[Ffmpeg-devel-irc] ffmpeg.log.20120903

[burek]

[01:11] <buhman> how can I override the input framerate?
[01:19] <jaquez> i have 50MB music file; how do i make sample of it under 256KB file
[01:27] <buhman> jaquez: use -ss and -t, and twiddle with -ab
[01:31] <jaquez> what does -ss and -t do
[01:33] <jaquez> ffplay.exe won't play .mlp properly;  is this a bug?
[01:40] <ubitux> buhman: -r before the input?
[01:40] <buhman> ubitux: I tried that
[01:41] <ubitux> jaquez: with dd i guess
[01:41] <buhman> "
[01:41] <buhman> Option framerate not found."
[01:41] <ubitux> (assuming the format doesn't have the header at the end of something)
[01:41] <buhman> http://ffmpeg.org/trac/ffmpeg/ticket/403 seems to be relevant
[01:42] <ubitux> buhman: " Resolution set to invalid"
[01:43] <buhman> ubitux: because he couldn't provide a sample or something
[01:43] <buhman> "
[01:44] <buhman> The -r input option is expected to work on inputs where fps is unknown."
[01:44] <buhman> what about in the case where the input fps is invalid?
[01:44] <buhman> (would you like a sample?)
[01:44] <buhman> (or the verbatime file for that matter)
[01:46] <buhman> ubitux: http://sprunge.us/HGOc
[01:47] <buhman> jaquez: that would allow you to select a smaller portion of your file, the start position and total duration
[02:12] <fatAgnes> i have mjpeg movie
[02:12] <fatAgnes> every frame is a full jpeg
[02:13] <fatAgnes> I want to do temperal compression, with mpeg4
[02:13] <fatAgnes> just have few keyframes, not that every frame IS a keyframe (my situation now)
[02:13] <fatAgnes> how do I accomplish this?
[02:27] <buhman> fatAgnes: erm, encode it?
[02:28] <fatAgnes> i dotn see ffmpeg command line where i can make this from descrete jpeg
[02:31] <buhman> fatAgnes: you have a pile of jpeg files that you want to make a video out of?
[02:31] <fatAgnes> basicly
[02:31] <buhman> that's actually in the manpage; let me google for you
[02:32] <fatAgnes> the jpeg,.., show slight movement and rotatation but the differences between jpegs is quite small
[02:32] <fatAgnes> ooh cool
[02:32] <buhman> ffmpeg -r 24 -i %03d.jpg foo.mpg
[02:39] <ubitux> buhman: dunno, maybe re-open the ticket with your sample
[02:41] <buhman> ubitux: sure
[02:52] <buhman> done
[03:54] <GRMrGecko> How do I use timecode files with FFmpeg?
[04:37] <barhom> ffmpeg -re -loop_input -i error503.jpeg -vcodec mpeg2video -f mpegts udp://239.192.200.200:3301 < this works beautifully, but it doesn't slow down to something like 25frames/s
[04:38] <barhom> because the CPU is 99% when I do that,
[06:42] <buhman> so I'm trying to multiplex audio/video together
[06:42] <buhman> I do ffmpeg -f h264 -i crf28-veryslow-tc63-29f.h264 -r 29.97 -vcodec copy -i audio-96k.aac -acodec copy output.mp4
[06:42] <buhman> apparently this is invalid.
[06:43] <buhman> http://sprunge.us/iHJO
[06:43] <buhman> "Unknown decoder 'copy"
[06:48] <buhman> I fiddle around with the order of arguments, then ffmpeg makes me thing something very wrong is going on: http://sprunge.us/YJMP
[06:55] <catsup> so...  does ffmpeg source build libavcodec.so.* ?  if so, how?  i got ffmpeg binaries but what i actually need is libavcodec
[07:27] <catsup> ok...  make install-libs builds the .so files
[07:27] <catsup> er, no, it installs the .a :/
[07:27] <catsup> not sure if that will work
[07:37] <catsup> make install-libavcodec-shared !
[07:38] <catsup> of course
[08:47] <fling> I can't ff/rewind aac files with mpd
[08:47] <fling> do I need to convert or just to mux?
[08:48] <fling> do I need another container like ogg?
[08:54] <fling> looks like I can mux into mp4, do I need it?
[08:58] <fling> [mp4 @ 0x1338780] malformated aac bitstream, use -absf aac_adtstoasc
[08:58] <fling> av_interleaved_write_frame(): Operation not permitted
[09:00] <fling> ffmpeg -i input.aac -absf aac_adtstoasc -acodec copy output.mp4
[09:00] <fling> ^ this works, now I can ff/rewind with mpd
[09:01] <fling> but output file is smaller somewhy
[09:01] <fling> thanks
[09:01] <fling> fling: yw
[09:22] Action: buhman encountered this -absf aac_adtstoasc nonsense earlier when he was muxing aac in mp4; not sure what that means
[09:22] <buhman> doesn't seem to be well documented for that matter.
[09:23] <fling> buhman: right, do I need mp4? :p
[09:29] <buhman> fling: do you?
[09:29] <fling> buhman: idk which container is better to use?
[09:30] <fling> looks like mpd's ff is not very stable with mp4
[09:31] <buhman> O.o why are you doing mp4 in mpd anyway?
[09:31] <buhman> mpd would be ..erm.. music by definition.
[09:31] <fling> buhman: I have some music in aac files
[09:32] <fling> mpd can't ff/rew aac somewhy
[09:32] <fling> so I have muxed it into mp4, now it works but not so stable
[09:33] <buhman> so why not jusy keep using that? no need for a more complex container format.
[09:34] <buhman> fling: oh; why aac?
[09:34] <buhman> I mean, if you're so concerned about audio quality that mp3 isn't sufficient, you should be using flac.
[09:35] <fling> buhman: source is aac, idk why, I hate it, I always use flac/pcm16le
[09:36] <buhman> can you find flac sources somewhere?
[09:36] <fling> maybe, idk
[09:36] <buhman> or, worst case, transcode to flac :S
[09:36] <buhman> it'll be lossess from the aac at least
[10:16] <AlMalaak> hi please help
[10:16] <AlMalaak> i need to convert a .webm to another format for facebook but maintain the quality
[10:16] <AlMalaak> if i do ffmpeg -i foo.webm foo.avi it degrades quality big time
[10:16] <AlMalaak> how do i maintain quality but change format?
[10:17] <Tjoppen> -vcodec copy -acodec cpoy
[10:17] <Tjoppen> *copy
[10:17] <fling> buhman: but it will take more space
[10:17] <AlMalaak> thanks sir!
[10:17] <AlMalaak> that's fine, for upload
[10:18] <AlMalaak> ffmpeg -i foo.webm -vcodec copy -acodec copy foo.avi
[10:18] <AlMalaak> that wont work
[10:18] <AlMalaak> oh, i had to move something
[10:18] <AlMalaak> same thing:(
[10:18] <AlMalaak> it changed fps
[10:19] <Tjoppen> try using a decent container like mov
[10:19] <AlMalaak> ok
[10:19] <AlMalaak> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[10:19] <AlMalaak> if i specify .mov instead
[10:20] <Tjoppen> of course, the most important question is: why do you want to change container at all?
[10:21] <AlMalaak> Tjoppen, if i upload .webm to facebook it wont accept that format:(
[10:21] <AlMalaak> for some stupid reason
[10:21] <AlMalaak> they require you use a diff format
[10:21] <AlMalaak> but cheese uses .webm
[10:22] <Tjoppen> try encoding with x264 instead then, with a high quality setting
[10:22] <Tjoppen> like -crf 20
[10:22] <AlMalaak> i cant do copy
[10:22] <AlMalaak> what is the extension of x264?
[10:22] <AlMalaak> .mov?
[10:23] <Tjoppen> or mp4. -vcodec libx264 -crf 20  should hopefully work. and -acodec copy of course
[10:23] <AlMalaak> http://www.askmeaboutlinux.com/?p=1693
[10:23] <AlMalaak> that does not work
[10:23] <AlMalaak> ok
[10:24] <Tjoppen> the avi thing should really be a bug ticket. it uses really crappy defaults
[10:24] <AlMalaak> -acodec copy?
[10:24] <AlMalaak> it messes up the fps, the avi thing
[10:25] <Tjoppen> nm that
[10:25] <Tjoppen> just thinking out load
[10:25] <Tjoppen> *loud
[10:25] <AlMalaak>  ffmpeg -i love.webm love.mp4  -vcodec libx264 -crf 20 -acodec copy
[10:25] <AlMalaak> error
[10:25] <AlMalaak> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[10:25] <Tjoppen> uh, maybe that's the audio stream? in that case the video is OK
[10:26] <AlMalaak> ?
[10:26] <Tjoppen> try -vcodec copy -acodec aac -ab 256k
[10:26] <AlMalaak> ok
[10:26] <Tjoppen> and -strict -2  before -acodec
[10:26] <AlMalaak> Unrecognized option 'strict-2'
[10:27] <Tjoppen> space..
[10:27] <AlMalaak> Failed to set value '-acodec' for option 'strict-2'
[10:27] <AlMalaak> oh
[10:27] <AlMalaak> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[10:27] <AlMalaak> ffmpeg -i love.webm love.mp4  -vcodec copy -strict -2 -acodec aac -ab 256k
[10:27] <Tjoppen> no other errors in the log?
[10:27] <Tjoppen> derp
[10:27] <Tjoppen> love.mp4 on the end
[10:27] <Tjoppen> and -acodec copy again
[10:27] <Tjoppen> and mov
[10:28] <Tjoppen> :)
[10:28] <AlMalaak> im lost:O
[10:28] <Tjoppen> ffmpeg -i love.webm -acode copy -vcodec copy love.mov
[10:28] <AlMalaak> [mov @ 0x1034a80] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 44 >= 44
[10:28] <AlMalaak> av_interleaved_write_frame(): Invalid argument
[10:29] <Tjoppen> is stream 1 audio or video?
[10:31] <AlMalaak> ?
[10:31] <AlMalaak>    Stream #0:1(eng): Audio: vorbis, 44100 Hz, mono (default)
[10:32] <Tjoppen> -acodec aac -ab 256k
[10:32] <AlMalaak> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[10:32] <Tjoppen> gah
[10:33] <Tjoppen> try using an editor like kdenlive or something
[10:34] <Tjoppen> else I'm a bit out of ideas
[10:35] <AlMalaak> looking for f17 rpm for kdeinlive
[10:35] <AlMalaak> found
[10:36] <Tjoppen> then you can also trim the video and stuff :)
[10:37] <AlMalaak> ah ok
[10:37] <AlMalaak> thanks sir
[10:40] <AlMalaak> i installed a windows converter;)
[10:41] <Tjoppen> I wonder where the encoding guide went.. there used to be one in /topic
[10:43] <AlMalaak> wow, how is it 45m?
[10:43] <AlMalaak> not bad
[10:44] <AlMalaak> oo, this may be horrible
[10:44] <AlMalaak> it loses 9fps
[10:45] <Tjoppen> do actually look at the file. sometimes such statistics may be lying
[10:46] <AlMalaak> oic
[10:46] <AlMalaak> ok, im good for now, thanks sir!
[10:47] <Tjoppen> np
[13:17] <zizzu> hello, i am using this command, ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0 \
[13:17] <zizzu>  -f alsa -ac 2 -i pulse -vcodec libx264 -preset fast -crf 27 -acodec libmp3lame -ab 128k -ar 44100 -ac 2 -s hd480 -aspect 16:9 -f flv "$RTMPURL/$STREAM_KEY flashver=FMLE/3.0\20(compatible;\20FMSc/1.0)" but i get no sound on my stream, how can i solve this, help pls?
[13:19] <zizzu> i have libmp3lame installed, but no way to get sound :(
[13:20] <zizzu> \o/ hello
[13:24] <zizzu> also this give a file with no sound ffmpeg -f alsa -ac 2 -i pulse -acodec libmp3lame -ab 128k -threads 0 -f mp3 ./test.mp3
[13:30] <zizzu> can someone help? :)
[13:32] <thresh> michaelni: here?
[14:54] <burek> what is the proper way to change the brightness of the video, using ffmpeg?
[14:56] <burek> zizzu can you please use a pastebin site (like www.pastebin.com) to show your ffmpeg command and the complete console output?
[15:18] <michaelni> thresh, would it be ok with you/vlc if the seperate libpostproc repositories are killed and Daemon404 and me make sure that libpostproc from ffmpeg can be build standalone against system libavutil headers?
[15:19] <thresh> michaelni: of course, that's why I'm here
[15:19] <thresh> michaelni: I can make you a clean libpostproc.git repo on git.v.o with full access
[15:20] <thresh> clean == no commits, no files
[15:20] <michaelni> iam not sure we need that
[15:20] <michaelni> i just now have spoken with Daemon404 ... and i dont know exactly what vlc needs/wants
[15:22] <michaelni> thresh, so the question is does vlc need a seperate repo for this at all ?
[15:22] <michaelni> i mean if we get standalone build against system installed headers working for libpostproc ?
[15:23] <michaelni> standalone build is working already but not against system headers
[15:23] <thresh> michaelni: we need a standalone repo indeed
[15:24] <michaelni> ok, can you explain why ?
[15:25] <thresh> sure, we as an upstream must support building VLC against both ffmpeg and libav; the reason for the separate libpostproc.git (which is now a mirror of Daemon404 work) is to enable VLC to be built using libav with libpostproc from ffmpeg
[15:26] <thresh> the easiest way to do that (without checking out full ffmpeg tree) is to have a separate repo for libpostproc
[15:27] <michaelni> where is the problem with checking out the full tree and --disable-... --enable-postproc ?
[15:28] <michaelni> i mean if we get that to build against system (libav possibly) headers ?
[15:28] <thresh> ffmpeg.git isnt exactly small to checkout
[15:29] <michaelni> true but if the libpostproc would be a full clone of ffmpeg for easy merging it would be the same size
[15:30] <thresh> right, I didnt think of it
[15:30] <thresh> another option is having libpostproc/ as a submodule
[15:31] <thresh> why don't you like having it separate, btw? fate?
[15:32] <michaelni> fate and more work to maintain
[15:36] <thresh> I guess it will work for us, though ugly
[15:36] <thresh> if we had --disable-all --enable-postproc or something like that.
[15:36] <thresh> michaelni: let's move to #ffmpeg-devel, so Derek can read us
[15:37] <michaelni> thresh, ok
[16:00] <GRMrGecko> how can I work with variable frame rate?
[16:07] <burek> easily
[16:14] <natrixnatrix89> Hi guys. I'm really struggling to encode a png sequence into flv with alpha channel..
[16:14] <natrixnatrix89> what I have is ffmpeg -pix_fmt rgb32 -vcodec png -i <my-sequence> -an -b:v 1000k output.flv
[16:14] <natrixnatrix89> the result has very good quality, but it lacks transparency
[16:15] <natrixnatrix89> is there some option I'm missing?
[16:17] <JEEB> rgba is the pix_fmt with alpha, and then you have not specified which video codec you want to use and thus you most probably and end up with either VP6 or H.264 depending on the sanity of your ffmpeg compile
[16:17] <JEEB> also neither of those support alpha
[16:18] <natrixnatrix89> I tried libx264
[16:18] <natrixnatrix89> it didn't work
[16:19] <natrixnatrix89> and also I have to import it into adobe flash.. and it didn't accept libx264
[16:19] <JEEB> well libx264 = H.264
[16:19] <natrixnatrix89> I know
[16:19] <JEEB> and uh... flash should work JustFine with H.264 unless you used 4:4:4 or something
[16:19] <natrixnatrix89> What I want to know.. is there some way to define background color to be transparent?
[16:19] <JEEB> in the output?
[16:19] <JEEB> no
[16:19] <natrixnatrix89> yes
[16:20] <JEEB> because H.264 or VP6 and friends that go into flv don't support alpha
[16:20] <natrixnatrix89> so the only thing that I could try playing with is -pix_fmt
[16:21] <natrixnatrix89> and maybe vcodec qtrle
[16:21] <JEEB> which is the input pix_fmt
[16:21] <JEEB> does qtrle go into flv?
[16:21] <JEEB> settings before -i set the decoder/input settings
[16:21] <JEEB> settings after -i set the encoder/output settings
[16:21] <natrixnatrix89> rgba
[16:21] <natrixnatrix89> so I have to set px fmt for both?
[16:22] <natrixnatrix89> or for input will suffice?
[16:22] <JEEB> you shouldn't really have to set for either in case of png as it should get taken in as RGB(A) by default
[16:23] <lakcaj> Hello.  I have an IP security camera and I've used openRTSP to output the h264 UDP stream to an mpeg4 file, but I just can't get anything to play it without corruption.  The camera is actually using ffmpeg to do the encoding (I think).  I've uploaded a sample here:  http://www.sendspace.com/file/u0fe9l
[16:23] <lakcaj> Any help would be greatly appreciated - I'm at my wits end.
[16:23] <JEEB> so the only thing you would have to I think is output, but with that in most cases it sets it correctly
[16:24] <GRMrGecko> How about this. How can I convert to Variable Frame Rate?
[16:24] <lakcaj> If I view the streaming video through their windows only interface, it plays just fine
[16:24] <JEEB> GRMrGecko, if you have VFR input timestamps should be used as well as possible with the output and output should be VFR as well unless you use something a la AVI that has to be CFR
[16:27] <GRMrGecko> JEEB: I have used Handbrake to make a variable frame rate MKV, however handbrake doesn't seem to do it after I converted to Prores or YUV. Is it possible to take the timecodes from the good handbrake encode and use it with FFmpeg to take the constant and make it variable?
[16:27] <JEEB> wat
[16:27] <natrixnatrix89> JEEB: Hmm. looks like the problem is with input.. I just tried using only ffplay with the parameters.. and even there the background is very messy..
[16:27] <GRMrGecko> well& I would think not because the timecodes are for each individual frame...
[16:28] <GRMrGecko> And FFmpeg would not know which frame
[16:28] <GRMrGecko> than, the question would be is it possible to output variable from constant in ffmpeg?
[16:29] <GRMrGecko> or could you recommend a software package which could output variable H264
[16:29] <GRMrGecko> mencoder?
[16:30] <JEEB> uhhh, what the flying fuck are did you do to your VFR content and why is it CFR now?
[16:30] <JEEB> I have no fucking idea what you did and your sentences aren't exactly all making sense
[16:31] <GRMrGecko> The original video is a constant frame rate VOB file. I need to edit the output, however no editor works with variable frame rate.
[16:32] <GRMrGecko> I would like my end result to be variable frame rate.
[16:35] <JEEB> so you basically have 30i with 24p and 30i/p in there?
[16:35] <GRMrGecko> the original video is 29.97FPS
[16:35] <JEEB> sigh....
[16:35] <JEEB> that's 30i
[16:36] <GRMrGecko> ok
[16:36] <JEEB> 30/1.001fps, interlaced
[16:36] <GRMrGecko> yup
[16:36] <JEEB> anyways, I'd say your best bet would be to correctly IVTC/deint the actual separate parts of it that have different rates
[16:36] <JEEB> and just edit them separately
[16:37] <JEEB> and/or take the bullet and just add extra frames or whatever, because the editor won't be able to deal with your shit anyways and I'm pretty sure you're not up to the math to do the VFR on your side and just tell the editor they're all of the same frame rate
[16:38] <JEEB> the only way after editing to get it to "VFR" would be by using dedup or something but you really, really don't want to do that
[16:38] <GRMrGecko> JEEB: Mencoder has frame_drop_ratio
[16:40] <JEEB> which is like dedup
[16:40] <JEEB> not gonna be any fucking better
[16:40] <GRMrGecko> ok...
[16:42] <natrixnatrix89> Hmm so weird.. now I tried converting a video file, and when I specified pix fmt for input, ffmpeg said "Option pixel_format not found."
[16:43] <JEEB> it's pix_fmt
[16:43] <natrixnatrix89> I know.. that's what I wrote..
[16:43] <JEEB> "Option pixel_format not found."
[16:43] <JEEB> > pixel_format
[16:43] <natrixnatrix89> yes.. it's so weird.. I wrote pix_fmt
[16:44] <natrixnatrix89> and it replies about pixel_format
[16:44] <natrixnatrix89> and that pix_fmt is deprecated.. and use pixel format instead
[16:44] <JEEB> no idea, anyways -- finally got home and will get food >_>
[16:44] <natrixnatrix89> and then pixel_format not found
[16:44] <natrixnatrix89> right thing to do :)
[16:46] <lakcaj> avprobe reports yuvj420p - could the yuvj be the issue?  Again, thanks for any help.
[17:02] <lakcaj> Hello.  I have an IP security camera and I've used openRTSP to output the h264 UDP stream to an mpeg4 file, but I just can't get anything to play it without corruption.  The camera is actually using ffmpeg to do the encoding (I think).  I've uploaded a sample here:  http://www.sendspace.com/file/u0fe9l
[17:24] <lakcaj> Can someone at least confirm that you can see what I'm typing :)
[17:32] <microchip_> lakcaj: yes, we can see
[17:33] <lakcaj> microchip_, thanks :)
[17:33] <microchip_> but i have no idea about your problem
[17:36] <cbsrobot-> lakcaj: see the output of:
[17:36] <cbsrobot-> ffmpeg -i Downloads/vid_error.mp4 -vf showinfo -f null /dev/null
[17:36] <cbsrobot-> looks like your video has only P frames
[17:36] <cbsrobot-> http://en.wikipedia.org/wiki/Video_compression_picture_types
[17:36] <cbsrobot-> Pframes can use data from previous frames to decompress and are more compressible than Iframes.
[17:37] <cbsrobot-> so either you have to record a longer sample
[17:37] <cbsrobot-> or there is a bug somewhere
[19:23] <zizzu> can someone help? why this command give a file with no sound ffmpeg -f alsa -ac 2 -i pulse -acodec libmp3lame -ab 128k -threads 0 -f mp3 ./test.mp3?
[19:24] <zizzu> i want record sound of my deaktop, can someone help?
[19:24] <zizzu> desktop
[19:28] <zizzu> ok
[19:30] <zizzu> this is the paste http://pastebin.com/sYquVXa6, i would know why i get a file with no sound please
[19:33] <zizzu> there is noise but no sound of my desktop
[19:38] <cbsrobot-> zizzu: I guess you are sure the pulse audio works
[19:38] <cbsrobot-> else try:
[19:39] <cbsrobot-> ffmpeg -f alsa -ac 2 -i hw:0,0 -acodec libmp3lame -ab 128k ./test.mp3
[19:39] <cbsrobot-> or
[19:39] <cbsrobot-> ffmpeg -f alsa -ac 2 -i /dev/dsp -acodec libmp3lame -ab 128k ./test.mp3
[19:40] <zizzu> same thing cbsrobot, i hear a noisy sound with the music very very in the background, like volume is to zero
[19:41] <cbsrobot-> seeing & Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
[19:41] <cbsrobot-> maybe you need to add -ar 44100
[19:41] <cbsrobot-> ffmpeg -f alsa -ac 2 -ar 44100 -i hw:0,0 -acodec libmp3lame -ab 128k ./test.mp3
[19:43] <zizzu> cbsrobot, its the same
[19:44] <cbsrobot-> zizzu: have you tried to record with audacity or any other tool ?
[19:44] <zizzu> no i havent
[19:44] <zizzu> what should i do? :)
[19:45] <zizzu> i just found the command on the web and was trying to stream to ustream it works ok but no sound
[19:46] <cbsrobot-> try alsamixer on the command line
[19:48] <cbsrobot-> I'd make sure my sound card is correctly installed end everything works as expected
[19:48] <zizzu> i see only a single "Master" line on alsamixer
[19:49] <zizzu> sounds all works ok, i can listen to mp3 etc with vlc etc
[19:49] <cbsrobot-> and on input ?
[19:50] <zizzu> what you mean?
[19:50] <cbsrobot-> http://acervulus.info/2011/alsa-mixer-and-input-source-on-ubuntu-11-04/
[19:54] <zizzu> csbtobot i cant increase the volume on the capture line to more then 18 :\
[19:55] <zizzu> 19<>18
[19:58] <zizzu> i think thats the problem, on the mp3 file i can hear the sound very in low... how can i move the input to higher value??
[19:58] <lakcaj> cbsrobot-, sorry for the late reply - was outside working.  I ran your showinfo command and all i see are type:P
[19:59] <cbsrobot-> lakcaj: working outside is surely healthier than working with video :P
[20:07] <lakcaj> I had a stream recording for an hour and I don't see a iskey:1 or a type:I     <- I assume that's what I should be looking for?
[20:07] <cbsrobot-> lakcaj: yes I think so
[20:07] <cbsrobot-> can you jump in front of your cam ?
[20:07] <cbsrobot-> lakcaj: what cam is it ?
[20:07] <lakcaj> http://www.aliexpress.com/store/product/EC-IP5913-5-Megapixel-progressive-CMOS-sensor-1080P-HD-IP-camera-outdoor-wireless-wifi-ip-camera/400700_604665650.html
[20:07] <lakcaj> It's actually running redhat on ARM
[20:07] <lakcaj> and using ffmpeg I believe
[20:07] <lakcaj> it has an iframe interval which I reduced to 10 (which is also the framerate) and still no joy
[20:07] <cbsrobot-> ah - outdoors installing your webcam &. :-)
[20:07] <lakcaj> cbsrobot-, yard work... which the wife just reminded me I have to get back to.  I appreciate your help!
[20:24] <cbsrobot-> lakcaj: btw you could also use ffmpeg directly to access the rtsp stream &.
[20:31] <rud> hi, wondering, has anyone noticed issues while reading h264 streams on latest Master ? i get tons of "error while decoding MB 46 2, bytestream (1254)" and totally scrambled output .. i've simplified my line to ffmpeg -i rtmp://some/app/stream -f file.flv  (source is h264/aac) ...
[20:37] <Mavrik> hmm, anyone have any good media info program for windows?
[20:46] <rud> hmm seems it could be due to me using gcc4.2, checking with 4.4 ..
[20:47] <zizzu> fuck fuck fuck cant register my fukin desktop sound fuck -.-
[20:48] <zizzu> i moving up and down all channels in the alsamixer and all i get is a noise -.-
[20:50] <zizzu> btw, is it possible to record only 1 virtual screen with ffmpeg?? not all gnome screens but only 1??
[20:51] <cbsrobot-> zizzu: sure
[20:52] <cbsrobot-> you can specify the screen size
[20:52] <cbsrobot-> zizzu: try audacity to record your audio
[20:52] <cbsrobot-> just to make sure it works
[20:53] <zizzu> i want stream on ustream.tv
[20:53] <zizzu> i can get the video going but not the sound!
[20:53] <rud> zizzu: do as cbsrobot-  suggests, ensure you audio works first
[20:54] <zizzu> i hear a noise and the sound of music very very in the background, so i suppose it is some volume which is low but cant find it <.<
[20:54] <zizzu> in the mp3 file i mean
[20:56] <zizzu> i have 3 options as imput source in alsamix, front mic, line and rear mic, which one should i use to rec all the sounds of desktop?
[20:56] <cbsrobot-> zizzu: maybe this works: http://ubuntuforums.org/showpost.php?p=9668787&postcount=8
[20:59] <cbsrobot-> zizzu: I guess you need to record the "audio out" port
[20:59] <cbsrobot-> it's not in the capture list of alsamixer
[21:11] <ylluminate> seeing an issue with uploading a vid created from some snapshots where there's a grey overlay that appears to be some kind of corruption. it seems to happen regularly in flash player on os x safari, and flash on fullscreen on windows with chrome (but not in smaller sized players on windows)
[21:11] <ylluminate> here's a quick screencast of it to illustrate: http://screencast.com/t/QOv6THaD
[21:12] <ylluminate> i'm not 100% convinced it comes from ffmpeg as the left and right vertical boxes on the side are pure black
[21:12] <zizzu> cbsrobot, i can show you my desktop on ustream, can u please guide me step to step to get the fuckin audio working on it
[21:12] <zizzu> :\
[21:13] <ylluminate> oddly though facebook uploads look just fine and so do regular playing of the resultant original vid file uploaded to youtube
[22:21] <trysten> i bet this channel gets a lot of silly questions
[22:22] <trysten> i was just wondering what it means when i'm trying to mux and AVI and SRT into an MKV and it says Can't write packet with unknown timestamp
[22:26] <rud> if you ever get crappy decoding of h264 in Master, make sure you are not using gcc 4.2, it's the culprit to bad h264 decoding.
[22:32] <JEEB> rud, reminds me of the 4.2 mingw compilers being borked and f.ex. certain VLC releases' libx264 being bonkers as the result
[22:32] <JEEB> it didn't even crash, just what came out of the encoder was.... truly subpar and, if I may say, psychedelic
[22:32] <trysten> i wanna see
[22:36] <JEEB> trysten, try getting one of the windows 0.9.x releases of VLC and compare with 0.8.6
[22:37] <JEEB> I don't think any streams got saved from it on my side
[22:38] <rud> JEEB: hehe nice :)
[22:39] <rud> JEEB: fun is i'm using FreeBSD-9 which uses 4.2 as default compiler, couldn't figure why *some* Master pulls i had were working fine, some other not ..
[22:39] <rud> here i the associated trac: http://ffmpeg.org/trac/ffmpeg/ticket/1464
[22:39] <rud> also note, i've just tested the new HLS patch
[22:40] <rud> and i'm glad to say it WORKS !
[22:40] <rud> https://ffmpeg.org/trac/ffmpeg/ticket/1642
[22:40] <rud> -f segment is now fully HLS compliant :)
[22:40] <JEEB> I really recommend you update to a newer GCC or I guess current clang (3.1+) if you've got an old compiler like that
[22:41] <rud> indeed, i'm using jails to host my ffmpeg encoding part, so i've already re-compiled everything using 4.6
[22:41] <saste> rud: greta
[22:41] <JEEB> any reason why you didn't go 4.7?
[22:41] <saste> *great
[22:41] <rud> saste: hi !!
[22:41] <rud> :)
[22:41] <JEEB> in the range of things I've tested it with it hasn't given any trouble
[22:41] <saste> rud: so should I close the ticket after I apply the patches?
[22:41] <saste> rud: hi :)
[22:41] <rud> saste: yes :)
[22:42] <saste> good, will do soon
[22:42] <rud> JEEB: well, no reason, only that i usually keep 1 rev behind, i guess i could go with 4.7
[22:42] <JEEB> as long as it doesn't have a zero at the end of its version string :D
[22:42] <rud> hehe, true
[22:43] <rud> have to admit HLS works pretty well on iOS.
[22:43] <saste> rud: -segment_list_type m3u8 is not required anymore, since the list type is guessed by the suffix (.m3u8 in this case)
[22:44] <rud> oh,  nice
[22:58] <buhman> fling: so?
[23:19] <rud> saste: i'm unsure if  you should add  EXT-X-ENDLIST when using -segment_list_live, and when the encoder stops .. hmm
[23:20] <rud> imagine you're transcoding an RTMP stream, assume the broadcaster looses connectivity, the RTMP stream will stop, so will ffmpeg -f segment, and it will add the  EXT-X-ENDLIST .. resulting in HLS clients to think the stream is reaching an end, and not refresh the playlist anymore
[23:20] <rud> so HLS clients need to "refresh" in order to get new playlist (when the broadcaster comes back online)
[23:23] <rud> indeed, this is an issue, since the playlist was cached by the client, if you just refresh, it will fetch the (locally) cached playlist, thus the one with EXT-X-ENDLIST
[23:32] <rud> saste: see updated trac comment when you can ;)
[00:00] --- Tue Sep  4 2012