[01:24] <smjd> are you sure FLV doesn't support that? [01:26] <smjd> or is there also a codec called FLV? [01:28] <sacarasc> There is a codec called FLV. [01:31] <smjd> is it Sorenson Spark? [06:48] <elBradford> Does anyone have experience cross compiling ffmpeg with librtmp? [06:50] <elBradford> Everything seems to be going well, however I get "undefined reference" errors in librtmp.o [06:50] <elBradford> Then it stops going well, obviously. [07:37] <taqattack> is there a way command to reduce latency / control packet size in ffmpeg for streaming? [07:45] <EvanDotPro> i am pushing a tcp h264 stream from ffmpeg to a socket server and parsing the NAL units... if i stream it over 127.0.0.1, the NAL units are ~20k-30k bytes each, and the video decodes fine on the other end, but if i try to go over my router / modem where the MTU size is affected, the NAL units are tiny, like 1400-1500 bytes, and the video doesn't seem to want to decode. [07:47] <EvanDotPro> piping to netcat and then over 127.0.0.1 seems to cause the problem too, except that the NAL units seem to be 8191 bytes in that case [08:26] <skyroveRR> Hello, I'd like to display metadata for an audio/video file using ffmpeg, I'm using slackware, I tried http://stackoverflow.com/questions/9464617/retrieving-and-saving-media-metad... but haven't been successful yet, the metadata.txt file is empty. [08:28] <skyroveRR> I'd like to also edit and view the title, comments and other metadata on a particular media file, so what's the command for that? [08:41] <skyroveRR> Hello, I'd like to display metadata for an audio/video file using ffmpeg, I'm using slackware, I tried http://stackoverflow.com/questions/9464617/retrieving-and-saving-media-metad... but haven't been successful yet, the metadata.txt file is empty, and I also want to edit the metadata itself, any ideas? [09:28] <elBradford> Has anyone compiled ffmpeg with librtmp for Android? Looking to pay for help at this point, kind of desperate. [10:22] <BlackBishop> relaxed, burek, sacarasc: mhmm .. same .. tried -preset ultrafast, -vcodec flv | libx264 .. none get past 13 fps :| weird .. [10:27] <BlackBishop> Mhmm, I do have a raspberry pi that is said to have an encoder for mkv/wmv not just decoder .. [10:31] <BlackBishop> actually .. vc1/mpeg2 .. still, good enough .. [11:11] <kms_> there are any key for realtime encoding on low CPU with libtheora? [11:21] <IamTrying> http://paste.fedoraproject.org/42905/38044632/ - why the is giving "could not decode stream" ?? [12:10] <luc4> Hi! I wrote this function: http://paste.kde.org/pb44c1418/. It implements a pixel format conversion from yuyv to argb manually or using ffmpeg. What I noticed, and surprised me a little is that the manual conversion is considerably faster than the ffmpeg algorithm. This is pretty weird. Any idea why? Maybe I'm using ffmpeg wrong? Maybe there is a faster algorithm than SWS_FAST_BILINEAR? [13:17] <luc4> Anyone using ffmpeg for pixel format conversion? [13:18] <zap0> luc4. do you really care? or you have an actual question? [13:19] <JEEB> zap0, he has [13:19] <JEEB> see an hour before [13:19] <luc4> I already asked some hours ago, just trying again to see if someone new connected: Hi! I wrote this function: http://paste.kde.org/pb44c1418/. It implements a pixel format conversion from yuyv to argb manually or using ffmpeg. What I noticed, and surprised me a little is that the manual conversion is considerably faster than the ffmpeg algorithm. This is pretty weird. Any idea why? Maybe I'm using ffmpeg wrong? Maybe there is a faster algorithm than [13:19] <luc4> SWS_FAST_BILINEAR?. [13:19] <JEEB> also you should most probably ask about this on #ffmpeg-devel, although there could be various reasons [13:19] <JEEB> you could check the code in swscale [13:19] <JEEB> for this conversion path [13:20] <zap0> how is it ffmpeg being slower makes it wrong? isn't wrong a measure of the correctness of pixel calculations? [13:20] <JEEB> He is asking if it's possible he's using ffmpeg wrong [13:20] <JEEB> not that ffmpeg is wrong [13:20] <JEEB> not to mention that the compilation options generally don't tend to do too much optimization because that can "get in the way of hand-written assembly optimizations" so if you compile your own code with various optimizations enabled it can end up being faster [13:21] <luc4> Both seems to be correct from what I see. [13:21] <JEEB> I've not looked at that code path myself [13:21] <zap0> ffmpeg would be doing per pixel math. not some lookup table [13:22] <luc4> I can look at the code, but it is unlikely I'll get why it is slower in reasonable time. [13:23] <JEEB> hmm, a look-up table [13:23] <vlt> Hello. I ffmpeg'd an AC3 5.1 stream to wav and got a six channel file. Do you know which channel is which? [13:23] <JEEB> anyways, there can be various reasons [13:23] <JEEB> you're not getting any non-aligned data warnings, right? [13:24] <zap0> vlt, there is no OFFICIAL standard ordering. but variuos configs are "mostly" consistant. [13:25] <luc4> JEEB: I got a warning related to software conversion being run. But I'm not getting it at the moment. [13:25] <zap0> vlt, 6 channels can mean: channels of which there are 6. or it could mean 5.1 or it could mean quad+2-sides. [13:27] <zap0> vlt, if ffmeg was able to detect the 6 channels as meaning "5.1" then it will consistantly use it's 5.1 layout. [13:27] <vlt> zap0: Thsi is what `file` says: ATSC A/52 aka AC-3 aka Dolby Digital stream, 48 kHz,, complete main (CM) 3 front/2 rear, LFE on,, 640 kbit/s reserved Dolby Surround mode [13:28] <zap0> vlt, when playing it back, you will likely have to manually tell the playback device the meaning of this 6 channel file is "5.1" and in 90% of situations it should then work correctly. [13:29] <zap0> vlt, the original file or the WAV file you converted? [13:29] <vlt> zap0: orig [13:29] <zap0> ok, so ffmpeg should comprehend that as meaning 5.1 (3f,2r+LFE) [13:29] <zap0> it will then use 5.1 layout. [13:30] <zap0> but WAV doesn't store the meaning. so the playback device has to be told! [13:30] <zap0> the playback device is only going to see a WAV that has 6 channels. [13:31] <zap0> *TYPICALLY* due to market forces... many playback devices seeing a 6 chanel file will presume 5.1 orientation [13:31] <vlt> zap0: I converted it to that 6 ch wav to be able to edit the audio (in ardour). And now I don't know which channel is which. [13:31] <zap0> vlt, i *i know* sucks, don't it ?!? ;) [13:32] <vlt> zap0: ? [13:32] <zap0> get one of the 5.1 sample AAC files with human talk on each channel. convert it, open it in your app. then listen back, you'll hear what is what. [13:33] <vlt> zap0: Good idea. Thank you. [13:34] <luc4> JEEB: trying in the devel channel. Thans for your advice. [14:46] <funyun> hi. is there anyway to add an image to a movie with ffmpeg? so there's like a thumbnail when it's in itunes? [14:48] <sacarasc> Does iTunes not just take a frame of the movie to use as a thumbnail? [14:49] <funyun> sacarasc: itunes/other apps show a thumbnail if someone has added one. if not it just shows a frame. i'm just wondering if i can add one with ffmpeg [14:49] <sacarasc> I do not know much about meta data adding with ffmpeg... [14:49] <sacarasc> You probably could with MP4box, though. [14:50] <funyun> sacarasc: okay, and i should look under meta data? [14:50] <sacarasc> I would assume so. It would be like album art... [14:51] <funyun> sacarasc: thanks [16:58] <burek> !pastebin BlackBishop [18:32] <zap0> if i generate some 16bit PCM audio, will it be bigendian or little endian? [19:40] <DorjePy> what is the easiest, accurate way learn filters [19:46] <klaxa> read sourcecode? :x [19:51] <BlackBishop> burek: ? [19:52] <durandal_1707> klaxa: why? [19:52] <klaxa> durandal_1707: it's probably the most accurate way to learn how filters work [19:52] <klaxa> unless that's not what he wanted and we misunderstood each other [19:54] <durandal_1707> klaxa: what he wanted? [19:54] <klaxa> <DorjePy> what is the easiest, accurate way learn filters [19:54] <durandal_1707> that could mean how to learn to use filters [19:55] <klaxa> the question is asked rather vaguely [19:55] <durandal_1707> why would i waste time writing filter documentation [20:18] <defaultro> is it possible to crop specific pixel on top and bottom, different size? [20:22] <ztane> hi y'all, I am trying to segment an h264+aac flv stream into individual h264+aac mp4 files, but only the first file works right; [20:23] <ztane> in the past there used to be reset_timestamps option, but it does not seem to be there anymore [20:27] <defaultro> this isn't working, -croptop 88 -cropbottom 88 [20:27] <defaultro> ok [20:29] <defaultro> http://pastebin.com/M2RH2KC3 [20:30] <defaultro> I have 2376x1584 video and I want to remove 187 pixels on top and 62 pixels at the bottom [20:32] <durandal_1707> defaultro: have you read last few lines? [20:32] <defaultro> yeah, use crop filter instead [20:32] <defaultro> isn't cropbottom the cropfilter? [20:33] <defaultro> foudn this, i hope it works - http://ubuntuforums.org/archive/index.php/t-1690664.html [20:35] <defaultro> it works but I am not sure if it's going to remove the pixels properly [20:35] <defaultro> :( it didn't work [20:35] <defaultro> i mean, cropping didn't work [20:36] <durandal_1707> defaultro: read documentation of crop filter [20:36] <defaultro> -vf crop=2376:1584:187:62 [20:36] <defaultro> that didn't do a thing [20:38] <ztane> is it possible to segment a h264/aac video with just vcodec copy, acodec copy? [20:38] <ztane> I still get ever increasing space in the beginning of each file, alas I have to segment the video using raspberry pi, so I cannot re-encode it [20:39] <durandal_1707> defaultro: you set out width and height to same value as input, and expect output to be different .... [20:39] <defaultro> i'm understanding it now :) [20:39] <defaultro> it's working, just need to calculate now [20:39] <defaultro> the first value are the new output dimension [20:40] <defaultro> "Width of the output video. It defaults to iw. This expression is evaluated only once during the filter configuration."his should have been well written as [20:40] <defaultro> Width of the new output video. It defaults to iw. This expression is evaluated only once during the filter configuration. [20:41] <ztane> ah works, I didn't have reset_timestamps 1 :"> [20:44] <defaultro> looks like the new crop filter is not as flexible as croptop and cropbottom. Now I can't specify a specifc height of pixels I want to remove on top and bottom. [20:45] <defaultro> you know what, maybe it did work, 2536:1334:187:62 :) [20:50] <durandal_1707> defaultro: really, you can use expressions to leave filter to calculate it for you [20:52] <defaultro> oh [20:52] <defaultro> that is cool [20:53] <defaultro> btw, when I use -qp 0, the video on youtube sometimes stops [21:13] <defaultro> durandal_1707: just finished reencoding it with -qp 0 and I'm currently uploading to youtube. I'm uploading a 2376x1334 video as compared to my other upload which is 1920x1080. Here is the old one - http://www.youtube.com/watch?v=NSJBYi8RdHs [21:14] <defaultro> brb, going to the gym :) [22:28] <EvanDotPro> can anyone point me to the place in where the x264 NAL unit size is determined based on the MTU or something? [22:37] <EvanDotPro> when i set the output to -f h264 tcp://127.0.0.1:1234 it makes the NAL unit size like 30k bytes on average. [22:38] <EvanDotPro> but when i make the output stdout and pipe to netcat it's like 8k bytes, then if i set it to another IP on my LAN or a remote server, it seems to adjust to the MTU and make them 1440 bytes [22:44] <EvanDotPro> also, i'm a little confused as to if slice-max-size is related to the NAL unit size or not [00:00] --- Mon Sep 30 2013