[Ffmpeg-devel-irc] ffmpeg.log.20140606

[burek]

[00:34] <rawrmonster> I am making a script to convert all video files to mpeg2 to be burned to dvd but what is the best flag to keep pretty much the exact same quality. I know i cant just copy the streams because some of the codecs in some containers are not compatible
[00:35] <rawrmonster> i have looked on google but all i am seeming to get is a bunch of people that have no idea what they are talking about because the flags they are suggesting are not in the ffmpeg man page
[00:48] <bencc> how can I check if a video is mpeg4 or h264?
[00:48] <c_14> bencc: ffprobe it
[00:49] <sacarasc> Obligatory pedantry: H264 is a type of MPEG4.
[00:50] <bencc> I have several videos that don't play on Chrome on Windows 8
[00:50] <bencc> but play on Chrome on Windows 7
[00:50] <bencc> all are .mp4
[00:50] <bencc> http://html5test.com/ says chrome supports h264 but not mpeg4
[00:51] <pzich> a square is a rectangle, but not all rectangles are square
[00:53] <bencc> http://dpaste.com/30SM2QN/
[00:53] <bencc> this is what ffprobe gives me. I don't understand why it won't play on Chrome on Windows 8
[01:02] <sacarasc> bencc: Maybe ask Google?
[01:02] <sacarasc> I mean the company, not the search engine.
[01:09] <bencc> sacarasc: when disabling video hardware acceleration on Chrome the video plays
[01:09] <bencc> so it's a chrome bug
[04:09] <beekar> Okay, I'm stuck.  I'm trying to stream using:   ffmpeg -i *Pokopo.avi  -async 1 -ss 00:00:05 -tune zerolatency  -s 480x360 -c:v libx264 -c:a libfaac -re -f flv    with an rtmp string behind it.  Anything I've tried ends with a framerate of like 80-160.
[04:10] <beekar> the -re doesn't seem to work, neither does the -r 30.
[04:10] <sacarasc> Add -re to the input
[04:10] <sacarasc> Not output.
[04:12] <beekar> dang.
[04:12] <beekar> i owe you a thank-you and a handjob.
[04:14] <llogan> beekar: consider using VBV with -maxrate and -bufsize (and do you really need -async 1?)
[04:16] <beekar> i probably do not.  i had gotten into the mode of randomly trying things.
[04:17] <beekar> a last ditch attempt before asking for help.
[04:24] <llogan> beekar: for more info http://trac.ffmpeg.org/wiki/EncodingForStreamingSites
[04:24] <beekar> ooh yeah that page might have saved some time.  nice.  Thankya.
[04:27] <llogan> beekar: and -vf scale=480:-2 is the lazy method
[08:28] <sla3k> Hi people, could any one please let me how can I convert .mp4 (playable on the phone but not in a car DVD player) to .avi using ffmpeg so that it plays in th car player. PS, I use a pen drive to play stuff in car player.
[08:33] <sacarasc> What does your car player support?
[08:33] <relaxed> sla3k: ffmpeg -i input.mp4 -c:v mpeg4 -q:v 3 -vtag XVID -c:a libmp3lame -ac 2 -b:a 128k out.avi
[08:34] <sacarasc> That would probably work, thoguh.
[08:34] <sla3k> sacarasc: I am not sure, just that it does not play .mp4 with error (Format not Supported), relaxed: Thank a lot for that, going to try that.
[10:55] <anarsoul|2> hi
[10:56] <anarsoul|2> I'm trying to fix audio playback in ffmpeg tutorial (https://github.com/chelyaev/ffmpeg-tutorial/blob/master/tutorial07.c)
[10:56] <anarsoul|2> it doesn't work properly for some reason for most stereo files
[10:57] <anarsoul|2> I've spent some time with gdb and it appears that avcodec_decode_audio4() allocates less memory for a buffer than required
[10:58] <anarsoul|2> so could anyone give me some tips for further debugging, 'cause I'm not familiar with ffmpeg internals
[11:22] <anarsoul|2> ah, got it, audio format is planar, so there're several planes...
[15:27] <RoyK> hi all. I have some output files from a Lenel CCTV system. They're all 100MB, and I suspect it can be some sort of MPEG-TS. Any idea how to find out?
[15:29] <sacarasc> ffprobe -i file
[15:55] <aaronds> Hi all, I'm getting an error when trying to convert .wmv to .webm "encoder setup failed". This command works fine when the input file is of type .mov, but not .wmv. Full command and output here: http://pastebin.com/mGAzvatT Is anyone able to help? I'm sure this worked on an older version of ffmpeg
[15:57] <c_14> Pretty sure you aren't supposed to have a '\' between the 480 and the ','
[16:00] <aaronds> c_14: the \ is necessary. Without it you'd get a "No such filter: 'iw'"
[16:01] <aaronds> The earlier build I spoke of seems to be from February this year
[16:01] <c_14> If it works with an earlier build, it might be a regression.
[16:06] <c_14> Could you maybe try doing a bisection to see which commit brings in the error?
[16:07] <aaronds> c_14: sorry how exactly can I do that?
[16:08] <c_14> If the encoding doesn't work with current git head, but does work with the February release, try encoding with ffmpeg versions from between those time points until you find out at which point it breaks.
[16:11] <c_14> Once you know that you can open a ticket with the problem and the faulty commit. If you don't have the time to do a full bisect, you can also open an issue saying that it worked with the February version (and include the version number) but doesn't work with current git head. It would also be helpful if you included of a sample video and the command that doesn't work so the devs can test it.
[16:56] <aaronds> c_14: well I've got a workaround for now, using q:a 5 instead of b:a 128k works fine
[18:01] <Gregor> OK, looking for an ffmpeg guru answer: I have a bunch of files recorded on the same device at different times, and I have the timestamp each file was recorded. I want to create a file which contains all of the video with the appropriate PTS corresponding to the real timestamps. If I concat them together, the PTS are wrong because there are gaps between the videos. If I set their PTS in a filterchain before concatting, I have to reencode. Is there a way to do this
[18:01] <Gregor>  without reencoding?
[18:19] <Plorkyeran> you could create filler videos of the correct length to stick in each of the gaps between interesting segments
[18:19] <Plorkyeran> but that's kinda clunky
[18:20] <Plorkyeran> proper discontinuous gaps requires an output container format that supports those
[18:20] <Plorkyeran> and a muxer which supports them (and I have no idea if any of ffmpeg's muxers do)
[18:21] <Plorkyeran> for containers supporting vfr you could simply have the last frame of each segment last until the beginning of the next one
[18:40] <Gregor> Plorkyeran: Actually, filler videos is as clunky as it is perfect for my circumstances :)
[18:40] <Gregor> I could just take the first frame of the next video and set its framerate to something appropriately stupid.
[18:40] <Gregor> (All the formats I'm using support vfr, btw, so that much should be a nonissue)
[18:40] <Gregor> Thanks!
[19:07] <max246> hello
[19:08] <max246> I was looking in the documentation how to make spacing between letters when I draw a text on a video
[19:08] <max246> but I couldn-t find any parameters that will do it
[19:08] <max246> is ffmpeg able to change  spacing?
[19:24] <hi117> what is the flag for ffmpeg that muxes video with the mp4 streaming hack?
[19:25] <c_14> -movflags +faststart
[19:27] <hi117> ah ty
[20:31] <DeadSix27> totally weird question: If you had to store a video for a long time, up to 10years, would you use dts or flac?
[20:35] <Plorkyeran> crazy unexpected legal developments would be the only reason for it to matter
[20:35] <Plorkyeran> and those are rather difficult to predict usefully
[20:43] <DeadSix27> Plorkyeran, lost conn, did i miss anything?
[20:55] <llogan> DeadSix27: "Plorkyeran | crazy unexpected legal developments would be the only reason for it to matter"
[20:55] <llogan> " and those are rather difficult to predict usefully"
[20:59] <DeadSix27> ah
[20:59] <DeadSix27> well i got that parts
[20:59] <DeadSix27> thought something else showed up meanwhile
[20:59] <DeadSix27> thanks anyway :)
[21:29] <rsevero> Hi. I'm having sync problems in ffmpeg when capturing audio and video simultaneously. I this example: http://pastebin.com/iWCVLeQQ I got a 5 seconds delay between audio and video. This 5 seconds is exactly the difference of the stat: field of both input streams. My problem is that this difference changes each time I call ffmpeg. How can I remove this difference or automatically correct it?
[21:29] <rsevero> "This 5 seconds is exactly the difference of the start: field of both input streams."
[23:30] <theekoz> Is there anyway using -newaudio to add it to an existing transport stream maintaining ts characteristics without remuxing
[23:30] <theekoz> or using any command to get that outcome
[23:40] <llogan> theekoz: there is no -newaudio command anymore. use -map and -codec.
[23:40] <llogan> s/command/option
[23:41] <theekoz> thanks
[23:43] <theekoz> Do you know if there is anyway to prevent FFMPEG from remuxing and just add the audio to the transport stream. Losing all my PID'
[23:43] <theekoz> PID's from the original TS when adding audio via -c copy
[00:00] --- Sat Jun  7 2014