[Ffmpeg-devel-irc] ffmpeg.log.20140711

[burek]

[00:06] <benlieb> c_14:  any idea how I'd go about a dram-by-frame comparison? Would I have to export every frame to an image?
[00:10] <Muchoz> I don't really know where to ask this question, but since there are a lot of experienced users here related to audio codecs. Perhaps someone could answer my question: so I use Pafy (a Python library for downloading YouTube video/audio streams). So I got to download the m4a audio stream from a video and it plays fine in VLC. It tells me that it is a MPEG AAC Audio as expected. But when I try to open this in iTunes (which supports AAC and ALAC), it
[00:10] <Muchoz>  adds it to the list but I cannot play it for some reason (the duration row says "Unavailable", but in another language so it could be different in English). Does someone know what could be the cause of this?
[00:14] <Muchoz> It also plays fine on Google Play (Android). Does someone know why?
[00:16] <gaffa> * Cannot join #space (You are banned).
[00:16] <gaffa> And I never even went to #space!
[00:17] <iive> and you never will...
[00:18] <gaffa> that sucks.. :/
[00:18] <iive> well, i hope spacex can do something about it.
[00:22] <gaffa> Those magazines I read as a kid about populating Mars was bogus anyway. I had my hopes up that I would experience it happening in my lifetime. I want my subscription fee back! With interest!
[00:25] <gaffa> Well to be fair, I'm not dead yet.
[00:26] <gaffa> Maybe you're right if anyone can do it it is probably Elon Musk.
[02:02] <Dark-knight> haha muchoz i was talking about that yesterday
[02:03] <Dark-knight> be careful about saying m4a is mp4. people in here might get butthurt
[04:57] <sim590> I have a chromecast and it cannot read this file : RIFF (little-endian) data, AVI, 636 x 264, 29.97 fps, video: XviD, audio: Dolby AC3 (stereo, 48000 Hz)
[04:57] <sim590> I want to encode it in another video format
[04:57] <sim590> what should I do ?
[04:57] <sim590> here's the supported format by the chromecast https://developers.google.com/cast/docs/media
[05:02] <Hello71> ffmpeg -i input.avi output.webm
[05:02] <sim590> when I try this : ffmpeg -i <file>.avi <out_file>.mp4, it fails
[05:03] <sim590> I'll try webm
[05:03] <sim590> it says : http://ix.io/dl4
[05:05] <Peter_Occ> I am capturing a stream from my camera with a cronjob. Every hour it stops and starts again with a new video file. However, according to the video player, all the videos are 1 hour and 14 minutes long. What do I  do to make the video time closer to real time?
[05:06] <sim590> Hello71: sorry, it works with webm. I forgot to remove the flag -f mp4.
[05:06] <Hello71> rarely is -f a good idea
[05:07] <Hello71> Peter_Occ: your fps is probably off
[05:07] <Hello71> if you're starting from rawvideo
[05:09] <sim590> Hello71: the process is kind of slow. Do you think I can speed it up ?
[05:10] <sim590> hmmm. I don't think so after all.. `top` says my cpu is 100% used.
[05:10] <Hello71> I don't know if vpx has any encoding parameters
[05:10] <Hello71> it's a tradeoff between speed and quality here
[05:11] <Hello71> (and size obviously)
[05:11] <Hello71> with h264 you can use -preset fast or something like that
[05:12] <sim590> what is h264 ?
[05:12] <sacarasc> H264 is a video encoding standard. It is used by many people and is one of the best we have around at the moment.
[05:12] <Peter_Occ> I'm actually hoping to decrease the quality so I can store more video
[05:13] <sim590> is it a good trade in order to gain speed ?
[05:14] <Hello71> https://trac.ffmpeg.org/wiki/vpxEncodingGuide
[05:14] <Hello71> oh, it moved
[05:14] <Hello71> hm, ffmpeg doesn't expose speed options
[05:15] <sim590> just like that, I'm using avconv because ffmpeg says itself that it's depcrecated on debian.
[05:16] <Hello71> !libav
[05:17] <Hello71> !avconv
[05:17] Action: Hello71 grumbles
[05:17] <Hello71> http://stackoverflow.com/a/9477756/335964
[05:19] <Peter_Occ> f is set to 8. I'm not sure but I think my camera is set at 10, so that might explain it. Thanks for the tip.
[05:19] <Peter_Occ> I mean r
[05:25] <Hello71> yes, 60*5/4 is 75
[07:32] <Peter_Occ> Ever since I changed the fps to match the camera, the capture seems to go well and when I cut cut out 30 seconds of the video it seems to be working but then it quits and never makes the cut. toward the end of the out put is a bunch of zeros. Any idea what the problem is? Here is the output http://pastebin.com/LZpkWSN9
[10:33] <Mavrik> ugh
[12:07] <asherawelan> I would like to live feed my desktop to a video tag with webm, has anyone done something similar?
[12:20] <Dark-knight> i just figured out that i could stream youtube from my computer to my console
[12:20] <Dark-knight> just upload it to YT and do that
[12:21] <Dark-knight> YT has live streaming
[12:21] <Dark-knight> anyway, im out. hope someone helps you. :)
[15:28] <termos> how can I limit the number of threads used by h264 in ffmpeg? Seems the default is one process per core and I have a 40 core machine. The number of threads reduces the quality of the encoding
[15:32] <termos> right now it's saying threads=60, that's too manyu
[15:35] <JEEBsv> termos: when you say "h264" one usually thinks of the decoder, since it's called like that
[15:35] <JEEBsv> but it seems like you're talking of the libx264 encoder
[15:35] <JEEBsv> set -threads after -i
[15:35] <termos> ah yes, i mean the encoder
[17:55] <hizzmo> I've got a downmixing question, all... It's known that, for example, 5.1 -> 2.0 yields audio with a huge range in volume. But what can be done about that?
[17:56] <hizzmo> My Roku (or rather TV since it passes it through) does not support ac3 so I have to convert it. Nedless to say, changing the volume from 60 during dialog and 15 during action scenes is annoying and I can't grab the remote fast enough.
[17:58] <Mavrik> hmm
[17:58] <Mavrik> hizzmo, how are you doing the downmix_
[17:58] <Mavrik> ?
[18:00] <hizzmo> ffmpeg -i <input> -c:v copy -c:a libvo_aacenc -ab 192k -ac 2 <output>
[18:01] <Mavrik> well, for one, you managed to choose the worst of the three available AAC encoders :P
[18:02] <Mavrik> hizzmo, note, since you have AC3, you should probably try:
[18:02] <Mavrik> -request_channels 2 first (this will use AC3 feature where studio adds metadata for stereo downmixing)
[18:03] <Mavrik> if that doesn't work (your tracks aren't properly mastered), use -filter:a aformat=channel_layouts=stereo should give a better downmix than -ac 2
[18:03] <hizzmo> I'm using a Zeranoe FFmpeg build for Windows so VisualOn was all that was immediately available :/
[18:03] <Mavrik> note that AC3 carries downmix info which may interfere what you're doing, -drc_scale 0 MIGHT help with that
[18:05] <hizzmo> Thank you so much for the help Mavrik
[18:06] <hizzmo> What kind of interference does -drc_scale 0 prevent?
[18:07] <hizzmo> Bonus question: do you think ffmpeg's experimental aac encoder is superior and safe to use?
[18:07] <klaxa> no
[18:08] <klaxa> https://trac.ffmpeg.org/wiki/Encode/AAC#NativeFFmpegAACencoder
[18:09] <JEEB> it is better than vo-aacenc
[18:09] <JEEB> but not better than fdk
[18:09] <JEEB> that said, fdk can't be distributed :P
[18:09] <JEEB> as binary
[18:09] <JEEB> so you'd have to build it yourself
[18:10] <JEEB> also there's a VeryLongThread on the trac regarding improvements to the lavc aac encoder, and with the current patch it does get quite a bit better. I hope it will get merged soon, since even if it still has a bug or two, it's still much better than what we have right now
[18:13] <hizzmo> Windows is such a pain to work under (for a Unixman). I hadn't wanted to, but I suppose I'll do the fdk build at your suggestion.
[18:16] <hizzmo> Thanks, all.
[20:11] <hay> hi... I am adding internet radio stream to my live video and sometimes radio "stops" for a short period, is it possible to buffer input audio signal and how to do it? thanks!
[20:20] <sbujnak> Hi, is there a way to retrieve frame metadata using ffmpeg command line tool? There are multiple video filters that set frame metadata accordingly (cropdetect, blackdetect) and I need to obtain those values.
[20:31] <mca64> hi, libx265 + dshow input produces black video. Any idea?
[20:31] <c_14> Just libx265, what about other encoders?
[20:32] <mca64> try diffrent for audio, also diffrent containers
[20:32] <mca64> tried*
[20:37] <Fjorgynn> why
[20:38] <mca64> ok addin -pix_fmt yuv420p  fixed problems
[20:38] <mca64> x264 doesnt required it
[21:09] <libx264-max> hi
[21:09] <libx264-max> where do i get libx264-max for ffmpeg ? seems not in the fedora repos
[21:09] <libx264-max> also, is it true that ffmpeg package on fedora is compiled without faac ?
[21:10] <libx264-max> I'm trying to convert an .mpg into an .flv per some instruction found on the web
[21:11] <libx264-max> and get this 'File for preset 'libx264-max' not found'
[21:32] <libx264-max_> fedora ffmpeg package has no faac. How to add it?
[21:32] <c_14> compile ffmpeg yourself
[21:33] <c_14> https://trac.ffmpeg.org/wiki/CompilationGuide
[21:39] <sacarasc> libx264-max_: If you're going to compile, though, use libfdk_aac.
[21:42] <libx264-max_> sacarasc: ok, so it compiles only with one of those libs at a time? can't all of them be used at once?
[21:42] <sacarasc> No, you can use them all, but libfdk_aac is better than faac.
[22:02] <hay> is it possible to add some "input buffers" for an audio input or am I asking for a stupid thing? :)
[23:07] <Mista_D> How can I follow concat filter with a split? ` ./ffmpeg -i $input1 -i $input2  -filter_complex '[1:0] $drawtext_filter [tmp]; [0:0] [0:1] [tmp] [1:1] concat=n=2:v=1:a=1 [concat_out]; [concat_out] split=2 [out1] [out2]' -map '[out1]' -vcodec libx264 -s 320x176 $preview.mp4 -qscale:v 1 $lossless.ts  `  The error is: ` Too many inputs specified for the "split" filter. `
[23:23] <AlexRussia> Hi! I need to concatenate some sountracks in one file, how i could do it?
[23:24] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20media%20files
[23:24] <AlexRussia> thx
[00:00] --- Sat Jul 12 2014