[Ffmpeg-devel-irc] ffmpeg.log.20130705

burek burek021 at gmail.com
Sat Jul 6 02:05:01 CEST 2013


[00:45] <pfs35> Hello all. I'm trying to reencode a video I had problems playing using my Raspberry Pi yesterday, and I have a read error (most likely what caused the problems in the playback also). Is there any way to force ffmpeg to skip the problematic frames? I've tried using "-ec:v guess_mvs," but it hasn't solved the problem.
[04:11] <diego> hi
[04:11] <diego> how do i encode a video and discard the audio from the encoding?
[04:12] <diego> -an i think
[04:12] <diego> sorry
[04:35] <diego> http://pastie.org/8111621
[04:35] <diego> i'm trying to encode some video but i get this
[04:35] <diego> any ideas please?
[04:36] <diego> it doesn't write anything to the file
[04:36] <diego> except 4kb or something
[04:37] <diego> if i remove -ss the encoding works
[04:38] <diego> but i'm trying to tell ffmpeg "encode only from minute 24 until the end of the video"
[04:38] <diego> i wonder what i'm doing wrong
[04:47] <osmodiar> The file says: Duration: 00:01:30.18
[04:47] <diego> oh
[04:47] <osmodiar> So you can't start from 24 minutes into it.
[04:47] <diego> right
[04:47] <diego> sorry
[04:47] <diego> i just noticed that
[04:47] <diego> i'm an idiot
[04:47] <diego> sorry
[04:47] <diego> it should be 00:00:24 i think
[04:48] <diego> yep that's it, thanks
[04:49] <diego> ffmpeg -i VID_20130704_224456.mp4 -b:v 1000k -an -ss 00:00:24 out.webm
[04:51] <diego> this works, thanks
[04:52] <the1_> hello, im restreaming an RTSP stream to 3 bitrates and 1 audio only. my command looks like this:
[04:52] <the1_> ffmpeg -i rtsp://<rtsp.ip>:1935/live/aksyon3 -i /tmp/logo.jpg 			-vcodec libx264 -vprofile baseline -level 3 -b:v 250000 -g 45 -s 320x240 -acodec libfdk_aac -ar 44100 -ab 48k -ac 1 -r 15 -f flv rtmp://localhost/live/high                         -vcodec libx264 -vprofile baseline -level 3 -b:v 200000 -g 45 -s 320x240 -acodec libfdk_aac -ar 44100 -ab 48k -ac 1 -r 15 -f flv rtmp://localhost/live/medium                         -vcodec lib
[04:53] <the1_> my questions is, how do i add logo.jpg to audio stream so that it will have a static image
[04:56] <osmodiar> I think you'd have to encode it as a video using -loop_input...
[04:57] <the1_> and mapping?
[04:57] <mark4o> add a logo to the audio stream?
[04:57] <the1_> yes thats correct
[04:57] <mark4o> text to speech?
[04:57] <the1_> no just a logo
[04:58] <mark4o> so you want to add a video stream
[05:00] <mark4o> http://ffmpeg.org/ffmpeg-filters.html#Examples-49
[05:00] <the1_> im not sure if we call that a "video stream" if i add just a logo to the audio stream
[05:03] <the1_> would it still be considered as video by ffmpeg?
[05:04] <mark4o> do you have an audio file, and you want to create a video file?
[05:04] <the1_> the audio will be coming from an rtsp source
[05:05] <mark4o> and you are recording it to a file?
[05:05] <the1_> and im publishing to 3 bitrates and 1 audio-only
[05:05] <the1_> but i need to have the audio-only stream with a logo
[05:05] <the1_> im recording it just for HLS purposes, it gets purged after a short time
[05:06] <mark4o> oh you are sending rtsp, not receiving?
[05:07] <the1_> receiving rtsp and sending rtmp
[05:08] <the1_> :)
[05:08] <mark4o> I don't know rtsp, but I would imagine that you'd send it as a video stream if you want images
[05:08] <the1_> ic
[05:11] <mark4o> the1_: for a file it would be something like ffmpeg -loop 1 -i image.jpg -i audio -map 0:v -map 1:a -pix_fmt yuv420p output
[05:11] <the1_> in my example above where publishing to 4 outputs, how do i make the logo go to the last output
[05:12] <mark4o> the1_: sorry, don't know how to do it for rtsp
[05:14] <mark4o> oh and -c:a copy if you want it to copy the audio and not transcode it
[05:14] <the1_> i think its the same as file input, i just need to map the inputs to outputs
[05:17] <mark4o> ok then maybe replace "output" with something like -c:a copy -f rtsp rtsp://server/video.mp4
[05:18] <mark4o> and replace audio with your input audio source
[19:46] <enBrianNano> hey, what preset for libx264 do you guys think give you the best combination of speed of size?
[19:47] <Mavrik> um
[19:47] <Mavrik> probably default ;)
[19:49] <sdl420> depends on source?
[19:50] <enBrianNano> x264 source, but I'm burning in the subtitles
[19:50] <atiti> hey guys, I'm trying to decode single h264 NALUs with avcodec
[19:50] <sdl420> why do you want to burn in subtitles?
[19:50] <osmodiar> enBrianNano: Use the slowest you can bare to use. You only have to encode once but you keep the file for longer.
[19:50] <enBrianNano> so I can watch the videos on my ps3
[19:51] <sdl420> burn in means hardcoeded, right?
[19:51] <enBrianNano> yep
[19:51] <sdl420> which container?
[19:51] <enBrianNano> mkv to mp4
[19:52] <sdl420> so you need mk4 for ps3 i guess. i cant say any about mp4:/
[19:52] <enBrianNano> I've done a bunch already; now, I'm just messing with the preset to see if I can get smaller files
[19:52] <sdl420> but i would try just du demux the mkv and mux it into a mp4
[19:53] <Mavrik> enBrianNano, preset doesn't affet filesize
[19:53] <osmodiar> Mavrik: They do a bit...
[19:53] <Mavrik> sdl420, what are you talking about? PS3 doesn't support soft subs so you have to reencode the video.
[19:54] <sdl420> ok
[19:54] <sware> anyone have a windows config.h ?
[19:54] <Mavrik> osmodiar, unmeasurable amount when using crf, a very tiny amount when using bitrate. they do affect quality at that bitate, yes.
[19:54] <sware> from latest version
[20:00] <rager> so like... I just want to do some very basic reencoding with libavcodec and such
[20:01] <rager> but I don't know the first place to start - any good reading I should do?
[20:01] <Mavrik> well :)
[20:01] <Mavrik> rager, doc/examples and the doxygen documentation is pretty much the only place with documentation for that :/
[20:01] <rager> my issue is partially that I haven't done a single practical thing in C
[20:02] <Mavrik> then you're f'ed.
[20:02] <Mavrik> using libav* libraries requires you to at least be basically competent in C development since they're written to be fast, not pretty
[20:03] <rager> alternately
[20:05] <rager> is there a way to just invoke main() from another C file rather than calling a binary file?
[20:05] <rager> I'm really just trying to do some relatively simple stuff with JNI in android
[20:06] <Mavrik> no, because it would collide with your own main()
[20:13] <rager> I saw one attempt using dlsym and dlopen
[20:14] <Mavrik> rager, that's just doing dumb stuff from ignorance
[20:14] <Mavrik> instead of just renaming main() to a function name and linking it to your binary.
[20:14] <rager> I can see some people not wanting to alter ffmpeg for licensing reasons
[20:15] <rager> though I guess I really should just rename main() to whatever JNI likes with... configure? make?
[20:15] <rager> you can rename main() without actually renaming it in ffmpeg.c/.h, right?
[20:46] <downwater> hello
[20:46] Action: rager waves
[20:49] <downwater> i'm looking for relevant param to convert into wmv format which fits Windows Media Player "video for local playback (pal 2,1 Mbps/s)" ... i tried : -vcodec wmv2 -acodec wmav2  -b:v 1400k  -r 25
[20:49] <downwater> but i'm not sure it matches
[20:49] <downwater> err s/1400/2100/
[20:50] <Mavrik> rager, no.
[20:50] <rager> ok
[20:50] <Mavrik> you cannot rename function without actually renaming it.
[21:25] <MachinaeWolf> How did audacity capture the right tempo of the system sound but ffmpeg didn't??? How do I fix that?
[21:36] <durandal_1707> MachinaeWolf: alsa?
[21:36] <MachinaeWolf> Yeah I have alsa and pulse because I don't think the card I have supports the mixing deal to be able to just run with alsa
[21:41] <durandal_1707> MachinaeWolf: what ffmpeg version?
[21:42] <MachinaeWolf> 1.2.1
[21:43] <durandal_1707> and what samplerate is reported?
[21:44] <MachinaeWolf> hmm multimedia repo tookover there, maybe I need to fix that
[21:44] <MachinaeWolf> um don't know how to check that
[21:57] <rager> Mavrik: what is the dlopen and why is it just stupid?
[22:07] <MachinaeWolf> durandal_1707?
[22:09] <rager> Mavrik: it looks to just be used to load the ffmpeg lib, then call the function "main" with the given parameters
[22:19] <Chat9999> Hi
[22:21] <durandal_1707> MachinaeWolf: what multimedia repo?
[22:26] <MachinaeWolf> durandal_1707, the http://www.deb-multimedia.org repo
[22:28] <MachinaeWolf> durandal_1707, I didn't do a giant paste of anything...???
[22:28] <MachinaeWolf> wtf?
[22:28] <MachinaeWolf> I typed out that url that's it...
[22:29] <durandal_1707> give exact command you use for recording
[22:29] <durandal_1707> and its output
[22:29] <MachinaeWolf> oh
[22:32] <MachinaeWolf> http://pastebin.com/ijUWA25i
[22:35] <durandal_1707> hmm, and whats wrong with sound?
[22:36] <MachinaeWolf> it's distorted or at least tempo is way slower
[22:48] <durandal_1707> well its either wrong samplerate or number of channels
[22:52] <llogan> MachinaeWolf: does an output by arecord also sound weird?
[22:52] <MachinaeWolf> idk, never recorded with that
[23:01] <MachinaeWolf> So how do I do that llogan?
[23:07] <MachinaeWolf> well idk how to do it properly so it didn't record anything...
[23:11] <MachinaeWolf> I'm about to just rid myself of the debian multimedia repo and just use the regular versions
[23:21] <llogan> MachinaeWolf: i don't know. i've never used it.
[00:00] --- Sat Jul  6 2013


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