[Ffmpeg-devel-irc] ffmpeg.log.20131108

[burek]

[00:39] <hailwood> Hey guys, I am using PHP-FFmpeg - should I ask here about issues with it or on #PHP? My specific issue is that one format is double running a command, but all the others are fine.
[00:42] <llogan> hailwood: we don't have anything to do with that
[00:42] <hailwood> Sweet as - to #php it is!
[00:43] <llogan> hailwood: this might be useful for you http://trac.ffmpeg.org/wiki/Using%20FFmpeg%20from%20PHP%20scripts
[00:43] <llogan> see first paragraph in "conclusion"
[00:47] <hailwood> llogan, Yeah; I'm not using ffmpeg-php instead I am using this: https://github.com/alchemy-fr/PHP-FFmpeg which is actively developed
[01:06] <paulfm> Hey everyone
[01:08] <paulfm> I'm having trouble with ffmpeg on osx - finding libmp3lame. I have the lib/headers installed and am using --enable-libmp3lame... and still get the following error:
[01:08] <paulfm> ERROR: libmp3lame >= 3.98.3 not found
[01:09] <paulfm> Any advice is much appreciated
[01:09] <paulfm> libs are in /usr/local/lib
[01:09] <paulfm> config.log says is couldn't find <lame/lame.h>
[01:10] <paulfm> This file is in /usr/local/lib/lame/lame.h
[01:10] <paulfm> Lastly... libmp3lame is 3.99.5
[01:13] <paulfm> correction: header is in /usr/local/lame/lame.h
[01:14] <hailwood> Hmm, what's the effect of running a command twice with the only difference being -pass 1 or -pass 2 ?
[01:14] <paulfm> ob wow... I mean file is in /usr/local/include/lame/lame.h
[01:15] <paulfm> Is there any way to tell configure the actual path of libmp3lame
[01:17] <triggerhapp> How well would ffmpeg deal internally with, say, a video source that can change dimensions? I dont suspect there's much support right?
[01:19] <paulfm> looks like --disable-yasm is what I needed...
[01:19] <paulfm> and leaving out --enable-libmp3lame...
[01:31] <llogan> paulfm: --disable-yasm is another way of saying "--enable-slowness"
[01:31] <paulfm> haha
[01:31] <llogan> did the tail of config.log have any useful info?
[01:31] <paulfm> llogan: well - it compiled.. with mp3lame... weirdly enough
[01:32] <paulfm> all it said was that it couldn't find the header
[01:32] <paulfm> As far as I can remember
[01:32] <llogan> looks like someone else had similiar issues (I guess) lately http://ffmpeg.org/pipermail/ffmpeg-user/2013-November/018364.html
[01:33] <paulfm> make check passed all tests
[01:33] <llogan> might be useful for you: https://trac.ffmpeg.org/wiki/MacOSXCompilationGuide
[01:34] <paulfm> Yeah... odd.. I even tried older version of the libs.. but still no luck. I find it odd that adding disable-yasm, and not even mentioning to enable libmp3lame... build ffmpeg with mp3 support
[01:35] <paulfm> Ahaa.. handy guide.
[01:35] <fundies> any way can I play a mp4 with a missing trex / header?
[01:35] <paulfm> will try it with those options... thanks llogan
[01:38] <paulfm> I get same build error with OS X build guid options
[01:38] <paulfm> --disable-yasm makes it all work
[01:39] <llogan> fundies: sure: ffplay -f rawvideo -s 320x240 -i input.mp4
[01:40] <paulfm> actually... something is wrong here... checking
[01:41] <fundies> llogan, ffplay -f rawvideo -s 640x480 ~/Downloads/CCLENG0001-480-1000K.mp4Seg1-Frag3
[01:41] <fundies> i got colored bits
[01:41] <llogan> that was a bad joke.
[01:42] <paulfm> hmmm... was too hasty.. mp3 decode is enabled(go figure).. no encode :-(
[01:42] <fundies> llogan, http://pastie.org/8464180
[01:43] <paulfm> llogan: do you know if there is a way to tell ffmpeg where exactly to look for libs/headers?
[01:43] <Danathus> hey
[01:44] <Danathus> I am trying to use ffmpeg (on windows, if it matters) to take a bunch of png files in order and put them in a video file, playing at 1 fps
[01:44] <llogan> fundies: i would guess not with ffmpeg, but i'm not sure. maybe JEEB does.
[01:45] <Danathus> I have it working except for the 1 fps part, and whenever I try to specify a frame rate I get videos with just one frame throughout
[01:45] <fundies> JEEB ?
[01:45] <Danathus> has anyone here had any success doing something like what I'm trying to do? Or possibly knows that it's not possible?
[01:46] <llogan> Danathus: http://ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_003f
[01:46] <Danathus> my command line looks like this: ffmpeg.exe -y -f image2 -i img%%03d.png video.mov  (without attempting to set fps, works, but plays much faster than I want)
[01:46] <Danathus> thanks, I'll take a look
[01:47] <llogan> do you want it to show one frame per second, and/or do you want the output to actually be 1 fps?
[01:47] <Danathus> hmm that doesn't seem to cover setting a frame rate
[01:48] <Danathus> ideally the output to be 1 fps, but I'm not sure I'm entirely clear on the difference
[01:48] <llogan> for example, you can set input -r 1 and output -r 25. it will duplicate frames to reach -r 25 but the same image will be shown for one second
[01:48] <Danathus> I'd like it so that the images stored in the final output are just one image every second of video, so it's a smallish file
[01:48] <Danathus> ah I see
[01:48] <llogan> then add -r 1 as an output option
[01:48] <llogan> i mean input
[01:49] <llogan> note that some players may not work with -r 1
[01:49] <Danathus> -r 1 as input, and what for output?
[01:49] <llogan> nothing, since the output will inherit the input -r
[01:49] <Danathus> maybe that is my problem, the player not working -- testing with vlc
[01:49] <Danathus> okay
[01:50] <llogan> then add an output -r and increase the value, starting with 2, until it works if you are trying to limit output frame rate
[01:50] <Danathus> when I use -r 1 for input I get a video that is the correct length but when I play it it's just one still frame for the entire duration
[01:50] <Danathus> trying the -r 2 and up
[01:51] <llogan> paulfm: --extra-cflags --extra-ldflags
[01:51] <llogan> example http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[01:52] <paulfm> llogan: thanks - taking a look
[01:52] <Danathus> when I try with -r 24 on the output, it takes a lot longer to encode, I guess that's a good sign :/
[01:52] <Danathus> okay, output looks like it's correct then
[01:52] <Danathus> thans llogan
[01:52] <Danathus> *thanks
[01:53] <llogan> also see https://trac.ffmpeg.org/wiki/x264EncodingGuide
[01:53] <Danathus> seems well compressed still I think, I am satisfied
[01:53] <Danathus> hmm
[01:54] <Danathus> I think it will take me a bit to sift through that (:
[01:54] <llogan> if your output is not yuv420p then it won't work with dumb players like QuickTime
[01:54] <Danathus> I have a follow-up question
[01:54] <Danathus> alright
[01:54] <Danathus> works with vlc anyhow, that's what I care about now (ultimately to be read with custom software anyway)
[01:55] <llogan> ok. then don't worry about that
[01:55] <Danathus> my follow-up question is, really my input of png files are a temp intermedia format, and I am wondering if I might be able to have ffmpeg create the output from my source format
[01:55] <Danathus> I'll explain
[01:56] <Danathus> I am starting with a file which is a sequence of raw image data frames, one after another, no header information of any kind -- but I know the bit depth, dimensions, etc
[01:57] <Danathus> I am using other code to break this into a bunch of .raw files, one for each frame, and then I loop through making an ffmpeg call to convert each one to a png file, and then have my final ffmpeg command line call to put all the png files together in the mov file
[01:58] <paulfm> llogan:  --extra-cflags="-I/usr/local/include/"  worked!!.. thank you
[01:58] <Danathus> if I didn't have to convert from the .raw files to the png files that would be an improvement, and if I could get ffmpeg to read from the big fat sequential raw file, that would be even better
[01:59] <Danathus> but where img%03d.png seems to work, img%03d.raw doesn't, even if I am specifying -vcodec rawvideo -f rawvideo -pix_fmt rgb32 -s 480x640
[02:00] <Danathus> I *should* be able to get the .raw files to work, right?
[02:01] <llogan> i'd assume so, but i think the %03d syntax is specific to the image2 muxer
[02:02] <llogan> meaning i don't know if that would work with raw.
[02:02] <Danathus> hmm I see
[02:02] <llogan> you can ask on ffmpeg-user mailing list
[02:03] <llogan> did you try the big raw file as an input?
[02:06] <Danathus> I have
[02:06] <Danathus> it complains that it can't find the second frame
[02:07] <Danathus> okay
[02:08] <Danathus> for the big raw file attempt?
[02:09] <llogan> yes
[02:10] <Danathus> http://pastebin.com/7f7v8sSR
[02:10] <Danathus> sorry for the delay
[02:10] <Danathus> oh I'll include the error too
[02:10] <Danathus> my bad
[02:13] <Danathus> here http://pastebin.com/Ssm3fqwn
[02:15] <Danathus> updatefirst didn't seem to help either
[02:15] <llogan> -f image2 with mov container doesn't make sense to me
[02:16] <Danathus> okay, should I just omit the -f image2?
[02:16] <Danathus> oh wow it's working
[02:16] <llogan> isn't the ultimate goal to make a video out of the raw input? why even output to frames?
[02:17] <Danathus> only because that's how I was getting it to work
[02:17] <Danathus> but this is much better
[02:17] <Danathus> thanks a lot
[02:17] <Danathus> now I just need to combine this with the framerate bit...
[02:18] <llogan> by default it assumes the raw input is 25 frames/s
[02:18] <Danathus> alright
[02:19] <llogan> perhaps you can pipe the raw to ffmpeg from whatever makes the raw file
[02:19] <Danathus> the raw data isn't generated sequentially
[02:19] <Danathus> that's the only problem there
[02:20] <Danathus> okay I have a line that does exactly what I want now, and works swimmingly
[02:40] <fundies> noone can help me play this file?
[02:48] <Danathus> thanks again llogan
[02:48] <Danathus> I'm off
[02:48] <Danathus> good luck to you fundies
[02:48] <Danathus> ciao everybody
[03:28] <fundies> JEEB dead?
[05:06] <phunyguy> hello, I am trying to encode flac to mp3 in a script, which mostly works, but the -map_metadata option doesn't bring over the DATE tag.
[05:07] <phunyguy> it brings over everything else, including musicbrainz data, which makes this so strange.
[05:34] <phunyguy> le sigh.  It was writing ID3v2.4
[05:39] <DANtheBEASTman> how does one convert m4a to mp3 v0?
[05:39] <DANtheBEASTman> looking through the wiki all I can find is video examples
[05:42] <DANtheBEASTman> nvm, figured it out!  for f in *.m4a; do; avconv -i "$f" -qscale:a 0 "${f[@]/%m4a/mp3}"; done
[05:42] <DANtheBEASTman> thanks anyway!
[06:14] <kcm1700> I am creating mp4 file with some code. I'm feeding a packet to the AVFormat with av_write_frame. Which does AVPacket.pts use as a time base, AVStream.time_base or AVStream.codec->time_base ?
[06:21] <ChannelZ> Anyone familiar with the guts of h264/aac in an mp4?
[06:23] <ChannelZ> As in, I'm trying to help someone fix a corrupt mp4 (quicktime really) that crashed while recording... so no moov atom.  But I've got the data in the mdat atom which should be the actual media streams, but am not sure what "format" it's in such that I could feed it to ffmpeg or anything to try to salvage what is there.
[09:36] <tiksa> hey guys, could you please tell what's wrong with my ffmpeg command? My intention is to just trim video so that only video+audio between seconds 2..4 are left. the command is:
[09:36] <tiksa> ffmpeg -i input.mp4 -filter_complex "[0:0]trim=start=2:end=4[video];[0:1]atrim=start=2:end=4[audio]" -map '[video]' -map '[audio]' output.mp4
[09:38] <tiksa> it does not throw any error but what happens, the first 2 seconds are silent and the frame at 0:02 is shown. then, the rest 2 secs are as it should, from 0:02 to 0:04
[09:42] <noobed> could some1 give me example source code of decoding an Vorbis audio stream and encoding it into MP3 one
[13:02] <jay_> why does a video player say the file(.mp4) contains no playable stream but .flv, .avi etc play when the file is not 100% complete?
[13:04] <ubitux> because the header is likely at the end
[13:04] <JEEB> flv has no index, and avi can be parsed without one :P
[13:04] <JEEB> and most people and apps create mp4 files that have a single index, at the end
[13:04] <JEEB> because the index has to contain everything in the file
[13:06] <jay_> hmm, thanks for the replies ubitux , JEEB. I'll have to read a little on that
[13:07] <JEEB> mp4 can have smaller indexes around the file, but not many things use that feature
[13:07] <JEEB> "movie fragments" the feature was called, IIRC
[13:08] <jay_> JEEB, to sum it up, mp4 requires an index unlike flv and avi and for some reason it is usually at the end of the file
[13:08] <JEEB> not for "some reason"
[13:08] <JEEB> but for the reason that you have to have the whole file created when you make the index
[13:08] <JEEB> usually it's put in the beginnig by first creating the file, and then rewriting it so that you write the index first :P
[13:09] <jay_> oh ok
[13:16] <jay_> is it possible to make a new index with part of a .mp4 file?
[13:17] <JEEB> no
[13:17] <JEEB> since you have no idea what's in the file if you didn't make it yourself
[13:17] <JEEB> and if you made it yourself you could just do that again and use movie fragments
[13:17] <JEEB> (or just put the index in the beginning if you want that)
[13:18] <jay_> interesting. Thanks JEEB . I think i have to go read about indexing mp4
[17:51] <exfalso> Hi, i'd like to write a small program that simply takes a single picture and audio file and creates a video out of it. what encoding is best for this? what libraries should i use? (i'm a total noob, not even sure i'm asking in right channel)
[17:52] <exfalso> or is there an existing way of doing this with ffmpeg?
[18:13] <exfalso> this was easier than i thought... now just need to figure out how to keep audio quality the same
[18:17] <exfalso> was easy too... ffmpeg rocks
[18:47] <uehuehue> What does ffmpeg return on error/success?
[18:48] <uehuehue> Is it OK to do "ffmpeg <blah> && rm file"?
[18:52] <ubitux> yes
[18:52] <ubitux> is file the output file?
[18:52] <uehuehue> file would be the input file
[18:53] <ubitux> k
[18:53] <uehuehue> As in, convert file to something and remove the input file
[18:53] <ubitux> i hope you always keep a copy of your original somewhere though :)
[18:54] <uehuehue> Heh, converting from mp3 to ogg. Not sure what could go wrong
[19:02] <ubitux> well it's lossy to lossy so quality will go down
[19:02] <ubitux> ah he left.
[19:28] <ac_slater> hey guys. Is there an x264 channel? If not, maybe someone knows how to cross compile x264 (for ARM7) AND enable asm. I can compile only with disable-asm. Do I need and arm yasm? I'm confused. Thanks!
[19:29] <JEEBsv> #x264 is the x264 channel
[19:29] <JEEBsv> and no, you don't need yasm for ARM compilation
[19:29] <JEEBsv> the asm is generally inline for that, the little there is of it
[19:29] <ac_slater> oh. Weird I joined it and it was empty, it isnt now. Thanks though.
[19:29] <cortexman> ffmpeg looks like it supports NIST Sphere files http://ffmpeg.mplayerhq.hu/doxygen/trunk/nistspheredec_8c_source.html
[19:29] <ac_slater> JEEBsv: So you dont think it'll be beneficial? I was hoping there was NEON support, etc
[19:29] <JEEBsv> let's move the discussion there
[19:32] <cortexman> can ffmpeg output raw audio?
[19:32] <JEEBsv> cortexman: yes
[19:32] <JEEBsv> see ffmpeg -codecs:a
[20:14] <cortexman> i have a NIST Sphere file with these headers: sample_coding -s4 ulaw, sample_rate -i 8000, sample_n_bytes -i 1, sample_sig_bits -i 8
[20:14] <cortexman> no matter what options i specify the output is 16k which makes the audio sound pitch shifted into a chipmunk cartoon
[20:25] <durandal_1707> cortexman: sample, link to it?
[20:27] <cortexman> durandal_1707: http://filebin.ca/11FdZnYT7BCH/sw02001.sph
[20:27] <cortexman> i'm trying to get the first channel rate converted to 16k without becoming chipmunky
[20:27] <cortexman> short ints
[20:28] <cortexman> raw format
[20:46] <cortexman> figured out a way to do it with sox!
[20:49] <durandal_1707> cortexman: it works fine here
[20:49] <durandal_1707> and it is ulaw
[20:50] <cortexman> durandal_1707 you were able to convert the file?
[20:50] <durandal_1707> with ffmpeg? sure
[20:50] <cortexman> i was not able to convert it to 16k and have it properly rate converted with ffmpeg
[20:51] <cortexman> https://gist.github.com/brianmingus/b8ca6659c541a0fb2e72/raw/7b991ecb053274f42370afda64bc36bf4ab3e008/gistfile1.txt
[20:53] <durandal_1707> cortexman: you are not using nist demuxer but raw one
[20:53] <cortexman> the output should be raw
[20:54] <durandal_1707> and you are not giving full uncut console output but just worthless commands
[20:54] <cortexman> that's the only console output..
[20:56] <cortexman> i figured ffmpeg would choose nist automatically based on .sph
[20:56] <durandal_1707> cortexman: what you use to play raw file?
[20:56] <cortexman> `play`
[20:56] <cortexman> it's part of sox
[20:57] <durandal_1707> and do you set number of channels?
[20:57] <durandal_1707> what is 'play' command ?
[20:57] <cortexman> it plays an audio file
[20:57] <durandal_1707> yes but full command is?
[20:58] <cortexman> play file
[20:58] <durandal_1707> > play /tmp/o.raw
[20:58] <durandal_1707> play FAIL formats: bad input format for file tmp/o.raw': sampling rate was not specified
[20:58] <durandal_1707> raw can not work
[20:58] <cortexman> right.. there is play-sample-rate
[20:58] <cortexman> you can figure out what the sample rate of the file is by tweaking that
[20:58] <cortexman> until it sounds right
[20:58] <durandal_1707> but if you do not set channel number it will not be correct
[20:59] <cortexman> yeah i think that's the problem
[21:00] <durandal_1707> play -c 2 -r 8k -e signed -b 16 /tmp/o.raw
[21:00] <durandal_1707> this works fine
[21:02] <cortexman> ahhh! thanks!
[21:02] <cortexman> i used that on fmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le sw02001.raw
[21:02] <cortexman> i already know how to extract a channel with sox also
[21:04] <durandal_1707> you can do same with filters
[21:07] <durandal_1707> fmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le -af pan="mono|c0=FR" sw02001.raw
[21:08] <cortexman> oh man that would have taken me forever
[21:08] <cortexman> i had already tried like 10 methods:)
[21:09] <durandal_1707> like typing random stuff?
[21:10] <cortexman> interestingly you can still hear the second channel in the background. it's just very faint.
[21:12] <cortexman> ffmpeg must be doing something weird because the voices are encoded on separate channels
[21:13] <durandal_1707> what?
[21:13] <durandal_1707> you want to extract single channel or remix?
[21:15] <cortexman> ffmpeg -i /Volumes/SONDER/SwitchboardCorpus1/SW1P1_01/swb1/sw02001.sph -f s16le -acodec pcm_s16le -map_channel 0.0.0 sw02001_1.raw -f s16le -acodec pcm_s16le -map_channel 0.0.1 sw02001_2.raw
[21:15] <cortexman> :)
[21:15] <cortexman> thanks for your help
[22:51] <maujhsn> I'm using an older version  of ffmpeg version git-2012-01-21-9e53f62 Copyright 2012
[22:51] <maujhsn> i would like to know how to control the "rec" volume when I am recording mydesktop without using an external mic?
[22:52] <maujhsn> I thought I could do this in "pavucontrol" but its not increasing record volume!
[22:53] <share> do you recommend using ffmpeg to convert DTS track to AC3?
[22:54] <maujhsn> share Whom are you talking to?
[22:55] <share> to the users on this channel
[22:55] <maujhsn> Ok
[22:56] <DexterF> share: http://www.videohelp.com/tools/PopCorn-MKV-AudioConverter - off topic, but might be what you need. guess will run in wine if there's no native linux code
[22:57] <DexterF> that said I'd be interested, too, if ffmpeg can be used to convert and how it then handles the different dynamic ranges etc. usually dts means "fumble with the volume controls for two hours"
[22:59] <maujhsn> DexterF Was this message intended for me?
[22:59] <share> DTS to AC3 5.1
[22:59] <share> could be
[22:59] <DexterF> maujhsn: to the channel
[23:00] <maujhsn> Ooops!
[23:01] <share> EAC3toGUI ?
[23:01] <share> there is this....... https://github.com/JakeWharton/mkvdts2ac3/
[23:11] <share> Application provided invalid, non monotonically increasing dts to muxer in stream 0 ********* ?
[23:16] <maujhsn> I'm using an older version  of ffmpeg version git-2012-01-21-9e53f62 Copyright 2012
[23:16] <maujhsn>  i would like to know how to control the "rec" volume when I am recording mydesktop without using an external mic?
[23:16] <maujhsn> I thought I could do this in "pavucontrol" but its not increasing record volume!
[00:00] --- Sat Nov  9 2013