[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-w... ... 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