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June 2014
- 1 participants
- 64 discussions
[01:14] <cone-261> ffmpeg.git 03Rong Yan 07master:ab12373956e9: ppc: Fix the bug of fft for little endian environment on POWER7 and later
[01:46] <j-b> a1m
[04:39] <nevcairiel> man gcc and clang are annoying, why do they only support intrinsics of a certain instruction set if the whole binary is compiled for that instruction set ....
[04:41] <Plorkyeran> isn't it only that TU that has to be compiled for that instruction set?
[04:42] <nevcairiel> annoying enough
[04:42] <nevcairiel> msvc allows me to use any intrinsics i want :p
[04:42] <Plorkyeran> quite annoying, but possible to deal with at least
[04:43] <nevcairiel> gcc apparently at least has pragmas to even do it per-function
[04:43] <nevcairiel> clang does not
[04:43] <nevcairiel> well screw mac users
[04:44] <nevcairiel> you get slower builds!
[04:46] <Plorkyeran> if you aren't being all efforty and releasing universal builds, requiring sse 4.1 on os x isn't all that unreasonable
[04:46] <nevcairiel> hm right, mac osx is generally all intel
[04:47] <nevcairiel> no stupid AMD cpus which introduced instruction sets way too late
[04:47] <nevcairiel> and i only want ssse3 right now even
[04:47] <Plorkyeran> and the first model with a 64-bit processor can't actually run any 64-bit versions of the OS
[04:48] <nevcairiel> i shall discuss it with our mac expert tomorrow
[05:00] <jamrial> i wonder what the hell was AMD thinking back in the Barcelona days. they refreshed that arch like three times and at no point they considered that maybe ssse3 and sse4.1 were worthy additions to reduce the gap with conroe/penryn/nehalem
[05:01] <jamrial> they obviously changed the strategy now with Bulldozer, where each refresh adds like two or three new intruction sets, but the damage is done nonetheless
[05:05] <Compn> was that right about when they bought ati ?
[05:06] <jamrial> no idea, but could be
[05:17] <cone-746> ffmpeg.git 03Diego Biurrun 07master:e74433a8e6fc: dsputil: Split clear_block*/fill_block* off into a separate context
[05:17] <cone-746> ffmpeg.git 03Michael Niedermayer 07master:2b05db4f8102: Merge commit 'e74433a8e6fc00c8dbde293c97a3e45384c2c1d9'
[06:00] <cone-746> ffmpeg.git 03Michael Niedermayer 07master:042a82ca374b: avcodec/x86/lossless_videodsp: Fix size of values read for left/left_top
[06:22] <cone-746> ffmpeg.git 03Christophe Gisquet 07master:e7fc5d53a02c: huffyuv: change statistics initialization
[13:14] <cone-797> ffmpeg.git 03James Almer 07master:fe782233aa6a: x86/blockdsp: move asm code out of dsputil
[13:24] <cone-797> ffmpeg.git 03James Almer 07master:454c019cb536: x86/hevc_idct: fix movd parameter size in DC_ADD_INIT
[13:24] <cone-797> ffmpeg.git 03James Almer 07master:60a06400a9fa: configure: fix NASM output format on Win64
[13:31] <cone-797> ffmpeg.git 03James Almer 07master:4c9ee1590feb: lavf/Makefile: fix standalone compilation of live_flv demuxer
[13:51] <BBB> michaelni: Ive got resample_common_float_sse ported to yasm, I can push to github for review& cant merge yet, until avx/linear are also done, but that should be relatively minor (theyre just trivial variations of the same theme afaics)
[13:55] <michaelni> you could post a patch with [WIP] or so, i suspect more people read the mailing list than look at github
[14:08] <BBB> ok done
[14:09] <BBB> Ill do the linear version next
[14:09] <BBB> then avx last (that should be very trivial)
[14:57] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:08c5859f17ed: avcodec: add simpleauto idct
[15:17] <cone-797> ffmpeg.git 03Dirk Ausserhaus 07master:b2290bf10b12: i263: skip dummy frames
[15:17] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:47c8d1e174b8: Merge commit 'b2290bf10b12b7772f55fcab014a47e18e9fdef2'
[18:06] <cone-797> ffmpeg.git 03Diego Biurrun 07master:852c329a69dd: configure: Drop duplicate pkg-config entry from help output
[18:06] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:a68c05cf13c9: Merge commit '852c329a69ddfd635807f60454df07d021c64901'
[18:31] <wm4> does decoding on AVSubtitle always set a copy?
[18:31] <wm4> or can the AVSubtitle have pointers to data that will be freed or reused on the next decode call?
[18:39] <cone-797> ffmpeg.git 03James Almer 07master:a5ce608fc7ea: x86/blockdsp: restore author attribution
[19:17] <cone-797> ffmpeg.git 03Carl Eugen Hoyos 07master:891307b4d194: s86/scale: Do not return the result of a (void) function from a void function.
[19:27] <J_Darnley> I am shocked, SHOCKED(!), to discover that ffmpeg doesn't have signed 8-bit audio.
[19:31] <J_Darnley> Now then, I wonder if I am allowed to modify the input AVFrame in a filter...
[19:39] <J_Darnley> OMFG! Damn docs! Will you just stay disabled!
[19:48] <wm4> J_Darnley: well yes
[19:48] <wm4> J_Darnley: you must make it writable first
[19:49] <wm4> J_Darnley: there's a function in frame.h for this
[19:49] <wm4> it'll copy the frame if needed
[19:50] <J_Darnley> Oh yeah, silly me. I've read about that.
[19:50] <J_Darnley> I'll just write into my own buffer then.
[19:52] <wm4> write into your own buffer, and then write again into the output frame?
[19:53] <J_Darnley> No, I'm talking about the audio input of an a->v filter
[20:17] <nevcairiel> audio frames should always be writeable, they dont really have references much
[20:17] <nevcairiel> but of course there is the API to tell you
[20:26] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:5bf5e6b1c0fd: avcodec/h264: Use named identifier for single thread fall-back return from ff_h264_decode_slice_header()
[20:54] <cone-797> ffmpeg.git 03Gaullier Nicolas 07master:650ef180787d: avformat/utils: better probing for duration in estimate_timings_from_pts()
[21:10] <tab1293> I am trying to learn how to use libavcodec to encode a video stream in an application I want to write. I am very new to ffmpeg and it doesn't seem like there is much documentation on the website. Anyone have any suggestions on how I should go about learning how to use libavcodec?
[21:11] <nevcairiel> there are a bunch of examples in the source tree
[21:11] <nevcairiel> ie in doc/examples
[21:11] <Daemon404> doxygen too
[21:11] <Daemon404> on the siye
[21:11] <Daemon404> site
[21:12] <tab1293> okay let me check that out. But any resources available that provide more of a conceptual overview of the framework and how to use it?
[21:13] <nevcairiel> i think i saw something like that in the docs
[21:14] <tab1293> alright I'm grabbing the source now
[21:18] <Daemon404> ubitux, i have no works for the "bash" fix
[21:18] <Daemon404> (he uses "bash" to mean "shell" i think)
[21:21] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/2.2:f378636d90af: avformat/mp3enc: drop redundant and uninitialized variable
[21:28] <tab1293> how do I build the docs? I tried running make doc in the doc folder but that doesn't work
[21:31] <Daemon404> http://ffmpeg.org/doxygen/trunk/
[21:33] <tab1293> but what about the texi files how do I generate the documentation from those?
[21:33] <Daemon404> configure and make?
[21:33] <Daemon404> theyre also all on the website
[21:33] <Daemon404> all of this is linked from the documentation page
[21:33] <Daemon404> also this belong in #ffmpeg
[21:41] <nevcairiel> speaking of examples, wasnt there an invalid api usage patch floating around
[21:41] <nevcairiel> whatever happened to that
[21:45] <nevcairiel> oh right it was about re-using the context
[21:47] <Daemon404> i wrote it
[21:47] <Daemon404> its been ignored since feb
[21:52] <michaelni> Daemon404, pinged the question you ignored
[21:55] <wm4> AFAIK the plan on libav side seems to be to remove codeccontexts completely from libavformat?
[21:56] <wm4> also, I can't understand how reusing the codec context can work
[21:56] <wm4> what if parameters change?
[21:57] <nevcairiel> thats why its not safe
[21:57] <nevcairiel> changes to parameters are out of sync
[21:57] <Daemon404> michaelni, ? my client shows it as responded
[21:57] <nevcairiel> well technically you never answered the question other then "anton said so" :P
[21:57] <michaelni> Daemon404, yes with "According to Anton, we should never reuse it from there.
[21:57] <michaelni> He give me the reason why, but it is buried in my IRC logs
[21:57] <michaelni> somewhere (old patch is old)."
[21:58] <michaelni> but thats no way to explain what a patch does
[21:58] <nevcairiel> iirc, the reason is that a demuxer/parser can change the info during demuxing, while the decoder is still a couple frames behind and that would get messy
[21:59] <Daemon404> worst case is the example shows more api usage
[21:59] <michaelni> such a demuxer / parser is broken and violating API
[21:59] <Daemon404> which is hardly bad
[22:00] <Daemon404> using copy_context fixed a crash/bug i had in my code
[22:00] Action: Daemon404 looks for that
[22:01] <Daemon404> i think it was what nevcairiel said
[22:01] <michaelni> showing more api usage is a mixed bag, yes i agree its good to have more stuff covered by examples but then the doing X example should show X and not additionally things that confuse the reader
[22:02] <michaelni> examples should be minimal
[22:03] <michaelni> or it should be clearly documented what is optional in them
[22:04] <michaelni> the context split / copy is recent so it surely worked fine for a long time without it
[22:06] <Daemon404> it was never explicitly documented to work for decoding iirc
[22:06] Action: Daemon404 stabs grep
[22:07] <michaelni> there sadly was alot that wasnt documented
[22:08] <michaelni> but comparing long time ago to now i think docs have improved quite a bit
[22:09] Action: J_Darnley pasive-agressively mentions his emails about filters
[22:12] <michaelni> J_Darnley, maybe try to submit the code as git patch, might result in more comments
[22:14] <J_Darnley> I'll try that, sometime soon(ish)
[22:52] <cone-797> ffmpeg.git 03James Almer 07master:05d95c53a70f: alpha/blockdsp: move clear_blocks out of dsputil
[23:30] <J_Darnley> michaelni: thanks for your reply
[23:40] <miep1> I know I should ask this question on #ffmpeg, but nobody there seems to be awake. Is there anyone here who can point me to the right direction on how to use the #ffmpeg libraries to implement an audio-stripper? The idea is to take a video and strip the audio in an android app. I can NDK compile the libraries and call them with JNI, but now I'm left to implement the C-part, and I can't really find my way through the doc/example(s)
[23:44] <michaelni> just drop audio packets
[23:45] <nevcairiel> yeah that should be pretty easy, look at the demuxing and muxing examples, and just drop the audio in the middle
[23:53] <llogan> [matroska @ 0x181c400] Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead
[23:53] <llogan> that's going to freak out users like that other deprecated message
[23:54] <nevcairiel> easy enough fixed
[00:00] --- Fri Jun 20 2014
1
0
[00:05] <Spacexplosion> ok, one moment
[00:32] <Spacexplosion> ok c_14, sorry for the delay
[00:32] <Spacexplosion> error output: http://pastebin.com/dTNMVZVB
[00:33] <Spacexplosion> codecs list: http://pastebin.com/Eda4ahrn
[00:35] <Spacexplosion> I'
[00:36] <c_14> I think the problem is that ogg doesn't support subrip subtitles
[00:36] <Spacexplosion> eh, really?
[00:38] <c_14> So according to wikipidia (cause I couldn't find a better source), ogg supports writ, continuous media markup language, annodex and oggkate
[00:39] <Spacexplosion> huh, that didn't even occur to me
[00:39] <Spacexplosion> I'm looking at another ogg I have, and it has subs encoded as just "text"..
[00:39] <Spacexplosion> Stream #0:3(English): Subtitle: text
[00:39] <llogan> why theora (and ogg)? there are usually better alternatives these days
[00:40] <Mephistos> hi guys, any idea what output format/codec I can use with -pix_fmt rgb48le. I think that will give me more info converting from a bayer pixel format than rgb24.
[00:40] <Spacexplosion> llogan: I know; not my choice. it's for a special purpose. ;)
[00:41] <Spacexplosion> so if I set the sub codec to something else, like "text" it should work?
[00:42] <Spacexplosion> not sure how raw text has any timing info...
[00:42] <Mephistos> The input file is a MLV which has bayer_rggb16le as the pixel format and I think that I'm losing a lot of info when it's converting to rgb24, 16 bits per color to 8 bits.
[00:42] <c_14> technically yes, practically it depends on if the source codec is picture or text and if ffmpeg has an encoder for the output format
[00:46] <Mephistos> Hmm... I think I just found the issue... using r210 as the codec...
[00:46] <llogan> Spacexplosion: you could always make hardsubs if you really need subtitles and if ogg is being dumb for subtitles (i don't know shit about ogg 'n subtitles)
[00:47] <llogan> Mephistos: ffmpeg -h encoder=<name> will list supported pixel formats for each encoder
[00:47] <Spacexplosion> well, I was having trouble with that too. I was trying to use the "subtitles" video filter, but it claims that doesn't exist
[00:47] <Mephistos> oh yeah?
[00:47] <Mephistos> is there a way to do the reverse, see which codecs fit the pixel formats?
[00:48] <llogan> i'm not sure
[00:49] <Mephistos> I just did that and I get the options for ffmpeg but not the pixfmts
[00:49] <llogan> Mephistos: also the format and/or noformat filters may be use(ful|less) at some point
[00:50] <Mephistos> yeah, I've got a pastebin for you
[00:50] <Mephistos> I found r210 is compatible with rgb48le
[00:50] <llogan> <- paste addict
[00:50] <Mephistos> but I encountered a completely different issue, which is probably because we're all in new territory when it comes to raw video.
[00:50] <Mephistos> You'll like this one then... it's new.
[00:51] <Mephistos> http://pastebin.com/FrMcJX5a
[00:52] <Spacexplosion> what you want me to paste?
[00:52] <Mephistos> They want to see the output and full command line used
[00:52] <Spacexplosion> the error about subtitles filter?
[00:53] <Mephistos> They want the full output and command line of ffmpeg, that should also include that error.
[00:54] <Mephistos> take a look at my pastebin, it will give you an idea
[00:55] <llogan> Mephistos: i was referring to your -h issue. unfortnately I don't know what to suggest for "full chroma interpolation for destination format 'rgb48le' not yet implemented" unless you know C and want to give a go at implementing it. some users on ffmpeg-user ML might know more
[00:55] <llogan> Spacexplosion: yes. subtitles filter issue.
[00:56] <Spacexplosion> k, I don't have the command I was using handy. I'll have to fiddle with it
[00:56] <llogan> Mephistos: ffmpeg -pix_fmts might interest you
[00:57] <Mephistos> Oh I know, I've been using that to see what's baked into my compile of a recent snapshot, but the pix_fmts list doesn't tell me which codecs work with them.
[00:58] <Mephistos> I was able to google one that worked for testing purposed, r210, and it no longer converts it to rgb24...
[00:58] <Mephistos> no idea why that even works
[00:59] <Mephistos> I was testing using huffyuv, which
[00:59] <Mephistos> causes it to convert to rgb24
[00:59] <Mephistos> but bayer formats are weird
[01:00] <Mephistos> they are grgrgrgrgrg on one row, then gbgbgbgbg on the next row. So the conversion isn doesn't work, it's taking each pixel and converting it to
[01:00] <Mephistos> whoops
[01:00] <Mephistos> let me try that line again
[01:01] <Mephistos> they are grgrgrgrgrg on one row, then gbgbgbgbg on the next row. So the conversion isn't being done in a sane way, it's taking each pixel and converting it to 8bits or red green or blue... But the catch is there's no demosiac process being done!
[01:02] <Spacexplosion> llogan: subtitle filter error: http://pastebin.com/6HZMB9t7
[01:02] <Spacexplosion> filters: http://pastebin.com/9aQ9Ve7u
[01:02] <Spacexplosion> sure enough, "subtitles" isn't listed as a filter
[01:03] <Spacexplosion> all I have support for is ssa apparently
[01:03] <Spacexplosion> I wouldn't think the generic "subtitles" filter would be conditionally built
[01:04] <Spacexplosion> I'm running Gentoo, and it's not a USE flag
[01:05] <c_14> you're running ffmpeg version 1.08
[01:05] <c_14> that version is ancient
[01:06] <Spacexplosion> and Gentoo is supposed to be so bleeding edge :P
[01:06] <c_14> try unmasking 2.2.3-r1 or 9999
[01:06] <c_14> Always use the newest ffmpeg possible, preferably git head.
[01:07] <Spacexplosion> wonder why it is masked back to such an old version...
[01:10] <c_14> according to the bugtracker because some optional dependencies are masked
[01:21] <Mephistos> I think I see what's going on. The bayer conversion thing is probably going to be a huge new direction for the ffmpeg project, and there's tonnes of code out there from other projects, specifically darktable, which is the best and most complete raw image editing project out there, for stills.... but they have everything ready made and work within a 4x32 floating point pixel buffer space, which is insane.
[01:29] <paxromain> Hey everyone!
[01:29] <paxromain> anyone seen c_14 around?
[01:29] <llogan> Mephistos: peter ross would probably be the guy to talk to: https://trac.ffmpeg.org/wiki/FFmpegSummerOfCode2014#BayerRGBcolorspaces but i'm not sure what the current status of any beyer stuff is
[01:30] <paxromain> hey llogan what's going on?
[01:30] <llogan> procrastinatin
[01:33] <Mephistos> llogan, it's probably at a very initial primitive state, thanks for the link, this is cool. I'll message Michael. I have a feeling that a different kind of overview needs to be looked at for how bayer is handled.
[01:36] <Spacexplosion> oh wow, boy does ffmpeg 2.2.3-r1 ever depend on a lot of other masked stuff
[01:37] <RadSurfer> avidemux appears to be a sweet program, working great here so far... Win8.1/quadcore
[01:37] <RadSurfer> works great with ffmpegs .mkv's
[01:37] <BtbN> never got avidemux to do much more than creating broken videos and/or crashing.
[01:38] <RadSurfer> well, have U tried latest version? also have 8gb ram, I'm sure that helps
[01:38] <RadSurfer> 2012... last version?
[01:39] <RadSurfer> I'll try it this week and maybe get back to you, what little testing I've done is impressive for a freeby
[01:45] <RadSurfer> If I wanted ffmpeg to merely append/combine .vob files into one (and/or create .mkv), can someone please suggest a command line syntax
[01:45] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med…
[01:56] <Mephistos> llogan, I don't see a way to contact Peter Ross, only Michael, his backup mentor.
[04:03] <RadSurfer> Where is the documentation to control increasing the playback rate of the input file, when encoded to "normal rate" for output file?
[04:04] <c_14> playback rate as in fps?
[04:05] <RadSurfer> yeah... I must increase -i file by 2.4375
[04:05] <RadSurfer> times its normal speed
[04:06] <RadSurfer> https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20…
[04:06] <c_14> the setpts filter is documented in ffmpeg-filters, if that's what you're looking for
[04:07] <c_14> But if you want to speed up by 2.4375, setpts=(1/2.4375)*PTS should do the trick
[04:07] <RadSurfer> Thank you.
[04:16] <llogan> atempo for audio if you need that
[04:28] <RadSurfer> Hmm. wasn't expecting that
[04:28] <RadSurfer> aspect ratio is all wrong
[05:24] <RadSurfer> WHERE is the ffmpeg doc page that describes -aspect please? I found -scale
[05:25] <sacarasc> http://ffmpeg.org/ffmpeg-all.html
[05:27] <RadSurfer> I was hoping it would shed some light on " scale vs aspect ratio "
[05:27] <sacarasc> Scale sets how many pixels there are, aspect sets what shape it is.
[05:28] <RadSurfer> It's shape I am concerned about.
[05:28] <RadSurfer> I thought I used to know this, but haven't worked with it in ages
[08:14] <zzw> hello everyone, who can tell me the difference of "mpegts" and "mepgtsraw" demuxer?
[08:21] <zzw> i want to perform real-time demuxing and playing mpegts ,but i find the avformat_open_input function waste too much time
[08:47] <zzw> hello everyone, who can tell me the difference of "mpegts" and "mepgtsraw" demuxer?
[08:47] <zzw> i want to perform real-time demuxing and playing mpegts ,but i find the avformat_open_input function waste too much time
[09:08] <Guest68998> ive got some issues when trying to compile ffmpeg , getting error relocation R_ARM_THM_MOVW_ABS_NC against `ff_avg_pixels16_x2_neon' can not be used when making a shared object; recompile with -fPIC libavcodec/arm/hpeldsp_init_neon.o: could not read symbols: Bad value collect2: error: ld returned 1 exit status make: *** [libavcodec/libavcodec.so.55] Erreur 1
[09:08] <Guest68998> any clue what this might be?
[11:25] <decembersoul1> Hello, is there a way that ffserver + ffmpeg only transcode the stream if a consumer is connected.
[11:25] <decembersoul1> Mjpeg stream -> ffmpeg -> h265 -> ffserver -> listener
[11:25] <decembersoul1> If no one listen to the stream we don't need to create the stream
[11:26] <Mavrik> no.
[11:28] <decembersoul1> in some cases (fast entrance into stream) it is usefull, but in some cases it is a waste of CPU Power
[14:33] <Swoxxx> Hi, any of you know an opensource solution that would allow us to **render** videos server-side and headless? With a template video that would have placeholders, where we would insert some pictures and then render the video
[14:33] <Swoxxx> Serverside, dynamically, headless if possible
[14:34] <Swoxxx> If you have any idea about this, I'm more than willing to know about it
[14:43] <Esya> Hi, does anyone know where I can ask a question about headless video rendering on unix? (Not encoding, actual rendering)
[14:43] <Esya> or compositing actually.
[14:44] <klaxa|work> ffmpeg video filters maybe?
[14:44] <Esya> klaxa|work Hmm.. Not sure if that would work but let me give a quick example
[14:45] <Esya> Lets say we have a first video with characters moving but they have no faces. We also have user pictures, and we would need to compose their faces onto the characters
[14:46] <klaxa|work> and you want that to be done headless?
[14:46] <Esya> Not necessarily headless, I can use a X server, but automatically for sure
[14:46] <Esya> A bit like facebook lookback videos
[14:47] <klaxa|work> maybe you can use some tracking stuff with cinelerra
[14:47] <Esya> Hmm, it does not necesarrily imply tracking as it will be done only once and the source/background video will always be the same
[14:47] <Esya> So we can feed whatever we will use the right coordinates to use if needed
[14:48] <Esya> (But I am going to look at cinelerra, which I didnt know, and thanks for that)
[14:51] <Esya> klaxa|work Actually someone has suggested using blender in headless/server mode and said that even if it's not exactly 3d, it could work well
[14:51] <klaxa|work> oh yeah i heard such things about blender too
[15:31] <JEEBsv> I recommend you drop libaacplus, and instead use fdk-aac
[15:31] <JEEBsv> aand here I was way up in the buffer
[15:31] <JEEBsv> herp derp
[15:45] <Mavrik> :))
[16:13] <Fjorgynn> i see
[16:13] <Fjorgynn> JEEBsv: fdk aac ftw
[18:21] <maksimkaaa> I have a normal 4 seconds OGV movie generated from 4 images, whenever I try to generate an MP4 out of it, the resultant mp4 is just 2 seconds long with 2 frames.. here is the log of the ffmpeg command http://pastebin.com/H1ryAFFH The command itself was just ffmpeg -i 7_4499.ogv test1.mp4
[18:34] <llogan> maksimkaaa: can you provide the input?
[18:40] <maksimkaaa> llogan you mean the ogv file itself?
[20:23] <RadSurfer> ffmpegs default .mkv file video plays aok on my dvd player _except_ for "audio is not supported", I will have to over-ride encoding audio for that to be compatible apparently
[20:26] <RadSurfer> I wonder what the common-most audio format for dvd-players/internet-tv devices would be?
[20:28] <c_14> DVDs usually have somethnig alonng the lines of pcm,dts,mp2 or ac3.
[20:29] <RadSurfer> I am trying to reduce size of my camcorders .mp4 files; video is displaying just fine!
[20:29] <RadSurfer> So what would you recommend I try to audio options, please?
[20:30] <RadSurfer> /s/to/for
[20:31] <c_14> What's your source audio format?
[20:32] <RadSurfer> isn't there a way to have ffmpeg tell me?
[20:33] <c_14> ffprobe $file
[20:34] <RadSurfer> audio encoder : Samsung AVC Encoder Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 24000 Hz, stereo, fltp, 12
[20:34] <RadSurfer> 3 kb/s (default)
[20:35] <RadSurfer> so its just aac?
[20:35] <c_14> I'd probably use AC-3 if you're looking for compatibility and filesize requirements.
[20:57] <RadSurfer> this seems to work for me now:
[20:57] <RadSurfer> ffmpeg -i "file.MP4" -aspect "16:9" -vf yadif -c:a copy -c:v libx264 "movie0.mkv"
[20:58] <RadSurfer> although I don't really need -aspect in there
[20:58] <RadSurfer> 1/5th original size! lol
[20:59] <RadSurfer> nearly
[23:17] <miep1> I need some help on how to achieve something with libav*. I'm trying to set up something that filters out all audio from a regular video
[23:18] <miep1> I can't seem to find the relevant example in doc/example
[23:18] <miep1> Any pointers on where to look?
[00:00] --- Fri Jun 20 2014
1
0
[00:00] <llogan> WebUrine
[00:00] <wm4> that's what it is
[00:03] <Youka> Ok, as far as i see there's no nanosleep in MinGW. There were attempts in past, but at the end, Sleep & usleep remained.
[00:06] <Youka> So detecting mingw and consider in configure to avoid HAVE_NANOSLEEP would be needed (in my case, i change it in config.h for now).
[00:07] <wm4> uh the configure script already checks whether nanosleep exists
[00:08] <Youka> But the missing struct timespec while compilation tells another thing :/
[00:13] <jamrial> HAVE_NANOSLEEP should not be true in your configuration. something went wrong
[00:13] <Youka> I think too, but not on my side ^^"
[00:14] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:f31fd067130f: vf_sab: use av_malloc_array()
[00:14] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:eb0fc73dcc85: avfilter/vf_spp: Use av_malloc_array()
[00:19] <jamrial> it is on your side. configure checked for nanosleep and the test succeeded, which can't happen with mingw
[00:19] <jamrial> since for starters it's missing librt
[00:25] <Youka> Looking at the test, it uses nanosleep with an extern reference and without struct timespec, so the missing of both in headers is right, but the function is in a linked library it seems.
[00:26] <Youka> So the test could succeed, but the compilation fails.
[00:30] <Youka> Officially nanosleep is not part of mingw, but broken experiments are left in binaries.
[00:31] <wm4> too bad
[00:32] <wm4> time to paste your config.log?
[00:32] <Youka> y
[00:32] <Youka> mom
[00:33] <Youka> `http://pastebin.com/gZjzZrJm
[00:34] <Youka> With HAVE_NANOSLEEP=0 the build completed now
[00:35] <wm4> seems like a pretty bad idea what the configure script is doing
[00:35] <wm4> _especially_ if by default implicit function declarations are an error
[00:36] <Youka> ############################
[00:36] <Youka> 1 extern int nanosleep();
[00:36] <Youka> 2 int main(void){ nanosleep(); }
[00:36] <Youka> ############################
[00:36] <Youka> struct timespec ts = { usec / 1000000, usec % 1000000 * 1000 };
[00:36] <Youka> while (nanosleep(&ts, &ts) < 0 && errno == EINTR);
[00:36] <Youka> ############################
[00:36] <Youka> First is configure, second the part in libavutil/time.c
[00:37] <wm4> you know what, this is a dev channel
[00:37] <wm4> for user questions go to #ffmpeg
[00:39] <Youka> It's not about using ffmpeg or developing with libav, it's a build error, so isn't it right here?
[00:42] <Daemon404> winpthreads has nanosleep
[00:42] <wm4> so why does ffmpeg configure check for the symbol only
[01:14] <nevcairiel> because thats enough for 99.9% of things, and makes it much easier to abstract
[01:15] <nevcairiel> never had the problem myself with mingw
[01:16] <nevcairiel> of course its been years since i used mingw32, but admittedly its been more years since mingw32 was updated :p
[01:17] <Youka> My old ffmpeg didn't throw this error, so this problem came with newer versions.
[01:18] <Youka> Same compiler, new ffmpeg.
[01:19] <nevcairiel> personally i would call mingw32 unsupported already and push people to upgrade their old environments
[01:38] <cone-460> ffmpeg.git 03Tristan Matthews 07master:49106844929b: tiffenc: fix packet size calculation
[01:38] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:9f028b3f3af5: Merge commit '49106844929b7b71ac719064d640f8aa56c89b1f'
[01:51] <Timothy_Gu> Youka nevcairiel: I've never encountered any problems with mingwrt 3.20 and w32api 3.17
[01:51] <Timothy_Gu> Youka: what is your MinGW (not gcc) version?
[01:51] <cone-460> ffmpeg.git 03Josh Allmann 07master:74bc9458057f: flvenc: Don't over-count metadata.
[01:51] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:3f2f87cd5a84: Merge commit '74bc9458057f07c0820c3d3264396c4c64442e47'
[02:01] <cone-460> ffmpeg.git 03Rafaël Carré 07master:9affa784a074: vp9 encoder: use a decent default speed setting
[02:01] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:cb037b5db49f: Merge commit '9affa784a0743b95bde135b5acfce2183afc35d8'
[02:06] <llogan> Timothy_Gu: He's dead, Jim.
[02:14] <Timothy_Gu> llogan: yeah he is. didnt see that earlier
[02:16] <cone-460> ffmpeg.git 03Lou Logan 07master:d9a61ae31e8c: presets: remove moldering iPod presets
[02:17] <michaelni> matrixbench_mpeg2.mpg -vf scale=320:240,format=monow,spp=6:63,hqx=4
[02:17] <Timothy_Gu> ubitux: i wrote a patch fixing the underline mess on FATE
[02:18] <Timothy_Gu> you can see if that meets you expectations at http://goo.gl/yCqilT
[02:19] <Timothy_Gu> * http://goo.gl/yCqiIT
[03:21] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:28bf11150346: avcodec/wavpack: fix () in macros
[03:21] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:aab23f5ddd3d: avcodec/vp9: fix () in macros
[03:21] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:c89be82c986f: avcodec/vp8: fix () in macros
[03:21] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:0608bc65025a: swresample/audioconvert: fix () in FMT_PAIR_FUNC()
[04:11] <BBB> yay I managed to write resample_common_float_sse in yasm
[04:11] <nevcairiel> and is it 200% faster?
[04:22] <BBB> didnt measure yet
[04:22] <BBB> probably not, I didnt optimize anything yet, just basic stuff which is probably similarish to disassembly
[04:22] <BBB> the optimizations would come after that
[04:26] <Compn> always wondered if a machine translation from c > asm would be faster
[04:26] <Compn> i've heard it would be :P
[04:26] <Compn> 'asm is always faster'
[04:56] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:c932f12806d2: avcodec/cavsdec: simplify % 512 operations
[04:56] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:7f5296005c39: swscale/swscale: replace potentially slow % by &
[05:07] <nevcairiel> Compn: if its a machine translation, why didnt the compiler do it
[05:12] <Compn> nevcairiel : well. 'why didnt gcc optimize this?'
[05:12] <Compn> because its gcc.
[05:47] <Timothy_Gu> llogan: thanks for the reminder. I will send a patch in probably 2-3 days
[07:32] <ubitux> Timothy_Gu: LGTM, thx
[07:35] <ubitux> < michaelni> matrixbench_mpeg2.mpg -vf scale=320:240,format=monow,spp=6:63,hqx=4 // hehe pretty nice
[07:42] <llogan> about how big is the fate suite now?
[08:03] <ubitux> fate-samples is 854M here
[08:06] <llogan> ubitux: thanks. i was just curious because my innernet is slow.
[12:36] <BBB> well this uses clang, but anyway
[12:36] <BBB> (mac doesnt use gcc anymore)
[12:59] <cone-261> ffmpeg.git 03Roman Savchenko 07master:44671b57866a: AVFrame: factor out freeing the side data
[12:59] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:40c9d4308689: Merge commit '44671b57866aab8dd36715ff010e985e25baaf19'
[13:47] <cone-261> ffmpeg.git 03Niels Möller 07master:f2ce63246f5c: dcadec: replace ldexpf with a multiplication by a constant
[13:47] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:118d84b7e937: Merge commit 'f2ce63246f5c934429f9cb857a794e07624d7912'
[13:47] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:a740cae4a080: avcodec/dcadec: remove unsignedness from constant
[14:09] <cone-261> ffmpeg.git 03Carl Eugen Hoyos 07master:076ab9f7e3b1: Fix all VDPAU decoders.
[14:22] <cone-261> ffmpeg.git 03Ronald S. Bultje 07master:cbf21628a5e5: swr: remove div/mod from DSP functions.
[14:32] <cone-261> ffmpeg.git 03Carl Eugen Hoyos 07master:99a8a3de8c65: Revert "Fix compilation on ppc64 and ppc with pic if gas-preprocessor is installed."
[15:32] <ubitux> I suggest we replace the "largely broken" changelog entry about ffserver in the Changelog by a call for maintainer in the introduction (assuming we go with the new layout i'm suggesting)
[15:34] <iive> ubitux: do it.
[15:36] <ubitux> iive: well sure, but my changelog patchset is not oked
[15:36] <ubitux> and the av/ffserver drop is not merged yet
[15:36] <ubitux> (it was just pushed a few minutes ago)
[17:04] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:5439959ef013: avcodec/huffyuvdec: do not leave alpha values uninitialized
[17:19] <cone-261> ffmpeg.git 03Anton Khirnov 07master:894682a9739e: Remove avserver.
[17:19] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:daa7bccad227: Merge commit '894682a9739eb8d4bfc024b0d4e5757fcfe47378'
[17:27] <cone-261> ffmpeg.git 03Anton Khirnov 07master:cfbdd7ffbd9f: rtpenc: base max_frames_per_packet on avg_frame_rate, not codec timebase
[17:27] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:bd148ce07de0: Merge commit 'cfbdd7ffbd9fe14d110fd1bb89bf52f0f7bde016'
[17:31] <nevcairiel> you know, carl will never let that pkg-config patch go through anyway, no matter how many people agree
[17:32] <Daemon404> fuck carl
[17:32] <Daemon404> he is literally the only one.
[17:32] <ubitux> i sent my latest mail on this issue
[17:33] <nevcairiel> should just commit it and see what happens
[17:33] <ubitux> no
[17:34] <ubitux> not really in the mood for a commit/revert war, that won't really help
[17:34] <nevcairiel> he doesn't listen to logical arguments, you're just giving into his insanity?
[17:34] <cone-261> ffmpeg.git 03Anton Khirnov 07master:f792d3cbb8e8: lavf: add the notimestamps flag to the muxers missing it
[17:34] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:95398aa9499b: Merge commit 'f792d3cbb8e8e35c54a9358a55dd596b7a40f228'
[17:34] <Daemon404> ubitux, at what point do you reach a logic consensus
[17:34] <Daemon404> 100 people vs 1
[17:34] <ubitux> well i just want to make sure there is no misunderstanding first
[17:35] <ubitux> and that everyone is pretty aware of the problem
[17:35] <Daemon404> i am not sure carl is capabale of understanding some concepts
[17:35] <Daemon404> given the years of his emails ive read
[17:35] <wm4> so uh can't you just overrule him
[17:35] <ubitux> i don't want that
[17:36] <wm4> time to fork?
[17:36] <ubitux> we're just not agreeing on a technical issue
[17:36] <ubitux> it's not a that simple issue
[17:36] <Daemon404> a lot of carl's agrument is "it will break build scripts"
[17:36] <Daemon404> which is bollocks
[17:37] <ubitux> that's the only one
[17:37] <ubitux> and it's true somehow
[17:37] <Daemon404> i know
[17:37] <Daemon404> but pkg-config > setting cflags/ldflags
[17:37] <Daemon404> i welcome it
[17:37] <ubitux> but that's why i proposed the documentation
[17:37] <ubitux> and all the changelog stuff
[17:38] <ubitux> given all the advantages pkg-config provides, i think that's what we should
[17:38] <ubitux> oh well
[17:38] <nevcairiel> an argument of "it breaks peoples old obscure setups" can be made for many many changes, that doesn't mean we should never do them
[17:38] <wm4> it's a ridiculous argument compared to all the API changes
[17:38] <Daemon404> yes it is
[17:38] <j-b> lol
[17:38] <Daemon404> but how often does carl deal with API issues on bug tracker?
[17:38] <Daemon404> hardly ever
[17:38] <j-b> still fighting on pkg-config?
[17:38] <Daemon404> compared to "im retarded and configure isnt working"
[17:38] <wm4> j-b: yes
[17:38] <Daemon404> which seem to be like 40% of bugs filed
[17:39] <j-b> wm4: it's a joke, right?
[17:39] <Daemon404> his perspective is skewed
[17:39] <wm4> j-b: no, it's cehoyos
[17:39] <ubitux> j-b: hey i mentioned vlc, i hope you're happy ;)
[17:39] <j-b> ubitux: did you
[17:39] <j-b> ?
[17:39] <ubitux> of course
[17:39] <j-b> I skipped over some emails, in the middle, maybe
[17:39] <Daemon404> "vlc usus pkg-config, email j-b for more information"
[17:39] Action: Daemon404 runs
[17:39] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2014-June/159001.html
[17:39] <j-b> I thought it was a joke
[17:40] <ubitux> or well, it was in my shell path
[17:40] <Daemon404> j-b, dont underestimate aspergers patients
[17:41] <j-b> I hate pkg-config as much as anyone, but there are no solutions.
[17:41] <ubitux> what's the problem with pkg-config, aside the glib dep?
[17:41] <j-b> it's ugly :)
[17:41] <j-b> it's not well documented
[17:42] <Daemon404> the --static business is a design oversight
[17:42] <Daemon404> thats my main issues
[17:42] <Daemon404> -s
[17:42] <j-b> PKG_CONFIG_PATH PKG_CONFIG_LIBDIR is a mess
[17:42] <ubitux> ah?
[17:42] <Daemon404> j-b, rule of thumb: PKG_CONFIG_PATH is pretty much never the right env var to set
[17:42] <j-b> ${triplet}-pkg-config is not standard and a mess
[17:42] <Daemon404> despite what google says
[17:42] <j-b> and glib
[17:42] <ubitux> seems it works pretty fine for me
[17:42] <Daemon404> circular glib dep ftw
[17:42] <j-b> oh, yes --static
[17:43] <j-b> but, besides that, it's better than the rest
[17:43] <ubitux> seems you guys agree with cehoyos then
[17:43] <ubitux> you should reply in that thread
[17:43] <ubitux> ;)
[17:43] <Daemon404> nah
[17:43] <j-b> nah
[17:43] <Daemon404> its still less bad than what you have now
[17:43] <j-b> 17:43 < j-b> but, besides that, it's better than the rest
[17:43] <j-b> it's bad, but the other options are 10x worse
[17:44] <ubitux> ok so
[17:44] <ubitux> who's going to update our lib probing with pkg-config?
[17:44] <j-b> michael
[17:44] <ubitux> i mean it's been months that we try to have pkg-config for libx264
[17:44] <ubitux> i wonder how long it will take for the other libs
[17:45] <nevcairiel> i'm fascinated the x265 patch was actually applied :p
[17:45] <ubitux> i really don't mind updating the other libs supporting .pc
[17:45] <ubitux> i just don't want to fight for years to get the patch applied
[17:45] <ubitux> nevcairiel: cehoyos doesn't use it probably
[17:45] <ubitux> it's also pretty recent so it can be changed without the historical background we have for libx264
[17:45] <j-b> what I love is configure scripts that hardcode -lpthread
[17:46] <ubitux> :)
[17:47] <nevcairiel> pthread is a mess tho, the section for supporting pthreads in ffmpeg configure is like 20 lines
[17:47] <j-b> yep
[17:47] <nevcairiel> because of 20 different alternatives on different systems
[17:47] <j-b> pthreadGC2 <3
[17:48] <ubitux> btw, downstreams opinions on the Changelog patch welcome
[17:48] <nevcairiel> why would downstream care about the changelog
[17:48] <nevcairiel> i never read the thing
[17:48] <nevcairiel> :D
[17:49] <ubitux> well, developers don't, downstreams probably do
[17:49] <ubitux> if they don't then... that's probably because the current one sucks
[17:49] <ubitux> i mean, if we document behaviour changes, have the library versions, and some release notes in it alongside the features
[17:49] <ubitux> maybe people will have a quick look to it after a release?
[17:49] <nevcairiel> i follow git more or less on a daily basis to get my change info, but then i'm probably not the most usual downstream
[17:50] <j-b> But, but, who is the true downstream?
[17:50] <ubitux> you are j-b!
[17:50] <ubitux> you are the one this time!
[17:50] <nevcairiel> so does vlc read the changelog?!? :)
[17:50] <ubitux> we'll soon know
[17:54] <cone-261> ffmpeg.git 03Anton Khirnov 07master:d754ed41727b: riffenc: take an AVStream instead of an AVCodecContext
[17:54] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:2d7028225483: Merge commit 'd754ed41727b1fcbab335b510248a9758a73320c'
[18:04] <ubitux> ooh i finally see why he wants this hack
[18:06] <wm4> ooh?
[18:06] <ubitux> If I need FFmpeg with a particular x264 build, I
[18:06] <ubitux> compile x264 and prefer not to run "make install".
[18:06] <ubitux> basically he doesn't want to --prefix=/tmp/x264build
[18:06] <Daemon404> ... thats a stupid reason
[18:06] <ubitux> but actually he doesn't
[18:06] <ubitux> PKG_CONFIG_PATH focusing on the source directly might actually works
[18:06] <ubitux> since the .pc is here
[18:07] <ubitux> not sure though
[18:07] <wm4> ubitux: yes
[18:07] <wm4> mpv-build for example installs everything into a sub dir, and points PKG_CONFIG_PATH there
[18:08] <ubitux> yeah that's obviously the right way of doing this
[18:08] <ubitux> do you see another method?
[18:08] <ubitux> CE insists on having the old way still available
[18:08] <wm4> the configure script could have a generic way to force a lib
[18:08] <nevcairiel> i usually use make install into a custom prefix dir and point everything else at that as well, but of course thats still make install, never tried without
[18:10] <ubitux> should we somehow ignore the probing when --extra-{c,ld}flags are set?
[18:10] <ubitux> sounds like a bad heuristic to me
[18:10] <nevcairiel> thats terrible
[18:10] <ubitux> yes
[18:10] <nevcairiel> i set extra cflags for totally unrelated things
[18:11] <nevcairiel> or you know, the one or two libs that dont use pkg-config
[18:11] <nevcairiel> most old encoders, apparently
[18:11] <wm4> obviously it would need a new per-package flag
[18:11] <wm4> autoconfig might have an existing convention here
[18:13] <ubitux> yes they do, i mentioned it in the previous thread
[18:13] <ubitux> ^thread^mail
[18:16] <nevcairiel> ew he wants his hack to be applied to all other libs as well
[18:22] <Compn> Anton Khirnov master:894682a9739e: Remove avserver.
[18:22] <Compn> whats up with that
[18:22] <Compn> ubitux / wm4 / Daemon404 / nevcairiel : carl said all he wants is a failback if pkg-config fails
[18:23] <Compn> thats not that unreasonable is it ?
[18:23] <nevcairiel> its a ugly hack, so kinda is
[18:23] <ubitux> we do that to avoid a configure hell
[18:23] <ubitux> i think it's almost worth than keeping current hack
[18:25] <Compn> nevcairiel : how many lines of ugly is it ? :P
[18:26] <ubitux> twice as currently
[18:26] <cone-261> ffmpeg.git 03Clément BSsch 07master:5039aadf68de: avcodec/ass: explicit PlayRes[XY]
[18:26] <ubitux> wm4: ^
[18:26] <jamrial> michaelni: seems that your recent changes to libavcodec/bitstream.c are triggering some compiler bug in gcc 4.2.x
[18:27] <jamrial> other versions are unaffected
[18:27] <michaelni> jamrial, any idea why ?
[18:29] <michaelni> also 4.2 is kinda old does this affect something still maintained (aka should this be reported to gcc or is it just us using old compilers? )
[18:29] <jamrial> no, just noticed it on fate. every gcc 4.2.x slot is triggering it, regardless of arch
[18:30] <jamrial> and i doubt gcc is maintaining it
[18:31] <wm4> ubitux: thanks
[18:31] <jamrial> no, seems that 4.7 is the oldest they are maintaining
[18:31] <wm4> ubitux: you could have moved down the PlayRes headers, though (like I suggested)
[18:32] <michaelni> jamrial, ok so we just need to workaround it
[18:36] <ubitux> wm4: ah sorry, you mean below ScriptType?
[18:40] <wm4> yes
[18:40] <ubitux> wm4: will fix in a moment
[18:41] <ubitux> wm4: could you check if some stuff is badly supported in srt decoder in comparison to the mp* one?
[18:44] <wm4> uh, dunno
[18:45] <ubitux> well you said that the mp* code could be better than the ff one for some reasons
[18:46] <ubitux> since i originally wrote the one in mplayer i feel honored, but i'd love to see a better code maintained in ffmpeg nowadays :)
[18:46] <cone-261> ffmpeg.git 03Clément BSsch 07master:88514378bac9: avcodec/ass: move playres parameters below scripttype
[18:46] <ubitux> wm4: ^ fixed
[18:47] <wm4> thanks... it probably doesn't really matter, but good
[18:47] <wm4> the worse issue is that lavc still generates non-standard ASS
[18:48] <ubitux> ah, yeah that one
[18:48] <ubitux> grumbl
[18:48] <ubitux> i suddenly feel extremely tired
[18:48] <wm4> is it that hard to fix?
[18:49] <wm4> I think I even looked at it once, but don't remember
[18:49] <ubitux> yes you did
[18:49] <ubitux> and you realized it was a hell to fix
[18:49] <ubitux> so you looked at utf-16
[18:49] <ubitux> and abandonned too
[18:50] <ubitux> :(
[18:50] <wm4> heh
[18:50] <wm4> with utf-16 it's kind of annoying that it's all a hack
[18:50] <wm4> maybe it would actually be better to read the whole file into memory and convert that...
[18:51] <wm4> what formats did we say need utf-16 support?
[18:51] <ubitux> srt, sami and ass
[18:51] <ubitux> iirc
[18:52] <Compn> working on utf16 now ?
[18:52] <ubitux> wm4 wrote a cool patch already a while ago
[18:53] <ubitux> but it wasn't exactly completely good for applying
[18:53] <Compn> jamrial / michaelni : should we have a txt like apichanges but for known gcc versioned bugs ?
[18:53] <wm4> seems like everyone has his list of gcc versions that break his project
[18:53] Action: Compn looks at mplayer list :D
[18:53] <Compn> ehehe
[18:53] <Compn> anything gcc .0 breaks
[18:56] <Compn> i think they do it on purpose. maybe to see if they still have to fix whatever mplayer keeps breaking
[18:57] <wm4> I think it's more like mplayer breaks gcc
[18:58] <Compn> 4.4.0 and maybe 4.6.0 had problems
[18:59] <Compn> which is why i still think we should hack tcc up to compile ffmpeg and include it in the tree :P
[19:00] <Compn> me and my crazy ideas
[19:00] <ubitux> you mean "stupid" ideas, right? :)
[19:06] <gerion> ehm, short question, when writing a video filter; it is possible, that the resolution changes in the middle of the video stream? (can I use init() to change things that depends on the resolution or do I have to use filterframe etc.)
[19:07] <wm4> AFAIK not for filters
[19:07] <wm4> filters must be reinitialized for that
[19:09] <gerion> does the filter reinitializes itself or is this an extra method? :)
[19:13] <wm4> I assume the external code calls config...
[19:15] <gerion> hmm ok thx
[19:56] <cone-261> ffmpeg.git 03Anton Khirnov 07master:194be1f43ea3: lavf: switch to AVStream.time_base as the hint for the muxer timebase
[19:56] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:ac293b66851f: Merge commit '194be1f43ea391eb986732707435176e579265aa'
[19:56] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:75511c293add: dump_stream_format: fix division by 0
[20:16] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:1d54f5108477: avformat/mux: simplify ff_choose_timebase()
[21:49] <J_Darnley> Does C not let you assign values to a variable like this? point = { x, y };
[21:49] <J_Darnley> Assuming that the type of point is appropriate.
[21:51] <nevcairiel> i believe you need to shove a cast to the matching struct type in front
[21:51] <J_Darnley> oh yes, thank you
[22:07] <Daemon404> and that is c99-only
[22:20] <nevcairiel> well, i assumed its for ffmpeg, where its fine to be used
[22:26] <ubitux> so who is going to add a way of handling CE use case in the configure?
[22:26] <wm4> CE
[22:27] <Daemon404> thats a good one
[22:28] <ubitux> i don't think he plans to do it
[22:28] <wm4> then he'll have to deal with that
[22:35] <ubitux> well since he opposes to the patch unless a hack is available..
[22:47] <nevcairiel> ask him to provide it
[22:47] <nevcairiel> ball in his court etc
[22:47] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:eec55e622a73: ffmpeg_opt: Dont exit on avformat_find_stream_info() failure if there are streams
[22:48] <ubitux> yeah well when i'll get some motivation back
[22:49] <ubitux> i've really better to do
[23:00] <cone-261> ffmpeg.git 03Lou Logan 07master:eba453e03ffe: doc/bitstream_filters: document imxdump
[23:05] Action: llogan trivial_patch_man
[23:22] <cone-261> ffmpeg.git 03Luca Barbato 07master:49a242687cf4: doc: Document segment_list_type
[23:22] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:1ebc77bc7d68: Merge commit '49a242687cf44f86570b706db3c5912ff06bc6c2'
[23:26] <cone-261> ffmpeg.git 03Luca Barbato 07master:c9c1e00f9535: rtsp: Factor out fmtp parsing
[23:26] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:be94f5c171d9: Merge commit 'c9c1e00f95350e7a36b0d05a03939c7f6438e371'
[23:32] <cone-261> ffmpeg.git 03Joshua Kordani 07master:c385313d2590: rtsp: Support misordered fmtp rtpmap
[23:33] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:dacc4cb11513: Merge commit 'c385313d2590fd26e26bb9bdacddef78162ab490'
[23:37] <cone-261> ffmpeg.git 03Luca Barbato 07master:869fc416f7c7: hevc: Mark num_bins_in_se as unused
[23:37] <cone-261> ffmpeg.git 03Michael Niedermayer 07master:8e2ade43f1d0: Merge commit '869fc416f7c78ed4e397e0208acd1545771c0502'
[00:00] --- Thu Jun 19 2014
1
0
[00:45] <Smashcat> Hi, anyone know how I should convert a webm to mp4 with ffmpeg? If I try with ffmpeg i.webm o.mp4 the video kind of works, but audio frames seem to be out of order. Same webm plays perfectly in a browser (Firefox)
[00:47] <Smashcat> llogan: Here's the session - http://pastebin.com/ukGA1Bdn
[00:47] <llogan> you're using a counterfiet "ffmpeg" from a fork.
[00:47] <llogan> you'll have to get help from the fork, or use the real ffmpeg
[00:48] <Smashcat> llogan: Oh right, wasn't aware of that
[00:48] <llogan> that was their intention
[00:51] <Smashcat> Got it thanks. Got loads more errors in the output. Just capturing to a file...
[00:52] <llogan> you're probably wasting your time using that
[00:53] <Smashcat> er, the version you linked?
[00:53] <llogan> no. i assumed you were referring to the fake one
[00:54] <llogan> if you're using a recent static build you can show your command and console output again
[00:54] <Smashcat> Yep that's what I used.
[00:54] <llogan> and providing the input file will be good too
[00:55] <Smashcat> llogan: I can't provide that unfortunately - it's from another person and I don't have permission
[00:55] <kamil> Hello, is there any way to transcode video with Intel® Xeon Phi Coprocessor 5110P (8GB, 1.053 GHz, 60 core)?
[00:55] <kamil> Or maybe in Nvidia Tesla coprocessor?
[00:56] <llogan> Smashcat: then we'll just have to look at the command and console output
[00:56] <Smashcat> Yup, here it is: http://pastebin.com/yyXQvawC
[00:57] <Smashcat> The webm does play perfectly in Firefox. No stuttering or audio issues
[00:58] <llogan> what are you using to play the MP4 output?
[00:58] <Smashcat> vlcplayer
[01:00] <Smashcat> Oh, actually if I drop the mp4 into Firefox it plays fine. Sorry, seems to be a vlcplayer issue. I'll try uploading it to Youtube. This is the reason I was transcoding,as YT seems to have problems with the webm too.
[01:01] <llogan> that's ironic
[01:01] <llogan> how was the webm created?
[01:01] <Smashcat> There's a page for recording webms through the browser. I'll try to find it...
[01:01] <llogan> does ffplay also have trouble playing the output?
[01:02] <Smashcat> I can't use ffplay as I'm running Linux on a headless server.
[01:04] <llogan> if you need to re-encode for whatever reason, use -crf 18 (roughly visually lossless) and/or -qp 0 (lossless) and upload the smaller of the two to YouTube.
[01:04] <Smashcat> Here's the page that has the recorder. Works in Firefox only: https://www.webrtc-experiment.com/RecordRTC/AudioVideo-on-Firefox.html
[01:04] <Smashcat> After recording a clip you can download the webm
[01:06] <llogan> their example video doesn't even play for me in FF 29
[01:06] <llogan> oh, user error.
[01:07] <Smashcat> I'm on FF 30 here, but I think it works in earlier versions
[01:08] <llogan> produced a black video.
[01:41] <Smashcat> Hmm, seems that Quicktime and Firefox play it fine on Windows. VLCPlayer plays with corrupted audio, and Windows Media Player just looks a mess - corrupted video and no audio.
[03:24] <rindolf> Hi all.
[03:25] <rindolf> Hi all! ffmpeg -i ./Buffy.The.Vampire.Slayer.S03E01.AC3.DVDRip.x264-iND.mkv -ss 35:33 -to 42:08 ~/buffy-is-not-a-nobody.mkv ==> why does this command get stuck?
[03:27] <rindolf> sacarasc: OK, sorry, thanks.
[03:27] <rindolf> sacarasc: i'll do that.
[03:30] <rindolf> sacarasc: http://paste.debian.net/105565/ - here y'all go.
[03:34] <rindolf> Can anyone tell me what's wrong?
[03:34] <sacarasc> rindolf: Maybe add -c copy just after the -to 42:08
[03:34] <rindolf> sacarasc: thanks! Let me try.
[03:37] <rindolf> sacarasc: great! It appears to have worked.
[03:37] <rindolf> sacarasc++
[03:38] <rindolf> sacarasc: if we ever meet (I am http://www.shlomifish.org/ ) - please let me know and I can buy you a drink.
[03:38] <sacarasc> Heh, okay.
[03:38] Action: rindolf also gives 1,000,000 love points to sacarasc . e!
[04:13] <rindolf> sacarasc: http://www.shlomifish.org/Files/files/video/buffy-is-not-a-nobody.mkv - yay! Thanks. Took a while due to my crappy upstream.
[04:21] <rindolf> Now I can write a G+ post or maybe reduce the quality and youtube it.
[06:46] <benlieb> can anyone help me get the output of an ffprobe command to actually go to standard out?
[06:46] <benlieb> ffprobe -print_format json -hide_banner 999.mp4
[06:47] <benlieb> prints the json to the terminal, but when I try to pipe or store in a var, I get an empty { }
[06:57] <klaxa> try: ffprobe -print_format json -hide_banner 999.mp4 2>&1
[06:57] <klaxa> that will redirect stderr to stdout
[06:58] <benlieb> klaxa: yeah, figured that out. Why does it go there to begin with?
[06:59] <klaxa> programming convention i think
[06:59] <klaxa> also because it probably uses the built-in functions of av_*
[06:59] <klaxa> which output to stderr
[06:59] <benlieb> klaxa: i thought the convention is for output to go to stdout and errors to stderr. this isn't an error
[07:00] <klaxa> you wouldn't be able to use ffmpeg's output for pipes then
[07:00] <klaxa> because binarystreams would be mixed with console output
[07:01] <benlieb> but this is ffprobe, not ffmpeg.
[07:01] <BtbN> but it's the same library.
[07:01] <benlieb> anyway, the json output is making my ruby choke :(
[07:01] <benlieb> unexpected token at 'Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/Ben Lieb's Red External HD/storage/idance/previews/mp4/86.mp4'
[07:02] <benlieb> probably because of my dir name containing a ' ?
[07:02] <klaxa> probably
[07:02] <BtbN> just look at the json yourself and see?
[07:02] <klaxa> can you pastebin everything?
[07:02] <klaxa> command and output
[07:03] <benlieb> BtbN: well the json is a bit confused:
[07:03] <benlieb> https://gist.github.com/pixelterra/6420bddd1ead063b10ff
[07:03] <benlieb> the { and } come after the text
[07:04] <klaxa> that looks broken indeed
[07:04] <klaxa> can you pastebin the complete command and output? not only the output
[07:04] <benlieb> updated: https://gist.github.com/pixelterra/6420bddd1ead063b10ff
[07:04] <klaxa> it will also print your version information and stuff i think
[07:04] <benlieb> yeh just a sec
[07:07] <benlieb> It's somewhat confused because it's in a ruby script:
[07:07] <benlieb> https://gist.github.com/pixelterra/6420bddd1ead063b10ff
[07:09] <benlieb> this is kinda messed up. I'm just trying to get the duration for a movie clip? What's the easiest way to do that programmatically.
[07:10] <benlieb> klaxa: ^
[07:10] <klaxa> phew... not sure, ffprobe should actually do the job
[07:11] <klaxa> maybe try mediainfo instead?
[07:11] <klaxa> but that won't change the parsing error regarding the filename
[07:11] <benlieb> when I use the command on the command line the { and } are in the right place
[07:11] <benlieb> ffprobe -print_format json -hide_banner 999.mp4 2>&1 | echo -
[07:12] <klaxa> have you tried it with a different path?
[07:12] <benlieb> but when I pipe: ffprobe -print_format json -hide_banner 999.mp4 2>&1 | vi -
[07:12] <klaxa> maybe it is the '
[07:12] <benlieb> the { and } are in the wrong place
[07:12] <benlieb> I think there's two issues.
[07:12] <klaxa> that's weird
[07:12] <benlieb> I manually made the string that's produced by ffprobe and tried to parse it but got the unexpected token error
[07:13] <drv> that doesn'
[07:13] <drv> t look like the json output
[07:14] <drv> it should look like http://stackoverflow.com/questions/7708373/get-ffmpeg-information-in-friend…
[07:14] <benlieb> https://gist.github.com/pixelterra/2c13a50f1cf8f181c9ea
[07:14] <klaxa> regarding the curly braces being wrong, i can confirm that, but let me check something first
[07:15] <klaxa> okay so ffprobe is passing the braces through stdout
[07:15] <klaxa> and the information through stderr
[07:15] <klaxa> which is kinda... stupid
[07:16] <benlieb> the show_streams options in your link seems to help...
[07:20] <benlieb> ok I think I've gotten it close to working. Is this the easiest way to get the duration?
[07:20] <benlieb> klaxa: ^
[07:21] <klaxa> i don't know, probably?
[07:21] <klaxa> i don't know of any easier way
[07:21] <llogan> you only want the duration value?
[07:22] <benlieb> llogan: for now, yes.
[07:22] <llogan> some examples here: https://trac.ffmpeg.org/wiki/FFprobeTips
[07:22] <llogan> see the basic example first
[07:22] <benlieb> there's actually a very good ruby gem that furs this in to OOP, but it's ruby version is incompatible with mine..
[07:24] <benlieb> *turns* this into...
[07:53] <benlieb> llogan: yay. and 2 hours later I have: https://gist.github.com/pixelterra/51bbfdcf0d39cd02b98c
[07:53] <benlieb> which lets me do FfprobeObject.new(f).duration
[07:53] <benlieb> and later when I want another attribute, I can just beef up that class.
[07:54] <benlieb> I do have a question about stream order, though.
[07:54] <benlieb> is the video stream always first?
[07:54] <llogan> i've seen files where ffmpeg shows the audio first.
[07:55] <llogan> in fact you can make them like that with -map: ffmpeg -i input -map 0:a -map 0:v output. (IIRC)
[07:56] <llogan> if you only want video you can use -select_streams
[07:56] <benlieb> llogan: smart, that. will make a note
[07:57] <llogan> and then use a stream selector like 0:v:0, and remember that an input may contain several video streams
[07:58] <llogan> s/selector/specifier
[07:58] <benlieb> llogan: yeah, that's what worries me about this class, but I'll probably only be dealing with single stream video
[07:59] <benlieb> I can always cross that bridge when I come to it. All the vids should only have one stream anyway, but there may be some before my time that have more. I'd likely want to modify the video and not the class, because extra streams are useless for us.
[08:42] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ?
[10:17] <Guest30724> hello
[10:18] <Guest30724> I wish to just restream a mp4 file as it is over http protocol
[10:18] <Guest30724> Can anyone help
[10:26] <pagioss> hi guys
[10:26] <pagioss> i am trying to stream my desktop+audio using ffmpeg, i need them to be synch'd so thought of using RTP any idea how to do that?
[10:44] <sacarasc> Guest30724: Where is the mp4 file currently?
[10:47] <sacarasc> Guest30724: If it is a normal file on your computer, chances are that you'd need to do something to it... (faststart or segmenting)
[10:51] <pagioss> sacarasc: help
[10:52] <sacarasc> I have never done anything with RTP.
[10:52] <sacarasc> And for desktop streaming, I use OBS anyway... https://obsproject.com/
[10:52] <sacarasc> Not sure if it works on *nix, though.
[10:53] <pagioss> sacarasc: does it work on mac?
[10:53] <sacarasc> It might if you compile it...
[10:54] <sacarasc> They are working on making it cross platform in the future.
[10:54] <pagioss> sacarasc: it does not using RTP
[13:47] <decembersoul> Hello
[13:47] <decembersoul> I'm trying to set up an mjpeg->ffserver->h264 stream.
[13:47] <decembersoul> Basicly i have taken the example from https://trac.ffmpeg.org/wiki/Streaming%20media%20with%20ffserver#StreamingH…
[13:47] <decembersoul> I don't have audio so i have set NoAudio
[13:47] <decembersoul> But i have some problems:
[13:47] <decembersoul> 1)
[13:47] <decembersoul> How can i define that the h264 stream has the same VideoSize then the input?
[13:47] <decembersoul> If i don't set the VideoSize within my ffserver.conf the stream will take 160x128 and not the input size 1280x720
[13:47] <decembersoul> 2)
[13:47] <decembersoul> I tried to switch the Format to mpegts, but with mpegts vlc did not show anything.
[13:58] <azk> Afaik, you will have to specify a VideoSize in the ffserver and scale to that. I had no luck with any other kind of setup.
[14:05] <decembersoul> that is how it looks like.
[14:05] <decembersoul> But i want the same videoSize then the input. No scale.
[14:23] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ?
[14:29] <saste> decembersoul, i don't think that's possible
[14:29] <saste> decembersoul, but you might try with -override_ffserver
[14:29] <saste> in this case ffmpeg will ignore the ffserver parameters
[14:30] <decembersoul> ok, i will try
[14:30] <saste> indeed that's the trick used to allow filtering
[14:43] <decembersoul> it works. Grate job!!!
[14:49] <mdjp> Apologies if this is a daft question. I'm wondering if there is any way to disable the low pass filter applied to 5.1 encodes using ffmpeg. My simple command line is currently ffmpeg -i test_source.wav -c:a libfdk_aac -ac 6 -b:v 448k out2.mp4
[14:49] <mdjp> It would be great if I could preserve the LFE channel as a full bandwidth channel.
[15:15] <termos> I'm getting "Input picture width ... is greater than stride" error messages when I use the fps filter in my filter graph. Is there something I should look out for when using it?
[16:41] <Guest68998> hi im trying to cross compile ffmpeg with libaac plus and im stuck with the configure
[16:41] <Guest68998> im doing : ./configure --enable-cross-compile --cross-prefix=${CROSS_COMPILE} --arch=arm --target-os=linux --prefix=/home/sam/arm-build/ --enable-nonfree --enable-libaacplus --extra-cflags="-I/~/arm-build/include" --extra-ldflags="-L/~/arm-build/lib" --extra-libs=-ldl
[16:42] <Guest68998> I get: ERROR: libaacplus >= 2.0.0 not found
[16:42] <Guest68998> even if i installed 2.1.1
[16:45] <Guest68998> with : wget http://217.20.164.161/~tipok/aacplus/libaacplus-2.0.2.tar.gz tar -xzf libaacplus-2.0.2.tar.gz cd libaacplus-2.0.2 ./autogen.sh --with-parameter-expansion-string-replace-capable-shell=/bin/bash --host=arm-linux-gnueabihf --enable-static --prefix=~/arm-build make make install
[16:46] <Guest68998> i meant 2.0.2
[16:50] <klaxa|work> cross compiling is black magic to me
[16:55] <Mista-D> has anyone ported Sox volume filter to FFmpeg with a limiter (without the compand)?
[18:17] <llogan> mdjp: why do you want to do that? more users keep wanting to do this
[18:24] <mdjp> llogan glad to hear I'm not the only one! In this case I want to use a 5.1 stream to provide audio to a browser, specifically the web audio api for processing. However the audio is a combination of stereo and mono feeds, not a surround feed
[18:24] <mdjp> currently I send silence over the LFE but having the extra channel would make things much easier
[18:26] <JEEB> if you have ffmpeg understand that you have LFE, then, if your output has LFE as well, it will be kept
[18:26] <JEEB> only when you mix into a channel mapping that has no LFE, then it will not get mixed in because that's the standard
[18:40] <mdjp> JEEB yes I was just wondering if there was a way to overide the filter. 6.0 would be a channel layout that would work but doesn't seem to be supported by the libfdk_aac encoder
[18:48] <llogan> mdjp: maybe look at lfe_mix_level (i've never tried it). or you can use the pan audio filter but watch for overflows/clipping and may result in lowered volumes
[19:04] <mdjp> llogan thanks for the advice, I'll give it a go
[19:41] <lacx> hey all X)
[19:42] <lacx> anyone familiar with RTSP streaming with ffmpeg? Think I may have come across a bug
[19:43] <JEEB> feel free to say more
[19:43] <JEEB> also there's a trac issue tracker on ffmpeg.org
[19:45] <lacx> well Im doing some testing for an app im making, basically it streams from the device to a remote server with crtmpserver and ffmpeg, Im seeing diff behaviour in diff circumstances (i.e. wifi vs 3g) I took a WireShark trace and in the one its failing on (3g), there is an extra PAUSE command coming from ffpeg
[19:46] <lacx> ..ffmpeg* which is not incorrect by itself, other than the next PLAY command does not include the session id, and so the connection fails
[19:46] <lacx> yea saw that thanks JEEB
[19:46] <lacx> thought I would ask here first in case I was being stupid lol X)
[19:47] <lacx> I have set it up so that once the stream is established, ffmpeg is invoked to record the stream to the server
[19:48] <lacx> checking the spec/RFC, it seems to suggest the PLAY command (following a PAUSE) should contain the session id for the stream
[19:49] <JEEB> lacx, have you tested with current ffmpeg btw?
[19:49] <JEEB> or some older version only?
[19:49] <JEEB> just in case it has changed lately
[19:49] <lacx> http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-40#section-13.6
[19:50] <lacx> havent tested with latest source no
[19:50] <JEEB> http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=rtsp
[19:50] <JEEB> searched the git repo with 'rtsp'
[19:50] <lacx> ok thanks
[19:50] <lacx> I have: ffmpeg version N-58822-g6f7dbb9
[19:50] <JEEB> december 2013
[19:51] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=6f7dbb9
[19:51] <JEEB> since there have been some changes, you could check current code
[19:51] <JEEB> and then poke ffmpeg-devel and/or the trac
[19:51] <JEEB> both are useful
[19:51] <JEEB> #ffmpeg-devel I mean
[19:52] <lacx> thanks X)
[19:52] <lacx> very helpful JEEB
[19:53] <lacx> FOSS FTW! X)
[19:53] <JEEB> (also if you get dumb replies from cehoyos on the trac if and when you make an issue, just take a deep breath and poke #ffmpeg-devel again ;) )
[19:55] <lacx> hehe cheers
[19:55] <lacx> whats the diff between git.videolan.org|github/ffmpeg ?
[19:56] <JEEB> github is a mirror
[19:59] <lacx> ah cool
[21:09] <RadSurfer> I'm looking for a way to convert my camcorders .mp4 files to .avi or .flv but in HIGHEST possible quality, any suggestions please?
[21:10] <JEEB> just try to copy the original bit stream over? :P
[21:10] <iive> -c copy
[21:10] <JEEB> since the other alternative is lossless coding, and that most probably will be much bigger than the original (probably lossy compressed) clip
[21:10] <iive> btw, flash can play .mp4 directly
[21:10] <RadSurfer> hmm. I've just been using "-i file.mp4 file.avi", and its crap
[21:11] <JEEB> welcome to ancient defaults
[21:11] <JEEB> aka "never trust ffmpeg to do the right thing if it's actually re-encoding"
[21:11] <RadSurfer> I may want to re-size for sake of file size
[21:11] <RadSurfer> I hope it handles 16:9 .mp4 camcorder formats
[21:11] <JEEB> you get mpeg-4 part 2 at 200 or 300kbps or so with lulzy MPEG-1 level defaults I think?
[21:11] <JEEB> if you don't specify anything with AVI
[21:12] <iive> defaults that ensure realtime encoding on 400mhz P2
[21:12] <JEEB> (back in like 2000 or so it seemed to be a good idea to default all encoders to the same feature level)
[21:12] <RadSurfer> I'm attempting .mkv now, just to see how it compares...
[21:12] <JEEB> well, matroska will get H.264 selected by default (the libx264 encoder) if that's available
[21:12] <JEEB> and that one has actual special code paths
[21:12] <JEEB> so that libx264's defaults are sane'ish
[21:13] <JEEB> so you can't really compare them like that since it just depends on what is selected by default by ffmpeg :P
[21:13] <RadSurfer> camcorder produced 798mb for a 20min .mp4
[21:13] <RadSurfer> 16:9, default camera settings, yada-yada.
[21:14] <RadSurfer> I'd like to try to get that down to as much as possible, w/@least same quality
[21:14] <RadSurfer> converting to .mkv certainly taking longer :-) quadcore 3.6GHz system
[21:15] <JEEB> yes, if it picked libx264 encoder for H.264 it is one of the few lossy encoders with sane defaults
[21:15] <RadSurfer> I installed 64-bit ffmpeg
[21:15] <JEEB> "at least same quality" is not possible, because you most probably want lossy compression and that by definition won't be of the same quality, and the second mistake is that you can't have more than the quality that you already have
[21:16] <JEEB> visually similar quality is of course possible, but depending on your wishes it might be larger or smaller. libx264 + slowest preset that is still fast enough for you + higher -crf value that still looks good for you (defaults are -preset medium and -crf 23)
[21:17] <RadSurfer> it reduced it to 233mb :) how does it look?
[21:17] <RadSurfer> some mild interlacing? what can I do about that? otherwise, its not too bad!
[21:18] <JEEB> if it's interlaced you need to add yadif there, -vf yadif after the input (-i)
[21:18] <RadSurfer> thank you.
[21:18] <JEEB> and yes, crf 23 can be good enough for quite a few use cases
[21:18] <JEEB> you can tweak the crf value up and see if you can take it higher (-crf XX after input)
[21:19] <JEEB> if it looks good, go higher. if it looks bad, go lower
[21:19] <JEEB> that way you should find the highest crf value that still looks good enough for you for that kind of content
[21:19] <JEEB> also you can poke the -preset option (default = medium), set after input as well, for speed VS compression
[21:20] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset
[21:20] <JEEB> listing of the presets
[21:20] <JEEB> do note that the presets do switch various calculations to be less or more accurate, so the result of the same CRF value between them is not static
[21:21] <RadSurfer> Would anyone here have a good recommendation of a program that can properly edit .mp4 files? editing/slicing, etc?
[21:21] <JEEB> (for example it often is that while medium->veryslow ends up smaller, veryslow->placebo actually ends up growing, because the encoder "sees" more things)
[21:23] <llogan> RadSurfer: if you just want to trim shit: ffmpeg -ss <offset time> -i input -t <duration> -c copy -map 0 output
[21:25] <llogan> if your editor is giving you The Troubles with .mp4 then a temporary lossless intermediate may be an option: ffmpeg -i input -c:v utvideo -c:a pcm_s16le output
[21:27] <RadSurfer> trying that recommendation...
[21:28] <llogan> depending on your editor of course, which is unknown, you may have to perform an additional step (such as installing UT video then restarting Premiere)
[21:28] <llogan> or it will probably "just work" with KDenlive, AFAIK
[21:29] <llogan> with PP you can probably output to AVI i guess
[21:30] <llogan> i mean for PP
[21:34] <RadSurfer> yeah, copy produces a file of equal or greater size, but quality is there.
[21:34] <RadSurfer> its H264 of course... hmmm
[21:34] <llogan> copy copys so it is just copying
[21:35] <RadSurfer> its a different file extension: is it transcoding?
[21:35] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-copy
[21:37] <llogan> note that you can't use filters (-vf, -af, -filter:v, -filter:a, -filter_complex, or whatever other alias) when stream copying
[21:37] <RadSurfer> I wish this documentation was in PDF format...
[21:37] <RadSurfer> but I'll make time to read docs from http://ffmpeg.org/ffmpeg.html
[21:38] <llogan> the docs refer to the latest version (they are synced daily), so you should refer to your local copy if you are using something slightly older
[21:39] <llogan> the *online* docs
[21:40] <llogan> as for pdf here is something stolen from relaxed: http://pastebin.com/C4advjBM
[21:40] <RadSurfer> just now d/l 64-bit version for Windows
[21:40] <llogan> (i've never tried it)
[21:41] <RadSurfer> that is a bash command? I doubt that works in win8, lol
[21:41] <llogan> i assumed linux of course
[21:42] <RadSurfer> sadly, not this time.
[21:42] <RadSurfer> Win8.1 is nothing @all to write home about....
[21:43] <RadSurfer> I should be timing how long it takes to do this....
[21:44] <llogan> what editor are you using?
[21:45] <RadSurfer> notepad++
[21:45] <RadSurfer> text?
[21:46] <RadSurfer> I was searching for a decent video editor, don't have one yet
[21:46] <llogan> video. you mentioned an editor before.
[21:46] <RadSurfer> there used to be something called "Virtual Dub", is that still around?
[21:46] <llogan> you could give this a try: http://www.shotcutapp.com/
[21:48] <RadSurfer> http://sourceforge.net/projects/mp4cam2avi/
[21:51] <RadSurfer> what did you do! Using your options went from 798mb .mp4 down to 147mb .mkv!!!
[21:52] <RadSurfer> that is beautiful! Thank you!
[23:53] <Spacexplosion> Why could it be that when I do "ffmpeg -codecs" I see both srt and subrip having DE (decoding and encoding) support, but when I just try to copy a subrip track I get a "Unsupported codec" error?
[00:00] --- Thu Jun 19 2014
1
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[00:02] <cone-497> ffmpeg.git 03Michael Niedermayer 07master:2fbdfba0f2d1: avformat/flvdec: Support live flv / NGINX RTMP streams
[02:37] <cone-497> ffmpeg.git 03Michael Niedermayer 07master:61df0819d49b: avcodec/cavsdec: fix low delay decoding
[03:27] <jamrial> mmh, plepere is not here
[03:27] <jamrial> just read the logs and found the avx2 discussion
[03:27] <jamrial> he wants vpbroadcastw to do a SPLATW with ymm registeres
[03:27] <jamrial> i guess we should expand the macro to use that
[09:14] <plepere> got SDE working ! I can check my AVX2 code. :D
[09:17] <jamrial> plepere: you need to use vpbroadcastw if you want to do the equivalent of SPLATW for ymm regs with avx2
[09:17] <plepere> jamarial : ok thanks
[09:18] <jamrial> no prob
[09:19] <jamrial> we probably should expand SPLATW to use vpbroadcastw when cpuflags == avx2 (it works for both xmm and ymm regs)
[09:20] <jamrial> that way you can avoid some ifdeffery
[09:20] <plepere> jamrial, it would be better if someone could patch SPLATW to avoid self modification.
[09:46] <durandal_1707> i can't get expansion like metadata in drawtext filter to work
[09:46] <durandal_1707> this worked when i added that feature
[09:57] <ubitux> durandal_1707: we have no test for drawtext :(
[09:58] <ubitux> we need a "bitexact" mode with freetype
[09:58] <ubitux> it might even exists
[09:59] <durandal_1707> how you otherwise could display values from signalstats filter...
[09:59] <ubitux> ffprobe
[10:01] <durandal_1707> how?
[10:04] <ubitux> with movie source
[10:05] <ubitux> ffprobe -f lavfi movie=bla,signalstats -of flat -show_entries frame=tags
[10:05] <ubitux> or something like that
[10:05] <ubitux> don't remember exactly
[10:07] <ubitux> -show_entries frame_tags
[10:09] <plepere> ok, that's a new one. Ubitux can you take a look at my code quickly ?
[10:09] <ubitux> maybe :p
[10:09] <plepere> please !!!
[10:10] <ubitux> ;)
[10:10] <plepere> http://pastebin.com/p382ppvA
[10:10] <plepere> there's the SSE2 idct16 that(s working, but the AVX2 isn't
[10:11] <plepere> I get a very strange error
[10:11] <plepere> if I pass the %rep to 5, it's just an error.
[10:12] <ubitux> what error?
[10:12] <plepere> but if I go beyond, I get munmap_chunk(): invalid pointer or double free or corruption (out)
[10:12] <plepere> md5 errors
[10:13] <ubitux> did you look at the generated load/store instructions?
[10:13] <plepere> (which is logical, since by doing rep 5, I'm doing a 24x32 block
[10:14] <plepere> no. :/
[10:14] <plepere> but even by forcing vmovdqu I have these errors
[10:16] <ubitux> prototype is weird
[10:16] <ubitux> isn't it missing a digit?
[10:17] <plepere> owait
[10:17] <plepere> the 8 got me confused, indeed.
[10:18] <plepere> but it doesn't change the bug.
[10:18] <plepere> do you think it's code size that is bugging things ?
[10:19] <plepere> since just changing the %rep makes or breaks the build
[10:23] <ubitux> plepere: i odn't think so
[10:23] <ubitux> mmh "breaks the build"? how?
[10:25] <plepere> when I have %rep 5, there is no real error. it's just the decoding that is incorrect. When %rep is 6 or 7, I get the errors.
[10:27] <plepere> munmap_chunk(): invalid pointer or double free or corruption (out)
[11:13] <plepere> ubitux, I've found the problem. Next time I'm coming for help, remind me that I'm an idiot and that I missed something really obvious
[11:17] <plepere> in this case, I was calling the idct32 instead of the idct16. so it's logical that increasing the %rep was the problem since it was bringing me out of bounds
[11:32] <plepere> michaelni, is my patch OK ? I've added idct_dc_add avx2 functions (conformance takes a long time since I'm on an emularor) do you thinkI should do a 2nd patch with only them or re-submit the whole thing ?
[13:02] <michaelni> plepere, 2nd patch with only them is probably better as it keeps the patch smaller
[13:04] <michaelni> plepere, is the latest patch on the ML or there are some changes that wherent submited (except the avx2 idct stuff)
[13:05] <plepere> michaelni, the last fix was sent on monday
[13:05] <michaelni> ok
[13:07] <cone-605> ffmpeg.git 03Anton Khirnov 07master:71d6551e674c: avconv: use the correct variable in comparison
[13:07] <cone-605> ffmpeg.git 03Michael Niedermayer 07master:fa0d0fb42ecd: Merge commit '71d6551e674c86d07c165439c2bf6613b3892307'
[13:27] <cone-605> ffmpeg.git 03plepere 07master:92cccb7bcd79: avcodec/hevc: new idct + asm
[15:55] <ubitux> > Support Ogg/Vorbis in avi
[15:55] <ubitux> ._.
[15:57] <wm4> ubitux: mpv is not using the srt converter, because AFAIK there are fixes to the mpv one that are not in the ff srt one
[15:57] <wm4> these were added back in mplayer2 times
[15:57] <wm4> I'm not sure if ff has them
[15:57] <wm4> also, the ff converter doesn't set the subtitle resolution, which is annoying
[16:00] <ubitux> i didn't write the srt decoder in ffmpeg, Aurelien did
[16:00] <ubitux> and he did that after i wrote it in mplayer iirc
[16:00] <wm4> oh, annoying
[16:00] <ubitux> but afaik it's pretty complete
[16:01] <ubitux> what's this subtitle resolution thing?
[16:01] <ubitux> ah, yeah
[16:01] <ubitux> mmh that's an issue with all the decoders right?
[16:01] <ubitux> should we add an option for that?
[16:02] <wm4> why an option?
[16:03] <ubitux> how else would you control it?
[16:05] <wm4> why control it?
[16:07] <wm4> the resolution controls font size and how \pos tags are used
[16:08] <wm4> if you don't specify a resolution, libass will pick a static fallback value that is consistent with scripts from 10 years ago that lacked the resolution
[16:09] <ubitux> yes but what resolution should be set?
[16:10] <wm4> the fallback resolution, since apparently the code was designed for it
[16:10] <wm4> 384x288 AFAIK
[16:12] <ubitux> what would be the difference with the static fallback?
[16:13] <ubitux> http://pastie.org/9299198 ?
[16:13] <ubitux> sth like that would be fine for you?
[16:14] <ubitux> i'm not sure how that would help
[16:15] <ubitux> avoid a warning maybe? :P
[16:16] <iive> in theory, the pos should have been in the [0;1) range, or % for convenience
[16:19] <wm4> yes avoid a warning
[16:20] <wm4> I'm not sure if headers are order dependent (maybe not), but normally the playres comes after the script type
[16:20] <wm4> iive: that wouldn't change much
[16:28] <wm4> is there a function that can split avoptions out of a flat string, set them on some context, _and_ put missing options into a AVDictionary?
[16:29] <wm4> it's what you typically need for opening some things (like avio contexts or demuxers)
[17:09] <Mista-D> anyway to boost volume with a limiter?
[17:09] <Mista-D> sorry wrong window.
[17:38] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:a50c6f99b44e: ffplay: remove -ec handling code
[19:35] <llogan> Timothy_Gu: you forgot to send your patch for #3616
[19:36] <llogan> *3613
[21:36] <cone-460> ffmpeg.git 03Diego Biurrun 07master:fe4d5fe93611: jpeg2000: Mark static data init functions as av_cold
[21:36] <cone-460> ffmpeg.git 03Michael Niedermayer 07master:f9cd317223cb: Merge commit 'fe4d5fe9361162f9033ff1bd84bfc1b2091ba785'
[23:07] <ubitux> oh, git didn't like the unicode 7 character :(
[23:11] <ubitux> i guess i'll go for
[23:19] <wm4> what does git reject and why?
[23:20] <ubitux> no idea, didn't look much further
[23:20] <ubitux> https://ffmpeg.org/pipermail/ffmpeg-devel/2014-June/158969.html
[23:20] <ubitux> 🖔 was the character
[23:21] <wm4> also git normally doesn't reject anything, so this is some ffmpeg specific hook?
[23:21] <ubitux> dunno where the failure(s) occur, maybe that's not even in git
[23:22] <wm4> well, what part blows up?
[23:22] <nevcairiel> my font doesnt have your character anyway
[23:23] <ubitux> wm4: well at least on the mailing list it's broken
[23:23] <nevcairiel> but it does the star
[23:23] <jamrial> why did you delete all the entries from older releases?
[23:23] <ubitux> because it doesn't matter in the release itself
[23:33] <Youka> ah, cmon: http://pastebin.com/Rs4mpD7r
[23:34] <Youka> Building ffmpeg 2.2.3 on windows with mingw (tdm-gcc 32bit) results in error above.
[23:34] <Youka> btw.: with msvc it worked.
[23:48] <Youka> Could it be that it expects gcc only features which mingw doesn't have? Even TDM-GCC with GCC v4.8.1 hasn't nanosleep?
[23:50] <wm4> that depends on the headers, not gcc
[23:50] <wm4> I suspect there's something wrong with the configure script though
[23:50] <Youka> Ok, i found out that MinGW has no timespec or nanosleep, it remains unique for unix gcc.
[23:51] <wm4> again this has nothing to do with gcc
[23:51] <Youka> so ffmpeg configure should switch to another HAVE_*** for the an available sleep function with mingw
[23:51] <wm4> hm the timespec use is guarded with #if HAVE_CLOCK_GETTIME && defined(CLOCK_MONOTONIC)
[23:52] <wm4> and <time.h> is included
[23:53] <wm4> so, the mingw headers must somehow be inconsistent
[23:53] <wm4> though there are more correct ways to check for the presence of clock_gettime
[23:54] <jamrial> Youka: what version of mingw are you using. mingw32? mingw-w64 v2, v3, trunk?
[23:54] <wm4> in my own code I'm using: #if defined(_POSIX_TIMERS) && _POSIX_TIMERS > 0
[23:54] <Youka> TDM-GCC (MinGW) 32bit with GCC 4.8.1
[23:55] <jamrial> that doesn't say much. it could be using either mingw32 or mingw-w64 compiled for x86_32
[23:56] <Youka> mingw32
[23:57] <wm4> and the nanosleep thing shows that the configure script did something wrong
[23:57] <wm4> or the one running the configure script
[23:57] <Youka> I don't think, i reloaded ffmpeg 2.2.3 to be sure all is clean, then run configure and make like shown in pastebin
[23:58] <Daemon404> ... you dont happen to have messed with include / lib dirs or the env have you
[23:58] <Youka> no, i haven't
[23:58] <Youka> all like on install
[00:00] --- Wed Jun 18 2014
1
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[00:00] <ninjas_> http://pastie.org/9296689
[00:00] <ninjas_> any brilliant insight into this
[00:00] <ninjas_> before I murder a village?
[00:00] <c_14> sh does not have () syntax, use bash
[00:01] <ninjas_> I hate to ask this, but can you give me an example back? I'm about to have a heart attack
[00:01] <c_14> In line 1, replace sh with bash
[00:02] <ninjas_> I'm such an idiot
[00:03] <ninjas_> I would have never even thought to look there
[00:04] <c_14> Also, I think the < should be a $
[00:05] <c_14> unless you want the fifo
[00:05] <c_14> which you actually might, looking at it
[00:10] <ninjas_> here we go
[00:12] <ninjas_> wow
[00:33] <ninjas_> do images have to start wtih 000.jpg
[00:33] <ninjas_> or can it start with 001.jpg and be sequential
[00:33] <c_14> You can start with 001.jpg
[00:34] <ninjas_> I just renamed my youngest son to c_14
[00:37] <c_14> I'm sure he'll be very happy.
[00:40] <ninjas_> lol
[00:56] <ninjas_> so can I use bash and sh in the same file?
[00:56] <c_14> Yes, but that'll get real ugly real fast, why?
[00:56] <ninjas_> lol
[00:56] <ninjas_> well
[00:56] <ninjas_> I'm not getting my concat comoplete
[00:56] <ninjas_> and its so freaking simple
[00:56] <c_14> error output?
[00:57] <ninjas_> http://pastie.org/9297163
[00:58] <ninjas_> I don't know if its counting too high or what
[00:59] <c_14> you'll probably have to replace the printf with a for loop
[01:00] <ninjas_> can I do that with an unknown number of files?
[01:00] <c_14> <(for f in /home/video/images/$1/*.mpg; do echo "file \'$f\'"; done)
[01:00] <c_14> something along those lines should work
[01:00] <ninjas_> omg
[01:00] <ninjas_> thank you
[01:00] <ninjas_> here I go
[01:02] <c_14> those '\'es might not belong there
[01:02] <ninjas_> same herror
[01:03] <c_14> <(for file in /home/$var/$var*.png; do echo "file '$file'"; done)
[01:03] <c_14> ^that seems to produce decent output for me
[01:03] <ninjas_> thats all I'm going for brother, just decent output lol
[01:04] <ninjas_> and some sleep
[01:06] <ninjas_> http://pastie.org/9297214
[01:06] <ninjas_> telling me an output has to be specified, I just copy pasted what you gave me out of blind faith, I can add the output
[01:07] <c_14> you're missing the second half of the command
[01:07] <c_14> >-i 001.mp3 -c:v copy -shortest -strict -2 -movflags faststart output.mp4;
[01:07] <c_14> ^that part
[01:09] <ninjas_> yeah, I thought you had some super way of tleling it all of that
[01:09] <ninjas_> FYI
[01:09] <ninjas_> I "taught" myself FFMPEG since wendesday and don't do C programming at all
[01:09] <ninjas_> PHP and MySQL
[01:09] <ninjas_> and I didn't know I could do PHP FFMPEG
[01:10] <ninjas_> ok, running it from the top
[01:11] <ninjas_> you got a paypal email you want me to drop some loot in?
[01:11] <c_14> It's fine. I do this mainly for the kicks.
[01:11] <ninjas_> I'd be honored
[01:11] <ninjas_> because I may have committed a neighborhood wide genocide with you to calmly and respectfully talk me through this
[01:12] <ninjas_> its running, and its taking an awful long time, that is a good thing
[01:14] <ninjas_> omg
[01:14] <ninjas_> it claimed my audio wasn't there
[01:15] <c_14> File does not exist?
[01:15] <ninjas_> No such file or directory
[01:15] <c_14> can you paste the output?
[01:16] <ninjas_> http://snag.gy/IfroO.jpg
[01:16] <ninjas_> here is the directory
[01:16] <c_14> for most of it you're using the absolute path but for the audio you're using the relative path, maybe that's the issue?
[01:17] <ninjas_> http://pastie.org/9297272
[01:17] <ninjas_> ah
[01:17] <ninjas_> I swear man
[01:17] <ninjas_> thisi s 101 stuff
[01:17] <ninjas_> what about output?
[01:18] <ninjas_> should I use absolute as well?
[01:18] <c_14> depends where you want it, if you want it to always go in that directory, then yes
[01:18] <c_14> If you always want it in the current dir, don't.
[01:19] <ninjas_> ok
[01:19] <ninjas_> here we go
[01:22] <ninjas_> no way
[01:22] <ninjas_> it just f-ing worked
[01:23] <ninjas_> please let me send you something
[01:25] <Timmy_L|NB> Hai. I'm having some issues with an .mkv-file. It supposedly has three subtitle streams (at least according to mediafile), but -sst 0-2 does't seem to display them. Any pointers?
[01:25] <Timmy_L|NB> I can share the mediainfo output if you want.
[01:25] <ninjas_> http://pastie.org/pastes/new
[01:25] <c_14> ninjas_: It's fine. If you really want to though, pm me and we can talk about it there.
[01:25] <ninjas_> they like you to use
[01:25] <ninjas_> I don't know how to PM
[01:25] <ninjas_> I haven't used irc since the 1990s
[01:26] <Timmy_L|NB> http://pastie.org/9297313
[01:26] <c_14> Normally it's just /query nick
[01:26] <c_14> Timmy_L|NB: what are you trying to do?
[01:27] <c_14> according to ffmpeg-all -sst doesn't seem to be a valid option.
[01:27] <Timmy_L|NB> I'm trying to view the movie with the English subtitles (text-stream number three), so I thought ffplay -sst 3 (or maybe 2) would do it, but nope.
[01:27] <Timmy_L|NB> Oh? :o
[01:27] <Hello71> IRC hasn't changed since 1993 since rfc 1459 came out
[01:28] <Timmy_L|NB> man ffplay says -sst is where I pick subtitle stream number
[01:28] <c_14> ah, -sst is an ffplay option, now I see
[01:28] <c_14> I'm pretty sure you can only display one subtitle stream at a time.
[01:29] <Timmy_L|NB> Uhm, yeah? I only want one at a time. :3
[01:30] <Timmy_L|NB> Wouldn't specifying -sst 2 disable all other than stream number 2?
[01:31] <c_14> Have you tried pressing t while ffplay is playing to switch streams?
[01:32] <Timmy_L|NB> Doesn't seem to work.
[01:32] <c_14> I can't seem to get ffplay to display subtitles for me right now either...
[01:35] <Timmy_L|NB> Alright, VLC shows the subtitles and also fixed my 5.1 -> 2.1 audio issues. So at least I know the file is okay. ^^
[01:35] <Timmy_L|NB> I'll try to fix it with ffplay after I've seen the movie, but I'll stay here if you guys come up with anything. :3
[01:42] <c_14> hmm, ffplay really doesn't want to show subtitles...
[01:51] <llogan> Hello71: nerd.
[02:10] <fjen_> gerion: try libav
[03:34] <Freakshow> anyone around that has experienced an issue with -list_devices returning as an unrecognized option?
[03:35] <Freakshow> in such that
[03:35] <Freakshow> ffmpeg -list_devices true -f dshow -i dummy
[03:35] <Freakshow> ffmpeg version 2.2.3 Copyright (c) 2000-2014 the FFmpeg developers
[03:35] <Freakshow> built on Jun 5 2014 17:14:36 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
[03:35] <Freakshow> Unrecognized option 'list_devices'.
[03:35] <Freakshow> Error splitting the argument list: Option not found
[03:35] <Freakshow> or am I doing something completely unexpected?
[03:36] <Ross> Hello, is anyone active in this chat??
[03:38] <klaxa> not now at least
[03:40] <Ross> Should I use the mailing ilst instead??
[03:40] <Ross> list*
[03:41] <klaxa> how about you ask your question first
[03:41] <klaxa> also it's the middle of the night in most of europe
[03:42] <klaxa> not sure how the channel is geographically split though
[03:43] <Ross> I am using FFMPEG compiled from GIT version N-63716-g763e714 with --enable-gpl --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-libaacplus running this command line:
[03:43] <Ross> screen -d -m -S TV3 ffmpeg -i "udp://@224.2.12.1:1234?fifo_size=10000&overrun_nonfatal=1" -vcodec mpeg2video -pix_fmt yuv420p -me_method epzs -threads 16 -r 25 -g 45 -bf 2 -trellis 2 -cmp 2 -subcmp 2 -s 720x576 -b 1500k -bt 300k -acodec mp2 -ac 2 -ab 192k -ar 44100 -async 1 -y -f mpegts udp://224.2.89.3:1234
[03:43] <Ross> works perfectly and wonderfully for like 1 week solid
[03:43] <Ross> then a frame sort of got stuck in the video. you can see it here http://103.14.142.201/ecast/wtfmoment.mpg
[03:44] <Ross> at about 1 minute 35 you can see that dog with "dont go away" frame stays throughout the transcode
[03:45] <Ross> i was wondering if anyone had experiance with something like this??
[03:46] <klaxa> well there is nothing obvious in the command line
[03:47] <Ross> I should of compiled with enable pthreads too
[03:49] <klaxa> wow that looks bad
[03:49] <klaxa> like every frame is referencing that key-frame
[03:49] <Ross> lol yeah
[03:49] <klaxa> no idea, sorry
[03:50] <klaxa> maybe make the sample smaller and ask the mailing list
[03:50] <Ross> thats ok, thanks for looking. I will ask the mailing list
[03:59] <llogan> someone tell Freakshow next time that dshow is for Windows, AFAIK
[04:21] <Mephistos> hi folks
[04:25] <Mephistos> quick question, does is anyone familiar with bayer_rggb16le. I believe they've only just added this as part of support for MLV (Magic Lantern's version of raw).
[04:54] <Mephistos> I get errors trying to translate the files from bayer_rggb16le to yuv444 or something similar?
[04:56] <Freakshow> not sure if anyone saw my question about dshow
[04:56] <Freakshow> I know it's late in EU but I had disconnected for a bit
[04:59] <llogan> Freakshow: "<llogan> someone tell Freakshow next time that dshow is for Windows, AFAIK"
[05:00] <Mephistos> no problems
[05:00] <Mephistos> give me a sec
[05:10] <Mephistos> http://pastebin.com/zzPrYGfu
[05:10] <Mephistos> This is one that failed
[05:11] <Mephistos> And one that worked, but gets converted to rgb24
[05:12] <Mephistos> http://pastebin.com/xrcFjL3k
[05:56] <llogan> Mephistos: might want to try the ffmpeg-user mailing list for that one
[05:57] <Mephistos> sounds like a plan, it's very new.
[06:12] <_genuser_> hey guys, it appears that when I encode a 20minute video, it's completed in X amount of time. However, when I encode a 40minute video, the first 20minutes take X+Y amount of time.
[06:12] <_genuser_> using ffmpeg -i source.mpg -vcodec h264 -acodec aac -strict -2 output.mp4
[06:12] <_genuser_> has anyone else noticed this?
[09:23] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ?
[09:31] <yusif> hi how can i put c99convert in the path
[09:33] <yusif> any one here
[11:19] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ?
[12:14] <t4nk909> hi, can i LIVE stream four video + 4 audio to a tablet device that is in the same LAN without delay - with all in sync
[12:47] <liuqiang> should I call avpicture_free on code like this: " avpicture_alloc((AVPicture *)dst_frame, PIX_FMT_YUVA420P, iDstWidth, iDstHeight);"
[13:10] <tfirg> should i be able to do x11grab with uhd resolution w/o buffer underruns?
[13:11] <kamillo1888> is there any way to transcode video with Intel® Xeon Phi Coprocessor 5110P (8GB, 1.053 GHz, 60 core)?
[13:11] <tfirg> ive been using ffmpeg on full hd for a while with out problems but with this ultra hd screen i get all kinds of issues. like buffer underuns and blue flickers
[13:21] <vl4kn9> Could someone please explain how to compute 2D DCT? Do I compute DCT for each row and then for each column of the already computed DCT?
[16:19] <vl4kn0> Hi, I'm getting segfault at ff_dct32_float_avx that is automatically selected by the av_dct_init(5, DCT_II). When running the same code with DCT_III it works fine. Is that a known bug or am I missing something?
[16:40] <Freakshow> llogan: so I can't access dshow filters at all on linux?
[16:41] <Freakshow> btw, thanks for having someone mention that to me while I was disconnected, I appreciate that
[16:43] <Mavrik> that would be hard considering the fact directoshow doesn't exist on linux.
[17:05] <Freakshow> ok, that makes sense& I honestly haven't even needed to try before
[17:06] <Freakshow> I have a blackmagic card I'm trying to pull bits from and that's what I find consistently referenced for accessing it
[17:06] <Freakshow> grrr! thanks Mavrik
[17:09] <Freakshow> looks like I'm bound for bmdtools then
[17:09] <Mista-D> anyway to boost volume with a limiter?
[17:13] <needmorespeed> Are there any buffer or queue utilities for holding decoded frames?
[18:05] <vl4kn0> Does ffmpeg have boxing filter? Often also called smoothing, averaging, mean..., one that takes kernel of for example 7x7 and averages the value into currently computed pixel.
[18:20] <saste> vl4kn0, there is a boxblur filter
[20:05] <hendry> an MP4 i created doesn't playback on an iphone or ipad. What am I missing? http://mp4.dabase.com/
[20:09] <llogan> hendry: you're missing your actual, unscripted ffmpeg command and the complete ffmpeg console output
[20:10] <llogan> and which devices are you targeting in particular?
[20:11] <llogan> you can encode just a segment with -t instead of the whole thing if the segment also has the same issues
[20:12] <hendry> llogan: doesn't take too long to run from the start
[20:14] <hendry> llogan: http://ix.io/d0I
[20:18] <llogan> hendry: which iphone and which ipad versions?
[20:21] <hendry> llogan: very latest 7.x
[20:22] <hendry> wonder what -ar is http://stackoverflow.com/a/10748735/4534 something audio related and not applicable here?
[20:24] <llogan> 7 is the iOS version. I meant the hardware
[20:24] <sfan5> hendry: -ar is the audio bitrate
[20:24] <hendry> llogan: iphone 5s and ipad air
[20:24] <sfan5> s/bitrate/sample rate/
[20:34] <llogan> hendry: try "ffmpeg -i input -c:v libx264 -vf scale=1920:-2 -movflags +faststart output.mp4" also your build may not have fix for ticket #3307
[20:42] <llogan> hendry: oh, it can handle level 4.1 and my scale was wrong: ffmpeg -i input -vf scale=-2:1080 -level 4.2 output
[20:42] <llogan> damn it, i meant 4.2
[20:43] <llogan> you may have to add -refs 3. your console output without -refs may confirm this (or i could not be a lazy ass and look at git log)
[20:44] <llogan> to summarize, don't use the crappy old ipod preset
[20:56] <hendry> llogan: pity the preset isn't updated?
[20:57] <llogan> hendry: no, and I'm going to remove them. see https://trac.ffmpeg.org/wiki/Encode/H.264#iOS instead
[21:12] <hendry> llogan: python -m http.server <--- i think was my problem
[21:15] <hendry> what does: -vf scale=-2:1080 mean? esp the -2?
[21:17] <llogan> hendry: it's explained here under the w and h options: http://ffmpeg.org/ffmpeg-filters.html#scale
[21:20] <Phlarp> So whats the best practice for concatenating three videos together? can I do this in a single command or will I need intermediate files?
[21:22] <Phlarp> Also I have an audio track that I need to lay over all three videos. The audio is longer than the videos combined so it will need to trim / fade out at the end. I would welcome any advice on these tasks
[21:23] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med…
[21:24] <c_14> and for the audio, just add it as a regular input file and use -shortest as an output option
[21:24] <mark_t> The build of ffmpeg I'm using on this server doesn't have any aac encoders except for the built-in. I'm not very concerned about size or quality, but is it stable and does it produce compatible output?
[21:24] <llogan> it should be fine. just give it enough bitrat so it sounds good enough
[21:24] <llogan> damned bit rats
[21:26] <mark_t> Thanks, llogan.
[21:29] <hendry> is there a "cheap" way of appending a jpeg frame to an MP4?
[21:29] <llogan> before or after or during?
[21:30] <hendry> err after?
[21:30] <llogan> i mean do you want the image to show, after, or during the main content?
[21:30] <hendry> llogan: after, like an append
[21:35] <hendry> would folks here recommend a tune parameter for a time lapse?
[21:37] <mark_t> Is there an option to tell ffmpeg to print an error instead of asking questions? E.g. File 'test.mp4' already exists. Overwrite ? [y/N]
[21:37] <mark_t> Or, failing that, to tell it to always use the default answer?
[21:38] <Phlarp> Wow, you guys are _good_
[21:38] <Phlarp> This whole thing works just like that! Thank you! Is there a built in option to fade the audio out at the end with -shortest?
[21:41] <mark_t> hendry, what do you mean by "tune parameter"? Options for your video encoder? (And if so, which encoder are you using?)
[21:41] <hendry> mark_t: yes, h264, from https://trac.ffmpeg.org/wiki/Encode/H.264#
[21:43] <llogan> mark_t: use -y or -n global options
[21:44] <mark_t> Thanks, llogan.
[21:44] <llogan> hendry: you might be able to figure out something with concat filter for your append
[21:44] <mark_t> hendry, I use CRF 20 (ffmpeg -crf 20) for my timelapse. Use lower numbers for higher quality (larger files), and higher numbers for the reverse.
[21:45] <hendry> iirc "-crf 20" is slightly better quality mark_t ?
[21:46] <llogan> Phlarp: see afade filter
[21:46] <mark_t> hendry, "-crf 20" is what I suggest starting with.
[21:47] <mark_t> But it also depends on exactly what you're encoding them for. If you want to watch them on low-bandwidth connections, CRF may cause buffering in some parts of the video (high complexity/detailed scenes), but sail smoothly through others.
[21:48] <mark_t> Other encoding options will never buffer, but the high detail/complexity scenes will just get lower quality.
[21:50] <hendry> bit annoying how many "Multiple ff_thread_finish_setup() calls" messages I get back from ffmpeg
[21:52] <llogan> webpee
[21:53] <JEEB> <mark_t> But it also depends on exactly what you're encoding them for. If you want to watch them on low-bandwidth connections, CRF may cause buffering in some parts of the video (high complexity/detailed scenes), but sail smoothly through others. <- whatever rate control you are using, if you are encoding for limited bandwidth, YOU USE VBV
[21:53] <JEEB> (maxrate and bufsize)
[21:53] <JEEB> if you don't use it, then whatever rate control mode you use, it's incorrect
[21:54] <JEEB> general rate control + VBV (maxrate and bufsize) is the only correct way to handle limited bandwidth scenarios
[21:54] <Phlarp> llogan: With afade I have to pass in a start time and duration, but since I'm going to be using -shortest to trim the audio at a variable location is there anyway to say "however long the final output ends up being, fade the last 5 seconds of audio"?
[21:54] <JEEB> and yes, you need BOTH maxrate and bufsize
[21:54] <mark_t> Thanks for the info, JEEB.
[21:55] <JEEB> maxrate is pretty self-explaining and bufsize is the amount the client will buffer, which is usually set somewhere either has seconds (which would be maxrate * seconds) or size (which you then can set as-is)
[21:55] <JEEB> if those two are set, then after buffering once you can be sure that your player will never have to buffer again
[21:56] <llogan> hendry: i don't know why those messages show up, but using -threads 1 as an input option will remvome them, or you can change your loglevel.
[21:56] <hendry> llogan: i want to use max threads in my 8cpu machines
[21:57] <llogan> i was referring to the decoding
[21:58] <llogan> Phlarp: looks like you have to explicitly define the times. you can script it though
[21:58] <llogan> https://trac.ffmpeg.org/wiki/FFprobeTips
[21:59] <llogan> mark_t: http://trac.ffmpeg.org/wiki/EncodingForStreamingSites has some VBV examples
[22:27] <maksimkaaa> hello, I generated a 10 second MP4 video using ffmpeg, it plays only 2 seconds then jumps to end, but when i take this same generated MP4 video and convert it to ogg video it plays the entire 10 seconds! any hints about what could be the reason?
[00:00] --- Wed Jun 18 2014
1
0
[00:16] <J_Darnley> I need to apologize if I have ever said someone's API is crap.
[00:17] <J_Darnley> Only now do I see how much there is to think about (and I am basically copying one that already exists).
[00:17] <ubitux> hopefully you're not copying zlib or openssl
[00:18] <J_Darnley> Oh no.
[00:18] <J_Darnley> (much worse)
[00:18] <ubitux> that's hard to believe
[00:18] <J_Darnley> A microsoft-style, c++ API
[00:19] <J_Darnley> I can't decide on names, namespaces, data structures
[00:22] <J_Darnley> I'm not even sure how the original gathers it various AVFilter-like objects together.
[00:23] <wm4> what API is it?
[00:23] <J_Darnley> Nullsoft's Advanced Visualisation Studio
[00:23] <wm4> ow
[00:24] <wm4> I don't know that API but it sounds like it's bad :)
[00:25] <J_Darnley> It is Winamp's classic visualisation plugin
[00:28] <J_Darnley> I guess that might not mean much to you either.
[00:30] <nevcairiel> if you deal in multimedia, you better know what winamp is
[00:52] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:02bffc560f5e: huffyuv: fate: add a BGR24 test
[01:03] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:f6577bd9cf3e: huffyuvdec: use unsafe bitstream reader
[01:32] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:20c7baf832f4: avcodec/huffyuvdec: Fix fill value for truncated bitstreams
[01:43] <cone-859> ffmpeg.git 03Carl Eugen Hoyos 07master:08235708d84a: Do not suggest to use gas-preprocessor on aix where it doesn't work.
[01:43] <cone-859> ffmpeg.git 03Ronald S. Bultje 07master:edf930472bbd: swr: reindent.
[01:43] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:fdb6f66defac: Merge remote-tracking branch 'cehoyos/master'
[03:07] <BtbN> Timothy_Gu, in the header itself is a license header which says "do not redistribute this"
[03:28] <michaelni> Daemon404, do you want to review the fateserver patches ?
[04:24] <Compn> wm4 / nevcairiel : there is mplayer winamp filter wrapper :D
[04:24] <Compn> af wadspa ?
[04:24] <Compn> perhapse
[04:36] <nevcairiel> iirc thats for dsp plugins, not visualization
[05:09] <Timothy_Gu> XV, another half-baked Chinese crappy proprietary container
[05:13] <cone-429> ffmpeg.git 03Michael Niedermayer 07master:b152152df3b7: Add nointra AVDiscard level
[09:24] <j-b> 'morning
[09:31] <plepere> hi
[09:40] <plepere> what's the difference between HAVE_SSE and HAVE_SSE_EXTERNAL ?
[09:55] <plepere> I'm going to start AVX2 coding. Do I have to do anything specific or is INIT_XMM avx2 enough ?
[09:55] <plepere> ubitux,
[09:56] <ubitux> i think you need %if HAVE_AVX2_EXTERNAL as well
[09:56] <ubitux> to support old assemblers
[09:57] <ubitux> (within the asm)
[09:57] <ubitux> also, you probably want INIT_YMM avx2
[09:58] <plepere> but I'm using ints
[09:58] <plepere> or is YMM not only for floats
[09:59] <plepere> YMM = 256 registers ?
[10:00] <ubitux> 256bits regs yes
[10:00] <plepere> yes
[10:01] <ubitux> ymm is only for float in avx iirc
[10:01] <ubitux> that's the point of avx2, to be usable with ints
[10:01] <plepere> yes
[10:02] <thardin> ccache rules
[10:02] <plepere> so I do INIY_YMM avx2 and I can use all avx2 instructions.
[10:02] <plepere> ubitux, I just saw in x86inc that in INIT_YMM, %define mova movdqa. shouldn't it be vmovdqa to do a 256b move ?
[10:04] <plepere> or should I use movnta ?
[10:05] <plepere> I'm confusing myself. I'll keep things simple.
[10:14] <ubitux> plepere: i don't remember the details, you might need to explicit the 'v' for the load/store instructions
[10:14] <ubitux> at least when you want 256, otherwise it might only load 128
[10:15] <ubitux> but check anyway
[12:14] <cone-506> ffmpeg.git 03Vittorio Giovara 07master:5a69ded12aaf: vf_showinfo: print display transformation values
[12:14] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:593dc5aed907: Merge commit '5a69ded12aaf2bd3baed5d1d644ef23cfecef87f'
[12:33] <BBB> ubitux: x86inc prefixes v to all instructions for avx
[12:33] <BBB> plepere: ^^
[12:33] <BBB> plepere: so dont prefic the v yourself, it defeats the use of macros
[12:34] <BBB> i.e. you cant share code between pre-avx and post-avx anymore
[12:34] <BBB> thats also why m%d macros for xmm1/mm1 exist, to share code macros between mmx/xmm
[12:35] <kierank> plepere: I think you should also add 3-operand to your existing functions
[12:36] <kierank> it would explain the macro stuff better
[12:36] <kierank> if not I'll try with your mc functions this evening
[12:42] <ubitux> BBB: even for loads?
[12:48] <BBB> I believe so?
[12:50] <BBB> yeah, check e.g. the avx2 intrapred functions in vp9intrapred.asm
[12:50] <BBB> they use mova and need 32bytes of data
[12:52] <BBB> and disass shows vmovdqa
[12:54] <cone-506> ffmpeg.git 03Vittorio Giovara 07master:2dc265619a2f: lavf: group dump functions together
[12:54] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:83d100baea7a: Merge commit '2dc265619a2fc9c6f9aff7ac2bcdbcb90e9610cb'
[12:54] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:7b7afaf849ef: avformat/dump: put copyright back from the file from which dump.c was split out
[13:03] <cone-506> ffmpeg.git 03Vittorio Giovara 07master:711aa2a82727: lavf: dump stream side data when probing
[13:03] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:f0fbd9105bc1: Merge commit '711aa2a82727907f778fb8aa9a93aff2120170f2'
[13:13] <cone-506> ffmpeg.git 03Vittorio Giovara 07master:766f2d965ff1: gitignore: add dylib and swp entries
[13:13] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:60058c1880f2: Merge commit '766f2d965ff1948749c37f26a714adb8aa89c4a9'
[13:16] <kierank> plepere: can you explain what this does http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/x86/hevc_mc.asm;h…
[13:16] <kierank> the punpcklwd
[13:18] <cone-506> ffmpeg.git 03Luca Barbato 07master:83aa4fc3feec: configure: Directly support Android as target OS
[13:18] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:8a4a84a70a2e: Merge commit '83aa4fc3feec7389ac781fece1e994f2dfd7ebdb'
[13:19] <plepere> kierank, it's to prepare for pshufd m2, m2, 0
[13:20] <plepere> could be replaced by SPLATW
[13:20] <plepere> that + shuf
[13:20] <kierank> so is the value in the int guaranteed to be only 16 bits
[13:20] <kierank> ?
[13:21] <plepere> yes
[13:21] <kierank> ah ok
[13:22] <kierank> you may have seen I got a bit confused and submitted a patch with splatd
[13:24] <plepere> wx could even fit in 8 bit IIRC
[13:26] <kierank> as far as I can tell those are just the block size
[13:26] <plepere> kierank, is there somewhere to see the list of 3-operand functions ? I tried to used a 3 operand paddusb and it was a no-go
[13:26] <kierank> plepere: intel manual
[13:27] <kierank> plepere: you actually already use 3 operand
[13:27] <plepere> ok. I thought it was a x264 thing. my bad
[13:27] <kierank> but you don't declare avx2 functions
[13:27] <kierank> avx*
[13:27] <kierank> plepere: http://www.intel.com/content/www/us/en/processors/architectures-software-de…
[13:28] <plepere> I like to use https://software.intel.com/sites/landingpage/IntrinsicsGuide/
[13:28] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:9140d37af2fa: avcodec/h264: use the correct level for droping non intra frames
[13:28] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:9025072e6c25: avcodec/h264_slice: support skipping loop filtering for non key frames
[13:29] <plepere> BBB : yes, found out that instructions like movd won't work. :(
[13:29] <BBB> what do you want movd to do?
[13:30] <plepere> have a value in a register, do an add, a shift, move to 256b register and splat it.
[13:31] <BBB> dont you use xmm instructions for lower-half operations?
[13:31] <plepere> so I do it on xm%d and do the plat on m%d .
[13:31] <plepere> ?
[13:32] <plepere> *splat
[13:34] <plepere> what I want is : I have idct16_dc_add_sse2, and I use it in a wrapper for 32 block. I want to have the idct32_dc_add_avx.
[13:37] <plepere> avx2
[13:38] <BBB> ...
[13:38] <BBB> you want to use it in a wrapper?
[13:38] <BBB> but htey do fundamentally different things
[13:39] <plepere> I have idct32_dc_add_sse2 which calls idct16_dc_add_sse2. I want a stand alone avx2 idct32
[13:39] <BBB> so movd xm%d, r%d
[13:39] <BBB> and then splat as usual, yes
[13:40] <BBB> also try to spend time on the full idct
[13:40] <BBB> dc-only isnt very interesting
[13:41] <plepere> yeah, but now I must leave the transform and see how much speed-up I can gain from AVX2 functions. I'm using the idct_dc as a simple first function
[13:41] <plepere> BTW, SPLATW doesn't work in AVX2
[13:42] <plepere> going to do splatW on xm then unpack128
[13:45] <plepere> meh, even splatw on xm doesn't work.
[13:49] <plepere> BBB ; vmovdqa m3, [%2+%3 ] is unauthorised ?
[13:49] <plepere> the compiler doesn't like the '+'
[13:51] <BBB> I dont know what %2 and %3 is
[13:52] <plepere> r2 and r3
[13:54] <BBB> first of all, use mova
[13:54] <BBB> not vmovdqa
[13:54] <BBB> second, what INIT_ statement did you use?
[13:54] <BBB> i.e. what is m3?
[13:54] <BBB> or show full code?
[13:55] <BBB> Im purely guessing as to what this code will actually expand to
[13:55] <plepere> wait, I think precising q at the end of the register name fixes it
[13:56] <plepere> no, I'm just an idiot. I forgot I changed the arguments of the macro.
[13:57] <plepere> BBB: with INIT_YMM avx2, mova will load 265b ?
[14:06] <BBB> yes
[14:06] <BBB> we just discussed that
[14:07] <BBB> also simple questions like that are best answered by trying it (write the code, compile it, disassemble it in db)
[14:10] <BBB> ok work, bbl
[14:15] <plepere> bbl
[14:28] <Youka> Is adding "/MD" to --extra-cflags enough to build ffmpeg for msvc with dynamic linking to default libraries?
[14:50] <J_Darnley> Youka: not many people here use msvc so you might need to stay here for a while if you want an answer.
[14:51] <Youka> ok, thanks for the answer
[14:58] <J_Darnley> What do you mean by "default libraries"? The c standard library or ffmpeg's libav* libraries?
[14:59] <nevcairiel> Youka: yes that should be enough
[14:59] <Youka> i mean the runtime
[15:00] <J_Darnley> well, there you go.
[15:00] <plepere> EXTERNAL_AVX2(mm_flags) doesn't seem to work. am I missing something ?
[15:00] <nevcairiel> maybe nooone build that macro yet
[15:00] <nevcairiel> -o
[15:00] <BtbN> I'd recommend building ffmpeg with gcc, even if you want to use it with msvc. The resulting binaries are faster and more feature complete.
[15:00] <Youka> Ok, with "/MD" it not works. The resulting (partly german) error message: "c1 : fatal error C1083: Datei (Quelle) kann nicht geöffnet werden: "E:/MinGW/msys/1.0/MD": No such file or directory
[15:00] <Youka> ffconf.pauYOIwJ.c"
[15:00] <plepere> there's no warnings though...
[15:01] <nevcairiel> try -MD
[15:01] <nevcairiel> slashes can confuse some things
[15:01] <Youka> with "-MD" i have the problem with the following make: "make: Nothing to be done for 'all'"
[15:02] <Youka> So both no options :/
[15:02] <BtbN> make sure you start your mingw/cygwin command prompt from within a msvc shell, and it will generate .lib files, even when building with gcc.
[15:03] <J_Darnley> What is your full configure line then?
[15:04] <Youka> configure --prefix=build/msvc/ --disable-programs --disable-manpages --disable-podpages --disable-txtpages --disable-avdevice --disable-postproc --disable-avfilter --disable-network --disable-encoders --disable-muxers --disable-devices --disable-filters --enable-avisynth --toolchain=msvc --disable-debug --extra-cflags="-MD"
[15:05] <Youka> Without these extra cflags it's successfully compiling, but i need to use it in a project built with /MD
[15:05] <nevcairiel> run make distclean before re-running configure to ensure no stale files exist, that option should work just fine
[15:06] <plepere> welp, the i7-4930K has no AVX2 support. :/
[15:06] <J_Darnley> ... and you were wondring why the branch was not used?
[15:06] <nevcairiel> plepere: of course not, its an ivy bridge core
[15:06] <nevcairiel> only haswell has AVX2
[15:07] <plepere> I thought 4K were haswell
[15:07] <plepere> FML
[15:07] <nevcairiel> Ivy Bridge-E also uses 4k numbers, apparently
[15:07] <Youka> ok, cleaned and reconfigure with -MD, let's see...
[15:15] <Youka> Status report: make runs now, so distclean really helped.
[15:41] <ubitux> so what's the plan for ffserver?
[15:59] <merbanan> ubitux: wanna nuke it ?
[15:59] <ubitux> the opposite
[16:00] <ubitux> libav is dropping it, but since it's still used in production and has its users...
[16:01] <nevcairiel> are you volunteering to improve it then? :d
[16:01] <ubitux> not really
[16:01] <ubitux> the problem is that the drop seems to be motivated by some api breakage to come, and that will probably be a pain to maintain for us
[16:01] <wm4> <ubitux> libav is dropping it, but since it's still used in production and has its users... <- drop it and maintain an old copy of it against $$$
[16:02] <ubitux> well... you can come with the same reasoning for every tool
[16:02] <ubitux> ffserver could become an alternative to wowza & crtmpserver
[16:02] <ubitux> but well
[16:03] <ubitux> yeah we need at least a maintainer
[16:03] <Daemon404> who the fuck uses ffserver in production
[16:03] <Daemon404> they should be fired
[16:04] <Daemon404> [15:02] <@ubitux> ffserver could become an alternative to wowza & crtmpserver <-- LOL
[16:04] <Daemon404> do you have N developers and M man-years of labour to spare
[16:04] <nevcairiel> that depends on N and M
[16:04] <ubitux> depends on features you want to implement
[16:05] <Daemon404> ubitux, to be anywhere even close to competeti e with wowza
[16:05] <Daemon404> Good Luck (TM)
[16:05] <ubitux> we already kind of have some network and video layers
[16:05] <Daemon404> i think you are underestimating the amount of work it needs
[16:05] <ubitux> yeah well, at least be an alternative for a few common use cases
[16:05] <Daemon404> "kinda having" some layers isnt exactly close.
[16:06] <Daemon404> well also nobody has maintained it in years
[16:06] <Daemon404> most seem to run screaming
[16:18] <Youka> After building ffmpeg on msvc with cflag "-MD" for dynamic runtime linkage, i got my binaries, but using these for my msvc project results in link warnings like following one:
[16:18] <Youka> libavutil.a(timecode.o) : warning LNK4049: Local defined symbol "_avpriv_snprintf" was imported.
[16:18] <Youka> Now a library is created beside my application.
[16:20] <nevcairiel> its just a warning
[16:21] <Daemon404> Youka, thats known, and normal
[16:21] <Youka> ok, then i have to life with these exports.
[16:22] <kierank> Daemon404: you assume that wowza works
[16:23] <Daemon404> kierank, ive tried it personally
[16:23] <kierank> i'm sorry
[16:23] <Daemon404> it's the best working solution for multiplexing a stream to N otehr tpyes of protocls
[16:24] <kierank> it doesn't work at scale
[16:24] <kierank> it's too bursy
[16:24] <kierank> bursty
[16:24] <Daemon404> oic
[16:24] <Daemon404> we never actually ended up using it
[16:29] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:7eae8cd87054: avcodec/h264: fix frame skip code
[17:35] <ubitux> Timothy_Gu: are these underlined links all over fate wanted?
[17:36] <Timothy_Gu> Yeah, the links are wanted
[17:36] <ubitux> i'm talking about the underline
[17:36] <ubitux> everything is now underlined it's almost unreadable
[17:36] <Timothy_Gu> well links == underline
[17:36] <ubitux> yeah but no
[17:37] <Timothy_Gu> i also know it masked the green or red status
[17:37] <Timothy_Gu> which is bad
[17:37] <ubitux> btw, there is something extremely annoying
[17:37] <ubitux> look at the darwing failure
[17:37] <ubitux> "(TB)"
[17:37] <ubitux> try to unroll it
[17:37] <ubitux> it create a freaking horizontal scroll
[17:37] <ubitux> +s
[17:38] <ubitux> it's annoying as hell
[17:38] <ubitux> would it be possible to overflow-y inside the report block
[17:38] <ubitux> instead of the whole page?
[17:39] <Timothy_Gu> i was trying to fix that with the fate.css patch
[17:39] <Timothy_Gu> but obviously it doesnt work so well...
[17:39] <ubitux> anyway, please remove the underline, it really sucks :p
[17:41] <Timothy_Gu> the fate.css patch only changed the diff box, not the log box. I'll change that when i have time
[17:45] <Timothy_Gu> I can probably fix the underline just like the link in thr table headers
[18:12] <cone-506> ffmpeg.git 03Carl Eugen Hoyos 07master:189ffeafd423: Force aix strip to work on 32 and 64 bit by default.
[20:07] <kierank> llogan: wtf rlynettet1(a)aol.com
[20:07] <kierank> didn't know ffmpeg-devel was general tech support
[20:09] <llogan> kierank: i also wtf'd, then set modrator flag, then s/he emailed -devel-owner and i explained the situation.
[20:10] <kierank> do you know this person
[20:10] <llogan> no
[20:11] <llogan> "I joined the ffmpeg site because another young lady asked for the same help & she was in contact with ffmpeg the site answered her question and posted everything. So l in turn followed."
[20:11] <kierank> wtf
[20:12] <llogan> must be a crazy person.
[20:12] <Daemon404> http://theoatmeal.com/comics/email_address
[20:12] <Daemon404> this is relevant
[20:12] <llogan> i forwarded her to libav-tools.
[20:12] Action: llogan is lying
[20:15] <llogan> Daemon404: nice, but needs to add: * enters their own email address as subject
[20:16] <Daemon404> lol
[20:16] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:600cbf367231: avcodec/bitstream: zero vlc tables on allocation
[20:17] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:f7f96cf4bcc3: avcodec/bitstream: fill invalid vlc tables entries as last pass instead of first
[20:17] <cone-506> ffmpeg.git 03Michael Niedermayer 07master:e6f9fc4adc16: avcodec/bitstream: try to make vlc init code inherently thread safe
[20:26] <cone-506> ffmpeg.git 03Simon Thelen 07master:389d45351e06: version string: add copyright line to version string
[20:46] <kurosu> "another young lady" <- I sense an oxymoron
[23:03] <Mista-D> why is it that when using "-vf null", then "-c copy" is forbidden? It should be just a pass through no?
[23:04] <wm4> video filters only work on uncompressed video frames
[23:34] <cone-497> ffmpeg.git 03Michael Niedermayer 07master:293d5d7a8e12: swscale/input: fix harmless integer overflow
[00:00] --- Tue Jun 17 2014
1
0
[00:57] <aarobc> Hello. Is there a way to use the -map option to copy all streams, but then append another stream on top of it (other than manually specifying all the streams)
[00:57] <c_14> What do you mean with append another stream?
[00:58] <c_14> -map 0 will map all the streams from input to output
[00:59] <aarobc> c_14: well, I want to preserve all the existing streams, but then also add another stream that is transcoded from the AC3 audio, but in vorbis. so you have two audio streams. one for high quality, and one for backup in case you're playing on something that doesn't support ac3
[01:01] <aarobc> The reason I'm wanting to preserve the existing stream structure is in some cases you have a file that has multiple languages or commentary. I'd want that to be preserved.
[01:04] <aarobc> okay, just tested it and it works
[01:05] <aarobc> if you just use -map 0 -map 0:1 -c copy -c:a:1 libvorbis
[01:51] <morenoh153> never used ffmpeg before. Can it be used to make a screen recording? if so how?
[01:52] <c_14> https://trac.ffmpeg.org/wiki/Capture/Desktop
[01:52] <aarobc> morenoh153: https://trac.ffmpeg.org/wiki/Capture/Desktop
[01:52] <aarobc> lol
[01:55] <morenoh153> ok. How can I get a browser extension to execute that?
[01:57] <c_14> what?
[01:57] <c_14> explain
[01:57] <morenoh153> I want to make a chrome extension that will record the window. I have to bundle ffmpeg i think.
[01:58] <c_14> I have no clue how chrome extensions work.
[02:20] <aarobc> Is there still no multi-threded vp9 encoding?
[02:21] <c_14> not as of when I last checked
[02:21] <aarobc> for shame :(
[02:21] <aarobc> I thought that it was all finalized. How do they exepect anyone to use it if there isn't a usable encoder for it?
[02:27] <llogan> aarobc: they don't care about general users encoding. only decoding via watching in youtube
[02:28] <llogan> by "they" i mean Google
[02:28] <aarobc> How the hell does google do that anyway? they just have servers of fpgas doing hardware encoding?
[02:29] <llogan> $$$$$$$$$ & basements full of chinese orphan children manually encoding
[02:29] <aarobc> lol
[02:30] <llogan> but yes, lots of hardware. why should they care about how long it would take a normal person to encode when 99.8% of users will only be watching and not encoding with this format?
[02:33] <llogan> but that is just what I think is the situation. reality may differ
[02:34] <frikimaster> Hi! Any recommends to encode 1080i to 720p? I've tested yadif with poor results
[02:36] <llogan> you can add -t 10 to encode a short segment. and input sample could be helpful too.
[02:38] <frikimaster> i can't repeat the process now, sorry. Is a motorsport video with fast horitzontal moves
[02:38] <c_14> frikimaster: ffmpeg has at least 3-5 deinterlacing filters. Maybe try one of the other ones?
[02:39] <frikimaster> the best one in my case and opinion is yadif. In fact it isn't soooo bad
[02:42] <frikimaster> mm i need to upload some samples :-P
[02:42] <frikimaster> i'll go back (after testing -t option, thanks)
[02:42] <llogan> it has several modes. 1 (or send_field) may look better for your content than the default 0 (send_frame)
[02:46] <frikimaster> command used: http://pastebin.com/JEpyLA31
[02:48] <llogan> that's not ffmpeg
[02:49] <llogan> you're going to have to get help from avconv people
[02:49] <llogan> or use ffmpeg
[02:49] <llogan> (the real one)
[02:50] <frikimaster> i've readed just a bit about this topic. OK, i'll test mpeg also
[02:51] <llogan> the version in the Ubuntu repository is an unsupported counterfeit
[02:51] <llogan> see http://ffmpeg.org/download.html#LinuxBuilds or http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu
[02:52] <llogan> if you want ffmpeg (and therefore suggestions and help from this channel)
[03:06] <ElDimentio> does ffmpeg support Xunlei XV files?
[03:06] <ElDimentio> I'm trying to find a converter to FLV but all I find keeps bringing up virus alerts :/
[03:07] <llogan> i never heard of Xunlei XV. got a sample?
[03:08] <ElDimentio> well, I do, but it's nearly 200MB :/
[03:09] <ElDimentio> doesn't seem convenient to share
[03:09] <ElDimentio> it's a Chinese format
[03:09] <llogan> is it too big for you to share? 200 MB is fine for me to download
[03:09] <ElDimentio> a lot of Chinese TV shows are in this format, a program called Xunlei can download TV shows in that format but only Xunlei can play them
[03:10] <ElDimentio> where would I go to share that file then? any suggestions?
[03:10] <ElDimentio> my internet speeds are okay
[03:10] <llogan> dropbox, google drive, datafilehost
[03:10] <ElDimentio> ah yes
[03:11] <ElDimentio> dropbox it is
[03:12] <ElDimentio> being able to convert this would be great ... I find Xunlei too annoying (bloated) to be restricted to its media player, but their resources are immense
[03:13] <ElDimentio> https://www.dropbox.com/s/0rlzjm9b1apg0j5/1.xv
[03:13] <ElDimentio> does that work?
[03:15] <llogan> yes, thanks.
[03:15] <ElDimentio> no, thank you!
[03:28] <llogan> ElDimentio: does not appear to do so, but it might not be hard to implement. go to the bug tracker and make a feature request
[03:29] <llogan> provide the input file. also download a recent ffmpeg and provide "ffmpeg -i 1.xv" and the complete output from that command.
[03:32] <llogan> it is H.264 video.
[03:33] <ElDimentio> okay I'm registering right now
[03:34] <llogan> probably in FLV container that they crapped on
[03:34] <ElDimentio> so I just do "New Ticket" ?
[03:35] <llogan> yes. summary "Support decoding of Xunlei XV file" or something like that
[03:36] <llogan> type: enhancement, priority: wish, version: git-master (if you're using recent ffmpeg), keywords: xv
[03:37] <llogan> give description of file type, how it was created or what created it, and info on what can play it properly
[03:39] <llogan> i have to go now, but i'll be here tomorrow. the bug tracker will email you if there is ticket activity, so you don't have to add yourself to CC (if i can recall correctly)
[03:39] <ElDimentio> okay, thanks!
[03:45] <ElDimentio> ok done: https://trac.ffmpeg.org/ticket/3720#ticket
[04:44] <DeadSix27> is it possible to add x ammount of MS of blank audio to an existing audio file?
[08:37] <b_jonas> I can encode audio so that this mobile phone can play it, and I can encode video too, but now I have to figure out how to combine the two :)
[08:41] <sacarasc> b_jonas: Give both commands and I'll use my copy/pasting skills to make it work.
[09:00] <b_jonas> sacarasc: thanks, I'll try to combine them first
[09:01] <b_jonas> I might come back if I fail, but now I have other work to do
[09:38] <ekem> hi guys, im attempting to run ffserver, and everything appears to start fine, but the file it makes in temp is owned by root
[09:39] <ekem> while the server is being run as my user
[09:39] <ekem> if i try and run ffserver as a root, it seg faults, anyway, when i try and use ffmpeg to input a video i get a permission denied to the .ffs
[09:39] <ekem> er .ffm, does anyone have any suggestions?
[16:16] <termos> is there a way for ffmpeg to reduce the mpegts header overhead? I have several mpegts files when converting to hls and I would like the overhead to be at a minimum
[16:19] <termos> I'd pretty much like something like apples mediastreamsegmenter, only built into ffmpeg
[17:29] <GEEGEEGEE> Is there a way to send the same output to multiple RTMP servers? My command at the moment is:
[17:29] <GEEGEEGEE> ffmpeg -i \"http://source:9609/stream.ts\" -vcodec copy -acodec libfaac -ar 44100 -f flv rtmp://server1.com/live/test\
[17:31] <c_14> https://ffmpeg.org/ffmpeg-formats.html#tee
[17:31] <c_14> look at the tee muxer
[17:32] <GEEGEEGEE> ok thanks c_14
[19:44] <ninjas_> oh wow
[19:44] <ninjas_> I am so glad I found this
[19:44] <ninjas_> I'm trying to run a BASH command for FFMPEG out of PHP ... then I just learned that I can call an FFMPEG command right from PHP without using shell_exec()
[19:45] <ninjas_> is there a php user on?
[19:51] <ninjas_> I'm trying to pass a variable into an ffmpeg string
[20:07] <GEEGEEGEE> Im trying to output to multiple RTMP servers with ffmpeg -i "http://source:9609/stream.ts" -vcodec copy -acodec libfaac -ar 44100 -f tee "[f=flv]rtmp://paris2/live/test|[f=flv]rtmp://paris3/live/test"
[20:07] <GEEGEEGEE> but its not working :/
[20:08] <GEEGEEGEE> http://pastie.org/9295782rf
[20:08] <GEEGEEGEE> http://pastie.org/9295782
[20:08] <c_14> I wanted the output.
[20:10] <GEEGEEGEE> oh sorry
[20:11] <GEEGEEGEE> http://pastie.org/9295787 here
[20:13] <c_14> Just a question, does this command work when outputting to a file/when outputting to a single stream? Because it looks a lot like ffmpeg is failing at the h264 stream.
[20:14] <GEEGEEGEE> yeah ive been using it for weeks when just outputting to a single server
[20:15] <c_14> can you try adding -map 0 as an output option?
[20:17] <GEEGEEGEE> I did this: ffmpeg -i "http://source:9609/stream.ts" -vcodec copy -acodec libfaac -ar 44100 -f flv rtmp://paris2/live/test -vcodec copy -acodec libfaac -ar 44100 -f flv rtmp://paris3/live/test and it seems to be working
[20:18] <c_14> ye, but you'll be encoding the audio twice
[20:31] <llogan> c_14: i applied your patch
[20:48] <ninjas_> does anyone use PHP and FFMPEG?
[20:49] <Fjorgynn> nope noone
[20:49] <ninjas_> riiiiiiiiiiight
[20:49] <ninjas_> I'm trying to determine if I'm using shell_exec() right or not
[20:50] <ninjas_> or if I need ot quit using that and just use a direct command?
[20:57] <llogan> ninjas_: there are a few simple examples here: https://trac.ffmpeg.org/wiki/Using%20FFmpeg%20from%20PHP%20scripts
[20:57] <Mista_D> when using -vf null, -c copy
[20:58] <ninjas_> yeah
[20:58] <ninjas_> I think what I'm running into is
[20:58] <Mista_D> why is it that when using "-vf null", then "-c copy" is forbidden? Isn't "
[20:58] <ninjas_> my putenv isn't there
[20:58] <Mista_D> Isn't null filter just passes bitstrem through?
[20:59] <Mista_D> sorry for typing errs.
[21:02] <ninjas_> bool putenv ( string $setting ) I see this
[21:04] <ninjas_> I don't understand how to use it
[21:04] <ninjas_> I'm freaking lost - can anyone help me use putenv to tell my PHP file where ffmpeg is installed?
[21:05] <llogan> try #php
[21:06] <ninjas_> gotta be invited
[21:06] <ninjas_> are you in it to invite? :)
[21:18] <llogan> no
[21:20] <llogan> apparently there is "no official [PHP] IRC channel"
[21:22] <ninjas_> llogan any idea on what I can do?
[21:25] <llogan> http://php.net/support.php
[21:25] <llogan> other than that, no, sorry
[21:30] <billyshambrook> Does anyone know of any good c api examples for using the metadata injection feature in silencedetect?
[21:37] <nijm> Does anyone know what the silencedetect audio filter output is supposed to look like? I'm trying it on a video and it doesn't seem to be producing any output.
[21:37] <nijm> billyshambrook, I'm guessing you might?
[21:40] <ninjas_> putenv("PATH=" . $_ENV["PATH"] . ":/opt/asotechs/ffmpeg-2.2.3");
[21:40] <ninjas_> this was solve to my problem
[21:44] <nijm> Here's the command I used when I tried to use silencedetect and the output I got: http://pastebin.com/pXNm9N7V
[21:44] <nijm> Am I doing something wrong?
[21:45] <nijm> I'm trying to get 200ms silences less than -50dB.
[21:54] <llogan> nijm: if it does not detect the provided parameters it won't output anything, so you may have to adjust your d or n
[21:55] <nijm> llogan, is there anything in the output that acknowledges the filter is being used?
[21:56] <nijm> llogan, if I try to use a fictional 'jibberish' filter is still happily processes the file: ffmpeg -i Derek_20130205_2311.ts -vn -sn -filter:a jibberish=n=-10dB:d=0.01 -f null -
[21:57] <llogan> you can try adding "-loglevel debug" as a global option
[22:08] <ninjas_> I'm getting "could find no file with path" ffmpeg
[22:08] <ninjas_> and I'm using absolute path
[22:09] <ninjas_> llogan you know anything for that?
[22:20] <ninjas_> my bad guys, I'm new to IRC
[22:21] <ninjas_> http://pastie.org/9296136
[22:23] <c_14> You forgot to paste your ffmpeg command
[22:23] <llogan> why no line breaks?
[22:24] <ninjas_> llogan it is error log
[22:25] <ninjas_> http://pastie.org/9296174
[22:25] <ninjas_> sorry, here is my command
[22:27] <c_14> If I'm reading the ffmpeg output correctly, bash isn't expanding the glob correctly.
[22:27] <ninjas_> the *
[22:28] <c_14> precisely
[22:29] <ninjas_> ok, so i should rename my image files ot 001.jpg 002.jpg 003.jpg
[22:29] <ninjas_> so they are very uniform?
[22:30] <ninjas_> or is there anotehr way to do?
[22:30] <c_14> What are you trying to do?
[22:31] <ninjas_> make a video for each image 12 seconds long in the folder
[22:31] <ninjas_> random number of files in the folder
[22:35] <c_14> You might want to test this in 3 steps.
[22:35] <ninjas_> Sure
[22:35] <c_14> The ffmpeg commandline on one picture until that works.
[22:35] <ninjas_> ok
[22:35] <c_14> The for loop with an echo or an ls until that works.
[22:35] <c_14> And then combine them.
[22:35] <ninjas_> ok, here I go
[22:35] <ninjas_> thanks for the inspiration
[22:54] <yusif> how i can use ffmpeg with vs 2012
[22:55] <JEEB> it's in the fine documentation, although vs2013 is recommended
[22:55] <JEEB> since it has actual support for C99 features
[22:55] <JEEB> so it doesn't need the converter
[22:56] <yusif> i have vs 2012 i downlod the converter but i dont know how to install it so help me
[22:56] <JEEB> read the documentation or watch dae's vimeo clip on it :P
[22:57] <JEEB> also I'm not sure how long the converter will be supported, so I recommend you update to 2013 as soon as it's possible
[22:59] <yusif> JEEB can you recommend tutorial or book explain how to use ffmpeg in devloper way not cmd line
[23:01] <JEEB> if you know what the components more or less do (esp. lavf and lavc), then the doxygen is your friend
[23:01] <JEEB> also there are examples in the source tree
[23:01] <JEEB> consider most other examples out of date
[23:55] <ninjas_> c_14
[23:55] <ninjas_> you did it
[23:55] <ninjas_> thank you
[23:55] <c_14> I didn't do _that_ much.
[23:55] <c_14> Thanks anyway.
[23:56] <ninjas_> nah man
[23:56] <ninjas_> having someone to talk to about this stuff
[23:56] <ninjas_> that has a brain
[23:56] <ninjas_> and isn't a stack overflow jerk
[23:56] <ninjas_> (not that all of them are)
[23:56] <ninjas_> makes a big difference, thank you very much
[23:56] <ninjas_> I'm compiling the second command now
[23:56] <ninjas_> cross your fingers!
[23:56] <ninjas_> its a concat, should be easy
[23:56] <ninjas_> just taking a while
[23:57] <ninjas_> *explode*
[23:57] <ninjas_> famous last words
[23:59] <ninjas_> http://pastie.org/9296687
[00:00] --- Tue Jun 17 2014
1
0
[00:01] <J_Darnley> Better instruction order?
[00:02] <kierank> inlining perhaps
[00:07] <kurosu> dead code elimination
[00:07] <kurosu> I suspect the macros have written to achieve their goal, not to be efficient
[00:08] <kurosu> *the asm macros were
[00:08] <kurosu> eg the filter coeffs loaded into specific regs without much care about the fact other regs were unused
[00:09] <kurosu> but I can understand pierre-edouard
[00:10] <kierank> I need to find an avx2 machine
[00:10] <kierank> I think I have one somewhere
[00:11] <cone-576> ffmpeg.git 03Ronald S. Bultje 07master:ada8f9c046c8: swr: remove obsolete function prototypes.
[00:21] <smarter> kierank: don't expect me to comment on asm stuff :p
[00:21] <kierank> smarter: i mean comment on the patch to play the broken sample
[00:21] <smarter> ah I didn't see that
[00:21] <kierank> there were also comments on #libav-devel
[00:25] <smarter> kierank: looks OK to me
[00:25] <smarter> didn't see any comment on #libav-devel
[00:26] <kierank> from elenril there was a comment
[00:26] <smarter> It might be a good idea to do something similar for other checks that do not prevent decoding, but maybe it's better to annoy people into fixing their encoder :p
[00:26] <kierank> kurosu: any idea how to do packusdw in sse2
[00:27] <kierank> because then i think at least one function is convertable
[00:28] <kierank> also found some double splats
[00:29] <kierank> hmm he does that a lot for some reason
[00:29] <jamrial> https://stackoverflow.com/questions/11024652/simulating-packusdw-functional… first answer
[00:30] <kierank> jamrial: yeah but *stackoverflow* :)
[00:30] <jamrial> it's intrinsics, but changing that with actual asm is trivial
[00:30] <jamrial> hey, if it works :P
[00:30] <jamrial> didn't try it, though
[00:31] <J_Darnley> kierank: I think I wrote one in yadif
[00:33] <kierank> i wonder if it's worth moving it to x86util
[00:33] <jamrial> if it's used in more than one file then yes
[00:35] <J_Darnley> Yes: libavfilter/x86/yadif-16.asm
[00:35] <kierank> yeah i found it
[00:35] <J_Darnley> line 64
[00:51] <kierank> hmm my patch is broke
[00:59] <kierank> I think he's actually trying to do splatw
[01:07] <BBB> Im wondering whats uglier, non-static functions or static dead code
[01:07] <BBB> ...
[01:13] <jamrial> if you mean swr, prototypes are going to be needed for yasm, so might as well add them now as a solution for the msvc failures
[01:14] <BBB> I suppose
[01:15] <BBB> I would just give them ff_ prefixes, not swri_
[01:15] <BBB> since theyre local within the lib and we want them to be stripped
[01:28] <jamrial> michaelni suggested swri. i can resend the patch using ff_ if that's better
[01:35] <michaelni> swri should be removable
[01:35] <michaelni> or am i missing something ?
[01:35] <michaelni> i mean removeable like ff_
[01:35] <michaelni> swr_* should be exportet
[01:36] <michaelni> swresample_dsp* should be renamed
[01:36] <michaelni> it matches neither suggestion
[01:39] <cone-576> ffmpeg.git 03Michael Niedermayer 07master:86a9370e2b91: avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it
[01:40] <cone-576> ffmpeg.git 03James Almer 07master:7f4dfbd0809d: swr: add prototypes for resample dsp functions
[01:41] <michaelni> currently: "global: swr_*; ff_*; swresample_*;"
[01:41] <michaelni> so it wont export swri*
[01:45] <BBB> oh swri is internal
[01:45] <BBB> then its ok
[01:46] <BBB> kinda weird but ok
[01:47] <BBB> who calls vzeroupper in inline avx?
[01:49] <BBB> anyway I dont feel like removing inline today, maybe some other time
[01:49] <BBB> Ill go watch football for another few minutes :)
[04:28] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:103f9c261a68: avcodec/indeo5: Fix infinite loop in skip_hdr_extension()
[04:28] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:73d820ee1eb0: avcodec/xbmdec: remove dependancy on zero padding on input packet
[04:33] <michaelni> jamrial, as you are maintainer of a few things, you should have op in in #ffmpeg-devel too
[04:40] <jamrial> michaelni: thanks!
[05:39] <michaelni> kurosu, you have contributed quite a bit of code and asm to ffmpeg, do you want to maintain some of that? if so dont hesitate to send a patch that adds you to maintainers
[05:57] Action: Timothy_Gu thinks that michaelni is becoming a MAINTAINERS maniac.
[06:09] <llogan> now that's an odd message in -devel
[06:09] <llogan> setting mod bit
[10:48] <wm4> did that yadif relicensing some time ago actually work out, i.e. did the contributors get paid?
[10:49] <kurosu> michaelni, I appreciate the consideration, but I'm not sure about it
[10:49] <kurosu> I may be kind of unreliable as a maintainer, in particular for reviews in due time
[10:53] <kurosu> though you probably need the relief, comic or not
[11:11] <J_Darnley> wm4: yes
[11:11] <J_Darnley> well I did
[11:15] <wm4> nice
[13:14] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:56a795e34fb7: aandcttab: fix spelling
[13:14] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:2a1158ff3b29: fate: yadif: add >8 bit tests
[13:18] <michaelni> kurosu, maybe you want to add yourself as co maintainer for something ? like x86 asm ? then being sometimes/often unavailable shouldnt be a problem
[13:31] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:91076128185e: x86util: add and use RSHIFT/LSHIFT macros
[13:32] <kurosu> michaelni, ok, why not
[13:32] <kurosu> I'll see about that later today
[13:33] <michaelni> ok, thanks
[14:17] <cone-859> ffmpeg.git 03Ronald S. Bultje 07master:083cd3d1f771: swr: compile mmx2 s16p functions only on x86-32.
[14:48] <BBB> kurosu: did you fuzz the unsafe bitstream reader stuff for huffyuv under valgrind or so?
[14:48] <BBB> kurosu: or asan
[14:48] <BBB> kurosu: Im fine with that if were convinced we have adequate manual bitstream overread protection
[14:48] <BBB> kurosu: but we want to make sure were not missing anything obvious I guess
[14:50] <kurosu> BBB: nope, though the loops check every 2 samples if end of bitstream was reached
[14:50] <kurosu> but indeed I haven't checked and I don't feel completely safe there
[14:50] <BBB> whats the max size of a sample?
[14:50] <kurosu> 32bits or 3*11 depending on how you look at it
[14:50] <kurosu> so within padding
[14:51] <BBB> hmk
[14:51] <BBB> should be ok
[14:51] <kurosu> yeah, should
[14:51] <kurosu> but fuzzing might be better
[14:51] <BBB> maybe ask michealnis contacts at google to fuzz it for you at their servers
[14:52] <kurosu> I just felt encouraged when mentionning this to michaelni and he hadn't noticed and it made sense
[14:53] <kurosu> michaelni, googlz plz fuzz ? ^
[14:57] <BBB> hehe :)
[15:00] <kurosu> I bet with the experiment on huffyuv, some lossless codecs could be made 10-30% faster
[15:00] <kurosu> huffyuv is 30-40% faster with all patches applied, compared to some weeks ago
[15:01] <kurosu> *huffyuvdec
[15:07] <kurosu> well, when the data is hot - because obviously the hard drive is the limiting factor here
[15:34] <wm4> there's no xml parser in use in ffmpeg yet, right?
[15:35] <wm4> I see mov.c does some strstr() to "parse" xml, but apparently there's nothing else
[15:41] <Daemon404> please dont give them ideas, wm4
[15:45] <kurosu> why not? What's a few extra MBs when your debug binary is already 100+ ?
[15:46] <Daemon404> because it is a solved problem, NIHing it is retarded.
[15:47] <kurosu> Daemon404, I was ironical
[15:47] <Daemon404> oh
[15:47] <nevcairiel> I guess we're just lucky that no mainstream format needed XML parsing yet
[15:48] <Daemon404> ttxt!
[15:48] Action: Daemon404 runs
[15:48] <wm4> there are some xml subtitle formats
[15:48] <kurosu> configure and link are the most time consuming things when I work on ffmpeg
[15:48] <wm4> they're all handled with ad-hoc custom parsers
[15:48] <Daemon404> kurosu, configure is, by far, for me
[15:48] <nevcairiel> I said mainstream! :p
[15:48] <kurosu> but i deserve what I get for having a slow laptop under win64 without ssd
[15:48] <Daemon404> configure is like 2x the build time
[15:48] <wm4> smil is pretty mainstream in some parts
[15:48] <wm4> well, in korea (lol)
[15:49] <nevcairiel> Asians also use real media
[15:49] <nevcairiel> They don't count
[15:49] <Daemon404> not asians
[15:49] <kurosu> I was surprised there were no patch for hevc-in-rm
[15:49] <Daemon404> chinese, specifically.
[15:49] <Daemon404> kurosu, nah
[15:49] <kurosu> oh crap, this channel is logged
[15:49] <Daemon404> it will be real video 9
[15:49] <Daemon404> or whatever
[15:49] <kurosu> what have I unleashed on the world
[15:49] <Daemon404> which is conveniently similar to hevc
[15:49] <wm4> kurosu: what have you done
[15:50] <kurosu> rv10 probably died when real networks codec department was bought by intel
[15:50] <Daemon404> ah true
[15:50] <kurosu> well, rv50 in ffmpeg's slang
[16:21] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:9dc17919777a: huffyuvdec: swap code blocks
[16:21] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:e9c477059d78: avcodec/huffyuvdec: assume vlcs can be 32 instead of 31 bits max
[16:23] Action: Daemon404 wonders why huffyuv is getting so much love after a decade
[16:32] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:eb6f6f25dc9f: MAINTAINERS: Add Co maintainer for huffyuv*, rv4*, vc1*
[17:25] <BtbN> How would i add an encoder to ffmpeg that needs an external header which cannot be distributed with ffmpeg because of its license?
[17:29] <Daemon404> BtbN, in configure, make it require "non-free"
[17:29] <Daemon404> to be enabled
[17:29] <BtbN> No, the final binary can be freely distributed. The header can't
[17:30] <Daemon404> huh?
[17:30] <BtbN> The NVENC sdk does not allow redistributing any of its headers
[17:30] <Daemon404> why would you add their header to the git repo anyway
[17:30] <Daemon404> i.e. why would it be an issue
[17:31] <BtbN> because it's not an installable dependency on any distribution.
[17:31] <Daemon404> thats not a good enough reason
[17:31] <BtbN> There is no way to search for it. And it's only a header, no library
[17:32] <Daemon404> its part of the official sdk, and i dont think it is unreasonable to think people who want to compile with said encoder need said sdk
[17:36] <BBB> BtbN: you would carefully document how people can get access to this sdk
[17:37] <BBB> BtbN: and if its really that hard, you probably want to make binaries available that have the sdk dependency in it
[17:37] <Daemon404> the other option is to do a cleanroom rewrite or whatever
[17:37] <BBB> (as in, were built against that header)
[17:37] <Daemon404> like mingw-w64 has for dshow
[17:37] <Daemon404> and stuff
[17:37] <BBB> even then the header would live in its own project, not in ffmpeg
[17:37] <BBB> we dont ship libfdkaac headers
[17:38] <BtbN> BBB, well, the SDK can be freely downloaded from nvidias site. But it's nothing that can be normaly installed via some package manager, just some zip file.
[17:38] <BtbN> Daemon404, hm, how would i rewrite a header without basicaly copying the original?
[17:39] <Daemon404> BtbN, i think its reasonable to expect the compiling user to dl the sdk
[17:39] <Daemon404> if they want that feature
[17:40] <BtbN> would be nice to have the feature included in normal builds though. It doesn't have any dependencies except for that header, so it would just work if the user of a distribution built ffmpeg has a recent enough driver
[17:41] <BtbN> I think i'll just mail nvidia if i can re-license that header into something gpl compatible
[17:41] <Daemon404> lol
[17:41] <Daemon404> good luck with that
[18:01] <cone-859> ffmpeg.git 03Christophe Gisquet 07master:29f427c2397b: huffyuvdec: remove somewhat deprecated code
[18:25] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:153b5fb2fdef: avformat/framecrcenc: print the checksum and size of extradata as well
[18:27] <ubitux> strange vqf-demux test
[18:27] <ubitux> fate-vqf-demux: CMD = md5 -i ... -f framecrc
[18:37] <jamrial> so it's doing an md5 with the framecrc output?
[18:39] <jamrial> it's silly, but technically it still works as intended. a change in the framecrc output will be reflected in the md5 value :P
[18:40] <ubitux> i was wondering because of latest commit
[18:40] <ubitux> (there was a diff in md5 for some reason)
[18:43] <jamrial> the framecrc output for that test is about 2600 lines, so i guess that's why it's using the md5 protocol
[18:43] <jamrial> it probably should instead be "fate-vqf-demux: CMD = crc -i ..." like most other demux tests
[18:45] <ubitux> ah, okay
[19:27] <Timothy_Gu> BtbN: i just read the NVENC license https://developer.nvidia.com/nvidia-video-codec-sdk-license-agreement
[19:28] <Timothy_Gu> It says: Subject to Licensees compliance with the terms of this Agreement, NVIDIA grants to Licensee a nonexclusive, non-transferable, worldwide, royalty-free, fully paid-up license and right to install, use, reproduce, display, perform, modify the Source Code of the Software
[19:28] <nevcairiel> doesnt really change the fact that an external header like this should not be part of ffmpeg git
[19:29] <nevcairiel> and i'm sure someone would find fault in that for not being GPL compatible in some way
[19:46] <ubitux> ok hq3x working
[19:47] <ubitux> vf_hqx.c 538L, hqx project 12201L
[19:51] <BBB> ubitux: very cute
[19:51] <BBB> what about speed?
[19:51] <BBB> (or is that not relevant?)
[19:52] <ubitux> it might be relevant, but this is going to be used for stuff like 320x240 or similar
[19:52] <ubitux> i don't really want to bother
[19:52] <ubitux> i'm adding threading right now
[19:52] <ubitux> that should do.
[19:53] <ubitux> and i still have a bug and a few other things to do
[19:53] <ubitux> also, it might be possible to make the code even simpler
[19:53] <ubitux> i'll probably the final version this week or next w/e
[19:53] <ubitux> +submit
[20:15] <kurosu> ubitux, would you or someone close to you use it, or was it for the fun ?
[20:16] <ubitux> more like a challenge, i don't think i'll use it
[20:16] <ubitux> :p
[20:17] <kurosu> so that was fun
[20:17] <ubitux> mmh
[20:17] <kurosu> my definition of fun stops at simd inloop filter
[20:17] <ubitux> it got me too much angry to actually call it fun but yeah, somehow
[20:43] <cone-859> ffmpeg.git 03Anshul Maheshwari 07master:9a11b33a2d9d: avcodec/dvbsubdec: restructure version check
[20:43] <cone-859> ffmpeg.git 03Anshul Maheshwari 07master:77ade55fe52e: avcodec/dvbsubdec: add AVClass to context
[20:43] <cone-859> ffmpeg.git 03Anshul Maheshwari 07master:fbb59a3bf4c8: avcodec/dvbsubdec: Split save_subtitle_set() out
[21:05] <wm4> ubitux: I know you probably still hate subtitles, but I think this supports some tags your code doesn't: https://aur.archlinux.org/packages/mplayer-microdvd/
[21:05] <ubitux> wat
[21:06] <ubitux> http://b.pkh.me/p0-MicroDVD-36279.patch
[21:06] <ubitux> :/
[21:07] <ubitux> it looks like feature we're supporting
[21:07] <ubitux> mmh
[21:07] <ubitux> + if (*p == '#') ++p; // Correction By Xaweryz; for subs like {c:$#007FFF}
[21:07] <ubitux> maybe this?
[21:07] <ubitux> (wth)
[21:08] <wm4> lol
[21:08] <ubitux> did you get a bad subtitles file or sth?
[21:08] <wm4> no, https://github.com/libass/libass/issues/110
[21:08] <wm4> someone wants to add microdvd parsing to libass
[21:09] <ubitux> lol
[21:09] <ubitux> well anyway the microdvd code already supports color & stuff
[21:09] <ubitux> i could add the '#' workaround easily if you get me a sample
[21:11] <ubitux> wm4: mpv uses microdvd decoder, right?
[21:12] <wm4> currently the internal one is still preferred
[21:12] <wm4> I'll push a commit to change this
[21:25] <ubitux> great, i found a bug in the original hqx code
[21:34] <wm4> ubitux: copy&paste error?
[21:34] <wm4> or whatever the process is which was used to create the source
[21:34] <wm4> because it's so highly unrolled
[21:34] <ubitux> unrelated to the unrolling
[21:35] <ubitux> it's the rgb->yuv table
[21:35] <wm4> aw
[21:35] <ubitux> it's missing the init for the latest element
[21:35] Action: wm4 still doesn't understand why there's yuv in this
[21:35] <ubitux> so with white in an input i had surprises, sometimes :)
[21:36] <ubitux> like: http://b.pkh.me/pixelart0.png-hq4x.png vs http://b.pkh.me/ff-pixelart0.png-hq4x.png
[21:36] <ubitux> (i let you find the glitch)
[21:36] <ubitux> wm4: because it has difference thresholds
[21:37] <ubitux> (to check if there is a difference with surrounding pixels)
[21:37] <wm4> so it basically uses luminance difference as metric?
[21:37] <ubitux> yes, and also chroma
[21:38] <ubitux> 48 diff in luma, 7 and 6 in u/v
[21:38] <wm4> and I can't spot the glitch
[21:38] <ubitux> look at the skeleton
[21:38] <wm4> oh.
[21:39] <ubitux> btw, i could add an option to make it use ffmpeg internal scaling and use yuv only (zomg faster) but it won't be bitexact with the reference
[21:40] <ubitux> it seems it's already reported... https://code.google.com/p/hqx/issues/detail?id=8 heh.
[21:41] <jamrial> sep 2012. status new. no replies
[21:41] <jamrial> well...
[21:41] <ubitux> :)
[21:51] <michaelni> kierank, ok if i apply the hevc/4k patch or should we keep waiting ?
[21:54] <kierank> Ask smarter
[22:01] <kurosu> kierank, is there a way to identify the encoder that produced that file?
[22:01] <kurosu> because otherwise it seems to let invalid bitstream be decoded
[22:01] <JEEB> what kind of derp did that bitstream seemingly have?
[22:01] <kurosu> and I don't know what that may cause if the origin is not that encoder
[22:02] <kurosu> incorrect vps info
[22:02] <JEEB> ah
[22:02] <JEEB> funny enough the Japanese 4K channel was able to actually do that shit right :P
[22:02] <JEEB> so much that I'm even damn surprised
[22:02] <kurosu> related to reference frames iirc, but I don't think it's related to rps
[22:03] <kurosu> so it is a Japanese DVB-S stream or something ?
[22:03] <JEEB> no
[22:03] <JEEB> kierank has his own sources
[22:03] <JEEB> I've only seen some Japanese ISDB-S samples
[22:04] <JEEB> and those decoded fine with lavvc
[22:04] <JEEB> *lavc
[22:04] <kurosu> ok
[22:04] <JEEB> you'd be surprised with the features they used :3 https://drive.google.com/folderview?id=0BzkZOi5-Ka62NG1aa2RkaUFldzg&usp=sha…
[22:04] <JEEB> (check the file that ends with A)
[22:05] <JEEB> 2160p60, HEVC, 10bit, BT.2020 \o/
[22:08] <ubitux> michaelni: could you upload http://b.pkh.me/pixelart0.png and http://b.pkh.me/pixelart1.png to fate-samples/filter?
[22:09] <kurosu> JEEB: is that a H.264 encoder with HEVC syntax ?
[22:10] <michaelni> kierank, i asked smarter long ago if he wants to maintain hevc and he didnt want to. But well, i surely would be very happy if that changed
[22:10] <JEEB> I haven't checked the bit stream too much, but I wouldn't be surprised if it was using 64x64 blocks
[22:11] <michaelni> smarter, is "0614 20:26 Kieran Kunhya (2.0K) [FFmpeg-devel] [PATCH] hevc: Fix 4K sample video" ok to apply ?
[22:15] <kierank> kurosu: came from a European football match
[22:18] <michaelni> ubitux, uploaded
[22:19] <ubitux> michaelni: thx
[22:21] <llogan> it has Golden Axe, but is missing Altered Beast.
[22:26] <kurosu> michaelni, about the patch I didn't send, I'm holding onto it because of the end psnr value
[22:27] <kurosu> I would have expected infinite psnr, but seeing 38.xdB, I'm wondering if that's due to colorspace conversions here or there
[22:28] <kurosu> going back some commits still yield this, so I'm inclined to think it's the reaseon
[22:28] <michaelni> does it compare yuv420 or bgr24 ?
[22:29] <kierank> kurosu: do you know of any good free hevc analysers
[22:30] <kurosu> michaelni: http://pastebin.com/LRVTc9eF
[22:31] <kurosu> kierank, no
[22:32] <kierank> I am curious now to see if JEEB's samples are just h264 encoders writing an hevc bitstream
[22:32] <kurosu> kierank, just a humorous remark
[22:32] <kierank> that happened in the early days of h264
[22:32] <kurosu> actually I don't have a way to check either
[22:32] <llogan> i don't understand why carl told a user to not use -an in first pass encoding w/ libx264.
[22:32] <kierank> there is one very expensive encoder that did that
[22:33] <kurosu> but people may not had time to actually create a real hevc encoder
[22:33] <kierank> excactly
[22:34] <llogan> why would encoding (or not encoding) the audio in the first pass matter? (i haven't read the ticket he mentined though since it was so long)
[22:35] <kurosu> I know some companies boasting H264 devices, where the motion was only halfpel - but that may have been like 5+ years ago
[22:35] <kierank> i've seen that too and with only 8x8 blocks
[22:35] <kurosu> nowadays, some do the same with hevc
[22:36] <kurosu> I even heard of a company showing their encoder, while it was actually an offline encoding reencoded to H.264 lossless or something
[22:37] <kurosu> at IBC or NAB or whatever such trade show
[22:37] <kierank> many do that
[22:43] <JEEB> anyways, I need to grab more samples of that Japanese channel, since they're going to be airing the world cup
[22:48] <kurosu> probably I shouldn't be asking but... is that not encrypted ?
[22:48] <JEEB> it uses basic Japanese DTV encryption
[22:49] <JEEB> (all of the DTV in Japan except for 1seg is encrypted)
[22:49] <JEEB> so no, it's not a paytv channel
[22:50] <kurosu> I really meant encrypted only
[22:50] <kurosu> I didn't know there were different encryptions with different levels of complexity
[22:50] <JEEB> well, encryption usually means that something is paytv
[22:50] <JEEB> at least in the west
[22:51] <JEEB> which is what this is not
[22:51] <JEEB> it's just that everything except for 320x180(a)15/1.001 mobile broadcast is encrypted
[22:51] <JEEB> (1seg)
[22:52] <JEEB> the non-paytv Japanese DTV encryption was fully software'ized a bit more than a year ago
[22:52] <JEEB> http://repo.or.cz/w/soft_arib_std_b25.git
[22:52] <wm4> wait, so even public broadcast is encrypted?
[22:52] <JEEB> yes
[22:53] <JEEB> _everything_ is encrypted, thanks to hollywood pressure
[22:53] <JEEB> except that crappy mobile broadcast that even mobiles are moving away from
[22:53] <kierank> wm4: there is drm but not CA I guess
[22:54] <wm4> terrible
[22:54] <JEEB> also funny enough the official devices and software are meant to keep that content under extra locks on users' devices
[22:54] <JEEB> even the mobile airing
[22:54] <JEEB> thankfully it didn't take too long for "research only" hardware to pop up
[22:55] <JEEB> and given that B-CAS cards are just normal smart cards it wasn't too hard to implement the descrambling
[22:55] <kurosu> Nice, all-zeros encryption keys
[22:55] <kierank> JEEB: what channel is that?
[22:55] <kierank> which you recorded
[22:55] <JEEB> kierank, channel 4k
[22:56] <kurosu> except for NHK
[22:56] <JEEB> ch502
[22:56] <JEEB> kurosu, I think those without valid keys are the paytv channels
[22:56] <kurosu> yea, still I guess someone was able to bruteforce at least NHK key
[22:57] <JEEB> nah, those just weren't hardened in any way
[22:57] <JEEB> paytv actually means you have to poke at the internals
[22:57] <JEEB> since the key is passed upon you via the card ID
[22:57] <JEEB> and then stored on the card
[22:58] <JEEB> kierank, this one http://www.nextv-f.jp/information/index.html
[22:58] <kierank> ok i think it's an ntt
[22:59] <JEEB> NHK is participating, yes
[22:59] <JEEB> it seems to be a group effort
[23:00] <JEEB> NTV, SkyPerfect TV, TBS, TV Tokyo, NHK and WOWOW are giving out content at least
[23:02] <JEEB> and yeah, starting member companies are listed on the index page and include a whole lot of corps
[23:02] <JEEB> including those I just mentioned
[23:02] <JEEB> and hardware makers such as Sharp, Sony, Toshiba, Panasonic
[23:03] <JEEB> Fujitsu, NEC, NTT
[23:03] <JEEB> and yes, I totally read your NTT as NHK :P
[23:03] <JEEB> getting late >_>
[23:10] <michaelni> kurosu, the psnr looks like colorspace convert related, yes, i think you can post the patch and ill apply it unless someone wants to add support for bgr based tests
[23:20] <smarter> kierank: why did you submit a different patch to ffmpeg and libav?
[23:20] <kierank> smarter: there some additional checks in ffmpeg iirc
[23:21] <kierank> the behaviour was different
[23:21] <smarter> ah, maybe ffmpeg already has a patch to tolerate the error on the sps
[23:22] <smarter> indeed
[23:23] <smarter> michaelni: the hevc patch is OK
[23:23] <michaelni> smarter, thanks
[23:28] <cone-859> ffmpeg.git 03Carl Eugen Hoyos 07master:b67bcd784dde: Fix compilation on ppc64 and ppc with pic if gas-preprocessor is installed.
[23:28] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:c1b15c16ef74: avformat/smoothstreamingenc: Use av_mallocz_array()
[23:28] <cone-859> ffmpeg.git 03Kieran 07master:bc21260e643c: hevc: Fix 4K sample video
[23:29] <smarter> kierank: I'll send an updated patch to libav with the missing change from ffmpeg
[23:29] <smarter> (which already checks sps->temporal_layer[i].num_reorder_pics > MAX_DPB_SIZE - 1)
[23:29] <kierank> thanks
[23:34] <kurosu> michaelni, confirmed it, when having an intermediate rgb file, input/output perfectly match
[23:38] <jamrial> kurosu: same happens with other ffhuff tests. also zlib
[23:39] <kurosu> jamrial, I hadn't noticed
[23:39] <kurosu> the yuv422 test case confused me
[23:39] <kurosu> the psnr is perfect, but I suspect this is because of point resampling
[23:40] <kurosu> had I look one file further, I wouldn't have needed to confirm it
[23:40] <kurosu> *looked
[23:41] <jamrial> i noticed the other day when i added a cinepak test to play with vsynth, and the result wasn't lossless. then started checking other ref files and quite a lot of formats were the same
[23:42] <jamrial> for that matter, cinepak enc maintainer should probably add a vsynth test
[23:42] <jamrial> though it's so absurdly slow i don't know who would use it as is
[23:44] <kurosu> what, not tented to add dsp assembly for a 25yo codec ? :)
[23:45] <kurosu> (probably more related to the r/d exploration than anything, but I was kidding)
[23:45] <jamrial> heh
[23:47] <jamrial> actually, i'm confusing things. cinepak is lossy. i'm not sure how i ended up looking at lossless refs and finding that about ffhuff by playing with that
[00:00] --- Mon Jun 16 2014
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[00:01] <waressea1cher2> now its less messy
[00:01] <waressea1cher2> at first I was shocked
[00:02] <llogan> yeah, and using testsrc as an input makes it a little more verbose
[00:02] <llogan> althought easy to test
[00:03] <waressea1cher2> c_14: that web page says "trac powered", maybe you haven't made any mistake ?
[00:04] <llogan> basically all you need are: pad (or some other method to make a "canvas"), setpts, and overlay. and additional other filters for audio depending on what you want to do
[00:05] <c_14> trac is just the wiki software. >Trac is an enhanced wiki and issue tracking system for software development projects.
[00:08] <waressea1cher2> its always better to use one ffmpeg command with long filter chain rather than few with one filter in each ? like "ffmpeg -i input -vf filter1,filter2 output" is better then: "ffmpeg -i input -vf filter1 output1" and "ffmpeg -i output1 -vf filter2 output2" ?
[00:09] <c_14> better is subjective, but you will spare yourself a lot of encoding and potential quality loss (depending on the codec used)
[00:43] <smo_> hi
[00:44] <smo_> i m trying the libx265 encoder and get lot of non-existing PPS (i try to open files with gst-launch and lide265)
[00:44] <smo_> can do something for this ?
[04:02] <causasui> can ffmpeg segment an audio file into files of an arbitrary length? or would I have to loop through the time stamps?
[04:02] <causasui> e.g. take this 7 hour mp3 and segment it into 10 minute chunks
[04:03] <c_14> look at the segment muxer
[04:09] <_genuser_> dang, ffmpeg encoding same file seems to take longer.
[04:09] <_genuser_> proably because normally I see it doing pass 1 really fast and second pass. this one seems to be reporting time much slower and not showing which pass it's on.
[08:39] <isomorphismes> I'm following the Ubuntu ffmpeg compilation instructions
[08:39] <isomorphismes> but at the end I got some errors on the ffmpeg make and make install.
[08:39] <isomorphismes> There's no ~/bin/ffmpeg, probably as a result.
[08:44] <isomorphismes> ok hold on
[08:44] <isomorphismes> dpaste.de/UfTj
[08:46] <isomorphismes> dpaste.de/0M7H is the longest i can get
[08:47] <sacarasc> I have never seen strip say "blah is not a normal file" before. :|
[09:03] <Fyr> guys! I run the following command:
[09:03] <Fyr> ffmpeg -i 1.wmv -vcodec h264 -pass 1 -passlogfile n1 -threads 12 -psnr -vstats -profile:v baseline -preset veryslow -an -y -crf 18 -f rawvideo /dev/null && ffmpeg -i 1.wmv -vcodec h264 -pass 2 -passlogfile n1 -threads 12 -psnr -vstats -profile:v baseline -preset veryslow -crf 18 -acodec libfdk_aac -aq 8 -y 1.mp4
[09:03] <causasui> gesundheit
[09:03] <Fyr> and get error: wrong bitrate, width, height or something else.
[09:04] <Fyr> I always get this error when I try to run second pass.
[09:05] <Fyr> I changed -crf on -b, tried -an, removed one-by-one presets, profiles.
[09:05] <Fyr> recompiled fmpeg.
[09:05] <Fyr> anyway, the same error.
[09:18] <isomorphismes> sacarasc: what do you think that means?
[09:18] <sacarasc> I don't know, you're going to have to wait for someone who might know.
[09:18] <isomorphismes> ok cool no problem
[09:18] <isomorphismes> TIA
[09:21] <Fyr> fine
[09:21] <Fyr> any ideas how to solve it?
[09:21] <causasui> pastebin.com sux :\
[09:39] <sacarasc> Fyr: Can we have your paste?
[09:42] <Fyr> how to make ffmpeg output to a file?
[09:42] <sacarasc> Which OS?
[09:42] <Fyr> I have a big part of output cut because of tmux.
[09:42] <Fyr> Scientific Linux 6.3
[09:43] <sacarasc> ffmpeg blah &> cheese.txt
[09:43] <Fyr> ok
[09:44] <a141982112> when I press t while playing video subtitle switch from actual stream to -1, what does -1 stream mean
[09:44] <a141982112> i am using ffplay to play
[09:52] <Fyr> sacarasc:
[09:52] <Fyr> http://pastebin.com/isTujsP3
[09:55] <sacarasc> Could be: [libx264 @ 0x3442f40] constant rate-factor is incompatible with 2pass.
[09:55] <sacarasc> Try using 1 pass or a bitrate.
[09:56] <Fyr> is crf constant-rate factor?
[09:56] <sacarasc> Yeah.
[09:57] <Fyr> how to choose bitrate?
[09:58] <Fyr> the file has vbr with peak 10Mbps and average 8Mbps.
[09:58] <sacarasc> Why do you want to do 2 pass?
[09:58] <Fyr> it decrease filesize.
[09:58] <Fyr> it means better compression.
[09:59] <sacarasc> Do you have a specific size you want?
[09:59] <Fyr> no
[09:59] <Fyr> I need higher quality and lower file size.
[10:00] <sacarasc> Then just use crf and single pass. Run a test of a high action scene and a low action scene, find the crf number you want and then use that for the whole file. (Or, just go for crf 20 straight off.)
[10:00] <Fyr> how to find crf number?
[10:01] <sacarasc> Higher is worse quality, but lower size, lower is higher quality but larger size.
[10:01] <Fyr> so, how to find the proper value?
[10:02] <sacarasc> ffmpeg -i blah.wmv -ss 10:00 -t 1:00 -an -c:v libx264 -crf X output.mp4
[10:03] <sacarasc> Do that with different values of X until you find the highest number you want.
[10:03] <Fyr> and how to use psnr for it?
[10:03] <Fyr> and analyze vstats?
[10:04] <sacarasc> Just add them in.
[10:23] <Fyr> PSNR Mean Y:47.956 U:51.159 V:51.581 Avg:48.823 Global:48.803 kb/s:5210.03
[10:24] <Mavrik> don't use PSNR to measure video quality ffs -_-
[10:26] <Fyr> how to measure video quality?
[10:26] <sacarasc> Eyes.
[10:27] <Mavrik> Eyes.
[10:27] <Fyr> I have a bad squint and dyslexia.
[10:28] <Mavrik> measuring quality of video produced by codec with psycho-visual optimizations is totally wrong
[14:24] <Fjorgynn> hey
[14:24] <sacarasc> Hello.
[14:24] <Fjorgynn> I did ffmpeg -i hey.wav to hey.flac
[14:24] <Fjorgynn> but it lowered the bitrate from 700 kbps to 250 kbps
[14:25] <Fjorgynn> why
[14:25] <sacarasc> Because FLAC is lossless *COMPRESSION*. It makes wave files smaller.
[14:25] <sacarasc> Without losing any of the quality.
[14:25] <Fjorgynn> ok
[14:29] <Fjorgynn> I am digitizing my music casettes and think it might be a good idea to save the lossless to, if mp3 isn't the right way for them
[14:29] <Fjorgynn> or if I want to increase volume and such
[18:50] <luc4> Hello! Anyone who can advice on a way to resample a video file to reduce the fps?
[18:50] <luc4> Keeping quality unaltered I mean.
[18:56] <c_14> I'm not sure you can change the fps without reencoding the video, and if you are using a lossy codec you will lose quality even if it isn't noticeable.
[18:57] <luc4> c_14: ah ok, I see that probably by adding -vcodec it actually re-encodes and sets the frame rate. Thanks.
[20:55] <waressearcher2> I found that page http://wjwoodrow.wordpress.com/2013/11/22/adjusting-video-contrast-brightne… it suggest to use "-vf mp=eq2=gamma:contrast:brightness:saturation:rg:gg:bg:weight" for changing "gamma,contrast,brightness,saturation" is it right way ? because it uses "mp" for mplayer
[20:58] <llogan> waressearcher2: that's one method, I suppose. there are many video filters available that can do similar stuff like the "curves" filter
[23:47] <needmorespeed1> I've got frames and playback. Does ffmpeg have any queueing feature like a decoded picture buffer (DPB)?
[00:00] --- Mon Jun 16 2014
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