[Ffmpeg-devel-irc] ffmpeg.log.20130503

[burek]

[00:20] <Zarx> im trying to encode only a certain frame range of a video, and also encode the audio from that same frame range. I had thought the command to trim it would look something like this:   -vf select='gte(n\,100)*lte(n\,200)',aselect='gte(n\,100)*lte(n\,200)'
[00:20] <Zarx> am I wrong?
[00:21] <Zarx> the video part works, but when I add the part to trim the audio, that breaks it
[00:23] <durandal11707> Zarx: because aselect is audio filter and thus cant work with -vf
[00:23] <Zarx> haha
[00:23] <durandal11707> use -filter_complex
[00:23] <Zarx> ok, thanks
[00:25] <durandal11707> also n for aselect means number of samples, not frames
[00:26] <Zarx> oh... am I going to have to use timecodes then?
[05:01] <smj> using a webcam overlay drops streaming fps to 3-5 from 60
[05:04] <smj> to me that sounds like it's waiting to get a fresh frame from the webcam
[05:04] <smj> is there a way to do this asynchronoysly?
[05:11] <smj> it might be this http://ffmpeg.org/pipermail/ffmpeg-user/2011-May/000934.html
[05:26] <smj> in overlay filter documentation... "Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as it does the example for the movie filter."
[05:26] <smj> what's a PTS? some sort of time stamp?
[05:31] <relaxed> smj: http://en.wikipedia.org/wiki/Presentation_time_stamp
[05:31] <smj> yeah, found that
[06:03] <smj> that setpts thing doesn't do much...
[06:06] <smj> so there's https://lists.ffmpeg.org/pipermail/ffmpeg-user/2011-April/000346.html https://lists.ffmpeg.org/pipermail/ffmpeg-user/2011-April/000395.html and http://ffmpeg.org/pipermail/ffmpeg-user/2011-May/000939.html
[06:16] <smj> and https://ffmpeg.org/trac/ffmpeg/ticket/1339?cversion=3&cnum_hist=2
[06:46] <cryptopsy> how to trim duration a-b out of a file that's a-c in duration?
[06:47] <cryptopsy> nvm
[07:16] <Mavrik> Gandalfar, wat
[11:46] <EvilTengil> I'm trying to figure out how I can bundle some data with each frame. Is this possible with "data streams" (AVMEDIA_TYPE_DATA)?
[12:07] <zap0> i too would like to know.
[12:08] <durandal_1707> what data?
[12:08] <EvilTengil> 6 doubles
[12:08] <EvilTengil> the video is supposed to be sent from a phone with sensors
[12:45] <rvnikauj> hi ffmpeg users and dev
[12:46] <rvnikauj> i have an issue with cli usage under archlinux
[12:46] <rvnikauj> i have some .VOB which i want to convert to theora/ogg
[12:46] <rvnikauj> but only the 2.0 track, which is the third one
[12:47] <rvnikauj> so i try this from the wiki
[12:47] <rvnikauj> at trac
[12:47] <rvnikauj> http://pastebin.com/UiXm4ser
[14:49] <sacredchao> Hi, can I use ffmpeg to extract an AC-3 stream from a decrypted M2TS file that came from a BR?
[14:50] <sacredchao> I want to leave all streams alone, and simply extract only one of the 2 AC-3 audio streams in the M2TS files
[14:54] <Magicking> sacredchao: You might be looking for the -map option http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option
[14:56] <dolevo> hi all
[14:57] <dolevo> I have a question related to hardware accelerated decoding on AMD radeon. Is there anybody who can help me?
[15:02] <durandal_1707> no
[16:08] <theos> hey
[16:09] <theos> is there a way to reduce the size of mp3 files?
[16:10] <durandal_1707> no
[16:11] <durandal_1707> unless you want to transcode them which is really bad idea
[16:11] <theos> like reduce bitrate etc?
[16:13] <theos> no?
[16:16] <durandal_1707> yes you can - its called transcoding - and its bad idea
[16:18] <theos> why is it bad idea?
[16:27] <relaxed> theos: you lose quality
[16:28] <sacarasc> theos: It's like taking photo of a photo. It won't be as good as the original photo and only a shade of the original.
[16:29] <theos> got it
[16:31] <sacarasc> If you have the original wave files, though, you could reencode that and it would sound better than reencoding the mp3 ¶ mp3.
[16:58] <EvilTengil> Does anyone know what the side_data is on a AVFrame? Can I add my own data there?
[17:07] <tempusfol> Hello, make fails with ffmpeg's last git with "/usr/bin/ld: libavutil/lls.o: relocation R_X86_64_PC32 against undefined symbol `memset@@GLIBC_2.2.5' can not be used when making a shared object; recompile with -fPIC". May I proceed pasting my ./configure here?
[17:10] <tempusfol> well, I was asking if this was the correct channel to discuss about this, I'll assume "yes"
[17:18] <atog> When I tell ffmpeg to record audio from an HTTP stream for a specific duration, I receive a warning that it's estimating duration from the bitrate. How do I get rid of this warning and make it record based upon the actual time it's been recording?
[17:31] <tempusfol> Last full log (only last error showed) http://fpaste.org/10255/75950281/  btw, I've tried with --enable-static --disable-shared as well. It's part of a script that rebuilds for me mplayer,ffmpeg,x264,xvidcore,fr3ior and that has served me well for over two years.
[18:00] <atog> anyone have any suggestions for my duration issue?
[18:01] <Magicking> atog: You download the file first and don't use http
[18:02] <atog> Magicking: it's an audio stream
[18:03] <atog> Magicking: What I want to do is copy the a portion of a live audio stream for a specific duration
[18:04] <atog> Magicking: I just don't see why ffmpeg is trying to estimate the duration. If I tell it to record for 10 minutes, I expect the output to be 10 minutes long
[18:08] <klaxa> atog: have you tried doing that?
[18:08] <klaxa> i mean just recorded and checked?
[18:08] <klaxa> the "estimating duration from the bitrate" doesn't really apply on streams i think
[18:11] <Fjorgynn> Where is my cake?
[18:11] <Fjorgynn> what audio stream is that?
[18:13] <Fjorgynn> happy friday
[18:15] <Magicking> atog: You are just anoyed by the warning ?
[18:16] <atog> klaxa: I've got several streams that I used cron to start at a specific time and told ffmpeg to record for a fixed amount of time. The cron job usually starts recording a few seconds after the minute.. and the file time indicates that ffmeg stopped recording 8 or more seconds before it should have
[18:17] <klaxa> that's weird, indeed
[18:19] <atog> klaxa: I can only assume the difference is due to the fact that ffmpeg is due to the initial duration estimate
[18:19] <klaxa> how do you invoke the command?
[18:20] <klaxa> and what kind of stream is it?
[18:20] <atog> klaxa: it's an mp3 stream
[18:20] <klaxa> fixed bitrate?
[18:20] <atog> no idea
[18:20] <atog> presumably
[18:21] <klaxa> still, what command do you use to record the stream?
[18:21] <atog> current command is: ffmpeg -i <http stream> -acodec copy -t 300 output.mp3
[18:22] <klaxa> what ffmpeg version are you using?
[18:22] <atog> klaxa: 1.0.6
[18:23] <klaxa> humm..
[18:24] <klaxa> that's weird indeed
[18:26] <atog> the stream is supposed to be a 64kb mp3 stream
[18:41] <atog> klaxa: time ffmpeg -i <http stream> -acodec copy -t 60 test.mp3 ran for 50s
[18:43] <sine`> audio vorbis is that oog ?
[18:43] <sine`> or ogg
[18:44] <klaxa> atog, pretty weird :S
[18:44] <klaxa> sine`: it's called ogg
[18:45] <sine`> ok so i have a video that has the vorbis 44100 hz, stereo ,s16 default
[18:45] <sine`> how do i just rip or demux the audio without reencoding it
[18:46] <klaxa> ffmpeg -i my_file.mkv -c:a copy audiostream.ogg
[18:46] <klaxa> hmm...
[18:46] <klaxa> maybe add -vn in there too
[18:46] <klaxa> so: ffmpeg -i my_file.mkv -c:a copy -vn audiostream.ogg
[18:46] <Fjorgynn> yeah -vn is awesome
[18:47] <Fjorgynn> if it's ogg by default
[18:47] <klaxa> if you want to make it super fancy use -map
[18:47] <Fjorgynn> and if you want mp3 use lame to convert it from wave
[18:48] <sine`> i got the ogg
[18:48] <sine`> hopefully my phone will play that
[18:48] <Fjorgynn> :)
[18:48] <Fjorgynn> maybe
[18:48] <sine`> dont want to reencode it again uggh
[18:49] <Fjorgynn> what phone you have?
[18:50] <sine`> note 2
[18:50] <sine`> should be ok
[18:52] <sine`> yes
[18:52] <sine`> fine
[18:52] <sine`> schweet
[18:52] <sine`> thanks
[18:52] <sine`> time for a run bye
[18:55] <Fjorgynn> :D
[19:36] <voidah> hi guys
[19:37] <voidah> Is there general tips concerning sound/video out-of-sync issue when using x11grab?
[19:37] <voidah> or a place where I could find the answer
[19:37] <voidah> I tried so many thing
[19:38] <klaxa> editing the file afterwards would be easiest i think
[19:39] <voidah> this is what I would like to avoid if possible
[19:40] <voidah> in despair, I tried recordmydesktop
[19:40] <voidah> there is not sync issue with it
[19:40] <voidah> but I _need_ to use ffmpeg
[19:40] <klaxa> doesn't recordmydesktop just call ffmpeg?
[19:40] <voidah> I don't know, and their website doesn't specify that
[19:41] <voidah> out-of-sync issue with ffmpeg only is visible on long video
[19:41] <durandal_1707> what codec you use?
[19:41] <voidah> libx264
[19:41] <voidah> acodec: mp3
[19:41] <voidah> (tried flac too for acodec)
[19:42] <durandal_1707> tried ultrafast preset?
[19:42] <voidah> yes
[19:42] <klaxa> oh it does in fact not simply call ffmpeg
[19:42] <voidah> I'm now trying veryfast
[19:42] <voidah> to see if it makes a difference
[19:42] <voidah> I record audio and video separately
[19:43] <voidah> the duration is not even the same
[19:43] <voidah> on a ~10 minutes record, the sound is ~10 seconds longer
[19:43] <voidah> the sound is perfectly in sync in the first ~5 minutes of the video
[19:43] <atog> klaxa: I'm going to switch to streamripper since it seems to record for the correct amount of time
[19:43] <voidah> then the longer the video is, the bigger the gap is
[19:43] <klaxa> atog kk
[19:44] <voidah> I tried going though pulse, of directly with hw:0,0
[19:44] <voidah> trought
[19:44] <voidah> same difference
[19:45] <klaxa> hw:0,0 is alsa though
[19:45] <voidah> s/of/or
[19:49] <voidah> last test:
[19:49] <voidah> video is exactly 10:00 long
[19:49] <voidah> sound is 10:02
[19:49] <voidah> both capture were started at the same time
[19:50] <voidah> and stopped at the same time
[19:51] <voidah> the sound and video are perfectly in sync at the beginning
[19:51] <voidah> but out of sync at the end
[19:51] <voidah> I'm sure I made something wrong, and if someone have tips, they are welcomed
[19:53] <voidah> is there a way to merge the video and audio files, shrinking a little bit the audio file so that it fits the length of the video file?
[19:53] <voidah> with ffmpeg of course
[19:56] <klaxa> did you look at -async?
[19:56] <voidah> nope, I google it
[19:58] <voidah> should I use it when I merge the 2 files?
[19:59] <voidah> and does it works with any files types?
[20:00] <klaxa> wait... are you recording audio and video with two ffmpeg instances?
[20:00] <voidah> yes
[20:00] <voidah> I tried doing it when the same instance too
[20:01] <klaxa> but?
[20:01] <voidah> I get the same problem
[20:02] <voidah> what method would you recommend?
[20:04] <klaxa> try using one instance and apply -async and see if that helps
[20:04] <voidah> Ok I'm trying it now
[20:05] <voidah> I get many of these message when using async
[20:05] <voidah> [matroska @ 0x84ce60] st:0 PTS: 67854 DTS: 67854 < 67892 invalid, clipping
[20:08] <voidah> it's mayby a good sign
[20:08] <voidah> still 5 minutes to record
[20:14] <voidah> klaxa: man.. perfectly in sync
[20:14] <voidah> with async
[20:14] <voidah> I'll test it again just to be 100% sure
[20:21] <durandal_1707> tempusfol: you did make distclean?
[20:27] <voidah> klaxa: it does'nt seems to work finally...
[20:27] <voidah> here is the command used: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 20 -s 1920x1080 -i :0.0 -acodec pcm_s16le -async 1000 -vcodec libx264 -preset ultrafast -threads 0 output.mkv
[20:27] <tempusfol> durandal_1707: yes
[20:32] <tempusfol> durandal_1707: now I always get "libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_132'"
[20:33] <durandal_1707> tempusfol: what libx264 version?
[20:34] <tempusfol> durandal_1707: http://paste.fedoraproject.org/10306/67606024
[20:37] <durandal_1707> x264 is build shared?
[20:38] <tempusfol> I've tried right now once again --enable-static --disable-shared, it ends with "libx264.c:(.text.unlikely+0xb08): undefined reference to "x264_encoder_open_132"
[20:38] <tempusfol> x264 is build.... let me check
[20:39] <voidah> tempusfol: remove everything that contains static of dynamic
[20:39] <voidah> in your configure options
[20:39] <voidah> it will work
[20:39] <voidah> I once had a similar proble
[20:39] <tempusfol> x264 is built with "./configure --enable-shared --enable-visualize --enable-strip --extra-cflags='-O2 -march=native -mtune=native' --prefix=/opt/bleedingvideo"
[20:39] <voidah> m
[20:40] <tempusfol> voidah: I'll try that as well
[20:43] <tempusfol> voidah: ...no luck. It always ends with `undefined reference to `x264_encoder_open_132'.. I'll guess I'll grab a different version of x264 and try again?
[20:44] <voidah> ok then I don't know
[20:45] <voidah> try your luck in #ffmpeg-devel
[20:45] <tempusfol> ehm I started here, went there, and then durandal_1707 replied to me here
[20:46] <durandal_1707> instead of pinging there, ping here
[20:46] <durandal_1707> but whatever, what recently changed that caused such breakage?
[20:46] <tempusfol> (that's what I'm doing)
[20:47] <tempusfol> nothing peculiar, it's a script I've used for over two years
[20:47] <durandal_1707> so nothing? did not installed new x264, instealled something other that changed those libs or ....
[20:47] <tempusfol> today, it has recompiled x264, mplayer, frei0r, x264, ffmpeg is the last one...and ffmpeg failed
[20:48] <tempusfol> when the scripts starts, it changes libs of *everything*
[20:48] <tempusfol> ...start fresh.
[20:48] <durandal_1707> reboot
[20:48] <tempusfol> d'oh
[20:48] <cbsrobot> tempusfol: maybe yyou have 2 libx264 - in different places
[20:49] <durandal_1707> disable ccache
[20:49] <tempusfol> ...I could try that as well
[21:18] <tempusfol> durandal_1707: reboot didn't help, ccache disabled didn't help, obliterating all the local mirrors and re-git cloning everything solved. I don't know why it was solved though.
[21:20] <durandal_1707> re-git clonining should really not be needed
[21:20] <tempusfol> I think so
[21:49] <JodaZ> is there something like a -re * some_factor
[21:53] <JodaZ> does anyone know how/if i can produce individual HLS segments from input video on demand ?
[21:58] <JodaZ> :/ i used to get answers here
[22:01] <llogan> JodaZ: wot's your question? i probably just missed it.
[22:01] <JodaZ> llogan, its 3 lines up
[22:02] <JodaZ> o wait, you joined after *_*
[22:02] <JodaZ> does anyone know how/if i can produce individual HLS segments from input video on demand ?
[22:09] <hughmanwho> Hello!  Anyone on here know how to stream rtsp via ffmpeg and save it to a file?  Here is a pastebin of what I've done and my results: http://pastebin.com/j6N0WVX0
[22:10] <tmatth> hughmanwho: try saving it to another container, maybe mp4 can't store your audio
[22:14] <JodaZ> audio: none ?_?
[22:15] <JodaZ> hughmanwho, is there audio ?
[22:17] <hughmanwho> no audio
[22:20] <hughmanwho> Seems to be doing something now, or at least it's missing packets which I assume means it's also getting some.  What format would you recommend
[22:20] <hughmanwho> ?
[22:24] <hughmanwho> Switched over to mjpg and am now getting a bunch of this: http://pastebin.com/j6N0WVX0
[22:27] <tmatth> hughmanwho: can you pastebin the sdp?
[22:29] <hughmanwho> ffmpeg -i rtsp://192.168.14.212/live.sdp -t 10 outputFile.mjpeg
[22:31] <hughmanwho> That one did produce a short clip.. only it seemed like it was only a frame or two, it flashed something then disappeared
[22:31] <hughmanwho> Is mjpeg a valid extension or only valid for streaming or something like that?
[22:35] <Rajeev> what is the significance in  a MP4 files non-standard atom 'yqoo'
[22:39] <Rajeev> is this a standard type of atom now?
[22:45] <hughmanwho> What would be a good file type to use for saving videos?
[22:55] <trose> I'm trying to pipe images directly into ffmpeg from python as in this example http://stackoverflow.com/questions/13294919/can-you-stream-images-to-ffmpeg-to-construct-a-video-instead-of-saving-them-t
[22:56] <trose> I'd like to use png instead of jpeg for better quality though
[22:56] <trose> doesn't seem like ffmpeg supports png for pipe2image
[22:56] <trose> is there a work around?
[23:18] <hughmanwho> Thanks tmatth and JodaZ, for some reason I don't have any problem saving it as a .gif which probably does what I need it to do for now
[23:23] <hughmanwho> Any idea why I keep getting 'non-existing PPS referenced' errors?
[23:23] <hughmanwho> Here's a pastebin: http://pastebin.com/j6N0WVX0
[23:45] <hughmanwho> How can I change the 'analyzeduration'? I'm getting a 'max_analyze_duration 5000000 reached at 5000000 microseconds error.. I call
[23:45] <hughmanwho> C:\Tools\ffmpeg-20130428\bin>ffmpeg -i rtsp://admin:admin@192.168.14.25/defaultPrimary?streamType=u -t 60 -analyzeduration 10000000 outputFile.gif
[23:45] <hughmanwho> yet still get  the same error without the max_analyze_duration having been changed
[23:46] <relaxed> Are you actually getting an error?
[23:48] <relaxed> hughmanwho: That message meant to be informative. It's not an error message.
[23:49] <hughmanwho> I'm getting a bunch of 'non-existing PPS referenced', 'non-existing PPS 0 referenced', 'decode_slice_header error
[23:49] <hughmanwho> ' and 'no frame!' errors
[23:49] <hughmanwho> see this pastebin to see the entire output: http://pastebin.com/j6N0WVX0
[23:49] <relaxed> does it stop encoding?
[23:50] <hughmanwho> Good to know regarding that msg
[23:50] <hughmanwho> I also get a 'Output file #0 does not contain any stream' error
[23:50] <relaxed> where is that in the pastebin?
[23:50] <hughmanwho> Very bottom
[23:51] <hughmanwho> Never mind.. that got cut off
[23:51] <hughmanwho> just a sec
[23:52] <Jorky> Hello Guys
[23:52] <Jorky> anyone here at the moment?
[23:52] <sacarasc> Yo.
[23:52] <Jorky> I just want to know what is the command to convert whole album to another format
[23:53] <sacarasc> What OS?
[23:53] <Jorky> for instance album of mp3 songs to wma at the same time
[23:53] <hughmanwho> http://pastebin.com/j6N0WVX0
[23:53] <Jorky> linux
[23:53] <Jorky> archlinux
[23:53] <sacarasc> for music in *.mp3; do ffmpeg -i "$music" "$music".wmv; done
[23:53] <sacarasc> Or some variation upon that.
[23:54] <Jorky> with music you mean name of the album?
[23:54] <sacarasc> No, that's just a variable name.
[23:54] <sacarasc> Do that in the directory with the files.
[23:54] <hughmanwho> I have to run.. relaxed I appreciate it, if you have any thoughts, I'll keep this open and check on your reply or you can email me at hughmanwho at gmail.com  Thanks again relaxed!!
[23:55] <Jorky> yes but this means that I must do it for every song in album?
[23:55] <relaxed> Jorky: for i in *mp3; do ffmpeg -i "$i" -c:a wmav2 "${i%.*}".wma; done
[23:55] <sacarasc> No.
[23:55] <sacarasc> That's an even better one!
[23:55] <sacarasc> That will do it for each file called whatever.mp3.
[23:56] <sacarasc> I really need to remember the ${i%.*} thing... I never can.
[23:56] <relaxed> Jorky: study this http://mywiki.wooledge.org/BashGuide
[23:57] <Jorky> hm
[23:57] <Jorky> I still don't get it
[23:58] <relaxed> It's a for loop that takes each mp3 in the current dir and encodes them one at a time using ffmpeg.
[23:59] <Jorky> can you type me more specific script
[00:00] --- Sat May  4 2013