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May 2014

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ffmpeg-devel.log.20140510
by burek 11 May '14

11 May '14

[00:11] <cone-85> ffmpeg.git 03Andrey Utkin 07master:8cc6e91f6950: Document smoothstreaming muxer [00:51] <brimestone> hey guys... i'm getting frustrated with AVFoundation and wants to add ffmpeg into my Objective-C project, where shold i start.. any pointers? [03:00] <jamrial> libav keeps postponing the native opus decoder because any fix or improvement seemingly needs to be squashed into the first big commit [03:09] <iive> yes... it must be perfect on the first commit or the project would crash in flames. [03:10] <iive> and all copyright assignments should be scrubbed clean, so nobody knows who changed what. [03:12] <iive> It's not like (L)GPL requires proper copyright attribution. [03:12] <iive> ~~ [03:32] <jamrial> I mean, i could see them wanting to do that with show stopping bugs, but to factor out code for performance reasons and remove an unused variable? [03:33] <jamrial> v5 will probably not be posted until like monday, then the cycle will begin anew [03:34] <jamrial> because that "int j" line must not make it into the tree under any circumstances [03:40] <compn> jamrial : why do you want it rushed into libav ? thats now how they roll [03:41] <jamrial> well, I wouldn't call it rushed at this point [03:45] <iive> well, it is not like they have a competition or something... [03:45] <iive> ;) [04:04] <cone-551> ffmpeg.git 03Michael Niedermayer 07master:dc6b99d6b20e: avformat/movenc: Check ff_put_wav_header() return code [04:23] <jedir0x> does avpicture_fill copy the data - or just copy the pointers to the data? [12:42] <funman> http://ks.chui-pas.net/~fun/ffmpeg_chocolate.jpg [13:44] <ubitux> michaelni: with which subtitles/cmd did you trigger that issue? [13:58] <ubitux> i'm surprised ubsan doesn't have an abort mode [14:41] <michaelni> ubitux, you mean for "avformat/mux: 2 subtitle packets could have the same DTS" ? [14:41] <ubitux> yes [14:41] <michaelni> IIRC the one from carl in the referenced ticket [14:42] <ubitux> ah, missed that, my bad [14:43] <michaelni> funman, i just woke up and looked at that pic, got me confused for a second ;) [14:46] <BBB> hcocolate yummy [14:46] <BBB> do we get some? [14:54] <cone-56> ffmpeg.git 03Michael Niedermayer 07master:2e55f3e90715: avfilter/vf_lut3d: check size in parse_dat() [14:55] <ubitux> michaelni: does it have the same pts too? [14:55] <michaelni> -debug_ts should show it [14:56] <michaelni> I dont remember [14:58] <michaelni> demuxer -> ist_index:10 type:subtitle next_dts:426311 next_dts_time:0.426311 next_pts:426311 next_pts_time:0.426311 pkt_pts:7783637667 pkt_pts_time:86484.9 pkt_dts:7783637667 pkt_dts_time:86484.9 off:-86484364989 off_time:-86484.4 [14:58] <michaelni> demuxer -> ist_index:10 type:subtitle next_dts:497978 next_dts_time:0.497978 next_pts:497978 next_pts_time:0.497978 pkt_pts:7783637667 pkt_pts_time:86484.9 pkt_dts:7783637667 pkt_dts_time:86484.9 off:-86484364989 off_time:-86484.4 [14:59] <michaelni> so yes, look the same on the input side already [15:01] <cone-56> ffmpeg.git 03James Almer 07master:fa23190a7ac9: hevcdsp: add missing header include [15:13] <funman> michaelni: ;) [15:14] <funman> BBB: there is lots of chocolate in Berlin [15:36] <cone-56> ffmpeg.git 03Matt Oliver 07master:c6bf660bef8f: opencl: Make init_mtx static with volatile lock [15:37] <BBB> funman: for me to eat chocolate, it needs to be in the same location as me [15:45] <funman> BBB: where are you? [15:45] <BBB> funman: ny [15:45] <BBB> can you send me some? [15:45] <funman> http://shop.borgodoro.it/products/libav-chocolate can [15:45] <funman> if you were in mountain view i could have brought some next month [15:47] <BBB> but I'm not in mtv :-p [15:47] <wm4> doesn't this libav chocolate use the ffmpeg logo [15:48] <BBB> anyway maybe next time you guys are in the proper city [15:48] <BBB> :-p [15:48] <funman> wm4: http://ks.chui-pas.net/~fun/ffmpeg_chocolate.jpg clearly does [15:53] <nevcairiel> wm4: its the same logo :p [18:54] <cone-56> ffmpeg.git 03James Almer 07master:8e078000010b: hevcdsp: include stddef.h for ptrdiff_t definition [19:38] <ubitux> michaelni: why not add the TS_NONSTRICT fmt flag to teletext instead? [19:38] <ubitux> ah my bad it's for formats meh [19:38] <michaelni> yes, meh [19:38] <ubitux> well i haven't work on dvb teletext [19:38] <ubitux> i have no idea if that's normal to have subs at the same pts/ts, but i guess it makes sense [19:39] <ubitux> but isn't pos or something suppose to make a different in order? [20:09] <michaelni> all reordering code should maintain the order of packets with equal dts within one stream, if thats what you menats [20:18] <ubitux> ok [20:18] <ubitux> well i see no objection, but as i said i'm probably not the best person to review that kind of stuff [20:28] <ubitux> seems i'll have to wait for the next gcc version to use -fno-sanitize-recover with ubsan :( [20:29] <ubitux> it's almost useless now [21:46] <cone-56> ffmpeg.git 03Michael Niedermayer 07master:6b9e0bf42fb3: avfilter/vf_removelogo: fix memleak on failure [21:58] <cone-56> ffmpeg.git 03Reimar Döffinger 07master:97bcb0588916: Add missing "const" to pointer arrays. [22:05] <Daemon404> wm4, yes fuck eurovision [22:05] <Daemon404> gf is watching it by me [22:05] <Daemon404> kill me now. [22:05] <wm4> lol [22:06] <ubitux> as long as she is just watching and not listening everything is ok [22:06] <Daemon404> lol [22:06] <Daemon404> i dont get eurovision [22:06] <nevcairiel> ubitux: actually, i wouldn't agree with that [22:07] <Daemon404> seemingly normal europeans suddenly like what seems to be x factor [22:07] <Daemon404> but with N countries [22:07] <ubitux> nevcairiel: ah, visual are insane as well? [22:07] <nevcairiel> here, have this .. guy http://shrani.si/f/P/sK/2lp1q32o/3.png [22:07] <ubitux> ahah [22:08] <Daemon404> nevcairiel, russia and belaruse wanted him kicked off [22:08] <Daemon404> somthing about a wasteland of sodomy [22:08] <wm4> interesting [22:08] <wm4> I guess in some obscene way eurovision might be interesting for the "politics" [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:8713b8e3fb76: avformat/mux: Check for and remove invalid packet durations [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:3d5c38ca7d90: avformat/h263dec: Fix h263 probe [22:09] <iive> do russia and belaruse participate in eurovision? I thought they are not in EU. [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:e8411c3b6d92: swresample: fix AV_CH_LAYOUT_STEREO_DOWNMIX input [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:d2ceca6e8bf0: ffmpeg_filter: fix pointer to local outside scope [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:cb0d255e2fc5: avcodec/mjpegdec: Fix undefined shift [22:09] <Daemon404> iive, what has it got to do with EU? [22:09] <nevcairiel> its a european thing, not a EU thing really [22:09] <Daemon404> eurovision is much older than the EU [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:3af977d4d644: avfilter/graphdump: Fix pointer to local outside scope [22:09] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:8e8dc210cd5a: sws: dont use the optimized 410->420 unscaled conversion when height%4 [22:09] <nevcairiel> cant throw someone out of the continent :P [22:10] <iive> never heard of it before EU [22:10] <JEEB> it even has israel and turkey there [22:16] <cone-56> ffmpeg.git 03Michael Niedermayer 07release/1.1:03bda4468991: Update for 1.1.11 [23:15] <cone-56> ffmpeg.git 03Reimar Döffinger 07fatal: ambiguous argument 'refs/tags/n1.1.11': unknown revision or path not in the working tree. [23:15] <cone-56> Use '--' to separate paths from revisions [23:15] <cone-56> refs/tags/n1.1.11:HEAD: Add missing "const" to pointer arrays. [00:00] --- Sun May 11 2014

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ffmpeg.log.20140510
by burek 11 May '14

11 May '14

[00:18] <brimestone> hey guys.. im trying to write something in Cocoa for Desktop ( Non iOS ) and i wondering if there is a API for ffmpeg? [00:19] <sacarasc> The libav* stuff is the API. [00:19] <brimestone> or framework [00:19] <sacarasc> See the libraries documentation on http://ffmpeg.org/documentation.html [00:20] <sacarasc> There's a bunc hof them. [00:20] <brimestone> really, im been trying my lunch with AVFoundation, but very limited to ProRes and H264... [00:20] <sacarasc> And the API docs is a little below that. [00:22] <brimestone> so, basically, does this mean i can use ffmpeg without using NSTask? [00:23] <sacarasc> If the very brief description of what NSTask means what I think, yes. [00:23] <sacarasc> You don't need to run the ffmpeg binary yourself. [00:24] <brimestone> i know im starting to be a bug.. but where should i start reading? the Doxygen? [00:24] <sacarasc> Yeah. [00:25] <sacarasc> For whichever branch of ffmpeg you're using. [00:25] <sacarasc> I can't really help much, as I am only a user, not a coder. :D [00:26] <brimestone> so if i want to decode and encode.. ill look at libavcodec [00:35] <brimestone> Jesus! this is sooo way over my head [00:38] <rafael2k> brimestone: taka a look at ffplay.c source code, inside ffmpeg source. [00:44] <brimestone> yikes [00:44] <brimestone> "u1 += u;" yikes!!! [00:45] <brimestone> variable/object name short as 2 character [00:58] <rafael2k> brimestone: this is C code, usually coding style differs from Objective C... [01:06] <brimestone> rafael2k.. i took C back in Computer Engineering.. but that was more than 10 years ago [01:07] <brimestone> rafael2k: do you know how i can start writting OSX tools using the libavcodec? [01:17] <Bray90820> So While trying to covert a VOB to FLAC in ubuntu i get an error [01:17] <Bray90820> http://pastebin.com/raw.php?i=Kq6gaR9T [01:19] <c_14> Can you play the input file? [01:19] <c_14> Also, you do know it's pretty useless converting a lossy source to lossless, right? [01:27] <Bray90820> c_14: I can play it [01:28] <Mavrik> brimestone, look into doc/examples of ffmpeg source [01:29] <c_14> Does ffmpeg actually error out, or does it finish encoding? [01:32] <Blue_Knight> woof woof [01:33] <Bray90820> c_14: it finishes [01:33] <Blue_Knight> is dvb supported as input? [01:33] <c_14> Bray90820: Then as long as you can play the output it should be fine? [01:33] <Bray90820> But why am i getting that error [01:34] <sacarasc> Bray90820: It was a warning, not an error. [01:35] <c_14> The source isn't perfect. Shit happens. [01:35] <Bray90820> Ahh then nevermind [01:35] <Bray90820> Actually it was an error [01:35] <Bray90820> frame sync error [01:36] <sacarasc> If it didn't end the encode, it was a warning. [01:37] <Bray90820> ok [03:17] <eristisk> I made a recording of a lecture directly from a PA system and it has this annoying, constant buzz that I'd like to remove as much as possible. What can I use to do that? [03:37] <Hello71> audacity. [03:37] <Hello71> or possibly sox, but that isn't likely to end well [04:20] <jedir0x> does avpicture_fill copy the data - or just copy the pointers to the data? [10:06] <leggo> hello [10:06] <Ardonik> leggo: I usually do my framerates with -r <desired_ofps>. [10:07] <Ardonik> Keep in mind, now, that I'm not an expert! These guys are. [10:07] <leggo> (I'll repaste the options) [10:07] <leggo> -threads {threads} {offset} {realtime} -i "{infile}" -threads {threads} -f flv -vcodec libx264 -bufsize 1024k -b:v 2125k -bt 2250k {framerate} -map 0:0 -map 0:1 -vf "yadif, scale={scalex}:{scaley}" -preset {vpreset} -tune film -vprofile baseline -level 30 -acodec libmp3lame -ab 256k -ar 44100 -ac 2 -async 1 -y "{outfile}" [10:07] <leggo> this apparently sets the bitrate to 2560. how? I don't see 2560 anywhere in there. [10:08] <leggo> also the framerate doesn't appear to be set anywhere, but it doesn't seem to be 30 fps. [10:08] <Ardonik> -b:v is the output video bitrate (2125 kbits/s.) [10:09] <leggo> so that would be 2125 kbits/s, not 2560? [10:09] <Ardonik> Add that to -b:a (audio stream) and -b:s (subtitle stream) and you get the total output bitrate. [10:09] <leggo> ah [10:10] <Ardonik> I think you're running into a default framerate or something. Try -r 25 or something. [10:10] <Ardonik> Yes, I said "or something" twice. [10:10] <leggo> is deinterlace set in there somewhere? the video feels as if it just throwing away half the frames. [10:10] <Ardonik> Ugh...now you're getting into the deep magic. My ffmpeg interlace-fu is weak. [10:12] <leggo> what is {framerate} by the way? [10:12] <Ardonik> The one you have just sitting there? I dunno. [10:12] <Ardonik> It seems to be missing a flag. [10:14] <leggo> perhaps the program that runs ffmpeg fills that in with something [10:15] <Ardonik> Meh, just put a "-r" before that {framerate} and see what happens. [10:22] <leggo> ok that did make it appear smoother! [10:22] <Ardonik> Excellent! [10:28] <leggo> the program uses ffmpeg to reencode live tv and then stream it to tablet/phone. it looks great now [10:29] <Ardonik> What about the interlacing? [10:29] <leggo> I see no interlacing artifacts, so it must be doing it. [10:29] <Ardonik> Curious, but so much the better. [10:33] <leggo> thanks for help. I'll dwell more into ffmpeg one of these days, so maybe I could further tweak it. right now the bottle neck is my CPU, it is already at 100%, and if I increase the bitrate the video starts to buffer. [10:34] <Ardonik> leggo: Yeah, that usually happens when you do massive encodes. [10:35] <Ardonik> ffmpeg is one of the few modern processes that is actually CPU-bound these days! [10:36] <leggo> having ancient core 2 duo here, haven't upgraded in 7 years. [10:36] <Ardonik> Not that cane be done to improve the speed, then! (Except for sacrificing encoding quality, naturally.) [10:37] <Ardonik> And nicing the ffmpeg process will just make the encode slower. [10:37] <Ardonik> (Which you may still want to do in order to actually have a usable system while the encoding takes place.) [10:40] <leggo> yeah, I will see what I can do. [15:33] <Waddles> Hi there. I'm a bit stuck on something so I was wondering if anyone here could help. At the moment I'm working on an application that just reads a file from any old source, (hopefully) identifies the format and starts outputting PCM audio into a buffer I have. The problem I am having is that, av_read_frame seems to report AVERROR_EOF before the end of the file. [15:44] <Waddles> http://pastebin.com/WpedtGxr [16:35] <Waddles> oh, don't worry. I fixed it. I wasn't calculating the correct number of samples properly. Thankyou. [17:41] Action: haspor slaps jarainf around a bit with a large trout [17:41] <haspor> sorry, ignore that [17:41] Action: jarainf slaps haspor around a bit with a large trout [17:42] <haspor> ok we are even now [17:42] <Ardonik> Turnabout is fair play! [17:43] <haspor> who could help me abit with ffmpeg lib, i need to decode at3 format, but get bad output [17:44] <Ardonik> haspor: See the paste link in the topic. [17:44] <haspor> all right [17:47] <Ardonik> No, wait. The paste link isn't in the topic at all! [17:47] <Ardonik> http://ffmpeg.pastebin.com/ [17:48] <haspor> http://pastebin.com/gkDAa0Mh [17:49] <haspor> the output is playing too fast and it should be more than 4 seconds [17:49] <Ardonik> Oh, it's a library call. Not my area of expertise, alas. [17:49] <haspor> :( [17:55] <Aiena> I am on linux but I want to help a freind on windows. I am wondering which builds I should get fro windows from the Zeranoe page. Should I use the git version on the stable release versions from the site ? [18:01] <c_14> Unless you need some feature/bugfix that isn't in stable yet, it doesn't really matter. I usually go for git head though. [18:02] <Aiena> c_14 thanks [20:29] <allengreen> anybody? [20:30] <c_14> Just ask and if somebody can help you, they will. [20:32] <allengreen> ok, I just call ffmpeg function to decode mpeg2 file, and I got a lot of AVPackets, then I muxe the packets to mpeg2ts, the ts file can't played. [20:32] <allengreen> I believe the muxe code is correctly. [20:34] <allengreen> a es file to a ts file, then demux ts file, then muxe to ts file [20:34] <allengreen> the result file can't be displayed. [20:35] <c_14> I'm not very good with the ffmpeg libraries, you'll have to wait for someone who is. [00:00] --- Sun May 11 2014

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ffmpeg-devel.log.20140509
by burek 10 May '14

10 May '14

[00:15] <cone-159> ffmpeg.git 03plepere 07master:63832e01c3c7: hvcodec/x86/hevcdsp: make macros more modular to support functions that are not sse4 [02:11] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:6df33c4926c8: avcodec: include GET_RL_VLC() in trace output [02:11] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:e54c052beedb: avcodec/mpeg4videodec: print run/level/index values [02:43] <michaelni> BBB, Daemon404, do you have comments on "[FFmpeg-devel] [PATCH 2/2] configure: Use intel math.h header" ? (its about intel/ms compilers) [03:35] <Snowleaksange> sorry for irate comments earlier. ffmpeg is pretty sweet. was just frustated cuz i hit single deadend. but it didnt even take me long to implement libpng, libjpeg and bmp which is all i needed [04:13] <BBB> michaelni: I don't really know, admittedly kinda very hacky but I can't really easily come up with something clean [04:30] <michaelni> Snowleaksange, np, just if you find a fixable problem and have an idea on how we can fix it, dont hesitate to tell us [04:31] <michaelni> about image2, for multiple image support with %d or otherwise something capable to open multiple "things" is needed [04:34] <michaelni> BBB, so i should apply it ? or do you think it makes sense to wait (i agree wholeheartly that its ugly ...) [04:57] <michaelni> nevcairiel, it seems ffmpeg_dxva2.c doesnt build on or fate box and also isnt disabled by configure (see http://fate.ffmpeg.org/history.cgi?slot=x86_32-ubuntu-mingw32-gcc) [05:21] <cone-529> ffmpeg.git 03Michael Niedermayer 07master:341cacb9ac1b: avcodec/x86/hevcdsp_init: fix build failure with --disable-mmx [07:39] <ubitux> should i add an asan and ubsan fate instances? [07:42] <ubitux> mmh we don't have a tsan either? [11:09] <ubitux> mmh it's strange i have a clang/tsan instance but i don't see it on fate.ffmpeg.org [11:14] <ubitux> oh wow it's "locked" meh [11:36] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20140509093218&slot=x86_64-archlinux… [11:36] <ubitux> > libavcodec/mpegaudiodec_template.c:219:33: runtime error: left shift of negative value -1 [11:37] <ubitux> > libavcodec/golomb.h:75:13: runtime error: shift exponent -31 is negative [11:37] <ubitux> i added gcc-asan and gcc-tsan too, they're still running [11:38] <JEEB> nice [11:38] <wm4> indeed [11:42] <nevcairiel> michaelni: apparently that fate box has a broken dxva2api header. iirc classic mingw32 didn't ship that header at all, so it must be a third party install. if you have access to that box, can you send me the dxva2api.h, so i can check it out and build a test to check it? [11:51] <nevcairiel> (not that i consider classic mingw32 still to be relevant in the real world) [11:55] <JEEB> yeah, it has been very well made irrelevant [12:02] <wm4> wow these responses are so assholish [12:02] <wm4> apparently you can nest extern"C" blocks [12:03] <wm4> so it should actually be no problem to add them to the headers [12:03] <nevcairiel> the argument pro is equally stupid though, its not really anoying or stupid, its well documented that you need to, and thats that [12:04] <wm4> there's no technical reason to require every API user to do this, other than being assholes [12:22] <ubitux> ubsan/asan/tsan fate instances available [12:23] <ubitux> just like helgrind & drd, tsan detects has a shitload of problems, probably always the same issue [12:24] <ubitux> i hope nicolas' patch is applied soon [12:34] <michaelni> nevcairiel, http://pastebin.com/v770na2m (its debians mingw IIRC) [12:55] <BBB> tsan lol [12:56] <BBB> http://blogs.gnome.org/rbultje/2014/01/12/brute-force-thread-debugging/ ftw :-p [13:06] <j-b> oh god, that seems like a HACK [13:13] <BBB> hi j-b [13:17] <compn> we may be looking at different failures at the same time (as is demonstrated by the different outputs for the 2 shown failures). [13:17] <compn> nooo [13:21] <nevcairiel> michaelni: as i thought, the header is just broken, it uses wrong variable names in some defines .. guess that needs a test [13:21] <j-b> creating invalid files because an IE plugin does not support it... [13:23] <compn> you're just upset internet explorer is turning into a media player :P [13:24] <compn> who needs vlc, it plays right in the browser ;) [13:25] <j-b> compn: your trolls are less strong those days [13:25] <kierank> less of a troll and more of the truth =p [13:27] <BBB> Compn: well it was different failures caused by the same bug [13:28] <BBB> Compn: so the patch created from one test case fixed all of the bad md5s in that particular file [13:28] <compn> yes, luckily it wasnt multiple bugs :) [13:28] <BBB> that would've sucked, yes [13:28] <BBB> but other files showed different bugs [13:29] <compn> are we recording fate for random test failures like that ? [13:29] <BBB> not systematically [13:29] <compn> i mean does fate test 1233 fail once in 100 runs ? [13:29] <BBB> right, not systematically [13:29] <BBB> we probably should, esp. for multithreaded codecs and fate boxes, but we don't [13:30] <BBB> maybe someone wants to hack on fate and extend it to do this? [13:31] <BBB> also... someone broke fate comment strings for darwin [13:31] <BBB> "gcc 4.6.4 (MacPots gcc46 4.6.4_3+univesal)" [13:31] <BBB> MacPoRRRRts univeRRRRRsal? [13:31] <BBB> someone hates the r [13:31] <compn> maybe is broken key [13:32] <compn> on an old broken keyboard :P [13:33] <BBB> indeed [13:33] <BBB> is someone fixing hevc asm on windows? [13:44] <cone-85> ffmpeg.git 03Michael Niedermayer 07master:0be95996d0a0: avcodec/mpegaudiodec_template: make shift unsigned to avoid undefined behavior [13:44] <cone-85> ffmpeg.git 03Michael Niedermayer 07master:2201d1a0f8c5: avcodec/hevc: Fix undefined shifts [14:02] <ubitux> mmh [14:02] <ubitux> 56ee3f9de7b9f6090d599a27d33a392890a2f7b8 caused a regression it seems [14:04] <ubitux> btw, didn't we have some kind of ffbisect at some point? [14:04] <nevcairiel> should still exist [14:04] <Daemon404> tools/ ? [14:04] <nevcairiel> you need to use tools/bisect-create to create the script [14:05] <Daemon404> ah yea [14:05] <ubitux> ah that's because i'm in a bisect branch obviously [14:05] <ubitux> in the other one state [14:05] <nevcairiel> because otherwise it would overwrite itself [14:08] <ubitux> yeah well i confirm that's the regression, meh [14:08] <ubitux> i have some trouble with a flv because of this [14:12] <ubitux> without -ss 0: http://pastie.org/pastes/9159067/text [14:12] <ubitux> with -ss 0: http://pastie.org/pastes/9159069/text [14:13] <ubitux> if -ss 0 is supposed to drop the negative ts, i don't see how that applies here [14:13] <nevcairiel> hrm how do I debug a h264 clip that outputs no frames unless i set the showall flag? debug log outputs not much :( [14:13] <ubitux> it looks like it's picking a random ts [14:14] <ubitux> and obviously it's not a kf so the output is garbled [14:18] <Voicu> hello, how do I use a filter programmatically? [14:18] <ubitux> Voicu: see doc/examples/filtering_video.c [14:19] <Voicu> thanks [14:26] <Voicu> hmm, I'm not sure [14:26] <Voicu> I need to use the aac_adtstoasc filter [14:26] <Voicu> for converting an AAC stream [14:26] <Daemon404> thats a bitstream filter [14:27] <Daemon404> i.e. not libavfilter [14:27] <Voicu> yeah my bad [14:27] <Daemon404> also careful, that particular filter isnt perfect [14:27] <Voicu> OK, so I have the bitstream initialized, how do I use it? [14:27] <Daemon404> (i dont think its possible to be) [14:27] <Daemon404> i actually dont know -- never had to [14:27] <Voicu> well ye, different fields I suppose [14:27] <Daemon404> nevcairiel maybe [14:28] <ubitux> gcc: error: unrecognized command line option -fno-sanitize-recover [14:28] <ubitux> :( [14:28] <nevcairiel> i never used bitstream filters [14:28] <Daemon404> o ok [14:40] <nevcairiel> bitstream filters are more for remuxing, not required for decoding [14:40] <nevcairiel> so.. never touched the stuff [14:44] <cone-85> ffmpeg.git 03Hendrik Leppkes 07master:bc47801968fc: configure: check for recent dxva2api headers with fixed COBJMACROS defines [14:50] <Voicu> nevcairiel, I am remuxing [14:50] <Voicu> i.e. I have 2 streams, one h264, one aac and I am streaming them to a rtmp server in flv format [14:51] <Voicu> but I did, I just had to use av_bitstream_filter_filter [15:11] <nevcairiel> apparently streams without IDRs and without recovery SEIs don't play at all without the showall flag, that can't be as desired .. it shows absolutely corruption free, too. Just no IDRs, plenty I frames though. [15:21] <ubitux> libav receiving patches to fix things already fixed in ffmpeg makes me really sad for the time wasted by these non aware contributors... [15:22] <ubitux> (https://lists.libav.org/pipermail/libav-devel/2014-May/059501.html - http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=5db49fc38d9132e134de92…) [15:34] <cone-85> ffmpeg.git 03Diego Elio 'Flameeyes' Pettenò 07master:a7b554f86399: rtpdec: make the NTP time values unsigned. [15:37] <ubitux> patch from 2010? are you time traveling michaelni? :) [15:39] Action: kshishkov puts a nice FFmpeg sticker on his laptop [15:39] <kshishkov> feel free to use that exact quote [15:40] <ubitux> ahaha [15:44] <nevcairiel> ubitux: the interesting part about that patch is that its only half the fix, there is a follow up commit with more corrections [15:44] <ubitux> the guy might be still debugging [15:49] <ubitux> does anyone have a sample with negative ts? [15:50] <ubitux> (at the beginning) [15:50] <ubitux> in flv if possible [15:50] <ubitux> or any idea how to generate one [16:05] <Voicu> can you just demux and remus a file and alter the .pts of the packets? [16:05] <Voicu> *remux [16:14] <ubitux> ok so it seems the initial seek fails in my case (-ss 0 with a flv) [16:15] <ubitux> but it changes something in the context [16:15] <ubitux> which makes everything else crumble [16:15] <ubitux> ff_read_frame_flush() is the responsible [16:18] <ubitux> which is called just before s->iformat->read_seek() in seek_frame_internal() [16:20] <ubitux> flush_packet_queue() to be more specific [16:27] <ubitux> why is this code called before seek? [16:27] <ubitux> shouldn't it be called after the seek if it succeed instead? [16:27] <DHE> I have a patch I'm considering submitting to ffmpeg. It takes A53 subtitles (currently collected by the mpeg1/2 decoder) and puts them into libx264 output. But it modifies the mpeg1/2 collector a bit in a way that might not sit well with others. [16:29] <J_Darnley> What do you mean by "puts them into libx264 output"? Anyway, you can ask for comments here or send a patch to the devel mailing list and ask for comments there. [16:30] <DHE> J_Darnley: I have a Set Top Box for IPTV. It can show closed captions from a feed transcoded from MPEG2 to H264 using ffmpeg now [16:32] <DHE> http://www.dehacked.net/ffmpeg-a53-to-libx264.txt This is the patch (git format-patch output) [16:33] <compn> you force mpeg subs into side data and pass it to x264 eh [16:33] <DHE> mpeg12dec.c is already collecting it. I just modified it to grab full headers (some metadata was previously uncollected and I am not sure I can reliably regenerate it right now) [16:34] <DHE> but it's working. I have it running on my desk right now. [16:34] <DHE> Haven't run regression suites on it yet [16:34] <compn> ah [16:34] <J_Darnley> oh yes, I forgot x264 had that feature [16:35] <ubitux> no one familiar with the seek code? :( [16:35] <ubitux> fate doesn't seem to like to have the flushing post seek [16:36] <DHE> so that's my patch [16:36] <DHE> I can see a white-space error in libx264.c but otherwise... [16:37] <compn> i think michael is mpeg maintainer , wait for him [16:50] <JEEB> gcc sure is "quick" in backporting fixes [16:51] <wm4> it's a GNU project after all [16:51] <JEEB> commits from either 23rd or 28th that fix the golomb optimization derp [16:51] <JEEB> are not yet in [16:51] <JEEB> (in the 4.9 branch) [16:52] <JEEB> wm4, that is true indeed [16:53] <michaelni> DHE, doesnt this break the ABI of libavutil, the frame side data format is defined in it [16:59] <DHE> it does? I checked and all I found (in ffmpeg itself) using it was a diagnostic that just reports its existance [17:00] <DHE> there's 1 or 2 bytes of metadata I was having trouble regenerating from the existing code. Maybe I just didn't try hard enough.../ [17:03] <DHE> well, back to the drawing board then [17:04] <ubitux> why is flac_read_timestamp() marked as av_unused? [17:04] <ubitux> it looks very used to me [17:04] <wm4> maybe it's not used under all configurations [17:05] <ubitux> doesn't seem so [17:08] <Daemon404> blame? [17:09] <ubitux> first commit introducing the function (it's recent) [17:10] <Daemon404> o [17:14] <Voicu> is there a tutorial or something that explains how presentation timestamps and timebases work? [17:15] <Voicu> also, is it normal for every AAC packet to have a duration of 655360 ? [17:15] <Voicu> the numbers don't add up in the end [17:16] <Daemon404> Voicu, pts * timebase = time in ms (in ffmpeg's case) [17:18] <Voicu> hmm, it still doesn't add up [17:18] <Voicu> I have ~11s of video and I get 30s when doing pts * timebase [17:19] <Daemon404> i get the feeling youre pulling the wrong number from somewhere [17:20] <Voicu> well I have an AAC stream that ffmpeg reads [17:20] <Voicu> the pts values I get go up to 30s worth [17:20] <ubitux> funny, ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0); doesn't work (and breaks everything) while ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, INT64_MAX, 0); works fine [17:21] <ubitux> i guess that's because the seek succeed in the second case [17:25] <wm4> ubitux: I'm simply using the old seek API [17:25] <wm4> except when it doesn't work [17:25] <ubitux> it's the same thing [17:25] <wm4> then I revert to the new one [17:26] <ubitux> the problem is deeper [17:26] <ubitux> but i'm pretty confused at the internals of the seeking code [17:26] <Daemon404> Voicu, get from where [17:27] <Daemon404> ubitux, pre-indexing master race [17:27] <Voicu> Daemon404, the packets I was reading from the stream [17:27] <Voicu> Daemon404, but now I almost got it [17:27] <Voicu> I fixed the timebases and I get timestamps which are roughly /2 [17:27] <Voicu> so, I missed something simple [17:29] <wm4> optional preindexing in libavformat would be nice [17:30] <Daemon404> i dont think it would "fit" in very easily [17:30] <Daemon404> api wise [17:31] <wm4> I don't really see a problem [17:31] <wm4> but it'll make utils.c uglier [18:14] <cone-85> ffmpeg.git 03nu774 07master:9880a0d4b131: pcm-dvd: Fix 20bit decoding [18:14] <cone-85> ffmpeg.git 03Michael Niedermayer 07master:8a21613821fd: Merge commit '9880a0d4b131ef36694d62f78060350a81f08b80' [18:15] <kierank> who is at linuxtag? [18:35] <cone-85> ffmpeg.git 03Matt Oliver 07master:3554c2fafc46: libmpcodecs: Fix compilation due to missing static in suncc. [19:11] <michaelni> ubitux, is "[FFmpeg-devel] [PATCH] avformat/mux: 2 subtitle packets could have the same DTS" ok ? [19:12] <michaelni> its just a 1 line patch [19:45] <DHE> I updated my A53 CC patch. New version at: http://www.dehacked.net/ffmpeg-a53-to-libx264.txt [19:45] <DHE> (old version intentionally overwritten - it had a memory error) [19:46] <DHE> I think I can drop some of the last few bytes from the extra sei data, still experimenting. [19:51] <Daemon404> it should probably be a private option [19:51] <Daemon404> it doesnt seem wise to hard enable it [20:06] <cone-85> ffmpeg.git 03Michael Niedermayer 07master:398e3a591fb8: avcodec: replace uses of deprecated avcodec_set_dimensions() [20:09] <DHE> Daemon404: makes sense [20:38] <DHE> Refreshed the patch, added some comments, removed some bytes from the stream output (WorksForMe) and made it an option that defaults to off [22:14] <cone-85> ffmpeg.git 03Matthew Lindner 07master:b372f673427c: avcodec: better level/index printing [00:00] --- Sat May 10 2014

1 0

ffmpeg.log.20140509
by burek 10 May '14

10 May '14

[00:06] <kolizer> a -ss use? c_14 [00:11] <c_14> -ss is usually only helpful when you don't want part of the input, not when the a/v is out of sync [00:13] <kolizer> Show an example of how to use -itsoffset ? c_14 [00:15] <c_14> ffmpeg -i audio.mp3 -itsoffset 25 -i video.mp4 output.mp4 [05:44] <allengreen> I use "ffmpeg -i input_file.mp4 -vcodec copy -an output_file.mp4" to remove audio track from mp4, but that does not keep the frame pts, anybody know how to do this? [05:54] <Bray90820> So i seem to be getting a strange error while converting a vob to flac on ubuntu [05:54] <Bray90820> http://pastebin.com/w9ySbLZH [05:55] <Bray90820> Here is the raw text which is easier to read [05:56] <Bray90820> http://pastebin.com/raw.php?i=w9ySbLZH [10:24] <anshul> !log [11:33] <kolizer> Picture with a web cam behind the video screen http://pastebin.com/SRUqapCP [11:35] <kolizer> how to fix? [11:46] <kolizer> help [11:48] <xagaba> I'm using FFplay version SVN-r0.5.10-4:0.5.10-1 that states --enable-vdpau in configuration. How I can tell ffplay to use vdpau for playback ? [11:49] <JEEB> it's not implemented in ffmpeg or ffplay (esp. at such an old version), not sure if supports it even now [11:49] <JEEB> also man, that is an old version :D [11:51] <xagaba> it's squeeze version.. [11:52] <xagaba> but the configuration shows that vdapu it's enabled .... [11:52] <JEEB> yes, it probably has some kind of vdpau support in libavcodec [11:52] <JEEB> that does NOT mean it can be just used from ffmpeg or ffplay [11:53] <JEEB> hw accels generally need a framework around them [11:53] <xagaba> using squeeze mplayer i can use vdpau, don't mplayer use ffmpeg libs ? [11:54] <JEEB> yes, they use the hwaccel, so they have the surrounding code there [11:54] <JEEB> ffmpeg and ffplay haven't had it before late 2013 [11:54] <JEEB> or at least ffmpeg now seems to have VDPAU hwaccel support [11:55] <xagaba> with an actual ffplay version it's vdpau supported ? [11:56] <JEEB> not that I can see [11:56] <JEEB> I can see support added in ffmpeg [11:56] <JEEB> no mention of ffplay [11:57] <xagaba> ok thanks for clarifications JEEB ;-) [11:57] <JEEB> np [11:57] <JEEB> if you want playback btw, you might want to try out the mpv-build repository so you can build your own up-to-date mpv with newer components :) [11:58] <JEEB> https://github.com/mpv-player/mpv-build [11:58] <kolizer> Picture with a web cam behind the video screen http://pastebin.com/SRUqapCP [11:58] <kolizer> how to fix? [12:39] <fridus> Hi [12:39] <fridus> I've problem with build on mac os x [12:40] <fridus> Undefined symbols for architecture x86_64: [12:40] <fridus> "_ff_h263p_decoder", referenced from: [12:40] <fridus> _avcodec_register_all in allcodecs.o [12:40] <fridus> could you help me ? :S [14:15] <fridus> Hi [14:16] <Voicu> hello, how do I use a filter programmatically? [14:27] <fridus> i don't know [14:28] <fridus> I've a question too [14:44] <klaxa> fridus: just state your problem as precise as possible [15:04] <Voicu> maybe it's not translatable into human language [15:05] <Voicu> like that feeling you get when you talk about something deep and it seems like you're close to understanding something great but you can't just express it [15:23] <fridus> sorry [15:23] <fridus> I try to compile ffmpeg on mac os x [15:24] <fridus> i would like make a shared lib ( .so) [15:24] <fridus> But when it build, I see this error: [15:24] <fridus> Undefined symbols for architecture x86_64 [15:25] <fridus> "_ff_h263p_decoder", referenced from: [15:25] <fridus> _avcodec_register_all in allcodecs.o [15:25] <fridus> "_ff_h264_set_erpic", referenced from: [15:25] <fridus> _ff_h264_decode_slice_header in h264_slice.o [15:26] <fridus> "_ff_mpegvideo_decoder", referenced from: [15:26] <fridus> _avcodec_register_all in allcodecs.o [15:26] <Voicu> lol [15:27] <Voicu> fridus, don't paste in here [15:27] <fridus> where ? [15:27] <Voicu> fridus, this is true for all support channels - you use a pastebin like pastebin.com [15:27] <Voicu> so you don't clutter the chat with random code [15:28] <fridus> oh .. [15:29] <fridus> No issues on github ? [15:29] <Voicu> fridus, hm? [15:30] <fridus> I don't know where and how send my problem/bug to the support [15:32] <c_14> https://ffmpeg.org/bugreports.html [15:32] <c_14> But first, have you tried: https://trac.ffmpeg.org/wiki/CompilationGuide/MacOSX [15:35] <fridus> thanks [17:13] <blz> Hello, I have a collection of frames taken from a webcam (png images) that I would like to store as a video file. Can such a thing be achieved with ffmpeg? [17:13] <c_14> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images [17:18] <blz> c_14 that seems promising indeed! Thanks! [18:41] <Phlarp> I have a question: Why does the apt-get package for ffmpeg install V 0.8.10? [18:44] <c_14> The ffmpeg that is installed under debian and debian-derivatives is actually libav, and usually a rather outdated version thereof. [18:45] <reventlov> Hello. [18:47] <c_14> Hi, if you have a question just ask and if someone can help you, they will. [18:49] <reventlov> I'm using ffserver on archlinux. When a client disconnect from ffserver, it segfaults. [18:49] <reventlov> Version: http://sprunge.us/AdPF [18:49] <reventlov> Is this a bug known ? [18:50] <c_14> Do you have the ffserver output? [18:51] <reventlov> Fri May 9 18:50:56 2014 127.0.0.1 - - [POST] "/rtsp.ffm HTTP/1.1" 200 1138688 [18:51] <reventlov> zsh: segmentation fault (core dumped) ffserver -f /etc/ffserver.conf [18:51] <reventlov> using : ffmpeg -i *.mp4 -override_ffserver -vcodec mpeg2video -b:v 3000 http://127.0.0.1:8090/rtsp.ffm [18:51] <reventlov> ffserver.conf: http://sprunge.us/WMRY [18:51] <xreal> Is there something like re-quantizising for x264 ? [18:52] <reventlov> it's not automatic, but I can reproduce it by opening/closing ffmpeg and/or one client [18:52] <reventlov> (like i just did) [18:53] <JEEB> xreal, re-quantization? please explain [18:53] <reventlov> I have one strace, if you want [18:54] <xreal> JEEB: In the past, it was possible to re-quantizize a 9 GB DVD-video to fit in a DVD-R with little quality loss. [18:54] <c_14> reventlov: You might want to submit a bugreport: https://ffmpeg.org/bugreports.html [18:54] <reventlov> c_14: ok. [18:55] <JEEB> xreal, ok -- so either just straightforward re-encoding or abusing the fact that MPEG-2 and DVDs were structurally so simple [18:55] <xreal> JEEB: http://en.wikipedia.org/wiki/DVD_Shrink [18:56] <JEEB> yeah, I know that piece of software [18:56] <JEEB> anyways, it's just re-encoding and (possibly) abusing the fact of how simple MPEG-2 was as a format + how limited DVD Video is [18:57] <JEEB> depending on your needs and how well the original file you're trying to compress is already compressed, you may be able to get some kind of results from re-encoding something [18:57] <JEEB> (as the case is always) [18:57] <kolizer> i overlay webcam on video and when i make screencast webcam no work. what problem? [18:57] <JEEB> also, x264 is just an encoder, H.264 is the format [18:58] <JEEB> (or AVC) [18:58] <JEEB> so yeah, one can use the CRF rate control and the usual ways of encoding with libx264 to re-encode something [18:58] <JEEB> I will have to say, though, that in many cases stuff encoded in H.264 already is quite compressed. Unless your source is a blu-ray or something. [18:59] <xreal> JEEB: But re-quantizition (I think, that was the term used) was very, very quick [19:00] <JEEB> I have no idea what that marketing term actually technically means [19:00] <kolizer> Somebody help me? [19:00] <JEEB> also as I said already, depending on what exactly that feature did [19:00] <JEEB> it might have abused features of the MPEG-2 Video format, or the fact of how simple/limited DVD Video has to be [19:01] <JEEB> also libx264 can be as fast or slow as you want, but the efficiency will of course vary [19:02] <kolizer> i overlay webcam on video and when i make screencast webcam no work. what problem? [19:04] <rafael2k> people, any knows if there are any free software MMT (MPEG 23008-1) out there? [19:05] <JEEB> I tried to read through the spec and I didn't understand what exactly it was trying to be :P [19:05] <JEEB> it's just some random crap on top of either MP4 or MPEG-TS? [19:05] <rafael2k> MPEG-TS substitute [19:05] <rafael2k> no [19:05] <JEEB> doesn't seem to be a substitute [19:05] <rafael2k> it's the evolution of MPEG-TS [19:05] <JEEB> at least looking at the spec [19:05] <rafael2k> it's much easier than MPEG2-TS [19:06] <JEEB> have you read the spec? [19:06] <JEEB> it's short as hell [19:06] <rafael2k> don't carry that craps of legacy ATM stuff [19:06] <JEEB> it doesn't specify a new format [19:06] <rafael2k> I have the 2nd working draft [19:06] <JEEB> I think I have the DIS or so [19:07] <Phlarp> So, I have compiled / installed the 2.2.2 build and the commands I'm trying to use work when executed on the command prompt, but fail when called by shell_exec(). error log says sh: 1: ffmpeg: not found. Anyone know how to make this work? [19:08] <c_14> ffmpeg isn't in your path, either use the absolute path to the binary or put it in your path [19:08] <Phlarp> how do I put it in my path? [19:10] <klaxa> export PATH=$PATH:/path/to/dir/containing/ffmpeg [19:10] <klaxa> iirc [19:11] <c_14> That'll only work for the current shell though. [19:11] <klaxa> so if it is in /home/user/build/ffmpeg/ffmpeg add /home/user/build/ffmpeg to your PATH [19:11] <klaxa> yeah put it in your .profile or .bashrc or whatever [19:11] <klaxa> or copy/link/symlink ffmpeg to /usr/local/bin/ or see if make install installs it in the system [19:12] <Phlarp> I need to make it executable by apache. [19:12] <Phlarp> so I need to copy/symlink ffmpeg to apaches? /usr/local/bin? [19:13] <reventlov> Reading symbols from ffserver...(no debugging symbols found)...done. [19:13] <reventlov> is that a problem for a bugreport ? [19:13] <klaxa> Phlarp: did you try running "make install" or "sudo make install" ? [19:13] <reventlov> (there is no ffserver debug bin) [19:13] <kolizer> i overlay webcam on video and when i make screencast webcam no work. what problem? [19:13] <klaxa> that should install it in /usr/local/ or whatever directory you specified with --prefix= during ./configure [19:14] <Phlarp> Yes, I have it made and installed already; it works fine when invoked from a commandline but not when triggered by a webpage script [19:14] <c_14> reventlov: You need to compile ffmpeg/server with debug symbols enabled/without strip. [19:14] <reventlov> ok, let's go. [19:14] <reventlov> without --disable-debug [19:15] <reventlov> nothing related to "strip" in the options I pass. [19:15] <reventlov> --disable-stripping so. [19:16] <reventlov> that's it ? [19:16] <klaxa> Phlarp: then you need to either specify the full path in the script (i.e. /usr/local/bin/ffmpeg) or add /usr/local/bin to that... "shell"'s path [19:16] <klaxa> using the full path might be easier [19:16] <JEEB> rafael2k, I went through the spec once again and I see MPUs ("An MPU shall be encapsulated as a single ISOBMFF file") and something called MMT payload (?) [19:17] <c_14> reventlov: think so [19:17] <reventlov> Do i need --enable-memalign-hack emulate memalign, interferes with memory debuggers ? [19:17] <c_14> don't think so... [19:18] <JEEB> and actually.. looking forward in this, an awful lot of the stuff is just putting stuff into ISOBMFF [19:18] <JEEB> I don't see a "replacement for MPEG-TS" here at all [19:19] <rafael2k> no? [19:19] <rafael2k> how? [19:20] <rafael2k> there is packet, MPUs, system clock, FEC... [19:20] <JEEB> yeah. I can see those packets [19:21] <JEEB> this seems more like a (network) protocol than what MPEG-2 Systems is [19:21] <rafael2k> the packet can have arbitrary size, not hardcodec 188bytes, what I didn't understand is the use of HTML5 in the transport standard... [19:21] <JEEB> yeah, this thing seems to be quite overcomplicated :P [19:22] <JEEB> (also whatever you sent me sure doesn't seem to have gone anywhere) [19:22] <rafael2k> no no, indeed, all the clock stuff is much easier [19:22] <rafael2k> it uses NTP clock syntan [19:22] <rafael2k> also the fragmentation and buffer model seems easier [19:22] <rafael2k> (I just passed the eyes in the standard) [19:24] <rafael2k> but they mixed the presentation definition in it, seems not very correct [19:25] <JEEB> yeah, it's a rather messy spec :P [19:25] <rafael2k> true [19:25] <JEEB> first they give you some crap that is supposed to go into an ISOBMFF file [19:25] <rafael2k> I'll try to get the final draft to check if things got better [19:25] <JEEB> then they give you something more [19:25] <JEEB> and then there's the MMT Protocol [19:25] <rafael2k> the ISOBMFF AU goes inside the MPUs, as I could understand [19:27] <JEEB> but yeah, the spec is a mess and it truly doesn't make me feel like I want to touch it at all [19:28] <rafael2k> eheheheh [19:28] <rafael2k> I'll try to play a little bit [19:28] <rafael2k> <- now working w/ hybrid broadcast [19:29] <JEEB> if you can explain me later WTF this thing is supposed to actually be, I'd be rather thankful. Because so far it just seems like a mess [19:30] <JEEB> it's not simply a container, it's some kind of a hackjob to also communicate between a client and a server or whatever [19:33] <JEEB> ahaha [19:33] <JEEB> jesus christ [19:33] <JEEB> http://mpeg.chiariglione.org/standards/mpeg-h/mpeg-media-transport/use-case… [19:33] <JEEB> I checked out this document [19:33] <JEEB> first two Use Cases: Ultra-HD Content, 3D Video Content [19:34] <JEEB> THOSE ARE NOT CONTAINER SPECIFIC THINGS FOR FUCK'S SAKE [19:34] <JEEB> > Interactive 3D content [19:34] <JEEB> ok, this makes sense, but... this smells like the most unused parts of MPEG-4 [19:35] <JEEB> and the most convoluted etc as well [19:36] <JEEB> and a lot of this crap is something that IMHO is application-side stuff [19:36] <JEEB> not _container_ side stuff [19:36] <JEEB> that said, it already defines a goddamn protocol [19:36] <rafael2k> JEEB: I'm discussing here in the lab, it uses ISOBMFF [19:36] <JEEB> why does this remind me of the crappiest parts of MPEG-4? [19:36] <JEEB> yes, that's what I gathered so far [19:36] <JEEB> crap on top of ISOBMFF [19:36] <rafael2k> so, lots of redundancy in MPEG-2 (lots of tables and so on) were removed [19:37] <rafael2k> btw, I don't like MPEG-4 file format also.. [19:37] <reventlov> Anyone on archlinux could compile ffmpeg with the debug flags for x86_64 ? Ty. [19:37] <JEEB> well, at least the MPEG-4 file format was a container [19:37] <JEEB> and not some... [19:37] <rafael2k> the hierarquical FEC is nice [19:37] <JEEB> I don't even know what the flying fuck this is so far :D [19:37] <reventlov> c_14: i'm using archlinux, and it seems there is a bug related to gcc 4.9. Is this a problem for my bugreport ? [19:38] <reventlov> Oh, it's fixed. [19:38] <reventlov> Nevermind. [19:39] <rafael2k> for broadcast multiple layers for protection is nice [19:39] <JEEB> but yeah, I probably wouldn't call this a new container :P [19:39] <rafael2k> for example, when using SVC, and can protect the base video stream with a more robust FEC, and the upscale stream in a less robust FEC level [19:40] <JEEB> > someone actually using SVC [19:40] <JEEB> wow.jpg [19:40] <rafael2k> :P [19:40] <rafael2k> JEEB, I'm only doing research, not real use yet... [19:40] <JEEB> well, yeah [19:40] <rafael2k> the idea is to use this in the next generation of hybrid broadcast systems [19:41] <JEEB> the whole MMT thing sounds like something reasearchers came up with [19:41] <rafael2k> (I'm doing this for Brazil) [19:41] <JEEB> with all due respect to researchers [19:41] <JEEB> just like certain very unused things in MPEG-4 [19:41] <rafael2k> true [19:41] <rafael2k> like BIFS [19:41] <JEEB> oh yes [19:41] <JEEB> yes [19:41] <rafael2k> and many other parts of MPEG-4 that are _very_ complicated [19:41] <JEEB> "let's not have a thing for chapters, instead people can implement them in BIFS!" [19:42] <rafael2k> btw, there is no Brazilian in MPEG board... [19:42] <rafael2k> :P [19:42] <JEEB> and then no-one implemented them in BIFS because Nero had one hack, and others just derped with the MOV way [19:42] <JEEB> (and the MOV CHAP atom is now actually noted as something you shouldn't use) [19:43] <rafael2k> the new standards from mpeg I'm interested in understanding better is the xHE-AAC one, the HEVC and the 3-D audio one [19:43] <JEEB> HEVC is nice [19:43] <JEEB> simplified AVC with features for extra compression [19:44] <JEEB> no PAFF/MBAFF \o/ [19:44] <rafael2k> good! [19:44] <JEEB> the 3-D Audio one is not around at all yet [19:44] <rafael2k> how can you do interlaced content? [19:44] <JEEB> rafael2k, normal coding modes and just in the parameter sets you note that this is a field [19:44] <rafael2k> good [19:45] <rafael2k> 3-D audio is nice, I'm extending the Ginga middleware to add 3-D audio, not that complicated [19:45] <JEEB> last I checked the audio part of MPEG-H wasn't yet done [19:45] <rafael2k> that is the good part [19:46] <rafael2k> I'm not doing my work based in MPEG-G [19:46] <rafael2k> MPEG-H part 3 [19:46] <rafael2k> ; ) [19:46] <JEEB> heh [19:46] <rafael2k> KISS aproach [19:46] <JEEB> but yeah, so far my opinions on MPEG-H vary from "this is pretty good" (Part 2) to WTF (Part 1) [19:46] <JEEB> and other parts aren't done yet [19:46] <JEEB> :D [19:46] <rafael2k> good to know part 2 is good [19:47] <rafael2k> and that SVC and MVC is compatible with HEVC also? [19:47] <JEEB> well, good on the ISO/IEC and ITU-T level [19:47] <JEEB> scalable extensions are probably gonna get in this year or so [19:47] <JEEB> range extensions also [19:47] <JEEB> range extensions will have actually useful stuff [19:47] <JEEB> such as 4:2:2/4:4:4 and >10bit [19:47] <JEEB> scalable extensions has some random shit corps came up with [19:48] <rafael2k> I'm very interested in SVC [19:48] <JEEB> it looks cool on paper [19:48] <JEEB> but in the end it doesn't get any usage if at all [19:48] <JEEB> I know like a couple of things that use SVC and that's it [19:48] <rafael2k> it is important for backward compatibility, at least in theory, with millions are already sold TV sets... [19:49] <JEEB> and in reality you just switch streams :P [19:49] <rafael2k> but this is the web, the broadcast enviroment is much more restrictive [19:49] <JEEB> also yeah, seems like 3-D Audio has only started http://itscj.ipsj.or.jp/sc29/29w42911.htm#MPEG-H [19:50] <rafael2k> true, this is why is important to come up fast with a good 3D audio standard before MPEG [19:50] <rafael2k> mainly fraunhofer is doing it... [19:51] <JEEB> also regarding HEVC, http://www.f265.org/f265/static/txt/h265_companion.html [19:51] <JEEB> you can smirk at the rant :P [19:53] <rafael2k> tks! [19:53] <rafael2k> ; ))) [19:54] <JEEB> but yeah, you'll get more scalable shit into HEVC this year [19:55] <rafael2k> good [19:55] <rafael2k> it's not shit! [19:55] <rafael2k> :P [19:56] <JEEB> but yeah, all of the first three amendments are now past DIS, only FDIS left [19:59] <reventlov> ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers [19:59] <reventlov> is this the latest developpment version ? [20:00] <reventlov> Because i made all my logs with that :') [20:00] <JEEB> no [20:00] <JEEB> that's the latest release [20:00] <reventlov> fuck. [20:00] <JEEB> latest development version means current git master HEAD [20:00] <reventlov> so, i can just throw all my logs, i guess. [20:01] <reventlov> and i must recompile ffmpeg. [20:01] <reventlov> -_- [20:01] <iive> it is 4-5 days old release [20:01] <JEEB> based on an older branch :) [20:01] <JEEB> how new/old a release is, is kind of deceptive [20:02] <reventlov> so, do i *have* to recompile using git or not ? [20:02] <JEEB> if you want to check a checkbox that says "I've tested with the latest development version" then yes, otherwise just note that you've only tested with the latest 2.2 branch release [20:04] <reventlov> 2.2.2 is not listed on trac, btw [20:04] <JEEB> probably no-one just input it :) [20:04] <iive> cut from master on 2013.03.01 yeh, old branch. [20:05] <JEEB> like the topic of this channel, which still has 2.2.1 as the latest release ;) [20:05] <iive> reventlov: btw, gcc 4.9 is known to miscompile ffmpeg [20:06] <iive> http://gcc.gnu.org/bugzilla/show_bug.cgi?id=60902 [20:06] <JEEB> that of course only matters if you built the ffmpeg with 4.9.0 GCC [20:06] <reventlov> iive: fixed with the last gcc package here [20:06] <reventlov> https://bugs.archlinux.org/task/40256?string=ffmpeg&project=1&type[0]=&sev[… [20:07] <JEEB> I guess they did the same as I did with my test mingw-w64 toolchain, just took current 4.9 branch and stuck those two commits on top [20:10] <reventlov> « I have files to attach to this ticket » [20:10] <reventlov> where do i put these files ? [20:13] <reventlov> https://trac.ffmpeg.org/ticket/3630 [20:14] <reventlov> ok, done. [20:15] <JEEB> nice. also just a note for the future: generally the importance of tickets is only set by developers, so in the future you could just leave it to "normal" or whatever it was :) [20:15] <reventlov> ok, sorry. [20:43] <rafael2k> I don't know if this is of interest, but I'll be maintaining a debian 8/ubuntu 14.04 repo of ffmpeg with 'ffmpeg' library suffix here: www.telemidia.puc-rio.br/~rafaeldiniz/ginga4linux/debian/amd64/ [20:44] <rafael2k> I really don't understand why debian and ubuntu keeps using libav... [20:46] <JEEB> when libav forked off it kind of seemed promising, and the debian/ubuntu ffmpeg maintainer is a libav developer [20:47] <rafael2k> wtf [20:47] <rafael2k> any hope things will going to change? [20:47] <JEEB> basically, both ffmpeg and libav are better than ffmpeg-old [20:48] <JEEB> I'm not seeing it changing in general [20:48] <rafael2k> and how about make ffmpeg and libav not conflit in terms of library names? [20:49] <rafael2k> so then both could make it's way in debian/ubuntu repos [20:50] <JEEB> probably not going to happen either [20:50] <rafael2k> :/ [20:50] <rafael2k> So I'll keep shouting to keep ffmpeg instead of libav [20:50] <rafael2k> :P [20:53] <rafael2k> btw, library suffix solved my issue with the software I'm developing, but I need to maintain ffmpeg, but not problem.. next thing is to make static linking work here... [23:25] <xreal> JEEB: I figured out about DVDshrink. It doesn't need to do calculation of the motion vectors and other stuff. So transcoding is much faster. [23:25] <xreal> I think, this won't work for x264, does it? [23:25] <JEEB> you basically have to poke the encoder's internals inside out [23:25] <JEEB> for that [23:26] <JEEB> also it's not always something that works [23:26] <xreal> JEEB: I think, some video cutting applications can do this. [23:26] <xreal> JEEB: Smart Transcoding [23:26] <JEEB> that's another thing [23:26] <JEEB> that usually means that they let you cut on specific pictures even if not a random access point [23:26] <JEEB> and re-encode what's needed [23:27] <JEEB> otherwise trying to be "smart" about re-encoding can just end up kicking you in the arse [23:27] <JEEB> I'm not sure it always is worth it [23:27] <JEEB> or well, I'm not sure if it really is worth it ever :s (esp. with the speed of fast encoding in general nowadays, with f.ex. libx264) [23:28] <xreal> JEEB: So I think. there isn't a fast way to resize 1920 to 1080 :) [23:30] <JEEB> xreal, if you're not being bandwidth-limited between VRAM<->RAM transfer I see you being able to resize pictures rather well with OpenGL/Direct3D/etc [23:31] <xreal> JEEB: Oh, I want to create a 1080 file actually, not in realtime playback. [23:31] <JEEB> xreal, you want to resize something, no? [23:31] <xreal> JEEB: Yes, but can I do it with my GPU? [23:32] <reventlov> What options would you use to stream any mp4 found online using ffserver ? [23:32] <reventlov> I use ffmpeg -i foobar.mp4 -override_ffserver -vcodec mpeg2video -b:v 3000 http://127.0.0.1:8090/rtsp.ffm [23:32] <JEEB> xreal, why not? picture stuff is actually what that hardware was meant for after all [23:32] <reventlov> and it's ugly. [23:32] <xreal> JEEB: Does ffmpeg support this? [23:32] <JEEB> of course VRAM<->RAM transfer can be a bottleneck depending on how you implement the stuff [23:32] <JEEB> xreal, of course not [23:32] Action: xreal is confused [23:32] <JEEB> write your own stuff \o/ [23:32] <xreal> :D [00:00] --- Sat May 10 2014

1 0

ffmpeg-devel.log.20140508
by burek 09 May '14

09 May '14

[00:34] <cone-429> ffmpeg.git 03Michael Niedermayer 07master:9e5867743828: avcodec/huffyuvenc: Make version 3 of ffvhuff non experimental [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:351f6118c7e3: lavu/fifo: add av_fifo_freep function [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:70b63419ee33: lavfi: use av_fifo_freep [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:75a19f5eec66: lavd/jack_audio: use av_fifo_freep [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:a5f2f33a6794: ffmpeg: use av_fifo_freep [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:ce1eefe691ef: lavf: use av_fifo_freep [04:11] <cone-884> ffmpeg.git 03Lukasz Marek 07master:9b60d907ae6d: lavc: use av_fifo_freep [04:11] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:c71f8d40beb3: Merge remote-tracking branch 'lukaszmluki/master' [07:50] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:6c23a85000fd: avfilter/af_compand: Silence "may be used uninitialized" warning [09:20] <TimNich> lunux man pages [16:41] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:364e8904ce91: avcodec/cinepakenc: Fix "may be used uninitialized in this function" warnings [16:59] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:7e79d72c2871: avcodec/dcadec: remove always true if() [16:59] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:62a9725bc95e: avcodec/dcadec: add an assert to silence a uninitialized variable warning [17:44] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:ce4ae1840578: avcodec/takdec: Always initialize hsize, silence "may be used uninitialized" warning [17:44] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:aff352be6339: avcodec/mjpegdec: zero gb to silence warning about it being possibly uninitialized [19:39] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:0c2a6dabced5: avformat/rtmpproto: add FFMIN() to av_strlcpy() [21:53] <cone-159> ffmpeg.git 03Gonzalo Garramuno 07master:cd3daad77ea4: avcodec/exr: Add a gamma flag to exr loader to avoid banding [21:53] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:126927c58e16: avcodec/exr: mark gama option as experimental / unsupported [22:28] <Snowleaksange> seems image2 format doesnt work with custom AVIOContext? find_image_range() seems to rely on local filename, and the AVFMT_NOFILE flag is set [22:29] <ubitux> michaelni: no need to review 5/5, i'll drop it [22:30] <ubitux> i think it might cause some issue with pad filter [22:30] <ubitux> i'm not able to find a way of triggering the issue, but i'm not comfortable with that change anymore [22:30] <ubitux> i'll apply the 4 others though, th [22:30] <ubitux> thx [22:31] <michaelni> ok [22:33] <michaelni> Snowleaksange, it should work with a custom URLProtocol, though beware of ABI/API compatibility issues if you use that [22:38] <Snowleaksange> ah i didnt know about urlprotocols [22:39] <Snowleaksange> any recommended resource? can i replace all my old aviocontext with urlprotocols? [22:45] <cone-159> ffmpeg.git 03Martin Storsjö 07master:0bacfa8d3771: rtmpproto: Check the buffer sizes when copying app/playpath strings [22:45] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:996fffbbb3c2: Merge commit '0bacfa8d37710b904897e7cbeb8d6f96fbf75e2e' [22:45] <Snowleaksange> hmm yeah doesnt really look usable [22:45] <Snowleaksange> ok im just going to directly to all the codec libs myself [22:45] <Snowleaksange> basically reimplemented ffmpeg [22:48] <Snowleaksange> shouldve done this long ago [22:48] <Snowleaksange> you guys should just stop developing ffmpeg [22:48] <Snowleaksange> its never gonna be more than trojan horse [22:49] <Snowleaksange> no way to fix the code [22:49] <Snowleaksange> burn it [23:09] <compn> did i miss something ? [23:09] <compn> great troll. [23:20] <ubitux> ah, question for the timeline in case of phase filter [23:20] <ubitux> currently, when the filter is in "enable mode", it have to have a 1-frame delay [23:20] <ubitux> this might not be what the user is expecting [23:21] <ubitux> the other way is to always keep a reference to the previous frame [23:21] <ubitux> so it can be reuse immediately for when the filter is enabled again [23:21] <ubitux> that sounds better right?? [23:21] <ubitux> -? [23:27] <michaelni> id say yes [23:30] <ubitux> ok [23:47] <ubitux> #define DIFF(a, as, b, bs) (t = ((*a - b[bs]) << 2) + a[as << 1] - b[-bs], t * t) [23:47] <ubitux> pdif += DIFF(nptr, ns, nptr, ns); [23:47] <ubitux> i just realized how awesome this is [23:47] <ubitux> :) [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:362ab940fc5d: avfilter/phase: add comment on the context cached frame. [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:687119aa9d2d: avfilter/phase: do not pass the whole context to analyze_plane(). [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:8de600de553a: avfilter/phase: avoid a memcpy per frame. [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:69d72140accb: avfilter/phase: add timeline support. [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:5df10150d4a8: avfilter/phase: add some constness for RO data pointers. [23:58] <cone-159> ffmpeg.git 03Clément BSsch 07master:5623a10526ff: avfilter/phase: make the main loop readable. [00:00] --- Fri May 9 2014

1 0

ffmpeg.log.20140508
by burek 09 May '14

09 May '14

[01:01] <ChannelZ-Wk> Anyone good with stream/channel mapping? I can't get my head around something. [01:01] <c_14> State your problem and maybe someone will be able to help you. [01:02] <ChannelZ-Wk> I have a Quicktime with an audio stream which has 3 channels. I want to take that and tag the first two channels as Left and Right respectively, and the third as a mono track [01:05] <ChannelZ-Wk> So I guess I want to actually break it into 2 audio streams; a stereo one, and a mono. [01:10] <c_14> Can you try: ffmpeg -i input -map 0:0 -map 0:0 -map_channel 0.0.0:0.0.0 -map_channel 0.0.1:0.0.1 -map_channel 0.0.2:0.1 output [01:13] <c_14> Assuming the audio stream is stream 0 that is... [01:15] <ChannelZ-Wk> 0:0 is video, 0:1 is the 3 channel audio. But let me adapt and try.. thanks [01:16] <FreezingCold> -c:v libx264 -preset slow -crf 18 -c:a libvorbis -q:a 5 -pix_fmt yuv420p [01:16] <FreezingCold> I tried using those settings but it claims it's not a valid preset [01:18] <c_14> !paste FreezingCold [01:34] <ChannelZ-Wk> ah, think I got it. My problem seems to be that if I also try to use -filter_complex asetnsamples=n=16384 it ignores my stream mapping. [01:37] <c_14> Ye, that's probably because (according to the ffmpeg output) -map_channel is forwarded to lavfi as -af ... and I'm guessing -filter_complex then overrides that. [01:41] <ChannelZ-Wk> How I despise Quicktime. [01:43] <ChannelZ-Wk> c_14: Thanks for your help [01:43] <c_14> np [01:44] <ChannelZ-Wk> Making a bunch of ProRes. Quicktime Player on PC is a POS and unless you have a longer mux rate for the audio, it will randomly drop frames on playback. asetnsamples is the only way I've found to lengthen the mux. [01:44] <ChannelZ-Wk> oh well [01:53] <ChannelZ-Wk> One thing, it tells me "-map_channel is forwarded to lavfi similarly to -af pan=...." followed by a "This syntax is deprecated. Use "|" to separate the list items." Is it telling me the -map_channel syntax I used is deprecated, or the -af pan syntax _it_ turned it into is? :) [01:56] <c_14> Excellent question. I have no idea. [02:10] <ChannelZ-Wk> I'll update my build (it's a little stale) and see what it says. Heading home for now. Thanks again. [02:11] <l_r> can ffmpeg restart itself under some conditions? [03:59] <DOZO> does anyone know a way to identify unknown audio formats/codecs? using ffmpeg or anything else [03:59] <DOZO> ffprobe just tells me "Invalid data found when processing input" [04:00] <sacarasc> Try mediainfo, maybe? [04:01] <DOZO> will do [04:04] <DOZO> nope, all it reports is the file size [04:06] <sacarasc> What do you think it should be? [04:09] <DOZO> well, i'm trying to work with a program for an old computer that doesn't support WAV files, so i used it to convert a sample sound file to formats described as "YM2608 ADPCM", "MSM6258 ADPCM", "RF5C68 PCM", "Macintosh PCM" and "Unsigned 4-bit PCM" and am looking for a way to open/save/convert one of those on a modern computer [04:10] <DOZO> the "Unsigned 4-bit PCM" is what I actually need but i might be able to convert from one of the other formats to it [04:11] <DOZO> ffmpeg's adpcm_yamaha doesn't seem to be YM2608 ADPCM unfortunately [04:27] <helsinki> why `avformat_open_input` would block(wouldn't return forever) when openning some bad streams rather than return failure? [06:20] <joseph_> using -vcodec libx264 gives me "Unknown decoder 'libx264'", even though I have libx264 installed. What's going on? [06:42] <relaxed> joseph_: did you compile ffmpeg? [06:42] <relaxed> oh, libx264 isn't a decoder [06:43] <relaxed> you probably put -vcodec libx264 before the input, instead of after [08:32] <maksimkaaa> Hello, I have around 3000 frames as follows: first 1000 with frame rate of 5fps, next 2000 with 12 fps. I am looking for any documentation to handle multiple frames rates in a single video, could you please advice? I couldn't find the topic or examples on this in the docs.. [08:39] <DeadSix27> i there any way in which ffmpeg is able to identify if a video in the mp4 container has the "faststart" flag? [08:40] <DeadSix27> actually, is there way i can ientify it by using ffmpeg* [10:40] <Mavrik> DeadSix27, I doubt it, but a simple byte check should do [10:40] <Mavrik> the whole thing about faststart is just that mp4 MOOV atom is at the start [10:41] <Mavrik> so you can easily write a script to check if FTYP is followed by MOOV [11:29] <ergZay> I want to add -DTRACE to the compile of ffmpeg but it doesn't seem to work [11:29] <ergZay> I'm trying to do ./configure --extra-cflags="-DTRACE=1" [11:32] <ergZay> nvm seems to work now [13:55] <lxer> i'm trying to cut off the first 0.5 sec of some video framents, and then concatenate them into 1 video again. However, this is causing sync problems and a lot of these erors: [mp4 @ 0x349f6c0] Non-monotonous DTS in output stream 0:0; [13:57] <lxer> I'm using -ss 00:00:00.5 to trim the fragments, but I guess that isnt right. What is the correct way? [14:27] <hsch_web> hello [14:29] <hsch_web> is it possible to use ffmpeg to record audio to files that stores the audio of a hour and each file is created on the hour [14:33] <[A3G1S]> hey guys, I am trying to concat 2 videos of different sizes/resolution [14:34] <[A3G1S]> can anyone help me with that, I am able to concat videos which have same size/resolution [14:37] <lxer> how is it not working? [14:40] <[A3G1S]> i am trying to figure that out [14:40] <[A3G1S]> I have used steps from http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20medi… [14:40] <[A3G1S]> and is able to merge two files of same size [14:41] <[A3G1S]> *videos of same resolution* [14:44] <klaxa|work> maybe use a combination of scale/crop and the concat video filter in a filter complex [14:45] <lxer> perhaps add -qscale:v 1 [14:47] <[A3G1S]> trying [14:48] <sebastian__> Hello anyone out there ? [14:48] <[A3G1S]> lxer that didn't work [14:49] <[A3G1S]> i probably need to do something like this [14:49] <[A3G1S]> http://stackoverflow.com/a/17610511/938012 [14:49] <sebastian__> I have an Issue using Spotify, is there anyone out there who can help ? [14:50] <lxer> [A3G1S]: i was just reading this (for some other issue) http://stackoverflow.com/questions/12847262/ffmpeg-video-concatenation-audi… [14:50] <sebastian__> I get an response like: "12:44:00.455 W [audio_file_decompressor_ffmpeg.cpp:387] Failed to open stream" what does that mean? what is going wrong there ? [14:51] <[A3G1S]> ok k [14:56] <sebastian__> is everyone busy ? --- btw I am using that chat the very first time, sorry for inconviniences [14:56] <lxer> tell what you try to do. what is giving errors [14:57] <sebastian__> Thanks ,.. I am only trying to play an mp3 file stored on my harddisc with spotify [14:58] <sebastian__> Hahaaaa ,..... sooory ,.. fixed it [14:58] <sebastian__> OMG ,..... somebody shoot me ,.... [14:59] <sebastian__> of course I have to mount the drive before using it ,.. [15:01] <sebastian__> OKay ,... thanks for making my brain work ,.. bye [15:09] <lxer> I have many (small) mp4 files where the audio starts playing about 0.5 sec after the video started. Does this have something to do with syncing to keyframes or something? what causes this / what can I do about it? [15:12] <ac_slater> so I guess h264 isnt a supported RTP payload format for libavformat? It's not listed in the rtp_payload_types array :( [15:16] <ac_slater> hmm but it does with via -vcodec libx264 [15:17] <ac_slater> and is it true that ffmpeg's rtp only allows for 1 stream? [18:38] <Snowleaksange> any recommend example for how to av_probe_input_format a buffer? [19:40] <Snowleaksange> i can open a file fine [19:40] <Snowleaksange> but cant for the life of me open a buffer [19:41] <Snowleaksange> i remember i actually modified a previous version of av_probe_input_format() [19:50] <Snowleaksange> sucks how buggy this still is. the whole ffmpeg api tbh is horrendous by modern standards [19:54] <kolizer> ? [20:13] <Snowleaksange> ah hmm i guess none of my my custom aviocontexts have worked since i upgraded ffmpeg [20:13] <Snowleaksange> so my issue not necessarily related to av_probe_input_format [20:13] <Snowleaksange> thats encouraging [20:54] <kolizer> hello [20:55] <kolizer> what overlay webcam on video? [20:55] <kolizer> how overlay webcam on video? [20:56] <lxer> que? [20:57] <kolizer> what lxer [20:57] <kolizer> ? [21:02] <SpartanWarrior> hey guys, I have a problem converting a sequence of pngs to a video, apparently the first frame in the output doesnt seem to follow the fps I provided, here is what I put: http://pastebin.com/iuJNd8jS [21:03] <SpartanWarrior> I saw it was a reported bug, but the workaround I found never worked :/ [21:17] <Snowleaksange> maybe the image2 format just doesn twork with custom AVIOContext [21:18] <Snowleaksange> ffmpeg is possibly the worst software in the world by metric = trojan timesuck potential * number of users [21:21] <lxer> true. mostly because documentation is old/wrong/missing. [21:23] <kolizer> how overlay webcam on video? [21:35] <c_14> kolizer: Look at the overlay filter [21:36] <c_14> SpartanWarrior: I'm not sure what to tell you, but your command works for me. [21:37] <SpartanWarrior> c_14, is the first frame duration correct? it should be exactly 1 second as the other ones [21:38] <c_14> Wait, you're right. [21:38] <c_14> I lost track of which frame was first. [21:39] <c_14> That only happens when I output to mp4 though... [21:50] <c_14> SpartanWarrior: https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images… [21:50] <c_14> I used -vf fps=11 instead of -r 11 and the first frame is shown corretly. [21:52] <SpartanWarrior> c_14, using the fps would modify the video sound? I'm doing a slideshow [21:52] <SpartanWarrior> fps filter* [21:53] <c_14> It shouldn't. [21:53] <SpartanWarrior> i'll give it a shot [21:58] <SpartanWarrior> indeed it doesn't affect the sound, but now the video is all black and only one frame appears in the middle x_x [22:01] <c_14> what? [22:01] <c_14> That wasn't supposed to happen. [22:01] <SpartanWarrior> forget it, it's working [22:01] <SpartanWarrior> BUT [22:01] <SpartanWarrior> now it's the last frame what is not appearing :/ [22:07] <c_14> Well, if it's the last frame, the cheap answer would be to copy the last frame to last frame + 1... [22:08] <c_14> It is apparently a known bug: https://trac.ffmpeg.org/ticket/2674 [22:10] <c_14> Might I ask why you even bother setting an output framerate? [22:11] <c_14> If I just set the input framerate via -r 1, the output steps through 1 picture per second and skips neither the first nor the last frame. [22:11] <kolizer> -f x11grab [22:11] <kolizer> f v4l2 [22:11] <kolizer> i use -filter_complex overlay= [22:11] <kolizer> c_14 [22:12] <kolizer> and drop( [22:13] <kolizer> whay? [22:13] <SpartanWarrior> i tried using only -r 1 but the output was a video with anything else than 1fps [22:14] <SpartanWarrior> (it was faster, way faster) [22:16] <c_14> kolizer: I'm afraid I don't understand what you're trying to tell me, could you paste your complete command and all output? [22:16] <SpartanWarrior> i gues i'll stay with the nasty hack of cloning a frame :( [22:16] <kolizer> ffmpeg -y \ [22:16] <kolizer> -f x11grab -video_size 1280x720 -framerate 30 -show_region 1 -i :0.0+0,49 \ [22:16] <kolizer> -f video4linux2 -input_format mjpeg -i /dev/video0 -c:v copy \ [22:16] <kolizer> -filter_complex overlay=main_w-overlay_w:main_h-overlay_h \ [22:16] <kolizer> -vcodec libx264 -preset veryfast -maxrate 1400k -bufsize 1400k -g 60 \ [22:16] <kolizer> -threads 0 \ [22:17] <c_14> SpartanWarrior: if you ffprobe your output file what does it say for fps [22:17] <c_14> kolizer: next time use a pastebin service please [22:17] <kolizer> ok [22:18] <c_14> kolizer: So what's the problem? [22:18] <kolizer> drops [22:18] <kolizer> low fps [22:19] <SpartanWarrior> http://pastebin.com/iSxfYqxJ [22:19] <c_14> That's the one where you only specified the -r 1 on the input file? [22:20] <SpartanWarrior> yes [22:20] <SpartanWarrior> the original file [22:21] <c_14> Weird, why does ffmpeg think the fps is 11... [22:22] <kolizer> 5 fps( [22:22] <c_14> kolizer: Try adding -r 30 before -i /dev/video0 [22:24] <kolizer> /dev/video0: Invalid argument c_14 [22:24] <kolizer> oh no [22:25] <kolizer> fps=0.5 [22:25] <c_14> That's just the encoding fps. [22:25] <c_14> Ie how many frames ffmpeg is encoding per second. [22:26] <c_14> That has nothing to do with the fps of the actual output. [22:26] <kolizer> frame= 35 fps=0.3 q=23.0 size= [22:27] <Hello71> you're encoding at 0.3 fps. [22:27] <Hello71> meaning it will take you approximately 200x the duration of the video to encode [22:27] <Hello71> assuming 60fps. [22:29] <kolizer> what to do? [22:29] <kolizer> http://pastebin.com/59bcVM2e [22:29] <c_14> Normally, this isn't that much of an issue. If you want to encode faster, you can change your preset to superfast or ultrafast [22:31] <jamesaustin> Question: How do I convert a 240 fps video to 60 fps but retaining all the frames, ie. making it 4 times longer? [22:31] <SpartanWarrior> weird :/ [22:32] <jamesaustin> I tried: ffmpeg -r 240 -i in.mp4 -filter:v "setpts=4.0*PTS" -r 60 -an out.mp4 [22:34] <c_14> jamesaustin: What happens if you leave out both the -r options? [22:36] <jamesaustin> c_14: It seems to not alter the fps. 240fps in, 240 fps out. [22:36] <jamesaustin> c_14: As if the "setpts" is ignored. [22:37] <c_14> But the video is 4x as long? [22:38] <kolizer> why do I have a low FPS? [22:38] <jamesaustin> Ok, my mistake. I had a second -filter:v on the command. Which appears to overwrite the first. [22:39] <c_14> ye, they do that. just separate the filters with commas or use -filter_complex [22:39] <jamesaustin> Thanks - sorry for the stupid question. [22:39] <c_14> Happens to the best of us. [22:40] <kolizer> I do not understand c_14 [22:41] <c_14> kolizer: Encoding video takes time, depending on the input and output video and the computer it can take more or less time. The fps=0.3 output ffmpeg is giving you is telling you that ffmpeg is currently encoding the source video at 0.3 frames per second. [22:43] <c_14> If you want the video to encode faster, you need a less complex source, -preset {superfast,ultrafast}, a lower resolution, a faster compression algorithm/implementation, or a faster computer. [22:44] <kolizer> i use superfast,ultrafast, low fps [22:49] <kolizer> CPU 5% over c_14 [22:49] <c_14> That's strange. [22:49] <c_14> try removing -threads 0 [22:51] <kolizer> no [22:54] <kolizer> oh( [22:58] <kolizer> http://pastebin.com/T01rZRRY c_14 [22:58] <kolizer> fps 15 why? [22:59] <c_14> The fps of the file or the fps ffmpeg is telling you while it's encoding? [23:00] <kolizer> while it's encoding [23:08] <kolizer> maybe need webcam setting c_14? [23:10] <c_14> I'm testing with my own right now. [23:20] <c_14> So I just tested it with my webcam, ffmpeg says it's encoding with 15fps, but when I play the output file, it has 30fps and it seems to be encoding at the same speed that I'm playing the output, which I gander is what you're after. [23:25] <kolizer> and? [23:27] <c_14> So what I'm saying is just try the command and while the command is working, play the output and see if the encoding keeps up with the speed you're playing the file at. [23:32] <kolizer> Video is too quick c_14 [23:33] <c_14> Is the video too fast or is the encoding too slow? [23:34] <kolizer> video too fast [23:35] <c_14> Try adjusting the framerate until the video plays at normal speed. [23:53] <kolizer> ok works overlay c_14 [23:54] <kolizer> bat Sound late on webcam [23:56] <c_14> Try using -itsoffset to adjust until it's in sync [00:00] --- Fri May 9 2014

1 0

ffmpeg-devel.log.20140507
by burek 08 May '14

08 May '14

[00:48] <cone-318> ffmpeg.git 03Uwe L. Korn 07master:7ce3bd961471: rtmpproto: Support alternative slist parameter in rtmp URLs [00:48] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:d68ed9f257bd: Merge commit '7ce3bd9614717e545af8fb8455032c807e389b78' [01:16] <Daemon404> ffmpeg now has exatly 1000 stars on github. [01:16] <Daemon404> yay? [01:17] <wm4> lol [01:17] <J_Darnley> Yay 1000 internet points! [01:17] <J_Darnley> What do we win? [01:18] <BBB> 2 youtube dollars [01:19] <BBB> (context: https://www.youtube.com/watch?v=idZOVqdcqno) [01:20] <J_Darnley> South Park? [01:25] <iive> we got internet money!!! [01:40] <cone-318> ffmpeg.git 03James Almer 07master:4cdea9297679: swresample/resample: add missing xmm clobbers [02:21] <Compn> what movie do you want to watch on youtube ? [04:09] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:84655bdaed9c: avcodec/hevc: remove unused variable [04:09] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:326463893beb: avcodec/wavpack: initialize pointers to silence warning about them possibly being uninitialized [04:09] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:04b15a6055fc: avformat/h263dec: shift data in state to make more bits available to probe [04:09] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:3ad21c50af04: avformat/h263dec/h263_probe: Check PSC bit 9 and 13 in [04:09] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:cd20b93e2f51: avformat/h263dec/h263_probe: Check TR [06:15] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:8dd435f0cf2f: cmdutils: include config.h, fix "is not defined" warning due to CONFIG_OPENCL [06:15] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:134206ca7a3d: avformat/utils: fix const warnings [07:42] <anshul> hi in nb_streams what does nb mean [07:42] <ubitux> number of [07:43] <anshul> thanks [07:48] <anshul> there is a problem in ffprobe, nb_streams_packets are allocated by looking only part of the file, due to which size of nb_streams_packets is less if numbre of stream increase later in video [07:50] <anshul> so is there any ffprobe context which save nb_stream at the time of allocation of nb_streams_packets, or should i store one more variable globally beside nb_streams_packets [07:55] <anshul> packet.stream_index start from 0 or 1 [08:02] <anshul> its start from 0 [09:10] <ubitux> anshul: yes there is a ticket opened about that issue i believe [09:19] <anshul> yes #3603 , i am on it. [09:28] <anshul> is it safe to give pointer from av_calloc to av_realloc [09:29] <anshul> or should i av_calloc memory and assign previous memory value to newly assigned memory [09:34] <ubitux> that's called av_realloc [09:34] <ubitux> make sure first call is a realloc too though [09:39] <plepere> hi all [09:39] <plepere> got the feedback on the hevc patch [09:40] <plepere> I'm doing some changes on the hevcdsp_init.c to give it more modularity [09:53] <ubitux> grumbl archlinux rebuilding ffmpeg with 4.9 [09:53] <ubitux> (https://bugs.archlinux.org/task/40256) [09:53] <JEEB> yeah, there was a guy on #ffmpeg yesterday [09:55] <JEEB> seems like this aphirst dude didn't care to report it further than the IRC channel :P [10:03] <anshul> ubitux, actually in ffmpeg memory is allocated using mmap(depend on configuration), while realloc has to be given pointer which was given by malloc, realloc or calloc, so I was wondering whether it is safe or not. [10:03] <ubitux> it's not, it's in the doxy [10:04] <ubitux> see @warning [10:07] <anshul> there is place where av_calloc is used to allocate according to number of stream, but after some time streams are increased, i need to realloc it so if i use av_realloc that would be safe or not. [10:08] <ubitux> replace the original av_calloc with a av_realloc [10:09] <ubitux> (use a NULL) [10:11] <anshul> ok thanks [10:27] <anshul> clement, as u asked "mark the fix in the description, and describe the fix in the title" I marked it fixed, but i didnt got the second part. [10:28] <ubitux> i meant to use something like "ffprobe: fix crash because of new streams occuring" and 2 lines below "Fix ticket #3603" [10:29] <ubitux> anshul: also while at it, please fix your spacing [10:29] <ubitux> another thing i don't like is that i don't see any overflow check [10:30] <ubitux> and i'm not sure about of the new static int nb_stremas: fmt_ctx->nb_streams should be enough [10:30] <ubitux> anshul: do you want me to state all of this on the ml? [10:31] <anshul> no i am doing it [10:31] <ubitux> beastd: i'll reply to your mail later, sorry about the anger unleash btw [10:32] <beastd> Hey ubitux! [10:33] <beastd> No problem and no hurry. I will be travelling in about 1h and probably get to look at mails again somewhen tonight. [10:34] <beastd> now to sth completely different... [10:38] <beastd> i have been experimenting with Trac and found a way to make the view tickets from the top menu bar just use the query module and output a formatted table with the result. i think it is more comfortable than how things are working now where you get a list of reports first and than need find the report you want to execute and do another click. [10:40] <beastd> ubitux: i think you have mentioned that you find it cumbersome too. somewhen later this month i will create a ticket and ask if the behaviour should be changed. [10:41] <anshul> ubitux, i have used int nb_streams for saving number of stream at the time of allocating nb_stremas_packets, since size of nb_streams_packet is always equal to this new static nb_streams [10:42] <anshul> this static nb_streams(size of array) is checked for checking overflowing [10:42] <ubitux> beastd: yeah definitely [10:45] <ubitux> anshul: sounds clumsy since there is nb_streams_{frames,packets,...} [10:45] <ubitux> anshul: btw, you could drop the initial av_realloc altogether [10:45] <ubitux> oh wait my bad the "nb_" are tabs [10:46] <ubitux> well, what about calling av_realloc unconditionnally? :) [10:47] <ubitux> like, drop all the nb_streams_* = av_calloc(), and before nb_streams_packets[...] you just realloc with fmt_ctx->nb_streams [10:47] <ubitux> mmh that won't do for the zero padding, which you're not handling anyway [10:48] <ubitux> oh well, i'll wait for your next iteration [10:49] <ubitux> anshul: space before * and (, look at the surrounding style please [10:50] <ubitux> where do you memset to zero the new added stream(s) counters? [10:50] <anshul> where should nb_streams_* free all the reallocated thing [10:50] <anshul> ok i will look at * and ( [10:51] <ubitux> the memset is more important [10:51] <ubitux> static nb_streams should be fine, i just missed that originally [10:52] <anshul> ok i will add memset and done * and ( locally [10:52] <ubitux> but since it's zero i still think you should just have the realloc code once (and simply drop the av_calloc) [10:53] <anshul> i agree with you [10:54] <anshul> but allocating in read_interval_packets and free in probe_file looks odd to me [10:55] <anshul> is it a better idea to remove those static pointer and move them to avformat [10:57] <anshul> ubitux, is cehoyos your name in trac or is he carl Euegon [10:58] <ubitux> "ce" from "cehoyos" stands for carl eugen [11:01] <anshul> if i remove realloc completely from probe file, then nb_streams_frames and selected_streams may be used before [11:04] <ubitux> ok [11:52] Action: beastd is leaving [11:52] <j-b> to? [11:53] <beastd> Heading to Berlin now :) [11:53] <beastd> j-b: Do you know who will be there from VideoLAN? [11:54] <j-b> beastd: EVERY-ONE [11:54] <j-b> beastd: like EVERY-fucking-ONE [11:56] <beastd> j-b: oha. So you will be on LinuxTag too? [11:57] <beastd> anyways, i am out. bye... [13:34] <anshul> What does analyse by developer mean in ffmpeg trac, after analysing ticket i marked the ticket analyse, does it mean something else. [13:38] <Daemon404> ubitux, remind me why av_calloc is incompatible [13:38] Action: Daemon404 always forgets the malloc/realloc/calloc business [13:40] <kierank> alignment probably [13:40] <anshul> av_calloc use mmap function, av_realloc use realloc function, in linux realloc manual it is specified that there is undefined behaviour if pointer passed to realloc is not given by malloc calloc or realloc [13:42] <Daemon404> i guess it uses mmap for 'performance reasoons [13:42] Action: Daemon404 is not sure why calloc would use mmap [13:42] Action: Daemon404 checks source [13:42] <kierank> yeah i doubt it uses mmap [13:43] <kierank> realloc isn't guaranteed to get you an aligned pouinter [13:43] <Daemon404> it doesnt [13:43] <Daemon404> i mena [13:44] <Daemon404> av_calloc calls av_mallocz which calls av_malloc [13:44] <Daemon404> which calls posix_memalign on most systems [13:44] <Daemon404> which != malloc [13:46] <Daemon404> i dont actually see that we've documented it anywhere that you cant use av_malloc and av_realloc [13:46] <Daemon404> and the names suggest you can. [13:47] <anshul> its documented in doxy [13:48] <anshul> actually av_calloc use av_malloc [13:48] <Daemon404> indeed i already said that [13:49] <Daemon404> where in doxy is it [13:49] <anshul> sry , didnt came till last message [13:49] <Daemon404> ok i see it in doxy [13:49] <Daemon404> doesnt mention calloc though [13:50] <anshul> I dont seen doxy but in source code above av_realloc defination [13:51] <anshul> calloc is not having any problem it av_realloc which should we not called after av_malloc and av_calloc [13:51] <anshul> What does analyse by developer mean in ffmpeg trac [13:52] <Daemon404> it means a developer has looked at it [13:53] <anshul> so what define developer [13:53] <anshul> since every time i mark a ticket as analysed then cehayos unset it. [13:56] <Daemon404> carl is an asshole [13:56] <Daemon404> thats the only reason [13:59] <anshul> hmmmm [13:59] <Daemon404> anshul, i wouldnt worry [14:00] <Daemon404> just mark it as fixed one the fix is pushed [14:00] <Daemon404> he cant really argue with that [14:01] <anshul> yup, i dont worry about that, urs idea is good [14:07] <iive> anshul: do you write the result of the analysis as a comment in the ticket? [14:08] <J_Darnley> anshul: perhaps the "analysed by developer" flag keeps getting unset because carl doesn't recognise you on trac and thinks some random newbie keeps setting a flag he (the newbie) doesn't know the meaning of [14:27] <anshul> iive, yes i wrote my analysis [15:04] <ubitux> can't we do something better than link->time_base = av_inv_q(s->framerate); in the fps filter? [15:05] <ubitux> i have serious problems here because of this [15:05] <Daemon404> how could somerthing be "better" [15:06] <ubitux> well, the typical problem i'm having here is that i'm trying to extract regular thumbnails doing sth like ffmpeg -i in.mp4 -vf fps=0.45,scale=120:80,tile=10x3 -y -frames:v 1 out.jpg [15:07] <ubitux> fps is calculated using the duration and the number of frame we want (here 30) [15:07] <ubitux> oh mmh wait [15:08] <Daemon404> i dont think you should be gettign fps involved to extract thumbs [15:15] <ubitux> i was doing something wrong, it actually works fine [15:15] <Daemon404> o ok [15:15] <ubitux> Daemon404: well, do you have another solution to get regular frames? [15:16] <ubitux> i could use select, but it's the same as fps typically [15:16] <Daemon404> >vfr [15:16] <ubitux> ? [15:16] <Daemon404> time based is what you want [15:16] <Daemon404> oh wait, yorue changing teh fps [15:16] <ubitux> fps is time based, select can be as well [15:16] <nevcairiel> Daemon404: in the real world, people don't care about vfr :P [15:16] <Daemon404> nevcairiel, i dunno [15:16] <Daemon404> i get paid to care about vfr [15:17] <nevcairiel> many companies dont live in the real world [15:17] <Daemon404> i cant really ignore vfr [15:17] <Daemon404> no more than youtube can [15:17] <nevcairiel> because 1 in 10000 uses it? :d [15:17] <Daemon404> significantly mroe [15:17] <Daemon404> iphones capture vfr [15:17] <JEEB> well, youtube just converts everything to specific video rates [15:17] <Daemon404> and many MANY professional editors output vfr [15:18] <Daemon404> either using edit lists or just pts/dts [15:19] <ubitux> iphone captures vfr but randomly [15:19] <Daemon404> yes [15:19] <Daemon404> so do some android phones [15:19] <ubitux> it's cfr unless luma change :') [15:19] <ubitux> and then suddenly surprise [15:19] <Daemon404> ubitux, and it looks horrible [15:19] <Daemon404> :D [15:20] <nevcairiel> phones typically only capture vfr when they notice their crappy hardware can't keep up [15:20] <nevcairiel> at least thats one random factor in androids capturing, not sure about iphones [15:21] Action: Daemon404 goes to poek swscale, weeps inside [15:21] <Daemon404> adding new pix fmt is not very fun [15:42] Action: Daemon404 wonders why new hevc stuff is beign submitted while the old stuff is still broken [15:45] <plepere> Daemon404, it's because I've been working on ASM for some time now, and the performance boost is really necessary for everyone. [15:45] <Daemon404> it would be even better if it didnt crash [15:45] <Daemon404> which is kind of also important [15:45] <plepere> obviously. [15:45] <plepere> you're referring to the win32 crashing ? [15:45] <Daemon404> yes [15:46] <plepere> ok, I'm checking it out. hopefully it's only an ASM problem. [15:46] <Daemon404> well, win64. [15:48] <Daemon404> [14:45] < plepere> Daemon404, it's because I've been working on ASM for some time now, and the performance boost is really necessary for everyone. <-- fyi though, 32-bit windows is the largest consumer of ffmpeg, technically :P [15:48] <Daemon404> and no such asm exists [15:49] <smarter> plepere: I have a cleaned up version of the deblocking asm at https://github.com/smarter/libav/tree/hevc_dbl_asm if you're planning on sending it to the ML (otherwise I'll do it once I have the time) [15:50] <ubitux> mmh why ffmpeg -i in.mp4 -vf fps=0.5 -frames:v 1 a.png and without -vf fps doesn't give the same thing? [15:50] <plepere> smarter : I'll try to get win64 compilation working first, but I'm keeping your adress in my favorites. thanks [15:50] <ubitux> mmmh i mean with a seek [15:51] <ubitux> jeez my brain is off or what [15:53] <smarter> Daemon404: chrome should distribute a binary that contains both a 32 bits and a 64 bits version of itself, problem solved :p [15:53] <Daemon404> ;p [15:53] <Daemon404> smarter, the other big distributer is steam [15:53] <Daemon404> because it uses chromium [15:54] <smarter> though neither of them care about hevc [15:54] <Daemon404> yet [15:55] <Daemon404> nobody *really* cares about hevc yet [15:55] <Daemon404> since nothing can beat x264 consistently [15:55] <smarter> I don't see hevc in <video> ever happening [15:55] <plepere> Daemon404, steam could use hevc in their streaming@home thing. [15:55] <JEEB> plepere, after the implementations get somewhere [15:56] <Daemon404> smarter, i do [15:56] <Daemon404> if not solely for MS [15:57] <kierank> Daemon404: they do care [15:57] <Daemon404> kierank, not for any legiitmate reasons [15:57] <Daemon404> marketing isnt legitimate [15:57] <cone-904> ffmpeg.git 03Matt Oliver 07master:1898c2f49da3: inline asm: fix arrays as named constraints. [15:58] <kierank> smarter: why not [15:58] <Daemon404> because smarter works on vp9 [15:58] <Daemon404> /troll [15:58] <kierank> if iphone supports it [15:58] <kierank> it'll happen [16:06] <kierank> i think hevc could go the way of mpeg-4 asp [16:06] <kierank> patent hell [16:07] <Daemon404> fun [16:07] <Daemon404> not that it ever stopped pirates form using it [16:28] <nevcairiel> people didnt want mpeg2 to live for ever, so what was there instead! [16:38] <nevcairiel> I really wonder why someone would bother with ICL, it seems to cause new problems every day [16:38] <nevcairiel> and the code isn't really faster, at least not fast enough to warrant it [17:13] <cone-904> ffmpeg.git 03Michael Niedermayer 07master:efbf107f5b28: avfilter/aeval: silence "may be used uninitialized" warning [17:13] <cone-904> ffmpeg.git 03Michael Niedermayer 07master:eeb48353abaa: avcodec/ac3enc: silence 2 warnings about "possibly uninitialized" variables [17:16] <michaelni> ubitux, did you post the coverage.ffmpeg.org link to the ML ? if not i think you should, it would make sense if people see where test coverage is lacking and its imho important that we improve test coverage [17:18] <ubitux> michaelni: i believe i did on various occasions [17:18] <ubitux> michaelni: but i think it should be part of the website [17:19] <ubitux> that's the reason i wanted a new website but... well it's stalled [17:19] <ubitux> we juste need a stupid left menu with categories to move all the shit in trees in it [17:19] <ubitux> so we can add random entries [17:19] <ubitux> i think i'll end up taking 2-3 hours to do that [17:20] <ubitux> but fuck, i hate web :( [17:25] <ubitux> michaelni: db0 was stalled because of that last thread, i told her she should just continue to do it anyway, so here we go [17:26] <michaelni> i agree it should be part of the website, also, if you see something that i could do to unstall this or otherwise ... [17:27] <michaelni> also we maybe could remove some stuff from the top naviatio line or make it 2level with a second line based on what is selected in the first [17:28] <michaelni> or drop down menu style [17:28] <michaelni> but either it should have some fallback for people who disabled javascript if possible [17:28] <michaelni> if it uses javascript that is [17:30] <ubitux> db0 was going to do a drop down menu; i would personally prefer something like http://www.musicpd.org/ simply [17:31] <ubitux> so you don't have to care about various compat issue with js & shit [17:33] <michaelni> agree, that looks nice too [18:20] <cone-904> ffmpeg.git 03Michael Niedermayer 07master:a779602584b4: avcodec/eamad: silence uninitialized variable warnings [18:20] <cone-904> ffmpeg.git 03Michael Niedermayer 07master:a6e9448dc697: avcodec/cinepakenc: Zero AVPictures, silence "may be used uninitialized" warnings [18:20] <cone-904> ffmpeg.git 03Michael Niedermayer 07master:54ae58802e98: avcodec/cinepakenc: drop coded_frame init [00:00] --- Thu May 8 2014

1 0

ffmpeg.log.20140507
by burek 08 May '14

08 May '14

[02:44] <Na_Klar> why does a convert to dpx rgb48le image sequence fails with "Video encoding failed"? (ffmpeg 0.7.6-4:0.7.6-0ubuntu0.11.10.3) [02:51] <iive> it is probably libav fork and it is very old. [02:52] <afflicto> Hello everyone. How can I instruct ffmpeg to record a pulseaudio null-sink? [02:53] <afflicto> Right now I do "-i hw:0,0" to grab the default input device. [02:55] <Na_Klar> iive, would you be kind enough to check your ffmpeg version for dpx rgb48le encoding? [02:55] <iive> what is your command line? [02:56] <Na_Klar> ffmpeg -i test.mov -an -vcodec dpx -r 24 -pix_fmt rbg48le -f image2 ./%04d.dpx [02:58] <iive> ffmpeg -pix_fmts |grep rgb48le [02:58] <iive> IO... rgb48le [02:58] <Na_Klar> i have that too [02:58] <Na_Klar> rgb [02:58] <Na_Klar> yeah, typo (here not in my cli) [03:00] <iive> Unknown pixel format requested: rbg48le [03:00] <iive> let me take a look. [03:00] <iive> rgb ? [03:00] <Na_Klar> yeah it's rgb .. not rbg .. that was a type (here in the iirc, not in my cli) [03:00] <iive> rgb48le works [03:00] <Na_Klar> really? [03:01] <Na_Klar> that's good to know .. then it's my ffmpeg version .. fine [03:01] <Na_Klar> ty [03:01] <iive> well, mine is 2.2.0 yours is 0.7.6 [03:02] <Na_Klar> it's actually a bit newer .. ubuntu has some weird versioning for its ffmpeg forks .. [03:02] <Na_Klar> but older that 2.2 obivously [03:03] <iive> libav 9.13 [03:03] <iive> seems to be the latest fork release. [03:04] <iive> i guess, they dropped the leading zero [03:05] <Na_Klar> yeah, so i have 7.6 here .. [03:06] <Na_Klar> besides ubuntu moved to avconv since 12.04 [03:06] <sacarasc> It's only since (0.)9.x that they've done that. You've got good old 0.7.6... [03:06] <sacarasc> That's the fork iive's been talking about. [03:07] <Na_Klar> ic [03:07] <iive> btw, try avconv, they (libav) keep an older ffmpeg build and update their renamed program instead. [03:07] <afflicto> I need some help with audio glitches, I'm streaming it here: http://twitch.tv/dreamvoid - here's the ffmpeg command I use: http://pastebin.com/Wh6tdnmC [03:11] <bencc> what are the settings I need for converting raw video to be played on html5 on ios and android? [03:11] <Na_Klar> iive, yeah, i'll try. btw, is there a pix_fmt to convert to a plain 10bit per channel dpx? rgb48 are 16bits per channel. [03:11] <bencc> this should work? [03:12] <iive> afflicto: this is quite interesting glitch. [03:12] <bencc> ffmpeg.exe -ss 00:04:00 -t 1:44:55 -i output.mkv -c:v libx264 -preset medium -crf 23 -vf crop=iw:ih-120:y=0,format=yuv420p -vf scale=1024:672 -c:a aac -strict experimental -b:a 96k -movflags +faststart output.mp4 [03:12] <afflicto> iive: mhm. I believe it works fine if I just stream hw:0,0 [03:13] <iive> afflicto: then it is most likely pulse audio bug. I blame it for every audio related problem, until proved otherwise :| [03:13] <afflicto> iive: hm :\ [03:13] <Na_Klar> bencc, to have a playable html video file for all (the most major) systems, you'll need a mp4, ogg and webm file. mp4 profile 4.1 is fine for the most. webm with vp8, ogg as usual. [03:13] <iive> I was going to eliminate the variables. [03:14] <sacarasc> bencc: Does that not work? [03:14] <iive> e.g. try to capture without video. try to save directly to pcm file . [03:14] <afflicto> iive: Audio works fine using SimpleScreenRecorder though [03:14] <bencc> mp4 works for me. I don't know what resolution I need [03:15] <bencc> and if I also need to mess with the mp4 profile [03:15] <afflicto> iive: what a dilemma. audio is glitchy with ffmpeg and video is glitchy with simplescreenrecorder :( [03:16] <bencc> sacarasc: when I tried 1280x840 it didn't play on iphone 4S. [03:16] <Na_Klar> bencc, for you means not for everybody .. but whatever. The resoloution is irrelvant for the playability. [03:16] <iive> just try to isolate the problem. [03:16] <sacarasc> bencc: http://developer.android.com/guide/appendix/media-formats.html https://developer.apple.com/library/ios/documentation/miscellaneous/concept… [03:16] <bencc> Na_Klar: are you sure that iphone 4S can play 1280x840? it didn't work in iphone and galaxy 3 [03:17] <Na_Klar> bencc, my samsung galaxy s4 can play 1920x540 mp4 h264 profile >=5 [03:17] <Na_Klar> s/1920x540/1920x810/ [03:17] <bencc> Na_Klar: so resolution does matter. you said 810 height but I tried 840 [03:17] <bencc> so is there a safe resolution I can use? [03:18] <bencc> sacarasc: not sure how to translate this info to ffmpeg parameters [03:18] <Na_Klar> bencc, in principal it's irrelevant. but if you want to go sure use factor of 16 or 4 or at least 2 for x and y [03:19] <bencc> Na_Klar: what do you mean? [03:19] <bencc> does 1024x672 supposed to work? [03:19] <sacarasc> H.264 video, up to 1.5 Mbps, 640 by 480 pixels, 30 frames per second, Low-Complexity version of the H.264 Baseline Profile with AAC-LC audio up to 160 Kbps, 48 kHz, stereo audio in .m4v, .mp4, and .mov file formats [03:19] <sacarasc> That's the highest Apple say works. [03:20] <bencc> ok. so I need to encode in much lower resolution. I'll try it now [03:21] <bencc> sacarasc: I need baseline profile? [03:21] <sacarasc> That's what Apple says. [03:21] <bencc> "Low-Complexity version of the H.264 Baseline Profile" [03:21] <bencc> how do I enforce it with ffmpeg? [03:21] <Na_Klar> bencc, it's devideable by 4. but - honestly - it should work anyways. resoloution is irrelevant. [03:22] <Na_Klar> [apple is sh*t anyways .. just saying] [03:22] <klaxa> bencc: you can enforce it in ffmpeg by using: -profile:v baseline [03:22] <bencc> https://trac.ffmpeg.org/wiki/x264EncodingGuide#Compatibility [03:23] <bencc> this says iPhone 4 supports main 3.1 [03:23] <bencc> but apple's docs says baseline profile. what should I believe? [03:23] <bencc> what is the default profile ffmpeg use? [03:26] <sacarasc> bencc: It depends how far back you want compatibility, I guess. [03:27] <Na_Klar> 3.1 .. gosh .. that's far back [03:28] <klaxa> 3.1 is level, not profile, right? [03:28] <Na_Klar> right [03:28] <sacarasc> Yes. [03:28] <Na_Klar> profiles are like slow, very-slow, placebo ... [03:28] <klaxa> no [03:28] <klaxa> those are presets [03:28] <Na_Klar> ups [03:28] <Na_Klar> my fault [03:28] <klaxa> profiles are baseline, main, high, high10, high444 [03:28] <Na_Klar> right [03:29] <bencc> this sets the correct resolution but doesn't crop [03:29] <iive> afflicto: seems fixed now [03:29] <bencc> ffmpeg.exe -ss 00:04:00 -t 1:44:55 -i output.mkv -c:v libx264 -preset medium -crf 23 -vf crop=iw:ih-120:y=0,format=yuv420p -vf scale=1024:672 -c:a aac -strict experimental -b:a 96k -movflags +faststart output.mp4 [03:29] <afflicto> iive: it is? =D [03:29] <afflicto> iive: YESS ;D [03:29] <iive> no [03:29] <bencc> is it wrong to use -vf crop... -vf scale ? [03:29] <afflicto> iive: nope. it's not :( [03:29] <klaxa> probably? put it in one -vf argument [03:29] <iive> it goes fine for a while, then distorts. [03:29] <klaxa> just do: -vf crop=iw:ih-120:y=0,format=yuv420p,scale=1024:672 [03:29] <bencc> klaxa: how [03:30] <bencc> trying [03:30] <afflicto> iive: yeah :\ [03:30] <iive> have you tried renice on pulse sound deamons? [03:31] <iive> something else. try without -ar 44100 [03:31] <afflicto> renice? [03:31] <afflicto> I can try without the -ar 44100 I guess [03:32] <azk> Does FFserver check for X-Forwarded-For headers? [03:33] <afflicto> iive: without the "-ar 44100" part, I get an error "incorrect codec parameters?" [03:33] <iive> try with 48000 [03:34] <afflicto> iive: I get: Could not write header for output file #0 (incorrect codec parameters ?) [03:34] <afflicto> iive: with 48000 [03:34] <iive> just a moment. [03:35] <iive> try with -af aresample=44100 [03:36] <iive> most ac97 cards work at 48kHz, if you request 44,1kHz someone should resample the pcm. [03:37] <iive> if pulse resample is buggy, or worse, it doesn't resample, then it might just drop samples. [03:37] <afflicto> iive: I see [03:38] <afflicto> so I replace "-ar 48000- with "-af aresample=44100" ? [03:38] <iive> yes, try it. [03:39] <afflicto> iive: unrecognized option 'af' [03:40] <iive> hum, -af stands for audio filter... [03:40] <iive> you might have old ffmpeg, or worse. [03:40] <afflicto> iive: hm [03:41] <iive> latest is 2.2.2 (or 9.13 in case of the fork) [03:41] <iive> do you have avconv ? [03:42] <afflicto> I dno [03:42] <afflicto> yes [03:42] <iive> just type it in the terminal. [03:42] <afflicto> afflicto@afflicto-linux:~$ avconv [03:42] <afflicto> ffmpeg version 0.10.9-7:0.10.9-1~raring1 [03:43] <iive> the libav fork provides old ffmpeg version that is not updated, instead they update their avconv utility. try replacing the ffmpeg with avconv in the script. [03:44] <afflicto> ok [03:44] <iive> I do recommend you to use latest real ffmpeg, but that might be quite a headache. [03:44] <afflicto> It still won't recognize the -af option [03:45] <iive> try with the old command. [03:45] <afflicto> k [03:46] <iive> so far so good. [03:46] <afflicto> iive: I think it works =D [03:46] <afflicto> iive: so what's this avconv thing? another version? [03:46] <iive> nope. [03:46] <afflicto> iive: oh god no :( [03:46] <iive> it works for a few seconds. [03:47] <afflicto> iive: yea :( [03:47] <iive> try without threads 4 [03:47] <iive> x264 would use threads by default. [03:47] <afflicto> k [03:48] <iive> libav is fork of ffmpeg. they renamed the programs, aka ffmpeg-> avconv, ffplay->avplay etc.. [03:48] <afflicto> ok [03:49] <afflicto> still glitchy [03:49] <iive> but they left original ffmpeg in the package... that they don't develop or update. [03:49] <afflicto> huh ok [03:50] <iive> try -filter:a aresample=44100 [03:50] <afflicto> where should I put it? :P [03:50] <iive> before mp3lame [03:51] <afflicto> ok and remove -ar 44100 or? [03:51] <iive> i mean, before -acodec [03:51] <iive> yes, remove it. [03:52] <afflicto> so like this: -filter:a aresample=44100 -acodec libmp3lame [03:52] <iive> parameter position matters, maybe if you move -ar right after -i pulse. but let's test this later. [03:52] <afflicto> It says "flv does not support that sample rate" >.< [03:53] <iive> :O [03:53] <afflicto> it continues: "flv does not support that sample rate" [03:54] <afflicto> Oops, I mean "choose from (44100, 22050, 11025) [03:55] <iive> strange, it does work for me :O [03:55] <klaxa> try using a static build? [03:55] <klaxa> or are you using x11grab? [03:55] <afflicto> I am using x11grab yes [03:56] <klaxa> yet another avconv bug? [03:56] <iive> there is sound distortion. [03:57] <iive> and pulse audio is involved. [03:57] <iive> afflicto: put the -ar back and keep the aresample command. [03:58] <iive> i mean, option... [03:58] <afflicto> ok that works [03:59] <iive> so far so good.... [03:59] <iive> nope... [04:00] <iive> sorry, I got to go. [04:00] <afflicto> ok [04:00] <afflicto> well thanks for helping anyway! [04:00] <iive> i hope somebody would be able to help you. [04:01] <afflicto> I would guess someone has a solution.. somewhere [04:02] <iive> try to capture just the audio, without encoding it and streaming it. [04:02] <iive> just to make sure the input part is working. [04:02] <iive> n8 ppl. [04:23] <bencc> sacarasc: Na_Klar: lower resoultion works on the iPhone. thanks [04:27] <allengreen> how to pass "continuity counter" when I muxing mpeg2 ts file? [04:29] <allengreen> I am using av_write_frame and avio_open [05:03] <allengreen> hey, fellows! [07:42] <anshul> in nb_streams what does nb mean [07:43] <anshul> got answer, didnt knew wherther it is user or developer [07:43] <anshul> its number of [07:54] <procfile> Hello all [07:55] <procfile> I have a very high definition [probably 1080p] .mkv file sized at 1.5GB, I'd like to lower its quality to about 720p, and convert it to .ogg file, may I get some pointers for that? [08:30] <relaxed> procfile: ffmpeg -i input -vf scale=1280:-1 -c:v libtheora -b:v 2000k -c:a libvorbis -ac 2 -b:a 128k output.ogg [08:30] <procfile> I'll try that out, thanks! [08:33] <procfile> relaxed: btw, does the above command work with .avi files too? [08:34] <relaxed> change libtheora -> mpeg4 and libvorbis -> libmp3lame [08:34] <procfile> No no, I meant from .avi to .ogg. [08:34] <relaxed> oh, then yes [08:35] <procfile> Ok [09:55] <allengreen> hi [09:57] <allengreen> crystalhd.c:746: undefined reference to `DtsProcOutputNoCopy' [10:53] <allengreen> http://www.pasteall.org/47301 [11:06] <BlackBishop> Trying to convert from a .iso ( mpeg dvd ) to a mkv .. I did this http://pastebin.com/py5wSCvs .. but the audio gets left behind... any ideas on what I did wrong ? [11:12] <relaxed> BlackBishop: install a recent version of ffmpeg [11:14] <BlackBishop> hmm, relaxed http://packages.gentoo.org/package/media-video/ffmpeg says that's the latest stable in their mind .. which one do you recommend ? [11:15] <relaxed> ffmpeg.org says the latest stable is 2.2.2 [11:16] <JEEB> welcome to distro issues [11:16] <relaxed> gentoo shouldn't have this issue [11:17] <JEEB> new ffmpeg is most likely not yet installed by default because some packages just never get updated to new APIs [11:17] <JEEB> no, gentoo is /not/ free of issues [11:17] <JEEB> it makes some things simpler, but you still can't fix some shit that just hasn't gotten updated [11:17] <relaxed> BlackBishop: Try this http://johnvansickle.com/ffmpeg/ [11:17] <JEEB> plus gentoo has some weird shit going on with the default 'stable' setup [11:18] <JEEB> a lot of "By all means stable" stuff is not marked as such mostly because of "just because" [11:18] <JEEB> I mostly saw it a few years ago with firefox [11:18] <JEEB> they would just keep current - X version as stable [11:27] <JEEB> BlackBishop, basically update your ffmpeg and see if the package management tells you that anything breaks [11:27] <JEEB> I'd guess in most cases it shouldn't break anything [11:27] <JEEB> unless you have some package that depends on the old lavc/lavf APIs [11:29] <relaxed> BlackBishop: it could be because you're using an iso as input. [11:33] <BlackBishop> relaxed: well, the content are video_ts/*.VOB and IFO and BUPs [11:34] <BlackBishop> I'm trying with the one in the link you gave me now .. [11:37] <relaxed> dump the stream with, mplayer -dvd-device input.iso -dumpstream -dumpfile output.vob [11:37] <relaxed> then encode that with ffmpeg [11:37] <relaxed> er, mplayer -dvd-device input.iso dvd://1 -dumpstream -dumpfile output.vob [11:38] <relaxed> or use some other dvd ripper [11:38] <BlackBishop> dump: 12288 bytes written to 'output.vob'. [11:39] <relaxed> try just dvd:// [11:41] <BlackBishop> one sec .. I'l try with the vobs in the video_ts folder .. just the ones I need/want [11:41] <BlackBishop> The others are menus and stuff [11:50] <BlackBishop> alot better [11:57] <clever> i use cpdvd to copy a dvd, simple [14:18] <plepere> if I want to transcode a H.264 video to HEVC, should I simply use : ffmpeg -i input.mp4 -vcodec hevc output.mp4 ? [14:45] <spaam> plepere: replace -vcodec hevc with -c libx265 [14:45] <spaam> if you have support for libx265 [14:46] <plepere> isn't it in the main branch ? [14:47] <spaam> plepere: yes. but did you enable it when you built ffmpeg ? [14:48] <klaxa|work> maybe there is --enable-libx265? [14:49] <plepere> didn't do that. [14:49] <plepere> Thank you [14:50] <plepere> I usually just stick with the default configure. [14:51] <klaxa|work> the "default" configure doesn't enable any shared libraries afaik [14:51] <klaxa|work> s/shared/external/ [14:55] <plepere> is there a simple command to grab libx265 in there or should I find it myself ? [14:56] <klaxa|work> where is "in there"? [14:57] <klaxa|work> your distribution should offer a package called x265 which should include the libraries needed to build ffmpeg against them [14:57] <plepere> I've got a "x265 not found" error, so I guess I need to manually put x265 in shared/external [15:02] <BlackBishop> should ffmpeg -i vts_08_1.vob -i vts_08_2.vob -i vts_08_3.vob -vcodec libx264 -acodec ac3 -sn -preset slower Test.mkv create an mkv with all 3 vobs ? [15:03] <spaam> BlackBishop: concat ? [15:05] <spaam> BlackBishop: ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" .... [15:05] <BlackBishop> ow [15:05] <BlackBishop> ok [15:22] <plepere> ok, I've got libx265. Compiled it, installed it, put the .so in usr/lib. in my ffmpeg folder, I did a make dist-clean, configure --enable-gpl --enable-libx265. and I've got a invalid encoder tpye 'libx265' [15:23] <plepere> what did I miss ? [15:23] <JEEB> I will have to guess that the configuration step went through well, and then you compiled [15:23] <plepere> (I did do a make -j after the configure) [15:23] <JEEB> did the configure script actually finish successfully :P [15:23] <JEEB> and did the compilation finish successfully? [15:23] <JEEB> if they did, then check if you're running the correct binary [15:24] <plepere> the configure didn't say anything bad at me [15:24] <JEEB> and the configure script did output libx265 as an encoder? [15:25] <plepere> in the x265 build/linux folder, I have .a, .so and the x265 executable [15:26] <plepere> should I put them all in the external/shared ? [15:27] <JEEB> ... [15:27] <JEEB> did you get the encoder listed in configure script's output? [15:27] <plepere> it's in the external libraries lis [15:27] <plepere> list [15:28] <plepere> oh, might have forgotten --enable-encoder=x265 ? [15:29] <Milos_SD> Hello... I pulled ffmpeg from git just now, and tried compiling it, but it fails with this error: libavutil/opencl.c:329:12: error: no previous prototype for init_opencl_mtx [-Werror=missing-prototypes] [15:29] <Milos_SD> inline int init_opencl_mtx(void) [15:29] <Milos_SD> what can be the problem here? [15:29] <JEEB> plepere, no -- that's OK [15:29] <JEEB> --enable-libx265 should be enough [15:29] <JEEB> if it's successfully listed in the enabled libs listing [15:29] <plepere> JEEB, yes, libx265 is in the enabled encoders list [15:29] <JEEB> ok [15:29] <JEEB> now check if compilation finished correctly [15:29] <JEEB> if all went fine [15:30] <JEEB> if yes, then you are just running the wrong binary :P [15:30] <plepere> even after doing a make distclean ? [15:31] <plepere> ok, compiling done [15:31] <JEEB> now you can run the just compiled thing with dot-slash-ffmpeg [15:31] <JEEB> from the same directory [15:32] <JEEB> otherwise it's up to how you install the binary and if you install it at all [15:32] <JEEB> have fun [15:32] <plepere> same [15:33] <plepere> ./ffmpeg -i input.mkv -c libx265 output.mkv [15:33] <plepere> invalid encoder type 'libx265' [15:34] <JEEB> pastebin full ffmpeg output [15:34] <JEEB> and link here [15:34] <plepere> http://pastebin.com/r7kgQN7J [15:34] <smo_> hi who knwo how to compile a full libffmpegsumo.so for chromium with all latests codecs (xvid/avi,opus/h264/ac3/vp9...) ? [15:35] <JEEB> plepere, what about -c:v libx265 :P [15:35] <plepere> better. [15:35] <plepere> :D [15:36] <plepere> 5.7fps. not bad [15:36] <JEEB> yeah, libx265 is after all a video encoder [15:36] <JEEB> yeah, x265 is not slow with the defaults, too bad it just plain sucks :P [15:36] <plepere> what do you mean ? [15:36] <JEEB> quality-wise [15:36] <JEEB> you'll have to use preset placebo and max out refs and bframes to get some performance out of it [15:37] <JEEB> and even so, as soon as you leave the "so low bit rate that both kind of suck" area [15:37] <JEEB> x264 actually can start winning in some cases :P [15:37] <JEEB> in PSNR and SSIM x265 will most probably win, though [15:37] <plepere> isn't it possible to choose the wanted bitrate ? [15:38] <JEEB> yes, but I don't recommend 1pass ABR [15:38] <JEEB> the best quality rate control mode that is ABR is currently CRF with x265 [15:38] <JEEB> 1pass bitrate based ABR just sucks [15:38] <JEEB> and x265 has no 2pass [15:39] <plepere> ok [15:40] <JEEB> the last time I did testing was --preset placebo --crf 32 and then I maxed both refs and bframes to 16. Then I matched x264 by the resulting bit rate [15:40] <JEEB> for whatever reason preset placebo didn't max out those two [15:40] <plepere> and subjective quality, was HEVC better ? [15:41] <JEEB> the result was ~415kbps with 720p24, and they both more or less sucked. But yes, at very low bit rates x265 beats x264, but is also ~10x slower (than preset placebo) [15:42] <JEEB> Also there's no question about HEVC being better in the end, it's a newer format. What I'm trying to say is that unfortunately the use cases for HEVC encoders _right_now_ is rather small [15:42] <plepere> x264 has the benefit of having much more iterations for quality and performance optimizations. [15:42] <JEEB> naturally [15:42] <JEEB> x265 will probably always be more slow, though, since it is after all a more complex format (while the specification is simpler than AVC's in many places) [15:43] <plepere> well I'm working on just HEVC decoding, so it's not much of my problem. :) I've got the easy part. [15:43] <JEEB> yeh [16:08] <Voicu> what could be the reason for a stream not having a proper codec after I open it? [16:09] <Voicu> i.e. I'm following the tutorial here http://www.codeproject.com/Tips/489450/Creating-Custom-FFmpeg-IO-Context and I managed to get the stream open but the context doesn't have the codec setup (it's NULL) [16:09] <Voicu> reading packets works [16:11] <Voicu> they get proper DTS and PTS and probe_format produces the proper format [16:11] <Voicu> so I don't know why the codec itself is NULL [16:16] <Voicu> here's the code: http://pastebin.com/rEi01vdp [16:56] <Voicu> no idea from anyone? [16:57] <Voicu> I've been working on this streaming thing for almost a month [17:24] <BtbN> Is there a recommended ppa to get a recent ffmpeg version(>=2.1) to ubuntu? [17:30] <sacarasc> I think that's what's usually recommended, BtbN. [17:30] <BtbN> those are not debian packages [17:31] <sacarasc> Either that or compiling yourself, because of all the things that use libav as dependencies, it would be a lot of trouble to do it. :p [17:31] <BtbN> i need them as dependency in my ppa [17:31] <BtbN> there are ffmpeg ppas out there, a lot of them. [18:07] <tlhonmey> Hello, I'm trying to recover some video data where the header information was stored in one set of files and the actual stream data was stored elsewhere. The headers have since been lost, so I'm looking for a way to figure it out from the stream. Before I start on a script to just apply brute-force to it, I figured I'd ask if anyone here knows of any good analysis software. [18:07] <tlhonmey> Google is kind of failing me as there's far too much noise in every search I've been able to think of. [19:20] <I_can_FLY> I am trying to set up a livestream with ffserver and ffmpeg. The problem is I am trying to use webm, and I need to bun in subtitles, which are located in a mkv. I am not quite sure how to do this, but here is my attmept at it: https://gist.github.com/FLY-chan/2379a35023a424dd2538 and here is my output: https://gist.github.com/FLY-chan/2f0a25e2b967c28399af [19:21] <I_can_FLY> If someone could help, that would be great. [19:49] <smo_> is it possible to just build a ffmpegsumo.so from ffmpeg src ? [19:49] <smo_> (for node-webkit/chromium/chrome) [20:42] <Maverick|MSG> is there a master list of all pixel_formats that'll work with ffmpeg? [20:49] <Mavrik> Maverick|MSG, ffmpeg -pix_fmts will list them [20:49] <Maverick|MSG> thanks [20:49] <Mavrik> of course, each codec has it's own list of formats it supports [20:55] <Maverick|MSG> trying to follow the instructions here https://trac.ffmpeg.org/ticket/3614 [20:55] <Maverick|MSG> but having a hard time following what ceyoyos said in the last few replies [21:12] <Maverick|MSG> is he saying I should use -f image2pipe -vcodec rawvideo ? [23:10] <Maverick|MSG> anyone know how to calculate a frame_size based on the width and height of an image? [00:00] --- Thu May 8 2014

1 0

ffmpeg-devel.log.20140506
by burek 07 May '14

07 May '14

[01:46] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:5fc4c0097239: avfilter/af_bs2b: Fix incompatible pointer type warnings [01:46] <cone-764> ffmpeg.git 03Michael Niedermayer 07master:2baf1c8c6080: avcodec/snowenc: Fix "incompatible pointer type" warning [05:44] <cone-542> ffmpeg.git 03Michael Niedermayer 07master:141a84688f58: avcodec/mpeg4videodec: fix mcbpc error messages [05:44] <cone-542> ffmpeg.git 03Michael Niedermayer 07master:7db0f9bef16d: avcodec/mpeg4videoenc: fix alternative implementation of mpeg4_encode_dc() [09:44] <ubitux> i didn't know ioq3 was using opus [09:44] <ubitux> that's pretty fun [10:47] <j-b> Compn: as far as I know, I received, from Dolby, DTS, VIA, Bluray, One-Red, One-Blue, Philips, MPEG-LA [14:07] <plepere> BBB, do you have any feedback to give me about my patch ? [14:07] <BBB> not yet... I looked at earlier one and that was pretty good, so I don't mind if we just commit it [14:07] <BBB> let's look now [14:10] <BBB> what is the mcbuffer argument to the dsp functions? [14:10] <BBB> I guess that's gone right? [14:10] <BBB> so that's not an issue [14:11] <BBB> lol @ intptr_t vs ptrdiff_t (that was a big flamewar back then and you're doing exactly what I wanted to do - use intptr_t - and the decision was to use ptrdiff_t :-p) [14:11] <BBB> don't fix/change it, it's fine [14:11] <BBB> just funny :) [14:11] <plepere> ok ok. haha [14:11] <BBB> c patch is ok [14:11] <plepere> :) [14:12] <plepere> the ASM is my work, so it's the most interesting for me [14:13] <plepere> also, I've got the AVX2 machine back so I'll try to get on that if the current ASM is validated [14:13] <BBB> I think it looks very similar to what you send privately a while ago, I see most of the comments were addressed, e.g. the weird 12/24 px functions are gone, etc. [14:13] <BBB> so this is basically ok [14:14] <plepere> thanks. :) [14:14] <BBB> can we get it 1-2 instructions faster? sure, most probably, but this is a massive improvement, and the foundation (api/structure) is good, so no comments from me [14:14] <BBB> what's next? lf or idct? [14:14] <plepere> the LF is already done, I'll have to see if the author wants it patched. [14:15] <BBB> lf isn't in ffmpeg yet is it? [14:15] <plepere> no [14:15] <plepere> we have it in openHEVC [14:15] <BBB> ok, look forward to seeing that move over [14:16] <plepere> yes [14:16] <BBB> nice work, I really like this code [14:16] <plepere> but once we've got the base MC, the rest of the ASM will come smoothly [14:16] <BBB> do you guys have speed comparisons of other hevc decoders vs. your code? [14:16] <BBB> or planning to get that once all asm is done? [14:16] <JEEB> HEVC asm, nice [14:17] <BBB> seeing same-quality or same-size hevc vs vp9 or hevc vs x264 would be nice also [14:17] <BBB> but that's stuff for later I guess [14:17] <plepere> I don't have other decoder speeds. I heard that the competitors were surprised to see that we could do 2 4K@60fps decoding and display on a single PC. [14:17] Action: BBB run to work now [14:17] <BBB> bye [14:17] <BBB> well that's mt right? [14:17] <BBB> most decoders aren't mt [14:18] <BBB> (frame-mt, not wpp/tile/slice) [14:18] <plepere> yes, but it's still quite a burden on PCs to decode so much data [14:18] <BBB> right [14:18] <BBB> that's a good number, nice - I'll compare a little in a few weeks [14:18] <BBB> for now - work! [14:18] <plepere> tell me wxhen the patch is validated, the rest will follow shortly. [14:18] <plepere> thank you [14:18] <plepere> good lucj [14:19] <plepere> *luck [14:37] <cone-318> ffmpeg.git 03Lukasz Marek 07master:38b722e7d71c: gitignore: add tools/uncoded_frame [14:37] <cone-318> ffmpeg.git 03Matt Oliver 07master:ac9869ffb070: x86/mpegaudiodsp.c: msvc compilation error without sse/avx_external [15:47] <cone-318> ffmpeg.git 03Marton Balint 07master:a82e87618fc0: mpegts: remove uneeded buf_size check [15:47] <cone-318> ffmpeg.git 03Marton Balint 07master:9ba8debce994: mpegts: only emit new packets if data buffer exists [15:47] <cone-318> ffmpeg.git 03Marton Balint 07master:e2752de34458: mpegts: fix indentation after last commit [15:47] <cone-318> ffmpeg.git 03Marton Balint 07master:d7ca91498843: mpegts: factorize pes packet state reset function [15:48] <cone-318> ffmpeg.git 03Marton Balint 07master:a78274329c0b: mpegts: unref buffer in reset_pes_packet_state [15:48] <cone-318> ffmpeg.git 03Marton Balint 07master:6c5375100c4f: mpegts: always reset pes packet state on new packet [15:48] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:760a9ef45184: Merge remote-tracking branch 'cus/stable' [18:26] <cone-318> ffmpeg.git 03Mickaël Raulet 07master:83976e40e896: hevc: C code update for new motion compensation [18:26] <cone-318> ffmpeg.git 03plepere 07master:7a2491c436af: HEVC : added assembly MC functions [18:26] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:3b3db02f2e5c: avcodec/x86/hevcdsp_init: fix build on 32bit [18:26] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:7be230b5fa58: avcodec/x86/Makefile: remove duplicate line [18:29] <jamrial> michaelni: you missed my email about hevc asm [18:30] <JEEB> oh, someone wrote asm that works on both 32bit and 64bit? [18:30] <JEEB> kind of surprised [18:31] <nevcairiel> nah, its all 64-bit [18:31] <nevcairiel> although a bunch of the functions look like they could just work on 32-bit [18:32] <cone-318> ffmpeg.git 03Michael Niedermayer 07master:fc7d0d82017d: avcodec/x86/hevcdsp_init: fix SSE4 checks [18:33] <jamrial> Thanks [19:13] <j-b> move DSD-to-PCM conversion from libavcodec/dsddec into libswresample [19:14] <j-b> what does this mean? [19:20] <av500> j-b: its tech talk [19:44] <j-b> av500: sure, but aren't all decoders a converter to PCM? [19:48] <nevcairiel> one could argue that, but DSD is technically special, as its a true alternative to PCM and not just a way to store PCM compressed [19:52] <j-b> nevcairiel: meaning that I need libswresample then to decode it [20:01] <jamrial> seems that a big chunk of the hevc asm is either ssse3 or sse2 after all [20:01] <jamrial> hevcdps_init.c macros will need quite a lot of changes to get that working, lol [20:05] <jamrial> and like nevcairiel said, many can work just fine in x86 [20:07] <jamrial> just found one that's using registers xmm0 to xmm4, then xmm9, xmm14 and xmm15 for no reason [20:08] <jamrial> this is terrible for win64 [20:08] <jamrial> i may be missing something, though [20:08] <nevcairiel> a lot of the functions declare way high xmm usage counts, i've been wondering if they really all need that many [20:11] <jamrial> for example http://pastebin.com/Ywt7iWhh [20:14] <jamrial> it only needs seven xmm registers, but it's using x86_64 registers [20:50] <wm4> shouldn't this DSD thing be a codec? [20:52] <j-b> wm4: thanks :) [20:55] <kierank> http://www.euclidiq.com/euclidiq-announces-the-results-of-recent-mpeg-2-com… [20:55] <kierank> lol comparing to ffmpeg mpeg2 [20:55] <thardin> best encoder [20:55] <iive> what happened with x262? [20:55] <JEEB> lavc mpeg-2 might be good, but it sure means you'd need to know all of those non-documented options :D [20:56] <JEEB> that said, I've never tested it and have no idea about how good it actually is [20:56] <wm4> why can't it have sane presets? [20:56] <JEEB> because no-one cares about it I guess [20:56] <JEEB> just like the mpeg4 encoder [20:58] <Skyler_> I think it had the problem of having so many options and being so modular and trying to do everything [21:00] <kierank> iive: kind of works ok, some people use it [21:00] <kierank> needs some quality improvement though [21:22] <j-b> jb [21:22] <j-b> oops, wrong window :) [21:25] <kierank> Ed balls [21:26] <iive> people, stop pasting your passwords in this publicly logged and indexed channel. [21:26] <iive> ;) [21:47] <nevcairiel> i totally expected msvc to break with the new asm, oh well [21:47] <JEEB> sometimes surprises do happen [21:47] <j-b> iive: nice pwd :) [21:47] <nevcairiel> its not only msvc though, also mingw on windows [21:48] <nevcairiel> and only 64-bit, so it must be causing issues with the x64 abi [21:48] <nevcairiel> on the other hand, the asm isnt in 32-bit.... :P [21:48] <JEEB> :D [21:49] <nevcairiel> i would guess the number of regs is wrong on a bunch of functions, which only showed up on windows in previous cases [21:57] <Daemon404> :D [00:00] --- Wed May 7 2014

1 0

ffmpeg.log.20140506
by burek 07 May '14

07 May '14

[02:20] <storrgie> I want to capture audio playing in flash (firefox on fedora) is there a way to attach to the pulse playback? [02:21] <klaxa> ffmpeg -f alsa -i pulse out.ogg [02:21] <klaxa> then open pavucontrol to select the correct recording device for ffmpeg in the recording tab [02:21] <klaxa> change it to the monitor device of your soundcard [02:22] <storrgie> pavucontrol? is that the gnome default sound manager? [02:23] <storrgie> it appears to be listening on my microphone [02:26] <klaxa> yes now with the volume control of pulse (pavucontrol) you can change the input for ffmpeg [02:26] <storrgie> thanks klaxa [02:26] <storrgie> that actually is really awesome [02:26] <storrgie> I'm super impressed [02:27] <storrgie> I love you [06:43] <sony> Hi. [06:43] <Guest72492> Is there a way to use Ffmpeg to use an analog TV tuner as the input? [07:02] <kode54> does the Apple Intermediate Codec decoder support interlaced video? [07:02] <kode54> I tried using it to decode some interlaced video a month ago or so, and it had horrible ghosting artifacts around things [07:23] <kode54> nm [07:23] <kode54> maybe I used the wrong settings when I was deinterlacing this last time [07:24] <kode54> I made the mistake of trying to compress it to huffyuv, and apparently the huffyuv version is incompatible with whatever the mac version of avidemux bundles [07:27] <kode54> huh [07:27] <kode54> the white text in this video still seems to have dark shadows around it [07:49] <kode54> and it has some sort of banding effect every 8 scanlines or so [07:53] <kode54> I'll just upload the video so someone here can poke and prod it :D [08:03] <alsu> trying to transcode, I get "Unable to parse option value (null) as sample format" [08:03] <alsu> transcoding just the video (by passing -na) lets it work properly [08:05] <alsu> is there a way to get ffmpeg to just conitinue? liek by forcing the sample format, or telling it to just drop the audio, or whatever [08:05] <alsu> *continue [11:03] <kolizer> hello people [11:04] <kolizer> I need help [11:05] <kolizer> how overlay image on video? [11:05] <AGSPhoenix_> it might be possible with the overlay filter: https://ffmpeg.org/ffmpeg-filters.html#overlay-1 [11:06] <AGSPhoenix_> I don't know how though [11:22] <Mavrik> yes, overlay filter is used to make overlays :) [11:34] <kolizer> ok [11:36] <Caterpillar> I use FFMPEG on Fedora 20. Why in video streaming using a computer with a AMD Phenom II X4 965, FFMPEG uses SSE2 cpu capabilities instead of SSE4a provided by cpu? Could it be caused by compilation flags used in making the ffmpeg package? [12:01] <JEEB> Caterpillar, if you mean the stats that x264 outputs when you initialize it, those list human-written optimized code paths, so either there is no fitting code paths, or the CPU is noted as one where the optimized code path is not faster than the SSE2 code path, or it's a bug :) [12:04] <Caterpillar> JEEB: yes I meant the x264 output [12:23] <anshul> what does -of xml option mean [12:27] <ubitux> it's for ffprobe [12:27] <ubitux> and it allows dumping the infornmation in xml [12:27] <ubitux> it stands for "output format" [12:29] <anshul> thanks ubitux [12:39] <anshul> In one of report its written "regression since 29b9aee4", is this number represent commit number [12:42] <anshul> are they first 8 digit or last 8 digit of commit [12:48] <anshul> I got one matching commit with first 8 digit matching with that number,i am compiling code with that commit and with code previous to that commit [12:48] <kolizer> -vf "movie=logo.gif [logo];[in][logo] overlay=10:10 [out]" [12:48] <kolizer> what gif no movie? [13:09] <kolizer> as overlay gif on video? [13:19] <rsdrsdrsd> What is the best method to create a correct aspect ratio screenshot of a video [13:19] <rsdrsdrsd> I have different aspect ratio videos, with different par, sar and dar. Now I need a command so that it always outputs a correct screenshot with a fixed height [13:20] <rsdrsdrsd> I tried scale = ceil(height*dar):height:-1 but it seems that this doesn't always is good [13:21] <sunny_> hello sir [13:21] <sunny_> i want to run ffmpeg library at online server [13:21] <sunny_> how can i do it [13:21] <sunny_> at hostgator [13:22] <anshul> my guess was correct "regression since 29b9aee4" means first 8 byte of commit [13:24] <sunny_> Is there anyone help me regarding ffmpeg php running on hostgator [13:24] <anshul> sunny_, start with compiling ffmpeg for your server [13:35] <sunny_> how can i do it?? [13:36] <AGSPhoenix_> It's probably just be easier to download a precompiled binary [13:36] <AGSPhoenix_> It'd* [13:37] <AGSPhoenix_> http://ffmpeg.gusari.org/static/64bit/ffmpeg.static.64bit.latest.tar.gz [13:38] <sunny_> it will work? [13:38] <AGSPhoenix_> Probably [13:45] <sunny_> i'm not able to do that [13:45] <sunny_> is there any another [13:45] <sunny_> or guideline [13:45] <AGSPhoenix_> Why does it not work? [13:45] <AGSPhoenix_> That should be the easiest way [13:46] <sunny_> i dnt knw [13:47] <sunny_> it will work on the online server [13:48] <anshul> your server have linux or windows [13:48] <sunny_> linux [13:48] <sunny_> server [13:49] <anshul> 64 bit or 32 bit [13:49] <sunny_> 32 [13:51] <anshul> is your kernel 3.2.x+, [13:51] <anshul> to check uname -a [13:52] <sunny_> 3.2.45 [13:52] <sunny_> linux kernel 3.2.45 [13:52] <AGSPhoenix_> oh, I linked him the 64 bit version [13:52] <AGSPhoenix_> sorry [13:52] <sunny_> Architecture x86_64 [13:52] <AGSPhoenix_> so, it is 64 bit? [13:53] <sunny_> its a 32 i guess [13:53] <sunny_> its architecture x86_64 [13:55] <anshul> how do you guess? for 32 bit my uname show Linux daku_daddy 3.7.10-1.28-pae #1 SMP Mon Feb 3 14:11:15 UTC 2014 (c9a2c6c) i686 i686 i386 GNU/Linux :) [13:58] <sunny_> its x86_64 [13:59] <sunny_> ok may be i'm wrong now what i have to do [14:09] <anshul> have u downloaded binary file AGSPhoenix_ asked [14:10] <sunny_> yes i do [14:10] <anshul> if yes then run that the executable using ./ffmpeg [14:10] <anshul> why cany you do that [14:14] <sunny_> where i have to run? [14:51] <anshul> where you run uname -a [14:53] <anshul> you need to get in the directory where ffmpeg is there [14:53] <anshul> I think this is not the irc to teach linux, please ask help on linux question or type what you need on google [15:47] <ale345354> Hello. I upgraded Ubuntu from 13.10 to 14.04 (maybe I should have not). Couldn't find ffmpeg so I found your guide in /http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu/. I followed it 'till the end and all was ok, I converted some files perfectly. Now, after a couple of hours, I'm trying to do another conversion but it says again "Command not found". Why? Here's my pastebin log: http://pastebin.com/R48JJtKG [15:48] <ale345354> Notice I did nothing, not even restarted the system or gnome session. [15:51] <JEEB> probably your ffmpeg binary isn't in your PATH [15:51] <JEEB> also you generally shouldn't have to set threads with x264 [15:52] <JEEB> and no, the components in 14.04 are better than in 13.10, so that upgrade is fine by itself :P [15:53] <JEEB> and the reason it doesn't have ffmpeg by itself is because back in '11 or so it switched to libav from ffmpeg, and libav no longer has an ffmpeg binary. avconv is the binary that comes with libav [15:53] <JEEB> the ffmpeg you had up to 13.10 was so old anyways, you would've not wanted to use it :) [15:56] <ale345354> Hello JEEB and thanks for answerring me so quickly. What should I do to simply open a terminal, browse to the folder where ai have a file I want to convert it with ffmpeg and run a ffmpeg command? [15:56] <JEEB> where's your binary? [15:57] <JEEB> compare where your binary is against what `echo $PATH` (without the `s) tells you [15:58] <ale345354> I think this is the real problem: after I followed your guide and converted some files on my desktop, I deleted the folders the installation created on my desktop (they are still in my bin), one of those is ffmpeg-git-20140506-64bit-static [15:58] <JEEB> as long as you have the binary, that should be fine :P [15:58] <JEEB> just compare where the ffmpeg binary is [15:58] <JEEB> against your PATH [15:58] <JEEB> if it's not in PATH, you can open the dot-bashrc file in your home directory with a text editor [15:59] <JEEB> and add "export PATH=/where/your/ffmpeg/binary/is:${PATH}" to the end of it :P [15:59] <ale345354> wait a second, I'm not that good but I think the solution is at my hand. I wrote echo $PATH in my terminal and it said /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/games:/usr/local/games [16:00] <JEEB> yeah, that's your PATH [16:00] <JEEB> and I guess the directory that has your ffmpeg binary is not in it, right? [16:00] <JEEB> time to modify your dot-bashrc to add it to it :) [16:00] <ale345354> yes because it is on my desktop, inside the folder ffmpeg-git-20140506-64bit-static [16:01] <JEEB> and then open a new terminal, and the new PATH should be set [16:02] <ale345354> ok, I think I'll move the folder where my ffmpeg binaries are, into my /home [16:04] <ale345354> and now I'm adding "export PATH=/home/ale/ffmpeg-git-20140506-64-bit-static:${PATH}" without the " 's at the end of my .bashrc file which is in my /home [16:04] <ale345354> right? [16:06] <JEEB> yes, dot-bashrc [16:06] <JEEB> it shows up in the directory browser if you enable "show hidden files" [16:06] <JEEB> essentially stuff that starts with a dot is "hidden" [16:07] <ale345354> Ah-aaahhhh IT WORKS :DDDDDDDDDD [16:07] <ale345354> Thank you so so so so much JEEB [16:07] <JEEB> np [16:07] <JEEB> basics on all systems :) If something is not in your PATH, it has to be called with the full path to the binary [16:08] <JEEB> (or relative path, yes) [16:08] <ale345354> time to do some conversions :) thanks again and have a nice day [16:41] <damorrison> Hi [16:41] <damorrison> Anyone know a way of using an output files variables (i.e. height, width etc) in the output file name? [16:45] <Hello71> call ffprobe [16:50] <damorrison> thanks - I'll take a look [16:55] <watsonkp> Im having a problem transcoding to H.264 in mp4 programmatically. The resulting file will play properly in Chrome and VLC, but refuses to open in Safari/Quicktime. Ive tried to use the transcoding.c and remuxing.c examples, but the problem persists. [16:55] <watsonkp> Ive parsed out the atom structure of the resulting file and compared it to a functioning file from ffmpeg. The difference seems to be the avcc atom, particularly the SPS fields. [16:55] <watsonkp> This atom seems to be written with ff_isom_write_avcc during the av_write_trailer call, but I have so far failed to determine where the data it is writing is coming from, and how Im getting that wrong. [17:40] <f0lder> how do i mux h264 video in a mp4 container with aac audio in another mp4 container to a mp4 container? [17:41] <f0lder> i tried "ffmpeg -i audio.mp4 -i video.mp4 -acodec copy -vcodec copy mux.mp4" but its async [17:52] <c_14> Statically async, or dynamically async? [17:53] <c_14> If it's static, try -itsoffset [17:57] <f0lder> hard to say, the difference is huge and throughout the video [17:59] <f0lder> looks like the reason for that is the source vid being too fast however that happened [18:15] <c_14> How big is huge? [18:19] <another> c_14: very big [18:49] <Voicu> hello, I see that av_open_input_stream is not usable any more. What is the alternative? I'm trying to read a stream of in memory data [18:51] <Voicu> never mind [19:01] <JEEB> Voicu, you probably noticed how you create your own io context and the read/seek functions :) [19:01] <Voicu> yes yes :D [19:02] <JEEB> funny enough the first thing I coded with lavf/lavc contained just that [19:02] <JEEB> mapping Windows IStreams so that they can be read with lavf :D [19:03] <Voicu> yeah, there are tutorials for just that [19:03] <Voicu> my thing is more convoluted [19:30] <f0lder> just re-encoded the video with another encoder, that fixed the issue c_14 [19:30] <f0lder> thanks anyway [21:08] <RoyK> hi all. anyone that knows how I can use ffmpeg/avconv to garble audio files for (somehow) anonymizing them? [21:14] <RenatoCRON> RoyK, why can't just crypt and keep the file name equals? or I don't understand what you want to do. [21:15] <aphirst> Hi there, I just mentioned this to #archlinux and they thought it would be worth mentioning here too. Since updating to 2.2.2, some videos using FLAC audio (i.e. which use high CPU amounts) have audio skipping and popping. [continued] [21:16] <aphirst> I thought it was just an ffmpeg bug, because it worked again when I downgraded to 2.2.1; BUT if I build 2.2.1 from source I get the same problematic behaviour. [continued] [21:16] <sacarasc> RenatoCRON: I think they want to do voice distortion. [21:16] <aphirst> I believe the issue is in fact due to being compiled with GCC 4.9.0, which hit the main Arch repos recently. [21:17] <JEEB> aphirst, gcc 4.9.0 is broken [21:17] <RenatoCRON> sacarasc, oh! now it make sense! [21:17] <aphirst> But I'm in the sticky situation of not having one single person to blame. [21:17] <RoyK> RenatoCRON: not likely. I want to distort voice so that the file can be stored for later use without the person being identified again [21:17] <JEEB> blame your distro for switching to it [21:17] <aphirst> JEEB, yeah, I read that a lot of things didn't compile properly with 4.9.0 [21:17] <JEEB> yeah, but this was poked at gcc [21:17] <JEEB> and this doesn't seem to be a case of bad code [21:17] <aphirst> JEEB, Oh, have others mentioned this already? [21:17] <aphirst> (or something analogous) [21:17] <JEEB> yeah, when 4.9.0 was released @ GCC [21:18] <aphirst> specifically ffmpeg having issues? [21:18] <JEEB> yes, ffmpeg's flac gets miscompiled [21:18] <JEEB> gcc fixed it [21:18] <JEEB> but of course they don't fix their releases out of release schedule :P [21:18] <alsu> heaven forbid things work for the duration [21:18] <JEEB> ffmpeg has broken many times because of bad coding practices; this seems to be one of the few cases where the compiler side says they're wrong [21:19] <sacarasc> Isn't just about every x.x.0 GCC broken? [21:19] <JEEB> sacarasc, yeah -- they're pretty much betas [21:19] <JEEB> and any distro that uses a .0 release is bound to have trouble [21:19] <aphirst> GCC fixed it you say? So I "just" have to wait for a point release? [21:19] <JEEB> aphirst, or make your distro merge that patch :P [21:19] <JEEB> for the gcc package [21:19] <JEEB> http://gcc.gnu.org/bugzilla/show_bug.cgi?id=60902#c25 [21:20] <JY`_> don't you have a gcc-config for switching compiler ? [21:20] <JEEB> found the gcc ticket [21:21] <RenatoCRON> RoyK, I never used ffmpeg to do anything with audio, but I guess that you can get the audio, speed-up a little bit (like 1.4x),but keep the time the same, then run again, slowing the voice the same amount. [21:21] <aphirst> JEEB, I'll just downgrade ffmpeg from the repo for now, then [21:21] <JEEB> yeah [21:22] <JEEB> never update to .0 releases with GCC if only possible [21:22] <JEEB> both projects and GCC itself usually have problems [21:22] <JY`_> my teacher used to say "bad tool, bad worker" [21:22] <RoyK> RenatoCRON: that won't garble it. it's reversable [21:22] <RenatoCRON> when you speed up, save, the voice will be more 'thin, high voice', so you just need back to normal speed, and the voice will be disfigured. [21:22] <RenatoCRON> hmm, =/ [21:23] <iive> RoyK: you may want to try audacity [21:23] <JEEB> but yeah... kind of surprising that this time it was GCC that was to blame, generally you just find lulzy code where GCC starts breaking things, and you fix your code :) [21:23] <RoyK> iive: know any plugins there that can do it? [21:23] <aphirst> JEEB, well, Arch updates almost everything, my impression is that (outside of specific project's developers) a lot of the people finding early bugs are Arch or Gentoo users [21:24] <aphirst> so I suppose there's some utility in there being early adopters [21:24] <RenatoCRON> i vote to audacity too, even dunno what you need to do to really annoyn it. [21:24] <JEEB> true, but you should do that knowingly [21:24] <JEEB> so you don't end up with broken software :) [21:25] <iive> RoyK: audacity comes with lots of different plugins, give them a try. [21:26] <RoyK> just wanted to know if you knew which one could do it :P [21:26] <aphirst> JEEB, "knowingly" [21:26] <JEEB> aphirst, as in, if you updated to a .0 release of gcc, you must note that something will most probably break [21:27] <aphirst> JEEB, I did note it, which is why I'm not spazzing the fuck out :P [21:27] <JEEB> :) [21:27] <aphirst> I just thought I would report what I saw [21:27] <JEEB> also, #ffmpeg-devel is generally a better channel to poke regarding things like that [21:27] <JEEB> just for future reference [21:27] <aphirst> fair, without (slowly) loading channel lists its not always obvious that there's a -devel channel too [21:28] <JY`_> gentoo is still un 4.7 stable and 4.8 unstable btw [21:28] <RenatoCRON> RoyK, https://whyweprotest.net/community/threads/how-to-make-your-own-voice-filte… and http://wiki.audacityteam.org/wiki/Automation [21:28] <iive> maybe we should add gcc x.y.0 check in the configure? [21:28] <JEEB> JY`_, and that on the other hand is taking it way too seriously :D [21:29] <aphirst> I was generally quite looking forward to 4.9 since a couple of Fortran bugs which were irritating me were fixes [21:29] <aphirst> *d [21:29] <JEEB> I think I moved to 4.8 either during 4.8.1 or 4.8.2 [21:29] <JEEB> not sure [21:30] <JY`_> last 4.8 is even masked [21:30] <aphirst> 4.8.2 felt like it was around for *ages* [21:31] <JY`_> well multislot gcc installation is useful [21:31] <JEEB> JY`_, well gentoo's "stable" definitions are often rather loaded :) [21:31] <JEEB> heavy preference or just no-one caring [21:32] <RoyK> RenatoCRON: the intonation isn't changed. it's identifiable [21:33] <RoyK> RenatoCRON: that's what makes anonyfying audio so hard [21:33] <RenatoCRON> RoyK, well, so I dunno what to do! I will be grade to know how you solve this when you're done! [21:33] <JEEB> did you mean: anonymizing [21:34] <RenatoCRON> just for knowledge! [21:34] <RoyK> RenatoCRON: will post if I make it [21:34] <RoyK> it's so much in speech that's identifiable, it's almost impossible [21:34] <RoyK> you have the accent/dialect, the tone of voice, the intonation, the changes of level [21:35] <RoyK> et cetera [21:35] <RoyK> et cetera et cetera ad infinitum :P [21:39] <RenatoCRON> RoyK, now i'm only thinking and taking pity for the people who were on TV talking ~anonymously~, sitting back in a dimly lit room. [21:39] <pisto> how do I negate a condition in -filter select? [21:40] <RoyK> RenatoCRON: so do I [21:52] <tclarke> trying to mux 2 h.264 annex b streams into a single mpegts and I'm getting "first pts value must be set" [21:53] <tclarke> command line: ffmpeg -i stream1.h264 -i stream2.h264 -vcodec copy -an -map 0:v -vcodec copy -an -map 1:v -f mpegts -fflags +genpts out.ts [21:53] <kolizer> how to create news ticker and overlay him on video ? [21:56] <tclarke> kolizer: you could create an srt file and add it as a subtitle stream [21:57] <tclarke> also look here: http://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo [21:57] <kolizer> cool info [22:12] <kolizer> tclarke: it works) thanks [22:28] <kolizer> a how overlay gif on video? [22:32] <kolizer> help [22:37] <Caterpillar> http://superuser.com/questions/750519/ffmpeg-webcam-stream-to-youtube-drops… [22:37] <Caterpillar> if you have some suggestions.. [22:39] <sacarasc> Can you record from webcam to file? [22:40] <kolizer> a how overlay gif on video? [22:41] <sacarasc> You need to use the overlay filter, I think, kolizer. Which is beyond my understanding. http://trac.ffmpeg.org/wiki/FilteringGuide might be able to help you. [22:44] <kolizer> sacarasc: no( [22:45] <alsu> during transcoding, I get [abuffer @ 0x7fef0ac13080] Unable to parse option value "(null)" as sample format [22:46] <alsu> is there a way to force the sample format, or simply discard the audio when (and only when) this happens? [22:46] <alsu> the file does correctly transcode with -na, but I only want to do that if the sample format detection has failed [22:54] <kolizer> People help! how overlay gif on video? [22:58] <kolizer> -vf "movie=logo.gif [logo]; [in][logo] overlay=main_w-overlay_w:main_h-overlay_h [out]" [22:59] <kolizer> NO WORKS~! [23:02] <kolizer> Sorry it works! IT WORKS!!!!!!! AAAAAAAAA [23:07] <BlackBishop> for libx264 what does a -preset slower ofer ? ( compared with fast for example ) in terms of quality and filesize ? [23:08] <JEEB> everything slower than medium enables step by step various functionality that enables for better compression [23:09] <JEEB> you should generally just pick the slowest preset that still is fast enough for your use [23:10] <BlackBishop> so .. everything < medium is better compression .. that means lower file size and (same or good) quality, right ? [23:10] <BlackBishop> (same or better)* [23:10] <JEEB> well, one of those things, yes. I will just have to note that the same CRF value only means the same thing with the same settings [23:11] <JEEB> so you can't compare the results with the same CRF value across presets [23:11] <BlackBishop> I didn't set any crf :) [23:11] <JEEB> if you didn't set any rate control option, the default is crf 23 [23:12] <JEEB> which is the libx264 default [23:12] <JEEB> if you set some rate control option(s), the last of those gets used :P [23:13] <BlackBishop> nope, only set preset (medium) acodec vcodec and that's about it [23:13] <JEEB> then it's crf 23 [23:19] <BlackBishop> oh well, since it's a mpeg2 iso .. the source isn't so great at visual quality .. [23:19] <sacarasc> MPEG2 iso? So a DVD? [23:19] <BlackBishop> yeah [23:28] <BlackBishop> oh well, from 1.9G iso to 520M mkv much better .. with the normal preset .. and it didn't seem to loose much detail .. ( or at all infact ) [23:28] <BlackBishop> I'm trying the "slower" preset now see how that goes [23:28] <JEEBsv> well, just don't expect slower presets automagically giving you smaller file sizes [23:29] <JEEBsv> because the definition of the CRF value changes with settings used [23:29] <BlackBishop> ok [23:44] <kevix> newbie here: is this a place to ask about rtmpdump issues? or not, where? [23:44] <kevix> "ERROR: HandleCtrl: SWFVerification Type 2 request not supported! Patches welcome... [23:44] <kevix> " [23:45] <sacarasc> It is quite obvious, that type of verification isn't supported. [23:46] <kevix> I dont know if it might be related to a missing dependency or its code that is not yet written [23:47] <kevix> I was using get-flash-videos before with Hulu and it now gives this. [23:48] <sacarasc> Try #mplayer, as I think rtmpdump is an MPlayer side project. [23:48] <kevix> ok thanks [00:00] --- Wed May 7 2014

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