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February 2012

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ffmpeg-devel.log.20120223
by burek 24 Feb '12

24 Feb '12

[02:19] <durandal_1707> michaelni: cehoys did channel layout for flac decoder wrong [02:37] <michaelni> durandal_1707, elaborate please [02:37] <michaelni> what commit & whats wrong [02:37] <michaelni> ? [03:33] <Zeranoe> I see FFmpeg supports v412, I'm not sure what this lib is, is it linux video or something? Is there a home page I'm missing? [03:41] <durandal_1707> Zeranoe: v4l2 [03:41] <durandal_1707> aka video for linux 2 [03:46] <Zeranoe> durandal_1707: Thanks, a silly mistake on my part with the 1 and l [03:48] <Compn> v4l is like dshow for linux :P [05:03] <CIA-17> ffmpeg: 03Diego Biurrun 07master * r72b95764d0 10ffmpeg/libswscale/ (rgb2rgb.c rgb2rgb_template.c): swscale: variable declaration and placement cosmetics [05:03] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r0a7ce3caa8 10ffmpeg/libavformat/flvdec.c: [05:03] <CIA-17> ffmpeg: flvdec: Fix indentation [05:03] <CIA-17> ffmpeg: Also split a long line. [05:03] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [05:03] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r383a3b64cb 10ffmpeg/libavformat/ (isom.h mov.c): (log message trimmed) [05:03] <CIA-17> ffmpeg: movdec: Restart parsing root-level atoms at the right spot [05:03] <CIA-17> ffmpeg: If parsing moov+mdat in a non-seekable file, we currently [05:03] <CIA-17> ffmpeg: abort parsing directly after parsing the header of the mdat [05:03] <CIA-17> ffmpeg: atom. If we want to continue parsing later (if looking to [05:03] <CIA-17> ffmpeg: parse later fragments), we need to skip past the content of the [05:03] <CIA-17> ffmpeg: mdat atom, otherwise we end up parsing the content of the mdat [05:03] <CIA-17> ffmpeg: 03Alex Converse 07master * r0ad522afb3 10ffmpeg/libavformat/isom.c: mov: Add support for MPEG2 HDV 720p24 (hdv4) [05:03] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r0c7b8b758a 10ffmpeg/libavformat/ (movenc.c movenc.h): (log message trimmed) [05:03] <CIA-17> ffmpeg: movenc: Buffer the mdat for the initial moov fragment, too [05:03] <CIA-17> ffmpeg: This allows writing QuickTime-compatible fragmented mp4 (with [05:03] <CIA-17> ffmpeg: a non-empty moov atom) to a non-seekable output. [05:03] <CIA-17> ffmpeg: This buffers the mdat for the initial fragment just as it does [05:03] <CIA-17> ffmpeg: for all normal fragments, too. Previously, the resulting [05:03] <CIA-17> ffmpeg: atom structure was mdat,moov, moof,mdat ..., while it now [05:03] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r58700edb94 10ffmpeg/libavformat/cdxl.c: [05:03] <CIA-17> ffmpeg: cdxl: correctly synchronize video timestamps to audio [05:03] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [05:03] <CIA-17> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [05:03] <CIA-17> ffmpeg: 03Tim Walker 07master * r147715ca10 10ffmpeg/libavutil/audioconvert.c: [05:03] <CIA-17> ffmpeg: Add channel names to channel_names[] array for channels added in b2890f5 [05:03] <CIA-17> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [05:03] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r0584e3ca97 10ffmpeg/ (libavformat/utils.c tests/ref/fate/wtv-demux): [05:03] <CIA-17> ffmpeg: lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields(). [05:03] <CIA-17> ffmpeg: It is not supposed to be done outside lavc. [05:03] <CIA-17> ffmpeg: This is basically a revert of 818062f2f346df30f4ec0c0c1f54e8025cc3a80a. [05:03] <CIA-17> (64 lines omitted) [05:04] <overflow_0f8b> static void* array[] = { &&foo, &&bar, &&hack }; goto *array[i]; [05:31] <CIA-17> ffmpeg: 03Compn 07master * rb18c27def8 10ffmpeg/libavformat/riff.c: riff: IPJ2 decodes with j2k decoder. samples can be made from mencoder and jp2avi.dll codec. [05:37] <Compn> yay, a commit! [05:38] <Compn> wonder if theres any other j2k fourccs [09:11] <CIA-17> ffmpeg: 03Nicolas George 07master * r69a653cac3 10ffmpeg/configure: [09:11] <CIA-17> ffmpeg: configure: add lavfi dependencies to .pc files. [09:11] <CIA-17> ffmpeg: lavfi have optional filters that depends on some components: [09:11] <CIA-17> ffmpeg: it is necessary to test which one is enabled to set the correct [09:11] <CIA-17> ffmpeg: dependencies. [09:11] <CIA-17> ffmpeg: Signed-off-by: Nicolas George <nicolas.george(a)normalesup.org> [10:00] <CIA-17> ffmpeg: 03Nicolas George 07master * r42655b24c2 10ffmpeg/libavfilter/avfiltergraph.c: [10:00] <CIA-17> ffmpeg: avfiltergraph: avoid harmful auto audio conversions. [10:00] <CIA-17> ffmpeg: With the current code, an automatically inserted aconvert necessary [10:00] <CIA-17> ffmpeg: for format change would usually convert to mono for no good reason. [10:00] <CIA-17> ffmpeg: The new code will not avoid all conversions, but at least will keep [10:00] <CIA-17> ffmpeg: them among the layouts common to both filters. [10:32] <kriegerod> ubitux, need help on asetframesize [10:33] <kriegerod> can't figure out the meaning of filter_samples(), and can't understand where which logics part should be [10:35] <kriegerod> ubitux, https://gist.github.com/1891931 [12:26] <ubitux> kriegerod: filter_samples() is simply the callback called everytime there are samples; you seem to have understood that by looking at your code [12:26] <ubitux> i'm not sure to understand what you want [12:28] <kriegerod> ubitux, i looked at af_volume, that had _only_ filter_samples, and initially all work was done in filter_samples in my filter. That didn't work (probably that function did not even got called i think, coz i can't find where from it could be called) [12:28] <kriegerod> i need large explanation what should i do in .poll_frame, .request_frame, .filter_samples [12:28] <kriegerod> and which of them this filter needs [12:30] <kriegerod> ubitux: could you explain me that right now? i need to move on with my task, and this filter is part of it [12:31] <ubitux> ok so, you can't have only filter_samples() because you are not sure you can return samples all the times (your fifo might be incomplete) [12:32] <ubitux> so poll_frame should tell if your fifo can return a buffer of size N (basically if your fifo has enough data) [12:33] <ubitux> and request_frame() should transfer that data (call the next filter_samples) [12:33] <kriegerod> when filrer_samples ever gets called? seems only from avfilter_filter_samples() that is external API call, which has no examples of usage [12:33] <ubitux> your filter_samples should only keep adding to the fifo [12:34] <ubitux> it should be called regulary; isn't it the case? [12:34] <ubitux> everytime the previous filter has data to send actually [12:34] <kriegerod> maybe, but i wasn;t able to track the call ot it [12:34] <kriegerod> looking at sources [12:35] <ubitux> you may want to have a look to vf_thumbnail [12:35] <ubitux> vf thumbnail accumulates N frames [12:35] <ubitux> and returns only one [12:35] <ubitux> it's only for video, but i guess it could apply to audio somehow [12:36] <kriegerod> i can't look at video filters coz i don't know what's draw_slice, end_frame, start_frame [12:39] <kriegerod> ubitux: so my filter_samples and poll_frame look ok, right? [12:40] <ubitux> from my point of view it looks like it's the correct way; of course i could be wrong, or there might just be bugs in it [12:41] <kriegerod> i'd be grateful if you look through my request_frame, which looks wrong according to your above explanation. [12:41] <kriegerod> i'm not confident what should it look like [12:44] <kriegerod> btw in which order should these functions be called? [12:45] <kriegerod> e.g. i av_asrc_buffer_add_buffer() one large frame, then avfilter_poll_frame(). But my filter says it has 0 bytes in fifo [12:47] <ubitux> mmh give me a few minutes, i'm going to have a look [12:58] <ubitux> kriegerod: when you call avfilter_poll_frame(), it should call go up to the top filter to call the first filter poll_frame [12:58] <ubitux> the first one should be abuffer [12:59] <ubitux> check if the poll_frame() from abuffer is actually called, and raise something [12:59] <kriegerod> so i guess in poll_frame i should request frames from upper layer, and accumulate new data in fifo? [12:59] <kriegerod> to give answer if i can give new frame [13:00] <kriegerod> or...? [13:01] <ubitux> it's likely you indeed have to request_frame() from your poll_frame() if you don't have enough data but the source has [13:02] <ubitux> so then if you still have not enough data, you can return 0 in poll_frame(), or otherwise that you actually have some now [13:03] <ubitux> i wonder if that couldn't be improved [13:03] <ubitux> it's quite tricky [13:04] <kriegerod> so in my poll_frame, after calling avfilter_request_frame(), i should get AVFilterBufferRef using avfilter_get_audio_buffer(), right? [13:05] <kriegerod> ah no, wrong [13:06] <kriegerod> yep, where from should i pull data, if avfilter_request_frame succeeds? [13:06] <kriegerod> if my poll_frame param is outlink, then outlink->src->outputs[0]? [13:09] <ubitux> in your poll_frame, you call avfilter_request_frame from the input link (likely abuffer if your filtergraph is simple); if you look at the abuffer request frame callback, it will calls avfilter_filter_samples [13:10] <ubitux> look at the poll_frame() callback in vf_thumbnail, it should be something similar [13:12] <ubitux> i need to go, 'be back later [13:12] <kriegerod> ah, so when i call avfilter_request_frame(outlink->src->inputs[0]), it triggers _my_ filter_samples, so i can probably get new data in my fifo after that? [16:01] <CIA-17> ffmpeg: 03Nicolas George 07master * rb6ff81d63b 10ffmpeg/libavformat/sbgdec.c: sbgdec: read_seek was undeprecated. [16:16] <durandal_1707> there should be nicer way to maninpulate planar pcm formats [16:18] <Daemon404> while i knoww hat it is, planar is suck a crap name for it [16:18] <Daemon404> :P [16:18] <Daemon404> too bad i dunno what else to call it [16:18] <av500> i think it comes from the video world [16:19] <Daemon404> it does [16:19] <av500> so, call it channular [16:19] <Daemon404> sounds liek tubular... :# [16:19] <Daemon404> :3 [16:22] <durandal_1707> heh, i'm proposing killing s8 pcm codec and crating s8 sample format [16:28] <durandal_1707> and extend libswresample to handle planar so that direct conversion is possible between them (something like how it is done in sws) [16:57] <kriegerod> ubitux, posted the new audio filter, please comment when you can [16:58] <Daemon404> microchip_, i have a patch to ffmpeg, but i cant send it, as you havent merged changes from libav that it requies. [17:16] <durandal_1707> Daemon404: ? [17:17] <Daemon404> kostya's wavpack fix wasnt merged [17:17] <Daemon404> so my final little fix wont apply ofc [17:17] <durandal_1707> Daemon404: it was reverted is better word [17:17] <Daemon404> i see [17:17] <Daemon404> which reason is this [17:17] <durandal_1707> np, michael will revert revert :)) [17:17] <Daemon404> my little 2-line patch has passed all my stress tests on it [17:18] <Daemon404> all 5 types of lossy and lossless, mutilple chan layouts, the "special" files in teh test suite [17:18] <Daemon404> all work now! [17:19] <Daemon404> er s/\!/\~/ [17:19] <Compn> j-b : another j2k fourcc IPJ2 [17:20] <Compn> imagepower j2k encoder creates such files [17:21] <durandal_1707> Compn: just add IPJ(0-9) and be done with it :) [17:21] <j-b> Compn: I hate j2k with force [17:23] <av500> may the force be hate you [17:23] <Daemon404> is vlc tryingto be an image viewer? [17:23] <av500> think animated J2K [17:23] <Daemon404> motion j2k sounds saner :P [17:24] <Daemon404> name-wise [17:24] Action: av500 likes animated H264 intra [17:27] <Compn> Daemon404 : j2k is also a video codec :P [17:27] <Daemon404> any image format can be a video codec [17:27] <Daemon404> just means it be intra-only [17:28] <Compn> j-b : dont worry, everyone else does too. [17:28] <j-b> Compn: weren't they supposed to write a new openjpeg, a bit less slow? [17:28] <Compn> durandal_1707 : i think the j2 stands for jpeg2000... so its not like ipjx would be a fourcc [17:29] <Compn> j-b : i dont follow openjpeg news [17:30] <Compn> j-b : anything going on in vlc land ? [17:30] <av500> 2.0 hate [17:30] <Compn> or 'videolan' land as they call it now [17:30] <av500> videoland [17:34] <j-b> Compn: 2.0 got released. Some issues, but not much on Linux/win32. Complete crazy on Mac. [17:34] <j-b> Compn: mxf crash in libavformat [17:34] <j-b> 2.0.1 on the way [17:35] <av500> j-b: why did you not test on mac? [17:35] <j-b> because I hate steve jobs [17:35] <Compn> can we piss on his grave yet? :P [17:35] <av500> j-b: you hate a dead man? [17:35] <j-b> I hope so. [17:36] <av500> you a dead man hatin' [17:36] Action: durandal_1707 mutes haters [17:36] <j-b> mxf crash is annoying [17:36] <j-b> http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=570f6aaab12e2bc;sf=tgz is what we use [17:36] <j-b> aka jump-to-54 commit - 1 [17:41] <durandal_1707> and mxf crash is reproducible with latest git version? [17:43] <j-b> no idea [18:24] <cbsrobot> guess somebody needs to create x2k project then [18:26] Action: durandal_1707 needs :time: [18:27] <cbsrobot> durandal_1707: here you go: http://www.youtube.com/watch?v=vtveZ0t48p4 [18:32] <Daemon404> ... disappointing [18:32] Action: Daemon404 was expecting pink floyd [18:33] <durandal_1707> i'm not fan of either [18:34] <durandal_1707> cbsrobot: http://www.youtube.com/watch?v=0I8N74t2x50 [19:30] <cbsrobot> durandal_1707: reminds me of http://www.youtube.com/watch?v=pFwR1c-cdFw [19:38] <durandal_1707> cbsrobot: when that was composed? [19:41] <ubitux> Koyaanisqatsi, 1982, with Sigur Ros music e [19:42] <ubitux> (iirc) [19:44] <ubitux> my bad, maybe not sigur ros [19:45] <Daemon404> michaelni, libav dropped libpostproc now. you have access to standalone libpostproc repo on videolan servers. [19:45] Action: Daemon404 showers [19:48] <cbsrobot> nah - philip glass [19:51] <ubitux> yah, totally brainfart :) [20:09] <michaelni> <durandal_1707> and extend libswresample to handle planar so that direct conversion is possible between them (something like how it is done in sws) [20:09] <michaelni> swr supports planar [20:10] <michaelni> since long time [20:10] <durandal_1707> michaelni: there are no encoders [20:36] <ubitux> saste: any idea why lavfi device doesn't call poll_frame()? [20:36] <ubitux> it seems to be troublesome for the asetframesize filter recently proposed [20:54] <durandal_1707> flashsv2 encoder is buggy [20:55] <ohsix> there's a moth in the tubes! [21:00] <Compn> that task failed , according to 2011 soc [21:01] <Compn> encoders are hard [21:01] <Daemon404> encoders are not [21:01] <Daemon404> good ones are [21:02] <Daemon404> or even "not crap" ones [21:02] <Daemon404> i guess [21:02] <Compn> wonder how much diego was able to code [21:02] <Compn> i dont remember him being a programmer [21:07] <vadim__> sorry to interrupt, guys... but, who could take a look at 2 tickets regarding one problem with udp.c, and maybe solve it? :) [21:08] <vadim__> https://ffmpeg.org/trac/ffmpeg/ticket/1017 and https://ffmpeg.org/trac/ffmpeg/ticket/609 [21:17] <Compn> maybe wbs can take a look at that vadim__ :) [21:20] <vadim__> Compn: thanks a lot! :) btw, there is one more ticket regarding this problem on bug tracker nr. 1006. there are some listings, maybe it could be helpfull... [21:25] <iive> Compn: he can take an existing code and rework it into different code. So far I haven' seen him write something new. [22:14] <michaelni> vadim__, how can it be reproduced ? [22:20] <vadim__> michaelni: use any mpegts live stream broadcasted through udp multicast as input... the listings in tickets 1017 and 1006 contains the used cli [22:21] <vadim__> michaelni: is this for help?.. or you need something more specific?.. [22:24] <michaelni> vadim__, ideal would be something that i can just copy and paste and that shows the bug [22:26] <vadim__> michaelni: do you mean the cli command? [22:26] <michaelni> neither of the 2 commands from the 2 tickets work for me, ffmpeg just waits forever [22:27] <michaelni> vadim__, also i suspect the -re is causing the overrun [22:27] <michaelni> like in reading data slower than its received [22:28] <vadim__> michaelni: ok... i can check without -re [22:30] <vadim__> michaelni: same behavior without -re :( [22:31] <durandal_1707> vadim__: i can't reproduce your issue with the command you provided [22:31] <vadim__> michaelni: will be a new listing for any help? [22:31] <durandal_1707> vadim__: not at all [22:32] <vadim__> michaelni: i see.... [22:32] <michaelni> iam sure i can fix it if i can reproduce it but i cant atm [22:33] <vadim__> michaelni: how else could i help?.. sorry, but i can't give you access to my testing machine )) [22:35] <michaelni> vadim__, if you know C you could look at udp.c add a few av_log() to print what happens and why it overflows [22:37] <michaelni> hmm [22:37] <michaelni> maybe your box is too slow to encode at realtime [22:37] <michaelni> and thats why the buffers overflow [22:37] <vadim__> michaelni: if so, why does it hangs? [22:38] <vadim__> michaelni: by the way, do you get any data from your udp source? [22:41] <burek> are there any news about forum? [22:41] <burek> I hadn't check mails for days [22:45] <vadim__> michaelni: sorry, but i'm not a prof in C... could you please insert av_log() in udp.c at desired places and send it to me?.. [22:51] <pasteeater> burek: what forum? [22:51] <beastd> burek: Unfortunately not. I am preparing for CLT currently. so i don't have time for poking at Kyle currently. [22:51] <burek> ffmpeg forum [22:52] <burek> :/ [22:52] <burek> can I help somehow [22:52] <burek> I got the server [22:52] <burek> I know how to work with phpbb [22:53] <burek> can we just install the kyle's copy and build a new one around it [22:53] <beastd> burek: maybe you can ping kyle. or maybe he will say you do the work because he has no time [22:55] <burek> it's taking us 3 months so far to get a single forum up and running :S [22:59] <michaelni> burek, why is kyle doing it and not you ? [23:00] <michaelni> if you have time and kyle not ? [23:01] <michaelni> vadim__, i succeeded to reproduce it, ill try to fix it [23:01] <vadim__> michaelni: thank you a lot! [23:12] <burek> michaelni, I don't know.. I didn't want to interfere, but this is taking really too long :( [23:13] <burek> we could just put up a forum online and give out admin accounts [23:13] <burek> to build it up and sort things on the fly [23:13] <burek> I'm not blaming anyone, it's just taking too long, that's why I'm saying that [23:18] <beastd> burek: agree. imho just build up a new forum from scratch now. the layout is kind of worked out already. so just make a new one and tell each of your user communities to register and join there. [23:19] <pasteeater> whose forum is this? http://ffmpeg.test-lab.ch/ [23:20] <burek> mine [23:20] <beastd> burek: just propose to kyle to do it this way. then you have to decide if Kyle can give you the access you need to his server or we should just use your server and you give him whatever access he needs (i think at least a phpbb admin account). [23:23] <burek> ok, I'll send him a message right away [23:34] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r29a8f6b8d4 10ffmpeg/libavformat/udp.c: [23:34] <CIA-17> ffmpeg: udp: Fix sign of error codes. [23:34] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:35] <beastd> Bye... [23:44] <pasteeater> vadim__: check your bug report [23:46] <vadim__> pasteeater: thx, i'll test it [23:46] <pasteeater> thank michaelni, not me. [00:00] --- Fri Feb 24 2012

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ffmpeg.log.20120223
by burek 24 Feb '12

24 Feb '12

[00:08] <tbws> ty vadim_ i didnt have the partner's listed [00:08] <tbws> :( *sigh* [00:09] <tbws> installed it from a debian package now anyways [00:30] <pasteeater> you don't need partners for libfaac-dev [00:31] <pasteeater> it's in the "multiverse" repository, and installing from a deb package isn't recommended. [00:54] <tbws> pasteeater: then the question of why it wasn't available to me is still up in the air, i have the multiverse reps listed. [00:54] <tbws> All is installed now [00:54] <tbws> just having a problem executing it [00:54] <tbws> i'm trying to use ffmpeg to encode a video for mobile safari [00:54] <pasteeater> you could paste the message you get when you tried to: sudo apt-get install libfaac-dev [00:54] <pasteeater> use pastebin.com [00:54] <tbws> Couldn't find package [00:54] <pasteeater> that's it? [00:54] <tbws> yup [00:54] <pasteeater> what about your sources.list and anything in /etc/apt/sources.list.d/ [00:54] <tbws> all i have are the universe and multiverse reps listed [00:54] <tbws> my soources.list looks like vadim_'s only i've stripped the comments out. [00:54] <pasteeater> did you run "sudo apt-get update" after editing the file? [00:54] <tbws> yep [00:54] <pasteeater> what about /etc/apt/sources.list.d/ [00:54] <vadim_> tbws: there was no errors while updating? [00:54] <tbws> nope [00:54] <vadim_> tbws: may be you have broke some links when stripped the comments? [00:54] <tbws> vadim_ i checked it against yours [00:54] <tbws> they are correct [00:54] <tbws> its just plain freakin wierd [00:54] <tbws> but i'm having an encoding problem at the moment http://pastebin.com/chR96YAh [00:54] <pasteeater> how can we guess what you may have done wrong if you don't show your command as well? [00:55] <tbws> true [00:55] <tbws> "ffmpeg -i 19-2.flv -vcodec libx264 -preset hq -preset ipod640 -b 250k -bt 50k -acodec libfaac -ab 56k -ac 2 -s 480x320 19-2.mp4" [00:55] <pasteeater> there is no preset named hq. there is no preset named ipod640 (you use that via -vpre). [00:57] <tbws> i have installed those presets [00:57] <pasteeater> that's not how presets work [00:57] <pasteeater> you don't need to grab preset files from random locations. they probably won't work. [00:58] <pasteeater> ...libx264 -preset slow -vpre libx264-ipod640 ... [00:59] <tbws> ok cool, thank you very much [00:59] <tbws> that seems to be encoding at the moment [01:05] <tbws> Is there a way to maintain the 16:10 aspect ratio of the video being encoded? [01:06] <PovAddict> is 16:10 the DAR or SAR? [01:07] <tbws> no idea [01:07] <tbws> lets say i wanna crop the video down until the width fits the width that i want the video to be converted to. [01:07] <PovAddict> DAR is the aspect ratio of the video as displayed, SAR is the aspect ratio of the pixels forming the stored image; they aren't necessarily the same [01:08] <PovAddict> erm [01:08] <PovAddict> SAR is the aspect ratio of the stored image, width/height in pixels [01:08] <PovAddict> (if they aren't the same, then the pixels are not square) [01:09] <tbws> ah ok, the DAR in that case [01:09] <pasteeater> tbws: you can use the scale filter to preserve aspect. you give it your desired width or height and it will provide a width or height to preserve aspect. [01:10] <pasteeater> scale=480:-1 for example. [01:10] <pasteeater> or scale="480:trunc(ow/a/2)*2" if -1 returns an odd number which x264 dislikes. [01:11] <tbws> How would you write that into my command string though? "ffmpeg -i 19-2.flv -vcodec libx264 -preset slow -vpre ipod640 -scale="480:trunc(ow/a/2)*2" -b 250k -bt 50k -acodec libfaac -ab 56k -ac 2 -s 480x320 19-2.mp4" ?? [01:12] <PovAddict> it's -vf "scale=480..." [01:12] <PovAddict> vf means video filter [01:13] <tbws> thank you so much guys, you've been really helpfull [01:13] <pasteeater> libx264 ignores -bt [01:13] <PovAddict> what's -bt? [01:15] <pasteeater> "bitrate tolerance" [01:16] Action: PovAddict plays around with the x264 CRF [01:16] <pasteeater> tbws: unless you're trying to get a specific output file size, then you should probably use -crf instead of -b [01:16] <PovAddict> crf seems a bit non-linear... [01:16] <PovAddict> how does it work, roughly? [01:17] <PovAddict> using 40 instead of 20 produces something much much worse than what I'd call "half quality" [01:18] <pasteeater> CRF is an arbitrary constant quality: a ratecontrol mode without a target bitrate [01:19] <pasteeater> it's logarithmic, as opposed to qscale for mpeg-* [01:19] <pasteeater> which is linear [01:20] <pasteeater> +6 is roughly half the bitrate with crf when using 8-bit x264. i'm not familiar with 10-bit so i'm unsure how it changes for that. [01:20] <PovAddict> so 20 has be N times the quality of 21? (where N may very well be < 2) [01:20] <PovAddict> s/be// [01:21] <pasteeater> basically just use the highest value that still gives you an acceptable quality. [01:21] <pasteeater> a good start is 24. [01:22] <PovAddict> I wish I had a good way to compare the quality of original and encoded [01:24] <PovAddict> pasteeater: 23 is the default, so that gives me an idea already [01:24] <PovAddict> pasteeater: my source material is low quality to begin with, but I don't want to make it any worse [01:24] <pasteeater> youll just have to look at it and compare for yourself. [01:25] <PovAddict> re: comparison, I meant I wish there was a way to play them in sync and switch between them with one click :) [01:28] <PovAddict> hmm... starting two players at the same time gave me surprisingly good sync [01:30] <PovAddict> wow, it's encoding faster than realtime [01:30] <PovAddict> faster than mpeg2 too, due to better threading :) [01:41] <PovAddict> pasteeater: trying crf 25, the encoded video looks better than the source because it smoothed out the analog color noise :D [01:43] <pasteeater> what is the original format? [01:44] <PovAddict> 8mm analog tape, which I transcoded into deinterlaced .dv around a year ago [01:46] <nextime> is there any particular option to tell ffmpeg to use neon on omap cpu? [01:46] <pasteeater> PovAddict: i have some of that here. really noisy due to low light. yadif and hqdn3d helped a bit. [01:47] <PovAddict> unfortunately DV is not lossless [01:48] <PovAddict> and removing noise *might* have been easier on the interlaced data [01:48] <PovAddict> but going back to the source tapes isn't an option, I don't have a working device for that anymore :P [01:49] <pasteeater> might find one for $5 at a thrift store. [01:50] <PovAddict> our camera died around 10 years ago, so last year I borrowed one to do these conversions, gave it back already [01:50] <PovAddict> there is also one particular tape that took me several tries to rip [01:51] <PovAddict> after a few minutes the image slowly started to get garbled; somehow I'd get it to work again and copy another piece [01:52] <PovAddict> concatenating the pieces was some work too :) [01:53] <PovAddict> I'm afraid that if I touch that tape again it will suddenly break down into fine rust [03:18] <slackyman> hi [03:19] <slackyman> is there a mode to reverse a strip command? [03:19] <slackyman> is there a way tyo revert stripping? [03:23] <slackyman> anyone? [04:02] <slackyman> is there a way tyo revert stripping? [04:04] <PovAddict> what's stripping? [04:09] <slackyman> strip an executable, a binary file [04:11] <PovAddict> err no [04:11] <PovAddict> strip deletes stuff from inside the executable [04:12] <slackyman> ok [04:12] <PovAddict> you undo it by restoring your pre-strip backup :P [04:12] <slackyman> thanks [04:12] <PovAddict> or compiling the binary again [04:12] <slackyman> I'm "making-install" again :) [04:13] <slackyman> thanks [04:13] <slackyman> bye [04:13] <Ultimatespirit> So this is probably a stupid question? but does ffmpeg include libx264? [04:14] <PovAddict> ffmpeg is distributed as source code, and it doesn't "include" anything that is elsewhere (it's bad practice to embed other libraries since they'd go inevitably out of date) [04:14] <PovAddict> if you mean a ffmpeg binary you got somewhere, that's not something we'd know :) [04:15] <Ultimatespirit> ah, since i'd seen examples of using libx264 as a codec and I was wondering if I can actually use it [04:15] <PovAddict> yes, ffmpeg can use libx264 [04:15] <PovAddict> and many other libraries [04:15] <PovAddict> use != include [04:15] <Ultimatespirit> then I guess i messed something up since it gives me errors when i try it... [04:15] <Ultimatespirit> != include [04:15] <Ultimatespirit> !=include [04:16] <PovAddict> I meant "use isn't the same as include" [04:16] <Ultimatespirit> ah [04:16] <PovAddict> are you compiling ffmpeg from source? [04:16] <Ultimatespirit> yes [04:16] <Ultimatespirit> i compiled it from freedom [04:16] <Ultimatespirit> but can use a git build too if it's something that wasn't in freedom [04:16] <PovAddict> what's "freedom"? [04:17] <Ultimatespirit> 0.10 release [04:17] <PovAddict> ah [04:17] <PovAddict> well [04:17] <PovAddict> install libx264, pass --enable-libx264 to the configure script when compiling ffmpeg :) [04:17] <Ultimatespirit> that gives error saying it isn't found [04:17] <Ultimatespirit> which is why I was worried that I was missing something [04:17] <PovAddict> what linux distro do you have? [04:17] <Ultimatespirit> using mac osx so that's probably a problem? [04:18] <PovAddict> ah okay [04:18] <PovAddict> so you installed libx264 from source as well? [04:18] <Ultimatespirit> ah, so i need to do that? because that gave me entirely different errors that I cannot figure out [04:18] <PovAddict> ffmpeg doesn't include x264 [04:18] <PovAddict> it can use x264 if you have it [04:19] <Ultimatespirit> ah, that explains a lot. Thanks. [04:19] <Ultimatespirit> Now to find out what went wrong wiht libx264 huh... [04:23] <edrx> I am puzzled by this behavior: when I run this to extract the 1-minute [04:23] <edrx> segment between 0:30 and 1:30 from a very heavy .MOV file (20 mins, almost 1GB), [04:23] <edrx> "ffmpeg -i DSC_0004.MOV -ss 0:00:30 -t 0:01:00 /tmp/clip.MOV" [04:23] <edrx> I am getting a low-quality .MOV, with about 3.7MB, instead of a .MOV with the original quality and about 50MB... [04:23] <edrx> the manpage says "By default ffmpeg tries to convert as losslessly as possible: It uses the same audio and video parameters for the outputs as the one specified for the inputs" - so this is a case where the default is _not_ to be lossless. What is happening? Any hints (esp. on how to do lossless extraction)? 8-( [04:23] <PovAddict> edrx: huh, that quote seems crap :P [04:24] <edrx> is there an option to ask for losslessness? =) [04:24] <PovAddict> if you want it to be *really* lossless, use -vcodec copy -acodec copy, but then you can only extract down to nearest keyframe, which may be quite a few seconds away from the time offsets you want [04:25] <PovAddict> otherwise, just increase the video and audio bitrate; the default video bitrate in ffmpeg is ridiculously low [04:27] <edrx> how do I increase it to, hmm, 50MB per minute -> 1MB per second? [04:27] <PovAddict> -b 8000k [04:27] <PovAddict> (8 megabits per second) [04:28] <edrx> trying [04:43] <edrx> PovAddict: excellent, thanks =)! and I've just solved my other problem too, which was to find a way to reduce the framerate to 6 FPS to save more space... I am converting the recordings of a maths talk, so 6 FPS is ok, but the blackboard needs to appear in hi-res [04:44] <edrx> this did the job: "ffmpeg -i DSC_0004.MOV -b 10000k -r 6 -ss 0:00:30 -t 0:01:00 /tmp/clip-high6.avi" [05:26] <tbws> i'm trying to write a bash script that will encode a file for all of the various html5 video formats and i'm having difficulty choosing what the params should be [05:26] <tbws> http://pastebin.com/ [05:26] <tbws> could someone please help me out? [05:26] <tbws> *http://pastebin.com/TYj2qBa9 [05:28] <relaxed> tbws: "ls $1 | while read file; do" is wrong, use --> for i in "$@"; do [05:29] <edrx> I prefer this: "for file in *; do ...; done" [05:30] <edrx> oops, that was practically the same as relaxed's [05:31] <edrx> (I am a newbie, I can say very little about parameters) [05:36] <relaxed> tbws: also, that's a sh script [05:38] <CarlFK> how do I see what the codec names are? [05:39] <zap0> ffmpeg -codecs [05:39] <zap0> or --codecs [05:39] <CarlFK> thanks [06:09] <kevin_> hi all. so i have a video file and i need to extract the video stream to one file and the audio stream to another file and i want the video to have no audio in it. if i use the -an flag, it errors because then the audio stream doesn't exist to put in.. any ideas? command-line https://gist.github.com/1890453 [06:09] <kevin_> using that will cause cut/$file.mpg to have both the audio and video streams, but i only need it to have the video stream [06:13] <relaxed> kevin_: why do you want mpeg2 video and mp3 output? [06:14] <kevin_> relaxed: i'm normalizing the audio using normalize-mp3 [06:15] <kevin_> then rejoining the two streams [06:16] <relaxed> output to wav and use normalize-audio instead [06:16] <relaxed> you won't lose as much quality [06:20] <relaxed> I would encode both streams together (with raw audio), dump the wav, normalize it and the mux the normalized audio and mpeg2 video together. [06:20] <kevin_> hmm [06:22] <relaxed> well, encode the normalized audio to ac3/mp2 while muxing with the video. [06:23] <kevin_> ok yeah [06:23] <relaxed> the day ffmpeg gets a normalization filter will be a wonderful day indeed. [06:27] <kevin_> well, any idea on how to just have the intermediate video file not have the audio stream? seems unnecessary to have ffmpeg to have to encode it to ac3 when it's not gonna be used [06:29] <relaxed> not ac3, use pcm_s16le so there won't be any quailty lose. This will help you maintain sync as well. [06:30] <relaxed> s/lose/loss/ [06:30] <kevin_> which is the raw audio correct? [06:30] <relaxed> yes [06:31] <kevin_> even if the intermediate uses mpeg2video+raw, it still seems unnecessary to have the audio in there... [06:32] <kevin_> ie, why have one mpeg2video+wav, then dump that wav, when i could just have one mpeg2video and one wav...? [06:32] <kevin_> (if i'm explaining that right) [06:34] <relaxed> then use -f mpeg2video and -f wav to separate the two [06:36] <kevin_> ah, ok. great. [11:12] <rainmaker1> Hello I am trying to caputre HD live stream (h264) to disk using ffmpeg -re -i udp://231.0.0.124:5000 -vcodec copy -acodec copy -y testing.mpg [11:13] <rainmaker1> using vlc there is no video just audio [11:13] <rainmaker1> Is h264 capturing supported using ffmpeg? [11:14] <kriegerod> yes [11:14] <rainmaker1> kriegerod: any hint? [11:14] <kriegerod> i didn't understand exactly what's your problem [11:15] <rainmaker1> kriegerod: When I capture live video stream (h264) to disk using ffmpeg -re -i udp://231.0.0.124:5000 -vcodec copy -acodec copy -y testing.mpg command [11:15] <rainmaker1> kriegerod: and when I try to play that file using vlc, there is no video and I can hear audio [11:16] <Mavrik> rainmaker1: provide ffmpeg output in a pastebin as ChanServ notified you when you joined please. [11:17] <kriegerod> makes sense also to publish the output file of duration ~10 seconds [11:17] <Mavrik> he's probably just remuxing it wrong [11:17] <kriegerod> rainmaker1: did you try to playback the file using ffplay? [11:18] <kriegerod> Mavrik: you mean container may be incompatible? Or just VLC support of mpg format has issues? [11:18] <Mavrik> probably incompatible container [11:19] <kriegerod> rainmaker1: try saving with extension .ts, or mp4 [11:21] <rainmaker1> Here is pastebin http://pastebin.com/ewLfTPsv [11:22] <Mavrik> rainmaker1: your input is a MPEG2-TS stream which you're remuxing to MPEG, which could be the cause of your problems [11:22] <Mavrik> try using -f mpegts output.ts for output file [11:22] <Mavrik> also it seems your input is rather lossy :) [11:25] <rainmaker1> Mavrik: When I try to play live stream (not from file) it says it's H264-MPEG-4 AVC(part10)(h264) [11:26] <Mavrik> yeah [11:26] <Mavrik> packaged into a MPEG-2 TS [11:26] <Mavrik> that's standard for IPTV [11:27] <Mavrik> also, "-bufsize" after "-i" is the wrong way to set input buffer size [11:29] <rainmaker1> Mavrik: hmmm nope, still cirucula buffer overun error :( http://pastebin.com/nwcCccn8 [11:30] <rainmaker1> Mavrik: now I have video but everytime I get cirucular buffer overun error [11:30] <Mavrik> ok good [11:31] <Mavrik> that's probably because your datarate is too high [11:31] <Mavrik> try using [11:31] <rainmaker1> Mavrik: you mean I need a faster cpu? [11:31] <Mavrik> udp://231.0.0.1?pkt_size=1316 [11:31] <Mavrik> for input [11:32] <Mavrik> the circular buffer error seems quite common (And no you don't need a faster CPU) [11:33] <rainmaker1> Mavrik: same :( http://pastebin.com/zt0i90uV [11:33] <Mavrik> also try using ?buffer_size=150000 or something like that [11:34] <Mavrik> to see if you still get buffer overruns [11:35] <rainmaker1> nope [11:35] <rainmaker1> same [11:36] <rainmaker1> should I set -bufsize or this is the same as ?buffer_size ? [11:37] <Mavrik> no [11:37] <Mavrik> -bufsize is for the encoder and means something totaly different [11:38] <kriegerod> try udp://231.0.0.1?fifo_size=0buffer_size=10000000 [11:38] <kriegerod> try udp://231.0.0.1?fifo_size=0&buffer_size=10000000 [11:38] <Mavrik> UDP input reads buffer_size tag after URL [11:41] <rainmaker1> so far so good, but bitrate is now about 2000k and was 8000-11000k [11:43] <rainmaker1> I could not imagine it's this hard to make for living from video :) [11:43] <Mavrik> it's harder :P [11:43] <rainmaker1> :D [11:43] <Mavrik> 2Mbit/s is pretty standard bitrate for IPTV H.264 broadcasting [11:43] <Mavrik> if it's SD [11:44] <rainmaker1> no, it's HD [11:44] <Mavrik> we even went as low as 1Mbit [11:45] <rainmaker1> that's exactly what I want to do, but as far as I can't capture sample video to make some benchmarks I want go far away :) [11:53] Action: nextime ha issues with mpeg4 decoding [11:53] <nextime> *has [11:54] <Mavrik> which mpeg4? :) [11:58] <nextime> Mavrik : from an ip cam [11:58] <nextime> http://nopaste.info/88d2a1bda6_nl.html [12:00] <nextime> but, it has those errors only on a omap3 cpu ( with neon ) [12:00] <nextime> on x86 it work [12:00] <nextime> ( same version of ffmpeg and whole things ) [12:01] <rainmaker1> Mavrik: strange, but when I remove -re I have no circular buffers errors... [12:01] <Mavrik> -re? [12:02] <Mavrik> nextime: hmm... mpeg4 decoder tends to spew out those errors when it can't keep up with the input [12:02] <Mavrik> nextime: try specifying a bigger buffer... is your CPU fast enough for that transcode? [12:02] <rainmaker1> Mavrik: yes, -re: Read input at native frame rate. Mainly used to simulate a grab device. [12:02] <nextime> Mavrik : so, do you think it is a performances issue> [12:02] <nextime> ? [12:02] <Mavrik> nextime: it's possible, why aren't you passing speed preset to libx264? [12:03] <Mavrik> -preset ultrafast [12:03] <nextime> Mavrik : i was, this is a "simplified" command line [12:03] <nextime> cause i was trying to understand the bug [12:04] <Mavrik> well, the encoder is complaining about broken input [12:04] <Mavrik> I've had those errors when the buffers were overrun [12:04] <Mavrik> which could be your problem since you're switching to an obviously slower CPU [12:05] <Mavrik> do you get correct input if you just dump it to a file? [12:05] <nextime> Mavrik : it isn't so slow [12:05] <nextime> as it is 1Ghz clock [12:05] <nextime> and also, it work well with more huge streams that aren't mpeg4 [12:06] <Mavrik> 1GHz ARM is slow when it comes to video encoding. [12:06] <Mavrik> especially if you're trying to use x264 [12:06] <nextime> Mavrik : on the x264side it work great with neon [12:07] <nextime> i can encode in realtime from an mjpeg stream that is as double resolution than the mpeg4 one [12:07] <nextime> and also from the same resolution with an mpeg2 stream [12:07] <nextime> with very low cpu load [12:07] <Mavrik> hmmf [12:07] <Mavrik> interesting that [12:07] <Mavrik> there IS a chance that mpeg4 decoder is broken on ARM [12:08] <Mavrik> however I'd check input buffers first [12:09] <nextime> ( and -preset ultrafast doesn't solve ) [12:10] <nextime> what make me think to an mpeg4 decoder issue is than with every other stream than mpeg4 with the very same options on the encoding side [12:10] <nextime> it work well enough [12:10] <Mavrik> mhm [12:10] <Mavrik> nextime: do you have the same transport for all? meaning, could it be a demuxer issue? [12:11] <nextime> Mavrik : yes, it could be [12:11] <nextime> the transport isn't the same [12:12] <nextime> wait, i have the same stream on rtsp over http inside an afs file other than the rtsp i'm trying [12:13] <Mavrik> (I've had rather bad experience concerning FFMpeg's network muxers/demuxers) [12:15] <nextime> uhmm [12:15] <nextime> same stream but with rtsp encapsulated in http and asf, it seems to work [12:17] <Mavrik> bbl, need to install some hardware to my workstation :P [12:17] <nextime> ok no [12:18] <nextime> it *seems* to work at first [12:18] <nextime> but same issue on the second try [14:19] <Qcoder00> Morning [14:19] <Qcoder00> Or Afternoon [14:19] <Qcoder00> :) [14:20] <Qcoder00> Does anyone know how you set up ffmpeg on Windows XP? [14:20] <Qcoder00> I get an error from Windows Media Player saying it can't play DVD's [14:20] <Mavrik> uum [14:20] <Qcoder00> And I am sure this is a missing codec issue [14:20] <Mavrik> ffmpeg isn't what you want probably [14:20] <Qcoder00> What do I need then? [14:21] <Mavrik> try using an alternative player (Media Player Classic Home Cinema, VLC, etc.) or install a decoder (ffdshow is a popular one) [14:22] <Qcoder00> The error I get in Windows Media Player is that it claims there is no 'compatible' decoder [14:22] <Qcoder00> which is silly as I thought XP could play DVD out of box [14:22] <Mavrik> afaik it doesn't [14:22] <Mavrik> you're using an obsolete old OS anyway [14:22] <Mavrik> so grab a player that can do it [14:23] <Qcoder00> VLC works [14:23] <Qcoder00> but it takes a VERY long time to decode stuff [14:24] <Qcoder00> I'll try upgrading VLC and see it that cures the problem [14:25] <Qcoder00> In the meantime where do I ask for ffdshow support? [14:26] <JEEBsv> doom9's thread [14:26] <JEEBsv> clsid and friends do not IRC [14:27] <Mavrik> puny humans. [14:34] <Qcoder00> Is doom9's thread a 'safe' site? [14:34] <JEEBsv> doom9 is one of the oldest DVD ripping/video encoding related sites :P [14:34] <JEEBsv> and doesn't have ads as far as I know [14:34] <Qcoder00> OK [14:35] <Qcoder00> So it's not a site that's going to get busted for showing people how to break the encryption NASA uses on downlinks or anything like that? ;) [14:35] <JEEBsv> they only do video-related DRM research [14:35] <JEEBsv> and it hasn't been busted yet [14:35] <JEEBsv> during the... probably over 10 years of existence it has had? [14:36] <Qcoder00> OK [14:36] <Qcoder00> I have ethical issues with DRM in any case [14:36] <Qcoder00> DRM =Don't Rip me!! [14:37] <Qcoder00> DRM= doom9 reversed me ;) [14:37] <Qcoder00> (Joke) [14:42] Action: nextime hates gstreamer, but it seems that i'll have to use it instead of ffmpeg for what i'm doing... [14:43] <ayaka> does ffmpeg support acceleration in some arm cpu like msm [14:44] <nextime> ayaka : it has "some" support for neon on omap [14:44] <nextime> but i'm going to gstreamer cause it doesn't support the TI DSP [14:46] <ayaka> nextime, but i think i can't run a x with acceleration in it [14:47] <ayaka> nextime only omap why neon is obly in it [14:47] <nextime> ayaka : of course no, you need accelleration support in the X driver [14:48] <ayaka> nextime, do you have sone info about that [14:48] <nextime> ayaka : i don't know if it is only omap, i just know omap cause i'm using it [14:49] <nextime> ayaka : just what you can find on google about linux on beagleboard and similar things [14:50] <ayaka> nextime, it means you have install a linux on mobile, do you have time and offtopic channel, i want to ask something [14:57] <nextime> ayaka : i just have some beagleboards and similar boards with omap and my lovely n900 :P [15:05] <ayaka> nextime, sorry yaaic shock down, mine is htc evo 3d, i'm installing X for it in chroot, do you know fastboot? [15:35] <nextime> uhmm [15:36] Action: nextime has found some posts in the developement ml of ffmpeg about the ti dspbridge support [15:58] <julienT_> hello, i'm just doing to make ./configure on the ffmpeg project but nothing [15:59] <julienT_> how install the git project on my pc? [15:59] <julienT_> not "./configure" "./make install" ? [16:06] <drno_> julienT_: ./make? [16:06] <drno_> looks like a path issue. [16:07] <julienT_> oh yes i forgot ./make [16:07] <julienT_> ty :p [16:11] <julienT_> return error with make [16:11] <julienT_> cmdutils.c line 48 [16:11] <julienT_> print_program_info [16:11] <drno_> what version of ffmpeg are you trying to compile? [16:12] <julienT_> last [16:12] <julienT_> git clone [16:12] <julienT_> from git://source.ffmpeg.org/ffmpeg.git [16:13] <andrees> hi [16:13] <julienT_> hi [16:14] <andrees> j^ [16:14] <andrees> julienT_ [16:14] <julienT_> drno_: does it correct git? [16:14] <andrees> do you know how to encode avi file to flv with vbr using 2 pass? [16:15] <andrees> do you know how to encode avi file to flv with vbr using 2 pass? in windows ffmpeg :P [16:15] <julienT_> no i'm trying to compile ffmpeg but i don't [16:15] <julienT_> you do? [16:15] <andrees> ok [16:15] <andrees> and in linux bulid? [16:15] <julienT_> no windows [16:16] <andrees> ok [16:16] <andrees> i want to learn how to encode .avi or any other file to .flv or .mp4 (avc,aac) with vbr using two pass [16:17] <andrees> but i still dont understand how two pass works :S [16:20] <andrees> if someone wants to help me, i'd be super gretful :) [16:22] <julienT_> how build FFmpeg on windows plz [16:22] <andrees> so julienT_, you need help with something? [16:23] <julienT_> yes [16:23] <andrees> can you tell me? i may help you [16:23] <julienT_> how build on windows [16:23] <andrees> you men, you have the source code, and you want to build it to use in windows? [16:24] <julienT_> yes [16:24] <andrees> mmm, i have no idea [16:24] <andrees> lol [16:24] <julienT_> and there is some errors [16:24] <andrees> why dont you just donwload an already buit windows version? [16:27] <andrees> barhom_ hi [16:46] <julienT_> in which file we can found the main plz? [16:50] <andrees> hi scoobertron codewrangler [16:51] <andrees> i want to learn how to encode .avi or any other file to .flv or .mp4 (avc,aac) with vbr using two pass [16:54] <codewrangler> Andrees: I'm just a beginner using ffmpeg& Lately I've been using iVI on my Mac for video conversion lately. [16:54] <andrees> ok [16:54] <andrees> lol [16:55] <andrees> so codewrangler you need help with anything? i may help [16:55] <codewrangler> andrees: Just my 2011 taxes is all. ;-) [16:55] <codewrangler> I'm pretty set at the moment for video, but thanks so much for asking. [17:15] <andrees> hi [17:15] <andrees> is there anyone that can help me? [17:15] <andrees> i want to learn how to encode .avi or any other file to .flv or .mp4 (avc,aac) with vbr using two pass [17:28] Last message repeated 1 time(s). [18:10] <Alasdairr> Hi all [18:11] <Alasdairr> Trying to build ffmpeg 0.10, getting a gcc error about redeclaration of enumerator 'VAR_SAR' - this is with gcc 4.4.5 [18:11] <Alasdairr> There are indeed two occurances of VAR_SAR in the var_name enum [18:12] <lkiesow> Hi everyone, I have a problem with ffmpeg 0.10 [18:12] <lkiesow> After calling "ffmpeg -i INPUT -acodec aac -vcodec libx264 OUTPUT.mp4" I get the error messages "[NULL @ 0x1dc8da0] Codec is experimental but experimental codecs are not enabled, see -strict -2" and "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height" [18:12] <lkiesow> Adding "-strict experimental" or "-strict -2" changes nothing [18:13] <lkiesow> On another system with ffmpeg version 0.8.9 it worked just fine [18:13] <lkiesow> Has anyone an idea what might be wrong? [18:17] <relaxed> lkiesow: what does this return? ffmpeg -codecs 2>&1| grep ^..E.*aac* [18:20] <lkiesow> "ffmpeg -codecs 2>&1| grep ^..E.*aac*" returns nothing, but "ffmpeg -codecs 2>&1| grep aac" will give you "DEA D aac Advanced Audio Coding" and " D A D aac_latm AAC LATM (Advanced Audio Codec LATM syntax)" [18:21] <lkiesow> relaxed: Thus the codec should be there if I'm not mistaken [18:22] <relaxed> lkiesow: yes, I was hoping you had a better aac encoder to use [18:22] <relaxed> like libvo-aacenc [18:23] <lkiesow> I build it myself I can build it with other encoders. Bt as I said: It worked with the old ffmpeg& [18:24] <relaxed> I know but you would be better off using libvo-aacenc anyway [18:25] <lkiesow> Nevertheless: Which aac encoder would you recommand? faac, libvo-aacenc (are there more) [18:25] <lkiesow> Why? :-) [18:27] <teratorn> are there any established image file formats that store uncompressed or compress YUV data? [18:27] <relaxed> ffmpeg's aac encoder still needs work, libfaac is non-free and unmaintained, so that leaves libvo-aacenc [18:29] <lkiesow> relaxed: ok, I try it. It vannot get worse anyway ^^ [18:30] <relaxed> lkiesow: I use it and have been happy with it. [18:31] <relaxed> teratorn: using the output "%03d.Y" will output 001.Y, 001.U and 001.V if that's what you're after. [18:32] <teratorn> relaxed: ah no, I need to produce said files in code [18:32] <teratorn> such that some kind of image viewer could view them [18:33] <relaxed> are you talking lossless? [18:40] <andrees> i want to learn how to encode .avi or any other file to .flv or .mp4 (avc,aac) with vbr using two pass [18:40] <andrees> is there anyone that can help me? [19:08] Action: nextime cry and want to have tidspbridge support in ffmpeg [19:08] <jamiel> Hi all, is it possible to apply an avfilter at specific time intervals within the stream rather than have it applied all the time? [19:08] <jamiel> Can't find any documentation on this [19:12] <relaxed> jamiel: look at the movie filter in the man page [19:16] <relaxed> I'm not sure if it gives you the control you're looking for. [19:20] <jamiel> Hmm, it looks like I can only specify a single time interval ... I'm using the overlay filter and I would like it to appear every X number of seconds [19:20] <jamiel> Anyone know a better way to achieve this if there is no way to make the filter repeat itself? [19:46] <jamiel> I see there is a select filter ... can I combine the select filter and color filter ... so apply the color filter to the bits of the stream that match the select? [20:27] <Mattias> I have a headset, and I want to stream, with ffserver/ffmpeg, what I hear. It's an usb soundcard and it only has "Speaker" and "Capture" which can be seen in alsamixer. How do I use "Speaker" as audio input? is that possible? [20:37] <pasteeater> burek probably can figure it out if he's around. i'm alsaignorant. [21:04] <o_0oo> hi [21:04] <jamiel> hi o_0oo , how can we help? [21:04] <vadim__> hi 2 all [21:06] <o_0oo> well, i was wondering - i have setup a filtergraph which splits and in one of the split chains i do a select - when i overlay it back with the original, the whole in->out chain seems to be trimmed to what's in the select. I would have expected the in->out chain to be unaffected, and only the selected part of the second chain would be overlaid [21:08] <o_0oo> weird, huh? [21:08] <jamiel> hmmm [21:09] <jamiel> what's your -av line ? [21:10] <jamiel> I mean -vf [21:11] <o_0oo> essentially, it is [in] split [nobox], fifo, [withbox] overlay [out]; [nobox] fifo, drawbox, select [withbox] [21:11] <o_0oo> the select gets just seconds 10-20 [21:11] <o_0oo> so i would have thought that the drawbox would be overlay'd for just seconds 10-20 [21:12] <o_0oo> but, it doesnt do tht [21:16] <o_0oo> what i want to achieve is to get a colour'd overlay on top of an rtmp at specified intervals (e.g., every 20 seconds). [21:22] <o_0oo> if anyone could offer any advice on how to do that, i'll give them a hug [21:25] <andrees> i want to learn how to encode .avi or any other file to .flv or .mp4 (avc,aac) with vbr using two pass [21:25] <andrees> is there anyone that can help me? [21:28] <JEEB> are you trying to get the exact same file size for same length, or do you just want to do good quality video? [21:48] <andrees> jeeb are you asking me? [21:48] <JEEB> whom else? [21:48] <andrees> lol [21:48] <andrees> yes [21:49] <JEEB> so you would rather have exactly the same file sizes compared to stable quality? [21:49] <andrees> i want to convert an avi file to flv, that keep all the quality but the less size posible [21:49] <JEEB> ...which means you want some certain quality .-. [21:49] <JEEB> and not exactly the same file size [21:49] <andrees> yes [21:50] <andrees> the same quality than the avi, but the less size posible [21:50] <andrees> i thought that with vbr the file size could be less [21:50] <JEEB> it's pretty hard to get cbr with libx264 :P [21:50] <andrees> ok, i want to vbr [21:50] <JEEB> (possible but definitely not what you'd want in 99% of all cases) [21:51] <JEEB> anyways, the same algorithm is used for two vbr rate control modes in libx264 [21:51] <andrees> is it posible to encode to vbr flv? [21:51] <andrees> ok im listening [21:51] <JEEB> crf is basically "quality level" [21:51] <JEEB> and then there's a bitrate mode for "I want this exact size/average bit rate" [21:52] <andrees> ok [21:52] <JEEB> for bitrate mode you need two passes for best quality, but for crf you only need one because you aren't really limiting the encoder in a similar way [21:52] <andrees> ok i understand [21:53] <andrees> but [21:53] <JEEB> ffmpeg -i input.derp -vcodec libx264 -preset:v slowest_libx264_preset_you_can_take -crf highest_crf_that_still_looks_good_to_you -acodec libfaac (out.flv or out.mp4) [21:53] <JEEB> ^ this is what you want [21:54] <andrees> with this, will a have a variable bitrate? [21:54] <JEEB> yes [21:54] <andrees> o a constant bit rate? [21:54] <JEEB> ... [21:54] <JEEB> <JEEB> it's pretty hard to get cbr with libx264 :P [21:54] <andrees> ok [21:54] <andrees> good [21:54] <JEEB> you'd have to actually work harder to get CBR out of x264 :P [21:54] <andrees> its ok [21:54] <andrees> i dont want cbr [21:54] <JEEB> because it's certainly NOT RECOMMENDED and NOT WANTED for most cases [21:54] <andrees> i want the variable bit rate [21:55] <pasteeater> just do what JEEB said and you'll get what you want. [21:55] <andrees> yes i will [21:55] <andrees> but first [21:55] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [21:55] <JEEB> ^ list of presets [21:55] <pasteeater> also add '-q:a 100' for audio VBR when using libfaac [21:56] <andrees> why do i need a slow preset? [21:56] <JEEB> you don't need a slow preset [21:56] <andrees> i mean i have this ones [21:56] <JEEB> medium is default [21:56] <JEEB> and then if you want it to compress less and be faster you go faster [21:56] <JEEB> and if you want it to compress more and be slower you go slower [21:56] <JEEB> which is why you use the slowest preset YOU CAN TAKE [21:56] <andrees> ok [21:56] <andrees> i understand [21:56] <andrees> so can i use this preset?: libx264-slow_firstpass.ffpreset [21:57] <JEEB> ... [21:57] <JEEB> use the libx264's internal presets [21:57] <JEEB> I just listed them [21:57] <JEEB> also start with crf around 24 or so [21:57] <JEEB> if it looks good, raise it [21:57] <JEEB> if it looks bad, lower it [21:57] <JEEB> also [21:58] <JEEB> you most probably don't want to use the fastest and slowest presets in libx264 [21:58] <JEEB> because those are mostly for testing [21:58] <JEEB> placebo is lolslow [21:58] <JEEB> and ultrafast sacrifices everything to be fast [22:02] <andrees> ok [22:02] <andrees> it is ocnverting now [22:02] <andrees> it is converting now [22:02] <andrees> another question, when using this :ffmpeg -i input.derp -vcodec libx264 -preset:v slowest_libx264_preset_you_can_take -crf highest_crf_that_still_looks_good_to_you -acodec libfaac (out.flv or out.mp4) [22:02] <andrees> is one pass or two pass? [22:03] <JEEB> one pass [22:03] <JEEB> crf only needs one [22:03] <JEEB> ...which, btw, I already said [22:10] <pasteeater> andrees: you forgot '-q:a 100'. [22:11] <pasteeater> ...but default at least isn't bad. [22:11] <pasteeater> so either way... [22:12] <andrees> it keep giving me this error [22:12] <andrees> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [22:15] <andrees> pasteeater no i dont forgot it, i include it in the cmd, but it give me arror [22:15] <andrees> i believe its because im using -acodec aac insted of libaac [22:15] <andrees> but i dont have libaac library [22:16] <andrees> libfaac [22:18] <andrees> JEEB pasteeater are you there? [22:19] <JEEB> -acodec aac is the ffaac encoder (ff* are internal ones), which is still in experimental status [22:19] <JEEB> are you sure it doesn't tell you to add the experimental enabling setting before that final error? [22:20] <andrees> it doesnt askme to write -strict experimental [22:20] <andrees> but i just try and it works thanks [22:21] <andrees> i will try con encode now and tell how it gon [22:31] <andrees> jeeb [22:31] <Mattias> JEEB: ! [22:31] <Mattias> You are everywhere :) [22:31] <andrees> at first look it seems it work good [22:32] <andrees> so you said, lower cfr will result in more quality and higer cfr in worse quality, is that correct? [22:32] <andrees> mattias do you speak spanish? [22:33] <Mattias> andrees: I'm Swedish [22:33] <Mattias> I only speak Japanese [22:33] <andrees> ok [22:33] <andrees> i asked because of your name [22:33] <andrees> it is a very common name here in argentina [22:34] <andrees> but with one t only [22:34] <Mattias> Very common in Sweden with 2x t's or th combo [22:35] <Mattias> ok, so, I'm trying to stream with ffserver + ffmpeg, when I run ffmpeg I get this: http://pastebin.com/aqdFAgTy broken ffmpeg something -.- Did I forget a conf file somewhere? [22:35] <JEEB> <andrees> so you said, lower cfr will result in more quality and higer cfr in worse quality, is that correct? <- yes, thus the thing we're aiming is the highest crf value that looks good for you [22:36] <Mattias> I have x264 library installed and such -.- [22:36] <JEEB> which is why I said, if it looks good -> go higher [22:36] <andrees> yes, thanks a lot :) [22:37] <Mattias> JEEB: any experience with ffserver + ffmpeg? well, I guess this error is more related to ffmpeg alone [22:37] <JEEB> ffserver is someething deeply generally unknown and unused [22:38] <JEEB> arcane black magic [22:38] <Mattias> Got any other ideas on how to stream on linux? [22:38] <Mattias> to pages like twitch.tv etc [22:38] <Mattias> I want to stream me playing dwarf fortress :P [22:38] <JEEB> I'd say the most luser-usable thing for that [22:39] <JEEB> is VLC [22:39] <burek> Mattias, if your audio card doesn't support recording of "what you hear" then try with jack audio [22:39] <o_0oo> does anyone know anything about filter graphs? [22:39] <burek> o_0oo, yes, what do you need [22:40] <Mattias> burek: yeah, shitty usbsoundcard, got nothing but a "Speaker" and "Capture" for mic.. I'll take a look at jack audio after I get at least video running [22:40] <burek> use -f x11grab if you are on linux [22:41] <Mattias> burek: http://pastebin.com/aqdFAgTy <-- I get this error [22:41] <Mattias> [libx264 @ 0x205f400] broken ffmpeg default settings detected -.- [22:41] <o_0oo> burek, i need to overlay a colour source onto a stream, but with a select applied to the chain with the colour, so that it's only displayed for the period that the select is valid for [22:42] <burek> broken ffmpeg default settings detected [22:42] <burek> that means your ffmpeg is old [22:42] <Mattias> eeh [22:42] <burek> o_0oo, I see.. let me think [22:42] <o_0oo> thanks dude [22:43] <Mattias> burek: 20120127-3 <-- this is the version I'm using, how old can that be? [22:43] <Mattias> I'll try dev version [22:43] <burek> Mattias, it doesn't matter when has it been compiled, it matters from when the source code is [22:44] <Mattias> getting git versions [22:45] <burek> o_0oo, first read this please: http://ffmpeg.org/ffmpeg.html#Filtergraph-syntax [22:45] <Mattias> both for x264 and ffmpeg [22:45] <burek> so I can explain it to you more easy [22:46] <o_0oo> ok [22:46] <burek> o_0oo, also this: http://ffmpeg.org/libavfilter.html#Tutorial [22:46] <burek> it's not too much, but it'll explain the basics [22:46] <o_0oo> i've read that one :-) [22:46] <burek> Mattias, that's the smartest idea :) [22:47] <burek> o_0oo, oh, so you are familiar with chains [22:47] <o_0oo> to an extent, yes [22:49] <andrees> jeeb another question, is it posible to encode using variable frame rate? and if it is, does it reduce file size? [22:50] <o_0oo> ive tried a number of approaches, firstly by creating a colour source, applying the select and then overlaying with the original - -vf "color=#8EABB8@0.9:480x208,select='gte(t,10)*lte(t,20)' [over]; [in][over] overlay [out]" [22:50] <burek> o_0oo, so, use the split, then in one branch, use just the fifo and overlay, just like in that libavfilter example image [22:50] <burek> in another branch use select [22:51] <o_0oo> i tried that too - but it seemed like the select was applied to the primary chain [22:51] <burek> if I get you correctly, you need the normal video and overlayed video only when select allows it [22:51] <o_0oo> or, to both chains [22:51] <o_0oo> burek - exactly [22:51] <burek> ill paint you a diagram :) [22:51] <o_0oo> :) [22:54] <andrees> pasteeater, when using -q:a 100, how do i change quality, what happen if i use less than 100 or grater than 100? [22:54] <burek> o_0oo, http://pastebin.com/3DRedLJf [22:55] <o_0oo> awesome, hold on [22:56] <Mattias> burek: latest git and x264 git, exact same error [22:57] <burek> o_0oo, this is a more complete version: http://pastebin.com/FE6tyFuP [22:59] <o_0oo> ok, cool, this is what I have - it looks the same, but the stream seems to freeze after 10 seconds - http://pastebin.com/bznGMUCF [23:00] <Mattias> burek: ah, just found on google... apparently x264 thinks ffmpegs encoding settings are broken and refuses to run just like that O.o x264 and ffmpeg aren't friends? [23:07] <pasteeater> andrees: see 'man faac' [23:08] <o_0oo> well, to be clear, it doesn't freeze, the output is just the 10 seconds of the select [23:09] <pasteeater> Mattias: ffmpeg and x264 work fine together. you just have to get thr right syntax for your version(s). [23:11] <Mattias> I noticed x264 wouldn't run on git version, reverted back to stable arch linux package [23:12] <Mattias> hm, I wonder what I'm missing -.- [23:12] <pasteeater> ioni does a good job at keeping ffmpeg updated. probably better than any other sane distro. [23:13] <burek> Mattias, -vpre is deprecated [23:14] <burek> ffmpeg now uses native x264 profiles/presets/tunes [23:14] <Mattias> can't find any presets for x264 either :/ [23:14] <burek> using -preset -profile -tune [23:14] <burek> -vpre used some sort of files, containing a lot of x264 specific options [23:14] <burek> that's now deprecated, that's why you get that error [23:14] <Mattias> but I haven't used -vpre, it just tells me to use it [23:15] <relaxed> jamiel: output all the frames and process them based on their sequence number(s) [23:15] <pasteeater> i'd assume 37208-g01fcbdf is new enough to use -preset instead of -vre [23:15] <pasteeater> not sure why is suggests vpre [23:15] <Mattias> ah, tried profile, not preset... -preset ultrafast works [23:16] <Mattias> getting a lot of VBV underflows though [23:23] <o_0oo> burek, any further ideas? [23:23] <burek> just a sec please [23:23] <burek> im on the phone [23:29] <rainmaker1> Does anyone know why what's the purpose of pkt_size? Yes, I know set the size of udp packets in bytes, but is only because of network MTU or there is something else? [23:29] <rainmaker1> me and my fat fingers, sorry for typos :) [23:29] <pasteeater> you need a dialing wand. [23:30] <rainmaker1> :D:D:D [23:32] <pasteeater> rainmaker1: did you see 'man ffmpeg'? Under PROTOCOLS there is a udp section. [23:34] <o_0oo> i think filtergraphs limit the output to the length of the shortest chain [23:35] <o_0oo> so if one chain has a select, that will determine the length of the whole output [23:35] <Mattias> woho, I streamed :D [23:35] <Mattias> got it to work :) [23:35] <nextime> lucky you. [23:37] <Mattias> just followed this guide and changed vpre to preset = http://michaelchinen.com/2011/06/27/justin-tv-streaming-in-linux/ ^.^ [23:37] <Mattias> works like a charm ^.^ [23:38] <rainmaker1> pasteeater: yes, but I don't understand the point? I know what pkt_size option do, but I dont know WHEN I should use it... I guess Ill need a little help from tcpdump [23:40] <burek> im here [23:40] <burek> sorry for keeping you waiting [23:40] <burek> o_0oo, well, you need to tell select [23:41] <burek> to pass empty frames or something, just to continue the output [23:41] <burek> im not sure, but i think it stops because it waits for the 2nd branch to get the input from [23:43] <nextime> Mattias : i insist, lucky you that you can solve your issue with some doc [23:43] <nextime> my options is to write code or to use gst instead of ffmpeg ( and i hate gst and love ffmpeg ) [23:44] Action: nextime sad [23:45] <burek> nextime, what's your issue [23:46] <burek> rainmaker1, the packet size is usually prioritized on routers [23:46] <nextime> burek : the lack of support for tidspbridge on ffmpe [23:46] <nextime> ffmpeg [23:46] <burek> the smaller the packet, the bigger priority will be [23:46] <burek> it's just a tweaking option, the stream should work even without touching it [23:46] <nextime> there are some patches in some ml posts of 2 years ago [23:46] <burek> what is tidspbridge :) [23:46] <nextime> and nothing more apparently [23:47] <nextime> burek : the support for encoding and decoding using the texas instruments DSP that exists in a lot of cortex a8 and a9 based soc [23:47] <rainmaker1> burek: I am not quite sure, the packet is prioritized based on CoS/QoS values not the size AFAIK [23:47] <pasteeater> nextime: we are interested in orphaned patches. link? [23:48] <nextime> pasteeater i found some about that here [23:48] <nextime> http://ffmpeg.org/pipermail/ffmpeg-devel/2010-September/100622.html [23:49] <nextime> pasteeater : at least in debian from debian-multimedia they appears to be not present [23:49] <nextime> i'm looking at the latest sources if the support exists [00:00] --- Fri Feb 24 2012

1 0

ffmpeg-devel.log.20120222
by burek 23 Feb '12

23 Feb '12

[00:09] <overflow_0f8b> hi [00:10] <overflow_0f8b> amiga is reloaded ! http://a-eon.com/x1000.html http://en.wikipedia.org/wiki/AmigaOne_X1000 [00:10] <overflow_0f8b> 2010 summer with programmable logic coprocessor [00:25] <buzz_> overflow_0f8b, overpriced, underpowered. [00:26] <overflow_0f8b> yeah overpriced, but programmable logic coprocessor ownz all instruction sets [00:26] <buzz_> not really [00:27] <buzz_> dave haynie doesnt think its much use.. some extra speed on the main cpu would be more useful [00:27] <overflow_0f8b> buzz_<< your CPU is just a large macro of logic gates. [00:28] <buzz_> are you talking about the xcore ? [00:29] <buzz_> because that isnt a programmable logic coprocessor [00:29] <buzz_> also, what has any of this got to do with ffmpeg ? :) [00:30] <buzz_> well. it is a coprocessor with an instruction set - but i mean it isnt like some reprogrammable device like an fpga [00:32] <buzz_> (afaik). anyway. david haynie (who knows more about hardware than either of us im sure). didn't think it would be much use [00:34] <buzz_> also see: http://www.osnews.com/thread?431085 [00:34] <overflow_0f8b> well i was just thinking about the possibility of having a programmable logic coprocessor, don't know what the Xena actually does yet, i just read the page that advertised it as programmable logic coprocesor [00:35] <buzz_> read the link i just gave then. [00:35] <overflow_0f8b> it'd be cool to program some FPGA like thing for any signal processing function, then do it at 500MHz rate for example [00:35] <ohsix> how do you schedule an fpga [00:35] <ohsix> the problem is not so much its presence but it's use [00:35] <overflow_0f8b> the program with FPGA's is, they need to boot :( [00:36] <overflow_0f8b> unless they make them on the fly modifyable... [00:36] <ohsix> they are on the fly modifiable, but the dev tools don't really support it, you need to do it by hand [00:36] <overflow_0f8b> wow srsly? [00:37] <ohsix> if you have 2 blocks, you need to test either of them, then both; if you have 16 blocks ... [00:39] <overflow_0f8b> can you point out some low level xilinx hacking resources? [00:40] <overflow_0f8b> i'd like to build an advanced system that is self modifying [00:41] <buzz_> fpga arcade .. minimig [00:42] <buzz_> minimig is open so you can get some info from it [00:43] <buzz_> http://code.google.com/p/minimig/ [01:01] <overflow_0f8b> burek<< wow thanks that looks a great start [01:01] <overflow_0f8b> i'm gonna build a mini amiga [03:01] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r3ab9a2a557 10ffmpeg/libavcodec/ (7 files in 3 dirs): [03:01] <CIA-17> ffmpeg: rv34: change most "int stride" into "ptrdiff_t stride". [03:01] <CIA-17> ffmpeg: This prevents having to sign-extend on 64-bit systems with 32-bit ints, [03:01] <CIA-17> ffmpeg: such as x86-64. Also fixes crashes on systems where we don't do it and [03:01] <CIA-17> ffmpeg: arguments are not in registers, such as Win64 for all weight functions. [03:01] <CIA-17> ffmpeg: 03Martin Storsjö 07master * ra5c50913a8 10ffmpeg/libavformat/mov.c: (log message trimmed) [03:01] <CIA-17> ffmpeg: movdec: Adjust keyframe flagging in fragmented files [03:01] <CIA-17> ffmpeg: For video, mark the first sample in a trun which doesn't have the [03:01] <CIA-17> ffmpeg: sample-is-non-sync-sample flag set as a keyframe. [03:01] <CIA-17> ffmpeg: In particular, the "sample does not depend on other samples" flag [03:02] <CIA-17> ffmpeg: isn't enough to make it a keyframe, since later frames still can [03:02] <CIA-17> ffmpeg: reference frames prior to that one (the flag only says that that [03:02] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r08630d25a2 10ffmpeg/libavcodec/ (pngdec.c pngenc.c): [03:02] <CIA-17> ffmpeg: png: make .long_name more descriptive [03:02] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [03:02] <CIA-17> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [03:02] <CIA-17> ffmpeg: 03Aneesh Dogra 07master * r33510e86b1 10ffmpeg/libavformat/gif.c: [03:02] <CIA-17> ffmpeg: gif: K&R formatting cosmetics [03:02] <CIA-17> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [03:02] <CIA-17> ffmpeg: 03Paul B Mahol 07master * rca085e667b 10ffmpeg/libavcodec/mlpdec.c: [03:02] <CIA-17> ffmpeg: mlpdec: use av_log_ask_for_sample() [03:02] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [03:02] <CIA-17> ffmpeg: Signed-off-by: Janne Grunau <janne-libav(a)jannau.net> [03:02] <CIA-17> ffmpeg: 03Janne Grunau 07master * re48a70e6da 10ffmpeg/avplay.c: [03:02] <CIA-17> ffmpeg: avplay: fix -threads option [03:02] <CIA-17> ffmpeg: The AVOptions based default to threads auto in 2473a45c8 [03:02] <CIA-17> ffmpeg: works only if avplay does not use custom option handling [03:03] <CIA-17> ffmpeg: for -threads. [03:03] <CIA-17> ffmpeg: CC: <libav-stable(a)libav.org> [03:03] <CIA-17> ffmpeg: 03Alex Converse 07master * rb0f29db5c2 10ffmpeg/ (18 files in 5 dirs): Mark mutable static data const where appropriate. [03:03] <CIA-17> ffmpeg: 03Diego Biurrun 07master * r24d150b773 10ffmpeg/LICENSE: Clarify licensing information about files borrowed from libjpeg. [03:03] <CIA-17> (38 lines omitted) [03:58] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8a08503b78 10ffmpeg/libavcodec/pngdec.c: [03:58] <CIA-17> ffmpeg: pngdec: Print error if signature is wrong. [03:58] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:58] <CIA-17> ffmpeg: 03Paul B Mahol 07master * rd00bc6a8fd 10ffmpeg/libavcodec/ (r210enc.c y41penc.c): [03:58] <CIA-17> ffmpeg: r210enc and y41penc: don't set same pkt->size twice [03:58] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [03:58] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:38] <durandal_1707> owner of x86_64-linux-gnu-gcc-4.6 N-38118-gd00bc6a should finally rsync fate samples [04:53] Action: michaelni thinks bcoudurier is owner of x86_64-linux-gnu-gcc-4.6, but iam not 100% sure [08:34] <Zeranoe> Could anyone explain to me the package stagefright-h264? Should it be included in the Windows builds? Is it just a different h264 decoder? [08:43] <Zeranoe> Is it only really helpful on Android? That seems to be the main search result. [08:45] <ohsix> it's from android [11:02] <CIA-17> ffmpeg: 03Nicolas George 07master * r4217dfe87b 10ffmpeg/libavfilter/af_pan.c: [11:02] <CIA-17> ffmpeg: af_pan: remove dual double/int storage of gain. [11:02] <CIA-17> ffmpeg: libswresample takes care of that now. [11:47] <CIA-17> ffmpeg: 03Nicolas George 07master * r9cbf17e9af 10ffmpeg/ (4 files in 2 dirs): lavc: introduce av_get_pcm_codec. [11:47] <CIA-17> ffmpeg: 03Nicolas George 07master * r534a82a3fd 10ffmpeg/libavdevice/lavfi.c: lavd/lavfi: support all sample formats. [11:57] <ubitux> michaelni: any reason tests/copycooker.sh is not used? [11:57] <ubitux> it seems we don't have any copy tests [11:58] <ubitux> mmh my bad, there are actually a few [13:11] <kriegerod> is it possible to pull from audio filtergraph the needed number of samples, but not that many as there was in pushed AVFrame? [13:11] <kriegerod> oh, sorry, wrong chat, that's for #ffmpeg [13:11] <ubitux> i was talking about something like that recently with saste [13:12] <ubitux> kriegerod: no it's fine here :) [13:12] <kriegerod> so what was the result? [13:12] <ubitux> actually, it would be nice to have an audio filter to act as buffering [13:12] <ubitux> something like adding fixedbuffer=512 at the end of the filtergraph [13:13] <kriegerod> ubitux, i could implement it if you direct me [13:13] <kriegerod> seems to me like just creating AVFifo inside the filter instance [13:14] <ubitux> yes, and you might also need to pad with silence at the end [13:14] <kriegerod> i think better name is 'setframesize' [13:14] <kriegerod> well, "asetframesize" [13:15] <ubitux> it was just a random guess [13:15] <ubitux> this could be discussed on ffmpeg-devel, feel free to do it [13:16] <ubitux> this would have various usages btw [13:16] <ubitux> i was thinking of auto inserting such filter if some filters require a fixed size [13:16] <ubitux> (for example fft filters) [13:16] <ubitux> also, it could simplifies the usage of the API [13:17] <kriegerod> yep [13:17] <kriegerod> i thought about such filter to avoid avfifo management inside app [13:17] <kriegerod> could you advise to start from another audiofilter as a template? [13:17] <ubitux> i encourage you to discuss this on ffmpeg-devel ml, i'm sure Nicolas and saste would like to comment [13:17] <ubitux> mmmh [13:18] <ubitux> lemme see [13:19] <ubitux> af amerge changes the number of samples, so you might want to have a look [13:19] <ubitux> but it's a complicated filter [13:19] <ubitux> also, af volume is a nice simple filter which deal with various sample format [13:26] <ubitux> you will need to implement request_frame() btw (to tell if the current samples window is complete, and thus if there is data to pull) [13:27] <ubitux> i mean poll_frame sorry [13:28] <ubitux> i don't think any audio filter use it atm btw [13:28] <ubitux> actually, abuffer does [14:53] <kriegerod> is that right that for audio, AVFilterBufferRef.format holds value from enum AVSampleFormat? [14:54] <ubitux> iirc, yes [15:21] <ubitux> heh, that filter could also fix a bug i have in my WIP -af [15:22] <ubitux> we could use it to request the output stream frame size if codec has not a variable frame size cap [15:40] <kriegerod> in avfilter.h, in comment on poll_frame, there's noted "Output video pads only". Is that correct? [15:40] <kriegerod> also on request_frame [15:40] <ubitux> since they're in use in audio filters, i guess it's not accurate anymore [15:41] <ubitux> a leftover from where there was only video support in lavfi i guess [15:43] <CIA-17> ffmpeg: 03Nicolas George 07master * rf16c950f32 10ffmpeg/libavutil/common.h: lavu: document av_reverse. [15:55] <ubitux> can't we do something about the assert warnings on openbsd? [15:56] <ubitux> and maybe the external declarations with icc [15:58] <durandal_1707> ubitux: what exact warnings? [15:59] <ubitux> http://fate.ffmpeg.org/log.cgi?time=20120222094717&log=compile&slot=x86_32-… [15:59] <ubitux> http://fate.ffmpeg.org/log.cgi?time=20120222045648&log=compile&slot=x86_64-… [15:59] <ubitux> oh and the fpic for mingw: [15:59] <ubitux> http://fate.ffmpeg.org/log.cgi?time=20120221211007&log=compile&slot=x86_64-… [15:59] <ubitux> these configs are flooded with unwanted warnings [16:01] <michaelni> ubitux, dont remember why copycooker isnt used, maybe something didnt work or it was just unfinished+forgotten [16:01] <ubitux> ok [17:14] <overflow_0f8b> is this the simplest and best way getting the fractional part of a float ? ;/ (a).f = ( abf ) - floorf( abf ); [18:56] <Plorkyeran> modf [20:34] <overflow_0f8b> Plorkyeran<< but floatingpoint division is slow [20:35] <overflow_0f8b> #define fractf( f ) ( copysign( fmodf( fabsf( f ), 1.0 ), f ) ) [20:35] <overflow_0f8b> #define fractf( f ) ( fmodf( f, copysign( 1.0, f ) ) ) [20:35] <overflow_0f8b> also... [20:36] <Plorkyeran> what does modf have to do with floating point division [20:36] <Plorkyeran> modf != fmodf [20:39] <overflow_0f8b> what does modf have to do with my floating point rounding? [20:39] <overflow_0f8b> i don't see it in tgmath.h btw [21:11] <overflow_0f8b> Plorkyeran<< lol thanks, i was doing the exact same thing as modf ;/ [21:12] <overflow_0f8b> it breaks down a float to an integer and fractional part [21:23] <overflow_0f8b> hopefully it won't start floatingpoint dividing ... [21:27] <overflow_0f8b> fi = modff( abf, &abf); [23:00] <durandal_1707> ffplay seems to ignore pkt->duration [23:15] <michaelni> durandal_1707, probably best if you mail marton or report it on trac [23:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r43b1943a55 10ffmpeg/ (doc/eval.texi libavutil/eval.c tests/ref/fate/eval): [23:19] <CIA-17> ffmpeg: eval: Add taylor series evaluation support. [23:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r094673ff1b 10ffmpeg/libavutil/eval.c: [23:19] <CIA-17> ffmpeg: eval: support 3 parameter functions. [23:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Thu Feb 23 2012

1 0

ffmpeg.log.20120222
by burek 23 Feb '12

23 Feb '12

[00:33] <pasteeater> nextime: you mean you want two output files (one video only, one audio only) from one input? [00:52] <Guest90713> hey guys [00:53] <Guest90713> i installed macports, then did sudo port install ffmpeg [00:53] <Guest90713> and my terminal isn't displaying any new lines [00:53] <Guest90713> but the last lines say "activating ffmpeg @0.7.8_0" and something about non free GPLed stuff [00:53] <Guest90713> what should i do? >< [00:56] <sacarasc> Is the CPU being used? [00:57] <Guest90713> idk [00:57] <Guest90713> how can i tell [00:57] <Guest90713> and does it matter [00:57] <sacarasc> If it's using your CPU, then it's compiling. [00:57] <sacarasc> Otherwise something went wrong. [00:58] <sacarasc> Also, 0.7.8 is very old. [00:58] <sacarasc> It's up to 0.10, now. [00:58] <Guest90713> this is what i got http://pastebin.com/i0c3eb0Z [01:03] <Freakshow> well, macports isn't up to date by any means... [01:03] <Freakshow> maybe ffmpeg-devel instead... [01:04] <Freakshow> I've recently confirmed that macports is exactly what Guest90713 has referenced [01:04] <Guest90713> huh? [01:04] <Guest90713> wowow what do i have to do [01:05] <Freakshow> hmm... I'm not entirely sure, maybe ping the macports maintainer? [01:05] <Freakshow> ask them to update? I don't know if that's an option [01:06] <Guest90713> can i close my terminal and come back tomorrow it's 1:06 am here [01:06] <Guest90713> without breaking any packages or whatever of course [01:10] <Guest90713> guys i gtg [01:41] <Freakshow> ok [02:05] <nextime> argh. [02:05] <nextime> [mp4 @ 0x25ba4a0] muxer does not support non seekable output [02:05] <nextime> how to convert for a stream that has to be embed on html5 by <video> tag? [02:06] <sacarasc> The easiest way is ffmpeg -i input output.webm [02:06] <Shimmy> Hello! [02:06] <Shimmy> Anyone willing to help me with my issue? [02:06] <Shimmy> https://ffmpeg.org/trac/ffmpeg/ticket/747 [02:06] <nextime> sacarasc : i have to support ALSO webm [02:07] <sacarasc> Webm plays in <video> tags, so win/win! [02:07] <nextime> sadly not all browser support webm or theora [02:07] <nextime> sacarasc : no, webm play in <video> tags if you are using firefox, opera, chrome [02:07] <nextime> but not with IE or safari [02:07] <nextime> not on iphone [02:07] <sacarasc> H264/? in MP4 is the other one? [02:08] <nextime> mp4 is a container [02:08] <nextime> and inside mp4 the codec usually is h264 [02:08] <sacarasc> That's why I said *IN* MP4. [02:08] <pasteeater> yeah, sacarasc. duhhh. [02:08] <sacarasc> And what audio? [02:08] <nextime> aac [02:08] Action: pasteeater annoys sacarasc [02:10] <sacarasc> ffmpeg -i input -acodec aac -b:a 60k -vcodec libx264 -preset blah -profile:v blah2 -b 500k output.mp4 && qtfaststart output.mp4 output2.mp4 [02:10] <nextime> sacarasc : it doesn't work if you are using an output that is a pipe [02:11] <nextime> it's a realtime converted network stream [02:11] <nextime> i can't do it on a file [02:11] <pasteeater> this is why you use pastebin to provide your complete command and the complete console output when you encounter errors [02:11] <sacarasc> Or say exactly what you want first. [02:12] <pasteeater> although we can becoming more psychic with experience... [02:12] <nextime> http://pastebin.com/sGnLK9ED [02:12] <sacarasc> I suppose the [mp4 @ 0x25ba4a0] muxer does not support non seekable output should have given it away that you were trying to stream ffmpeg produced mp4... [02:12] <nextime> sacarasc: exactly [02:13] <nextime> and now, as the mux can't work on a pipe [02:13] <nextime> i need to find an alternative container [02:13] <nextime> that work on <video> for IE and safari [02:13] <nextime> and iphone and ipad [02:29] <pasteeater> vo-aacenc 0.1.2 released today, and opencore-amr & vo-amrwbenc, but who gives a rat's ass about those. [02:29] <pasteeater> ...not that i really care for vo-aacenc. [02:44] <nextime> ok, webm, ogg, and flv for flash are working [02:44] <nextime> only the mp4 issue remain... [02:52] <Freakshow> mp4, who gives a rat's ass about that [02:53] <Freakshow> >.< [02:53] <assem> trying to compile under cygwin and at linking avdevice i get undefined references to _sws_scale, _sws_freeContext, ..., _swr_convert, _swr_free [02:54] <assem> any help? using both ffmpeg-0.10 release and the git repo [02:56] <assem> hm nm i appear to have just fixed my problem; added swresample and swscale to avdevice/Makefile's FFLIBS=... [06:22] <maujhsn> Would enabling the enviromental variable "--shlibdir=DIR" in ffmpeg be the way to go for a new plug-in like "the gimp-plugin which is independent of ffmpeg? [06:27] <maujhsn> Mavrik Any thoughts on my topic?:) [06:28] <mgeary> hi folks [06:28] <mgeary> is there a way to get libfaac to not have a problem with mono audio? [09:59] <pom> This issue http://ffmpeg.org/trac/ffmpeg/ticket/336 is marked as fixed. How can I figure out if this resolution is included in the git repo, and if not, how can I contact "michael"? [10:03] <ubitux> it's most likely in; did you expect a similar issue? [10:05] <pom> ubitux: Don't understand what you mean by expect, but I seem to have this exact symptom now. [10:06] <pom> I just noticed that it can be worked-around by specifying format wih -f "mp3" before the -i. [10:06] <ubitux> could you share the sample? [10:07] <pom> Yes. Just wanna make sure I'm using a recent version on this machine. Brb. [10:09] <pom> My version is built at feb 1st, but it's only version 0.7. I'm gonna have a look at this first and get back to you if I still experience it. [10:29] <dipanjan> hi, i have converted some m4v files to vob files using ffmpeg. but dvdauthor says No VOBus found. Can anyone help? used `ffmpeg -y -sameq -i "$i" -target pal-dvd "$name.vob"` [11:48] <mrfree> hi all [11:48] <mrfree> how can I show all available presets in ffmpeg? [11:48] <mrfree> I read something about "x264 --help" but I don't know... well... where use it :) ffmpeg x264 --help doesn't work [12:00] <mrfree> using strace I've found the problem... ffmpeg search for .ffpreset file but it should use .avpreset, shouldn't it? [12:01] <mrfree> adding a symlink works [12:03] <JEEB> libx264 has its own internal presets now [12:03] <JEEB> which should be used [12:04] <JEEB> as they're kept up-to-date on the libx264 side [12:04] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [12:04] <JEEB> if you have no x264 binary, you can check this listing for a list of presets [12:06] <Mavrik_> morning everyone :) [16:06] <Iszak> Is there any progress on supporting Intel QuickSync in FFMPeg? [17:35] <mgeary> hi folks [17:36] <mgeary> so, i'm still struggling with some sort of aspect ratio issue. Quicktime is interpreting the video dimensions differently than ffmpeg. Would manually changing the pixel aspect ratio possibly solve this issue? [18:37] <minihydra> Im trying to downscale fullhd screencast, 4x (2x width and 2x height) to get smaller filesize, however i get considerably worse quality and file size remains almost the same... Maybe someone could give me some advice? [18:41] <minihydra> http://pastebin.com/EVgvP9xt some output [18:42] <minihydra> i specify -q:v 1, so according to manual this should imply vbr, but while encoding q holds about 27, so this seems strange, and how can file take almost same space with 4x smaller resolution and lower quality.... [20:59] <kevin> so if i use ffmpeg to generate ntsc-dvd compatible mpgs (using -target ntsc-dvd), can i join the mpgs using 'cat' to create a larger mpg for use in dvdauthor? [21:00] <kevin> or do i have to re-encode the joined mpg file? (not sure about headers joining from the individual files, etc) [21:03] <Mavrik> afaik .mpg files can be joined [21:05] <kevin> yeah, that's what the ffmpeg FAQ says regarding joining multiple video files (convert each to mpg, cat the mpgs, convert back to original format) , so i'm thinking it should work. i'll just have to try it out [21:07] <kevin> second question: can you hardcode subtitles using ffmpeg into the actual video stream? i read somewhere that it cant and i'd need to use like mencoder. i'm looking at ffmpeg man pages and all subtitle-related options seem to talk about the subtitle stream, not actually writing it in the video stream [21:12] <Mavrik> kevin, check libavformat documentation for a subtitle video filter [21:12] <Mavrik> but I doubt it's possible at the moment [21:18] <ubitux> kevin: with libass video filter you can [21:18] <ubitux> it requires you to build with --enable-libass, and also have the subtitles in a file, in ASS format [21:18] <ubitux> it's meant to be a temporary solution for hardsub, but it should work [21:26] <kevin> hrmmm alright thanks [21:38] <VooDooNOFX_> kevin, ffmbc can burn in subtitles [21:43] <vadim_> hi, guys [21:45] <vadim_> ffmpeg hangs on reading from udp source... it throws a "circular_buffer: OVERRUN" and hangs after a few seconds... have somebody solved this?.. [21:48] <pasteeater> vadim_: i know nothing of udp, but generally when you encounter an error your command and the complete console output is useful. use a pastebin service. [21:49] <rexbron> Hey everyone, ffmpeg is giving garbage output on some ProRes 4444 samples I have and ffplay is also wacked. Is ProRes4444 suported by the decoder? [21:50] <rexbron> Also, does ffmpeg support copying of mov timecode tracks into other movs/wrappers? [22:04] <PeraZdera> Hi [22:06] <PeraZdera> can anyone help me with circular buffer overun problem? I want to transcode live mpegts to h264 but after some time I get circular buffer overrun message and ffmpeg get 100% cpu usage [22:06] <PeraZdera> here is pastebin [22:06] <PeraZdera> http://pastebin.com/9Y9ZNPhD [22:07] <vadim_> pasteeater: i was talking about the same problem as PeraZdera does... [22:10] <PeraZdera> vadim: which is circular buffer overun? [22:10] <cbsrobot> rexbron: can you share a prores 4444 sample ? [22:10] <rexbron> cbsrobot: sure [22:10] <cbsrobot> a few seconds is enough .... [22:11] <vadim_> PeraZdera: this is a bug... ( [22:11] <rexbron> cbsrobot: the file has a lot of preroll (bars and a sync countdown), should it be actual image data? [22:11] <cbsrobot> rexbron: yes please [22:11] <cbsrobot> btw. I think copying timecode is not well supportet [22:12] <cbsrobot> but try ffmbc [22:12] <PeraZdera> vadim_: are you sure? I can't find any open bug about this the last one is close in 2008 IIRC [22:12] <PeraZdera> vadim_: s/close/closed/ [22:12] <vadim_> PeraZdera: https://ffmpeg.org/trac/ffmpeg/ticket/609 [22:13] <vadim_> PeraZdera: and it is open for 4 months :( [22:13] <rexbron> cbsrobot: should I upload it to incomming or my own hosting? [22:14] <cbsrobot> i don't care [22:14] <vadim_> PeraZdera: i have tested with latest ffmpeg 0.10, the problem is still there [22:14] <cbsrobot> maybe soemwhere I can dl it [22:14] <cbsrobot> if I remember well ffmpeg even crashes trying to copy tc tracks in mov [22:15] <cbsrobot> I was working on it a few weeks ago - but I don't have time to finish it atm :-(. [22:16] <cbsrobot> relaxed: are he samples form an alexa ? [22:16] <PeraZdera> vadim_: hmm I am using the latest version from git repo and still the same bug, let me see if I can find the older bug I have found to be reported as fixed... [22:18] <vadim_> PeraZdera: i have analyzed all major versions starting with 0.6.1... each introduces new modifications but the result is the same :( [22:19] <rexbron> cbsrobot: No, it is the master from a short film I shot [22:19] <vadim_> PeraZdera: the ticket i have referenced, IMHO, describes the problem the best possible... [22:19] <cbsrobot> what device recorded it ? [22:20] <PeraZdera> vadim_: http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7c4e4c6a0664bf668… [22:20] <vadim_> pasteeater: who should we pray to, to solve this problem ASAP?.. i'm not the only person which need this module stable... [22:21] <rexbron> cbsrobot: Created by apple compressor (From Final Cut Studio 3 iirc) [22:22] <cbsrobot> hmm - why 4444 then ? [22:26] <rexbron> cbsrobot: Onlining from HDCAM SR after colour grading [22:26] <rexbron> I think [22:26] <rexbron> it was about three years ago [22:26] <rexbron> :P [22:27] <rexbron> cbsrobot: Noticed something odd [22:27] <rexbron> cbsrobot: http://pastebin.ca/2120881 [22:28] <rexbron> cbsrobot: copying both streams results in the new file probing as the wrong pixel format [22:29] <rexbron> also, duration is incorrect [22:29] <vadim_> PeraZdera: what version of ffmpeg do you use? [22:30] <cbsrobot> rexbron: not sure qhat is going on [22:30] <cbsrobot> there are now 2 enoders and 2 decoders in ffmpeg ... [22:30] <pasteeater> vadim_: if you think it's a bug then check the reports here: https://ffmpeg.org/trac/ffmpeg/ [22:30] <cbsrobot> I need to run [22:31] <cbsrobot> just post the link and I'll see later [22:31] <pasteeater> if you don't see an existing bug related to your issue(s) then make a new, detailed report [22:32] <vadim_> pasteeater: this one describes the situation https://ffmpeg.org/trac/ffmpeg/ticket/609 [22:33] <rexbron> cbsrobot: http://files.aehunter.net/DIVISION_SAMPLE.mov [22:34] <pasteeater> vadim_: looks like they are using ffmpeg programtically, and not ffmpegcli. you might want to post a response to the report with a cli example so others can duplicate the issue easier. [22:34] <pasteeater> if you believe it's the same issue you are experiencing [22:38] <vadim_> pasteeater: ok... i'll open a new ticket with the console listing... but IMHO it's the same trouble... btw, take a look at this http://pastebin.com/9Y9ZNPhD I'm going to send something similar... [22:39] <vadim_> pasteeater: the link i sent you was posted by PeraZdera [22:46] <rexbron> cbsrobot: reported as https://ffmpeg.org/trac/ffmpeg/ticket/1015 [22:51] <PeraZdera> vadim_: can you please post your pastebin link? I would like to see how ffmpeg was compiled [22:52] <vadim_> PeraZdera: compiling right now... will post it in 10-15 minutes [22:56] <PeraZdera> vadim_: Do you use live mpeg2-ts or you are transcoding file? [22:57] <vadim_> PeraZdera: live [22:58] <TlatoSMD> hi there [22:58] <TlatoSMD> hey, I'm trying to encode an SRT subtitle into a file [22:58] <TlatoSMD> so far, I've been using VDub and Avidemux, to no avail [22:58] <TlatoSMD> how do I do with ffmpeg? [22:59] <TlatoSMD> in any case, I suppose hard subs (that are always on) will do [22:59] <kevin> TlatoSMD - are you trying to create it as a new stream or overlay/bake it in the video stream? [23:00] <TlatoSMD> the process will also involve transcoding it from DivX to x264, so I suppose that makes it a new stream? [23:00] <kevin> i asked about the latter earlier today and got this response: https://gist.github.com/1887712 [23:02] <TlatoSMD> sounds like I'll need an ASS file in the first place? [23:03] <undercash> hi [23:03] <undercash> what s max crf value? [23:06] <kevin> TlatoSMD: for the hard subs i believe so. i personally haven't tried it yet [23:11] <CarlFK> melt (build on ffmpeg libs) uses acodec=libmp3lame ... vcodec=libx264 which no longer works. melt dev says "The codec names probably changed." anone know what I should be using ? [23:12] <vadim_> CarlFK: do you use '=' sign in your cli? [23:13] <CarlFK> yes [23:13] <vadim_> replace with ' ' (space) [23:13] <CarlFK> that was cut/paste from what I run at the $ prompt [23:13] <CarlFK> k [23:22] <vadim_> PeraZdera: http://pastebin.com/0gK9JSw4 here it is... sorry for the length... [23:27] <tbws> hi everyone [23:28] <tbws> i'm having a problem compiling ffmpeg on lucid. Apparently it cannot find my installation of libaac [23:28] <tbws> 2 secs im just uploading my log [23:31] <CarlFK> vadim_: melt syntax wants the = replaced with space I get: Failed to load "acodec"; Failed to load "libmp3lame" [23:31] <sacarasc> CarlFK: Full command line and full output to a pastebin, please. [23:34] <CarlFK> http://pastebin.com/vbR9XZgv [consumer avformat] audio codec libmp3lame unrecognised - ignoring [23:35] <tbws> http://textuploader.com/?p=6&id=OlQSI : just to recap i'm having problems compiling ffmpeg on lucid as it can't find libfaac [23:35] <vadim_> CarlFK: sorry, my mistake... never used melt... [23:38] <vadim_> tbws: have you installed libfaac-dev package?.. [23:38] <tbws> yep, its reffered to as libfaad-dev on lucid [23:40] <vadim_> tbws: i,ho, you're wrong... install libfaac-dev... follow this http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 [23:41] <tbws> couldn't find package [23:41] <tbws> i don't think i am [23:44] <vadim_> tbws: take a look here http://pastebin.com/72bv5RVT this from my machine [23:46] <pasteeater> undercash: 51 is the highest crf value (for 8-bit x264 at least) [23:47] <undercash> hi pastee thx [23:47] <tbws> ty vadim_ what does your sources.list look like? [23:49] <pasteeater> tbws: maybe a better quesiton is what does _your_ sources.list look like? [23:55] <vadim_> pasteeater: lol, it's not a big deal to share the sources.list ))) [23:55] <vadim_> pasteeater: any way, thx ) [23:55] <vadim_> tbws: http://pastebin.com/X7mE6d33 [00:00] --- Thu Feb 23 2012

1 0

ffmpeg-devel.log.20120221
by burek 22 Feb '12

22 Feb '12

[01:13] <bove> I'm trying to understand what determines the coefficients when converting rgb to yuv. I see some mentions in yuv2rgb.c, but I don't know how it's treated the other way [03:32] <durandal_1707> lol michaelni MatildaHarris is still active and spamming [03:34] <Compn> on mailing list ? [03:34] <durandal_1707> wiki on trac [03:35] <Compn> ah [05:29] <CIA-17> ffmpeg: 03Panagiotis H.M. Issaris 07master * r2b3d041cdc 10ffmpeg/libavformat/hls.c: (log message trimmed) [05:29] <CIA-17> ffmpeg: applehttp: Do seeking within segments, too [05:29] <CIA-17> ffmpeg: Enhance seeking by demuxing until the requested timestamp is [05:29] <CIA-17> ffmpeg: reached within the segment selected by the seek code using the [05:29] <CIA-17> ffmpeg: playlist info. [05:29] <CIA-17> ffmpeg: Some mpegts streams don't have dts set for all packets though, [05:29] <CIA-17> ffmpeg: this seeking method doesn't work well for that case. [05:29] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * rb18f8cbf3d 10ffmpeg/libswscale/ (swscale_internal.h x86/swscale_mmx.c x86/swscale_template.c): (log message trimmed) [05:29] <CIA-17> ffmpeg: Revert two swscale commits. [05:29] <CIA-17> ffmpeg: Revert "swscale: update context offsets after removal of AlpMmxFilter." [05:29] <CIA-17> ffmpeg: (commit a95e3fa90b4190381b65d180eec5a4027075e2da) [05:29] <CIA-17> ffmpeg: and [05:29] <CIA-17> ffmpeg: Revert "swscale: Remove some write-only variables related to alpha handling." [05:29] <CIA-17> ffmpeg: (commit 9d03cb9fc5ddf914920ab0dbe13f19a34c754966). [05:29] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r8fb26950ed 10ffmpeg/libavcodec/x86/h264_deblock.asm: [05:29] <CIA-17> ffmpeg: h264: don't use redzone in loopfilter on win64. [05:29] <CIA-17> ffmpeg: Red zone usage is not allowed in the Win64 ABI. [05:29] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r11505f39e1 10ffmpeg/libavcodec/zmbvenc.c: zmbvenc: switch to encode2(). [05:29] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r8c8c7b5e37 10ffmpeg/libavcodec/zmbvenc.c: [05:29] <CIA-17> ffmpeg: zmbvenc: move header writing to the end of encode_frame(). [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r0b42a9388c 10ffmpeg/ (4 files in 2 dirs): avutil: add av_rescale_q_rnd() to allow different rounding [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * re9cda85351 10ffmpeg/ (doc/APIchanges libavcodec/avcodec.h libavformat/utils.c): [05:30] <CIA-17> ffmpeg: avcodec: add duration field to AVCodecParserContext [05:30] <CIA-17> ffmpeg: This will allow parsers to export the duration of the current frame being [05:30] <CIA-17> ffmpeg: output, if known, instead of using AVCodecContext.frame_size. [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rc7f3f1c91e 10ffmpeg/libavcodec/flac_parser.c: flac parser: set duration instead of frame_size [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r16e54ac725 10ffmpeg/libavcodec/aac_ac3_parser.c: (e)ac3 parser: set duration instead of frame_size [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r7575ffac8a 10ffmpeg/libavcodec/mpegaudio_parser.c: mpegaudio parser: set duration instead of frame_size [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r2460b168b4 10ffmpeg/libavcodec/gsm_parser.c: gsm parser: set duration [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rb3a4c7e0f1 10ffmpeg/libavcodec/mlp_parser.c: mlp parser: set duration instead of frame_size [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r41ac9bb253 10ffmpeg/libavcodec/adx_parser.c: adx parser: set duration [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r91a28b0e8e 10ffmpeg/libavcodec/ (internal.h libspeexenc.c libvorbis.c utils.c): avcodec: add ff_samples_to_time_base() convenience function to internal.h [05:30] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r770a5c6d02 10ffmpeg/ (libavcodec/adpcmenc.c tests/ref/acodec/adpcm_yam): (log message trimmed) [05:30] <CIA-17> ffmpeg: adpcmenc: Use correct frame_size for Yamaha ADPCM. [05:30] <CIA-17> ffmpeg: Output packet size should match avctx->block_align. The target output packet [05:30] <CIA-17> ffmpeg: size is 1024 bytes. [05:30] <CIA-17> ffmpeg: Before: [05:30] <CIA-17> ffmpeg: mono - 1024 samples -> 512 bytes [05:30] <CIA-17> ffmpeg: stereo - 2048 samples -> 2048 bytes [05:30] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * readd4264ee 10ffmpeg/: (log message trimmed) [05:30] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [05:30] <CIA-17> ffmpeg: * qatar/master: (36 commits) [05:30] <CIA-17> ffmpeg: adpcmenc: Use correct frame_size for Yamaha ADPCM. [05:30] <CIA-17> ffmpeg: avcodec: add ff_samples_to_time_base() convenience function to internal.h [05:30] <CIA-17> ffmpeg: adx parser: set duration [05:30] <CIA-17> ffmpeg: mlp parser: set duration instead of frame_size [09:39] <AdamWorld> Hi! I run "ffserver.exe" in CentOS with command:"./ffserver -f doc/ffserver.conf" and "ffmpeg.exe" in windows like "ffmpeg -i 1.avi http://192.168.0.128:8090/feed1.ffm" but "http://192.168.0.128:8090/feed1.ffm: Operation not permitted" occurred. What's wrong with me? [13:25] <ubitux> michaelni: is there a reason the "audio sync method" is not part of libswresample? [13:25] <Tjoppen> ls [13:26] <Tjoppen> oops :) [13:26] <ubitux> :) [13:27] <ubitux> i guess it needs to much pts stuff& [13:34] <Tjoppen> btw, the way ffmpeg does sync change depending on stream order. this seems wrong to me [13:35] <Tjoppen> if I capture my webcam to stream 0 and mic to stream 1 the streams end up ~1 sec off. if I reverse the order (mic = 0, webcam = 1) then it works fine [13:36] <ubitux> i also expected this issue, wasn't aware about that workaround :) [13:36] <ubitux> i have a lot of a/v desync when extracting samples too [13:37] <ubitux> (if i don't re-encode) [13:37] <Tjoppen> youtube has similar issues [13:37] <ubitux> isn't youtube using ffmpeg? :p [13:37] <Tjoppen> yes, they are [13:38] <Tjoppen> which explains the problems I had a while back with uploading emulator captures that I encoded just to not have to upload raw footage [13:39] <Tjoppen> the emulator (MESS) doesn't do a particularly good job of its AVI output, which when passed through virtualdub caused various problems [13:40] <ubitux> nice emulator name [13:43] <Tjoppen> it's actually a collection of emulators. I use it for capturing and uploading videos of demoscene productions [13:44] <Tjoppen> I should make a youtube upload guide with some tips and caveats like that sync thing [13:46] <ubitux> or fix ffmpeg and wait a few years to get it it libav and then youtube [13:46] <ubitux> or both :) [13:48] <Tjoppen> I've started to nearest-point supersample every 8-bit thing I upload. gives very crisp video and audio (since for some reason youtube shits on audio bitrate at 240p) [13:53] <Tjoppen> case in point: www.youtube.com/watch?v=CO_HcjzU4SI [14:31] <ubitux> it seems the audio sync is not tested at all [14:32] <ubitux> lcov seems to say nothing ever pass through the sync stuff [14:33] <Tjoppen> is frame accurate cutting supported yet? my guess is no [14:34] <ubitux> not afaik [14:38] <michaelni> ubitux, moving the audio_sync_method code into swr may be interresting, i havnt thought about that ... [14:39] <michaelni> Tjoppen, we maybe should change the async default, this should fix the stream order sync issue i think [14:40] <michaelni> i remember considering to change the default a while ago already bit didnt do it cuz we where before a release [14:40] <michaelni> seems i forgot about it [14:43] <Tjoppen> michaelni: sounds good [14:43] <ubitux> michaelni: this is actually giving me headaches while hacking do_audio_out to integrates -af :p [15:11] <ubitux> it would be nice to add a few -async fate tests :p [15:12] <ubitux> if the option actually work :p [16:29] <gnafu> Hehe: <AdamWorld> [...] What's wrong with me? [16:30] <ubitux> saste: i found the cause of the issue when changing channels, and fixed it [16:30] <ubitux> saste: so well, -af now "works" with ffmpeg [16:30] <ubitux> but it breaks a few other things [16:30] <saste> ubitux: what was the problem? [16:31] <ubitux> the problem is simply that do_audio_out uses the dec->channels to define the size to send to encode frame [16:31] <ubitux> but dec->channels is not the number of channels which outputs the filtergraph [16:32] <ubitux> output of* [16:32] <ubitux> *gets out* (english beeh) [16:33] <saste> ah funny [16:33] <ubitux> i'm going to send the WIP, and all the issues that need to be fixed [16:33] <saste> i wonder how many of these assumptions there are still in ffmpeg.c [16:33] <ubitux> this is not an assumption, this is normal [16:33] <ubitux> do_audio_out() is actually doing the right thing in this case [16:33] <ubitux> but not when using -af [16:34] <saste> yes i mean is correct unless you have a filtergraph between the decoder and the encoder [16:34] <ubitux> not even -af actually, the use of audio filters [16:34] <ubitux> yes right [16:34] <ubitux> but using the filtergraph between decoder and encoder causes a lot of problem in do_audio_out :) [16:35] <ubitux> so i just disabled almost everything in it [16:35] <ubitux> (hoping the af aresample and aconvert will deal with it fine) [16:35] <ubitux> but there is still two unsolved problem even if that works: [16:35] <ubitux> -async [16:35] <ubitux> and -map_channel (i can deal with this one) [16:35] <ubitux> -async is the real issue :p [16:36] <ubitux> also, aresample will need some rework too, iirc it requires a flush [16:36] <ubitux> a lot of things to do :( [16:36] <saste> yes a flush API is missing (also for video) [16:37] <saste> the more you do the more you know there is to do [16:37] <saste> it's normal [16:37] <ubitux> well the issue is, the more there is to do to finish just a feature :p [16:37] <ubitux> it's normal to get more and more things to do, but when they're all blocking one single issue, it's a nightmare :D [16:38] <ubitux> all of this is just to get the merge in -map_channel ffs! :D [16:38] <saste> eheh... that's because we're pushing to the extreme the design of ffmpeg.c [16:39] <ubitux> i'd like to auto insert the volume filter too :-° [16:46] <CIA-17> ffmpeg: 03Clément BSsch 07master * r241f8465d0 10ffmpeg/libavfilter/asrc_abuffer.c: (log message trimmed) [16:46] <CIA-17> ffmpeg: lavfi/abuffer: init the data planes with NULL pointers. [16:46] <CIA-17> ffmpeg: Samples buffer ref is allocated and loaded with the uninitialized data [16:46] <CIA-17> ffmpeg: pointers: [16:46] <CIA-17> ffmpeg: av_asrc_buffer_add_buffer() [16:46] <CIA-17> ffmpeg: -> av_asrc_buffer_add_samples() [16:46] <CIA-17> ffmpeg: -> avfilter_get_audio_buffer_ref_from_arrays(data, ...) [16:46] <CIA-17> ffmpeg: 03Clément BSsch 07master * raecf0cf5ed 10ffmpeg/doc/examples/ (Makefile filtering_audio.c): [16:46] <CIA-17> ffmpeg: doc/examples: add audio decoding/filtering example. [16:46] <CIA-17> ffmpeg: Mostly based on doc/examples/filtering.c. lavfi API is still limited to [16:46] <CIA-17> ffmpeg: "buffer feeding" instead of "frame feeding" at the moment, so this [16:46] <CIA-17> ffmpeg: example code sticks with it. [17:34] Action: Daemon404 pokes kierank [20:35] <Freakshow> what's a supported format for -loop [20:39] <Freakshow> nm... I think I found out [22:39] <kriegerod> Is this a sane idea to implement video filter that gives output pictures with given framerate? So it can force constant framerate from vfr input, or just change fps. [22:47] <CIA-17> ffmpeg: 03Clément BSsch 07master * r2bdac29360 10ffmpeg/doc/examples/ (Makefile filtering.c): doc/examples: rename filtering.c into filtering_video.c. [22:48] <overflow_0f8b> hi [23:06] <burek> kriegerod, do you mean ffmpeg -i input -vcodec ... -acodec ... -r X -f image2 out%02d.png ? [23:07] <kriegerod> No, i mean avfilter [23:07] <iive> i think he means something like filter that enforces constant framerate [23:08] <kriegerod> That allows to transform, e.g., vfr flv video to constant 25fps [23:09] <kriegerod> Avfilter will be handy for apps [23:09] <kriegerod> I need some mentorship to implement it [23:59] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r68fac5c2b8 10ffmpeg/libavutil/avutil.h: [23:59] <CIA-17> ffmpeg: doxy: Disable the main index page. [23:59] <CIA-17> ffmpeg: Theres no usefull or even remotely complete information on it currently. [23:59] <CIA-17> ffmpeg: Which just leads to confusion. [23:59] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:59] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * re6380afa6a 10ffmpeg/libavformat/avidec.c: [23:59] <CIA-17> ffmpeg: avidec: Reduce log level for out of index error message. [23:59] <CIA-17> ffmpeg: Its otherwise spaming every time one tries to seek to outside [23:59] <CIA-17> ffmpeg: the file. [23:59] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:59] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r92b5f71a7d 10ffmpeg/libavformat/utils.c: [23:59] <CIA-17> ffmpeg: lavf: Reimplement new seek API emulation [23:59] <CIA-17> ffmpeg: This fixes seeking to before and after files with ffplay. [23:59] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Wed Feb 22 2012

1 0

ffmpeg.log.20120221
by burek 22 Feb '12

22 Feb '12

[01:12] <lucas^> ffmpeg says this file has 50356 video frames [01:12] <lucas^> comskip says it has 59258 [01:12] <lucas^> which do I trust? [03:12] <AdamWorld> Hi! [03:14] <AdamWorld> I run ffserver in CentOS and ffmpeg in windows like ffmpeg -i disney.avi http://192.168.157.129:8090/feed1.ffm but in ffmpeg "Operation not permitted" error occurrd. In localhost It succeeds. What wrong with me? [03:15] <AdamWorld> The correct err is "http://192.168.157.129:8090/feed1.ffm: Operation not permitted". Help me. [03:53] <AdamWorld> no one anwers... [03:56] <grepper> gee, one minute and no one answers :) [09:04] <AdamWorld> Hi! I run "ffserver.exe" in CentOS with command:"./ffserver -f doc/ffserver.conf" and "ffmpeg.exe" in windows like "ffmpeg -i 1.avi http://192.168.0.128:8090/feed1.ffm" but "http://192.168.0.128:8090/feed1.ffm: Operation not permitted" occurred. What's wrong with me? [10:38] <elkng> are "ffmpeg" and "mencoder" competitors same way as "vim" and "emacs" or "linux" and "windows" ? [10:40] <dirtycookie> hello people, back in november last year i tried to install ffmpeg from source onto my ubuntu, and i get this message:"ffmpeg: relocation error: /usr/local/lib/libswresample.so.0: symbol av_opt_set_int, version LIBAVUTIL_51 not defined in file libavutil.so.51 with link time reference" [10:40] <dirtycookie> i left it because i didnt have the time to fix it. i wanted to ask howto uninstall it [10:41] <ubitux> elkng: mukashi mukashi, there was mplayer+mencoder, then ffmpeg spawned, and was meant to replace mencoder. mencoder isn't maintain much anymore for years, and all dev is meant to be done on ffmpeg (even/especially from a mplayer point of view) [10:42] <ubitux> dirtycookie: make uninstall from the source directory? [10:42] <dirtycookie> ubitux: i would if i only had the folder [10:43] <ubitux> do you still have your ./configure in mind? [10:43] <ubitux> mind/history [10:43] <cbreak-work> elkng: ffmpeg is a subset of mplayer [10:44] <cbreak-work> (not organisational but in code, because mplayer uses ffmpeg as library) [10:44] <ubitux> cbreak-work: mmh, better say ffmpeg provides libraries used in mplayer [10:44] <cbreak-work> the last time I checked, ffmpeg was added as subrepository to mplayer [10:45] <ubitux> ton simplifies things, but using the term "subset" is a bit confusion imo, since the goal aren't really the same [10:46] <ubitux> s/ton/to/ [10:46] <ubitux> but well, whatever :) [11:23] <dirtycookie> ubitux: history couldn't give me something useful to work wit [11:23] <dirtycookie> h [11:24] <ubitux> where is it installed? [11:24] <ubitux> /usr/local? [11:30] <dirtycookie> ubitus:/usr/bin /usr/local/bin /usr/local/share usr/share [11:31] <ubitux> you have two installs then [11:31] <ubitux> one with --prefix=/usr and another one with --prefix=/usr/local (the default) [11:32] <ubitux> you may check out the repository, ./configure --prefix=... and then make uninstall [12:18] <kriegerod> ubitux, could you tell me personally, how should i pull AVFrame with audio data from AVFilterBufferRef? [12:23] <ArmandoDolce> hi at all [12:23] <ArmandoDolce> can i use ffmpeg to convert wma to mp3 ? [12:23] <ArmandoDolce> or i must use LAME ? [12:24] <Mavrik> ArmandoDolce: you can use FFMPEG with libmp3lame :) [12:27] <ArmandoDolce> but lame convert only wav to mp3 [12:27] <ArmandoDolce> lame don't convert wma to mo3 [12:27] <ArmandoDolce> lame don't convert wma to mp3 [12:27] <ArmandoDolce> is it right ? [12:28] <Mavrik> ArmandoDolce: wouldn't know about that [12:28] <Mavrik> but ffmpeg with libmp3lame can do those conversions [12:28] <ArmandoDolce> ok [12:28] <ArmandoDolce> listen [12:28] <ArmandoDolce> what options i must use to have low format ? [12:28] <Mavrik> low format? [12:28] <ArmandoDolce> example 22050 hertz [12:28] <Mavrik> something like [12:29] <Mavrik> ffmpeg -i <input_file> -acodec libmp3lame -ar 22050 output.mp3 [12:29] <Mavrik> -ab 128k to set bitrate [12:29] <Mavrik> -ac 2 to set channels [12:55] <ubitux> kriegerod: you don't "pull" an avframe given a avfilterbufferef [12:55] <ubitux> you create an avframe, and then fill it with the settings [12:55] <ubitux> look at encode_audio_frame() in ffmpeg.c [12:56] <kriegerod> but is there an analog to avfilter_fill_frame_from_video_buffer_ref()? [13:00] <ubitux> no but i have one locally [13:00] <ubitux> http://pastie.org/3426550 [13:00] <ubitux> which do this [13:00] <ubitux> you can inline something similar [13:01] <kriegerod> could you please post it to devel ml? it is good to have in API [13:01] <ubitux> it will be part of the -af patchset [13:01] <ubitux> i'm working on it [13:01] <kriegerod> great, when do you feel the patchset to be ready? [13:02] <ubitux> i have one major issue, so i don't know [13:03] <kriegerod> what is the issue? do you have your changes in some public git fork, so someone can contribute? [13:03] <ubitux> it's a local WIP, there is no need to contribute; i just need to figure out the best way to fix a bug [13:05] <ubitux> the issue is that it doesn't deal with a change in number of channels [13:06] <kriegerod> ok. thanks again for your WIP, i guess it must simplify ffmpeg.c a lot, because resampling and so on can be moved into audio filtering [13:06] <ubitux> well, ideally yes, but at the moment no :) [13:06] <ubitux> it actually complicates things a lot [13:06] <ubitux> for instance if you yes -af ... and then -ac [13:06] <ubitux> and -ar [13:07] <ubitux> also, the audio compensation code in ffmpeg.c is still needed [13:07] <ubitux> and that's the hard part [13:07] <ubitux> s/yes/have/ (dafuck.) [13:08] <ubitux> well anyway, better continue this discussion on #ffmpeg-devel [13:41] <spacepluk> hi, is it possible to filter arbitrary sections from an mpegts using libavformat? I've seen some code to add sectionfilters but I'm not sure if/how it can be used from an application [14:01] <kriegerod> spacepluk: possible, but tricky. i can sell you the code that does that (apply video filters to certain time interval, remuxing without re-encoding as much as possible) [14:05] <kriegerod> spacepluk: if you want to just cut specific piece of video, then you could say it simpler and i'd answer you to use -ss and -t options [14:06] <kriegerod> spacepluk: or, from API, to seek the input AVFormatContext to needed place, then remux all frames to output until end of time interval is reached [14:36] <spacepluk> kriegerod: I meant sections as in DVB-SI tables [14:37] <kriegerod> i don't know what's this at all [14:37] <kriegerod> maybe you mean you have multi-program MPEG TS? [14:39] <spacepluk> yes, I'd like to be able to fetch sections from arbitrary tables like EIT not just PAT/PMT and then parse the tables on my application [14:42] <spacepluk> I guess that's not possible [14:44] <kriegerod> sorry, i don't understand that in terms of tables, i can say that you can manage to read only the program you need by looking AVFormatContext.programs[], and figuring out which AVStreams belong to needed program [14:47] <spacepluk> I'm not interested in AV data here, only Service Information tables embedded in the transport stream [14:49] <spacepluk> mpegts_open_section_filter in libavformat/mpegts.c seems to be what I want, but I'm not sure if I can access that functionality from the application [14:49] <kriegerod> sure can't - its internal, static function [14:49] <kriegerod> did you look at libdvbpsi? [14:51] <spacepluk> yes, but I think I still need a way to fetch the sections [14:52] <kriegerod> example apps in libdvbpsi distribution show how to open the stream and feed it to analyzing functions [14:53] <kriegerod> then, as i recall, it'll trigger some callback func you provide, when it finds particular table [14:54] <spacepluk> oh, I've looked at libdvbsi++ which is a different library I'll check this one, thank you very much [15:28] <littlebit> hi im trying to resize a 720 mp4 file to 480 with ffmpeg, but I get the error message "Unsopported codec for output stream #0.1", how can i enable x264. I'm using debian in a vbox. [18:03] <grepper> what is stream 0.1 ? Check the output of ffmpeg -i FILE [19:35] <slackyman> hi [19:36] <sacarasc> Hi. [19:36] <slackyman> I'm cross compiling ffmpeg with libass [19:36] <slackyman> I get an error regarding fribidi [19:36] <slackyman> I compiled fribidi, fontconfig and libxml2 [19:37] <slackyman> all went fine [19:37] <slackyman> but when I compile ffmpeg I got "libass.a(ass_shaper.o):ass_shaper.c:(.text+0x140): undefined reference to `__imp__fribidi_get_joining_types' [19:37] <slackyman> " [19:38] <slackyman> I moved all the private libs in libs in every *.pc pkg-config file [19:44] <mgeary> hi folks [19:44] <mgeary> i'm struggling with pixel formats [19:46] <mgeary> i've got a video that is 720x480 with pixel format rgb24. I'm trying to shrink it to 480x320, but when i try using -s 480x320 or -vf scale=480:320, the result is 432x320 [19:46] <mgeary> ffmpeg output follows [19:47] <mgeary> http://dpaste.com/706453/ <-- i suspect at least one issue is the warning, "Incompatible pixel format 'rgb24' for codec 'libx264', auto-selecting format 'yuv420p'" [19:47] <mgeary> but i'm not sure where to go from here [19:51] <GA-Flix> Hi everyone :) [19:51] <GA-Flix> I was in here a few days ago and asked whether i'm allowed to include ffmpeg.exe into my project. [19:51] <GA-Flix> I'm still not sure whether i am or not, so i ask you again, this time with screenshots: [19:51] <GA-Flix> http://95.113.142.212/Upload%20System/files/127.0.0.1/Interface%20with%20in… [19:51] <GA-Flix> http://95.113.142.212/Upload%20System/files/127.0.0.1/Project%20as%20it%20i… [19:51] <GA-Flix> Maybe someone has a little bit time for my problem :) [19:51] <GA-Flix> Thanks [19:56] <GA-Flix> (additional info: the project is freeware) [20:05] <slackyman> GA-Flix: is the source of your project distribuited or is your software provided with a "written offer" for the source? [20:06] <GA-Flix> i don't know the difference (my english is not the best) [20:06] <GA-Flix> i will not publish the code :) [20:07] <hugogee> greets all [20:08] <slackyman> it's not so easy, GA-Flix [20:08] <GA-Flix> ouh :( [20:08] <slackyman> ffmpeg is gpl-ed [20:08] <hugogee> why would converting flac to wav cause the sound to repeat 4-6 timex? [20:08] <slackyman> but.. [20:08] <slackyman> wait [20:08] <hugogee> drivin me mad! [20:09] <slackyman> http://www.gnu.org/licenses/gpl-2.0.html [20:11] <GA-Flix> slackyman, isn't that like only for people who publish a modified ffmpeg.exe file? [20:11] <GA-Flix> (my ffmpeg.exe is original) [20:11] <slackyman> read the point #10 [20:12] <GA-Flix> slackyman, so i need to contact the owner of ffmpeg.org and ask for permission? :) [20:12] <slackyman> read also the last sentence [20:12] <slackyman> This General Public License does not permit incorporating your program into proprietary programs. If your program is a subroutine library, you may consider it more useful to permit linking proprietary applications with the library. If this is what you want to do, use the GNU Lesser General Public License instead of this License. [20:13] <slackyman> so [20:13] <slackyman> LGPL is applyed to libraries [20:13] <slackyman> like libavcodec [20:14] <slackyman> you can make a program linking that library [20:14] <slackyman> it's a difficult question [20:14] <GA-Flix> but i don't use the library (ffmpeg.dll), do i? ô.0 [20:14] <slackyman> yep [20:14] <slackyman> you are not using that [20:14] <slackyman> but you can do something [20:14] <slackyman> like... [20:15] <slackyman> your program can be distribuited WITHOUT ffmpeg [20:15] <slackyman> but at first launch it download ffmpeg.exe [20:15] <slackyman> :) [20:15] <GA-Flix> uuh, nice [20:15] <slackyman> so you are not DISTRIBUTING a copy of ffmpeg [20:15] <slackyman> and you are gpl-free [20:16] <slackyman> or... I think so [20:16] <slackyman> :) [20:16] <GA-Flix> okay :) that's really good news for me ^^ thank you very very much :) [20:16] <slackyman> maybe you can email someone here at ffmpeg to know something more [20:17] <slackyman> I think that asking is always the best way [20:17] <slackyman> or maybe you can contact fss [20:17] <slackyman> free software foundation [20:18] <GA-Flix> ehm ^^ okay ^^ [20:18] <GA-Flix> i think i'll contact a ffmpeg member [20:18] <GA-Flix> cuz i don't know whether there's a german "free software foundation" ^^ [20:19] <slackyman> I think there's a european section [20:19] <slackyman> GA-Flix: but why are you thinking not to distribute the code? [20:19] <slackyman> which language are you using? [20:20] <GA-Flix> C# [20:20] <slackyman> omg [20:20] <slackyman> O.o [20:21] <slackyman> I'm italian, by the way :) [20:21] <GA-Flix> i know that everyone who wants can easily decompile my file, but i don't want all the unskilled guys who just use copy&paste do get it ^^ [20:21] <hugogee> what is the best codec to convert form flac to wav? [20:21] <GA-Flix> ah, ok ^^ [20:21] <slackyman> GA-Flix: you're missing the advantages of free softwaer with open source [20:21] <GA-Flix> sounds interesting ^^ [20:22] <slackyman> hugogee: I think libflac [20:22] <GA-Flix> slackyman, can u tell me more (or do you have a link?) [20:22] <hugogee> ty slackyman [20:22] <slackyman> as decoder [20:22] <Bisqwit> I used to do "ffmpeg -t 200 -loop_input -i <jpegfilename> -vcodec mpeg2video -qscale 1 -g 600 -r ntsc -f mp2 output.mpg" to convert a static image into video using ffmpeg. Nowadays, it complains that -loop_input is deprecated and one should use -loop. I replaced "-loop_input" with "-loop 1", but I get "Output file #0 does not contain any stream" error. Any remedy? [20:22] <hugogee> slackyman, is that what is used by default? [20:23] <slackyman> GA-Flix: if you publish the source maybe someone can steal your project, but some other people can find bugs and fix them [20:23] <hugogee> pcm_s16le is causing echos is there a bug? [20:24] <slackyman> hugogee: you can download flac.exe somewhere [20:24] <GA-Flix> slackyman, the program was created as quick-'n'-dirty program for a few people [20:25] <GA-Flix> i don't think that bugs are that hard in this case ^^ [20:25] <hugogee> i am on linux [20:25] <slackyman> GA-Flix: I know the case, but you can publish it on the web (even if there's a billion of youtube-downloader) and make money by donations or banners or ads [20:25] <Bisqwit> I found solution. Insert "-map 0:0" and it works. I wonder why it did not do that by default. [20:26] <slackyman> hugogee: you can get flac binary [20:26] <hugogee> i wrote a script to convert a bunch of files and some just echo. WTF so i tried to convert a single file from cli. same. [20:26] <slackyman> hugogee: http://flac.sourceforge.net/download.html [20:26] <slackyman> download flac [20:27] <slackyman> flac can convert flac files to waves [20:27] <hugogee> Is there something wrong with this??? ffmpeg -i one.flac -acodec pcm_s16le -ac 1 -ab 64k -ar 8000 one.wav [20:27] <slackyman> hugogee: my mistake :) FLAC is used to convert TO flac [20:27] <hugogee> ok i will get flac, just annoyed this is happening. :D [20:28] <hugogee> doh [20:28] <slackyman> hugogee: try ffmpeg -i one.flac -ab 64k one.wav [20:28] <hugogee> k [20:29] <Freakshow> hmm... -loop_input seems intriguing... does that actually work on static input files (e.g. .flv or mp4)? [20:29] <mgeary> does the pixel format have anything to do with pixel aspect ratio? [20:29] <hugogee> same sheeit. i guess it must be the files, although i recorded them on the same system, same time, same app.?! [20:30] <slackyman> hugogee: give me a moment: I make a try [20:30] <hugogee> slackyman: can i send you the file? [20:31] <slackyman> hugogee: ok [20:31] <slackyman> I just tried to convert a flac file to a wav one without problems [20:34] <Freakshow> looks like -loop_input is deprecated... and I don't see any additional info in the docs for -loop [20:34] <Freakshow> any clues? [20:34] <Bisqwit> "-loop 1" is apparently what it's replaced with. I don't know what else it can be [20:35] <Freakshow> that's what I'm going to assume as well [20:35] <Freakshow> now, what's a supported format? [20:35] <Freakshow> :P [20:35] <Bisqwit> It only works on image2 inputs [20:36] <Freakshow> ah... boooo [20:44] <GA-Flix> bb slackyman and thank you very much :) [20:46] <slackyman> by GA-Flix [20:47] <slackyman> I cannot cross compile ffmpeg with libass [20:48] <slackyman> it keeps giving me error about fribidi [21:47] <mgeary> so, about these pixel aspect ratios.... [21:53] <pasteeater> mgeary: i see no mention of 432 in your paste. i assume you mean some player is resizing it. [21:55] <mgeary> pasteeater: sorry. here is the command that generated that output: ffmpeg -i in.mov -s 480x320 out.mp4 [21:56] <mgeary> and yes, when i view the result in Quicktime, it reports it as 432x320 [21:56] <pasteeater> i had a feeling it was quicktime [21:56] <mgeary> but you are correct that ffmpeg -i out.mp4 does report an apparent size of 480x320 :S [21:57] <mgeary> i'm also confused by this "Incompatible pixel format" message [21:57] <nextime> hello all [21:58] <nextime> is there a way to get a network stream and dump audio and video to two separate pipes concurrently? [22:00] <pasteeater> mgeary: ignore that message. the libx264 encoder doesn't support rgb, so it converts to yuv420p (the libx264rbg encoder does support rgb, but you probably want yuv420p anyway) [22:00] <mgeary> pasteeater: aha, okay. cool [22:01] <pasteeater> and most players only support YUV anyway [22:01] <pasteeater> AFAIK [22:02] <mgeary> works for me. i just was gathering from the output message that it wasn't using H.264, but i do see now that it is [22:02] <mgeary> so if i could just get it to play back at the expected size, i'd be set [22:05] <pasteeater> does ffplay look as you would expect? [22:07] <mgeary> i don't have ffplay on my machine (Mac). Lemme try in VLC... [22:10] <mgeary> well, VLC reports it as 480x320, but it is actually displaying at ~480x355 :S [22:13] <pasteeater> you can force a desired aspect with Video > Aspect Ratio [22:13] <pasteeater> in VLC [22:15] <mgeary> but clearly something is unusual (to me), presumably in my source media, because this kind of thing doesn't usually happen (in Quicktime). If i specify or change the PAR, would that make it work the way i expect? [22:42] <Franz> hi everyone [22:42] <mgeary> bongiorno Franz [22:44] <Franz> i need your help... I'm trying to convert a .mp4 (H.264 with AAC) file to avi format, so i can play the video on my moviestation... But right now, nothing has worked... Can you please help me? [22:44] <Franz> thank you and sorry for my bad english [22:45] <mgeary> Franz: do you get an error when you try to use ffmpeg to do this? [22:46] <Franz> mgeary: nope, i've tried thing like: ffmpeg -i file.mp4 -ab 128k -ar 48000 -acodec libmp3lame -vb 866k -async 1 -vtag XVID file.avi [22:47] <Franz> no errors, wmp play it well, but my moviestation wont read it [22:48] <mgeary> ah. i'm afraid i'm neither an ffmpeg expert, nor do i know anything about moviestation. Sorry. Maybe someone else here can help [22:49] <Franz> ah ok, thanks anyway mgeary! [23:22] <Freakshow> btw, thanks kriegerod for that assist with the multi output, works nicely [23:23] <kriegerod> Np [23:26] <Freakshow> I'm curious if there's a way to split the log output per output stream [23:28] <Freakshow> because I can't define the first output rtsp://blah.foo.com/output/stream 2> log -b:500k -bt 500k so on and so forth [23:28] <Freakshow> as it decides it's done and just starts the first output only [23:29] <Freakshow> meanwhile the log complains that -b is an unknown command [23:29] <Freakshow> :P [23:38] <kriegerod> Try newer ffmpeg [23:41] <bove> Anyone know how coeficients are decided when converting rgb to yuv? [23:48] <Freakshow> :P 0.9.1 isn't new enough? [23:55] <vadim__> hi everyone [23:56] <vadim__> how can "circular_buffer: OVERRUN" error be solved?.. [23:59] <nextime> #2 try: is there a way to say to ffmpeg to dump audio and video on two separate pipes in output? i know about -an and -vn, but i need to work in realtime on a network stream, so, i can't do it in two passes [00:00] --- Wed Feb 22 2012

1 0

ffmpeg-devel.log.20120220
by burek 21 Feb '12

21 Feb '12

[01:42] <CIA-48> ffmpeg: 03Anton Khirnov 07master * r5455384219 10ffmpeg/libavcodec/mpegvideo_enc.c: [01:42] <CIA-48> ffmpeg: mpegvideo_enc: force encoding delay of at least 1 frame when low_delay=0 [01:42] <CIA-48> ffmpeg: This allows the following commit to extrapolate better dts for the first [01:42] <CIA-48> ffmpeg: frame. Pts difference between the first two frames is reused as the [01:42] <CIA-48> ffmpeg: difference between pts and dts of the first frame. [01:42] <CIA-48> ffmpeg: 03Anton Khirnov 07master * r445a7d48b1 10ffmpeg/libavcodec/ (10 files): mpegvideo_enc: switch to encode2(). [01:42] <CIA-48> ffmpeg: 03Aneesh Dogra 07master * r1a58daed0f 10ffmpeg/libavcodec/sunrastenc.c: [01:42] <CIA-48> ffmpeg: sunrastenc: set keyframe flag for the output packet. [01:42] <CIA-48> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [01:42] <CIA-48> ffmpeg: 03Aneesh Dogra 07master * r566df2eea2 10ffmpeg/libavcodec/sunrast.c: [01:42] <CIA-48> ffmpeg: sunrast: use RLE trigger macro inplace of the hard coded value. [01:42] <CIA-48> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [01:42] <CIA-48> ffmpeg: 03Paul B Mahol 07master * re04ca1d4ce 10ffmpeg/libavcodec/tta.c: [01:42] <CIA-48> ffmpeg: ttadec: cosmetics: reindent [01:42] <CIA-48> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [02:56] <overflow_0f8b> no way telling gcc not to jump around like a stupid monkey? [02:57] <overflow_0f8b> tried to force gcc make cmovs, but it likes to jump [03:22] <Compn> overflow_0f8b : -O0 ? ;P [03:22] <Compn> unroll all loops ? [03:22] Action: Compn no clue [03:31] <overflow_0f8b> used -o3 [09:24] <CIA-17> ffmpeg: 03Clément BSsch 07master * r6e6bdeb137 10ffmpeg/doc/filters.texi: [09:24] <CIA-17> ffmpeg: doc: fix channel layout value in abuffer example. [09:24] <CIA-17> ffmpeg: The channel layout must start with 0x, otherwise it is parsed as a [09:24] <CIA-17> ffmpeg: number of channels. [10:34] <ubitux> michaelni: how do you notice the timefilter divergence issues you're talking about with Nicolas? [15:04] <mateo`> hi, it seems that this merged commit from qatar (9ce2a91b842bdb99fae0f153c5317d786a3f38d8), disable the use of some mpeg2 flag such as ivlc and non_linear_q. [15:05] <mateo`> is there a new method to specify them ? [15:05] <av500> try bahrain? [15:05] <av500> or the emirates [15:20] <michaelni> mateo`, there are codec specific options, look at the mpeg2_options[] and mpeg1_options[] tables [15:20] <michaelni> havnt checked if they are docuemented [15:20] <michaelni> if not a patch that adds them to the docs would be welcome [15:22] <mateo`> michaelni: thanks [15:33] <Daemon404> michaelni, in swscale, what does readChrPlanar refer to? [15:33] <Daemon404> i.e.it's a function pointer, but what is said function supposed to do [15:34] <cbsrobot> maybe read chroma planar ? [15:34] <Daemon404> hmm perhaps [15:34] <Daemon404> looks like i might need to extend swscale to do what i want... [15:35] <cbsrobot> what are you working on ? [15:35] <Daemon404> currently, GBRP is -always- converted to yuv (and scaled too mught i add), and then converted back to packed rgb [15:35] <Daemon404> when doing a GBRP -> RGB conversion. [15:36] <Daemon404> obviously this is Bad (TM) [15:36] <cbsrobot> says who ? [15:36] <Daemon404> what do you mean? [15:36] <Daemon404> says who what? [15:37] <cbsrobot> "this is Bad" [15:37] <michaelni> Daemon404, you are planing to add a RGB scaling codepath ? [15:37] <Daemon404> maybe [15:37] <michaelni> :) [15:37] <Daemon404> cbsrobot, if youre doing gbrp to rgb conversion, it's simpyl reordering bytes [15:37] <Daemon404> converting to yuv in between is BAD [15:37] <Daemon404> fucks up color [15:37] <cbsrobot> ah I get it [15:37] <Daemon404> andd introduces roundign errors/. [15:38] Action: cbsrobot understood adding gbrp->rgb is bad [15:38] <michaelni> btw, strictly speaking it shouldnt fuck up color (not counting rounding) [15:38] <Daemon404> it's getting scaled. [15:38] <Daemon404> to tv levels, i think [15:38] <Daemon404> (limited range yuv) [15:38] <cbsrobot> btw do we support rec709? [15:39] <michaelni> we should use more bits [15:39] <cbsrobot> I only found rec609 conversion [15:39] <michaelni> we partially support 709 [15:39] <Daemon404> michaelni, it's using limited range always [15:39] <Daemon404> from what i udnerstand [15:39] <Daemon404> and never full range [15:39] <cbsrobot> michaelni: where ? [15:39] <Daemon404> [09:37] < michaelni> Daemon404, you are planing to add a RGB scaling codepath ? <-- BBB said he was going to, Eventually (TM) [15:40] <Daemon404> im merely impatient :x [15:40] <michaelni> cbsrobot, yuv->rgb has user selectable tables [15:40] <Daemon404> since this means GBR avc decoding will always have an unneeded yuv conversion when used with e.g. swscale + avisynth [15:41] <cbsrobot> ah ff_yuv2rgb_coeffs ? [15:41] <cbsrobot> could we add xyz to that ? [15:43] <michaelni> cbsrobot, see sws_setColorspaceDetails() for the user API of setting yuv variant [15:44] <michaelni> about adding xyz, i suspect it would need some additional changes [15:44] <cbsrobot> michaelni: just found it [15:44] <cbsrobot> I think so too [15:44] <cbsrobot> I'm just digging - you know the details [15:44] <michaelni> as rgb<->yuv have some 0 and 1 in the matrix that i belive are not in xyz [15:45] <michaelni> and we probably have optimizations for all 0 and 1 multiplies [15:48] <cbsrobot> also rgb -> xyz would be useful ... [15:54] <michaelni> yes [16:35] <ubitux> would anyone be interested in a history for code coverage? [16:36] <ubitux> let's say 10 or 20 old code coverage reports [16:36] <ubitux> or it doesn't really matter [16:36] <ubitux> ? [16:39] <Compn> mike maybe, ask himmmmmm [16:39] <Compn> :P [17:18] <ubitux> Compn: done :) [17:19] <ubitux> saste: we don't have any audio filter to "buffer" a given number of samples, right? [17:20] <saste> ubitux: no, unless you count abuffer_src which is not a "filter" [17:20] <ubitux> i was wondering for the FFT visualization filter [17:21] <ubitux> because we need windows of a given number of samples [17:21] <ubitux> also, it could help user app which are running similar routines [17:22] <ubitux> the could get buffer of N samples each time, easy to deal with that doing the buffering themselves [17:22] <ubitux> any thoughts about this? [17:22] <ubitux> s/the could/they could/ [17:22] <ubitux> (this would suggest buffering with silence for the last frame) [17:22] <cbsrobot> Compn: I had another idea [17:23] <Compn> yes? [17:23] <cbsrobot> for gsoc, but maybe it's too wired [17:23] <cbsrobot> maybe someone is interested in a audio filter doing similar stuff to http://www.dolby.com/us/en/professional/software/broadcast/dolby-media-mete… [17:23] <cbsrobot> I would be at least [17:24] <ubitux> oh that would be great [17:24] <av500> BS.1770-1 [17:24] <ubitux> cbsrobot: a lot of "music information retrieval" stuff could be done in the same idea [17:24] <cbsrobot> http://www.google.com/url?sa=t&rct=j&q=itu-r%20bs.1770-1&source=web&cd=1&ve… [17:24] <ubitux> such as pitch, beat or key extraction [17:25] <cbsrobot> ubitux: why not [17:26] <cbsrobot> maybe kierank would be interested in ^ aswell [17:26] <kierank> cbsrobot: stuff exists already to do that [17:26] <kierank> libebu128 [17:26] <kierank> etc [17:28] <Daemon404> kierank, you forget the great NIH policy [17:28] Action: Daemon404 runs [17:29] <kierank> Daemon404: coming to bristol this weekend? [17:29] <Daemon404> well its my last weekend here with my the gf... [17:29] <Daemon404> weekdays work better for me :/ [17:29] <cbsrobot> kierank: you mean http://sourceforge.net/projects/r128gain/ ? [17:29] <Daemon404> sitll waiting on my google uk results too... sooo ;) [17:29] <kierank> cbsrobot: yeah and similar [17:30] <kierank> erm i'm around tomorrow and wednesday [17:30] <kierank> and friday afternoon i guess [17:30] <kierank> tomorrow and wednesday afternoons [17:30] <kierank> [16:28] Daemon404: kierank, you forget the great NIH policy --> duh [17:31] <Daemon404> kierank, oic [17:31] <Daemon404> i could do a day-trip [17:31] <Daemon404> it's a short train ride away after all [17:32] Action: Daemon404 checks train prices [17:32] <kierank> bbiab [17:32] <Daemon404> k [17:32] <cbsrobot> kierank: thanks - i wasn't aware of it .... [17:38] <kierank> ok back [17:39] <kierank> oh bleh you have to go to bristol parkway from exeter [17:39] <Daemon404> not temple meads? [17:39] <kierank> oh you can [17:39] <kierank> never mind [17:39] <kierank> temple meads is better [17:40] <Daemon404> yes [17:40] <AdamWorld> hi! [17:40] <kierank> hello [17:40] <Daemon404> kierank, well then wahts a good time to show up / what would we be doin' around bristol? [17:41] <kierank> well to be honest i was just thinking of getting a beer [17:41] <kierank> but i dunno what you want to do in bristol [17:41] <Daemon404> beer is fine :) [17:41] <AdamWorld> I run ffserver as in document and in client's command I run ffplay http://192.168.157.129:8090/test.asf but it fails to play. [17:41] <AdamWorld> what's the problem? [17:41] <AdamWorld> In other player like VLC works well. [17:42] <Daemon404> AdamWorld, i think you want #ffmpeg [17:42] <AdamWorld> Yes. [17:42] <AdamWorld> Should I ask in #ffmpeg? [17:42] <Daemon404> yea [17:43] <AdamWorld> Then What's this #ffmpeg_devel for? [17:43] <kierank> developing ffmpeg [17:43] <AdamWorld> Sorry. [17:44] <kierank> Daemon404: not entirely sure what we can do apart from drink [17:45] <kierank> and i need to think of a decent place to drink [17:45] <Daemon404> kierank, drinking is fine [17:45] <Daemon404> it's actually what i expected :P [17:45] <kierank> problem is after about 2hrs we won't have much to talk about [17:46] <Daemon404> well i should probably be back here by ~6:30pm latest [17:46] <Daemon404> or i will be murdered, etc [17:46] <kierank> well as long as it doesn't seem like a crap day trip to you [17:46] <Daemon404> nah [17:46] <Daemon404> meeting people is never a crap day trip [17:48] <saste> ubitux: sure a buffer audio filter may be useful [17:49] <ubitux> could we consider to auto insert with the help of query_formats? [17:54] <kierank> Daemon404: i think the only places open during the day will be wetherspoons [17:55] <Daemon404> lol i dont know what that is [17:55] <Daemon404> but im sure it's better than anything in canada. [17:55] <kierank> it's a big chain of pubs [17:55] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r3293b1adca 10ffmpeg/libavcodec/ (utils.c version.h): [17:55] <CIA-17> ffmpeg: lavc: set bit_rate in the decoder context, just after the init stage [17:55] <CIA-17> ffmpeg: Also move the get_bit_rate() function, in order to avoid an undefined [17:55] <CIA-17> ffmpeg: symbol reference in avcodec_open2(). [17:55] <kierank> the beer is usually not crazy expensive but the pubs are a bit corporately [17:55] <CIA-17> ffmpeg: 03Matthieu Bouron 07master * r278d6ab900 10ffmpeg/ (doc/ffprobe.xsd ffprobe.c): [17:55] <CIA-17> ffmpeg: ffprobe: report bit rate in stream description [17:55] <CIA-17> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [17:56] <kierank> corporatey* [17:56] <Daemon404> oic [17:56] <Daemon404> but better beer than bud light [17:56] <Daemon404> re: north america [17:56] <kierank> oh yes [17:56] <Daemon404> ;) [17:57] <kierank> the beer is good and usually they have nice ales [17:57] <Daemon404> lovely [17:57] <kierank> http://www.jdwetherspoon.co.uk/home/pubs/v-shed [17:57] <Daemon404> what's a preferabke time for me to show up then? [17:57] <kierank> which day? [17:57] <Daemon404> tomorrow or next [17:57] <Daemon404> i can do either [17:57] <kierank> about 2ish [17:58] <kierank> that place doesn't look too bad [17:58] <Daemon404> ok lets looky at tickets [17:58] <kierank> and is on the waterside [17:58] <Daemon404> getting around Uk is so easy... [17:58] Action: Daemon404 notes back home is has a 800 km radius of forest on all sides... [17:58] <kierank> if you want a pub closer to the station or further feel free to say so [17:58] <Daemon404> it* [17:58] <kierank> the famous ones won't be open during the day unfortunately [17:58] <Daemon404> thats fine [17:59] <Daemon404> i dont care much for "famous" [17:59] <Daemon404> i care for good beer and friends [17:59] <kierank> well the pub i had in mind is famous because it has its own beer [17:59] <kierank> but it's quite far [17:59] <Daemon404> oic [17:59] <kierank> and small so not easy to talk [17:59] <Daemon404> well ill leave it in your hands [17:59] <Daemon404> im rather impartial [18:00] <Daemon404> but i do know your bus system as confused me before [18:00] <kierank> it may be best to walk from the station [18:00] <kierank> depends on the pub i guess [18:01] <Daemon404> yea [18:02] <kierank> let's see what facebook has to say about bristol central pubs [18:04] <kierank> bbl [18:04] <Daemon404> k [18:04] <Daemon404> poke me when ya get back. [18:12] <Daemon404> an anytime return is cheaper than a scheduled one.... logic. [18:15] <Daemon404> kierank, looks like i have to go weds... tesco deilvery is tomorrow.. lol [18:15] <kierank> ok [18:17] <ubitux> is Andrey Utkin on IRC? [18:17] Action: Daemon404 books trains [18:19] <Daemon404> man, what is it with british places and charging me extra if i want to use a credit card [18:20] <av500> use her majesties huge coins [18:20] <kierank> Daemon404: really? [18:20] <kierank> you should be able to buy the ticket from whichever train company you please [18:20] <kierank> http://www.moneysavingexpert.com/travel/cheap-train-tickets#find [18:20] <Daemon404> im using thetrainline.com [18:21] <Daemon404> anyway, never seen this practice in north america [18:22] <kierank> well if you haven't finished already use one of the ones it suggests [18:22] <kierank> because thetrainline is a ripoff [18:22] <Daemon404> ok [18:29] <ohsix> how much extra, .25-.50c is pretty normal at places that only do small transactions, to offset the 3% visa or whoever their merchant is gets [18:29] <Daemon404> 3.5 pounds [18:29] <ohsix> ouch [18:30] <ohsix> on top of currency conversion? [18:30] <Daemon404> yea [18:30] <Daemon404> and yes. [18:30] <ohsix> could come from a lot of places then [18:30] <Daemon404> kierank, booked [18:30] <Daemon404> i arrive wed, ~1:30pm (cheapest), open return. [19:32] <kierank> Daemon404: ok [19:33] <kierank> all my friends suggest wetherspoons [19:34] <Daemon404> all right [19:34] <kierank> the 8 or 9 bus and get off at the big university tower [19:34] <kierank> (park street)/berkeley squre [19:34] <overflow_0f8b> hellou [19:34] <kierank> the bus should stop right outside the pub [19:34] <Daemon404> ok [19:34] Action: Daemon404 saves that on his phone [19:37] <kierank> the university tower is called the wills memorial building [19:37] <kierank> you won't miss it [19:38] <Daemon404> aight [19:59] <knoppix> Hello.I am not sure whether I should ask questions on developing libavfilter here or at #ffmpeg. [19:59] <kierank> here [20:01] <knoppix> O.k.....I had a look at http://wiki.multimedia.cx/index.php?title=FFmpeg_filter_HOWTO .However,it seems to be outdated.I am interested in learning how to add filters to libAVFilter.What should I do? [20:02] <ubitux> audio or video filter? [20:03] <knoppix> Video. [20:04] <ubitux> what kind of processing is it? modifying the video, or deleting/adding frames? [20:04] <knoppix> I am interested in both. [20:06] <ubitux> you have various kind of filters, better look at simple ones [20:06] <ubitux> vf_scale is "simple" for instance, as a content modifying example [20:06] <ubitux> (changing props too) [20:06] <ubitux> vf yadif/select/thumbnail do "selection" [20:07] <ubitux> if you're looking for filering out frames [20:07] <ubitux> you also have filters like vf_blackframe just to analyze content, and not modify it [20:08] <Compn> knoppix : ask saste about filters :) [20:08] <knoppix> I see...I will eventually understand it by studying the code,but I want documentation. [20:08] <Compn> saste should update the wiki docs ;\ [20:08] <knoppix> Ohh...I see............thanks Compn. [20:10] <knoppix> Hello saste. [20:11] <ubitux> knoppix: also, asking specific questions about the issues you expect is a good way of learning about the internals [20:11] <saste> work calls for more work (and no profit)... i should have never start to work on lavfi :/ [20:12] <ubitux> imo, more efficient that incomplete (or deprecated) doc :) [20:12] <saste> knoppix: yes i'll eventually document it (or write a tutorial) but only $gods know when [20:12] <ubitux> saste: hey, you already have nicolas & me to work on the filters to help you ;) [20:12] <saste> ubitux: yep :) [20:13] <ubitux> (or ask you for reviews) [20:13] <ubitux> (and help) [20:13] <knoppix_> O.k.........thank you. [20:13] <saste> knoppix: better thing is to ask on ffmpeg-devel if you want to contribute, or ask on libav-user if you want help for using the API [20:17] <saste> gottago now [20:18] <ubitux> cya :) [20:43] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r016c7bb762 10ffmpeg/libavformat/matroskadec.c: [20:43] <CIA-17> ffmpeg: matroskadec: remove H.264 special-case that disabled header parsing. [20:43] <CIA-17> ffmpeg: The parser was fixed so this workaround should no longer [20:43] <CIA-17> ffmpeg: be necessary. [20:43] <CIA-17> ffmpeg: This allows using stream-copy to fix files with keyframes [20:43] <CIA-17> ffmpeg: incorrectly marked. [20:43] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:44] <overflow_0f8b> took another a few hours to reverse GCC's logic and make it generate 2 cmovs :) though it still does mov edx,edx;mov ebx,ebx; ;>>>> but hey, it works [20:44] <Daemon404> this sounds painful... [20:46] <overflow_0f8b> this is probably because gcc is not able to realize ebx == ebx [20:46] Action: Daemon404 does all his development with clang :D [20:48] <overflow_0f8b> i should do a second pass on my asm output and remove mov eax, eax and similar lines [20:52] <Daemon404> why are you pokign gcc-generated asm anyway? [20:53] <overflow_0f8b> coz i like to synthesize C code instead of inline asm hacking [20:54] <overflow_0f8b> it is possible to code in C using logic [20:54] <Daemon404> not sure what you mean by synthesize [20:54] <Daemon404> so "code in such a way that you minimize gcc retardisms" [20:54] <overflow_0f8b> yes [20:54] <overflow_0f8b> but i have hit a wall now with these mov eax,eax type things [20:55] <overflow_0f8b> gcc does not realize pointer a == uintptr_t a [20:55] <kierank> those are normal, no [20:55] <overflow_0f8b> and used -O3 [20:55] <Daemon404> i think optimizing for a certain compiler is a tad silly,imho [20:56] <Daemon404> in fact, youre optimizing for that specific version [20:56] <overflow_0f8b> hoping 4.4.4 and newer generates using similatr logic... [20:58] <CIA-17> ffmpeg: 03Nicolas George 07master * r560b224f53 10ffmpeg/ (5 files in 2 dirs): libswr: allow to set custom matrices. [20:58] <CIA-17> ffmpeg: 03Nicolas George 07master * ra44b510d56 10ffmpeg/ (doc/filters.texi libavfilter/af_pan.c): af_pan: use libswr for rematrixing. [20:58] <CIA-17> ffmpeg: 03Nicolas George 07master * ra297856b8c 10ffmpeg/libavfilter/af_pan.c: af_pan: reindent after last commit. [21:23] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r6a3f1726af 10ffmpeg/ffmpeg.c: [21:23] <CIA-17> ffmpeg: Add missing av_free_packet. [21:23] <CIA-17> ffmpeg: Fixes memory leak when encoding at least with mpegvideo using the [21:23] <CIA-17> ffmpeg: new encode2 function. [21:23] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:52] <kierank> Daemon404: we could go here actually: http://en.wikipedia.org/wiki/Llandoger_Trow [22:52] <kierank> but it's more of a food place [22:53] <kierank> worth seeing apparently [23:01] <pasteeater> interesting looking place. [23:03] <pasteeater> better than the shithole across my street: http://en.wikipedia.org/wiki/Alaskan_Hotel_and_Bar [23:05] <Daemon404> kierank, cool [23:05] Action: Daemon404 must tend to his other half atm,so bbl! [23:16] <overflow_0f8b> Daemon404<< i like this style, i just leave the dirty asm scheduling and logistic thing to C, and write a C-code that can be logically translated into an ASM code [23:16] <overflow_0f8b> in most of the cases i can get away with this [23:38] <CIA-17> ffmpeg: 03Nicolas George 07master * ra923b6b8f4 10ffmpeg/libavfilter/af_pan.c: [23:38] <CIA-17> ffmpeg: af_pan: move query_format before config_props. [23:38] <CIA-17> ffmpeg: That is the logical order in which the functions are called, [23:38] <CIA-17> ffmpeg: and there is no longer any need of forward declarations. [23:38] <CIA-17> ffmpeg: 03Nicolas George 07master * reb9ba527e7 10ffmpeg/libavfilter/af_pan.c: [23:38] <CIA-17> ffmpeg: af_pan: remove the indirect function call. [23:38] <CIA-17> ffmpeg: libswresample is used in all cases. [00:00] --- Tue Feb 21 2012

1 0

ffmpeg.log.20120220
by burek 21 Feb '12

21 Feb '12

[04:36] <T3mp3sT> I use to use this cmd for xgrab. Now I am getting lossless_untrafast not found [04:36] <T3mp3sT> ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 "$outputdir"/"$1" [04:37] <T3mp3sT> Did something change? [04:37] <T3mp3sT> Was about a year ago I was using lossless_ultrafast [04:41] <T3mp3sT> Looking under /usr/local/share/ffmpeg at the presets. Seems to be some missing. [04:41] <T3mp3sT> I compiled all from git with x264 [04:41] <T3mp3sT> and the config was right afaik [04:53] <T3mp3sT> Using Debian wheezy and have xgrab before compiling. If that old cmd does not work how do I use xgrab? [04:53] <T3mp3sT> pulled all gits from today [05:21] <burek> "I'm used to" [05:21] <burek> :) [05:22] <burek> also "lossless_untrafast" does not exist [05:22] <T3mp3sT> This looks like crap ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 output.mkv [05:22] <burek> more on that [05:22] <burek> vpre is deprecated [05:22] <burek> you should type x264 --help to see what presets are valid [05:22] <burek> but iirc those are: ultrafast, faster, fast, ... [05:23] <burek> also, if you want lossless, why don't you use RLE compression [05:24] <burek> because -crf 0 and "ultrafast" don't go together not even in real life :) [05:26] <T3mp3sT> DarkShakra or what ever his was was a few years ago gave me some tips on this. Turned out there was a codec (MicroSoft?) that had a really nice color space and was darn nice. [05:27] <T3mp3sT> Yeah we used ffmpeg the files were darn big....kike 50GB [05:28] <T3mp3sT> burek so what would you recommend if you did not care at all about size and have a computer faster then God [05:29] <burek> just a sec [05:33] <burek> well [05:33] <burek> it depends on your goal [05:34] <burek> if your goal is not size nor speed [05:34] <burek> then use vcodec with uncompressed codec [05:34] <burek> ffmpeg -codecs [05:34] <burek> to see all [05:34] <burek> so you will get pretty much something like bitmap images :) [05:38] <T3mp3sT> To make a crystal clear copy of what I see on my screen. A 30" HD 1020 screen. My disks can keep up with huge files ( raid 0 OCZ Vertex 3 Max IOPS) so thats all good even with bitmap [05:38] <burek> so, what is actually your goal [05:38] <burek> i see [05:38] <burek> well, then get the software that was written for that [05:38] <burek> like camtasia [05:38] <burek> or camstudio [05:39] <T3mp3sT> Never heard of that. Only use the main repo of debian testing. I can take a look though. [05:39] <burek> its a windows software [05:39] <T3mp3sT> lol.... [05:39] <T3mp3sT> no thx [05:39] <burek> so google for something that works on linux [05:39] <burek> or spend your time having hard time with ffmpeg [05:39] <T3mp3sT> ArchLinux or Debian is OK though [05:39] <burek> saying "no thx" [05:39] <burek> :) [05:39] <burek> its your life, your choice :) [05:40] <T3mp3sT> So you saying this is not possible with this project? [05:40] <burek> I didn't say that [05:40] <burek> you can wait for somebody, who had more experience with your problem, to come online and help you, though [05:41] <T3mp3sT> prolly should go to bed. Its 5:41am here in Sweden. Been coding since...emmm. I think yesterday...or maybe the day before =P [05:42] <burek> get some sleep or read on [05:42] <burek> wikipedia about polyphasic sleep [05:42] <burek> it might help :) [05:42] <T3mp3sT> or some big fat roofies so I fall asleep on my keyboard. [05:43] <burek> :))) [05:43] <burek> or that :) [05:43] <T3mp3sT> Later. I work this out 2mRO [05:43] <T3mp3sT> Cya [05:43] <burek> :beer: :) [05:43] <burek> o/ [06:19] <JCDenton> I'm getting an error when trying to add a watermark to my video using ffmpeg. I'm using the same command I used on another box, and as far as I remember, it always worked without a problem. The error I get is: [06:19] <JCDenton> [NULL @ 0x14a0540] Unable to find a suitable output format for '[watermark]' [06:19] <JCDenton> [watermark]: Invalid argument [06:19] <JCDenton> ffmpeg -threads 8 -i input.mp4 -vf movie=watermark-small.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out] output.mp4 [06:19] <JCDenton> Is what I'm running. [06:38] <JCDenton> Any idea why? [07:31] <grepper> JCDenton: "movie=...[out]" quoted ? [07:32] <grepper> should be if it isn't [07:43] <JCDenton> grepper: Oh wow, I'm an idiot, thanks [07:43] <qop> is it possible to get ffmpeg to use an iso or dvd as input? [07:46] <grepper> np [07:47] <JCDenton> DVD maybe, ISO I'm not so sure. [07:47] <qop> how? cant find out [07:48] <qop> and if dvd is possible I can just mount the iso [07:48] <qop> ot trick through script [09:32] <grepper> qop: you could take a look at http://godromo.com/gmt/vgtmpeg [09:40] <AdamWorld> Hi! When I update FFmpeg git-6cb2085 32-bit Dev (Latest) but "'UINT64_C': identifier not found" error occured. what's the problem. [09:59] <qop> grepper: thanks!! ^^ [11:34] <elkng> how to procees only first 2 minutes of file ? there is and option "-ss" to seek to position, is there similar option ? [11:40] <kriegerod> should i rescale pts/dts of incoming AVPacket to decoder timebase before decoding, or i can just rescale the decoded frame pts to encoder timebase after decoding? [11:40] <kriegerod> elkng, -t [12:32] <qop> ffmpeg still encoding after 3 days :P [12:32] <qop> ~23000 files (its 50%) [12:33] <qop> I have a flight Wednesday! wonder if it will end [12:33] <qop> \in time [13:10] <al4711> Hi all [13:12] <al4711> I try to insert some frames into a movie which was calculated by previous frames => slow motion effect. [13:12] <al4711> I have try to search with google but I'am not sure if I search for the right syntax "ffmpeg slow motion" [13:13] <al4711> thanks for help [14:24] <elkng> there is an example in "man ffmpeg": "ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi", why "snatch" ? why isnt there "lord_of_the_rings.avi" or "star_wars.avi" ? [14:25] <sacarasc> Maybe the devs liked Snatch. [14:29] <ubitux> originally committed in 2002 [14:30] <ubitux> (3e12d48a89e3385a7da76cc086333359911c2131) [14:30] <ubitux> not much about it, maybe in the ml [15:14] <MindSpark> hello, anyone into some serious debugging? [15:15] <MindSpark> bind(port 8090): Address family not supported by protocol [15:15] <MindSpark> that's the output I get when I start ffserver [15:15] <MindSpark> does anyone know what causes this? [15:19] <kriegerod> first of all tell us what actually you're doing [15:19] <ubitux> we don't have a maintainer for ffserver at the moment [15:19] <ubitux> so i don't think anyway has a gdb open on ffserver now :) [15:20] <MindSpark> I am trying to convert an rtmp stream to rtsp [15:21] <MindSpark> I got ffserver to run on different machines. For some odd reason, I just tried on a different one (the computer I am on right now) and I get this error [15:21] <MindSpark> however even on the other machines, that I got it to run on, I failed to correctly transcode rtmp to rtsp [16:01] <stryjan_> is there a way to use ffmpeg only to copy metadata between two files without any conversion? [16:02] <djook> hye guys [16:03] <djook> I try to use loop input and drawtext at the same time but drawtext doesnt work [16:04] <djook> I loop a single jpeg picture for video source [16:04] <djook> and would like to insert text aswell [16:05] <djook> http://pastebin.com/yAbMELEm [16:05] <djook> here is my command line [16:06] <djook> video comes out without the text [16:06] <djook> is itnormal behaviour ? [16:07] <djook> any body awaken? [16:07] <djook> 16h00 in France ... [16:08] <cbreak-work> hmm [16:10] <djook> I feel it is not taking the filter into account while looping the image [16:10] <djook> but doesnt complain [16:10] <cbreak-work> have you moved the filter after the input? [16:10] <djook> seems logicalanyway [16:10] <cbreak-work> I think the order of arguments in ffmpeg is very important [16:10] <djook> yes I tried to change [16:12] <djook> wherever I placed it ,itdidnt appear on the final output [16:13] <djook> I'll try with a non looping input [16:15] <cbreak-work> maybe your whole drawtext is incomplete [16:15] <cbreak-work> the one that works for me specifies a font [16:18] <djook> it complains when taken into account too [16:18] <djook> and asks for fontfile [16:19] <djook> when using a non loop input [16:20] <djook> so I do it in two times [16:21] <djook> first time,from loop input without drawtext, then again with drawtext from the video file I just obtained [16:26] <djook> okguys see you [16:29] <fj_> hi [16:29] <fj_> I am using the old version of ffmpeg but can' t use pstrcpy [16:29] <fj_> whats up with that [16:45] <Freakshow> anyone have examples of how I would multiple bitrates out of a single input? [16:45] <Freakshow> looking at this as an example [16:45] <Freakshow> http://smorgasbork.com/component/content/article/35-linux/92-ffmpeg-the-mot… [16:45] <Freakshow> but was curious how applicable it was. [16:46] <Freakshow> given recent release changes... [16:47] <Freakshow> holy crap... just saw the date on that... [16:48] <Freakshow> nevermind, I'll cross reference and see what I can come up with today [16:48] <Freakshow> and then ask again [16:48] <kriegerod> is there any usage example of av_asrc_buffer_add_samples() on earth? [16:51] <kriegerod> Freakshow, that example is over-complicated. You should try "ffmpeg -i in.file -b:v 100k 100k.file -b:v 200k 200k.file" - you get the idea [16:51] <kriegerod> -b is for bitrate [16:51] <kriegerod> input file options go before -i option, output file options go before output file name [17:00] <Freakshow> kriegerod: what if I want to specify multiple rtsp outputs? does the same principle apply? [17:01] <kriegerod> yes [17:02] <Freakshow> hmm... is the simple notion of the additional -b:v 200k indicating that it's another output? [17:03] <Freakshow> so like -b:v 100k -c:v libx264 -profile:v baseline -f rtsp rtsp://output -b:v 200k -c:v libx264 -profile:v baseline -f rtsp rtsp://output [17:03] <kriegerod> simply options are split by places where input and output file names/URLs are located [17:04] <kriegerod> your line is right [17:04] <Freakshow> ok ok [17:04] <Freakshow> cool [17:04] <Freakshow> I'll give that a shot, thanks! [17:04] <MindSpark> I think I just missed the beginning of this conversation [17:04] <MindSpark> can someone pastebin/send it to me? [17:05] <Freakshow> sure, hang on [17:05] <MindSpark> this is probably very similar to what I am trying to achieve [17:13] <Freakshow> done [17:43] <AdamWorld> I run ffserver as in document and in client's command I run ffplay http://192.168.157.129:8090/test.asf but it fails to play. [17:43] <AdamWorld> what's the problem? [17:43] <AdamWorld> In other player like VLC works well. [17:52] <MindSpark> AdamWorld, what OS? [17:52] <MindSpark> what is the output? [17:55] <MindSpark> can someone tell me how to convert this rtmp stream to rtsp? [17:55] <MindSpark> rtmp://aljazeeraflashlivefs.fplive.net/aljazeeraflashlive-live/aljazeera_ar… [17:58] <Velmont> Hmm. FFmpeg doesn't support old Ogg Vorbis files. I tested with ogg123, and that works. -- Is it intended? [17:59] <sacarasc> Output to a pastebin, etc. [18:03] <ubitux> Velmont: a sample would be welcome too [18:03] <Velmont> http://dpaste.org/3rNO0/ [18:04] <ubitux> [mp3 @ 0xf0a1e0] [18:04] <ubitux> what :D [18:04] <Velmont> looked kinda strange [18:04] <ubitux> can you share the file? [18:05] <Velmont> http://odin.s0.no/tmp/Marte%20Wulff%20-%2001%20-%20Run.ogg [18:05] <ubitux> btw, you are using a pretty old version [18:05] <ubitux> plays fine here [18:05] <ubitux> try upgrading your ffmpeg [18:06] <Velmont> Hmm. Strange, - it worked in older versions, - maybe the one Ubuntu 11.04 has is just broken. [18:06] <sacarasc> Also, using libav, not ffmpeg. [18:06] <Velmont> sacarasc: ffplay? [18:07] <Velmont> Oh, yes, it uses it as well. [18:07] <sacarasc> ffmpeg was forked a few months back, Ubuntu ships the fork. [18:07] <ubitux> Velmont: i'm testing with ffplay from upstream, and it works [18:07] <ubitux> and indeed this is libav, not ffmpeg [18:08] <ubitux> (they still use the ff* tools name? :/) [18:08] <ubitux> seriously& :( [18:08] <ubitux> Velmont: nice music btw ;) [18:08] <sacarasc> I think that's for backwards compatibility. [18:08] <ubitux> or backward stabbing! [18:09] <Velmont> ubitux: Yes, I like her. Suddenly wanted to listen to some old stuff, and in Ubuntu's fork it didn't work. [18:09] <ubitux> https://launchpad.net/~jon-severinsson/+archive/ffmpeg [18:09] <ubitux> maybe check this out [18:10] <ubitux> if it still does not work, i can only suggest to update to upstream [18:11] <Velmont> ubitux: Well, I've been thinking about switching back to arch linux. Was always easier to get upstream stuff etc then. [18:11] <Velmont> But sysadmin had installed Ubuntu on my work computer when I came here already, so I just went with the flow. [18:12] <ubitux> :) [18:14] <ubitux> i'm not sure changing its OS because of a packaging issue is the correct answer to the problem; maybe you should complain on lp (or even better do sth more productive) about that :p [18:14] <ubitux> though, archlinux doesn't have (yet?) the "libav issue" :) [18:37] <MindSpark> I can confirm [18:37] <MindSpark> ubuntu's ffmpeg in 11.04 is just broken [18:37] <MindSpark> and for me neither ffserver nor ffmpeg can connect to anything [18:56] <Freakshow> MindSpark: is that stream even working atm? [18:57] <Freakshow> rtmp://aljazeeraflashlivefs.fplive.net/ [18:58] <Freakshow> nm... mplayer got it [19:15] <twinther> hi, I have a problem with a rtsp stream where the video freeze after a while (~80 secs) [19:16] <twinther> this is logged just before the stream stops: [rtsp @ 0x20658e0] CSeq 9 expected, 1 received. [19:17] <twinther> full log here http://pastebin.com/zTrhxs7N [19:26] <MindSpark> Freakshow, sorry for the delay, was afaik [19:26] <MindSpark> Freakshow, yes, it should be working. I am trying to restream it over rtsp, but ffserver is not playing it as far as vlc is loggin [19:26] <MindSpark> logging [19:28] <Freakshow> no worries, I was trying to determine if the source feed itself was actually up [19:29] <Freakshow> flowplayer wasn't picking it up [19:29] <Freakshow> sorry, my test flowplayer [19:29] <MindSpark> :) [19:29] <Freakshow> but mplayer latched onto it quickly [19:29] <Freakshow> can you paste the actual ffmpeg command line you're using? [19:30] <MindSpark> yes, playing it directly hasn't been a problem for me. I just don't know what to do to get ffserver to restream it [19:31] <MindSpark> ffmpeg -i rtmp://aljazeeraflashlivefs.fplive.net/aljazeeraflashlive-live/aljazeera_ar… -vcodec libx264 -b 300000 -vpre default -vpre baseline -sameq -s 480x270 -acodec copy -re -f rtsp rtsp://127.0.0.1/jazeera.sdp [19:31] <MindSpark> this is one way I read about that should work [19:32] <MindSpark> but I get Could not write header for output file #0 (incorrect codec parameters ?) [19:32] <Freakshow> which version of ffmpeg? [19:32] <Freakshow> looks like e 0.7.1 [19:32] <Freakshow> but < 0.9.1 [19:33] <MindSpark> 0.6.4-4:0.6.4 [19:34] <Freakshow> ugh... can you update to something newer? head perhaps? [19:34] <MindSpark> that's the binary I got from ubuntu's package manager, do you think that will solve the problem? [19:35] <Freakshow> I mean, I understand if not... but yikes, I had some issues with rtmp/rtsp output on the older versions [19:35] <MindSpark> it says built in jan 2012 [19:35] <Freakshow> not sure about ubuntu's main line [19:35] <MindSpark> Freakshow, ok, makes sense I guess, maybe I need to recompile [19:35] <Freakshow> you would leave it be and install ffmpeg-devel [19:35] <Freakshow> just to test [19:35] <Freakshow> if you have an alt machine to use, that might be worthwhile [19:36] <Freakshow> devel on bsd is 0.9.1 [19:36] <Freakshow> not sure what ubuntu devel port is at though [19:36] <Freakshow> pasteeater may know [19:38] <MindSpark> alright, I'll give this a shot [19:40] <MindSpark> I'll just try to compile on my local machine and see from here [19:40] <MindSpark> but does the above line look valid? [19:42] <Freakshow> yes, that seems valid... [19:43] <MindSpark> so I don't need to run ffserver to restream? [19:43] <MindSpark> using rtsp://link as an ouput would be enough to open up rtp/rtsp ports? [19:43] <Freakshow> well... I presume you have ffserver on the local machine [19:43] <Freakshow> you may need to declare a port [19:44] <Freakshow> but I'm not sure... I stream to wowza, not ffserver [19:44] <Freakshow> but that does seem valid from what I recall [19:44] <Freakshow> with the newer version you'll probably want to force rtsp over tcp [19:44] <Freakshow> it defaults to udp, and it's not pretty [19:46] <MindSpark> Freakshow, yes, do you know how this is done? [19:46] <Freakshow> sure... -f rtsp -rtsp_transport tcp rtsp://rtsp.server.com:1935/my/stream/path.sdp [19:47] <Freakshow> would be what I would use [19:47] <Freakshow> and which is also 'what' I use [19:47] <Freakshow> ;) [19:49] <Freakshow> but again, this is to wowza... [19:49] <MindSpark> :) alright, sounds good. Is wowza hosted somewhere or do I download and install? [19:52] <Freakshow> oh boy... that's an entirely different ball of wax [19:52] <Freakshow> try with ffserver first [19:52] <Freakshow> and yes, you DO need to specify the protocol to localhost [19:52] <MindSpark> yea, I am tight on time, I am supposed to find a solution to this by the end of the week [19:53] <MindSpark> and I've already spent a week trying to get things done with ffserver [19:53] <Freakshow> well, spend the hour or so getting the newer version installed and see where that gets you [19:55] <MindSpark> alright, thanks. I'll be back later I guess [19:55] <MindSpark> Freakshow, what's your timezone? [19:55] <Freakshow> PST [19:55] <Freakshow> you? [19:56] <MindSpark> EET [19:56] <MindSpark> I guess that's like 9 hours apart [19:56] <Freakshow> looks like [19:59] <MindSpark> I am going to install a new version on some amazon cloud instance [20:00] <MindSpark> actually, 0.8-4:0.8 [20:00] <MindSpark> is that an "ok" version? [20:02] <Freakshow> well, I'm not necessarily sure that the previous version was not ok, but it's always best to use closer to head as possible [20:02] <Freakshow> security fixes, etc. [20:37] <pasteeater> Velmont: there is a ubuntu compile guide link in the channel topic [20:46] <pasteeater> i finally found "open command window here" in win7. [21:00] <ago> Hi folks [21:02] <ago> I have just a pair of doubt about cpu/arch optimizations [21:03] <ago> When starts configure phase, I see: ARCH x86 (core2) but this is an x86_64 environment, it is ok or should be x86_64/amd64/other? [21:04] <ago> I I enable cpudetenction, in configure phase I see always --cpu=host, is expected? [21:04] <ago> *If I enable [21:07] <burek> ago [21:07] <burek> can you please use pastebin.com, to show your command line and its output? [21:10] <ago> burek: as you can see I didn't make it manually, is a gentoo environment http://paste.pocoo.org/show/554139/ [21:10] <ago> and I see --cpu=host when I use ffmpeg [21:11] <burek> does it work? [21:12] <ago> burek: yes [21:13] <ago> burek: there is a list of argument that --cpu accepts? [21:13] <ago> e.g. as -march core2 corei7 prescott etc. [21:13] <burek> I don't see --cpu in your paste? [21:13] <burek> just type ./configure --help [21:13] <burek> you'll get all the info you need I guess [21:14] <ago> burek: http://paste.pocoo.org/show/554143/ [21:14] <cbreak> trying to cross compile? [21:14] <ago> no [21:14] <cbreak> did you compile the app? [21:14] <ago> help does not show what I need [21:14] <ago> cbreak: yes [21:14] <cbreak> then use file [21:14] <cbreak> file ffmpeg [21:14] <cbreak> it should tell you which architectures are in the binary [21:15] <abec0> Hi there. I gott this nice command line that records my x11 actions, but I'd like to stream this video instead. First on LAN, then on the Internet. [21:15] <burek> --cpu=CPU select the minimum required CPU (affects... [21:15] <burek> it does show [21:15] <cbreak> well... at least it does on OS X :) [21:16] <abec0> So if someone has a link to a working example, I would really like to hear from it. [21:16] <burek> abec0, instead of output file, put -f mpegts udp://dest.ip.addr:port [21:17] <abec0> burek: That would stream to one host on the LAN ? [21:17] <ago> cbreak: no, I mean a different thing [21:17] <burek> abec0, yes, for example to a vlc, which can act as a http server [21:17] <burek> to rebroadcast it [21:17] <cbreak> ago: what does it show? [21:18] <ago> cbreak: when you use --enable --disable you can activate or not an 'option' but when you have --foo=bar I think you should have a list of supported arguments that can be $bar [21:18] <ago> where I can see the syntax to pass to --arch options? [21:18] <ago> e.g [21:18] <abec0> burek: OK. So that would be vlc that does the broadcasting. I should go check the man page to understand what mpegts means. [21:19] <cbreak> ago: arch accepts things like ppc [21:19] <ago> cbreak: you know it but i'm asking where is documented ;) [21:19] <cbreak> Don't know. I read the configure script [21:20] <ago> cbreak: e.g for ppc could be obvious, but for amd64, it can be: amd64, em64t, x86_64 [21:20] <ago> and there is no explanation about it [21:20] <cbreak> I've only used armv7, armv5, i386 and the default [21:20] <burek> mpeg-ts = transport stream, abec0, but you can also use ffserver [21:20] <burek> its just the matter of choice [21:20] <cbreak> ago: but you said you don't want to cross compile [21:20] <cbreak> so why do you want to change it? [21:21] <abec0> ffserver. I'll go do some reading/testing and report back. Thanks. [21:21] <burek> :beer: :) [21:22] <abec0> That might compromise the reading, but... ;-). [21:26] <barhom_> I'm looking for a developer that is available for hire that has good knowledge of transcoding video, mainly from mpeg2-ts to h264-ts. Im wishing to build some "dynamic" transcoder that would serve the transcoded video+audio as a HTTP stream. What I mean with dynamic is that whilst there are users watching the stream, then transcode, when last one quits then stop transcoding and free up CPU power. One could use lib curl (f [21:26] <barhom_> fetching HTTP mpeg2-ts), transcode with libavcodec (h264) and then serve the stream with libmicrohttpd [21:27] <barhom_> If there is anyone that is interested please private message me so we can discuss it in more detail. If anyone can refer me to someone else that isn't on this IRC, please do that as well. [22:10] <PeraZdera1> Hi [22:12] <PeraZdera1> I am trying to find out why ffmpeg does not 'like' some switches, although I compiled the last git version [22:12] <abec0> burek: about «instead of output file, put -f mpegts udp://dest.ip.addr:port», I tried -f localhost:1234, but can't find a way to read the thing. [22:14] <PeraZdera1> here is pastebin http://pastebin.com/BMF711ix [22:15] <abec0> Well I tried -f mpegts udp://localhost:1234, to be precise... but don't know if I should do something before with vlc, or after, or what I should do. [22:16] <pasteeater> PeraZdera1: is there a reason you are not using the libx264 encoding presets? declaring each x264 option is not supported. [22:18] <PeraZdera1> pasteeater: Well thats another "problem", all the tutorials I have been reading does state that there are .ffpresets files, but I can only find libx264-ipod320.ffpreset and libx264-ipod640.ffpreset in ffmpeg presets directory [22:19] <burek> open vlc and go to media - open network stream and type: udp://@:1234 [22:20] <pasteeater> PeraZdera1: they are now "internal" rahter than text files. [22:21] <PeraZdera1> pasteeater: Thank you for your answer, can you tell me then is this valid anymore? https://sites.google.com/site/linuxencoding/x264-ffmpeg-mapping [22:21] <PeraZdera1> pasteeater: I mean can I create my own preset files? [22:22] <pasteeater> that's a waste of time, generally [22:23] <PeraZdera1> Ok, thank you very much for your answers [22:23] <pasteeater> PeraZdera1: a good start would be... [22:23] <pasteeater> ffmpeg -re -i udp://230.0.0.7:5000 -r 25 -c:v libx264 -preset slow -level 30 -b:v 1000k -c:a libfaac -ar 44100 -b:a 96k -f flv rtmp://192.168.1.10/ffmpeg [22:23] <pasteeater> -preset slow contains most of the options you previously were using [22:24] <pasteeater> -threads 0 is now automatically applied, so you don't need that anymore [22:25] <PeraZdera1> Aha, I will try to find which options are in slow profile [22:25] <PeraZdera1> Thank you :) [22:26] <pasteeater> note that there is a difference between presets and profiles [22:26] <PeraZdera1> Huh? [22:27] <pasteeater> see x264 --fullhelp for a list of options (in addition to default options) that are applied for each preset [22:27] <pasteeater> a preset is not a profile. there is a -preset option and a -profile option [22:28] <pasteeater> if you want to overwrite any preset options then see "ffmpeg -h" under the "libx264 AVOptions" section [22:29] <pasteeater> ...for a list of libx264 specific options. [22:29] <abec0> burek: Output #0, mpegts, to 'udp://localhost:8512' [22:31] <abec0> But vlc says: [22:31] <abec0> [0x8fcd47c] main stream error: cannot pre fill buffer [22:31] <abec0> When I use STOP. Using PLAY did not play a thing. [22:35] Action: abec0 regrets screencasting is not as simple as screenshooting... [22:35] <abec0> I'll try again later. [22:36] <barhom_> (out of topic from you guys) the output ramp:// to what media server are you posting the transcoded stream to? [22:36] <barhom_> *rtmp:// [22:37] <pasteeater> screencasting to justin.tv isn't too hard [22:40] <barhom_> btw, does anyone know if you can pipe the output from ffmpeg to vlc so that you can stream it from an http link, or should I use something else? [22:42] <barhom_> curl -s http://10.0.0.101:2004/bysid/5 | ffmpeg -i - -acodec libfaac -b:a 96k -vcodec libx264 -b:v 900k -preset superfast -threads 4 -bt 2500 -qcomp 0.7 -deblockalpha 2 -deblockbeta 2 -deinterlace -f mpegts 1.ts < something like this, but instead of saving to 1.ts make it available at http://something/1.ts without saving to file [22:50] <PeraZdera1> pasteater: what is the best aac encoder for ffmpeg? I can see that many people say that internal is still unstable and that is best option to use libfaac, but also it is very slow. Does libaac have some bugs when very little people use it or it is something else? [22:53] <pasteeater> PeraZdera1: i'm not sure about any libfaac bugs. [22:54] <pasteeater> you should try the interal encoder and if it sounds good enough for you then use it. [22:54] <pasteeater> there is also libvo_aacenc and libaacplus, so you have 4 AAC type encoders to choose from. [22:55] <PeraZdera1> pasteeater: and libaacplus nobody use? [22:56] <pasteeater> it's probably good for low bitrates. i've never used it. [22:56] <PeraZdera1> pasteeater: I see, I am just looking for the most stable encoder, as I am trying to transcode live stream and stability is pretty much everything [22:57] <PeraZdera1> pasteeater: Is it ok if I use daemontools for ffmpeg monitoring? I mean I would like in a case of crash to start ffmpeg again [22:58] <pasteeater> i don't know. i have no experience with that. [22:59] <PeraZdera1> tnx, you helped me very much [22:59] <PeraZdera1> and sorry for bothering :) [22:59] <pasteeater> i know ffmpeg has the -d option to enable "daemon mode", but i know nothing of it and there is little documentation as far as i know [22:59] <pasteeater> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=39aafa5ee90e10382e2f991e4d… [23:03] <PeraZdera1> well, it works, thats for sure, I can see the stream [00:00] --- Tue Feb 21 2012

1 0

ffmpeg-devel.log.20120219
by burek 20 Feb '12

20 Feb '12

[00:00] <Compn> mike(a)multimedia.cx [00:00] <Compn> hes not on irc [02:00] <ubitux> ok i think i automated the coverage stuff [02:19] <CIA-48> ffmpeg: 03Martin Storsjö 07master * r5302633919 10ffmpeg/libavformat/movenc.c: [02:19] <CIA-48> ffmpeg: movenc: Write the unknown duration as 64 bit fields in ismv [02:19] <CIA-48> ffmpeg: This is required for the files to play back properly in [02:19] <CIA-48> ffmpeg: windows media player. [02:19] <CIA-48> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [02:19] <CIA-48> ffmpeg: 03Martin Storsjö 07master * r99a357f4c5 10ffmpeg/libavformat/movenc.c: [02:19] <CIA-48> ffmpeg: movenc: Write track durations with all bits set if duration is unknown [02:19] <CIA-48> ffmpeg: According to 14496-12, the duration should be all 1s if [02:19] <CIA-48> ffmpeg: the duration is unknown. This is the case if writing a moov [02:19] <CIA-48> ffmpeg: atom without any samples described in it (e.g. as in ismv files). [02:19] <CIA-48> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [02:19] <CIA-48> ffmpeg: 03Martin Storsjö 07master * r89baf35996 10ffmpeg/libavcodec/dct-test.c: [02:19] <CIA-48> ffmpeg: dct-test: Remove a stray declaration of a nonexistent function [02:19] <CIA-48> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [02:19] <CIA-48> ffmpeg: 03Martin Storsjö 07master * r0733375059 10ffmpeg/libavcodec/dct-test.c: [02:19] <CIA-48> ffmpeg: dct-test: Add the missing ff_ prefix to the altivec functions [02:19] <CIA-48> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [02:19] <CIA-48> ffmpeg: 03Ronald S. Bultje 07master * rc0994e39d7 10ffmpeg/libavcodec/mpc7.c: mpc7: assign an error level + context to av_log() msg. [02:19] <CIA-48> ffmpeg: 03Ronald S. Bultje 07master * r9d3050d3e9 10ffmpeg/libavcodec/wmadec.c: [02:19] <CIA-48> ffmpeg: wma: don't return 0 on invalid packets. [02:19] <CIA-48> ffmpeg: Return 0 means "please return the same data again", i.e. it causes an [02:19] <CIA-48> ffmpeg: infinite loop. Instead, return an error. [02:19] <CIA-48> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [02:19] <CIA-48> ffmpeg: CC: libav-stable(a)libav.org [02:19] <CIA-48> ffmpeg: 03Kostya Shishkov 07master * r6d702dc072 10ffmpeg/libavcodec/proresenc.c: [02:19] <CIA-48> ffmpeg: proresenc: force bitrate not to exceed given limit [02:19] <CIA-48> ffmpeg: Apple ProRes Format Specifications mentions target data size for every frame, [02:19] <CIA-48> ffmpeg: so make sure frame meets it. This also allows encoder to demand much smaller [02:19] <CIA-48> ffmpeg: packet sizes for output. [02:19] <CIA-48> ffmpeg: 03Martin Storsjö 07master * r3eec23f3cd 10ffmpeg/libavformat/ (isom.h mov.c movenc.c): [02:19] <CIA-48> ffmpeg: mov: Use defines for trun flags [02:19] <CIA-48> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [02:19] <CIA-48> (64 lines omitted) [02:21] <CIA-48> ffmpeg: 03Clément BSsch 07master * rf075537561 10ffmpeg/doc/examples/filtering.c: examples/filtering: fix implicit declarations and function mis-usage. [02:21] <CIA-48> ffmpeg: 03Clément BSsch 07master * rb117c39237 10ffmpeg/doc/examples/filtering.c: examples/filtering: fix deprecated function calls warnings. [02:21] <CIA-48> ffmpeg: 03Clément BSsch 07master * r2b97be5d57 10ffmpeg/doc/examples/filtering.c: examples/filtering: fix unused variable and return value in open_input_file(). [02:21] <CIA-48> ffmpeg: 03Clément BSsch 07master * r624c413d55 10ffmpeg/doc/examples/Makefile: examples: add -Wall to CFLAGS. [03:25] <durandal_1707> looks like mono decoding for pam is broken [03:27] <michaelni> how can i reproduce it ? [03:27] <durandal_1707> michaelni: easy, pam encoder for mono is not broken [03:28] <Compn> durandal_1707 : michaelni wants to know command line that fails [03:28] <Compn> durandal_1707 : so ffmpeg -i file.wav -options... [03:28] <durandal_1707> create mono file: ffmpeg -i image -pix_fmt monob /tmp/test.pam [03:28] <Compn> so he can reproduce and fix problem [03:28] <durandal_1707> play it with ffplay [03:29] <durandal_1707> file is correctly displayed with display (ImageMagick) [03:33] <michaelni> is this a regression ? [03:34] <durandal_1707> dunno [03:37] <durandal_1707> michaelni: lol, something is weird, monow for xwd is using whole CPU time when using ffplay to display created image, gimp shows it just fine [03:38] <durandal_1707> michaelni: so i think it is bug in sws, and it is regression [03:39] <j-b> michaelni: FYI, I have pushed the postproc.git repo on git.v.o [03:41] <durandal_1707> michaelni: something is really screwed up, ffprobe is using whole CPU too [03:50] <durandal_1707> michaelni: nah, it is not sws [04:01] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * re98a76d3c2 10ffmpeg/libavcodec/h264.c: [04:01] <CIA-48> ffmpeg: h264: use av_fast_padded_malloc() [04:01] <CIA-48> ffmpeg: This zeros all the memory once and avoids valgrind warnings. [04:01] <CIA-48> ffmpeg: alternatively the warnings could be suppressed. [04:01] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:01] <CIA-48> ffmpeg: 03Paul B Mahol 07master * rb6a08665d6 10ffmpeg/libavformat/img2.c: [04:01] <CIA-48> ffmpeg: img2dec: add im32 extension, sunrast with 32bit depth [04:01] <CIA-48> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:01] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:22] <michaelni> j-b, and others, are there any bugreports about postproc ? [04:22] <michaelni> i remember some people flaming but i havnt found any bug reports on our tracker [04:22] <j-b> not that I know of. [04:22] <j-b> complaints were that it was useless [04:22] <j-b> not bus [04:22] <j-b> bugs [04:23] <j-b> http://git.videolan.org/?p=libpostproc.git;a=summary . If you do not have write access in it, please tell me. [04:47] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * r5d468edb2e 10ffmpeg/libavcodec/pamenc.c: [04:47] <CIA-48> ffmpeg: pamenc: move allocation down. [04:47] <CIA-48> ffmpeg: This way the size that is calculated is known and can be used in a following [04:47] <CIA-48> ffmpeg: commit. [04:47] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:47] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * rd3f81a6700 10ffmpeg/libavcodec/pamenc.c: [04:47] <CIA-48> ffmpeg: pamenc: allocate the correct packetsize for mono. [04:47] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:54] <durandal_1707> michaelni: that fix is wrong [04:55] <durandal_1707> encoding is fine, decoding is broken, you just broke encoding [05:13] <overflow_0f8b> to branch or not to branch? 2 memory writes, or one jump above them ? [05:16] <durandal_1707> michaelni: actually that 2 last commits could not break pam mono encoding but i dont see what they really fix [05:32] <michaelni> they allocate a bigger bitstream buffer [05:33] <durandal_1707> but that should not really be needed [05:33] <durandal_1707> and that have nothing to do with broken decoding [05:33] <michaelni> we dont support black&white PAM decoding [05:34] <durandal_1707> that is funny considering that pbm works [05:36] <durandal_1707> michaelni: how you got crash? [05:41] <michaelni> accesses over end of array in valgrind [05:41] <michaelni> iam not 100% if i tried without valgrind [05:47] <durandal_1707> that n change for mono does nothing, except increasing packet size [05:48] <durandal_1707> heh, your change indeed fix bug [05:48] <durandal_1707> for mono [05:51] <durandal_1707> but that branch for mono encoding is brain dead [05:51] <michaelni> i got mono decoding working [05:52] <durandal_1707> michaelni: make first shure that pam mono encoding is actually correct [05:53] <michaelni> imgmagick can decode it [05:53] <michaelni> ive not found anything else yet [05:53] <durandal_1707> pam = 300k vs pbm = 37 kb [05:54] <michaelni> yes iam aware of that [05:54] <michaelni> when iam happy with my patch ill post it to ffmpeg-dev [05:54] <durandal_1707> what happened to their brain..... [05:54] <overflow_0f8b> i need to swap 2 variables if a certain condition is met, but cmov does not have a mem,reg version ;< [05:58] <durandal_1707> michaelni: there is no nice way to swap bits in byte? [05:59] <michaelni> if its 1 byte then a LUT can be used as a generic solution [06:00] <michaelni> if some specific reshuffling is done there might be a nicer way [06:00] <durandal_1707> lut is what? [06:00] <michaelni> look up table [06:00] <durandal_1707> where it is? [06:01] <michaelni> outputbyte = table[inputbyte] [06:01] <michaelni> table is 256 entries [06:01] <durandal_1707> i need to generate one? [06:02] <michaelni> well if you want to use that suggestion yes, but i dont even know what you are tryig to do exactly so maybe the idea is nit the best [06:03] <durandal_1707> i need to swap bits in byte (one byte for now) it is for xbm (X BitMap) [06:03] <michaelni> overflow_0f8b, one random way is set X to 0 or var0-var1; var0 -= X; var1 += X [06:04] <durandal_1707> the way how mono plane is stored is incompatible(and brain dead) with sws [06:04] <michaelni> what does "swap" do exactly? [06:05] <durandal_1707> 00000101 => 10100000 [06:06] <michaelni> sounds like av_reverse[] [06:06] <overflow_0f8b> #define __swap__( a1, a2 ) { \ [06:06] <overflow_0f8b> tmp = (a1) ); \ [06:06] <overflow_0f8b> (a1) = (a2); \ [06:06] <overflow_0f8b> (a2) = tmp; \ [06:06] <overflow_0f8b> } [06:06] <overflow_0f8b> swap means exchange [06:07] <overflow_0f8b> and i can do it, i'm concerned about the jump penalty. [06:08] <overflow_0f8b> i can do it without jump if i do 2 unnecessary memory writes ;/ [06:08] <overflow_0f8b> memory writes are slow too (compared to nothing) [06:09] <michaelni> a1 & a2 are in memory ? [06:09] <overflow_0f8b> yes [06:09] <overflow_0f8b> it is data [06:10] <michaelni> cant you just change the algorithm to not need any swap at all ? [06:11] <michaelni> i mean no conditional swap at least [06:11] <overflow_0f8b> hmm, i'm open for suggestion on how to swap 2 variables in memory without swapping them [06:12] <overflow_0f8b> so, if i do swap the 2 variables, it's cool, but if the swap condition is not met, then it does 2 unnecessary memory writes [06:12] <overflow_0f8b> or one jump above. [06:13] <michaelni> overflow_0f8b, what is it that you are doing that needs these condiitonal swaps ? [06:13] <overflow_0f8b> michaelni<< can you suggest another method to let the first operand be the smaller one, and second be the larger ? [06:14] <overflow_0f8b> #define __minmax0__( min, max ) ( if ( (min) > (max) ) __swap__( (min), (max) ) ) [06:14] <durandal_1707> michaelni: thanks [06:14] <michaelni> durandal_1707, np [06:15] <michaelni> overflow_0f8b, what is that minmax0 used in ? [06:15] <overflow_0f8b> a sorting algorithm? [06:16] <overflow_0f8b> michaelni<< btw i believe a table is not needed for the bitswapping. [06:17] <michaelni> use radix sort [06:19] <michaelni> that is unless the array is very small or nearly sorted already [06:34] <overflow_0f8b> michaelni<< will you recommend a table for swapping all 64 bits in a 64 bit variable? [06:38] <michaelni> when you want to bit reverse a 64bit variable, its probably faster without a table [06:39] <overflow_0f8b> and you say it will be faster to read a table for 8 bit? [06:39] <overflow_0f8b> then using bit magic? [06:40] <michaelni> i dont really know where what is faster, best you try both [14:50] Action: Daemon404 swears michaelni may as well be on north american time, given the hours he is awake... [14:52] <durandal_1707> ffprobe/ffmpeg/ffplay still hangs for me on xwd [14:52] <Daemon404> infinite loop? [14:53] <durandal_1707> it takes to much time so yes it is probably (nearly) infinite [14:54] <durandal_1707> after say few seconds I just use ctrl+c [14:54] <Daemon404> it could just be incredibly inefficient [14:54] <durandal_1707> it hangs before decoding [15:03] <durandal_1707> lol, removed my experimental encoder/decoder and problem is gone.... [15:04] <durandal_1707> heh, pebkac [15:29] <CIA-48> ffmpeg: 03Reimar Döffinger 07master * r3fa6d20533 10ffmpeg/libavcodec/h264_parser.c: [15:29] <CIA-48> ffmpeg: H264 parser: enable header parsing/frame type detection for is_avc. [15:29] <CIA-48> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [16:54] <ubitux> oh nice, the two valgrind boxes in the green [17:07] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * rd9b3097ba2 10ffmpeg/libavcodec/ (pnm.c pnmdec.c): [17:07] <CIA-48> ffmpeg: pamdec: fix mono support. [17:07] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:41] <CIA-48> ffmpeg: 03Benjamin Larsson 07master * r7f215e3e72 10ffmpeg/libavcodec/wmalosslessdec.c: Use fixed-length array in revert_mclms() [19:41] <CIA-48> ffmpeg: 03Mashiat Sarker Shakkhar 07master * ra520a9946f 10ffmpeg/libavcodec/wmalosslessdec.c: Perform inter-channel decorr. only if both channels are coded [19:41] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * r04f7eee709 10ffmpeg/: [19:41] <CIA-48> ffmpeg: Merge remote-tracking branch 'shariman/wmall' [19:41] <CIA-48> ffmpeg: * shariman/wmall: [19:41] <CIA-48> ffmpeg: Perform inter-channel decorr. only if both channels are coded [19:41] <CIA-48> ffmpeg: Use fixed-length array in revert_mclms() [19:41] <CIA-48> ffmpeg: Merged-by: Michael Niedermayer <michaelni(a)gmx.at> [22:12] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * r1060e9ce54 10ffmpeg/ (cmdutils.c cmdutils.h cmdutils_common_opts.h): [22:12] <CIA-48> ffmpeg: cmdutils: add -cpuflags. [22:12] <CIA-48> ffmpeg: This allows globally forcing specific cpuflags (or lack thereof) [22:12] <CIA-48> ffmpeg: Useful for debugging and benchmarking [22:12] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:12] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * rb240866c5b 10ffmpeg/libavcodec/wmalosslessdec.c: [22:12] <CIA-48> ffmpeg: wmalossless: remove tab [22:12] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:12] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * r886b40b94a 10ffmpeg/libavcodec/wmalosslessdec.c: [22:12] <CIA-48> ffmpeg: wmalossless: Switch to new audio API [22:12] <CIA-48> ffmpeg: Partially fixes Ticket1000 [22:12] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:12] <CIA-48> ffmpeg: 03Michael Niedermayer 07master * rf4f386dd00 10ffmpeg/libavcodec/wmalosslessdec.c: [22:12] <CIA-48> ffmpeg: wmalossless: Fix infinite loop. [22:12] <CIA-48> ffmpeg: Fixes more of Ticket1000 [22:12] <CIA-48> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Mon Feb 20 2012

1 0

ffmpeg.log.20120219
by burek 20 Feb '12

20 Feb '12

[00:13] <hotwings> using debian testing here.. how do i build ffmpeg-dev from git? i dont see any option for it in ./configure [00:16] <burek> what exactly do you need it for [00:20] <hotwings> building the VDR softhddevice plugin [00:26] <matteowiz> hi all, anyone can help me with a cross compilation problem? [00:36] <burek> hotwings, well, I'm not sure, but, if you have all the headers and you've built your ffmpeg than you have your -dev [00:36] <burek> matteowiz, [00:36] <burek> can you please use pastebin.com, to show your command line and its output? [00:37] <matteowiz> the problem is related to x264 [00:37] <matteowiz> if i build it with --disable-asm [00:37] <matteowiz> no problem. [00:37] <matteowiz> but with asm enabled [00:37] <matteowiz> i got: [00:37] <matteowiz> i486-openwrt-linux-uclibc-gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC -O2 -pipe -march=i486 -fno-caller-saves -fhonour-copts -Wno-error=unused-but-set-variable -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/usr/include -I/usr/src/trunk/bin/x86/OpenWrt-S [00:37] <matteowiz> DK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/include -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_uClibc-0.9.33/usr/include -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_uClibc-0.9.33/include -fpic -fno-strict-aliasing [00:37] <matteowiz> -std=c99 -fomit-frame-pointer -fPIC -pthread -E -o /tmp/ffconf.TfiqPYxq.o /tmp/ffconf.EiXMx4Ai.c [00:37] <matteowiz> In file included from /tmp/ffconf.EiXMx4Ai.c:1:0: [00:37] <matteowiz> check_func x264_encoder_encode -lx264 [00:37] <matteowiz> check_ld cc -lx264 [00:37] <matteowiz> check_cc [00:37] <matteowiz> BEGIN /tmp/ffconf.EiXMx4Ai.c [00:37] <matteowiz> 1 extern int x264_encoder_encode(); [00:37] <matteowiz> 2 int main(void){ x264_encoder_encode(); } [00:37] <matteowiz> END /tmp/ffconf.EiXMx4Ai.c [00:37] <matteowiz> i486-openwrt-linux-uclibc-gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC -O2 -pipe -march=i486 -fno-caller-saves -fhonour-copts -Wno-error=unused-but-set-variable -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/usr/include -I/usr/src/trunk/bin/x86/OpenWrt-S [00:37] <matteowiz> DK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/include -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_uClibc-0.9.33/usr/include -I/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_uClibc-0.9.33/include -fpic -fno-strict-aliasing [00:37] <matteowiz> -std=c99 -fomit-frame-pointer -fPIC -pthread -c -o /tmp/ffconf.TfiqPYxq.o /tmp/ffconf.EiXMx4Ai.c [00:37] <matteowiz> i486-openwrt-linux-uclibc-gcc -L/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/usr/lib -L/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/target-i386_uClibc-0.9.33/lib -L/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_ [00:37] <matteowiz> uClibc-0.9.33/usr/lib -L/usr/src/trunk/bin/x86/OpenWrt-SDK-x86-for-Linux-i686-gcc-4.6.2_uClibc-0.9.33/staging_dir/toolchain-i386_gcc-4.6.2_uClibc-0.9.33/lib -Wl,--as-needed -o /tmp/ffconf.lAgnQI9K /tmp/ffconf.TfiqPYxq.o -lx264 -lmp3lame -lfaac -lm -pthread -lz [00:37] <cbreak> oh god [00:37] <cbreak> stupid spammers attack us! [00:38] <matteowiz> sorry for long lines [00:38] <cbreak> go use a pastebin [00:39] <matteowiz> http://pastebin.com/QjB90cih [00:42] <cbreak> and what's the problem? [00:42] <cbreak> you can't link ffmpeg? [00:42] <matteowiz> after that ffmpeg configure says libx264.so not found [00:43] <matteowiz> because of several undefined reference to functions _sse|_mmx|_mmx2 etc [00:43] <matteowiz> as u see in the pastebin [00:43] <hotwings> burek, ok thanks [00:44] <cbreak> matteowiz: check if the x264 headers you use match the library you try to link to [00:44] <matteowiz> cbreak: yes cause i zeroed my toolchain several times [00:45] <cbreak> and you don't have any other headers in the include path? (like /usr/include or so) [00:45] <matteowiz> if i build in the same toolchain x264 with --disable-asm everything works fine [00:45] <burek> matteowiz, did you build yasm [00:45] <cbreak> hmm. [00:45] <matteowiz> yasm installed and uptodate [00:45] <cbreak> you build for some weird platform? [00:45] <matteowiz> i build for openwrt-i486 [00:46] <cbreak> hmm... [00:46] <cbreak> so check why the symbols are not in the library [00:46] <matteowiz> the main problem is that with libx264 with disable-asm performance is horrible [00:47] <matteowiz> creak i run nm libx264.so.120 [00:47] <matteowiz> and the symbols are there [00:47] <burek> maybe they are not exactly "there" [00:48] <burek> if the linkage type is not the same [00:48] <burek> meaning int @@main is not the same as int __main [00:48] <matteowiz> for example: [00:48] <burek> althought they both show as "main" [00:48] <matteowiz> 0001e360 t x264_pixel_ads1 [00:48] <matteowiz> U x264_pixel_ads1_avx [00:48] <matteowiz> U x264_pixel_ads1_mmx2 [00:48] <matteowiz> U x264_pixel_ads1_sse2 [00:48] <matteowiz> U x264_pixel_ads1_ssse3 [00:49] <cbreak> that means they are undefined. [00:49] <cbreak> not in there [00:49] <matteowiz> ah [00:49] <matteowiz> great to know [00:49] <cbreak> T would mean that they are in the text segment [00:49] <matteowiz> why are U then? [00:49] <cbreak> (code) [00:49] <cbreak> it's an external dependency [00:49] <cbreak> it didn't link to code that contained it [00:50] <matteowiz> i use no stripping btw [00:50] <cbreak> did you manage to create an x264 executable? [00:50] <matteowiz> yes [00:50] <cbreak> so ... that executable must have those symbols defined [00:51] <cbreak> wonder from where. [00:53] <matteowiz> mmm if i try to run x264 seg fault [00:53] <matteowiz> :) [00:55] <matteowiz> so is yasm messing up things here? [00:56] <matteowiz> here is gdb core processing: [00:56] <matteowiz> This GDB was configured as "i486-openwrt-linux". [00:56] <matteowiz> (no debugging symbols found) [00:56] <matteowiz> Core was generated by `x264'. [00:56] <matteowiz> Program terminated with signal 11, Segmentation fault. [00:56] <matteowiz> [New process 1720] [00:56] <matteowiz> #0 0xb76b8857 in ?? () [00:57] <matteowiz> and here the run program: [00:57] <matteowiz> This GDB was configured as "i486-openwrt-linux"... [00:57] <matteowiz> (gdb) r [00:57] <matteowiz> Starting program: /usr/bin/x264 [00:57] <matteowiz> Program received signal SIGSEGV, Segmentation fault. [00:57] <matteowiz> 0xb77df857 in parse_enum () [00:57] <cbreak> stop spamming [00:58] <cbreak> and check your x264 executable wether it has the symbols defined [00:58] <matteowiz> sorry but i don't think it's spam [01:01] <matteowiz> nm shows that are undefined even in the executable [01:02] <cbreak> more than two lines in a single second is spamm :) [01:02] <cbreak> seems you never linked them [01:03] <matteowiz> oh sorry for spam then :) [01:03] <matteowiz> mmm [01:04] <matteowiz> due to the complexity of openwrt buildroot [01:04] <matteowiz> i modified the Makefile in x264 [01:05] <matteowiz> cause there was a strange trailing space missing, causing AR to miss 'rc' flags [01:05] <matteowiz> i think i better have to recheck LD flags [01:07] <cbreak> maybe the makefile generation aborted prematurely? [01:08] <matteowiz> no, it was saying something like: i486-openwrt-linux-uclibc-arlibx264.a ... i486-openwrt-linux-uclibc-arlibx264.a: Command not found [01:11] <matteowiz> and then : i486-openwrt-linux-uclibc-ld x264 x264.o input/input.o .... i486-openwrt-linux-uclibc-ld: cannot find x264: No such file or directory [01:11] <matteowiz> ... bingo... [01:11] <matteowiz> ld x264 with no flags prior the x264? [07:38] <^qop^> hi there, can I write tags to wav? [08:54] <magn3ts> Can I ask a lot of possibly annoying questions? I hope they're not annoying because I've spent like two weeks trying ot figure these things out on my own and I'm stumped. [08:54] <magn3ts> How does one best transcode video and stream it "live"? I would like to be able to transmit duration information since it is known to the encoder, even though the transcoded video will be streamed live as soon as it has begun. [08:55] <magn3ts> Is it possible to get any muxer to properly encode duration length at the beginning of the file when in streaming mode? [12:36] <NonFish_> how come osd frame numbers are always skewed.. if i divide frame number by framerate of source, the time that results is not the same as the time mpchc shows [12:40] <NonFish_> or, how are the frame numbers properly converted to a timecode in simple math? [12:45] <JEEB> it's only simple in case of stuff like avisynth where there is no such thing as variable frame rate [12:47] <NonFish_> the source file is xvid+mp3 [12:48] <NonFish_> nothing vfr [12:48] <NonFish_> vbr mp3 [12:50] <JEEB> let's say you have 24000/1001fps content [12:50] <JEEB> (framenum * 1001)/24000 , I would guess [12:51] <NonFish_> ehh 1001 hm [12:54] <NonFish_> is (framenum * 1001)/23976 right for 23.976fps [12:55] <JEEB> 23.976 is 24000/1001 [12:55] <NonFish_> oh. lol sorry. [13:18] <NonFish_> thanks tho it is still coming out wrong [13:18] Action: NonFish_ sleeps [13:22] <JEEB> well, what is "wrong"? [13:22] <JEEB> it gives you seconds and something that's usable as milliseconds :P [13:22] <JEEB> and the results in python look very much like what I get with ffmpegsource [13:49] <Diogo> hi, one question please, first this is possible use ffmpeg for real time encoding, i need to build a file upload in php (send file via POST http) and the server when is getting the video start to encode..any tutorial on the web? like transloadit.com.. [13:49] <Diogo> i can see a solution http://transloadit.com/demos/video-encode/encode-a-video-in-realtime [13:49] <Diogo> but i don't know how they do this.. [13:49] <Diogo> thanks for your help [13:52] <Tjoppen> you somehow get the upload as a pipe, then send that to ffmpeg [13:53] <Tjoppen> a CGI script could do it I think [14:59] <Medi> Hi, I want to set several http headers, I found out I can do it with "-headers" option in command line, my question is how to set several headers, Should I use several -headers option, or I should use , / | .... to split values ? [15:09] <Medi> I want to set several http headers, I found out I can do it with "-headers" option in command line, my question is how to set several headers, Should I use several -headers option, or I should use , / | .... to split values ? [15:09] <Medi> I will be grateful for you help [15:09] <Medi> please [15:17] <Medi> is there anyone who can help me ? [18:51] <MindSpark> hi, I am going crazy over this. Can someone please tell me how to convert rtmp to rtsp using ffserver? [18:55] <maujhsn> Can someone show me how to incorporate this command: -timestamp now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z]) [18:55] <maujhsn> Looking for a practical example! [19:11] <maujhsn> Can someone show me how to incorporate this command: -timestamp now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z]) [19:16] <sacarasc> -timestamp now [19:16] <sacarasc> There you go. :p [19:19] <maujhsn> sacarasc "ffmpeg -i input.mpg -timestamp now"? [19:19] <sacarasc> You'd need an output file, too. [19:19] <sacarasc> But that's what it seems to do. [19:22] <maujhsn> sacarasc I will give it a shot. [19:26] <q|o|p> with all honesty... how well does the ffac aac encoder compares to the nero encoder? I rather convert all my files to oga but I've got a free ipod so... [19:28] <q|o|p> btw, and fmpeg -i "$file" -acodec libvorbis -aq 9 #should output similar quality to oggenc -q 9 right? [19:28] <maujhsn> sacarasc When I run the the command: ffmpeg -i cab_A.mpg -timestamp now cab_B.mpg" I see no difference in the output. I also see this yellow tag: "timestamp is deprecated, set the 'creation_time' metadata tag instead." [19:43] <quebre> hello [19:43] <quebre> how can i effectively convert the mp4 avc720p to avi ? [19:43] <q|o|p> olleh [19:43] <quebre> -vcodec libx264 -preset slow -crf 18 -acodec libfaac -ac 2 -ab 128k [19:44] <quebre> i use this [19:44] <quebre> ;_) [19:44] <Mavrik> quebre, you'll have to find out what "avi" do you want [19:44] <q|o|p> I wouldnt [19:44] <quebre> xvid [19:44] <quebre> i want the same or similar DVDRip-XviD [19:44] <quebre> can i get some syntax for ffmpeg [19:44] <q|o|p> he just want to change the container [19:44] <quebre> maybe some of you have it [19:44] <quebre> but what i use now [19:44] <quebre> is bad quality [19:44] <quebre> and im pretty sure i did it in the lame way [19:44] <quebre> ;/ [19:44] <Mavrik> you're not even creating xvid with that. [19:45] <quebre> yeah, i noticed [19:45] <Mavrik> crf 18 should be awesome quality actually [19:45] <quebre> thats why im here ;p [19:45] Action: q|o|p would use mkv instead [19:45] <quebre> i can't [19:46] <Mavrik> first of all you need to make sure your ffmpeg is compiled with libxvid [19:46] <Mavrik> (ffmpeg -codecs) [19:46] <quebre> i am converting it so i can run it on old laptop IBM Thinkpad T20 [19:46] <quebre> thats why i wnt to make it run smoothly [19:46] <quebre> on older computers [19:46] <quebre> so i thought decerasing quality will improve perfomance.. [19:46] <quebre> but.. [19:47] <q|o|p> http://ffmpeg.org/faq.html#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_0… [19:47] <quebre> you guys know better so im open for advices [19:47] <q|o|p> read the two contiguous ones [19:47] <quebre> i need the one-liner ;f [19:47] <quebre> coz im very tired... [19:47] <q|o|p> it is one line [19:47] <quebre> but not link ;/ [19:47] <quebre> ffmpeg [options] [19:47] <quebre> like this, if you would be so kind.. [19:47] <q|o|p> there is lazy... and there is fking lazy [19:47] <quebre> no, im just tired [19:47] <quebre> pretty please ;/ [19:48] <q|o|p> click it! it is one line [19:48] <quebre> i need it to finish the work [19:48] <Mavrik> first hit on google "ffmpeg encode xvid" gives your full commandline [19:48] <Mavrik> stop wasting our time. [19:48] <quebre> did the pharse earlier [19:48] <quebre> on google [19:48] <quebre> but it jumped out with nonsence [19:48] <quebre> like: [19:49] <q|o|p> whatever... so... how well does faac compare to neroaccEnc? [19:49] <q|o|p> I cant fnd any good comparisions beyond perceived [19:49] <hi117> faac's docs do say its not as good as some [19:51] <q|o|p> quebre: if you were to at least check the FAQ before anything, you could have read "ffmpeg -i input.mp4 -same_quant output.avi" was your answer with fewer effort than getting up in irc [19:51] <q|o|p> hi117: yeah... but how good is not as good lol [19:52] <hi117> the standard ogg encoder my distro comes with did better then faac at same bitrate IMO [19:52] <q|o|p> ogg? [19:52] <hi117> vorbis [19:52] <q|o|p> I am talking of aac [19:53] <hi117> i know but from what i hear aac>vorbis in terms of quality at same bitrate [19:53] <q|o|p> I thought fmpeg -i "$file" -acodec libvorbis -aq 9 #should output similar quality to oggenc -q 9 [19:53] <q|o|p> hi117: the study I saw said that was true at low level bitrates [19:53] <q|o|p> and had ogg as winer in higher bitrates [19:54] <hi117> heh that might be why since i was doing a 300kb/s+ encode [19:54] <hi117> it was a -q 10 [19:54] <q|o|p> -q 10 is 500 k [19:54] <Mavrik> q|o|p, neroaacenc should be noticably better [19:55] <hi117> the input was crap though so it only needed 300 [19:56] <q|o|p> Mavrik: then there is a problem, to convert using neroaacenc I'd have to convert to wav before and I'd lose the tags in the process >.> [19:56] <hi117> convert to flac? doesnt that keep tags? [19:56] <Mavrik> q|o|p, yep [19:56] <Mavrik> q|o|p, that's why we just use libfaac anyway [19:56] <q|o|p> yeah but neroaacenc only accepts wav as ainput [19:56] <hi117> ah [19:57] <q|o|p> meh, I'll use aac [19:57] <q|o|p> err ogg [19:57] <Mavrik> quality gain isn't work d*cking around with wav conversion and offhanding to neroaacenc [19:57] <q|o|p> :) [19:57] <q|o|p> yeah [19:57] <cbreak> FLAC ftw. [19:58] <q|o|p> yeah except I only have 8 gbs of space, and 800 gbs of mostly flacs dont fit in :p [19:58] <Mavrik> 3TB drives are like 180¬ :P [19:58] <cbreak> 3tb hard disk: 200$ [19:59] <q|o|p> yeah, do you know any 3TB cellphone? [20:00] <cbreak> it's the cloud! [20:00] <cbreak> or so I was told [20:01] <q|o|p> oh yeah, not in the US, no googleMusic and wireless internet is stupidly expensive here [20:01] <Mavrik> just throw 128k mp3s on the phone ffs [20:01] <q|o|p> never!!!!!!!!!!!!!!!!!!! [20:01] <cbreak> 128k aac would be better [20:02] <Mavrik> running around city with traffic noise you'll never hear the difference between 128k and 320k anything [20:02] <cbreak> although, with the DAs in a mobile phone [20:02] <cbreak> you'd probably not hear the difference [20:02] <q|o|p> 256 aac min [20:02] <Mavrik> so you're just causing yourself problems with compilcations [20:02] <q|o|p> well I am not a gifted listener but 128 is crappy enough for me to notice it :) [20:03] <Mavrik> 160 VBR then [20:03] <q|o|p> 192 [20:03] <q|o|p> :) [20:03] <Mavrik> wasting space on a device when it's gonna be destroyed by ambient noise is just silly :P [20:04] <q|o|p> I have noise blocking headphones! [20:04] <cbreak> just change your music style to industrial techno [20:04] <cbreak> then the ambient noise would be an added benefit [20:05] <Mavrik> q|o|p, so now you even have headphones which add additional noise [20:05] <q|o|p> http://www.bjs.com/webapp/wcs/stores/servlet/ProductDisplay?catalogId=10201… [20:05] <Mavrik> cbreak, ^^ [20:05] <q|o|p> lol [20:06] <q|o|p> lol gtg :) [20:21] <maujhsn> Trying learn how to use the avfilter in ffmpeg! This page gives one practical corny example "http://ffmpeg.org/libavfilter.html" does anbody know of other url's that are more user friendly? [20:25] <mapreduce> In http://dranger.com/ffmpeg/tutorial08.html I see static struct SwsContext *img_convert_ctx; I believe swscale isn't thread-safe, what's the minimum I could do to make this thread safe? [20:25] <mapreduce> I tried using gcc's __thread, but got whole application freezes that I couldn't debug, and logging didn't show that I was creating lots of threads. [20:26] <mapreduce> Less than 10. [20:38] <cbreak> if thread safety were easy, everyone would do it. [20:39] <cbreak> in general, if you want something to be thread safe, you have to ensure that either each thread acts in completely different data [20:39] <cbreak> or that access is serialized, and correct [20:40] <Mavrik> mapreduce, keeping separate SwsContexts with their own datasets SHOULD be enough [20:40] <Mavrik> (for each thread) [20:41] <cbreak> unless there's global state [20:41] <cbreak> (like non-const function statics, or globals, or static members if this were c++ which it isn't :) [20:42] <Mavrik> yeah, but IIRC there isn't [20:42] <Mavrik> (but it was awhile ago since I used swscale) [20:42] <kriegerod> offtop: does somebody knows existing technical approach to do live, realtime online video translation of airsoft game? [20:43] <kriegerod> (with points of views of players) [20:43] <cbreak> airsoft? [20:43] <cbreak> translation? [20:43] <kriegerod> sorry? [20:44] <Mavrik> kriegerod, it's expensive and hard [20:44] <Mavrik> and you need video professionals [20:50] <NonFish_> is airsoft a videogame or a realworld offline game with airguns? [20:54] <NonFish_> well, if a videogame, and this is totally a proprietary, windows only method, kriegerod.. but you could download xsplit and stream the screen to a www.justin.tv account. [20:57] <kriegerod> NonFish_, nope, it's real-world game [21:00] <NonFish_> what did you mean by translation? [21:01] <cbreak> translating languages probably [21:01] <cbreak> dubbing? [21:20] <kriegerod> broadcasting [21:20] <kriegerod> "translation" word in russian means sth close to "broadcasting", thus i mis-used this word [21:21] <NonFish_> oh good. realtime language translation is hard. [21:22] <vadim> hi guys )) [21:23] <vadim> i may be in role of "real time translator" for a while ))) [21:23] <vadim> english-russian-english [21:29] <NonFish_> broadcasting is easier. xsplit will let you use any attached camera as a video source. maybe you can find a wireless webcam and mount it to player helmets, kriegerod. [21:30] <cbreak> vlc does broadcasting [21:30] <cbreak> out of the box [21:33] <NonFish_> yes though, never quite understood how to access the output over the web.. but maybe that was many versions ago [21:35] <kriegerod> NonFish_: most used option for accessing over the web is publishing to Wowza, as i know [21:35] <kriegerod> wowza gives rtmp, rtsp, apple hls [21:36] <cbreak> vlc gives that and more [21:36] <kriegerod> vlc can _serve_ rtmp? [21:37] <cbreak> yes. [21:37] <kriegerod> wow, since when? [21:37] <cbreak> and http live streaming [21:37] <cbreak> and a bunch of other stuff [21:37] <cbreak> no idea. [21:37] <cbreak> http live streaming since 1.12 or so [21:37] <cbreak> I use that at work [21:37] <JEEB> I've done http streaming with mpeg-ts as the container since 0.8.6 or so [21:37] <JEEB> lol [21:38] <cbreak> what kind of http streaming? [21:38] <cbreak> http live streaming is a specific protocol [21:38] <JEEB> mpeg-ts pushed into a http port [21:38] <cbreak> that's not it. [21:38] <JEEB> yes, you're talking about HLS [21:38] <JEEB> :3 [21:38] <JEEB> so yes, we are talking of different things [21:39] <JEEB> although both in the end boil to a very similar concept of mpeg-ts over http [21:39] <JEEB> IIRC [21:39] <cbreak> http live streaming is a pull protocol [21:39] <cbreak> (of course it is, it's based on http...) [21:39] <kriegerod> cbreak, sorry i'm just so surprised. Please tell me again, you launch VLC and it receives connections from clients, and gives them media by HLS? Just as far as i know HLS even client-side support is not available in vlc 1.1 at all [21:39] <cbreak> it uses chunked .ts files [21:40] <cbreak> and apparently the splitting was new in 1.12 [21:40] <JEEB> yeah [21:40] <NonFish_> one thing is though, justin.tv or wowza type services save you bandwidth with many viewers. [21:40] <cbreak> kriegerod: no, I write the output into some directory, and apache (or an other web server) serves it [21:40] <mapreduce> Mavrik: I hoped the same, but the mechanism I tried to use to keep separate SwsContexts, gcc's __thread, caused freezes. [21:40] <kriegerod> cbreak, so vlc just segments it, right? [21:40] <cbreak> in the mode of operation I use it yes [21:41] <cbreak> but if you do rtp/udp/udpbroadcast or a bunch of others it actually does stream [21:41] <kriegerod> cbreak, returning to your previous reply, are you confident VLC is usable as rtmp server? [21:41] <JEEB> this all reminds me of the hacks I did to stream stuff from my receiver in Japan to my place [21:41] <JEEB> lol [21:41] <cbreak> kriegerod: I know it can do it [21:42] <cbreak> I don't know if it's worth anything or only supports a single digit number of clients [21:42] <JEEB> UDP from the receiver to VLC on localhost, then VLC streaming out mpeg-ts into http [21:42] <JEEB> lol [21:44] <cbreak> http://news.slashdot.org/story/12/02/19/0127204/vlc-20-twoflower-released-f… [21:44] <cbreak> kriegerod: do you have a VLC available to you? [21:45] <NonFish_> does streaming with vlc require viewers to have vlc too? [21:45] <cbreak> depends [21:45] <JEEB> there is no single way of streaming in VLC [21:45] <cbreak> I Stream to iPads without VLC [21:45] <JEEB> there are multiple [21:45] <cbreak> (with http live streaming) [21:46] <cbreak> the whole point of using httpls is to be able to stream to them in my case :) [21:46] <kriegerod> cbreak: sorry? [21:47] <cbreak> kriegerod: can you start VLC now? [21:48] <cbreak> if you can, you can try out the streaming :) [22:33] <mape> I'm dumping images from a video and was wondering if there is any way of defining what %d in img-%d.jpg starts on. I have multiple videos of one timestamp and want to pick up where the previous left off [22:36] <mape> sorry, multiple videos but one "moment", so they are cut up but I can't merge them as new ones keep coming in [00:00] --- Mon Feb 20 2012

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