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July 2013
- 1 participants
- 62 discussions
[00:38] <llogan> what's a use case for ffmpeg daemon mode? (-d)
[00:53] <durandal_1707> isn't it obvious?
[01:19] <cxo> Is ffmpeg compatible with gcc 4.7?
[01:20] <durandal_1707> it should be
[01:20] <cxo> Why is build_stagefright hard coded to use 4.4.3?
[01:21] <durandal_1707> perhaps it was tested with only that one
[03:50] <cone-796> ffmpeg.git 03Michael Niedermayer 07master:cb6d02df4c03: asfdec: avoid using AVStream->priv_data
[03:50] <cone-796> ffmpeg.git 03Michael Niedermayer 07master:465aa53f21f3: asfdec: move packet_obj_size to ASFStream
[03:50] <cone-796> ffmpeg.git 03Michael Niedermayer 07master:6df20d0d3507: asfdec: remove "ignoring invalid packet_obj_size" code
[09:56] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:55c4cba2a703: huffyuvenc: BGRA support
[09:56] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:947f2d4d34f2: Merge commit '55c4cba2a7037161ea56b2d55b347c8381df7f1e'
[10:08] <cone-766> ffmpeg.git 03Martin Storsjö 07master:e1d5b244761c: hls: Store first_timestamp in units of AV_TIME_BASE
[10:08] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:5a270296cb01: Merge commit 'e1d5b244761cf69db655ad7ece1dbf2c13dd4fce'
[10:18] <cone-766> ffmpeg.git 03Martin Storsjö 07master:c44191039944: hls: Store all durations in AV_TIME_BASE
[10:18] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:4971551fd215: Merge commit 'c44191039944526dd7eb6e536990b555837961f5'
[10:24] <cone-766> ffmpeg.git 03Martin Storsjö 07master:a2b7eeeb0647: hlsproto: Store all durations in AV_TIME_BASE
[10:24] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:ef1544740e02: Merge commit 'a2b7eeeb06471979ee39fd3075a04633222678a6'
[10:30] <cone-766> ffmpeg.git 03Martin Storsjö 07master:06205b5efdcf: hls: Free packets when skipping packets when seeking
[10:30] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:7dfe798d148b: Merge commit '06205b5efdcf0bc4c5463bfdd02f09b5f79fc4cd'
[10:41] <cone-766> ffmpeg.git 03Diego Biurrun 07master:c47037c7a17e: fate: Add timefilter test
[10:41] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:4cf6e0008187: Merge commit 'c47037c7a17e4026d38dfa41da27b9f91506725b'
[10:48] <cone-766> ffmpeg.git 03Diego Biurrun 07master:10db1a9bca46: avutil/tree: K&R formatting cosmetics
[10:48] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:93b3a5edffa3: Merge commit '10db1a9bca46b0f760a1263d47142b2f57e195d7'
[11:12] <cone-766> ffmpeg.git 03Diego Biurrun 07master:d99a2b6d0a4d: avutil/tree: Drop debug messages that clutter the output
[11:12] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:647836608c9f: avutil/tree: change debug output to AV_LOG_DEBUG level
[11:12] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:db75537e2519: avutil/tree: allow adjusting the log level
[11:12] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:1f5755d869b5: Merge commit 'd99a2b6d0a4d0b2f5ba6a1f1637d6cb88c5c9184'
[11:19] <cone-766> ffmpeg.git 03Diego Biurrun 07master:711c4da1af71: fate: Add tree test
[11:19] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:dc737f0bfa80: Merge commit '711c4da1af71e0d26ca93626a3c2dd48821f1cc7'
[11:24] <cone-766> ffmpeg.git 03Diego Biurrun 07master:4822ee3ca620: configure: Detect AIX ar command instead of hardcoding it in the OS section
[11:24] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:5fae4678949b: Merge commit '4822ee3ca620a92cd2b0a9a03ea9e34288192c79'
[11:30] <cone-766> ffmpeg.git 03Diego Biurrun 07master:0e8c6f221a8d: avisynth: Fix a "AVISynth" vs. "AviSynth" struct name typo
[11:30] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:c7d17179571d: Merge commit '0e8c6f221a8ddb7dfb3c9e9bd0b33cb12e9391b8'
[11:38] <cone-766> ffmpeg.git 03Ed Torbett 07master:3357bccc5cb3: udp: Allow specifying multicast include/blocks as host names as well
[11:38] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:15ca3ab82531: Merge commit '3357bccc5cb31795f248cd72dc480025f3075a5b'
[11:50] <cone-766> ffmpeg.git 03Ed Torbett 07master:7d99c92945e4: udp: Keep track of include and exclude sources separately
[11:50] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:42f7c45d7b85: Merge commit '7d99c92945e4b2d12ed2126365482e1900340da0'
[11:57] <cone-766> ffmpeg.git 03Ed Torbett 07master:749722209097: rtpproto: Support more than one SSM include address, support excludes
[11:57] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:0b712f5d5e5f: Merge commit '74972220909787af5a3ffe66f7fa8041827c2bd2'
[12:02] <cone-766> ffmpeg.git 03Ed Torbett 07master:1f57d60129b0: rtsp: Support RFC4570 (source specific multicast) more properly.
[12:02] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:870f506cfeec: Merge commit '1f57d60129b0e297cd197c6031c4439b30a6b503'
[12:10] <cone-766> ffmpeg.git 03Diego Biurrun 07master:78b4bfdb84b7: Replace remaining obsolete PIX_FMT names with AV_PIX_FMT equivalents
[12:10] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:000f8bcd0360: Merge remote-tracking branch 'qatar/master'
[12:57] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:e5526a4baea1: avformat/gxfenc: replace deprecated PIX_FMT constants
[12:57] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:231e33f05038: avutil/pixfmt: dont refer to deprecated PIX_FMT in the documentation
[12:57] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:66487d73c3a9: avdevice/timefilter: cleanup the formating of the test output
[15:36] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:d6686149e2d7: asfdec: Skip to keyframe after seeking
[16:24] <cone-766> ffmpeg.git 03Timothy Gu 07master:6b255e5e70c7: doc/encoders: add libxvid doc
[16:35] <durandal_1707> lets see what is this about....
[16:36] <wm4> durandal_1707: btw, feature request: support mid-stream tags for mp3 and ogg (webradios do this)
[16:37] <durandal_1707> $$
[16:38] <BBB> I thought shoutcast did that
[16:38] <wm4> durandal_1707: I just transferred $10000000 open source dollars to your non-existent fantasy account
[16:38] <BBB> it takes 3 minutes to implement that in the http layer
[16:38] <BBB> or if you want a separate demuxer, it might take 5 minutes
[16:38] <wm4> BBB: shoutcast metadata sucks to hell, so support for that is disappearing and mid-stream tags take its place
[16:38] <BBB> oh I see
[16:39] <BBB> well I agree they suck
[16:39] <BBB> but usually end users don't care whether something sucks technically
[16:39] <wm4> yeah, but now the shoutcast info they send is totally insufficient
[16:46] <nevcairiel> so the shoutcast patch you send earlier is basically useless now? :D
[16:46] <wm4> nevcairiel: yes
[16:46] <wm4> on some streams
[16:47] <nevcairiel> so what do the others do, just a random idv3 block midstream?
[16:47] <wm4> not sure, but I think I've seen it at least with ogg
[18:08] <thefarcry118> I was just wondering if GCC for ARM has improved recently? For example, loop unrolling caused some issues with DSP functions.
[19:30] <BBB> an id3 block midstream sounds like a bad idea
[19:40] <Daemon404> BBB, im officially unemployed from aug 1 to aug 18
[19:40] <Daemon404> re: vp9
[19:49] <durandal_1707> wow how many will work on vp9 I expect vp10 by end of this year
[19:54] <Daemon404> 3 afaik
[19:54] <Daemon404> end of teh year? seems reasonable
[20:13] <burek> is this a known issue or is it just me: http://pastebin.com/wftJW3m2
[20:18] <Daemon404> burek, wtf i with random lisp
[20:18] <Daemon404> is*
[20:18] <burek> well, it's just a part with errors, during regular static builds process
[20:18] <burek> for 32bit version
[20:19] <Daemon404> with what compiler
[20:19] <burek> gss
[20:19] <burek> debian
[20:19] <burek> gcc*
[20:19] <Daemon404> never seen that before
[20:20] <burek> I'll try to provide full logs
[20:20] <burek> i was just thinking to solve it kinda easily ^^
[20:20] <Daemon404> is this latest head
[20:20] <Daemon404> that file had changes under 48 hrs ago
[20:21] <burek> yes
[20:21] <Daemon404> looks like it ma be related
[20:21] <Daemon404> poke michaelni
[20:22] <burek> ok, let me prepare full logs first
[20:28] <Daemon404> holy output_count deprecation warnings batman
[20:29] <burek> :)
[20:30] <durandal_1707> vote for anti-warning log!
[20:34] <BBB> Daemon404: 2 weeks? that's all? uh
[20:34] <BBB> ok
[20:34] <BBB> Daemon404: moving within vimeo or elsewhere?
[20:36] <Daemon404> within vimeo
[20:39] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:43411f416f6d: avcodec/get_bits: init_state is used for thread sync, make it volatile
[20:46] <nevcairiel> you would wish if a ticket is closed as a dup, you would link to the original
[20:56] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:121fc05b4dad: avcodec/get_bits: place volatile at the right spot
[21:00] <burek> nevcairiel +1
[21:17] <durandal_1707> but there is already to many open tickets
[22:29] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:1bbf94d84fa0: avfilter/af_compand: %f is for printing doubles
[22:32] <cone-616> ffmpeg.git 03Paul B Mahol 07master:d49252721a18: lavf/webvttenc: use proper printf format for int64_t values
[23:13] <cone-616> ffmpeg.git 03Romain Beauxis 07master:38536e592750: Support for shine 3.0.0
[00:00] --- Wed Jul 31 2013
1
0
[00:01] <alexschomb> exactly. not sure whether this is a problem anyway. right now it is still compiling ;)
[00:03] <alexschomb> wait, if I understood you right you mean that I should modify the PKG_CONFIG_PATH variable? I already tried .../output/lib/pkgconfig, but that didn't remove the error
[00:05] <roxlu> hi
[00:05] <alexschomb> hi
[00:05] <roxlu> can ffmpeg get input from a named pipe? (e.g. raw rgb images or audio)
[00:05] <llogan> alexschomb: probably don't need to. it looks like the external libraries were located anyway.
[00:07] <alexschomb> roxlu: I guess that answers your question? http://stackoverflow.com/questions/7605388/named-pipes-and-can-they-stream-…
[00:07] <alexschomb> llogan: no errors while compiling so far. so I guess you're right. Thanks for your fast help! :)
[00:09] <roxlu> alexschomb: partially :-) I'm looking for a solution where I write into a named pipe
[00:14] <alexschomb> http://ffmpeg.org/ffmpeg-protocols.html#pipe
[00:14] <alexschomb> replacing pipe:0 with your named pipe doesn't work?
[00:15] <llogan> there are some examples in the FAQ: http://ffmpeg.org/faq.html#Concatenating-using-the-concat-protocol-_0028fil…
[00:23] <roxlu> alexschomb: yeah I think I can just create a named pipe and use that as an input
[00:24] <roxlu> I only need to find a way to tell ffmpeg what kind of input I'm providing (raw rgb in this case)
[00:32] <alexschomb> llogan: finally compiled and copied the contents of my arm-directory to /usr/local/ on my RPI. but libaacplus is said to be missing although I followed the guide: http://pastebin.com/AXUgSbLJ
[00:36] <llogan> alexschomb: i've never used libaacplus. if i test that guide i may switch it to libfdk-aac from libaacplus
[00:41] <alexschomb> llogan: my fault. had to add /usr/local/lib to library path, http://www.cyberciti.biz/faq/linux-setting-changing-library-path/
[03:10] <Guest83225> I'm trying to use the encoder h264 with ffmpeg. I installed x264 from source and with few, and I configured ffmpeg with ./configure --enable-gpl --enable-shared --extra-ldflags="-ldl" --enable-libx264. Yet the config.log tells me fatal error: 'x264.h' file not found
[03:10] <Guest83225> and I get the error libx264 not found
[03:29] <klaxa> Guest83225: you need to add --extra-cflags="-I/path/to/include" and maybe replace --extra-ldflags="-ldl" with --extra-ldflags="-L/path/to/lib"
[03:29] <Guest83225> thanks! to which command?
[03:29] <klaxa> configure for ffmpeg source
[03:30] <klaxa> /path/to/include should be the path to the directory that contains x264.h and /path/to/lib should be the directory that contains libx264.so.XXX
[03:30] <klaxa> whereas libx264.so.XXX should probably be libx264.so.134
[03:30] <klaxa> *135
[03:41] <Guest83225> Thanks! The configuration seems to have worked. Yet my application still shows this error when I try to transcode a video: Unknown encoder 'h264'
[03:43] <Guest83225> How can I verify that it worked? In the output of the configure command, I see 'bzlib libx264' in the list of External libraries
[03:44] <Guest83225> oh, I also see libx264 in the list of enabled encoders, so yes, it worked
[03:54] <relaxed> Guest83225: the codec encoder name is "libx264", -c:v libx264
[03:54] <relaxed> -codec
[03:56] <Guest83225> now I get Unknown encoder 'libx264'
[03:59] <Guest83225> at this point, I think I can figure it out, thanks!
[04:20] <Guest83225> nope, ffmpeg -codecs | grep 264 shows only " D.V.LS h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10" which is the decoder. I need the encoder, which is libx264
[04:23] <Guest83225> I'm running this command on the ffmpeg executable in the same directory I ran the ./configure
[04:25] <relaxed> Guest83225: you need to compile ffmpeg with libx264 support, or use http://goo.gl/DPrRY
[04:30] <Guest83225> thanks! I don't see any installation instructions or configure or setup files. How do you recommend I install it?
[04:52] <Guest83225> nvm I'm on a mac. I'm looking for another build
[05:11] <relaxed> Guest83225: look at the ffmpeg.org download page
[05:38] <Guest83225> I ran through every step of https://trac.ffmpeg.org/wiki/MacOSXCompilationGuide, with only the error that libaacplus not found, yet when I ran make & make install, I got errors, beginning with this: Makefile:2: Makefile:2: config.mak: No such file or directoryconfig.mak: No such file or directory
[06:12] <Guest83225> nvm I got it up and running
[09:54] <nlight> what behavior can I expect from swscale if the input and output formats are the same?
[10:08] <b6> the AVFrames i'm reading have pkt_pts seemingly in milliseconds. i think those frames were encoded with microsecond precision. how can i tell?
[10:18] <elkng> I have a question related using ffmpeg libraries, anyone know something about ffmpeg libraries ? I think they can be used to work directly with wideo files on a low level, am I right ?
[10:18] <nlight> elkng, you are right
[10:18] <elkng> all right, so the question is:
[10:19] <elkng> I have a program like some sort of video editor but its more like frame manager, to edit video I convert video into bunch of png images and that program then uses SDL to show them and I can use "Left" "Right" to change images left and right like in some video editor, that program is better than image viewer like xnvies because if you use xnview
[10:19] <elkng> when you change images they blink and its very tiresome for eyes, also that program can do some other things, but now I want to know is it possible to somehowe use ffmpeg libraries or something to load frames directly from video file ? now I need to convert entire video to png images first then use something like "image=IMG_Load(path_to_file);
[10:19] <elkng> SDL_BlitSurface(image)" to load image from file and show it on the screen, I really would like not to convert video to png files because 5 minutes of video will consume 3GB of disk space, and its also time consuming operation, so is it possible to somehow skip part of converting video to images and use some functions from ffmpeg libraries to load
[10:19] <elkng> images from video file ?
[10:20] <nlight> yes, that's what ffmpeg is for
[10:21] <nlight> go here http://dranger.com/ffmpeg/tutorial01.html
[10:25] <elkng> nlight: but how fast it would be ? I have celeron 500, and in case I allready have converted video to png files, that program can show them at a rate about 10 frames per second because all it needs to do is to open png file, and its fast, but if I instead would use ffmpeg library how fast that library could access needed frame load it and give it
[10:25] <elkng> to my program for it to eventually show it ?
[10:28] <elkng> also my program uses SDL function "image = IMG_Load()" and it gets that image in "SDL_Surface" format, that also could increase time processing because of conversion, isn't it ?
[10:28] <elkng> beceuse I would need to convert from ffmpeg format to SDL_Surface format
[10:35] <elkng> any ideas ?
[10:35] <elkng> worth to try ?
[10:36] <nlight> celeron 500?
[10:36] <elkng> yes
[10:37] <nlight> what?
[10:37] <nlight> really?
[10:37] <elkng> yes
[10:37] <nlight> I don't think you can decode video in real time on that but dunno
[10:37] <elkng> "on that", was that offence ?
[10:38] <nlight> it's ancient hardware, so yes :D
[10:38] <nlight> it's a 7 years old cpu
[10:42] <elkng> I got it on 20 april in 2000
[10:43] <elkng> nlight: that PC also has video: Riva TNT, 400MB RAM 133Mhz, CDROM 48X speed, 2x80GB HDD
[10:43] <elkng> impressive ? monster
[10:43] <elkng> Riva TNT 16MB
[10:43] <nlight> wow
[10:44] <elkng> and any video I need to convert to 650xXXX so it will not be too slow to watch
[10:45] <nlight> well, go ahead and try it out
[10:46] <nlight> what is the difference between AVInputFormat and AVOutputFormat?
[10:46] <nlight> and why does AVFormatContext have both?
[10:46] <nlight> should I expect only one of iformat/oformat to be available at any time?
[11:07] <hendry> anyone tried streaming to icecast using ffmpeg? A "PUT" IIUC?
[12:27] <jcath> hi, friends. does ffmpeg support the blu-ray directory as input?
[12:28] <Fjorgynn> blu-ray direcotry?
[14:13] <pyBlob> I've got an avi file that contains 2 mjpg-streams
[14:15] <Fjorgynn> aha
[14:15] <pyBlob> ffmpeg -i HNI_0001.AVI -vcodec copy -f image2 dump/out%05d.jpg
[14:15] <pyBlob> that extracts the single jpg-frames
[14:16] <pyBlob> problem is that only the first 2 frames contain a HuffmanTable
[14:16] <pyBlob> > other jpg-files can't be read by standard image viewers
[14:20] <relaxed> pyBlob: try with -bsf:v mjpeg2jpeg
[14:23] <pyBlob> relaxed: that works, thanks =)
[14:24] <relaxed> you're welcome
[14:34] <b4u> is there a way to set a crf with libx264 but also have an upper bound on the bitrate rather than an average?
[14:36] <JEEB> bitrate is always an average of something
[14:36] <JEEB> amount of bits over something
[14:36] <JEEB> anyways, you will have to set BOTH -bufsize and -maxrate
[14:36] <b4u> is it possible to tell the encoder not to encode any frames higher than a specific bitrate though?
[14:36] <b4u> at all
[14:36] <b4u> but don't I need to know what will be buffered for bufsize?
[14:36] <JEEB> do you really need per-picture VBV?
[14:37] <JEEB> you really need to have a specific use case to need per-picture VBV
[14:38] <b4u> bandwidth limited situation where I need to know it won't go over a certain bitrate but can't guarantee a certain amount will be buffered in advance either
[14:39] <JEEB> well single picture VBV still means you will have to have that picture's amount buffered at the very least
[14:40] <b4u> okay, how do I do something like that?
[14:40] <b4u> I've not heard of it before
[14:40] <JEEB> I won't tell you unless I'm sure you need it :P
[14:40] <JEEB> because it means that encoder will be really limited and thus you are pretty much derping quality that way
[14:41] <b4u> well my other way of doing it was to set an arbitrarily low bufsize
[14:41] <b4u> of like 200k
[14:42] <JEEB> anyways, it sounds like you are dealing with something you could very well fix somewhere else
[14:43] <JEEB> instead you're just opting to *heavily* derp the encoder
[14:44] <b4u> does a low bufsize do that also?
[14:44] <JEEB> yes
[14:46] <b4u> is there a way to do this even with 2 pass?
[14:47] <JEEB> uhh, what? VBV works the same way no matter what rate control you are using (except maybe constant quant, although no-one should in general be using that rate control mode except for development purposes)
[14:51] <b4u> the problem is I can't guarantee a certain amount of data will be buffered
[14:51] <b4u> I can do a certain amount of seconds
[14:51] <b4u> which may vary depending on the video
[14:53] <JEEB> it can vary of course, but in general it should be OK. Just use <maxrate> * seconds_to_buffer in that case for bufsize?
[14:53] <bencoh> wow, two vbv-days in a row :)
[14:53] <JEEB> and match them that wise
[14:53] <b4u> uhh but that won't work will it?
[14:53] <JEEB> why wouldn't it
[14:53] <b4u> or could produce bad buffer times
[14:54] <b4u> because right now people wait a number of seconds worth of buffer to view a video rather than data
[14:54] <b4u> so I guess small videos will start more slowly than normal
[14:54] <b4u> perception wise
[14:55] <JEEB> I don't follow you
[14:56] <b4u> well right now a video with 100k bitrate takes not long to start
[14:56] <b4u> because 6 seconds of that is not much
[14:56] <JEEB> and it still wouldn't be because flash would then see that not much data is going on?
[14:56] <b4u> what I mean is
[14:57] <b4u> ok so you said set bufsize to seconds * maxrate?
[14:57] <b4u> that means the encoder assumes I will buffer that much before playback
[14:57] <b4u> yes?
[14:57] <JEEB> yes
[14:57] <b4u> my player only buffers in seconds
[14:57] <JEEB> yes, most probably some kind of dynamical thing
[14:58] <JEEB> so it looks how much data is flowing over and tries to adapt to that
[14:58] <JEEB> so it should still buffer the same amount?
[14:58] <b4u> it's a constant in seconds, I think
[14:58] <JEEB> seconds of content I would guess :P
[14:59] <JEEB> or dunno
[14:59] <b4u> but I mean the first seconds of the video might not be at that maxrate
[14:59] <b4u> so surely I will not necessarily buffer enough data
[14:59] <b4u> to meet what the encoder expects
[14:59] <JEEB> as far as I know flash shouldn't be that dumb
[14:59] <JEEB> as to not be adaptive related to your use of bandwidth
[14:59] <b4u> how would it know when to start playing?
[15:00] <b4u> short of having a time or data limit
[15:00] <JEEB> anyways, tons of people are using it well
[15:00] <JEEB> so you are most probably looking at it too bad :P
[15:00] <JEEB> and yes, I hate things that don't let you set the amount of data to buffer
[15:01] <JEEB> anyways, single picture VBV totally was the wrong thing for you, as I thought
[15:01] <b4u> so what kind of buffer limit should I set do you think? I feel like normalseconds * maxrate is too much
[15:01] <JEEB> just use bufsize = <seconds to buffer> * maxrate
[15:01] <b4u> I could maybe do 1 or 2 seconds at maxrate
[15:04] <JEEB> anyways, for all of the things I've so far seen if you have to set a time on the decoder side to buffer, the bufsize to be calculated with that should be <time_to_buffer> * maxrate
[15:04] <nlight> anyone seen these kind of artifacts - http://i.imgur.com/tob4Bm5.jpg ?
[15:17] <JEEB> b4u, I just checked the blog of a certain x264-using streaming thingy, and it uses the exact same thing to calculate the matching between seconds of buffering and bufsize
[15:18] <JEEB> and this is with CRF rate control, yes
[15:25] <luc4> Hi! I noticed that libavformat needs to seek when using callbacks muxing a stream. Is this always necessary? Can I avoid this somehow or is it impossible to avoid?
[15:26] <JEEB> b4u, btw I would say that such VBV-specific questions with x264 might be better off being asked on #x264 rather than here
[15:28] <b4u> sorry, was afk
[15:30] <Sum> Afternoon chaps - I'm having trouble getting ffmpeg to install in ubuntu 13.04 64 bit using the guide on http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide. Running ffmpeg 2>&1 | head -n1 gives an error saying I need to install ffmpeg via apt-get install. Any ideas?
[15:30] <JEEB> Sum, sounds like you're just not having wherever you compiled your ffmpeg binary in your PATH
[15:32] <b4u> you didn't forget to "make install" right? :P
[15:33] <Sum> yes - i did 'make install'
[15:33] <b4u> probably what JEEB says then
[15:34] <Sum> @ubuntu:~/ffmpeg_sources/ffmpeg$ find ~ -name ffmpeg -print gives:
[15:34] <JEEB> uhh
[15:34] <JEEB> you set a prefix as well as a bindir when configuring
[15:34] <JEEB> IIRC the guide says to do that
[15:34] <JEEB> the bindir would be the place where the binary gets installed
[15:35] <Sum> ah - i just copied and pasted the commands - will read a bit further
[15:40] <Sum> ah - i see I think. I guess running the following will fix the path to the executable? echo 'export PATH=$PATH:/home/sum/bin' >> ~/.bashrc
[15:40] <JEEB> basically yes
[15:40] <JEEB> in the next opened terminal that is
[15:40] <JEEB> since that file is run every time you open bash
[15:41] <JEEB> for currently open terminals you would have to run that export part in the terminal
[15:42] <Sum> brill! thanks :-)
[17:25] <Sum> Hey folks. I'm wondering if you can help with this: I don't appear to have the 'libfaac' encoder installed and following https://help.ubuntu.com/community/Medibuntu hasn't appeared to install that encoder. any hints on where to get this encoder from and how to install it?
[17:26] <microchip_> Sum: did you ask in #ubuntu ?
[17:26] <JEEBsv> That's because faac was found to be nonfree, aka not distributable. Which means you will have to compile it yourself. That said, now we have something better than that, so you will most probably want to build fraunhofer's encoder instead :)
[17:27] <JEEBsv> (fdk-aac)
[17:29] <JEEBsv> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide there's always this, but this includes way too many encoders in it, so feel free to read through and compile only the external encoders you need
[17:38] <Sum> ok - is this another instance where the fdk-aac 'file' (executable?) isn't in the path environmental variable (or whatever it's called in unix? As per the Ubuntu Compilation Guide I ran make, make install adn make distclean adn there is a file that looks like the one I need - /home/sum/ffmpeg_sources/fdk-aac - is this the file that should be in a path statement?
[17:41] <JEEBsv> Sum: ffmpeg -codecs:a |grep "aac"
[17:42] <JEEBsv> see if it's listed there
[17:42] <JEEBsv> if it is, you can just use its name with -c:a
[17:45] <Sum> JEEBsv: - it isn't I think - output is DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfdk_aac ) D.A.L. aac_latm AAC LATM (Advanced Audio Coding LATM syntax)
[17:45] <Sum> do i need to recompile ffmpeg with this encoders support added?
[17:47] <klaxa> >encoders: aac libfdk_aac
[17:47] <klaxa> that looks good though
[17:48] <JEEBsv> Sum: libfdk_aac is fdk-aac :)
[17:48] <JEEBsv> so -c:a libfdk_aac will use it
[17:48] <JEEBsv> instead of -c:a libfaac for example
[18:04] <Sum> JEEBsv - fantastic! Now transcoding at 110fps instead of 8 - I'm a very happy man. Thanks for your patience. :-)
[20:21] <phr3ak> how could I join several asf?
[20:28] <durandal_1707> phr3ak: with concat demuxer
[22:31] <roxlu> hi guys, I'm a bit cross posting, but I hope someone here knows this ....
[22:31] <roxlu> when I combine a video and audio file, how can I set that the resulting video takes the duration of the video, not of the audio?
[00:00] --- Wed Jul 31 2013
1
0
[00:00] <durandal_1707> what I should do instead?
[00:01] <cehoyos> You wrote above that you don't want to revert but instead fix the issue properly.
[00:02] <cehoyos> (I would of course have been able to revert but you said you are against reverting.)
[00:07] <durandal_1707> i thought it was possible to fix bug without reverting, but i found it revert is better
[00:07] <durandal_1707> i will rewrite this code anyway
[00:07] <cone-858> ffmpeg.git 03Carl Eugen Hoyos 07master:2c046c718aef: Revert "Avoid overflows when reading pgm files with maxval != 255 and != 65535."
[00:07] <cone-858> ffmpeg.git 03Paul B Mahol 07release/1.1:ef81f55ec7a9: Revert "pnm: remove nonsense code"
[00:07] <cone-858> ffmpeg.git 03Paul B Mahol 07release/1.2:9f2d73168f54: Revert "pnm: remove nonsense code"
[00:07] <cone-858> ffmpeg.git 03Paul B Mahol 07release/2.0:0047a310905c: Revert "pnm: remove nonsense code"
[00:25] <durandal_1707> if filter_ctx->input/output_count is deprecated what is used instead?
[00:28] <cone-858> ffmpeg.git 03Paul B Mahol 07master:329221eee7d6: pnm: fix 255 < maxval < 65535 for pam with depth 1
[00:28] <cone-858> ffmpeg.git 03Paul B Mahol 07master:a5155294e571: pnm: do not set avctx->coded_frame
[00:43] <cone-858> ffmpeg.git 03Paul B Mahol 07master:e999f2339ab0: truemotion2: make code independent of sizeof(AVFrame)
[00:43] <cone-858> ffmpeg.git 03Paul B Mahol 07master:19da86f3ff43: targaenc: remove unused TargaContext
[01:04] <durandal_1707> hey http://ffmpeg.gusari.org/viewtopic.php?f=8&t=1014
[02:07] <michaelni> is jon severinsson on irc ?
[02:18] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:f3c300d0a4d3: update all trac links to use the trac subdomain
[02:44] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:73746237a188: Prepare for 0.8.7 Release
[02:44] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:a563e4af9f56: oma: Validate sample rates
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:2eaf8698a3bb: avfiltergraph: check for sws opts being non-NULL before using them.
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:c65fb5b41b8e: xmv: do not leak memory in the error paths in xmv_read_header()
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:0f6364b62bb5: bmv: check for len being valid in bmv_decode_frame().
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:9aa2eee31389: xmv: check audio track parameters validity.
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:4c7f40c6df83: dfa: check for invalid access in decode_wdlt().
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:881526744eea: lavf: make sure stream probe data gets freed.
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:74753cf1a99b: indeo3: fix data size check
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:c5084a176549: rv10: check that extradata is large enough
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:9b2af4d080c7: indeo3: check motion vectors.
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:c579d4283edb: indeo3: switch parsing the header to bytestream2
[02:44] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:43c0a87279e7: qdm2: check that the FFT size is a power of 2
[02:44] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:be90f0279d07: Merge commit '43c0a87279e717c1384314c6da7155c306ee7c60' into release/0.10
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:d2d2ddf9a34e: oggdec: fix faulty cleanup prototype
[03:26] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:7bf6a86f2e49: update Changelog
[03:26] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:a590979988b1: aac: check the maximum number of channels
[03:26] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:6742f0408dba: indeo3: fix off by one in MV validity check
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:f4bb72d33db2: id3v2: check for end of file while unescaping tags
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:31ed79af7f47: wav: Always seek to an even offset
[03:26] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:2f61e4090820: update Changelog
[03:26] <cone-858> ffmpeg.git 03Michael Smith 07release/0.10:4941dfb4f6de: proresdec: support mixed interlaced/non-interlaced content
[03:26] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:e6617580e3f2: update Changelog
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:a03dcec36474: mjpegdec: properly report unsupported disabled features
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:2ebabfff4804: jpegls: return meaningful errors
[03:26] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:ca4a25acf89b: jpegls: factorize return paths
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:d26bc6c6b69f: jpegls: check the scan offset
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:6711d410dc13: wavpack: validate samples size parsed in wavpack_decode_block
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:da5cf7e45263: ljpeg: use the correct number of components in yuv
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:33492ad81000: mjpeg: Validate sampling factors
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:7ca8d8223db2: mjpegdec: validate parameters in mjpeg_decode_scan_progressive_ac
[03:26] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:42fed7f433e6: wavpack: check packet size early
[03:26] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:7118358a950e: Merge commit '42fed7f433e6d2167ffd4aae31905b583a53b988' into release/0.10
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:5a6af4fd74a4: wavpack: return meaningful errors
[04:02] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:ea7ba1d8717d: apetag: use int64_t for filesize
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:96de1c5ed90b: tiff: do not overread the source buffer
[04:02] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:e98f95670bf1: Prepare for 0.8.8 Release
[04:02] <cone-858> ffmpeg.git 03Kostya Shishkov 07release/0.10:e96aaa5622ed: smacker: fix an off by one in huff.length computation
[04:02] <cone-858> ffmpeg.git 03Kostya Shishkov 07release/0.10:d7b7b10518cc: smacker: check the return value of smacker_decode_tree
[04:02] <cone-858> ffmpeg.git 03Kostya Shishkov 07release/0.10:db0c8061fe54: smacker: pad the extradata allocation
[04:02] <cone-858> ffmpeg.git 03Kostya Shishkov 07release/0.10:b40870e63640: smacker: check frame size validity
[04:02] <cone-858> ffmpeg.git 03Alexandra Khirnova 07release/0.10:9248f789d1fa: vmdav: convert to bytestream2
[04:02] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:078e68d2617f: 4xm: don't rely on get_buffer() initializing the frame.
[04:02] <cone-858> ffmpeg.git 03Anton Khirnov 07release/0.10:e797b7787b25: 4xm: check the return value of read_huffman_tables().
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:284ac9191b75: 4xm: use the correct logging context
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:e5679444fd60: 4xm: reject frames not compatible with the declared version
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:6a4f1e784e39: 4xm: check bitstream_size boundary before using it
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:0c943d1cdd18: 4xm: do not overread the source buffer in decode_p_block
[04:02] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:b20004b2e616: lavc: move put_bits_left in put_bits.h
[04:02] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:4ff5167ee7fd: wmapro: make sure there is room to store the current packet
[04:02] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:a3539d26ecee: Merge commit '4ff5167ee7fdee6d35c1bb2558172329ae6ec770' into release/0.10
[04:40] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:4e1999ebcb67: wmapro: check num_vec_coeffs against the actual available buffer
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:46e09894acb2: wmapro: return early on unsupported condition
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:bd5ff335ec55: wmapro: check the min_samples_per_subframe
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:f16aa5843ff9: wmapro: error out on impossible scale factor offsets
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:7658333c1755: indeo: Properly forward the error codes
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:7eff48029f99: indeo: use proper error code
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:0a1d02ca77c9: indeo: check for reference when inheriting mvs
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:e6a365b5d2fc: lavc: check for overflow in init_get_bits
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:c9d8424395fc: indeo: use a typedef for the mc function pointer
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:e22a5d490de1: kmvc: use fixed sized arrays in the context
[04:41] <cone-858> ffmpeg.git 03Luca Barbato 07release/0.10:79edb9adf619: kmvc: Clip pixel position to valid range
[04:41] <cone-858> ffmpeg.git 03Reinhard Tartler 07release/0.10:8c62082b5127: Changelog for 0.8.8
[04:41] <cone-858> ffmpeg.git 03Justin Ruggles 07release/0.10:e786cc333120: swfdec: do better validation of tag length
[04:41] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:62d9d4d9d4be: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10
[05:14] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:4e17e9f8afc4: update for 0.10.8 Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at>
[05:14] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:5bb347a3b4a8: dct-test: fix order of evaluation bug
[05:14] <cone-858> ffmpeg.git 03Michael Niedermayer 07release/0.10:e3e25777944e: ffserver: fix order of evaluation bugs
[11:14] <cone-858> ffmpeg.git 03Luca Barbato 07master:e13a929314e4: xl: K&R formatting cosmetics
[11:15] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:688331c51ce6: Merge commit 'e13a929314e4a9a09d3146648f876f162f46cc32'
[11:21] <saste> durandal_1707, you should not consider user ffmpeg-related incompetence as a personal attack
[11:22] <saste> about workarounds, most people prefer that way, just because they can fix it quickly
[11:22] <saste> it will broke apart the next time they upgrade, but in the short time workarounds represent the shortest path solution
[11:23] <saste> the weird thing is that the guy found the time to document all that stuff, for no obvious reason
[11:32] <cone-858> ffmpeg.git 03Luca Barbato 07master:f1cb490d6d73: xl: Fix the buffer size check
[11:32] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:03853b1091e4: Merge commit 'f1cb490d6d7391ff7e28cc376908cc98a652228d'
[11:38] <cone-858> ffmpeg.git 03Luca Barbato 07master:0e78ef0f94cf: flac: use meaningful return values
[11:38] <cone-858> ffmpeg.git 03Luca Barbato 07master:45ee556d51ef: qdm2: Whitespace cosmetics
[11:38] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:5f312139dff4: Merge commit '45ee556d51ef04d79d52bf6b0b7f28a4d231cb0c'
[11:42] <saste> log level bikeshed
[11:50] <cone-858> ffmpeg.git 03Luca Barbato 07master:62b1e3b1031e: aasc: Check minimum buffer size
[11:50] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:0c6f382e8557: Merge commit '62b1e3b1031e901105d78e831120de8e4c3e0013'
[11:52] <durandal_1707> michaelni: if i prefer to merge every single commit stop doing merges
[12:06] <cone-858> ffmpeg.git 03Paul B Mahol 07master:b09d86c6366f: utvideoenc: use av_image_copy_plane()
[12:06] <cone-858> ffmpeg.git 03Zhang Rui 07master:0f51c398beac: http: Support reading gzip/deflate compressed data
[12:06] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:39a69d9dfbc7: Merge commit '0f51c398beac87682b2249662b97e30512f7868c'
[12:20] <cone-858> ffmpeg.git 03Luca Barbato 07master:3ca1dd2502a8: xl: Make sure the width is valid
[12:20] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:6561aae8a955: Merge commit '3ca1dd2502a860989ee5289316af6b9816cee78f'
[12:33] <cone-858> ffmpeg.git 03Luca Barbato 07master:4f03a77e5259: oma: refactor seek function
[12:33] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:bc672a016fa1: Merge commit '4f03a77e52596cbe9ec179666ddb3e0345a8133a'
[12:46] <michaelni> mergig comits one at a time makes later bisecting and debuging much easier
[12:48] <michaelni> git can bisect of course over arbitrary complex merges but that doesnt help if one ancestor branch is 2+ years outdated and missing the feature one wants to debug
[12:49] <michaelni> also it avoid having to rebase huge merges when others push changes while confilcts are being resolved
[12:51] <cone-858> ffmpeg.git 03Luca Barbato 07master:9d0b45ade864: oma: check geob tag boundary
[12:51] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:9d8fb2374723: Merge commit '9d0b45ade864f3d2ccd8610149fe1fff53c4e937'
[12:51] <ubitux> BBB: i really don't know what is the timeline and workload required for each thing; as long as i learn about video codecs i'm happy with anything
[12:59] <durandal_1707> michaelni: cherry-pick, actually pushing real commits and not merges actually show what is changed
[12:59] <durandal_1707> how i'm supposed to see what really changed?
[12:59] <durandal_1707> by using merges you helping noone
[13:01] <cone-858> ffmpeg.git 03Luca Barbato 07master:2219e27b5b17: oma: correctly mark and decrypt partial packets
[13:01] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:5fb652dddb79: Merge commit '2219e27b5b17d146e4ab71a3ed86dfc013fb7a93'
[13:02] <michaelni> the gitlog script could be changed to show the changes to the first parent on merges
[13:03] <michaelni> also somethig like "git log --first-parent -p -m" might work
[13:04] <durandal_1707> this is far from useful
[13:20] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:c5b46a064d99: lavf: Don't interpret just slightly broken timestamps as wraparound
[13:20] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:f7bc03bcaff5: Merge commit 'c5b46a064d9991f2cd045c90179fccf35ecffc34'
[13:28] <michaelni> if you think cherry picks work better (against everything linus says about rebasing others history), noone stops you from using them to "merge" changes from libav into your private git repo. If they (against my expectations) do work better nothing would be in the way to reconsider things (given there are volunteers, lack of technical issues ...)
[13:30] <durandal11707> michaelni: i read what linus said, and what you are doing is definitely not following him
[13:32] <michaelni> what libav is doing is not following him really, and thats where all the problems stem from
[13:32] <durandal11707> this is like tallking with wall
[13:35] <michaelni> i cant change libav, nor can i change your oppinon on merges but cherry picks dont work as replacement for merges, you are welcome to show me wrong on any point i said
[13:36] <durandal11707> i'm not looking for replacement of merges, i do not want merges at all
[13:36] <durandal11707> just commit normal way
[13:38] <michaelni> normal way is to use merges (see linux or any other large project)
[13:38] <nevcairiel> merges allow the history to be used for conflict resolving, ie. with a 3-way merge, only if the merge history is known it can use the full power of this merge and resolve more cases automatically
[13:38] <durandal11707> but merges: hides actuall changes, introduces new changes, introduces subtle bugs
[13:40] <durandal11707> nevcairiel: you cleary do not know what you are talking about, should ffmpeg merge itself to 'speed' development and help conflict resolving?
[13:40] <nevcairiel> i know what i talk about, but you clearly dont know the full power of git
[13:41] <durandal11707> if the full power of git is to create subtle bugs than its really nice feature
[13:42] <nevcairiel> this wouldnt change if you apply the changes any other way
[13:43] <durandal11707> not at all, who check that merges are actually doing the right thing?
[13:43] <nevcairiel> who checks if a cherry-pick does the right thing?
[13:44] <nevcairiel> a cherry-pick is just a rebased fast-forward merge, it uses the same mechanism internally
[13:53] <durandal11707> with merges, person that do merges usually just let git do it and do not care single bit if result is optimal
[14:03] <cone-858> ffmpeg.git 03Martin Storsjö 07master:eb33ba04e03d: hls: Return all packets from a previous variant before moving on to the next one
[14:03] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:07440c938073: Merge commit 'eb33ba04e03d9f36e23fffd442510c824be709c3'
[14:08] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:82bf8c878345: hls: Copy the time base from the chained demuxer
[14:08] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:cdd2d73d315e: hls: Don't check discard flags until the parent demuxer's streams actually exist
[14:08] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:d31c0fcfa55a: Merge commit 'cdd2d73d315ecaf19ff49e64c91923275f1bda68'
[14:15] <cone-858> ffmpeg.git 03Martin Storsjö 07master:c11e33a3d966: hls: Set stream offset before opening a chained demuxer
[14:15] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:2f9e97b32a7a: Merge commit 'c11e33a3d9665dd1fc5dbdecdd03a4860ac6a622'
[14:29] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:9d64f236292b: hls: Respect the different stream time bases when comparing dts
[14:29] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:65dcb5449344: Merge commit '9d64f236292ba28018dd9afd2d57f8f944b33f81'
[14:36] <cone-858> ffmpeg.git 03LYF 07master:c110cbf6b59d: hls: Create an AVProgram for each variant
[14:36] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:994e09345ef9: Merge remote-tracking branch 'qatar/master'
[16:38] <durandal_1707> michaelni: why no merges have bunch of See: .... i'm not interested what commits are being duplicated in commit history
[16:38] <durandal_1707> *now
[16:46] <cone-858> ffmpeg.git 03Paul B Mahol 07master:2786ac777d2c: zmbv: use av_image_copy_plane()
[17:00] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:a9903f7ec123: Revert "bitstream: make vlc init of static tables thread safe."
[17:00] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:6af8326354ed: avcodec/ff_init_vlc_sparse: use a spinlock for thread sync
[17:00] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:f80ccf074a3b: avcodec/ff_init_vlc_sparse: use a local array for static vlcs avoiding malloc()
[17:07] <durandal_1707> ubitux: why your site is dead?
[19:41] <cone-858> ffmpeg.git 03Carl Eugen Hoyos 07master:f2118d44c9f1: Fix compilation with --disable-everything --enable-shared.
[20:42] <cxo> On Android if I use stlport_static, at runtime I get a Zdlpv symbol could not be found. If I use gnustl_port, at compile I get undefined reference to _M_allocate/deallocate. What's going on?
[20:44] <cxo> (Trying to use build_stagefright)
[20:58] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:3cd8aaa2b2e7: avcodec/kmvc: fix MV checks
[21:13] <cone-858> ffmpeg.git 03Paul B Mahol 07master:6b68e2a43b34: lavfi: add compand filter
[21:24] <llogan> durandal_1707: you missed the me/be typo
[21:37] <cxo> I think I got disconnected. Any body know if build_stagefright works?
[22:06] <durandal_1707> cxo: give exact build failure
[22:08] <cone-858> ffmpeg.git 03Paul B Mahol 07master:c96b3ae8db35: doc/filters: fix typo
[22:21] <cxo> durandal_1707, sure
[22:28] <cxo> I'm just getting the output now.... By the way, is ffmpeg not compatible with google's stlport_static?
[22:30] <durandal_1707> i dunno what is that
[22:33] <cxo> https://raw.github.com/flyskywhy/android-ndk-r7b/master/docs/CPLUSPLUS-SUPP…
[22:36] <cxo> I just did a git pull and remake and it compiles now with gnustl_port (it didnt yesterday). I get a different runtime error (compared to when using stlport) http://pastebin.com/Ljk3rnt1
[22:38] <durandal_1707> cxo: i see you disabled bunch of stuff, it happens with normal setup too?
[22:39] <cxo> What is a normal setup?
[22:40] <durandal_1707> the one you do not disable everything
[22:41] <cxo> But I am not using --disable-everything
[22:42] <durandal_1707> but you use --disable-decoders
[22:42] <cxo> Do you want me to remove that and rebuild?
[22:43] <durandal_1707> note this just to find out if the source of problem is bug in Makefile
[22:43] <durandal_1707> cxo: yes, if noone give some better solution
[22:44] <cxo> I dont understand c++ symbols, is OMXCodec::create the same as the missing "_ZN7android8OMXCodec6CreateERKNS_2spINS_4IOMXEEERKNS1_INS_8MetaDataEEEbRKNS1_INS_11MediaSourceEEEPKcj"
[22:44] <cxo> If its the same, then that symbol is not part of FFmpeg
[22:46] <durandal_1707> i can't dechiper that error at all, thus way i proposed ....
[22:50] <cxo> No difference, same error
[22:52] <durandal_1707> perhaps shared is broken, what if you try static?
[22:54] <cxo> where do i go for static stagefright libs?
[22:54] <cxo> And back to my original question; has anyone know if tools/build_stagefright works?
[22:55] <durandal_1707> well it worked for someone
[22:55] <cxo> who?
[22:56] <cxo> maybe they can help me
[22:56] <durandal_1707> you should also open bug report
[22:58] <cxo> I will submit a bug report
[00:00] --- Tue Jul 30 2013
1
0
[00:55] <alsu> how do I print the Lavf version ffmpeg is using?
[05:21] <kidx> I have a question how do i get the latest FFMPEG and also why does it say outdated on my Lubutu based distro and one last thing how can i screen cast and broadcast?
[05:24] <nownot> question on avconv, how would I convert flac to alac (mp4) ? I keep getting errors like "utomatic encoder selection failed for output stream #0:0. Default encoder for format ipod is probably disabled. Please choose an encoder manually."
[05:25] <kidx> I love ffmpeg just dont know why my distro has it outdated when a version was release this year?
[05:26] <kidx> I cant make youtube videos or broadcast with ffmpeg not working with oput bugs?
[05:26] <sacarasc> kidx: Get one of them ^
[05:27] <kidx> how do i install latest one
[05:27] <kidx> i want latest ffmpeg
[05:27] <sacarasc> Just get a static build and you won't need to build/install.
[05:27] <kidx> what is that a deb
[05:27] <sacarasc> nownot: avconv is not part of ffmpeg. Try #libav.
[05:27] <kidx> where i get that from
[05:28] <kidx> can i get a 2.0 static build?
[05:28] <nownot> sacarasc: oh, sorry. well, do you know the command I would use for ffmpeg? I'm pretty impartial as long as the job gets done
[05:29] <sacarasc> kidx: The second link is newer than 2.0...
[05:30] <kidx> how do i install this i am new here and love this but new?
[05:31] <kidx> I just wnat that stupid thing to goaway where it says please use libav I dont want to
[05:31] <sacarasc> Just unarchive it, then use ./ffmpeg in the directory you put the binary in.
[05:32] <kidx> oh
[05:33] <kidx> thanks you gusy are very helpfull unlike libav they have attitudes not supporting bash scripts and stuff question do you guys know how to capture the screen or broadcast or have a fourm i ca look for the topics in?
[05:34] <sacarasc> !forum
[05:34] <sacarasc> Bah.
[05:34] <sacarasc> The link is in the topic, anyway.
[05:36] <kidx> thansk alot guys my dad has used this all the time and now my turn hehe well thansk for every thing one thing before i go is .ffmpeg in the home folder or no casue i dont see it?
[05:37] <sacarasc> I don't think ffmpeg makes one.
[05:37] <kidx> so where is the directory of ffmpeg go once install i am on lubuntu
[05:38] <sacarasc> The static build shouldn't install.
[05:38] <sacarasc> It's just a binary you run when you want to.
[05:38] <kidx> oh ok
[05:38] <kidx> well this is confusing now casue i have ffmpeg installed on here and its saying please use avconv
[05:39] <kidx> and all i wanted to do was update that to latest
[05:39] <sacarasc> In the directory where you put the ffmpeg binary, run ./ffmpeg to run it.
[05:39] <kidx> let me try
[05:39] <sacarasc> If you want to install it, add that directory to your path BEFORE where ffmpeg is installed.
[05:40] <kidx> i am new here all i wanna do is make youtube videos with out lag.
[05:40] <sacarasc> And then it should find the static build before the avconv thing.
[05:40] <kidx> what you are saying is unarchive the latest to the place its inatalled on my HD
[05:41] <kidx> overright the existing one and voila
[05:42] <sacarasc> That would probably work, but could cause problems with your package manager. Put it somewhere like ~/bin/, then add $HOME/bin to the $PATH BEFORE the rest of it... (So it would looke like PATH=$HOME/bin:$PATH or similar.)
[05:43] <kidx> make a bin folder in my home folder and extract it there correct and cd to the driver and use./ffmpeg
[05:43] <kidx> sounds simple
[05:44] <nickeest> I can't seem to convert an mkv file to an avi. It's an h264 and ac3 audio. I get error: [mp2 @ 0x8331a90]encoding 6 channel(s) is not allowed in mp2
[05:45] <kidx> also i hope ffmpeg stays forever
[05:46] <sacarasc> nickeest: -ac 2
[05:46] <nickeest> sacarasc, can you give me the full command please, I am not so literate
[05:46] <sacarasc> nickeest: What command do you have so far?
[05:46] <kidx> nickeest add -ac 2 to your command you have i believe
[05:47] <nickeest> sacarasc, yep its working now
[05:47] <nickeest> kidx, thx
[05:47] <kidx> np
[05:47] <nickeest> sacarasc, thx
[05:47] <kidx> i am here to help as well as get help
[05:47] <kidx> this is why i love ffmpeg
[05:53] <kidx> question how do I do fullscreen broadcast with our the game going black?
[05:56] <kidx> also it is not grabing my screen
[05:56] <kidx> giving me errors and this avconv thing is ticking me off i am gona uninstall ffmpege
[06:04] <kidx> Unknown input format: 'x11grab'
[06:04] <kidx> thats the error i am getting on linux
[06:08] <nickeest> How do I estimate completion time? I see: frame=22601 fps=18 size= 84343kb time=958.34 bitrate=721.0kbits/s
[06:08] <sacarasc> I'm not sure the static builds are built with X stuff.
[06:09] <nickeest> oh oh, now it printing that line I mentioed on new lines and scrolling the page.
[06:10] <sacarasc> Make your terminal larger.
[06:10] <kidx> nickeest you jut tryign to screen cap
[06:10] <nickeest> sacarasc, it is, but when it would keep updating the same line, now it is printing new lines
[06:10] <kidx> want a full code for that
[06:11] <kidx> also this updates in lines
[06:11] <nickeest> kidx, what do you mean? I have an mkv video, but my computer is slow, so I am converting it to avi so maybe it will play smooth
[06:11] <kidx> thats why you dedicate a terminal to it
[06:11] <kidx> why
[06:11] <nickeest> oh wait, it went back to just one line again
[06:12] <kidx> mkv opens with VLS
[06:12] <kidx> VLC
[06:12] <kidx> and is not slow
[06:12] <nickeest> kidx, I can open it and play it, but its slow, I have a really old computer
[06:12] <kidx> it depends on your sets
[06:12] <nickeest> its only 2ghz
[06:12] <kidx> me to
[06:12] <kidx> thats why i run LXLE
[06:12] <nickeest> and you can play it. I have debian
[06:12] <nickeest> I am using xfce
[06:12] <kidx> only uing 2 cpu
[06:13] <kidx> 27/30 max on recording
[06:13] <nickeest> and you can play mkv 720p video?
[06:13] <kidx> nw lets try playing a game hehe
[06:13] <kidx> yup
[06:13] <nickeest> I am surprised
[06:13] <kidx> i use terminal
[06:13] <kidx> casue i am on LXLE
[06:13] <kidx> what distro you on
[06:13] <nickeest> ok, I can play it from terminal and try then
[06:13] <nickeest> debian
[06:13] <kidx> yuck
[06:14] <nickeest> kidx, and yourself?
[06:14] <kidx> its good but not that good if you want small and less pc usage use LXLE
[06:14] <kidx> by the way i have 2 2.0ghz
[06:14] <kidx> i have a server pc
[06:14] <nickeest> lxle or lxde?
[06:15] <nickeest> but the video is still on gonna play in one of the two cpu's
[06:15] <kidx> Lubuntu Xtended Life Edition
[06:15] <nickeest> I trust debian guys more than anyone else
[06:15] <kidx> runs greate for me here
[06:15] <kidx> i dont trust them casue lthey lack updates
[06:15] <kidx> I only trust them for a base
[06:16] <nickeest> they are slow, no doubt about that
[06:16] <kidx> as for my main os I would not reconmend them
[06:17] <nickeest> I am running old stable, but I gonna have to upgrade to stable if I want to use google chrome with flash, cause youtube videos are playing really slow in firefox 3.5
[06:17] <nickeest> Once I get youtube vids going fine, I am set, thats all I need, everything else is fine for me
[06:18] <kidx> back to the x11 grab thing
[06:18] <kidx> whay is x11 grab an error
[06:18] <nickeest> How long will it take me to convert this mkv video you'd guess that is 1gig in size to avi
[06:18] <kidx> a while
[06:18] <nickeest> hour?
[06:18] <kidx> almost not worth recording in mkv if your gona jsut convert
[06:19] <kidx> just record in avi
[06:19] <nickeest> its at 130mb now, since I started when you recall I asked for the -ac 2 command
[06:19] <nickeest> I downloaded the vid
[06:19] <kidx> ply it in VLC
[06:19] <nickeest> the avi will be larger or smaller than the original
[06:19] <kidx> no need to convert and openshot can open it as well as kdenlive
[06:19] <nickeest> kidx, I tried that first, it isn't smooth play, it isn't watchable
[06:20] <kidx> casue you prolly lagged when recording it
[06:20] <kidx> witch that is not gona fix your issue but wast time
[06:20] <nickeest> kidx, I downloaded it, I didn't record it myself
[06:20] <kidx> then the vid is junk
[06:20] <kidx> not your pc
[06:21] <kidx> if it plays like junk then its prolly junk
[06:21] <nickeest> but on my old computer which was much faster, these same vids played well
[06:21] <kidx> link me to thsi vid
[06:21] <kidx> i wanna test it
[06:21] <nickeest> kids criten network, elitewarez
[06:21] <kidx> oh
[06:22] <kidx> i dont bother with them
[06:22] <kidx> all i can say is run it on somthing else
[06:22] <kidx> or upgrade
[06:23] <nickeest> yeah, I am gonna wait, and in the meantime find a cheaper version
[06:23] <nickeest> I am also gonna upgrade to wheezy, but I'd they didn't include compiz on wheezy. Do you use compiz?
[06:24] <kidx> no
[06:24] <kidx> i just game
[06:24] <kidx> and make youtube vids
[06:24] <kidx> but seems like all the distros are junk except LXLE
[06:26] <nickeest> you play games on lxle, I though you need windows for games, I thought linux wasn't good for gaming
[06:27] <kidx> it is
[06:27] <kidx> people lie
[06:27] <kidx> steam on linux naitive
[06:27] <kidx> no need for windows
[06:27] <nickeest> steam?
[06:29] <kidx> yes
[06:29] <kidx> i am getting alot of errors
[06:29] <kidx> wow
[06:30] <kidx> ffmpeg -f x11grab -r 25 -s 1280x1024 -i :0.0+100,200 -f alsa -ac 2 -i pulse -vcodec libx264 -crf 0 -preset fast -acodec pcm_s16le output.mkv
[06:30] <kidx> thats my code but no cast is working
[06:30] <kidx> cant grab x11
[06:31] <kidx> also shared memory found x error failed request?
[06:34] <kidx> could it be this --enable-indev=x11_grab_device and if show how do i enable that so i can use x11grab
[06:41] <kidx> well i am outthanks for everything guys
[08:32] <dedeeer_> http://paste.ubuntu.com/5924298/
[08:33] <dedeeer_> whats wrong
[08:38] <JEEB> I can see a report for an older ffmpeg version and GCC 4.8.1
[08:39] <johnny22> whats the best way to syncronize audio and video from a live audio source and from a live video source with the ffmpeg libs ?
[08:39] <JEEB> https://trac.ffmpeg.org/ticket/2756
[08:39] <JEEB> that was fixed by updating ffmpeg
[08:40] <dedeeer_> to me ?
[08:40] <JEEB> yes
[08:41] <dedeeer_> I just complie the xbmc android builds its refer to ffmpeg builds
[08:42] <JEEB> yes, xbmc internally contains some random version of ffmpeg :)
[08:42] <JEEB> I tried looking through dca-related commits but I couldn't really notice anything that would clearly fix your problem http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=dca
[08:43] <JEEB> one way or another that problem has been already fixed, but I have no idea which commit did it :P
[08:43] <dedeeer_> you mean i should to update the ffmpeg modules for xbmc builds
[08:43] <dedeeer_> ou mean i should update the ffmpeg modules for xbmc builds
[08:51] <dedeeer_> does ffmpeg is optimized x264 on NEON ?
[09:05] <JEEB> dedeeer_, I mean that one way or another the ffmpeg you're building should be updated, or you downgrade your toolchain :D I just checked http://git.videolan.org/?p=ffmpeg.git;a=shortlog;h=refs/heads/release/1.2 and I don't see any related fixes in there after the 1.2.0 release. so welp
[09:05] <JEEB> that is, if you are using the 1.2 release branch :P
[09:06] <JEEB> (which is what that error was reported with)
[10:20] <nlight> I want to build ffmpeg for windows using visual studio 2013
[10:20] <nlight> any tips?
[10:20] <JEEB> expect breakage? :D
[10:21] <JEEB> anyways, I don't think all of those features noted at BUILD are yet in, so you will most probably have to wait until RTM for "pure" compilation
[10:21] <JEEB> so I guess follow the usual steps?
[10:21] <nlight> ok, scratch that then
[10:22] <nlight> I'll build it with mingw
[10:22] <JEEB> :)
[10:22] <JEEB> yes, c99wrap is not exactly fast
[10:22] <JEEB> anyways, this is the usual way http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b-or-Intel-C_002…
[10:23] <JEEB> this way you can get MSVC debug symbols and all, but unfortunately c99wrap is needed
[10:23] <nlight> is it worth to do this if I can use mingw?
[10:24] <JEEB> if mingw isn't any worse for you use case (see: debug symbols not needed to be usable from MSVS), then mingw-w64 generally is less of a hassle
[10:25] <nlight> is this guide up to date - https://trac.ffmpeg.org/wiki/MingwCompilationGuide ?
[10:25] <xlinkz0> why would you compile it yourself for windows?
[10:26] <JEEB> lol, that compilation guide really tries to add everything, and only notes mingw (the original project), while nowadays mingw-w64 is definitely the better choice
[10:26] <JEEB> xlinkz0, why would you not?
[10:27] <xlinkz0> because zeranoe
[10:27] <JEEB> because zeranoe only gives GPLv3 compiles?
[10:27] <JEEB> there are plenty of reasons for various people to compile their own thing
[10:27] <JEEB> and nothing bad or weird in that
[10:27] <xlinkz0> just asking
[10:28] <JEEB> also I guess fdk-aac is one of the main reasons to build yourself otherwise
[10:28] <JEEB> since it can't be distro'd
[10:28] <JEEB> (in binary form)
[10:28] <xlinkz0> didn't say it was bad, i genuinely was curious why would someone do that :)
[10:28] <JEEB> or a limited build for a specific purpose? there's plenty of reasons
[11:07] <Pet0r> Is there any way to force ffmpeg to obey a maximum bitrate even when -crf is set?
[11:08] <Pet0r> For x264 I am using crf 18 which is proving great for 99% of videos, but on some videos with a high input bitrate (10MBit/s+) it's overshooting on bitrate hugely and I would prefer it to cap out at the bitrate rather than try to maintain crf 18
[11:08] <JEEB> If you mean the equivalent of -b:v, no. You can limit bandwidth within a buffer with -bufsize and -maxrate, though.
[11:10] <Pet0r> I currently have -b:v 3M and -maxrate 3M, and it overshoots to ~7.5MBit/s bitrate
[11:10] <Pet0r> on average
[11:10] <JEEB> 1) you didn't set bufsize 2) -b:v is over the whole clip
[11:11] <JEEB> also if you want one-pass encoding -crf + VBV (bufsize + maxrate) is IMHO better than setting a bit rate and VBV
[11:12] <Pet0r> am I right in thinking bufsize is what it buffers in order to make a better decision about the upcoming video?
[11:12] <Pet0r> or am I totally wrong there
[11:13] <Pet0r> not sure what it does
[11:13] <JEEB> bufsize is the amount of buffer within which the maxrate is calculated
[11:13] <JEEB> in theory if you are not streaming over limited bandwidth and just want to limit the overall file size you could set this rather large
[11:13] <Pet0r> I need to assume bandwidth is limited
[11:13] <JEEB> well then you should always be using VBV :P
[11:13] <Pet0r> The overall file size cannot exceed duration * 3mbps
[11:13] <JEEB> pronto
[11:14] <JEEB> eh
[11:14] Action: JEEB sighs
[11:15] <JEEB> anyways, it's the amount of buffer within which the maxrate is calculated and kept within limits (and if x264 cannot do that it will scream at you loudly, although that really doesn't happen much nowadays with all the emergency modes etc. in place)
[11:15] <JEEB> aka the amount which the player has to buffer when starting the playback to make sure that it never has to buffer again as long as it has maxrate amount of bandwidth
[11:15] <Pet0r> So is there a good way to calculate a decent value for that?
[11:15] <Pet0r> ah
[11:15] <JEEB> a decent value is something you decide, and then match up on the encoder and the player :P
[11:16] <Pet0r> So is that value in bit/s or byte/s?
[11:16] <JEEB> ffmpeg's is bits/s
[11:16] <JEEB> and bits
[11:16] <Pet0r> oh
[11:16] <Pet0r> *ok
[11:17] <JEEB> it will then divide by eight when passing it to the x264 API
[11:17] <JEEB> ah, no
[11:17] <JEEB> it will divide by 1000
[11:17] <JEEB> because x264 takes kilobits
[11:17] <Pet0r> thing is, will this negatively affect other videos which do not require the whole bitrate?
[11:17] <Pet0r> because right now
[11:17] <Pet0r> if I encode a video way under 3mbps with these settings
[11:18] <Pet0r> the bitrate will come out below 3mbps
[11:18] <Pet0r> which is what I want
[11:18] <JEEB> it will affect all videos that you encode, to make sure that nothing goes over maxrate calculated over a bufsize buffer
[11:18] <JEEB> it will not make anything use more bits
[11:19] <JEEB> also if you have encoded without maxrate + bufsize before you have pretty much failed as a VOD provider unless you expected your users to buffer a lot :P
[11:19] <Pet0r> So I guess my question is then, what is the point of bufsize? If I set a max bitrate, why isn't that just obeyed over the whole video? and if I do set a bufsize, why does that make any difference since isn't it always limited to that?
[11:20] <Pet0r> I feel like I am fundamentally misunderstanding bufsize and I do want to understand it :P
[11:20] <JEEB> normal rate control controls the bit rate over the whole clip, bufsize controls rate control within a constantly moving buffer so that at all points if you have buffered bufsize amount of video and have at least maxrate amount of bandwidth, you will never have to buffer again
[11:21] <Pet0r> okay, I see
[11:21] <Pet0r> so what you're saying is, in the video player, you tell it to download a certain amount of the video (bufsize), as opposed to a duration of the video, before it starts playing
[11:21] <Pet0r> at which point as long as the bandwidth remains available, the video should always play
[11:22] <JEEB> I don't understand the "as opposed to a duration of the video" part, but yes -- it will make sure that if those parameters are true, then the viewer will have a smooth playback experience
[11:23] <JEEB> you just match up the player with the encoder with regards to how much bufsize you used, and encode accordingly
[11:24] <JEEB> in theory if you encoded with just bit rate rate control, if the beginning of a clip is very very bandwidth needing, you could end up exhausting all of your bits there (esp. if the stuff after that needs only nominal bit rate)
[11:24] <JEEB> thus if you are giving out video over a limited bandwidth, you *need* VBV
[11:25] <JEEB> the beginning in that case will look worse, but it will be within the VBV limits
[11:25] <Pet0r> I see, so would setting bufsize quite low force the video to always obey maxrate as well?
[11:25] <Pet0r> like I don't know, if you set it to something stupidly low like 100k
[11:25] <JEEB> huh?
[11:25] <Pet0r> what would happen?
[11:25] <JEEB> it would just make it very hard for the encoder to fluctuate anything
[11:26] <JEEB> it would be a very small buffer in that case
[11:26] <JEEB> usually such are only used in low-latency things where you can't buffer much
[11:27] <JEEB> in your case I would at the very least recommend having a maxrate amount of bufsize, possibly more
[11:27] <Pet0r> well, the problem for me is not just the buffering, the reason I was asking originally about the maxrate etc. is that I have a file size constraint
[11:27] <Pet0r> now I know 2 pass is usually best for that
[11:27] <JEEB> well, maxrate + bufsize is even more limiting than that :P
[11:28] <Pet0r> but the problem with 2 pass is that it seems to do the opposite of what it does now, it perfectly makes this 10mbps video 3mbps for the duration
[11:28] <Pet0r> but it then upscales smaller videos to a higher bitrate than necessary
[11:28] <Pet0r> well, I use "upscales" wrongly, it doesn't upscale
[11:28] <Pet0r> but you know what I mean
[11:28] <JEEB> yes, because you've asked for X, you got X
[11:28] <Pet0r> yeah, is there a way to say, like
[11:28] <Pet0r> don't exceed the input bitrate
[11:29] <Pet0r> short of looking that up in advance and passing it in
[11:29] <JEEB> the input bit rate is irrelevant as soon as you start re-encoding
[11:29] <JEEB> forget about the input bit rate, basically
[11:29] <JEEB> if you have a limitation on the output, that is a limitation on the output
[11:30] <Pet0r> a lot of the videos we get are passed in containers which have crappy codecs, like FLV
[11:30] <Pet0r> so we can usually rely on being able to produce a smaller output with comparable quality
[11:30] <JEEB> anyways, maxrate + bufsize will keep the average bitrate under something because it even has a smaller buffer within which it limits the clip :P
[11:30] <Pet0r> so input bitrate is sometimes useful
[11:30] <Pet0r> ok
[11:30] <Pet0r> so a maxrate of 3M with a buffer size of say... 1M?
[11:30] <JEEB> ugh
[11:31] <JEEB> use AT LEAST maxrate amount of bufsize
[11:31] <JEEB> I mean, you would be OK with a bufsize of the whole damn clip, if you were using -b:v without VBV before, which is of course completely incorrect for any kind of limited bandwidth video provision
[11:32] <Pet0r> well, I didn't write any of the previous stuff, I am trying to improve it
[11:33] <JEEB> I would say even 2-3 "seconds" worth of buffer would be usable
[11:33] <JEEB> if not even more
[11:33] <JEEB> maxrate * 2 or 3 that is
[11:33] <JEEB> then set the player to buffer the same amount, and enjoy
[11:35] <Pet0r> what about -bt, does that have any effect as well?
[11:35] <Pet0r> or not worth using?
[11:35] <JEEB> what is bt in ffmpeg again? I'm more knowledge'able in x264 settings than ffmpeg's
[11:35] <Pet0r> bitrate threshold, apparently
[11:35] <JEEB> keep away from all that stuff
[11:35] <JEEB> x264's defaults are GOOD
[11:36] <JEEB> all you should in general be touching are: preset (defaults, speed vs compression), some kind of rate control mode (bitrate or CRF), VBV (maxrate and bufsize)
[11:37] <Pet0r> okay so with a bufsize set, the output video bitrate (average) is much lower
[11:37] <Pet0r> but it's actually way under 3mbps even
[11:37] <Pet0r> (I set bufsize = maxrate = 3M)
[11:38] <Pet0r> average bitrate with those settings is 1600kbps
[11:38] <JEEB> if the crf is so that it always ends up using the maximum amount of buffer then it will be pretty much 3M in the end, otherwise it will just be smaller
[11:38] <JEEB> because VBV only limits
[11:39] <JEEB> and the larger the bufsize the more playing ground the encoder will have
[11:39] <Pet0r> one sec
[11:40] <Pet0r> so with identical settings (crf 18, b:v 3M, maxrate 3M)... bufsize: 3M = 1600kbps avg, no bufsize = 3750kbps
[11:41] <JEEB> because maxrate does nothing without a bufsize
[11:41] <JEEB> maxrate is only applied with a bufsize
[11:42] <Pet0r> but why is bufsize 3M producing such a low bitrate encode? surely if you can buffer 3M the average bitrate could be much higher than 1600K
[11:43] <JEEB> basically some part would have taken more bit rate, but it was limited, and CRF didn't need as much rate in other places so it kept the average down
[11:44] <JEEB> and if you keep making bufsize bigger the encoder will be able to fluctuate more
[11:44] <JEEB> thus getting closer and closer to that unlimited result
[11:45] <Pet0r> presumably it's not a good idea to have a static bufsize for every input video though?
[11:45] <JEEB> why not?
[11:45] <Pet0r> ok
[11:46] <JEEB> you have the certain maximum bit rate, and a bufsize for it
[11:46] <Pet0r> well, I guess, yeah, it would be alright
[11:46] <Pet0r> I was thinking if you had a really low quality input video that ended up with a low bitrate output, it would have to buffer a lot more of the video
[11:46] <Pet0r> and of course, it would
[11:46] <Pet0r> but it would take the same time to do so
[11:46] <JEEB> that's why I kept telling you that at the very least you should use at least a 1sec bufsize
[11:46] <Pet0r> in the player, I mean
[11:47] <JEEB> not related to your last lines
[11:47] <JEEB> but you basically have a case where you can have the user wait a couple of seconds at the beginning
[11:47] <JEEB> and then just keep to the maxrate under that buffer
[11:50] <Pet0r> okay
[11:51] <Pet0r> that's been very helpful, thanks
[11:51] <Pet0r> I now understand what those settings mean and how they work
[11:51] <JEEB> I'm still surprised that people who actually get paid know less of this than I do :P
[11:51] <Pet0r> I didn't get hired specifically to do this :P
[11:52] <Pet0r> I'm a software engineer
[11:52] <Pet0r> lol
[11:53] <JEEB> and I'm a random coder
[11:54] <JEEB> also, just so that there are no misunderstandings, <JEEB> and then just keep to the maxrate under that buffer <- this doesn't mean that the maxrate always has to be under the bufsize (that is needed in low-latency cases over limited bandwidth), it was written as "within the buffer"
[11:54] <JEEB> *that
[11:56] <Pet0r> I see
[11:59] <Keshl> Before I make a dumb mistake while we're on the topic of bufsize.. I have a video that has a lot of repeating frames over the course of 5-10 seconds, and it's running at 120 FPS. If I increase the bufsize to some rediciously large amount will it actually pick up those repeating frames and offer better compression than it does now, oÉo?
[12:00] <JEEB> bufsize is not that kind of buffer
[12:00] <JEEB> that's a job for rc-lookahead and amount of refs/bframes
[12:00] <Keshl> Okay, good thing I asked first ^É^; What should I be doing them, oÉo?
[12:00] <Keshl> But rc-lookahead is limited to 250.
[12:00] <JEEB> yes
[12:00] <Keshl> At 120 frames per second, 250's.. Very inadiquite. XD
[12:01] <JEEB> well, it should capture enough pictures being the same, just keep keyframe interval long
[12:01] <Keshl> How do I do that exactly, oÉo? I'm not terribly familier with ffmpeg's magical commands. xwx
[12:02] <JEEB> -g?
[12:03] <Keshl> I have this feeling that I won't get it right. o_o Does -g 1200 sound reasonable if I want it to have around 10 seconds of lookahead/behind?
[12:04] <JEEB> that will mean that the maximum interval between keyframes that also cut the pictures future pictures can reference off of will be 1200
[12:04] <JEEB> for 120fps content that would be 10 seconds, yes
[12:04] <Keshl> Awesome, oÉo.
[12:07] <Keshl> o Éo Is -g all I need?
[12:08] <JEEB> and naturally a preset that lets you use enough refs/bframes
[12:08] <Keshl> Like medium? Or OH.
[12:08] <Keshl> ... -Facedesk-
[12:08] <Keshl> Kay, yeah, now I see my mistake, thanks.
[12:10] <Pet0r> ok yeah that seems to be working well, even with a bufrate = maxrate, the average bitrate is half maxrate, but I see it spiking as high as 133% of the maxrate during actual playback
[12:15] <Keshl> JEEB: How do I do -no-scenecut of -keyint? http://mewiki.project357.com/wiki/X264_Settings seems outdated, and the video in question has no scenecuts.
[12:16] <Keshl> *or
[12:16] <JEEB> -g = keyint
[12:16] <Keshl> Ohhhh.. oÉo..
[12:17] <Keshl> .... And how do I set -bframes and -b-adapt? x.x
[12:24] <microchip_> Keshl: https://sites.google.com/site/linuxencoding/x264-ffmpeg-mapping
[12:25] <Keshl> Dankye, oÉo
[12:30] <Keshl> ... Motion estimation depends on color, doesn't it? So if I only have two colors, black (the background) and orange (And I don't mean different shades, I mean two colors, /period/), motion estimation becomes next to impossible and results in inordinately large files? xwx
[12:35] <ubitux> BBB: i really don't know what is the timeline and workload required for each thing; as long as i learn about video codecs i'm happy with anything
[12:49] <saste> ubitux, wrong ML
[12:49] <saste> bah wrong channel...
[12:50] <ubitux> oups
[12:51] <ubitux> thx
[13:01] <nlight> I'm trying to compile using mingw and I get the following error during compilation "libavutil/atomic.c:101:2: error: #error "Threading is enabled, but there is no implementation of atomic operations available"
[13:02] <nlight> any ideas?
[13:02] <nlight> I have explicitly passed --enable-w32threads
[13:03] <nlight> here is my whole configure statement $ ./configure --prefix=/mingw --enable-gpl --enable-nonfree --enable-postproc --enable-avfilter --enable-w32threads --enable-runtime-cpudetect --enable-memali
[13:03] <nlight> gn-hack --enable-zlib --enable-libx264 --enable-libxvid --enable-static --disab
[13:03] <nlight> le-shared --disable-debug
[13:03] <nlight> sorry for the spam :|
[13:04] <spaam> nlight: :(
[13:05] <Pet0r> http://trac.ffmpeg.org/ticket/2363
[13:05] <Pet0r> did you try that?
[13:05] <nlight> I could do that, but I wanted native threads, well..
[13:06] <Pet0r> you can have native threads
[13:06] <Pet0r> if you specify a --cpu val
[13:07] <nlight> it compiles with --enable-pthreads, that's ok for now I guess
[13:07] <nlight> thanks
[13:07] <Pet0r> look at comment 3
[13:07] <Pet0r> if you want native threads
[13:08] <JEEB> uhh
[13:08] <JEEB> native threads should work out-of-box
[13:09] <nlight> I guess I should switch to mingw64
[13:09] <JEEB> also disable-shared is the default
[13:09] <JEEB> yes, yes you should
[13:09] <JEEB> http://files.1f0.de/mingw/
[13:09] <JEEB> grab one of those and switch your /mingw
[13:09] <nlight> will do, thanks
[13:11] <JEEB> also runtime-cpudetect is default now too
[13:11] <JEEB> memalign hack is only needed for mingw32
[13:11] <JEEB> not mingw-w64
[13:12] <JEEB> zlib is enabled by default if found
[13:12] <JEEB> (is visible in configure's output)
[13:12] <JEEB> also you haven't enabled anything nonfree
[13:13] <JEEB> generally I'd say you only need possibly prefix, enable-gpl, enable-libx264, enable libxvid
[13:13] <nlight> ok
[13:13] <JEEB> also as a hint for msys
[13:13] <nlight> it's the first time I'm building ffmpeg so any tips are super welcome
[13:13] <JEEB> you can switch between mingws with etc/fstab
[13:13] <JEEB> first tip: start with pure ./configure first
[13:13] <JEEB> then look if it passes
[13:14] <JEEB> after that enable those external things you need, and possibly some internal ones
[13:14] <JEEB> enable-gpl is pretty much the only thing that enables something inside ffmpeg itself
[13:14] <JEEB> everything else is just for third party libraries
[13:15] <nlight> aha, so --enable-nonfree does nothing by itself?
[13:15] <JEEB> yes
[13:18] <JEEB> also, funny enough pretty much the only thing that prefers pthreads to native threads for whatever reason in its configure is x264
[13:18] <JEEB> and I think that one is a mistake
[13:18] <Fjorgynn> I am
[13:19] <JEEB> basically x264's configure first tries native threads, and then if it finds pthreads and they are usable it will switch to that
[13:19] <JEEB> so there you will have to specify the native threads
[13:19] <JEEB> ffmpeg and friends default to native threads
[13:22] <nlight> isn't there some python script that automatically downloads and builds everything?
[13:22] <nlight> like ffmpeg + xvid + x264 + aac + lame
[13:22] <nlight> or something
[13:22] <JEEB> you could create one rather easily, or a shell script
[13:22] <nlight> that's what I'm currently doing
[13:23] <nlight> I wish one was already available but meh
[13:27] <Pet0r> there are windows builds released very often @ zeranoe.com
[13:27] <nlight> do they include xvid + x264?
[13:27] <Pet0r> yes
[13:27] <Pet0r> http://ffmpeg.zeranoe.com/builds/
[13:28] <nlight> oh, figures
[13:28] <nlight> I should just use those then
[13:28] <nlight> thanks
[13:30] <Fjorgynn> :)
[13:32] <nlight> I'm still setting up my own ffmpeg build but these will do the trick for testing
[13:33] <Pet0r> fair enough :P
[13:33] <nlight> is there a 'static library' version?
[13:34] <nlight> i'm not sure if it's good practice to link ffmpeg statically
[13:36] <JEEB> if you are following the license there's no real problem with static linking. it's a bit funky with mingw-w64 tho :)
[13:37] <elkng> I use command: "ffmpeg ../1.avi -c:v png "%3d.png"" to make video into set of png images, but they are too dark, when I watch video I can increase bringtness with some key, but how to make those images brighter while converting using that command ?
[13:37] <JEEB> (although it's only funky when you are linking the static lib from MSVC, if you are using mingw64 all-around then there's no problem)
[13:38] <nlight> JEEB, funky in what way?
[13:38] <nlight> as in "don't do this unless you're damn sure you need it"?
[13:39] <JEEB> when you link a static mingw-w64 lib from MSVC there are some functions that aren't found, and thus you need to define that one or those two somewhere as a workaround in your MSVS code
[13:39] <elkng> should I use "-vf mp=eq2=1.0:2:0.5" ?
[13:39] <Pet0r> why not give it a try?
[13:39] <Pet0r> :P
[13:39] <JEEB> officially it's not really supposed to be used, but it's found to be rather usable because people have been doing it for years :P
[13:39] <Pet0r> just limit your video to a time when you know it's dark
[13:39] <JEEB> dll is preferable of course
[13:39] <Pet0r> and see how they come out
[13:40] <JEEB> since dll is less :effort:
[13:47] <elkng> Pet0r: I mean there is also lutrgb='r=1.1*val:g=1.1*val:b=1.1*val' or lutyuv=y=val*1.1
[13:47] <Pet0r> there are multiple ways to do it with different filters
[13:47] <Pet0r> give it a try and just use a slice of your video where you know it's dark
[13:47] <Pet0r> like if you know it's dark in 30-35 seconds then use -ss 00:30:35 -t 00:00:05
[13:47] <Pet0r> so just cut that section of video
[13:48] <Pet0r> and see if they come out dark or not
[13:48] <Pet0r> * -ss 00:00:30
[13:48] <Pet0r> lol
[17:07] <hackeron> hey, any ideas what is wrong with this command? < "ffmpeg -rtsp_transport tcp -i 'rtsp://192.168.55.3/media/video1' -analyzeduration 0 -map 0 -dn -codec:v copy -f alsa -ac 2 -i hw:0,0 -codec:a libfdk_aac -flags +qscale -global_quality 1 -afterburner 1 -reset_timestamps 1 -ar 44100 -y video.mkv" -- I'm getting Option map (set input stream mapping) cannot be applied to input file hw:0,0
[17:09] <hackeron> durandal_1707: http://pastie.org/8186961
[17:10] <sacarasc> hackeron: You have input and output options mixed up.
[17:11] <hackeron> sacarasc: do I move the -f alsa -i hw:0,0 straight after the -i 'rtsp://..."?
[17:11] <sacarasc> Yes.
[17:12] <hackeron> sacarasc: then I get this: http://pastie.org/8186968 - no audio in output file :(
[17:23] <Fjorgynn> I disagree
[19:56] <t4nk795> hi, I have been trying to find the correct command to overlay two png images and keep the alpha channel, but the second image always loses it and has a black background instead, can anyone help me? This is the command line I've been using ffmpeg.exe -f image2 -i image000.png -vf "[in] scale=iw:ih, pad=2*iw:ih [left]; movie=image000.png, scale=iw:ih [right]; [left][right] overlay=main_w/2:0 [out]" -vcodec png Output.png
[20:03] <relaxed> t4nk795: first make sure you're using a recent version of ffmpeg, then look at -filter_complex
[20:08] <t4nk795> thanks!
[21:50] <griddy> Hi guys, i get the following error when sending a video stream fom ffmpeg to ffserver: "av_interleaved_write_frame(): Connection reset by peer". Any tips?
[21:51] <griddy> *from
[21:52] <Fjorgynn> Sit Now :)
[21:52] <griddy> ok sitting:P
[21:55] <griddy> Hi guys, i get the following error when sending a video stream from ffmpeg to ffserver: "av_interleaved_write_frame(): Connection reset by peer". Any tips?
[23:42] <alexschomb> hello there :)
[23:43] <alexschomb> I'm trying to compile FFMPEG via cross-compiling for Raspbian following the wiki entry: http://trac.ffmpeg.org/wiki/How%20to%20compile%20FFmpeg%20for%20Raspberry%2…
[23:44] <alexschomb> In the last step FFMPEG is not being compiled because "libx264 not found". I suppose this happens because there is no libx264.so created in /my/path/were/i/keep/built/arm/stuff/lib/
[23:45] <alexschomb> (I change the path to my own one of course)
[23:46] <alexschomb> I'd love some advice, already tried building with --enable-static, not helping.
[23:50] <llogan> alexschomb: try adding --extra-libs=-ldl to ffmpeg configure
[23:50] <llogan> those non CamelCase urls are annoying
[23:51] <alexschomb> llogan: thanks that helped! :)
[23:51] <llogan> i'll update that page then
[23:52] <alexschomb> it configured although i got an potential error about pkg-config not found
[23:53] <alexschomb> thank you very much :)
[23:54] <llogan> alexschomb: what's the exact error? (if it is long use pastebin.com)
[23:54] <alexschomb> i guess i found another small error. you'd need to make && make install in the step "Compiling ALSA library" too
[23:55] <llogan> ok, thanks.
[23:56] <alexschomb> llogan: http://pastebin.com/HdjzWV8w/
[23:59] <llogan> ah, i see it already mentions PKG_CONFIG_PATH, but it could probably be moved.
[00:00] --- Tue Jul 30 2013
1
0
[00:39] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/1.2:930d035e72ca: update for 1.2.2
[01:25] <KGB> 01[13FFV101] 15michaelni pushed 2 new commits to 06master: 02 http://git.io/JUTNUg
[01:25] <KGB> 13FFV1/06master 146eeb731 15Michael Niedermayer: ffv1/3.1: Fix punctuation to make the text understandable
[01:25] <KGB> 13FFV1/06master 14924e9d3 15Michael Niedermayer: ffv1/3.1: The border is about slices not pictures
[03:06] <cone-656> ffmpeg.git 03Marton Balint 07fatal: ambiguous argument 'refs/tags/n1.2.2': unknown revision or path not in the working tree.
[03:06] <cone-656> Use '--' to separate paths from revisions
[03:06] <cone-656> refs/tags/n1.2.2:HEAD: lavc: do not override format if neither text nor bitmap codec prop is set
[03:35] <cehoyos> mateo: Regarding vaapi with ffplay, I don't think SDL supports any hardware acceleration, I would have to get OpenGL support to ffplay first (which would also allow VDPAU support)
[04:10] <cone-656> ffmpeg.git 03Nicolas Bertrand 07release/2.0:e0d88cfd1818: jpeg2000: Initialize only once mqc arrays
[04:10] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:fa6b6dad3d78: jpeg2000: fix overflow in dequantization
[04:10] <cone-656> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:09b33f9a8251: Fix pix_fmt detection in the native jpeg2000 decoder.
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:9da9b36435c9: jpeg2000dec: parse CDEF
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:8b221d60fa5e: jpeg2000dec: Support non subsampled 8bit planar pixel formats
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:9a6d3eee5995: jpeg2000dec: silence unused variable warning
[04:11] <cone-656> ffmpeg.git 03Nicolas Bertrand 07release/2.0:e0d88cfd1818: jpeg2000: Initialize only once mqc arrays
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:d73ce6cb5609: jpeg2000dec: Support non subsampled 9-16bit planar pixel formats
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:fa6b6dad3d78: jpeg2000: fix overflow in dequantization
[04:11] <cone-656> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:09b33f9a8251: Fix pix_fmt detection in the native jpeg2000 decoder.
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:9da9b36435c9: jpeg2000dec: parse CDEF
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:8b221d60fa5e: jpeg2000dec: Support non subsampled 8bit planar pixel formats
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:9a6d3eee5995: jpeg2000dec: silence unused variable warning
[04:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07release/2.0:d73ce6cb5609: jpeg2000dec: Support non subsampled 9-16bit planar pixel formats
[11:34] <cone-858> ffmpeg.git 03Diego Biurrun 07master:966689442ed8: buffersink: K&R formatting cosmetics
[11:34] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:edab63a4067a: Merge commit '966689442ed843019dc0722a49bfb0ac51755d19'
[11:53] <cone-858> ffmpeg.git 03Diego Biurrun 07master:16c22122c788: h264: K&R formatting cosmetics
[11:53] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:102397d2c14d: Merge commit '16c22122c788b5e54a2f2e224bd0106429f0714c'
[11:58] <cone-858> ffmpeg.git 03Diego Biurrun 07master:37063714c0a0: build: Only check FATE dependencies when running FATE tests
[11:59] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:1ecf380223b0: Merge commit '37063714c0a064808f9671ec4d376955d664f463'
[13:12] <cone-858> ffmpeg.git 03Luca Barbato 07master:10aa44aa675e: avidec: K&R formatting cosmetics
[13:12] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:d78122099113: Merge commit '10aa44aa675e05067845e3e55fac37642cbbdae4'
[13:23] <cone-858> ffmpeg.git 03Luca Barbato 07master:c8f0b20b4a6b: avidec: Let the inner dv demuxer take care of discarding
[13:23] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:3afcddcff2bc: Merge commit 'c8f0b20b4a6bb6691928789d83e4b02896969848'
[13:29] <mateo`> hi guys
[13:30] <mateo`> in which case s->oformat can be NULL in the mov muxer ? (lines 3541, 3550)
[13:34] <cone-858> ffmpeg.git 03Diego Biurrun 07master:6ff15cd569e1: Remove unreachable returns
[13:34] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:2655c1ac1282: Merge commit '6ff15cd569e1345bc3612fb69ad3003b104fe50d'
[13:43] <cone-858> ffmpeg.git 03Diego Biurrun 07master:5b097399baa2: eval: Explicitly ignore return value of strtod() in parse_db()
[13:43] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:8ff23f0e3c9a: Merge commit '5b097399baa2e38cc513939cfab3a9b6fdbc33df'
[13:51] <cone-858> ffmpeg.git 03Diego Biurrun 07master:bf4b0ed1d5d3: Add missing deprecation attributes
[13:51] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:b0edd5999404: Merge commit 'bf4b0ed1d5d323050a87c9f0ad1dd40860eb3da2'
[14:04] <mateo`> s->oformat seems not to be check in matroskaenc
[14:21] <cone-858> ffmpeg.git 03Diego Biurrun 07master:270d7e3a1802: doc: cosmetics: Consistently format list and table items
[14:21] <cone-858> ffmpeg.git 03Michael Niedermayer 07master:f0308af5fac9: Merge remote-tracking branch 'qatar/master'
[14:35] <michaelni> mateo`, maybe the oformat checks are unneeded
[14:36] <mateo`> it seems like, i will send a patch
[15:51] <saste> any XML expert around?
[16:27] <BBB> ubitux: Daemon404: let's talk logistics, when do you guys want to start? next week monday (i.e. in 8 days)?
[16:28] <BBB> ubitux: Daemon404: then second, I've written a skeleton (keyframe bitstream parsing and some minor parts of keyframe reconstruction), shall I upload that to my github so we can work on such a branch together?
[16:40] <cone-858> ffmpeg.git 03Stefano Sabatini 07master:18df69d238fb: doc/ffprobe.xsd: specify tag elements in stream after disposition element
[16:40] <cone-858> ffmpeg.git 03Stefano Sabatini 07master:2fcd400669bc: ffprobe: do not treat array elements as nested in the compact writer
[16:40] <cone-858> ffmpeg.git 03Florent Tribouilloy 07master:2186a7e547fd: ffprobe: add -show_programs option
[16:47] <Daemon404> BBB, that depends when i can mov e into my flat in london
[16:47] <Daemon404> which i should find out tomorrow at latest
[17:05] <BBB> Daemon404: ok I guess.. any interest in me pushing an early nonworking branch to github?
[17:06] <Daemon404> BBB, actually im technically unemployed for 2 weeks or so in aug (while i transfer)
[17:06] <Daemon404> so i should have time
[17:06] <Daemon404> soon... sh
[17:06] <BBB> lol
[17:06] <Daemon404> ish*
[17:06] <Daemon404> i also mostly finished The H.264 Book
[17:06] <Daemon404> thought i got lost in the CABAC chapter
[17:07] <Daemon404> though*
[17:07] <BBB> that's ok, cabac is already implemented
[17:07] <BBB> so is vp8/9/5/6's rac coder
[17:07] <Daemon404> entropy coding didnt change?
[17:07] <BBB> not the base part
[17:07] <BBB> the use in the codec did, of course
[17:07] <BBB> and it added adaptivity at a higher layer
[17:07] <BBB> but the lowlevel part is identical
[17:07] <Daemon404> k
[17:12] <cone-858> ffmpeg.git 03Matthieu Bouron 07master:ee4ef139e361: lavf/movenc: remove useless checks on AVOutputFormat
[17:12] <cone-858> ffmpeg.git 03Matthieu Bouron 07master:8a0919554502: lavf/movenc: improve psp check
[17:25] <kurosu_> BBB / Daemon404: what codec are you discussing about ?
[17:26] <JEEB> vp9
[17:26] <BBB> kurosu_: we want to write a native ffmpeg decoder in august
[17:27] <kurosu_> ah ah ok, you want to beat hevc on another topic then :)
[19:27] <cone-858> ffmpeg.git 03Carl Eugen Hoyos 07master:ae4dc0b37aa8: Set bits_per_raw_sample when decoding mjpeg / ljpeg.
[19:27] <cone-858> ffmpeg.git 03Carl Eugen Hoyos 07master:7e8e8ba9cc2b: Assume gray8 if 1 < maxval <= 255 in pgm.
[19:27] <cone-858> ffmpeg.git 03Carl Eugen Hoyos 07master:7651c0e49b1f: Avoid overflows when reading pgm files with maxval != 255 and != 65535.
[20:47] <cehoyos> It is possible that both 7651c0e and a0348d0 are wong
[20:47] <cehoyos> I am not sure how to test
[20:50] <durandal_1707> uhh 7651c0e looks like workaround for bug in libjpeg
[20:50] <cehoyos> "The most significant byte is first." - does that imply big endian?
[20:52] <cehoyos> See http://netpbm.sourceforge.net/doc/pgm.html "9."
[20:54] <durandal_1707> cehoyos: pnm code have big-endian variants only
[20:54] <durandal_1707> so more 1 byte its in big-endian
[20:55] <cehoyos> Which on of these is big-endian (assuming 12 bit): "02 d0" or "d0 02"?
[21:02] <durandal_1707> big endian system reads bytes from left to right
[21:22] <durandal_1707> did we get bankrupt?
[22:00] <cone-858> ffmpeg.git 03James Almer 07master:ac8e70d7358e: oggenc: Write stream metadata if available
[22:11] <cehoyos> durandal_1707: a0348d0 definitely breaks some pgm's on little endian, do you want to fix it? I will then revert 7651c0e if you don't do it.
[22:12] <cehoyos> av_le2ne16() cannot be correct on both le and be if the pix_fmt is gray16be in both cases
[22:12] <cehoyos> av_be2ne16() cannot be correct on both le and be if the pix_fmt is gray16be in both cases
[22:15] <durandal_1707> cehoyos: what pgms?
[22:16] <durandal_1707> your "patch" that get commited did not fix anything, it just introduced regression, because now ffmpeg decoder incorrectly some pnm files
[22:24] <alsu> I have an rtmp stream which works with flvstreamer but not ffmpeg. should I file a bug with ffmpeg?
[22:24] <cehoyos> The patch was not correct but It did not introduce a regression, the regression was there since a0348d0
[22:25] <alsu> happy to try and debug it too, if I can be of use
[22:25] <cehoyos> There are no pgm's that were decoded correctly on le since a0348d0, current FFmpeg decodes libjpeg's broken pgm's correctly.
[22:25] <cehoyos> No pgm
[22:25] <cehoyos> No pgm's with 255 < maxval < 65535
[22:25] <durandal_1707> nonsense
[22:26] <durandal_1707> as i already said ffmpeg does correctly decodes netpbm images with maxval != 255 & maxval !=65535
[22:30] <cehoyos> Yes, on be; on le, it is not correct. Note that av_be2ne16() cannot be correct on both be and le for the same colourspace
[22:31] <durandal11707> and where is av_be2ne called?
[22:33] <cehoyos> grep av_be2ne libavcodec/pnmdec.c ? (Note that the call is visible in my patch that you don't like.)
[22:34] <durandal11707> cehoyos: your patch is wrong
[22:34] <cehoyos> YES
[22:35] <durandal11707> and code in pnmdec is worse than it was before
[22:35] <cehoyos> Please fix the regression you introduced, you can either revert my patch, or I will do it when I backport your fix.
[22:35] <durandal11707> and upgrade =2 silly obscure crap is wrong too
[22:35] <cehoyos> I don't disagree, but before a0348d0 the "crap" worked, it is broken since
[22:35] <durandal11707> cehoyos: your claim that i break pgm decoder for maxval != 65535
[22:36] <cehoyos> No for 255 < maxval < 65535
[22:36] <durandal11707> i'm telling you that i broke nothing that was not already broken
[22:37] <durandal11707> cehoyos: there is ticket on trac that explains why maxvals: 255<maxval<65535 & 0<maxval<255 are not decoded correctly
[22:39] <cehoyos> Wow, that's cool: You first broke decoding, then you reported it and nobody tested it for a regression;-)
[22:40] <cehoyos> otoh: This ticket is about ppm, I was only talking about pgm, so I don't think these are related.
[22:41] <durandal11707> not at all, decoding was never correct
[22:41] <cehoyos> To shorten this: Do you want to fix the regression or not?
[22:41] <cehoyos> It worked fine before (and it still works fine with avconv)
[22:41] <durandal11707> also calling av_be2ne16 for each pixel is smart too
[22:43] <durandal11707> cehoyos: do whatever you want, but be aware I may rewrite anything stupid i find
[22:47] <cehoyos> That is exactly why I ask you: Do you want to fix the regression you introduced?
[22:47] <cehoyos> (And additionally: Do you want to send patches for review?)
[22:51] <durandal_1707> cehoyos: i'm against revert, as setting pix_fmt and than changing it is bad practice
[23:00] <durandal_1707> nice, i get enormous spam when building
[23:01] <durandal_1707> bunch of input/output_count
[23:07] <cehoyos> durandal_1707: I am not sure I understand: Do you mean we should not revert my patch? Or we should leave the regression because fixing bugs is "bad practice"? (Sorry, I really don't understand.)
[23:08] <durandal_1707> cehoyos: i was talking about reverting my patch
[23:08] <cehoyos> So why are you against reverting your patch?
[23:09] <durandal_1707> because seting pix_fmt, than checking pix_fmt and seting it again is bad practice
[23:09] <durandal_1707> it should be set in single place
[23:09] <cehoyos> So do I understand correctly that you want to fix the regression differently? That is perfectly fine with me.
[23:10] <durandal_1707> yes
[23:10] <cehoyos> Great, when will you do it?
[23:11] <durandal_1707> i can do it, and give whole commit authorship to you
[23:11] <cehoyos> Why? (I don't understand.)
[23:13] <durandal_1707> because you found it
[23:14] <cehoyos> Then please give me creadit for finding the issue if you want, just fix it;-)
[23:15] <durandal_1707> but you also provided solution for problem
[23:16] <cehoyos> Really? (I intentionally didn't.)
[23:23] <durandal_1707> can pnms be read with netpbm ?
[23:30] <durandal_1707> cehoyos: also what about pams with maxval < 65535 ?
[23:30] <durandal_1707> pams with depth 1
[23:32] <cehoyos> What about pams? I am not sure I understand.
[23:32] <cehoyos> (I thought a0348d0 only affected pgm's)
[23:33] <durandal_1707> 7e8e8ba9cc2b074b988e8ab28f69cfc5d454d674 changed pam code
[23:35] <cehoyos> It only affects images with maxval < 255 but the regression I mentioned above only appears for maxval > 255
[23:36] <durandal_1707> regression only broke pgms not pams
[23:37] <cehoyos> That is what I am saying, yes. And 7e8e8ba also only affected pgm's, so how is this related to pam's ?
[23:37] <durandal_1707> no 7e8.. affects only pams
[23:38] <cehoyos> Afaict, it only affects pgm's, but if you have a pam that gets affected, I'd love to see it.
[23:39] <durandal_1707> cehoyos: it changes only pam's code
[23:40] <durandal_1707> pgms are type 5
[23:45] <cone-858> ffmpeg.git 03Paul B Mahol 07master:768e40b451a4: Revert "pnm: remove nonsense code"
[23:51] <cehoyos> You are right, the commit message is not correct, it fixes pam decoding.
[23:59] <cehoyos> Your commit message is only marginally better;-( (255 < ...)
[23:59] <cehoyos> Why did you revert the patch?
[00:00] --- Mon Jul 29 2013
1
0
[02:37] <alsu> I have a video file which seems to have odd timing information. there are no frames between 21:21 and 34:40. playing the file with VLC correctly skips ahead in time instantly.
[02:37] <alsu> is there a way I can re-process this video with ffmpeg to remove that oddity?
[02:37] <alsu> preferably without re-encoding the video
[02:39] <alsu> (that is, I want to collapse time, not insert still frames)
[02:41] <Aprel> alsu: you could try `-acodec copy -vcodec copy`. That will simply take the file and put it in a new container. Maybe ffmpeg will be smart with the timing in the process.
[02:41] <Aprel> Dunno if it will work, but might be worth a shot.
[02:42] <alsu> Aprel: nope. same property in the output file
[02:46] <alsu> converting to another format either keeps the artifact or inserts strange (transparent?) frames for the duration of the gap
[05:16] <salamanderrake> seems like the best place to ask, what is better webm or mp4 as far as quality or are they just container?
[05:31] <defaultro> hey folks, -loop_input doesn't work anymore
[05:34] <defaultro> -loop 1 worked
[05:38] <defaultro> what's wrong with this, it's only creating one frame? ffmpeg -i M000077.jpg -t 10 -loop 1 -r 24000/1001 -vcodec libx264 -crf 0 -preset veryslow last.mp4
[05:38] <defaultro> -t 10 from my old notes works
[05:41] <defaultro> got it working: ffmpeg -loop 1 -i M000077.jpg -r 24000/1001 -vcodec libx264 -crf 0 -preset veryslow -t 10 last.mp4
[07:53] <johnny22> does ffmpeg support timed metadata frames ?
[07:53] <johnny22> i basically would like to have ffmpeg create a mp4 out of 3 pipes: video, audio and data
[07:57] <johnny22> not sure what type of data, maybe it's side data or metadata
[08:37] <blippyp> I'm applying a filter to the same video, but I keep getting this error: "First input link top parameters (size 704x480, SAR 1:1) do not match the corresponding second input link bottom parameters (704x480, SAR 40:33)". Even though it's the same file. The filter is doing something to it or something. How can I fix this? The scale option for filtering is suppose to have a sar parameter, but when I use it, it complains that this option doe
[08:38] <johnny22> is it possible to pipe a live subtitle stream as input to ffmpeg ?
[08:38] <blippyp> FYI - the video is used as two input sources for the filter - I'm blending them both, one on top of the other (same file).
[08:39] <blippyp> Sorry johnny22, I don't know anything about that... :(
[08:40] <johnny22> m'ok
[08:41] <blippyp> To help elaborate on my issue - The video was recorded on my hand-held video camera, and even though it was recorded at 16:9 at a resolution of 640x480 - if that makes any sense to anyone - but when I watch it, the video is 704x480....
[09:32] <alsu> hm, ffmpeg -i rtmp://... doesn't seem to work nearly as well as flvsteamer -r rtmp://...
[09:33] <alsu> is rtmp support still maintained?
[10:23] <ilmenite> what inbuilt variable is available which returns the length of the [0:v] input ?
[10:25] <Mavrik> none afaik.
[10:26] <ilmenite> ah, nvm, i don't actually need to specify the length. I tried my command without the -t options
[10:26] <ilmenite> option*
[16:59] <blippyp> when doing any overlay with an alpha layer, there is a 'blending' effect that happens to the transparent areas - anyone know how to prevent this 'blending' and instead be completely transparent?
[17:00] <Fjorgynn> thank you
[17:00] <Fjorgynn> blippyp: blending?
[17:00] <blippyp> I'm using alphamerge/alphaextract...
[17:01] <blippyp> when I add the alpha image to a background, the 'transparent areas' have a blending effect - I'm doing a overlay filter...
[17:03] <blippyp> ffmpeg -i $backdrop -map 0:1 -i $backdrop -vf "movie=tmpCamera.mov, scale=$cameraX:$cameraY [vid2]; [in][vid2] overlay=$cx:$cy" -c:v libx264 -crf 17 -preset ultrafast -to $testTime -c:a ac3 $base
[17:04] <blippyp> that's the final overlay command that I'm using - The alpha transparency is working, except the transparent edges aren't 'completely' transparent...
[17:17] <blippyp> fixed it: http://ffmpeg.org/pipermail/ffmpeg-user/2013-April/014562.html
[19:16] <Ener2> hello, how do I scale a video with ffmpeg, if video has two streams and I want to scale both?
[19:19] <blippyp> Ener2: Not sure, my guess would be to use the -vf (scale) option... You will likely need to map the video streams
[19:19] <Ener2> I tried something like this:
[19:19] <Ener2> ffmpeg -i HNI_0020.avi -map 0:0 -map 0:1 -map 0:2 -s 480x240 -acodec copy output.avi
[19:20] <Ener2> but alas it has only one stream included
[19:20] <Ener2> and it even changed the video codec...
[19:20] <JEEB> unspecified = container-specific default that can be pretty much anything
[19:21] <JEEB> also you can't scale without re-encoding, just FYI
[19:21] <Ener2> I dont mind reencoding
[19:22] <Ener2> also it seems to be missing audio stream even though it is mapped...
[19:24] <blippyp> This is suppose to work: ffmpeg -i input1 -i input2 -c copy -map 0 -map 1 output.mkv - Change your containers
[19:24] <Ener2> blippyp: but I have only one source
[19:26] <blippyp> I just did it and it worked...
[19:26] <Ener2> huh?
[19:27] <Ener2> you have two inputs in your command, I have only one input video containing three streams, audio which I want to copy and two video streams which I want to scale...
[19:27] <blippyp> do this: ffmpeg -i HNI_0020.avi -copy -map 0:0 -map 0:0 -map 0:1 out.mkv (this should duplicate your video stream twice and add the audio as well I believe)
[19:28] <Ener2> unrecognized option copy
[19:28] <Ener2> but again, I dont want to duplicate stream
[19:30] <blippyp> then what do you want exactly - because your previous command suggest that...
[19:30] <Ener2> I have a video
[19:30] <Ener2> it has one audio stream and two different video streams
[19:30] <Ener2> (it is a 3D video)
[19:30] <Ener2> what I want is to scale both videos to 480x240
[19:31] <Ener2> because now they are 800x240
[19:31] <blippyp> k - hold on...
[19:31] <Ener2> sure
[19:33] <blippyp> k - I just did this and it worked: ffmpeg -i in.avi -map 0:0 -map 0:1 -map 0:2 -s 480x240 out.avi
[19:33] <Ener2> blippyp: it does, but it changes the codec...
[19:33] <blippyp> basically, exactly what you said previously...
[19:34] <blippyp> it worked for me fine
[19:34] <Ener2> it changes codec for me
[19:34] <blippyp> (i forgot to included the -c:a copy in my example - I used it when I tested it...
[19:34] <blippyp> yes it has to
[19:34] <Ener2> why?
[19:34] <blippyp> or at least, re-encode it - because you changed the video size
[19:34] <Ener2> sure but reencode to same codec...
[19:35] <blippyp> doesn't matter, it's not a copy
[19:35] <Ener2> http://pastebin.com/5rzjLrZp
[19:35] <blippyp> it has to re-encode it
[19:35] <Ener2> well codec matters to me otherwise video will be unplayable on the device
[19:35] <blippyp> then you have to re-encode them
[19:35] <Ener2> ugh three reencodes will make video ... unwatchable...
[19:36] <Ener2> cant I specify which codecs?
[19:36] <Ener2> to use
[19:36] <JEEB> yes
[19:36] <blippyp> yes
[19:36] <blippyp> c:v libx264 will encode in x264
[19:36] <JEEB> s/in x264/in H.264/
[19:36] <Ener2> hmm
[19:37] <Ener2> there is no codec mjpeg/mjpg...
[19:37] <blippyp> type ffmpeg -codecs to see a list of codecs you can use
[19:37] <blippyp> yes there is - it's mjpeg I believe
[19:37] <blippyp> Hmm, maybe not - I thought for sure I read about that being one...
[19:38] <Ener2> [NULL @ 04d601c0] Unable to find a suitable output format for 'mjepg' mjepg: Invalid argument
[19:38] <Ener2> err
[19:38] <Ener2> this
[19:38] <Ener2> NULL @ 04d201c0] Unable to find a suitable output format for 'mjpeg' jpeg: Invalid argument
[19:38] <blippyp> Maybe you have to re-compile ffmpeg to get mjepg working... I'm sure I read about it working... You probably just need to include it....
[19:39] <JEEB> wat
[19:39] <JEEB> Ener2, pastebin your full command line and output and link here
[19:39] <Ener2> d:\export>ffmpeg -i HNI_0020.avi -map 0:0 -map 0:1 -map 0:2 -s 480x240 -c:a -c:v mjpeg out.avi
[19:39] <Ener2> sure
[19:39] <JEEB> uhh
[19:39] <JEEB> see -c:a and -c:v right after it
[19:39] <Ener2> http://pastebin.com/gAGf0EFs
[19:40] <Ener2> hmm
[19:40] <JEEB> -c:a copy -c:v mjpeg maybe? :P
[19:40] <Ener2> hmm
[19:40] <Ener2> okay seems to be reencoding but
[19:40] <Ener2> I get this error:
[19:40] <Ener2> [avi @ 048ef420] Non-monotonous DTS in output stream 0:1; previous: 1485, curren
[19:40] <Ener2> t: 492; changing to 1486. This may result in incorrect timestamps in the output
[19:40] <Ener2> file.
[19:41] <Ener2> and no audio stream at all
[19:41] <Ener2> video streams are fine though
[19:41] <JEEB> once more pastebin kthx
[19:41] <Ener2> uhh I will try to catch it
[19:42] <JEEB> just push stdout and stderr to a file
[19:42] <JEEB> Ener2, http://stackoverflow.com/questions/1420965/redirect-stdout-and-stderr-to-a-…
[19:42] <Ener2> god knows how you do it in windows
[19:42] <Ener2> ah okay thanks
[19:43] <blippyp> got it working - I see what you did wrong - include an audio encoder
[19:43] <blippyp> you didn't in your example...
[19:43] <JEEB> I think we're like a mile past that :P
[19:43] <blippyp> ffmpeg -i video.avi -c:a ac3 -c:v mjpeg out.avi
[19:43] <Ener2> I would need
[19:43] <Ener2> adpcm_ima_wav
[19:44] <Ener2> to be specific
[19:44] <Ener2> adpcm_ima_wav ([17][0][0][0] / 0x0011), 48000 Hz, stereo
[19:44] <JEEB> which is -c:a copy
[19:44] <Ener2> which gives that error...
[19:44] <JEEB> thus, pastebin pl0x
[19:45] <Ener2> http://pastebin.com/QcfLJ3by
[19:45] <Ener2> stdout
[19:45] <JEEB> well yeah
[19:45] <JEEB> just use what I linked
[19:45] <JEEB> :P
[19:45] <JEEB> because ffmpeg outputs most stuff to stderr
[19:45] <Ener2> that is what I used...
[19:46] <JEEB> and if you got your warnings during encoding why did you exit when it asked about overwriting... -__
[19:46] <Ener2> d:\export>ffmpeg -i HNI_0020.avi -map 0:0 -map 0:1 -map 0:2 -s 480x240 -c:a copy
[19:46] <Ener2> -c:v mjpeg out.avi > a.txt 2>&1
[19:46] <JEEB> how am I supposed to help you for eff's sake
[19:46] <JEEB> ok, so you had stdout and -err there
[19:46] <Ener2> hmm well I am not sure when to exit actually
[19:46] <Ener2> I just exit after some delay
[19:46] <Ener2> but I guess I have to put in "y", sorry
[19:47] <Ener2> JEEB:http://pastebin.com/3HpCnnLt
[19:47] <JEEB> ok
[19:48] <Ener2> also I dont like those changes between reencodings
[19:48] <Ener2> like SAR DAR for mjpegs....H
[19:48] <Ener2> but might not be fatal for playback I suppose
[19:48] <JEEB> well
[19:48] <JEEB> you changed the aspect ratio of the picture
[19:48] <Ener2> okay
[19:48] <JEEB> thus the metadata will have to change as well
[19:49] <JEEB> that tells it to keep the original ratio
[19:49] <JEEB> naturally, with avi most things don't read that stuff from the video stream :P
[19:49] <Ener2> also video quality is horrible
[19:49] <JEEB> welcome to ffmpeg defaults
[19:50] <Ener2> along with no sound stream
[19:50] <JEEB> also those messages pretty much mean that the decoding timestamps (DTS) of the audio stream are somehow wonky o_O
[19:50] <Ener2> video has only 20fps though
[19:50] <Ener2> maybe that can throw it off?
[19:50] <JEEB> I see it writing the audio tho
[19:50] <Ener2> yeah but media player sees no audio stream
[19:50] <JEEB> audio:2462kB
[19:50] <Ener2> Video: MJPG 480x240 20.00fps [Stream 00]
[19:50] <Ener2> Video: MJPG 480x240 20.00fps [Stream 02]
[19:51] <JEEB> I think AVI had per-stream info for that
[19:51] <JEEB> so no, the frame rate shouldn't really matter
[19:51] <JEEB> it's a per-stream thing
[19:51] <Ener2> original file:
[19:51] <Ener2> Video: MJPG 800x240 20.00fps [Stream 00]
[19:51] <Ener2> Video: MJPG 800x240 20.00fps [Stream 02]
[19:51] <Ener2> Audio: IMA ADPCM 48000Hz stereo 128kbps [Stream 01]
[19:51] <JEEB> also if you want to make sure random players can handle it, I recommend you take it through virtualdub with copying audio and video
[19:51] <JEEB> that muxes in a way that MS things generally like
[19:52] <Ener2> actually it is for 3DS ...
[19:52] <Ener2> but I encoded original, which I already deleted to wrong size...
[19:52] <Ener2> thus the need for scale
[19:53] <JEEB> anyways, can't really say more about why the audio is wonky, and if it's an ffmpeg bug, or if it's something else
[19:53] <JEEB> you're in general doing it all right command-line wise
[19:56] <JEEB> but it most definitely is writing the audio stream there :)
[19:56] <Ener2> :(
[19:56] <Ener2> cat hear anything though
[19:56] <JEEB> so do see what remuxing that with vdub + audio and video set to copy does
[19:57] <JEEB> what did you create that thing you're inputting with btw?
[19:57] <JEEB> ffmpeg or something else?
[19:57] <JEEB> as in, should ffmpeg's avi output work as is on that nintendo thing?
[19:57] <Ener2> well it should
[19:57] <Ener2> I found a website with ffmpeg commands
[19:58] <Ener2> and it was fine except they had wrong resolution...
[19:58] <JEEB> funky
[19:58] <Ener2> https://javaz.org/2011/12/ffmpeg-convert-3d-video-for-nintendo-3ds/
[19:58] <JEEB> might want to report it over at the trac then I guess?
[19:58] <JEEB> with a sample that creates those DTS messages when you read it in and try to copy the audio
[20:00] <blippyp> do you guys have any experiene with alphamerge dropping frames - do you know any possible fixes? like switching to a different codec maybe? I'm usuing qtrle in a mov container currently, and it keeps dropping frames and messing up my final video...
[20:03] <blippyp> my only current solution at this time is to record a longer video sequence for that section and then re-encode when I'm done specifying a time before the video freezes, everything looks fine in the video, but that section just 'freezes' after a while...
[20:42] <alsu> can I overwrite the timecode in a file?
[20:43] <alsu> I think the timecode in this represents the live stream, but I tuned in about 3 hours in, so it thinks 3 hours of video is missing :/
[23:06] <Aprel> hi is there anyway to truncate an audio/video file in ffmpeg without de/reencoding it?
[23:07] <durandal_1707> Aprel: yes, not that you can truncate audio packets only not exact samples
[23:07] <durandal_1707> *note
[23:07] <Aprel> for ex: if I have a 4-hour aac-encoded mp4 file, and I'm only interested in the last hour, can I do that?
[23:08] <Aprel> durandal_1707: do you know the command?
[23:08] <Mavrik> use "copy" as codec.
[23:09] <Aprel> Mavrik: okay, then how do I specify to ffmpeg with portion of the file to cut out?
[23:09] <Aprel> ^what
[23:11] <Aprel> Okay, I'll check the docs, but is there a term I can search for for what I want to do? I didn't get far on google because 'cropping' seems to mean to cut out pixels, not frames. What is cutting frames called?
[00:00] --- Mon Jul 29 2013
1
0
[00:28] <durandal_1707> sorry for spam
[00:35] <durandal_1707> michaelni: if i want to mux qtrle into nut, shoult i add qtrle tag to nut.c or isom tags table to nut.c ?
[00:41] <michaelni> id say table to nut.c if this has no negative sideeffects
[00:42] <michaelni> also nut4cc.txt needs to be updated which needs a patch to be sent to nut-devel for discussion there
[00:45] <durandal_1707> i'm not going to subscribe to yet another mailing list nobody use
[00:46] <durandal_1707> also nut does not store bits_per_coded_sample anywher
[00:46] <llogan> nobody? then you won't have to worry about getting any messages from it
[00:46] <durandal_1707> and adding extradata handling in encoder/decoder and muxer smells wrong
[00:47] <durandal_1707> in this case extradata would just store bits_per_coded_sample
[00:48] <durandal_1707> llogan: why you do not comment other, bigger patch?
[00:49] <llogan> which one is the bigger patch?
[00:49] <durandal_1707> very big patch
[00:51] <llogan> sorry, but i don't know which one that is.
[00:52] <wm4> too bad
[00:52] <durandal_1707> llogan: one i sent the other day not this day
[00:53] <llogan> compand?
[00:57] <durandal_1707> and perspective, and whole doc/*
[01:42] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:8efe96ee6f3f: swscale/fill_rgb2yuv_table: fix default detection
[01:42] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:8720d3ac217e: avfilter/vf_scale: set in_color_matrix default to "auto"
[01:47] <wm4> michaelni: would it be possible to add ycgco to swscale?
[01:47] <wm4> that's really just a "why not", not a serious feature request
[02:03] <Compn> wm4 : got a sample ?
[02:14] <michaelni> everything is possible, i suspect its not trivial though
[08:59] <cone-368> ffmpeg.git 03Hendrik Leppkes 07master:779e6c2b985f: lavfi: add attribute_align_arg to all public entry points
[08:59] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:2fffc05f424f: Merge commit '779e6c2b985f2ad461a1ae704160822f97709ba7'
[09:05] <cone-368> ffmpeg.git 03Martin Storsjö 07master:1297f7b87f8a: rtpenc: Fix some odd comments
[09:05] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:5cb3d35dcc23: Merge commit '1297f7b87f8a84930a23eca705765c7c353dfcd5'
[09:12] <cone-368> ffmpeg.git 03Hendrik Schreiber 07master:b512360184ad: avio: Don't set the seekable flag if no seek function is provided
[09:12] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:103b8d876a86: Merge commit 'b512360184ade835fba621f5042d643fc9e2ee9d'
[09:18] <cone-368> ffmpeg.git 03Michael Kostylev 07master:80ade7985cd9: AIX: add support for shared builds
[09:18] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:e64dc86a13c0: Merge commit '80ade7985cd95156e2156f50adc7b86d0e43a07a'
[09:25] <cone-368> ffmpeg.git 03Martin Storsjö 07master:86f042dcabde: wtv: Make WTV_SECTOR_BITS a 64 bit constant
[09:25] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:902792852279: Merge commit '86f042dcabde2a5386dbd95ab0451b274987d253'
[09:52] <cone-368> ffmpeg.git 03Martin Storsjö 07master:979e9e8f3671: wtv: Drop some casts that now are unnecessary
[09:52] <cone-368> ffmpeg.git 03Michael Niedermayer 07master:e8023dbaf0db: Merge remote-tracking branch 'qatar/master'
[10:24] <ThePuppetMaster_> i try to recode a avi file into a mpeg file. but i get a error: Insufficient thread locking around avcodec_open/close() ... thats the full output: http://pastebin.com/SGn1UH3j .... can anyone tell me whats wrong, and how i can fix it?
[10:33] <michaelni> ThePuppetMaster_, check what is calling avcodec_open/close at the same time
[10:35] <michaelni> and if these calls are valid & intended to occur at the same time then set a lock manager with av_lockmgr_register
[10:35] <ThePuppetMaster_> well ... how i can check this? ... i'm only use mencoder ... i'm not develing in it ... but, i search in the source the point, where calling this output, but foudn a lot and i don't know whats the right
[10:36] <ThePuppetMaster_> and, avcodet lib is realy big :P
[10:37] <michaelni> a gdb backtrace from where the message is printed will tell you one of the 2 conflicting uses of avcodec_open/close
[10:37] <ThePuppetMaster_> can u give me a searchpoint for me?
[10:37] <ThePuppetMaster_> ok
[10:37] <ThePuppetMaster_> i try it
[10:49] <michaelni> also to find the 2nd one, you can make gdb print stacktraces on calls to avcodec_open/close, something like break avcodec_open // commands 1 // bt // continue
[10:56] <ThePuppetMaster_> well .. gdb is not my favorite .. .. its hard for me to use it
[10:56] <ThePuppetMaster_> but, it looks, my mencoder was compiled without -debug flags
[10:57] <ThePuppetMaster_> i only get adresses of the break
[10:57] <ThePuppetMaster_> like this #0 0x00007fffefa221b5 in ?? ()
[13:00] <durandal_1707> perhaps those failures with clang 2.9 are because of clang bug?
[13:16] <ThePuppetMaster> which?
[13:20] <ubitux> hey
[13:20] <ubitux> i'm alive
[13:20] <ubitux> i'm soon back on ffmpeg
[13:20] <durandal_1707> who are you?
[13:21] <ubitux> BBB / BBB-work is the vp9 prect still ok?
[13:21] <ubitux> durandal_1707 :D
[13:23] <ubitux> s/prect/project/
[13:23] <ubitux> anyway, 1st august i'm on the plane on my way back
[13:53] <saste> ubitux, hey!
[13:53] <saste> why, where are you this time?
[13:55] <ubitux> japan again saste
[13:55] <ubitux> lost in hokkaido :)
[13:55] <saste> eheh
[13:55] <ubitux> i love this country so much seriously...
[13:57] <JEEB> hokkaido seems nice
[13:57] <JEEB> TT is currently in Japan, too
[13:57] <durandal_1707> japan doesn't have internet?
[13:57] <JEEB> very little free wifi
[13:57] <JEEB> the only one I know is at the tokyo metro
[13:58] <JEEB> basically if you need internets your best bet is to buy a b-mobile (virtual operator on the NTT DoCoMo network) prepaid SIM card for data
[13:58] <JEEB> you don't need papers that you live in Japan for getting a data-only line, thank goodness
[13:58] <saste> JEEB, probably he has better things to do there than checking his emails
[13:59] <JEEB> haha, most probably :)
[13:59] <JEEB> still, people in Japan don't SMS
[13:59] <JEEB> they e-mail
[13:59] <JEEB> (because of the differing types of networks and the fact that you couldn't even send SMS to other networks within the same technological system until very lately)
[14:00] <JEEB> So, even if DoCoMo and SoftBank f.ex. both used W-CDMA for 3G, you couldn't still send SMS between these two until very recently IIRC :)
[14:00] <JEEB> and then you have au with CDMA2000rev0
[14:01] <JEEB> and willcom with le HPS
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:5efa5103b0e7: sgidec: simplify return path
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:997e2b59e354: sgidec: return meaningful error codes
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:86e722ab97d7: sgidec: safer check for buffer overflow
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:573018c61fad: lclenc: cosmetics: reindent
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:6dbcecd78e02: lclenc: allocate FF_INPUT_BUFFER_PADDING_SIZE for extradata
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:3ea5d01a12ed: escape130: make "chroma_vals" static
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:d9954ccff07b: targa: set pict_type
[14:05] <cone-656> ffmpeg.git 03Paul B Mahol 07master:88b071a47392: exr: set pict_type
[15:31] <cone-656> ffmpeg.git 03Michael Niedermayer 07master:5e763920a44f: avcodec/mpegvideo: update disabled assert() to av_assert0()
[15:31] <cone-656> ffmpeg.git 03Michael Niedermayer 07master:3adf054b22d6: avcodec/vorbisenc: change 6 asserts to av_asserts()
[15:45] <mateo`> is ffplay enable to use vaapi for decoding ?
[15:46] <JEEB> if it's a hwaccel, no -- those need extra code to use it from the using side
[15:47] <mateo`> would it be a good idea to add such code to ffplay ?
[15:54] <durandal_1707> JEEB: ut patch sent
[15:56] <JEEB> ah
[15:57] <JEEB> at a quick glance looks OK
[15:57] <JEEB> have to run to the Koff's park soon
[15:57] <JEEB> free sausages
[15:58] <ubitux> < JEEB> still, people in Japan don't SMS // there is some kind of gateway between the two afaict
[15:58] <ubitux> which is cool
[15:58] <JEEB> yeah, lately
[15:58] <ubitux> and for free internet mmh, guest houses & hostel have them :)
[15:58] <JEEB> well, yeah :)
[15:59] <JEEB> I just need e-mail when I'm mobile generally in Kantou
[15:59] <ubitux> i actually have internet access almost every day, but i'm busy wondering about where i can go the next day :)
[15:59] <JEEB> since phone calls are lolexpensive and I was stupid enough not to make a softbank prepaid back when I had the papers
[15:59] <JEEB> also I should have definitely made a bank account in '05
[15:59] <durandal_1707> ubitux: are you near sea or vulcano or nuclear reactor?
[15:59] <JEEB> they made the rules harsher in '06 and now it's quite herp derp
[16:00] <ubitux> durandal_1707: i did a few island, and passed along fukushima
[16:00] <ubitux> right now, i'm at sapporo, going down to tokyo, yukuri~
[16:00] <ubitux> no volcano this time
[16:01] <ubitux> but i went to aso last time
[16:02] <ubitux> not a single earthquake yet
[16:02] <ubitux> almost disappointed :(
[16:03] <JEEB> you get used to them rather quickly
[16:03] <JEEB> and most are like shindo 1-3
[16:03] <ubitux> i didn't get any
[16:03] <ubitux> well it's not like i've lived in a earthquake-area for 5 years but well
[16:04] <ubitux> anyway, gtg 'see ya
[16:24] <durandal_1707> how much log() is slow?
[16:51] <cone-656> ffmpeg.git 03Stefano Sabatini 07master:b4bd21b7fe2a: doc/codecs: fix dangling reference to codec-options chapter
[17:38] <wm4> has anyone tried this lavfi opencl stuff? does it actually help?
[19:01] <cehoyos> Compn: Hi, in case user hotwings reappears and asks about vdpau again, please ask him to test current MPlayer and send him to one of the mailing lists. (I don't even understand his question.)
[19:02] <BBB> ubitux: yes
[19:02] <BBB> ubitux: we start in about a week right?
[20:18] <BBB> durandal_1707: pretty awful
[20:18] <BBB> durandal_1707: you can get faster if you know limitations on use
[20:19] <BBB> e.g. lower precision, lower range, ec.
[20:19] <BBB> *etc.
[20:35] <durandal_1707> hmm, filter does log+exp for every sample which is pretty costly...
[20:35] <BBB> lol that sounds awful yes
[20:35] <BBB> try profiling it
[20:35] <mateo`> michaelni, durandal_1707: hi, do you think it would be acceptable to add a write_apic_packet function to the muxers ?
[20:51] <durandal_1707> BBB: for 44100 sample rate its fast enough....
[20:52] <durandal_1707> but 96000 would already be slow here
[21:07] <michaelni> mateo`, which muxers and what would the function do ?
[21:10] <cone-656> ffmpeg.git 03Tim.Nicholson 07master:84e345b38e58: Forward interlaced field information from mov to v210 decoder.
[21:10] <mateo`> michaelni: it would be implemented in all muxers which handle cover arts (mp3, aiff, flac, ...)
[21:11] <mateo`> michaelni: all packets from streams with AV_DISPOSITION_ATTACHED_PIC will be pushed into it
[21:12] <mateo`> so we have a "generic" way to handle thoses streams in muxer (and avoid a if block)
[21:12] <michaelni> sounds reasonable ...
[21:13] <mateo`> but as I said on the mailing list, since my goal is still to add this support in the mp4 muxer, it won't simplify the support for this muxer
[21:14] <mateo`> or at least all the instrusive part
[21:14] <michaelni> i think something should be done about the intrusive stuff
[21:17] <mateo`> me too, but without addind a mapping between stream indexes and track indexes, the only solution left, is to not consider apic streams as normal streams (and not reference them in s->streams)
[21:17] <mateo`> (afaik)
[21:18] <michaelni> if theres no good solution then we have to accept a not so good one :(
[21:18] <michaelni> iam not objecting to the intrusiveness if its the least problematic path
[21:20] Action: michaelni afk
[21:20] <mateo`> it seems to be the least problematic path without modifying the way attached pics are handled
[21:21] <durandal_1707> the current (iirc only?) patch on ml?
[21:22] <mateo`> durandal_1707: yes
[22:08] <cone-656> ffmpeg.git 03Paul B Mahol 07master:60a7fac61b81: utvideoenc: use av_image_copy_plane()
[22:23] <cone-656> ffmpeg.git 03Paul B Mahol 07master:0909f3edbdfa: zmbv: return meaningful error code
[22:54] <cone-656> ffmpeg.git 03Paul B Mahol 07master:9d099b9ae4b6: targa_y216dec: remove empty function
[22:54] <cone-656> ffmpeg.git 03Paul B Mahol 07master:6a574942040c: dpxenc: cosmetics: reindent
[22:54] <cone-656> ffmpeg.git 03Paul B Mahol 07master:94ad38e2e8a5: dpx: use av_image_copy_plane()
[23:09] <cone-656> ffmpeg.git 03Marton Balint 07master:63c0113588d6: lavc: do not override format if neither text nor bitmap codec prop is set
[23:09] <BBB> durandal_1707: define fast enough
[23:10] <BBB> durandal_1707: you make it sound like there is a specific threshold above which it is 100% ok and below which it is 0% ok
[23:10] <BBB> durandal_1707: I don't think that's how these things work
[23:10] <nevcairiel> fast enough = still works in realtime :P
[23:10] <BBB> durandal_1707: we should make it afap (as fast as possible), so that it can be used in combination with as many features combined on as many systems combined that are as low-power as possible
[23:10] <durandal_1707> well its 40x fast for 44100
[23:10] <Compn> cehoyos : hotwings is on irc, send him a pm
[23:10] <Compn> cehoyos : hes just not on this channel...
[23:11] <durandal_1707> BBB: i just removed log/exp and it was 10x slower (I did not scaled other numbers to get correct output)
[23:12] <BBB> you are trying to say log/exp made it faster?
[23:12] <BBB> or you mean log/exp is 90% of its runtime?
[23:13] <durandal_1707> no, i think its because long float multiplications are slower or faster depending on input
[23:13] <durandal_1707> i did not looked how much log/exp costs
[23:13] <BBB> hm...
[23:14] <durandal_1707> or its CPU...
[23:36] <saste> mmh saturday night, time for nerds to send patches...
[00:00] --- Sun Jul 28 2013
1
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[00:12] <linemimic> Sorry, that script is broken. This one works: http://pastebin.com/q4X3GdM64
[01:49] <cxo> If I --enable-decoder=aac, I get undefined reference to av_dct_init/calc/end when make reaches ffmpeg_g
[01:50] <cxo> (using git as over yesterday)
[02:05] <cxo> hmm i did --enable-dct and that compile error stopped, but at runtime (ffprobe) i get the same error
[09:26] <Belliash> hi
[09:26] <Belliash> is it possible to pipe output of ffplay to ffmpeg ?
[10:19] <ThePuppetMaster_> hi @all
[10:19] <ThePuppetMaster_> i try to recode a avi file into a mpeg file. but i get a error: Insufficient thread locking around avcodec_open/close() ... thats the full output: http://pastebin.com/SGn1UH3j .... can anyone tell me whats wrong, and how i can fix it?
[15:23] <Eduard_Munteanu> Hi.
[15:25] <Eduard_Munteanu> Could anyone suggest a near-lossless video codec, other than H264? Somebody's going to send me a video, I'll convert it to H264, but quality needs to be reasonable and the file shouldn't be too large.
[15:26] <Eduard_Munteanu> I'll convert to H264, so I don't care about playback details like hw acceleration.
[15:26] <klaxa> h264 is probably the best lossly compression you'll get
[15:26] <JEEB> unfortunately x264's lossless is most probably gonna end up the smallest (a small exception with not so changing content is to just 7z raw YCbCr or something)
[15:27] <JEEB> also with lossy x264's H.264 at low crfs can be quite "visually similar" if lossless in any case is gonna be too big
[15:27] <Eduard_Munteanu> I see... they've already sent me a huge, 1GiB msvideo1-encoded crappy quality video. :)
[15:28] <JEEB> also this reminds me of asking for a lossless clip and I got raw RGB in mov :P
[15:28] <JEEB> packed into rar
[15:28] <klaxa> haha
[15:28] <Eduard_Munteanu> JEEB: in my experience something like crf 16 is manageable wrt size.
[15:28] <Eduard_Munteanu> And quite lossless.
[15:29] <zap0> quality is subjective.
[15:30] <JEEB> crf 16 would really depend on the resolution and content
[15:30] <Eduard_Munteanu> zap0: sort of... if you can prove humans can't distinguish, that sort of quality is rather objective
[15:30] <JEEB> in the end I don't like taking chances so I end up just using lossless (Ž
[15:30] <JEEB> f.ex. I remember encoding DVD rips with crf 16 at one point
[15:31] <JEEB> although the problem is general in the fact that when you encode SD and your screen resolution is way higher than that
[15:31] <JEEB> even small artifacts start popping up
[15:31] <zap0> HDD space is 20G/$
[15:31] <Eduard_Munteanu> JEEB: it's just a 30s video, lossless should be fine I guess
[15:31] <Eduard_Munteanu> Hopefully they manage to do that, I'm talking about a non-technical person.
[15:31] <JEEB> :D
[15:31] <zap0> sometimes putting it into pespective like that, loosing your work to safe 20c of HDD space doesn't make sense.
[15:32] <Eduard_Munteanu> (and using some Windows thingy)
[15:32] <JEEB> well, on windows most editors at least can handle VFW output
[15:32] <JEEB> so in the worst case you can use ut video or something
[15:33] <Eduard_Munteanu> JEEB: what's "ut video"?
[15:33] <JEEB> a Japanese lossless video format that has decoders and encoders in pretty much every imaginable framework
[15:33] <Eduard_Munteanu> Ah.
[15:33] <JEEB> supports 4:2:0, 4:2:2 and RGB
[15:34] <klaxa> i didn't know video codecs had nationalities :>
[15:34] <JEEB> well, it's made by a guy called umezawa, thus ut
[15:34] <klaxa> ah
[15:34] <JEEB> originally to work with captured game content
[15:35] Action: JEEB implemented the encoder in libavcodec, and fixed single-symbol mode in the decoder last year
[15:35] <Eduard_Munteanu> Can webm go lossless, I wonder?
[15:35] <JEEB> webm is a container which is a subset of matroska
[15:35] <JEEB> not sure if VP8 had a lossless mode
[15:36] <JEEB> at least it didn't have at start
[15:36] <Eduard_Munteanu> Er, VP8
[15:36] <JEEB> ...and I'm poking umezawa because his dyndns host seems to be down
[15:37] <JEEB> (I mostly like Ut Video purely because it's open source and has stuff for so many frameworks, VFW, DS, MF, QT (OS X)... and libavcodec)
[15:37] <durandal_1707> its libavcodec decoder/encoder slow?
[15:37] <JEEB> thus you can use it out of many video editors and thus just tell folk to use it and get lossless :D
[15:38] <JEEB> durandal_1707, I don't think it has much SIMD, but I wouldn't call it *slow* per se
[15:38] <Eduard_Munteanu> Hm, I see.
[15:38] <JEEB> I could frame thread the encoder I guess, but people want sliced threading
[15:38] <JEEB> which they then found out wasn't a good idea because of :reasons:
[15:39] <Eduard_Munteanu> I wonder if I should ask for raw video in a zip.
[15:39] <JEEB> the official windows version at least has a *lot* of SIMD
[15:39] <durandal_1707> also Ut Video have only key frames
[15:40] <JEEB> yeah, it's not meant for high compression
[15:40] <JEEB> but the fact it's available pretty much everywhere makes it viable
[15:40] <JEEB> as in, you can ask a random video editor guy to give you stuff in it in most cases
[15:40] <Eduard_Munteanu> Only keyframes is useful for synced playback, I ended up encoding stuff like that.
[15:40] <durandal_1707> compared to ffvhuff?
[15:40] <JEEB> because that guy just has to install the component for that OS and be done with it
[15:40] <JEEB> uhh
[15:41] <JEEB> ffdshow-tryouts is dead
[15:41] <JEEB> and there are no things for MF or QT (OS X)
[15:41] <JEEB> and there is no DShow encoder
[15:41] <JEEB> umezawa has been pretty busy with implementing his stuff on pretty much all the things
[15:42] <JEEB> also what colorspaces did ffvhuff support?
[15:42] <JEEB> but yeah, compression-wise it's similar to ffvhuff
[15:43] <durandal_1707> JEEB: you wrote ut encoder in libavcodec?
[15:43] <JEEB> yes
[15:43] <durandal_1707> then why it does not use memcpy?
[15:44] <JEEB> excuse me?
[17:21] <griddy> Hi guys, what is the purpose of providing video and audio config settings for a input feed of FFServer?
[17:22] <zap0> why not?
[17:24] <griddy> I understand the settings for reincoding your feed to a suitable output stream but why the feed settings. I am a noob:P
[17:45] <zap0> why not feed settings?
[17:46] <zap0> your quetsion is like, why is the sky above us
[17:50] <t4nk452> hi
[17:51] <t4nk452> getting the following error
[17:51] <t4nk452> Unrecognized option 'directpred'. Error splitting the argument list: Option not found
[17:52] <t4nk452> full error http://pastebin.com/gkRDenfV
[18:02] <Angelaaaaa> hola
[19:14] <sacarasc> What kinds of subtitles does MP4 support?
[19:28] <Plorkyeran> mpeg-4 timed text
[19:29] <Plorkyeran> afaik muxing anything else into mp4 involves converting it to ttext or stuffing it in a userdata atom
[23:47] <blippyp> durandal_1707 - thx for pushing me to you use blend yesterday - I finally figured it out - much faster - just wanted to say thanks...
[00:00] --- Sun Jul 28 2013
1
0
[10:43] <cone-300> ffmpeg.git 03Diego Biurrun 07master:6c145ecf785d: twinvq: K&R formatting cosmetics
[10:43] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:e28fabb4cbe5: Merge commit '6c145ecf785dc3d26ba3fed3ea9892cc80244625'
[10:49] <cone-300> ffmpeg.git 03Diego Biurrun 07master:4a2ef39442bf: cosmetics: Add '0' to float constants ending in '.'.
[10:49] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:af7949fdea2f: Merge commit '4a2ef39442bf7f0150db07a1fbfcf8286e4d44a3'
[10:57] <cone-300> ffmpeg.git 03Rainer Hochecker 07master:582963a81565: vdpau: Fix VC-1 interlaced mode
[10:57] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:bbc7d33a0d7d: Merge commit '582963a8156522582e55466be4a59974a8d909a5'
[11:08] <cone-300> ffmpeg.git 03Rémi Denis-Courmont 07master:93a51984a27f: mpeg12: Ignore slice threading if hwaccel is active
[11:08] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:5dc2c99055ae: Merge commit '93a51984a27f3ba84d4e6f13d0c704ee9891603e'
[11:20] <cone-300> ffmpeg.git 03Diego Biurrun 07master:03039f4c8cdb: miscellaneous typo fixes
[11:20] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:46ad2d9e4463: Merge remote-tracking branch 'qatar/master'
[12:07] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:6663205338f5: avformat/dtsdec: Improve probe, reject things looking like analoge signals
[12:34] <cone-300> ffmpeg.git 03Michael Niedermayer 07release/1.0:34dfb907b222: avformat/dtsdec: Improve probe, reject things looking like analog signals
[12:34] <cone-300> ffmpeg.git 03Michael Niedermayer 07release/1.1:a1ac3c2d9cff: avformat/dtsdec: Improve probe, reject things looking like analog signals
[12:34] <cone-300> ffmpeg.git 03Rémi Denis-Courmont 07release/1.2:70127070dd9f: mpeg12: Ignore slice threading if hwaccel is active
[12:34] <cone-300> ffmpeg.git 03Michael Niedermayer 07release/1.2:875649bfaef8: avformat/dtsdec: Improve probe, reject things looking like analog signals
[12:34] <cone-300> ffmpeg.git 03Rémi Denis-Courmont 07release/2.0:ccf470fdb6e3: mpeg12: Ignore slice threading if hwaccel is active
[12:34] <cone-300> ffmpeg.git 03Michael Niedermayer 07release/2.0:18043e3d2282: avformat/dtsdec: Improve probe, reject things looking like analoge signals
[13:35] <durandal_1707> shit some guy started to send our code to fork
[13:44] <Compn> gasp!
[13:45] <durandal_1707> they can change random thing and break compability when merged by michaelni, as it already happened multiple times
[13:47] <cone-300> ffmpeg.git 03Paul B Mahol 07master:2a1a599bcb2b: lavfi/buffersrc: remove unused item from BufferSourceContext
[13:52] <Compn> did they rename jp2k twice ?
[13:56] <Compn> looks like i have to find samples of imc mode 1 for kostya :)
[14:05] <cone-300> ffmpeg.git 03Hendrik Leppkes 07master:af2a196e669c: lavfi: add attribute_align_arg to all public entry points
[14:47] <wm4> ffmpeg can't decode a simple 16384x16384 png
[14:47] <wm4> Picture size 16384x16384 is invalid
[14:50] <durandal_1707> wm4: how you got such sample?
[14:50] <wm4> http://dl.dropboxusercontent.com/u/5321377/Sierpinski%20Gaskets/sierpinski1…
[14:53] <durandal_1707> you can actually display it with some other app?
[14:53] <nevcairiel> there is some odd overflow check in imgutils which i dont really understand
[14:54] <nevcairiel> it checks if (w+128)*(h+128) is > INT_MAX/8
[14:54] <nevcairiel> not sure why
[14:54] <nevcairiel> probably for the image buffer size
[14:54] <durandal_1707> remove that check, and try to display it....
[14:54] <durandal_1707> good luck
[14:55] <nevcairiel> i see no reason it doesnt work
[14:55] <durandal_1707> on 64bit computer and bunch of RAM, maybe
[14:55] <wm4> it did work with some image viewer
[14:55] <wm4> on a 32 bit system
[14:55] <wm4> (ok maybe there's some luck involved)
[14:56] <durandal_1707> what image viewer?
[14:56] <wm4> simple gtk based thing
[14:56] <durandal_1707> and it have whole decoded image in memory?
[14:56] <nevcairiel> rgba 16384x16384 only uses a gigabyte of memory
[14:56] <wm4> yes
[14:56] <nevcairiel> should be well in thel imits
[14:56] <durandal_1707> yes,
[14:56] Action: nevcairiel removes check
[14:57] <durandal_1707> function is flawed
[14:57] <wm4> anyway, I don't understand that check either
[14:57] <wm4> maybe 8 because there's a maximum of 8 bytes per pixel?
[14:57] <nevcairiel> possible
[14:57] <durandal_1707> rgba64 would take 2 gb
[14:58] <durandal_1707> but design is flawed anyway
[14:58] <nevcairiel> thats near the signed integer overflow, which it will try to protect against i guess
[14:58] <durandal_1707> as monow would take even less memory
[14:58] <wm4> int sws_scale(struct SwsContext *c, const uint8_t *const srcSlice[],
[14:58] <wm4> const int srcStride[], int srcSliceY, int srcSliceH,
[14:58] <wm4> uint8_t *const dst[], const int dstStride[]);
[14:58] <wm4> oh... ok.
[14:59] <wm4> so that's why INT_MAX
[14:59] <durandal_1707> is sws_scale able to scale it on 32/64 ?
[15:00] <wm4> it does support RGBA64 if you mean that
[15:01] <durandal_1707> no, i mean is it able to scale down that png...
[15:01] <nevcairiel> at least the decode error went away, sadly my video player refuses ebcause it wants to allocate 3 or 4 image buffers already, which obviously doesnt work on 32-bit :p
[15:01] Action: nevcairiel builds 64-bit
[15:03] <durandal_1707> another nice feature would be decoding and displaying only part of image (if bitstream allows it)....
[15:03] <durandal_1707> it could be useful with tiled tiffs and exrs and others....
[15:05] <nevcairiel> ha on 64-bit it displays, but my players uses 6gb of memory :P
[15:05] <JEEBsv> :D
[15:06] <nevcairiel> so yeah the check is a bit overzealous
[15:06] <nevcairiel> sadly it doesnt have the correct information in that function to do it properly
[15:08] <wm4> my player manages to display hat png (after removing the check), but only with vo_x11 (the most crappy output)
[15:09] <nevcairiel> all my outputs try to allocate too many image buffers to hold in 32-bit
[15:09] <nevcairiel> so they bail
[15:09] <nevcairiel> but i'll worry about that once a video actually triggers that problem :p
[15:11] <durandal_1707> switch to 64bit and get min 1024GB of RAM ASAP
[15:53] <cone-300> ffmpeg.git 03Paul B Mahol 07master:cf5ab8b6f716: vmnc: make code independent of sizeof(AVFrame)
[15:53] <cone-300> ffmpeg.git 03Paul B Mahol 07master:94372592767f: vmnc: check return values of reallocation
[15:53] <cone-300> ffmpeg.git 03Paul B Mahol 07master:3838637bbb14: yop: check return value of av_frame_alloc()
[16:05] <cone-300> ffmpeg.git 03Paul B Mahol 07master:2868035d951b: pcxenc: get rid of unused PCXContext
[17:04] <durandal_1707> ahh random failures
[17:07] <cone-300> ffmpeg.git 03Paul B Mahol 07master:a4b899f588bd: lavfi/gradfun: remove nv21/nv12 as they are not supported
[17:28] <saste> ffprobe writers are painful
[17:29] <cone-300> ffmpeg.git 03Paul B Mahol 07master:0addc8288045: avdevice/x11grab: use AV_OPT_TYPE_VIDEO_RATE
[17:29] <cone-300> ffmpeg.git 03Paul B Mahol 07master:2634af575c2d: avdevice/fbdev: use AV_OPT_TYPE_VIDEO_RATE
[17:29] <cone-300> ffmpeg.git 03Paul B Mahol 07master:7ac736afbd18: lavfi/frei0r: use AV_OPT_TYPE_VIDEO_RATE
[17:34] <durandal_1707> saste: how?
[17:35] <wm4> saste: how does an application handle libavfilter EOF correctly? calling av_buffersink_get_frame() in a loop until no more data is available doesn't seem to do the right thing
[17:46] <nevcairiel> wm4: i flush the graph with a NULL frame for av_buffersrc_write_frame and then poll get_frame until it doesnt give me anything anymore, seems to work
[17:49] <wm4> just av_buffersrc_write_frame(s, NULL)?
[17:49] <nevcairiel> yes
[17:50] <wm4> ok thanks
[17:50] <wm4> of course this is not documented
[17:50] <nevcairiel> i think i read the code to figure this out
[18:39] <durandal_1707> descriptions of frei0r filters is really obscure
[19:17] <cone-300> ffmpeg.git 03Paul B Mahol 07master:be7d6710b45c: lavfi/frei0r: use AV_OPT_TYPE_IMAGE_SIZE
[20:24] <cone-300> ffmpeg.git 03Paul B Mahol 07master:fdf9296df7f5: sgienc: remove unused SgiContext
[20:33] <durandal_1707> michaelni: mpegvideo.c still calls assert
[20:35] <cone-300> ffmpeg.git 03Paul B Mahol 07master:1fa0284c57fd: qcelpdec: change asserts to av_asserts
[22:38] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:c88bbc01eb7f: avcodec/mpegvideo: replace assert about interlaced EC by debug av_log()
[22:38] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:205f423e28bc: libavcodec/mpegvideo: change some asserts to av_assert0()
[22:41] <wm4> so... why NIH assert()?
[22:51] <michaelni> wm4, is there an existing system that could be used ?
[22:52] <michaelni> some asserts are in speed critical code, some are checking conditions that the author really wants to be checked because failure may have security implications and theres no speed effect of te extra check
[22:53] <michaelni> so more than just 1 assert function is needed
[22:55] <wm4> what's the security issue?
[22:57] <michaelni> wm4, i dont understand the question
[22:58] <wm4> also, any reason why vf_scale uses AVFrame.color_range but not AVFrame.colorspace?
[22:58] <wm4> well you said "failure may have security implications"
[22:58] <wm4> if the assert is not compiled?
[22:58] <michaelni> well an example of a security related assert would be "av_assert0(pics <= MAX_DELAYED_PIC_COUNT);"
[22:59] <michaelni> you would write out of array if that fails and isnt stoped there
[22:59] <wm4> ok...
[23:01] <michaelni> vf_scale does use AVFrame.colorspace
[23:01] <michaelni> inv_table = parse_yuv_type(scale->in_color_matrix, av_frame_get_colorspace(in));
[23:03] <wm4> hm, maybe it doesn't work for me
[23:04] <michaelni> you have to specify "auto" as colorspace, which i wanted to make default after more tests
[23:06] <wm4> ok it does work with auto
[23:06] <wm4> why does vf_scale has its own array of coefficients instead of using the swscale functions?
[23:14] <cone-300> ffmpeg.git 03Michael Niedermayer 07master:f3f4e133d6a0: avfilter/vf_scale: use sws_getCoefficients()
[23:26] <durandal_1707> checking pointer that points out of array is safe or not?
[23:42] <durandal_1707> do you want to review my cosmetics patches?
[23:44] <wm4> durandal_1707: AFAIK out of bound pointers are never safe... unless they point right at the end of the valid range
[23:46] <durandal_1707> i have 10 patches here
[00:00] --- Sat Jul 27 2013
1
0
[00:12] <salamanderrake> Ok I got it to work but now its just black
[00:14] <salamanderrake> terminal output http://pastebin.com/tTAHg7eq
[00:14] <salamanderrake> command http://bpaste.net/show/117407/
[00:14] <llogan> add -pix_fmt yuv420p as an output option
[00:16] <salamanderrake> thanks that helped
[00:17] <llogan> you should suggest that to the script maker
[00:30] <salamanderrake> I did
[00:30] <salamanderrake> still need to adjust the quality because of my isp
[00:31] <brontosaurusrex> salamanderrake, i don't think you should use crf and b:v in the same sentence
[00:32] <brontosaurusrex> its either one or another
[00:33] <salamanderrake> crf?
[00:34] <klaxa> i uh... salamanderrake, i would change your twitch-tv api key now
[00:34] <klaxa> since you pasted it publicly on pastebin
[00:34] <brontosaurusrex> salamanderrake, http://bpaste.net/show/117407/ , line 21?
[00:34] <salamanderrake> I know
[00:34] <klaxa> and it expires never
[00:34] <salamanderrake> how do I change it?
[00:34] <klaxa> there's some option for it, let me find it
[00:37] <salamanderrake> I'm good now
[00:38] <klaxa> did you find where to change it?
[00:53] <salamanderrake> klaxa, yah
[00:54] <salamanderrake> but now the script is broke again. lol
[01:25] <salamanderrake> the problem this time was positional
[08:16] <posciak> would anyone be able t ohelp me with open gop setting please? I can't seem to be able to produce a stream with SEI recovery instead of IDRs, which I think this is for.
[08:16] <posciak> I tried: ffmpeg -i in.mp4 -c:v libx264 -x264opts open_gop=1 -c:a copy out.mp4 and some variations of open_gop argument, but x264 keeps reporting "open_gop=0"... what am I doing wrong?
[09:09] <posciak> l
[10:42] <Belliash> hello
[10:44] <Belliash> id like to ask whats wrong with ffplay, or what do i do incorrectly?
[10:44] <Belliash> when i try to play a stream using rtmpdump, its playing fine using this command: rtmpdump -r rtmp://95.211.186.67:1936/live/nick?token=gxqZr6H70ga1UQ5HOMoRrA -W http://goodcast.tv/jwplayer/player.swf -p http://goodcast.tv
[10:45] <Belliash> but when i try to use ffplay for this, then i got error... here is the command: ffplay "rtmp://95.211.186.67:1936/live/nick?token=gxqZr6H70ga1UQ5HOMoRrA swfUrl=http://goodcast.tv/jwplayer/player.swf pageUrl=http://goodcast.tv"
[10:45] <Belliash> and here is the output: http://wklej.org/hash/92fa8d131ba/
[11:04] <Belliash> http://wklej.org/hash/1785aaa92f5/
[11:12] <pyBlob> http://pastebin.com/BJfc2bMf
[11:13] <pyBlob> getting a "operation not permitted", why?
[11:15] <spaam> you are not allowed to read that file..
[11:16] <pyBlob> can't think of having set up a read-restriction
[11:18] <spaam> its something with ffprobe.
[11:18] <spaam> not your file :)
[11:18] <spaam> i guess it just show the wrong error code.
[11:19] <pyBlob> what was the option to make ffprobe more verbose?
[11:20] <praveenmarkandu> hi. for some reasons my hls segments are not the same from .ts file to ts file
[11:31] <Belliash> any1?
[11:33] <saste> pyBlob, report on trac
[11:34] <saste> what the file is supposed to contain? what is "srf"?
[11:34] <saste> i mean in terms of stream formats
[11:34] <pyBlob> saste: it's probably because the file is encrypted or something like that ^^
[11:34] <saste> pyBlob, a return value of -1 is weird anyway
[11:35] <saste> especially since ffmpeg gives no indication of what's wrong with the file
[11:35] <pyBlob> it's the recording of my new samsung-tv
[11:36] <saste> pyBlob, create a ticket with the sample, at least we can improve the feedback
[11:36] <pyBlob> ... but I already thought that it isn't as easy as giving the file to ffmpeg/ffprobe
[11:37] <pyBlob> the sample is ~1.4GB ... curently I don't have a smaller one
[11:37] <saste> you can cut only the first few megs
[11:38] <pyBlob> I'll do that, when I've got time for that
[11:39] <pyBlob> ... the format itself seems to be an mpeg-transport-stream
[13:44] <blippyp> Hi everyone - I'm trying to join two video clips (basically, one is a background 'video' image and the other is smaller video centered on the screen) - I can do this fine, but I'm trying to set the top video (the smaller one centered on the screen) to have transparency... I can use the blend filter and what not, but I'm basically looking for a way to 'blend' the two images with the top one having a specified % of opacity - Anyone know h
[13:46] <blippyp> I haven't tried the fade filter - but my guess is that it's something like that, only without a fade in/out time - I want the fade to be permanent and constant...
[13:53] <blippyp> I think that the blend filter is what I want to use - but I can't figure out how to use the all_opacity option - so that I can set the alpha bit...???
[14:51] <zap0> how do you say "don't copy the meta data to the result file" >
[14:51] <zap0> ?
[15:04] <zap0> how do you create a .wav file?
[15:04] <durandal_1707> !doc
[15:07] <zap0> can anyone help?
[15:08] <blippyp> zap0: I don't think it does copy metadata by default - what command are you using?
[15:09] <zap0> blippyp an mp3 to a wav file. it copied meta data. so i'm trying to go via raw PCM, but now i'm having trouble getting it to make a wav file, i say out.wav as filename, yet it makes raw pcm file.
[15:11] <blippyp> hmm, not sure about that - if you man ffmpeg and search for 'meta' you will see a way to 'clear' each meta tag, but you will have to specify each tag...
[15:13] <zap0> too complicated. i'll just convert to raw, then back to wav.
[15:13] <durandal_1707> so you are still not going to read documentation?
[15:19] <zap0> thanks blip. i think i got it now.
[15:22] <durandal_1707> there is switch to disable metadata and it is mentioned in documentation
[16:13] <blippyp> if anyone was curious about my problem, the solution I just found was with using the geq filter...
[16:13] <durandal_1707> blippyp: geq is generic and thus very slow....
[16:14] <blippyp> I'm all ears if you have a better solution - geq is working....
[16:14] <durandal_1707> and you can also use overlay filter
[16:14] <blippyp> how?
[16:14] <blippyp> overlay doesn't have the option for opacity???
[16:16] <durandal_1707> its opacity is derived from alpha plane
[16:16] <durandal_1707> and you can set opacity with blend filter ....
[16:16] <blippyp> yes - so how do you set that using the overlay filter?
[16:17] <durandal_1707> i already told you, you modify alpha plane....
[16:18] <durandal_1707> alpha plane specify opacity for each pixel
[16:18] <blippyp> I played with the blend filter - no matter what I did this image was only 'barely' visible - I couldn't get one to show more than the other...
[16:18] <blippyp> I need an example - I obviously don't know what you mean...
[16:19] <durandal_1707> i do not know what you are actually really doing
[16:19] <blippyp> blending two videos - with one showing thru more than the other...
[16:20] <blippyp> Basically - two videos, one as a backdrop - the other to be resized as I want with opacity so that I can still see the background video thru it (but only barely)
[16:20] <durandal_1707> that is what blend filter do
[16:20] <durandal_1707> so your need give more precise description
[16:21] <blippyp> blend would only allow me to use a number between 0 and 1 and no matter what I set it to, both images were barely visible... maybe I was using it wrong, but I don't think...
[16:21] <blippyp> think so...
[16:22] <durandal_1707> you can do that with crop: crop what images you want to blend (and how to blend) and than overlay that overlays blended image on top of bigger one....
[16:22] <blippyp> that's what I did - i cropped it out and blended it with geq
[16:22] <durandal_1707> you can't blend wit geq...
[16:22] <blippyp> yes you can - I just did it
[16:22] <durandal_1707> geq can only "blend" with color
[16:23] <blippyp> there ya go...
[16:23] <durandal_1707> so you just blend something with background color, that is not much ....
[16:24] <blippyp> blend doesn't work right
[16:24] <durandal_1707> really?
[16:24] <blippyp> I literally tried pretty much everything...
[16:24] <blippyp> seriously
[16:24] <blippyp> both images were extremely transparent - I couldn't get one to be the 'parent' and show thru more
[16:25] <durandal_1707> then you did not cropped at all
[16:25] <blippyp> although I didn't play around with all_expr too much, perhaps that would have helped...
[16:25] <blippyp> not with blend I didn't
[16:25] <blippyp> maybe that was the 'key'
[16:25] <durandal_1707> also blend filter have expressions too, so you can do same as with geq, but much more than blending just static color
[16:27] <durandal_1707> but using combination of crop,blend and overlay may be faster
[16:27] <blippyp> brb, my kids up - I gotta go change here diaper and give her a bottle...
[16:31] <salamanderrake> the error trying to capture output from a window to put in a webm file http://pastebin.com/BBz0aMNT the script http://bpaste.net/show/117599/
[16:31] <salamanderrake> is the -s flag for video out a scale option?
[16:31] <pyBlob> I'm searching for an option that tells ffmpeg to start decoding at a specific frame ... but "-ss" or the other options seem to accept a timestamp only, not frames
[16:33] <zap0> pyBlob, what is a 'frame' ?
[16:34] <pyBlob> it's one image of the video
[16:34] <zap0> what is an image of the video?
[16:34] <durandal_1707> salamanderrake: that error is nonsense - there should not be such error at all....
[16:35] <salamanderrake> durandal_1707, really
[16:35] <pyBlob> zap0: so how would you call it?
[16:35] <salamanderrake> would it be a libvpx limitation?
[16:35] <zap0> pyBlob, call what?
[16:35] <salamanderrake> an image of a video is a frame
[16:36] <durandal_1707> you build it yourself?
[16:36] <salamanderrake> durandal_1707, gentoo
[16:36] <durandal_1707> salamanderrake: one thing i dont unerstand: why you scale to 1x480 image?
[16:36] <zap0> pyBlob, not being difficult. you really do need to think quite hard about this. the reason there is no 'frame', is because you define it as a point in time.
[16:37] <salamanderrake> thats what I got from here https://www.virag.si/2012/01/webm-web-video-encoding-tutorial-with-ffmpeg-0…
[16:37] <durandal_1707> salamanderrake: 1x480 is not watchable at all
[16:37] <salamanderrake> durandal_1707, what should I set that as?
[16:37] <salamanderrake> or drop it
[16:37] <durandal_1707> it is -1 not 1
[16:37] <pyBlob> zap0: that's the point ... I don't specify a framerate, so I also can't use the time to get the n-th frame of the video
[16:38] <salamanderrake> durandal_1707, so it should be -1:480 ?
[16:38] <durandal_1707> yes
[16:38] <durandal_1707> use your brain cells
[16:39] <salamanderrake> thanks
[16:39] <salamanderrake> and learn to read
[16:39] <salamanderrake> cause its right there
[16:39] <zap0> pyBlob, you can't get the "what?-th" of the video? you are going in circles cause you still think a 'frame' is something a video has.
[16:40] <blippyp> pyBlob: http://superuser.com/questions/459313/how-to-cut-at-exact-frames-using-ffmp…
[16:40] <pyBlob> -.-
[16:41] <blippyp> Also, you cannot copy, you MUST re-encode the video to do precision cuts...
[16:42] <zap0> pyBlob, do you `get it` yet?
[16:44] <pyBlob> that's what I've figured out too, so there is no straightforward way to do that, even if there is no audio in the video?
[16:45] <zap0> audio is irrelevant.
[16:45] <pyBlob> nice
[16:49] <salamanderrake> with the script I showed you and the fix I got no audio
[16:49] <pyBlob> with "-vframes" you can specify how many frames should be recorded, why isn't there something similar to specify the start-offset using a framecount?
[16:50] <pyBlob> theoretically I could use "-r 1 -ss <framecount>"?
[16:50] <durandal_1707> salamanderrake: that is probably not ffmpeg issue
[16:51] <salamanderrake> I get audio with other scripts
[16:51] <salamanderrake> if anything its vorbis
[16:52] <salamanderrake> the streaming script http://bpaste.net/show/117609/
[16:53] <blippyp> -ss num is the miliseconds - not framecount...
[16:54] <pyBlob> ooops ... so I have to set -r to 1000
[16:55] <blippyp> -r is your framerate, you don't want it at 1000
[16:56] <salamanderrake> and the video is like altra fast
[16:57] <durandal_1707> salamanderrake: with rtmp or with webm?
[16:59] <salamanderrake> ohh sorry with webm
[16:59] <salamanderrake> I changed the fps and it made it wors
[16:59] <salamanderrake> *worse
[17:01] <durandal_1707> salamanderrake: set -framerate for x11grab
[17:01] <durandal_1707> default one is ntsc
[17:01] <salamanderrake> oh ok
[17:01] <durandal_1707> so if you record at ntsc and than trascode to !ntsc funny things happen
[17:02] <salamanderrake> what about this -quality option I can not find it in the wiki
[17:02] <salamanderrake> sorry this is the webm wiki not the ffmpeg wiki http://wiki.webmproject.org/ffmpeg
[17:03] <durandal_1707> look at libvpx wrapper documentation...
[17:08] <salamanderrake> what is the scale for 16:10 ?
[17:09] <salamanderrake> the -vf scale=-1:1.6
[17:09] <durandal_1707> salamanderrake: there is documetnation for scale filter
[17:10] <durandal_1707> where -1 thing is explained....
[17:10] <durandal_1707> and bunch of others nice features
[17:13] <salamanderrake> reading the wrong page then http://trac.ffmpeg.org/wiki/Scaling%20(resizing)%20with%20ffmpeg
[17:13] <durandal_1707> what is wiki
[17:14] <salamanderrake> https://ffmpeg.org/ffmpeg-filters.html#scale is what I am looking fore
[17:14] <salamanderrake> *for
[17:15] <salamanderrake> so I basicly don't need it unless I am resizing it
[17:16] <durandal_1707> yes
[17:29] <salamanderrake> I set the fps back to 25 fps for the x11grab and its still plays fast.
[17:32] <salamanderrake> here is the new script http://bpaste.net/show/117623/
[17:48] <durandal_1707> salamanderrake: you still do not set framerate at wich you record
[17:49] <durandal_1707> default framerate at which x11grab records is ~30
[17:49] <durandal_1707> reported as 29.97
[17:50] <durandal_1707> i already said that -framerate cotrols fps at which x11grab records
[17:50] <durandal_1707> and same one is already explained in, guess where, documentation
[18:02] <gerion> https://github.com/OpenHEVC/openHEVC just discovered, is this known? maybe this could be used to implement hevc nativly in ffmpeg
[18:03] <gerion> openHEVC is an open source compliant HEVC video decoder created as a fork of Libav for research purposes
[18:03] <JEEBsv> gerion: openhevc's decoder is smarter's :P
[18:04] <JEEBsv> they just hacked on it :D
[18:04] <JEEBsv> and smarter is on #ffmpeg-devel and #libav-devel
[18:04] <JEEBsv> so yes, it is known
[18:04] <gerion> JEEBsv: ah ok
[18:04] <JEEBsv> the problem with openhevc folk was that they basically only cared about their herpy derpy API
[18:04] <JEEBsv> thus breaking compatibility with normal libavcodec in all ways possible
[18:05] <JEEBsv> lately they finally started backporting their changes into proper libavcode
[18:05] <JEEBsv> *libavcodec
[18:05] <JEEBsv> (the libav repo under openhevc)
[18:07] <gerion> nice to know, that there are efforts ;)
[18:21] <salamanderrake> thanks I missunderstood what you said earlyer
[18:22] <salamanderrake> oh and there is audio now.
[18:42] <sekh> hi
[18:42] <sekh> can you help me, I'm trying to use the alsa output of ffmpeg
[18:43] <sekh> I'm not sure of how to use it
[18:43] <sekh> ffmpeg -f lavfi -i aevalsrc="0::duration=10" -f alsa -
[18:44] <sekh> thanks, but I already read it
[18:47] <sekh> https://pastee.org/dmb2h
[18:51] <pfloyd> When I use -vf "transpose=1" the resulting video is all black. full command line is: ffmpeg -y -i in.mp4 -vf "transpose=1" out.mp4 and this is the latest ffmpeg as of a few mins ago and a fresh git clone (ffmpeg version N-54954-g7ac736a)
[19:00] <durandal_1707> pfloyd: what you use to play?
[19:00] <pfloyd> it's just windows media player. I suppose I could try vlc
[19:02] <pfloyd> it does play in vlc, hmm
[19:09] <sekh> I finally found how to do it, but it's not clear in the documentation http://ffmpeg.org/ffmpeg-devices.html#alsa :-)
[19:10] <sekh> maybe just a link to the input alsa format can be useful
[19:10] <sekh> (about the syntax)
[23:05] <linemimic> I made a simple script for streaming a video file with ffmpeg or avconv: http://pastebin.com/4ypbkNDh3
[23:29] <Fjorgynn> Dang!
[23:29] <Fjorgynn> linemimic: :)
[23:31] <linemimic> :)
[00:00] --- Sat Jul 27 2013
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