[Ffmpeg-devel-irc] ffmpeg.log.20160403

[burek]

[01:34:07 CEST] <varu-> having an interesting issue with processing an mpegts stream with ffmpeg
[01:35:08 CEST] <varu-> the source is an rtp udp stream & i get a ton of "max delay reached. need to consume packet" and RTP: missed packets
[01:35:29 CEST] <varu-> as well as constant reference errors, [h264 @ 0x55775b23cc20] Reference 2 >= 2
[01:35:29 CEST] <varu-> [h264 @ 0x55775b23cc20] error while decoding MB 37 2, bytestream 2127
[01:36:12 CEST] <varu-> this is with -vcodec libx264; if i do -vcodec copy, i only get the rtp missed packets
[01:36:58 CEST] <varu-> the result is a stream whose audio is fine, but video cycles 2-3 corruptly-decoded frames for a few seconds until a few new ones are captured, then the cycle repeats
[01:37:14 CEST] <varu-> packetloss is not an issue, this is on the same vlan
[01:38:50 CEST] <Ben321> I need some help with the msvideo1 encoder in FFMPEG.
[01:39:31 CEST] <Ben321> How do I set the quality to maximum. It uses minimum quality by default. -q:v 100 has no effect. -q:v 1 has no effect.
[01:39:46 CEST] <JEEB> see if it uses quantizer for anything
[01:40:05 CEST] <JEEB> libavcodec/encodername_enc.c
[01:40:28 CEST] <JEEB> libavcodec/msvideo1enc.c to be exact
[01:41:02 CEST] <Ben321> I don't think msvideo1 encoder uses quantizer, but it does have the ability to set quality, I just don't know how to do it in FFMPEG.
[01:41:19 CEST] <JEEB> well -q is not quality, it's quantizer :P
[01:41:45 CEST] <JEEB> and quality variable in the encoder seems to be internal
[01:42:28 CEST] <JEEB> ok, as far as I quickly could see
[01:42:33 CEST] <JEEB> it's pretty uncontrollable :P
[01:42:34 CEST] <Ben321_> But yes, the official Microsoft msvideo1 codec most certainly supports manual setting  of quality. I know, because I've used it a number of times in Windows video software (not FFMPEG) for encoding video in an AVI file.
[01:42:55 CEST] <TD-Linux> now THAT's an encoder
[01:43:14 CEST] <TD-Linux> it even has rdo!
[01:43:33 CEST] <Ben321_> So hopefully FFMPEG will support (if not now, then at some point in the future) the ability to set the quality, just like the official Microsoft msvideo1 codec does.
[01:43:46 CEST] <TD-Linux> I doubt anyone will write it unless you do
[01:43:51 CEST] <J_Darnley> A separate piece of software has different settings?  I am Socked(!)
[01:44:50 CEST] <furq> out of interest, what terrible crime did you commit which has resulted in you being forced to use msvideo1
[01:44:52 CEST] <JEEB> also whatever "quality" means in MS's encoder can mean something completely different than in libavcodec's internal "quality" variable in the encoder
[01:45:11 CEST] <JEEB> and yes, you can make a feature request to be able to set something
[01:45:22 CEST] <Ben321_> Well, msvideo1 is supposed to have the ability to set quality. I know it does, because the official codec for it from Microsoft has this ability. If FFMPEG were to correctly implement an msvideo1 encoder, then it would most certainly support a quality setting.
[01:45:25 CEST] <JEEB> but unfortunately I think nobody cares about the encoder
[01:46:14 CEST] <JEEB> you are still missing the point of definition of "quality". given a quick guess for the encoder it probably is something like the quantizer :P
[01:46:23 CEST] <JEEB> but in any case, you can make a feature request on trac
[01:46:42 CEST] <JEEB> but unless you either pay someone or do it yourself, I don't see anyone caring enough
[01:48:22 CEST] <Ben321> I didn't get forced to use it, I just wanted to use it for making some nostalgic video files, that would have all the artifacts of this codec (but not to the degree it does on low-quality settings) just like I used to do years ago. It has decent compression, and on highest quality setting (in the official MS codec) decodes to a decent quality image.
[01:49:42 CEST] <Ben321_> Why did I get spontaneously disconnected just now?
[01:49:56 CEST] <scoofy> why not
[01:50:00 CEST] <J_Darnley> Blame your ISP
[01:50:03 CEST] <scoofy> the funz of the internet
[01:50:13 CEST] <scoofy> or your wifi rooter is too far
[01:50:21 CEST] <Ben321_> And just in case my last post didn't make it onto the IRC, here it is again:
[01:50:24 CEST] <Ben321_> I didn't get forced to use it, I just wanted to use it for making some nostalgic video files, that would have all the artifacts of this codec (but not to the degree it does on low-quality settings) just like I used to do years ago. It has decent compression, and on highest quality setting (in the official MS codec) decodes to a decent quality image.
[01:50:54 CEST] <TD-Linux> Ben321_, if you can compile ffmpeg yourself, you can just twiddle the quality number in the code
[01:51:21 CEST] <TD-Linux> maybe you can even make improvements to make ffmpeg the best msvideo1 encoder ever :^)
[01:51:32 CEST] <varu-> http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2746 is very similar to what i'm seeing, as the ts stream renders perfectly fine in vlc
[01:54:49 CEST] <varu-> https://trac.ffmpeg.org/ticket/3405 this is basically what i'm seeing
[01:55:40 CEST] <Ben321> Connection just reset again, so I'm not sure if the last thing I typed actually got posted, so here it is again:
[01:55:44 CEST] <Ben321> According to this webpage https://ffmpeg.org/pipermail/ffmpeg-devel/2009-March/072802.html (which is titled "[FFmpeg-devel] [PATCH] MS Video 1 encoder, take 2") the feature I'm now suggesting already was patched into FFMPEG, and that post was back in 2009. The part of the title of that post "[FFmpeg-devel] [PATCH]" indicates that a developer patched this feature into the official FFMPEG...
[01:55:45 CEST] <Ben321> ...release. But now that feature isn't present, so I assume another developer got the bright idea to REMOVE that feature.
[01:56:40 CEST] <Ben321> Can you still read what I'm posting? Or are all my posts disappearing?
[01:57:03 CEST] <furq> it could just mean that the patch never made it into ffmpeg
[01:57:22 CEST] <furq> judging by the rest of that thread, it didn't
[01:57:58 CEST] <Ben321> I thought that that thread was a developer's own blog, where he stated what he was adding into FFMPEG. So it should be in there.
[01:58:37 CEST] <Ben321> Hello?
[01:58:52 CEST] <J_Darnley> Huh?  It is the ffmpeg devel mailing list.  You send patches there to be discussed and reviewed by others.
[01:59:09 CEST] <furq> https://ffmpeg.org/pipermail/ffmpeg-devel/2009-March/072861.html
[01:59:09 CEST] <J_Darnley> When they are approved they get commited.
[01:59:43 CEST] <J_Darnley> The thread: https://ffmpeg.org/pipermail/ffmpeg-devel/2009-March/thread.html#72802
[02:01:04 CEST] <Ben321_> J_Darnley, could you repeat everything up to your last post? My connection dropped again.
[02:01:19 CEST] <Ben321_> So I may have missed a BIG chunk of this conversation.
[02:01:19 CEST] <J_Darnley> Huh?  It is the ffmpeg devel mailing list.  You send patches there to be discussed and reviewed by others.
[02:01:29 CEST] <J_Darnley> When they are approved they get commited.
[02:01:36 CEST] <J_Darnley> EOF
[02:03:04 CEST] <Ben321_> Is their a source distribution of FFMPEG that is designed to be compiled in Microsoft Visual C++ 20xx?
[02:03:47 CEST] <Ben321> I have Visual Studio 2010 on my PC, and would be able to compile, if there's a distribution of the source code that includes the required project file for Visual Studio 2010.
[02:04:14 CEST] <J_Darnley> AFAIK the official source works with some MSVCs
[02:04:32 CEST] <furq> https://trac.ffmpeg.org/wiki/CompilationGuide/MSVC
[02:04:36 CEST] <furq> that's as close as you'll get
[02:04:51 CEST] <furq> you still need to use msys
[02:05:28 CEST] <Ben321> What's msys?
[02:05:56 CEST] <furq> it looks like the zeranoe dev builds include msvc static libraries, so you could use those
[02:07:30 CEST] <furq> or the import libraries, rather
[02:08:12 CEST] <furq> actually nvm i forgot what you were asking. you just want to compile ffmpeg don't you
[02:09:10 CEST] <Ben321> Unfortunately, it also looks like a crapload of external libraries are required (such as bzip2), and these aren't included in the source distribution, but rather are linked to separately on the zeranoe website.
[02:09:22 CEST] <furq> you don't need any of those libraries
[02:12:05 CEST] <Ben321> Of the prebuilt copies on the zeranoe website, there's Static, Shared, and Dev. I assume Static means it doesn't require any additional DLL files, while shared does require additional DLLs. But what's in the Dev version? I assume it has something for developers. But just what is the difference?
[02:12:26 CEST] <furq> the dev version contains the import libraries so you can link against the dlls
[02:12:52 CEST] <furq> those are no use if you want to compile a patched ffmpeg, though
[09:33:31 CEST] <zzz> I'm trying to decide what deinterlacing filter to use for video playback. I'm seeing bwdif, kerndeint, mcdeint, w3fdif, and yadif. Is there one of these generally regarded as highest quality?
[10:06:51 CEST] <hanshenrik> in files containing video and audio
[10:07:02 CEST] <hanshenrik> should stream 0:0 always video and 0:1 always be audio
[10:07:06 CEST] <hanshenrik> or do i have to check first?
[10:07:19 CEST] <hanshenrik> if i want to only select the video stream, eg -map 0:0
[10:10:36 CEST] <wiistriker> Hello!
[10:11:18 CEST] <wiistriker> trying to implement such thing: get stream from IP cam (rtp), get audio from microphone on local machine and send stream to rtmp
[10:12:25 CEST] <wiistriker> http://pastebin.com/qi1HQigP
[10:13:11 CEST] <wiistriker> it actually works, but i need to scale video stream from rtp
[10:13:13 CEST] <wiistriker> -vf scale=640x360
[10:14:02 CEST] <hanshenrik> yup, scaling dont work?
[10:15:04 CEST] <wiistriker> Option vf (set video filters) cannot be applied to input file audio=@device_cm_{33
[10:15:18 CEST] <wiistriker> it try to apply scale for audio stream
[10:23:10 CEST] <wiistriker> should i run two ffmpeg process?
[10:58:03 CEST] <c_14> hanshenrik: -map 0:a:0 -map 0:v:0 etc. 0:0 does not have to always be video
[10:58:19 CEST] <hanshenrik> oh thanks!
[10:58:36 CEST] <c_14> wiistriker: can you pastebin the command and output with the vf as well?
[11:01:30 CEST] <wiistriker> i just move -vf option after -f dhow - auido="..."
[11:01:32 CEST] <wiistriker> and it works
[11:02:13 CEST] <wiistriker> bad thing that my rtsp stream delay for about 3 sec but i get audio in realtime
[11:50:10 CEST] <momomo> anyone here today?
[11:58:42 CEST] <momomo> -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -reconnect_delay_max 2 -i url is not working
[11:58:47 CEST] <momomo> not recognized options
[11:58:49 CEST] <momomo> why ?
[12:01:12 CEST] <nichego> hello. i have dvd content with soft telecine. is it possible to produce a hard telecine video from this using ffmpeg?
[12:18:57 CEST] <durandal_1707> no
[12:19:24 CEST] <nichego> so no equivalent to mencoder's softpulldown filter?
[12:21:21 CEST] <durandal_1707> repeatfields maybe
[12:21:59 CEST] <nichego> good pointer. i'll test now
[12:25:23 CEST] <nichego> that seems to be doing the job, durandal_1707. many thanks!
[12:34:03 CEST] <nichego> goodbye
[13:40:00 CEST] <momomo> furq, i switched to using your suggested version of ffmpeg, static build .. but now I am getting: Decoder (codec dvb_teletext) not found for input stream #0:2
[13:40:13 CEST] <momomo> what could that be due to ?
[14:36:44 CEST] <Fyr> guys, is there a C(++) library for simple FFT? I need simple include to my project, just like: #include "fft.h" and usage like double * fft(double *)
[14:46:05 CEST] <J_Darnley> FFmpeg has a "simple" fft interface
[14:46:17 CEST] <J_Darnley> Other than that the common one people suggest is fftw
[14:46:46 CEST] <J_Darnley> And you need more than "double * fft(double *)" for a good fft
[14:47:06 CEST] <J_Darnley> (for starters you need a size argument)
[15:06:40 CEST] <effractur> Fyr: fftw
[15:06:47 CEST] <effractur> Fyr: http://www.fftw.org
[16:05:34 CEST] <karab_44> hello!
[16:06:06 CEST] <karab_44> How can I vspipe frames with ffmpeg to png?
[16:06:26 CEST] <karab_44> I'd like to keep as much original png properties as possible
[16:07:11 CEST] <karab_44> All I want to do is to process some png sequence and save as png sequence
[16:59:29 CEST] <squack998> hello
[16:59:41 CEST] <squack998> i would like to know how to stack commands
[17:00:04 CEST] <squack998> for example
[17:00:12 CEST] <squack998> ffmpeg -i "http://l3md.shahid.net/media/l3/2fda1d3fd7ab453cad983544e8ed70e4/a86fcc92c90d4c64bee9e4a2bf4d4e69/e64c8758a20d430e8f5f82cabc291773/nawaya_s01_e14.mpegts/playlist-f91888035ecaf2bd5c1ad2ee644e6ca9c3327393.m3u8" -c copy nawaya_s01_e14.ts
[17:00:33 CEST] <squack998> then it runs this ffmpeg -i "http://l3md.shahid.net/media/l3/2fda1d3fd7ab453cad983544e8ed70e4/560305293d15485aa1ff6f26b59bf881/4b7ecd99522241efb2eddeefd1dc04cf/nawaya_s01_e15.mpegts/playlist-0aa2830c410cee418496502efba4a2e843360f79.m3u8" -c copy nawaya_s01_e15.ts
[17:00:49 CEST] <squack998> how do i stack them?
[17:04:11 CEST] <bencoh> what do you mean by "stack"?
[17:04:45 CEST] <bencoh> it sounds like a shell script question
[17:09:43 CEST] <squack998> i mean do the first then followed by the second
[17:09:48 CEST] <squack998> and so on
[17:10:29 CEST] <squack998> is it possible
[17:10:32 CEST] <squack998> ?
[17:11:08 CEST] <squack998> cause right now i open several ffmpeg and paste same commands of each episode
[17:11:38 CEST] <squack998> if i can just run one command to grab ep 1 ep2 and so on easier
[17:59:49 CEST] <jfmcarreira> hey guys
[17:59:59 CEST] <jfmcarreira> are the GSoC application finished?
[18:11:10 CEST] <Filarius> hello, I need to fix issue with A/V sync on livestream recording
[18:11:41 CEST] <Filarius> Where is errors for recording, and or re-encoding what I recorded http://pastebin.com/jtYnTB58
[18:12:07 CEST] <flargon> Hi, I have a question about https://trac.ffmpeg.org/wiki/Encode/AAC#NativeFFmpegAACencoder
[18:12:40 CEST] <flargon> For AAC (vbr) it says "Effective range for -q:a is around 0.1-2."
[18:12:57 CEST] <Filarius> Sample of recoded https://www.dropbox.com/s/yrvlvy2nry1izwk/record.flv (issue at first 6 seconds. Webplayer do not show this issue, try PC player)
[18:13:00 CEST] <flargon> But each value gives the same bitrate
[20:33:43 CEST] <karab_44> guys I combined some vapoursynth on png 32bit RGBA image sequence and and I have output 64bit RGBA
[20:34:08 CEST] <karab_44> how to do that correctly? I want to keep original bitdepth
[20:42:10 CEST] <pzich> you want a 64bit video?
[20:44:12 CEST] <pzich> by 64bit do you mean 64 bits per channel, or 16 bits per channel, 64 bits per pixel?
[20:50:42 CEST] <scoofy> per pixel i guess
[20:50:53 CEST] <karab_44> pzich I have originally RGBA 32bit PNG
[20:51:11 CEST] <karab_44> after vapoursynth this PNG is RGBA 64bit
[20:51:28 CEST] <karab_44> and if I don't include alpha channel it's like RGB 48bit
[20:52:05 CEST] <karab_44> and it's originally around 1MB big and after processing it's like quadroople it's size
[20:52:23 CEST] <scoofy> definitely.
[20:53:26 CEST] <karab_44> why it's 16bit per channel?
[20:54:57 CEST] <scoofy> why not
[20:55:09 CEST] <karab_44> I want it to keep original preferences because there is no gain or loss on quality I want to keep 32bit RGBA
[20:55:22 CEST] <scoofy> sure converting it to 64 bit would mean no improvement
[20:55:30 CEST] <karab_44> there is no any
[20:57:01 CEST] <karab_44> what can impact on these settings?
[20:57:54 CEST] <karab_44> fft = core.fmtc.bitdepth(src, flt=True).fmtc.matrix(mat="601", col_fam=vs.YUV).fmtc.bitdepth(bits=8)
[20:57:54 CEST] <karab_44> input = core.fmtc.bitdepth(src, flt=True).fmtc.matrix(mat="601", col_fam=vs.YUV).fmtc.bitdepth(bits=16)
[20:58:31 CEST] <karab_44> this is first conversion to YUV css
[20:58:58 CEST] <karab_44> then I do some processing and after that I revert the conversion
[20:58:59 CEST] <karab_44> ret = core.fmtc.matrix (clip=ret, mat="601", col_fam=vs.RGB)
[20:58:59 CEST] <karab_44> ret = core.fmtc.bitdepth (clip=ret,flt=1, dmode=7)
[22:10:51 CEST] <dexikiix> Hi all. I have a video file that was corrupted, ran it through "video repair tool" and got an output of good video, but the audio cuts out every second or so, and runs longer than the video and timecode... I'm looking for a way to fix the corrupted bit of the audio, and bring it all in sync with the video, if possible. Any tips?
[22:11:11 CEST] <scoofy> extract original video from original file?
[22:11:16 CEST] <dexikiix> its an mov and im on win 7 if that helps/matters
[22:11:17 CEST] <scoofy> i mean, original audio
[22:11:30 CEST] <scoofy> and combine with new video
[22:11:47 CEST] <dexikiix> from the corrupted one that wouldn't play?
[22:11:52 CEST] <scoofy> yes.
[22:11:55 CEST] <dexikiix> ok let me try that
[22:11:59 CEST] <scoofy> why not
[22:12:10 CEST] <scoofy> at worst, you get an error
[22:12:15 CEST] <dexikiix> i tried to extract audio from the newer one, but got a 0kb file, let me grab my command and see what i'm doing wrong
[22:14:48 CEST] <dexikiix> ok so i did ffmpeg -i broken.mov -acodec copy fixed.mov and I get "moov atom not found broken.mov: Invalid data found when processing input"
[22:15:11 CEST] <scoofy> sounds fux0red
[22:15:18 CEST] <dexikiix> prob want an mp3 file anyway, do I need to adjust the command for that or just the filename?
[22:15:25 CEST] <scoofy> yet that program could restore it, somehow
[22:15:38 CEST] <scoofy> with errors
[22:16:12 CEST] <dexikiix> yeah... if only i could get the audio out I could even go into audacity and cut it myselfg
[22:16:22 CEST] <scoofy> what format was the audio in?
[22:16:24 CEST] <dexikiix> myself* if that was possible
[22:16:33 CEST] <scoofy> probably
[22:16:35 CEST] <dexikiix> the original file is from a dashcam that saves in mov
[22:16:41 CEST] <scoofy> audio codec?
[22:16:44 CEST] <dexikiix> hmm
[22:16:47 CEST] <dexikiix> let me see if i can find that
[22:16:50 CEST] <scoofy> maybe you gotta find out
[22:17:00 CEST] <scoofy> and i'll tell you what you might do
[22:17:25 CEST] <dexikiix> is that what the probe is for?
[22:17:36 CEST] <scoofy> i think that can tell you, yes.
[22:18:43 CEST] <dexikiix> pcm_s16le ?
[22:19:19 CEST] <dexikiix> 32000 hz 1 channel s16, 512 kb/s (default)
[22:19:55 CEST] <dexikiix> that's what I get from the probe...
[22:20:50 CEST] <dexikiix> a bit of googling says pcm is "traditional wave format"
[22:20:59 CEST] <scoofy> yep. sounds like a format you can read with Audacity!
[22:21:08 CEST] <scoofy> maybe you can just open the raw file in audacity
[22:21:12 CEST] <scoofy> and try to locate the audio part
[22:21:16 CEST] <furq> dexikiix: if you have a working video from the same source you can try using untrunc
[22:21:28 CEST] <scoofy> you need to open it as signed, 16 bits - you can try offset 0 or 1
[22:21:31 CEST] <furq> https://github.com/ponchio/untrunc
[22:21:35 CEST] <scoofy> set sampling rate to 32 kHz
[22:21:40 CEST] <dexikiix> furq, i do, and I will thanks. scoofy, i never even considered that, that could work too
[22:22:03 CEST] <scoofy> dexikiix: there may be an 1 byte offset, so you may need to offset the file when opening it as raw audio
[22:22:20 CEST] <scoofy> make sure you open it as signed, 16 bit integer. one program that can surely do this, is an old audio editor called 'cool edit'.
[22:22:52 CEST] <scoofy> s16 pcm just means raw 16 bit audio.
[22:22:59 CEST] <dexikiix> thanks
[22:23:06 CEST] <bencoh> furq: it tries to rebuild moov/atom?
[22:23:23 CEST] <furq> looks like it
[22:23:25 CEST] <furq> i've never had to use it
[22:23:37 CEST] <dexikiix> ah crap furq ... hate to be this guy but im not on linux... >.>
[22:23:49 CEST] <bencoh> I like the idea behind it (similar working file + broken file)
[22:24:10 CEST] <scoofy> dexikiix: in case you open the file in audacity, anything resembling like an audio signal would look not like a block from top to bottom. so look for something that looks different from the rest.
[22:24:28 CEST] <furq> you might be able to compile it under msys if you're lucky
[22:24:30 CEST] <dexikiix> checking it out now, will do, thanks
[22:24:36 CEST] <furq> but it'd probably be quicker to just install linux in a vm or something
[22:24:41 CEST] <dexikiix> furq, never tried but i'm willing :)
[22:24:47 CEST] <dexikiix> haha fair point
[22:24:58 CEST] <dexikiix> can I run something like that through a usb version?
[22:25:21 CEST] <furq> if you mean a bootable linux usb stick then sure
[22:25:34 CEST] <furq> there's no good reason to not do it in a vm though
[22:25:34 CEST] <bencoh> but the debian in virtualbox way is probably simpler
[22:25:37 CEST] <furq> yeah
[22:26:53 CEST] <furq> compiling linux stuff with msys/cygwin is a bit of a crapshoot
[22:27:05 CEST] <dexikiix> alright i'
[22:27:17 CEST] <dexikiix> i'll look into that after audacity here. I just got some static from the import
[22:27:51 CEST] <furq> i take it your dashcam can't save to a better format which isn't useless if it gets truncated
[22:27:58 CEST] <furq> for the future, obviously
[22:28:15 CEST] <dexikiix> yeah its a pretty cheap piece of junk
[22:28:26 CEST] <dexikiix> but it works
[22:28:43 CEST] <dexikiix> does endianness matter?
[22:28:48 CEST] <scoofy> yes!
[22:28:58 CEST] <scoofy> endianndess and offset both matter. you can try 1 byte offset.
[22:29:04 CEST] <scoofy> try all the combinations.
[22:29:07 CEST] <dexikiix> ok
[22:32:30 CEST] <karab_44> OK I happily found the solution! Thanks guys!
[22:33:01 CEST] <dexikiix> dangit, everythings coming out static in audacity
[22:33:55 CEST] <dexikiix> ok next step is a vm
[22:34:05 CEST] <scoofy> how big is the file?
[22:34:26 CEST] <dexikiix> 502 megsd
[22:34:27 CEST] <dexikiix> megs*
[22:35:16 CEST] <scoofy> i wonder where .mov puts the audio in the file
[22:35:26 CEST] <furq> it looks like the arch package builds against ffmpeg 2.8, so a recent debian should work fine
[22:35:35 CEST] <furq> i was a bit concerned that the github readme says you need libav
[22:36:29 CEST] <furq> https://www.debian.org/devel/debian-installer/
[22:37:14 CEST] <nadermx> is there a config option I have to compile with for ffmpeg to have http_proxy option?
[22:37:35 CEST] <c_14> probably just don't disable http
[22:39:24 CEST] <furq> nadermx: what ffmpeg version
[22:39:29 CEST] <furq> that option was only added in december
[22:40:10 CEST] <nadermx> I'm running the most recent version of ffmpeg
[22:40:25 CEST] <nadermx> git version 03-01
[22:41:10 CEST] <nadermx> yeah i figure it must be calling a older version so removing and re installing to see why it wasn'tw orking
[22:46:16 CEST] <nadermx> its kind of weird, i compiled and ran ffmpeg but when i try to use the proxy its not using it
[22:46:19 CEST] <nadermx> ffmpeg version git-2016-03-01-c3bb616 Copyright (c) 2000-2016 the FFmpeg developers
[22:47:30 CEST] <nadermx> when i run ffmpeg -protocols shows 'httpproxy'
[00:00:00 CEST] --- Mon Apr  4 2016