[Ffmpeg-devel-irc] ffmpeg.log.20131106

[burek]

[00:02] <ac_slater> llogan: right I needed a little than strings could give me thanks mate
[00:03] <ac_slater> (ie - nice output :p)
[00:31] <IFN10EBrain> Hello #ffmpeg. I'm trying to make a screencast using ffmpeg but the sound dies after a few seconds of recording. How can I fix this? The mic is a digital integrated one on a notebook.
[00:31] <triggerhapp> I'm trying to work out how to livestream with synchronized audio... currently no matter what I try the audio runs out of sync (specifically, behind) the video - is there something else I could be doing? http://pastebin.com/UPagVsP9
[00:31] <triggerhapp> sounds fun IFN10EBrain, what are you doing for a command atm ?
[00:33] <IFN10EBrain> Its this: "ffmpeg -f alsa -i pulse -f x11grab -r 25 -s 1024x600 -i :0.0 -acodec pcm_s16le -vcodec huffyuv -sameq Screencast.avi". The sound works for a few seconds and dies.
[00:34] <triggerhapp> why use alsa, couldn't you just '-f pulse'?
[00:35] <IFN10EBrain> triggerhapp: pulse errors: pa_simple_new failed: No such entity
[00:36] <IFN10EBrain> triggerhapp: it's an input/output error.
[00:36] <IFN10EBrain> That's when I use -f pulse.
[00:36] <IFN10EBrain> -f alsa kinda works. But the sound is just a few seconds.
[00:47] <IFN10EBrain> Oy oy... I found a solution. It works :)
[00:54] <triggerhapp> Ooh, what'd you try?
[00:54] <IFN10EBrain> triggerhapp: I just got sound to work :s
[00:54] <IFN10EBrain> I need to get video With sound.
[00:55] <IFN10EBrain> I'm trying a mischmash of different things I found on the net.
[00:55] <triggerhapp> I meant using what method :P
[00:55] <triggerhapp> but I can't say much, I'm doing the same trick to try to get my sound to sync :)
[00:56] <IFN10EBrain> triggerhapp: setting -i hw:0,0 worked for sound. But in combination with video...it doesn't work either.
[00:59] <IFN10EBrain> The worst thing about using an extremely powerful tool is not knowing how to use it.
[01:00] <triggerhapp> My issue seems worse. I'm using a powerful tool that seems to be butchered into unusable :)
[01:01] <triggerhapp> having to deal with avconv instead of ffmpeg
[01:07] <llogan> don't use -sameq. it does not mean same quality, makes no sense with a lossless encoder, and has been removed from ffmpeg anyway
[01:07] <triggerhapp> I... did use pastebin?
[01:07] <llogan> ah...i missed it
[01:07] <triggerhapp> :P
[01:08] <llogan> avconv is not from FFmpeg
[01:08] <llogan> if you want help here you'll have to use ffmpeg from FFmpeg
[01:12] <IFN10EBrain> I'll get back to it later. I need to go to sleep. Nite nite.
[01:26] <triggerhapp> llogan, using a static build and it doesn't recognise x11grab as an input format. Short of compiling (since this is intended to be a python app) what could I do?
[01:27] <llogan> triggerhapp: the static builds don't have x11grab support
[01:28] <triggerhapp> that one I could gather :P Short of compiling it myself how can I get a version for ubuntu which will support it?
[01:28] <triggerhapp> any advised PPA's at all?
[01:29] <llogan> you could try https://launchpad.net/~jon-severinsson/+archive/ffmpeg but it only offers 0.10 & 0.7 release branches
[01:30] <llogan> ...or compile http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[01:30] <triggerhapp> The issue with compiling is to then get a compatible version for every user of my script I either drop them a lot of binary libraries and exec, or make them compile too
[01:30] <triggerhapp> In theory I love compiling... just not for a python app
[01:34] <llogan> maybe it would be easiest for you to include a binary so you don't have to deal with various versions, forks, and option changes
[01:34] <llogan> ...and hopefully x11grab will work as expected for everyone.
[01:35] <triggerhapp> And people running on arm? :P
[01:37] <llogan> sure, if they capture 180x120
[01:39] <triggerhapp> The raspberry pi manages 1080p just fine, I'm targetting desktops and pi's right now
[01:41] <triggerhapp> next question... which version is it that filter_complex is in? seems to be missing from this 0.10
[01:44] <triggerhapp> ... I'm gunna sleep actually. Morning shift
[04:59] <hailwood> Hi Guys, is there a concensus on how to best web optimize a video with ffmpeg?
[09:09] <plepere> hello
[14:48] <sspiff> modifying ffmpeg.c to keep the clear packet in a big, dirty static AVPacket dvbsub_clear, and only sending it out as soon as a subtitle packet with a higher PTS occurs "fixes" the transcoding
[14:49] <sspiff> obviously, the last clear will be missing
[14:49] <sspiff> which is (probably? you tell me...) unacceptable
[14:53] <sspiff> oops, wrong channel again!
[16:44] <dharriso> hi, im using the ffmpeg API to transcode AAC video over HTTP in a proxy.  this works well.  I have a need to specify the AAC sample size up front
[16:44] <dharriso> in an index (moov box) thats sent to the client.  However sometimes i need to pad the AAC sample up to the size I initaly specified in the index.
[16:44] <dharriso> In my tests I have padded frames with hex zeroes and it seems to play on some desktop player.
[16:44] <dharriso> Is padding with hex zeroes the best way to acheive this?
[16:49] <dharriso> should i post this question on ffmpeg-devel, though the ffmpeg.org website indicates this is  the correct place
[17:16] <sspiff> dharriso: I think this is the right place. Also, people from #ffmpeg-devel are also in this channel, so they'll see it eventually, I hope :)
[19:51] <noobed> hello everybody - there're only 2 IRC channels and since I'm making software that uses FFMPEG I'm writing here.
[19:51] <noobed> Could I ask a couple of questions?
[19:51] <noobed> 1) I think I've extracted the audio stream from an .flv file and successfully wrote the raw audio - it's a 2 channel audio stream and i've used the resampling audio example
[19:51] <noobed> so is there any way I can test it's properly written on my hdd ?
[19:51] <noobed> 2) after extracting the raw audio I want to encode it into mp3 but I get
[19:51] <noobed> "Invalid data found when processing input "
[19:51] <noobed> when running
[19:51] <noobed> avformat_open_input(&avFormatContext, argv[1], NULL, NULL); // where avFormatContext is NULL at that point and argv[1] contains the raw audio file name]
[21:59] <memand> Hey guys, can I make ffplay stream to stdout instead of just streaming to everything on the soundcard?
[22:06] <relaxed> memand: isn't that a job for ffmpeg?
[22:07] <memand> might be, the thing I want to do is to be able to take a music file and output it into aplay so I can choose what sound port it comes out from
[22:10] <memand> relaxed: ^
[22:10] <memand> I tried this 'ffmpeg -i 3938033.mp3 - | aplay -f cd -D front'
[22:11] <memand> but it says:
[22:11] <memand> Unable to find a suitable output format for 'pipe:'
[22:11] <memand> pipe:: Invalid argument
[22:12] <sacarasc> memand: ffmpeg -i blah.mp3 -f pcm -
[22:15] <roothorick> so, curious, why does ffmpeg want yasm/nasm and doesn't just use GNU as?
[22:17] <ShadowJK> syntax
[22:18] <roothorick> what syntax specifically?
[22:19] <ShadowJK> it's less of a headache to write code for nasm syntax than gnu as
[22:19] <roothorick> is it a preprocessor thing?
[22:20] <roothorick> my curiosity was mostly spurred by a bizarre errorspam I've been getting while building 2.1 in mingw-w64, but only on one certain machine which just boggles my mind
[22:20] <relaxed> I don't think ffmpeg needs nasm, only yasm.
[22:22] <roothorick> it's either/or
[22:22] <roothorick> and I couldn't get yasm to compile under mingw so nasm it is
[22:27] <roothorick> the errorspam seemed to suggest that it was trying to assemble 64bit code in a 32bit mode
[22:27] <roothorick> but I'm having no issues on a 32bit mingw on a different 64bit machine
[22:31] <dbro> Can anyone help debug some video encoder output: https://s3.amazonaws.com/dbro/h264_madness/ffmpeg_1383772856149.ts
[22:31] <dbro> [h264 @ 0x7f9be2818400] corrupted macroblock 11 29 (total_coeff=-1)
[22:31] <dbro> [h264 @ 0x7f9be2818400] error while decoding MB 11 29
[22:31] <dbro> working on a FOSS Android library to mate Android's hardware encoder to ffmpeg: https://github.com/OnlyInAmerica/FFmpegTest
[22:36] <noobed> Could some1 help me - I'm getting a raw audio from an .flv file.... afterwards I try to open that raw audio with avformat_open_input() and I'm getting an error " Invalid data found when processing input " ... I've posted a thread on SO http://stackoverflow.com/questions/19823260/c-ffmpeg-decode-into-raw-audio-which-cannot-be-read-afterwards ... this's been bugging me for days now :(
[22:38] <mikepack> Does anyone have experience converting .3gp files? I'm having trouble aggregating 3gp files into a compilation with other files types (mainly mp4).
[23:19] <noobed> several ffmpeg/libavdevice/sdl.c:64: undefined reference ..... any clues what library to add in my makefile?
[00:00] --- Thu Nov  7 2013