[Ffmpeg-devel-irc] ffmpeg.log.20120710

[burek]

[00:53] <Orbixx> Is it normal to get an absolute ton of "invalid dropping" output when converting a dvr-ms file?
[00:55] <newl> is the resulting file looking good?
[00:56] <Orbixx> av sync could be better
[00:56] <Orbixx> and the resolution is incorrect
[01:10] <Orbixx> Hmm, seems to be alright aside from the width/height being way off
[01:12] <newl> see about -s and -aspect
[01:25] <Orbixx> newl: For some reason the width keeps defaulting to 1000 pixels
[01:26] <Orbixx> Even though the source video file is nowhere near 1000.
[01:26] <Orbixx> Even when I specify the resolution with -vf "scale"
[01:30] <Orbixx> Right, changes to the -s and -vf are actually occurring
[01:31] <Orbixx> Media players just seem to not recognise the actual resolution
[01:31] <Orbixx> newl: Any idea? ^
[01:32] <newl> -s
[01:33] <Orbixx> newl: Could be a problem with my filters actually...
[01:33] <newl> strip the cli down  and see what basic gives you
[01:34] <newl> exit
[02:43] <mika__> hi there
[02:44] <mika__> anyone is available to help me figure a piping issue from flash-AIR to ffmpeg?
[02:45] <mika__> or not :) http://omgitszooey.com/
[02:49] <mika__> right now i m rendering frames from a flash app and saving to a raw argb bytearray in a file, after i render those ~2000 i render to video with ffmpeg
[02:50] <mika__> where can i find info on how to pipe that in a straight process?
[02:51] <mika__> udp?
[02:52] <mika__> not many answers today :)
[02:54] <newl> or to many the other days
[03:32] <mikos> hi there again
[04:10] <marcosmlopes> Hello, someone can help me? http://stackoverflow.com/questions/11334568/live-video-streaming-with-node-js-doent-work
[04:32] <marcosmlopes> please, anyone can help answer my question on stackoverflow? http://stackoverflow.com/questions/11334568/live-video-streaming-with-node-js-doent-work
[04:38] <newl> exit
[06:03] <train_h8> im so drunk rite now
[06:06] <train_h8> http://www.youtube.com/watch?v=YLKyuBSaCMw
[06:08] <train_h8> nuclear energy is out of the question
[06:16] <train_h8> http://www.youtube.com/watch?v=YN0PI5b4Nww
[06:19] <train_h8> what
[07:04] <piercedwater> can someone please tell me the correct configure flags to enable flac and alac
[07:11] <piercedwater> whoops
[07:14] <train_h8> http://www.youtube.com/watch?v=jXNkf08ObhU&feature=related
[07:42] <JonOomph> Hello!  Is it possible to encode a file (i.e. write_header, av_interleaved_write_frame, write_trailer) and return the binary chunks as the encoding happens, instead of writing it to a new video file?  Thanks!
[08:05] <rohankasat> #include <libavcodec/avcodec.h>
[08:05] <rohankasat> #include <libavformat/avformat.h>
[08:05] <rohankasat> #include<string.h>
[08:05] <rohankasat> #include <stdio.h>
[08:05] <rohankasat> #include <SDL/SDL.h>   // simple directmedia layer header files
[08:07] <rohankasat> m getting an error "Exactly one scaler algorithm must be chosen"  can somebody help me
[08:07] <rohankasat> ?
[08:46] <train_h8> http://www.youtube.com/watch?v=TZi18OBMhgw
[13:18] <catchvibhor> hello
[13:42] <shibby> Hi
[13:43] <shibby> I was wondering is the following is doable with a filter chain
[13:43] <shibby> I want to take the audio from a movie a change its volume at certain timestamps
[13:45] <shibby> for example, between 00:00:02.500 and 00:00:10.000 i want the volume to be doubled, and between x1,y1 i want it to be half
[13:46] <shibby> is there any way to do this using a combination of amovie, avolume, and other audio filters?
[13:48] <ahhughes> I have some questions regarding codecs, not ffmpeg exactly.... but, Im encoding some mpeg2 video, to h.264 avc... thing is.. I was under the impression that h.264 would highly compress the video... but the size works out to be the same as the mpeg2. Am I barking up the wrong tree with this codec?
[13:48] <catchvibhor> @shibby : u should try at http://ffmpeg.gusari.org/
[13:49] <catchvibhor> @shibby : it contains the similar questions &  their solutions
[13:51] <ahhughes> I got the same result with mpeg4 output :'(
[13:53] <shibby> @catchvibhor i know how to apply the volume filter to amovie, my question is whether is it possible to apply different volumes to subsets of an amovie using a chain
[14:12] <shibby> with this i can reduce the volume of the whole audio to half ffmpeg -f lavfi -i "amovie=my_movie.flv, volume=0.5" -c:a libmp3lame test.mp3
[14:12] <shibby> but, is it possible to take subsamples of the audio extracted with the amovie filter
[14:13] <shibby> ?
[14:13] <shibby> say like what we do outside the filterchain with -ss timestamp1 -t duration1
[14:48] <natrixnatrix89> What could be causing the fact that sometimes seeking of the video doesn't work, when played back on my phone?
[14:48] <natrixnatrix89> It's quite interesting.. I use the same settings, but on some videos seeking works just fine, but on others the video freezes when I seek, but audio playback continues..
[14:48] <natrixnatrix89> I'm using libx264 and libfaac
[14:50] <natrixnatrix89> could it be that it has something to do with keyint?
[14:54] <natrixnatrix89> hmm. gotta try reducing keyint it to 25
[16:08] <natrixnatrix89> Hmm. That didn't change anything..
[16:08] <natrixnatrix89> I'm wondering, what could this be related to..
[16:10] <natrixnatrix89> hmm. my only guess is that the bitrate or size could be too high.. but the video is playing normally..
[16:20] <natrixnatrix89> Anyone? Is this caused by the framerate or by the encapsulation?
[16:37] <varaderoguy> Hello all
[16:37] <varaderoguy> Need some advice on FLV media
[16:38] <varaderoguy> I've got a problem with audio on all of my clips gradually becoming slow
[16:39] <varaderoguy> I suspect that I need to use the -async flag;
[16:39] <varaderoguy> Need to understand how this works?
[16:42] <varaderoguy> okay - maybe -asyncts
[16:43] <varaderoguy> The issue I've having is that the audio/video are correct at the start of the game, but gradually become further apart with the audio becoming slow
[16:43] <Spideru> Hi. I would to open an rtp stream generated with: avconv -f alsa -ar 16000 -ac 1 -i default -f rtp -f s16le rtp://localhost:12345. What should i use? Thank you (avplay rtp://localhost:12345 didn't worked
[16:43] <varaderoguy> this means that if I have a person kicking a football, then I get the sound effect (at extremis) 3 seconds late
[16:44] <varaderoguy> the further the clip goes on, the worst the problem becomes
[17:13] <Spideru> still have problem on reproducing rtp stream
[17:23] <mbrit> dont you need an sdp file to play an rtp stream?
[17:23] <Spideru> how i can generate it? the command wrote up doesn't do it
[17:24] <Spideru> *doesn't
[17:24] <Spideru> I need to generate s16 16khz stream
[18:33] <pespin> hi, any idea on what should I set to get an h264 output with num_ref_frames = 1 ?
[18:33] <pespin> I tried setting context->refs=1 but I always get num-ref_frames=3 on the encoded h264 stream
[20:04] <jShaf> -ss option states that the "position may be either in seconds or in hh:mm:ss[.xxx] form."
[20:04] <jShaf> but i want milliseconds
[20:04] <jShaf> can go with 0.25 for example?
[20:05] <Tjoppen> hat do you think ss[.xxx] means? :)
[20:05] <Tjoppen> oh, you mean plain seconds? yeah, should work
[20:41] <Sashmo> hey everyone, has anyone noticed any significant benfit by using ffmpeg on a 64 bit machine?  Im using 64 bit ubuntu.
[20:48] <cbreak> as opposed to not using ffmpeg?
[20:50] <Sashmo> no oppsed to not using 32 bit vs 64 bit
[20:51] <newl> i doubt ffmpeg has limit to 2G when writing a file
[21:06] <jShaf> Tjoppen: right, just the seconds, not the timestamp
[21:07] <jShaf> actually
[21:07] <jShaf> hh:mm:ss[.xxx] starts from 00:00:00 position?
[21:07] <jShaf> meaning that i can use 00:00:00.250 ?
[21:17] <raptor67682> do you know if ffserver works on a current Squeeze distro?
[21:34] <Tjoppen> jShaf: should work, assuming the file starts at zero
[21:35] <raptor67682> sounds that not...
[21:35] <raptor67682> ffserver happens not to work at all on squeeze stable. You should try to install it and run it with qmeu
[23:17] <undercash> hello
[23:17] <undercash> if i update rtmpdump do i need to recompile ffmpeg?
[23:19] <newl> what makes you think that?
[23:19] <newl> they don't even share libs
[23:22] <undercash> but ffmpeg use it to output to rtmp right?
[23:23] <undercash> so it s cool i just update rtmpdump and i m fine
[23:23] <newl> do a ldd on each to see what they each have or don't have in common
[23:55] <seablade> Quick question, I have a list of image files named DSC_****.jpg I need to make into a stop motion video... my ffmpeg command on Linux is `ffmpeg -f image2 -i 'DSC_%04d.jpg' -r12 -vcodec libx264 ani.m4v` and I am getting an error of `DSC_%04d.jpg: No such file or directory`... what am I doing wrong?
[00:00] --- Wed Jul 11 2012