[Ffmpeg-devel-irc] ffmpeg.log.20130329

[burek]

[00:17] <Keshl> So, so, so so I has a video, .ts encoded with h.264. I'm trying to convert it to .mp4 using the x264 encoder, but any time I try the audio is horribly desynced with the video. The pitch is correct, leading me to believe that ffmpeg can't handle the framerate I expect. (59.94 FPS) The video is ahead of the audio. Here's an ffprobe of the .ts file, http://pastebin.com/diRyJpTZ . I've tried commands similar to "ffmpeg -r 59.94 -i foo.ts -
[00:17] <Keshl> vcodec libx264 -preset medium -crf 25 bar.mp4" with no success. The .TS file, however, plays just fine. Oddly, when I ran "ffmpeg -r 60000/1001 -i foo.ts -vcodec libx264 -preset medium -crf 25 -acodec copy -t 30 bar.mp4", it produced a 35KB file VLC can't recognize and gave me an error reading "[mp4 @ 000000000037f720] malformated aac bitstream, use -absf aac_adtstoascav_interleaved_write_frame(): Operation not permitted". Does anyone has
[00:17] <Keshl> any suggestions? .É.?
[00:30] <llogan> Keshl: that's a lot of text, but i think it's telling you to try adding "-absf aac_adtstoasc"
[00:32] <Keshl> Oh. .... OH, see, spaces are important. <.<
[00:32] Action: Keshl tries.
[00:33] <Keshl> That makes it work.. Hopefully it syncs the audio too..
[00:33] <Keshl> Nope.
[00:33] <Keshl> x.x
[00:34] <Keshl> Same as pretty much every other try. x.x
[00:39] <llogan> maybe someone can help, but now i must go
[00:42] <Keshl> http://pastebin.com/0Fvxcf0v for whoever needs it, oÉo
[00:49] <smus> asked this yesterday, but i wasn't in here long enough. small riff on the main video_encode_example: http://pastebin.com/M5VuHM9b except the outbuf that's populated by avcodec_encode_video is all zeros. out_size > 0 though. ideas?
[00:50] <smus> (this is running in Native Client, btw -- trying to make a video encoder that can be invoked from JavaScript!)
[02:24] <Muon> hi
[02:25] <Muon> I'm trying to record my desktop's audio, but it's getting recorded slowed
[02:25] <Muon> I'm running PulseAudio
[02:25] <Muon> neither -f pulse -i default nor -f alsa -i pulse works properly
[02:26] <Muon> I've tried setting -sound_rate on -f pulse to 44100, but that didn't change anything
[02:36] <Sanky> hello, I use ffmpeg to stream my desktop, but the audio desyncs (about five seconds from the beginning, but it falls behind even more)
[02:37] <Sanky> what should I use to keep audio and video in sync?
[02:39] <Muon> also, when recording from the desktop audio monitor, PA's CPU usage massively jumps
[02:39] <Muon> around 30% in top
[02:39] <Sanky> this is the command I've put together: http://pastie.org/7157324
[03:22] <Muon> hmm
[03:22] <Muon> it was a problem with the sample rate on the PA side
[03:23] <Muon> now to figure out why I can't record both the mic and the desktop audio at the same time
[03:24] <klaxa> there's two ways to that
[03:24] <klaxa> either you record two audio-tracks with ffmpeg, or you mix both audio-streams in pulseaudio and record only one
[03:24] <klaxa> the first approach is cleaner though since you can do proper post-production on the different tracks
[03:24] <klaxa> *post-processing
[03:25] <Muon> klaxa: hmm?
[03:25] <Muon> klaxa: I'm currently just specifying two pulse inputs
[03:26] <Muon> but only the last one seems to be going through
[03:26] <klaxa> how can you tell?
[03:27] <Muon> hm, actually, no
[03:28] <Muon> klaxao: if I put the system audio monitor in at all, it takes over
[03:36] <Muon> ok
[03:39] <Muon> klaxa: http://pastie.org/private/9llsgymvlnupncngbh0ga
[03:40] <klaxa> Muon: try adding -map 0 -map 1 to your command line
[03:42] <Muon> klaxa: nope, that just reintroduced the pitch problem
[03:42] <klaxa> do you have two audio-tracks though?
[03:42] <Muon> klaxa: ?
[03:43] <klaxa> either the output of the command line you ran, or ffprobe <output>
[03:43] <klaxa> on a pastebin-like site please :)
[03:43] <Muon> sec
[03:44] <Muon> klaxa: http://pastie.org/private/h6tr6igezihv0qv21n9ka
[03:44] <klaxa> Stream mapping:
[03:44] <klaxa>   Stream #0:0 -> #0:0 (pcm_s16le -> libvorbis)
[03:44] <klaxa>   Stream #1:0 -> #0:1 (pcm_s16le -> libvorbis)
[03:44] <klaxa> at least you have both audio streams :X
[03:45] <klaxa> that's weird though
[03:45] <klaxa> it doesn't look like there is any resampling going on even
[03:45] <Muon> I can record both with parecord
[03:45] <Muon> FWIW
[03:45] <Muon> (at the same time)
[03:46] <klaxa> yes, ffmpeg is recording both inputs now too (only on two different tracks, you will have to extract them individually and mix them again if you want to have them in one track)
[03:48] <Muon> klaxa: right, but I want them to be mixed
[03:48] <klaxa> ah you want to mix them without post-processing? (are you sure? your mic would probably create a lot of noise you might want to filter)
[03:49] <klaxa> in that case however, you can use pulseaudio to mix the two streams to one single stream and record that one
[03:49] <Muon> this is just for fun :)
[03:49] <klaxa> to do that you have to load a null-sink module and one loopback-module in pulseaudio
[03:49] <Muon> klaxa: can't ffmpeg do that?
[03:50] <klaxa> hmm... i'm sure there is some mixing audio filter, but you might still have this weird pitch-issue
[03:50] <Muon> I want to try
[03:52] <klaxa> https://ffmpeg.org/ffmpeg-filters.html#amix
[03:53] <klaxa> Muon: ^
[03:57] <Muon> klaxa: I love you :D
[03:57] <Muon> thanks
[03:57] <klaxa> :)
[03:57] <Muon> it actually doesn't sound that bad
[03:58] <Muon> maybe with a low-pass filter
[03:58] <klaxa> Sanky: you still there?
[03:59] <klaxa> you could try to add -async 1 to your command line
[03:59] <klaxa> or a higher value
[03:59] <klaxa> see the ffmpeg manpage for more details, it tries to keep the audio synced by adding or removing samples from the audio-stream
[04:00] <klaxa> ah uh...
[04:00] <klaxa> wait... 1 is a special case, also that option is deprecated :|
[04:01] <klaxa> Sanky: https://ffmpeg.org/ffmpeg-filters.html#asyncts
[04:04] <Muon> klaxa: http://pastie.org/private/1mv7hzdm9snckt9ptipkg
[04:04] <Muon> I tried using aresample
[04:07] <klaxa> okay from what i take from the output ffmpeg doesn't know which input to resample
[04:07] <klaxa> hmm no wait
[04:08] <klaxa> mh sorry dunno
[04:08] <Muon> ah
[04:08] <Muon> fixed it
[04:08] <Muon> need to put the filter string in one filter command
[04:08] <Muon> -filter_complex 'foo, bar, baz'
[04:09] <klaxa> ah
[04:13] <Muon> klaxa: it's better, but I still think I need a low pass filter to get rid of the hum produced by the computer
[04:14] <Muon> and it claims that lowpass is not a filter
[04:14] <klaxa> lolwut
[04:14] <klaxa> paste it?
[04:15] <Muon> klaxa: http://pastie.org/private/nyfknxdwlye9czm0qtbjw
[04:16] <klaxa> weird
[04:16] <klaxa> see if it shows up in "ffmpeg -filters"
[04:16] <Muon> it doesn't
[04:17] <klaxa> then you don't have it... and neither do i, i wonder why
[04:17] <Muon> hrm
[04:18] <klaxa> you might have to recompile and add --enable-filter=lowpass to ./configure
[04:44] <stevenbell> Hi, I'm trying to simultaneously webcast and record a single video input (v4l2, in this case).  I can easily do this by having two output files for one command, but I want to be able to start and stop the recording, while leaving the webcast running.
[04:45] <stevenbell> Right now I'm doing this by having one instance that sets up the web stream (flv to Justin.tv) and another RTP stream, which is read by a second instance of ffmpeg.  Is there a better way to do this?
[10:33] <Zeeflo> how do I define srt (subtitles) playresx and y ?
[10:33] <Zeeflo> and, can I define scale (size) of subtitles?
[10:33] <Zeeflo> im using bureks build
[10:34] <Zeeflo> ubitux,
[10:44] <relaxed> Zeeflo: you probably want to convert them to ass
[10:45] <Zeeflo> it takes out the ease of use with SRT and not having to convert them..
[10:45] <Zeeflo> Is it cause of some vobsub setting?
[10:45] <relaxed> maybe you can find some software to edit them to your liking
[10:46] <Zeeflo> nah.. I dont wanna edit all my subtitle files.
[10:46] <Zeeflo> time saved = money saved
[10:46] <Zeeflo> i was just wondering if I could specify the playres for srt
[10:47] <relaxed> well, I know with ass it's specified in the subs
[10:47] <Zeeflo> yea it is
[10:47] <Zeeflo> as well as font etc
[11:49] <PG12> so my question is can we output from  ffmpeg to http ?
[11:50] <PG12> something like ffmpeg -i RTP_H264 -o http://ip:80/file.vp8
[11:50] <PG12> is this possible?
[11:50] <Mavrik> is there a server on ip:80?
[11:50] <PG12> yes
[11:52] <PG12> Mavrik: my concern is mainly is if ffmpeg can accept rtph64 and output to an http address
[11:52] <Mavrik> iirc yes
[11:53] <Mavrik> you will have to pass "copy" as codecs if you don't want to reencode
[11:53] <PG12> i just heard that ffmpeg cant output  http
[11:53] <JEEB> yes, it cannot serve http
[11:53] <JEEB> only ffserver can
[11:54] <JEEB> VLC or ffserver are the alternatives with ffmpeg
[11:54] <PG12> which one would be easier to get it running
[11:54] <Mavrik> PG12, that's why I asked you
[11:54] <Mavrik> if you have a server on the receiving side
[11:54] <Mavrik> and you said "yes"
[11:55] <Mavrik> there's a difference between being a server and sending to a server
[11:55] <PG12> Mavrik: i do have a streaming server wuilling to accept the vp8 sent from ffmpeg
[11:55] <Mavrik> then ffserver isn't a solution for you
[11:55] <Mavrik> ffmpeg should work.
[11:56] <JEEB> I remember trying to do http streming with apache once
[11:56] <JEEB> wouldn't take my named pipe :V
[11:57] <PG12> i didnt get it
[11:57] <Mavrik> JEEB, I streamed VP8/vorbis to stream-m with ffmpeg over http with no problems
[11:58] <JEEB> yeah, but what was serving http in that case?
[11:58] <JEEB> and how did you feed it to your http server in that case?
[11:58] <Mavrik> stream-m is a HTTP streaming server.
[11:59] <JEEB> just out of interest, how did you feed the stream to it?
[11:59] <PG12> Mavrik: how did you feeed the theora/vorbis from ffmpeg into the stream-m ?
[11:59] <Mavrik> JEEB, with http :)
[12:00] <Mavrik> JEEB, see the "publishing on linux" example https://code.google.com/p/stream-m/
[12:00] <PG12> Mavrik: how with http?
[12:00] <Mavrik> stream-m had some stability problems though
[12:00] <PG12> Mavrik: you used ffserver?
[12:00] <Mavrik> no.
[12:00] <JEEB> waait, I don't get it
[12:00] <JEEB> does ffmpeg have http output now?
[12:00] <PG12> -f webm http://example.com:8080/publish/first?password=secret ?
[12:01] <Mavrik> JEEB, yeah.
[12:01] <Mavrik> :)
[12:01] <JEEB> ...
[12:01] <JEEB> since when?
[12:01] <Mavrik> had it for awhile afaik
[12:01] <Mavrik> I tested this like a year ago
[12:01] <Mavrik> (maybe less)
[12:01] <JEEB> I've been pretty sure it couldn't do http serving
[12:01] <Mavrik> it can't
[12:01] <Mavrik> it can connect to a HTTP server
[12:01] <Mavrik> it cannot act as a http server
[12:01] <JEEB> aahhh
[12:01] <Mavrik> (that's why you need ffserver for that)
[12:01] <JEEB> so http post instead of get?
[12:01] <JEEB> or whatever
[12:02] <Mavrik> yeah
[12:02] <JEEB> gotcha
[12:02] <Mavrik> stream-m open HTTP socket where it receives video and then acts as a server
[12:02] <Mavrik> *opens
[12:02] <PG12> Mavrik: did you get it working with ffmpeg?
[12:02] <PG12> or ffserver?
[12:02] <PG12> i am confused
[12:03] <JEEB> you need ffserver if you have no other streaming server, if you have a separate streaming server that takes in post
[12:03] <JEEB> then you can use ffmpeg
[12:03] <PG12> yes i have icecast
[12:03] <PG12> ffmpeg -re -i univac.webm -vcodec copy -acodec copy \
[12:03] <PG12> -f webm http://ICECAST:8080/publish/first?password=secret
[12:03] <PG12> that should work then?
[12:09] <JEEB> PG12, I have no idea if icecast is supposed to work, but yes -- in general like that
[12:10] <Mavrik> yeah, haven't tried icecast
[12:30] <PG12> do you think the Raspberrypi can handle vp8 transcoding from a webcam and send to icecast?
[12:50] <Sanky> klaxa: late thanks!
[13:07] <Sanky> hm, I can't get asyncts nor aresample to work
[13:08] <Sanky> and I've been using async (it's in the paste I provided)
[13:10] <relaxed> Sanky: url?
[13:11] <Sanky> relaxed: http://pastie.org/7157324
[13:11] <Sanky> I'm trying to get audio and video in sync, they start desynced and shift away even more with time
[13:12] <relaxed> This is a common issue
[13:13] <relaxed> try using a faster preset and/or less fps
[13:13] <Sanky> that doesn't sound like a solution :/
[13:13] <Sanky> I'm already using the "fast" preset
[13:14] <relaxed> do you need to change the audio's sample rate?
[13:14] <Sanky> not sure, maybe I don't
[13:14] <relaxed> well your paste shows medium, and that's all I have to go by
[13:15] <Sanky> oh, geez, I have two presets set
[13:15] <Sanky> FLV does not support sample rate 48000, choose from (44100, 22050, 11025)
[13:15] <Sanky> yes, I need to set it
[13:16] <relaxed> ok, try again with -tune zerolatency
[13:16] <JEEB> it does support the sample rate of 48kHz -- but with AAC :P
[13:16] <Sanky> yeah, I have to use flv
[13:17] <JEEB> I didn't look at your command or what format you are using, but you just said something incorrect
[13:17] <JEEB> FLV supports 48kHz
[13:17] <JEEB> with AAC
[13:17] <JEEB> if you are using something else but AAC, then yes
[13:17] <Sanky> I said exactly what vlc spewed at me
[13:17] <JEEB> then VLC is incorrect
[13:17] <JEEB> you can go look at the FLV specification
[13:17] <Sanky> relaxed: the audio is still behind with that option, but I don't know if it will or won't shift more with time
[13:17] <JEEB> the VLC specification says that 48kHz is all OK for AAC in FLV
[13:17] <JEEB> ugh
[13:18] <JEEB> not VLC specification
[13:18] <Sanky> if it's behind by a fixed amount I assume that could be fixed
[13:18] <JEEB> FLV
[13:18] <JEEB> I know libavformat got fixed regarding this some time ago (within the last year or so)
[13:18] <Sanky> JEEB, oh, well I also use -acodec libmp3lame
[13:18] <JEEB> yes, for MP3 it doesn't have wider support
[13:18] <JEEB> it's only for AAC
[13:19] <JEEB> also FLV is otherwise pretty much limited to stereo, but once again AAC is the exception
[13:19] <relaxed> I'm not sure what to tell you, maybe someone with more screencasting experience will chime in.
[13:20] <JEEB> for AAC it just says that you should write the values of 2ch/44.1kHz into the container, and then write your stream into it. And then the reader/player will ignore those container values and parse the data from the actual AAC stream
[13:20] <JEEB> (that's what the actual FLV specs say :D  )
[13:20] <Sanky> it's okay, I'll try streaming with this option the next time and see how it performs; thanks
[13:21] <relaxed> most of the time it's best to simplify your command as much as possible
[13:22] <relaxed> start with low bitrates and then introduce more complexity until it fails.
[13:37] <Muon> hello again!
[13:37] <Muon> this time I'm piping in data from a named pipe
[13:38] <Muon> however, what I get recorded is not what I see
[13:38] <Muon> (I hacked primusrun to dump frames to a named pipe)
[13:39] <Muon> to be specific, it's not quite running at the same rate
[13:40] <Muon> e.g. the loading screen apparently doesn't generate many frames, so in the combined output, the thing blurs by
[13:49] <Muon> so, uh, yeah
[13:50] <Muon> I've got variable rate raw video input
[13:50] <Muon> how do I tell ffmpeg to put it together properly?
[13:50] <Fjorgynn> what?
[13:51] <Muon> Fjorgynn: ?
[14:07] <Muon> Fjorgynn: I have variable frame rate input, how do I make ffmpeg output it with timestamps corresponding to when each frame was consumed?
[14:07] <Muon> I can play it back using ffplay just fine
[14:07] <Muon> (from the pipe)
[14:25] <Muon> hmm
[14:25] <Muon> some of the frames are skipped, but the playback appears to be slower than reality
[14:45] <Muon> OK, seems it's treating it as a stream going at 25 FPS o_O
[14:46] <Muon> I guess what I want is padding?
[14:49] <Mavrik> ?
[14:49] <Mavrik> padding?
[14:50] <Muon> Mavrik: frames that are padding
[14:50] <Muon> as in, duplication
[14:51] <Muon> if we don't have a frame yet, and we need one, copy in the last one
[14:51] <Mavrik> ah, padding usually means something else
[14:51] <Muon> *shrug*
[14:51] <Muon> this is just ridiculously annoying
[14:52] <Muon> I was so happy when I managed to hack primusrun to dump frames to a pipe :(
[16:12] <Muon> wait, I'm being silly
[16:12] <Muon> I shouldn't be using a pipe
[16:13] <Muon> a plain old file which is rewritten for every frame is enough
[16:13] <Muon> the possibility of tearing does not concern me
[16:13] <Muon> now how do I get this to re-examine a file all the time...
[16:27] <bparker> what kind of video/image file has the header "frmh," ?
[16:27] <bparker> I also see frma and frmv in the file
[16:27] <bparker> it's an image sequence, but I think it might also contain audio
[16:51] <Muon> right
[16:51] <Muon> how do I limit the output FPS?
[16:51] <Muon> uh
[16:54] <Muon> http://pastie.org/private/ozurbzmy17xcbdmolvedq
[16:54] <Muon> ^current command line and output
[16:55] <Muon> as you can see it's running at about 42 fps even though -r 10
[16:59] <LithosLaptop> is shows 10...
[16:59] <LithosLaptop> *it
[17:00] <Muon> LithosLaptop: frame=  336 fps= 42 q=31.0 Lsize=    1693kB time=00:00:33.60 bitrate= 412.9kbits/s
[17:02] <LithosLaptop> thats how fast it is encoding fps at, not the output fps
[17:03] <Muon> LithosLaptop: which is important
[17:03] <Muon> LithosLaptop: it's going to try to shove that down the internet stream at the same rate
[17:07] <LithosLaptop> sorry I don't really understand, maybe somebdoy else could help. to me it seems like you are outputting to a file right?
[17:08] <Muon> that's just for testing
[17:09] <LithosLaptop> if somebody downloads it faster they still will playback the file at 10fps
[17:09] <Muon> yes
[17:09] <Muon> which means they will see slowed video
[17:09] <Muon> the framebuffer can be sampled as much as you like
[17:10] <LithosLaptop> does it look slow when you playback the file?
[17:10] <Muon> yes
[17:11] <LithosLaptop> ah
[17:12] <LithosLaptop> you are saying your source is 10 fps
[17:12] <Muon> my source is whatever FPS
[17:12] <Muon> I just want it synced to my sound
[17:12] <LithosLaptop> maybe move -r 10 after the input
[17:13] <LithosLaptop> near the end of the command line
[17:13] <Muon> tried that
[17:13] <LithosLaptop> oh
[17:13] <Muon> yeah
[17:14] <Muon> it's running way to quickly
[17:15] <Muon> *too
[17:15] <Muon> but x11grab respects the settings
[17:16] <LithosLaptop> wuthout -t 10 it works fine?
[17:16] <LithosLaptop> I mean the duration
[17:16] <LithosLaptop> *-r 10
[17:16] <Muon> no, -r 10 is my attempt at fixing this
[17:17] <LithosLaptop> how many fps is the source?
[17:17] <Muon> infinite
[17:18] <Muon> I am just piping in a framebuffer-like file on loop
[17:20] <LithosLaptop> its piping too fast
[17:20] <Muon> obviously
[17:20] <Muon> but fixing the rate doesn't help
[17:20] <Muon> audio desynced
[17:20] <LithosLaptop> so any ffmpeg settings would help
[17:20] <LithosLaptop> *wont
[17:21] <Muon> well, if you could somehow tell ffmpeg to read the framebuffery thing on loop
[17:22] <LithosLaptop> yeah maybe
[17:22] <Muon> this isn't very useful
[17:22] <Muon> :(
[17:23] <LithosLaptop> sorry I am not a ffmpeg guru
[17:23] <LithosLaptop> I don't know where all the other people are
[17:23] <LithosLaptop> maybe sleeping
[17:23] <Muon> get the cattleprod
[17:24] <LithosLaptop> can you maybe show how you are piping the raw video?
[17:25] <Muon> LithosLaptop: not gonna be useful
[17:25] <Muon> lemme see if I can hack up a loopy device
[17:26] <Muon> because altering ffmpeg at 3:36 AM is totally sane
[17:26] <LithosLaptop> haha
[17:38] <Muon> seems it's gonna be easy to modify
[17:38] <Muon> but sleepy
[17:38] <Muon> curse youuuuu
[17:38] <Muon> good night
[19:23] <Sashmo> anyone have any luck with decklink SDI, HD sources and ffmpeg?
[19:25] <Mavrik> hmm
[19:25] <Mavrik> I only used HDMI source
[19:25] <Mavrik> it worked on Windows through directshow
[19:37] <Sashmo> any command line examples?
[19:37] <Sashmo> my CPU is at 90% and its a monster machine, 12000 cpu benchmark points... and the video looks like crap
[20:33] <Sashmo> does anyone know if there is a decomb filter for ffmpeg?  google tells me nothing....
[20:34] <durandal_1707> decomb filter does what?
[20:34] <Sashmo> like deinterlace
[20:34] <Sashmo> but selective
[20:35] <durandal_1707> selective on what?
[20:36] <JEEB> IVTC you mean?
[20:36] <Sashmo> http://pastebin.com/MqCBu6Sb
[20:36] <JEEB> ubitux is porting vivtc from vapoursynth (which is a reimplementation of the avs TIVTC filter)
[20:36] <JEEB> oh
[20:36] <JEEB> Handbrake's decomb
[20:37] <JEEB> that's an adaptive field matcher / deint filter IIRC
[20:40] <durandal_1707> Sashmo: there is deinterleave filter in libavfilter/ffmpeg, its called il
[20:41] <Sashmo> thanks I'll look into it
[20:41] <JEEB> I /think/ yadif atm is the closest to what he wants
[20:41] <Sashmo> is it the same as decomb? or similar?
[20:41] <durandal_1707> JEEB: yadif is deinterlacer
[20:41] <JEEB> yes, and that's what he wants, an adaptive deint/field match
[20:42] <JEEB> and yadif is atm the closest to get that
[20:42] <durandal_1707> no, from explanation he only wants deinterleaver
[20:42] <durandal_1707> which is faster than deinterlacer
[20:42] <JEEB> well decomb filter that he pastebin'd is slower
[20:42] <JEEB> it's handbrake's EEDI2 + combing finding filter, which tries to be adaptive between field matching and deint
[20:43] <JEEB> :V
[20:43] <JEEB> so looking at that pastebin the closest ffmpeg has is yadif
[20:43] <JEEB> but if you know better than I do...
[20:43] <JEEB> sure
[20:43] <JEEB> I don't know what his source is anyways (truly interlaced/hard telecine/whatever)
[20:44] <durandal_1707> yes, you are right, there is no deinterleaving at all, i failed at reading
[20:45] <durandal_1707> so he is actually looking for deinterlacer
[20:45] <Sashmo> the byadif deinterlacer leaves ghosting
[20:45] <Sashmo> yadif*
[20:46] <llogan> are you using default settings?
[20:47] <llogan> think of it as a feature. i hope it's a horror movie.
[20:49] <Sashmo> umm, I've used in many ways, and really I get the same sort of result each time but filter:v yadif is the best that otuput
[20:49] <Sashmo> BUT, its still leaving me some ghosting, specially on HD sources
[20:50] <durandal_1707> what version you are using?
[20:50] <durandal_1707> and what kind of HD sources is that (resolution, pixel format)
[20:52] <Sashmo> SDi input HD 1080i, cant recall the pixel format
[20:53] <durandal_1707> ffprobe output of some sort
[20:59] <Sashmo> yuv420p is the format
[21:03] <Sashmo> durandal_1707: what do you think?
[21:04] <durandal_1707> fill bug report for better filter
[21:04] <durandal_1707> otherwise nobody complains and code is left in status quo
[21:05] <Sashmo> durandal_1707: thanks... I gues ;)
[21:36] <grepper> The following works for me to output a frame every 5 minutes from an ntsc dvd video (for a static chapter menu):  -filter:v select="not(mod(n\,8991))",setpts="N/(29.97*TB)" .  What I am wondering is if its possible to output say 300 frames at interval N rather than just 1 frame ?
[21:37] <durandal_1707> you mean each 5 minutes pick 300 frames?
[21:38] <grepper> durandal_1707: sure, for example
[21:38] <durandal_1707> yes its possible
[21:39] <durandal_1707> read Expression Evaluation documentation
[21:40] <grepper> I was afraid you would say something like that :)  - I'm lost understanding filter theory, I was glad to find the above example with google
[21:40] Action: grepper goes to look
[21:43] <grepper> math is not my strong suit by a long shot, any hints durandal_1707 ?
[21:51] <grepper> what is 'n' in the above filter: 'mod(n'?  Current frame # ?
[21:52] <durandal_1707> grepper: there are ld() - load variable and st() - store variable
[21:52] <durandal_1707> grepper: 'n' is documented in select filter documentation
[21:53] <durandal_1707> n - is the sequential number of the filtered frame, starting from 0
[21:54] <grepper> I see
[21:55] <durandal_1707> and selecting 300 frames every 5 minutes could be done with some kind of periodic function like sin/cos
[21:56] <durandal_1707> but that is far from user friendly - requiering non-trivial math knowledge is not feature
[21:57] <durandal_1707> but you could do same with ifs,st,ld - which is ugly somehow
[21:58] <durandal_1707> also there may be better solution ....
[22:03] <grepper> okay thanks, I'll see if I can make heads or tails out of it
[23:17] <esdf> hi, how can i convert from flv, mp4, avi and wmv to .webm at same quality? (sameq ain't really same quality)
[23:22] <sacarasc> Same quality as what?
[23:37] <smus_> trying again, small riff on the main video_encode_example: http://pastebin.com/M5VuHM9b except the outbuf that's populated by avcodec_encode_video is all zeros. out_size > 0 and seems valid though. any ideas what's going on?
[23:38] <esdf> sacarasc as source
[23:38] <esdf> well, at least higher then youtube quality
[23:44] <UukGoblin> hi
[23:44] <UukGoblin> I'm trying to set up some streaming of my desktop
[23:44] <UukGoblin> my current problem is that ffplay doesn't seem to play anything
[23:44] <UukGoblin> I've set up a simple avserver
[23:45] <UukGoblin> ffmpeg seems to grab fine (grabs especially well to a file)
[23:45] <UukGoblin> is avserver the way to go? Man page says it's unmaintained and broken :-/
[23:45] <gmaxwell> UukGoblin: perhaps your video needs better timestamps?
[23:45] <UukGoblin> Hi gmaxwell :-)
[23:46] <UukGoblin> I've no clue about any timestamps :-)
[23:46] <UukGoblin> let me paste my stuff
[23:47] <llogan> esdf: https://ffmpeg.org/trac/ffmpeg/wiki/vpxEncodingGuide
[23:47] <llogan> would be a good start
[23:48] <UukGoblin> http://pastebin.com/SeWj77P0 <- these are my attempts so far
[23:48] <llogan> UukGoblin: avserver is not from FFmpeg
[23:48] <UukGoblin> oh :-/
[23:49] <UukGoblin> gmaxwell, I've just got your joke, lol
[23:50] <esdf> ty llogan
[00:00] --- Sat Mar 30 2013