[Ffmpeg-devel-irc] ffmpeg.log.20130812

[burek]

[11:14] <ItsMeLenny> is there any program in linux thats similar to a jack audio monitoring program which shows spectrums and waveforms but does it for videos
[11:14] <ItsMeLenny> but not for videos, for video input i should say, webcam, blackmagic (if i can even get it working)
[11:46] <cbsrobot> ItsMeLenny: see https://trac.ffmpeg.org/ticket/1624
[11:48] <ItsMeLenny> i just thought up a way to calibrate this actually
[12:04] <ItsMeLenny> is cbsrobot a bot?
[12:06] <relaxed> half man, half machine
[12:35] <_dan_> hallo
[12:35] <_dan_> av_interleaved_write_frame(): Invalid argument
[12:35] <_dan_> what does it mean?
[12:53] <jure> it means stop using -vsync drop
[12:53] <jure> it doesn't work
[15:28] <elmargol> Hi I'm using ffmpeg+zoneminder to moinitor a parkingspace. Somehow my RTSP frames are corrupted.
[15:28] <elmargol> maybe related to "https://ffmpeg.org/trac/ffmpeg/ticket/285". Any ideas how to fix this?
[16:33] <cirenyc> How can I add a 5s image to the front of a video?
[17:27] <raduu> hi, can anyone give me a hand with this one: http://stackoverflow.com/questions/18132342/ffmpeg-rtmp-streaming-process-exit
[17:34] <raduu> echo
[17:34] <klaxa> that behaviour is not expected i think
[17:35] <raduu> thought so.. but any causes for it to behave like this?
[17:48] <spaam> raduu: define crash
[17:49] <spaam> i dont see " process exits " as a crash
[17:49] <raduu> video:5264kB audio:0kB subtitle:0 global headers:0kB muxing overhead -100.000408%
[17:49] <raduu> and then I get the bash $
[17:49] <spaam> okey so just an exit then.
[17:50] <raduu> right
[17:52] <stefanha> I have a reference .ogg file with n samples of audio.  I'd like to generate a silence file with n samples of audio using ffmpeg -i /dev/zero.
[17:52] <raduu> so basically I try to run ffmpeg -threads 4 -i rtmp://..../chat/mp4:<variable>.mp4
[17:52] <raduu> -q:v 0.6 -r 15 -s 320x240 /frames/10021237_data/frame-%0999d.jpg
[17:52] <stefanha> I've tried setting the duration in seconds, using ffmpeg -t <secs> but that isn't sample-accurate,
[17:52] <raduu> in 2 terminals, with 2 different streams
[17:53] <stefanha> and I'm noticing that the length of the reference file and the silence file are not identical.
[17:53] <raduu> and when one ends (or I kill the process)... the other one exits
[17:53] <stefanha> Any suggestions on generating an .ogg file with an exact number of samples of silences?
[17:53] <spaam> raduu: try lower frame-%0999d.jpg the number there to something lower
[17:53] <spaam> raduu: and you will get some pictures
[17:53] <spaam> and i guess muxing thing is beacuse of .jpg
[17:54] <spaam> since there is no "container" for .jpg
[17:54] <raduu> hwo do I do that?
[17:55] <spaam> what ?
[17:55] <raduu> lower the frame
[17:56] <spaam> why do you have -r 15 and the frame number?
[17:56] <raduu> I tried with different frame rates
[17:56] <raduu> but same behaviour..
[17:56] Action: stefanha tries dding an exact number of /dev/zero bytes and then .ogg encoding that
[17:56] <raduu> I didn t get why you pasted frame-%0999d.jpg
[17:57] <raduu> I thought that is just a tweak to produce the same image name always
[17:57] <spaam> i dont see the problem.
[17:58] <raduu> :)
[17:58] <raduu> <spaam> raduu: try lower frame-%0999d.jpg the number there to something lower
[17:58] <raduu> what do you mean?
[17:58] <spaam> i didnt know that you chould do that to get the frame number
[18:01] <raduu> that gives me frame-0000000000000000000.jpg
[18:01] <raduu> all the time
[18:07] <raduu> spaam: I m not really sure what I m doing.. I m just trying to produce images from a stream
[18:08] <raduu> and it works really well for 1 stream
[18:08] <raduu> I have no clue what a muxer is in the first place
[18:08] <raduu> so if there is something wrong with the command please let me know
[18:27] <zap0> raduu,  get a container (avi etc),  stick in some audio (wav/pcm/mp3)  and some video (h264,mp4,yuv).. mux it all up, and BLAMMO!  you have a video
[18:27] <raduu> I do not want a video.. I want a sequence of images
[18:27] <raduu> from an mp4 adobe fms stream
[18:28] <zap0> that's nice dear.
[18:28] <raduu> ?
[19:26] <relaxed> !burek
[19:27] <burek> !beer :)
[19:28] <relaxed> burek: Could you update the bot to point to http://johnvansickle.com/ffmpeg ?
[19:30] <relaxed> also, the kernel requirement is now 2.6.32+
[19:31] <burek> ok
[19:31] <burek> just a sec
[19:32] <relaxed> Thanks, a lot of businesses block dropbox now.
[19:33] <burek> I can also give you a hosting place if you need
[19:33] <burek> could maybe setup a cron
[19:33] <burek> to download the build and publish it on the web root
[19:34] <relaxed> It's what I do now but I've been too lazy to make a cron job for it.
[19:35] <relaxed> We should probably scrape fate to check whether it's green or not.
[19:35] <burek> :)
[19:36] <burek> i think i made mine with netcat or something
[19:37] <relaxed> Oh, you do check fate?
[19:37] <burek> which acts like a dummy web server that only serves the given file on a command line
[19:37] <burek> er no. not quite :)
[19:37] <burek> i was just trying to say that there is no need to install a full-featured web server like apache
[19:37] <burek> just to be able to get your build
[19:38] <stefanha> I am trying to create an .ogg file containing silence.  It should be just as long as a reference file.
[19:38] <relaxed> I've already purchased hosting for this and other things, so that's not an issue.
[19:38] <relaxed> stefanha: what have you tried?
[19:38] <stefanha> It looks like the encoding process has some kind of block size, the silence file does not end up with the exact same number of samples as the reference:
[19:38] <stefanha> $ dd if=/dev/zero bs=2 count=352874 | ffmpeg -y -ar 44100 -ac 1 -f s16le -i - -acodec libvorbis out.ogg
[19:38] <stefanha> $ ffprobe -show_streams -i out.ogg | grep duration_ts
[19:38] <stefanha> duration_ts=352896
[19:39] <stefanha> 352896 != 352874
[19:39] <relaxed> you could use -i /dev/zero, fyi
[19:39] <relaxed> but that's not your problem
[19:39] <stefanha> relaxed: I tried that but only found ffmpeg -t <secs>, which isn't sample-accurate either
[19:40] <stefanha> Later on I mix together multiple audio files, and if their length is not identical, things get out of sync.
[19:40] <stefanha> That's why I'm trying to create a file with a precise length.
[19:41] <relaxed> right, give me a second
[19:42] <stefanha> relaxed: thanks!
[19:43] <stefanha> I just noticed another weird thing: converting an .ogg file to .wav results in a different duration_ts.
[19:43] <stefanha> That's just decoding to wav, no ogg/vorbis encoding.
[19:44] <stefanha> The .wav has a larger duration_ts value.
[19:45] <durandal_1707> by how much?
[19:47] <stefanha> durandal_1707: 598 samples (ogg: 352874, wav: 353472)
[19:48] <durandal_1707> you use vorbis, and it may add silence for last frame
[19:48] <durandal_1707> so no you can not use vorbis to encoder exact number of samples
[19:49] <stefanha> durandal_1707: I am trying to match a reference .ogg file, so it doesn't need to be arbitrary numbers of samples,
[19:49] <stefanha> it just needs to be the same as an existing .ogg file.
[19:52] <durandal_1707> that is certainly useless objective, if file and not decoded data must be bit by bit same
[19:53] <stefanha> durandal_1707: It's about timing.  I will concatenate and mix several tracks.
[19:53] <stefanha> If the silent piece has a different length, then the tracks will be out of sync.
[19:54] <burek> why not creating identical length wav files as a start, and do whatever magic you need
[19:54] <relaxed> maybe use raw formats until you have to encode to lossy
[19:54] <stefanha> The input is already .ogg and my goal is to concat ogg without reencoding
[19:54] <stefanha> I don't have raw input here.
[19:55] <durandal_1707> yes, but you can not have arbitary duration with vorbis
[19:55] <stefanha> durandal_1707: The fact that the reference file exists suggests vorbis can encode that length :)
[19:56] <stefanha> It must be possible but I guess there is some buffer size parameter somewhere.
[19:56] <durandal_1707> stefanha: hmm, it could be really possible, but not atleast with libvorbis from ffmpeg
[19:57] <burek> it's like "I want to create an mp3 file of specific size in bytes"... how will you accomplish it easily?
[19:57] <durandal_1707> but if you can compile ffmpeg yourself i could give you patch which you could try
[19:58] <stefanha> durandal_1707: That sounds interesting but maybe I should explain what I'm trying to do first.  Perhaps I'm doing it in a silly way :)
[19:58] <stefanha> There are several tracks of audio, like vocals, drums, guitar.  They come in .ogg except only in 8 second pieces.
[19:58] <stefanha> So you have guitar_1.ogg, guitar_2.ogg, etc.
[19:59] <stefanha> And I'm mixing them down into a single audio file for listening (that part works).
[19:59] <stefanha> The tricky part is that there is also silence sometimes, so guitar_4.ogg, guitar_6.ogg <--- #5 is missing (silent)
[19:59] <stefanha> durandal_1707: So what I'm doing at the moment is to find those silent parts and generate silence files with ffmpeg.
[20:00] <stefanha> Then I can concat all the guitar tracks (including silent files I generated) into guitar.ogg
[20:00] <stefanha> And finally I can mix guitar.ogg, drums.ogg, vocals.ogg down into a single audio file.
[20:00] <relaxed> can you concat oggs losslessly?
[20:00] <stefanha> relaxed: If they have the same number of channels, sample fmt, etc yes.
[20:01] <durandal_1707> good, but I still wonder why you use ogg and vorbis for intermediate step
[20:01] <stefanha> I want to keep the concat .ogg tracks so they are lossless.
[20:01] <stefanha> So you can have the guitar.ogg file and it's not reencoded.
[20:02] <stefanha> That's nice to have, but I can drop it if necessary.
[20:02] <durandal_1707> well if you must use vorbis, and need to create silence of arbitary length in vorbis that is fine
[20:02] <stefanha> Are you suggesting decode it all, add sample-accurate silences, and then mix + encode?
[20:03] <durandal_1707> stefanha: no, concatenate, without any transcoding should be enough
[20:04] <durandal_1707> so yes it perhaps should be possible to encode vorbis with arbitary samples, i just need to try it
[20:10] <durandal_1707> or you can try it right a way if you can compile ffmpeg
[20:13] <stefanha> durandal_1707: Sure, I can try building from source
[20:13] <durandal_1707> you tried it already?
[20:13] <stefanha> durandal_1707: Nope, just cloning ffmpeg.git now
[20:14] <durandal_1707> you just add CODEC_CAP_SMALL_LAST_FRAME to libvorbisenc .capabilities
[20:18] <stefanha> durandal_1707: thanks, testing now
[20:24] <stefanha> durandal_1707: Unknown encoder 'libvorbis'
[20:24] <stefanha> durandal_1707: I must have built without libvorbis support...
[20:24] <stefanha> I did ./configure && make
[20:26] <stefanha> Yeah, config.mak says "!CONFIG_LIBVORBIS_ENCODER=yes" :(
[20:28] <ubitux> external libraries need explicit enable switch
[20:28] <ubitux> --enable-libvorbis --enable-libx264 ...
[20:28] <stefanha> ubitux: Thanks, just enabled it
[20:34] <durandal_1707> hmm it seems it does not help...
[20:38] <stefanha> durandal_1707: It makes a difference for encoding here.  Now I'm able to encode 352874 samples exactly.
[20:38] <stefanha> $ dd if=/dev/zero bs=2 count=352874 | ffmpeg -y -ar 44100 -ac 1 -f s16le -i - -acodec libvorbis out.ogg
[20:39] <durandal_1707> hmm...
[20:39] <stefanha> durandal_1707: It seems to work for encoding, thanks! :)
[20:39] <durandal_1707> what libvorbis you have installed?
[20:40] <durandal_1707> perhaps it works only for silence ........
[20:40] <stefanha> durandal_1707: libvorbis 1.3.3-4.fc19
[20:41] <stefanha> I also tried 352875 (+1) and that worked too.
[20:42] <stefanha> durandal_1707: Still working for /dev/urandom here
[20:42] <durandal_1707> i'm puzzled it does not work with random flac i tried
[20:42] <durandal_1707> and i also have 1.3.3
[20:42] <stefanha> durandal_1707: What if you try raw s16le input instead of flac?
[20:43] <durandal_1707> it should not matter ...
[20:44] <stefanha> durandal_1707: Here decoding an .ogg test file with 352874 samples to .wav still results in 353472 samples.
[20:45] <stefanha> durandal_1707: So I wondered if flac decoding has the same issue.
[20:48] <durandal_1707> flac decoding is fine
[20:56] <stefanha> durandal_1707: Thanks for your help.  Got to go now.
[21:41] <jstackhouse> Hello! I've got a weird problem where I'm converting a video into jpegs, and around ~6-7k jpegs in, ffmpeg goes to 0% CPU usage and just sits. It doesn't exit, or throw an error. Anyone heard of that? Or have an idea what may be happening?
[21:42] <Keshl> jstackhouse: Just a hunch, your disk isn't filling up, is it?
[21:42] <jstackhouse> Nope, it's running on EC2, still got ~29GB on the drive. :S
[21:43] <durandal_1707> what is command? and what ffmpeg version?
[21:43] <Keshl> No idea, then. .É. Wait for someone who actually knows what they're talking about like durandal_1707, oÉo.
[21:44] <jstackhouse> The command is.. ffmpeg -i /path/to/mp4 -s 384x212 -f image2 -qscale 1 -vf fps=fps=24 /path/to/out%d.jpg
[21:44] <jstackhouse> ffmpeg version git-2013-08-12-d4ab129 Copyright (c) 2000-2013 the FFmpeg developers
[21:44] <jstackhouse>   built on Aug 12 2013 16:27:28 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
[21:45] <durandal_1707> perhaps limit on number of files in directory?
[21:45] <durandal_1707> i just thinking loud what 0% CPU means
[21:45] <durandal_1707> stalling  is far from optimal solution....
[21:46] <jstackhouse> The odd part is that it works fine doing like 6 videos at a time that are of like length 30 seconds to 4 minutes.
[21:46] <jstackhouse> but this one video that is over 9 minutes, causes this issue.
[21:46] <jstackhouse> Happens on my local machine too.
[21:46] <jstackhouse> I'm running OSX, the server is Ubuntu 12.04
[21:47] <durandal_1707> hmm, so you want to say it depends on input and not number of outputs?
[21:47] <jstackhouse> I suppose yea, initially we thought HTTP might be timing out, so we tried wget and run ffmpeg locally, but it still happens.
[21:48] <durandal_1707> well i would try another file and see if creating 6-7k files will work....
[21:49] <Keshl> It's definitely not a limit on the number of files per folder. I've had far more.
[21:49] <Keshl> Heck. Right now I'm looking at a folder with 32,189 items in it.
[21:50] <jstackhouse> ffmpeg doesn't hold open the file descriptors does it? maybe it is hitting ulimit?
[21:50] <durandal_1707> ahh or it may be recently reported issue of max opened files?
[21:50] <Keshl> jstackhouse: If it was it couldn't even do 20.
[21:50] <durandal_1707> though it should close file when writting
[21:50] <Keshl> The limit most OS's stick on open files by a single program is 20.
[21:51] <Soe1en> hello everyone! I just started to use ffmpeg and I have lots of questions if I may ask!
[21:51] <Keshl> Soe1en: Go ahead, dun ask to ask, oÉo
[21:52] <Soe1en> heh : ) I tried to take a look what de- and encoders are available with ffmpeg -decoders and ffmpeg -encoders, but I get Missing argument for option 'de- encoders' for some reason, how come?
[21:53] <Keshl> No idea, I get that too. x.x (Obviously not an expert here <É<)
[21:54] <Soe1en> well, shit, the documentation is suggesting this, yet it is not working
[21:54] <durandal_1707> Soe1en: that works fine here
[21:54] <Soe1en> durandal_1707: in fact a lot of things doesn't seem to work
[21:54] <Keshl> durandal_1707: You seem incredibly blessed, then.
[21:55] <durandal_1707> in fact perhaps you are not using FFmpeg at all
[21:55] <Keshl> No, we are.
[21:55] <durandal_1707> Keshl: because I'm omnipotent
[21:55] <Keshl> Nu you's not. <É<
[21:55] <Soe1en> durandal_1707: what do you mean?
[21:55] <red6m> is it possible to convert video/quicktime; to wav?
[21:56] <Keshl> red6m: Strictly speaking, no. As far as you need to care, yes. Just ffmpeg -i thingy.mov sound.wav
[21:56] <red6m> Keshl, awesome. awesome to the max!
[21:56] <Keshl> Welcomes, oÉo
[22:02] <Fjorgynn> "We used to have people in the industry, but they are basically gone" - Fredrik Reinfeldt, 2013.
[22:02] <Fjorgynn> wrong channel
[22:03] <Soe1en> how come ffmpeg is renamed avconv on ubuntu?
[22:03] <Keshl> It's not.
[22:03] <Fjorgynn> it's not
[22:03] <Keshl> avconv is a fork, not related to ffmpeg.
[22:03] <Keshl> Well. It's as related to it as forks go. But that's it.,
[22:04] <llogan> Fjorgynn: "Corn? When did I eat corn?" - Fredrik Reinfeldt, 1987
[22:04] <Fjorgynn> llogan: lol
[22:04] <llogan> back in university i guess
[22:04] <Fjorgynn> anyways
[22:05] <Fjorgynn> they trashed the Swedish industry programme now
[22:05] <Fjorgynn> canned maybe
[22:06] <Soe1en> thanks for the link!
[22:07] <durandal_1707> llogan: why corn?
[22:07] <llogan> because it does not always digest well.
[22:08] <red6m> Keshl, can I also set it to 8bit when converting?
[22:08] <jstackhouse> durandal_1707: So I ran ffmpeg directly, not by spawning it from Node, and it ran without issue
[22:08] <llogan> Soe1en: also refer to the link within the link
[22:08] <jstackhouse> durandal_1707: As well, I didn't put it on an EBS drive on Amazon. So now I move my output to the EBS drive.
[22:09] <red6m> Keshl, is this correct: ffmpeg -i in.mov -ab 8 -ar 11025 -ac 1 out.wav
[22:09] <red6m> ?
[22:09] <Keshl> red6m: Dunno. Really, all you need is -i thing.mov out.wav
[22:10] <red6m> Keshl, thanks.
[22:10] <llogan> meteor shower tonight
[22:10] <durandal_1707> red6m: bitrate is in bits 8 bits is nice
[22:10] <Keshl> Welcomes, oÉo
[22:10] <Soe1en> llogan: gotcha
[22:10] <red6m> durandal_1707, I don't follow. is this a sarcasm?
[22:10] <durandal_1707> red6m: so -ab 8 means 8bits/sec
[22:11] <red6m> durandal_1707, is that normal for .wav?
[22:11] <Keshl> No. Not at all. XD
[22:11] <Keshl> 8-bit is.. Remember how the OLLLD Gameboy sounded?
[22:11] <Keshl> Black and white one?
[22:11] <Keshl> That's 8-bit.
[22:11] <llogan> Keshl: are we that old now?
[22:12] <red6m> hmm. is bitrate or bits the same thing?
[22:12] <Keshl> ...Yes?
[22:12] <red6m> ...ok
[22:12] <Keshl> red6m: no.
[22:12] <Keshl> Was saying yes to llogan.
[22:12] <Keshl> Bitrate is how many bits are read per second. It depends on the bit depth ("bits") and sample rate.
[22:13] <mark4o> -ab is not valid for .wav (pcm)  (although ffmpeg won't complain)
[22:13] <Keshl> Bit depth is how many different positions the speaker can be told to move. Samples is how many times a second it's told to move.
[22:13] <red6m> ahhh. so I think I'd like to convert that thinkg into 8 bit dept please.
[22:13] <red6m> lol
[22:13] <Keshl> Generally, if you want high fidenlity, you need at least 44,100 samples per second and I forget the depth, but way more than 8.
[22:13] <durandal_1707> red6m: use pcm_s8 as audio encoder
[22:13] <red6m> Keshl, it's for single word pronunciation files.
[22:14] <Keshl> red6m: You DEFINITELY need more than 8, then.
[22:14] <Keshl> Far more than 8.
[22:14] Action: Keshl finds what a gameboy sounds like. Appearntly red6m's too younge.
[22:14] <red6m> Keshl, geesh. thanks internet.
[22:14] <Keshl> red6m: http://videospielmusik.de/Nintendo/game%20Boy/Pokemon%20(Red,%20Blue,%20Yellow)/PRBLAVEN.mid
[22:15] <Keshl> That is literally the most fidelity you can possibly get with 8-bit.
[22:15] <durandal_1707> for extra high fidelity that Keshl prefers use pcm_f64le
[22:15] <Keshl> For voice you pretty much need at least 32 bits, preferably more.
[22:16] <red6m> hmm. i see.
[22:16] <durandal_1707> ugh, 16 bits is enough
[22:16] <Keshl> durandal_1707: Really? o.O'
[22:16] Action: Keshl looks for SNES music.
[22:17] <Keshl> Oh. Yeah. my bad, 16 bits'll work.
[22:17] <durandal_1707> the highest audio i managed to find is 24bit flac at 19200 sample rate
[22:17] <durandal_1707> * one 0 missing
[22:17] <Keshl> durandal_1707: Really? You've never heard over 19200 sample rate? o.O'
[22:18] <mark4o> Keshl: http://people.xiph.org/~xiphmont/demo/neil-young.html
[22:18] <durandal_1707> Keshl: it's typo was supposed to be 192k
[22:18] <Keshl> Oh.
[22:19] <Keshl> mark4o: This I know.
[22:19] <red6m> thinking about it - I think I mean 8 bits per sample. Does that make it ok?
[22:19] <Keshl> Well, about the sampling rate anyway. Bit depth, I'm sure I either hear a difference.
[22:20] <Keshl> red6m: That's what bit depth is. Trust me, no, for voice it's not. You won't be able to understand anything.
[22:20] Action: Keshl just records his voice and uploads it.
[22:20] <durandal_1707> 8 bits per samples is limited, usually you use it one you do not have enough space
[22:22] <red6m> hmm. im looking at this python module: sndhdr and they list sampling_rate and bits_per_sample) as totally different thins: (type, sampling_rate, channels, frames, bits_per_sample)
[22:22] Action: Keshl is definitely doing something wrong here..
[22:22] <durandal_1707> next worse thing are digital rips of analog recordings
[22:24] <Keshl> Oh. No. Okay. Wow, that actually sounds more reasonable than I thought.
[22:24] <red6m> im pretty sure those are different things: http://www.voxforge.org/home/docs/faq/faq/what-are-sampling-rate-and-bits-per-sample
[22:24] <Keshl> red6m: They are. I explained that earlier.
[22:25] <Keshl> red6m: http://www.youtube.com/watch?v=XUZOzEBAIww 8-bit samples of human voices. Let a few play through.
[22:25] <red6m> Keshl, so - I can just use pcm_s8 ?
[22:26] <Keshl> No idea.
[22:26] <Keshl> If you're asking if it'll work right, no clue. If you're asking if the sound quality will be acceptable, to me, no way. Not a shot.
[22:26] <Keshl> To you? Listen to that video and decide for yourself.
[22:27] <red6m> Keshl, lol. thanks. i'll give it a try.
[22:27] <mark4o> red6m: http://xiph.org/video/vid1.shtml  - skip to 13:14 for sample formats
[22:33] <durandal_1707> Keshl: what sample rate you use?
[22:34] <Keshl> 44,100. Same as anyone else on a modern-day system, oÉo
[22:34] <Soe1en>  "Libav is totally ignoring FFmpeg", sounds like the way ubuntu acts to debian, heh
[22:35] <Keshl> Other way around. Ubuntu's a deritave of Debian. <.<
[22:36] <Soe1en> so is libav to ffmpeg isn't it?
[22:36] <Keshl> No.
[22:36] <Keshl> Er, yes.
[22:36] <Keshl> But you still have the order backwords on one, oÉo.
[22:37] <Keshl> Oh.
[22:37] <Keshl> Misread.
[22:37] <Keshl> Kay.
[22:37] <Keshl> >w>
[22:38] <Soe1en> heh ^^
[22:39] <durandal_1707> anybody have 16bit float (aka half-float) files?
[22:41] <Soe1en> so let me get this straight: ffmpeg != ffmpeg @ ubuntu
[22:42] <durandal_1707> ugh, whoever wrote this scenario should get to hollywood
[22:44] <durandal_1707> Soe1en: look what ffmpeg outputs when its run
[22:44] <durandal_1707> if it says something in caps lock, than its virus, you should remove it asap
[22:44] <Soe1en> durandal_1707: what are you talking about?
[22:45] <Soe1en> durandal_1707: you mean this? : *** THIS PROGRAM IS DEPRECATED ***
[22:45] <Soe1en> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[22:45] <durandal_1707> yes
[22:45] <durandal_1707> that is 'ffmpeg' from libav
[22:46] <Soe1en> how do I get the real thing?
[22:46] <Soe1en> this is so confusing, even after reading those articles
[22:46] <durandal_1707> do you  really need real thing?
[22:47] <Soe1en> perhaps
[22:51] <Soe1en> alright, I will try to remove avconv and install ffmpeg instead
[22:51] <durandal_1707> well other things are connected with it
[22:52] <durandal_1707> so perhaps you should download static
[22:52] <Soe1en> what does download it static mean?
[22:52] <durandal_1707> there are dynamic and static builds
[22:52] <Keshl> Static means all the code is built into the executable. No dependancies. Dynamic has dependancies.
[22:53] <Keshl> Dynamic, however, can re-use code already existing in RAM and your processor's cache to speed up. Static can't, but there's less complications.
[22:53] <meekohi> I am having trouble with ActiveSupport::MessageVerifier after upgrading to Rails 4. Was there something changed about how Cookies are encrypted?
[22:54] <durandal_1707> if you only will encoder 8 bit pcm files, you can live with avconv (but -decoders/-encoders will not work)
[22:56] <Soe1en> I downloaded and unpacked ffmpeg, having problems already
[22:56] <Soe1en> ./ffmpeg -version
[22:56] <Soe1en> Illegal instruction (core dumped)
[22:56] <Keshl> Soe1en: Get the 32-bit version.
[22:57] <Soe1en> it is the 32-bit version!
[22:57] <Keshl> o.O'
[22:58] <Soe1en> ./ffmpeg -version
[22:58] <Soe1en> Illegal instruction (core dumped)
[22:58] <Soe1en> ah crap, stupid paste function
[22:58] <durandal_1707> Soe1en: proably wrong kernel version
[22:58] <Soe1en> I see
[22:58] <Keshl> I don't -- Mind explaining that, oÉo?
[22:59] <durandal_1707> downloaded version for newer/older kernel and do not have cap to run never/older app
[22:59] <Keshl> I thought that only affected drivers..
[23:00] <durandal_1707> he did not said what he downloaded and what kernel version he have
[23:00] <Soe1en> I'm using ubuntu 12.04 lts
[23:00] <Soe1en> no clue what kernel I apparently use! But it must be less than 3.2
[23:01] <Keshl> Ahh, ubuntu, never teaching Linux users important things.. x.x
[23:02] <Soe1en> Keshl: which distro do you prefer
[23:02] <Keshl> Sabayon. Gentoo.
[23:02] <Keshl> Although, literally anything besides a *buntu is better.
[23:03] <Keshl> Linux may be a kernel, but there's a lot of design principles behind it. Ubuntu doesn't follow them, so some people don't even consider it Linux-based anymore.
[23:03] <Keshl> One of the main ones is "Linux doesn't stop you from doing stupid stuff, cuz that'd stop you from doing smart stuff too".
[23:04] <Soe1en> somehow it does make sense
[23:05] <Keshl> Here's a somewhat crude example, but it's what I mean.
[23:05] <Keshl> So, by default on windows, there's no utility to control your fan speeds.
[23:05] <Soe1en> well I get your point no worries heh
[23:05] <Keshl> There isn't on Linux either, but that's regardless, I'm just showing how something stupid can be used for smartness.
[23:06] <Keshl> Now, let's say you're on a laptop and you spill water into your case.
[23:06] <Keshl> You don't know where it is, but by some miracle nothing shuts down or shorts.
[23:06] <Keshl> At this point you have two choices: Either shut your computer off and try to let it dry into air, which depending on the area you live in might not even be reasonably possible..
[23:07] <Keshl> Or B, turn your fans off, force them to stay off, and then run benchmarks to get your system to warm up so it evaporates.
[23:07] <Keshl> Normally that'd be stupid. But in a situation like that, it's /really/ smart. I've actually had to do it before.
[23:07] <Keshl> (With careful tempeture monitoring, obviously)
[23:07] <Soe1en> ow
[23:07] <Keshl> Totally worked, too. Using that laptop now. <.<
[23:08] <durandal_1707> and causing that whole are lost electricity in such scenario
[23:08] <Keshl> durandal_1707: I coudln't understand that, mind rewording your post?
[23:09] <durandal_1707> are you seriusly telling that if one spills water he should just wait?
[23:09] <Soe1en> Keshl: how did you know that the water stopped to float?
[23:09] <Keshl> If you don't know where it is, and you live in a really humid environment, yes. You kinda have to.
[23:09] <Keshl> Unless you feel like running your laptop hot and trying to evaporate it.
[23:10] <Keshl> Soe1en: Stopped to float? What do you mean?
[23:11] Action: durandal_1707 lemme try it...
[23:11] <Keshl> ...
[23:11] <Keshl> Do not spill water in your laptop purposely.
[23:11] <Keshl> I got VERY lucky and it happened to land in an area with no electronics.
[23:12] <Keshl> In general, this won't happen and you'll fry something instead.
[23:12] <Keshl> I just didn't want to risk it being knocked around accidently, that's why I tried to evaporate it quickly rather than let it sit and evaporate very slowly on its own.
[23:13] <Soe1en> I see why you stopped the fan to rotate: 1. to stop spreading more water inside the laptop in case water landed in or near the fan and 2. for the overheat effect
[23:13] <Keshl> It landed nowhere near the fan, but yeah, that's another point.
[23:14] <Soe1en> but how did you know that the water inside your laptop stopped to float into idk, parts which could cause a defect?
[23:14] <Keshl> The effect would've been instant if it did. I woudln't be typing right now.
[23:14] <Keshl> Simply by it seeping inside and surviving more than two or three seconds, I knew it had to have landed in a spot with no electronics.
[23:15] <Soe1en> I see, interesting thought process
[23:15] <Keshl> Based on where I saw it seeping in, I also had a general idea of how long it'd take to heat the area and how long I'd need to maintain it.
[23:16] <Keshl> And in retrospect, I probbaly didn't need to turn the fans off. It was by the battery so I probably could've just unplugged it and let the battery heat up; the processors weren't anywhere near it.
[23:16] <Keshl> Still. It worked.
[23:18] <Soe1en> not bad dude, ingenious indeed, on the other hand, I would never have done that just because I would have considered this situation as too complex to predict what would happen in the next seconds, I would just had shutted down the system and waited some time
[23:19] <Keshl> It depends on how well you know your laptop vs your environment.
[23:20] <Soe1en> I guess that's true heh
[23:20] <Keshl> In my case, I know my laptop /very/ well. I know how long parts take to heat up depending on what I'm doing and where the heat generrally flows inside it. I also knew that the area it happened to be in had a lot of traffic so there's a chance that someone could've caused enough vibration to shake the water onto something electric.
[23:20] <Keshl> Besides that, it probably woudl've been me that did it. <.< I'm in an area that you cannot physically get out of unless you move the laptop.
[23:23] <Keshl> But yeah. My point in all this is that Linux lets you do stuff like that and ubuntu tries to stop you from doing that.
[23:29] <Soe1en> well yeah but I think ubuntu was never targeting senior users
[23:29] <Soe1en> if you know what I mean
[23:29] <Keshl> It wasn't, but it shouldn't be flying under the Linux banner if that's the case.
[23:30] <Keshl> From what I understand (and I may be entirely wrong about this), Linux is intended for very serious users, not newbies.
[23:31] <Keshl> Stuff just tends to be more efficent when it operates under the assumption that the end-user knows everything about the system. If it tries to be friendly, all the stuff that makes it friendly ends up getting in the way as soon as you try to get more serious,
[23:31] <Keshl> *.
[23:31] <Mista-D> Is there any way to for ffprobe to tell if timecode is drop-frame or non-drop-frame?
[23:32] <Soe1en> to be honest I never thought about what linux is intended for!
[23:33] <Soe1en> I just watched an interview about richard stallmann a few weeks ago and understood that I always missunderstood the meaning behind gnu all the time until that point
[23:34] <Keshl> Which interview, oÉo? I'm pretty sure I misunderstand it too <É<
[23:35] <Soe1en> Keshl: http://www.youtube.com/watch?v=uFMMXRoSxnA
[23:35] <Keshl> Danks, oÉo
[23:35] <Soe1en> no pasa nada!
[23:43] <Keshl> .... -Kicks Stallman's balls.- <.<
[23:43] Action: Keshl does not like 4:20.
[00:00] --- Tue Aug 13 2013