[Ffmpeg-devel-irc] ffmpeg.log.20150618

[burek]

[00:01:27 CEST] <Nolski> ahh i see
[02:10:06 CEST] <podman> hey guys, anyone know what would cause me to get an error like this? "Input picture width (640) is greater than stride (0)"
[05:01:14 CEST] <Tecan> not sure why but there is issues converting wma files, sounds staticy
[05:01:25 CEST] <Tecan> vlc playes them nice
[05:01:59 CEST] <durandal_1707> sample, what codec?
[05:02:19 CEST] <Tecan> ffmpeg -y -i 01a01.wma -f wav out.wav
[05:02:28 CEST] <Tecan> i was trying to play some mozart one sec
[05:04:12 CEST] <Tecan> trying to compile mplayer to see if that works
[05:05:03 CEST] <kepstin> in theory, mplayer, vlc, and ffmpeg command line tool are all using the same decoder from ffmpeg, maybe just different versions of it
[05:06:39 CEST] <chungy> yeah they should all be using libavcodec for it, although depending on a few things (mainly your OS), you might be using a Libav version instead of FFmpeg.
[05:10:30 CEST] <Tecan> http://www.xup.in/dl,14782458/01a01.wma.7z/  <- educational use only
[05:10:38 CEST] <Tecan> delete after testing ;)
[05:13:48 CEST] <chungy> are you able to provide another file to demonstrate what it should decode to (wav or flac would be best)
[05:14:12 CEST] <Tecan> with the static ?
[05:14:36 CEST] <chungy> the normal decoding, without static I should assume
[05:14:51 CEST] <Tecan> its probably just too high of volume
[05:15:03 CEST] <Tecan> or the gain is too high ?
[05:15:25 CEST] <Tecan> the normal decode has the static, one sec
[05:16:02 CEST] <chungy> doesn't seem staticy for me, with ffmpeg 0.7
[05:16:17 CEST] <chungy> 2.7, I mean.
[05:16:41 CEST] <Tecan> ffmpeg version N-71229-gacbb904
[05:16:52 CEST] <Tecan> wonder how old it is
[05:19:47 CEST] <chungy> I kind of wish more typical git describes were used, yeah...
[05:20:07 CEST] <chungy> $ git describe N-71229-gacbb904
[05:20:07 CEST] <chungy> n2.7-dev-740-gacbb904
[05:20:14 CEST] <chungy> a bit before the final 2.7 release.
[05:20:30 CEST] <c_14> Apr 2, This year
[05:22:21 CEST] <chungy> Tecan: random suggestion since I had it happened to me recently, make sure your system volume isn't set over 100% :P
[05:22:44 CEST] <chungy> I accidentally had mine set to like 140% recently and almost thought my speakers were going bad because everything was clipping
[05:23:06 CEST] <chungy> still don't know how that happened accidentally, haha
[09:19:33 CEST] <rweichler> eff eff em peg
[10:01:05 CEST] <damnlie> hiii, do you guys have documentation about how can i set up a separate ffmeg server, i have a separate webserver (where apache en this sort of things are running) and a separate staging server. I never used ffmeg so im a bit lost
[10:18:51 CEST] <BtbN> ffmpeg server?
[10:18:56 CEST] <BtbN> you mean ffserver?
[10:20:04 CEST] <damnlie> i think so :) my coworker installed the whole thing from git so i even do not find the config files... :/
[10:27:51 CEST] <damnlie> http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu#ffmpeg i found this but im not get what de path are?
[10:42:44 CEST] <Polochon_street> hello
[10:43:28 CEST] <Polochon_street> does anyone know why sometimes ffprobe shows duplicate titles with semicolon?
[10:43:44 CEST] <Polochon_street> lik this:
[10:43:47 CEST] <Polochon_street>     TITLE           : Controlling Crowds;Controlling Crowds
[10:44:38 CEST] <Polochon_street> (but maybe this is a dev question?)
[13:35:06 CEST] <gurparta> I am facing problem while saving stream from ip camera and capturing audio from laptop microphone.there are synchronization issues in it? Please somebody help.
[13:36:07 CEST] <gurparta> this is commandline:  ffmpeg  -f alsa  -ac 2  -async 1 -i default   -i rtsp://192.168.0.127   acodec libfaac -v "debug"  -y -vcodec copy  output.mp4
[13:37:39 CEST] <gurparta> Whiskey:  this is commandline:  ffmpeg  -f alsa  -ac 2  -async 1 -i default   -i rtsp://192.168.0.127   acodec libfaac -v "debug"  -y -vcodec copy  output.mp4
[13:41:03 CEST] <gurparta> relaxed:  this is commandline:  ffmpeg  -f alsa  -ac 2  -async 1 -i default   -i rtsp://192.168.0.127   acodec libfaac -v "debug"  -y -vcodec copy  output.mp4
[15:49:35 CEST] <proserpine-> hi. is there a way for ffmpeg to cut out frames where the video is still? i'm trying to edit very long camera footage but it's quite tedious.
[15:53:28 CEST] <saste> proserpine-, select filter, check the scene_change parameter
[15:54:12 CEST] <proserpine-> saste: excellent, exactly what i needed. thank you
[16:00:42 CEST] <ads34123423> Quick question, is an mpeg transport stream containing an H.264 encoded video stream likely to contain H.264 NAL units complete with start codes (0x000001), or would they be redundant and not used?
[16:48:26 CEST] <Mavrik> afaik TS streams have to use AnnexB format
[20:22:39 CEST] <ldiamond1> I want to use ffmpeg for twitch.tv streaming. I want to add a webcam overlay to the stream. So far it's all easy and there's plenty of examples online. However, I want to know if it's possible to send commands to the ffmpeg process to "alter" options, such that I can move the webcam overlay around and remove/add it on the fly (for example if it's obstructing the content).
[20:37:12 CEST] <daan_> Hello! I have ffmpeg installed on my Raspberry Pi but my terminal gives me as output: Failed to set value '[1:v]scale=1920:1080,setdar=16/9,boxblur=10[bg];[1:v]scale=1080:1080[fg];[bg][fg]overlay=(W-w)/2:(H-h)/2[v]' for option 'filter_complex' how can i fix it?
[20:49:04 CEST] <lucascastro> I'm trying to setup ffserver, but when I stream my /dev/video0 to the server, on server show /feed1.ffm not found
[21:22:35 CEST] <BW^-> ffmpeg -i 00058.MTS jake.mts -pass 3 -b:v 4000k -acodec copy
[21:22:55 CEST] <BW^-> for audio input it says: "   Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s"
[21:23:17 CEST] <BW^-> for the output though it says "  Stream #0:1 -> #0:1 (ac3 (native) -> mp2 (native))"
[21:23:24 CEST] <BW^-> and "Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s"
[21:23:29 CEST] <BW^-> so that clearly looks like recoding to me!?
[21:24:50 CEST] <BW^-> hmm with the latest FFMPEG, I have a     "Stream #0:0: Video: mpeg2video, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 50 fps, 90k tbn, 50 tbc" 26383kb/sec input stream that takes ~150MB per minute
[21:25:00 CEST] <BW^-> I just want to get it down to maybe 15-30MB per minute
[21:25:15 CEST] <BW^-> what codec do you suggest for preserving maximum quality?
[21:25:50 CEST] <BW^-> um.. wait.. no this is my input stream: "    Stream #0:0[0x1011]: Video: h264 (High) (HDPR / 0x52504448), yuv420p, 1920x1 080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc"
[21:26:24 CEST] <BW^-> now i just did "-pass 3 -b:v 4000k" and that generates a less than superb output
[21:27:41 CEST] <BW^-> aha let's try "-c:v libx264 -crf 22 "
[21:34:08 CEST] <BW^-> :D
[21:34:10 CEST] <BW^-> yey was perfet!
[21:34:18 CEST] <BW^-> ok this is good enough so no need to go into details - thanks :))
[22:29:38 CEST] <anoop_r> hi
[22:30:31 CEST] <anoop_r> any plans to include Apple AAC support for ffmpeg
[22:37:20 CEST] <durandal_1707> anoop_r: what? Is it open source?
[22:37:30 CEST] <JEEBsv> nope
[22:38:26 CEST] <durandal_1707> then answer is simply NO
[22:38:31 CEST] <JEEBsv> it has an API, to use which you have to install the API parts of QT. It is not GPL compatible anywhere else than on OS X (because there QT is a system component), and I don't see it being really better than fdk-aac
[22:40:02 CEST] <anoop_r> thanks
[22:50:01 CEST] <gchristensen> Hi, if someone wanted to donate infrastructure, is that something which would be useful?
[22:50:47 CEST] <durandal_1707> Absolutely yes
[22:50:58 CEST] <gchristensen> what would be beneficial, and who do I talk to?
[22:51:44 CEST] <durandal_1707> Ask in -devel
[23:16:15 CEST] <anoop_r> any way to add  	FAAC to ffmpeg
[23:22:27 CEST] <durandal_1707> decoder?
[23:38:01 CEST] <anoop_r> encoder
[23:42:41 CEST] <durandal_1707> IIRC it is already there
[23:57:20 CEST] <anoop_r> anyway to add QAAC to ffmpeg
[00:00:00 CEST] --- Fri Jun 19 2015