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February 2013

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ffmpeg-devel.log.20130212
by burek 13 Feb '13

13 Feb '13

[00:30] <saste_> wow av_url_split() sucks [00:35] <ubitux> unlimited free wifi at airport \o/ [00:35] <ubitux> i don't want to leave :( [00:42] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:3b1f037df56b: ffmpeg: fix streamcopy of lsf mp3. [00:43] <michaelni> BBB-work, stuttering should be fixed [00:43] <saste> ubitux, too bad for you [00:43] <saste> better for us ;-) [00:44] <ubitux> maybe i forgot how to write code [00:44] <saste> ubitux, now you write code in kanji [00:44] <ubitux> :) [00:45] <wm4> ubitux: btw., what's the status of the subtitle stuff? [00:45] <ubitux> untouched for one month? [00:46] <wm4> maybe I'll feel the need to kill of mplayer's subreader.c on a whim or so in the near future [00:46] <ubitux> wm4: i'll be back to it as soon as i'm home [00:46] <wm4> there was something about codepages still [00:47] <ubitux> but i need to catch up with one mo nth of semi hibernation [00:47] <ubitux> wm4: yeah there is a pending patch for this [00:47] <ubitux> codepage, char encoding etc [00:47] <wm4> I found libass doesn't even read utf-16 natively :( [00:47] <ubitux> it seems to be in need of some more thinking for the utf-16 shit [00:47] <ubitux> wm4: yeah well of course [00:48] <ubitux> SSA/ASS is utf-8 only afaik [00:48] <wm4> well I've seen utf-16 files in the wild [00:48] <wm4> http://code.google.com/p/libass/issues/detail?id=89 [00:48] <ubitux> lol [00:48] <wm4> chinese stuff [00:48] <wm4> fortunately not multibyte something :) [00:57] <saste_> i wonder if there is some way to avoid all the MP3 application-level hacks [01:02] <saste> michaelni: i summon you on "v4l2: bug #1570 and possible solution" [01:03] <saste> after some thinking i can't find a better solution of the one proposed [01:03] <saste> even if it is somehow hackish [01:12] <J_Darnley> What the... Who thought 99162f8d46db7ec02facae035c4ff573d2d8d612 was a good idea? [01:22] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:365270aec5c2: sanm: add forgotten check for decoded_size in old_codec37() [01:24] <saste_> J_Darnley, I don't like it very much, but what's the problem with it? [01:28] <J_Darnley> No real problem. I just think it is ugly. [01:28] <J_Darnley> Plus it missed one line. [01:29] <J_Darnley> All that to silence one warning? A warning that I thought an existing void pointer cast was there to hide. [01:42] <BBB-work> michaelni, are timestamps assigned to all non-b-frames now? can you check that? [01:44] <BBB-work> michaelni, we had issues in chrome where non-b-frames had AV_NOPTS_VALUE assigned to them for that video [02:04] <BBB-work> michaelni, and new patch sent that should apply cleanly on head for that other patch [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:f0d9ccac8937: sanm: reset frame sizes on buffer destruction [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:4ee88db840b3: sanm: init sizes with the full frame size [02:14] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:5260edee7e5b: sanm: Use the correct height variable in the decoded_size checks [02:23] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:7ff1a4b10f2d: Add add_pixels4/8() to h264dsp, and remove add_pixels4 from dsputil. [02:34] <michaelni> BBB-work, you need "-fflags genpts" or equivalent for pts from avi with b frames [02:34] <michaelni> this will add latency/delay though [02:47] <BBB-work> michaelni, in decoder? [02:49] <michaelni> latency in demuxer to delay packets till the next non B frame whichs dts is the previous frames pts [02:57] <creep> »! [03:41] <Plorkyeran> @ubitux> SSA/ASS is utf-8 only afaik <-- according to the spec it's ANSI, except for when you use \fe to override the encoding for a specific line [03:41] <Plorkyeran> everyone ignores that these days because it's retarded [04:32] <mfg> Hi, I have an mp4 file that wont play (its by Neulion) and I think they screw up the header on purpose to protected their content. Is there someone that can have a look at it? [04:39] <llogan> mfg: wrong channel. try #ffmpeg. [12:26] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=faac85005cadb5065d218b… [12:26] <ubitux> http://git.libav.org/?p=libav.git;a=commitdiff;h=fd1abf42693308ff99b9473c23… [12:26] <ubitux> rofl [12:27] <durandal_1707> you mean E1 thing? [12:27] <ubitux> yeah [12:27] <ubitux> E1 is obviously better [12:28] <av500> it saves storage space :) [12:28] Action: durandal_1707 blames nevcairiel [12:29] <nevcairiel> clearly i should've suggested E1 instead of EXPAND [12:48] <cone-102> ffmpeg.git 03Martin Storsjö 07master:ab8f1a698990: arm: Fall back to runtime cpu feature detection via /proc/cpuinfo [12:48] <cone-102> ffmpeg.git 03Derek Buitenhuis 07master:375ef6528c9d: libfdk-aacenc: Actually check for upper bounds of cutoff [12:48] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:3ccda2b02ba6: Merge commit '375ef6528c9dd2db7f9881e232cb0ec3aa16970d' [13:07] <iive> nevcairiel: actually E7 is better, E is short of Exp (like in the float format), 7 is the same key as & that is user for AND, so e7 reads as EXPAND. [13:08] <av500> iive: on your keyboard [13:09] <iive> av500: on most keyboards :P [13:10] <av500> iive: so maybe git post-checkout hook that modifies it for the locale :) [13:11] <cone-102> ffmpeg.git 03Diego Biurrun 07master:6fb296e97edc: avcodec/rectangle: Remove nonsense assert [13:11] <cone-102> ffmpeg.git 03Diego Biurrun 07master:62a43d30cc3b: crc: Move static data table from header to .c file [13:11] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:e2b703f53612: Merge commit '62a43d30cc3b9b2460f2c926beb522df32493625' [13:12] <durandal_1707> wasn't there patch on ml that fix VC1 interlaced? [13:15] <burek> just to mention this, we could integrate something like this into our wiki: https://github.com/mpenkov/ffmpeg-tutorial [13:15] <burek> A set of tutorials that demonstrates how to write a video player based on FFmpeg [13:16] <burek> :D [13:17] <burek> did you manage to get the list of cmds? :) [13:21] <durandal_1707> burek: i have private correspodence with ffloger [13:21] <durandal_1707> now if only it could write code .... [13:22] <burek> :) [13:25] <cone-102> ffmpeg.git 03Martin Storsjö 07master:d192ac3e03d1: swscale: Disallow conversion to GBRP16 [13:25] <cone-102> ffmpeg.git 03Martin Storsjö 07master:4c51fe48ba6d: h264: Copy h264chroma dsp context to slice thread copies [13:25] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:fe54f6cb453c: Merge commit '4c51fe48ba6dde059360b7451db7f6bbf2f11db3' [13:34] <cone-102> ffmpeg.git 03Daniel Kang 07master:b3f2a3fe3fe4: x86: mpeg4qpel: Make movsxifnidn do the right thing [13:34] <cone-102> ffmpeg.git 03Diego Biurrun 07master:304b806cb524: build: Make library minor version visible in the Makefile [13:34] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:7491356111e7: Merge commit '304b806cb524fb040f8e09a241040f1af2cb820b' [14:05] <cone-102> ffmpeg.git 03Diego Biurrun 07master:69dca3a4f45e: openbsd: Add minor number to shared library install name [14:05] <cone-102> ffmpeg.git 03Diego Biurrun 07master:56dea32f09e7: doc: developer: Fix wording in "naming conventions" section [14:05] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:08db519e1275: Merge commit '56dea32f09e71460add14a1b6d0fbe6eb16f02a4' [14:21] <cone-102> ffmpeg.git 03Diego Biurrun 07master:3658d2dfd48a: doc: developer: Add a note about reserved system name space [14:21] <cone-102> ffmpeg.git 03Luca Barbato 07master:fd1abf426933: lavu: avoid clashing definition of E [14:21] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:a63dc846b502: Merge remote-tracking branch 'qatar/master' [14:42] <durandal_1707> why transpose filter does not work for h != v chroma subsampling? [15:17] <divVerent> 14:42:28 @durandal_1707 | why transpose filter does not work for h != v chroma subsampling? [15:17] <divVerent> because then transposing would be lossy [15:17] <divVerent> and that's simply not implemented (one can always convert to 4:4:4 first though) [15:34] <durandal_1707> ok -> no reply -> applying patches at will [15:39] <durandal_1707> divVerent: ahh, because of spam i did not see you reply, sorry [15:43] <cone-102> ffmpeg.git 03James Almer 07master:0957041221fa: lavc/tta: Add support for encrypted streams [15:43] <durandal_1707> yes, that makes sense [16:02] <cone-102> ffmpeg.git 03Daniel Verkamp 07master:1ecce266fa5b: WAV muxer: add RF64 support [16:21] <ubitux> michaelni: "visible" [16:22] <ubitux> (commit message) [16:22] <av500> better than invisible commit message [16:23] <durandal_1707> ubitux: what/where? [16:24] <ubitux> "[FFmpeg-devel] [PATCH] developer.texi: replace "externally vissible" by public" [16:25] <ubitux> just a typo [16:25] <ubitux> 9h in the transit is a nightmare [16:25] <ubitux> in about ~15 hours i'm back for new adventures [16:26] <av500> going round the world? [16:26] <durandal_1707> you need 24h of sleep to sync [16:26] <ubitux> no, i'll be in sync directly [16:28] <ubitux> i just need to stay away for about 8 extra hours, which is easy with all the movie stuff available [16:29] <ubitux> i wonder what they use for the movies btw [16:29] Action: durandal_1707 removed base64 from its system [16:29] <ubitux> it's interlaced as hell [16:29] <ubitux> also, they display multiple subtitles at once [16:29] <ubitux> which is kinda interesting [16:29] <ubitux> the system is a linux but that's common knowledge [16:29] <ubitux> i wonder if someone has dumped a copy or two of these [16:30] <ubitux> av500: no, just back from a month in a country at the other side of the planet [16:31] <ubitux> and i'm waiting at the transit right now :P [16:31] <av500> australia? [16:31] <ubitux> transit is HK [16:35] <ubitux> (fr->china->jpn and th other way around) [16:36] <durandal_1707> hmm, why china? [16:43] <durandal_1707> nut doesnt support qtrle by default [17:00] <aballier> maybe i've missed something, but doesnt the AVFrame.channel change from int64 to int implies sizeof(AVFrame) decreases ? [17:00] <aballier> and in this case, accessing AVFrame.pkt_size will break [17:03] <aballier> emailed, better ask there directly [17:04] <nevcairiel> these fields are not part of the public API, you are supposed to use the getter functions [17:05] <aballier> ah ok, makes sense, i was missing that part [18:30] <durandal_1707> saste: i may be stupid but what is way to pad silence at start of stream? [18:30] <durandal_1707> sox have delay efffect which delays each channel in audio to some value [18:49] <saste> durandal_1707, aresample=min_comp=0.001:min_hard_comp=0.1 is one way [18:49] <saste> then there is a swr option which can be used to achieve the same effect [18:49] <saste> and yes adding an example may be useful [18:49] <saste> I don't know if apad can be used as well [18:54] <durandal_1707> what that numbers means and what ammount of silence it adds? [18:56] <saste> durandal_1707, wait, you want to *ADD* silence at the beginning of the stream? [18:58] <durandal_1707> yes [18:58] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/43315/focus=43318 [18:59] <durandal_1707> that resets pts to some 0 value [18:59] <saste> durandal_1707, plus, you can add an initial offset with asetpts=PTS+TIME_OFFSET/TB [19:00] <saste> I didn't try the first_pts option yet [19:00] <durandal_1707> well i would certainly need working command [19:02] <saste> durandal_1707, e.g. if you want to add 100 seconds of silence, you do: asetpts=PTS+100/TB,aresample=min_comp=0.001:min_hard_comp=0.1 [19:02] <saste> I suspect first_pts may be a more robust solution [19:03] <saste> apad could be extended to achieve the same effect [19:03] <saste> also now the audio filters order is messed up [19:04] <saste> what about factorizing documentation for all biquad filters? [19:05] <saste> you could have section and a subsection for all variants [19:05] <durandal_1707> saste: i'm more interested in writing new filters [19:05] <saste> you could have *a main section* and a subsection for all variants [19:05] <saste> i can do the work, just wanted to know your opinion [19:05] <durandal_1707> also I really do not know doxygen [19:05] <saste> texinfo i guess [19:05] <saste> not very much to know [19:09] <durandal_1707> saste: feel free to do whatever you feel is needed, my only opinion about documentation is that it should be present & correct and not cryptic [19:11] <saste> durandal_1707, about mp and optims, yes they are disabled [19:11] <saste> don't ask me why [19:12] <durandal_1707> saste: i'm trying to write VV->V filter where both inputs are readonly and i create new output buffer, i get buffer queue overflow errors [19:12] <durandal_1707> and nothing in output [19:12] <durandal_1707> ffmpeg -f lavfi -i testsrc -f lavfi -i mptestsrc -filter_complex blend /tmp/out.png [19:46] <Compn> http://torrentfreak.com/open-source-giant-vlc-mulls-bittorrent-support-1302… [19:51] <durandal_1707> seed!!! [19:57] <Compn> j-b talking to torrentfreak :D [19:58] <durandal_1707> saste: or should i have another queue for filtered buffers? [20:01] <durandal_1707> i really dunno what should I do and is that even possible [20:07] <michaelni> it should work to dereference the input and create new output [20:08] <durandal_1707> yes, but i'm clearly missing some code [20:11] <durandal_1707> https://github.com/richardpl/ffmpeg/tree/blend [20:28] <michaelni> I think you are missing a request_frame() implementation but i dont know if this is related to the problem [20:36] <durandal_1707> michaelni: and i would request frame for each input? [20:37] <durandal_1707> s/for/from [20:38] <durandal_1707> i just need template and i did 99% of filter already :D [20:47] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:1ac0fa50eff3: pngdec/filter: dont access out of array elements at the end [20:57] <durandal_1707> yay [23:52] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:6f9ae391deea: faxcompr: Dont read ref when the end has been reached in pass mode [23:52] <cone-102> ffmpeg.git 03Michael Niedermayer 07master:e1219cdaf9fb: tiff: Check buffer allocation and pointer increment more carefully in shorts2str() and double2str() [00:00] --- Wed Feb 13 2013

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ffmpeg.log.20130212
by burek 13 Feb '13

13 Feb '13

[00:50] <sam___> any suggestions .ffmpeg [libx264 @ 0x670520e0] Unable to parse option value "flags2 +bpyramid+wpred+mixed_refs-dct8x8" [00:51] <sam___> o.ffmpeg [Eval @ 0x7f8f4fffd4c0] Undefined constant or missing '(' in 'flags2' o.ffmpeg [libx264 @ 0x522f8140] Unable to parse option value "+bpyramid+wpred+mixed_refs-dct8x8" [00:53] <sam___> whats the problem with ffmpeg am i missing something? [00:53] <sam___> Undefined constant or missing '(' in 'flags2'. [00:54] <saste> sam___, use libx264 presets [02:44] <mishehu> greetings folks. I was wondering if openjpeg 2.0.x is supported in ffmpeg currently, or only the older 1.5.x branch. [03:07] <kode54> fuck, more distributions switching to avconv? [03:10] <llogan> kode54: which one(s) are you referring to? [03:11] <kode54> friend on another network caught my whining [03:12] <kode54> I spotted a capture and broadcast to twitch.tv tutorial [03:12] <kode54> which I first complained about because it was a freaking youtube video instead of a document [03:12] <kode54> then I complained because he was displaying a bash script in the video for viewers to transcribe, without actually linking to it in plain text form [03:12] <kode54> then I noticed he was using avconv instead of ffmpeg [03:13] <kode54> friend said he doesn't want to get into the whole ego war between the projects [03:13] <sacarasc> Which distro is your friend using? [03:13] <kode54> debian [03:13] <kode54> but he says gentoo is "switching" to avconv as the primary of the two as well [03:14] <kode54> his complaint is that ffmpeg gets more important features faster than avconv does [03:14] <kode54> but he's stuck with people making decisions that favor using the other [03:16] <demonjester> hey everyone, I was wanting to know if someone can help me with audio recording on my openwrt router [03:19] <llogan> kode54: at least the ffmpeg package in gentoo is actually FFmpeg upstream unlike debuntu [03:19] <kode54> that's cool, at least [03:20] <llogan> debian has dropped the "ffmpeg" package from experimental apparently [03:21] <llogan> so a real ffmpeg package may be one step closer to a reality...assuming the DDs don't simply disregard it [03:28] <demonjester> can anyone help me? i am using these comands and it wont save the file after I ctrl + c "ffmpeg -i /dev/audio test.mp3" [03:35] <llogan> relaxed: interested in changing the CentOS guide to static build instructions? [03:36] <defaultro> good evening folks. Is it possible to fix the color of my video? Looks like there is a a little cyan cast. The blue sky is like not blue [03:38] <llogan> defaultro: in all players? [03:38] <defaultro> yes [03:38] <defaultro> i remember reading an issue with this video camcorder [03:39] <defaultro> but Panasonic never made a fix for it :( [03:40] <llogan> there are probably several filters that can do this [03:40] <llogan> http://ffmpeg.org/ffmpeg-filters.html#hue [03:40] <defaultro> cool. Checking it out now [03:40] <demonjester> here is one with using oss http://pastebin.com/pGg9CnYA [03:41] <demonjester> default its not for you [03:41] <demonjester> sorry [03:41] <demonjester> here is the one without using oss http://pastebin.com/LfXsZ7rp [03:41] <defaultro> k [03:41] <llogan> demonjester: you have no mp3 encoder [03:42] <llogan> missing --enable-libmp3lame [03:42] <demonjester> so in my command i add --enable-libmp3lame? [03:42] <llogan> no. in your ffmpeg ./configure [03:42] <llogan> did you compile it? [03:42] <demonjester> kk [03:43] <demonjester> no i used opkg install in openwrt it did the configuring [03:43] <llogan> if you have the lame binary you could possibly pipe to lame [03:44] <llogan> ffmpeg -i input -f wav - | lame - output.mp3 [03:44] <demonjester> i downloaded a package where i thought it was lame but said lame is not found and idk what the package is in openwrt for lame [03:46] <llogan> you'll need to install lame headers first before you compile ffmpeg with --enable-lame [03:46] <llogan> *--enable-libmp3lame [03:48] <relaxed> llogan: Hey. What kernel does it ship with? [03:48] <demonjester> i cant fine headers for lame so it seems ffmpeg is out of the question. [03:48] <demonjester> thanks anyways [03:49] <llogan> https://dev.openwrt.org/browser/packages/sound/lame/Makefile [03:49] <defaultro> is it possible to send output to mplayer? This way I don't have to create a temporary file [03:50] <llogan> why do you want to do that? so you can see the effects of the filter? [03:50] <defaultro> yup [03:51] <llogan> ffplay -i input -vf hue [03:51] <defaultro> ah [03:54] <defaultro> llogan, can ffmpeg manipulate each RGB channel? [03:56] <defaultro> here is what I did. I took a snap of the problematic video and opened it in Photoshop. I modified hue settings and I was able to correct the sky. :) FFMPEG's hue filter should be able to do it too :) [03:59] <relaxed> llogan: I've been working on a distro agnostic guide for compiling ffmpeg to a custom locations, and avoiding package managers entirely. I could add a blurb about building it statically too. [04:00] <llogan> relaxed: sounds good to me. are you going to add it to the FFmpeg wiki? [04:01] <llogan> defaultro: i'm not sure. [04:01] <relaxed> I may, at first it will be a simple txt file in dropbox. [04:02] <relaxed> If work is slow tonight I'll finish it up and let you look at it. [04:05] <llogan> relaxed: thanks. [04:11] <defaultro> llogan, I can't get this to work with vf, mp=eq2=1.0:2:0.5 [04:11] <defaultro> it's contrast brightness [04:13] <ivor> I'm looking at http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide and notice many errors making it obvious nobody uses it [04:13] <ivor> Is there a better page for this? [04:14] <defaultro> llogan, i got it to work [04:17] <ivor> What does one do when one wants to compile the latest version of ffmpeg on linux mint? [04:19] <llogan> ivor: what are the many errors? [04:20] <llogan> defaultro: what did you do? [04:20] <ivor> Well, take a look at the page at http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide and notice how it deletes yasm and then pulls down a new version but never sets the path [04:21] <defaultro> i wrapped it with double quotes [04:21] <llogan> yasm in the repo is too old [04:21] <defaultro> however, i can't combine hue and contrast/brightness [04:21] <ivor> llogan: exactly, so the script attempts to download a new version but then never uses it [04:21] <llogan> the path defaults to /usr/local [04:22] <ivor> llogan: well the version that is pulled down only gets installed into the ~/yasm-1.... directory [04:23] <ivor> llogan: so if the script is used then it complains that there is no yasm on the system [04:23] <ivor> llogan: that error can be fixed if one updates their path to include the location the script installs yasm but that's just the first error with the script/directions [04:25] <defaultro> llogan, I can't get this to work. ffplay -i 00107.MTS -vf "mp=eq2=0.0:1:-.1:hue=h=26" [04:25] <defaultro> the hue is not working [04:26] <defaultro> ah. I got it to work using comma :) [04:27] <llogan> ivor: works for me. sounds like you skipped the checkinstall command. [04:27] <llogan> that guide will be changed soon (relatively) so there is no system installation [04:27] <llogan> and no screwing around with checkinstall/package manager. [04:28] <llogan> defaultro: which Panasonic model? [04:28] <defaultro> TM700 [04:28] <defaultro> Quality is amazing but the cyan/green cast is so annoying [04:29] <defaultro> I'm not sure if I should sell it and buy another video camera [04:29] <llogan> it occurs with default settings? are you sure it's not set to tungsten white balance when shooting in daylight? [04:29] <defaultro> yup, it's AWB [04:29] <llogan> do you have a small sample? [04:29] <defaultro> so many people have the same issue [04:29] <llogan> lame. [04:29] <defaultro> i can extract few frames [04:30] <defaultro> there are so many blogs about it [04:30] <llogan> i have a HMC150 laying around here that seems fine [04:30] <ivor> llogan: I'm not familiar with what checkinstall does but I did copy it in as shown... [04:31] <ivor> llogan: I'm reading the man pages on checkinstall now [04:32] <llogan> you can see if it installed with: dpkg -L checkinstall [04:33] <llogan> or: dpkg -s checkinstall | grep Status [04:33] <ivor> llogan: What would I be looking for in there, yasm, ffmpeg, or something else? [04:33] <llogan> duh... i meant yasm, not checkinstall [04:34] <llogan> dpkg -s yasm | grep Status [04:34] <llogan> should show "Status: install ok installed" [04:35] <ivor> llogan: no, it did not get installed [04:35] <ivor> llogan: Do I need to run the script with sudo? [04:35] <llogan> what script are you referring to? [04:36] <ivor> llogan: oh, sorry, I put the directions into a script so I could be sure I got everything right [04:36] <defaultro> llogan, where can I upload the 5 sec extracted clip? [04:37] <llogan> you don't have your own server? [04:37] <defaultro> not anymore, I shut it down llast week [04:37] <llogan> let me guess...you got rid of the neckbeard too? [04:37] <defaultro> :D [04:37] <llogan> get. out. [04:37] <defaultro> let me turn on httpd here [04:37] <llogan> datafilehost or mediafire are fine [04:38] <defaultro> oh ok [04:38] <llogan> i'd like to shut mine down, but i still have some legacy clients. im bored of server admining [04:39] <defaultro> :) [04:39] <defaultro> i didn't want to pay $120/year [04:40] <defaultro> upload should be done soon [04:40] <llogan> we're paying more than that per month [04:40] <llogan> not for long though [04:42] <llogan> ivor: as i mentioned earlier, it seems as if checkinstall was skipped or did not run successfully [04:43] <defaultro> :) [04:44] <defaultro> i'm only uploading 4.5mb but it's taking a long time [04:44] <defaultro> http://www.datafilehost.com/download-48faa237.html [04:45] <defaultro> maybe the reason why I'm having a hard time correcting the sky is because I used a circular polarizer [04:45] <llogan> where is that? [04:45] <ivor> llogan: I guess I have to go back and look at my script [04:46] <defaultro> Playa del Carmen, Mexico [04:47] <llogan> it looked familar but i couldn't think of the name [04:47] <defaultro> Blue something. Saw nude women there [04:48] <llogan> were they cyan? [04:48] <defaultro> yes [04:48] <defaultro> beautiful water [04:48] <defaultro> or turquoise [04:49] <defaultro> but look at the sky, it's not blue but i remember it was a really nice blue sky [04:50] <defaultro> i'll check some new videos. That mexico was from 2010 [04:54] <defaultro> some of my videos have green/cyan cast [04:54] <defaultro> on some videos, I remember using custom white balance. The skies looked fine [05:10] <fenduru> Anyone know how to avoid any padding at the end of each segment when using -f segment [06:27] <bigmac> i have a question about hosting lots of large images on my local webserver... current format is jpg, can i convert them to load faster and not loose much quality? [06:44] <klaxa> you could create thumbnails and have the user click on those before downloading the real thing [06:46] <fenduru> is there a way to do a 2 step conversion in one command? i.e. mp3 > wav > ogg [06:49] <klaxa> you want to have wav and ogg? [06:49] <klaxa> in one command line, yeah with many pipes though [06:50] <fenduru> well, my problem is that using -f segment [06:50] <fenduru> playing the outputted segments in sequence has gaps [06:51] <fenduru> but it seems to be an encoder or container issue, as straight PCM isn't giving me this issue [06:51] <klaxa> ffmpeg -i mp3_file.mp3 -c:a pcm_s16le -f wav pipe: | tee somefile.wav | ffmpeg -i - -c:a libvorbis somefile.ogg [06:52] <fenduru> so I'm trying to see if decoding to PCM.wav and then splitting that guy will alleviate my problems. [06:52] <fenduru> klaxa, thanks. I had considered that I was just trying to avoid multiple ffmpeg processes [06:52] <fenduru> but I'll give that a shot, ty [06:52] <klaxa> i didn't test it, but it should work [06:57] <fenduru> well, the pipe worked. still getting gaps though [10:56] <Bor0> could anyone help me with http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe… ? I am experiencing the same exact problem [11:59] <wahaha__> There is a file named input.txt file '/home/1.flv' file '/home/2.flv' [12:00] <wahaha__> I can use the command to Concatenate 1.flv and 2.flv into a mp4 file. [12:00] <wahaha__> ffmpeg -f concat -i input.txt -c copy output.mp4 [12:00] <durandal_1707> and? [12:00] <wahaha__> When i use the following command [12:01] <wahaha__> ffmpeg -i "concat:1.flv|2.flv" -c copy output.mp4 [12:01] <wahaha__> can run ,but wrong result ,only one file 1.flv in the output.mp4 [12:01] <wahaha__> ffmpeg -f concat -i input.txt -c copy output.mp4 is ok [12:02] <wahaha__> ffmpeg -i "concat:1.flv|2.flv" -c copy output.mp4 ,problem [12:02] <durandal_1707> perhaps it needs full path [12:04] <wahaha__> wait a minute ,let me try [12:09] <wahaha__> 'full path' can not solve it [12:10] <durandal_1707> with -v debug [12:11] <wahaha__> wait more moment [12:12] <wahaha__> i am in china , www.pastie.org was blocked by chinese gfw [12:15] <wahaha__> i have to prepare a ladder to break the blocked gfw [12:15] <durandal_1707> wahaha__: couldn't you just find paste site that is not blocked? [12:25] <wahaha__> please wait a moment [12:28] <relaxed> wahaha__: mkvmerge -o temp.mkv 1.flv +2.flv; ffmpeg -i temp.mkv -c copy done.mp4 [12:29] <wahaha__> let me try [12:30] <relaxed> ffmpeg's concat only works well with mpeg containers from what I've seen. [12:31] <durandal_1707> relaxed: but op issue is that one syntax works and another one do not [12:32] <wahaha__> mkvmerge -o /tmp/temp.mkv /tmp/1.flv +/tmp/2.flv bash: mkvmerge: command not found [12:32] <wahaha__> apt-get install mkvmerge Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package mkvmerge [12:34] <wahaha__> now i can get pastie.org,but i can't use it,it is difficult to use [12:35] <wahaha__> do you have email?i can send you my screen-grap picture [12:36] <relaxed> durandal_1707: oh :) [12:38] <durandal_1707> wahaha__: pastebin is blocked too? [12:39] <wahaha__> my screen-grap picture is on my desktop in png file,how can i paste it into the pastie.org [12:39] <durandal_1707> paste text [12:39] <durandal_1707> copy -paste text [12:40] <wahaha__> i have break it with some tools for several minutes,i can link pastie now [12:41] <wahaha__> had never used pastie.org [12:41] <durandal_1707> pastie* sites is for copy-paste of text [12:41] <durandal_1707> for uploading images there is bunch of other hosts [12:43] <wahaha__> here it is [12:43] <wahaha__> http://pastie.org/6124324 [12:46] <wahaha__> http://pastie.org/6124377 [12:46] <wahaha__> please see http://pastie.org/6124377 [12:50] <wahaha__> are you still there durandal_1707? [12:54] <leosaeba> hi to all. I'm trying to compile a self contained (not system installed) opencv (with ffmpeg video capabilities enabled) on raspberrypi to be used by an application I'm writing. [12:54] <leosaeba> I have problems compiling it [12:55] <leosaeba> I managed to compile and install x264, ffmpeg and opencv on x86_64 [12:55] <leosaeba> then I tried to do the same on the raspberry [12:55] <leosaeba> but opencv fails with: lib//libavcodec.so.54: undefined reference to `ff_flacdsp_init_arm' [12:56] <wahaha__> durandal_1707,what is the matter? [12:57] <durandal_1707> wahaha__: open bugreport [12:58] <burek> leosaeba, are you sure you really want to use libx264 on RPi? [12:59] <leosaeba> I'm pasting the whole compilation process on http://pastebin.com/G2GQx1y7 [12:59] <burek> no need to look at that [12:59] <leosaeba> burek: for now I'm just trying to get things up [12:59] <burek> most probably you'll get unusable system [12:59] <wahaha__> think you [12:59] <burek> and probably shouldnt waste your time with it [13:00] <burek> RPi is just to weak for any kind of video encoding [13:00] <leosaeba> I just need opencv to append jpeg frames on a video (after some computations) [13:00] <burek> find some better hardware for such purpose [13:00] <leosaeba> (my main constrain is money....) [13:00] <burek> well, ok, good luck with it, what can i say.. [13:01] <burek> i assume you'll use some usb webcam [13:01] <burek> so most probably you'll have serious usb issues too [13:01] <burek> im just trying to save your time=money here [13:01] <leosaeba> burek: [13:02] <leosaeba> no, actually I have remote nodes (with usb camera) that mount raspi filesystem [13:02] <burek> your RPi will generate video frames? [13:02] <leosaeba> they write frames there and raspi do some little assembly. [13:03] <burek> what do you mean by "remote nodes" ? [13:03] <leosaeba> rpi will append received frames to a video [13:03] <leosaeba> tcp/ip [13:03] <burek> i see [13:03] <burek> that might maybe work [13:03] <burek> but then you dont need libx264 on RPi [13:03] <leosaeba> I think so [13:03] <burek> you can remove it [13:04] <leosaeba> and ffmpeg will allow opencv to use VideoWriter? [13:04] <burek> just so you know, on RPi, ethernet controller is usb-based, so you won't have 100 mbps if you expect that too [13:04] <burek> the most you can hope for is < 5-6 mbps [13:05] <leosaeba> the problem I got on raspi (not on pc) is about flac-arm something [13:05] <burek> im not familiar with opencv, so you just need to try and test it [13:05] <burek> flac is an audio codec [13:05] <leosaeba> I've done some math, and for the bandwidth required it should suffice.. [13:05] <burek> avoid any kind of encoding on RPi.. [13:07] <burek> things might seem to be working at first, but you'll notice that at some point your RPi got frozen [13:07] <burek> rebooted itself and stuff [13:07] <leosaeba> I actually could work with uncompressed video, but my problem now is just to set opencv up. [13:07] <burek> so don't consider is as a cheap version of less-powerfull x86 machine [13:07] <burek> it* [13:07] <burek> what is the image size of the video [13:08] <burek> it shouldn't be a problem to set opencv as long as it doesnt require ffmpeg's encoding abilities [13:08] <burek> if you only need muxing, that might work [13:08] <leosaeba> burek: I'm not happy about broadcom gpu too....but I accepted the compromise on openness/cost [13:09] <burek> low price comes at the cost of everything else [13:09] <burek> you'll experience it sooner or later [13:09] <burek> i just wouldnt like that you spend a lot of time setting all up just to realize what i said above [13:09] <leosaeba> imagesize is not a constrain...I will start with HD (720 I believe) and gracefully downgrade until the system works. [13:10] <burek> forget hd.. [13:10] <burek> i had a camera with internal h264 encoder [13:10] <burek> and all RPi was supposed to do was to mux the stream and re-stream it [13:10] <burek> and it failed.. [13:10] <burek> so.. i think the math is pretty obvious [13:11] <leosaeba> I see... [13:11] <burek> cpu could manage it all, but the data bus couldn't [13:11] <burek> everything was usb-based and all the data travel over that usb bus [13:11] <leosaeba> My remote node feeds is jpeg frames [13:11] <burek> so conflicts are numerous at least [13:11] <burek> that's why i noted the ethernet usb controller too [13:11] <leosaeba> i understand [13:12] <leosaeba> well..anyway I will continue to try (at leas until I will reach the bottle-neck you are referring to) [13:12] <burek> ok [13:12] <leosaeba> ..I've got no alternatives right now [13:13] <Bor0> could anyone help me with http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe… ? I am experiencing the same exact problem [13:13] <burek> you have an rpi compilation guide on our wiki though [13:13] <burek> it might help [13:13] <leosaeba> actually I have set up slackwarearm, and everything is ok. [13:14] <burek> this "lib//libavcodec.so.54: undefined reference to `ff_flacdsp_init_arm'" means that you need a specific version of ffmpeg [13:14] <leosaeba> compilation is slow, but yesterday I compiled ffmpeg (stripped of a lot of things) in a few hours. [13:14] <burek> which was in mind of opencv programmers at the time they decided to use ffmpeg [13:14] <leosaeba> I compiled ffmpeg-1.1.2 [13:14] <burek> and you probably got latest [13:14] <burek> so it failed [13:14] <leosaeba> ah! [13:15] <leosaeba> I didn't think of that. [13:15] <leosaeba> on opencv documentation they refer to minimum version of ffmpeg, so I thought to use the lates version. [13:16] <burek> well see what they recommend [13:16] <burek> if it is 0.x then use 0.x [13:16] <burek> 1.x had a lot of changes [13:16] <burek> so it might be no wonder why it doesnt work [13:16] <leosaeba> I see [13:17] <leosaeba> I got tricked by the fact that everything worked on x86_64....but the error is arm related. [13:17] <burek> hmh.. most probably [13:18] <burek> on your x64, you have both ffmpeg on your system [13:18] <burek> one installed through package management [13:18] <burek> and another (unneeded) your own compiled [13:18] <leosaeba> Now I will try with the latest 0.x version (in the meantime I investigate which version of ffmpeg opencv 2.4.3 used) [13:19] <leosaeba> ...to avoid this chance I unistalled system ffmpeg ( but I think something could still be present with vlc) [13:19] <burek> well, dpkg -l | grep libav [13:20] <leosaeba> dpkg is debian/ubuntu? [13:20] <burek> yes [13:20] <leosaeba> I will check slackware way, just a sec [13:22] <leosaeba> I've looked in /usr/lib64 for any libav* (none present) [13:22] <leosaeba> then I checked if vlc used some libav* somewhere else [13:23] <leosaeba> and I found this: /usr/lib64/vlc/plugins/codec/libavcodec_plugin.la and /usr/lib64/vlc/plugins/demux/libavformat_plugin.la [13:24] <leosaeba> but I don't think it is it. [13:24] <burek> Bor0, it might be easier to ask the author of those tutorials, rather than us here, because i dont believe anyone here has got time to read those coding samples just in order to tell you what the issue was [13:25] <burek> libavcodec_plugin.la is vlc's wrapper for libav* libraries [13:25] <Bor0> I see. is there any official tutorial for this? [13:25] <burek> installed libav* libraries [13:25] <burek> Bor0, also see docs/examples in your ffmpeg dir [13:26] <Bor0> alright, will check those, thanks. [13:26] <burek> :beer: :) [13:28] <Bor0> yoghurt with burek ;) [13:28] <burek> +1 :) [13:37] <leosaeba> burek: Ok, now I'm trying to check if using ffmpeg-0.11.2 (without x264) works on my pc, then I will try the same on raspi. In the meanwhile thank you for helping. [15:49] <jarno> Hi all, I have a question about ffmpeg/libx264...am I in the right place to make such a question? [15:50] <durandal_1707> yes [15:52] <jarno> ...great...I'm using ffmpeg - libx264 to write video (actually it's OpenCV that's calling ffmpeg) and I get the infamous "broken ffmpeg default settings detected" [15:53] <jarno> I'm using Kubuntu 12.10 and OpenCV, ffmpeg, x264 etc. is all git:ted today (should be stable) and I was wondering from where (which source code file) the default settings get written from [15:55] <jarno> ...so that I could experiment with different settings...I wonder if it is the static const AVCodecDefault x264_defaults[] in libx264.c [15:56] <jarno> ....I would really appreciate if someone could set me on the right track..... [15:56] <durandal_1707> you should see what OpenCV use/calls and what expects [15:57] <durandal_1707> "because broken ffmpeg default settings detected" in nonsense [15:57] <jarno> I was checking out the OpenCV part and as far as I can see, it would seem that so called "ffmpeg" defaults are being used [15:57] <durandal_1707> than call ffmpeg with other options [15:58] <jarno> ...hmmm, what do you mean that "broken ffmpeg default settings detected" is nonsense? [16:00] <durandal_1707> broken in what sense? what is broken and why it is broken? .... [16:00] <jarno> ...I mean that I get the following error message from libx264: [libx264 @ 0x12c7440] broken ffmpeg default settings detected [16:04] <durandal_1707> what ffmpeg version are you using? [16:06] <jarno> ffmpeg version N-38355-ge2b703f Copyright (c) 2000-2013 the FFmpeg developers [16:06] <jarno> ...I just git:ted it today... [16:08] <Mavrik> huh [16:08] <Mavrik> that "broken ffmpeg settings" is a message from x264 [16:08] <Mavrik> very old [16:08] <Mavrik> jarno: default settings aren't read from files [16:08] <Mavrik> you set them via context options [16:08] <jarno> ...yes, I know, but as far as I understand it, the default settings are written by libffmpeg, but I might be mistaken... [16:08] <Mavrik> however if you get that error the OpenCV bindings must be ancient [16:09] <jarno> ...ok...that might be the case [16:10] <jarno> ...since I am not an expert (very far from it), from where I should search for examples etc. in order to get a better idea of how to use libffmpeg? [16:11] <Mavrik> hmm, no idea [16:12] <Mavrik> since libffmpeg seems like some 3rd party product :\ [16:12] <jarno> really? [16:12] <Mavrik> yep, ffmpeg libraries are named libav* (libavcodec, libavformat, libavutil) [16:12] <Mavrik> and have such prefixes when calling them (avformat_, avcodec_) [16:13] <Mavrik> libffmpeg seems like some wrapper around them to ease programming [16:13] <jarno> ...ok... [16:13] <Mavrik> sadly, i gotta run now, hopefully you find a solution :) [16:13] <jarno> thanks!!!! [16:13] <jarno> ...hmmm...perhaps I should stick to gstreamer then...it seems to be a more straight forward solution... [16:22] <JEEB> jarno, depends on what you're doing [16:23] <jarno> ...I'm reading images from uEye cameras... [16:23] <jarno> ...and then I just need to write them to a video without loosing too much information... [16:24] <jarno> ...I'm not really an expert when it comes to gstreamer nor ffmpeg, but I have done some simple stuff using gstreamer and it seems to be fairly well documented [16:24] <jarno> ...and "easy" to use... [16:24] <JEEB> well, I've just seen way too many people derp at it over at #x264 , but I have no idea what's the best API to do the camera reading for you to be honest :s [16:25] <jarno> ....actually, I'm getting the images using the API from uEye.... [16:25] <jarno> but writing videos does not function in Linux with the same API [16:25] <JEEB> Basically I'd use whatever is the best API to grab the pictures, depending on the colorspace push it through swscale or something to convert it if your output needs to be something != input, then encode with libx264, and then mux into a container with libavformat or something [16:27] <jarno> I'll check out what you are proposing [16:27] <jarno> ...and see what is the "easiest" / "fastest" way of doing it for the time being [16:29] <jarno> but I quess that what Mavrik was saying about the OpenCV bindings is right [16:29] <jarno> ...they are just damn old...but I'll check that out tomorrow [16:30] <JEEB> I have no idea of your use case, but I am kind of reading through the lines that you might get raw video frames from the camera, in some colorspace -> if the colorspace is something you don't want to encode as-is (f.ex. RGB with libx264 is possible, but you won't really be capable of playing it with anything but FOSS, no flash or hardware decoder will like that) you feed the raw frames to swscale, or another converter of your choice [16:30] <JEEB> that converts the picture to the colorspace you want -> libx264 itself to encode the pictures into raw H.264 (Annex B) -> take the data that libx264 outputs and put it through libavformat to put the raw H.264 streams into a container [16:30] <JEEB> and that would be it [16:30] <JEEB> libswscale and libavformat are libraries that ffmpeg provides, and libx264 is well, WhatItSaysOnTheTin [16:33] <jarno> ...thanx a lot for the description man!!! [16:34] <jarno> I really appreciate it!!! [16:43] <jarno> ...since I'm pretty noob when it comes to this, from where I should/could start learning about using libav:s?... [16:44] <JEEB> depends on what exactly you want to do? [16:44] <JEEB> I mean, the trunk doxygen is as good as it gets [16:45] <JEEB> (you will rip your hair off at some point, but on the other hand that unfortunately generally is better than the alternatives) [16:45] <jarno> ...just encode and write into a container... [16:45] <jarno> ...so there are no good "tutorials" going around as normally? [16:46] <JEEB> I would probably recommend just using libx264 itself for the encoding, thus x264.h is the thing to read through [16:46] <JEEB> and then libavformat, well that is in the libavformat doxygen [16:46] <jarno> ok [16:46] <JEEB> and yes, there are samples but if they are out-of-date you get the fun time of screaming at the developers [16:46] <JEEB> http://libav.org/doxygen/master/index.html [16:47] <jarno> ...hahahahah :D [16:47] <JEEB> the doxygen is auto-generated so you at least have it be generally up-to-date [16:48] <JEEB> also do make sure what the picture format of what you're getting from the camera and what you want to feed to x264 are the same thing, otherwise you'll have to stick libswscale in the middle [16:48] <JEEB> or something else [16:48] <JEEB> which pain to take is up to you [16:49] <jarno> ...ok...happily I know what the image format is and can actually change it using the API supplied by the vendor... [16:49] <JEEB> okies, as long as that is something you a) can and b) want to stick to x264 :) [16:50] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=tree;f=doc/examples;h=36f7e2da0aa85… <- and the samples are in doc/examples btw [16:50] <JEEB> I would guess scaling_video is swscale stuff and muxing is well, that [16:51] <jarno> ...thanx...I'll check it up tomorrow...gotta go now...and once more, thanx for your help... [16:51] <jarno> cheers!!!!! [19:00] <Sam_> yesterday i had asked one question about high CPU usage in ffmpeg, one fellow member said update ffmpeg to latest version. I did migrate to latest version but still i see during flv transcoding CPU reaches about 120% [19:01] <Sam_> any suggestions what could be the reason behind it? [19:03] <Sam_> why ffmpeg is eating up so much CPU during transcoding? [19:03] <sacarasc> Because it's trying to go as fast as possible. [19:03] <sacarasc> You could make it go real time, which would presumably use less CPU... [19:03] <stqn> you could use https://github.com/opsengine/cpulimit [19:03] <Sam_> <sacarasc> but this will reduce throughput right? [19:05] <sacarasc> If you want it as fast as possible, it's going to use as much CPU as possible. [19:06] <Mavrik> Sam_: well it's supposed to use ALL CPU [19:09] <Mavrik> your question is... strange. [19:09] <Sam_> i am limiting it to 1 thread [19:09] <Sam_> <sacarasc> Correct but while doing 20 parallel transcoding it reduce the throughput of the system by significant amount [19:09] <Sam_> <Mavrik> can you please put some light why question is strange [19:26] <framer99> I have a working ffmpeg cmdline that has jitter when run as part of initscripts on centos 6.3 [19:27] <framer99> I've tried many many things to figure it out, no luck. I can restart the process after bootup and it runs clean with no jitter. [19:27] <framer99> full details here : http://pastebin.com/L2wwX3iP [19:28] <framer99> I had it setup as a Upstart process/task and I could just ssh in and restart that Upstart task and it will run fine. I just can't figure out what is different about running during boot [19:29] <framer99> thansk in advance for any advice [19:29] <durandal_1707> that is so old, more than 2 years [19:30] <framer99> yeah i was goign to compile myself eventualy just never got to it.. let me do that and report back [19:55] <framer99> rrrr.. remaking with x11grab support [20:22] <charleszivko> in a program like Audacity I can see the waveforms of the each file and manually match the peaks of them. I'm probably using the wrong term , sorry. [20:22] <charleszivko> I know programs like final cut pro auto match two audio files [20:23] <charleszivko> trying to figure out how I can implement such a feature myself [20:24] <durandal_1707> you mean same data in several bytes? [20:25] <charleszivko> similar, sound files are recorded on different devices. A mic. B video from a iPhone [20:25] <charleszivko> so it's not exactly the same data [20:26] <durandal_1707> not currently, but nobody stops you to fill feature request on bug tracker for such audio filter [20:28] <charleszivko> cool. any pointers on a alternative library that may do this? [20:28] <durandal_1707> if i know one, i would already told you, i do not think sox have anything like that [20:29] <charleszivko> oh thanks, yeah I have been researching and already found sox. doesn't seem to handle what I am looking for either. [20:29] <charleszivko> appreciate your input [20:32] <durandal_1707> there is showwaves filter so you could just watch it ;) [20:33] <charleszivko> haha, yeah I found that in the docs. Was hoping I missed something that I might be able to leverage. [20:34] <durandal_1707> but you are basically looking for something like diff tool [20:35] <charleszivko> yeah. I know audio A and audio B have a match within a 1 - 2 second timestamp. So I need to analyze those to get the exact timestamp from each where they first match so I can use it as a offset to sync them up. [20:36] <charleszivko> B would replace A (video audio) but needs to match the video [21:00] <framer99> durandal_1707: build from latest git seems to be running smoother from initscripts now.... thanks for the pointer that I was way out of date, should have paid more attention [22:09] <FFmpeg> Hello friends :) [22:09] <fatpony> can somebody explain me why cropdetect constantly yields values that very clearly overcrop the source? [22:13] <FFmpeg> I am having some trouble with ffserver. I am trying to live stream via asf. Everything seems to be working, but when I try to open my stream in windows media player I get an error that it can't play the file. [22:48] <llogan> fatpony: adjust the limit threshold? [22:49] <fatpony> llogan: i tried to do that but i don't get realistic values [22:49] <fatpony> low threshold = less aggressive crop, right? [22:49] <llogan> yes, AFAIK [22:54] <llogan> you could use drawbox to get an idea using the values from cropdetect [22:54] <llogan> ffplay -f lavfi color=s=1024x720:c=blue -vf "pad=iw:ih+80:0:40,drawbox=0:40:1024:720:red@0.5" [22:54] <llogan> or similar [22:54] <llogan> of course your input will be a video and not a color source from lavfi [22:55] <llogan> ...and you won't need pad obviously [23:23] <fenduru> I have a .wav (pcm_s24le encoded) that ffprobe reports as 9.97s long, however when I reencode that to aac (.mp4), ffprobe reports 10.00 seconds [23:24] <fenduru> How can I avoid this? I need the .mp4 to be precisely the same as the wav in terms of duration [23:27] <fenduru> encoding to mp3 gives it a duratnoi of 10.03s [23:27] <fenduru> So I'm assuming it is a container issue (i.e. frame length is fixed, so the end is padded), but I need to force it to be precise [00:00] --- Wed Feb 13 2013

1 0

ffmpeg-devel.log.20130211
by burek 12 Feb '13

12 Feb '13

[01:11] <cone-654> ffmpeg.git 03Michael Niedermayer 07master:8491ac3ad2e9: mlpdec: check ch_assign more completely [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:55e35c98d5f3: h264 intra pred: add missing includes. [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:c7e3e55429cb: Move ff_emulated_edge_mc prototypes to videodsp. [01:11] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:5e74a2b68563: bit_depth_template: don't depend on dsputil.h. [01:12] <Darkarnium> Hey Compn, sorry, I didn't see your message last night until late [01:18] <Darkarnium> Pretty damned nasty, and only being used in a development environment as a PoC at the moment: http://pastebin.com/3LNcery6 [01:19] <Darkarnium> Worth noting, I'm using the snprintf and av_dict_set and 'headers' as 'cookie' does not seem to apply for my test. Whether I specify the options as per RFC6265 or as just an attribute value pair. [01:30] <wm4> someone is reporting broken decoding with h264_vda in recent ffmpeg git (might be my fault, but I don't know and it used to work fine) [02:36] <cone-654> ffmpeg.git 03Ronald S. Bultje 07master:fd6a021d8e3d: get_bits: return pointer to buffer that is the result of the alignment. [11:31] <Tjoppen> someone's improving my cinepak encoder toy. this makes me happy :) [11:50] <durandal_1707> i want that filter! - http://slowmovideo.granjow.net/videos.html [11:53] <Tjoppen> ooh [11:59] <durandal_1707> michaelni: when gbrp16 output? [12:09] <cone-615> ffmpeg.git 03Paul B Mahol 07master:5d7009dac28b: evrcdec: fix wrong condition [12:10] <michaelni> ill look into adding gbrp16 support if noone else does ... [12:17] <someone-noone> Hello! Question related to vda_h264 decoding. Does VDADecoderDecode in sync mode returns frames in display order? Docs are saying that frames are always in decode order, however ffmpeg reorders it only in async mode and do not reorder in sync mode. I'm confused. Is there smth I've missed in sources? [12:23] <michaelni> maybe ask dilaroga or xidorn quan [12:27] <someone-noone> thanks. How often are they online on this channel? [12:27] <someone-noone> or better try mail? [12:28] <cone-615> ffmpeg.git 03Diego Biurrun 07master:d46d87d1d3de: sparc: dsputil_vis: Fix silly variable name search and replace typo [12:28] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:c3a62d3507b2: Merge remote-tracking branch 'qatar/master' [12:31] <michaelni> probably better mail, ive seen dilaroga here but not common enough for waiting to be a good idea IIRC [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:272e7f6443b7: huffyuvdec: Check init_vlc() return codes. [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:ac476bfa9f90: huffyuvdec: Skip len==0 cases [12:41] <cone-615> ffmpeg.git 03Reinhard Tartler 07release/0.5:5c9d2d837783: Bump version number for the 0.5.10 release [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:0f6d4da8def2: bmpdec: only initialize palette for pal8. [12:41] <cone-615> ffmpeg.git 03Mina Nagy Zaki 07release/0.5:4475a7d88b89: lavfi: avfilter_merge_formats: handle case where inputs are same [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:6b97e76dfca8: avsdec: Set dimensions instead of relying on the demuxer. [12:41] <cone-615> ffmpeg.git 03Aneesh Dogra 07release/0.5:c28c631d29a8: bytestream: add a new set of bytestream functions with overread checking [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:d1729c3715af: avidec: use actually read size instead of requested size [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:4fac60d56863: cavsdec: check for changing w/h. [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:2ae6bdbb9b17: avidec: return 0, not packet size from read_packet(). [12:41] <cone-615> ffmpeg.git 03Anton Khirnov 07release/0.5:d4e4234147c9: yuv4mpeg: return proper error codes. [12:41] <cone-615> ffmpeg.git 03JindYich Makovi ka 07release/0.5:6731776795a8: h264: avoid stuck buffer pointer in decode_nal_units [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:5235db68c062: mpegaudiodec: fix short_start calculation [12:41] <cone-615> ffmpeg.git 03Alex Converse 07release/0.5:80f89a9b40cc: tiffenc: Check av_malloc() results. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:1f1b2f18067b: yuv4mpeg: reject unsupported codecs [12:41] <cone-615> ffmpeg.git 03Janne Grunau 07release/0.5:2e1474fd9988: lavf: avoid integer overflow in ff_compute_frame_duration() [12:41] <cone-615> ffmpeg.git 03Dale Curtis 07release/0.5:c3761b661874: Fix uninitialized reads on malformed ogg files. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:fe4409a396d7: oggdec: check memory allocation [12:41] <cone-615> ffmpeg.git 03Janne Grunau 07release/0.5:a49599b1255e: h264: check context state before decoding slice data partitions [12:41] <cone-615> ffmpeg.git 03Justin Ruggles 07release/0.5:7aeb281aa507: shorten: check for realloc failure [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:9def5c466648: shorten: Use separate pointers for the allocated memory for decoded samples. [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:4f8f4458a5a8: vp56: release frames on error [12:41] <cone-615> ffmpeg.git 03Luca Barbato 07release/0.5:b9500bf864e9: vp6: properly fail on unsupported feature [12:41] <cone-615> ffmpeg.git 03Michael Niedermayer 07release/0.5:10ec2308b0d4: Merge remote-tracking branch 'qatar/release/0.5' into release/0.5 [12:57] <durandal_1707> michaelni: what was idea of noise filter? and how addgrain (film grain) relates to it? [13:12] <cone-615> ffmpeg.git 03Carl Eugen Hoyos 07master:182e342fdbc7: Print mxf Universal Label if the user requested verbose logging. [13:19] <durandal_1707> shouldn't colormatrix filter actually use libswscale? [13:21] <michaelni> the idea of noise was to add different kinds of noise and grain ... [13:24] <durandal_1707> i don't think there is way in lavfi to overlay frames with differen opacity [13:52] <durandal_1707> how should i name blend filter? mix/merge/blend [13:56] <iive> mix sounds too audio related. [13:58] <durandal_1707> iive: its called amix [14:00] <iive> the filter may be called amix, but I do associate the mix word with audio. [14:12] <durandal_1707> ubitux: want to add different edge detectors (not just canny) to edgedetect? [15:23] <ubitux> durandal_1707: i called the filter "edgedetect" and not "canny" for that reason, so feel free to add some enhancement [15:26] Action: ubitux is going back tomorrow :( [15:44] <durandal_1707> ubitux: i already have bunch of other work, and you wrote edgedetect and are most qualified person to extend it [15:46] <cone-615> ffmpeg.git 03Paul B Mahol 07master:8160864b0027: lavfi/amerge: make description consistent with documentation [16:02] <ubitux> durandal_1707: i have no plans for it atm [16:21] <cone-615> ffmpeg.git 03Derek Buitenhuis 07master:c177f2ec4a21: libfdk-aacenc: Actually check for upper bounds of cutoff [17:14] <durandal_1707> ubitux: your opinion on blend filter? [17:14] <durandal_1707> overlay already have some blend code - very limited one though [17:16] <durandal_1707> also i dont understand this complicated logic with overlay/alphamerge request_frame thing [17:57] <durandal_1707> michaelni: i added yuv444 support to overlay and it is broken, works if i add format=yuv444p to begin and end any pix fmt of filter chain [17:58] <durandal_1707> this is strange [18:01] <durandal_1707> filter expect that overlay and main are of same format but it does not happen to actually be in config_input [18:10] <durandal_1707> negate/lut debug filter output is boring spam [18:11] <durandal_1707> it inserts scaler 444 -> 420 in middle [18:11] <durandal_1707> perhaps ffplay nonsense [18:15] <durandal_1707> indeed, it happens with ffplay only [18:25] <durandal_1707> so i want basically VV->V filter without pts nonsese stuff invading my code [18:26] <durandal_1707> is it doable? [18:30] <durandal_1707> another thing is that filters write to first frame, I dont find this much useful for blend and want to have two input read only frames and output writeonly [18:34] <michaelni> you can set permission on the inputs [18:34] <michaelni> or a filter can output read only buffers so they wont be written into [18:34] <durandal_1707> yes, I know that, but what about this PRESERVE and pts nonsense [18:35] <durandal_1707> i just read pixels from 2 inputs and write pixel to output [18:36] <durandal_1707> and both 2 inputs should have same format [18:37] <michaelni> for same format just make them use the same list and you should get the same format [18:41] <durandal_1707> michaelni: how? see bug with ffplay and overlay when i added yuv444p [18:44] <michaelni> i think vf_overlay uses seperate lists for its 2 inputs not sure thats the issue though [18:46] <durandal_1707> formats are set with query_formats and its global for all inputs/outputs [18:59] <durandal_1707> so filter_frame() can take two inputs at same time? [19:03] <durandal_1707> looks like i need to do some kind of buffering [19:04] <durandal_1707> this is nonsense, why would any filter would need to do this? [19:07] <durandal_1707> i will just copy-paste code [19:13] <durandal_1707> saste: about il: feel free to simplify options if you think that per plane options do not make sense [19:14] <durandal_1707> saste: what would be template of filter that takes 2 readonly inputs and 1 write only output? [19:14] Action: durandal_1707 thinks that having filter templates could help novice devs [19:20] <saste_> durandal_1707, overlay? [19:21] <durandal_1707> overlay writes to first input [19:21] <durandal_1707> that is useless for blend [19:23] <durandal_1707> well not always useless, but i imagine scenario where it would be useful to have original input [19:24] <durandal_1707> blend is just geq but instead on single frame it operates on VV->V [19:29] <cone-615> ffmpeg.git 03Michael Niedermayer 07master:0aa1d848ece6: ffv1enc: better heuristic to calculate initial states [20:14] Action: durandal_1707 now that is priority ^ [22:04] <llogan> why is rbg24 not automatically chosen for gif output instead of the current AV_LOG_ERROR message? [22:05] <llogan> it is confusing to users [22:05] <llogan> *rgb24 [22:18] <llogan> Compn: i'm not against your request, FYI. [22:31] <Compn> llogan : i know, i'm just pointing out -ss 30 is seconds, so -ss 30:30 is minutes+seconds [22:35] <llogan> gif appears to support pal8, (if you use -f image2) but that message does make that apparent. [23:00] <llogan> s/does/does not/ [23:07] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:972771dcf27d: h264chroma: remove duplicate 9/10 bit functions. [23:39] <BBB-work> michaelni, one more for you, a file encoded with "./ffmpeg -i bear_mpeg4_mp3.avi -acodec copy -vcodec mpeg4 -flags mv4+qpel+mv0+cbp+aic -trellis 1 -dia_size 1040 -g 250 -bf 8 -b_strategy 2 -cmp sad -subcmp rd -mbd 2 -precmp sad -last_pred 4 -subq 8 -bidir_refine 4 -qns 3 bear_mpeg4asp_mp3.avi" apparently fails to playback properly in ffplay (stutters a lot) because of a bug in compute_pkt_fields() for files with b-frames that are not [23:39] <BBB-work> h264 [23:39] <BBB-work> [let's continue that here] [23:40] <michaelni> ok [23:42] <cone-615> ffmpeg.git 03Ronald S. Bultje 07master:b582af1ed778: Use simple nop codes for <= sse (rather than <= mmx). [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:4c69cafceea5: doc/filters: fix position of il filter docs [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:af60e451509a: doc/filters: apply minor fixes to il filter docs [23:45] <cone-615> ffmpeg.git 03Stefano Sabatini 07master:64353be81135: doc/filters: rework idet filter description [23:56] <michaelni> BBB-work, i see the stuttering too without encoding with -bf 8 [00:00] --- Tue Feb 12 2013

1 0

ffmpeg.log.20130211
by burek 12 Feb '13

12 Feb '13

[00:34] <rbrauer> I'm piping raw video (rgba) to h264 in mp4 via Java which produces a valid video if I use -t to set duration. Does anyone know how I would signal ffmpeg to close the file without an explicit duration? I tried sending 0x1a, but that didn't work. I'm doing this on Windows 7. [01:00] <fenduru> rbrauer: I think 0x1A is only for text files. Are you closing the stream properly? [01:01] <rbrauer> I tired closing both input and output streams on the ffmpeg process, but it still didn't complete the file correctly [01:03] <rbrauer> after closing the video pipe it just sits there with no messages [01:03] <fenduru> are you closing the stream without sending an EOF character? [01:04] <FergusL> anybody with a mp3->ogg ffmpeg command at hand ?! I can't seem to find one working... [01:06] <rbrauer> fenduru: what byte am I supposed to send for EOF? [01:08] <fenduru> rbrauer: Honestly I'm not sure. But I would try just closing the stream without one. [01:08] <rbrauer> I tried closing it from the Java side (ffOut.close()), wasn't working [01:09] <rbrauer> I mean I get no repsonse from ffmpeg and the file isn't correct [01:09] <fenduru> What type of stream are you dealing with on the Java side? (so I can lookup javadoc) [01:10] <fenduru> FergusL: try using -acodec libvorbis and have your output file be NAME.ogg [01:10] <FergusL> fenduru: I have vorbis but not libvorbis, weird [01:11] <FergusL> oh wait [01:11] <FergusL> it's in avconv now... [01:11] <FergusL> Ubuntu here [01:12] <rbrauer> fenduru: its a plain outputStream coming out of a Process wrapped with a bufferedOutputStream [01:12] <rbrauer> I'm calling ffOut.flush() then ffOut.close() [01:22] <fenduru> I'm out of ideas. I'm not familiar with piping/stdin on windows. [01:22] <fenduru> Looking around it seems that there is no standard EOF in the spec [01:22] <fenduru> Others suggest using -1 [01:23] <fenduru> I'm piping data from node.js into ffmpeg's stdin [01:24] <fenduru> and using stream.end() to send the eof, which works great on Linux [01:24] <fenduru> Perhaps try that, and if it works take a look at node.js' source for what it sends for EOF? (or if not node.js something else that has a similar .end() function) [01:26] <fenduru> FergusL: did you figure out your problem? [01:26] <FergusL> fenduru: yes I used avconv and there I could use -acodec libvorbis [01:27] <FergusL> that's just crazy, avconv is a wrapper [01:27] <FergusL> could just be a symlink [01:27] <FergusL> to ffmpeg, but well... enough arguing over this I think [01:33] <rbrauer> I figured out what was happening I think. Apparently when I closed the BufferedOutputStream it didn't close the actual one. I did ffProc.getOutputStream.close() after closing and flushing the buffered stream and it seemed to work. [01:35] <rbrauer> well I thought that worked...maybe not [01:37] <fatpony> /join #avcon [01:37] <fatpony> duh [01:41] <rbrauer> yeah that worked my code is just really messy now and I keep putting stuff in the wrong place, thanks for the help fenduru [01:43] <fenduru> Sure thing rbrauer. [02:14] <fenduru> Any reason that ffmpeg would return successfully if it isn't segmenting the output as specified? [02:43] <fenduru> Oh, the ffmpeg in the Ubuntu repo is very out of date [02:44] <klaxa> the ffmpeg in the ubuntu repo isn't ffmpeg [02:46] <fenduru> yeah, but even the one listed on the ffmpeg site is old [02:50] <klaxa> huh? [02:50] <klaxa> fenduru: elaborate [02:52] <fenduru> klaxa: https://launchpad.net/~jon-severinsson/+archive/ffmpeg is listed as the repo to use for ffmpeg [02:52] <fenduru> but is at version 0.10 instead of 1.x [02:53] <klaxa> oh true that [02:54] Action: fenduru is compiling from source now [02:54] <klaxa> the only sane thing to do really :) [02:55] <fenduru> yep :) I've been spoiled with arch's rolling release / package system [02:56] <fenduru> although I should really just use one of the static builds that are online... this amazon EC2 server is quite slow [03:09] <asher^> is there a way to have ffmpeg add metadata to an existing file? [03:17] <klaxa> asher^: https://www.ffmpeg.org/ffmpeg.html#Main-options see: -metadata [03:17] <klaxa> to not touch the rest of the streams, do: -map 0 -c copy [03:18] <asher^> great, thanks, ill try that out [03:18] <fenduru> klaxa, any idea how to avoid choppiness when segmenting your output? [03:18] <klaxa> what do you mean with choppyness? [03:19] <fenduru> if i recombine sequential output segments, sometimes there are clicks between the segments [03:19] <fenduru> I assume there is some sort of padding going on here [03:19] <klaxa> i.e. audio clicks? [03:19] <fenduru> yep [03:19] <asher^> klaxa, Unrecognized option 'c'. Any idea why this would happen? [03:19] <klaxa> asher^: ffmpeg too old [03:20] <klaxa> fenduru: that's probably because the last sample of the prepending segment, or the first sample of the following segment are not at 0 amplitude [03:20] <klaxa> at least that's what i would think [03:21] <klaxa> those things produce audio-clicks [03:21] <klaxa> open the audio stream in a graphical audio editor and you should be able to see that [03:21] <klaxa> audacity is pretty well suited for that [03:22] <fenduru> I'll take a look. I'm segmenting for streaming on the web so I'd like to be able to reconstruct the segments losslessly [03:22] <klaxa> asher^: install a newer version from your distributions repository, if they don't offer a newer one, compile from source or grab a static build [03:22] <asher^> yeah ive got the latest from my distro, looks like ill have to compile. thanks [03:25] <asher^> also, anyone know an alternative to ffmpeg-php? it causes 502 errors in some circumstances on my machine [13:19] <veleno> hello everyone. is it possible to use ffmpeg/ffprobe to statically analyze a file and detect lost packets/lost frames ? [13:25] <durandal_1707> veleno: i guess it depends on container/codecs [13:25] <durandal_1707> which means more info is needed [13:26] <durandal_1707> some containers simply are not designed that way [13:26] <durandal_1707> also lost packets/lost frames is not same as corrupted packets/frames that fail to decode [13:27] <veleno> durandal_1707: what would be the ideal container/codecs to detect those info? i have the liberty to chhoose, a this point. [13:28] <veleno> lost packets/frame do have a mapping to corrupeted packets/frame, do they ? [13:28] <durandal_1707> corrupted/incomplete frames are reported by lavc when decoding fails [13:29] <durandal_1707> this depends on decoder, because some decoders may try to recreate data [13:29] <veleno> (when I refer to lost/corrupted packets, it imply an input file received by a network streaming, not rtp-based) [13:30] <durandal_1707> .. but than usually it is displayed in log what is happening .. [13:30] <veleno> i see. [13:31] <durandal_1707> veleno: than you need to look at that stream protocol/format [13:32] <veleno> durandal_1707: that i know that is udp. the stream is sent over the network with somehting like: ffmpeg -re -i video.mp4 -f mpegts rtp://127.0.0.1:60000 -loglevel debug -report -vstat [13:32] <veleno> with udp instead of rtp in the streaming url [13:35] <durandal_1707> then you need to find out if mpegts support some kind of packet drop detection [13:36] <mf_> I've read it somewhere, but has forgotten what it stated, but what does max-rate really make for a difference when you run 2-pass encoding? Did it distribute the bitrate more evenly across all frames, or did it "move" the unused bits from a non/barely-moving sequence to the one that has more movement? How can I optimize this further? (i.e. it it possible to look ahead a lot longer for it to calculate more accurately?). I have a s [13:36] <JEEB> in mpeg-ts there are continuity bits IIRC [13:36] <JEEB> 0-15 was it? [13:36] <JEEB> and if it doesn't go +1 then you've lost a packet [13:37] <JEEB> (an mpeg-ts packet) [13:37] <veleno> JEEB: that's useful indeed. i'm checking http://en.wikipedia.org/wiki/MPEG_transport_stream#Packet [13:37] <JEEB> then you've got the 192 and 204 byte mpeg-ts packets that contain checksums [13:37] <JEEB> I have no idea what libavformat supports and how [13:41] <veleno> and i guess there's a way to see which are the stream transport protocols available [13:41] <durandal_1707> there is some log under debug: "Continuity check failed for pid %d expected %d got %d\n" [13:42] <veleno> durandal_1707: under the logs of ffmpeg or ffprobe ? [13:42] <durandal_1707> veleno: it is in lavf, so anything that use lavf and care for its logs will get it [13:43] <JEEB> what about the checksums for >188 byte mpeg-ts? [13:47] <durandal_1707> i see some check_crc but it its result is not loged, i'm not mpegts expert and mpegts in lavf can be improved for sure [13:50] <JEEB> yeah, lavf lacks support for PID switching, too [13:53] <veleno> if a transport other than mpegts is preferrable for my goals, i'd rather switch to it. open for suggestions. [13:57] <Mavrik> JEEB: PID switching? [14:03] <JEEB> Mavrik, a PMT says that one or all of the streams will switch the PID they will be transferred over [14:03] <Mavrik> ah, yes, that's not supported :) [14:03] <JEEB> oh I know it's not supported [14:04] <JEEB> which is why I use tvtest for Tokyo MX recordings :) [14:04] <JEEB> (Tokyo MX switches between PIDs when they switch between SD and HD) [14:04] <JEEB> NHK on the other hand keeps the video PID the same afaik [14:06] <veleno> JEEP: in the mpegts packets i don't see CRC fields . where did you see the infos about checksum transported by byte 192 and 204 ? [14:06] <veleno> JEEB: ^ [14:06] <Mavrik> veleno: there's an option of adding CRC fields to the 188 byte packets [14:07] <Mavrik> if your MPEG-TS packets are 188B then you don't have those :) [14:07] <JEEB> http://wiki.multimedia.cx/index.php?title=MPEG-2_Transport_Stream#Packet [14:08] <JEEB> "Packets are most often 188 bytes in length, but some transport streams consist of 204-byte packets which end in 16 bytes of Reed-Solomon error correction data." [15:32] <iive> isn't that part of the FEC? [16:15] <skarm> Hi to all. Is it possible to stream /dev/video0 to local network, without using ffserver? [16:24] <veleno> I guess there are CRC infos in the TS packets of the stream being analyzed http://codepad.org/GrvX6094 .. [16:24] <veleno> i wonder if when using ffmpeg to stream the video it's possible to add those [16:26] <durandal_1707> veleno: yes, modify mpegts muxer and send patch [16:40] <veleno> durandal_1707: ok [17:53] <espr3ss0> hilo [17:54] <espr3ss0> need some help, is anyone here ? [17:55] <Fjorgynn> duno [17:55] <stqn> Cant help, but Im here. Hi. [17:55] <Fjorgynn> where's the cake? [17:56] <espr3ss0> lol [17:57] <Fjorgynn> espr3ss0: shoot? [17:57] <espr3ss0> anyone know how to set the timescale ? [17:57] <Fjorgynn> what? [17:57] <espr3ss0> mp4 works with QuickTime but to iPad complains [17:57] <espr3ss0> iTunes even plays fine :-/ [17:59] <klaxa> espr3ss0: a bit more information about the problem would be great [17:59] <klaxa> i.e. information about the produced file and ffmpeg output logs [17:59] <espr3ss0> " [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180e800] stream 0, timescale not set " [17:59] <klaxa> paste everything on pastebin [18:00] <espr3ss0> not wanting to do that soz ;-p [18:01] <espr3ss0> does ffmpeg have a timescale parameter ? [18:21] <durandal_1707> espr3ss0: yes [18:54] <iamtakin1iteasy> hi, which codec (and/or settings) do you recommend for recording desktop? [18:55] <iamtakin1iteasy> from x11grab for instance [18:55] <iamtakin1iteasy> so text won't get blurred [18:55] <JEEBsv> ffvhuff is a fast and simple lossless encoder [18:55] <iamtakin1iteasy> after a couple of seconds [18:55] <iamtakin1iteasy> thank you [18:56] <Guest94> in FLV transcoding using ffmpeg+libx264 i could see CPU usage is around 115% [18:56] <Guest94> any suggestions? [18:56] <Guest94> i am using single thread option [18:57] <durandal_1707> use faster preset/more threads/faster hardware/ .... [18:57] <Guest94> here is the commandline http://pastebin.com/nJpGpt26 [18:58] <Guest94> <durandal_1707> do you think using faster presets CPU usage will go down? [18:59] <durandal_1707> try it [19:00] <Guest94> let me give a shot [19:02] <Guest94> i tried superfast/ultrafast and veryfast but nothing seems to take the cpu usage below 100% [19:02] <Guest94> its always going >108% [19:05] <durandal_1707> those preset and without extra options? [19:06] <Guest94> here is the command i am using now ffmpeg -y -i /root/fGFNmEOntFA.flv -vcodec libx264 -preset superfast -threads 1 /tmp/test.3g1_super.flv [19:09] <klaxa> there is a superfast preset? [19:09] <klaxa> wtf? [19:11] <iamtakin1iteasy> also, can ffmpeg stream over the network? [19:11] <iamtakin1iteasy> without medium of local file [19:11] <klaxa> yes [19:11] <iamtakin1iteasy> how it looks like? [19:12] <Guest94> <klaxa> Read the article http://dev.gentoo.org/~beandog/x264_preset_reference.html [19:12] <klaxa> write to udp or rtp, if you have an rtmp server, you can stream to that too [19:12] <klaxa> oh cool didn't know that it had that many presets [19:13] <Guest94> now if you can suggest something which is causing high CPU usage would be great [19:13] <iamtakin1iteasy> hmm [19:14] <klaxa> well too much computation needed for compression, also ffmpeg will use as much cpu as possible to decrease encoding time [19:14] <Guest94> right but this will reduce the throughput of the system when we do multiple transcode [19:15] <klaxa> you can increase the nice-value for ffmpeg [19:15] <klaxa> however, it will still use as much cpu time as it can get [19:16] <durandal_1707> and need [19:17] <Guest94> any other better thought? [19:17] <klaxa> no [19:18] <klaxa> if you have multiple encodes at the same time obviously all encodes will be slower than if they are run sequentially [19:18] <klaxa> if you increase nice values for selected ffmpeg instances, those should work slower than those instances with unchanged nice values [19:18] <Guest94> all the encode in parallel, just like independent thread [19:19] <klaxa> yes, that doesn't increase productivity of a system [19:19] <iamtakin1iteasy> i am trying to stream x11grab over network, but apparently doing something wrong [19:19] <iamtakin1iteasy> here is my commandline output: http://paste.eientei.org/show/39/ [19:19] <iamtakin1iteasy> any ideas where i got wrong? [19:19] <iamtakin1iteasy> might it be wrong USE-flags i built ffmpeg with? [19:20] <klaxa> iamtakin1iteasy: you have to specify a container-format [19:20] <klaxa> for example: -f matroska [19:20] <klaxa> i *think* that should work, i'm not sure though [19:20] <klaxa> also, note that you will have rather high latency [19:20] <klaxa> and make sure you have high network throughput [19:21] <iamtakin1iteasy> i have added -f matroska to options list and now i am getting udp://192.168.1.4:6000: No such file or directory [19:21] <iamtakin1iteasy> 192.168.1.4 is the ip of current machine [19:21] <iamtakin1iteasy> should it be some remote listening server? [19:21] <klaxa> --disable-network [19:21] <klaxa> heh [19:22] <iamtakin1iteasy> ah [19:22] <iamtakin1iteasy> i see [19:22] <klaxa> that will be it [19:22] <iamtakin1iteasy> thank you [19:22] <klaxa> you could get it working with pipes and netcat though :V [19:22] <iamtakin1iteasy> nah, i'll just re-emerge the ffmpeg with proper use-flag [19:23] <iamtakin1iteasy> thanks for noticing [19:23] <klaxa> i can't say how reliable udp streaming is, i don't think i ever got it to work really [19:23] <iamtakin1iteasy> what about rtp servers? [19:23] <iamtakin1iteasy> is there any opensourced of them? [19:24] <iamtakin1iteasy> so i can setup it on my server with higher bandwitch [19:24] <iamtakin1iteasy> and stream to other users [19:24] <klaxa> dunno i haven't really found many good streaming solutions that are open-source [19:24] <klaxa> i think live555 is rather popular though [19:24] <Mavrik> red5? [19:25] <klaxa> oh yeah that sounds like what we are looking for [19:25] <klaxa> reading that it's implemented in java makes me think it must be bloated :V [19:26] <iamtakin1iteasy> i am java coder. i am getting paid for work with bloatware >:3 [19:26] <iamtakin1iteasy> (shouldn't be so proud of it) [19:27] <iamtakin1iteasy> but meh, there is no way i'll be scared by any kind of java monstrocity [19:27] <iamtakin1iteasy> thank you [19:57] <espr3ss0> ok ... had a look for a command to set the timescale but no luck :-( [19:58] <espr3ss0> and everybody has gone :-/ [19:59] <durandal_1707> timescale what? [19:59] <espr3ss0> an mp4 ? o_o [20:00] <durandal_1707> can you explain in finer detail [20:00] <espr3ss0> warning, " timescale not set " [20:00] <espr3ss0> i assume it's why i can't copy it to mi' iPad [20:01] Action: espr3ss0 shrugs [20:01] <espr3ss0> why need the whole info ? [20:01] <espr3ss0> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 2027 kb/s, 23.98 fps, 23.98 tbr, 96k tbn, 47.95 tbc [20:02] <durandal_1707> i cant help if i cant reproduce - is this so hard to understand? [20:02] <espr3ss0> oh, no :-) [20:02] <espr3ss0> how can u re-produce when u don't have the file ? [20:03] <durandal_1707> so instead of providing requested info you ask questions? [20:03] <klaxa> espr3ss0: simple: you can't, but you can try to encode a file with the same output parameters [20:04] <klaxa> that should produce similar encodes [20:04] <espr3ss0> ahh, i c [20:04] <klaxa> also, what durandal_1707 said, if you don't want to provide information, we can't help [20:04] <klaxa> "I cannot get my car to start, I think the battery might be empty, also it's read, how do i fix it?" [20:04] <espr3ss0> fair enuff ;-p [20:04] <klaxa> *red [20:05] <espr3ss0> i will post some info [20:05] <durandal_1707> klaxa: call 911 [20:07] <espr3ss0> http://pastebin.com/tdJ5tquf [20:11] <klaxa> looks like it's something with the input file rather than with ffmpeg [20:13] <espr3ss0> what's wrong with the input file ? [20:14] <klaxa> timescale is not set :V [20:14] <klaxa> no idea how to fix it myself [20:15] <klaxa> except maybe re-encode but you want to avoid that if possible [20:15] <durandal_1707> time scale is set from demuxer [20:15] <durandal_1707> or muxer [20:15] <durandal_1707> so perhaps png stream messed up something [20:16] <espr3ss0> tried from the original file, extract (copy) to an MKV then to MP4, which usually solves just about everything so far [20:16] <espr3ss0> oh, that's without completely re-encoding [20:16] <durandal_1707> perhaps it copy png "stream" too? [20:17] <durandal_1707> because timescale is set for video [20:17] <durandal_1707> and audio [20:17] <durandal_1707> it is just not set for cover png [20:19] <espr3ss0> stream 0.1 ? [20:19] <espr3ss0> unless that's the "Artwork" (meta), then it shouldn't be there :-/ [20:20] <klaxa> heh now that i see it... [20:20] <espr3ss0> listed in Subler is only Vid & Aud [20:20] <klaxa> 1080p video but 64 kbps 22050hz aac [20:20] <espr3ss0> ? o_O [20:20] <klaxa> the audio track is extremly low quality [20:21] <durandal_1707> espr3ss0: stream 0.2 is png [20:21] <klaxa> it's funny that one shouldn't get copied even [20:22] <espr3ss0> why not ? [20:22] <klaxa> but try: -map 0:0 instead of -an [20:23] <durandal_1707> anyway if you could make smaller sample (eg it happens when partially copied into mkv) that you could report it on trac and upload sample [20:24] <Pitfall> Hi folks [20:24] <Pitfall> I have a question about licensing [20:25] <Pitfall> I want to use FFMPEG in my project which is closed source [20:25] <Pitfall> Is that allowed? [20:25] <JEEBsv> see the LGPL license [20:26] <Pitfall> Is there a way to distinguish code between GPL and LGPL? [20:26] <Pitfall> I heard that it had some GPL too in it [20:26] <JEEBsv> yes, but if you do not do --enable-gpl ffmpeg will be LGPL [20:26] <JEEBsv> it will only compile code that is LGPL [20:26] <JEEBsv> (or LGPL-compatible) [20:26] <Pitfall> OK [20:27] <JEEBsv> you can then separately get a license from x264 LLC if you want to use x264 in a closed source app [20:27] <JEEBsv> because it otherwise is GPL [20:27] <Pitfall> We already have a x264 license [20:29] <JEEBsv> nice [20:29] <JEEBsv> anyways, just keep your ffmpeg configuration as LGPL, and you should be able to use it under the LGPL license :) [20:29] <Pitfall> OK thanks [20:29] <Pitfall> I do not want to be blamed for my actions :) [20:31] <JEEBsv> just keep away from enabling GPL or nonfree [20:31] <Pitfall> What if a library actually is GPL? [20:31] <JEEBsv> then you can't use it [20:31] <Pitfall> You cannot purchase a license for that? [20:32] <JEEBsv> no unless the developers give such out [20:32] <JEEBsv> you can always ask the developers [20:32] <JEEBsv> also, a quick go-over for how to be "OK" with LGPL [20:32] <durandal_1707> hmm, wasn't you can use gpl but using dynamic lib and not modifying code? [20:33] <JEEBsv> that's /very/ iffy [20:33] <JEEBsv> I'd rather keep away from that [20:34] <JEEBsv> LGPL: if you do dynamic linking, just the exact source code of the LGPL component(s) used in the binary [20:34] <JEEBsv> if you do static linking, the exact source code of the LGPL component(s) as well as the object files for the proprietary part [20:34] <JEEBsv> so that the user can switch around the librarie(s) if he/she wants [20:34] <JEEBsv> that is LGPL [20:34] <JEEBsv> more or less [20:35] <durandal_1707> Pitfall: read https://ffmpeg.org/legal.html [20:36] <Pitfall> I did read that, but it is pretty vague [20:36] <durandal_1707> than ask lawyer [20:37] <espr3ss0> just checked with a different file ... muxing (mp4) only a video (x264) track and audio (aac) shows as fine and i didn't realise adding meta artwork adds an extra track. but then it shows the timescale problem [20:38] <durandal_1707> it is definitely bug so report it anyway [20:38] <Pitfall> I guess I will first check if I can use FFMPEG in my project compiled as LGPL [21:08] <iamtakin1iteasy> hi, i hava a low-bandwitch netbook which screen i wish to stream over http protocol and i have a higher-bandwitch server which i want to do actual streaming to end-users. how do i best transfer the stream from netbook to server and restream for each user then? [21:08] <iamtakin1iteasy> i have tried various rtsp servers, but they don't seem to be capabel of http streaming [21:09] <iamtakin1iteasy> so you can't simply use mplayer on http stream they are [not] providing [21:09] <iamtakin1iteasy> maybe i missed something? [22:02] <Pinhole> How can I determine which extra libraries are actually being used (not compiled for)? I have two boxes with the same stream and the same ffmpeg command line. The resulting output files are very different in size. I'm trying to figure out why. [22:03] <Pinhole> Both are Fedora 17 with the same version of ffmpeg installed. One is 32 bit the other is 64 bit. The 64 bit seems right, the 32 bit is very bad. [22:03] <JEEBsv> what is being used should be visible from the terminal output, pastebin the commands and terminal output onto a pastebin provider of your choice, and then link them here [22:08] <Pinhole> Ok, I'll paste these and be back in about 40 minutes (they are in progress) [23:06] <fenduru> is there a way to use -f segment to segment audio and pipe the segments out to stdout? [23:08] <klaxa> use pipe: as an output file? [23:08] <klaxa> that will write to stdout [23:43] <fenduru> klaxa, I think I figured it out. I was overlooking the fact that aac needs a seekable output [23:43] <klaxa> kk [23:48] <humans63> !lista [23:48] <humans63> ciao [00:00] --- Tue Feb 12 2013

1 0

ffmpeg-devel.log.20130210
by burek 11 Feb '13

11 Feb '13

[00:44] <Zeranoe> Slightly off topic here but anyone know why patches are created with -up or "--git" instead of just running a standard diff command with no options? Don't the patches work the same? Is it just for readability purposes? [00:46] <beastd> Zeranoe: --git is what you will get from git naturally [00:48] <Zeranoe> beastd: Do the patches include the context around the modified area just for readability sake? [00:49] <beastd> Zeranoe: not only [00:50] <beastd> Zeranoe: it is also for finding the right spot to apply the patch (probably with "fuzz") [01:02] <J_Darnley> Surely patches are made that way because people ask git to make them rather than running diff on the files themselves... [01:50] <Compn> zera [01:50] <Compn> sfakgj [01:50] <Compn> J_Darnley : patches are made with git send email because you can apply these patches from your email client in one click / keystroke [01:50] <Compn> one or two anyways [02:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:13ae02d03fbb: imgconvert: add get_pix_fmt_score() [02:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:cf8d9b74f306: imgconvert: use get_pix_fmt_score in avcodec_find_best_pix_fmt_of_2() [05:34] <Darkarnium> Afternoon [07:05] <Darkarnium> Bit of a strange query regarding AVOptions. I patched one of the demuxers and added a few AVOptions that can be passed as CLI arguments. This works without issue under ffplay, all options are passed and the relevant calls therein are performed with the options specified via the flags. However, if I instead use the ffmpeg binary, the flags seem to be ignored and the demuxer never uses the CLI arguments. [07:05] <Darkarnium> Is there any major difference between the handling of options between these two applications? [07:06] <Darkarnium> I'm sure it must just be something I've missed. However, I've also taken the newly compiled binary and executed it directly - in case I was having issues with ffplay and ffmpeg being different versions, etc. [07:08] <Darkarnium> Any suggestions, documentation or mailing list threads would be appreciated - if I'm missing something here, which is most probable. [07:18] <Skyler_> what's your ffmpeg commandline? [07:57] <Compn> Darkarnium : wheres your patch ? [07:58] <Compn> Darkarnium : did you specify the options in the right order? it matters in ffmpeg for some reason [07:58] <Compn> eg ffmpeg -demuxeroptions filename is different than ffmpeg filename -options [08:02] <Compn> anyone rememeber what i was supposed to add to ffmpeg tracker ? [08:04] <Compn> Darkarnium : saste found some strange bugs with -user-agent and other options in ffmpeg. maybe there is bug... [08:04] <Darkarnium> I'll have a look at the order then, see how far I get :) [08:04] <Darkarnium> The patch isn;t something I'd be submitting to be honest, as it's for an internal application :) [08:05] <Darkarnium> I can't see there being another user who is doing something this broken to be honest :P [08:06] <Compn> someone remind me if bayer patch didnt get applied ... [08:06] <Compn> oh i think it still crashes [08:07] <Compn> Darkarnium : i meant we can check what you did [08:07] <Compn> to see if you did it wrong [08:07] <Compn> but you might want to ask saste [08:08] <Compn> a lot of option code got moved around, apparently [08:10] Action: Compn gone, night [08:11] Action: Compn pings pross about bayer decoder [08:11] <Compn> now night [10:53] <cone-23> ffmpeg.git 03Carl Eugen Hoyos 07master:f02f231e05e0: Do not write invalid matroska files. [11:12] <saste> may we remove the rfps spam from the log? [12:02] <durandal_1707> why ffplay outputs only yuv420p ? [12:02] <nevcairiel> because of SDL, more or less [12:06] <durandal_1707> and it still cant output 444 or rgb? [12:08] <durandal_1707> sdl 2 version i mean [12:18] <cone-23> ffmpeg.git 03Diego Biurrun 07master:8658e1c68253: mss4, ra288: Remove unused DSPContext local codec context members [12:18] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:36fab50e90d1: asfdec: silence a warning [12:18] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:0af4506e15a6: Merge commit '36fab50e90d15352e403e4cc210890810f2fb4e2' [12:18] <saste> durandal_1707, should we add a test for il? [12:19] <saste> durandal_1707, in fact SDL1.2 supports a few more formats, check lavd/sdl [12:21] <durandal_1707> saste: that are just packed variants [12:21] <durandal_1707> saste: feel free to add test for il,lut,negate, ... [12:38] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:61f9ad2dfcb3: asfdec: read the full Metadata Object, not just aspect ratio information [12:38] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:3600049dae9e: Merge commit '61f9ad2dfcb3f98b7ac5777d19d0e7b61d0be01e' [12:45] <saste> durandal_1707, currently il segfaults on rgb48 [12:45] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:84b721db366c: asfdec: also read Metadata Library Object [12:45] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:30085bd43e35: Merge commit '84b721db366c0734fdfd23c8daaa7da7da21f761' [12:45] <saste> so yes, we should add a test for it [12:49] <durandal_1707> saste: not when i tested rgb48 [12:49] <durandal_1707> saste: don't you have patch for it? [12:50] <saste> durandal_1707, it doesn't segfault with my patch [12:52] <durandal_1707> saste: it does not segfault here [12:52] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:db0a943266be: avplay: apply the stream sample_aspect_ratio to decoded video frames [12:52] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d80820655362: Merge commit 'db0a943266be29ff0596872ebb418dfed75d00de' [12:53] <saste> durandal_1707, -vf format=bgr48be,il=l=d:c=d,showinfo,format=bgr48be [12:53] <saste> anyway i don't mind, I'm going to visually test with my patch on [13:17] <cone-23> ffmpeg.git 03Vladimir Pantelic 07master:f5fac6f77752: asfdec: support reading ID3v2 tags in ASF files [13:17] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:9a714d871ac5: Merge commit 'f5fac6f77752931347ab302563802dcaa49c2419' [13:21] <saste> durandal_1707, can you explain this: ffplay -f lavfi color -vf format=monow,il=l=d:c=d,format=monow [13:21] <saste> check the first frame [13:21] <durandal_1707> saste: about your noise source, how you expected to use it with other videos? [13:22] <saste> with my patch it looks ok [13:22] <cone-23> ffmpeg.git 03Anton Khirnov 07master:1ef0e8a6bf89: asfdec: do not assume every AVStream has a corresponding ASFStream [13:22] <cone-23> ffmpeg.git 03Anton Khirnov 07master:81726a4f0b8a: FATE: add tests for additional flavors of asf cover art [13:22] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:e88f392733fe: Merge commit '81726a4f0b8a43e19898e2a36fdde80583bafff0' [13:22] <durandal_1707> saste: if it does not make sense, filter out bitstream formats in query_.. [13:22] <durandal_1707> saste: i said your patch is ok [13:23] <saste> durandal_1707, yes but i can't understand what's wrong [13:23] <durandal_1707> i think i just made it bug to bug compatible with mp=il [13:25] <saste> durandal_1707, noise source? which one? [13:25] <saste> the old and slow patch? [13:26] <durandal_1707> there was only one noise source afaik [13:28] <durandal_1707> saste: ffplay have strange buffer handling [13:28] <durandal_1707> use ffmpeg, ffplay hide several bugs i had with il [13:53] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d6d5ef5534d5: sws: GBRP9, GBRP10, and GBRP16 output support [13:53] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:42431a557e2d: Merge commit 'd6d5ef5534d582f9dbaf18ac2605cf5bb72cd821' [13:55] <saste> durandal_1707, i don't think it is a good idea to have separate modes for il [13:55] <saste> it was a (bad) idea ported from mplayer [13:55] <saste> for example what's about alpha mode and packed? [13:55] <saste> why should I want to set a different interleaving mode for the various components? [13:56] <saste> why do i need to set it up to make it work with a given format (and i have to account for the pixel format planar/packed layout)? [13:59] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:e58013dd8f00: sws: use planarRgbToRgbWrapper only for 8bit per component [13:59] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d1d9b93e9a21: Merge commit 'e58013dd8f00e17dd98ba64e41c72f1d152f6608' [14:03] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:63a99622876f: lavfi/il: simplify/generalize linesize computation [14:03] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:0b1896543446: lavfi/il: move query_formats() definition up [14:16] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:7b41c24c5fa2: sws: dont enable chrSrcHSubSample for planar RGB [14:16] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f2b89f21cc3d: Merge commit '7b41c24c5fa221b55a9302efaf1a9eeb38b12551' [14:27] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:813784a8bf90: sws: disable yuv2rgb warning for planar rgb. [14:27] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:02b3ac87f81a: Merge remote-tracking branch 'qatar/master' [14:34] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:2c5e4ae29180: Revert "Merge commit 'db0a943266be29ff0596872ebb418dfed75d00de'" [14:45] <cone-23> ffmpeg.git 03Carl Eugen Hoyos 07master:034a125c8c51: Support more atrac3-in-mkv samples. [15:18] <durandal_1707> so what is status of vf_psnr? [16:05] <durandal_1707> one cant generate pink/brown/.. noise with aevalsrc? [16:41] <durandal_1707> burek: why ubuntu packages are that old? [16:50] <creep> h [16:55] <wm4> durandal_1707: maybe because ffmpeg APIs change so often, and they can't get all packages to work with newer ffmpeg... that's generally the reason why ffmpeg packages are old [16:55] <burek> durandal_1707 ? [16:56] <nevcairiel> ubuntu doesnt have ffmpeg packages, it only has "fake" ones that actually are libav [16:56] <wm4> it probably has old libav packages then :) [16:58] <nevcairiel> someone should come up with a popular linux tool that only works with ffmpeg and not libav, and see what distributions do 0p [16:58] <durandal_1707> burek: i talk about one in link provided from ffloger [16:59] <durandal_1707> burek: also there could be static build for linux [17:04] <burek> nevcairiel, ffmpeg is already a popular tool ^^ [17:04] <burek> durandal_1707, can you provide more references? [17:04] <burek> so i could know what are you talking about [17:06] <burek> btw, searching for enhancements on trac and trying to implement the most easy/quick ones would make ffmpeg have more popular functionality than libav (which it already has though), but it would allow users to choose ffmpeg, because "it can do something that libav cant" or something [17:08] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:6462268e74fa: pcx: fix rounding in bytes_per_scanline check [17:13] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:e005697af685: tests: add fate-lavfi-il test [17:13] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:60950adc18fe: lavd/v4l2: implement consistent error handling [17:13] <durandal_1707> burek: read #ffmpeg [17:15] <durandal_1707> saste, michaelni, ubitux what is status of lavfi scritpting and dynamic/conditional filtergraph? [17:15] <burek> oh you mean Jon Severinsson's FFmpeg PPA [17:15] <burek> well, ask Jon Severinsson... [17:15] <burek> I just linked his ppa [17:15] <burek> because we don't have anything "ours" [17:15] <saste> durandal_1707, still [17:15] <burek> I mean, probably the guy is also here, so just ask him, i guess [17:16] <saste> durandal_1707, what features do you exactly need? [17:17] <saste> a lua filter is just part of it, in general we may need to add more events to the already implemented filters [17:17] <saste> for example to change position to an overlay (or disable it) [17:18] <durandal_1707> saste: crossfading for example, one apply fade out and mix/blend it with fade in of another input [17:18] <durandal_1707> no caching of any kind is needed [17:20] <saste> durandal_1707, concat+fade? [17:20] <saste> now the problem is that you don't know when a filter stream is going to end [17:21] <durandal_1707> concat is useless here .... [17:21] <saste> yes *cross*-fading i see [17:22] <durandal_1707> it already can be done if several steps - which is PITA [17:22] <durandal_1707> *in [17:33] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f41329c8aec0: pcx: use init_get_bits8() [17:33] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:d24de4596c3f: pcx: Add missing padding to scanline buffer [17:35] <michaelni> saste, libavdevice/v4l2.c:688:20: warning: ret may be used uninitialized in this function [-Wuninitialized] [17:39] <saste> Assertion bytes >= needed_bytes failed at libavformat/matroskaenc.c:166 [17:39] <saste> now i wonder why this happens... [17:50] <saste> ^^ anyone has ideas? [17:51] <durandal_1707> saste: how it happens [17:54] <saste> durandal_1707, ffmpeg -i IN.flv -an -codec:v copy -y OUT.mkv [17:54] <saste> this is happening at the end, when closing the file [17:56] <saste> apparently in end_ebml_master() [17:59] <durandal_1707> i ported noise filter [18:00] <saste> durandal_1707, do you plan to port all mp filters? [18:01] <durandal_1707> saste: not just mp useful filters but all useful filters [18:01] <durandal_1707> ilpack for examples is hardly useful [18:01] <wm4> but, 2002! [18:02] <nevcairiel> 2002 called, they want their filters back [18:02] <wm4> I think vf_pullup and vf_unsharp are relatively useful (though still a league below modern filters) [18:03] <wm4> vf_dsize doesn't make sense, you can delete it [18:03] <durandal_1707> so every new year I need to rewrite each decoder/encoder/demuxer/muxer? [18:04] <nevcairiel> every 10 years is enough [18:05] <durandal_1707> the idea is to have addgrain like filter [18:06] <saste> unsharp is already in, but it has some differences with the original unsharp [18:07] <durandal_1707> like what? [18:07] <wm4> uh, so you're keeping the mp one? [18:08] <saste> durandal_1707, like I wrote a patch long time ago, i sent it and it was approved, but then I was not satisfied with the patch and is still there [18:08] <saste> i wanted to read the papers and understand more about the algo, but never found the time/motivation to do that [18:10] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:151067bbc220: dpx: Fix rounding in 10bit total_size calculation [18:10] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:cb85779d459c: dpx: include offset in the total_size calculation [18:14] <durandal_1707> hmm, why i dont get rgba in filter_frame if original video is rgba? [18:23] <durandal_1707> another double ones are kerndeint and tinterlace [18:23] <durandal_1707> mp=tinterlace should be removed [18:23] <saste> durandal_1707, if you can convince michaelni to drop it [18:24] <saste> kerndeint can be optimized, ubitux was working on that [18:24] <saste> it should be easy, but i leave it to someone else [18:24] <saste> tinterlace is slower than mp=tinterlace [18:25] <saste> although the absolute difference should be meaningless [18:25] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:ea1d8465e6ec: Fix division by 0 due to audio frame size [18:25] Action: wm4 wonders why you would optimize these filters [18:27] <wm4> porting vivtc instead would be _actually_ useful: http://code.google.com/p/vapoursynth/source/browse/#svn/trunk/src/filters/v… [18:27] <saste> wm4, patches are... you know what [18:28] <wm4> now I don't want to be too offensive, "patches welcome" has to be the most stupid excuse ever [18:30] <saste> wm4, why? [18:30] <saste> it's not an excuse, it the truth [18:30] <saste> i won't work on that, but i won't prevent you to do so [18:31] <saste> that's how floss works [18:32] <wm4> I was just suggesting that porting vivtc would be a better use of your time rather than _optimizing_ barely useful filters (which probably not even mencoder users ever use) [18:33] <J_Darnley> Don't port it! Write a filter that passes video through Vapoursynth. [18:33] <J_Darnley> This is just the kind of stuff that thread on the ML was talking about [18:33] <wm4> J_Darnley: that certainly would be the best, though connecting these two APIs could be cumbersome [18:35] <saste> J_Darnley, getting rid of mp is more fun [18:35] <wm4> mp shouldn't have been added in the first place [18:36] <saste> about usefulness, it's not like i *personally* need il or any other interlacing filters [18:36] <saste> wm4, tell it to the users which use it [18:36] <saste> also it was useful for testing/porting stuff [18:36] <saste> that was the main reason for inclusion imo [18:45] <saste> that said, you should add ticket to trac/upvote, so we developers know what people really want [19:04] <Compn> durandal_1707 : going to port libvo from mplayer to ffplay ? [19:04] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:478fc7f57bac: pictordec: fix cga palette index limit [19:07] <Compn> wm4 / J_Darnley : why not rally vapoursynth and avisynth together to tie into ffmpeg ? [19:07] <Compn> surely theres plenty of *synth devs to make a measley wrapper [19:09] <Compn> michael did mp wrapper in less than a day ... [19:10] <Compn> of course [19:10] <Compn> you could also bribe michael into doing it :) [19:12] <durandal_1707> wrappers are bad [19:12] <Compn> i dont mean wrapper [19:12] <Compn> maybe i do [19:12] <Compn> external wrapper [19:12] <durandal_1707> same [19:12] <Compn> whats better? native api ? [19:12] <Compn> or porting everything to ffmpeg ? [19:13] <Compn> (or are you talking about libvo ? ) [19:13] <durandal_1707> Compn: there is libavdevice which is actually libavformat [19:16] <Compn> what [19:17] <Compn> oh libavdevice handles hardware and such ? [19:17] <Compn> so what are you saying ? [19:17] <Compn> sorry, i'm kind of dumb sometimes :) [19:31] <durandal_1707> Compn: answer to question about porting libvo [19:31] <durandal_1707> it is bad idea to port it to ffplay [19:34] <saste> who can help with matroskaenc? [19:43] <michaelni> saste, what exactly ? [19:43] <michaelni> about what [19:49] <durandal_1707> saste: having hwaccel pix fmts in supported inputs/outputs of filter is nonsense [19:58] <durandal_1707> saste: why has_alpha have double !! in vf_il [19:58] <durandal_1707> it is not needed at all [19:58] <saste> durandal_1707, flags & ALFA is a mask [19:58] <saste> I need to convert to 1/0 [20:00] <saste> michaelni, what are this cuepoints in matroska? [20:01] <saste> i have a (broken) file duplicated timestamps [20:01] <saste> i suspect this is the problem [20:01] <saste> it is trying to put too many entries into the same cuepoint, and asserting [20:14] <nevcairiel> cuepoints are a seeking index [20:14] <nevcairiel> more or less [20:22] <saste> in mkv_write_cues() is is setting an expectedsize in start_ebml_master [20:22] <saste> which is not honored, and which leads to an assert failure [20:29] <saste> michaelni: in other words, the MAX_CUEPOINT_SIZE is not assuming duplicated timestamps in the same stream [20:30] <saste> next question is, how to deal with that? [20:31] <saste> I could assume unexpected size, and increase filesize [20:32] <saste> now i'm not even sure how matroska copes with duplicated timestamps [20:36] <michaelni> saste, assuming duplicate timestamps arent disallowed how would they be usefull in the cue list ? [20:36] <michaelni> if you have 2 timestamps of 123 and you want to seek to 123 ... [20:37] <saste> michaelni, so what do you suggest to do? [20:37] <saste> the best thing would be to avoid to write duplicated cue points (for the same stream) [20:37] <saste> so we don't hit the bytesize limitation [20:38] <saste> do you agree? [20:41] <saste> michaelni, ^^ [20:45] <saste> durandal_1707, i can exclude the hwaccell from the il list, even if this is not going to do any difference [20:45] <michaelni> saste, yes, i agree dont wrote duplicate cue points [20:47] <durandal_1707> saste: doing it at higher level would be even better, even if lavfi will never use such formats [20:48] <durandal_1707> the noise filter makes it possible to select noise generation parameters per plane [20:48] <saste> durandal_1707, allright, direct commit or do you want to see the patch? [20:48] <durandal_1707> adding opts for all planes is pita [20:48] <saste> durandal_1707, in case of noise, i think it is useful for testing purposes [20:49] <saste> otoh the user should be able to specify with a single option the value for all the planes [20:49] <saste> and btw same for il [20:49] <durandal_1707> saste: it is usefull not only for testing but is there better way to handle this? [20:50] <durandal_1707> i start to hate opts tables. they clearly needs some improvement, like shortname or something.... [20:50] <durandal_1707> instead of duplicating each and every line [20:51] <durandal_1707> so it is 6*2*4 = 48 entries [20:51] <saste> durandal_1707, you can use macros [20:52] <durandal_1707> macros just hide issue [20:52] <saste> it's not like you need duplicated-similar options in most components [20:52] <saste> use them with caution [20:52] <durandal_1707> there should be type of list item [20:53] <durandal_1707> you set number of entries in list and size of each entry [20:54] <saste> for the constants? [20:54] <durandal_1707> no, for 0/1 and int [20:55] <durandal_1707> i could use AV_OPT_TYPE_BINARY [20:56] <durandal_1707> or actually not, but something similar [20:57] <durandal_1707> actually 0/1 items are just flags [20:58] <durandal_1707> and avopt don't care for this [20:58] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:f28043d0a34a: tiff: check for failure in search_keyval() [20:58] <cone-23> ffmpeg.git 03Michael Niedermayer 07master:0e3dacb11eac: tiff: dont leave geotag_count in an invalid state on errors. [20:58] <durandal_1707> or actually does ..... [21:00] <durandal_1707> oh so i get now 4*(1+1) = 8, i can handle this [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:f0703b6cba65: lavd/v4l2: fix error logic when enumerating standards [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:6b171f046c6f: lavf/matroskaenc: provide more feedback in case of unsupported codec [21:04] <cone-23> ffmpeg.git 03Stefano Sabatini 07master:3acaea26e5b6: lavfi/il: filter out hwaccels formats from the list of supported formats [21:41] <michaelni> an3k / Guest9410, why do you constantly change nick back and forth ? [21:42] <saste> he can't decide [22:16] <Compn> michaelni : hes got nick protect on, and since hes not here to identify, chanserv changes his nickl [23:19] <durandal_1707> Compn: buy how do you explain periodic changes? [23:27] <Compn> durandal_1707 : irc client script auto changes nick or maybe bnc script etc [23:28] <Compn> didnt mean to do that, oops [23:28] <Compn> ehe [23:40] <durandal_1707> michaelni: fine to relicense noise to lgpl? [00:00] --- Mon Feb 11 2013

1 0

ffmpeg.log.20130210
by burek 11 Feb '13

11 Feb '13

[00:10] <norbert_> hi, latest 32 bit ffmpeg binary, how can I specify the location on the screen from where I want the width+height to be recorded? [00:10] <norbert_> used to be -i :0.0+x,y [00:10] <norbert_> that now says "Protocol not found" [00:12] <norbert_> I'm willing to bet that 99% of all ffmpeg lines that Google has indexed no longer work with the latest version [00:12] <norbert_> even the ffmpeg wiki itself is outdated [00:13] <sacarasc> Which site did you get the binary from? Which OS? [00:13] <norbert_> Mint GNU/Linux and http://ffmpeg.gusari.org/static/32bit/ [00:15] <fenduru> I'm trying to decode and reencode an audio file into segments. Right now I am running ffmpeg -i pipe: -vn -b:a 192k -acodec libvorbis -map 0 -f segment -segment_time 10 output%d.ogg [00:16] <norbert_> and it's also funny how 99% of all users run into libx264 being unknown [00:16] <norbert_> gotta love that [00:16] <norbert_> plus the location in the string of input and output parameters [00:16] <fenduru> however, when I decode these files and play them sequentially I get occasional pops/clicks between segments. Is there a way to force ffmpeg to split on the nearest frame boundary? (I've tried -segment_time_delta but this hasn' helped) [00:17] <Samus_Aran> will this command re-encode the audio: ffmpeg -i input.aac -acodec copy output.mp4 [00:18] <Samus_Aran> I wouldn't think so, but it says it is encoding the audio [00:18] <Samus_Aran> I am just trying to put my aac files into a container that will provide the duration of the audio (so MPlayer doesn't incorrectly guess the duration) [00:19] <JEEB> setting encoding code to copy should just mux [00:19] <JEEB> so as long as the log says that it only copies, it doesn't [00:20] <Samus_Aran> it says it is using an AAC encoder [00:20] <Samus_Aran> there is no mention of copying [00:20] <JEEB> pastebin your full command and log, and link here [00:22] <norbert_> JEEB: http://pastebin.com/MEuMpkkn [00:22] <norbert_> :) [00:23] <JEEB> unfortunately I have no idea about screen capture [00:25] <beastd> norbert_: does -f x11grab before -i give a different response? [00:25] <norbert_> well, I tried that, but it gives me: Unknown input format: 'x11grab' [00:25] <norbert_> so I decided to just remove that x11grab thing and hoped that it would magically fix the problem [00:25] <norbert_> now I'm reading http://www.jedi.be/blog/2010/08/30/capturing-the-screen-of-your-virtual-mac… [00:26] <sacarasc> That sounds like they didn't compile with x11grab support. :o [00:26] <norbert_> which is complicated and goes on about what I could do and stuff [00:26] <beastd> norbert_: ffmpeg -formats | grep x11 [00:26] <norbert_> yeah, the guy didn't add --enable-x11grab [00:27] <norbert_> so back to the ancient ffmpeg on my system, I guess, no recent binary this time [00:27] <jure> so, to speed up the tempo of audio, without changing pitch, you would do something like "ffmpeg -i input.mkv -filter:a 'atempo=2.0' -vn output.mkv". how do you change the speed of the audio and get the change in pitch as well? [00:27] <Samus_Aran> JEEB: here's my command and the output: http://pastie.org/pastes/6106599/text [00:27] <norbert_> and then I run into: Unknown encoder 'libx264' [00:27] <norbert_> myeah... sigh [00:28] <beastd> norbert_: this is not ffmpeg's fault. seems you binaries miss a few popular features :( [00:28] <Samus_Aran> it shows libvo_aacenc and Lavf53.21.0 [00:28] <beastd> norbert_: which system are you on? [00:28] <norbert_> beastd: well, the binary doesn't have --enable-x11grab and my system (Mint GNU/Linux) has an ffmpeg without libx264 in its ffmpeg [00:29] <norbert_> which means I need to pick another codec or recompile the thing myself [00:29] <Samus_Aran> norbert_: Ubuntu for instance has repositories with more up-to-date ffmpeg releases [00:29] <JEEB> Samus_Aran, you seem to be using a ffmpeg binary from the libav project, but that is still weird... try with the 'avconv' binary that is more up-to-date in that project [00:29] <Samus_Aran> also more feature-enabled [00:30] <JEEB> avconv -i input -c:a copy out.mp4 [00:30] <norbert_> Samus_Aran: do you know what would be a good codec to pick for desktop recording? out of these: http://pastebin.com/kZFzxP84 [00:31] <norbert_> something that gives me good quality I mean, without grinding my system to a halt [00:31] <norbert_> I can reencode it later if necessary [00:31] <JEEB> ffvhuff is a nice quick video encoder [00:31] <JEEB> it's big, but it's lossless and supports various colorspaces [00:32] <JEEB> also quick to encode [00:32] <Samus_Aran> JEEB: with that command it becomes: Stream #0:0 -> #0:0 (copy) [00:32] <JEEB> ok [00:32] Action: juanmabc guesses game recording [00:32] <JEEB> ubuntu by default uses libav, so #libav would be more correct for that [00:33] <JEEB> basically they later removed the ffmpeg binary from their side and avconv is their up-to-date binary [00:33] <JEEB> but it seems to work now, so that's great :) [00:33] <norbert_> [matroska @ 0x8aa5020] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1 >= 1 [00:33] <norbert_> av_interleaved_write_frame(): Invalid argument [00:33] <norbert_> is what it says when I try: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 25 -s 1280x720 -i :0.0+1920,26 -acodec pcm_s16le -vcodec ffvhuff -crf 0 output.mkv [00:33] <Samus_Aran> norbert_: I would suggest using a dedicated screen capture app, then converting the video it produces. is this OpenGL or simple 2D apps? [00:33] <JEEB> norbert_, might as well use avi for those codecs, try it out :) [00:34] <norbert_> there is no avi or libavi [00:34] <Samus_Aran> JEEB: so you can confirm that the ffmpeg command and output I pasted is in fact re-encoding my audio files? [00:34] <norbert_> it seems [00:35] <JEEB> norbert_, no -- I mean set output file name to dot-avi [00:35] <JEEB> that should switch the container to avi [00:35] <JEEB> Samus_Aran, that's how it reads like :P [00:35] <norbert_> JEEB: hey thanks, now it records [00:37] <Samus_Aran> JEEB: thanks. need to copy over my audio again, then, as I erased several files after I thought I losslessly copied them. oops. [00:38] <Samus_Aran> blah @ re-encoding then re-encoding then re-encoding audio... [00:41] <norbert_> hey thanks again, gotta go [00:45] <Samus_Aran> JEEB: thanks again for your help. [01:52] <suddendad> hello [01:53] <suddendad> is it possible to stream a huffyuv encoded videostream over network? [02:02] <videoman> Can't fade in a png without it appearing as a black square, seemed to work in earlier versions http://pastebin.com/g9PUKXfp [02:20] <videoman> alpha is =1 as in fade=out:80:30:alpha=1 [11:05] <Ertes> Hi guys, I have a mkv with 2 different audio track, both AC3, 48,0 KHz, 16Bit, 6 Channel, but different Kbps...the first is 384, the second is 448 [11:06] <Ertes> My TV can read the second track, but not the first ( Audio not supported ) so i think the problem is the Kbps, ho to change the Kbps from 384 to 448? Thanks [11:10] <Ertes> I have try with this commnad --- ffmpeg -i input.mkv -vcodec copy -acodec copy -ab 448k output.mkv --- but the audio is always 384 after the complete process... [11:25] <Ertes> i have a warning for the channel layout, how to set up for the 6 channel? [13:19] <t4nk461> Hi, when running the ffmpeg configure script with --enable-libfdk-aac, the check of it fails --> /usr/bin/ld: error: cannot find -lfdk-aac ... undefined reference to 'aacEncOpen' ... But libfdk-aac.a is installed under /usr/local/lib [13:19] <Fjorgynn> and you have compiled libfdk-aac? [13:19] <t4nk461> yes [13:19] <Mavrik> t4nk461, did you run ldconfig after installing fdk_aac? [13:20] <JEEB> that shouldn't be needed for a static lib (.a) [13:20] <JEEB> only for .so (shared) [13:20] <Mavrik> ah yes [13:20] <t4nk461> as described under: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#FFmpeg [13:20] <JEEB> t4nk461, does your distribution by default look into /usr/local for headers/libs? [13:21] <t4nk461> how can I check this? [13:23] <JEEB> -print-search-paths for libraries it seems? And just -v when compiling a random c file for header search paths [13:34] <t4nk461> when compiling with -v LIBRARY_PATH does not contain /usr/local. How to set it up? I added LD_LIBRARY_PATH=/usr/local/lib to my .bashrc? [13:35] <JEEB> --extra-ldflags="-L/usr/local/lib" [13:35] <JEEB> when configuring [13:35] <JEEB> LD_LIBRARY_PATH is used for shared libraries when loading them at runtime, so that is unrelated [13:36] <t4nk461> But where do I set it permanently? [13:37] <JEEB> you really can't [13:37] <JEEB> unless you want to recompile your toolchain :P [13:37] <JEEB> you'll just have to get used to the fact that you will have to add -L/usr/local/lib to the linking stages of things [13:38] <t4nk461> ok, thanks [16:23] <Seablade> Real quick question, getting unknown input format concat event hough it is listed int he available filters with the command... ffmpeg -f concat -i temp.txt msg.mp4 [16:23] <Seablade> Any ideas? [16:26] <Seablade> The entire output is what I posted except the ffmpeg version number, which I have tried a HEAD compiled from a few months back and 0.8.5, built Feb5 [16:27] <Seablade> If there is nothing obvious from my command setup that is all I need to know, only have five minutes or so to get this done sadly [16:27] <durandal_1707> cant help yuo until i see full uncut output [16:28] <Seablade> durandal_1707, I repeat what I jsut meantioned but since you insist...http://pastebin.com/D4gdLHCc [16:29] <durandal_1707> Seablade: you are using wrong and old version [16:29] <Seablade> durandal_1707, Thus why I mentioned I have tried with a HEAD fversion from a few months back as well, this is on two different computers [16:30] <Seablade> durandal_1707, What I need to know, is is there anything wrong with the command, or should it work assuming my version is up to date [16:30] <durandal_1707> Seablade: what HEAD version? [16:30] <Seablade> durandal_1707, I compiled SVN head from a few months back, that computer is in a different room right now I can't get you the exact revision [16:30] <durandal_1707> no you are using ffmpeg from Libav [16:31] <durandal_1707> SVN head is old and not maitained/updated at all [16:31] <durandal_1707> you should use git head [16:31] <Seablade> durandal_1707, Ok good to know thanks, my question does still remain, is there anything wrong with the basic command structure? [16:31] <durandal_1707> command should work with latest ffmpeg from FFmpeg project [16:31] <Seablade> durandal_1707, Thank you [16:32] <Seablade> durandal_1707, That is what I needed to know [16:32] <durandal_1707> Seablade: you get list of supported formats with: ffmpeg -formats [16:33] <Seablade> durandal_1707, Ahh thank you that is what I was messing up [16:34] <Seablade> Yea this Ubuntu package deperecation is ebing annoying on this machine, my OS X machine apparently was from the deprecated version, trying to download and build hnow [16:35] <Seablade> durandal_1707, Yep installing that ppa right now in fact;) [16:37] <Seablade> durandal_1707, And that ppa doesn't seem to have an up to date version with concat format, yea;) [16:37] <Seablade> durandal_1707, At least not for precise [16:39] <durandal_1707> Seablade: that is 0.10.6 version? [16:39] <durandal_1707> concat is in 1.1 [16:39] <Seablade> durandal_1707, Yep [16:39] <Seablade> durandal_1707, And yep:) [16:39] <Seablade> durandal_1707, Gotta love it [16:40] <Seablade> durandal_1707, Ok well my time limit passed, was trying to fix something for a live event on the fly, thanks very much for the help [16:40] <Seablade> durandal_1707, Ill fight through setting up the build environment for something up to date later [16:49] <creep> h [17:37] <creep> somebody know where to get gtk+-2.0-dev? it should be here i think... ftp://ftp.gtk.org/pub/ [17:52] <creep> somebody know of a good qr code reader/generator for linux (opensource) ? [19:24] <karlox> hi all [19:25] <karlox> wich is the status of rtmp support in ffmpeg? [19:32] <durandal_1707> karlox: there is native and wrapper one [21:13] <njhbnjhbn> hi all! need help [21:14] <njhbnjhbn> i want to join a still image and an audio file to a video file [21:15] <creep> how do i make 1 image from webcam with ffmpeg and exit ? [21:15] <njhbnjhbn> i used the following command: ffmpeg -loop 1 -f image2 -r 1 -i dragon2.jpg -shortest -i a.mp3 -acodec copy -vcodec libx264 out.mp4 [21:15] <njhbnjhbn> but the -shortest option is ignored, it loops forever [21:15] <njhbnjhbn> the video keeps growing and growing until i cancel [21:27] <njhbnjhbn> i fixed it by using an older version from 2012 [21:42] <saste> creep: ffmpeg -f v4l2 -t 3 -i /dev/video0 -updatefirst out.png or something like that [21:42] <saste> you should keep the device open for a few seconds, so that it can adjust brightness/contrast [00:00] --- Mon Feb 11 2013

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ffmpeg-devel.log.20130209
by burek 10 Feb '13

10 Feb '13

[00:31] Action: Compn wonders why someone is encoding to qtrle in 720p [00:36] <kierank> why not [00:39] <Compn> whats qtrle good for ? seems like an old codec with poor compression ? [00:39] <Compn> maybe i dont know how good it is [00:40] <Compn> but most codecs from the year 2004 arent that great for hd. [00:40] <Compn> probably earlier, but going by copyright year :P [00:42] <kierank> it has alpha channel [00:42] <kierank> that's why its useful [01:02] <llogan> saste: any changes to "sort demuxers" other than moving the text? [01:19] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:a92816c4eb9d: get_se_golomb: fix decoding of very large values [04:06] <cone-778> ffmpeg.git 03James Almer 07master:53b2083f4433: sparc: Revert some parameter name changes in dsputil_vis.c [04:13] <cone-778> ffmpeg.git 03rogerdpack 07master:aa715cdcf5c6: dshow: attempt to build shared as well as static [04:13] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:a35f5c22339c: Merge remote-tracking branch 'rdp/dshow_shared' [05:31] <llogan> haven't seen a gcc seg fault in a while... [05:31] <llogan> i mean until now [06:17] <Compn> llogan : not an ICE then ? [11:26] <durandal_1707> michaelni: the il filter have option to (de)interleave chroma and luma separately [11:26] <durandal_1707> and alpha is completely forgotten [11:26] <durandal_1707> so i will fix this by making it more generic [11:27] <durandal_1707> keepeing old syntax though does not make sense [11:49] <cone-273> ffmpeg.git 03Diego Biurrun 07master:0b016eb99d38: dsputil: Move ff_block_permute to mpegvideo_enc [11:49] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:55151aa1013b: Merge commit '0b016eb99d38738e2c53e36549a4732a0f863b2e' [12:10] <cone-273> ffmpeg.git 03Diego Biurrun 07master:b5b7b75e91e2: dsputil: Move get_penalty_factor() to the only place it is used. [12:10] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:bcb15e66f7c4: Merge commit 'b5b7b75e91e291c810ebd01868a42b53f267a87c' [12:42] <cone-273> ffmpeg.git 03Mans Rullgard 07master:ed62e6e3c3ca: mjpegdec: use put_pixels instead of copy_block8 [12:42] <cone-273> ffmpeg.git 03Mans Rullgard 07master:b9ee5f2cab3f: indeo3: replace use of copy_block4 with put_pixels [12:42] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:e5212354eadd: Merge commit 'b9ee5f2cab3ffe1c962e542346b1ed61394864ec' [12:45] <durandal_1707> michaelni: do not push hidden stuff into merge, like this snowenc change [12:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:29a92c011480: histogram filter [13:13] <michaelni> durandal_1707, the code doesnt compile without that change [13:14] <durandal_1707> michaelni: yes, but hidden commits are worse [13:15] <michaelni> my commit messages suck i know, i should have explained it in the merge commit [13:15] <michaelni> its on my todo to see if they should be merged or [13:15] <michaelni> if we can gain something from seperate functions [13:17] <durandal_1707> michaelni: just commit it before merge [13:17] <durandal_1707> quatar removed snow so it should not matter [13:20] <durandal_1707> michaelni: why #1280 picks gray? [13:22] <michaelni> dunno [13:22] <michaelni> i could look into it later [13:22] <michaelni> libavfilter/vf_histogram.c:208:13: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] [13:24] <durandal_1707> when we will finally upgrade from ISO C90? [13:24] <burek> ..A.L. ilbc iLBC (Internet Low Bitrate Codec) [13:24] <burek> it doesnt say if it is encoder/decoder [13:25] <JEEB> not better than opus [13:25] <durandal_1707> burek: because you do not have either [13:25] <burek> if i do not have, why is it listed? [13:26] <burek> it* [13:26] <durandal_1707> others are listed tooo [13:26] <burek> even worse [13:26] <burek> why [13:26] <durandal_1707> to notify users of it existence [13:26] <burek> i see [13:26] <durandal_1707> you should still be able to demux files [13:27] <durandal_1707> and you would know what codec is it about [13:27] <burek> well, then maybe they should be sorted/groupped somehow at the end/beginning, to make it clear they are not available [13:27] <durandal_1707> burek: there is no native decoder, instead you enable wrapper... [13:27] <burek> yes, i understand that [13:27] <burek> using external library [13:28] <michaelni> durandal_1707, C90, when all compilers we support support that [13:28] <durandal_1707> michaelni: translation: never [13:30] <cone-273> ffmpeg.git 03Paul B Mahol 07master:e5670cb8a1c9: lavfi/histogram: make it C90 compliant [13:30] <iive> imho, declarations in any location in the code is misfeature. it solves a problem caused by another misfeature [13:31] <durandal_1707> iive: not really.... [13:37] <iive> i'm waiting for your explanation. [13:42] <someone-noone> Hello! I'm trying to pass "magic cookie"(extradata) data to mac os Core Audio framework for decoding AAC format. File is in m4a format. For this I'm using extradata from my AVCodecContext. But extradata_size is 7, however if I read "magic cookie" using OS X API, it's size is 44 for same file. Looks like magic cookie != extradata in ffmpeg (but should be). Any help how can I extract this data from ffmpeg? [13:44] <cone-273> ffmpeg.git 03Diego Biurrun 07master:6a4430407404: dsputil: Move ff_h264_idct function declarations to a separate header [13:44] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:0af07495eccc: Merge commit '6a4430407404bfe55581e66a86a496c9adf6fb44' [13:51] <cone-273> ffmpeg.git 03Diego Biurrun 07master:75d5156ac152: dsputil: Move ff_svq3 function declarations to a separate header [13:51] <cone-273> ffmpeg.git 03Diego Biurrun 07master:35685a3c2a1e: dsputil: Move ff_shrink* function declarations to separate header [13:51] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:ffbe19fc84fe: Merge commit '35685a3c2a1ec09f3c62dcfc4368fe9e92bcddf6' [13:52] <burek> does anyone have any idea what to use (in ffmpeg) to produce a low latency speech audio (for use in p2p audio chat)? [13:53] <burek> whatever i try, it doesnt work, except aac/mpegts (over udp) which has a significant delay :( [13:54] <someone-noone> burek, first of all you need low latency codec. The most talkable codec is OPUS for now. [13:56] <burek> well i tried opus, like: ... -acodec opus -f mpegts udp://localhost:5555 [13:56] <burek> but ffplay wont play it back [13:57] <nevcairiel> i dont think there is a defined way to store opus in mpegts [13:57] <someone-noone> mpegts doesn't support opus [13:57] <JEEB> did the "ogg container" have it? [13:57] <burek> no [13:57] <JEEB> then there's that matroska mapping that differs from the one that mkvmerge implemented [13:57] <JEEB> which might or might not have been removed from ffmpeg [13:57] <burek> well is there a standard way to stream voice/speech with ffpmeg (preferably over udp) [13:58] <nevcairiel> ogg is about the only thing that properly supports opus today [13:58] <JEEB> yeah [13:58] <nevcairiel> short of raw opus files, which may just be ogg, which i never checked =p [13:58] <JEEB> :D [13:58] <JEEB> I tried have him find 'opus' from the output of -formats [13:58] <JEEB> and there was none there [13:59] <wm4> is proper opus in mkv defined yet? [13:59] <JEEB> nope [13:59] <nevcairiel> no [13:59] <wm4> why? [13:59] <JEEB> so both mkvmerge and ffmpeg IIRC now have conflicting experimental ways of putting it there [13:59] <wm4> lol [13:59] <burek> well i might use speex i guess, i just need the hint for the output format used with udp [13:59] <nevcairiel> opus is problematic because it requires pre-roll data to start decoding [13:59] <JEEB> ^ [13:59] <cone-273> ffmpeg.git 03Diego Biurrun 07master:b9c2408b19dc: dsputil: Remove commented-out, unused function declarations [13:59] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:a8864615e6cc: Merge commit 'b9c2408b19dc9408e9dc0ce4a099af529bbfba3b' [14:00] <nevcairiel> and how to handle this has caused the difficulty defining the proper standard in mkv [14:00] <JEEB> and matroska, unlike mov/mp4, doesn't have that concept in the container until now [14:00] <JEEB> and they want to "define it properly" [14:00] <JEEB> somehow I feel like I'll be liking xiph next-gen container if matroska just doesn't start evolving >_> [14:01] <burek> is it possible there is no "standard" or preferred way in ffmpeg (so far) to have it send low latency audio to another peer? [14:01] <burek> i mean, it looks so advanced with everything [14:01] <nevcairiel> ffmpeg is mostly a transcoding application, streaming is not its main goal [14:01] <burek> i see [14:02] <burek> well, ok, thanks.. at least i know where not to look then i guess.. [14:10] <someone-noone> Okay, I found difference between magic cookie and extradata for AAC. OS X meaning of magic cookie is esds ATOM in mp4 file. However ffmpeg stores only part (in the middle, only 7 bytes) of it in extradata and it looks like enough for ffmpeg to decode content in AAC. Anyway how can, probably ffmpeg stores somewhere(while decoding) this atom in AVCodecContext structure ? [14:13] <Paranoialmaniac> someone-noone: libavcodec doesn't store atom [14:13] <Paranoialmaniac> atom is mp4/mov thing [14:14] <Paranoialmaniac> aac specific info is stored in extradata as the same format between mp4 and mov [14:15] <someone-noone> Paranoialmaniac, I see. But is there a way I can mux this data into my container that is a part of libavformat, that I could read it in another application that doesn't use libav*? [14:15] <nevcairiel> all the important data is stored in extradata, you just have to put all the atom data back around it [14:15] <nevcairiel> if you need it to be an atom [14:17] <Paranoialmaniac> someone-noone: i have no idea about libavformat muxing :P sorry [14:17] <someone-noone> I have an idead. Look at m4a format& It's somehow playing by OS X :) [14:21] <someone-noone> hm, looks like I can't find m4a format in libavformat folder. Doesn't it name should be mpegaudio or m4a ? [14:21] <nevcairiel> m4a is just a normal mp4 file with a different extension [14:21] <nevcairiel> apple just uses it for audio files, but its the same format [14:21] <Paranoialmaniac> mov.c [14:21] <someone-noone> didn't know that& thanks [14:22] <Paranoialmaniac> mp4 != m4a, m4v. pure mp4 requires object descriptor and bifs for presentation :P [14:24] <nevcairiel> its the same damn format, may just have different atoms somewhere =p [14:24] <Paranoialmaniac> well, yes. mp4, m4a and m4v is derived from iso base media file format [14:25] <someone-noone> well, I found mov_write_esds_tag that is doing some really crazy things :( [14:25] <Paranoialmaniac> esds atom includes not only mpeg-4 audio specific info but also bitrate info [14:26] <Paranoialmaniac> esds atom consists of mpeg-4 system structure [14:26] <someone-noone> I need smth like uint8_t *esds_atom = get_atom(); OSXOpenCodec(esds_atom+4) [14:26] <someone-noone> Really, there is no explanation in Apple's docs :( [14:27] <cone-273> ffmpeg.git 03Diego Biurrun 07master:bf6b3ec924b4: dsputil: Move rnd_avg inline functions to a separate header [14:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:81a11452f8e0: Merge commit 'bf6b3ec924b4fb64d14df33077f4d4541d525dbf' [14:27] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:5fd6d85d17d5: rnd_avg: fix author attribution [14:32] <cone-273> ffmpeg.git 03Diego Biurrun 07master:059866eb17cd: dsputil: Move WRAPPER8_16_SQ macro to the only place it is used [14:32] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:276c9c0577e9: Merge commit '059866eb17cd3c66422a61d3e9c3305f55830a50' [14:40] <cone-273> ffmpeg.git 03Diego Biurrun 07master:218aefce4472: dsputil: Move LOCAL_ALIGNED macros to libavutil [14:40] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:6b2e65078c02: Merge commit '218aefce4472dc02ee3f12830a9a894bf7916da9' [14:44] <durandal_1707> iive: for (int i ....) [14:45] <durandal_1707> michaelni: i get strange internal.h macro E redefintions warnings [14:46] <durandal_1707> #define E(x) x ???? [14:47] <nevcairiel> its for macro expansion [14:48] <nevcairiel> some compilers need an extra expansion to properly expand all parameters [14:48] <durandal_1707> but there is already E defined in bunch of other places for opt stuff [14:49] <durandal_1707> In file included from libavcodec/options.c:35: [14:49] <durandal_1707> libavcodec/options_table.h:39:9: warning: 'E' macro redefined [14:49] <durandal_1707> #define E AV_OPT_FLAG_ENCODING_PARAM [14:49] <durandal_1707> ^ [14:49] <durandal_1707> ./libavutil/internal.h:63:9: note: previous definition is here [14:49] <durandal_1707> #define E(x) x [14:49] <durandal_1707> ^ [14:49] <durandal_1707> libavcodec/options.c:111:5: warning: 'av_opt_set_defaults2' is deprecated [-Wdeprecated-declarations] [14:49] <durandal_1707> av_opt_set_defaults2(s, flags, flags); [14:49] <durandal_1707> ^ [14:49] <durandal_1707> 2 warnings generated. [14:49] <nevcairiel> apparently libav doesnt have those other E defines =p [14:50] <durandal_1707> nevcairiel: i doubt it [14:50] <nevcairiel> or they dont include that header there [14:50] <durandal_1707> this is the new thing recently happening [14:52] <durandal_1707> nevcairiel: and this is introduced when moving local aligned stuff [14:52] <durandal_1707> fairly recently [14:52] <nevcairiel> i saw the commit that caused it [14:53] <durandal_1707> good. i will just undefine macro [14:53] <nevcairiel> that macro is needed [14:54] <cone-273> ffmpeg.git 03Diego Biurrun 07master:5d3d39c72e57: dsputil: Move fdct function declarations to dct.h [14:54] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:5cfc0ae825e6: Merge remote-tracking branch 'qatar/master' [14:54] <durandal_1707> but if you did not yet notice this breaks when using own E in opt stuff before using LOCAL.... [14:55] <nevcairiel> maybe just rename E to EXPAND or something [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:111697170e14: doc/demuxers: sort demuxers by name [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:9b9b5a475b99: doc/demuxers: fix copy&paste error in "@c man end" line [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:765dbea9fe1f: lavf/rawvideodec: fix/extend option descriptions [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:bad82243cd83: doc/demuxers: add docs for rawvideo demuxer [14:59] <cone-273> ffmpeg.git 03Stefano Sabatini 07master:b46a8f055ee5: doc/filters: end table in histogram docs [15:00] <durandal_1707> beastd: so native resamples is fast as sox one? [15:01] <beastd> hi [15:01] <durandal_1707> welcome on board [15:02] <beastd> durandal_1707: i did not measure the speed of native vs sox resampling. i just found that new thread on HA somewhat misleading. should check if something was added [15:03] <cone-273> ffmpeg.git 03Carl Eugen Hoyos 07master:7a259a5fcdc3: Fix libpostproc compilation on x86_32 with disable-optimizations. [15:04] <durandal_1707> beastd: what thread, there is trollolo old few days [15:06] <beastd> i understood from michaelni that you talked about ffmpeg resampling on HA. Or did you mean the old thread? I found this one http://www.hydrogenaudio.org/forums/index.php?showtopic=99286 [15:06] <durandal_1707> i mean whatever thread that was recently active [15:07] <durandal_1707> yes that thread ..... [15:10] <beastd> durandal_1707: did you compare speed of native vs libsoxr resampling? [15:10] <durandal_1707> beastd: no, i'm busy with removing mp [15:12] <durandal_1707> and negate filter is really broken on big endian? [15:12] <durandal_1707> why is there no fate coverage? [15:13] <durandal_1707> saste: ping [15:15] <saste> durandal_1707, what? [15:16] <durandal_1707> saste: i ported il filter, did you mind if old syntax is not keept? [15:16] <durandal_1707> it does not make sense to keep it [15:16] <saste> durandal_1707, how hard would be to keep the old syntax? [15:16] <saste> lemme check [15:17] <durandal_1707> il filter does not copy alpha plane at all [15:17] <saste> about lut/negate, I cannot test [15:29] <cone-273> ffmpeg.git 03Paul B Mahol 07master:da726a8763d3: paf, sanm: remove unused header after b05830169d [15:30] <saste_> michaelni, do you want to review the il filter? [15:30] <saste_> also tell if you're ok with GPL -> LGPL [15:31] <durandal_1707> saste: read logs, he already approved change [15:31] <durandal_1707> it is few memcpy calls after all [15:31] <saste> durandal_1707, ok [15:31] <michaelni> saste, review for() memcpy ? no ;) [15:31] <durandal_1707> :)) [15:32] <saste> that's not only that [15:32] <saste> i'm going to review it, leave me some time [15:32] <durandal_1707> i said, i need to fix 4 and 2 plane situation [15:33] <durandal_1707> this means keeping old syntax is not possible at all [15:33] <saste> durandal_1707, i don't mind about that [15:35] <saste> durandal_1707, is mp=il broken or what? [15:36] <durandal_1707> saste: not it is working fine [15:36] <durandal_1707> saste: mp=il=i,mp=il=d [15:36] <saste> ffplay IN -vf mp=il=d [15:37] <saste> i got chroma artefacts [15:37] <durandal_1707> saste: mp=il=d:d [15:37] <saste> otoh i confess i don't know what the filter really does [15:37] <saste> yeah [15:42] <saste> durandal_1707, what's the plan with regards to mp=fil? [15:42] <durandal_1707> remove [15:43] <durandal_1707> it is just il that works correctly only when width match linesize [15:43] <saste> i mean it is possible to port such a filter (which according to docs, "does not always work" in unspecified circumstances)) [15:43] <durandal_1707> and fast when works [15:43] <wm4> haha [15:43] <wm4> saste: yeah that freaked me out when I saw it [15:44] <wm4> then I removed it from my mplayer fork [15:44] <durandal_1707> wm4: because you do not know what you are removing .... [15:45] <wm4> well, have fun rewriting it [15:45] <durandal_1707> saste: i could add option to il to use fil way when it will work ... [15:45] <saste> durandal_1707, seems a good idea [15:53] <durandal_1707> wm4: now one cant filter interlaced video with mpv and display it on tv [15:54] <wm4> durandal_1707: how would one achieve this with "fil"? [15:55] <durandal_1707> calling fil=d, fitlers, fil=i (note code is from 2002) [15:57] <wm4> filtering interlaced material and then playing it on a interlaced display...? [15:58] <durandal_1707> yes [15:59] <wm4> well, I guess there are quite some 2002 use cases lost by removing some of these filters, so if you care about these... [16:00] <wm4> I think extended filtering should be left to tools which are suitable for these tasks, such as avisynth (and hopefully vsynth) [16:06] <ramiro> michaelni: I don't remember having commit rights to the git repo. I don't think I ever sent my public key (even if I did, I've changed it a few times in the past two years) [16:22] <michaelni> ramiro, theres a key for Ramiro Polla on the server since Jan 19 2011 [17:03] <durandal_1707> saste: the nv21/nv12 works with il [17:25] <cone-273> ffmpeg.git 03Paul B Mahol 07master:4fa9defc7fde: lavfi/histogram: make waveform mode more useful [17:29] <cone-273> ffmpeg.git 03Paul B Mahol 07master:faac85005cad: lavu/internal: rename macro E(x) to EXPAND(x) [18:02] <ubitux> hey beastd [18:02] <ubitux> how is the subtitles thing going on? [18:02] <beastd> hi ubitux [18:02] <ubitux> you have 3 days left [18:02] <ubitux> ;) [18:03] <beastd> yeah, unfortunately it is not going so well. but that is not a reason to not enjoy your remaining 3 days on another planet ;) [18:03] <ubitux> alright, then i'm going to sleep [18:03] <ubitux> 'night all :) [18:03] <beastd> n8, see you [18:16] <cone-273> ffmpeg.git 03Carl Eugen Hoyos 07master:e149127bb217: Call the common 7.1 wide channel layout "7.1(wide)". [18:55] <durandal_1707> michaelni: what happened to dificulty tag for trac/ [19:03] <michaelni> durandal_1707, could be added but iam not sure its that usefull, open tickets with smaller numbers are harder [19:05] <durandal_1707> michaelni: not always, but yes ... [19:12] <durandal_1707> can lavfi open file and write to it and in another run another filter read file? [19:21] <durandal_1707> did performance dropped with dsputil split? [19:22] <nevcairiel> the functions just get moved to different files and different pointers, why would it get slower? [19:29] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:4401958fdc9a: motionpixels: Propagate errors in vlc table init [19:29] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:b5fc95e77f9e: motionpixels: Check that the vlc table has been fully inited [20:05] <durandal_1707> michaelni: what kind of data rv10 encodes? major/minor/micro/ version... [20:17] <michaelni> durandal_1707, dunno you would have to compare it with the decoder if theres no comment [20:18] <durandal_1707> michaelni: thing is rv10 encoders does not create extradata [20:18] <durandal_1707> and i stops here with invalid slice_count [20:18] <durandal_1707> and nothing sets avctx->slice_count [20:20] <durandal_1707> matroska expects extradata, and decoder too. the last one needs only last 8 bytes [20:22] <durandal_1707> actually i do same thing as rm muxer [20:24] <durandal_1707> which i think is hack because ignores extradata in case of -c copy i think [20:24] <durandal_1707> ^the rm muxer [20:35] <cone-273> ffmpeg.git 03Paul B Mahol 07master:350128b28fe1: rv10: remove check that is always true [20:39] <durandal_1707> the rm muxers adds some extra stuff to each video packet [20:49] <durandal_1707> i think that rv10/rv20 encoder & rm muxer are just wrong [20:50] <durandal_1707> eg muxer does encoder job [21:18] <cone-273> ffmpeg.git 03Nicolas George 07master:f43d09cd60a1: lavf: add tee pseudo-muxer. [21:20] <durandal_1707> michaelni: why ./ffmpeg -f lavfi -i testsrc -vf format=gray16 /tmp/out.bmp picks monob? [21:22] <cone-273> ffmpeg.git 03Michael Niedermayer 07master:796012af6c78: targa: Fix y check in advance_line [21:53] <durandal_1707> saste: i fixed il to work with >8 bpp pix fmts [21:53] <durandal_1707> also mp one crash with pal8 [21:56] <durandal_1707> so i will apply it soon and remove mp once [21:57] <durandal_1707> *one [22:48] <durandal_1707> michaelni: fine to remove ilpack? [22:51] <michaelni> i could only state my oppinon, you still should probably ask on the ML [22:52] <durandal_1707> michaelni: i asked for you opinion? [22:52] <michaelni> i suspect its ok but iam not sure [22:53] <durandal_1707> it is some hybrid scaler and deinterlacer [22:53] <durandal_1707> *reinterlacer [22:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:6e9abce5e2d2: lavfi: port MP il filter [22:56] <cone-273> ffmpeg.git 03Paul B Mahol 07master:c0a8918f2af7: lavfi: remove mp=il filter [23:00] <cone-273> ffmpeg.git 03Paul B Mahol 07master:969039eb6e10: lavfi/il: fix typo in Copyright [00:00] --- Sun Feb 10 2013

1 0

ffmpeg.log.20130209
by burek 10 Feb '13

10 Feb '13

[00:34] <stephanedev> hi everybody. i have issues when streaming a video file using rtp and playing it with ffplay. it works fine when the 2 hosts are on a LAN, but the video is very jerky and most frames are dropped when doing it over the internet [00:35] <stephanedev> the file contains a vp8 video stream 640*360, 25 fps, 500 kb/s [00:36] <stephanedev> ffplay is passed in the sdp file made of the ffmpeg output in the terminal [00:37] <stephanedev> command is: ffmpeg -re -i big_buck_bunny.webm -vcodec copy -an -f rtp rtp://<target_ip>:50004 [00:39] <stephanedev> in the ffplay output, there are a lot of messages like: [00:39] <stephanedev> [vp8 @ 026bc380] RTP: missed 3 packets [00:39] <stephanedev> [sdp @ 026a3e00] Missed unknown data, sequence broken [00:41] <stephanedev> any idea as to what may be wrong? am i missing some command-line option to ffplay? [00:42] <stephanedev> is it expected to receive the sdp error about a broken sequence? [00:49] <Mavrik_> stephanedev: well, obviously it's telling you that you're missing packets [00:49] <Mavrik_> either your bandwidth isn't large enough [00:49] <Mavrik_> or you're having ordering problems [00:59] <stephanedev> Mavrik_: thanks for the answer [01:00] <stephanedev> bandwidth shouldn't be an issue. downloading and video streaming using skype works just fine between the 2 endpoints [03:58] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [04:52] <st000> hello guys is anyone of you familiar with the problem that a from ffmpeg created (avi) file has the right aspect ratio in ffplay, but not in vlc or windows media player? [05:24] <Sam__> Guys i am doing simple flv transcoding using libx264+ffmpeg using single thread, when i see CPU usage using top i could see it cross 112% [05:24] <Sam__> and can see 4 instances of ffmpeg [05:25] <Sam__> 214358 root 20 0 300m 23m 4788 R 90.3 0.5 0:07.17 ffmpeg 214361 root 20 0 300m 23m 4788 S 5.0 0.5 0:00.35 ffmpeg 214359 root 20 0 300m 23m 4788 S 4.4 0.5 0:00.34 ffmpeg 214360 root 20 0 300m 23m 4788 S 4.4 0.5 0:00.35 ffmpeg [05:26] <Sam__> http://pastebin.com/TX5GqjWz [05:27] <st000> llogan: what do you mean with !pb ? [05:29] <llogan> st000: that is a shortcut to tell the #ffmpeg channel bot, fflogger, to tell you to, "please use a pastebin site..." [05:30] <Sam__> any suggestions please [05:30] <llogan> you could always... [05:31] <st000> llogan okay^^ [05:32] <st000> llogan: ehm here my pastbin http://pastebin.com/yHve1nzZ [05:35] <llogan> st000: not that it really matters, but your input is already 704 wide, so why add scale? [05:36] <Sam__> http://pastebin.com/fhqMPuRz [05:36] <Sam__> @ fflogger [05:36] <Sam__> @fflogger http://pastebin.com/fhqMPuRz [05:36] <elkng> anyone around ? [05:37] <llogan> fflogger is a bot, not a person [05:37] <st000> llogan: it was a try [05:37] <Sam__> and CPU usage in Pasebin http://webchat.freenode.net/ [05:37] <Sam__> http://pastebin.com/TX5GqjWz [05:38] <st000> even if i let it go it will not change anything [05:39] <llogan> what does vlc or WMP think the aspect ratio is? [05:39] <st000> 4/3 [05:39] <st000> or 1/1 [05:40] <st000> i think it is the last thing [05:40] <llogan> do you have a sample? [05:41] <st000> yes [05:41] <Sam__> @<fflogger> any suggestions? [05:42] <st000> it is crud it anything between 5:4 and 4:3 [05:45] <klaxa> Sam__: suggestion 1) update your ffmpeg 0.10 is pretty damn old [05:45] <klaxa> suggestion 2) add: -threads 1 [05:45] <Sam__> -threads 1 is already add in commandline [05:46] <klaxa> probably add it before -x264ops [05:46] <klaxa> *x264opts [05:46] <klaxa> also i'm not sure that's supported anymore [05:46] <klaxa> i could be wrong though [05:47] <Sam__> no effect adding it before x264opts as well [05:49] <klaxa> mmh dunno [05:49] <Sam__> @ <klaxa> I disagree with you here about options as these all are latest options for both [05:50] <Sam__> ffmpeg and libx264 [05:50] <klaxa> your ffmpeg version is 0.10 [05:50] <klaxa> that's, if i may say so, old as fuck [05:52] <Sam__> ok [05:54] <klaxa> grab the latest ffmpeg and take a look at this http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [06:02] <gudy> Hi, is it valid for a GOP to not start with an I frame, but with B frames which reference a future I frame. The I frame is first in the decode sequence, but not first in terms of presentation time ? [06:03] <gudy> (in h264 high profile) [06:12] <st000> llogan: do you want to have the sample ? [08:28] <jac1d> Hello all. I have a number of PCM streams coming from FM tuners to a single x86 CPU and I want to encode them to AC3 stereo in real time. What kind of CPU power would I need to do this? [09:42] <int> hello guys, I want to compress video file a bit without a big quality loss, but I receive the followint error: http://paste2.org/p/2841453 [09:42] <int> could somebody help me? [09:42] <int> maybe I should change codec? [09:45] <relaxed> int: ffmpeg -i avn1.mkv -c:a libmp3lame -b:a 128k -c:v libx264 -preset veryslow -crf 18 output.mkv [09:45] <sacarasc> Why are you using h263? [09:45] <int> sacarasc: I thought it is the best for compression [09:46] <int> relaxed: thank you very much [09:46] <sacarasc> H264 is what you wanted, using libx264. [09:46] <int> sacarasc: hm [09:46] <relaxed> The source is h264, so I doubt you'll gain much more compression. [09:46] <int> ok, I will try [09:47] <int> does redusing resoultion help me? [09:48] <sacarasc> Yes, it would. [09:53] <int> -s 1024x570? [10:17] <sacarasc> -vf scale=1024:570 I think is the proper way these days. [10:20] <int> ok, thank you [11:08] <creep> h [11:08] <creep> another nice nuclear meltdown in result of stupidity from japan, have a good appetite for fish from sea http://en.wikipedia.org/wiki/Lists_of_nuclear_disasters_and_radioactive_inc… [11:10] <elkng> creep: when was the latest ? in fukushima ? [11:10] <elkng> or there is new incident ? [11:12] <creep> elkng<< about a year ago, you know, it taskes some time to share this unimportatnt information :) [12:02] <sceriffo> hi , where can I find a code for generation extradata for AAC decoder ? [12:03] <creep> elkng<< soon, chinese will start selling radioactive flashlights, then they will be able to get rid of their radioactive waste legally, no recharge, 100 years of light, cheap, made in china... [12:13] <elkng> creep: "radioactive flashlights", its creepy [12:29] <iive> creep: funny. but not really accurate. [12:29] <iive> all flashlights are radioactive, because light is radioactive radiation :) [12:35] <Mavrik> hmm, does anyone know what "afterburner" parameter on Fraunhofer AAC encoder actually does? [12:36] <JEEB> seemingly optimizes more on quality than speed [12:36] <JEEB> what it *actually* does tho is something you'll have to ask wbs [12:38] <Mavrik> hmm [12:38] <Fjorgynn> seems like I [12:38] <Mavrik> basically I have a Motorola STB here which chokes as soon as I enable afterburner [12:39] <Mavrik> JEEB, wbs? [12:41] <JEEB> IIRC that's the IRC nickname for Martin Storsjö, who made fdk-aac from an Android thing to a library :P [12:41] <JEEB> and also who pushes fixes towards back to Android [12:41] <Mavrik> ah, ok :) [13:05] <DeLoreann> Hi! I have a problem with ffmpeg. While recording rtmp stream from ustream, quality decreases during recording. Anyone? (sorry for my english ;) [13:05] <DeLoreann> Here is a sample: http://bocik.pl/ss/player.php?v=njIdVI.flv [13:18] <burek> ive just set up ffmpeg as a p2p voice tool :) [13:18] <burek> btw, which audio codec is the most suitable for p2p voice chat [13:18] <burek> (the least delay, speech codec) [13:18] <JEEB> the one mumble uses, opus [13:21] <burek> hmh, it doesnt produce any errors [13:21] <burek> but sending it over udp with -f mpegts doesnt work not even in vlc [13:21] <burek> i mean, i cant hear anything [13:22] <JEEB> are you sure vlc is equipped with libopus decoding? are you sure its muxing into mpeg-ts is somehow supported by Various Things? [13:22] <burek> oh [13:22] <burek> how should i send it then? [13:22] <burek> i can use ffplay [13:23] <JEEB> to be honest I have no idea whatsoever, I think mumble just sends out a raw opus stream or whatever. [13:23] <JEEB> not sure if ogg has a set up mapping for opus, matroska's one is still in the works [13:25] <burek> ill try ogg too, thanks :) [13:25] <JEEB> as the container, that is [13:25] <JEEB> or just raw opus [13:25] Action: JEEB shrugs [13:26] <burek> how to specify -f raw [13:28] <DeLoreann> Mavrik: http://pastebin.com/SSjM5Gx9 [13:28] <JEEB> burek, see -formats for opus [13:28] <burek> thanks [13:29] <Mavrik> DeLoreann, your quality drops because you're re-encoding video to 200kb/s [13:29] <Mavrik> DeLoreann, use "-codec copy" as parameter [13:29] <Mavrik> to force only copy of stream, not reencode [13:30] <DeLoreann> "Unrecognized option 'codec'" [13:30] <Mavrik> DeLoreann, ah, you have an older version, it's "-vcodec copy" [13:31] <DeLoreann> [flv @ 0x808040020] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 100 >= 0 [13:31] <DeLoreann> av_interleaved_write_frame(): Invalid data found when processing input [13:32] <DeLoreann> output file: http://bocik.pl/ss/player.php?v=abfwcZ.flv [13:32] <DeLoreann> 51kB, (one frame?) [13:33] <Mavrik> hmm [13:33] <JEEB> DeLoreann, update your ffmpeg [13:33] <Mavrik> that shouldn't happen really... but you do have a rather old fmfpeg [13:33] <Mavrik> *ffmpeg [13:33] <JEEB> yes, it's /very/ old [13:34] <DeLoreann> im doing it on shell account, this will take time :) [13:34] <burek> JEEB, sorry for bugging you, but what exactly should i be looking in the -formats ? [13:34] <DeLoreann> anyway, thanks Mavrik! [13:34] <JEEB> it should have a raw format for opus, if there isn't... then there isn't :D [13:34] <burek> there isn't :) [13:34] <Mavrik> DeLoreann, grab static builds :) [13:35] <JEEB> Mavrik, he's on freebsd [13:35] <JEEB> lol [13:35] <Mavrik> oh. [13:35] <Mavrik> ah. [13:35] <Mavrik> that could be problematic then :D [13:35] <JEEB> burek, well then try -f ogg or something, can't really come up with anything better :D [13:36] <burek> ok, i'll try various test and see which one works then, thank you :) [13:42] <someone-noone> Hello! I'm trying to pass "magic cookie"(extradata) data to mac os Core Audio framework for decoding AAC format. File is in m4a format. For this I'm using extradata from my AVCodecContext. But extradata_size is 7, however if I read "magic cookie" using OS X API, it's size is 44 for same file. Looks like magic cookie != extradata in ffmpeg (but should be). Any help how can I extract this data from ffmpeg? [16:24] <elkng> anyone around [16:24] <elkng> ? [16:26] <elkng> that is the file http://sprunge.us/QQeM I'm trying to convert, that is the command http://sprunge.us/MAgd I use to convert it, that is the output http://sprunge.us/FHRa, it's length is 23:19 but it converts only first 4:57 and then stoped convert [16:26] <elkng> with errors, what is wrong ? [16:27] <elkng> and similar problems happened for last couple days about 4 times, it converts first few minutes and then stopped with errors [16:31] <durandal_1707> elkng: does number of channels really change to that big number? [16:31] <elkng> what number ? [16:31] <durandal_1707> in log, ich [16:32] <elkng> ffprobe says: "Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 192 kb/s" [16:32] <durandal_1707> [SWR @ 0x8a20640] Value 24.000000 for parameter 'ich' out of range [16:32] <durandal_1707> Unsupported number of input channels [16:32] <elkng> something wrong with video ? [16:33] <durandal_1707> maybe something get corrupted, does it play ok with other software? [16:35] <elkng> durandal_1707: no, seems like I need to redownload video, because mplayer crashes playing original mp4 file after 4 minutes [16:36] <durandal_1707> mplayer uses old lavf/lavc too [16:37] <durandal_1707> you need qt time or wmp that does not use lavf/lavc [21:12] <llogan> DeLoreann: did you figure it out. there is at least one other *BSD user here that may be able to provide assistance. [21:14] <DeLoreann> i managed to solve the problem by "-acodec copy -vcodec copy", here is sample: http://bocik.pl/ss/player.php?v=AiBeEC.flv [00:00] --- Sun Feb 10 2013

1 0

ffmpeg-devel.log.20130208
by burek 09 Feb '13

09 Feb '13

[00:15] <saste> anybody going to review the opencl deshake patch? [00:21] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:0d194ee51ed4: vc1dec: Fix destination pointers in context for fields [00:25] <michaelni> saste, iam lazy, you can review it [00:25] <saste> michaelni, i'm busy [00:25] <saste> i'll try to review it anyway, in the weekend [00:26] <saste> also i know no s*#t about opencl [00:27] <michaelni> anyone has any odd platforms/hw that are missing from fate.ffmpeg.org ? ... and wiling to setup fate clients ? [00:28] <michaelni> we now also have a SH4 client since a few days (qemu based though) [00:29] <saste> michaelni, i have a ZX spectrum, might that be helpful? [00:30] <michaelni> can it run ffmpeg / fate ? [00:42] <michaelni> saste, you could optimize and port ffmpeg to ZX spectrum ;) [00:43] <michaelni> maybe a striped down version that does support just one container and codec or so [00:43] <saste> uhm in 128KiB it's going to be hard [00:44] <michaelni> uhm why ? [00:44] <michaelni> there was a 3d first person shooter in less than 64k IIRC [00:44] <michaelni> not on ZX spectrum though id guess ;) [00:51] <gnafu> If anyone could do it, it would be Fabrice Bellard. And so I would think a project he started could be transformed to do it. [00:51] <gnafu> ;D [01:13] <saste> well at least we could have a player on zx [01:13] <saste> after all we have an encoder for c64 [01:48] <ubitux> hey [01:49] <ubitux> michaelni: a00c4b4db seriously? oO [01:49] <michaelni> yes seriously [01:49] <michaelni> no joke [01:50] <ubitux> why does this happen? [01:50] <michaelni> no idea, gcc i guess [01:51] <michaelni> but + is nicer for gcc [01:51] <michaelni> it allows o2*=a2; o2+= ... [01:51] <michaelni> with - that style would not be possible for gcc to assemble to [01:51] <michaelni> but i didnt check if gcc actually does this [01:52] <ubitux> fun [01:53] <ubitux> kyowa, tokyo desu!~ [01:53] Action: ubitux & [02:12] <Compn> michaelni : wei was asking in the dxva2 mail "Also I want to ask a question that is the chroma_format always stored in the ext sequence header?" [02:12] <Compn> he has some other questions in that mail too [02:27] <michaelni> Compn, what mail (subj/date) ? [02:54] <michaelni> Compn, found the mail & replied [03:00] <Compn> oop, ok good :) [03:00] <Compn> thanks for looking at it [11:45] <cone-778> ffmpeg.git 03Paul B Mahol 07master:810cd0dd9c03: riff: chunks must be at 2 byte boundary [12:14] <durandal_1707> michaelni: the fil filters is buggy, it interleaves incorrectly [12:51] <michaelni> durandal_1707, libmpcodecs vf_fil ? [12:51] <michaelni> does the isssue happen just with ffmpeg or also mplayer ? [12:51] <michaelni> its a mplayer filter and not maintained in ffmpeg [12:52] <durandal_1707> michaelni: i dont have mplayer installed [12:52] <durandal_1707> i just did fil=i,fil and expect to see same output [12:55] <michaelni> fil was designed to be fast (0 copy) IIRC it can be wrong sometimes [12:55] <durandal_1707> i want to port it [12:56] <durandal_1707> so i want to know what is expected result [13:00] <michaelni> i think these artifacts are expected for stride != width [13:06] <durandal_1707> michaelni: so its not useful at all? [13:07] <michaelni> might be usefull if you have a slow box and need it for realtime playback ... [13:08] <michaelni> but the copy would not be that slow so its probably not that usefull [13:08] <durandal_1707> michaelni: so what about making port that would do copy and be morae usefull [13:09] <michaelni> yes if it cant do it without copy & without artofacts it should copy [13:10] <michaelni> maybe with a user override switch that allows forcing non copy [13:10] <michaelni> but that would have very few users [13:10] <michaelni> also see vf_il [13:24] <durandal_1707> there is no filter that blends to images? [13:24] <durandal_1707> *two [13:36] <Compn> durandal_1707 : blend how ? [13:37] <Compn> durandal_1707 : the stereo3d puts images side by side [13:37] <Compn> or pad or that other filter [13:40] <iive> or interlace, or one above the other, or codes them red/blue .. ;) [14:16] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:47335126c049: ff_end_tag: assert that the start is aligned. [14:16] <cone-778> ffmpeg.git 03James Almer 07master:89946c840375: dxva2: Prevent redefinition of _WIN32_WINNT [16:14] <durandal_1707> ubitux: when you will become live again? [16:14] <ubitux> around wednesday [16:15] <durandal_1707> ok, don't drink too much blood [16:20] <durandal_1707> michaelni: so fil is just "fast" variant of il? [16:28] <michaelni> yes [16:28] <durandal_1707> ok, why they exist at all? [16:30] <michaelni> il/fil allow for example frame based filters to be easily applied to fields [16:30] <michaelni> probably they have other uses too [16:31] <durandal_1707> hmm, so i will just port il - could i pick less retarded name? [16:32] <durandal_1707> and fil will be just special case of il [16:43] <durandal_1707> michaelni: hmm mp=il=d gives obviously broken output here [16:55] <durandal_1707> nvm, i deinterleave only luma plane ... [16:55] <durandal_1707> michaelni: are you ok to relicense this trivial code to LGPL? [18:14] <michaelni> durandal_1707, the vf_il.c (4289byte) code (the parts i wrote) can be used under LGPL, its just some memcpies anyway [23:52] <cone-778> ffmpeg.git 03Michael Niedermayer 07master:b8820e7aa338: dsp: make put_no_rnd_pixels_tab 4x4 big [00:00] --- Sat Feb 9 2013

1 0

ffmpeg.log.20130208
by burek 09 Feb '13

09 Feb '13

[00:56] <stephanedev> saste: i just want to thank you again for your help. i am now setting "lag-in-frames" to zero for the vp8 encoder and latency is much lower. [00:56] <saste> stephanedev, of course you can't achieve the same performance with all encoders [00:57] <saste> libx264 -tune zerolatency is fu***ng brilliant [00:57] <saste> most other encoders may not be suited for that kind of use [00:57] <stephanedev> yeah i saw that option for x264 [00:58] <stephanedev> for now we settled on vp8 because we are not sure about x264 license [00:58] <JEEBsv> the software license or licensing of the H.264 format? [00:59] <JEEBsv> x264 at least is available for payware licensing if you don't want to touch GPL [00:59] <stephanedev> the format [00:59] <JEEBsv> yeah, that depends on how you're using it, don't remember where the MPEG-LA pdf about it was [00:59] <stephanedev> for ffmpeg we use lgpl [01:00] <JEEBsv> also IIRC MPEG-LA is rather open for talks, unlike VIA licensing f.ex. [01:00] <JEEBsv> so they should be rather easily contactable [01:01] <stephanedev> ok, thanks for the advice. we may contact them if vp8 does not do the job or customers want x264 [01:02] <stephanedev> by the way, i assume i need to build my own ffmpeg DLL with the proper options to remove everything that is not lgpl if this is the license i use, even if i don't use the glp bits? [01:02] <stephanedev> gpl* [01:03] <JEEBsv> yes [01:04] <JEEBsv> people have a tendency of checking the license you built/used your ffmpeg with [01:04] <stephanedev> thank you [01:04] <JEEBsv> and trolling you if you linked against a GPL ffmpeg [01:04] <JEEBsv> so yeah, building an LGPL library is a *good* idea [01:05] <stephanedev> it makes sense. we want to respect every third-party license, so i will build a lgpl version before we distribute our software [01:06] <JEEBsv> heh, I wish every developer was like you :P [01:07] <stephanedev> that's the least we can do, given how helpful some of these libraries are [01:08] <JEEBsv> people copying their ffmpeg confiuration from random GPL project X, and then subsequently using it with their non-open source app... [01:08] <JEEBsv> I wish that didn't happen as much as it does [01:10] <stephanedev> some developers don't bother to check the license text and just assume they can do anything with source code they find online [01:11] <JEEBsv> yup [01:42] <Keshl> stephanedev: Sorry for randomly going afk. Mind slipped. I swear I have early onset alzheimers. <É< http://pastebin.com/uN9RnWng [01:44] <stephanedev> hmm i usually get the fps on the video stream line, i.e. the one starting with Stream#0:0: Video [01:45] <stephanedev> let me try with a flv file on my system [01:49] <stephanedev> Keshl: i don't have fps either when probing a flv file. but it seems you can rely on the tbr: http://stackoverflow.com/questions/3199489/meaning-of-ffmpeg-output-tbc-tbn… [01:49] <Keshl> oÉo. [01:50] <Keshl> 25 tbs == 25 fps, or "about 25, maybe 24"? [01:53] <stephanedev> Keshl: i guess so, but check the last paragraph of the second message about tbr on: http://ffmpeg-users.933282.n4.nabble.com/What-does-the-output-of-ffmpeg-mea… [01:53] <Keshl> "I hope that helps a little.", oÉo? [01:53] <stephanedev> the one just before that :p [01:53] <Keshl> Ohhh oÉo [01:54] <Keshl> ... What kind of video is 25 or 12.5 fps? [01:54] <Keshl> o.O' [01:54] <stephanedev> i don't know what the field rate is, sorry [01:54] <Keshl> How does vlc know what rate to play it at? o_o' [01:54] <stephanedev> i am just a ffmpeg API user still struggling with some of the concepts involved :p [01:55] <stephanedev> it knows with the container rate and the codec rate [01:55] <Keshl> So how does I calculate that? [01:55] <stephanedev> the time bases in AVStream and AVCodecContext [01:55] <Keshl> o_o [01:56] <stephanedev> each packet is timestamped [01:57] <Keshl> <.< Can I get a dump of the first 60 timestamps [01:57] <Keshl> *? [01:57] <Keshl> =D [01:57] <stephanedev> the unit for the packet timestamp read from the file is in stream time base units (hope i am correct) [01:58] <stephanedev> not sure how to do it with ffmpeg command-line [01:58] Action: Keshl dies. [01:58] <Keshl> x.x [01:58] <stephanedev> but if you can program, the ffmpeg C API allows you to do that [01:59] <Keshl> o_o Anything beyond the standard library makes me wet myself. [01:59] <stephanedev> the demuxing.c example is a good starting point if you want to try [02:01] <stephanedev> you can easily modify demuxing.c (http://ffmpeg.org/doxygen/trunk/doc_2examples_2demuxing_8c-example.html) by adding some traces around the line calling avcodec_decode_video2 [02:02] <stephanedev> some printf's to show some of the members of pkt and frame (like pkt.pts, ...) [02:08] <stephanedev> Keshl: for ffmpeg command-line, "-debug_ts" may be useful [10:19] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [10:19] <portal`> would love for someone to chime in on why it might be. [10:19] <portal`> ffmpeg version N-49527-g985e93a [10:26] <portal`> running the same command usually will stop at around 8 seconds, but i just had it go to 48 seconds and stop there [14:07] <jeje34> hi to all [14:08] <jeje34> I have a problem using the last ffmpeg release (1.1.1) to decode h264 stream [14:10] <jeje34> when I want to enable multithreading to decompress, I always got an error when I open the codec (The maximum value for lowres supported by the decoder is 0) [14:10] <jeje34> But in the code of utils.c, this message appear when: if (avctx->codec->max_lowres < avctx->lowres || avctx->lowres < 0) [14:11] <jeje34> if I trace my code I have avctx->lowres=0 and avctx->codec->max_lowres=0 [14:11] <jeje34> I can't understand why I have this error [14:12] <jeje34> my codecctx have thread_count=3 (I have a dualcore) and I set thread_type = FF_THREAD_FRAME|FF_THREAD_SLICE [14:12] <jeje34> Thanks for your responses [14:15] <jeje34> for informations, I compile FFMPEG with Mingw with --enable-w32threads because I use ffmpeg libs in a windows application (using Visual Studio vc++ 2003) [14:29] <jeje34> the error code resulting from avcodec_open2 is -22 [14:52] <iive> jeje34: i don't think you are supposed to change max_lowres. [14:53] <iive> imho, if you don't use it, don't touch lowres too. [14:53] <iive> you may want to print the context after you get the error and see the value of lowres. if you think this is what caused the error. [14:54] <jeje34> I don't modify the value of max_lowres and lowres. When I look at the value, the two values are set to 0 (I think the default values) [14:54] <iive> aha. good... [14:55] <jeje34> I just call avcodec_alloc_context3 to get my context and so avcodec_get_context_defaults to set the default values [14:55] <iive> another possibility is that something got wrong with some of the other threads. [14:55] <iive> they have their own contexts, that are cloned from the main one... [14:55] <iive> most likely some tricky memory corruption. [14:57] <iive> just something for you to check. win32 doesn't have memalign, so ffmpeg have special memory allocation, that allocates more memory and then uses aligned memory into that allocation. [14:57] <jeje34> but what I don"t understand is why before calling avcodec_open2(codecCtx, codec, NULL) I have lowres=0 of my codecCtx and max_lowres=0 in my codec but I have the error message [14:57] <iive> be sure that memory allocated by ffmpeg function is free'd by ffmpeg functions and vice versa. [14:58] <iive> jeje34: put some debug printf's or breakpoints into the code that prints the error, and check the values there. [14:58] <jeje34> I use the ffmpeg calling to allocate and free memory [14:59] <iive> good. [15:00] <jeje34> I try to activate multithreading because I have a dualcore. When I decompress one stream, my CPU usage is near 1-2%. When I decompress 2 streams, I have 3-4%. But when I decompress 3 streams, i tkae me 25% of CPU usage. [15:00] <jeje34> Do you think activate Multithreading could decrease my cpu usage? [15:01] <iive> i didn't quite get that. [15:03] <jeje34> I'm going to rebuild FFMPEG with more traces I think [15:10] <iive> i would have recommended running it under valgrind, but I'm not sure if it is available under windows. [15:12] <JEEBsv> nope [15:12] <JEEBsv> there's a crap port, but that doesn't really work :S [15:14] <jeje34> I already have a look trying using GDB tools to debug FMMPEG under VisualStudio but I don't find naything working [15:14] <jeje34> opps anything sorry [15:15] <jeje34> Is someone have an idea how to configure the AVContext to use the lowest CPU usage when decoding H264 video streaming? [15:16] <jeje34> or perhaps in all the configurationof compilinf FFMPEG too [15:16] <jeje34> there's too much options... [15:22] <jeje34> my configure option are: ./configure --prefix=/mingw/i686-pc-mingw32 --enable-shared --disable-static [15:22] <jeje34> --enable-w32threads --enable-runtime-cpudetect --enable-memalign-hack --enable- [15:22] <jeje34> hwaccels --cpu=i686 --enable-debug=3 (I had enable-debug to try to see more reason) [15:33] <jeje34> for the CodecContext, I use error_concealment= FF_EC_GUESS_MVS | FF_EC_DEBLOCK, err_recognition = AV_EF_CAREFUL, debug = FF_DEBUG_THREADS [15:55] <flexnsniff> So i'm trying to stream RTMP to AMS5& streaming a movie to it. It works when i use the live/ app, but then is a green screen when i use the livepkgr/ app. Should i set ffmpeg differently or change how AMS works? [15:57] <jeje34> I add some trace in Utils.c of libavcodec, and so on, I have avctx->codec->max_lowres=0, avctx->lowres=3 [15:57] <jeje34> very strange [15:57] <jeje34> but it explain why I have this message when using avcodec_open2 [16:08] <Damik> Hi, I've problem with compiling ffmpeg under msvc toolchain. I've got - c99wrap cl is unable to create an executable file. [17:16] <mpfundstein> is here someone who has experience with sar/dar stuff? [17:16] <JEEBsv> in what way? [17:17] <mpfundstein> i have a video which is 720x480 (par 32/27, dar 16/9). all my calculations lead me to an actual width of 853x480 but quicktime reports 872x480 , which also works well in my formulas. i cant find any formula which leads to an actual width of 872 [17:17] <mpfundstein> i tried so many formulas, all of them say actual width = 853, but i need 872 :-) [17:18] <JEEBsv> are you sure there is no other metadata related? [17:18] <JEEBsv> also what video format is that in? [17:18] <mpfundstein> :( i dont know [17:18] <mpfundstein> video format? [17:18] <mpfundstein> mpeg2 [17:18] <mpfundstein> one sec [17:18] <mpfundstein> ill give you a ffprobe output [17:18] <JEEBsv> k [17:19] <mpfundstein> https://gist.github.com/MarkusXite/e7f3b032fe698041c985 [17:19] <mpfundstein> 720*((16/9)/1.5) = 853 [17:20] <mpfundstein> 1.5 = 720/480 [17:21] <JEEBsv> 875 or so is the DVD "16:9" aspect ratio (the whole size of the active+non-active picture areas), but unfortunately that really doesn't give us enough info on what the actual values either in the MPEG-2 video track or in the MOV container are [17:22] <mpfundstein> how can quicktime calculate it right [17:22] <mpfundstein> when i open the video in quicktime it says 720x480 (actual: 872x480) [17:22] <JEEBsv> depends on several things, none of which I can say anythin about definitely [17:22] <JEEBsv> because as I said, not enough info [17:22] <mpfundstein> ok [17:22] <mpfundstein> what do you think is missing? [17:22] <mpfundstein> :-) [17:23] <JEEBsv> I have no fucking idea, a full dump of the MPEG-2 Video headers and the full dump of data in the container would be needed for that [17:23] <JEEBsv> for dump of the MOV file, you can try http://code.google.com/p/l-smash/ 's boxdumper [17:23] <JEEBsv> (output boxdumper's output into a text file, then go through it) [17:24] <mpfundstein> ok [17:24] <mpfundstein> so its probably encoded with non visible data on the left and right [17:24] <JEEBsv> as for the MPEG-2 stream, you'd need something like elecard streameye to check the headers [17:24] <JEEBsv> mpfundstein: no [17:24] <JEEBsv> the 720x480 picture in DVDs is by the specs encoded with a SAR of 40:33 [17:24] <JEEBsv> which ends up being 875x480 or so [17:24] <mpfundstein> AH [17:25] <JEEBsv> (the active picture is 704x480) [17:25] <mpfundstein> that is really f**** up [17:25] <JEEBsv> (and 704x480 with a 40:33 AR is 853x480 or so) [17:25] <mpfundstein> mh [17:25] <JEEBsv> not really [17:26] <JEEBsv> so in the simplest case, QT just takes MPEG-2 Video in with these aspect ratios, but somehow I /really/ wouldn't like to think that's all it bases its guess upon [17:26] <JEEBsv> thus [17:26] <JEEBsv> go look at 1) the boxdumper's dump of the mov's metadata 2) the MPEG-2 Video headers [17:26] <mpfundstein> im screwed -.- [17:26] <Mavrik> hmm, how did I manage to miss the existence of AAC Main profile [17:27] <mpfundstein> ok thx JEEBsv, ill do more research on that [17:32] <casimiro_> Helloa all, is there anybody here who have combined ffmpeg with libev ? [17:35] <casimiro_> Or better, is there anyway that I can give a buffer with mpeg ts received from an tcp socket to libavformat so that I can extract the mpeg ts packets ? [18:27] <iive> casimiro_: ffmpeg merges most of libav, but it also adds its own stuff and fixes. [19:23] <casimiro_> thank you for your response iive. Do you know of anyway I pass a buffer containing an mpeg ts stream to some ffmpeg function and I get the mpeg ts packets ? [19:24] <iive> casimiro_: i'm not familiar with libavformat. but imho, mpeg-ts packets are the easiest to parse. they are always of same size and start with same sync byte. [19:28] <casimiro_> iive: In fact! But I have written an little app that receives mpeg ts from an TCP socket and send the stream through an UDP socket. In this case I need to splice the stream in order to send the udp packets correctly. I am able to find the sync bytes, but it seems that after some few packets I need to synchronize (find the sync bytes again..) [19:29] <casimiro_> iive: It seems that it's not only a matter of finding a sync byte and split every 188 bytes ahead [19:29] <iive> casimiro_: do you send exact multiples of the ts packet size? [19:30] <iive> udp could easily be lost or delivered out-of-order. [19:30] <Mavrik> casimiro_, so basically you want to read TCP stream [19:31] <Mavrik> casimiro_, and then forward it over UDP? [19:31] <Mavrik> without remuxing? [19:31] <casimiro_> Mavrik: yes [19:31] <Mavrik> hmm [19:32] <Mavrik> MPEG2-TS packetization is handled via MPEG-TS muxer in ffmpeg, so I'm not sure if there's an elegant way of doing that with ffmpeg libs [19:32] <Mavrik> (without remuxing that is) [19:32] <casimiro_> I see [19:33] <casimiro_> Should not it be just a matter of splicing the incoming stream on every 188 bytes ? [19:33] <Mavrik> casimiro_, yes, but you need to splice them on the packet boundaries [19:33] <Mavrik> casimiro_, basically you're looking for 0x47 sync byte at the start of each TS packet [19:35] <casimiro_> Exactly. And everything works fine at the beginning. But after a few packets, there is not anoter sync byte after 188 bytes from the current sync byte [19:35] <casimiro_> I understand that I will need to synchronize things after some time, but I am having to synchronize to often... [19:36] <casimiro_> I'm sorry if it's too much noob problem. =) [19:36] <Mavrik> not a noob problem really :) [19:36] <casimiro_> Thx! [19:39] <Mavrik> casimiro_, btw, are you checking continuity counters if your packets are in order? [19:41] <casimiro_> No...All I do for now is identify a sync byte send one UDP message for every 188 bytes ahead [19:42] <Mavrik> casimiro_, check those so you see if your packets are ordered properly on the input [19:42] <Mavrik> it's a 4-bit field in packet head [19:44] <casimiro_> But, an not ordered packet would implicate on bytes between packets ou packets longer or shorter ? [19:44] <casimiro_> Also, I am receiving the data from an HTTP request... [20:09] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds. [21:06] <shevy> quick question - can I easily combine .mp4 files via ffmpeg? [21:31] <shevy> ah well, the mencoder variant worked fine. cya. [21:32] <walker> Hi folks - I'm using ffmpeg to transcode a live video stream, from RTP to RTMP. [21:33] <walker> It works fine in good conditions, but I have an instance where the RTP sender starts with video muted. [21:33] <walker> So it doesn't start sending RTP until a few seconds in. During that time, ffmpeg starts up, can't analyze the video stream, and just does audio. [21:34] <walker> And won't do video after that (packets received from video stream are marked as NULL stream in ffmpeg -v debug) [21:34] <walker> How can I get around this? [21:34] <cheeseboy> with -f image2 -usefirst can i limit # of writes per second or something? [21:35] <beastd> cheeseboy: what do you want to do? [21:36] <cheeseboy> beastd, i want to load a preview frame ron my website but if its writting it 20 times a second its lil hard to load [21:36] <beastd> cheeseboy: you could use -re and probably probably set a different framerate [21:37] <cheeseboy> beastd, i want it to still encode at right rate thoo [21:37] <Fjorgynn> can I do -ss and copy and burn in the subs? [21:39] <beastd> cheeseboy: hmm. probably i still do not understand what you want. [21:39] <beastd> Fjorgynn: probably not [21:41] <beastd> cheeseboy: if you encode faster you will write out more frames per second and therefore you will be overriding the outputfile more often. [21:42] <beastd> Fjorgynn: I meant, you cannot use copy for the video stream you want to burn the subs onto. [21:52] <Fjorgynn> seems like I can't copy an ass file to that [22:15] <Youka> @shevy: have a look at <http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join,%20merg…> [00:00] --- Sat Feb 9 2013

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