[Ffmpeg-devel-irc] ffmpeg.log.20140613

[burek]

[00:00] <sanmarcos> ok
[00:00] <sanmarcos> formats now shows avfoundation
[00:00] <sanmarcos> it must be only in git
[00:02] <sanmarcos> http://pastie.org/9284895
[00:03] <sanmarcos> getting : Input/output error on listing devices
[00:03] <sanmarcos> both with qtkit and avf
[00:03] <sanmarcos> apparently it doenst support audio
[00:03] <sanmarcos> only video?
[00:04] <MDTech-us_MAN> c_14
[00:04] <llogan> nothing from: ffmpeg -f avfoundation -list_devices true -i ""?
[00:04] <MDTech-us_MAN> I'm confused
[00:05] <MDTech-us_MAN> is -ss the length from -t or length from the beginning of the audio
[00:05] <c_14> -ss seeks from the beginning of the file to the point specified.
[00:06] <MDTech-us_MAN> -ss is where to start
[00:06] <MDTech-us_MAN> cutting
[00:06] <c_14> yes
[00:06] <MDTech-us_MAN> what about -t ?
[00:06] <sanmarcos> http://pastie.org/9284899
[00:06] <sacarasc> How long to cut for, MDTech-us_MAN.
[00:06] <MDTech-us_MAN> Thats it!
[00:06] <MDTech-us_MAN> I was getting these very long files
[00:06] <MDTech-us_MAN> no I understand
[00:06] <MDTech-us_MAN> *now
[00:07] <MDTech-us_MAN> I was specifing where to stop instead of the length
[00:07] <sacarasc> To do that, you use -to
[00:13] <johny-b-goode> hello folks. I'm trying to extract a video out of a mp4 file. Is the following not the correct format? (it gives me an error).
[00:13] <johny-b-goode> ffmpeg -i Friends_The\ One\ With\ the\ Boobies.dvr-ms.mp4 -vcodec copy video.mp4
[00:14] <johny-b-goode> it generates an error: aac @ 0x9422900] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
[00:14] <c_14> add -an
[00:14] <c_14> If you only want the video that is.
[00:14] <johny-b-goode> c_14: does mean audio none?
[00:14] <c_14> ye
[00:15] <johny-b-goode> ok, thank you. that worked. :)
[00:16] <johny-b-goode> On my windows machine I extract the audio the same way by doing a -acodec copy without having to specify any video options.
[00:16] <llogan> sanmarcos: i'm not sure if it supports audio. docs don't explicitly mention it. might have to ask on ffmpeg-user or look at code.
[00:17] <c_14> johny-b-goode: ffmpeg by default usually maps one video and one audio track, but with the audio you're probably using a format that only supports audio so ffmpeg throws away the video
[00:17] <llogan> sanmarcos: and i wonder why it is not listed in ffmpeg -devices
[00:17] <johny-b-goode> c_14: that makes sense. I'm extract it to a .wav file.
[00:18] <c_14> If you were to extract it to something like .h264 then it should throw away the audio by itself.
[00:19] <sanmarcos> llogan: i think I Will give up for now
[00:19] <sanmarcos> llogan: file a bug with VLC and have them fix their crash on OS X
[00:20] <johny-b-goode> c_14: I see, I was extract video to mp4 so it probably kept the audio because of that.
[00:20] <johny-b-goode> *extracting
[00:21] <johny-b-goode> lol: [mp4 @ 0x93b0140] Malformed AAC bitstream detected: use audio bitstream filter 'aac_adtstoasc' to fix it ('-bsf:a aac_adtstoasc' option with ffmpeg
[00:22] <johny-b-goode> but the aac was formed by ffmpeg itself by converting from ac3. heh.
[00:30] <DelphiWorld> sup guuys
[00:30] <DelphiWorld> hi c_14
[00:30] <DelphiWorld> that work for some seconds then saying convertion failed
[00:30] <c_14> Can I have some context again?
[00:31] <DelphiWorld> c_14: pushing to rtmp
[00:31] <DelphiWorld> using that cmd
[00:31] <DelphiWorld> it works for some seconds
[00:31] <DelphiWorld> then x264 saying convertion failed
[00:34] <DelphiWorld> c_14: what the best way to redirect ffmpeg log to file?
[00:34] <c_14> -report
[00:34] <DelphiWorld> -report filename?
[00:34] <c_14> ye, without a filename it goes into ffmpeg-date or something
[00:35] <DelphiWorld> hehe, fun one
[00:35] <DelphiWorld> ok c_14 http://pastebin.com/J7izza9S
[00:37] <DelphiWorld> flv complain about malformed aac
[00:37] <c_14> What's your current commandline? Have you tried adding -bsf:a aac_adtstoasc to it?
[00:38] <llogan> DelphiWorld: how many times have you been here, yet you omitted your actual command
[00:38] <DelphiWorld> command:
[00:38] <DelphiWorld> i am changing it, yes
[00:38] <DelphiWorld> .../usr/local/bin/ffmpeg -re -f mpegts -i http://192.168.0.188:4271/bysid/3017 -r 30 -c:v libx264  -crf 18 -profile:v baseline -maxrate 400k -flags -global_header -c:a libfdk_aac -b:a 64k -ar 44100 -f flv rtmp://192.168.0.13:1935/live/sunna2
[00:38] <relaxok> my console output when encoding video, shows the bitrate i passed with -b, but when i check the output file that was created, it's a different bitrate
[00:38] <relaxok> blargh
[00:39] <relaxok> (the original one)
[00:40] <DelphiWorld> here you go c_14, /usr/local/bin/ffmpeg -re -f mpegts -i http://192.168.0.188:4271/bysid/3017 -r 30 -c:v libx264  -crf 18 -profile:v baseline -maxrate 400k -flags -global_header -c:a libfdk_aac -b:a 64k -ar 44100 -bsf:a aac_adtstoasc -f flv rtmp://192.168.0.13:1935/live/sunna2
[00:40] <DelphiWorld> but still shutting down
[00:40] <relaxok> http://pastebin.com/nSriuEiQ
[00:40] <relaxok> wtf?
[00:41] <c_14> DelphiWorld: does it work when encoding to a file?
[00:42] <DelphiWorld> c_14: good pointed, yes its ongoing into a file
[00:43] <c_14> So it only doesn't work with rtmp?
[00:45] <relaxok> ffmpeg.org ha been dying all day for me on and off
[00:46] <DelphiWorld> c_14: yep
[00:46] <llogan> relaxok: the bitrate should be set automatically depending, AFAIK, on your frame size, frame rate, and profile (proxy, LT, standard, HQ, 4444)
[00:47] <c_14> If you were to dump say 1 minute into a file and then use ffmpeg to stream that to the rtmp server with -codec copy, does that work?
[00:48] <DelphiWorld> c_14: mmmm, you're so vissious:P
[00:48] <DelphiWorld> you know what c_14 ?
[00:48] <DelphiWorld> flag
[00:48] <DelphiWorld> global header
[00:48] <DelphiWorld> not accepting it in rtmp
[00:52] <relaxok> llogan: -profile standard should work?
[00:52] <DelphiWorld> -crf is abr?
[00:52] <relaxok> -profile:v standard
[00:53] <relaxok> std
[00:53] <relaxok> i tried everything listed on the ffmpeg-codecs page
[00:53] <relaxok> as a profile for prores
[00:54] <relaxok> e.g. just tried: ffmpeg -i 42496EJ-00001_EJ0007.mov -an -c:v prores -profile:v proxy -y out.mov
[00:54] <relaxok> 'undefined constant or missing ( in proxy'
[00:54] <DelphiWorld> c_14: ok, stream up, now testing with client on hls
[00:55] <relaxok> ah with the integer value it works
[00:57] <llogan> standard is default
[00:58] <DelphiWorld> llogan: all is up now:P
[01:00] <DelphiWorld> llogan: 30fps is recomanded for hls, right?
[01:01] <llogan> i'm not familiar with the HLS specifications
[01:59] <ParkerR> Any idea what could be going on here? http://hastebin.com/itutafuxuv.pas
[01:59] <ParkerR> I've tried a few different audio condecs to no avail
[02:00] <ParkerR> Aha -c;a opus -strict -2
[02:11] <klaxa> if you want something more widely supported try vorbis
[02:11] <klaxa> libvorbis provides more flexible encoding than the native encoder i think
[05:06] <MDTech-us_MAN> hello
[05:08] <MDTech-us_MAN> c_14
[05:08] <MDTech-us_MAN> you there?
[05:09] <sacarasc> Just ask your question, there are more people than just c_14 here often.
[05:10] <MDTech-us_MAN> What would be the best way to do the following: concatinate say... about 20 files I will do something and then split them up again the same exact way
[05:11] <sacarasc> Write down the lengths and do it slowly? :D
[05:11] <MDTech-us_MAN> :|
[05:11] <MDTech-us_MAN> you are kidding?
[05:11] <MDTech-us_MAN> I have to do this many times
[05:11] <sacarasc> Why would you need to do that, may I ask?
[05:12] <MDTech-us_MAN> I want to put everything together and apply ReplayGain to all of them
[05:12] <MDTech-us_MAN> to average all the peaks out
[05:13] <MDTech-us_MAN> I can do it individually, but then the tracks will still be different volume
[05:15] <MDTech-us_MAN> sacarasc, sound ok?
[05:18] <sacarasc> You might be able to script it to make it work better, but I am not sure how.
[05:19] <MDTech-us_MAN> where would I go to get help on this?
[05:21] <MDTech-us_MAN> sacarasc
[10:25] <Darkfang1> Hi
[10:26] <Darkfang1> Quick question: Is it an expected behaviour that x265-params option are ignored without any warning if the syntax is not key=val:key=val? It seems it is the same for x264 btw.
[11:28] <wh-hw> ffmpeg -f video4linux2 -i /dev/video0 -an -vcodec libx265 cam.mkv
[11:29] <wh-hw> hi, all , why this command take  a [30536.203147] ffmpeg[17459]: segfault at 8 ip 00007f482a2ccd25 sp 00007fffaa618008 error 4 in libx265.so.1.1[7f482a1a0000+204000] error
[11:29] <wh-hw> >
[11:29] <wh-hw> ?
[11:29] <wh-hw> any body help me ?
[14:30] <KCLA5555>  Anyone looking for a new apple laptop? I got 2 17" Apple MacBook Pro's 2.3GHz Quad i7 new in box. $500 each MSG ME
[14:30] <sacarasc> KCLA5555: Please do not spam, you'll get banned from the network.
[15:49] <dericed> anyone got a working example of the metadata option in drawtext, i can seem to get it to work?
[16:41] <dericed> for instance this command doesn't show the last pix_fmt value: ffplay -f lavfi -i color=white -vf drawtext="fontfile=/Library/Fonts/Courier New.ttf:text='test %{pts\:hms} %{metadata\:frame=pix_fmt}'"
[17:15] <Moonlightning> sacarasc: what? o.o
[17:38] <tlsa> hi
[17:39] <tlsa> I want to skip certain frames from an input video to create an output video
[17:39] <tlsa> I can read frames with av_read_frame(), and determine whether I want to skip them
[17:40] <tlsa> what's the simplest way of putting the frames I want to keep into an output video?
[17:40] <tlsa> ideally I want to leave the format and codec unchanged
[17:59] <Mavrik> tlsa, you have to reencode the video
[17:59] <Mavrik> so pretty much the full encoding loop
[17:59] <Mavrik> you'll also have to renumber PTS/DTS to plug the holes
[18:09] <iive> tlsa: you can skip frames that are not used as reference. that's mostly b-frames (and no b-pyramid).
[18:11] <tlsa> what do you mean by renumber PTS/DTS?
[18:11] <iive> if you remove i/p frames, everything that depends on them would be broken.
[18:11] <iive> tlsa: he means to smooth it out.
[18:12] <iive> like decimate filter.
[18:13] <tlsa> by frames I mean "images", and I'm considering the "video" to be a series of "images", and I am skipping particular "images" in the output
[18:13] <tlsa> I don't mind that I have to reencode it
[18:13] <tlsa> if you see what I mean
[18:13] <tlsa> so I don't think I need to worry about PTS/DTS
[18:13] <iive> well, i thought you need something like stream copy and cut some parts.
[18:14] <tlsa> yeah, but I can do that the lossy way
[18:15] <iive> well, let's say that you remove every second frame. this would result of halving the fps. but you can pack them together, as result doubling the speed.
[18:15] <tlsa> what I need to do is simply set up an output format/codec that matches the input (same dimensions, same frame rate, ideally same codec)
[18:16] <tlsa> and I'm not sure how do do that
[18:16] <iive> well,  ffmpeg won't change image size and fps without explicit instruction to do that.
[18:17] <iive> the output container is usually guessed by the extension or set by -f option.
[18:17] <iive> as for the codec... i really don't know.
[18:18] <tlsa> I'm not using the ffmpeg program
[18:18] <iive> are you using ffmpeg libraries?
[18:18] <tlsa> I'm using the ffmpeg C libraries (avformat, and avcodec)
[18:18] <tlsa> yeah
[18:18] <iive> then it should be easier to handle that.
[19:23] <waressearcher2> ffmpeg -f concat -i list.txt -i audio.ac3 -ab 320k -ar 44100 -ac 2 -acodec libmp3lame -vcodec h264 -vb 3000k -s 1280x720 -r 30 -vf "fade=out:24900:360" -y -f avi video.avi
[19:23] <waressearcher2> I use that command to make file video.avi and it should be 14:02 long, now if I use mplayer on my PC it shows it is 14:02 long so its correct, but if I use ffprobe to check it's length it says its 15:07 long and also if I upload it on youtube it does show it is 15:07 long and after 14:02 there is black scree, where is the problem ? it should be
[19:24] <waressearcher2>  14:02 long there is no more video, mplayer shows it right, why youtube thinks its 15:07 ? also when I upload it on youtube it says "we can't recognize video codec and audio codec" but they still process the video ?
[19:24] <waressearcher2> or should I say use *.mp4 as output format ?
[19:25] <waressearcher2> s/scree/screen/
[19:27] <c_14> Try adding a -shortest, or a -to 14:02, and maybe pastebin the command and output so we can look at what ffmpeg is saying
[19:32] <waressearcher2> I will recode it again and paste output but it will take few hours
[19:33] <c_14> As long as you're just testing, you could add -preset ultrafast so that it doesn't take as long.
[19:52] <Fyr> how many cores can I utilize with ffmpeg?
[19:52] <c_14> Depends on the codec.
[19:52] <Fyr> libx264?
[19:53] <c_14> All of them.
[19:53] <c_14> Should do so by default.
[19:53] <Fyr> I set 12 cores, but I see only 600% cpu load.
[19:53] <c_14> using -threads ?
[19:53] <Fyr> yeah
[19:54] <c_14> With libx264 you don't have to bother setting threads, it should do that automatically.
[19:54] <Fyr> well, I set manually.
[19:54] <Fyr> on 16-cored system, I see 400-600% workload a swell.
[19:55] <c_14> It depends a lot on what you're doing. If the cpu isn't the limiting factor then you'll see 'low' cpu usage.
[19:55] <Fyr> ok, how to find what limits ffmpeg?
[19:57] <c_14> You'd probably have to check you system stats. It could be the IO.
[19:57] <Fyr> now I have 900% cpu load!
[19:58] <Fyr> IO isn't an option.
[19:58] <c_14> IO as in your disks, if they're too slow reading, writing or seeking.
[19:58] <Fyr> no, it's very fast.
[19:59] <Fyr> how hardware acceleration such as opencl and cuda can increase conversion speed?
[20:01] <c_14> opencl/cuda encoding for h264 is generally considered to be not worth it, but I'm not an expert
[20:07] <c_14> https://trac.ffmpeg.org/wiki/Encode/H.264#FAQ
[20:07] <c_14> 4th question
[20:10] <Fyr> I have two files - original video file and converted one. how to compare their quality?
[20:10] <Fyr> is there a tool to do that?
[20:10] <llogan> i never bothered to benchmark the opencl stuff
[20:11] <llogan> i guess i should. got 800 vids in the queue
[20:11] <c_14> Fyr: the only tool I know of that can give you an accurate analysis of the quality of a video file is the human eye
[20:17] <sfan5> <c_14> opencl/cuda encoding for h264 is generally considered to be not worth it
[20:17] <sfan5> why?
[20:17] <Fyr> c_14, no it's not accurate.
[20:18] <Fyr> I don't see any artifacts in picture.
[20:19] <Fyr> but I'm pretty much sure that signal/noise ratio is beyond the limits
[20:20] <Fyr> video is a signal. there were invented many methods to analyze it.
[20:21] <Fyr> how can I subtract pictures?
[20:21] <Fyr> the result will be a 2d-signal.
[20:22] <Fyr> I will see signal/noise ratio in Fourier image of the picture.
[20:25] <c_14> sfan5: I said that based mainly on internet sources I have found (which I can't find the links to right now) and the link I posted above where the faq for encode/h.264 states it.
[20:26] <c_14> Fyr: As I stated, "the only tool _I know of_ [..]". There might be something that can do that for you, but I don't know of it.
[21:11] <Fyr> c_14, just in case if someone ask you again:
[21:11] <Fyr> http://superuser.com/questions/338725/compare-two-video-files-to-find-out-which-has-best-quality
[21:12] <c_14> hmm, I'll look it over. Thanks.
[21:43] <mjuszczak> Are there other ways to see how ffmpeg might have been compiled other than using ldd on it and running "ffmpeg" without options to see a truncated ./configure string (limited to 1k unfortunately).
[21:52] <llogan> mjuszczak: what's wrong with "ffmpeg -version | grep configuration"?
[21:54] <mjuszczak> Oh, HOLY COW!  Thank you! :)
[22:09] <mjuszczak> llogan: Are there other things I might need in order to know how ffmpeg was setup/installed?  Use flags for instance?
[22:12] <llogan> mjuszczak: what are you trying to do, exactly?
[22:13] <llogan> maybe you'll want to know how any external libraries were configured?
[22:34] <llogan_> my power strip shorted out on this server. cheap ass shit.
[22:34] <llogan_> bad connection, not overload.
[22:41] <Fjorgynn> why is 22050 better than 44100?
[22:51] <iive> it is sampling rate, not max audio frequency.
[22:52] <iive> if you want to represent a sine wave at 22050Hz, you need 2 values, one maximum and one minimum. this means that sampling rate must be at least double of the maximum audible rate.
[22:53] <Fjorgynn> aha
[22:56] <Mavrik> hmm, I doubt 22050 is ever better than 44100 :)
[22:56] <Mavrik> well, except of generating half of data per second :)
[22:57] <Fjorgynn> btw sox <3
[23:02] <Fjorgynn> sox input.wav -n noiseprof | sox input.wav sox-output.wav noisered - 0.2 && ffmpeg -i sox-output.wav -c:a libmp3lame -b:a 128k podcast.mp3
[00:00] --- Sat Jun 14 2014