[Ffmpeg-devel-irc] ffmpeg.log.20130328

[burek]

[01:09] <smus> avcodec_encode_video(...) seems to return a valid byte size, but the output buffer I pass in is just zeros. Any ideas? Here's my code (very slightly modified from example): http://pastebin.com/M5VuHM9b
[02:59] <Jordan_> what would happen if you use an x86 ffmpeg build to run on an input file greater than 2GB?
[02:59] <Jordan_> is that a problem?
[03:02] <iive> Jordan_: do you mean 32bit build? It should not be a problem, as unix provides 64 seeking for 32 bit functions.
[03:02] <Jordan_> yes 32bit build and i'm on windows
[03:04] <Jordan_> i'd like to test it to be sure, can you append multiple files together?
[03:05] <iive> well, windows should also provide functions for 64 access, I hope the mingw uses them.
[03:06] <iive> there are some ways to concatenate files, but i'm not sure how and how good it is.
[03:07] <relaxed> use MP4Box to concat .mp4 and mkvmerge for everything else
[03:07] <relaxed> ffmpeg works for mpeg containers
[04:30] <relaxed> My static builds now have ffmpeg with 8bit & 10bit libx264 support.
[04:40] <sacarasc> Cool.
[05:01] <relaxed> I plan to include an x264 binary with lavf, swsscale, and gpac support as well.
[11:32] <Zeeflo> has anyone here experience with setting up a RTMP server?
[11:32] <Zeeflo> in other words, has anyone experience with setting up a CDN?
[11:41] <Mavrik> which one?
[11:41] <Mavrik> Wowza works very nicely
[11:43] <Zeeflo> wowza
[11:46] <Zeeflo> it looks like wowza is a service like amazon, where you pay to have them do your streaming?
[11:47] <Zeeflo> ah.. its server software
[11:48] <Zeeflo> Mavrik, do you use it yourself?
[11:49] <Mavrik> used it
[11:49] <Mavrik> bbl
[12:23] <brontosaurusrex> is there a way to embed a thumbnail into mp4?
[13:50] <__raven> hi
[13:50] <__raven> what is the right encoder for mpeg1 layer 2 audio?
[13:51] <durandal_1707> mp2?
[15:51] <Zeeflo> anyone here familiar with wowza?
[16:22] <ubitux> GRMrGecko: also, what configure flags are you using?
[16:23] <GRMrGecko> ubitux: http://p.webra.in/d5
[16:24] <ubitux> (--disable-shared --enable-static is not necessary)
[16:24] <ubitux> (--enable-pthreads as well afaik)
[16:25] <GRMrGecko> ubitux: Here is the full config with the result of my script http://p.webra.in/d1
[16:27] <GRMrGecko> I'm able to compile for x86_64 just fine
[16:28] <GRMrGecko> it is erroring in vf_yadif_init.c
[16:28] <GRMrGecko> I'm looking through code
[16:29] <ubitux> feel free to discuss on the ffmpeg-devel channel if you're willing to send a patch
[16:29] <ubitux> (sorry for the back & forth)
[16:29] <ubitux> it's likely a macro definition mess again
[16:30] <GRMrGecko> ubitux: ok
[16:32] <GRMrGecko> ubitux: Forgetting C& Because xmm_reg is defined twice in "typedef struct xmm_reg { uint64_t a, b; } xmm_reg;" isn't that the issue?
[16:33] <GRMrGecko> should it be something like "typedef struct xmm_reg_struct { uint64_t a, b; } xmm_reg;"
[16:33] <ubitux> i admit i didn't look yet
[16:33] <GRMrGecko> or maybe forget the last part of xmm_reg
[16:37] <GRMrGecko> ubitux: It's not fully compiled yet, but it seems to have passed the point where it errored out.
[16:37] <GRMrGecko> "typedef struct { uint64_t a, b; } xmm_reg;"
[16:39] <GRMrGecko> compiled
[16:55] <SenGlar> hi, i need to know if it is possible to set the encoding audio frame size to another size a little bit larger. For example, using 48kHz 2ch S16, the frame size should be 4096Bytes, but I'm getting from a camera a size of 7680. It is possible to encode such size per frame?
[16:58] <ubitux> do you still need to set the size with a recent api?
[16:59] <jure> hi
[17:00] <jure> can fdk-aac encode with a variable bitrate but at a nominal bitrate?
[17:03] <durandal_1707> SenGlar: latest api have no limit on frame size that is going to be encoded/decoded
[17:03] <SenGlar> @ubitux : I'm using the Magic version for implementing my encoder, and it seems that most encoders/codecs are not able to encode variable frame size, and I'm wondering if it could be easily done if I add the flag for example and/or so on.
[17:05] <durandal_1707> Magic version?
[17:05] <SenGlar> @durandal_1707 : but if I'm using an audio configuration of 48kHz stereo and S16 it seems that I can't encode 7680Bytes per frame,
[17:05] <SenGlar> ffmpeg 1.2 version :P
[17:06] <durandal_1707> SenGlar: you did not provided enought info, decoding/encoding >7680 bytes per frame is certainly possible
[17:07] <SenGlar> with AAC (the native one or libfaac) encoder it is also possible?
[17:08] <SenGlar> I want to encode 7680Bytes per frame that I got from a camera that is working at 25fps 48kHz stereo S16
[17:09] <Washu> Is there a way to concatenate all files in a directory as opposed to putting all the filenames in a file first?
[17:09] <durandal_1707> aac can not encode arbitary size, and that is codec limit and not api limit
[17:09] <SenGlar> OK!
[17:11] <SenGlar> I need to know if it was a technologic limit or API limit, because when I implemented this scenario with the oggvorbis API from xiph.org I could do that with no problem. So I was wondering if it could be done, also because of the comparison that using the vorbis encoder of the ffmpeg API it can't be done neither.
[17:11] <durandal_1707> Washu: via concat protocol but that probably does not work in your scenario
[17:12] <Washu> durandal_1707: doesn't that involve typing all the filenames into the command though?
[17:12] <durandal_1707> Washu: yes, if you want some feature open bug ticket
[17:16] <Washu> so anyone using ffmpeg put all the filenames in a file or in the command to concat? seems pretty complicated for it to always have been like that
[17:18] <durandal_1707> if nobody complains (open ticket) it will reamain in that state
[17:19] <Washu> Fine, just seems weird that a project like this wouldn't have thought of that which leaves me with the impression the bug will be ignored :P
[17:19] <SenGlar> @ druandal_1707 : so, do you mean that it is technologically impossible to re-implement the AAC codecs used in the ffmpeg API to reach the specs I've told?
[17:21] <durandal_1707> SenGlar: i already told you that aac does not support arbitary frame size, should API have support for some kind of fifo is different question
[17:25] <Washu> SenGlar: I think the answer to your question is yes if you wanted to write some AAC if it's even possible at all with that codec but ffmpeg doesn't support what you asked for because aac itself does not suppprt it
[17:25] <SenGlar> Ok! thanks!  Also to point out that I've implemented a fifo structure with the methods of the ffmpeg API but it breaks the 7680Bytes in 4096Bytes and encode it, then the residual 3584B are also encoded but with padding...
[17:25] <SenGlar> Thanks anyway people!
[17:25] <SenGlar> You are doing a good job! :)
[17:32] <durandal_1707> SenGlar: you mean with last frame?
[17:33] <durandal_1707> not all codecs supports trucated/small last frame
[17:34] <durandal_1707> but bad native aac encoder supports small last frame
[17:51] <SenGlar> yes, but I don't want a last frame with padding, I want to get a continue stream that is getting 7680 raw bytes per frame to encode
[18:11] <SenGlar> I'm leaving!
[18:11] <SenGlar> thanks anyway! :)
[18:11] <SenGlar> See yah!!
[20:55] <Bor0> what exactly does "crippled build" mean when specifying --disable-asm?
[20:55] <Bor0> yasm*
[20:58] <JEEB> it will be slow
[20:58] <JEEB> as in, the usefulness will be much much lower
[21:01] <Bor0> could that be the reason why video playback for some videos causes the player to freeze completely?
[21:01] <Bor0> and then eventually crash
[21:01] <bparker> o/
[00:00] --- Fri Mar 29 2013