[Ffmpeg-devel-irc] ffmpeg.log.20140922

[burek]

[00:00] <t4nk705> hi, I have a trueHD audio file in .mka container and i'd like to convert it to dts-hd ma or lpcm
[00:01] <t4nk705> how do i do this via command line using latest ffmpeg
[00:01] <t4nk705> not sure what arguments to use after ./ffmpeg
[00:36] <slimsag> Hi, I have a semi-legal question for you all. I know that you're not lawyers and can't give me any legal advice -- but I want to ask strictly of your *opinion*.
[00:36] <slimsag> I want to make Go bindings to (LGPL) FFMPEG. Go *really* encourages the idea of statically linking things. I don't want to statically link FFMPEG because the LGPL license would destroy any commercial application using my Go bindings.
[00:38] <slimsag> Dynamic linking *is possible* but makes deployment a pain. My idea is to make a Go library that *installs* FFMPEG for you silently, it would still be dynamically linking and would therefor fall under LGPL *technically*.
[00:40] <slimsag> e.g. it would extract FFMPEG dynamic libs into a folder like $HOME/.goffmpeg/
[00:41] <slimsag> Now my question: do you think from a legal standpoint (your opinion) that this is okay to do? Obviously installers already do this -- but it would be done silently when the application was ran rather than obviously with a GUI etc
[00:55] <Lac3rat3d> is there some metadata that determines default audio stream?
[00:56] <benlieb> Lac3rat3d: i think the first audio stream is selected
[00:56] <benlieb> if not specified
[00:57] <Lac3rat3d> ok
[00:59] <Lac3rat3d> i have some mkv's that are 264 dca. i converted them to 0:0 x264 0:1 aac 5.1 0:2 dca
[00:59] <Lac3rat3d> but my plex server still recognizes the file as x264/dca
[01:00] <c_14> Lac3rat3d: there is a flag-default for the matroska format, but FFmpeg does not currently support setting it
[01:00] <c_14> You can use mkvtoolnix to set it.
[01:00] <Lac3rat3d> hmm well i also changed the container to mp4
[01:00] <Lac3rat3d> so plex is recognizing it as mp4/x264/dca
[01:02] <eago> :( I don0t think is possibl to do what I want
[01:05] <eago> I can see how can I use the webcam as input.... but not from the browser with ffmpeg installed in a remote server
[01:07] <vmBenLub1r> I'm encoding some gifs into webm using the concat demuxer and the command line output looks like this: http://i.imgur.com/0VB4yoW.png
[01:07] <vmBenLub1r> the output video looks fine, but I'd like to disable whatever is giving that warning
[01:58] <eago> is possible to boradcast from a machine that isn't a server to a machine that has ffserverinstalled and is a server, so that second machine streams to viewers?
[02:07] <c_14> yes
[02:19] <eago> and is it possible to have ffmpeg in a remote server and capture from a local webcam?
[02:23] <c_14> if the webcam is forwarded to the network, yes
[02:25] <eago> ok
[02:44] <eago> you can think anyway I could send live video from a browser to ffmpeg on a server to then stream using ffserver?
[02:44] <c_14> "send from a browser" <- how is the browser accessing the webcam?
[02:45] <eago> using javascript
[02:45] <eago> getUserMedia
[02:46] <c_14> If you can send the video out using http,udp,tcp or any of the supported ffmpeg input protocols, yes.
[02:46] <c_14> But don't ask me how to do that in javascript.
[02:46] <eago> ok
[02:47] <c_14> You just need ffmpeg to read/wait for input on it's internet-accessable ip address and then do whatever you want with it.
[02:49] <eago> so... what happens if connection isn't great, it just pause and then resumes or it falls?
[02:49] <eago> I mean, crashes
[02:49] <c_14> if the connection breaks off, the ffmpeg process will stop
[02:50] <eago> ok
[02:52] <eago> thank you
[02:59] <eago> need some rest, thank you very much
[02:59] <eago> cya
[06:11] <active8> if I run -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a]" on an mp4 and out to mp4 twice to get 1/4 speed, all else being equal I should expect more loss in the audio than if I just ran -filter_complex "[0:v]setpts=4*PTS[v];[0:a]atempo=0.5,atempo=0.5[a]" once, right? leading up to another question, so I'm just trying to avoid an assumption
[06:13] <active8> getting pretty good sound either way, btw; but this might not always work - using -c:v libx264 -preset slow -crf 0 -c:a libmp3lame -q:a 0
[06:14] <active8> ok  -c:a libmp3lame -q:a 0 (since the question concerns the audio)
[11:30] <voxadam> Is it possible to use ffmepg/ffserver to stream adaptive bitrate DASH or FLS streams while transcoding to lower bitrates as needed on-the-fly?
[11:34] <voxadam> Ideally, what I'm interested in doing is to stream an MKV (h264/AAC) file via DASH while remuxing and transcoding on-the-fly as needed.
[11:43] <JazzCZ> Hi, is there a build including yesterday's updates? I wanna use TrueHD without Dolby Atmos
[11:44] <JazzCZ> can't build it myself though, hard as shit
[11:48] <JazzCZ> sigh
[14:25] <bearish> anyone familiar with DashCast or MP4Box?
[15:13] <pa> was looking at mp4box just now, but for the first time.. :)
[16:09] <bearish> pa: MPEG-DASH looks cool, but damn if i can get it to work
[16:40] <active8> if I run -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a]" on an mp4 and out to mp4 twice to get 1/4 speed, all else being equal I should expect more loss in the audio than if I just ran -filter_complex "[0:v]setpts=4*PTS[v];[0:a]atempo=0.5,atempo=0.5[a]" once, right? leading up to another question, so I'm just trying to avoid an assumption
[16:41] <active8>  getting pretty good sound either way with -c:a libmp3lame -q:a 0, btw; but this might not always work
[16:41] <active8> can I do better?
[16:43] <active8> if not using atempo, atempo (twice in the same filter string) maybe save to lossless and then run again and encode? Or what about doubling the samples so the ones that get dropped don't degrade the signal?
[16:43] <pa> bearwhat are you trying to do? :)
[16:44] <active8> [0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a] is half speed slow-mo and the other is 1/4 speed
[16:45] <active8> this is so I can see fingers flying and hear what they play. other apps mo0ght be differen't where I;d need better video, but this is about sound quality
[16:45] <active8> pitch is prevserved - couple places sound crappy
[16:46] <active8> actually, the fingers blur a bit, too, and that could be improved -one thing at a time, i guess
[16:47] <relaxed> pastebin your command
[16:48] <active8> relaxed, pa: http://pastebin.com/fzG5JALB
[16:49] <active8> g'morning 8)
[16:51] <eago> hello... I'm trying to find a protocol to stream using ffmpeg... I will have ffmpeg in a remot server and I want to do a live stream from a  local computer. I am able to post blobs from my local computer to an apache server
[16:51] <active8> rtmp?
[16:52] <eago> ok, cool, I'll try that, thank you
[16:52] <relaxed> active8: you could use -c:a pcm_s16le for lossless but it will increase the size
[16:53] <active8> eago: how are you sending the blobs to the server?
[16:53] <eago> from javascript  with a XMLHttpRequest
[16:53] <active8> ok relaxed - duh, i forgot and just did that for screencast capture.
[16:53] <relaxed> active8: with -crf 0 it should be lossless as well, so I don't know why you see blurring
[16:55] <active8> yeah. real fast acoustic intro. what if I doubled the samples into an intermediate file first? Think I'd have to -- not sure... increase fps and decrease rate?
[16:55] <active8> or vice versa
[16:57] <active8> btw -crf 0 and -crf 18 seem to be the same - like gold speaker cables don't do much, either 8)
[17:10] <active8> on mp4 to mp4 -- like if i'm just ctting something out with -ss and -t, why does q:a 1 always say [libfdk_aac @ 0x3100680] VBR quality 118 out of range, should be 1-5[libfdk_aac @ 0x3100680] Note, the VBR setting is unsupported and only works with some parameter combinations
[17:12] <active8> like here: http://pastebin.com/ppZwYNzc
[17:24] <sacarasc> active8: You can probably use -c:a copy
[17:42] <active8> sacarasc, I would, but it's not allowed with filtercomplex
[17:43] <sacarasc> That command didn't show that.
[17:43] <active8> it works with -c:a libmp3lame, but it's odd that -q:a 1 is in 1-5 and it tells me I'm asking for 118
[17:44] <active8> http://pastebin.com/fzG5JALB
[17:44] <sacarasc> Maybe try using a newer ffmpeg, yours is over a year old.
[17:44] <active8> line 31 of the paste
[17:45] <active8> ah, the rebuild mess. wonder if nux-dextop has a really recent build yet
[17:45] <active8> did it last month and still have to use my build for everything but VLC
[17:47] <active8> sacarasc, oh yeah. I used a quick example of the error. c:a copy works when I can get away with it 8)
[18:31] <pa> when transcoding to h264 using a mkv container, does mplayer know at once how many frames and cluster there will be in the output file (not the offset, obviously, but just the amount)
[18:33] <benlieb> what is the best way to denoise audio?
[18:58] <benlieb> Why doesn't my audio work in Quicktime (but does in other players) when combining an mp3 and video file using FFmpeg?
[18:58] <benlieb> I've written up a stack overflow question if anyone cares to take a stab at this:
[18:58] <benlieb> http://superuser.com/questions/815125/why-doesnt-my-audio-work-in-quicktime-but-does-in-other-players-when-combininhttp://superuser.com/questions/815125/why-doesnt-my-audio-work-in-quicktime-but-does-in-other-players-when-combinin
[19:00] <c_14> Does quicktime play the mp3?
[19:00] <voidDotClass> hey guys, i need to trim just the first second of a .3gp video. can anyone tell me the command for doing that?
[19:01] <c_14> You want the first second or you don't want it?
[19:02] <voidDotClass> i dont want it
[19:02] <benlieb> c_14: yes it does
[19:02] <voidDotClass> c_14: so it should trim the first second from the video
[19:02] <c_14> voidDotClass: ffmpeg -ss 1 -i file outfile (you can use -c copy if you don't want to reencode but the cut won't be as precise)
[19:04] <voidDotClass> file = input path, outfile = output path?
[19:04] <c_14> yep
[19:05] <voidDotClass> [h263 @ 0x9d5b3a0] Invalid pixel aspect ratio 65536/65536, limit is 255/255
[19:05] <benlieb> c_14: what do you think might be the problem
[19:05] <benlieb> ?
[19:05] <voidDotClass> Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height
[19:06] <benlieb> voidDotClass: is your height an odd number
[19:06] <benlieb> ?
[19:06] <benlieb> voidDotClass: as in odd/even
[19:06] <voidDotClass> no idea, its a video recorded by an iphone
[19:06] <benlieb> voidDotClass: it would say it in the output
[19:06] <benlieb> in red
[19:07] <voidDotClass> this is what it says in red: [h263 @ 0x9d5b3a0] Invalid pixel aspect ratio 65536/65536, limit is 255/255
[19:07] <c_14> benlieb: I'm guessing quicktime doesn't like the major brand/ compatible brands header. Or it just doesn't like mp3 in mp4, I don't know. Quicktime is weird.
[19:07] <c_14> Do you have any such files you could test, or other media players maybe?
[19:07] <c_14> voidDotClass: Is the input video h263?
[19:08] <benlieb> c_14: I've been using mp3 in mp4 for a while, and haven't seen any problem.
[19:08] <benlieb> What could I do to test your idea?
[19:08] <voidDotClass> c_14: http://pastebin.com/iRxfYQ5z
[19:08] <c_14> Try encoding the mp3 to AAC and merge that.
[19:08] <benlieb> I could change the codec on the audio from copy to something else...
[19:09] <benlieb> could I just specify that in the codec?
[19:09] <c_14> probably
[19:09] <c_14> voidDotClass: First of all, you're not using FFmpeg, you're using Libav. Either see #libav for support or download a recent copy of FFmpeg. ie one of the static builds from http://johnvansickle.com/ffmpeg/
[19:10] <voidDotClass> huh. but i'm using the ffmpeg command
[19:10] <JEEBcz> then you are using a lol old ffmpeg from the time when libav still had both ffmpeg and avconv binaries
[19:10] <JEEBcz> avconv being the updated one
[19:10] <JEEBcz> and ffmpeg being the old ffmpeg
[19:11] <voidDotClass> it says built in feb 2014
[19:11] <JEEBcz> yes
[19:11] <JEEBcz> but the version probably says 0.7 or 0.8
[19:11] <benlieb> c_14: arc is experimental but I'm trying libfaac
[19:11] <JEEBcz> the build date has nothing to do with how new/old the code is
[19:11] <voidDotClass> well, any chance you could help me with this issue? i just need to trim that 1 second.
[19:11] <voidDotClass> it is 0.8
[19:12] <JEEBcz> yes, the last Libav version with ffmpeg
[19:12] <voidDotClass> you can't help me with that issue?
[19:13] <JEEBcz> which was stuck in debian and ubuntu for a long long time due to the fact that there were large API changes in the next version and all packaged applications had to be updated :V
[19:13] <c_14> First, download the latest static build from that link I had. Then you can just use h264 instead of h263 and it should just work (probably).
[19:13] <voidDotClass> ok, trying that
[19:13] <JEEBcz> lol
[19:13] <JEEBcz> I'm just going to ignore all the things overlooked in that sentence
[19:14] <JEEBcz> because at this point I'm tired and want to eat
[19:15] <rickbol> I'm trying to tee the output of my transcoded video, but vlc launches two players for the resulting ts file, and...
[19:15] <rickbol> ffmpeg no longer "sees" the -pix_fmt yuv420p... directive in my -filter_complex definition.   http://pastie.org/9584834
[19:16] <c_14> rickbol: What's your command
[19:16] <rickbol> c_14: long, ugly and complex,  http://pastie.org/9584834
[19:16] <c_14> That's the autput
[19:17] <c_14> *output
[19:17] <c_14> Not the command.
[19:17] <rickbol> hmmm, I thought I included my sorry, one second
[19:18] <benlieb> voidDotClass: have you tried to reset the SAR or DAR manually?
[19:18] <benlieb> c_14: it seems to work when using libfaac. I wonder why id doesn't like the mp3
[19:18] <rickbol> http://pastie.org/9584899
[19:19] <benlieb> c_14: but all of this is just to clean up the audio. What do you recommend for audio denoising?
[19:20] <rickbol> benlieb: I haven't used ffmpeg for audio processing, but SoX might have filters and stuff for denoising, but I'm not sure.
[19:21] <benlieb> rickbol: right now my method is to split the audio from the vid, and then use audacity or an eternal program.
[19:21] <voidDotClass> benlieb: i haven't, how would i do that
[19:21] <benlieb> It would be great if I could do something right in ffmpeg. I have thousands of videos to deal with.
[19:22] <benlieb> voidDotClass: dunno if it will work but http://www.ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar
[19:24] <c_14> benlieb: I don't think there's anything in FFmpeg to denoise audio.
[19:24] <benlieb> c_14: that seems to be the case. in that case what would you recommend. The best denoiser I've found is in iMovie.
[19:24] <c_14> rickbol: What is exact problem again?
[19:25] <benlieb> but it won't do just audio.
[19:25] <c_14> benlieb: hmm, I'd probably try sox or audacity. There's a few others I've heard of, but none I've tried.
[19:25] <benlieb> it takes forever to import and then reexport, mostly for video processing, but I don't even need the vid processed
[19:25] <benlieb> c_14: i tried audacity. the quality of the denies pales compared to imovie
[19:26] <benlieb> which is probably using the same tech in iphone
[19:26] <benlieb> woh autocorrect
[19:26] <benlieb> the quality pales...
[19:27] <benlieb> so I'm importing the vid, denoising, then re-exporting to a very small vid size, extracting the audio, then recombining.
[19:27] <benlieb> not exactly efficient
[19:27] <benlieb> is sox command line?
[19:27] <benlieb> never heard of it.
[19:27] <rickbol> c_14: couple things:  1) with the "tee...-map..." in the last line, ffmpeg reports the -pix_fmt error; 2) in playback in vlc, two vlc instances are launched, one has video, the other mostly grey with some blotches...
[19:28] <c_14> benlieb: yes it is, might be a bit complicated but there should be guides on the internet
[19:28] <rickbol> c_14: ... in vlc (when playing the resulting ts file from disk)...
[19:29] <c_14> rickbol: 1) the -pix_fmt isn't an error, just a warning. 2) can you try playing the file with ffplay or some other video player?
[19:29] <benlieb> yeah, looks involved.
[19:29] <benlieb> maybe for another day.
[19:29] <benlieb> maybe another life, lol
[19:29] <pa> if i want to encode in h264 and control the generation of the container, how do i do? should i use libavcodec and libavformats directly?
[19:30] <rickbol> c_14: it plays from totem, but this vlc behavior doesn't happen if I don't tee the ffmpeg output. If I just write to a ts file only, vlc plays fine.
[19:31] <c_14> try manually setting the format for the ts file?
[19:32] <rickbol> c_14: the biggest problem seems to be that processing bit rate goes thru the roof, and my machine can't do it. But I'm really only taking the same transcoded output and writing to disk and a socket.
[19:33] <rickbol> c_14: if i don't use the -pix_fmt switch (without tee-ing), vlc won't play the ts file at all.
[19:34] <c_14> vlc might not like yuv422p
[19:34] <rickbol> c_14: it should be yuv420p
[19:35] <rickbol> c_14: that's why I'm using the switch.
[19:36] <c_14> You mean the format filter? If you only want to use the format filter and don't want to specify -pix_fmt then you'll have to place it after the overlay filter.
[19:36] <rickbol> c_14:  I'm thinking there's something going on with the -map statements between my -filter_complex and my output, but I jsut don't understand -map yet.
[19:37] <c_14> Your map is fine.
[19:37] <rickbol> c_14:  so what are the "[xxx]" things, just names?
[19:38] <c_14> I think they're called pads, you can visualize them as buckets.
[19:38] <c_14> You put things into the bucket.
[19:38] <c_14> And then you take it out again.
[19:38] <rickbol> yea, ok
[19:39] <rickbol> c_14: are you saying I need to move the yadif, denoise and format, elsewhere in my command?
[19:39] <c_14> just the format
[19:41] <rickbol> c_14:  ok, but I don't get the warning if I leave the "tee.." out of my output... which seems strange.
[19:42] <c_14> It does...
[19:45] <voidDotClass> c_14: so i've downloaded the static library and tried it again, here's my output this time: http://pastebin.com/4Sj65dCh
[19:45] <voidDotClass> any ideas?
[19:45] <vmBenLubar> what does "DTS -922328613750660, next:2260000 st:0 invalid dropping" mean?
[19:45] <c_14> voidDotClass: add -c:v libx264
[19:46] <rickbol> c_14: tee-ing for my case, since my file container is a transport stream (ts) and my I'm just routing that to a socket, that there's way to much processor being consumed
[19:46] <voidDotClass> ./ffmpeg -ss 1 -i ~/Vid*/VID*.3gp -c:v libx264 out.3gp ?
[19:47] <c_14> ye
[19:47] <voidDotClass> now i get this: http://pastebin.com/Pd1y8U7f
[19:48] <c_14> add -c:a copy
[19:48] <vmBenLubar> here's my command line and input and output: http://pastebin.com/wRmJA6zv
[19:49] <voidDotClass> replace the -c:v with the -c:a ?
[19:49] <c_14> nah
[19:49] <voidDotClass> or add a new one in addition to the last?
[19:49] <c_14> just add another one
[19:49] <voidDotClass> ok
[19:50] <voidDotClass> nice, finally seems to be working
[19:50] <c_14> vmBenLubar: As long as the output file[s] play, ignore it.
[19:50] <vmBenLubar> c_14: it does, but can I get it to not print that message a billion times?
[19:50] <c_14> You can probably decrease the loglevel.
[19:51] <vmBenLubar> I tried that, but it also removed "frame=  112 fps= 55 q=0.0 q=31.0 size=      63kB time=00:00:02.26 bitrate= 227.5kbits/s dup=1 drop=0"
[19:51] <rickbol> c_14: my purpose for tee-ing was so I could view the stream (confidence monitoring) of the file, But this tee-ing uses too much cpu. I can
[19:51] <rickbol> jsut open the file in mplayer w/o cpu problems. taht doesn't seem right.
[19:52] <c_14> It shouldn't use more cpu, it might burn a bit of io, but hmm.
[20:05] <rickbol> c_
[20:08] <rickbol> c_14: there's some interaction between my filter and tee-ing the output. I removed "filter=yuv420p" from  filter_complex, and added -pix_fmt yuv420p elsewhere, and now...
[20:08] <XHFHX> Hi there. I have a problem with the pipe command. At first I tried "-f avi pipe:" but the quality wasn't good enough. So now i changed it to "-c:v rawvideo -f avi pipe:" which works, but only until the file hits 360MB, after that the file doesn't grow anymore although it's still rendering. I'm piping to ffmbc
[20:08] <rickbol> c_14:  I see that the second vlc instance shows me the video, but w/o the overlay.
[20:09] <c_14> And the first instance shows the video?
[20:09] <rickbol> c_14: so my file has two video streams, one with overlay, and the second, without.
[20:09] <rickbol> both have the main video.
[20:10] <c_14> oh, wait, yes
[20:10] <c_14> You have too many maps
[20:10] <rickbol> ah, I had hoped so!
[20:10] <c_14> get rid of the -map 0:v and the -map 0:a
[20:11] <c_14> Didn't see those last time.
[20:11] <pa> is there a guide somewhere to transcode using libav* directly and not ffmpeg?
[20:11] <rickbol> c_14: in the output line?
[20:11] <pa> like for example to create the header and the trailer by hand
[20:11] <pa> and the raw data stream
[20:11] <llogan> XHFHX: why pipe to ffmbc?
[20:12] <pa> i tried to look into the source code a little, but its large and i got confused pretty quickly
[20:13] <llogan> pa: take a look in doc/examples. it may or may not have some useful stuff for you
[20:13] <pa> thanks!
[20:13] <XHFHX> because ffmpeg doesn't support XDCAMHD/xd5c
[20:13] <llogan> pa: also https://www.ffmpeg.org/doxygen/trunk/index.html
[20:14] <llogan> XHFHX: so why pipe from ffmpeg instead of using ffmbc to decode?
[20:15] <XHFHX> because ffmpeg doesn't support concatting of videos which I need
[20:15] <rickbol> c_14: much better!!! I'm able to write to file and monitor @ < 100%cpu ! thx. Now I need to tweak for max video quality while not maxxing cpu.
[20:15] <llogan> XHFHX: do you mean ffmbc in that last sentence?
[20:16] <XHFHX> llogan: sorry, yes :)
[20:17] <XHFHX> llogan: http://pastebin.com/aTGNtQMH
[20:17] <llogan> and if ffmpeg doesn't support something you can submit a feature request. it may be possible to port from ffmbc (despite the lack of a version control repo and the license)
[20:18] <llogan> XHFHX:  and the COMPLETE console output
[20:18] <llogan> many people ignore that part. i'm not sure why.
[20:20] <XHFHX> llogan: oh, sorry. but i don't know how to output the complete console output in windows, as the first lines begin to disapear after some time
[20:21] <XHFHX> currently im making a test with piping from ffmpeg to ffmpeg to see if it's an pipe error or a ffmbc error
[20:21] <XHFHX> because when I used the pipe without the -c:v rawvideo it worked but the quality was awful
[20:24] <_dunno_> ffmpeg by mistake considered my IN to be OUT - now the file still exists in the file table but is of zero-size. Is there some approved procedure which could be used to revive my movie?
[20:24] <llogan> did you encounter a bug that caused this to happen?
[20:25] <active8> if using setpts to slow down a video and atempo to keep the sound pitch the same - i get some blurring and maybe a little bit of sound degradation. mp4 to mp4. so those filters work on timestamps and I'm not sure if this will work, but how could I double - make redundant frames - and get that to play like the original, and then maybe if I slowed it down with setpts and atempo, I wouldn't be losing info.
[20:26] <_dunno__> sry - no - not a bug - but i ended up with an error msg
[20:26] <active8> or just fake the filter by changing bitrates and fps?
[20:27] <active8> i'd still need redundant info, i think
[20:30] <llogan> active8: sox may do a better job for audio, and others have mentioned using slowmovideo for video to perfrom interpolation
[20:32] <debianuser> _dunno_: no easy way, sorry. :( Try regular file recovery tools (or shut the PC down, disconnect a disk where your movie was and bring it to some file recovery service) (it's better to not write anything to that disk, that would increase a chance to recover a file)
[20:34] <active8> llogan: i wonder if openshot can do slow mo (it's not a built in effect) also wonder if syncing will be a chore after using two different tools
[20:34] <llogan> sorry. i don't know the answer to either of those questions.
[20:35] <debianuser> (for some systems if you notice a file loss and instantly cut of the power from the dist within ~5 seconds after file being overwritten there's a close to 100% chance to manually recover the filesystem with the original file on it)
[20:35] <active8> i heard sox is good and it''s in one of my trusted repos already
[20:44] <XHFHX> llogan: http://pastebin.com/G1yHAjKD hope this is ok, somehow it didn't store the console output into a file, so I only have the beginning and the ending of the log, but I think this is the intresting part
[20:46] <active8> i seem to be lacking a resource that explains the relation between fps, bitrate, etc for both audio and video - in case I want to tweak things (including speed up and slow down, maybe) Where's a good article? I saw some equation posted once. I hope the info isn't scattered across 100 bad articles
[20:47] <kepstin-laptop> well, there's no direct relation between fps and bitrate, except that in most codecs you need more bitrate to encode something with higher fps at the same quality
[20:50] <active8> bitrate is related to how fast it spits out to the screen, right? like bandwith in ntsc - pixels per second = bitrate/bits-per-pixel (roughly) ?
[20:51] <active8> so you'd need more bps to fill a hiher fps
[20:51] <kepstin-laptop> active8: only for raw video. For modern encoded video, it's more complicated
[20:52] <active8> yeah, that's the prob
[20:52] <active8> i would be converting to raw if it wasn't already raw, doing my thing, and reencoding
[20:53] <active8> just want the nuts and bolts engineering in one article to get a handle on it all.
[20:53] <active8> without the huffman coding explanation, wavelets, etc
[20:54] <kepstin-laptop> pretty much the only reason to set an output bitrate nowadays is that you need the file to fit a specific size, or to fit in a bandwidth restriction due to format (e.g. bluray drive speed, internet connection speed); for most uses, quality-based encoding (e.g. x264's crf) is preferred.
[20:55] <kepstin-laptop> but as far as bitrate goes, what happens is that you pick the bitrate you want, and then the codec is responsible for allocating bits to try to keep the best quality it can with the bits you're allowing it to use.
[20:56] <active8> what if I want to take each audio or video frame, duplicate it, and encode it at a different fps and have it play at the same "speed". if I can do that, I can them change the speed without messing with setpts dropping frames I need - maybe.
[20:57] <active8> say raw or mp4 to raw (and mp3 or aac to wav) and then mes with the info before reencoding at [insert equation i asked for] bps/fps
[20:58] <active8> double # of frames and set fps to oldfps*2 or same-fps to get half speed
[20:59] <active8> then VLC chokes on an off FPS and I'm back to how can it be done
[20:59] <kepstin-laptop> I'm not clear on what exactly your final goal is
[21:00] <kepstin-laptop> do you have a source video, and want to make a new video that's slow motion - i.e. it takes twice as long to play?
[21:00] <active8> not using setpts and atempo
[21:00] <active8> i should extract the pictures and flip them like a deck of cards
[21:01] <XHFHX> hi there, can anyone help me with this? http://pastebin.com/G1yHAjKD no matter how big the video is it renders, it doesn't grow beyond 360mb but still renders the whole video
[21:02] <kepstin-laptop> active8: it sounds like you had something that worked, but the final encoded video and audio quality was too low?
[21:03] <llogan> XHFHX: did it work as expected when you piped from ffmpeg to ffmpeg? what if you try a different encoder? what if you try a different format?
[21:03] <XHFHX> ffmpeg->ffmpeg is still rendering
[21:04] <llogan> if that works then you'll have to ask in #ffmbc I guess
[21:05] <active8> kepstin-laptop,  yeah. little bit of sound degradation, but blurring and it looks like the sync is off, but who knows with a youtube of a dude playing a lightning fast classical acoustic intro to the best song ever written (almost)
[21:05] <kepstin-laptop> well, you'll get some audio degredation with the audio stretch, but it sounds like your main issue is that you didn't set the quality on the encoders for the output video sufficiently high.
[21:06] <active8> try youtube rush la villa strangiatto acoustic intro (you'll see that from the thumbnail) cut the intro with -ss t and see what I mean - i'll post the command
[21:06] <active8> -c:v libx264 -preset slow -crf 0 -c:a libmp3lame -q:a 0
[21:06] <active8> -filter_complex "[0:v]setpts=4*PTS[v];[0:a]atempo=0.5,atempo=0.5[a]"	-map "[v]" -map "[a]"
[21:07] <active8> in reverse order, of course
[21:08] <active8> the docs talk about presets, but not really how they affect things other than compression and file size
[21:08] <active8> and they're not even clear on that
[21:08] <kepstin-laptop> hmm, yeah, those settings should give you pretty much the best audio and video you can
[21:09] <active8> "the choice is about this over that" but nothing about which does what
[21:09] <XHFHX> llogan: ffmpeg->ffmpeg is fine, so it must be a ffmbc problem :/
[21:09] <active8> that's why I want to try it the ugly hardcore way like
[21:12] <llogan> XHFHX: they may tell you to try 0.7.1 (if they even respond at all).
[21:12] <kepstin-laptop> if you're looking at the samve video I see, youtube id bOyEerX6k9M, the original video is pretty bad quality, so it's never gonna not look blurry
[21:12] <active8> not just for this project, either. i could maybe just slow down VLC and do alright, but there has to be some knowledge of how to do this without setpts and google isn't obeying me on this one
[21:14] <llogan> XHFHX: until then keep trying other stuff. -vcodec huffyuv. -f mov, -f matroska, etc. just shooting blind here...
[21:14] <active8> the only other one with an acoustic intro has the camera a bit far from the fretboard, but I might try that one, anyway. IT's motion blur big time at half and quarter speed.
[21:14] <kepstin-laptop> active8: but yeah, many videos that don't appear blurry at full speed will appear blurry when slowed down, simply because of how your mind perceives motion. Both the original and the slowed down have the same blur
[21:15] <active8> true
[21:16] <kepstin-laptop> and no matter what, you will get audio degredation when slowing down audio to ¼ original speed and attempting to keep pitch. Might be some tools that can do a better job than ffmpeg's atempo filter, but I dunno which.
[21:18] <active8> and there are two different positions to play this in. I might have to load XP on a machine and get the music printed out and just use my ear on the slowed down audio. I'd still like to find a way to avoid setpts/atempo. I know there was a way cause the docs refer to an old way. And I also know thre was an equation out there with bps fps kinda stuff in there maybe sample rate, fps, and speed.
[21:18] <XHFHX> llogan: i tried the same command again, but now I'm getting a "broken pipe" error message...
[21:18] <active8> sox might work on the audio
[21:19] <active8> a pipe's supply was terminated - just got that on ctrl-c from a pipe into less
[21:19] <kepstin-laptop> active8: you're not going to be able to get any better video than what that command is already giving you. But you might be able to use sox to do the audio, it might give better results.
[21:19] <XHFHX> llogan: nevermind, somehow the ffmpeg.exe file deleted it's concetnt itself so it was 0kb
[21:20] <kepstin-laptop> (the only way to get better video would be to use a higher quality source video)
[21:20] <active8> garbage in, garbage out
[21:20] <active8> next time alex lifeson gets locked up, i'll have the judge order community service right here
[21:21] <kepstin-laptop> all that setpts command does is make each video frame appear on the screen for 4 times as long, and the x264 settings are making it save the frame exactly as it was in the original video.
[21:21] <active8> ok. not sure why i was thinking it dropped frames. that would be fast motion
[21:23] <active8> well thanks, kepstin-laptop. At least I have something that will work on future endeavors with good video. think i'll go record my neighbor kicking his car and busting his knuckles with a wrench
[21:25] <mjuszczak> Are there ffmpeg 2.x packages for Ubuntu 14.04?  The PPA only goes up to 1.2.x
[21:29] <pa> llogan, but in theory does ffmpeg itself allow to produce only the "container" of a transcoded file? that is, without the av data itself?
[21:38] <XHFHX> @llogan: it seems -c:v rawvideo makes some problems. do you know what I could use instead? when I use nothing it's a damn ugly quality :/
[21:39] <XHFHX> looks like 256 colors with a resolution of 200x400 upscaled to 1080p
[21:54] <roninhack> hi
[21:54] <roninhack> anyone here? i have a question about building opencl
[21:55] <roninhack> anyone here?
[21:56] <MikeJoel> ffmpeg? told it would convert flash or avi to mpeg4?
[21:59] <MikeJoel> is ffmpeg an app or dll, or??
[22:04] <c_14> https://ffmpeg.org/
[22:06] <MikeJoel> thanks
[22:14] <llogan> XHFHX: i mentioned huffyuv before
[22:15] <llogan> pa: i don't know
[22:16] <pa> does it happen that av_write_trailer actually modifies the header too?
[22:20] <Sokolio> pa; yep
[22:21] <Sokolio> in mp4, when moving moov
[22:21] <pa> also in mkv i suppose
[22:22] <Sokolio> maybe
[22:22] <pa> do you know, however, if , for example in mkv, it is possible to know beforehand the number of cue points? (not their offset, obviously)
[22:23] <pa> it looks like that ffmpeg api are a little how to say
[22:23] <pa> inflexible , if one wants to do something weird
[22:24] <Sokolio> maybe libMatroska would help?
[22:24] <pa> is it used by ffmpeg?
[22:24] <Sokolio> I like to use faac or x264 directly
[22:25] <Sokolio> so why not use a muxing lib directly
[22:25] <Sokolio> i have no idea, I guess ffmpeg has its own parser
[22:25] <pa> you mean muxing raw x264 streams?
[22:25] <Sokolio> no, I meant encoding through x264 directly and not by using libx264 codec
[22:26] <Sokolio> sometimes it's not bad to walk around ffmpeg
[22:26] <pa> could be an option.. can x264 write mkv by itself?
[22:27] <Sokolio> the lib? nope
[22:27] <Sokolio> it's just an encoder
[22:27] <pa> and the output is? mp4?
[22:27] <Sokolio> the output is h264 bitstream, no container, no format
[22:27] <pa> so raw h264
[22:28] <Sokolio> yes
[22:32] <kepstin-laptop> x264 can optionally write to mkv, flv, or mp4 depending on how it was compiled.
[22:32] <kepstin-laptop> (the executable)
[22:33] <Sokolio> yep, the binary executable can mux several formats
[22:33] <Sokolio> but the lib is just for encoding
[22:34] <JodaZ_> kepstin-laptop, how does it do that?
[22:35] <kepstin-laptop> JodaZ_: for mp4 it uses gpac; it has built-in minimal muxers for the other formats.
[22:35] <pa> for mkv? libmatroska?
[22:36] <JodaZ_> k
[22:37] <Sokolio> I think I've never used x264 through CLI
[22:38] <Sokolio> I peeked ino the source
[22:38] <Sokolio> there;s a minimal muxer of their own design
[22:39] <Sokolio> so it's not libmatroska
[22:39] <Sokolio> there's also a flv muxer
[22:40] <pa> ah i see
[22:42] <Sokolio> but that won't probably solve your issues with cue points
[22:46] <pa> well what i need in fact is just the amount of necessary cue points and their position in time
[22:46] <JodaZ_> hmm, question, will ffmpeg get fame exact seeking any time soon? (so one can make HLS/DASH fragments with it)
[22:46] <pa> i guess this could be derived by the number of frames in the movie and selected number of frames per keyframe
[22:46] <kepstin-laptop> JodaZ_: seeking should be frame exact by default in current ffmpeg versions (unless you're using -v:c copy, of course)
[22:47] <JodaZ_> kepstin-laptop, having problems with audio
[22:48] <pa> kepstin-laptop, seeking as in -ss?
[22:48] <JodaZ_> with audiosamples split and filled with silence
[22:55] <pa> but is there a way to make ffmpeg seek by frame/keyframe instead of time offset?
[22:55] <pa> i mean either ffmpeg, or via ffmpeg apis in some custom example
[23:03] <JodaZ_> pa, you also interested in perfect splitting?
[23:07] <llogan> kepstin-laptop: or l-smash instead of gpac
[23:14] Action: Suchiman hears perfect splitting
[23:14] <Suchiman> yeah thats something i could also need :P cutting away ads from recordings
[23:14] <Suchiman> on the other Hand still searching for something making cutting as easy as possible
[23:20] <pa> yes, i need it for correct transcoding when seeking
[23:22] <Suchiman> pa: well, if you write -ss before the Input file, the Navigation will be keyframe based
[23:23] <Suchiman> pa: https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg#Fastseeking
[23:26] <JodaZ_> Suchiman, well for ads the cuts don't need to be perfect
[23:27] <JodaZ_> Suchiman, problem there is that its only keyframe based for the video, the audio is still cut inside frames
[23:27] <Suchiman> JodaZ_: is there a recommend technique? i'm currently cutting out only the movie material (copy), then put them back together using the concat demuxer
[23:28] <JodaZ_> don't think there is one
[23:29] <Suchiman> mhh.. a skip Video filter would be interesting
[23:29] <JodaZ_> that stuff exists
[23:29] <JodaZ_> but filters only work for reencoding i think not for copy codec
[23:30] <Suchiman> would be okay for the final stage, comming from raw / lossless compressed Input to cutted and filtered into transcoding
[23:35] <vmBenLubar> how do I find out what pix_fmt are supported with webm?
[23:35] <Suchiman> vmBenLubar: should be dependent on the codec, do you mean vp8 / vp9 ?
[23:35] <vmBenLubar> vp8
[23:36] <Suchiman> vmBenLubar: according to the RFC: VP8 works exclusively with an 8-bit YUV 4:2:0 image format.
[23:36] <Mavrik> https://www.ffmpeg.org/doxygen/trunk/libvpxenc_8c_source.html
[23:37] <Mavrik>   896     .pix_fmts       = (const enum AVPixelFormat[]){ AV_PIX_FMT_YUV420P, AV_PIX_FMT_NONE },
[23:37] <Mavrik> so that's that
[23:37] <Mavrik> oh sorry, that's VP9
[23:37] <Mavrik> VP8:   867     .pix_fmts       = (const enum AVPixelFormat[]){ AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUVA420P, AV_PIX_FMT_NONE },
[23:40] <vmBenLubar> thanks
[23:48] <asix3> hello everyone. I'm trying to grab video streamed over rtp from a raspberry pi -- the code I have to do so is here: http://pastebin.com/1ALcP0WH .  I get an error saying: Unable to receive RTP payload type 96 without an SDP file describing it
[23:48] <asix3> I'm streaming it from the pi using this gstreamer pipeline: raspivid -t 0 -h 720 -w 1080 -fps 25 -hf -b 2000000 -o - | gst-launch-1.0 -v fdsrc ! h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=131.156.68.41 port=5000
[23:48] <asix3> thoughts?
[00:00] --- Tue Sep 23 2014