[00:34] <active8> http://pastebin.com/jAnmAzAa - x11grab - Non-monotonous DTS in output stream 0:0; 4 examples with the multitude of warnings snipped to a few. 1. no audio playing. 2. VLC playing audio. 3. Audio coming from a browser video. 4. Full screencast with the browser video playing. [00:38] <llogan> what if you use -c:a copy instead? i wonder if using libpulse instead of alsa would make any difference. [00:40] <llogan> you would need to build with --enable-libpulse though [00:42] <relaxed> it appears he is using pulse [00:42] <llogan> i meant use -f pulse, not -f alsa [00:43] <okokokno> Hmm... Even though I now did avformat_find_stream_info, I still don't see some info I need for mp3 files [00:43] <okokokno> For example, how many packets there are in the file [00:43] <active8> it's not in the command output so I uess I would. ffmpeg -formats would output libpulse? [00:44] <llogan> no, because your build configuration lacks --enable-libpulse [00:44] <llogan> http://ffmpeg.org/ffmpeg-devices.html#pulse [00:44] <okokokno> Or can that be computed from the duration somehow? [00:44] <active8> but also notice that in the full screencap output, the message is all about the video stream: [matroska @ 0x2224460] Non-monotonous DTS in output stream 0:1; [00:45] <okokokno> (AVStream::nb_frames is still 0) [00:45] <llogan> active8: all this time i was thinking of "xruns". [00:46] <llogan> trying to do too much as once and i suck at multitasking and reading [00:46] <active8> that's what I meant by "output" --enable-libpulse isn't there [00:46] <relaxed> active8: you're using a 2013 build, which is pretty old [00:48] <active8> relaxed: roughly how often do they update ffmpeg. I'm not exactly ready to rebuild. Just asking. [00:48] <relaxed> every three months - or in git everyday [00:49] <active8> ffmpeg version git-2013-08-04-61af627 built on Feb 19 2014 04:33:30 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4) [00:49] <active8> funny. i pulled down those sources in feb and they were 6 months old [00:50] <relaxed> active8: try again with output.flac [00:51] <active8> same command, diff extension? [00:51] <relaxed> yes [00:51] <relaxed> loose -threads 0 [00:51] <relaxed> lose* [00:53] <active8> unsupported codec [00:54] <relaxed> pastebin [00:55] <active8> http://pastebin.com/JKx2rApg [00:56] <relaxed> active8: ffmpeg -f alsa -ac 2 -i pulse output.flac [00:58] <active8> now it's a red message : http://pastebin.com/6a0PK8Ji [00:59] <relaxed> what was the exit status? [01:00] <Hfuy> hihi [01:01] <Hfuy> ack [01:01] <Jan-> Problem: we have some AVC-Intra MXF files which seem to have eight streams of audio. [01:01] <Jan-> We're converting them (successfully) to ProRes quicktimes but the output files only have one mono audio channel in them. [01:02] <Jan-> Here's the ffmpeg analysis of the files, input first: http://pastebin.com/jwiSR6SR [01:02] <active8> relaxed: 2 if $! is exit status - can't remember [01:03] <active8> i mean $? [01:03] <relaxed> exit status is $? [01:03] <active8> linux [01:03] <Jan-> there should presumably be some way we can create stereo output files (using map?) but I'm not sure what the syntax would be. Any suggestions [01:03] <Jan-> ...would be most appreciated :) [01:03] <active8> the exit is at the bottom of the paste [01:03] <c_14> Jan-: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [01:03] <active8> exit is 2 if $? is exit status [01:03] <kepstin-laptop> Jan-: http://www.ffmpeg.org/ffmpeg-filters.html#amerge also [01:04] <relaxed> ok [01:04] <pzich> Jan-: have you been through https://trac.ffmpeg.org/wiki/AudioChannelManipulation ? [01:04] <relaxed> Jan-: you want to include all the audio streams and change them to stereo? [01:04] <active8> IIRC. I had this, or at least a messed up seek on VLC when making a slideshow. The time in VLC would be 00:00 - but later conversion from mpeg to mp4 or just mpeg to mpeg through c:v copy fixed it. I'd still like to not see the meaasage and don't really know what it means. Seems ffmpeg would need to guess the timestamps from the sample rate [01:04] <Jan-> As far as we know the input should only be stereo. [01:05] <Jan-> So there are probably only two useful channels of interest [01:05] <relaxed> Jan-: you're build is too old and that could be the problem [01:05] <relaxed> your* [01:05] <Jan-> we'd like to figure out which they are (probably by spitting out wave files to analyse which input streams we want) then put those into the stereo output. [01:06] <pzich> Jan-: sounds like you need -map_channel [01:06] <Jan-> uhhuh [01:07] <Jan-> The problem we found is that a lot of the examples deal with muxing wave files into videos to create multipoint surround etc [01:07] <llogan> or perhaps play in VLC and choose each individual stream. i imagine they would be in the same order as shown by ffmpeg [01:08] <Jan-> Oh hm you can do that [01:08] <Jan-> wait one we'll figure it out [01:08] <Jan-> I would be amazed if we didn't just want 0 and 1 [01:08] <Jan-> or rather 1 and 2, as 0 seems to be the video [01:08] <Jan-> MXF is such a PITA :/ [01:09] <Jan-> yeah, it's just 1 and 2 [01:10] <Jan-> for some reason VLC is interpreting the eight soundtracks as independent mono tracks, as opposed to assuming 1 and 2 are the stereo pair. [01:10] <Jan-> ffmpeg does the same [01:10] <llogan> -filter_complex "[0:a:0][0:a:1]amerge" -ac 2 [01:10] <llogan> or something like that [01:10] <relaxed> it could be a decoding issue. are you sure there are stereo streams? [01:11] <Jan-> wait one [01:12] <pzich> it may need to be [0:a][1:a] [01:12] <Jan-> llogan: Perfect [01:12] <pzich> oh nm, misread the output [01:12] <Jan-> man MXF Is horrible [01:12] <Jan-> also, how would we ever have figured that out :/ [01:13] <llogan> you may want to compare using -ac 2 with the pan example in https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a2stereostereo [01:13] <Jan-> Honestly it's just a simple channel assignment issue [01:13] <Jan-> your filter_complex line does it for a wave file [01:13] <Jan-> we just have to integrate it into a script [01:14] <llogan> i assumed pan and -ac 2 would provide the same output, but they don't for me [01:14] <Jan-> C:\Users\CobaltUser>c:\ffmpeg -i "E:\Roll 1\XDROOT\Clip\120_0017.MXF" -filter_co [01:14] <Jan-> mplex "[0:a:0][0:a:1]amerge" -ac 2 d:\test.wav [01:14] <Jan-> ...has the desired effect [01:14] <Jan-> where does 0:a:0 come from [01:15] <pzich> 0:a:0 would be channel 0 of the first audio stream [01:15] <Jan-> oh, right [01:15] <kepstin-laptop> in your ffmpeg output, there's a bunch of Stream # lines [01:15] <Jan-> there is [01:16] <kepstin-laptop> I think using [0:1] and [0:2], matching the stream lines in the output, would work as well? [01:16] <Jan-> fyi this MXF is the output of a Sony PXW-X180 camcorder in AVC-Intra 1080p25 mode. [01:16] <llogan> yes, but it's lazier to use 0:a:0, etc. [01:17] <Jan-> What does it mean by [mxf @ 00000000024d77e0] could not resolve sub descriptor strong ref [01:17] <Jan-> is that some weird mxf container stuff [01:17] <llogan> i need a new video camera... [01:17] <Jan-> don't get a pxw-x180 :) [01:18] <llogan> maybe i'll just start renting, but that's a bitch where i live [01:18] <Jan-> this is a review loaner. [01:18] <Jan-> and we scratched the display on one of the radio mic receivers :/ [01:19] <llogan> it was like that when you got it [01:19] Action: Jan- nods vigorously [01:20] <Jan-> sony's much-advertised "AVC Intra" just seems to be 10-bit h.264 with all I frames [01:20] <Jan-> at craptons of bitrate [01:21] <Jan-> what does it mean by "input channel layouts overlap" here: http://pastebin.com/Vz7xp2S9 [01:22] <kepstin-laptop> Jan-: both input channels are 'mono', rather than one being 'left' and one 'right' [01:22] <kepstin-laptop> Jan-: in this case it does the right thing, it assumes since there's two, that you meant stereo [01:22] <Jan-> fair enough [01:23] <Jan-> I wasn't aware a an audio channel could be marked with a position [01:23] <kepstin-laptop> well, an audio stream is marked with a layout [01:23] <Jan-> does this imply that the MXF isn't properly marking its channels [01:23] <Jan-> (and wtf is it doing with 8 channels of audio anyway?!) [01:24] <llogan> http://ffmpeg.org/ffmpeg-filters.html#amerge describes the input layout stuff [01:24] <Jan-> ffmpeg is a very very big piece of software isn't it... [01:27] <Jan-> Ah. I can't have -c:a copy with this audio filtering [01:28] <kepstin-laptop> no, but -c:a pcm_s24le will pass it through with no further encoding. [01:28] <Jan-> will just omitting the -c:a do the same [01:28] <kepstin-laptop> depends on the output format, some may autoselect a lossy codec. [01:29] <relaxed> probably not, be spcific [01:29] <Jan-> hm no I get AAC in that case :) [01:29] <Jan-> snksnksnk a prores quicktime with 128kbps AAC audio [01:29] <Jan-> that's hilarious [01:29] <kepstin-laptop> using ffmpeg's not particularly good aac encoder too, I bet ;) [01:29] <Jan-> woah, this camera uses 24bit audio!? [01:30] <llogan> Jan-: did you ever get to mess around with a Canon EOS C100 to C500? [01:31] <Jan-> a C300, and not me, my boyfriend [01:31] <Jan-> why d'you ask [01:32] <llogan> what did he think of it? i may rent one for a week for a project (and learn how to use it on the plane). [01:32] <Jan-> he says "incredibly nice sensor, shame about the gimped 8 bit output" [01:32] <Jan-> and "what're you gonna do for glass" [01:34] <llogan> more rental, but i haven't decided which lens(es) yet. i need to find more about the actual project if it's mostly interviews or more outdoor. [01:35] <Jan-> canon know how to make nice sensors [01:36] <Jan-> this fault had us worried though [01:36] <Jan-> suddenly both channels appear to be the same audio and neither of them is the radio mic [01:36] <Jan-> eeeeek [01:37] <Jan-> By the way, who's this famous Kostya guy who did the prores encoder? [01:37] <llogan> there are 2 prores encoders. [01:38] <Jan-> Yeah. [01:38] <Jan-> prores_ks seems to work quite a lot better, but it's a bit slower [01:38] <Jan-> the normal "prores" one creates files that go way over bitrate (though few things seem to care) [01:39] <kepstin-laptop> http://codecs.multimedia.cx/?p=388 has some discussion on speed [01:39] <Jan-> we even put some of the non-ks prores files on hardware recorders [01:39] <Jan-> and they played them fine [01:43] <Jan-> "probesize" tells it how much of the video to look forward at when estimating bitrate, right? [01:44] <Soul_keeper> I use it for streams that arent detected at the begging [01:45] <kepstin-laptop> and in some formats (mpeg ts in particular) to find streams that don't start at the start of the file [01:45] <Jan-> oh hm [01:45] Action: Jan- runs "convert.bat" and goes to bed [01:45] <Jan-> is "could not resolve sub descriptor strong ref" bad? [04:47] <fluke_> Hi every one. I want to ask how to calculate network speed while playing http video with ffplay? [04:47] <fluke_> My thought is to calculate in http.c [11:47] <fswings> Morning, how do I extract the DTS core from a DTS-HD MA track using ffmpeg? Or are there alternative Linux based tools which are recommended? [11:47] <fswings> I'm running via command line. [13:29] <okokokno> Can I somehow get the info from the XING header in an mp3 via libavformat? [13:38] <krasnayarsk> Hi. I have an avi video and its been converted from mp3. There is no image when listening/watching the avi file. What is the correct/best way to add a still image, perhaps a series of images, to the sound in the avi file? [13:40] <c_14> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images [13:42] <krasnayarsk> c_14: Thank you. [13:43] <krasnayarsk> part [15:27] <phw> Can ffmpeg play one file with video being redirected but audio playing locally? [15:28] <ubitux> try to map a stream to an alsa output or something [15:28] <ubitux> and map the video stream to another output [15:29] <ubitux> your question is weird though [15:30] <phw> well i am using piwall to display a video, while playing the same videos audio [15:32] <phw> This results in "ffmpeg -re -i big_buck_bunny_720p_surround.avi -vcodec copy -f avi udp://239.0.1.23:1234" to broadcast the video via stream [15:33] <phw> but i want ffmpeg to simultaniously play the audio of the file [18:04] <Joryn> I'm looking to get some of the functionality in the AC3 decoder from FFMPEG into a gstreamer application that I'm working on (specifically recent dialnorm and DRC changes), but not sure how compatible the current version is with gstreamer-1.0. What do I need to know? Thanks! [18:19] <lipizzan> is there a way to "tee" ffmpeg output to both write to ts stream to file disk and STDOUT (or socket?) as input to mplayer? [18:24] <kepstin-laptop> sure, have ffmpeg write to stdout, and pipe it to the 'tee' command to save a copy to disk [18:24] <kepstin-laptop> then pipe that to mplayer. [19:35] <c_14> lipizzan: https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Teepseudo-muxer [20:30] <jelle__> I have a origin.mkv with 1 fps and I would like to keep all the frames, but play them 60 fps. How do I do this? Keep all frames, but speed them up? [20:36] <__jack__> strange idea [20:40] <__jack__> you can still extract all frames as image, then copy all of them 60x, and recreate the movie [21:12] <jelle__> have a origin.mkv with 1 fps and I would like to keep all the frames, but play them 60 fps. How do I do this? Keep all frames, but speed them up? [21:13] <c_14> ffmpeg -i origin.mkv -r 60 out.mkv ? [21:14] <jelle__> no filters needed? [21:15] <c_14> shouldn't [21:19] <jelle__> ok, Ill fiddle with that, thanks [21:42] <Diogo> Hi this is possíble to generate hls m3u8 with fixes extinf? [21:49] <jjohn> Ahoy, guys. Just a quick question. [21:50] <jjohn> I got raw pcm_s16le streams and want to put them into a lossless container format. I read that FLAC might be appropiate, but the bitrate of the converted files is lower than that of the original ones. [21:50] <jjohn> I thought FLAC was lossless, wasn't it? Why is the bitrate lower then? [21:50] <c_14> pcm is uncompressed [21:50] <c_14> FLAC is compressed [21:51] <jjohn> But still lossless? And it's the ONLY difference? [21:51] <c_14> It's not the _only_ difference, but it is still lossless. [21:51] <c_14> Ever take a big file and compress it in rar/zip/7z/gz/bz2/xz ? [21:52] <c_14> Same concept. [21:52] <jjohn> No differences in quality either? [21:52] <jjohn> OK, I got it. Thank you very much then. [21:52] <c_14> No difference in quality (as long as the encoder doesn't start doing weird shit) [21:52] <jjohn> Thanks. You have been always a great help to me. [21:52] <jjohn> Bye then. [22:52] <ackwood> Hi everyone. What is "AVCODEC_MAX_AUDIO_FRAME_SIZE" known as these days? [22:52] <ackwood> Last time I remember being able to use it I had ffmpeg 0.8 [23:04] <Plorkyeran> it was removed entirely, not renamed [23:05] <ackwood> Plorkyeran, that sucks. I have a framework that depends on the existence of that variable. Is there a program way of coming up with that number? [23:05] <ackwood> I'm not exactly an expert on multimedia encdec, I'm just trying to maintain something useful [23:06] <Plorkyeran> it was removed because anything that used it was inherently broken and needs to be rewritten to do things correctly [23:07] <Plorkyeran> iirc doing so was fairly easy [23:07] <Plorkyeran> other than the utter lack of useful documentation... [23:09] <goulard> anyone know the rule of thumb on how many AAC ADTS packets to put in one TS packet? [23:14] <Sokolio> You mean PES packet? [23:14] <goulard> yeah [23:15] <goulard> I put one ADTS packet per PES, and quicktime and vlc wont play [23:15] <Sokolio> I guess there's no rule, even a rule of thumb [23:18] <Sokolio> PTS however must be set right according to sample rate [23:18] <Sokolio> does vlc complain in any way? [23:18] <goulard> it just says there is nothing to play [23:18] <goulard> i suppose the pts is correct, as the duration in quicktime is correct [23:19] <goulard> and media info reports properly [23:19] <goulard> pts = frame# * 1024 / samplerate [23:21] <Sokolio> does quicktime actually support adts packaging? [23:21] <goulard> yes [23:21] <goulard> you have to contain AAC in ADTS in TS [23:21] <Sokolio> in qt there is a 2 byte config [23:21] <Sokolio> and raw AAC [23:21] <Sokolio> yes, in TS that;s true [23:22] <Sokolio> well, I have no idea [23:23] <Sokolio> I've seen a single ADTS packet per PES and it sorta played [23:24] <goulard> do you know what buffer fullness is for adts? [23:24] <goulard> I had wondered if that had something to do with it [23:24] <Sokolio> you mean the header field [23:25] <Sokolio> well, I usually hardcoded it [23:25] <Sokolio> dunno if it wasnt all 1's in my case [23:27] <Sokolio> 7ff to be specific [00:00] --- Sat Sep 20 2014