[Ffmpeg-devel-irc] ffmpeg.log.20130702

[burek]

[02:38] <BrianNano> I have a question about hard coding subtitles.  If I don't have a version of ffmpeg with libass enabled, is my best approach to use mencoder, dump the file into an avi with hardcoded subs, and then use ffmpeg to make a mp4.  I'm concerned about the b_frames in the mkv that I'm converting.
[02:39] <klaxa> woah wait what? before converting 20 times, either compile ffmpeg yourself, or grab a static link
[02:40] <BrianNano> thanks, of course, to make things super difficult, I'm on obsd.
[02:41] <klaxa> oh haha...
[02:41] <BrianNano> I asked the port maintainer to add libass support to the next updated package, but if that doesn't work, I'll start hacking on the port.  I just want a solution that is reasonable in the mean time.
[02:41] <klaxa> mmhh... i think compiling yourself might be the best approach, if you can't do that, maybe use mplayer for decoding and applying video filters
[02:42] <klaxa> BrianNano: http://klaxa1337.blogspot.de/2012/09/streaming-videos-over-http-while.html
[02:42] <klaxa> that's totally horrid and i wouldn't do it like that anymore
[02:42] <klaxa> but the piping i used might be useful for you
[02:43] <klaxa> what it does is decode with mplayer2, apply subtitles, output as yuv4mpeg to a pipe, which then gets read by ffmpeg as yuv4mpeg input then encoded to whatever
[02:44] <BrianNano> thanks for the link
[02:45] <BrianNano> I just have a bunch of mkvs that I'd like to stream over rygel to my ps3. Yeah, I know, I should probably just get a media player for my TV, but it's just convenient
[02:46] <klaxa> you can play mkv files with the ps3 afaik
[02:46] <BrianNano> check out tmux as an alternative to screen: http://tmux.sourceforge.net/
[02:46] <klaxa> yeah i uh... tried out tmux
[02:46] <BrianNano> klaxa, yeah, rygel will transcode on the fly, but my subtitles go poof:)
[02:46] <klaxa> ah heh
[02:47] <BrianNano> and ass subtitles are so much nicer than srt; I tried converting to srt for rygel, but it doesn't look good at all since srt is super limited
[02:47] <klaxa> mhm
[02:48] <klaxa> well i guess compiling ffmpeg from source might be the "cleanest" solution
[02:49] <BrianNano> yeah, I agree, but the existing port patches around configuration are ugly: http://www.openbsd.org/cgi-bin/cvsweb/ports/graphics/ffmpeg/patches/ ,so I'll have to hack on that.  Let me give you a pastebin of what I'm trying to do, and you can tell me if I'm crazy.
[02:50] <BrianNano> http://pastebin.com/fKhq1hxu
[02:51] <klaxa> can avi even contain h264 video?
[02:51] <klaxa> last command might need -c copy
[02:51] <BrianNano> yep, it just can't handle bframes
[02:51] <BrianNano> I changed the last to ffmpeg -i out.avi out.mp4; do I need a codec all copy?
[02:52] <BrianNano> now, my theory is that since I'm muxing to mp4, I should be good with the bframes set to two, but I don't really know for sure.
[02:52] <klaxa> i would do that, otherwise ffmpeg will encode it again
[02:53] <klaxa> i'm not too sure about mp4's limitations, but i think just about any h264 stream should be okay
[02:53] <BrianNano> the last piece is just muxing from avi to mp4
[02:53] <klaxa> no it's actually encoding
[02:53] <klaxa> if you don't specify to copy the codecs, ffmpeg will encode with default settings for the specified container
[02:54] <BrianNano> so is this better: ffmpeg -i out.avi -c copy out.mp4
[02:54] <klaxa> also, do you really want pcm audio? can't you just copy the audio stream at least?
[02:54] <klaxa> yes
[02:54] <BrianNano> I don't want pcm; I want to stick with acc
[02:55] <klaxa> mencoder <input file> -ass -sub subtitles.ass -ovc x264 -x264encopts subq=4:bframes=2:b_pyramid=normal:weight_b -oac pcm -o <avi output file>
[02:55] <klaxa> change -oac pcm to -oac copy then
[02:56] <BrianNano> Audio format 0x4134504d is incompatible with '-oac copy', please try '-oac pcm' instead or use '-fafmttag' to override it.
[02:57] <klaxa> haha what
[02:57] <klaxa> skip audio copying then
[02:57] <klaxa> and copy it from the original mkv
[02:58] <klaxa> to do that: ffmpeg -fix_sub_duration -i input.mkv -codec copy -map 0 -map -0:a out.mkv
[02:59] <klaxa> then: ffmpeg -fix_sub_duration -i out.avi -i input.mkv -map 0 -map 1 -c copy out.mp4
[03:01] <BrianNano> so I'm grabbing just the audio from the new out.mkv with map?
[03:04] <klaxa> oh wait...
[03:04] <klaxa> ffmpeg -fix_sub_duration -i out.avi -i input.mkv -map 0 -map 1:a -c copy out.mp4
[03:04] <klaxa> like that
[03:05] <klaxa> first you are mapping everything, BUT audio
[03:05] <klaxa> then you use mencoder to encode the video + subtitles
[03:05] <klaxa> then you mux the video with the original audio
[03:06] <Lyude> hi klaxa
[03:06] <klaxa> hi Lyude
[03:11] <BrianNano> so I create a mkv with only audio and subtitles first?
[03:12] <klaxa> no you are creating an mkv with only video and subtitles
[03:12] <klaxa> we're not touching the audio
[03:12] <klaxa> with -map -<input number>[:stream specifier] you unmap streams
[03:12] <klaxa> as opposed to -map <input number>[:stream specifier]
[03:14] <BrianNano> now I'm really confused on map
[03:14] <klaxa> if you put a "-" before the thing you are mapping, you unmap it
[03:14] <klaxa> http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option
[03:16] <BrianNano> reading it now
[03:18] <BrianNano> great, I have 21 streams because of all the fonts for ass subtitles
[03:20] <klaxa> heh
[03:21] <klaxa> also, why exactly are you applying -fix_sub_duration?
[03:21] <klaxa> .ass is sophisticated enough to not mess up timing
[03:23] <BrianNano> early efforts of getting this work, I ended up with soft subs in a mp4, but to get that to work, I had to fix sub duration. I found the fix via googling a lot, but then I remembered that rygel can't handle soft subs that are ass.
[03:23] <BrianNano> https://ffmpeg.org/trac/ffmpeg/ticket/2212
[03:24] <BrianNano> so I have 21 streams, and I want all but one, is there a way to do that with out a very, very long commandline of mappings?
[03:24] <klaxa> is that bug relevant though?
[03:24] <BrianNano> my ffmpeg is from March
[03:25] <klaxa> i just provided it: ffmpeg -i out.avi -i input.mkv -map 0 -map -1:a -c copy out.mp4
[03:25] <klaxa> ah wait
[03:25] <klaxa> no
[03:25] <klaxa> ffmpeg -i input.mkv -codec copy -map 0 -map -0:a out.mkv
[03:25] <klaxa> that one
[03:25] <klaxa> map everything from 0 (-map 0), unmap all audio streams from 0 (-map -0:a)
[03:38] <BrianNano> so like this: http://pastebin.com/KmdHbdFT
[03:39] <BrianNano> this cleans up the names: http://pastebin.com/iRYyqtdV
[03:42] <klaxa> not exactly but we're close!
[03:42] <klaxa> you can combine the -fix_sub_duration and the unmapping of the audio, thus merging the first two ffmpeg commands
[03:43] <BrianNano> great
[03:43] <klaxa> you *have* to combine the last two ffmpeg commands because the first one's syntax is not entirely correct
[03:43] <klaxa> ffmpeg -i out.mkv -map 0 -map 1:a -c copy out_audio.mkv
[03:43] <klaxa> there is no input with id 1 ;)
[03:43] <klaxa> ffmpeg -i out_vs.avi -i input.mkv -map 0 -map 1:a -c copy out.mp4 will most likely work
[03:44] <klaxa> we are taking the original source and take the audio straight from there
[03:47] <BrianNano> gotcha
[03:47] <BrianNano> I'm running the long mencode part right now; I'll stop by later after going through these steps to let you know what happened.  *fingers crossed*
[03:48] <klaxa> good luck!
[03:48] <klaxa> depending on how long it takes i might already be asleep though
[03:48] <BrianNano> the file is only 24 minutes long, so it shouldn't take forever
[03:49] <relaxed> why not use ffmpeg's ass filter to hardsub?
[03:49] <BrianNano> because I have a version that doesn't have it enabled, and I'm not setup to hack on the port right now.
[03:50] <klaxa> openbsd madness
[03:50] <BrianNano> I asked the port maintainer to add it
[03:50] <BrianNano> but I don't have a local ports tree setup to hack on it, so I just needed a work around for the time being
[03:50] <relaxed> at least there's no remote hole in your ffmpeg for heck of a long time.
[03:50] <BrianNano> obsd is secure as long as you don't add software to it:)
[03:51] <BrianNano> and I've been using it for years, so I'm just used to it
[03:51] <relaxed> uou should learn to compile software on it locally :-P
[03:52] <relaxed> you*
[03:52] <BrianNano> I know how to; I just don't have space to pull down the ports tree locally
[03:53] <relaxed> Install the precompiled deps via pkg_add and grab ffmpeg's source.
[03:56] <BrianNano> it's not quite that simple; check out the current patches on ffmpeg: http://marc.info/?l=openbsd-ports&m=137125315310155&w=2
[03:57] <BrianNano> for now, I'm good; I already asked the ports maintainer to add support.  If that doesn't happen in the next package release for ffmpeg, then I'll setup a build box and start hacking on it.  But this is getting way off topic now.
[04:52] <cortexman> i'm using cinelerra which I guess is using ffmpeg, and when I open a video taken on my phone, I see this repeatedly on my console:
[04:52] <cortexman> [h264 @ 0x7f187ae76ae0]brainfart cropping not supported, this could look slightly wrong ...
[04:52] <cortexman> i know this isn't a cinelerra support channel.. but i'm curious what this means. i'm really curious as to how this could ever be a good error message also
[04:53] <cortexman> also the video is totally cropped and messed up. for some reason the software doesn't leave the aspect ratio alone.
[09:41] <mrAlmond>  /msg NickServ identify gianluca
[09:41] <mrAlmond> whoops
[10:41] <BrianNano> So I ended up having to rebuild my ffmpeg to support libass for hardsubs, and I tested it out and it works great on my PS3.  However, I have 50 files that I want to re-encode.  Does anybody have tips on how best to automate this process?  I'm comfortable using ksh, perl, and c.
[10:47] <BrianNano> let me clarify, I re-encoded 10bit MKVs to 8bit MP4 with hardsubs, and it worked great.  I even ended up with a smaller file size than the original mkv, and I used crf 18.
[10:49] <JEEB> BrianNano, keep resolution within whatever limits you want, level within 4.1 and keep a note on the refs (reference frame count) -- neither ffmpeg or libx264 will limit them to match the level you set.
[10:49] <JEEB> and that's more or less it
[10:49] <JEEB> keeping the reference frame count within the level limits is the hardest part of that, in my humble opinion
[10:51] <BrianNano> is there anyway to speed up adding the hard subs?  that was my bottleneck
[10:51] <relaxed> BrianNano: Use a for loop. I'm not sure of the syntax for ksh, but for bash it would be-  for i in *mkv; do ffmpeg -i "4i
[10:51] <relaxed> er, mistyped :/
[10:52] <JEEB> I don't think you can make the libass rendering/overlay any faster, and I'm surprised *that* is your bottleneck
[10:52] <BrianNano> so am I
[10:52] <relaxed> for i in *mkv; do ffmpeg -i "$i" <encoding options here> "${i%.*}".mp4; done
[10:52] <JEEB> usually it's the video encoder that's the bottleneck
[10:52] <JEEB> (in this case libx264)
[10:52] <BrianNano> I don't understand why it took longer for me to hardsub it than it did for me convert the 10bit to 8bit
[10:53] <BrianNano> yep
[10:53] <relaxed> Now you're doing both?
[10:54] <mrAlmond> Paranoialmaniac are you there?
[10:55] <BrianNano> here's a pastebin of what ended up working for me:  http://pastebin.com/drs4CuuZ
[10:56] <BrianNano> it's just manual
[10:57] <BrianNano> and I had to spend an hour figuring out the Makefile for my openBSD port of ffmpeg to enable libass, but now, I have a working patch I can submit for review tomorrow
[10:58] <relaxed> I don't see where you're downconverting to yuv420p
[10:59] <BrianNano> I can't play high10 h264 files on my ps3
[11:00] <JEEB> relaxed, default output colorspace for 10bit 4:2:0 and with 8bit libx264 is yuv420p
[11:01] <relaxed> Are you sure it wouldn't be  yuv420p10le or uv420p10be ?
[11:01] <relaxed> which is different than yuv420p
[11:02] <BrianNano> what is yuv420p?  any suggestions for what I can do better?  I'm new at this, and after fighting with mencoder all weekend, I'm liking ffmpeg a lot now.
[11:02] <JEEB> you shouldn't use mencoder in general
[11:02] <JEEB> relaxed, you can't output that with a standard 8bit libx264
[11:02] <BrianNano> I learned that the hard way
[11:02] <JEEB> also it wouldn't play on you-know-what
[11:03] <JEEB> because the 10bit AVC/H.264 hw decoders cost four digit sums at the very least
[11:03] <JEEB> yuv420p10le would be the default for *10bit* libx264
[11:03] <JEEB> naturally
[11:08] <BrianNano> so my subtitles are encoding at about 8fps, but my video encoded around 16fps.  Is this because of the aac audio that I had to use -strict -2 for?
[11:21] <mrAlmond> JEEB : About the issue I'm having with that MP4...using "mediainfo -f" it says codec is avc1 and the working one says codec is "AVC" what's the difference?
[11:21] <JEEB> just read 14496-15 and make sure your AVCC creation is correct :P
[11:23] <mrAlmond> ook
[11:23] <mrAlmond> so that means that my avcc creation is not perfect?
[11:24] <JEEB> not necessarily, but that's the first thing I remember from your yesterday's thing
[11:25] <JEEB> mediainfo is a crapshoot of a library/app, but if it can't find your profile/level you're doing something wrong
[11:29] <mrAlmond> ok tnx
[11:30] <JEEB> 14496-15 should be your primary thing that you're following
[11:31] <JEEB> but if there's something in that that you'd like an alternative implementation of visible, try L-SMASH's source code or so
[11:51] <makara> hi. I'm encoding a guy speaking. Its a lecture. The background doesn't change and the camera doesn't move, but in the encoded file the pixels for the background are flying around much better. I'm concerned this is adding unnecessary weight to the file. When I encode at RF 25 it happens a lot less, but then I want to use RF20 so that his face/body is clear. What should I do?
[11:55] <zap0> build a mask.
[11:55] <zap0> no 2D algo understands what a 'background' is.
[11:56] <zap0> can you tell it what a background is?     by capturing a reference background when the speaker is out of shot
[12:42] <makara> zap0, no
[12:44] <makara> ok, but the thing is, it doesn't sparkle in the MTS original which I understand is already in h264, so why should it come to life in the resultant?
[13:38] <makara> helloo?
[13:43] <paprika> Hi
[13:43] <paprika> I was wonder how to set the muxrate for a ts using this example https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/muxing.c
[13:43] <paprika> *wondering
[13:44] <paprika> I've tried passing av_dict_set(&dict, "muxrate", "20000000", 0); into avformat_write_header
[13:44] <paprika> but I get a dts < pcr warning
[13:59] <t4nk832> Hi! I need to reduce the frame rate of a network video stream but avoid frame duplication due to network delays. How can I do this?
[14:08] <zap0> where do i get a list of the  depth sizes?     s16le  etc..  what is u8 ?
[14:12] <zap0> nevermind its just u8
[14:13] <durandal_1707> u8 is unsinged char
[14:18] <zap0> yeah, thanks.  i know.  i meant what is the symobl for unsigned char.
[14:18] <zap0> its 'u8'
[14:48] <shocksock> What does av_opt_set(codec_context->priv_data, "muxrate", "20000000", 0); do? What is a codecs muxrate?
[14:49] <JEEB> a muxrate is the rate of the output container, aka padding etc. possibly added to make the mux be of a certain rate
[14:49] <JEEB> I think pretty much only MPEG-TS and maybe PS has that
[14:50] <shocksock> So in order to specify it I would have to apply the above command to each codec context?
[14:51] <bencc> encoding/decoding a video frame takes much longer than an audio frame?
[14:52] <JEEB> well, audio generally tends to be simpler and have less data in a single "frame" :P
[14:55] <bencc> JEEB: can I assume that audio takes roughly 1/10 to encode or decode?
[14:55] <JEEB> nfi
[14:56] <bencc> ok
[16:01] <bunniefoofoo> When should avcodec_free_frame() be used? It doesn't seem to free much. Should I always use av_frame_free ?
[16:24] <bunniefoofoo> I think I may have found a memory leak with avio's dyn_buf. I've looked at ffmpeg source and it seems like everything is being freed. http://pastebin.com/6czZyhx9
[16:31] <t4nk188> Hi, I got a qestion regarding the muximing of AAC into MP4 via FFmpeg
[16:31] <durandal_1707> so what?
[16:32] <t4nk188> How does FFmpeg decide how many AAC frames or "samples" (in MP4 jargon) get muxed into one "chunk" ???
[16:32] <t4nk188> I see no pattern
[16:33] <t4nk188> Ideally, I would like to force each sample into its own chunk
[16:33] <t4nk188> if *all* sampels go into a single chunk, that would be okay too
[16:33] <t4nk188> Any ideas?
[16:34] <t4nk188> BTW: I already tryied -chunk_size and -chunk_duration, but they have no effect :-(
[16:37] <t4nk188> So how can I set the chunk size in ffmpeg? Is there any way?
[16:39] <b_jonas> t4nk188: what is the full command you are using?
[16:39] <t4nk188> ffmpeg -i input.wav -c:a libfaac -b:a 128k output.m4a
[16:40] <klaxa> i would like to advise you to use libfdk-aac https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide
[16:42] <t4nk188> does it have an option to define the numbe of samples per Chunk?
[16:42] <t4nk188> Shouldn't that be a *muxer* option?
[16:42] <durandal_1707> nope
[16:42] <durandal_1707> why you need exact number of samples per "chunk"?
[16:42] <t4nk188> sorry, "nope" what?
[16:43] <durandal_1707> google it
[16:43] <t4nk188> I need to create multiple files and they need to have the *identical* chunk layout
[16:43] <t4nk188> I googled for hours, but no help
[16:43] <durandal_1707> what is "chunk layout" ?
[16:44] <t4nk188> file #1 and file #2 need to have the exactly same number of samples in each chunk
[16:44] <t4nk188> I mean, the first chunk of file #1 needs to have the same number of chunks as the first chunk of file #2
[16:44] <t4nk188> And so on...
[16:44] <durandal_1707> define "chunk"
[16:44] <t4nk188> chunk as in MP4
[16:45] <t4nk188> MP4 container stores "samples" and groups them to "chunks"
[16:45] <durandal_1707> i have no idea what are you talking about
[16:45] <t4nk188> that is basic MP4
[16:45] <t4nk188> MP4 stores data as "samples"
[16:45] <t4nk188> for each "sample" we have a size
[16:45] <t4nk188> but we do NOT have an offset
[16:45] <t4nk188> Offsets are given for "chunks"
[16:46] <t4nk188> also there is a table that maps "samples" to "chunks"
[16:46] <t4nk188> you can find all that in the MP4/ISOBaseMedia specs
[16:47] <durandal_1707> usually there are different "chunks" and each have distinct name and function
[16:47] <t4nk188> Now, when I check the M4A file that FFmpeg is creating, I see the number of samplesPerChunk is fluctuating
[16:47] <t4nk188> I have audio-only files, so there only is a single (audio) track. And thus there only are audio chunks
[16:48] <durandal_1707> what codec?
[16:48] <durandal_1707> are codecs alwayws same? as number of channels and sample rate?
[16:49] <zap0> anyone know about AAC vs MP3 audio encoding?
[16:49] <t4nk188> AAC, encoded with faac
[16:49] <t4nk188> Yes, all Stereo @ 44 KHz
[16:50] <t4nk188> Please see here:
[16:50] <t4nk188> http://i.imgur.com/9pLHDrK.png
[16:51] <t4nk188> As you can see, "samplesPerChunk" is always different :-O
[16:51] <t4nk188> But what is the reasoning behind that ???
[16:52] <t4nk188> And how can I force my own settings, if possible at all ???
[16:53] <t4nk188> Is this something to ask in the "dev" channel?
[16:56] <t4nk188> Nobody knows an answer?
[16:58] <durandal_1707> i see nowhere in code that this can be controlled
[16:58] <durandal_1707> and i don't know if its allowed at all
[16:58] <durandal_1707> why you need it anyway?
[16:58] <Paranoialmaniac> if constant sampling rate, -movflags frag_duration?
[16:59] <Paranoialmaniac> oops
[16:59] <samguyjones> I'm trying to extract EXIF data from PNGs and GIFs.  Does anyone know if that's something ffmpeg can do?
[16:59] <Paranoialmaniac> this is fragment thing
[16:59] <durandal_1707> t4nk188: anyway feel free to ask on -devel and/or open bug report
[16:59] <Paranoialmaniac> it seems ffmpeg libavformat has no option to specify samples per chunk
[17:00] <t4nk188> It's for a project I work on
[17:00] <t4nk188> I cannot go too much into detail, but we need the same number of samples per chunk in different files
[17:00] <t4nk188> Ideally, each sample would get its own chunk
[17:00] <t4nk188> The overhead would be okay
[17:01] <Paranoialmaniac> how about the last chunk in a file?
[17:01] <t4nk188> what is with the last chunk?
[17:01] <Paranoialmaniac> the same number of samples per chunk <
[17:01] <t4nk188> the last one too
[17:02] <durandal_1707> samguyjones: nope, EXIF extraction of any kind and any container is not implemented, if you need it open feature request
[17:03] <Paranoialmaniac> t4nk188: even if non-mod samples per chunk in the source material?
[17:04] <t4nk188> Well, if we force a chunk-size of N and the total sample count is not mod-N, the last chunk would be smaller than N
[17:04] <t4nk188> But as long as that is consistent between different files, it would be okay
[17:04] <t4nk188> and if N=1, there is no problem anyway
[17:04] <samguyjones> durandal:  Thanks.  I'm checking because another developer suggested it.  Might it be something the dev team would do, or is it too far off-mission?
[17:05] <durandal_1707> samguyjones: there is nothing off-mission, people works on what they have time/motivation
[17:05] <durandal_1707> if there is something not done, it means nobody yet come with patch
[17:06] <durandal_1707> usually because nobody cares
[17:06] <durandal_1707> and user either as they do not push for it
[17:58] <mark4o> t4nk188: ffmpeg does not do it but you can rechunk it with mp4box after encoding.  e.g. mp4box -inter 500 file.mp4  will rechunk it into 500ms chunks
[17:59] <mark4o> or .m4a - works fine with just audio
[18:04] <grr> When I use ffmpeg on Windows and it runs into a lot of errors during transcoding, it almost freezes the whole system for me, making it hard to impossible to kill the process. How can I work around that, for instance, is there an option to make it quit after encountering too many errors?
[18:37] <grr> found http://superuser.com/questions/588147/can-i-make-ffmpeg-stop-if-integrity-check-encounters-an-error
[22:31] <Macey> Hi All,
[22:31] <Macey> -i dd-htngu.mp4  -ss 00:00:00 -t 00:03:45  -acodec mp2 -b:a 128k -vcodec mpeg4 -b:v 3500k -r 25 -s 1024x768 -aspect 16:9 -flags +aic+mv4 -vf "drawtext=fontfile='C\:\\Windows\\Fonts\\arial.ttf': timecode='10\:00\:00\:00': r=25: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: fontsize=72 : box=1: boxcolor=0x00000000 at 1" -y ZX12345.mp4
[22:31] <Macey> is my command line... i'm not seeing errors but, the Burntin T/C is a good seconds behind
[22:31] <Macey> any idea?
[22:43] <Macey> *good 10 seconds behind
[22:59] <superware> I badly need to test the patchset http://ffmpeg.org/pipermail/ffmpeg-devel/2013-June/145286.html (and two following), can someone please help me build win32 avformat.dll so I can verify the patch works?
[22:59] <durandal_1707> what exactly you need?
[23:00] <superware> a compiled win32 avformat.dll with the above 3-parts patch (very small)
[23:01] <superware> durandal_1707: can you build win32?
[23:01] <durandal_1707> superware: no
[23:02] <durandal_1707> i was thinking i could help you building ffmpeg
[23:02] <durandal_1707> but looks like you are not interested in that
[23:03] <superware> I'm afraid I have no past experience with that, I've spend the last hour trying to build with MSVS but couldn't
[23:03] <superware> if you can guide me through the process, I'll really appreciate it
[23:08] <durandal_1707> superware: there is guide for building with MSVC
[23:08] <durandal_1707> there is documentation
[23:09] <durandal_1707> so read it and ask what you do not understand
[23:09] <durandal_1707> i cant waste my time for free giving you exact instruction by instruction
[23:09] <durandal_1707> even i'm now wasting my precious time replying to you
[23:13] <superware> I appreciate your intentions, but I'm afraid we both might waste our time just to check a single patchset. An alternative might be to provide someone with the "problematic" stream (udp unicast over the net) so we can verify the patchset.
[23:47] <superware> Mavrik: hi
[00:00] --- Wed Jul  3 2013