[Ffmpeg-devel-irc] ffmpeg.log.20140809

Sun Aug 10 02:05:01 CEST 2014

[00:14] <elfer> best dvd author software on linux suggestions?
[00:15] <spectralsun> is there any kind of test you can do to determine if a video has been qtfaststarted?
[00:23] <sacarasc> spectralsun: Copy the first half of it somewhere and see if it playsd.
[00:24] <Mavrik> spectralsun, "qtfaststarted"... a better term is "has MOOV atom at the start of file"
[00:24] <Mavrik> spectralsun, which also tells you the answer to your question :)
[01:07] <spectralsun> Mavrik: cool, thanks
[02:04] <luc4> Hello! I have a variable frame rate mp4 with h264. Can I convert that to fixed frame rate? I know this seems pointless, but it is for development purposes.
[03:44] <active8> refresh my memory, please. Was the copy codec ( e.g., -c:a copy) deprecated in favor of -q:a and -q:v ? Maybe I'm just mixed up and remembering that -qscale is deprecated and -q:x is the preferred option; or that -qscale is misunderstood (and I don't remember why
[03:45] <relaxed> -q is shorthand for -qscale
[03:46] <relaxed> -c:a copy is shorthand for -acodec copy
[03:46] <active8> yeah. i think the output of ffmpec to the command line just suggests using -q:
[03:47] <Dark-knight> what are you trying to do?
[03:48] <active8> oh I just stripped the audio from barney miller and trimed the guitar solo in one shot. just trying to remember things i learned from previous trials
[03:48] <active8> good learning. -vn vorbis to mp3 -ss and -t --- one shoit
[03:48] <active8> shot
[03:49] <active8> it was trimming off a second of applause where i used the copy codec which ...
[03:50] <active8> wait. is it needed when just trimming a bit of track from one file to another using the same format - like mp3 or does it decode and encode, thus losing quality.
[03:50] <active8> ?
[03:52] <relaxed> active8: you can copy the stream or use a lossless codec like flac
[03:53] <Dark-knight> does mp3 support flac?
[03:54] <active8> but would ffmpeg -i clip.mp3 -ss 00:01 -t 30 trimmed.mp3 cause it to decode/encode? That would force a lossy compression. So I ask if I should use the copy codec.
[03:56] <active8> seems to me ffmpeg is smart enough to pick the codec based on the output file extension and would use the codec if I don't -c:a copy
[03:57] <relaxed> active8: yes, try ffmpeg -i clip.mp3 -c copy -ss 00:01 -t 30 trimmed.mp3
[03:58] <active8> both work but can't hear the difference. just askin' 8)
[04:00] <relaxed> it's lossy unless you use -c copy
[04:03] <active8> ok. looks like the output text says "encoder : Lavf55.13.101" for both ways but it's different in other ways
[04:03] <active8> Stream mapping: Stream #0:0 -> #0:0 (mp3 -> libmp3lame)
[04:03] <active8> vs
[04:03] <active8> Stream mapping:  Stream #0:0 -> #0:0 (copy)
[04:04] <active8> great! I meant to ask about that and almost forgot. thanks for you help
[04:11] <Dark-knight> can ffmpeg do everything mp4box can?
[04:11] <Dark-knight> http://gpac.wp.mines-telecom.fr/mp4box/
[04:15] <Baked_Cake> well let me just quote the late, great, colonel sanders
[04:15] <Baked_Cake> he said: im too drunk, to taste this chicken
[04:15] <Baked_Cake> oops wrong chat again
[04:15] <Dark-knight> lol
[08:12] <Dark-knight> according to this from 2008, ps3 supports ALAC
[08:12] <Dark-knight> http://www.avforums.com/threads/flac-and-apple-lossless-alac-on-ps3.778932/
[08:13] <Dark-knight> is this still true?
[08:14] <Dark-knight> it seems there are differences in how the alac audio is made that affects the way its played
[08:20] <Dark-knight> so what is the difference between
[08:20] <Dark-knight> taking a .mkv with FLAC audio and converting it into .m4a with ALAC audio and then renaming the file ext. to .mp4
[08:20] <Dark-knight> and using -f ipod
[08:20] <Dark-knight> ffmpeg -i input -c copy -c:a alac -f ipod output.mp4
[08:20] <Dark-knight> ?
[08:21] <Dark-knight> renaming the extension vs. using -f ipod
[09:06] <Dark-knight> yes, im still here waiting
[09:06] <Dark-knight> i can read you mind
[09:19] <Dark-knight> seriously what is the difference?
[09:19] <Dark-knight> http://pastebin.com/FpBBccLk
[09:19] <Dark-knight> http://pastebin.com/27UVCBHM
[09:40] <relaxed> Dark-knight: what is your question?
[09:40] <Dark-knight> scroll up?
[09:41] <relaxed> the -f ipod question?
[09:41] <Dark-knight> yes
[09:42] <relaxed> -f ipod enables some options in mp4 that are apple centric
[09:42] <relaxed> We've been over this. Look at the source if you want to see the difference.
[09:43] <relaxed> And since apple created alac, it makes since that it works with -f ipod
[09:43] <relaxed> Do you have a real problem you need help with?
[09:43] <Dark-knight> i looked at both files using mediainfo and also used ffmpeg to get more info and i honestly can't see what the difference is
[09:44] <relaxed> Why do you give a shit?
[09:44] <Dark-knight> well i need make sure that i can play this file on the ps3 because some instances of ALAC dont work on ps3
[09:45] <relaxed> maybe the ps3 only supports 2 channel alac, and not 5.1
[09:52] <relaxed> I'm sure all these details are on the interweb
[09:52] <Dark-knight> i looked
[09:52] <Dark-knight> that why im asking here
[09:52] <Dark-knight> i've learned to google before asking dumb questions
[09:54] <Dark-knight> some people have converted flac to alac themselves but it wouldn't play on their ps3, but when they used itunes to convert the flac to alac, it worked
[09:54] <Dark-knight> what i want to know is, why. and if that is that same as rename vs -f ipod
[10:03] <relaxed> make small some samples and see what's what
[10:03] <relaxed> some small*
[10:06] <Dark-knight> i already poked around with those 2 i converted already, i dont know where else to look inside of them?
[10:07] <sfan5> you could convert two files
[10:07] <sfan5> one with -f ipod, one without
[10:07] <sfan5> and compare them in a hex editor
[10:07] <Dark-knight> -_- i all ready did
[10:07] <Dark-knight> oh hex editor
[10:08] <Dark-knight> and what would i look for?
[10:08] <sfan5> everything that differs
[10:08] <Dark-knight> ok
[10:08] <Dark-knight> got a link for hex editor?
[10:08] <sfan5> which OS?
[10:08] <Dark-knight> windows
[10:09] <sfan5> nope
[10:09] <sfan5> google is your fried
[10:09] <sfan5> friend*
[10:14] <Dark-knight> which one is best?
[10:14] <Dark-knight> http://en.wikipedia.org/wiki/Comparison_of_hex_editors
[11:28] <luc4> Hello! I have an h264 variable frame rate stream in an mp4 file. Can I convert it to a fixed frame rate?
[11:29] <sacarasc> You can, but it might look jerky.
[11:30] <sacarasc> -r sets the frame rate and will make it fixed.
[11:32] <luc4> sacarasc: tried that& but it doesnt seem to work&
[11:33] <relaxed> try -vsync 1
[11:33] <luc4> I mean that the fps is written in the container probably, ffprobe sees it. But if I set something like 20 or 40 I see no difference.
[11:34] <luc4> relaxed: you mean in addition? Or should I replace it?
[11:39] <relaxed> use it alonf with -r,  man ffmpeg| less +/'  -vsync'
[11:39] <relaxed> along*
[11:40] <relaxed> you are trying to re-encode the stream, correct?
[11:46] <luc4> relaxed: yes, I suppose that is necessary to change the frame rate.
[11:50] <luc4> relaxed: I tried 40fps but I see no difference&
[11:50] <relaxed> do you want to see a difference?
[11:50] <luc4> relaxed: yes, maybe ffmpeg replicates the frames?
[11:50] <relaxed> look at ffmpeg's console output for dup frames
[11:51] <luc4> dup=3493
[11:51] <luc4> so I suppose yes
[11:52] <luc4> Any way to avoid this? I would like to see the frames one after the other with fixed frequency.
[11:52] <luc4> I know it makes no sense usually, but it is for development purposes.
[11:55] <relaxed> I'm not sure how to do that. You could output all the frames to png(s) anf then use them as input using an arbitrary freame rate.
[11:57] <luc4> relaxed: ah ok, Ill try that, thanks.
[11:58] <relaxed> luc4: wait, I believe you can force decoding at a specific frame rate.
[11:59] <relaxed> ffmpeg -r $whatver -i input
[12:00] <relaxed> from the fine manual, "As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps."
[12:02] <relaxed> luc4: try that with stream copying
[12:03] <luc4> relaxed: Ill try this! Thanks!
[12:27] <Dark-knight> not a SINGLE hex editor for windows 8
[12:27] <Dark-knight> F***
[12:28] <Dark-knight> yeah im done looking for tonight
[12:43] <viperfx_> Hi all. I have posted a question on SO regarding compilation flags. If someone here could take a look and offer some help, that would be great. https://stackoverflow.com/questions/25217628/detecting-the-needed-decoder-muxer-demuxer-etc-for-a-file
[12:48] <relaxed> viperfx_: -enable-demuxer=matroska --enable-decoder=vorbis
[12:49] <viperfx_> Would you be able to help me understand what the purpose of demuxer/decode etc is. Or point me to a link?
[12:49] <viperfx_> Thanks, I will try those flags.
[13:03] <viperfx_> It works! Thanks relaxed
[13:05] <relaxed> viperfx_: you're welcome
[13:25] <viperfx_> relaxed: What about this one? http://pastebin.com/RrYn6We7
[14:11] <viperfx_> Is there any way to improve the speed of the av_open_input_file() call for an HTTP stream
[14:12] <viperfx_> Does FFmpeg need to read all of it before it can proceed?
[14:22] <viperfx_> *avformat_open_input()
[14:55] <anshul_mahe> How to set thread_count in avcodec_decode_video2 for libavcodec?
[14:55] <anshul_mahe> I am using libav library
[15:07] <anshul_mahe> I want to debug some part of code, if i put scheduler locking on in gdb then all threads hang to pthread_cond_wait, looks like deadlock cant exactly say since not verified all condition
[15:09] <anshul_mahe> so I am searching for some av_dict_set to change thread_count
[15:10] <anshul_mahe> !log
[15:10] <anshul_mahe> log!
[15:17] <anshul_mahe> where is ffmpeg-irc publicly logged?
[15:18] <Chaz6> /dev/null
[15:19] <anshul_mahe> ok, Chaz6
[15:20] <anshul_mahe> ohh i was missing 's' in !logs
[15:21] <anshul_mahe> thanks, sacarasc
[16:28] <phelps> how do arguments work in the expansion for a filter
[16:28] <phelps> %{pts}, how do I add hms as an argument
[16:28] <phelps> "The text between the braces is a function name, possibly followed by arguments separated by :."
[16:29] <phelps> %{pts} %{pts:hms} %{pts hms} %{pts}:hms etc.. none work
[16:58] <fajung> I'm trying to convert an avi/mp3 to mp4/aac, but I get this error:"Unknown encoder 'libfaac'". I'm on ubuntu 14.04,  ffmpeg version 2.3
[17:00] <Mavrik> fajung, your ffmpeg isn't built with libfaac support.
[17:00] <fajung> how can I do to make it support it ?
[17:03] <sacarasc> fajung: You'd have to compile yourself, but if you do that, you might as well use libfdk_aac instead, because it's a lot better than libfaac. (Just a moment, getting a link.)
[17:03] <sacarasc> https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu to compile.
[17:03] <sacarasc> https://trac.ffmpeg.org/wiki/Encode/AAC about AAC encoders.
[17:03] <Mavrik> fajung, you either use another AAC encoder (since libfaac is utter shit)
[17:04] <Mavrik> fajung, or you recompile your ffmpeg with libfaac support
[17:04] <Mavrik> if you're using ubuntu and you typed "apt-get install ffmpeg" you don't have ffmpeg installed anywa
[17:04] <Mavrik> y
[17:04] <fajung> ok, i''m going to try to compile it
[17:05] <Mavrik> fajung, take fdk-aac instead of libfaac then.
[17:05] <sacarasc> Yes, libfdk_aac is much better.
[18:49] <fajung> now i'm getting this error: Unknown encoder 'libfdk-aac'
[18:52] <fajung> ~/bin$ ./ffmpeg -i ./input.avi -acodec libfdk-aac -b:a 128k -vcodec mpeg4 -b:v 978k -flags +aic+mv4 out.mp4
[19:04] <nicholaswyoung> fajung: it's libfdk_aac
[19:05] <nicholaswyoung> assuming you've installed libfdk, anyway
[19:05] <fajung> I'm doing all the guide again to see where I did wrong
[19:08] <nicholaswyoung> fajung: what OS are you using?
[19:09] <fajung> ubuntu 14.04
[19:09] <fajung> I'm doing this: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu
[19:09] <fajung> trying...
[19:10] <fajung> in the ffmpeg step, I have to add the --enable-libfdk_aac &  --enable-libfdk-aac right?
[19:11] <nicholaswyoung> looking at that page now, to see if it's current
[19:12] <c_14> Just one of the two iirc.
[19:12] <c_14> Just --enable-libfdk-aac
[19:12] <nicholaswyoung> it looks like --enable-libfdk_aac is the correct formation: https://trac.ffmpeg.org/wiki/Encode/AAC
[19:13] <c_14> Not according to ./configure --help
[19:13] <nicholaswyoung> you might have to --enable-nonfree too
[19:13] <fajung>   --enable-libfdk-aac --enable-libfdk_aac  --enable-nonfree
[19:13] <fajung> just in case
[19:14] <fajung> and gpl
[19:16] <fajung> I got this: ERROR: libfdk_aac not found
[19:19] <c_14> Are you sure you're running the ffmpeg that you just compiled and that it's loading the correct libraries?
[19:20] <fajung> I deleted I've to recompile it again
[20:13] <fajung> now it is working, I dont't know how or why, but it is working with lib_aac
[20:13] <fajung> libfdk_aac
[00:00] --- Sun Aug 10 2014