[Ffmpeg-devel-irc] ffmpeg.log.20140316

[burek]

[01:57] <everdred> Hello all.
[01:57] <everdred> Does anyone have experience compiling ffmpeg on Slackware?
[01:57] <iive> i'm on slackware
[01:59] <everdred> Good to hear! I'm having trouble using the Slackbuild
[01:59] <everdred> I install the packages it asks for one by one as it asks for them...
[01:59] <everdred> but then it asks for X11.
[02:00] <everdred> I'm doing this on a server without X, but I'll install whatever I need to to get ffmpeg built.
[02:00] <everdred> Except I can't tell which actual package it needs.
[02:01] <relaxed> X is only required if you want ffmpeg's x11grab
[02:01] <everdred> Ah. I probably don't need that.
[02:04] <everdred> I don't see that among the "optional features" listed on the Slackbuild page for ffmpeg: http://slackbuilds.org/repository/14.1/multimedia/ffmpeg/
[02:05] <everdred> Does x11grab support fall under one of the options they have listed there?
[02:07] <everdred> Oh wait, I see there's a line in the Slackbuild script for x11grab, even though it isn't listed on the page.
[02:28] <everdred> Removing that line appears to have worked; it's actually building now.
[02:28] <everdred> Thanks, relaxed!
[03:36] <t4nk622> hola buena noches
[03:37] <t4nk622> soy usuario de Blag 140000 he instalado Gnash y me pide q instalee los codecs Gstreamer-ffmpeg alguien me puede dar un empujon por ahi
[03:38] Last message repeated 1 time(s).
[03:38] <t4nk622> alguien quien me pueda ayudar
[03:40] <t4nk622> soy usuario de Blag 140000 he instalado Gnash y me pide q instalee los codecs Gstreamer-ffmpeg alguien me puede dar un empujon por ahi
[03:41] <mmint> gstreamer-ffmpeg is not part of ffmpeg. It's just a plugin that interfaces the ffmpeg library with gstreamer. http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html
[03:45] <t4nk622> por favor los comandos para instalar y desempaquetar con blag140000, las disculpas del caso soy algo novato en esto
[03:48] <mmint> I don't know. Try asking in #blag on irc.indymedia.org
[03:51] <t4nk622> mil gracias <mmint> tratare de resolver por el otro medio
[03:52] <mmint> buena suerte
[14:08] <maister> Does the RTP reader in libavformat reorder packets? If so, is there a way to increase the buffer size?
[14:43] <roger21> hello, i get "Error getting first pass stats" while trying to do a 2 pass theora encode
[14:44] <roger21> here is the command : ffmpeg.exe -i "video.mkv" -map_metadata -1 -c:v libtheora -s 256x144 -aspect 16:9 -r:v 30000/1001 -q:v 10 -pass 1 -passlogfile lolilol -an -sn -f rawvideo -t 9 -y NUL &
[14:44] <roger21>  -&
[14:46] <roger21> i also have that : "Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height"
[14:48] <roger21> oh it seems to be the -q:v 10
[14:48] <klaxa> i'd be surprised at that
[14:48] <klaxa> i don't think you can specify -f rawvideo when using -c:v libtheora
[14:49] <klaxa> also given the resolution, are you really aiming for high quality? i would think a two-pass encode is unnecessary
[14:49] <klaxa> unless you have to match a certain filesize
[14:51] <feld> does anyone know how to get ffmpeg to encode alac files that are "optimized for streaming" ?
[14:52] <feld> there are a few tools out there that can do this. dbPowerAmp describes it as this:
[14:52] <feld> "m4a Layout can be 'Optimized for Streaming' (where the audio details and tags are placed before audio data)"
[14:52] <klaxa> is a lossless codec ever optimized for streaming?
[14:52] <klaxa> ah
[14:52] <klaxa> that's more of a container thing is it not?
[14:53] <feld> yeah, sort of, but I was hoping I could fix this as I convert things from FLAC to ALAC with ffmpeg
[14:54] <feld> the resulting files that ffmpeg produces do not work well with streaming to iOS devices, etc and any tool that doesn't require me to use a Mac or Windows machine to touch all these files would be ideal
[14:54] <feld> I'd like to be able to do it on the server itself
[14:54] <klaxa> i don't know a lot about alac and streaming lossless audio, sorry
[14:54] <klaxa> i usually encode to a lossy codec to stream
[14:54] <klaxa> i think it also depends on the streaming software you are using
[14:55] <feld> wait, i may have found this finally
[14:55] <feld> i should have looked for AAC first -- -movflags +faststart
[14:56] <feld> i think if I add that to my ffmpeg flags it will produce what I want. I'll give it a shot.
[14:56] <klaxa> sounds good
[14:57] <roger21> no of corse, it's test, what should i put to -f then ?
[14:57] <klaxa> ogg maybe?
[14:58] <klaxa> matroska should work too
[14:58] <roger21> no but it's the first pass
[14:58] <roger21> i put ogg at the second one
[14:59] <klaxa> you still need to specify a valid container i think
[15:12] <roger21> if i use -b instead of -q it seems ok
[15:12] <roger21> why no -q in two pass
[15:45] <Riso> hello
[15:45] <Riso> I am trying to "burn" in srt subtitles to a .mkv; I used "-i in.mkv -vf subtitles=in.srt out.mkv"
[15:46] <Riso> it created a "out.mkv" which is 1/2 of "in.mkv" and no subtitles in the video
[15:46] <Riso> can someone advise what is the correct way to achieve this?
[15:52] <Riso> fflogger: http://pastie.org/8935716
[15:53] <Riso> or klaxa ;)
[15:55] <klaxa> there are fontconfig errors
[15:55] <klaxa> i haven't seen them before
[15:55] <klaxa> i don't know if that makes a difference
[15:55] <Riso> all i downloaded was the ffmpeg binary - i am on MACOS
[15:55] <Riso> i followed http://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20video
[15:56] <klaxa> as to why the file is so much smaller, your encoding settings most likely don't match the input's quality
[15:56] <Riso> i want the video to be untouched
[15:56] <Riso> what should i use so it is the same?
[15:57] <klaxa> that's impossible
[15:57] <klaxa> at least with lossy compression, and you don't want lossless compression for production because of the resulting size
[15:58] <Riso> what does this mean?
[15:58] <Riso> Input stream #0:1 frame changed from rate:48000 fmt:fltp ch:2 chl:stereo to rate:48000 fmt:fltp ch:6 chl:5.1(side) Input stream #0:1 frame changed from rate:48000 fmt:fltp ch:6 chl:5.1(side) to rate:48000 fmt:fltp ch:2 chl:stereo
[15:58] <klaxa> anyway, back to the issue at hand, could you try using ass instead of srt? the encoding guide describes how to convert srt subs to ass subs
[15:58] <Riso> yes :) with ass its encoding
[15:59] <klaxa> is it rendering the subtitles correctly?
[16:00] <Riso> in vlc, mplayer always played them perfectly
[16:01] <klaxa> are you sure the subtitles are not rendered by the player, but actually burned into the video?
[16:01] <dvnl> Hi Everyone! Please, help me with this issue: how can I cross-compile ffmpeg to an ARM bare-metal (so no linux). I tried the following: ./configure --disable-everything (#--selecting only the needed decoders) --disable-asm --disable-yasm --enable-cross-compile --arch=arm --target-os=none --disable-optimizations. I'm trying to do this from Windows, MinGW. What other options should I add?
[16:02] <Riso> here is with "ass" http://pastie.org/8935733
[16:03] <klaxa> well is it burning in the subtitles correctly?
[16:03] <Riso> when i Play  the original .mkv it fetches the original corresponding .srt
[16:03] <klaxa> can you disable that?
[16:03] <Riso> when i played out.mkv there were no subtitles in the video
[16:03] <klaxa> that is not good...
[16:04] <Riso> the buring with "ass" it says [Parsed_ass_0 @ 0x101b06260] Added subtitle file: 'in.ass' (2 styles, 770 events)
[16:05] <Riso> it did not say that when i used srt
[16:05] <klaxa> you mean the styles and event counts? that's .ass specific
[16:05] <klaxa> although the event counts should be the same
[16:05] <Riso> ah hmm
[16:06] <Riso> if i play out.mk which is being written to, the video plays, no subtitles in the video :(
[16:06] <klaxa> :/
[16:07] <klaxa> without having the actual material, i'm out of ideas, sorry
[16:23] <c_14> Riso: about the output file being smaller than the input file, that's because you're reincoding the source video and audio. In order to burn the subtitles into the video you need to reencode the video, but you might want to try messing with the quality settings, for the audio you can use -c:a copy.
[16:23] <c_14> And about the subtitles not burning, I found this bug report: https://trac.ffmpeg.org/ticket/2100
[16:24] <c_14> Apparently the issue is that fontconfig can't find a config file and therefore doesn't have font paths.
[16:24] <c_14> Not sure if this is your issue, but it might help.
[16:28] <Riso> c_14: thanks for the tip, how do i get the fonts to be fixed?
[16:29] <c_14> You could try what the person with the trac ticket did. Second to last comment on the link I posted.
[16:29] <Riso> c_14: how do i leave audio untouched
[16:29] <Riso> ?
[16:29] <c_14> Just add -c:a copy to the commandline.
[16:37] <Riso> ok.. trying again. no warning with fonts seen this time..
[16:38] <Riso> Stream mapping:   Stream #0:0 -> #0:0 (h264 -> libx264)   Stream #0:1 -> #0:1 (vorbis -> libvorbis) Press [q] to stop, [?] for help [Parsed_subtitles_0 @ 0x101b03b00] Neither PlayResX nor PlayResY defined. Assuming 384x288 frame=  717 fps= 54 q=28.0 size=    3440kB time=00:00:30.13 bitrate= 935.2kbits/s
[16:38] <Riso> works!!!
[16:39] <Riso> c_14: thanks a lot!!
[16:39] <c_14> No problem.
[17:06] <namccarty> Alright, I am having a rather confusing issue, running the command ".\ffmpeg.exe -i .\song.mp3 -ab 96k -v 0 -f mp3 - > test-a.mp3" results in a 6MBish unplayable file, but ".\ffmpeg.exe -i song.mp3 -ab 96k -v 0 -f mp3 test-b.mp3" results in a 3MBish playable file
[17:06] <namccarty> what am I doing wrong here?
[17:06] <namccarty> I should also note that this issue is only affecting specific files
[17:11] <namccarty> hold on, i lied a little, apparently output to stdout isnt working at all for me now
[17:19] <namccarty> here is the output with loglevel set to verbose with the working version: http://pastebin.com/H2Ks5eRH , and the non-working version: http://pastebin.com/u0TnfmTX
[17:20] <namccarty> any help would be appreciated, as this problem has been plaguing me for a while
[17:23] <klaxa> namccarty: use pipe:1 instead of - for output
[17:24] <namccarty> mp3 is still unplayable and twice as big as the playable one
[17:25] <namccarty> command and output here:
[17:25] <namccarty> http://pastebin.com/dbpfaWhH
[17:26] <namccarty> also the file is a few KB larger than the one that i got when using - for the output
[17:27] <klaxa> that is weird
[17:27] <klaxa> also according to the output ffmpeg provides all files should be the same size
[17:27] <namccarty> yeah, that is not the result I am getting
[17:28] <klaxa> maybe windows handles pipes weirdly?
[17:28] <namccarty> the result is persisting and identical on another build i just swaped out
[17:28] <klaxa> what i can imagine is that some text-output is mixed with the audio output
[17:29] <klaxa> can you upload the resulting file?
[17:29] <namccarty> yes
[17:29] <klaxa> the larger one
[17:29] <namccarty> one moment
[17:30] <namccarty> https://dl.dropboxusercontent.com/u/42936303/help3.mp3
[17:31] <namccarty> I've been having the same problem on linux, but I've only done serious troubleshooting on windows so far
[17:31] <namccarty> let me get out the laptop and see if I can reproduce these results on linux
[17:33] <klaxa> it seems that almost every other byte is 0x00
[17:33] <namccarty> that is disturbing
[17:38] <namccarty> alright, testing on the laptop
[17:41] <namccarty> output appears to be correct for both - and pipe:1 on that particular file, let me locate one i was having trouble with in the past
[17:45] <namccarty> appernently the issue isnt happening on my laptop at all
[17:45] <namccarty> given that i was having this issue on linux before, and across two machines, i am now questioning my own sanity
[17:46] <namccarty> i dont have sound on my laptop though, so the files are the same size, i dont know if they are both valid though
[17:46] <Riso> if i run "~/ffmpeg -i in.mkv -c:a -vf subtitles=in.srt out2.mkv" it says
[17:46] <Riso> File 'subtitles=in.srt' already exists. Overwrite ? [y/N] ^C
[17:46] <Riso> if i ommit "-c:a" it runs fine
[17:47] <c_14> -c:a needs an option/value after it.
[17:47] <klaxa> use -c:a copy to copy the audio stream
[17:47] <klaxa> as in: ~/ffmpeg -i in.mkv -c:a copy -vf subtitles=in.srt out2.mkv
[17:47] <Riso> i want the audio to be untouched
[17:47] <Riso> "-c:a copy" ?
[17:47] <klaxa> yes, that will do that
[17:48] <namccarty> welp, laptop is doing everything fine, must have managed to get a janky build of ffmpeg multiple times
[17:48] <Riso> ah thanx
[17:49] <namccarty> oh wait, now i remember, i managed to get it working on my desktop right before i started encountering cripiling graphics driver issues
[17:51] <klaxa> maybe windows is just bad at piping
[17:52] <namccarty> possibly
[17:54] <namccarty> okay, turns out it is just powershell that is bad at piping
[17:54] <namccarty> now to figure out why madsonic is trying to ship my phone the version powershell is making
[17:58] <namccarty> okay, its shipping out a valid mp3 after the switch to pipe:1 from -
[17:59] <namccarty> but its not one my phone wants to play
[18:03] <namccarty> annoying, but i dont think it's ffmpegs problem at this point
[18:03] <klaxa> the 0x00 bytes in the file won't come from ffmpeg
[18:04] <klaxa> maybe switch to *nix as your default os? :>
[18:05] <namccarty> problem with that on my desktop is graphics drivers
[18:05] <namccarty> im waiting for the stable amd drivers to support the latest version of xorg
[18:06] <namccarty> but it appears that even though i now have ffmpeg outputting something that VLC can play
[18:06] <namccarty> for some songs it is still outputting an invalid mp3 on both windows and linux
[18:07] <namccarty> but
[18:07] <namccarty> only when it is being piped
[18:07] <namccarty> huh
[18:08] <namccarty> piped version contains one less mpeg frame
[18:08] <namccarty> both the linux and the windows produced versions
[18:08] <klaxa> huh
[18:08] <klaxa> what happens if you use lame without ffmpeg inbetween?
[18:11] <namccarty> hold on, let me see what happens when i try that
[18:16] <namccarty> same thing with the one less mpeg frame, let me see what happens when i try to play it on my phone
[18:16] <namccarty> works on my phone and the mp3 is totally valid
[18:17] <klaxa> well yeah that's the way mp3 works
[18:17] <klaxa> you take frames and concatenate them
[18:17] <klaxa> frame size is specified in the 4-byte header
[18:18] <namccarty> well, thing is though, the only diffrence is one file is being piped and another is being output as a named file, i know you could get the same data with a diffrent number of frames
[18:18] <namccarty> just strikes me as odd that we are getting a diffrent number here
[18:18] <klaxa> not really
[18:18] <klaxa> if the number of frames is different, the file is different
[18:18] <klaxa> and yeah it's pretty odd
[18:19] <klaxa> let me see if i can reproduce
[18:19] <namccarty> i meant data here as in what the person eventually hears
[18:20] <namccarty> but yeah, something about using ffmpeg with this particular song (and all the others i have by the same artist) appears to be corrupting the file when it is piped
[18:21] <namccarty> but in such a way that VLC is still able to play it, but the media player on my phone is not
[18:22] <namccarty> here is one of the songs that triggers issue for me
[18:22] <namccarty> https://dl.dropboxusercontent.com/u/42936303/all.mp3
[21:18] <hannes3> i encode images to a video with "-i %04d.png -vcodec libx264 -pix_fmt yuv420p out.mkv". now i would like to add an audio track. i have a mp3 but in the video the audio would need to be at 19.55s already. can i do that on the commandline or do i need to cut the mp3 beforehand?
[22:33] <llogan> hannes3: ffmpeg -i video.mkv -i audio.mp3 -c copy -shortest output.mkv
[22:33] <llogan> if i understand yuor question correctly
[00:00] --- Mon Mar 17 2014