[Ffmpeg-devel-irc] ffmpeg.log.20140808

[burek]

[00:00] <phelps> jrgill -c:a copy
[00:01] <phelps> should tell it to copy the audio codec and not transcode
[00:04] <jrgill> phelps, it seems to conflict, which was why I tried -c:v.  "Filtergraph 'pan=stereo:c0=c0' was defined for audio output stream 0:1 but codec copy was selected."
[00:05] <phelps> jrgill: hmm, dunno then, I've never used filters
[00:06] <jrgill> Ultimately trying to invert that right channel.  I think the only way is to multiply by -1 in pan audio.
[00:07] <jrgill> Will just try some other ways.  Was hoping it could be simple.  :/
[00:19] <llogan> jrgill: filters require encoding
[00:40] <jrgill> llogan, same with map_channel?  With -c copy it just ignores my mapping while -c:v copy transcodes audio.  ffmpeg -i in.m4v -map_channel 0.1.0 -map_channel -1 -c copy out.m4v
[01:02] <jrgill> llogan, http://privatepaste.com/download/590d7e8605
[01:08] <jrgill> Guess I could map the channels in QuickTime and use ffmpeg to switch from mov to m4v.
[01:40] <jrgill> Thinking maybe there's a way to use channel_layout to do the equivalent of QuickTime assigning the channels.  Any suggestions?
[03:01] <Dark-knight> Zeranoe: hey man
[03:01] <Dark-knight> Got some question to ask you?
[03:02] <Zeranoe> Whats up?
[03:02] <Dark-knight> just a sec
[03:12] <Dark-knight> I copy'd these from my notes.
[03:12] <Dark-knight> I was wondering if you could add the ALAC tag to the .mp4 container and add support for setting the default stream/track for a container?
[03:13] <Dark-knight> ...to the next build
[03:17] <Zeranoe> I try to avoid making any modifications to the source code for FFmpeg, so if it's something that requires patching the source it would be best to submit a feature request ticket
[03:20] <Dark-knight> those two things i suggested would be very beneficial to people that need a solution to certain problems.
[03:21] <Dark-knight> you might have more pull in the community and your request tickets might be taken more seriously.
[03:25] <Zeranoe> Are there already tickets out for those issues?
[03:25] <Dark-knight> as it stands right now. support for setting the default stream is very high on my want list, and Im sure that it would benefit many people who needed the feature, (as i have seen on google)
[03:26] <Dark-knight> if i had to choose between the 2, i would choose the feature to change the default stream
[03:30] <Zeranoe> Dark-knight: Where does that apply?
[03:32] <relaxed> Dark-knight: did you try -atag ALAC
[03:34] <Dark-knight> what do you mean?
[03:35] <relaxed> what did you mean by "add the ALAC tag to the .mp4 container"?
[03:37] <Dark-knight> FLAC audio doesn't like the .mp4 container. so I converted the .mkv container to .m4a and the FLAC audio to ALAC. Then I changed the name from .m4a to .mp4
[03:37] <Dark-knight> What im suggesting is, adding the alac tag to .mp4, so making all those steps pointless.
[03:38] <relaxed> .m4a and .mp4 are the same thing
[03:38] <Zeranoe> (except one is audio only)
[03:44] <Dark-knight> it would be a simple thing to change
[03:45] <Dark-knight> but im more concerned about the default stream.
[03:46] <relaxed> what kind of stream?
[03:51] <relaxed> you can use the .mov container with alac
[03:51] <Dark-knight> changing the default audio or subtitle stream
[03:52] <relaxed> The default audio stream probablu defaults to the first stream
[03:52] <relaxed> probably*
[03:53] <Dark-knight> it doesn't always
[03:53] <Dark-knight> which is why manual change is necessary
[03:56] <relaxed> do you know of another tool that can do these things? Like mp4box, l-smash, or something?
[04:02] <relaxed> Dark-knight: ffmpeg -i input -c:a alac -f ipod output.mp4
[04:04] <Dark-knight> i heard rumors that mp4box could change the default stream, but i found evidence to the contrary.
[04:05] <Dark-knight> I believe the flag that tells the container which stream is the default is here
[04:05] <Dark-knight> http://matroska.org/technical/specs/index.html Down by "Track" it's called "FlagDefault".
[04:05] <relaxed> matroska is not mp4
[04:05] <Dark-knight> i know this
[04:05] <relaxed> maybe it depends on the player
[04:05] <Dark-knight> is should be the same
[04:06] <relaxed> anyway, the command above sovles one your problems
[04:06] <Dark-knight> interesting
[04:06] <Dark-knight> does it have any loss in quaility? setting -f ipod?
[04:07] <relaxed> it controls the container
[04:07] <Dark-knight> or is that the same as changing the name from .m4a to .mp4
[04:13] <relaxed> it probably sets some apple centric things, since they insist on doing things their own way.
[04:13] <Dark-knight> so is it the same a a name change?
[04:13] <relaxed> No
[04:14] <relaxed> but it should still work on whatever console you're targeting
[04:14] <Dark-knight> how so?
[04:14] <Dark-knight> thanks
[04:15] <Dark-knight> now that, that is out of the way. how about we work on that default flag?
[04:16] <relaxed> go read the mp4 spec, find out what has to be done or if it's even possible, and then file a bug report.
[04:16] <relaxed> supply a sample of an mp4 with these features too
[04:17] <relaxed> first make sure there's not already a feature request for it.
[04:18] <Dark-knight> where do i check?
[04:18] <relaxed> google
[04:18] <relaxed> and the ffmpeg bug tracker
[05:26] <jjohn> Ahoy, guys. Can you help me out with two little questions that I got for you?
[05:26] <jjohn> I wrote it down here: http://pastebin.com/9PMsZ908
[05:29] <relaxed> jjohn: flac is lossless and it well supported, but the output will be larger than the original audio stream.
[05:30] <relaxed> it is*
[05:30] <jjohn> relaxed: It's not only flac, it's actually a wild mix of everything. And I basically just want to copy the audio stream.
[05:32] <relaxed> right, but wouldn't it be nice to have it all the same format for consistency?
[05:33] <jjohn> Actually, preserving the original audio quality is even more important than consistancy. :)
[05:33] <jjohn> consistency*
[05:33] <relaxed> you can achieve both by encoding everything to flac
[05:33] <jjohn> Yeah, well, I once tried that out as well.
[05:34] <jjohn> Turned a 200 MB video into a 800 MB audio file. m4a was about 123 MB.
[05:34] <jjohn> So flac is even worse.
[05:35] <jjohn> You already said that before, but I wanted to make the point that flac is really bad, and I have no real gains from it.
[05:38] <Zeranoe> jjohn: flac will provide the best "original audio quality". The file size will be the sacrifice
[05:39] <Dark-knight> is there more then one audio stream?
[05:40] <jjohn> Zeranoe flac is cool if the quality is to be preserved. But I just want to copy the audio stream, no more, no less.
[05:40] <Dark-knight> just "ffmpeg -i input -c:a copy output" should suffice
[05:40] <jjohn> Dark-knight no there isn't.
[05:40] <Zeranoe> ^
[05:41] <jjohn> Dark-knight that wasn't my question. ;)
[05:41] <Dark-knight> but it is your answer
[05:41] <jjohn> My question was: do you guys know any other good pure audio container formats?
[05:42] <Dark-knight> let my quote my notes
[05:42] <Dark-knight> FLAC and ALAC are lossless audio formats. (the files is compressed and no data is lost.)
[05:42] <Dark-knight> WAV and AIFF are uncompressed lossless audio formats. (the files are uncompressed and take up lots of space. not recommended)
[05:42] <Dark-knight> APE is a highly compressed lossless audio format (the files are more compressed and no data is lost. It's not very compatible.)
[05:42] <Zeranoe> jjohn: What are you trying to do?
[05:42] <Zeranoe> jjohn: and why
[05:43] <jjohn> As I stated on pastebin: I have video files with audio, and the videos suck, so I want to GET the AUDIO stream and put it into a new file, no encoding, no decoding, into an appropiate audio container format.
[05:44] <Zeranoe> jjohn: what is the format of the audio stream in the original?
[05:44] <Dark-knight> again
[05:44] <Dark-knight> just "ffmpeg -i input -c:a copy output" should suffice
[05:44] <jjohn> Getting, for example, the entire AAC stream, and put it into m4a. Worked fine, but I wanna know if I can improve that.
[05:44] <Dark-knight> improve how?
[05:44] <Zeranoe> jjohn: why? is it not working?
[05:45] <Zeranoe> if your input is AAC, your containers are m4a and aac
[05:45] <Dark-knight> Zeranoe: i didn't think aac was a container, i thought it was a format?
[05:46] <jjohn> Zeranoe: I am fairly new with all that encoding stuff, and I was actually sure my naive solution would not provide the best results AND saving all encoding and decoding stuff.
[05:47] <jjohn> But if you tell me that if I got a raw AAC stream I should put it into a m4a container, as I did from the very beginning, I am glad. Because then someone who knows more about it told me I did OK.
[05:47] <Zeranoe> Dark-knight: doesn't look like FFmpeg supports it anyway, muxing that is. It can demux
[05:48] <Dark-knight> ah
[05:48] <Zeranoe> jjohn: ffmpeg -i input -c:a copy -vn output.m4a will dump the exact AAC stream to a m4a file.
[05:48] <jjohn> Zeronoe that's what I did. I have been telling you this three times now. I just wanted to know if there is a better container for AAC.
[05:49] <jjohn> Or for pure audio in general.
[05:49] <Dark-knight> i really dont see how thats different from mine Zeranoe. :/
[05:49] <jjohn> Because, as I stated, I am absolutely not sure about all the features ffmpeg provides, and I was sure I did something stupid.
[05:50] <Zeranoe> Dark-knight: it's not
[05:50] <jjohn> Like, using a container which is outdated or not appropiate.
[05:50] <Dark-knight> jjohn: !google containers compatible with aac audio
[05:51] <Zeranoe> jjohn: there isn't a audio only container for every type of audio stream if that's what you mean. m4a is the container for a AAC stream, and is the best container for your purpose.
[05:52] <jjohn> Zeranoe: yep, that's what I meant. And since you told me it's the best for my purpose, I will accept that.
[05:52] <jjohn> And what is about the second question?
[05:54] <Dark-knight> probably not
[05:55] <jjohn> probably?
[05:55] <Dark-knight> 99.9%
[05:56] <Dark-knight> how about you just do it and listen to it for yourself
[05:56] <Dark-knight> should answer your own question
[05:56] <jjohn> So I can ignore the missing flag? Dark-knight I did that but it's 5:56 AM here and I am a bit tired, I don't trust my hearing right know. And again I wanted to be sure. :)
[05:57] <Dark-knight> paste an output of the error that you a receiving
[05:57] <jjohn> It sounded good, but I could just imagine any affect subconsciously.
[05:59] <Dark-knight> are*
[05:59] <Dark-knight> sometimes i forget to type out entire words and just type the first letter by accident
[05:59] <jjohn> ffloger I did so half an hour ago. Not the exact output, but what I used to extract the audio stream (first question) and what it gave me as flag on the audio output: http://pastebin.com/9PMsZ908
[06:00] <Zeranoe> jjohn: Time for bed, unless you woke up at that time, in which case, time for bed
[06:00] <jjohn> It's not an error Dark-knight. As I said it's just a *missing* flag.
[06:01] <Dark-knight> Zeranoe: lol
[06:02] <jjohn> And I wondered if that's OK. Because, as I stated like three times *again* that I am not experienced with ffmpeg and codecs.
[06:02] <Dark-knight> jjohn: can you post an output of the missing flag?
[06:06] <jjohn> Dark-knight: sure: http://pastebin.com/KRui4wNV
[06:07] <Dark-knight> looks fine
[06:07] <Dark-knight> just do it and listen
[06:08] <jjohn> OK, if you say so ... that missing flag has no meaning?
[06:08] <Dark-knight> i've never seen it when i converted/remuxed shit
[06:08] <Dark-knight> so nope
[06:09] <jjohn> OK ... then I'll do it. Thanks for the help, and sorry if I was hard to understand.
[06:09] <jjohn> Didn't mean to.
[06:09] <Dark-knight> nah you were fine
[06:10] <jjohn> Thanks again. And good night/morning/whatever. Bye
[06:30] <Dark-knight> just a quick question
[06:32] <Dark-knight> do i need to specify -map 0:v if there is only one video track or will -c:v copy suffice?
[06:33] <relaxed> the latter
[06:34] <Dark-knight> this is what i had
[06:34] <Dark-knight> ffmpeg -i input -map 0:v -map 0:1 -map 0:2 -map 0:4 -map 0:6 -c:v copy -c:a alac -c:s mov_text -f ipod output.mp4
[06:34] <Dark-knight> will this work just the same
[06:34] <Dark-knight> ffmpeg -i input -map 0:1 -map 0:2 -map 0:4 -map 0:6 -c:v copy -c:a alac -c:s mov_text -f ipod output.mp4
[06:37] <relaxed> do you only want one video stream in all your output?
[06:38] <Dark-knight> yes
[06:39] <relaxed> and is it always the first video stream?
[06:39] <Dark-knight> video track is stream 0:0
[06:40] <Dark-knight> if i specify -map once, do i have to include everything i want?
[06:41] <Dark-knight> using more -map
[06:42] <relaxed> what do you want?
[06:42] <relaxed> only one video stream and all audio and subtitle streams?
[06:44] <Dark-knight> http://pastebin.com/Lr3QJ9Lg
[06:44] <Dark-knight> contents of the file
[06:47] <relaxed> Both of those commands should work. Is there a problem?
[06:49] <Dark-knight> just trying to make it as simple as possible
[06:49] <Dark-knight> just wondering for future reference if leaving -map 0:v out of that command line will still copy over the video stream
[06:51] <relaxed> if there's more than one video stream it will copy them all
[06:51] <Dark-knight> i know that
[06:52] <relaxed> use -map 0:v:0 if you always want one, and you assume the first video stream is what you're after
[06:54] <Dark-knight> im wondering. because i already specified a lot of other streams using -map and was wondering if not including -map 0:v will still allow the video stream to be copyed
[06:56] <relaxed> oh yeah, you do have to specify all the streams if you specify any :)
[06:56] <Dark-knight> ok thanks
[06:56] <Dark-knight> ill ad that to my nots
[06:56] <Dark-knight> add*
[06:56] <Dark-knight> notes*
[07:03] <Dark-knight> wait... does that mean if i specify any of these -map 0:v or -map 0:a or -map 0:s. that i have to all the other streams
[07:05] <relaxed> you -map what you want in the output
[07:14] <Dark-knight> not always
[07:14] <Dark-knight> sometime -map is un-nessasry
[07:15] <relaxed> example?
[07:21] <Dark-knight> ffmpeg -i input -c copy output
[07:21] <Dark-knight> when there is only 1 of each stream
[07:21] <relaxed> yes, there are sane defaults
[07:22] <Dark-knight> so like i said
[07:22] <Dark-knight> this file has 1 video, 3 audio, 3 sub. Do i have to specify the 1 video file or can i just specify everything else
[07:23] <relaxed> as I said earlier, once you map one stream you have to map them all
[07:23] <Dark-knight> ok just making sure. i gotta cover all my bases
[07:24] <relaxed> or else how would ffmpeg know what you wanted?
[07:27] <Dark-knight> so the second command wouldnt do what i wanted
[07:28] <relaxed> oh, right
[07:28] <relaxed> too many numbers
[07:28] <Dark-knight> not what i meant
[07:29] <Dark-knight> by removing -map 0:v from the first command line, then the video stream wont be copyed
[07:29] <Dark-knight> right?
[07:29] <relaxed> correct
[07:30] <Dark-knight> even thought i have -c:v copy?
[07:32] <relaxed> try it
[07:38] <Dark-knight> im busy, just tell me
[07:40] <Dark-knight> nvm
[07:40] <Dark-knight> im just try it
[07:46] <Dark-knight> ahh shit
[07:47] <Dark-knight> your right
[07:48] <spectralsun> where did qt-faststart go?
[07:48] <spectralsun> it is not in tools anymore :(
[07:49] <relaxed> spectralsun: use ffmpeg -i input -movflags faststart ... output.mp4
[07:52] <Aero> Hai!
[07:52] <Aero> Is this some scamware?
[08:03] <Dark-knight> ok relaxed i have a file with 3 audio, 2 of them are flac and 1 is aac. i want to change the two flac into alac and keep the aac the same
[08:03] <Dark-knight> whats the command line?
[08:06] <relaxed> pastebin the ffmpeg -i input
[08:07] <Dark-knight> http://pastebin.com/Lr3QJ9Lg
[08:07] <relaxed> video too?
[08:08] <Dark-knight> just -c:v copy for the video
[08:10] <relaxed> ffmpeg -i input -map 0:v -map 0:a -map 0:s -c:v copy -c:a:0 alac -c:a:1 alac -c:a:2 copy -c:s mov_text -movflags faststart output.mp4
[08:10] <Dark-knight> Ahh thank you so much
[08:11] <Dark-knight> didnt know you could do that
[08:11] <Dark-knight> ill add that to my notes
[08:11] <relaxed> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1
[08:11] <Dark-knight> Aero: What are you talking about
[08:11] <relaxed> http://ffmpeg.org/ffmpeg.html#Advanced-options
[08:17] <Dark-knight> what does ffmpeg -movflags faststart do?
[08:18] <relaxed> moves the index to the beginning of the file.
[08:37] <Dark-knight> index?
[08:39] <relaxed> google moov atom
[08:39] <ParkerR> Dark-knight, AVI files larger than 4GB usually have to have an index built in order to skip through properly
[08:39] <ParkerR> Is my guess. may be talking out of my ass though :/
[09:08] <[A3G1S]> hey guys, I am trying to hardcode srt subtitles using ffmpeg, but it isn't working (I've build a static ffmpeg build and trying to use it for hardcoding subs)
[09:08] <[A3G1S]> any 1 can help ?
[09:42] <jrgill> Do the aac encoders each give some special treatment to the bitrate for mono streams?  Seems native aac sees >=289k as invalid and libvo_aacenc just forces 64k.
[10:20] <termos> how is ffmpeg handling multithreaded programs? I'm doing av_write_frame calls from different threads but to different rtmp streams and I'm getting some weird bugs. Could it be related?
[11:59] <joe90k> Can anyone help me remove a compiled version of ffmpeg or totally remove it from my CentOS box? I have the original compile instructions and have tried removing the dir that was created, but when I run 'ffmpeg' I get '-bash: /usr/local/bin/ffmpeg: No such file or directory '
[12:07] <joe90k> Sorted with reboot..
[15:04] <zlice> what am i looking for? when ffmpeg is given an input file - it normally shows config and then says 'input #0 type, from 'file.type' '
[15:04] <zlice> i do not get there, just stops at the config
[15:07] <zlice> hm...let me see what i can do
[15:08] <zlice> f it, i'll have to type it, h/o
[15:17] <zlice> http://pastebin.com/e6K7z0X6 phew
[15:18] <zlice> i've tried other file formats, hang different places on the loglevel but all just hang after build config. so idk what i'm looking for really
[15:23] <c_14> zlice: What exactly is the problem? Does the command freeze? Or do you just want the codec/format information?
[15:26] <zlice> c_14 : pretty sure it freezes
[15:27] <c_14> Can you try attaching an strace to the process or checking in something like htop what state the command is? Is it still taking cpu cycles?
[15:27] <zlice> i'd have to cross-compile that...probably couldn't til this afternoon/night
[15:28] <zlice> hm, opening up another ssh looks like ffmpeg is running at 100% - could it but stuck? trying to calc something? i didn't look at the code, where does it say 'input file #0' ?
[15:29] <c_14> It should be the next line after the '[mp3' lines
[15:30] <zlice> (running on qnx bb q10 btw)
[15:30] <c_14> Can you play the mp3?
[15:30] <c_14> Heck, can you cat it?
[15:30] <zlice> i'll try that
[15:30] <zlice> yyyep
[15:30] <c_14> ffprobe mymp3.mp3 ?
[15:31] <c_14> Oh, you disabled ffprobe...
[15:31] <zlice> ya
[15:32] <zlice> i guess, is there an easy way to tell if it's actually doing something? i thought verbose would help, maybe i'm not thinking of something simple
[15:34] <zlice> it could just be qnx, idk
[15:34] <c_14> You can try pressing h while ffmpeg is running.
[15:35] <c_14> That gives out more debug info.
[15:35] <c_14> Well, it starts dumping packets/hex which I guess counts as more debug info.
[15:36] <zlice> that's what i'm looking for :) but i don't see anything yet :(
[15:36] <c_14> What does `ps -o stat `pidof ffmpeg`' return?
[15:37] <c_14> If you had something like gdb, or strace you could try that as well.
[15:40] <zlice> pidin(ps) shows PID# ffmpeg 10r(the state) RUNNING
[15:40] <zlice> i know qnx has gdb, haven't used it yet really...
[15:41] <c_14> So, it's not stuck in IO-wait...
[15:43] <c_14> I'm guessing it might be an issue with the configuration/architecture.
[15:47] <zlice> hm ok, i will see if i can't get ntoarm-gdb working and try to look at it further, thans c_14
[15:48] <Neppy> ohi~ trying to loop a video+audio with ffmpeg... not haivng much luck though, ffmpeg -i "input.mkv" -c:v copy -c:a copy  -loop 20 "output.mkv" but the output i sjust 3 seconds, same as input.. or can i not use "copy" with that?
[15:49] <c_14> The loop option doesn't really do what you want it to.
[15:50] <c_14> Your best bet is to use the concat demuxer and just concat the video 20 times.
[15:51] <Neppy> ah :/
[15:59] <Neppy> hmm that works but messes up the audio xD
[16:01] <Neppy> though when cutting out the file i want to loop out of a bigger file i also had some weird audio issues? seems audio is in chunks  too and it cant cut "subchunks" or so..
[16:01] <Neppy> since i had to increase ms by .050 or it would cut out the audio early
[16:02] <c_14> yep, if you use -codec copy you can only cut in "chunks"
[16:03] <Neppy> hmm what if i use -c:v copy and -c:a libvorbis?
[16:11] <c_14> It should be able to cut precisely.
[16:12] <Neppy> hmm doesnt seem to :<
[16:17] <bunniefoofoo> in mpeg2video, does a b frame have to be between two p-frames for example for a 4-frame closed gop, IPBB is invalid but IPBP is OK ?
[16:18] <bunniefoofoo> the reason, I'm asking is because ffmpeg is generating files with a B frame at the end when using strict gop mode
[16:30] <Chaz6> Hey there, just wanted to pop by and say, I got a Logitech C930e as it's supposed to be a step up from the C920, but it does not support h264 encoding with directshow :/
[16:30] <Chaz6> So I'll stick with the C920
[17:00] <DdoubleU> Hey, is there any way to encode a video file and have the output name the same as input , so just replacing the original?
[17:01] <Chaz6> DdoubleU: i don't know but perhaps looking to COW (copy on write), e.g. btrfs/vss?
[17:01] <bunniefoofoo> should work on linux if you pipe the file into ffmpeg, on windows probably not since typically you can't overwrite an open file
[17:02] <DdoubleU> ok thanks
[17:03] <DdoubleU> Another thing, Is there a way to encode multiple inputs all to their own outputs in one command?
[17:06] <Bombo> DdoubleU: why not rename it before? source.mkv source_old.mkv then encode source_old.mkv to source.mkv, del old...
[17:06] <DdoubleU> ^ thats what im going to end up doing
[17:07] <Bombo> DdoubleU: you mean multiple inputs, as in 'encode *.mkv'?
[17:08] <bunniefoofoo> doubleU: you don't save any drive space either way... if the file is open for reading and you "overwrite" it as on linux, the file is not free'd up until you are done anyways
[17:09] <DdoubleU> yes, I have the paths of the inputs
[17:09] <DdoubleU> and i want each to encode seperately into its own output
[17:10] <Bombo> DdoubleU: just script it in bash or batch or whatever ;)
[17:10] <DdoubleU> ok
[17:26] <Bombo> how do i extract subtitles with ffmpeg? this is what i'm trying: http://bombo.jpe.gs/sub/ff.txt what codec parameters are missing? and why is the operation not permitted?
[17:26] <ubitux> there is no vobsub muxer in ffmpeg yet
[17:27] <ubitux> see http://trac.ffmpeg.org/ticket/2391
[17:27] <ubitux> Bombo ^
[17:28] <ubitux> dvd subtitles can only be extracted in a vobsub (sub+idx)
[17:28] <ubitux> .sub is the extension for MicroDVD subtitles, which are text (while dvd sub are bitmaps)
[17:30] <Bombo> ubitux: ah ok so it isn't possible right now...
[17:30] <ubitux> feel free to +1 the ticket, there is a vote system
[17:30] <Bombo> i just thought if ff is able to copy it it should be able to save it somehow
[17:31] <ubitux> it also needs to index it in an idx file
[17:31] <ubitux> it's not trivial, DVD are a real mess
[17:31] <ubitux> and vobsub is a nice pile of crap as well
[17:32] <ubitux> writing a demuxer for vobsub was a few orders of magnitude harder than what i expected
[17:32] <ubitux> so i'm not really motivated to write a muxer right now ;)
[17:33] <iive> i think you want to say that writing a demuxer for vobsub in theory is easy, but writing one that works correctly with all the available vobsub is nightmare :)
[17:36] <ubitux> well vobsub is an home made split file format based on one format that was unknown
[17:36] <ubitux> there is duplicated information, sometimes inconsistent, the segmentation is completely chaotic, and as no one understands any of the two, there are indeed all kind of broken files all around
[17:37] <ubitux> and i don't feel like i understand it enough to take the responsibility to write a muxer that will output more broken files
[17:38] <ubitux> but, i would love to have one ;)
[17:39] <Bombo> +1'ed ;)
[17:40] <Bombo> ok thx for the explanation
[17:40] <JEEB> just find a bastard who is ready to take the responsibility and then not do anything about it
[17:40] <JEEB> :3
[17:40] <ubitux> Nicolas seemed to be motivated
[17:40] <ubitux> but i think seeing me derping with the demuxer warned him about the curse
[17:55] <Bombo> hm doesn't work with mkvextract either
[17:55] <Bombo> probably this https://trac.ffmpeg.org/ticket/2035
[17:56] <Bombo> i think i'll give up now
[18:58] <LtHummus> I'm trying to encode a file for DVD, but DVD Studio Pro only allows for elementary mpeg streams...is there a way i can output both an m2v and an ac3 elementary file instead of doing run of ffmpeg to encode and an additional one to demux? now i'm running ffmpeg -i source.whatever -target ntsc-dvd <more random flags> output.mpg
[19:04] <amkrankruleuen> Hello i have problem with ffmpeg, someone alive?
[19:23] <azk> amkrankruleuen: Hi, your best bet is to state your problem and stick around
[19:25] <amkrankruleuen> azk: Ok, I have problem with android player, i download mp4 file to my linux on laptop and convert mp4 file to mp3 file using ffmpeg, when i download mp3 file to my phone from laptop i jave problem with moving file
[19:25] <amkrankruleuen> i have*
[19:25] <amkrankruleuen> all android version, on laptop is okay, on old symbian is okay but on android is not okay
[19:26] <azk> So, you're having problems moving the file within the android filesystem?
[19:27] <amkrankruleuen> Nooo, i have problem with moving audio in player
[19:27] <azk> Oh, seeking. I see.
[19:27] <amkrankruleuen> for example i move from 0:01 to 1:00 but player moving to random time ex: 0:10
[19:28] <dequid> Hello all, i'm having trouble encoding mov's with prores and pcm audio.
[19:28] <amkrankruleuen> azk: My coomand http://pastebin.com/3EjSerGP
[19:29] <azk> amkrankruleuen: Let me see if I can reproduce it
[19:29] <dequid> ffmpeg/ffprobe recognizes the audio as 'pcm_s16be', but when i examine the file in quicktime player inspector, it says for the audio channel: 32 bit, signed, big endian
[19:30] <dequid> the resulting audio (even when just playing with ffplay) sounds weirdly slowed down, like slow-mo, with click sounds
[19:31] <amkrankruleuen> azk: Ok
[19:31] <c_14> dequid: Have you tried forcing the codec to pcm_s32be?
[19:31] <dequid> Here is my output of ffprobe: http://pastebin.com/QQ1Ffbrh
[19:34] <dequid> c_14: Yes, here's my command: http://pastebin.com/45YrcPER
[19:35] <c_14> try your command with ffmpeg -c:a pcm_s32be -i [file] [options]
[19:35] <dequid> c_14: ok
[19:36] <c_14> That'll force the codec detection in case the audio really is pcm_s32be.
[19:37] <azk> amkrankruleuen: I can't reproduce it.
[19:38] <dequid> c_14: http://pastebin.com/Td1bp2KC
[19:39] <dequid> c_14: No change, audio still not good. Only change is resulting file is half size.
[19:39] <dequid> c_14: resulting file still recognized as s16be by ffprobe
[19:41] <c_14> Without the first -i input.mov, but not sure that'll change much in this specific use-case. How did you create the video/audio track?
[19:45] <dequid> c_14: Ops, my fault. Now we're getting closer. With: ffmpeg -c:a pcm_s32be -i input.mov -vn output.aiff the 'slo-mo' effect is gone, audio seems in the right pitch, but it plays very choppy, with click sounds
[19:46] <dequid> c_14: The files come straight from the Convergent Design Odyssey 7Q 422 recorder
[19:48] <c_14> Try adding -c:a pcm_s32be as an output option as well. ie: ffmpeg -c:a pcm_s32be -i input.mov -vn -c:a pcm_s32be output.aiff
[19:49] <c_14> The clicks _could_ be originating from the bit depth change from the 32bit source to the 16bit output. Not sure though.
[19:49] <c_14> (on my system the aiff muxer picks s16be by default)
[19:50] <dequid> c_14: Same thing with ffmpeg -c:a pcm_s32be -i input.mov -vn -c:a pcm_s32be output5.aiff just that filesize doubles again
[19:51] <c_14> The filesize is due to the bit depth. Do you have a player that can play the source without the clicking and without the slow-mo?
[19:52] <dequid> c_14: Strange thing is, that Quicktime can play it without problem (original file) and Adobe media encoder also
[19:53] <dequid> c:14: When i try to play it in VLC, picture works but i get a message that a decoder module for "   " is not found (empty string)
[19:54] <c_14> I'm guessing that's because ffmpeg can't correctly detect the audio format.
[19:55] <c_14> Does this happen with every file from that recorder or just that one?
[19:55] <dequid> c_14: With every one unfortunately.
[19:55] <c_14> Might be a bug.
[19:55] <c_14> I'd probably report it on the trac.
[19:55] <c_14> If you can provide a sample, that would be great as well.
[19:57] <dequid> My smallest file is 80 MB, i tried to make it smaller with 'dd' as described in the bug report howto, but the resulting file is unreadable for every player/ffmpeg.
[19:57] <amkrankruleuen> azk: I don't understand
[19:57] <amkrankruleuen> reproduce what?
[19:57] <azk> Your error
[19:58] <amkrankruleuen> maybe i show file?
[19:58] <azk> That exact command produced a seekable mp3 on my android device
[19:58] <azk> sure
[19:58] <c_14> dequid: ffmpeg -t duration -i input.mov -codec copy out.mov
[19:59] <dequid> c_14: duration = time in seconds?
[19:59] <c_14> ye
[19:59] <dequid> ok
[19:59] <amkrankruleuen> azk: http://193.187.65.6/audio.mp3
[19:59] <amkrankruleuen> not seekable
[20:00] <c_14> dequid: you could also use -fs limit_size , not sure how well it works though
[20:01] <dequid> c_14: Funny: when i do that, the audio is now corrupted even in QT
[20:01] <dequid> I think i'll best upload the whole thing to our server.
[20:02] <c_14> If you can just get it somewhere that the devs will have easy access to, it should be fine.
[20:02] <dequid> ok
[20:02] <c_14> Just say that you didn't upload it to the ftp because it was large and cutting it introduced additional errors.
[20:04] <azk> amkrankruleuen: Seems like it's the default Android
[20:04] <azk> 'Music' app that's not seeking
[20:04] <azk> Other players work fine.
[20:06] <spectralsun> what happens if you faststart a video multiple times?
[20:07] <amkrankruleuen> azk: any idea how to solve?
[20:07] <azk> No idea at all.
[20:07] <c_14> spectralsun: Nothing should happen. It should notice that the moov atom is already at the beginning and not do anything.
[20:07] <azk> Short term solution would be to use another player amkrankruleuen
[20:08] <amkrankruleuen> azk: But your file mp3 from mp4 work correctly?
[20:08] <amkrankruleuen> on 'Music' app
[20:08] <amkrankruleuen> ?
[20:09] <spectralsun> c_14: cool, thanks.
[20:09] <azk> Didn't actually check with that since it isn't my default.
[20:10] <azk> amkrankruleuen: Yeh, I can seek in Music app
[20:13] <dequid> c_14: Thanks for your help anyway
[20:19] <amkrankruleuen> azk: show me your command ffmpeg
[20:19] <azk> The exact same as your.
[20:19] <azk> yours
[20:19] <amkrankruleuen> :|
[20:27] <amkrankruleuen> azk: hmm, indeed your is seekable but my not
[20:29] <azk> pastebin your ffmpeg --version
[20:30] <amkrankruleuen> http://pastebin.com/1x8NxYsB
[20:45] <azk> Hmm, I'm running on 2.2.2, not sure if that has anything to do with it.
[21:07] <bunniefoofoo> is there a way to feed ffmpeg separate audio & video stream, using -acodec copy, truncate the audio stream to match the video?
[21:22] <relaxed> bunniefoofoo: the audio is longer?
[21:22] <bunniefoofoo> yeah
[21:22] <relaxed> ffmpeg -i input -shortest ...
[21:23] <bunniefoofoo> I am trying to use ffprobe to get the frame count, then use ffmpeg -t <duration>
[21:27] <relaxed> bunniefoofoo: -shortest
[21:29] <bunniefoofoo> is that going to use estimated duration?
[21:30] <relaxed> it will end encoding when it reaches the end of the shortest stream
[21:32] <bunniefoofoo> shortest based on  the packet pts then...
[21:59] <MykeBates> I've always had issues with doing this properly.  Hoping someone can help me out here!  Trying to convert MKV to MP4.  Simply running -i input.mkv output.mp4 works but the quality is poor.  Every option combination I seem to try -vcodec copy -acodec copy, etc do not seem to work for me.  Any pointers out there for how to convert mkv to mp4 while maintaining quality?
[22:00] <Mavrik> -codec copy is what you're looking for
[22:00] <Mavrik> (just remuxing without reencoding)
[22:00] <Mavrik> so... I suggest fixing your "does not work" issue.
[22:01] <MykeBates> so, for example - ffmpeg -i input.mkv -codec copy out.mp4    ?
[22:02] <Mavrik> yes.
[22:07] <MykeBates> Damn yes!  Forgive my ignorance on this....0_o  (mac specific question)would you happen to be able to explain why it would play fine in a video player(vlc) but not in Mac's preview?
[22:08] <Mavrik> no idea
[22:08] <Mavrik> Quicktime (mac default player) can be quite shitty with format support
[22:08] <Mavrik> and you really didn't provide any info for me to know what you have :P
[22:11] <MykeBates> I certainly can, didn't want to oberstay my welcome with bombarding with too much.  Check it:
[22:11] <MykeBates> Converting with just -i input.mkv output.mp4 results in a poor quality but playable in preview and here are some specs on the video - https://cloudup.com/cUZeWk3LD9Y
[22:11] <MykeBates> When running with -codec copy - mac preview(space bar on file) results in https://cloudup.com/cNbo6browLN as opposed to playing the video and when opening in the same file viewer app - https://cloudup.com/ct457dLuKSn does not seem to want to show the preview or video information.
[22:13] <Mavrik> uh
[22:13] <Mavrik> yeah, that's not helpful
[22:13] <Mavrik> do this:
[22:15] <MykeBates> http://pastie.org/9456510
[22:17] <Dark-knight> wow
[22:18] <Dark-knight> umm "EXACT ffmpeg command and the COMPLETE console output."
[22:18] <Mavrik> MykeBates, output too, it's the important bit - it shows what your source is and what's it encoding into ;)
[22:18] <MykeBates> damn, my bad -- friday -- brain fried.
[22:18] <MykeBates> one sec
[22:20] <MykeBates> http://pastie.org/9456521
[22:21] <Mavrik> ah
[22:21] <Dark-knight> and what is the problem with the output? what are you trying to accomplish?
[22:21] <Mavrik> MykeBates, my guess is preview/quicktime doesn't like the AC3 audio
[22:22] <Mavrik> Dark-knight, he has problems with remuxed mp4 files not playing in OS X preview
[22:22] <Dark-knight> let me check my notes, i got something for that
[22:22] <Mavrik> MykeBates, AC3 isn't really standard as long as MP4 wide support goes
[22:22] <Dark-knight> brb
[22:22] <Mavrik> also, audio is 5.1 :)
[22:23] <bunniefoofoo> MP4 proper needs to use AAC audio
[22:23] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a libfaac output.mp4
[22:23] <Dark-knight> that should do it
[22:24] <MykeBates> nailed it!
[22:24] <Dark-knight> you're welcome
[22:24] <MykeBates> Quite obvious I need some brushing up on a/v codecs... sorry for my ignorance and thanks a million for the help :)
[22:25] <Dark-knight> np
[22:26] <Mavrik> ugh
[22:26] <Dark-knight> what
[22:26] <Mavrik> well that just murdered all your audio quality :P
[22:26] <MykeBates> yeah?
[22:26] <Dark-knight> haha he could just try "-c:a aac" instead
[22:27] <Dark-knight> delete the libf
[22:27] <Mavrik> MykeBates, yes, since ffmpeg took defaults and libfaac doesn's support multichannel
[22:27] <Mavrik> so your audio probably went from 5.1 high-quality to shitty stereo :D
[22:27] <Dark-knight> well lets see
[22:27] <Dark-knight> post the output
[22:28] <MykeBates> cool, one sec
[22:28] <MykeBates> http://pastie.org/9456535
[22:29] <Dark-knight> looks like 5.1 to me
[22:29] <Mavrik> oh, libfaac now supports 5.1, nice
[22:30] <Dark-knight> so it seems that worked, but ill post this anyway
[22:30] <MykeBates> bonus!
[22:30] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a aac output.mp4
[22:30] <Dark-knight> If that doesn't work the first time use
[22:30] <Mavrik> anyway, you might wanna use the internal aac encoder (libfaac really is the worst of all possible AAC encoders)\
[22:30] <Dark-knight> ffmpeg -i input.mkv -c copy -strict -2 -c:a aac -b:a 32k output.mp4
[22:30] <Mavrik> and set bitrate to something like 256kbit
[22:30] <Baked_Cake> mine only works if i set it to libvo_aacenc
[22:30] Action: Mavrik slaps Dark-knight.
[22:30] <Dark-knight> what?
[22:30] <Mavrik> WHY are you giving him misleading advice
[22:30] <Mavrik> coding movie audio to 32kbps?!
[22:31] <Dark-knight> well it works for me
[22:31] <Baked_Cake> lol
[22:31] <Mavrik> jesus fuck
[22:31] <Mavrik> that's 1950s telephone quality :D
[22:31] <Baked_Cake> dark knight like to have his ears shredded by the devil
[22:31] <Dark-knight> fine then dont listen
[22:31] <Mavrik> Baked_Cake, yeah, that's because your ffmpeg is built with libvo_aacenc instead of libfaac
[22:32] <Baked_Cake> what do you guys think about the vbr function on libfdk_aaac
[22:32] <Mavrik> libfaac < libvo_aacenc < internal aac < fdk-aac
[22:32] <Baked_Cake> typo
[22:32] <Mavrik> that's pretty much the quality scale
[22:32] <Baked_Cake> ic
[22:32] <Dark-knight> Mavrik:  mind posting the command line using the internal aac encoder then?
[22:32] <Mavrik> fdk is currently still pretty much the best
[22:33] <Mavrik> Dark-knight, take what Dark-knight said but change -b:a to something sensible, 256k should probably keep enough of 5.1 quality for you to not notice the difference :)
[22:33] <Mavrik> er
[22:33] <Mavrik> yes, that made no sense
[22:33] <MykeBates> ffmpeg -i input.mkv -c copy -strict -2 -c:a aac -b:a 256k output.mp4 <--- worked really well
[22:34] <Dark-knight> i actually want Mavrik to answer my question
[22:34] <MykeBates> can notice the crispness in the audio :)
[22:34] <Baked_Cake> ive been giving -vbr 3 a try on my 5.1 flacs i think it comes out to about 224k on average
[22:34] <Baked_Cake> on 2 channel i get about 132k
[22:35] <Mavrik> mhm, that's ok
[22:35] <Mavrik> most AAC audio is about ~192k for nice quality
[22:36] <Mavrik> we went as low as 64k for mobile audio, but just with HE-AACv2 profile there
[22:37] <Dark-knight> Mavrik:  mind posting the command line using the internal aac encoder then?
[22:37] <Mavrik> em.
[22:38] <Dark-knight> and do one with the fdk-aac while your at it
[22:38] <Mavrik> didn't you read what I wrote couple of lines above?
[22:38] <Baked_Cake> u have to get a pirateed version of ffmpeg to get the libfdk_aac codec
[22:38] <Baked_Cake> or build it urself
[22:39] <Dark-knight> so just
[22:39] <Dark-knight> ffmpeg -i input.mkv -c copy -c:a internal aac output.mp4
[22:39] <Dark-knight> will do?
[22:40] <Baked_Cake> dark u can run ffmpgeg -codecs
[22:41] <Baked_Cake> erm
[22:41] <Dark-knight> just a simple yes or no
[22:41] <Baked_Cake> ffmpeg -codecs
[22:41] <Baked_Cake> pause
[22:41] <Mavrik> Dark-knight, you need to set desired bitrate
[22:41] <Mavrik> to something sensible
[22:42] <Baked_Cake> and that will give you a list of the codecsthat u can try
[22:42] <Dark-knight> what happens if i leave it blank?
[22:42] <Mavrik> also, you need the strict option for experimental codec
[22:42] <Mavrik> it'll default to something awful
[22:42] <Baked_Cake> mine never worked if i just put -c:a aac
[22:42] <Mavrik> Baked_Cake, mhm, but it's usually worth building your own ffmpeg if you do something n volume :)
[22:42] <Mavrik> yeah, because aac is experimental
[22:45] <Baked_Cake> i should take the time to learn how to do it
[22:46] <Mavrik> well, if you're on Windows you have a problem
[22:46] <Baked_Cake> ya thats my problem
[22:46] <Mavrik> on linux it's mostly just ./configure --enable-fdk-aac
[22:46] <Baked_Cake> i know almost nothing about linux
[22:46] <Mavrik> on OS X it's just brew install ffmpeg --enable-fdk-aac :)
[22:47] <Baked_Cake> i tried doing it myself with this guide that used a linux shell terminal in windows, but i didnt get vvery far
[22:47] <Baked_Cake> im probly better off trying to set up my virtual machine again
[22:49] <Dark-knight> i did ffmpeg -codecs but it gave me a list of all of them. is there a command that just shows me the audio codecs
[22:50] <Baked_Cake> i think generally the video codecs r listed first
[22:50] <Baked_Cake> then oudio, the availible aac codec-s shouldbe clumped together
[22:53] <Dark-knight> i know there is a command that just shows the audio codecs, i used it once before but now i can't remember
[22:54] <Mavrik> hmm, I don't think there is - probably something like "ffmpeg -codecs | grep A"
[22:55] <Dark-knight> i did a quick google but couldn't find the command
[22:56] <Baked_Cake> u can always right click + find
[23:05] <Dark-knight> ahhhh it was -encoders
[23:05] <Dark-knight> made it easy
[23:06] <Dark-knight> what is this
[23:06] <Dark-knight> libfaac is clearly supported but it is not shown in the output
[23:07] <Baked_Cake> is probly not enabled
[23:07] <Mavrik> it only shows up if you compiled ffmpeg with libfaac
[23:07] <Baked_Cake> or that
[23:08] <Baked_Cake> if ur not  linux or compiler savy u can find nonfree versions of ffmpeg but they are outdated and god knows what else
[23:08] <Dark-knight> ok i just converted a video with ac3 audio using libfaac as a test, and it worked
[23:08] <Dark-knight> but its not showing up in the list
[23:08] <Baked_Cake> o
[23:56] <bunniefoofoo> is there some trick to get ffprobe to show me the duration of an mpeg2video stream (.m2v) file?
[23:56] <bunniefoofoo> it says duration N/A
[00:00] --- Sat Aug  9 2014