streams
[18:28] <clever> so you basically have 2 entirely seperate h264 video streams
[18:28] <clever> which you are playing side by side
[18:28] <clever> or one above the other
[18:28] <devinheitmueller> clever: Ah, ok. A bit simplisitic, but certainly appropriate for streams which are already saved on disk.
[18:29] <clever> and it must be sliced at encode time
[18:29] <clever> once sliced, it can also use threads at decode time
[18:29] <devinheitmueller> Yeah, in my case Im interested purely in encode, and given the source is realtime then splitting into slices doesnt really help that much.
[18:30] <clever> you would have 2 encode threads, each thread takes half of the frame, and encodes it by itself
[18:30] <devinheitmueller> Ive read about other implementations which do MoCo in serial and DCT in parallel, but wasnt sure which category ffmpeg fell into.
[18:30] <clever> and then it somehow muxes the 2 h264 streams together with a special header to say its sliced
[18:31] <clever> so each thread can do a frame in half the time, since its encoding only half of the frame
[18:31] <devinheitmueller> Ah, I got it now.
[18:31] <clever> and the exact same thing at decode time
[18:32] <clever> i'm sure ffmpeg has a command line option to just do it
[18:32] <clever> the main issue is that you loose some compressability
[18:32] <devinheitmueller> I would assume support for sliced encoding would be dependent on the codec though, eh?
[18:32] <clever> yeah
[18:32] <clever> when something moves from one slice to the other, its an entirely different h264 stream
[18:32] <clever> so it cant encode it as moving
[18:33] <devinheitmueller> Right.
[18:33] <clever> it has to encode it as a new image entering the slice
[18:33] <clever> that limitation removes the need to share data between the 2 encode threads
[18:33] <clever> making the encode process much simpler
[18:33] <clever> the threads dont care about eachother
[18:33] <devinheitmueller> So a quick Google search suggests that sliced encoding is restricted to H.264/MPEG-4 (as opposed to MPEG-2). Do you concur?
[18:34] <clever> dont know if any other codecs support it
[18:34] <clever> ive only heard of it being used with h264
[18:34] <devinheitmueller> Gotcha.
[18:34] <devinheitmueller> Well, learn something new every day. :-)
[18:34] <clever> for single core decoders (or encoders), it can just do both slices on the same core, in the same ammount of time as a normal single slice file
[18:34] <devinheitmueller> sure.
[18:41] <JEEB> devinheitmueller, there are two main ways of threading, picture (frame) based and slice (or part-of-picture otherwise) based
[18:42] <devinheitmueller> Gotcha
[18:42] <JEEB> the first just happens to work even with references between the pictures with current formats it seems, and the latter is OK for devices with less RAM or if you need low latency
[18:42] <JEEB> because of course picture-based threading generally tends to bring latency, while the speed-up generally is greater than slice-based
[18:43] <devinheitmueller> Ah, nice.
[18:43] <devinheitmueller> Good to know there are a couple of different schemes available.
[18:43] <JEEB> and of couse you can do slice/whatever part-based stuff only when the file you're reading has that
[18:44] <JEEB> for example blu-ray requires you to have four slices per picture, but most random files you'll find on the internet will only have one per picture, since that gives you the best compression efficiency
[18:45] <devinheitmueller> Understood.
[18:50] <JEEB> with HEVC there's another way of possibly doing parallelization that most vendors seem to be enabling by default in the files, WPP (wavefront parallel processing). Basically it means that you have to decode two CTU blocks from a CTU row, and then you can decode the next CTU row. I think it costs a few % of compression eff. but the vendors and the standards body were OK with that.
[18:51] <JEEB> lavc currently doesn't support WPP multithreading for HEVC, only picture-based which works with all streams
[18:51] <devinheitmueller> HEVC is largely out of scope for me at this point, but thats certainly good to know whats coming down the pipe.
[18:51] <JEEB> well, at this point it seems like lavc will be using the picture-based threading model for that
[18:52] <JEEB> since that works in basically all cases
[18:52] <JEEB> instead of some, albeit the vendors seem to be pushing WPP rather hard so it will probably be a commonly usable way of multithreading
[18:57] <iive> <clever> when something moves from one slice to the other, its an entirely different h264 stream
[18:57] <iive> <clever> so it cant encode it as moving
[18:57] <iive> that's actually not true.
[18:58] <iive> slices still operate on same frames, output and reference.
[19:00] <iive> e.g. mpeg2 by standard have every row of macroblocks as separate slice. Since mpeg2 uses fixed huffman tables, it doesn't really affect compression. the only difference is that it waste 4 bytes on a slice header.
[19:11] <clever> ah
[19:21] <adi10289> hi guys i need help
[19:22] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key can someone help me how to download from such streaming
[19:34] <adi10289> somebody pls help me out :(
[19:46] <blahsd> Hello! I was wondering if anybody knew what is the best way to have the smart blur more intense
[19:49] <blahsd> pretty please? :D
[19:50] <blahsd> hey aard_ are you there?
[19:52] <aard_> i guess :)
[19:52] <blahsd> there's not many people actually online I guess :P Can I bother you for a second? Maybe you can give me a quick tip :)
[19:53] <aard_> if i can help
[19:53] <blahsd> thanks a bunch mate
[19:53] <blahsd> I'm trying to get a stronger blur going on a video, but raising chroma values don't get anywhere near where I want to be
[19:54] <blahsd> do you have any idea what I could do?
[19:54] <aard_> not my expertise sorry :/
[19:54] <blahsd> that's alright no worries :) It's a very specific question haha
[20:02] <adi10289> somebody help me too :(
[20:03] <blahsd> adi10289: what are you looking for?
[20:03] <adi10289> i m trying to download from a m3u8 hls streaming which is having https AES key
[20:04] <adi10289> but haven't got any success till now :(
[20:04] <blahsd> stop right there man that's waaaay off my league hahah
[20:04] <blahsd> I'm sorry dude
[20:07] <adi10289> ok blahsd, guess i have to switch to debian and start testing :(
[20:07] <blahsd> sorry man :(
[20:09] <blahsd> gotta go good day all!
[20:31] <dannixon> Anyone have any ideas what could be causing this: http://pastebin.com/nDAqKy69
[20:43] <c_14> You're using libav, see #libav
[21:09] <Spec-Chum> is there a way to get ffmpeg to tell you the video codec used?
[21:10] <Spec-Chum> I want something like: if (codec != x264) convertToX264(); else copyVideo();
[21:10] <Spec-Chum> in bash BTW, just horrible pseudo code above :)
[21:10] <c_14> ffprobe
[21:12] <Spec-Chum> bit drastic, isn't there an ffmpeg option that just returns "h264" string or somesuch?
[21:13] <Spec-Chum> Just doing a simple script to convert any file to stream on Chromecast: ffmpeg -i "$1" -c:a pcm_s16le -f wav - | neroAacEnc -if - -ignorelength -of "$2".mpa; ffmpeg -i "$1" -i "$2".mpa -c:v copy -c:a copy -map 0:v:0 -map 1:a:0 "$2".mp4
[21:14] <Spec-Chum> that does work, but it copies the video stream, I want to test and only convert if it's not h264
[21:14] <Spec-Chum> make sense?
[21:17] <c_14> ffprobe -show_entries stream=codec_name
[21:19] <Spec-Chum> ah, righto cheers!
[21:33] <Spec-Chum> ffprobe -v quiet "$1" -select_streams v -show_entries stream=codec_name
[21:33] <Spec-Chum> perfect, thanks c_14
[22:47] <Samian> so lets say you want to edit 10 seconds of a 1 hour video. Obviously, you wouldn't want to re-encode the entire video and thus lose quality. What's the best way to only re-encode the 10 second part? What's a good program that has that capability?
[22:50] <c_14> I don't know about the best way, but what I would do is split the video into 3 parts, mess around with the part in the middle and then just concat them together again.
[22:50] <c_14> I'd look at -ss -t -to and the concat filter.
[00:00] --- Sat Apr 12 2014
1
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[00:00] <lukaszmluki> Hi, I have question: is it possible that AVPacket.size may change for the certain stream or it is always the same?
[00:00] <wm4> of course it will change
[00:01] <lukaszmluki> inside one file, for the same stream, it may have different sizes, right?
[00:01] <wm4> yes
[00:01] <lukaszmluki> ok, thx
[00:03] <ubitux> lukaszmluki: ffprobe -show_entries packet=size -select_streams v:0 -of flat foobar.mp4
[00:04] <wm4> is ffprobe turing complete yet
[00:04] <ubitux> since it's the encoded video frames, packet sizes change every time, obviously
[00:04] <ubitux> wm4: eval should be
[00:04] <lukaszmluki> I think about audio packets, if thats matter.
[00:05] <ubitux> change 'v' into 'a'
[00:05] <lukaszmluki> oki
[00:05] <ubitux> and change 'video' into 'audio' in my next sentence
[00:05] <lukaszmluki> information that it is not constant is enough
[00:14] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:f703dae7aa76: avcodec/cngdec: use av_mallocz_array()
[00:14] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:968516cebf89: avformat/nut: add ff_nut_audio_extra_tags to demuxer too
[00:14] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:849d40dce62d: avcodec/mpegvideo_enc: change some asserts to av_asserts
[01:34] <cone-291> ffmpeg.git 03Carl Eugen Hoyos 07master:bf87d582b1cb: Do not allow writing invalid wav channel layouts by default.
[01:34] <cone-291> ffmpeg.git 03Carl Eugen Hoyos 07master:2a3ff9fbbacf: Remove an unneeded include of avassert.h from amr.c.
[01:34] <cone-291> ffmpeg.git 03Carl Eugen Hoyos 07master:cf79f202757b: Fix a typo in amr.c.
[01:34] <cone-291> ffmpeg.git 03Michael Niedermayer 07master:421c116fc498: Merge remote-tracking branch 'cehoyos/master'
[01:38] <wm4> michaelni_: so what's the deal, are all enums now supposed to use fourcc values?
[01:40] <michaelni_> our codec ids do
[01:41] <michaelni_> i dont think anyone suggested to use fourccs for pixfmts
[01:44] <wm4> they seem to pop up in a number of places - even when not needed
[01:49] <michaelni> which one are you thinking of ?
[01:50] <michaelni> i mean which did pop up and isnt needed ?
[01:51] <wm4> e.g. the ones in libavdevice/avdevice.h
[01:51] <lukaszmluki> probably my patch to avdevice.h
[01:51] <lukaszmluki> but that was Nicolas who wanted fourcc
[01:52] <wm4> but even the ones for codecids are not needed (though that damage is already done)
[01:52] <michaelni> damage ?
[01:53] <michaelni> fourccs should reduce the risk of conflicts
[01:53] <wm4> they make debugging harder etc.
[01:53] <wm4> they seem like an innecessary complication
[01:53] <wm4> *un
[02:00] <michaelni> iam not sure i agree, i mean a codec id of 123 tells me as much as 98765432 which is how they would look like when showen as literal numbers in a debuger
[02:01] <michaelni> ... and a good debuger should show the named identifer of an enum type
[02:01] <michaelni> either way
[02:09] <kierank> wm4: what about codecs in containers that lavc doesnt support
[02:09] <kierank> they need an identifier of some sort
[02:09] <wm4> kierank: codecids are never mixed with real fourccs
[02:09] <kierank> true
[03:06] <cone-291> ffmpeg.git 03Lukasz Marek 07master:fbf05759e38f: sws: fix warning about RGB_PACK24_B_OPERANDS redefinition
[04:03] <cone-291> ffmpeg.git 03Lukasz Marek 07master:e1c48e3593fb: lavf/utils: fix mem leak
[04:46] <cone-291> ffmpeg.git 03Reinhard Tartler 07release/2.2:a014b9614e61: Prepare for 10.1 Release
[04:46] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:bd553941ae51: Merge commit 'a014b9614e6165fb8df9e756f46a95d516794678' into release/2.2
[04:53] <cone-291> ffmpeg.git 03Martin Storsjö 07release/2.2:e0aa76d38a02: golomb: Fix the implementation of get_se_golomb_long
[04:53] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:b2583c2b6289: Merge commit 'e0aa76d38a02090245284fc157afb9074e9ff073' into release/2.2
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:a221c9bd7677: avcodec/libx264: move where x264opts is applied down so it isnt overridden by avctx & defaults
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:5cb2a1c3f0b9: avfilter/vf_pullup: fix gray8
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:02bae9f013e7: avcodec/h264: clear cur_pic structure instead of duplicating it in ff_h264_update_thread_context()
[05:08] <cone-291> ffmpeg.git 03Lukasz Marek 07release/2.2:fc8eb4c1f984: lavu/opt: validate range before dereference
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:f41622ecb4b3: avcodec/wma: use av_freep(), do not leave stale pointers in memory
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:544accc895f3: avformat/omadec: fix probetest failure
[05:08] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:34592d04fbea: examples/avio_reading: fix null dereference on error
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:36cab9c408bd: avcodec/pthread_frame: fix missing unlock on error
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:b479b42b26b9: avcodec/msrle: check return code for success before use
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:352b0969e23c: swresample/resample: Limit filter length
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:d20ac551a861: swresample/dither: use av_malloc_array()
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:dde95268cc96: swresample/resample: use av_malloc_array() where appropriate
[05:09] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:ce94955b3c0b: swscale/x86/swscale_template: loose hardcoded dstw_offset
[05:09] <cone-291> ffmpeg.git 03Anthoine Bourgeois 07release/2.2:dfddefa13afa: avcodec/dirac_arith: Fix build with PIC and stack-check options
[05:16] <cone-291> ffmpeg.git 03Michael Niedermayer 07release/2.2:e72c0a04664a: update for 2.2.1
[05:53] <cone-291> ffmpeg.git 03Lukasz Marek 07fatal: ambiguous argument 'refs/tags/n2.2.1': unknown revision or path not in the working tree.
[05:53] <cone-291> Use '--' to separate paths from revisions
[05:53] <cone-291> refs/tags/n2.2.1:HEAD: lavf/utils: fix mem leak
[07:43] <ubitux> heh koda is at it again
[07:47] <ubitux> > In a normal world one could envision ffmpeg and libav as two branches, one for the development/hacks/quickbugfix and the other for stable development.
[07:47] <ubitux> lol
[09:27] <t355u5> problem compiling 2.2.1 with x265:
[09:27] <t355u5> libavcodec/libx265.c: In function libx265_encode_init:
[09:27] <t355u5> libavcodec/libx265.c:124: error: x265_param has no member named bEnableVuiParametersPresentFlag
[09:27] <t355u5> libavcodec/libx265.c:125: error: x265_param has no member named bEnableAspectRatioIdc
[09:27] <t355u5> libavcodec/libx265.c:126: error: x265_param has no member named aspectRatioIdc
[09:27] <t355u5> libavcodec/libx265.c:127: error: x265_param has no member named sarWidth
[09:27] <t355u5> libavcodec/libx265.c:128: error: x265_param has no member named sarHeight
[09:29] <nevcairiel> 2.2.1 only works with the x265 version that was released when 2.2.1 was released, not with any newer versions
[09:29] <nevcairiel> thats how it is, and thats how it shall remain
[09:32] <t355u5> 2.2.1 was released today. and I'm using an x265 version from today
[09:32] <nevcairiel> well technically it should've said "when 2.2 was released"
[09:33] <nevcairiel> x265 changes its API once a week now, the release branch is not going to get updated for that. If you want to use x265, use ffmpeg git master
[09:34] <t355u5> which would be which version exactly?
[09:34] <t355u5> I'm creating static builds for git master and for releases
[09:40] <relaxed> t355u5: it fails to build with git master too
[09:41] <nevcairiel> like i said, they keep changing it once a week these days, its not always perfectly up to date :p
[09:42] <nevcairiel> i saw a patch for the recent breakage on the mailing list however
[09:42] <t355u5> e1c48e3 built just fine
[09:43] <t355u5> I guess I will have to match the dates from 2.2 and x265 and try again
[09:45] <t355u5> thx for the info
[10:20] <t355u5> just tried with x265 from March 21st, since 2.2 was released on March 23rd - still the same error. does anyone have an idea which x265 is supposed to compile with 2.2?
[12:52] <ubitux> BBB: no need to ping koda, you are an uninvited troll according to him in the thread
[12:53] <ubitux> "It's hard to reply without reminding sad past topics, arousing trolls (a few have already took part uninvited)", "Much less the troll attempts in the other emails that got duly ignored."
[12:56] <BBB> ah, duly ignored
[12:56] <BBB> everyone that disagrees with me is an E V I L T R O L L!
[12:56] <BBB> there
[12:56] <BBB> I said it
[12:56] <BBB> thus it is true
[13:52] <Compn> ubitux : want to submit an api patch to libav-devel to call their bluff? :)
[13:52] <Compn> :P
[14:08] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:8945dcbb52b6: libx265: Use x265_param_parse to set the SAR
[14:25] <pross-au> sigh
[14:29] <ubitux> pross-au: he has commit access
[14:29] <pross-au> the logical deduction still holds
[14:30] <pross-au> join libav, get commit rights? no?
[14:33] <av500> fill out the form in triplicate
[14:33] <Daemon404> my biggest question is
[14:33] <Daemon404> why do you all keep replyoimg
[14:33] <Daemon404> replying*
[14:34] <ubitux> well we don't want to let him give the illusion that libav is trying to cooperate while we ignore it
[14:34] <ubitux> because that's actually what they do
[14:37] <Daemon404> wut
[14:38] <Daemon404> michaelni, found another jpeg that fails with avcodec btw
[14:38] <Daemon404> looking into it
[14:38] <Daemon404> not cmyk this time
[14:38] <pross-au> (i dunno abou you, but i found the repeated advice to stop merging helpful)
[15:03] <Compn> Daemon404 : probably a million jpegs that fail with avcodec
[15:03] <Compn> its a gigantic file format with so many crazy specs
[15:03] <Compn> trillions of samples and broken software
[15:03] <Compn> :)
[15:09] <cone-593> ffmpeg.git 03Nicolas George 07master:ad8159e0fe2b: libavfilter: Add asettb filter for setting timebase for audio
[15:09] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:476471495265: Merge remote-tracking branch 'qatar/master'
[17:48] <cone-593> ffmpeg.git 03Matt Oliver 07master:d1e6e5c88733: avcodec/x86: Exclude broken get_cabac under icl.
[17:49] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:fcc43e02d302: libx265: Use 16-bit SAR
[17:49] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:cdd21f1f03c5: libx265: Only set the SAR if it is valid
[20:05] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:c486cfab792e: avcodec/imgconvert: fix nb_components and depth for PAL8 in get_pix_fmt_score()
[21:32] <cone-593> ffmpeg.git 03Anton Khirnov 07master:9a07fac678a8: mov: read hydrogenaudio replaygain information
[21:32] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:6e573b77597c: Merge commit '9a07fac678a8540d076e635061bbaa4ed09a9431'
[21:32] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:4d4bc89409fb: avformat/mov: merge mov_read_custom_metadata() and mov_read_custom()
[21:41] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:4a2226451e14: libx265: Use x265_param_parse to set the SAR
[21:41] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:cd0ac6f6e5c3: libx265: Use 16-bit SAR
[21:41] <cone-593> ffmpeg.git 03Derek Buitenhuis 07master:25fbfeed98d2: libx265: Only set the SAR if it is valid
[21:41] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:a4662cd6b11e: Merge remote-tracking branch 'qatar/master'
[21:52] <jedir0x> I'm using the ffmpeg libraries on Android to render video. On some devices when creating a 480x480 h.264 mp4 with aac audio the bottom 1/4 or so of the video is a large green bar.... does that mean anything to anyone? :)
[22:04] <iive> jedir0x: green means the area is full of zeroes.
[22:15] <jedir0x> i see
[22:15] <cone-593> ffmpeg.git 03James Almer 07master:09c16542085a: lavf/rtspcodes: add missing header include
[22:15] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:6290ba005296: avfilter/settb: remove duplicate #include
[22:18] <Compn> jedir0x : try creating a window larger than your video res ? :)
[22:18] <Compn> but no idea
[22:20] <jedir0x> i was passing the wrong stuff to sws_scale :0
[22:20] <jedir0x> :) *
[22:20] <jedir0x> thanks for the tip
[22:21] <Compn> lol how did you figure it out from my guess ?
[22:27] <cone-593> ffmpeg.git 03Carl Eugen Hoyos 07master:ecb80fe78c58: Cosmetics: Remove an outdated comment from lavc/rawdec.c.
[22:37] <cone-593> ffmpeg.git 03Michael Niedermayer 07master:517cc81e939f: ffmpeg: stop protocols on interrupt when no transcoding has happened yet
[23:09] <cone-593> ffmpeg.git 03Anssi Hannula 07master:39b192b8e1c1: avformat/hls: do not call ffurl_seek/ffurl_size on crypto protocol
[00:00] --- Fri Apr 11 2014
1
0
[02:20] <vl4kn0> Hi, is there a way to process online video over the https protocol? ffmpeg says it's unsupported protocol
[02:32] <vl4kn0> here's the paste then: http://pastie.org/9069185
[02:34] <salohcin> I was wondering if anyone had any pointers to make a video stream faster, right now I am running the command: ffmpeg -y -f rawvideo -vcodec rawvideo -s 1280x720 -pix_fmt rgb24 -i - -an -c:v libx264 -pix_fmt yuv420p http://localhost:8090/feed1.ffm which works well for streaming, but is a really slow framerate
[02:36] <salohcin> i have tried using the options -preset ultrafast -tune zerolatency, but I am getting an error "Option preset not found." or "Option tune not found.", any ideas?
[02:36] <salohcin> I am opening ffmpeg from within python and piping raw rgb frames to ffmpeg to encode and stream
[02:37] <klaxa> vl4kn0: that looks like a youtube video, i think those links are not valid for very long
[02:37] <klaxa> also you are missing the output completely
[02:49] <vl4kn0> klaxa: the youtube link's been valid while testing and I don't need to have output to see 'unsupported protocol'
[03:38] <Nomad145> Greetings. I get the following error while running ffmpeg. I'm trying to grab frames from an rtsp stream from an IP Based camera and the stream dies after about 30 frames with a sort of error. All information is included in the paste. Any help would be greatly appreciated. http://pastebin.com/xnb4Nfgd
[03:39] <klaxa> you are using avconv, see #libav
[03:39] <klaxa> or get a static build or compile from source
[03:44] <Nomad145> Mmk. I'll try that.
[04:04] <salohcin> I was wondering if anyone had any pointers to make a video stream faster, right now I am running the command: ffmpeg -y -f rawvideo -vcodec rawvideo -s 1280x720 -pix_fmt rgb24 -i - -an -c:v libx264 -pix_fmt yuv420p http://localhost:8090/feed1.ffm which works well for streaming, but is a really slow framerate
[04:04] <salohcin> i have tried using the options -preset ultrafast -tune zerolatency, but I am getting an error "Option preset not found." or "Option tune not found.", any ideas?
[04:04] <salohcin> I am opening ffmpeg from within python and piping raw rgb frames to ffmpeg to encode and stream
[04:06] <relaxed> salohcin: I'm guessing your ffmpeg is too old. What's the output of "ffmpeg | sed q"
[04:07] <relaxed> er, ffmpeg 2>&1| sed q
[04:09] <salohcin> that's what my guess was too, I have 1.1.5, but I got the latest from ffmpeg.org, version N-41838-g754f846-
[04:10] <salohcin> that one gives the same error, I'd like to note that the error isn't thrown at startup, but after I send it a single rgb frame
[04:10] <relaxed> do you have multiple versions installed, maybe?
[04:10] <salohcin> I've been on google and I think it might be an error when x264 is invoked
[04:11] <salohcin> I have the latest executable in my home directory and I am running it with a full path, can I not do that?
[04:11] <relaxed> you can
[04:12] <salohcin> are there libraries that it depends on that need to be updated too?
[04:12] <relaxed> run it from the command line and pastebin your complete command and console output
[04:17] <salohcin> http://pastebin.com/6zzDGEEt
[04:26] <salohcin> do i need to specify a video container format like flv?
[04:26] <salohcin> I havn't done streaming before, so I'm still learning how it all works
[04:30] <relaxed> salohcin: ah, you're using my build :)
[04:33] <salohcin> relaxed: bleeding edge?
[04:33] <relaxed> hmm, -preset ultra work with my test
[04:33] <relaxed> ultrafast*
[04:34] <salohcin> I can get it to work too with other configurations, it seems to be this particular way of streaming or something
[04:36] <relaxed> salohcin: what happens if you don't pipe it?
[04:36] <salohcin> for instance if I specify an output container format like "-f flv" it works
[04:37] <salohcin> I just haven't been able to stream an flv properly, I can only get the .swf example from the ffserver website working
[04:37] <relaxed> have you seen https://trac.ffmpeg.org/wiki/StreamingGuide ?
[04:40] <salohcin> if I don't pipe anything to it, it just hangs with no error, waiting for piped input. Yeah I have seen the streaming guide, but I wasn't able to find an example of capturing an flv on the client side, I guess I should do more research...
[04:41] <salohcin> out of curiousity, what happens if you do not specify a container format? Does ffmpeg pick a suitable one for you?
[05:02] <salohcin> relaxed: thank you for your help
[10:17] <tim-kos> hey all, is there a uniform way to preserve embedded audio artwork when converting audio files? I need to support conversions to mp3, flac and alac.
[10:24] <termos> How can I set the movflag in code using the C api?
[13:04] <anshul_> Hi my ffprobe output does not show PAR, is there some extra flag for knowing PAR of video stream
[13:05] <relaxed> anshul_: does it show SAR?
[13:05] <relaxed> they're the same thing
[13:06] <anshul_> no it does not show even SAR
[13:07] <relaxed> anshul_: I see "sample_aspect_ratio=1:1" with, ffpobe -show_streams input
[13:07] <relaxed> ffprobe*
[13:09] <anshul_> I also see that way, thanks
[13:17] <anshul_> my image show SAR as 0:1, is it possible for pixel having such a ratio 0 height to 1 width.
[13:17] <anshul_> even DAR = 0:1
[13:24] <luko28> Hi all, small question: I'm setting rtsp, and want to set multiple transport protocols by av_dict_set(), exactly 'tcp' and 'udp', in that order. How to achieve that?
[13:45] <anshul_> is there any way to dump dvb subtitle in text format.
[13:46] <plaz> Hi, quick question. Is there a matrix library within ffmpeg that provides basic utility functions such as transposing? I can't seem to find anything in the docs.
[13:46] <anshul_> is there any way to dump dvb subtitle in text format. using ffmpeg
[13:48] <anshul_> plaz, there is file libavfilter/vf_transpose.c in ffmpeg sr code so you can find some filter for doing same.
[13:55] <barnoux> Hello
[13:56] <barnoux> i would like to know if i can launch various ffmpeg conversion at the same time?
[13:56] <barnoux> on a server
[14:05] <plaz> anshul_: I don't need a filter, I have some internal data (that's really a matrix) represented by a flat array and I would like to transpose it. I wondered if ffmpeg already had the facility for doing commmon matrix operations.
[14:39] <anshul_> barnoux,yes you can
[14:42] <anshul_> plaz, nothing that I know out of that :)
[14:43] <barnoux> anshul, how many conversion process i can launch at the same time?
[14:43] <barnoux> processor is Intel Atom
[14:43] <barnoux> C2750 8 Threads
[14:48] <anshul_> barnoux, You might have to test it, There are lots of thing that matters, so easiest way would be to try manually
[14:49] <barnoux> ok
[15:36] <mintux1> i have two separated file audio .m4a and video .mp4 that i want put audio on video without any change to original quality . i test this command but doesn't work: $ ffmpeg -i 'video.mp4' -i 'sound.m4a' -vcodec copy -acodec copy TestVideoMux.mp4
[15:37] <mintux1> it creates but video file is 59MB while output is 3.6mb
[15:38] <jkli> hello, are green screens signs of low buffer or framerate drops?
[15:39] <mintux1> did you say to me?
[15:40] <mintux1> it's my output
[15:40] <mintux1> http://codepad.org/HZjLOlil
[15:42] <jkli> did you test if it reads m4a correctly?
[15:43] <mintux1> jkli: the output is only audio file
[15:43] <jkli> you might need to reencode the audio
[15:43] <jkli> also if the original already has audio
[15:43] <jkli> you might need to use the map directive
[15:44] <jkli> which is a bit hasslesome
[15:44] <relaxed> mintux1: try, ffmpeg -i 'Amir Farjam - Zendegi Ba To OFFICIAL VIDEO HD-JMOyX5cpy4o.mp4' -i 'Amir Farjam - Zendegi Ba To OFFICIAL VIDEO HD-JMOyX5cpy4o.m4a' -map 0 -map 1 -vcodec copy -acodec copy output.mp4
[15:45] <mintux1> relaxed: it's only audio file 3.6MB
[15:46] <mintux1> it's HD video from youtube 1080
[15:46] <mintux1> http://codepad.org/NCGIsmU7
[15:46] <mintux1> then video doesn't have audio so i download audio separately
[15:47] <jkli> if you just mux videos it can be unsynchron
[15:47] <jkli> depends on video file
[15:48] <mintux1> jkli: i download both audio and video from same source and same link on youtube.com but youtube mute sound on HD video when we download
[15:48] <jkli> yeah
[15:48] <jkli> they suck
[15:51] <mintux1> so what is your solution ?
[15:54] <phelix> I am fairly new trying to use ffmpeg via the command line. ffmpeg -i ./movie.mp4 -s 854x480 -aspect 16:9 -strict experimental ../Movie\ Converted/movie.mp4 This is giving me the correct resolution I am looking for but the video quality is just terrible. Anyone know how I can improve the video quality to be just as good as the original?
[15:55] <mintux1> solved
[15:58] <another> phelix: check -b
[15:59] <phelix> I am not to sure about bitrates. What should I set this to?
[15:59] <phelix> can't it use the same bitrate of the current video?
[16:00] <another> i think there is an option -sameq
[16:01] <phelix> k, let me try that thanks
[16:04] <phelix> There we go, I think that is going to do it. Thank you!
[16:04] <phelix> Will changing the resolution cut off some of the video do you know?
[16:07] <another> shouldnat cut off anything afaik
[16:09] <ubitux> another: http://ffmpeg.org/faq.html#Why-was-the-ffmpeg-_002dsameq-option-removed_003…
[16:09] <ubitux> phelix: if you transcode, it will never be lossless
[16:10] <ubitux> (unless you explicit it, and you'll get a huge output)
[16:10] <iive> phelix: add -c:v libx264 -crf 23 , you could remove the -strict experimental
[16:10] <ubitux> add -c copy so the other stream are copied losslessly
[16:10] <iive> most likely ffmpeg uses mpeg4asp instead of mpeg4 avc
[16:11] <ubitux> iive: x264 should be the default with mp4
[16:11] <iive> ubitux: it won't change the resolution with copy.
[16:11] <ubitux> iive: for the other streams, such as audio
[16:11] <ubitux> experimental was probably added because of that
[16:12] <ubitux> ffmpeg -i in.mp4 -vf scale=854x480,setsar=16/9 -c copy -c:v libx264 -crf 23 out.mp4 should do it
[16:13] <phelix> ok let me try that
[16:13] <phelix> thank you
[16:13] <ubitux> it won't be lossless since you rescale though
[16:13] <ubitux> lower crf for better quality
[16:14] <phelix> ok, I want the same quality just in a different resolution that fits the device I am encoding this for
[16:14] <iive> indeed, it even uses crf 23 by default.
[16:14] <iive> phelix: do you use recent ffmpeg version? no forks or stuff?
[16:14] <ubitux> "same quality" makes no sense if you rescale
[16:15] <phelix> I think so, just installed it from the ubunutu application mamager
[16:15] <ubitux> ah, probably not ffmpeg then
[16:16] <phelix> yah I am getting Unrecognized otion 'c' failed to set value 'copy' for option 'c'
[16:16] <ubitux> here you go ^
[16:16] <phelix> k
[16:17] <iive> phelix: try -vcodec instead of -c:v or use `avconv`
[16:17] <phelix> same commandline option except avconv instead?
[16:18] <iive> yes, -c:v is syntax that was introduces by avconv
[16:18] <iive> debian/ubuntu continued to ship ancient version of ffmpeg.
[16:19] <phelix> Unknown encoder 'libx264'
[16:19] <phelix> where do I get this at?
[16:19] <iive> try ffmpeg -vcodec libx264... etc
[16:20] <phelix> so i can remove the -c copy ?
[16:20] <ubitux> iive: you should not recommend to use the broken ffmpeg distributed by libav :p
[16:21] <iive> yeh...
[16:21] <phelix> yah still get unkown encoder for libx264 with ffmpeg and avconv
[16:21] <ubitux> that's because it's not build with libx264 (licenses ...)
[16:21] <iive> debian probably doesn't even link it
[16:21] <iive> x264 is gpl
[16:22] <ubitux> patents sorry
[16:22] <iive> but there are also the mpeg-la patents on h264
[16:22] <phelix> k let me first install the other version
[16:22] <iive> phelix: get static ffmpeg build.
[16:23] <phelix> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg ??
[16:23] <phelix> will this version work?
[16:24] <iive> phelix: you'd have to compile x264 with shared/static libraries first, before compiling your own ffmpeg.
[16:24] <phelix> ahh ok
[16:27] <phelix> ok seems to be running with those options now
[16:27] <phelix> thanks
[16:32] <phelix> ./ffmpeg -i in.mp4 -vf scale=854x480,setsar=16/9 -vcodec libx264 out.mp4 I checked this and the resolution is def NOT 854x480 any ideas?
[16:33] <phelix> it looks exactly like the source file
[16:36] <phelix> did I miss something?
[16:37] <iive> is the size different?
[16:38] <iive> what is the resolution of out.mp4 ?
[16:38] <phelix> Well I didn't let the whole thing finish. Its a 2 gig file I am sure its going to take a few hours
[16:38] <iive> still, the result should be playable.
[16:38] <phelix> Yes, thats how I was able to tell
[16:39] <iive> you can use -ss <start_time> -t <duration>
[16:39] <phelix> I don't know how to tell exactly what the out resolution is. I can just tell by looking at it that its not 854x480 its really thing like the original
[16:39] <iive> ffprobe could tell you resolution and other stuff.
[16:39] <phelix> it appears that its the same resolution as the in file
[16:40] <phelix> Stream #0.0(und): Video: h264 (High), yuv420p, 854x480 [PAR 16:9 DAR 427:135], 755 kb/s, 23.9
[16:40] <phelix> hmm
[16:40] <phelix> it says it is 854x480 with ffprobe
[16:40] <phelix> but it sure doesn't seem so. still seems really thin and wide.. very weird
[16:40] <iive> oh...
[16:42] <phelix> yeah there is something really weird going on. Its actually thinner than the original file.. Only able to tell by looking at the video but something isn't right
[16:51] <phelix> ffmpeg -i ./in.mp4 -s 854x480 -aspect 16:9 -strict experimental -sameq ../Movie\ Converted/out.mp4 using the other version of ffmpeg I had and this command line appears to give the desired output. Just seems that it may be cutting some of the picture off. But its much larger and fills the screen properly.
[16:54] <iive> phelix: is the original with correct aspect ratio?
[16:54] <iive> remove the setsar and just use the -aspect 16/9 as before.
[16:54] <phelix> k let me try that
[16:55] <phelix> the original is 1920x800
[16:57] <iive> if the original looks ok, then you shouldn't override the aspect info, scale would recalculate the aspect to compensate for change in resolution.
[16:58] <phelix> so to take the aspect from the source video to this new video resolution I am trying to get is actually blowing it up to fit and cropping some of the video off isn't it?
[16:59] <phelix> the way you had me doing it first was keeping everything the same but it seemed to look thinner because it changed the resolution to fit properly? which would probably make it appear to be thinner
[16:59] <phelix> if that made any sense lol
[17:22] <iive> there are a number of different ways to express aspect ratio...
[17:23] <iive> the video was thinner because you resized the image to be 16/9 then you said that the pixel is 16/9 that basically tells the player to rescale the image to 16/9 once again.
[17:25] <iive> usually the displays have square pixels, so the resolution and aspect are the same. e.g. 1920x1080 is (almost) 16/9
[18:56] <edoceo_> I have a video, it's 208.700 seconds long, and I want to stretch it to 214.607 seconds long, how could one do that?
[19:35] <synth_> could someone assist me in modifying my command to have two separate text overlays? i'd like to place part of my current overlay on the top and some on the bottom: http://pastebin.com/NZPwu9Xt
[19:42] <llogan> edoceo: see the setpts video filter
[19:43] <llogan> not sure if you can get the exact length that you want though
[19:43] <edoceo> llogan: I've been looking at that, but it's not clear to me how to adjust when I only know the times
[19:43] <edoceo> Like, my video is 98.4695% the length I want, so do I use a 1/N or N/1 kind of specifier?
[19:44] <llogan> setpts=1.02830378534*PTS or similar
[19:45] <llogan> i'm not sure how many decimals it will parse
[19:45] <llogan> synth_: just add another drawtext
[19:46] <llogan> a different project. you guys are busy.
[19:47] <edoceo> numbers >1 speed it up?
[19:48] <llogan> no. not when n*PTS
[19:55] <llogan> edoceo: 2*PTS and PTS/0.5 should do the same thing (which is to slow it down)
[19:55] <synth_> llogan: do i need to do anything special with the [0:v] tags?
[19:55] <edoceo> thanks!
[19:57] <llogan> synth_: no. see http://ffmpeg.org/ffmpeg-filters.html#Filtergraph-syntax-1
[19:58] <llogan> so you can do [0:v]drawtext,drawtext[bg];[bg][1:v]overlay........etc
[19:58] <galex-713> Hello
[19:58] <galex-713> Is there some option to remove a part of a video?
[19:59] <llogan> synth_: or if you prefer [0:v]drawtext[d1];[d1]drawtext[d2];[d2][1:v]overlay
[19:59] <galex-713> Or should I cut a first part, then cut a second part, then find how to join videos, then join parts, then test, etc.?
[19:59] <synth_> ok i actually got it
[19:59] <synth_> thanks llogan!
[20:00] <llogan> galex-713: http://video.stackexchange.com/a/10401/1760
[20:25] <RenatoCRON> wow, I just got one that ip-camera:554/cam/realmonitor?channel=4&subtype=01 but ?subtype=01&channel=4 get first channel !
[20:26] <RenatoCRON> these devices dvr maybe using regexp to parse params!
[20:45] <galex-713> llogan: ah ok, so I need to split and *then* join& ok thanks
[21:06] <phelix> Is there anyway with ffmpeg to have it convert faster than it takes to watch the entire movie?
[21:12] <llogan> phelix: that depends on many factors
[21:13] <llogan> the input, your hardware, if you're performing any filtering, the encoder being used, the settings of the encoder, other bottlenecks, etc
[21:21] <Ove_BV> Hello. I am trying to decode H264 video, but I want to decouple the demuxing from the decoding.
[21:21] <Ove_BV> Right now I am demuxing using av_read_frame to get the AVPacket, and then I am decoding using a manually-allocated AVCodecContext.
[21:21] <Ove_BV> I have noticed that decoding does not work unless I copy the extradata from the format's codec context to my manually-allocated codec context.
[21:21] <Ove_BV> Here is the code I am using:
[21:21] <Ove_BV> codec = avcodec_find_decoder(codec_id);
[21:21] <Ove_BV> codec_ctx = avcodec_alloc_context3(codec);
[21:21] <Ove_BV> codec_ctx->extradata = format_ctx->streams[stream_idx]->codec->extradata;
[21:21] <Ove_BV> codec_ctx->extradata_size = format_ctx->streams[stream_idx]->codec->extradata_size;
[21:21] <Ove_BV> avcodec_open2(codec_ctx, codec, nullptr);
[21:21] <Ove_BV> Is there any way I can make it work without copying the extradata from the codec context?
[21:24] <Ove_BV> (decoding an MPEG4 file using the same code works without copying the extradata, I only have a problem for H264)
[21:25] <galex-713> Hello, I want to cut off some part of a video (I have an english and german version, and the german version has precisely 10 seconds of additionnal video added, and I want to remove it) and seeing at the video image per image, I identified the precise frames of begin and end of the sequence to cut, but some doesnt are at a precise time in seconds (like things ending with .3333333&), so is there way to cut videos acc
[21:25] <galex-713> video stream instead of time in seconds/milliseconds?
[21:42] <llogan> galex-713: sorry, but i don't understand your question
[21:58] <galex-713> llogan: I want to cut off the part situated between frame 507 and 746 (included) of video
[21:59] <galex-713> I could do 507/24 and 746/24 to get times, but that cause some other problems: 746/24 is an irrational number, and however the fps is not precisely 24 but something like 23.97&
[22:00] <galex-713> So I would like to know if there is a way to do like -ss and -t but with frames instead of seconds
[22:01] <galex-713> llogan: also, for the concat of videos: the accurate method pointed by your link doesnt work with ffmpeg (which however is deprecated), it says Unknown input format: 'concat'
[22:09] <galex-713> and the other method output following error: Unrecognized option 'filter_complex'
[22:09] <galex-713> Failed to set value '[0:v]trim=0:21.125,setpts=PTS-STARTPTS[v0]; [0:a]atrim=0:21.125,asetpts=PTS-STARTPTS[a0]; [0:v]trim=31.083:3191,setpts=PTS-STARTPTS[v1];
[22:09] <galex-713> [0:a]atrim=31.083:3191,asetpts=PTS-STARTPTS[a1]; [v0][a0][v1][a1]concat=n=2:v=1:a=1[out]' for option 'filter_complex'
[22:11] <JEEB> galex-713, that ffmpeg is an old old binary that's still in an older version of libav.
[22:11] <JEEB> libav does have a newer binary, avconv, but libav doesn't have the concat "format"
[22:12] <JEEB> probably the simplest way for you to use it would be to grab ffmpeg's ffmpeg
[22:12] <JEEB> !bin static
[22:12] <JEEB> use one of the static builds available http://ffmpeg.org/download.html
[22:12] <JEEB> you can just extract and run
[22:13] <galex-713> Ok
[22:16] <galex-713> Oh, it seems I already downloaded it long time ago& ^^
[22:16] <galex-713> And yeah, it works now, thanks :)
[22:33] <galex-713> JEEB: ok, it worked, thank you :)
[22:33] <galex-713> llogan: thank you too
[00:00] --- Fri Apr 11 2014
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