[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-... [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@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