[Ffmpeg-devel-irc] ffmpeg.log.20140214

[burek]

[00:10] <eZpl0it> hey llogan
[00:10] <eZpl0it> http://pastebin.com/jTTYJgd4
[00:47] <houms> can ffserver take rtsp url and rebroadcast?
[00:55] <llogan> eZpl0it: what player shows it incorrectly?
[00:56] <llogan> why do you set -qscale:a and then -ab? they are mutially exclusive. i believe -qp is a libx264 private option
[01:02] <houms> say multicast?
[01:04] <eZpl0it> i only got a greyscale image when i set libx264 codec
[01:04] <eZpl0it> forgot to remove it
[01:04] <eZpl0it> mplayer
[01:04] <eZpl0it> i recorded it with ffmpeg
[01:04] <eZpl0it> mplayer shows it correctly when i watch directly via /dev/vide0
[01:04] <eZpl0it> videoß
[01:04] <eZpl0it> video0*
[01:20] <llogan> eZpl0it: ffmpeg -f v4l2 -i /dev/video0 -f alsa -ar 44100 -ac 2 -i hw:2 -vcodec libx264 -crf 18 -vf "yadif=1,format=yuv420p,setsar=1:1" -acodec libmp3lame -q:a 4 geburtstag.mkv
[03:05] <Magiobiwan> Hey guys. Has anyone ever used ffmpeg to transcode video coming from a TV tuner card (on Linux) to an RTMP stream?
[04:40] <funyun> hi. if a mediainfo says my video is interlaced, but i can't see a single line, should i deinterlace when encoding? or is the info incorrect?
[04:52] <z1lt0id> just wondering if there is a way to remux a .ts file so it doesn't relative audio to video delay in ffmpeg
[04:54] <z1lt0id> i have two .ts files one with 6ms and another say 16ms i want to encode both of these files to h264 and the paste them together in MP4Box
[04:54] <z1lt0id> the audio will demuxed to .wav file and concat together and then encoded to say aac
[04:54] <z1lt0id> and then muxed into the MP4
[05:13] <BtbN> z1lt0id, so, you have two ts file you want to concat and mux as mp4?
[05:15] <z1lt0id> i do
[05:15] <z1lt0id> but there is an audio delay sometimes it can go up to 30-40ms
[05:16] <z1lt0id> do i need to add an audio delay into the audio file
[08:37] <Keestu> could someone help me out. http://pastebin.com/4SVJGudD      i am unable to find the pts of frame.
[09:40] <relaxed> Now with libx265, hevc: http://johnvansickle.com/ffmpeg/
[10:45] <Keestu> can some one help me out the concept behind it please?  i dont get much information from the net. rather i would say i am unable to understand.    i tried getting the frames after  avcodec_decode_video2 (), by checking  one of the variable passed in it. i get the frames like IPPPPPPPPPPIPPPPPP.... Now mu question is Can i send the this 'P' frame directly to the UI?
[10:45] <Keestu> or do i need to do any calculation >
[10:52] <relaxed> Keestu: you might try the libav-user mailing list: https://lists.ffmpeg.org/mailman/listinfo/libav-user/
[10:52] <relaxed> oh, you already did :)
[11:00] <Keestu> relaxed,  if i ask there they are redirecting me here...
[11:00] <Keestu> it is happening third time for me :(.   Just because of the information/knowledge about the system, it takes huge amount of time.
[11:38] <relaxed> Keestu: It's fine to ask here. I was just giving you another avenue to pursue.
[11:53] <brontosaurusrex> playing around with davinci resolve and noticed that "4:2:0 avc > 4:2:2 prores" clips kinda grade better than original "4:2:0 avc" stuff, any rational explanation (I converted all using ffmpeg to 10bit prores 4:2:2) ?
[11:54] <zap0> brontosaurusrex, why wouldn't they?
[11:54] <zap0> 4:2:2 is more betterer than 4:2:0
[11:55] <zap0> 10 bit is 4x better than 8bit
[11:55] <brontosaurusrex> lol, yes, but this is upscaled 4:2:0 avc
[11:55] <brontosaurusrex> "upscaled"
[11:56] <brontosaurusrex> a fake 4:2:2 10 bit stuff
[12:04] <zap0> but it wont be  4:2:0  just duplicated horizontolly.. it'll have interpolation in the extra horizontal bits
[12:05] <zap0> and 10 bit will give you far more room to grade in
[12:06] <zap0> and is 10bit proress a linear 10bits.. or some wacky log scale?
[12:10] <zap0> brontosaurusrex,   if you have 0,1,2,3,4..256.. and just "dumb upscale" it to basically  0,4,8,12.. 1023.  you then say add 1/1024th..  that 0 will become 1.. but then you display it on a 8bit display it'll display as 0 again..   but with some interpolation, you would get an upscaling that is not just 0,4,8,12..  you get odd numbers in there too.
[12:11] <zap0> so when you then add 1/1024th..  some of those odds round upto next 8bit value..  this means even small changes like adding 1/1024 will actually show you something when displayed on a 8bit display
[12:12] <zap0> this is aliasing!
[12:14] <zap0> http://en.wikipedia.org/wiki/Quantization_error
[12:15] <zap0> brontosaurusrex, may i ask,  did you purchase a camera and get Resolve free ?
[12:24] <Keestu> thanks relaxed.  any input to my question ?.
[12:35] <brontosaurusrex> zap0, no, i use the free edition (lite)
[12:36] <zap0> ah.   does it have limitations? (being the lite ver)
[12:36] <brontosaurusrex> i'd expect from grading software to do wonderfull magical upscale to 10bit or whatever, but obviously this doesn't happen
[12:37] <brontosaurusrex> zap0, yes, like no noise reduction
[12:37] <brontosaurusrex> and probably more
[12:37] <zap0> oh :(
[12:38] <brontosaurusrex> its actually pretty powerfull
[12:38] <brontosaurusrex> but i'am not a pro-grader, so i have nothing to compare it to
[12:38] <zap0> if you look at it's set of DLL files.. it's mostly based on OSS code
[12:39] <brontosaurusrex> like what?
[12:42] <zap0> av*.dll  all the  lib*.dll
[12:43] <zap0> Qt*.dll
[12:47] <brontosaurusrex> and it doesn't like my aja card obviously, maybe there is a way to use hdmi somehow, but haven't actually test that
[20:01] <houms> can anyone help me get sytax right for getting rtsp to multicast? something like http://pastie.org/8734159
[20:02] <houms> the rtmp/rtsp is mpeg4 10
[20:03] <houms> it seems it is flv and 15 frames
[20:03] <houms> trying to figure out how to rebroadcast over udp
[20:03] <houms> not sure what params i am missing
[21:31] <znf> How could I loop a video used in an overlay? :-/
[21:40] <znf> -loop 1 seems to work only for images (so disappointing!)
[23:33] <Logicgate> hey guys
[23:33] <Logicgate> is there a way to accelerate watermarking
[23:36] <relaxed> buy the fastest haswell cpu you can afford
[23:37] <Magiobiwan> How do I tell ffmpeg to take audio input from a tuner card through ALSA?
[23:37] <Magiobiwan> The card shows up in arecord as a capture device, but I need to capture it via ffmpeg
[23:38] <relaxed> Magiobiwan: https://trac.ffmpeg.org/wiki/Capturing%20audio%20with%20FFmpeg%20and%20ALSA
[23:38] <Magiobiwan> Ooh, fancy. A wiki
[23:39] <Magiobiwan> That makes it so much easier
[23:40] <Magiobiwan> I was thinking I'd have to use arecord and pipe it to a socket as an oss output, then have ffmpeg pull from that input
[23:41] <Magiobiwan> Now that that's figured out... Is there a way to specify the input resolution from a tuner card to ffmpeg? It defaults to thinking it's 320x240 which is wrong; it's 720x480
[23:43] <Logicgate> relaxed, right now it takes about 3 hours to watermark an 125mb video.
[23:43] <Logicgate> on a small EC2 instance
[23:43] <Logicgate> will multithreading help?
[23:44] <clever> only if the file is being sliced i think
[23:46] <znf> Is there any way to make a source generated with the showwaves= filter to have a transparent background, or simply modify it's alpha channel?
[23:46] <znf> Been trying to figure this out for hours now
[00:00] --- Sat Feb 15 2014