[Ffmpeg-devel-irc] ffmpeg.log.20130313

[burek]

[01:38] <desperatede> is there a way to "persuade" ffmpeg to accept 48000 as an output audio sample rate for an FLV file when we know for a fact that the input is FLV ?
[01:39] <klaxa> ffmpeg -f flv -i <file> -ar 48000 <other options> <output>
[01:39] <desperatede> I want to output in FLV without transcoding
[01:41] <desperatede> ffmpeg -f flv -i http://89.149.156.210:80/flv/5117be75aa5e9/testat123 -f flv -c copy -
[01:41] <desperatede> the above command fails with [flv @ 02537020] FLV does not support sample rate 48000, choose from (44100, 22050, 11025)
[01:41] <desperatede> since "-f mpegts" produces freezes
[01:42] <desperatede> I want to just find a container that enables this "copy the input to the output" procedure
[01:42] <desperatede> mp4 does not work (non-seekable output)
[01:43] <desperatede> thing is, while the input is flv (with 48K audio), ffmpeg does not let me have the output at 48k
[01:44] <klaxa> hmm... well either choose a different container, or resample the audio only
[01:45] <desperatede> 1. can there be real-time resampling where the input is sent to the output in real time?
[01:45] <desperatede> 2. what other container can I use?
[01:45] <klaxa> if your CPU is fast enough
[01:45] <desperatede> (thanks for the help)
[01:45] <desperatede> CPU is not fast enough
[01:45] <desperatede> :(
[01:45] <klaxa> matroska is rather flexible
[01:46] <desperatede> [NULL @ 02537020] Requested output format 'mkv' is not a suitable output format
[01:46] <desperatede> no, sorry
[01:46] <klaxa> hmm...
[01:46] <desperatede> using "-f matroska" produces error av_interleaved_write_frame(): Broken pipe
[01:46] <klaxa> ah
[01:46] <klaxa> eh
[01:46] <desperatede> ffmpeg -f flv -i http://89.149.156.210:80/flv/5117be75aa5e9/testat123 -c copy -f matroska - | mplayer -
[01:47] <klaxa> yeah um... instead of using "-" as the output, you have to use: ":pipe"
[01:47] <klaxa> so try: ffmpeg -f flv -i http://89.149.156.210:80/flv/5117be75aa5e9/testat123 -c copy -f matroska :pipe
[01:47] <desperatede> :pipe: Protocol not found
[01:47] <desperatede> (win32)
[01:47] <klaxa> ugh
[01:48] <klaxa> can't mplayer play it directly?
[01:48] <klaxa> or for what reason do you pipe it through ffmpeg?
[01:48] <desperatede> only to change the magic (first three bytes of input)
[01:49] <klaxa> hmm sorry no idea how to do that on win32 :/
[01:49] <desperatede> mplayer produces cache errors and ultimately fails to even show a window
[01:49] <klaxa> what ffmpeg version do you have?
[01:49] <desperatede> N-50354-g2ecf564
[01:49] <desperatede>  built on Feb 27 2013
[01:50] <klaxa> sounds rather recent, yeah...
[01:51] <octe> to stream to an rtmp server with a delay i do ffmpeg capture into an output file, wait a minute, then start another ffmpeg process with the output file as input and the -re parameter and output that to the rtmp-server
[01:51] <octe> this works pretty good
[01:51] <octe> except it stops after a while
[01:51] <octe> any ideas why?
[01:52] <klaxa> tbh, that sounds awful :X
[01:52] <octe> i've not managed to do it any other way
[01:53] <klaxa> yeah i'm thinking about how to do it...
[01:53] <klaxa> hmm... no i can only come up with other awful ways
[01:54] <klaxa> anyways, maybe you run out of space?
[01:54] <octe> nope
[01:54] <octe> and it seems to be the rtmp-process stopping only
[01:54] <octe> so i wonder if ffmpeg gets confused somehow and thinks it's reached the end of the file?
[01:54] <klaxa> i'd think so too yeah
[01:55] <klaxa> well less ffmpeg, more the read() calls on the file
[01:56] <octe> with the -re it's supposed to read the file at it's framerate
[01:56] <klaxa> yeah
[01:56] <klaxa> is the other process fast enough for encoding?
[01:56] <klaxa> maybe the rtmp one catches up
[01:56] <octe> that's interesting, i suppose it could fall below the specified fps
[02:19] <octe>  klaxa is there someway to make ffmpeg display a log or something when it's not able to keep up with the requested framerate?
[02:19] <octe> or some other way i can verify that this is the problem
[02:19] <klaxa> hmm...
[02:19] <klaxa> you could keep track of the timer ffmpeg displays
[02:19] <klaxa> and compare the values of both instances
[02:20] <klaxa> but that's kinda... hard to do
[02:21] <octe> yeah
[02:21] <octe> that's the only thing i could think of
[02:22] <klaxa> what you *could* do, but that would require watching both instances again is use a utility to measure pipe throughput and put it between ffmpeg and the output-file and the output-file and the ffmpeg instance that streams to rtmp
[02:23] <klaxa> you could also run the ffmpeg streaming instance like this: ffmpeg <temp_file> rtmp://whatever; killall ffmpeg
[02:23] <klaxa> and then compare the time you streamed with the length of the temp-file
[02:23] <octe> true
[02:23] <octe> that's a good test :)
[02:33] <JDuke128> hi , i need vp6a encoder
[02:33] <JDuke128> can i encode with ffmpeg ?
[02:33] <JDuke128> vp6a
[02:34] <klaxa> nope
[02:34] <klaxa> only decode i think
[02:34] <klaxa> let me check the options of ./configure
[02:35] <klaxa> hmm i don't think ffmpeg can encode vp6a
[02:35] <JDuke128> ;(
[02:35] <JDuke128> its very bad news
[02:35] <JDuke128> i m making new javafx app
[02:36] <JDuke128> i need transparent video
[02:40] <llogan> ubitux: how do you upload images for GSoC wiki page?
[02:41] <ubitux> http://wiki.multimedia.cx/index.php?title=Special:Upload
[02:41] <ubitux> llogan ^
[02:51] <octe> klaxa, that seems to be the problem
[02:51] <octe> the streaming one catching up to the end of the file
[03:06] <creep> hi
[05:22] <woddf2> When I ran a command such as ffmpeg -i source.mpg -ss 00:42:07 -t 00:50:03 destination.mpg earlier, it worked, but now nearly-identical commands do not work. What is this?
[05:22] <woddf2> frame=    0 fps=0.0 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A
[05:23] <woddf2> "VBV buffer size not set, muxing may fail"
[05:23] <klaxa> does it just take long maybe?
[05:24] <woddf2> klaxa: The one that actually worked did not.
[05:24] <klaxa> because using -ss after -i causes ffmpeg to decode every frame
[05:24] <woddf2> http://superuser.com/questions/446107/extracting-a-clip-from-an-mp4-file-in-linux-on-kubuntu-11-10
[05:28] <klaxa> woddf2: have a look at this: http://ffmpeg.org/trac/ffmpeg/wiki/Seeking%20with%20FFmpeg
[05:28] <klaxa> skipping 42 minutes in a file will take pretty long
[05:28] <klaxa> also, are you sure you want to re-encode the video?
[05:29] <woddf2> klaxa: 42 was an example.
[05:29] <klaxa> well then paste the complete log including ffmpeg command on a pastebin-like site
[05:30] <woddf2> klaxa: http://sprunge.us/GcMN
[05:31] <klaxa> you are skipping over 2 hours
[05:31] <klaxa> like i said, ffmpeg will decode those 2 hours worth of video
[05:31] <woddf2> klaxa: Oh
[05:32] <klaxa> try running: ffmpeg -ss 02:35:00 -i is2008-tvgc20000609.mpg -ss 00:01:05 -t 265 -qscale 1 tvgc-20000609.mpg
[05:36] <woddf2> klaxa: That does the same thing.
[05:36] <klaxa> well...
[05:36] <klaxa> if you leave i can't help you
[06:05] <grepper> klaxa: do you know much about the fast seeking alternative shown there? I'd love to drop transcode and just use ffmpeg in my project, but there doesn't seem to be an 'officially' supported fast seeking method that will be frame accurate like transcode. If I knew it was robust across many formats I would use it ...
[06:14] <klaxa> i don't know the code in detail, but i think it's rather accurate? well if you use -ss after the -i again, it should be accurate, just run some tests to see
[06:16] <grepper> I remember the last time I tried it wasn't consistent across various ffmpeg versions, on one or two it didn't work at all
[06:17] <klaxa> i can't say anything for sure, i can only advise to test
[06:17] <grepper> okay
[06:18] <grepper> its too bad, I see they closed burek's wishlist/bug report
[06:18] <grepper> https://ffmpeg.org/trac/ffmpeg/ticket/1573
[06:18] <grepper> I don't understand the reasoning ...
[06:21] <klaxa> neither do i
[06:22] <grepper> strange that an project like ffmpeg doesn't support such a feature, I would have thought it to be useful to many, even essential.  Maybe just no developer with enough interest.
[06:23] <klaxa> maybe, if you are lucky you can find someone willing to implement it :P
[06:24] <klaxa> anyways, off to sleep! it's late/early enough!
[06:24] <starkline> maybe a dumb question, but is it possible to access the web interface remotely, say from two networked machines on a LAN.
[06:24] <starkline> oops, wrong chan
[07:53] <dmonjo> can someone explains why when running ffmpeg as root the conversion tfrom ogv to m3u8 works fine on the webserver but when under another user it doesnt work fine but the file plays fine?
[08:55] <dmonjo> is it normal to have an empty m3u8 file when only 1 ts is available?
[08:55] <dmonjo> when 2 or more the m3u8 starts growing
[08:55] <dmonjo> is this normal?
[08:55] <ubitux> "it doesnt work fine but the file plays fine" doesn't tell us what is the problem
[08:56] <ubitux> if it works as root but not as user, something is just wrong in your permissions
[08:56] <ubitux> your second question is also completely out of context and thus it's impossible to understand your problem
[08:58] <dmonjo> ubitux: i am trying to process m3u8 mpg-ts for iphones/ipads using this command:
[08:58] <dmonjo> ffmpeg -i http://127.0.0.1:8000:/event1.ogv -vcodec libx264 -acodec aac -strict experimental -b:v 128k -flags -global_header -map 0:0 -map 0:1 -f segment -segment_time 3 -segment_list_size 0 -segment_list /var/www/hipernation.com/wp-content/uploads/2013/03/testlist3.m3u8 -segment_format mpegts /var/www/hipernation.com/wp-content/uploads/2013/03/playn1%05d.ts
[08:59] <dmonjo> i am wondering if segments are fine with this confniguration
[08:59] <dmonjo> the ts file genertated are fine and play fine
[08:59] <dmonjo> but i am wondering if the m3u8 and the segment parameters of this command are ok
[09:01] <dmonjo> ubitux: this is my full code execution
[09:01] <dmonjo> http://pastie.org/6468938
[09:03] <ubitux> why would it be incorrect?
[09:03] <ubitux> and no need to highlight me, i don't know, i was merely suggesting how you could get potential help
[09:04] <dmonjo> sampling rate should be set to some variables for h264 hls
[09:04] <dmonjo> this is why i am asking
[09:30] <HorizonXP> hey guys
[09:30] <HorizonXP> ffmpeg can use Android's libstagefright for HW decoding
[09:30] <HorizonXP> any reason why it couldn't use it for encoding too?
[09:30] <HorizonXP> other than the fact that no one's written code for that
[10:03] <dmonjo> relaxed
[10:20] <dmonjo> how can i know if ffmpeg is using the latest lib9 tools?
[10:29] <relaxed> what is lib9?
[10:33] <dmonjo> i am using ffmpeg version git-2013-03-09-22cc8a1 is it the latest release?
[10:38] <relaxed> looks like it
[10:39] <dmonjo> how can i list what acodec ican use?
[10:40] <dmonjo>  DEA.L. aac                  AAC (Advanced Audio Coding) (encoders: aac libfaac libfdk_aac )
[10:40] <dmonjo>  D.A.L. aac_latm             AAC LATM (Advanced Audio Coding LATM syntax)
[10:40] <relaxed> -codecs
[10:41] <relaxed> use libfdk_aac
[10:41] <dmonjo> Unknown encoder 'libfdk-aac'
[10:41] <dmonjo> used -acodec libfdk-aac
[10:41] <dmonjo> ok _
[11:04] <dmonjo> leikl
[11:10] <jax> hello
[11:12] <jax> hm, why would I be losing the audio when converting something simple like: ffmpeg -i my.mpg -acodec copy my.mp4 ?
[11:12] <jax> the analysis of the input stream is: Stream #0:1[0x1c0]: Audio: mp2, 44100 Hz, stereo, s16, 224 kb/s
[11:26] <grepper> have you tried using -map to explicitly state what you want ?
[11:33] <jax> not yet
[11:44] <jax> works thx
[12:13] <dmonjo> is this error normal: Broken file, keyframe not correctly marked
[12:15] <jax> hello, trying to convert an uncompressed swf to mp4. the stream input is: Stream #0:0: Video: mjpeg, yuvj420p, 270x572 [SAR 1:1 DAR 135:286], 30 tbr, 30 tbn, 30 tbc
[12:16] <jax> when using ffmpeg -i out.swf -vcodec libx264 -an flash.mp4 it converts but i don't get a playable video. seems to be just 1 frame or something
[12:17] <jax> if i do a ffmpeg -i flash.mp4 it just shows: Duration: 00:00:00.03, Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 270x572 [SAR 1:1 DAR 135:286], 2292 kb/s, 30 fps, 30 tbr, 7680 tbn, 60 tbc
[12:25] <jax> ah probably because of Duration: N/A, bitrate: N/A in the input
[12:27] <jax> how can i figure out the duration / bitrate of the input swf?
[12:45] <MessedUpHare> Hi All,
[12:46] <dmonjo> how can i make sure my m3u8 converted file with its ts are working properly?
[12:46] <dmonjo> how can i test? avplay cannot play m3u8
[12:46] <MessedUpHare> I'm trying to apply a filter to part of a video output (by duration)
[12:46] <MessedUpHare> I think I might be able to do so by using a filter graph from the select function and i'm struggling to find any good examples of complex filtergraphs
[13:53] <fatpony> i've got a bluray with 6 channels dts hd-ma and when i extract the core, it only has 2 channels, is that normal?
[13:59] <PaperWings> #red5
[14:05] <nacron> Hello, I have a question about opening a RTSP Stream. I set av_log_set_level to debug but when I open a rtsp url only the received RTSP Packages are shown. Is there any way without sniffing packets to enable the logging of the sent packages?
[14:29] <dmonjo> hello
[14:29] <dmonjo> can anyone tell me if this is correct conversion
[14:29] <dmonjo> http://pastie.org/6471112
[14:30] <dmonjo> i am not able to play the ts file and m3u8 on ipad/iphones/ios.....
[14:30] <dmonjo> appreciate valid comments.
[15:15] <Sashmo> burek: are you around?
[15:25] <dmonjo> anyone here for assistnacE?
[15:25] <dmonjo> i am not able to convert HLS correclty
[15:29] <dmonjo> needs asssitance from someone expert with segments.......
[16:16] <dmonjo> can anyone tell me if this is correct conversion
[16:16] <dmonjo> http://pastie.org/6471112
[16:16] <dmonjo> not knowing how to convert hls correctly
[16:16] <dmonjo> doesnt read on ios
[16:52] <dmonjo> do i have to use -f hls or -f segement when doing an hls h264 transcodE?
[16:58] <Sashmo_> burek: did you get that link I sent?
[16:58] <burek> yes
[17:00] <ubitux> dmonjo: both should work in ffmpeg
[17:01] <ubitux> we added the hls muxing in segment, then the fork wanted to do it differently so the added a hls muxer with the same features
[17:01] <ubitux> and then we merged it
[17:01] <ubitux> so you should be able to do it with both hls and segment muxers
[17:06] <ubitux> dmonjo: you seem to have finally explain properly the problem on the other channel
[17:06] <ubitux> so the .ts plays properly
[17:06] <ubitux> but the .m3u8 included thos .ts doesn't
[17:06] <ubitux> is that right?
[17:06] <ubitux> are you sure the .ts links into the .m3u8 are correct?
[17:07] <ubitux> if they are, is there some explicit error in the player you are using?
[17:07] <ubitux> and if not, are you sure it's not a bug in that player?
[17:19] <Fjorgynn> test
[17:21] <dmonjo> ffmpeg -v debug -i http://127.0.0.1:8000:/event1.ogv -vcodec libx264 -acodec aac -strict experimental -b:v 128k  -flags -global_header -map 0:0 -map 0:1 -f hls -hls_time 10 -hls_list_size 999999  /var/www/hipernation.com/wp-content/uploads/2013/03/testlist12.m3u8
[17:21] <dmonjo>  is only playing for 30 sec ~ and than stopping the streaming
[17:21] <dmonjo> any hints?
[17:32] <dmonjo> it is only playing the first TS
[17:32] <dmonjo> not the whole list of m3u8
[17:39] <dmonjo> hello?
[17:43] <durandal_1707> dmonjo: your problem?
[17:43] <ubitux> dmonjo: please i asked a lot of things
[17:44] <ubitux> read what i said, or at least answer it, otherwise you won't get help
[17:44] <dmonjo> sorry i missed that
[17:44] <dmonjo> i apologize
[17:45] <ubitux> i higlighted you too times, doesn't your irc client support that?
[17:55] <dmonjo> ubitux: regarding your first question
[17:56] <dmonjo> this is my m3u8 file : http://pastie.org/6473563
[17:56] <dmonjo> the ts file sare in the same directory as the m3u8
[17:58] <ubitux> looks somehow correct, i don't remember the specs
[17:58] <ubitux> does it help if you add a '/' in front of them?
[17:59] <durandal_1707> huh, why would that help? what is actuall error?
[18:00] <dmonjo> durandal_1707: the error is that only 10 sec of the streams are read and then it hangs
[18:00] <ubitux> durandal_1707: completely random guess; also maybe it will generate a different error about the path, so that might give clues about if it's a path error or no
[18:00] <ubitux> oh then that's not it
[18:01] <ubitux> dmonjo: forget my '/' suggestion then
[18:01] <dmonjo> i think there is a problem with the m3u8 not switching ts
[18:01] <dmonjo> should i augment the hls size and time ? ;/
[18:04] <durandal_1707> each stream in playlist is longer than 10 seconds?
[18:06] <dmonjo> let me check
[18:07] <dmonjo>  -hls_time 10 means each ts should be 10 sec right
[18:07] <durandal_1707> isn't that option documented?
[18:07] <dmonjo> didnt find it
[18:08] <durandal_1707> ffmpeg -h full
[18:08] <dmonjo> not much documentation for hls
[18:08] <dmonjo> ok
[18:09] <dmonjo> -hls_time          <float>      E..... set segment length in seconds (from 0 to FLT_MAX)
[18:09] <durandal_1707> it is muxer option
[18:10] <dmonjo> will try to sec to 5
[18:10] <durandal_1707> aren't you having problem in demuxing ?
[18:11] <dmonjo> i set it now to 5sec and it played to 24sec
[18:11] <dmonjo> and cut
[18:13] <dmonjo> durandal_1707: you mean increase that value?
[18:14] <dmonjo> it is always stopping on 24sec no matter what i set it to
[18:46] <dmonjo> [ogg @ 0x3c79b20] Changing stream parameters in multistream ogg is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[18:46] <dmonjo> ogg @ 0x3c79b20] failed to create or replace streamhttp://127.0.0.1:8000:/event1.ogv: Not yet implemented in FFmpeg, patches welcome
[18:46] <dmonjo> what is this error?
[18:47] <durandal_1707> that is errror in playing multichained oggs
[18:47] <durandal_1707> it should be fixed in master/1.2 if you disable threads for libavformat
[18:48] <dmonjo> no fixes now?
[18:51] <durandal_1707> ^ what that means?
[18:52] <durandal_1707> actually ignore, this is still not supported at all
[18:57] <dmonjo> durandal_1707: you mean chanied ogg will not work because of this error?
[18:58] <durandal_1707> if they change parameters than yes, otherwise they will work
[19:00] <RobertNagy> so, I'm looking at the new frame allocation API  which is to replace the get_buffer/release_buffer API. However, it kind of breaks some stuff for me as I want ffmpeg to decode directly into OpenCL managed memory (without an extra unecessary copy)., i.e. custom memory management which was possible with get_buffer/release_buffer... any advice?
[19:01] <RobertNagy> doesn't rly look like I can achieve the same thing anymore...
[19:05] <RobertNagy> michaelni?
[19:20] <michaelni> RobertNagy, the old API should still work
[19:20] <michaelni> or is there some bug with the old get_buffer ?
[19:20] <dmonjo> ubitux: answering your second question, i noticed that my m3u8 when loaded in a player like mplayer shows 6 seconds  (which are equivalent to 1 TS file) then CUTS and then shows the nextTS so it is not harmonic, its like showing each TS alone and loading each time a new TS in the player
[19:21] <dmonjo> ffmpeg-linux64-20130308/ffmpeg  -v debug -i http://127.0.0.1:8000:/event1.ogv -vcodec libx264 -acodec aac -strict experimental -b:v 128k  -flags -global_header -map 0:0 -map 0:1 -f hls -hls_time 4 -hls_list_size 999999  /var/www/hipernation.com/wp-content/livefeeds/testlist.m3u8
[19:21] <dmonjo> i am using this command to create the m3u8
[19:23] <RobertNagy> miachelni: it works for now, but since the old api is deprecated I am a bit worried going forward
[19:27] <RobertNagy> michaelni: I've also noticed that release_buffer calls in some codecs have been removed recently
[19:27] <RobertNagy> haven't checked whether that breaks anything yet though
[19:28] <michaelni> from a user app point of view release buffer should still get called
[19:28] <michaelni> its just called differently now
[19:30] <RobertNagy> what do you mean? av_frame_unref doesn't call avctx->release_buffer...?
[19:30] <RobertNagy> as far as I can see
[19:32] <RobertNagy> it would be nice to be able to hook into the av_buffer api and do custom memory management...
[19:32] <michaelni> RobertNagy, see compat_free_buffer()
[19:32] <michaelni> it should call release_buffer()
[19:33] <michaelni> RobertNagy, hook in? like provide your own av buffers, this should be possible
[19:34] <michaelni> for decoders
[19:34] <RobertNagy> almost
[19:34] <RobertNagy> av_buffer_make_writable
[19:34] <RobertNagy> always uses av_buffer_alloc
[19:36] <RobertNagy> and realloc is a bit tricky...
[19:36] <michaelni> is anything relevant calling av_buffer_make_writable?
[19:37] <RobertNagy> checking...
[19:37] <RobertNagy> mpegvideo.c, make_tables_writable
[19:38] <RobertNagy> and even if nothing is currently, I am also thinking about the future...
[19:39] <RobertNagy> though for now, I guess the old api and compat_free_buffer should do
[19:42] <Mista_D> can't force FFmpeg to exit on a broken mpeg2ts stream. I get a lot of "max resync size reached, could not find sync byte" warnings and ffmpeg keeps going even though i set it at "-err_detect aggressive -xerror". Any advice?
[19:49] <dmonjo> guys
[19:49] <dmonjo> my m3u8 is only playing the first TS file and exiting
[19:49] <dmonjo> why is that ?
[19:49] <dmonjo> ffmpeg -v debug -i http://127.0.0.1:8000:/event1.ogv -vcodec libx264 -acodec aac -strict experimental -b:v 128k  -flags -global_header -map 0:0 -map 0:1 -f hls -hls_time 6 -hls_list_size 999999  /var/www/hipernation.com/wp-content/uploads/2013/03/testlist12.m3u8
[19:50] <dmonjo> if i ty to play the m3u8 in vlc it only plays the first 10 sec
[19:50] <dmonjo> and stops
[19:51] <llogan> dmonjo: hard to say without the the console output
[19:53] <dmonjo> llogan: http://pastebin.com/kGP1ApxS
[19:57] <alx-> hey - i followed all the steps on https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide but my ffmpeg version is still 0.8.5. how can i upgrade to the latest one?
[20:03] <dmonjo> is there like a '-segment_list_flags', '+live', ' for HLS ?
[20:11] <llogan> alx-: did you deviate from the guide at all?
[20:12] <alx-> llogan: yes, i removed --enable-libaacf --enable-libfdk-aac because they were causing errors
[20:13] <llogan> what kind of errors?
[20:14] <dmonjo> the pope is being nominated and i hope someone can assist me with my problem on this occasion :)
[20:15] <llogan> dmonjo: ffmpeg version N-37070-g532f31a appears old. current is N-50902-gde3e0ab. that's 13832 commits behind.
[20:15] <alx-> llogan: ERROR: libfdk_aac not found
[20:16] <llogan> did you install libfdk_aac?
[20:16] <llogan> if there is a problem with the guide ill fix it
[20:16] <dmonjo> llogan: shoudl  i install the git one?
[20:16] <llogan> it's usually recommended for general users to use the most recent code
[20:17] <llogan> and you can always just make and not install if you just want to test
[20:17] <llogan> (the compile guides will eventually be updated to not install)
[20:17] <alx-> i didn't think not installing libfdk_aac would have an effect on ffmpeg version but i'll try installing it
[20:17] <ezekiel> are constants like "iw", "ih", etc. totally unusable within filter_complex ?
[20:18] <ezekiel> or, do they just need to be qualified according to input stream?
[20:18] <ezekiel> I get:
[20:18] <ezekiel> Undefined constant or missing '(' in 'iw'
[20:18] <ezekiel> Error when evaluating the expression 'iw'
[20:18] <alx-> should i try to uninstall ffmpeg now, or just follow the steps in the guide again
[20:18] <ezekiel> yeah, I know, I know - but my question isn't really specific to any command
[20:18] <llogan> alx-: you can skip libfdkaac if you like, and then remove the --enable-libfdk-aac configure from the ffmpeg section
[20:19] <dmonjo> well note llogan
[20:19] <dmonjo> -acodec aac -strict experimental  can be used ?
[20:19] <dmonjo> or better to use something else?
[20:19] <ezekiel> generally, are constants like "iw" and "ih" available within filter_complex?
[20:19] <llogan> ezekiel: should be. i think it's more filter dependent, but i may be wong.
[20:19] <llogan> *wrong
[20:19] <alx-> llogan: yeah, that's what i did and my ffmpeg version id 0.8.5-4:0.8.5-0ubuntu0.12.04.1
[20:19] <llogan> that's from the repo and isn't ffmpeg from FFmpeg, but a fake, shitty version
[20:20] <llogan> dmonjo: see https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide
[20:21] <alx-> llogan: but i got ffmpeg from git://source.ffmpeg.org/ffmpeg
[20:22] <ubitux> llogan: can you contact the guy from the stackoverflow link?
[20:22] <ubitux> he seems to have shifted all his links
[20:22] <ubitux> (when adding the pic)
[20:22] <llogan> oops. i'll fix it
[20:22] <ubitux> message reworded ’ pic link, ffmpeg/libav situation ’ launchpad, etc
[20:22] <ubitux> llogan: oh, you're the guy? :)
[20:23] <llogan> would i have such a stupid name?
[20:23] <dmonjo> long live argentine
[20:23] <llogan> ubitux: don't answer that
[20:23] <ubitux> i won't
[20:24] <llogan> but the pic was worth it
[20:24] <llogan> alx-: i'm guessing that A) you didn't run the first command in the guide, and B) your compiled ffmpeg did not actually install
[20:25] <llogan> ubitux: fixed. want any other changes?
[20:25] <alx-> just checked history - i ran the first command
[20:26] <ubitux> llogan: thx
[20:26] <ubitux> llogan: no i guess that's ok
[20:28] <llogan> alx-: check your history for any errors during the ffmpeg configure, make, or checkinstall
[20:30] <llogan> alx-: or just start over, or simply use a static build
[20:31] <mogria> hey, i want to record the screen using ffmpeg and x11grab as input and encode it using libx264, but my command doesn't work, and I don't know why. Here's the command. http://pastebin.com/9nx3afdC command output follows.
[20:31] <alx-> llogan: thanks, im gonna start over
[20:32] <llogan> mogria: don't use -sameq. where is the console output?
[20:32] <mogria> here's the output: http://pastebin.com/4EPnhrfK
[20:33] <mogria> llogan, what else should I use to get 1080p quality?
[20:33] <ubitux> mogria: [ac3 @ 0xbd0240] invalid bit rate
[20:33] <ubitux> by the way, you're not using ffmpeg
[20:33] <llogan> mogria: ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers
[20:34] <ubitux> mogria: see this ^
[20:34] <llogan> what ubitux said
[20:34] <llogan> we don't serve their (libav) kind
[20:34] <llogan> Name the movie. Bonus points for the character.
[20:34] <ubitux> mogria: you may want to test with a real ffmpeg like distributed above ^
[20:35] <mogria> are these packages also available on the Ubuntu Repos, or on a PPA?
[20:37] <mogria> is this the official PPA? https://launchpad.net/~jon-severinsson/+archive/ffmpeg
[20:37] <llogan> not in official repo. there is an unofficial PPA, but it is a little graybeard. https://launchpad.net/~jon-severinsson/+archive/ffmpeg
[20:37] <llogan> by that I mean it is old
[20:38] <mogria> then I'll try a static build
[20:39] <ezekiel> for anyone else using "iw", and "ih" in their filters, here's my working output with hardcoded values: http://pastie.org/private/olv7t4ykfs0jqcfrdmc5bw, and the failure using "iw"/"ih": http://pastie.org/private/row0z50po2ov37994qtnsg
[20:40] <ezekiel> maybe I'm just doing it totally wrong and need some stream qualifiers in there or something
[20:41] <ezekiel> fwiw, same error if I name streams in the filter: http://pastie.org/private/z3nygabqxn4j8nkceprtyg
[20:42] <llogan> use w, not iw for overlay
[20:42] <llogan> http://ffmpeg.org/ffmpeg-filters.html#overlay-1
[20:43] <llogan> i guess that's a worthy nit that could be fixed
[20:43] <llogan> perhaps as a simple qualification task.
[20:47] <mogria> the latest static build doesn't recognize x11grab in the -f option? how come? I used the first example on this site: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(screen)%20with%20FFmpeg
[20:47] <mogria> this message appears:  Unknown input format: 'x11grab'
[20:47] <llogan> oh, i forgot the static builds don't support x11grab. IIRC.
[20:48] <mogria> that's a bummer
[20:49] <ezekiel> llogan: awesome, thank you - sometimes my eyes just quit serving their purpose - mixing up my use of constants between filters without even noticing
[20:49] <ezekiel> very much appreciated
[20:50] <dmonjo> N-50904-gfebd78e
[20:50] <dmonjo> installed
[20:50] <llogan> ezekiel: np. i'm procrastinating a server move with ~50 domains.
[20:50] <ezekiel> now, the real reason I'm re-working the overlay filters - once overlayed, the videos are out of sync!
[20:51] <ezekiel> is this almost assuredly because of timestamps in the source videos?
[20:51] <mogria> llogan, is there also a channel for the libav version of ffmpeg, to ask there?
[20:52] <sacarasc> #libav
[20:52] <ezekiel> aaahhh, the daunt of server moves, I know that feeling well
[20:52] <mogria> llogan, sacarasc, thank you
[20:53] <llogan> mogria: they'll probably tell you to compile git head
[20:53] <llogan> ezekiel: Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp
[20:53] <llogan> as mentioned in the overlay filter docs
[20:53] <llogan> see overlay examples for an example using setpts
[20:54] <ezekiel> haha, yep - righ there at the bottom
[20:54] <ezekiel> I could squeeze you
[20:54] <ezekiel> and the docs writers
[20:54] <ezekiel> for being rad, I mean
[20:54] <llogan> that's a first. usually people are angry at them
[20:54] <ezekiel> dealing with ffmpeg, I've definitely been impressed that the docs are MORE up-to-date than even some of the code :)
[20:55] <ezekiel> but still I find myself in old habits of thinking the docs are not so current
[20:55] <llogan> there is also the community wiki for more newuser centric guides
[20:55] <llogan> https://ffmpeg.org/trac/ffmpeg
[20:56] <dmonjo> llogan: http://pastebin.com/zBgWGmyr
[20:56] <dmonjo> still my m3u8 plays the first TS file
[20:56] <dmonjo> and not all the TS files sequentially
[20:57] <llogan> dmonjo: i have to go. i'll try to look later, but i may not. send message to ffmpeg-user mailing list if you don't get an answer here.
[20:59] <dmonjo> :(
[21:16] <dmonjo> btw how can i patch the system >?
[21:16] <dmonjo> https://ffmpeg.org/trac/ffmpeg/attachment/ticket/1842/segment.patch for example this patch
[21:16] <dmonjo> how can i apply it ?
[21:19] <dmonjo> i patch configure make and makeinstall ?
[21:23] <dmonjo> patch -p1 < segment.patch
[21:23] <dmonjo> patch unexpectedly ends in middle of line
[21:23] <dmonjo> : ? ? ??
[21:27] <dmonjo> what si this ? 9 out of 9 hunks FAILED -- saving rejects to file home/zcp/ffmpeg-1.0/libavformat/segment.c.rej
[21:30] <jpsantos> I have a question about exporting single images to gif, all I seem to be getting is grey, any suggestions?
[21:42] <jpsantos> anyone?  why does ffmpeg export all solid grey gifs?
[21:44] <jpsantos> it's like the gif palette isn't being saved
[21:46] <ezekiel> bah - I still can't get the two videos I'm overlaying to be in sync. makes me wonder if there's something wrong with the source videos
[21:51] <jpsantos> nice ffmpeg -i ../timelapse-14-24fps-mjpeg-960p-20110414183412.avi -an -pix_fmt rgb24 -f image2 -r 24 -t 1 -threads 2 gifs/test-`date +%Y%m%d%H%M%S`-%04d.gif
[21:51] <jpsantos> you want to see the output too yes?
[21:54] <jpsantos> http://sprunge.us/BjQc
[21:55] <jpsantos> the gifs seem to have the data, but the palette is missing - pure grey
[22:03] <jpsantos> llogan, anything?
[22:14] <dmonjo> anyone here?
[22:16] <llogan> jpsantos: only idea i have is to try a recent static build http://ffmpeg.gusari.org/static/
[22:17] <jpsantos> hmm
[22:17] <jpsantos> I will try another version
[22:17] <jpsantos> I have access to
[22:17] <jpsantos> but *sigh*
[22:18] <llogan> jpsantos: also see http://superuser.com/a/556031/110524
[22:19] <llogan> jpsantos: you want to try a recent version in case you are experiencing a bug. FFmpeg development is very active.
[22:19] <jpsantos> *nod*
[22:19] <jpsantos> thank you
[22:19] <jpsantos> this might actually do that I need it to do
[22:19] <jpsantos> if I can export to jpg
[22:19] <jpsantos> and convert will create a gif
[22:19] <jpsantos> then this will save me trouble
[22:20] <llogan> if you choose jpg then add -qscale:v 2 to improve output quality (but there was an old bug that may affect your version where -qscale was  ignored for jpg)
[22:26] <dmonjo> llogan: any idea about my problem ? :/
[22:30] <llogan> why are you applying that old patch?
[22:36] <dmonjo> llogan: it is already applied in the new release
[22:37] <dmonjo> llogan: i think what i am going through i s a bug
[22:37] <dmonjo> do you have time to go over it?
[22:37] <dmonjo> it is interresting
[22:43] <llogan> dmonjo: sorry, i have no experience with hls and segment stuff
[22:44] <dmonjo> anyone here aexpert in this?
[22:44] <llogan> if you think you found a bug then you can report it (but search for existing report first)
[22:44] <dmonjo> will do
[22:45] <dmonjo> where can i see the existing reportS?
[22:45] <llogan> if it is not considered a bug then it will simply be closed
[22:45] <dmonjo> not solved yet
[22:45] <llogan> http://ffmpeg.org/bugreports.html
[22:45] <dmonjo> i mean the list of opwnws reports
[22:46] <llogan> https://ffmpeg.org/trac/ffmpeg/report
[22:46] <llogan> or the search box in upper right
[22:48] <dmonjo> thank you
[22:50] <saste> dmonjo, how old is your ffmpeg?
[22:50] <saste> also what's the problem?
[22:50] <dmonjo> llogan: http://pastebin.com/zBgWGmyr
[22:50] <dmonjo> ok
[22:50] <dmonjo> i am converting from a ogv to an m3u8
[22:51] <dmonjo> ogv is a live stream fed to icecast first then fed to ffmpeg to convert it to st files
[22:51] <dmonjo> if i try to open the m3u8 file being generated on the spot using vlc or other players, i can only process the first few segments of the m3u8 and then it stops
[22:52] <saste> dmonjo, is the generated m3u8 correct?
[22:52] <dmonjo> how can i know if it is correct, it is playing part of the ts only not all of them seemlisly
[22:52] <dmonjo> i can point you to the m3u8 if you would like
[22:52] <eric__> If in case of MP4 moov atom at the end of the file, can we do online transcoding
[22:53] <saste> dmonjo, yes
[22:53] <dmonjo> Sashmo_: please give me a few minutes
[22:53] <saste> but m3u8 is human readable, it's a simple textual playlist
[22:54] <dmonjo> http://pastie.org/6479372
[22:55] <dmonjo> saste: ^^ the file
[22:55] <eric__> any suggestions? If in case of MP4 moov atom at the end of the file, can we do online transcoding
[22:55] <saste> dmonjo, seems pretty sane ...
[22:56] <dmonjo> saste: yes, the problem is when switching ts i think something is missing in the headers of the ts files :/
[22:57] <saste> dmonjo, -flags -global_header <- does it help?
[23:00] <dmonjo> saste: if i dont include the -flags -global_header i get an error
[23:00] <dmonjo> http://pastebin.com/JAL9yz0J
[23:02] <saste> dmonjo, typo
[23:03] <dmonjo> yea removing the -flags
[23:03] <saste> i see no global_header
[23:03] <dmonjo> saste please note that this is the main command and the output i am receiving:
[23:04] <dmonjo> http://pastebin.com/zBgWGmyr
[23:04] <dmonjo> i thought you asking me to ommit the headers
[23:04] <saste> dmonjo, i asked to try the option -flags -global_header
[23:05] <dmonjo> ok let me try it now and paste the output
[23:10] <dmonjo> saste: http://pastebin.com/fbeKV9px
[23:10] <dmonjo> same problem occuring if i point to the m3u8 file using vlc i can play only the first segment
[23:11] <saste> dmonjo, try with ffplay just for testing
[23:12] <saste> also. ffmpeg -f lavfi testsrc -vcodec libx264 -acodec aac -strict experimental -b:v 128k  -flags -global_header -map 0:0 -map 0:1 -f hls -hls_time 6 -hls_list_size 999999  /var/www/hipernation.com/wp-content/livefeeds/zac.m3u8
[23:12] <dmonjo> saste: it is the same problem with ffplay
[23:12] <saste> so you are sure it is not an issue depending on the input
[23:15] <dmonjo> saste: btw ffplay doesnt exit after playing the first TS, it plays it then stops hangs  for some time and and plays the second ts...
[23:16] <saste> dmonjo, which input?
[23:17] <dmonjo> ffplay http://...../<m3ua>
[23:19] <saste> dmonjo, start simple, then add complexity
[23:19] <saste> start with a local test
[23:19] <saste> ffmpeg -f lavfi -i testsrc -codec:v libx264 -b:v 128k -map 0:0 -f hls -hls_time 6 -hls_list_size 999999 out.m3u8
[23:19] <saste> dmonjo, that's for a start
[23:19] <saste> i used the hls muxer and i didn't experience those problems
[23:20] <saste> it may also be due to a bandwidth problem
[23:22] <saste> dmonjo, as alternative you may try the segment (more powerful and more complex), but i don't think it would make any difference
[23:23] <dmonjo> saste: the ffmpeg -f lavfi -i testsrc works fine without disconnection
[23:23] <dmonjo> plays only to 59 right
[23:23] <dmonjo> it stopped on 59
[23:26] <dmonjo> saste: playing out.m3u8 plays only from 30 to 59 and then stops is this normal? can we make it count more ?
[23:26] <dmonjo> i generated it on the server and tried playing it from my computer using ffplay http://....out.m3u8
[23:29] <llogan> eric__: yes
[23:30] <llogan> you can relocate the atom with "-movflags faststart" or by using the qt-faststart tool
[23:31] <llogan> it allows the video to begin playback before the client has completely downloaded the file
[23:32] <saste> dmonjo, -hls_list_size 999999?
[23:35] <Kion> how can I record my screen using avconv?
[23:44] <dmonjo> saste: made it hls_list_size 999 and still same problem
[23:47] <saste> dmonjo, you need to stop the encoding, or use -re otherwise it will continue to write the file forever
[23:47] <dmonjo> saste: ok now when i reduced it to 999 it counts from 0
[23:48] <dmonjo> when it was 9999999 the clock starting counting from 30 to 59
[23:48] <dmonjo> saste: the clock is exiting after few seconds
[23:49] <dmonjo> saste: ffmpeg -f lavfi -i testsrc -codec:v libx264 -b:v 128k -map 0:0 -f hls -hls_time 6 -hls_list_size 999999 out.m3u8
[23:49] <llogan> Kion: avconv isn't from FFmpeg.
[23:50] <dmonjo> if i try to ffplay that m3u8 the clock counts from 0 to 10 and exits
[23:50] <Kion> llogan: I thought that avconv was a fork of ffmpeg...
[23:53] <saste> dmonjo, ffmpeg -f lavfi -i testsrc=d=60 -codec:v libx264 -b:v 128k -map 0:0 -f hls -hls_time 6 -hls_list_size 999999 out.m3u8
[23:53] <saste> ffplay out.m3u8
[23:53] <saste> ^^ works fine here
[23:53] <dmonjo> it works now coz i exited the encoding
[23:53] <dmonjo> let me delete it and restart it
[23:55] <saste> same with vlc out.m3u8
[23:55] <dmonjo> ffplay http://www2.hipernation.com/wp-content/livefeeds/out4.m3u8
[23:55] <dmonjo> starts at 230
[23:56] <llogan> Kion: ^
[23:56] <dmonjo> saste: stops at 259
[23:56] <dmonjo> almost 20sec of playing only
[23:56] <dmonjo> saste: how is it playing on your end?
[23:59] <llogan> Kion: also http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html
[23:59] <saste> dmonjo, please check the options, in particular you need to specify -hls_list_size
[23:59] <dmonjo> saste: the testsrc is eating up my memory
[23:59] <dmonjo> it is getting killed
[23:59] <dmonjo> Killed10078 fps=105 q=7.0 size=N/A time=00:06:41.40 bitrate=N/A
[00:00] --- Thu Mar 14 2013