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March 2016

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ffmpeg-devel.log.20160315
by burek 16 Mar '16

16 Mar '16

[00:49:08 CET] <cone-128> ffmpeg 03Marton Balint 07master:67bef4cffa69: avcodec/aactab: do not use floats for constants [00:52:37 CET] <cone-128> ffmpeg 03Michael Niedermayer 07master:652173f63f02: avcodec/pthread_frame: Remove unused variable [01:02:40 CET] <cone-128> ffmpeg 03Marton Balint 07master:a740263d7e7b: avutil/dict: do not realloc entries when deleting a non-existing item [01:02:41 CET] <cone-128> ffmpeg 03Marton Balint 07master:c3c7a879baea: avutil/dict: add warning to docs about invalidating existing entries when adding a new entry [09:16:40 CET] <cone-567> ffmpeg 03Paul B Mahol 07master:a68d4bf2356b: avfilter/vf_waveform: add forgotten color and acolor filter to switch case [09:33:06 CET] <j-b> 'morning [10:36:36 CET] <cone-567> ffmpeg 03Matthieu Bouron 07master:31fe3c4d23aa: lavc/mediacodec: fix codec_name leak [10:36:37 CET] <cone-567> ffmpeg 03Matthieu Bouron 07master:03a6ed83b029: lavc/mediacodec: remove stray empty lines [11:42:10 CET] <nevcairiel> ubitux: all your x86_64 fate systems seem to have turned yellow recently, something weird is up there [11:43:05 CET] <ubitux> dammit [11:43:20 CET] <nevcairiel> not sure if a change caused this, i didnt see anything obviouss [11:43:22 CET] <nevcairiel> -s [12:35:35 CET] <ubitux> nevcairiel: it's weird, it seems the file checksum is fine [12:35:37 CET] <ubitux> d29d3e2c9d05cfb6d1d7c9ed08a83838 fate/fate-suite/aac/er_eld2000np_48_ep0.mp4 [13:17:00 CET] <nevcairiel> ubitux: and for the s16? [13:26:44 CET] <ubitux> nevcairiel: ah yeah, indeed... [13:27:23 CET] <ubitux> yeah one bitflip again [13:27:25 CET] <ubitux> i don't like this [13:27:35 CET] <nevcairiel> you sit under a funnel for cosmetic rays or something? [13:27:41 CET] <nevcairiel> cosmic* [13:28:08 CET] <ubitux> dunno :( [13:37:13 CET] <rcombs> comic rays [14:28:23 CET] <Daemon404> xmen joke could have been made, but i wasnt [14:28:25 CET] <Daemon404> im dissappointed [14:28:27 CET] <Daemon404> s/ss/s/ [14:33:45 CET] <Daemon404> so uh [14:33:50 CET] <Daemon404> what do we need a 64bit PRNG for [14:41:11 CET] <kierank> aac stuff I guess [14:42:41 CET] <atomnuker> well, AAC just requires the random numbers to be only somewhat random [14:43:06 CET] <atomnuker> you could probably replace the random numbers with just a static table of somewhat random numbers [14:43:30 CET] <atomnuker> and that probably won't really change much in terms of the output quality [14:43:56 CET] <atomnuker> though an integer RNG might be useful for some things I have in mind [14:44:18 CET] <Daemon404> we do have one. but it is not 64bit [14:44:55 CET] <atomnuker> yeah, a 32 bit one would be enough [14:45:17 CET] <nevcairiel> what does it use them for? noise generation? [14:45:25 CET] <nevcairiel> (aac that is) [14:46:03 CET] <Daemon404> all the uses outside of crypto i found in the DB ae for noise/encoding [14:46:14 CET] <Daemon404> and some coding stuff [14:46:29 CET] <Daemon404> the rest are all for fate tests [15:20:29 CET] <rcombs> if it's just for testing and doesn't need to be properly cryptographically secure (can't imagine why we'd need the latter internally), just stick together 2 32-bit randoms and call it a day [15:21:35 CET] <Daemon404> rcombs, we use it in some stuff like aes [15:21:44 CET] <Daemon404> maybe for hls or rtmpe [15:21:46 CET] <Daemon404> or something [15:21:58 CET] <Daemon404> i dont know how 'important' those are [15:22:08 CET] <rcombs> Daemon404: do we actually generate AES keys for any purpose other than testing, though? [15:22:20 CET] <Daemon404> good point [15:22:28 CET] <Daemon404> we have a lot of avoptions for passing in keys [15:22:52 CET] <Daemon404> i think it's mostly a case of "university student implements papers" [15:24:18 CET] <rcombs> I mean, we also use GCRY_WEAK_RANDOM when generating RTMPE keys [15:24:32 CET] <rcombs> but I don't think anyone actually gives a shit about RTMP crypto being strong [15:25:07 CET] <rcombs> it's pretty much just for show [16:11:44 CET] <wm4> lol [16:14:10 CET] <kierank> fair point [16:24:11 CET] Action: Compn runs [16:27:04 CET] <wm4> since when were you into comedy Compn [16:28:57 CET] <BtbN> what even is this report [16:29:32 CET] <BtbN> Kind of raises the general question what --help should provide. [16:29:49 CET] <BtbN> A short summary of how to do basic thing, or a full, searchable documentation [16:29:52 CET] <BtbN> +s [16:30:55 CET] <wm4> ffmpeg/ffplay/ffprobe -help is definitely extremely useless [16:31:05 CET] <wm4> it would be more helpful to output nothing at all [16:31:17 CET] <jkqxz> ffplay does suggest that it is grouped (offers the '-h topic' etc. options) like ffmpeg is, but doesn't actually support that and always dumps everything. [16:31:48 CET] <rcombs> step 1: install mpv [16:32:00 CET] <rcombs> step 2: https://mpv.io/manual/master/ [16:32:08 CET] <rcombs> afaik ffplay is an API example [16:32:24 CET] <Daemon404> i should hope not. [16:33:20 CET] <JEEB> lol [16:33:28 CET] <JEEB> well it's a proof of concept simple playback thingy [16:33:32 CET] <rcombs> well that's how I think of it :3 [16:33:56 CET] <wm4> it's sometimes useful for testing [16:34:08 CET] <cone-141> ffmpeg 03Michael Niedermayer 07master:7eedad946c30: avcodec/mediacodec_sw_buffer: remove redundant article [16:34:20 CET] <rcombs> and, like, if something works in ffplay and not in mpv then you can be reasonably sure it's an mpv bug and not a lavf/lavc one [16:34:59 CET] <rcombs> but if you're actually using it as a media player for non-test purposes I dunno what you're thinking [16:36:28 CET] <rcombs> jkqxz: btw, I'd expect a CoreImage hwcontext would be a useful thing [16:37:01 CET] <wm4> why would it need a hwctx? maybe a framectx at worst [16:37:41 CET] <rcombs> I guess I need to read up more on the relevant APIs [16:38:46 CET] <rcombs> but having lavfi be able to handle CoreImage formats well` [16:39:42 CET] <rcombs> things get a little bit awkward around conversions between CoreImage (which lavfi would need) and CoreVideo (which lavc would need) objects (which should largely be "rewrap a GPU surface handle") [16:39:45 CET] <wm4> on OSX you can use hardware surfaces without silly device contexts [16:40:22 CET] <wm4> aren't they both CVPixelBufferRef? [16:41:02 CET] <jkqxz> From the description it looked like it wanted a format and associated hwframes propagation. Unclear on the device part, but the current API requires one to assocuate the hwframes with. [16:41:21 CET] <rcombs> nah, gotta convert between CV* and CIImage [16:42:00 CET] <rcombs> `[CIImage imageWithCVPixelBuffer:&]` [16:42:08 CET] <rcombs> (oh did I mention the filtering is all obj-c) [16:42:42 CET] <rcombs> (probably less annoying than trying to do C++ work in lav* though) [16:42:43 CET] <jkqxz> The commentary in the associated bug does suggest it might want devices somehow to make the multi-GPU case work nicely, though there might be transparent magic already handling it. [16:43:27 CET] <rcombs> as I understand it, coreimage/corevideo involves a _lot_ of transparent magic [16:44:50 CET] <rcombs> like, applying a filter apparently produces a "promise" wrapped in a CIImage, which isn't "resolved" (i.e. filters all actually applied) until you do something with it that gets directly at the pixel data (or pass it to VT or whatever else) [16:45:19 CET] <eduarte_> hello [16:45:38 CET] <rcombs> and then the system figures out the most efficient path underneath it all [16:45:49 CET] <eduarte_> may i ask a question here ? [16:46:13 CET] <durandal_170> ask, don't ask to ask [16:47:08 CET] <rcombs> I've never seen a channel where people weren't OK with asking questions, but _were_ OK with asking if it's OK to ask a question [16:48:32 CET] <eduarte_> :D maybe its a dumb question tho,,,, I am trying to cut a video with ffmpeg, cutting it down to 30 milliseconds, tho the result always have more than 32 centiseconds, and always different sizes [16:48:58 CET] <thardin> use a video editor to edit video [16:48:59 CET] <eduarte_> is that influenced by the video quallity ? how can that be happening ? [16:49:20 CET] <wm4> eduarte_: #ffmpeg [17:48:46 CET] <cone-141> ffmpeg 03Martin Vignali 07master:69638517d159: lavf/segment: add increment_tc option [17:53:02 CET] <cone-141> ffmpeg 03Stefano Sabatini 07master:7725210e715d: lavf/segment: change type of increment_tc to BOOL [19:31:41 CET] <thardin> the hell.. I'm coding 720p webm at 3 fps [19:32:05 CET] <thardin> oh, vp9 [19:32:56 CET] <thardin> vp8 runs considerably faster. sort of [20:14:32 CET] <ethe> I'm not sure a JACK outdev would actually ever work, rate limiting the output, so that it doesnt sound distorted in JACK would result in the need for massive (or possibly infinite) buffers, and requiring -re isn't really a viable solution (not that it even works with -re due to the buffers not being able to stay full) [20:23:58 CET] <TD-Linux> thardin, also try setting -speed higher [20:24:20 CET] <TD-Linux> but yeah it's like using x264 in 2005 :) [20:25:23 CET] <nevcairiel> ethe: how is it different to any other realtime outputs [20:27:02 CET] <Daemon404> TD-Linux, in 2005, x264 had working ratecontrol, im pretty sure [20:27:06 CET] <Daemon404> libvpx is further back :D [20:27:41 CET] <wbs> doesn't libvpx-vp9 default to exhaustive search unless you set speed higher? [20:27:50 CET] <llogan> nobody actually knows how to use it [20:28:00 CET] <Daemon404> wbs, wut? [20:28:05 CET] <TD-Linux> wbs, no, it doesn't [20:28:13 CET] <Daemon404> llogan, so it's modeled after our mpeg4 encoder then [20:28:15 CET] Action: Daemon404 runs [20:28:16 CET] <TD-Linux> you're thinking of -best [20:28:40 CET] <wbs> TD-Linux: hmm, ok [20:29:02 CET] <TD-Linux> Daemon404, and it's true about the rate control. [20:29:06 CET] <Daemon404> ;p [20:29:12 CET] <TD-Linux> the closest thing to x264 CRF is 2-pass VBR in libvpx [20:29:57 CET] <TD-Linux> which is quite counterintuitive [20:30:44 CET] <nevcairiel> i suppose it sort-of makes sense [20:32:01 CET] <nevcairiel> but it would be nice if there was a decent 1-pass RC as well ;) [20:32:41 CET] <TD-Linux> xiphmont is writing one, we'll see how it goes. [20:33:13 CET] <Daemon404> hopefully better than theora's [20:33:14 CET] Action: Daemon404 runs [20:36:27 CET] <TD-Linux> Daemon404, well it's based on Theora's rate control :^) [20:36:47 CET] <Daemon404> oh dear. [20:37:37 CET] <TD-Linux> though I don't think theora's rate control was that bad. though releasing libtheora 1.2 would help [20:37:46 CET] <BBB> why bother [20:37:50 CET] <BBB> its theora... [20:37:58 CET] <BBB> lets release a new version of a vp3 encoder" [20:38:15 CET] <TD-Linux> BBB, people still use it. but yeah that's why no one has released it yet :) [20:38:28 CET] <BBB> people still use ffmpegs mpegvideo encoder [20:38:52 CET] <BBB> I would go one step further; if they still use libtheora to this day, quality is probably the last thing they care about [20:39:01 CET] <BBB> or rate control [20:39:02 CET] <Daemon404> TD-Linux, last i tested was like 10 years ago using ffmpeg2theora [20:39:05 CET] <Daemon404> for your reference. [20:39:38 CET] <Daemon404> BBB, you cant even use the freedom excuse now :P [20:40:10 CET] <BBB> its good that they merged all these scattered efforts into one project [20:40:23 CET] <BBB> I hope it goes somewhere [20:40:35 CET] <Daemon404> yes and that was called vpx... or was it daala... [20:40:38 CET] <Daemon404> oops. [20:40:47 CET] <nevcairiel> open-something-alliance? [20:40:57 CET] <TD-Linux> alliance for open media [20:41:10 CET] <TD-Linux> it's ok, I can't remember it half the time either [20:41:10 CET] <Daemon404> i still have my skeptical hat on for that [20:41:16 CET] <Daemon404> alliances formed by tech corps are uh... [20:41:16 CET] <Daemon404> :) [20:44:14 CET] <TD-Linux> Daemon404, you were about to say "super fun and a great time" right? [20:44:45 CET] <TD-Linux> this is much better than one company dropping codecs over a wall, though. [20:45:18 CET] <Daemon404> well uh... lemme know when that stops happening [20:45:33 CET] Action: Daemon404 stares at multiple codebases [20:50:07 CET] <TD-Linux> Daemon404, there is a "central" codebase now, currently on the webm infrastructure, which is basically libvpx with a bunch of stuff deleted [20:50:23 CET] <Daemon404> ic [20:52:16 CET] <TD-Linux> on a related note, for an external video codec, do you prefer a x264-style struct interface or a AVOption/opusctl style interface? [20:52:42 CET] <Daemon404> ive never used opusctl [20:52:53 CET] <Daemon404> but the struct style is a PITA [20:53:04 CET] <nevcairiel> a stable string-based interface is usually easier for ABI/API compat [20:53:10 CET] <Daemon404> thats why we have x264-params, which sets stuff via x264_param_parse [20:53:20 CET] <Daemon404> structs are an abi abi mightmare [20:53:24 CET] <TD-Linux> opusctl is ioctl-style in that it uses constants not strings [20:53:29 CET] <Daemon404> and also means you have to set everything manual [20:53:31 CET] <nevcairiel> we had to break x265 compat several times until we finally used a string interface for them [20:53:34 CET] <TD-Linux> but it's the same idea [20:53:45 CET] <wm4> Daemon404: it's interesting that MS seems to go pretty well with structs in its API [20:53:57 CET] <Daemon404> wm4, ms understands ABI [20:54:01 CET] <wm4> they just never try to extend structs (these days) [20:54:04 CET] <Daemon404> also it has some stuff to bend over backwards [20:54:08 CET] <Daemon404> also COM [20:54:23 CET] <wm4> yes, ISomeStuff5 [20:54:30 CET] <wm4> which derives from ISomeStuff4 [20:54:35 CET] <Daemon404> iirc they have special code to pack structs in special ways for certain apis [20:54:35 CET] <wm4> and adds 1 or 2 methods [20:54:38 CET] <Daemon404> or something [20:54:43 CET] <nevcairiel> MS has some flexible structs, which then typically have a size member as the first element in the struct, where the size defines the specific version of the struct you work with [20:54:55 CET] <wm4> nevcairiel: that seems to be an old convention [20:55:05 CET] <nevcairiel> (ie. those structs typically just grow, never change existing stuff) [20:55:09 CET] <wm4> haven't seen it in newer APIs [20:55:24 CET] <Daemon404> i personally abhor setting options in structs [20:55:25 CET] <nevcairiel> guess they just define stable structs these days [20:55:46 CET] <Daemon404> some apis make it very hard to know in between which calls to set stuff, or when accessing them is ok [20:55:55 CET] <Daemon404> i wonder whose api does that? [20:55:58 CET] <Daemon404> it is a myster. [20:55:59 CET] <Daemon404> y [20:56:26 CET] <TD-Linux> ok. libvpx currently has both styles of APIs, I wanted to delete one and now I know which :) [20:57:02 CET] <wm4> Daemon404: it scares me that Libav is in favor of adding public fields instead of accessors [20:57:52 CET] Action: Daemon404 puts fingers in ears, sings [21:13:45 CET] <dinux5> I get the following error while testing mlp encoder : [21:13:48 CET] <dinux5> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [21:14:04 CET] <dinux5> what should I do ? [21:17:10 CET] <ethe> nevcairiel: not sure really, which are the other realtime outputs, maybe I just havent done enough research [21:19:22 CET] <ethe> oh right. [21:19:49 CET] <ethe> nevcairiel: it's different because it *requires* a buffer, it sends data based on callback functions [22:17:20 CET] <thardin> my ffmpeg has no -speed [22:17:25 CET] <thardin> perhaps I should pull and recompile [22:18:03 CET] <Daemon404> he was probably talking about the tool from libvpx [22:19:44 CET] <nevcairiel> ffmpeg libvpx should have -speed, although it takes numbers, not tokens [22:19:52 CET] <nevcairiel> (and its a deprecated alias for cpu-used) [22:38:08 CET] <llogan> ffmbc is on github so the commits can now be publicly viewed. not sure how long it has been there [22:39:19 CET] <durandal_1707> who still cares about ffmbc? [22:41:25 CET] <thardin> hasn't ffmbc always been public? as in the source being available [22:42:04 CET] <llogan> IIRC just the source with no vcs [22:42:22 CET] <llogan> maybe i'm wrong. can't remember anymore. [22:46:26 CET] <wm4> wasn't ffmbc a problem because incompatible license [22:46:38 CET] <JEEB> yah, ffmbc's own stuff is GPL [22:46:45 CET] <nevcairiel> it uses the lgpl clause to upgrade to gpl [22:47:15 CET] <nevcairiel> i had an argument with a neckbeard once how that option only adds freedom [22:48:11 CET] <JEEB> :D [22:53:15 CET] <iive> this entirely depends on your definition of what free means :) [22:59:08 CET] Action: Daemon404 things of Freedom Isn't Free [22:59:11 CET] <Daemon404> thinks [00:00:00 CET] --- Wed Mar 16 2016

1 0

ffmpeg.log.20160315
by burek 16 Mar '16

16 Mar '16

[00:00:14 CET] <Dan0maN> my first example had a %05, this example has %03 [00:00:17 CET] <llogan> try the glob pattern. 3rd example that furq linked. hopefully it will order them correctly. [00:01:23 CET] <Dan0maN> k. ty [00:01:26 CET] <llogan> then add -pix_fmt yuv420p as an output option; otherwise it will try to preserve the chroma subsampling, but most non-FFmpeg based players only like YUV 4:2:0 [00:01:40 CET] <Dan0maN> is that a standard type of pattern descriptor? [00:01:48 CET] <llogan> and use -framerate instead of -r for image file demuxer [00:01:55 CET] <Dan0maN> k [00:02:16 CET] <furq> [mjpeg @ 0x5649cc0] Changeing bps to 8 [00:02:20 CET] <furq> where do i report this terrible bug [00:06:51 CET] <drv> the "Changeing" part? [00:06:58 CET] <furq> yes [00:07:33 CET] <llogan> was just about ti fix it, but some crazy old lady entered my office. [00:07:40 CET] <llogan> go away [00:10:19 CET] <llogan> ok. gave her coworkers number... must have been fixed before. not in git master. [00:11:25 CET] <daslicht> not ffmpeg related but maybe anyone like to help [00:11:31 CET] -daslicht:#ffmpeg- http://stackoverflow.com/questions/35999691/is-there-any-icecast-cli-source… [00:11:31 CET] -daslicht:#ffmpeg- http://stackoverflow.com/questions/35999718/is-there-any-icecast-cli-source… [00:12:52 CET] <llogan> yeah, 3885ef0 [00:13:45 CET] <Dan0maN> worked great. ty llogan. [00:14:52 CET] <llogan> Dan0maN: also add "-movflags +faststart" if your viewers will watch via progressive download [00:15:24 CET] <llogan> daslicht: those don't look like programming questions. SO is for programming questions. [00:15:41 CET] <daslicht> which is teh correct place ? [00:16:12 CET] <daslicht> stack exchange ? [00:16:16 CET] <llogan> within SE probably Super User or video.stackexchange.com [00:16:44 CET] <llogan> you can get the questions migrated instead of making crossposts which is annoying [00:17:00 CET] <daslicht> i move them [00:17:08 CET] <daslicht> video ? [00:17:16 CET] <daslicht> adio [00:17:17 CET] <daslicht> audio [00:18:06 CET] <daslicht> hm theer is no audio [00:20:41 CET] <daslicht> no idea [00:20:57 CET] <daslicht> I leave them there until we found teh correct place to migrate [00:58:34 CET] <cbsrobot> daslicht: tried using python ? [01:02:45 CET] <daslicht> nope [01:02:55 CET] <daslicht> I have very little expericne with python [02:49:46 CET] <nadermx> When running ffmpeg if I use the -threads option should I set it to number of threads on server (4) or 0? [02:51:39 CET] <c_14> 0 is the default [02:52:04 CET] <nadermx> but does 0 use more than one thread if its available? Or does 0 use just 1? [02:52:22 CET] <furq> it depends on the encoder [02:52:27 CET] <c_14> and decoder [02:52:36 CET] <nadermx> libmp3lame specifically [02:52:42 CET] <furq> i don't think you can multithread lame [02:52:44 CET] <c_14> If they support threading without additional options it should do it automatically. [02:53:05 CET] <furq> if you're encoding multiple files you can do them in parallel with something like xargs or parallel [02:53:59 CET] <nadermx> I'm calling ffmpeg as a subprocess in a golang program [02:55:08 CET] <nadermx> Just trying to figure out how to make it go faster I suppose for longer songs [02:55:33 CET] <c_14> Get a faster cpu [02:56:02 CET] <c_14> Or try cutting the file into parts and encoding them in parallel and then concatting them [03:03:28 CET] <nadermx> ok thank you [03:46:46 CET] <wellen> hi [03:47:46 CET] <wellen> can i check this result... ffmpeg -i test720pav.mp4 -c:a copy test192ka.m4a [03:48:48 CET] <wellen> is that good for just getting the audio from the input file eg quickest method? [03:52:56 CET] <c_14> add -vn and -sn [03:53:05 CET] <c_14> or -map 0:a:0 [03:53:11 CET] <c_14> Assuming you only want 1 audio stream [03:53:12 CET] <johnnny22-afk> In ffserver, what happens to a Feed's File once it reaches the FileMaxSize ? [03:53:16 CET] <c_14> or -map 0:a if you want all of them [03:53:42 CET] <c_14> johnnny22-afk: I assume it's a ring buffer, but I don't know [03:53:45 CET] <johnnny22-afk> I'm also pondering about how that Truncate option works.. it's not too clear. [03:54:12 CET] <wellen> the input file has audio adn video but only 1 of each and i am just trying to get it to output the ony audio it has as an m4a [03:54:56 CET] <c_14> Then either -vn or -map 0:a. In this case they'll do the same thing [03:55:33 CET] <wellen> based on my first attempt is that why it is encoding the video (because i need -vn ? [03:55:39 CET] <c_14> yes [03:56:03 CET] <wellen> thanks c_14 - i used to be a lot sharper than this...then i got ill doh! [04:01:25 CET] <wellen> im still pondering a ffmpeg gui btw [04:03:47 CET] <johnnny22-afk> c_14: a ring buffer on disk ? [04:04:34 CET] <wellen> im done, night all thanks again c_14 [05:50:49 CET] <jookiyaya> what is better audio codec ? vorbis or opus? [06:17:48 CET] <sling_> Hi [06:18:12 CET] <sling_> i am running the ffmpeg shell command using the java process builder [06:18:32 CET] <sling_> when ffmpeg fails with error how do i get the error code [06:18:43 CET] <sling_> bcaz in the log i dont find the error code [06:20:02 CET] <sling_> ProcessBuilder pb = new ProcessBuilder(command.split(" ")).inheritIO(); [06:20:17 CET] <sling_> final Process p; try { p = pb.start(); TransProgressThread transProgressThread = new TransProgressThread(transcodingContext); transProgressThread.start(); p.waitFor(); } [06:24:09 CET] <sling_> [aac @ 0000000002b71c80] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [aac @ 0000000002b71c80] Alternatively use the non experimental encoder 'libvo_aacenc'. [06:24:28 CET] <sling_> how do i can get that there is an error happened using above error [07:13:35 CET] <relaxed> sling_: if ffmpeg returns anything other than zero, there was an error. [07:14:51 CET] <johnnny22-afk> can ffmpeg skip holes in a sparse file ? [07:53:00 CET] <johnnny22-afk> if i wanted ffmpeg to simply append to a output file that already exists, should I output to stdout and >> to the file ? [07:53:16 CET] <sling_> @relaxed THeN I NEED TO PARSE THE LOGS TO GET THE ERROR STATEMENT? [07:53:47 CET] <sling_> do you have a list of error statements? [07:53:57 CET] <sling_> based on that i can handle errors differently [07:59:48 CET] <relaxed> sling_: you'll have to check the source [08:16:23 CET] <johnnny22-afk> lets say ffmpeg fails to read the input stream, is there any tricks to resume consumption of the input when reinvoking ffmpeg again after ? [08:16:53 CET] <johnnny22-afk> to resume at the same location [09:54:57 CET] <johnnny22-afk> what would be the best way to throttle ffmpeg externally ? [09:55:27 CET] <johnnny22-afk> ideally throttle it's bandwidth usage. [09:56:13 CET] <JEEB> depends on what you want to do [09:56:26 CET] <johnnny22-afk> I was thinking of using iptables possibly [09:57:35 CET] <johnnny22-afk> or possibly achieve limiting it's output, thus impacting it's input !? I'm guessing. [10:00:02 CET] <JEEB> output limiting is possible with most sane video encoders with maxrate and bufsize [10:04:05 CET] <johnnny22-afk> right, right, but i'm using ffmpeg to consume an online stream and writing it to disk with -c copy. I might even have two instance of that running. Ideally, I'd want to be able to set the maximum bandwidth that whole group of instances of ffmpeg are using as a group. [14:46:12 CET] <zuloyd> hi [14:47:39 CET] <zuloyd> Is it possible with ffserver to record a live stream, play it back via HTTP and be able to seek inside it? [14:48:06 CET] <zuloyd> I managed to get it to record an RTSP stream, and this RTSP stream is forwarded to a WEBM stream accessible with the built-in HTTP server [14:48:10 CET] <zuloyd> however no seeking is possible [14:48:23 CET] <zuloyd> I'd like to be able to jump back in time [14:54:30 CET] <flux> so it say on the webs that -crf drops video quality for high-action scenes, because it's difficult to see what's happening so clearly anyway [14:54:47 CET] <flux> however, my goal would be to use this for security video footage, where this might be counter-productive [14:55:23 CET] <flux> is my only recourse to use the x264 library interface directly to adjust the quality manually? or perhaps use the same quality value instead of trying to increase it for high-motino scenes? [14:56:10 CET] <J_Darnley> You could try constant quantiser as an alternative [14:57:39 CET] <flux> right, I suppose that's the only "pre made" solution? [14:58:23 CET] <J_Darnley> If you want decent frame-by-frame quality then yes (or just go for high quality) [14:58:58 CET] <flux> though for pictures that don't have a lot of going on (ie. only camera noise), I would prefer not to waste space :) [14:59:34 CET] <J_Darnley> const. quant. isn't going to waste much in those areas [15:00:01 CET] <J_Darnley> inter-frame prediction will reduce the cost of static scenes and areas towards nothing [15:00:05 CET] <flux> is the bitrate adjustment algorithms smart enough to understand it is compressing noise? [15:01:09 CET] <J_Darnley> x264 does have a perceptual feature to reduce the cost of high noise (usually film grain or dithering) [15:01:19 CET] <flux> nice [16:43:58 CET] <eduarte> hello [16:44:18 CET] <eduarte> am i alone here ? [16:44:34 CET] <eduarte> can i make a question ? [16:48:00 CET] <J_Darnley> Yes. Just ask your question. [16:49:34 CET] <eduarte> I am trying to cut a video with ffmpeg, cutting it down to 30 milliseconds, tho the result always have more than 32 centiseconds, and always different sizes [16:49:40 CET] <eduarte> is that influenced by the video quallity ? how can that be happening ? [16:52:26 CET] <kepstin> eduarte: are you re-encoding or using copy mode? [16:53:22 CET] <kepstin> keep in mind also that 30ms is (depending on the frame rate) only 1-2 frames, or sometimes not even that [16:55:46 CET] <eduarte> i mean -i videofile [16:57:33 CET] <eduarte> http://pastebin.com/LNMSErJ3 [16:58:08 CET] <J_Darnley> What's this about fractions of a second? [16:58:14 CET] <kepstin> eduarte: with that command, you're asking for 30 seconds of video... [16:58:16 CET] <J_Darnley> That is 30 seconds [16:58:22 CET] <eduarte> in the case it is not even 1-2 frames, it wont work ? [16:58:30 CET] <eduarte> oh rly ? [16:58:39 CET] <kepstin> yes, it's HH:MM:SS [16:58:42 CET] <J_Darnley> HH:MM:SS [16:58:46 CET] <eduarte> and can i add more : ? [16:58:50 CET] <eduarte> hh:mm:ss:cs:ms ? [16:58:55 CET] <kepstin> no, but you can use fractional seconds with a . [16:59:02 CET] <kepstin> so -t 0.030 is 30ms [16:59:09 CET] <eduarte> oh ok [16:59:14 CET] <eduarte> thank you :D [17:01:10 CET] <eduarte> another question is: [17:02:07 CET] <eduarte> imagine i join 1000 videos with 30 ms each with ffmpeg, does the quantity will affect the video performance? create any glitch or somth ? [17:03:04 CET] <kepstin> eduarte: you're probably going to have rounding errors that add up, due to the fact that 30ms is close to, but not quite, the same length as a frame. [17:05:04 CET] <eduarte> hm [17:05:42 CET] <eduarte> is there someway to be sure the minimum i can cut from a video with ffmpeg ? [17:06:03 CET] <J_Darnley> No [17:06:05 CET] <J_Darnley> Depends on the video format [17:06:12 CET] <eduarte> i see [17:06:13 CET] <J_Darnley> and the framerate [17:06:19 CET] <kepstin> eduarte: depends mostly on your desired output framerate, I think [17:06:34 CET] <J_Darnley> You can't cut a P or B frame out of the middle of a video [17:06:54 CET] <kepstin> J_Darnley: looks like eduarte is doing a re-encode, so grabbing an arbitrary frame is possible [17:06:58 CET] <eduarte> than again, i would still have a problem if i want to cut like 1 and half frames right ? [17:07:14 CET] <J_Darnley> What? The use the image2 muxer and save every frame with ease. [17:07:27 CET] <kepstin> eduarte: i'd recommend that you pick a framerate for the final result, then work based on frames rather than time [17:07:42 CET] <eduarte> hm i see [17:07:47 CET] <furq> eduarte: use -frames:v if you want a particular number of frames [17:08:58 CET] <eduarte> ok I will try that [17:10:45 CET] <eduarte> just to give you context, what im trying to do is, 1- i have videos of me playing each note of an instrument ,,, 2-i am interpreting midi files and generating with ffmpeg a final video cutting and concatenating the videos of each note, the reason of those questions is because i need to get the timings right :), and i get each notes timing in milliseconds [17:11:44 CET] <J_Darnley> I hope you're using a 1000fps camera then [17:11:44 CET] <eduarte> the final goal is to have an output of you playing a music given a midi file, kinda weird but soudns fun :) [17:12:09 CET] <eduarte> not at all :D [17:12:37 CET] <eduarte> i will see the max precision i can get from the camera in time and reduce the minimum note time to that [17:12:42 CET] <J_Darnley> Then good luck with 40/33/16 ms precision. [17:13:21 CET] <kepstin> yeah, you're probably just gonna have to make your conversion program drop frames to try to keep the video approximately in sync [17:14:43 CET] <eduarte> thanks [17:14:46 CET] <osense> hi [17:15:08 CET] <osense> is there some recent issue with the 'copy' codec? [17:15:16 CET] <osense> I'm getting 'Protocol not on whitelist 'none'!' [17:15:47 CET] <osense> when trying to concatenate videos with the demuxer [17:16:04 CET] <kepstin> osense: that's complete unrelated to the copy codec, but it rather a demuxer/format issue [17:17:36 CET] <osense> hmm, any idea what could be wrong? is it illegal to try to concat a single video? [17:17:38 CET] <osense> http://pastie.org/10760945 [17:19:10 CET] <kepstin> osense: you're hitting a security protection. by default, ffmpeg only allows playlists (and concat is a playlist, basically) to read local files if the playlist is itself a local file [17:19:22 CET] <kepstin> but you're sending the playlist on stdin, which isn't a file [17:20:41 CET] <kepstin> to override that, you can use the "-safe 0" input option, which disables the security check [17:21:33 CET] <kepstin> (or at least, I think that's the problem? could be wrong) [17:23:11 CET] <osense> hmm, nothing's changed [17:23:26 CET] <osense> (input option means after -i right?) [17:23:32 CET] <kepstin> no, before [17:23:46 CET] <osense> still nothing [17:23:51 CET] <kepstin> hmm. [17:25:32 CET] <kepstin> oh, the -safe option only prevents access to e.g. absolute paths. it's probably not needed here, actually [17:25:41 CET] <kepstin> I have no idea how to override that protocol issue [17:25:54 CET] <kepstin> (aside from writing the playlist to a file, which would probably work) [17:29:49 CET] <kepstin> ah, I was looking at the help for the wrong ffmpeg version [17:29:58 CET] <kepstin> try adding "-protocol_whitelist file" as an input option [17:43:56 CET] <osense> kepstin: now I get Protocol not on whitelist 'file'! [17:44:52 CET] <osense> bbl [17:53:23 CET] <mr_pinc> Hello I am trying to follow this example - https://trac.ffmpeg.org/wiki/Scaling%20%28resizing%29%20with%20ffmpeg#no1 to convert a 4:3 image into a resized 16x9 image with a matined aspect ratio. Thing is even though I specify 16x9 dimensions, the resulting image is still 4:3 - so if my input image is 1024x768 and I run the following scale="'if(gt(a,16/9),403,-1)':'if(gt(a,16/9),-1,2227)'" [17:53:23 CET] <mr_pinc> output_320x240_boxed.png - the result is an image that is 303x227 not 403x227 [18:06:45 CET] <mr_pinc> So the image does maintain the aspect ratio but it does not fill in the desired 16x9 (403x227) [18:27:53 CET] <J_Darnley> You need to pad if you want black bars [18:28:36 CET] <mr_pinc> Thank you - I just got it working using this suggestion - http://stackoverflow.com/questions/8133242/ffmpeg-resize-down-larger-video-… [18:39:11 CET] <cimbakahn> Hello! [18:39:20 CET] <cimbakahn> Does anyone know where there is a list of all the formats that mpv plays? [18:39:39 CET] <J_Darnley> ffmpeg -formats? [18:41:10 CET] <cimbakahn> You know like does it play .avi, .ogg, .mp4, .wax etc.. [18:41:59 CET] <cimbakahn> I know it plays ffmpeg formats, but what are those? [18:44:02 CET] <furq> cimbakahn: http://sprunge.us/CSBW [18:44:28 CET] <furq> note the first line [18:45:16 CET] <cimbakahn> Thank you furq! You are so sweet! [18:45:36 CET] <furq> i guess `ffmpeg -formats ^\ D` is a better answer [18:45:39 CET] <furq> er [18:45:45 CET] <furq> i guess `ffmpeg -formats | grep ^\ D` is a better answer [18:46:48 CET] <cimbakahn> HAHAHA! [18:50:02 CET] <J_Darnley> I doubt there is an E without a D [18:51:15 CET] <J_Darnley> oh well there are some [18:52:14 CET] <J_Darnley> oh now most of those come from slight naming differences [19:15:47 CET] <note> Hello there! I wish for ffmpeg transcoded videos from the cameras of Android and iPhones to be viewable in each other's devices, but so far whatever arguments I use to ffmpeg, the resulting videos are only viewable on the same phone/platform the video was captured from. Might you know what is the problem? [19:17:06 CET] <note> I should add that I would like them to be viewable in the HTML5 WebView component of each other's platform. [19:24:00 CET] <pzich> note: I unfortunately don't know too much about what might be causing that, but whoever does is going to want to see your commands and their full output, i.e. [19:27:00 CET] <note> Ok thanks, the commands ( http://pastebin.com/cQgNH1kz ) have always succeeded. It's just that the resulting videos don't work on the opposite platforms. [21:23:31 CET] <jookiyaya> what is better audio codec ? vorbis or opus? [21:25:30 CET] <kepstin> opus. [21:27:57 CET] <jookiyaya> why are ffmpeg people so bias to opus [21:28:15 CET] <furq> why do you keep asking the exact same questions and then complaining about the answers [21:29:00 CET] <jookiyaya> i am not complaining [21:29:18 CET] <J_Darnley> Is it not true that opus is better? [21:29:31 CET] <kepstin> based on listening tests that i've seen, opus provides better quality at the same bitrate as vorbis. therefore it's a better audio codec. [21:30:00 CET] <jookiyaya> j_darnley it seems like people here are very pro opus, which is not the same case for different #channel [21:30:59 CET] <furq> why are you listening to what people in #handbrake have to say [21:31:10 CET] <bencoh> haha [21:31:13 CET] <jookiyaya> i like to listen to everybody [21:31:22 CET] <jookiyaya> bencoh what is funny? [21:32:09 CET] <bencoh> the idea implied by furq's question :) [21:32:40 CET] <jookiyaya> furq people in #handbrake seem to respect people in #ffmpeg [21:32:58 CET] <furq> that's nice [21:36:40 CET] <jookiyaya> kepstin what about vorbis vs aac [21:37:10 CET] <furq> 02:53:25 ( furq) jookiyaya: http://listening-test.coresv.net/s/scores_by_tracks_en.png [21:38:00 CET] <jookiyaya> there is no ffmpeg-aac on that list [21:38:14 CET] <furq> i wonder if that guy who disagrees has been reading those garbage "soundexpert" listening tests with artifact amplification [21:38:17 CET] <jookiyaya> i don't think ffmpeg support qaac [21:38:37 CET] <kepstin> jookiyaya: qaac is quicktime's encoder, not open source. [21:38:49 CET] <furq> nobody can say with any authority how good ffaac is because nobody has done a reasonably wide-scale listening test [21:39:04 CET] <furq> if i had to guess i'd say it's probably a bit worse than vorbis but not enough to be noticeable at 128k [21:39:08 CET] <kepstin> jookiyaya: compared to fdk_aac, the (arguably) best aac encode ffmpeg supports, I think vorbis is slightly worse? [21:39:18 CET] <kepstin> hard to tell for most people yeah [21:39:33 CET] <furq> fdk was slightly worse than qaac in 2011 [21:39:43 CET] <furq> i've not seen any more recent tests [21:40:08 CET] <kepstin> I mean, if you're running all of these codecs at well above 128kbit (stereo), you're probably not gonna be able to tell them apart [21:40:31 CET] <furq> i don't see any reason to encode audio unless you need player compatibility [21:40:35 CET] <kepstin> (you'd probably be able to notice vo-aacenc tho, even at those bitrates. Don't use v-aacenc) [21:40:39 CET] <furq> in which case your choice is determined by the player [21:41:11 CET] <jookiyaya> is v-aacenc really that bad? [21:41:19 CET] <bencoh> (iirc ffmpeg dropped support anyway :) [21:41:21 CET] <furq> vo is even worse than faac [21:41:26 CET] <furq> and yeah vo isn't even in 3.0 anyway [21:41:34 CET] <jookiyaya> is vo worse than mp3? [21:41:34 CET] <bencoh> 3.x already? pfew [21:41:55 CET] <bencoh> oh right, it's been released [21:42:12 CET] <furq> someone should tell freebsd that it's been released [21:42:32 CET] <jookiyaya> people these days have smartphones now and they all have players like VLC< so why is player compatiblity an issue? [21:42:34 CET] <TD-Linux> this is the most recent listening test with ffmpeg's aac encoder https://hydrogenaud.io/index.php/topic,109716.0.html [21:42:45 CET] <kepstin> but yeah, if your source is e.g. a dvd, there's probably a 192-256kbps stereo ac3 track which you'd just want to passthrough. [21:43:12 CET] <furq> TD-Linux: that's from one guy isn't it [21:43:22 CET] <kepstin> that's just one listener, and an older version of the encoder, yeah [21:43:29 CET] <bencoh> jookiyaya: because people also have smartTVs/settopboxes/phones with hardware/DSP decoders [21:43:34 CET] <furq> and yeah that's a pre-3.0 release [21:43:44 CET] <TD-Linux> furq, yes, and it's out of date too. [21:43:57 CET] <bencoh> and in some cases you just have to use what your devices can play [21:44:23 CET] <furq> it still does better than faac [21:44:31 CET] <jookiyaya> bencoh you cannot install player like vLC on smarttv/settopbox ? [21:44:58 CET] <furq> jookiyaya: there's no opus/vorbis support in mp4 and no aac/mp3 support in webm [21:45:04 CET] <bencoh> even if you could, you often cant decode in software anyway [21:45:09 CET] <BtbN> VLC decodes in software, you don't realy want that on mobile devices. [21:45:21 CET] <furq> apparently opus is coming to mp4 but it's not going to be much use to anyone who hasn't got an up-to-date player [21:45:23 CET] <BtbN> And on SmartTVs the CPU might not be up to the task to decode anything [21:45:31 CET] <TD-Linux> in 2016 audio is nearly always decoded in software. [21:45:33 CET] <jookiyaya> btbn why? what's wrong with software decodes? [21:45:47 CET] <BtbN> They use way more energy than a DSP doing it for free. [21:45:48 CET] <bencoh> BtbN: software audio decoding on mobile isn't that bad actually compared to some hw pipeline [21:45:57 CET] <TD-Linux> DSPs don't decode for free [21:46:04 CET] <bencoh> but still :) [21:46:10 CET] <BtbN> More than a CPU having to stay awake [21:46:33 CET] <BtbN> *Less than [21:46:35 CET] <TD-Linux> BtbN, reminds me I need to redo my measurement of this [21:46:38 CET] <bencoh> BtbN: my tests on n900 says otherwise for sw vs dsp cpu decoding [21:46:42 CET] <bencoh> -s [21:46:50 CET] <bencoh> s/cpu/mp3/ wetf [21:47:04 CET] <bencoh> alright, looks like I cant type properly tonight [21:49:23 CET] <jookiyaya> does ffmpeg support converting 7.1 audio to 5.1 ? [21:51:49 CET] <jookiyaya> according to this: http://oi57.tinypic.com/15g2cxz.jpg Neroaac is better than opus [21:52:07 CET] <bencoh> neroaac, seriously? [21:52:16 CET] <furq> firstly, no it isn't [21:52:22 CET] <furq> secondly, that test is by one guy [21:52:26 CET] <furq> but thirdly, and most importantly, no it isn't [21:52:35 CET] <jookiyaya> furq nero is using 96k [21:53:13 CET] <jookiyaya> opus is using 102 [21:53:36 CET] <furq> well i guess you'd better use nero then [21:53:37 CET] <furq> hf [21:54:10 CET] <jookiyaya> hf? [21:55:35 CET] <pzich> it means "have fun" :) [21:57:20 CET] <jookiyaya> oh [21:57:26 CET] <kepstin> jookiyaya: look closer - the legend of that graph says that the average over multiple albums of opus was 94k, nero was 99k. Basically what happened is that opus makes better use of vbr to give extra bitrate to hard samples and save it on easy ones. [21:57:57 CET] <jookiyaya> oh [21:58:06 CET] <jookiyaya> is Nero better than aac-fdk ? [21:58:34 CET] <kepstin> jookiyaya: I haven't seen any well-done comparisons between the two [21:58:53 CET] <kepstin> i suspect they're fairly close. [21:59:27 CET] <furq> http://listening-tests.hydrogenaud.io/igorc/aac-96-a/nonblocked_means_all.p… [22:00:16 CET] <jookiyaya> what are "ct" "tvbr" [22:00:29 CET] <furq> cvbr and tvbr are qaac [22:00:34 CET] <jookiyaya> oh [22:01:06 CET] <furq> CT is another proprietary encoder [22:01:25 CET] <jookiyaya> and it stands for? [22:01:43 CET] <jookiyaya> never heard of it [22:02:14 CET] <furq> coding technologies [22:02:27 CET] <furq> it was the winamp aac encoder before they ditched it in favour of fraunhofer [22:02:36 CET] <furq> fdk is the open-source version of fraunhofer aac [22:03:25 CET] <jookiyaya> i love how fdk is open source yet it's not included in ffmpeg [22:03:42 CET] <furq> it has an incompatible license [22:06:48 CET] <TD-Linux> the license does not meet the OSI definition of open source [22:07:42 CET] <jookiyaya> i tried fdk-aac.exe , it doesn't even transcode [22:10:21 CET] <jookiyaya> does ffmpeg support converting 7.1 audio to 5.1 ? [22:16:22 CET] <kepstin> jookiyaya: of course. The question is whether it has a predefined matrix for it, or if you need to use the pan filter for that. You could just drop the extra channels of course... [22:17:03 CET] <jookiyaya> kepstin what does default option do? [22:18:15 CET] <jookiyaya> i just found a bug for opus. it's misreporting its bitrate [22:18:27 CET] <jookiyaya> or is that ffmpeg bug [22:24:34 CET] <kepstin> jookiyaya: not sure what you mean [22:25:08 CET] <jookiyaya> kepstin what is the difference betwen predefined matrix and pan filter [22:25:35 CET] <kepstin> jookiyaya: with pan filter, you can decide exactly how to downmix the channels manually. [22:25:44 CET] <jookiyaya> i see [22:26:33 CET] <kepstin> if you're using unconstrained vbr mode in opus (the default), the bitrate parameter is basically a quality level, calibrated such that if you encode a large number of files of varying types of audio, the average bitrate will be what you requested. [22:26:46 CET] <kepstin> average over all the files [22:27:11 CET] <jookiyaya> kepstin i understand that but it's reporting as 3072 kbps [22:27:21 CET] <jookiyaya> when it's close to 128kbps [22:28:07 CET] <kepstin> are you encoding audio + video? the bitrate ffmpeg prints in the status line is the combination of both. [22:28:21 CET] <jookiyaya> it's misreporting when it's inside mkv file [22:28:50 CET] <jookiyaya> audio only, it's fine [22:29:22 CET] <kepstin> jookiyaya: are you going to paste some command lines and example output? I have no idea what you are talking about here. [22:30:34 CET] <jookiyaya> http://www.imagebam.com/image/9db148471937577 [22:31:09 CET] <furq> that's mpc-hc's fault [22:31:09 CET] <jookiyaya> it should be 640/128/128 [22:31:22 CET] <furq> try looking at the mediainfo tab instead [22:31:31 CET] <jookiyaya> mpc-hc use mediainfo [22:31:39 CET] <furq> it does in the mediainfo tab [22:31:43 CET] Action: llogan loves images of text [22:31:56 CET] <furq> the main tab is from directshow or some other inaccurate bullshit [22:32:17 CET] <jookiyaya> it doesn't report the bitrate from mendiainfo tab [22:32:20 CET] <kepstin> yeah, that's not ffmpeg, what does ffprobe say the stream bitrates are? :) [22:32:34 CET] <furq> that's because mkv doesn't store bitrates in the metadata [22:32:44 CET] <furq> you can only get track bitrates from mkv if there's 0 or 1 vbr tracks [22:32:53 CET] <furq> or by probing the entire file [22:33:27 CET] <jookiyaya> you are right mediainfo does display correctly [22:34:58 CET] <jookiyaya> what is wrong with this tab [22:35:11 CET] <jookiyaya> it's misreporting it completely [22:35:14 CET] <furq> i guess it's there for legacy reasons [22:35:20 CET] <furq> it should really just use mediainfo for that [22:35:49 CET] <furq> the mediainfo tab is there specifically because the main tab is always either missing information or completely wrong [22:36:00 CET] <jookiyaya> latestversion of mediainfo finally fixed not reporting bitrate with mkv too [22:36:05 CET] <furq> oh that's nice [22:36:18 CET] <jookiyaya> it used to not report it for some reason [22:36:29 CET] <furq> it didn't report it for the reason i just said [22:36:37 CET] <furq> you need to scan the whole file if there are multiple vbr tracks [22:37:23 CET] <jookiyaya> it (always) works fine with mp4 though [22:37:36 CET] <furq> mp4 stores stream bitrates in the metadata, mkv doesn't [22:37:38 CET] <kepstin> jookiyaya: that's because mp4 is a different format that works in a different way [22:38:05 CET] <jookiyaya> problem with mp4 is , i cannot preview the video while it's encoding, with mkv, i can [22:38:17 CET] <jookiyaya> i hate that [22:44:43 CET] Action: kepstin keeps getting annoyed that mkv doesn't preserve video timebase [22:49:07 CET] <jookiyaya> kepstin what is that? [23:26:49 CET] <NeedHelp> Hello is there anyone available to answer a couple questions? [23:27:28 CET] <Guest33272> Is there anyone around who can answer a couple questions for me? [23:27:38 CET] <kepstin> generally works better on irc to ask your question, then hang around to see if anyone answers it. [23:27:57 CET] <Guest33272> Oh sorry, I'm not used to irc [23:28:00 CET] <Guest33272> I'll do that then [23:28:26 CET] <Guest33272> so I have a .flv file that I'm trying to put into lightworks, but the file extension isnt supported [23:28:38 CET] <Guest33272> can ffmpeg help me out with that? [23:28:50 CET] <Guest33272> I'm trying to convert the .flv file into a readable/importable format [23:29:04 CET] <kepstin> yes, ffmpeg can do help with that [23:29:08 CET] <J_Darnley> Just rename it if it doesn't like the filename [23:29:10 CET] Action: J_Darnley runs [23:29:21 CET] <pzich> my little brother used to think that would actually work :-/ [23:29:22 CET] Action: kepstin doesn't know offhand what containers and codecs lightworks supports [23:30:12 CET] <kepstin> it could be that something as simple as 'ffmpeg -i whatever.flv -c copy whatever.mp4' would be enough, depending on the codecs used in the flv file. [23:30:33 CET] <Guest33272> Yeah, see, It's a video I recorded with OBS [23:31:47 CET] <kepstin> probably h264 video and aac or mp3 audio then. give the simple ffmpeg copy command a try. [23:31:56 CET] <furq> http://www.lwks.com/index.php?option=com_content&view=article&id=100&Itemid… [23:32:34 CET] <furq> it looks like you'll have to either transcode it or use an editor which isn't openly hostile to non-professional formats [23:33:03 CET] <Guest33272> hm, I'll try what kepstin suggested [23:33:09 CET] <kepstin> hmm? that list includes h.264 [23:33:15 CET] <Guest33272> it's my first time using this thing [23:33:22 CET] <kepstin> does say avchd tho [23:33:28 CET] <kepstin> might need a different container [23:33:41 CET] <Guest33272> argh this is pretty complicated. [23:33:54 CET] <Guest33272> stupid obs... [23:33:56 CET] <kepstin> should handle h264 in mpeg-ts at the very least, i think [23:34:26 CET] <kepstin> really, lightworks should just use ffmpeg to read input files, then this problem wouldn't happen ;) [23:34:27 CET] <Guest33272> stupid question, how do I execute/run ffmpeg? [23:34:36 CET] <Guest33272> just extracted it to a sep folder but i dont see an executable file [23:34:47 CET] <kepstin> Guest33272: what did you download? [23:34:57 CET] <furq> if you got the zeranoe build then ffmpeg.exe is in bin/ [23:34:58 CET] <kepstin> I assume you're on windows, so you'd have to grab a binary [23:35:12 CET] <Guest33272> yeah i'm on windows [23:35:20 CET] <kepstin> https://ffmpeg.zeranoe.com/builds/ [23:35:25 CET] <furq> https://ffmpeg.zeranoe.com/builds/ [23:35:32 CET] <furq> bah [23:35:53 CET] <Guest33272> i'm running a 64 bit so... [23:35:58 CET] <Guest33272> which of the above 3 would I select? [23:36:01 CET] <Guest33272> all of them or..? [23:36:15 CET] <kepstin> Guest33272: read the page, it says right on it [23:36:23 CET] <furq> If you are confused about what build type you need just download a static build. [23:36:28 CET] <furq> it makes no difference whether you get 32 or 64 bit [23:36:38 CET] <Guest33272> Alright, thanks, ill do that first [23:37:06 CET] <jookiyaya> https://sourceforge.net/projects/ffmpeg-hi/ this claims it comes with fdkaac and 10bit x264 ? what is 10bit [23:38:04 CET] <jookiyaya> does regular build not come with 10 bit? [23:38:11 CET] <J_Darnley> 10 bit sample depth [23:38:14 CET] <jookiyaya> is that against license too? [23:38:26 CET] <J_Darnley> and it better not come with fdkaac [23:38:33 CET] <furq> someone's getting reported [23:38:48 CET] <furq> 10-bit isn't against license but it prevents you from encoding 8-bit x264 [23:38:48 CET] <jookiyaya> furq lol are you serious? [23:39:33 CET] <furq> and very few players support decoding 10-bit h264 [23:39:46 CET] <jookiyaya> so what does extra 2 bit exactly mean [23:39:53 CET] <J_Darnley> Only rubbish ones that don't use ffmpeg [23:40:01 CET] <furq> yeah but that's most of them [23:40:25 CET] <furq> on the desktop it'll be fine but if you need browser or mobile device support then good luck [23:40:25 CET] <jookiyaya> why can't they build one that does both? [23:40:32 CET] <furq> ask the x264 guys [23:40:39 CET] Action: J_Darnley points back to his last line [23:40:41 CET] <furq> libvpx and x265 support multiple sample depths [23:40:53 CET] <jookiyaya> no i will ask the ffmpeg guys [23:40:55 CET] <furq> in the same build, i mean [23:41:07 CET] <Guest33272> alright so I got the exe loaded @kepstin [23:41:08 CET] <J_Darnley> Because you haven't written a patch for it yet, jookiyaya [23:41:18 CET] <Guest33272> where do I go from here :O [23:41:41 CET] <jookiyaya> i don't see how ffmpeg cannot include 2 x264 when it can support like 50+ different encoders [23:42:04 CET] <furq> Guest33272: are you actually invested in using lightworks [23:42:19 CET] <Guest33272> I was recommended the software by a friend [23:42:22 CET] <furq> it might be easier to use an NLE which can actually import your files [23:42:26 CET] <Guest33272> but unfortunately, because I streamed on OBS for a while [23:42:27 CET] <furq> http://www.openshotvideo.com/ [23:42:28 CET] <furq> like that [23:42:43 CET] <Guest33272> it's saved all my stream VODs with .flv file extensions [23:42:55 CET] <Guest33272> so uploading to youtube is rather painful because youtube's editor is terrible. [23:43:08 CET] <furq> what are you actually hoping to do [23:43:24 CET] <furq> if you just want basic editing tools then avidemux is probably your best bet [23:43:26 CET] <Guest33272> I want to take my stream vods (which are .flv file extensions) and convert them into a format where [23:43:32 CET] <Guest33272> I can edit them with a video editing software [23:43:41 CET] <furq> i mean what do you want out of your video editing software [23:44:16 CET] <Guest33272> More or less something that offers me alot of flexibility in how I can customize my video [23:44:26 CET] <Guest33272> Lightworks seems pretty good, especially for a free software [23:46:05 CET] <Guest33272> Does that openshotvideo program support .flv file extensions though? cause otherwise I'd be back to square 1 [23:46:26 CET] <Guest33272> http://www.lwks.com/index.php?option=com_kunena&func=view&catid=23&id=11039… <- this link told me to use ffmpeg but now I'm pretty confused haha [23:46:40 CET] <jookiyaya> i don't understand the 8/10 bit: i thought rgb is 24bit [23:47:28 CET] <J_Darnley> And how many bits does each red, green, and blue sample use? [23:48:02 CET] <jookiyaya> oh okay: so 8 bit = 24bit [23:48:26 CET] <furq> Guest33272: i've never used openshot but it uses ffmpeg for decoding and encoding so it should support everything ffmpeg supports [23:48:27 CET] <jookiyaya> so 10 bit = 30bit RGB ? [23:48:47 CET] <J_Darnley> Basically, yes [23:48:55 CET] <furq> "Also, you can import and edit H.264 directly in Lightworks if you use the ultrafast preset of the OBS Studio H.264 encoder." [23:49:01 CET] <jookiyaya> but in windows, it claims it uses 32bit color [23:49:13 CET] <J_Darnley> 8 bits of padding or alpha! [23:49:14 CET] <furq> that's troubling [23:49:20 CET] <Guest33272> hm? [23:49:31 CET] <J_Darnley> Learn yourself some stuff [23:49:39 CET] <Guest33272> I'm kind of new with the whole video editing software thing, but I recognize the importance of it for my viewers [23:49:59 CET] <Guest33272> so I'm doing my best to work with what I have and I've got alot of footage in .flv format [23:50:13 CET] <Guest33272> that I'd like to be able to export to youtube after editing in a video editing software [23:50:32 CET] <Guest33272> if openshot can do that, then I'd use it gladly [23:50:48 CET] <TD-Linux> youtube can parse tons of stuff so that shouldn't be an issue [23:52:09 CET] <jookiyaya> what does 8 bits of padding or alpha exactly do? [23:52:20 CET] <c_14> magic [23:52:31 CET] <J_Darnley> Fills up the rest of the doubleword [23:52:31 CET] <Guest33272> I mean, I could put it into youtube but then [23:53:13 CET] <J_Darnley> and alpha is for transparency [23:53:40 CET] <jookiyaya> "doubtword"? [23:54:02 CET] <J_Darnley> no, doubleword [23:54:08 CET] <jookiyaya> what is doubleward? [23:54:12 CET] <J_Darnley> 4 bytes [23:54:12 CET] <jookiyaya> word* [23:54:58 CET] <J_Darnley> Wow, I could swear some people have never looked at assembly on x86(!) [23:55:57 CET] <jookiyaya> i remember in windows, you had option to choose 32bit color or 16bit color: how many bits of rgb does 16bit color use and how much of it were part of alpha [23:56:10 CET] <J_Darnley> Probably none [23:56:22 CET] <J_Darnley> Idon't remeber whether Windows used 565 or 555 [23:56:55 CET] <jookiyaya> so 16bit color = only 5.333 bit [23:57:25 CET] <jookiyaya> i could easily tell the difference between 16bit and 32 bit color settings [23:57:39 CET] <jookiyaya> in windows [23:58:00 CET] <jookiyaya> also it support 256 colors too [23:59:00 CET] <furq> jookiyaya: https://gist.github.com/l4n9th4n9/4459997 [23:59:28 CET] <furq> just ignore all the stuff about animes [00:00:00 CET] --- Wed Mar 16 2016

1 0

ffmpeg-devel.log.20160314
by burek 15 Mar '16

15 Mar '16

[10:21:17 CET] <cone-255> ffmpeg 03Matthieu Bouron 07master:33d69a90085d: lavc/ffjni: remove use of private JniInvocation API to retreive the Java VM [13:25:37 CET] <cone-788> ffmpeg 03Paul B Mahol 07master:9f6e63f6f2f2: avfilter/vf_waveform: add >8 bit support for other filters [13:25:37 CET] <cone-788> ffmpeg 03Paul B Mahol 07master:29d106e751be: avfilter/vf_waveform: remove mirroring from chroma filter [14:14:12 CET] <JEEB> hmm, has anyone else seen movenc giving the first sample a bit over a double of duration? [14:14:20 CET] <JEEB> in case of fragmented stuff [14:19:37 CET] <wbs> JEEB: yes [14:19:55 CET] <wbs> JEEB: and you should probably have a warning print in the log about it these days [14:20:38 CET] <wbs> JEEB: tl;dr, please set AVPacket duration when muxing, if possible. otherwise the muxer will try to use some heuristics to guess it, or if it can't, it will end up zero, which will force this kind of fixup for the start of the next fragment [14:24:08 CET] <JEEB> wbs: ID'm currently testing with ffmpeg cli, and it seems like I get this with fdk_aac and not with the aac encoder o_O [14:24:32 CET] <JEEB> basically it was leading to a new fragment not starting with a sync sample [14:24:48 CET] <JEEB> but it didn't happen if I had b-frames enabled [14:24:54 CET] <JEEB> this is weird :D [14:25:08 CET] <wbs> yes, your explanation is weird [14:25:14 CET] <wbs> please come back when you have something I can reproduce [14:25:36 CET] <JEEB> yeah, I was just minimizing my examples [14:36:58 CET] <JEEB> wbs: http://up-cat.net/p/a8ba0579 [14:37:25 CET] <JEEB> so I set to have 5 second fragments and I have a static GOP length of 125 pictures in x264 [14:48:14 CET] <JEEB> feel free to hit my head if I am doing something stupid, but this seems pretty weird [14:49:43 CET] <JEEB> (frag_keyframe enabled does the same, just for reference) [14:54:00 CET] <wbs> JEEB: nothing wrong here, move along [14:54:28 CET] <wbs> JEEB: the issue comes from the amount of timestamp adjusting/shifting due to b-frames and aac preroll [14:54:51 CET] <JEEB> kind of was guessing it was something like that... [14:55:05 CET] <wbs> say your stream starts at dts=-1, pts=0 [14:55:28 CET] <wbs> now due to avoid_negative_ts (since ismv doesn't really do edit lists), this is shifted into dts=0, pts=1 [14:55:39 CET] <wbs> or similar [14:56:25 CET] <wbs> now if the audio preroll is much larger than one frame, it would shift all of it into e.g. dts=0, pts=2, since it all starts from dts=0 unless you have edit lists [14:57:08 CET] <wbs> for ismv, without edit lists (and for dash) I guess your patch (a proper version of it, that is) that allows negative cts offsets and sync stuff to start with dts=pts=0, would be the simplest [14:57:24 CET] <wbs> together with dropping enough audio before the audio encoder, to counter the added encoder delay [14:57:54 CET] <JEEB> yeah... I tested my current patch and that was failing at something, so I moved to removing B-frames for now [14:58:05 CET] <JEEB> and then I got this funky business :D [15:02:07 CET] <JEEB> wbs: or you give a slightly different DTS for the audio in the "fragment's start DTS" thingy [15:02:34 CET] <wbs> JEEB: no, that just messes things up even more [15:03:01 CET] <wbs> JEEB: really, if you don't have edit lists, which you don't for ismv, make sure that your audio track starts at dts=pts=0 [15:03:17 CET] <JEEB> yeah [15:04:49 CET] <omerjerk> I've another doubt regarding writing the floating point data to the output file from the decoder. [15:05:01 CET] <omerjerk> This is where we write it to the output - https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/alsdec.c#L1554 [15:05:27 CET] <omerjerk> wm4 said, I can write float values to *dest. [15:06:10 CET] <wm4> of course you must fix the type and the output sample format [15:07:20 CET] <nevcairiel> it probably sets sample_fmt somewhere [15:07:33 CET] <nevcairiel> oh it already does [15:07:35 CET] <omerjerk> but the data type of *dest is int [15:07:36 CET] <omerjerk> https://ffmpeg.org/doxygen/2.7/structAVFrame.html#a1d0f65014a8d1bf78cec8cbe… [15:07:37 CET] <nevcairiel> even to float [15:07:42 CET] <omerjerk> yes it does. [15:07:44 CET] <omerjerk> I saw that. [15:07:56 CET] <nevcairiel> its an uint8_t*, which is our generic byte storage format [15:08:12 CET] <omerjerk> yes. [15:08:18 CET] <nevcairiel> you would still just write float samples in there [15:08:25 CET] <nevcairiel> with appropriate casts [15:08:26 CET] <omerjerk> so how would I write float value to it ? [15:08:49 CET] <nevcairiel> float *buffer = frame->data[0]; voila, you have a float buffer =p [15:08:59 CET] <omerjerk> oh. [15:09:07 CET] <omerjerk> but how is this working ? :o [15:09:19 CET] <nevcairiel> the interelave output macro would probably have to be re-worked to support floats [15:09:37 CET] <omerjerk> yes. I know that. [15:09:49 CET] <omerjerk> I'm more concerned about float *buffer = frame->data[0]; [15:09:51 CET] <nevcairiel> it could probably also be re-worked to use planar output and save a bit of re-shuffling, but thats a task for another day [15:10:29 CET] <omerjerk> so it'll work, right ? [15:10:37 CET] <nevcairiel> data[0] is just generic memory storage, it doesnt care what you put in there [15:10:44 CET] <nevcairiel> what is in there is identified by the sample_fmt [15:10:52 CET] <omerjerk> okay. [15:10:53 CET] <wm4> well ff_get_buffer does care [15:11:06 CET] <nevcairiel> if its AV_SAMPLE_FMT_FLT or whatever its name was, it will happily take float [15:11:10 CET] <wm4> it allocates the required amount of memory by looking at the sample_fmt [15:24:02 CET] <JEEB> wbs: so yeah, I guess I'm hosed for the time being until I get the negative CTS offsets properly implemented :D [15:34:54 CET] <burnrice> hi, is there any rule, what do to when demuxing pseudo-formats that are only disguising already implemented container ? [15:35:06 CET] <burnrice> e.g. kux (#4519), which in fact is flv, just with some offset [15:35:19 CET] <burnrice> it could be done by creating new demuxer and redirecting data like: [15:35:28 CET] <burnrice> int kux_read_packet(AVFormatContext *s, AVPacket *pkt) { return ff_flv_demuxer->read_packet(s, pkt); [15:35:38 CET] <burnrice> but this is ugly, and causes problems, and probably not the way it should be done [15:35:52 CET] <burnrice> changing original flv file, to make it sensitive for this type of files defnitely works better, especially that in fact only two ifs are added [15:36:38 CET] <burnrice> or maybe modify the original stream, and then somehow reinitialize whole open_input procedure (is it even possible?) [15:41:22 CET] <JEEB> huh [15:41:40 CET] <JEEB> -avoid_negative_ts 0 [15:42:44 CET] <JEEB> this seems to give me correct durations [15:42:59 CET] <JEEB> need to check how badly it mutilates some other values, though [15:50:42 CET] <jkqxz> Is that coreimage stuff actually wanting to use hwcontext? [15:57:20 CET] <cone-788> ffmpeg 03Hendrik Leppkes 07master:a38627be6d61: avcodec/cbrt_data: include avutil/libm.h to unbreak compilation on systems without cbrt [16:12:26 CET] <JEEB> huh, this actually seems to kind of do the thing in the perverse way of pushing down audio [16:12:44 CET] <JEEB> so I have audio start 426667 ticks before video [16:13:23 CET] <JEEB> need to see if this streaming server likes that [16:34:22 CET] <cone-788> ffmpeg 03Carl Eugen Hoyos 07master:58fa69497873: lavf/http: Add httpproxy to the default protocol whitelist. [16:38:26 CET] <cone-788> ffmpeg 03Carl Eugen Hoyos 07master:f1a3677c0f31: lavf/mxfdec: Fix Canopus essence element size. [16:39:28 CET] <cone-788> ffmpeg 03Carl Eugen Hoyos 07release/3.0:9da31a03738f: lavf/http: Add httpproxy to the default protocol whitelist. (cherry picked from commit 58fa694978735e30f433f8ede6c7f808b9409919) [16:46:12 CET] <JEEB> alright, this thing worked :D [16:46:33 CET] <JEEB> wbs: so yeah, it's a mess but with -avoid_negative_ts 0 I get it working (it seems) without B-frames [16:46:46 CET] <JEEB> which gives me time to implement the proper fix [16:47:11 CET] <JEEB> the audio starts 0.04 s before video, which I think is the encoder delay [16:58:56 CET] <kierank> I'm currently looking into storing WebVTT subtitles[1] in Matroska [17:03:03 CET] <durandal_1707> why? [17:03:38 CET] <kierank> that's a quote [17:03:41 CET] <kierank> from the CELLAR mailing list [17:05:04 CET] <wm4> but why [17:05:26 CET] <wm4> google somehow managed to make the webvtt matroska mapping buttmad insane [17:05:41 CET] <wm4> (the demuxer has to rewrite them to feed it to our "decoder") [17:06:28 CET] <wm4> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/matroskadec.c;h=… [17:11:28 CET] <atomnuker> why'd they choose something so limited like webvtt for archiving [17:17:04 CET] <nevcairiel> if google already defined webvtt in webm, it would probably be insanely mad to assume cellar would just adopt that, huh [17:23:21 CET] <kierank> oh wow he knows how to use the bugtracker [17:23:29 CET] <JEEB> wow [17:26:54 CET] <durandal_1707> now your worst nightmare have just begun [17:28:24 CET] <ubitux> maybe we could shadowban him [17:29:42 CET] <mateo`> as long as he stops spamming the ml ... [17:35:41 CET] <wm4> kierank: he appeared on the bug tracker first, where he insulted all ffmpeg developers [17:35:53 CET] <wm4> only then he moved on to the ml [17:38:50 CET] <JEEB> oh right [19:03:47 CET] <dinux5> If anybody could tell me what should I do with the variable "coded_frame" which I found in an encoder file and has been deprecated. [19:04:07 CET] <BtbN> remove it [19:04:55 CET] <dinux5> But it has been used in few line of code and can not be removed straight away i think . :/ [19:05:04 CET] <dinux5> lines* [19:05:18 CET] <nevcairiel> its usually kept for compatibility for some time [19:05:25 CET] <nevcairiel> so just ignore it [19:05:40 CET] <BtbN> an encoder shouldn't need it for anything, put it in deprecation-ifdefs if you want to stay compatible [19:09:08 CET] <dinux5> Alright. Also libavcodec/mlp.h has filterParams struct which does not contain the members coeff_bits and coeff_shift which are present in bulk in the encoder. [19:09:20 CET] <dinux5> How do I deal with those ? [19:10:14 CET] <dinux5> What alternative to be used ?or should the functions using them be deleted right away ? [19:10:29 CET] <nevcairiel> why do we even bother releasing release branches if the first thing every error reporter hears from carl is to get master instead? the latest release should still be considered a valid target for bug reports =p [19:11:29 CET] <fritsch> +1 (not that I have a vote) :-) [19:12:22 CET] <dinux5> I am actually working on my qualification task for OPW :) [19:12:30 CET] <dinux5> need help. [19:15:47 CET] <Compn> nevcairiel : no, releases are not valid for bugreports. in many cases bugs are fixed, especially from older releases [19:16:12 CET] <Compn> if someone wants to change it, then we'd need more people working on trac bugs [19:16:19 CET] <nevcairiel> Compn: thats nonsense, either we care about bugs in releases, or we shouldnt release them [19:16:21 CET] <BtbN> well, if the release is still maintained, I'd still considder it a valid report [19:16:41 CET] <nevcairiel> even more so if the release is what, a month old? [19:16:43 CET] <BtbN> just not against master. Kind of like a backport-request, in the most simple cases. [19:16:50 CET] <nevcairiel> and if its already fixed, you know what to backport [19:16:52 CET] <Compn> yeah but lets say you backport the fix [19:17:00 CET] <Compn> the distro would have to update the release [19:17:13 CET] <Compn> how often does that happen ? [19:17:19 CET] <nevcairiel> often [19:17:21 CET] <Compn> *for non security fixes* [19:17:21 CET] <BtbN> Don't they usualy do that? [19:17:36 CET] <nevcairiel> there is usually security related things in there as well [19:17:41 CET] <BtbN> I think Debian does pull in ffmpeg minor releases [19:18:06 CET] <Compn> what about releases older than 3 months ? [19:18:26 CET] <BtbN> If it's still maintained [19:18:27 CET] <nevcairiel> the most recent releases from the still maintained releases branches [19:18:57 CET] <nevcairiel> ie. 3.0 and 2.8.6, and whatever other branches are stiull up to date [19:19:18 CET] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=MAINTAINERS;hb=HEAD#l574 it even lists which releases are maintained [19:19:22 CET] <BtbN> aparently 3.0 is not [19:19:24 CET] <Compn> we could ask carl to ignore reports made on release branches, if thats your complaint? [19:20:16 CET] <BtbN> I don't see a problem with asking people to reproduce on master. But I'd still considder it valid if it works on master, but does not in some release branch. [19:20:42 CET] <BtbN> And i think that's how it's handled already? [19:20:51 CET] <nevcairiel> its a matter of timing as well, 3.0 is a month old, the likelyhood that something has been fixed is pretty low as it is =p [19:21:25 CET] <Compn> i've seen and reported bugs that were fixed within a month before :P [19:21:36 CET] <Compn> then told its unreproducable in master. updated to master... bug was fixed [19:21:54 CET] <Compn> it does happen! :P [19:22:26 CET] <Compn> but ok, i was just brainstorming , no problems [19:22:45 CET] <Compn> should we update that maintained release list in maintainers file ? [19:22:53 CET] <Compn> 'ave to ask michael :P [19:23:00 CET] <jamrial> sure does. but it doesn't change the fact it's broken in that release and the fix should be backported, especially if the bug is a regression [19:23:39 CET] <Compn> seems like a large maintainer burden [19:24:11 CET] <nevcairiel> if you dont wnat to do th at, then you shouldnt even bother making releases at all [19:24:11 CET] <Compn> merging, testing, backporting, fixing security bugs, issuing security fixes, testing again... [19:24:18 CET] <jamrial> then saying you're "maintaining" releases would be false, if you're not fixing bugs as they are reported [19:24:35 CET] <nevcairiel> having a branch thats essentially unmaintained but "appears" like a stable version is worse than just having master [19:25:05 CET] <Compn> its not like we test branches much before release [19:25:34 CET] <nevcairiel> first off, the rest of the world doesnt know that [19:25:45 CET] <jamrial> we test them as much as any master snapshot, using fate on several systems, etc [19:25:50 CET] <nevcairiel> and by limiting the amount of changes that go in to branches to be fixes, they get more stable over time [19:25:55 CET] <Compn> right so they are the same as master, just old [19:26:35 CET] <Compn> i dont have the answers [19:26:43 CET] <nevcairiel> you never do, thats ok =p [19:27:27 CET] <nevcairiel> but we as a project can't act like any other project and release "stable" versions, and then tell everyone that we dont really care about those versions [19:27:32 CET] <Compn> so we need release maintainers, and trac release maintainers [19:27:34 CET] <nevcairiel> then we just shouldnt release any [19:28:03 CET] <jamrial> Compn: yes, they have a frozen feature set, abi and api, which distros love. if you find a bug, you want that bug gone in the release and not get a "use master instead" as reply [19:30:11 CET] <Compn> would be nice to have an iso with a random sampling of popular downstream products , with some kind of automated test to make sure a new release didnt break those programs. i dont know what the distros use to test such things. [19:30:39 CET] <jamrial> just imagine if gcc every april said "we're making a release, branch has been made and is now frozen forever". you'd be stuck with crap like 4.9.0 for a whole year unless you use trunk which is essentially suicide until like december [19:31:21 CET] <Compn> gcc is probably not a project we should follow. seeing how many times gcc changes have broken our build [19:31:33 CET] <Compn> esp .0 releases [19:31:43 CET] <nevcairiel> why not, what do you imagine how often new releases break downstream software :D [19:32:20 CET] <Compn> gotta run, bbl [21:40:20 CET] <Carll_> Hi [21:40:23 CET] <Carll_> I would like to feed ffserver with some encoded packets.I'm using " av_interleaved_write_frame( format_out_ctx, outPacket ); " function to write a packet to an output media file. I can connect with the server with avformat_write_header( format_out_ctx, NULL ) function, and it is working fine (i think). If I pass that function, the server shows me " [POST] "/feed1.ffm HTTP/1.1" 200 4096 "" [21:40:28 CET] <Carll_> But my problem is, the av_interleaved_write_frame() function doesn't feed the server. [21:40:33 CET] <Carll_> Can anyone tell me what am i doing wrong? If somebody is willing to help me, I could send ffserver.conf and my C files. [22:11:08 CET] <Carll_> someone? [22:14:59 CET] <Compn> try ask in #ffmpeg , Carll_ ? [22:15:13 CET] <Compn> this channel is for development of ffmpeg [22:15:18 CET] <Carll_> tried :/ [22:15:22 CET] <JEEB> but in general ffserver is only known by very few, so good luck [22:15:32 CET] <Compn> try ask on mailing list ? [22:15:45 CET] <JEEB> if you depend on it, you should either learn it by code so you can maintain it yourself or just run away [22:16:08 CET] <Carll_> i would like to get the stream working on any media server, but i dont know if i am doing right. [22:16:49 CET] <ubitux> awi [22:16:51 CET] <ubitux> oups [22:22:01 CET] <Carll_> Can someone help me please? The following command is working "ffplay -rtsp_flags listen -i rtsp://127.0.0.1:8090/live.sdp". It streams my desktop. [22:22:05 CET] <Carll_> But i would like to feed the server [22:22:14 CET] <Carll_> and it doesn't work [22:23:20 CET] <JEEB> first of all this is the wrong channel for user support and second of all almost no-one knows anything about ffserver as already stated [22:23:26 CET] <JEEB> you're better off asking the users' mailing list [22:32:50 CET] <Mavrik> Should probably just delete ffserver from documentation, git and existence :P [22:36:54 CET] <JEEB> IIRC the incoming API changes will pretty much break it, so after it goes through the period of deprecation warnings it will probably get removed - unless someone comes up and picks it properly [22:37:37 CET] <wm4> Mavrik: +1 [22:37:45 CET] <wm4> but "they" will resist [22:38:10 CET] <wm4> also, codecpar is still not merged... [22:38:54 CET] <JEEB> yeah [22:39:55 CET] <jamrial> git retcon ffserver [22:44:54 CET] <nevcairiel> wm4: well you didnt fix it any further, so it sits there and wiats? :) [22:55:58 CET] <cone-128> ffmpeg 03Michael Niedermayer 07master:c34250178ba2: avfilter/af_sofalizer: Fix "warning: ISO C90 forbids mixed declarations and code" [23:14:13 CET] <cone-128> ffmpeg 03Paul B Mahol 07master:af5559ab6711: avfilter/vf_waveform: add graticule for chroma and flat filter [00:00:00 CET] --- Tue Mar 15 2016

1 0

ffmpeg.log.20160314
by burek 15 Mar '16

15 Mar '16

[04:45:35 CET] <jookiyaya> anybody have ffmpeg build that support aac-fdk and mp3 shine [04:47:46 CET] <furq> stop asking people to violate the gpl [04:48:25 CET] <furq> if you want an ffmpeg build with fdk support, build it yourself [04:49:13 CET] <jookiyaya> i have no idea how to [04:49:35 CET] <jookiyaya> and mp3 shine doesn't work either [04:51:14 CET] <furq> well then learn how to or use a different aac encoder [04:51:16 CET] <jookiyaya> and what version # is this: R:\>ffmpeg64 --version [04:51:16 CET] <jookiyaya> ffmpeg version N-78843 [04:51:39 CET] <jookiyaya> furq i bet you don't even know how to either [04:57:42 CET] <c_14> jookiyaya: there should be a -g<something> after that. that something is the git commit hash [04:59:54 CET] <jookiyaya> https://ffmpeg.zeranoe.com/builds/win64/static/ffmpeg-3.0-win64-static.7z [05:00:02 CET] <jookiyaya> what does it mean by "shared version" [05:00:07 CET] <jookiyaya> vs "static version" [05:01:10 CET] <c_14> The shared version has shared libraries, the static version has static libraries [05:01:39 CET] <jookiyaya> what does it mean by having 'shared libraries" and "static libraries" [05:03:25 CET] <PrestonL> Anyone know of a module or way for there to be an XML or JSON stat.html page so that another application can query the server status and display that info in another way? [05:04:11 CET] <c_14> jookiyaya: http://cs-fundamentals.com/tech-interview/c/difference-between-static-and-d… [05:04:58 CET] <jookiyaya> does it make a difference in final encoding product [05:05:09 CET] <jookiyaya> whether you used static or shared [05:05:20 CET] <c_14> no [05:05:41 CET] <jookiyaya> from which ffmpeg version did they removed fdk-aac [05:06:01 CET] <c_14> none [05:06:38 CET] <jookiyaya> i was told they had it before [05:06:44 CET] <c_14> It was never in a distributible build (but you can still compile ffmpeg with it yourself) [05:22:54 CET] <jookiyaya> what is correct file extension for audio-only aac file [06:21:43 CET] <jookiyaya> is there 2 different vorbis encoder in ffmpeg? [10:43:43 CET] <eightfold> hi there! i have a video that is 720x576px. i want to upscale it to 1920x1080 and put black borders on the left and right sides. [10:50:03 CET] <eightfold> i've used: ffmpeg -y -i input.mp4 -aspect 5:4 -vf "scale=1920:1080" output.mp4 [10:50:26 CET] <eightfold> but when i check media information in vlc it says "resolution: 1920x1090" [10:50:37 CET] <eightfold> and "display resolution: 1920x1080" [10:51:55 CET] <eightfold> can this be fixed? input files also says: "resolution: 720x578" and "display resolution: 720x576" [11:24:49 CET] <maxinminax> i need compile ffmpeg with ocr filter (use tesseract lib) but when do that, it have errror "ERROR: tesseract not found using pkg-config" [11:25:20 CET] <maxinminax> anyone can help me? [11:25:38 CET] <maxinminax> or anyone have pre-build version which have build options --enable-libtesseract [11:36:41 CET] <flux> is there some modern video format (supported by ffmpeg) that's playable while the file is being written? mpeg4 does not seem to fit the bill :) [11:37:12 CET] <flux> it also seems ffmpeg cannot write the required header information without releasing memory etc, so a program cannot just write the headers once a minute or so [11:38:58 CET] <Mavrik> flux, MPEG-TS can be read while written [11:39:12 CET] <Mavrik> It's used everywhere where live streaming is required for that reason ;) [11:54:09 CET] <Bluez_> hi [12:40:32 CET] <flux> mavrik, can I seek it with ffmpeg-based players? [12:44:01 CET] <flux> actually I think I can, though it might not work if I seek beyond the track length that existed when I start the player [12:44:11 CET] <flux> but yeah, MPEG-TS is probably the way to go. thanks! [12:44:40 CET] <flux> (and once the save is finished (I split it in one-hour chunks) I can just convert it to MPEG4) [12:46:20 CET] <Mavrik> MPEG4 is not a container [12:46:22 CET] <Mavrik> :) [12:46:55 CET] <flux> fine, ISO Media File Format then ;) [12:47:25 CET] <flux> actually is MP4 not a container format.. [12:48:12 CET] <Mavrik> MP4 is a container format but it's a tiny part of the spec :) [12:48:24 CET] <Mavrik> MPEG-TS is also part of MPEG spec etc. :) [12:48:29 CET] <Mavrik> That's why it can be confusing :) [12:49:30 CET] <flux> however, copying H264-encoded data from MPEG-TS to an MPEG4 is a completely sensible thing to do, no?-( [12:49:42 CET] <Mavrik> Well, depends :) [12:49:52 CET] <Mavrik> It can be done easily and quickly. [12:50:11 CET] <BurnerGR> I have troubles getting a decent quality when live encoding 1080p@60 using libx264, it does not seem to utilize much CPU regardless of the number of --threads, it just drop fps when increasing quality settings [12:50:28 CET] <Mavrik> BurnerGR, pastebin command + output if you can [13:20:57 CET] <BurnerGR> Mavrik, this is an example where encoding fps drops, but cpu load is low http://pastebin.com/z2r6rA9a [13:28:58 CET] <BurnerGR> Mavrik, oh, looking at my own paste, I see that cpu load is not low, ffmpeg is just "niced" [13:38:32 CET] <DHE> BurnerGR: libx264 sets some of its own encoding threads as low priority [13:39:03 CET] <DHE> also you're mixing -b:v with -crf [13:39:45 CET] <DHE> it does look like you're CPU-bound. maybe try veryfast? [13:40:12 CET] <BurnerGR> veryfast works, but its a horrible quality [13:41:55 CET] <BurnerGR> I'll assign some more cores and see if it helps [13:56:31 CET] <DHE> oh it's a VM [13:57:54 CET] <DHE> I'm inclined to say that for 60fps 1080p, you might want to raise the bitrate a little bit. depending on the media 4.5 megabits is a bit on the low side, possibly better suited for 30fps material [14:08:20 CET] <BurnerGR> DHE, media source is mixed, and can have a lot of action (fps gameplay), there is no problem with quality during interviews and such [14:08:46 CET] <DHE> ah, video games. sharp edges. yeah that'll be hard [15:14:17 CET] <Carl___> I have a problem with ffserver. I can feed the server in linux and play in windows. But i can't feed it under windows [15:14:43 CET] <Carl___> the command i'm using is: ffmpeg -i sample.mpg http://192.168.10.117:8090/feed.ffm [15:51:14 CET] <wizonesolutions> OS X: I'm trying to record system audio on Mac, and I've installed iShowU Audio Capture and set up a multi-output device to get system audio into a device usable by ffmpeg when using the avfoundation format. However, the audio comes in faster than the video, and I'm not sure where to start trying to troubleshoot that. [15:51:36 CET] <wizonesolutions> Or if something like Soundflower actually *would* be better (googling implies it doesn't work that great) [15:59:10 CET] <jarainf> "Audio comes in faster"? [15:59:21 CET] <jarainf> Is it delayed? [16:05:13 CET] <xeons> Having audio / video sync issues do to timestamps and offsets when I -ss crop out sections of h.264 video: http://stackoverflow.com/questions/35416110/ffmpeg-concat-video-and-audio-o… [16:05:56 CET] <xeons> I've tried apad, -c copy, -shortest, -avoid_negative_ts, make_zero, -fflags, +genpts, etc.. without luck [16:06:40 CET] <durandal_1707> apad adds audio at end [16:06:52 CET] <xeons> Yes, I was using it to try to make the audio last as long as the video [16:07:08 CET] <xeons> + -shortest [16:07:33 CET] <durandal_1707> you can't use it with copy [16:07:53 CET] <xeons> Good to know. I've tried it with an encode and it failed too. [16:08:54 CET] <xeons> I can do better than that. Here is the source video and commands: http://davidpennington.me/share/audio_sync_test_video.zip [16:13:36 CET] <Carl___> I'm getting an error [16:13:38 CET] <Carl___> HTTP error 503 Server too busy [16:13:47 CET] <Carl___> http://192.168.10.117:8090/feed1.ffm: Server returned 5XX Server Error reply [16:13:51 CET] <Carl___> How can i fix that? [16:17:21 CET] <Carl___> anyone? [16:29:01 CET] <lakis> Hello! Anyone have any examples of batch processing with the CLI? I have big folder with lots of subfolders of mostly mp3's and would like to batch convert them to small m4a/aac files [16:31:35 CET] <andrey_utkin> lakis, just traverse your folders in a loop, you can do that in any programming language you like [16:31:58 CET] <andrey_utkin> ffmpeg doesn't process many files at once in your case [16:34:53 CET] <lakis> yeah, i was looking to save some time thinking that someone must have done something similar before. I need a linux script that picks all mp3's and wma's in all subfolders and re-encodes them into smaller sized m4a's in place. Can't find any good examples online, have any links? [16:35:39 CET] <lakis> I guess something similar to this, for audio: for f in *.avi; do ffmpeg -i "$f" -c:v libx264 -crf 23 -preset medium \ -c:a libfdk_aac -vbr 4 -movflags +faststart -vf scale=-2:720,format=yuv420p \ encoded/"${f%.avi}.mp4"; done [16:36:00 CET] Last message repeated 1 time(s). [16:38:17 CET] <furq> find . $ -name "*.mp3" -o -name "*.wma" $ -exec ffmpeg -i "{}" ... "{}.m4a" \; [16:39:07 CET] <Carl___> Why does this wait_feed problem occurs? [16:39:12 CET] <Carl___> I thought it was repaired [16:39:36 CET] <furq> i wouldn't hold your breath for help with ffserver, it's practically abandoned [17:55:20 CET] <n1cky_> Can anyone shed some light on how I can use ffmpeg to buffer an http stream, such that I can rewind it ~60 seconds and make recordings? [17:56:03 CET] <n1cky_> The end goal is to be able to make short clips after something interesting happens in the stream. [17:56:39 CET] <teratorn> is there an easy way to say, cut the last 5 seconds off a video without having to do manual calculations? Ie. with a -5s parameter somewhere or something like that? [17:58:26 CET] <n1cky_> teratorn: man ffmpeg-utils "Time duration" [17:58:49 CET] <n1cky_> haven't tried it so I don't know, but there is a "optional - indicates negative duration." [17:59:37 CET] <teratorn> also whats a good ffmpeg front-end that runs on os x? :) [17:59:43 CET] <teratorn> n1cky_: thanks ill check it [18:11:37 CET] <flux> mavrik, ts works just out-of-the-box with my app, but apparently mpv gets easily confused by time stamps that don't start at 0. ffplay works fine, though. [18:12:13 CET] <flux> (even mpv works ok if said time stamps are inside an mp4) [19:02:37 CET] <Zucca> Do i need to specify -threads _after_ input files to make it function? I'm trying to encode vp9 video, but I'm only getting "one core hot". [19:03:15 CET] <BtbN> you don't need to specify it at all if you want it to use all cores. [19:03:17 CET] <DHE> yes, -threads is processed on a per-stream basis [19:03:40 CET] <Zucca> DHE: Thanks. [19:04:34 CET] <Zucca> So I guess if I specify it before inputs, it gets applied to decoding process. [19:05:30 CET] <DHE> but BtbN's point stands. unless you have specific requirements, the defaults are to autodetect number of threads to use [19:06:22 CET] <BtbN> libvpx isn't exactly good in using the CPU though [19:06:31 CET] <Zucca> Doesn't make sense. O.o If I specify threads before inputs I get one thread. [19:06:54 CET] <Zucca> So better not to specify it at all. [19:06:56 CET] <BtbN> no, that doesn't make sense. You get number-of-cores threads if you don't speficy it for the encoders. [19:19:25 CET] <Zucca> Dang... [19:19:38 CET] <Zucca> Anyway I do it, I get only one thread. [19:19:47 CET] <Zucca> At least one active. [19:20:48 CET] <kepstin> Zucca: it depends a bit on other encoding options, frame size, etc. whether having more threads is actually usable or will help [19:23:20 CET] <Zucca> I looked for the options from here: http://wiki.webmproject.org/ffmpeg/vp9-encoding-guide [19:24:02 CET] <Zucca> Multi-threaded encoding may be used if -threads > 1 and -tile-columns > 0. [19:24:26 CET] <Zucca> Last Modified: 2015-09-08 [19:25:00 CET] <Zucca> So I don't know if the information there is already obsolote/faulty. [19:27:29 CET] <kepstin> what size of video are you using? iirc, multithreaded vp9 has the biggest gains when you're doing HD or 4K video sizes. Playing around with the '-speed' setting might make a difference. [19:27:47 CET] <kepstin> could try explicitly setting -frame-parallel 1 as well. [19:29:09 CET] <Zucca> kepstin: -pass 1 -frame-parallel 1 -tile-columns 6 -speed 4 ... 1080p [19:29:28 CET] <Zucca> I wonder it -pass 1 makes the difference. [19:34:02 CET] <Zucca> Nope. [19:37:41 CET] <atomnuker> don't use frame-parallel, it reduces compression and using ffmpeg to decode actually reduces performance since it doesn't fully support that mode [19:38:04 CET] <atomnuker> IIRC it only works fine with libvpx [19:38:26 CET] <Zucca> I'm decoding from lossless format and encoding to vp9+opus, webm. [19:38:45 CET] <kepstin> hmm, fun. with x264, frame-parallel has long been the preferred mode :/ [19:39:43 CET] <kepstin> Zucca: be interesting to compare the performance of doing only the decode vs. doing the decode+encode. Some lossless codecs can be pretty heavy to decode iirc. [19:40:38 CET] <Zucca> kepstin: I'm using zlib video as input here. [19:41:22 CET] <Zucca> It's pretty tasking to decode, but I can playit real-time just fine. [19:42:50 CET] <Zucca> htop shows about 133% CPU usage in real time decoding. 600% being maximum (hexa-core CPU). [19:43:56 CET] <kepstin> hmm, so you're already saturating a single thread doing only the decoding. ffmpeg's core is single-threaded, so you might be hitting some bottlenecks due to the decoding? not sure. [19:45:00 CET] <Zucca> kepstin: input fps 35 and I'm gertting max of 1 fps encoding. [19:45:37 CET] <Zucca> With 1 fps decoding surely isn't the bottleneck. :\ [19:46:09 CET] <kepstin> but yeah, if doing frame threading in libvpx isn't recommended, the only other option would be slice threading, which of course *also* reduces the compression. [19:46:31 CET] <atomnuker> the frame-parallel variable for libvpx actually refers to decoding each tile in parallel, rather than each frame in parallel [19:46:45 CET] <kepstin> hmm. not sure how vp9 works actually [19:46:50 CET] <kepstin> i've only ever done vp8 stuff [19:46:59 CET] <atomnuker> ffmpeg supports only doing frames in parallel and will not do per-tile decoding in separate threads [19:47:42 CET] <Zucca> I removed it. Still getting one active thread. With -speed 4 I get maximum of 2.2 fps encoding with audio dropped out. [19:48:00 CET] <kepstin> Zucca: what's the video size? [19:48:04 CET] <kepstin> oh, 1080p [19:48:26 CET] <kepstin> if vp9 is doing some parallelization (per tile, i guess?) that should be big enough to show it [19:49:24 CET] <kepstin> but yeah, both vp9 and x265 are sort of 'next generation' codecs, and as such are both very slow [19:50:05 CET] <Zucca> Now there's something wrong... with vp8 (libvpx) and same settings I get 3.5 fps and one active thread. [19:50:10 CET] <kepstin> (or alternatively, if tuned to be the same speed as current generation codecs, don't really provide much in the way of quality improvements) [19:51:11 CET] <kepstin> Zucca: yes, vp9 is slower than vp8 at encoding :) [19:52:01 CET] <Zucca> kepstin: Yeah. I know. But I cannot get multithreaded encoding with vp8 either. And its speed is also _slow_. [19:53:22 CET] <kepstin> i seem to recall that libvpx-vp8 had worse multithreading ability than vp9 :/ [19:54:38 CET] <Zucca> I guess it's still libx264 that's most balanced in speed and quality. :( [19:55:13 CET] <Zucca> I just wished I could get bit better performance aout of thise free vp codecs. :| [19:55:30 CET] <kepstin> are you using "-quality good"? I forget what the default is, but you generally don't want to use "best" (it's akin to x264's placebo preset in that it makes it slower without any real gain in quality) [19:57:02 CET] <Zucca> My full cli line was/is: ffmpeg -y -i in.mkv -acodec opus -ab 48k -vcodec libvpx-vp9 -b:v 330k -tile-columns 6 -auto-alt-ref 1 -lag-in-frames 25 -speed 4 out.webm [19:57:29 CET] <Zucca> I've tried with -speed 1 also.- [19:57:45 CET] <Zucca> But that just slows it down (obviously). [19:57:48 CET] <kepstin> 330k for a 1080p video? that's probably not gonna give very good results... [19:58:04 CET] <Zucca> kepstin: Yeah. This it just for testing. [19:58:52 CET] <kepstin> how doe the performance compare when using a single-pass encode with -crf option set to something resonable? [19:59:01 CET] <Zucca> Giving it 3M doesn't affect the speed. :( [19:59:06 CET] Action: kepstin wouldn't expect much difference tho [19:59:31 CET] <Zucca> Let's try. [20:00:58 CET] <kepstin> (in general, i'd expect using -pass 1 shouldn't make the encode slower; in some codecs it can make it faster by disabling parts of the encoder that aren't needed for the first-pass bitrate analysis) [20:01:00 CET] <Zucca> Under 2 fps and single threaded. [20:01:23 CET] <Zucca> What it happening here..? :D [20:01:57 CET] <Zucca> I could try to compile libvpx instead of using prebuilt... [20:02:57 CET] <kepstin> what kind of cpu is it anyways? There might be some contention issues on AMD processors - video encoding is heavily using floating point, and every 2 "cores" on the amd chips share floating-point resources. [20:03:34 CET] <kepstin> although I'd still expect it to show up in top as using multiple cores [20:04:30 CET] <kepstin> but yeah, my general experience with the vpx codecs has also been "kinda slow and don't parallelize very well" :/ [20:06:17 CET] <kepstin> (even x264's frame-based threaded encoding mode isn't ideal in some cases; it reduces coding efficiency since it limits the area usable for reference, and adds latency such that it's not useful for "realtime" stuff) [20:06:32 CET] <Zucca> kepstin: I have AMD Phenom II X6 1055T 2.8GHz. The architecture in this CPU isn't like on those FX CPUs. Those six core don't share caches. [20:07:46 CET] <Zucca> I checked this from cpuworld. Each core is "independent". [20:08:07 CET] <kepstin> Zucca: yeah, i see why you want multithreading to work, that's a pretty slow/old cpu by modern standards :/ [20:08:14 CET] <jkqxz> Floating point what? Except for some legacy things, video encoding is entirely fixed point. [20:09:00 CET] <Zucca> kepstin: But you might be right about this... Since few of my friends managed to get multi-threaded encoding - they all have intel CPUs. [20:09:46 CET] <kepstin> jkqxz: ok, not floating point, but with assembly heavy use of the vector units which usually share resources with floating point hardware [20:10:37 CET] <Zucca> I'm gonna upgrade to ocata-core 4GHz FX CPU... But it just might make things worse because of the shared caches. Although all the specs and benchmarks indicate 20-60% boost in performance... [20:13:50 CET] <Zucca> One _really_ dirty thing to do would be to split the input and then run _seperate_ ffmpeg processes to encode raw streamable vp9 data. Fianlly stream copy all (concatenate) all the pieces and encode audio into a container. :P [20:14:37 CET] <Zucca> That would need some scripting. And woudn't be very good in compression... [20:14:54 CET] <Zucca> But hey! Doable! (Won't do it anyway). [20:16:31 CET] <Zucca> I did once a script that runs multiple optipng processes to speed up compressing multiple (or big in size) pngs. [20:16:41 CET] <Zucca> It was fun. [20:19:56 CET] <kepstin> Zucca: if you do it at intervals which are significantly longer than the keyframe interval, the compression loss would be somewhat minimal (assuming you're using "crf" mode for a quality based encode) [20:20:11 CET] <kepstin> obviously, two-pass bitrate encodes are out if you do that [20:22:11 CET] <Zucca> Yeah. Or well. You can use them, but the quality between chucks/parts might change very much. [20:23:36 CET] <Zucca> But yeah. Doing it using crf and seperate processes on a video file that's long enough... why not. I'm not just going to go there right now. [20:56:25 CET] <daslicht> hi, [20:56:49 CET] <daslicht> which data of a mp3 file do I need to be able to determine its bytes / second so that I can iterate over it in 1 second steps ? [20:58:35 CET] -daslicht:#ffmpeg- why does ffpmeg return 2 differnet bitrates for one file ? [20:58:35 CET] -daslicht:#ffmpeg- https://gist.github.com/daslicht/3d7e85fa2345655152b0#file-gistfile1-txt-L27 [20:58:35 CET] -daslicht:#ffmpeg- https://gist.github.com/daslicht/3d7e85fa2345655152b0#file-gistfile1-txt-L52 [21:17:54 CET] <kepstin> daslicht: one's from the container, the other's from the codec. In this case, it looks like the format one is just "filesize (bytes) / duration * 8" so it could be office, since it'll include the metadata headers, etc. [21:18:10 CET] <kepstin> so it could be off by a bit* [21:50:22 CET] <needmorespeed> I'm not finding any useful settings for H264 decoding speed. I've tried thread_count, doesn't seem to make a difference. Any suggestions? [21:51:20 CET] <TD-Linux> there aren't really any settings for it [21:51:59 CET] <kepstin> needmorespeed: it already decodes as fast as possible by default, since why would anyone want to make it slower? [21:52:02 CET] <Carll_> I would like to feed ffserver with some encoded packets.I'm using " av_interleaved_write_frame( format_out_ctx, outPacket ); " function to write a packet to an output media file. I can connect with the server with avformat_write_header( format_out_ctx, NULL ) function, and it is working fine (i think). If I pass that function, the server shows me " [POST] "/feed1.ffm HTTP/1.1" 200 4096 "" [21:52:06 CET] <Carll_> But my problem is, the av_interleaved_write_frame() function doesn't feed the server. [21:52:08 CET] <Carll_> Can anyone tell me what am i doing wrong? If somebody is willing to help me, I could send ffserver.conf and my C files. [21:53:32 CET] <kepstin> needmorespeed: if you can't cpu decode the video fast enough, you might want to look at a hardware decoder (vaapi, vdpau, qsv, whatever) [22:11:13 CET] <Carll_> someone? [22:21:29 CET] <Carll_> Can someone help me please? The following command is working "ffplay -rtsp_flags listen -i rtsp://127.0.0.1:8090/live.sdp". It streams my desktop. [22:21:37 CET] <Carll_> But i would like to feed the server [22:21:50 CET] <Carll_> but it doesn't work [23:49:50 CET] <Dan0maN> hello all. i have an IP camera that writes .jpg stills into my NAS. wanting to create an mpeg out of them. i found an example, but not working: http://www.itforeveryone.co.uk/image-to-video.html [23:50:31 CET] <Dan0maN> creating paste. one sec [23:52:17 CET] <Dan0maN> https://gist.github.com/anonymous/332c77ee54ffd25114a7 [23:52:59 CET] <Dan0maN> any help is appreciated. completely noob at ffmpeg [23:54:11 CET] <furq> Dan0maN: https://www.ffmpeg.org/ffmpeg-formats.html#Examples-1 [23:55:13 CET] <llogan> Dan0maN: you overwrote your inputs [23:57:05 CET] <Dan0maN> k. yeah, at least it prompted me, which why i copied them into a new directory first [23:57:10 CET] <Dan0maN> ty. i'll read up [23:58:59 CET] <Dan0maN> okay. i thought that %05d was an http encoding thing, but i guess those actually mean something to the cli? [23:59:42 CET] <llogan> 00001, 00002, 00003, etc [23:59:52 CET] <Dan0maN> yeah. trying to search for which one to use [00:00:00 CET] --- Tue Mar 15 2016

1 0

ffmpeg-devel.log.20160313
by burek 14 Mar '16

14 Mar '16

[00:18:29 CET] <ubitux> mmh it seems there is a subtitles format that look just like microdvd but uses [] instead of {} [00:18:35 CET] <ubitux> i wonder what's that format [00:20:51 CET] <wm4> vplayer? [00:22:25 CET] <ubitux> don't think so [00:23:56 CET] <ubitux> it really is similar to microdvd [00:23:57 CET] <ubitux> [3579][3607]Nie... piraci... [00:23:59 CET] <ubitux> [3672][3708]E.J. Lotterman|Redaktor naczelny [00:24:01 CET] <ubitux> [3728][3753]Nie teraz. [00:25:02 CET] <cone-869> ffmpeg 03Paul B Mahol 07master:4a7c705fde1a: avfilter/vf_vectorscope: make it possible to override colorspace [00:27:02 CET] <ubitux> they look quite common actually [00:27:48 CET] <ubitux> oh we actually support them, it's simply mpl2, i forgot i did this one as well [00:29:32 CET] <wm4> lol [00:30:14 CET] <wm4> if you make stats, I'd be interested what types of tags happen in which formats [00:30:24 CET] <wm4> ass, microdvd, html [00:49:47 CET] <ubitux> yep [00:52:08 CET] <Zeranoe> Lots of bugs for h264_qsv.... Another one was just posted on my forum [00:53:39 CET] <Compn> looks like mpl :P [00:53:50 CET] <Compn> surprised anyone uses it to be honest [00:54:02 CET] <Zeranoe> I am too [00:54:59 CET] <michaelni> jamrial, thx for e71096bb3ab4310999731dc53988cdcbd8aa4a72 [00:55:00 CET] <Zeranoe> Based on a recent email I got from Intel http://links.mail3.spopessentials3.com/servlet/MailView?ms=MjQ5MzY2NjQS1&r=… they're pushing it pretty hard, so that might be part of it. [00:55:17 CET] <Compn> i was talking about ubitux's subtitle ehehe [00:55:25 CET] <Compn> but yeah people want to use intel encoder [00:56:15 CET] <Zeranoe> From what I've found it produces worse quality than x264, but it does it faster [00:56:38 CET] <ubitux> mmh so the uncompressed dump of subtitles is 195G [00:56:44 CET] <Compn> nice [00:58:09 CET] <ubitux> which makes an average size of ~54kB per subtitles file [00:58:26 CET] <ubitux> this was the useless stat of the day [01:02:20 CET] <Compn> how many files total [01:02:33 CET] <Compn> must be thousands [01:02:39 CET] <Compn> :P [01:02:56 CET] <Compn> millions [01:06:06 CET] <ubitux> about 3.8 millions [01:08:16 CET] <Compn> i hope your harddrive fs can handle it [01:12:03 CET] <ubitux> well i'm not running on fat32 or worse (like hfs+) so it should be fine [01:12:31 CET] <Compn> are you going to write up a blog post detailing the great subtitle sample ? [01:12:36 CET] <Compn> i like reading such blog posts [01:14:43 CET] <ubitux> probably, but not anytime soon [01:15:02 CET] <ubitux> there are a lot of things to analyze, and probably a lot of things to fix in ffmpeg first :) [01:19:52 CET] <cone-869> ffmpeg 03Michael Niedermayer 07master:52e7f6b17eed: Revert "avfilter/vf_decimate: Check that input parameters match" [01:19:53 CET] <cone-869> ffmpeg 03Michael Niedermayer 07master:58cb1fb1fe92: avfilter/vf_decimate: Use the correct frame in difference calculation [01:33:27 CET] <ubitux> 1 [01:33:28 CET] <ubitux> 00:00:19,740 -> 00:00:22,740 [01:33:30 CET] <ubitux> omfg [01:33:42 CET] <ubitux> (supposed to be "-->", whole file like this) [01:34:04 CET] <Compn> haha [01:35:01 CET] <Compn> i think i've ran into a sub with that problem before [01:35:10 CET] <ubitux> 1 [01:35:11 CET] <ubitux> 00:00:-13,000 --> 00:00:04,881 [01:35:13 CET] <ubitux> lol [01:35:40 CET] <Compn> how do you know the video DOESNT start with -13 timestamp ? [01:35:42 CET] <Compn> how do you KNOW [01:36:13 CET] <ubitux> dammit this archive is a fucking gold [01:36:27 CET] <ubitux> 00:00:46,913-->00:00:49,006 [01:36:44 CET] <Compn> no spaces [01:36:46 CET] <Compn> i kind of want a copy. [01:37:01 CET] <ubitux> i can't share it [01:37:13 CET] <Compn> ah dang [01:37:18 CET] <ubitux> A [01:37:19 CET] <ubitux> 00:00:01,011 --> 00:00:04,074 [01:37:21 CET] <ubitux> omfg @!# [01:37:33 CET] <ubitux> 'A' [01:37:37 CET] <Compn> the next line isnt B is it ? [01:37:40 CET] <ubitux> no... [01:37:43 CET] <ubitux> next are 2, 3, 4... [01:37:48 CET] <ubitux> but the first one is a fucking A [01:37:48 CET] <Compn> how are you finding the non standard files btw ? [01:37:50 CET] <Compn> lol [01:37:58 CET] <ubitux> i'm feeding them to ffprobe [01:38:04 CET] <ubitux> and look at the failing ones [01:38:06 CET] <Compn> edited ffmpeg to output all non standard lines [01:38:06 CET] <Compn> ah [01:38:21 CET] <ubitux> [01:38:23 CET] <ubitux> Warning: filesize() [<a href='function.filesize'>function.filesize</a>]: stat failed for /24.401-med.srt in C:\Szerver\xampp\htdocs\downloadsubs.php on line 12 [01:38:25 CET] <ubitux> lol [01:39:27 CET] <ubitux> omg, some guy edited a file with wordpad [01:39:37 CET] <Compn> wordpad added cruft ? [01:39:39 CET] <Compn> ehe [01:39:41 CET] <ubitux> yeah [01:39:56 CET] <Compn> yep gonna be a lot of crlf vs lf no doubt [01:40:23 CET] <ubitux> well i think it's wordpad [01:40:37 CET] <ubitux> http://b.pkh.me/b44c42f400d6e45850d75543c3eb93a5 [01:41:09 CET] <ubitux> damn i really need to push my fix for vplayer [01:41:50 CET] <ubitux> i see a non negligible amount of srt files where the first chapter is simply absent [01:41:55 CET] <ubitux> and it starts right away with a timing [01:42:44 CET] <Compn> so many different subtitle softwares out there, you're going to see some weird stuff [01:43:07 CET] <Compn> country specific stuff. [01:43:24 CET] <ubitux> heh there are some webvtt files [01:44:52 CET] <Compn> any real text .rt ? [01:45:30 CET] <ubitux> Compn: none so far, but it's going to take a while to probe everything [01:45:32 CET] <ubitux> -1 [01:45:34 CET] <ubitux> 00:00:03,320 --> 00:00:39,845 [01:45:37 CET] <ubitux> omg this is a joke [01:45:38 CET] <Compn> haha [01:45:46 CET] <Compn> does it go to 0 or just 1 ? [01:45:57 CET] <ubitux> -1 0 1 2 ... [01:46:08 CET] <Compn> better than A > 1 then [01:46:16 CET] <Compn> at least its some kind of pattern [01:46:48 CET] <Compn> er A > 2 [01:47:16 CET] <ubitux> they really like negative stuff [01:47:18 CET] <ubitux> 00:00:-5,-601 --> 00:00:03,157 [01:47:32 CET] <ubitux> i have a lot of them like this [01:48:12 CET] <ubitux> nice, a file starting with microdvd subtitles [01:48:20 CET] <ubitux> and then suddenly it becomes srt [01:48:31 CET] <Compn> not at the 1cd / 2cd mark? [01:50:15 CET] <ubitux> no, just one line like this: http://b.pkh.me/555de38a7c4bddfe8c3c964948a9495b [01:51:23 CET] <ubitux> oh nice, twice the utf-8 bom [01:52:28 CET] <ubitux> some files are so broken [01:52:32 CET] <ubitux> they are detected as AAC [01:52:40 CET] <Compn> good for fixing aac probe then [01:53:02 CET] <Compn> are we allowed to add filesizes to the probes (when available) ? [01:53:20 CET] <ubitux> probing is only buffer based [01:53:31 CET] <ubitux> like, content only, that's all you get [01:54:00 CET] <wm4> more like aac probing is so broken it reacts to subtitle files [01:54:11 CET] <wm4> at least it's not mp3 [01:55:33 CET] <Compn> right [02:06:10 CET] <ubitux> http://b.pkh.me/3c127e35bcce41698fb9665e32df55ce lol [02:08:01 CET] <ubitux> http://b.pkh.me/3a034b410c18041904976930324f6562 [02:08:53 CET] <Compn> mixing html / ass type and srt type [02:08:56 CET] <Compn> hehe [02:10:11 CET] <ubitux> http://b.pkh.me/6f990b214cb514f35ff4de20c636a49a [02:10:23 CET] <ubitux> some people are basically thinking these files are free editable zones [02:11:08 CET] <ubitux> i actually see this pattern a lot... [02:11:19 CET] <ubitux> it's generally 2 or 3 lines [02:17:22 CET] <ubitux> i think i saw enough for now [04:47:03 CET] <alexandre_> Good night, I'm interested in participating gsoc and work on ffmpeg. Looking out for info with a possible mentor. Thanks in advance. [05:01:10 CET] <Compn> alexandre_ : did you see the ffmpeg gsoc wiki ? [05:01:31 CET] <Compn> alexandre_ : read this page https://trac.ffmpeg.org/wiki/SponsoringPrograms/GSoC/2016 [05:01:45 CET] <Compn> anyone want to write a gsoc/opw news entry for ffmpeg.org ? [05:01:50 CET] <Compn> we need one badly... [05:05:26 CET] <alexandre_> thanks for answering, i saw the wiki. I will try to reach a mentor directly and sorry for spamming [05:05:46 CET] <Compn> its ok, feel free to ask any questions here [05:05:57 CET] Action: Compn goes afk [08:15:58 CET] <ubitux> http://b.pkh.me/bb8bc4a6e0e845517943fb49be5465b6 [08:16:02 CET] <ubitux> i don't get it. [08:22:53 CET] <wm4> hilarious [08:23:36 CET] <wm4> just wtf must have been going on in this user' shead [08:24:58 CET] <wm4> and what program parses this... oh well the world is a bad place [09:06:08 CET] <JEEB> wow [09:42:36 CET] <cone-921> ffmpeg 03Carl Eugen Hoyos 07master:db4786f1cf03: lavc/hevc_ps: Support GBR decoding. [09:42:36 CET] <cone-921> ffmpeg 03Carl Eugen Hoyos 07master:d11d78facb1b: lavc/libx265: Support GBR encoding. [09:50:13 CET] <durandal_170> dev can be only mentor for one project [10:15:35 CET] <cone-921> ffmpeg 03Thomas Mundt 07master:5024a82e9548: avfilter/vf_bwdif: add x86 SIMD [11:29:56 CET] <cone-921> ffmpeg 03Paul B Mahol 07master:912fec3e54ae: avfilter/vf_waveform: add acolor filter [13:13:31 CET] <cone-921> ffmpeg 03Mats Peterson 07master:f680c8e41a9b: tests/ref/vsynth: Remove unused file [13:13:32 CET] <cone-921> ffmpeg 03Mats Peterson 07master:ed5d295a6983: lavf/riffenc: Improve spec compliance; Fix WMP playback of AVI with xxpc chunks [13:16:17 CET] <cone-921> ffmpeg 03Carl Eugen Hoyos 07master:4405ff2cae50: lavc/hevc_ps: Fix default display window offsets for yuv422 and yuv444. [16:06:53 CET] <cone-921> ffmpeg 03Mats Peterson 07master:9ee5b4096476: avformat/avienc: factor out avi_add_ientry() [16:06:54 CET] <cone-921> ffmpeg 03Mats Peterson 07master:77bc3e1944a9: lavf/avienc: Add xxpc entries to index [16:47:39 CET] <cone-921> ffmpeg 03Mats Peterson 07master:bdc798cc8d00: lavf/avienc: Clear whole tag in avi_add_ientry() [17:41:53 CET] <cone-921> ffmpeg 03Ganesh Ajjanagadde 07master:0c33ceda91dd: lavu/lfg-test: add simple sample mean, stddev checks [18:20:26 CET] <cone-921> ffmpeg 03Reimar Döffinger 07master:7c93f2c0b940: Move cbrt tables to separate cbrt_data(_fixed).c files. [19:27:27 CET] <omerjerk> I have a question regarding libavcodec. [19:28:12 CET] <omerjerk> Support there's a decoder (being specific to alsdec.c), will it always convert to .wav only ? [19:31:48 CET] <BtbN> decoders decode to raw audio, that's their job. [19:31:56 CET] <nevcairiel> it decodes to raw pcm [19:32:04 CET] <JEEB> libavcodec either decodes in some way encoded things into one of libavcodec's raw format [19:32:12 CET] <nevcairiel> which can then be processed into .wav, or re-encoded into any other format [19:32:12 CET] <JEEB> or encodes raw formats to some coded format [19:33:44 CET] <omerjerk> okay, so my qualification task is to add floating point support in the alsdec. [19:34:05 CET] <omerjerk> I know how to generate the floating number from the bitstream. [19:35:04 CET] <omerjerk> but, I'm not able to understand how to exactly return that sampled float data to ffmpeg for it to write. [19:36:03 CET] <omerjerk> there's this ALSBloackData struct - https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/alsdec.c#L244 [19:40:27 CET] <omerjerk> so my question is, if I write float value to *dest, will everything work ? - https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/alsdec.c#L1554 [19:40:44 CET] <omerjerk> or is there some other place too where I'll need to make changes. [19:40:51 CET] <omerjerk> Just asking in a general sense. [19:41:08 CET] <omerjerk> A general answer would suffice. [19:46:17 CET] <wm4> yes, but you have to set the sample format to a float one too [19:48:26 CET] <omerjerk> hey. I'm really sorry. My internet got disconnected. [19:48:42 CET] <omerjerk> I didn't see any messages if anyone replied after my last message. [19:48:49 CET] <fritsch> 19:46 <+wm4> yes, but you have to set the sample format to a float one too [19:48:53 CET] <fritsch> ^^ was the only answer [19:49:31 CET] <omerjerk> Thanks a lot. [19:49:35 CET] <omerjerk> Where do we have to set that ? [19:49:46 CET] <fritsch> :-) if it's made too easy [19:49:51 CET] <fritsch> you don't learn anything [19:49:54 CET] <fritsch> while implementing it [19:51:14 CET] <omerjerk> Okay. I'll see. Hopefully I'll be able to find it. :) [19:51:16 CET] <omerjerk> Thanks. [20:12:43 CET] <cone-921> ffmpeg 03Paul B Mahol 07master:80974559294f: avfilter/vf_vectorscope: copy frame props, mainly for bench filter [20:12:44 CET] <cone-921> ffmpeg 03Paul B Mahol 07master:55c872146d45: avfilter/vf_vectorscope: process alpha at end [20:12:45 CET] <cone-921> ffmpeg 03Paul B Mahol 07master:370cecc1e87d: avfilter/vf_vectorscope: less aggressive memory allocation [00:00:00 CET] --- Mon Mar 14 2016

1 0

ffmpeg.log.20160313
by burek 14 Mar '16

14 Mar '16

[01:53:26 CET] <jonga> witch codecs must i choose when i stream a webm video to icecast with ffmpeg ? [01:54:14 CET] <J_Darnley> vp8 or vp9 (and theora) are the only ones supported in webm [02:03:03 CET] <jonga> ok thanks [04:28:22 CET] <dongs> how do I downmix 16 channel wav into something usable? [04:33:27 CET] <kepstin> dongs: well, first you have to decide how to mix it, and how many output channels... [04:34:00 CET] <kepstin> dongs: you might be best off opening it in audacity or something so you can play with the channel gains and pan and whatnot [04:42:19 CET] <alexandre_> Good night, I'm interested in participating gsoc and work on ffmpeg. Looking out for info with a possible mentor. Thanks in advance. [04:48:45 CET] <dongs> kepstin: does audicity even open 16ch wav [04:48:59 CET] <dongs> i hope there's at least usable left/right stuff [04:49:47 CET] <kepstin> where'd you get this/what's in it? [04:49:57 CET] <kepstin> 16ch isn't normally something meant for direct playback [04:50:10 CET] <dongs> extract from mxf [04:50:34 CET] <kepstin> oh, fun, movie theatre sounds system stuff probably then [04:50:38 CET] <dongs> i have written stuff to save mxf>wav, but for final downmix i'll probly just use ffmpeg native mxf reader + downmix [04:50:41 CET] <dongs> yeah [04:50:51 CET] <kepstin> I have no idea what the channel layout for that even looks like [04:50:54 CET] <dongs> i guess ill download audishitty and see [04:51:15 CET] <kepstin> you'd probably want to use the pan filter in ffmpeg so you can do a custom mix with your own coefficients [04:51:19 CET] <dongs> would i be able to run ffmpeg to extract channel by channel? [04:51:20 CET] <kepstin> once you figure out something useful [04:51:27 CET] <dongs> if I cant get .wav opened [04:52:18 CET] <kepstin> dongs: you should be able to, either using the -map_channel option or some filters [04:53:37 CET] <dongs> hey hah, it can ope nit [04:53:55 CET] <dongs> and yea there's stuff on al l16 channels, haha [04:54:15 CET] <dongs> but looks like first 6 or so are strongest [04:54:24 CET] <dongs> so maybe its the usual 5.1 mix and then other shit is extra [05:00:33 CET] <kepstin> there's probably some documentation for the channel layout somewhere [05:05:08 CET] <kepstin> dongs: you may find https://tech.ebu.ch/docs/r/r123.pdf interesting, it some channel mappings used in mxf [05:06:31 CET] <kepstin> although that's more intended for tv broadcast stuff, so it might not match your file [06:03:24 CET] <dongs> kepstin: yea, t he firs6 channels are L R C MFE LS RS [06:03:31 CET] <dongs> er MFE=LFE [06:03:35 CET] <dongs> good enough, i can just ignore the other stuff [06:04:02 CET] <kepstin> dongs: should be easy enough to use the -map_channel option to turn that into a standard 5.1 layout with ffmpeg then [06:04:10 CET] <kepstin> and drop the extra channels [06:06:17 CET] <dongs> yep [06:06:35 CET] <dongs> there's also what looks like a bitstream in one of the channels [06:06:43 CET] <dongs> cuz its just high dB noise [06:07:00 CET] <dongs> maybe thats DTS or some other junk [06:09:48 CET] <johnnny22-afk> what are the file formats or file structures that would support a sliding window mechanism ? [06:10:39 CET] <johnnny22-afk> technically it's not possible with a single file structure right ? [06:11:05 CET] <dongs> sliding window of what [06:12:09 CET] <kepstin> johnnny22-afk: it's theoretically possible to do in a single file with linux's new "hole punch" mechanism, but i've never seen that implemented for video [06:12:33 CET] <kepstin> johnnny22-afk: usual way is to use segmented files (similar to what hls/dash do) and delete old segments while writing new ones [06:12:50 CET] <johnnny22-afk> dongs: to do what kepstin just described :) [06:13:22 CET] <johnnny22-afk> kepstin: "hole punch" mechanisme !? humm [06:13:55 CET] <dongs> o [06:14:00 CET] <kepstin> johnnny22-afk: linux has an api that lets you remove a section from the start or middle of a file, deleting it from disk [06:14:32 CET] <johnnny22-afk> kinda just started reading on it (googled). very interesting. [06:15:11 CET] <johnnny22-afk> I wonder how accessing such a file would be handled by a player like mplayer. [06:16:16 CET] <kepstin> I think there's a way to do it such that it actually moves the file offset so that the kept section will start at 0. THen it should just work if you're using a format like mpeg-ts which can be cut arbitrarily. [06:16:35 CET] <kepstin> if you don't reset the offset, it'll fail, because it'll be a sparse file starting with a long section of 0s [06:16:50 CET] <johnnny22-afk> humm, very very interesting [06:17:12 CET] <johnnny22-afk> I could limit a cache to be say 1 gig or so like that. [06:17:50 CET] <johnnny22-afk> I guess I'll have to try attempting to populate such a file and see how mplayer handles it :) [06:18:11 CET] <kepstin> you'd have to make sure you're using appending writes to the file, i'm not sure how that interacts with the hole punching. [06:18:47 CET] <johnnny22-afk> though, i'm not sure how or what you mean exactly by moving it's file offset, but I bet i'll encounter it while reading :) [06:20:45 CET] <kepstin> if you have a file like 'XXXAAAA', you should be able to do an fallocate hole-punch starting at byte 0 and 3 bytes long. That would normally give you a file like '000AAAA' where the 000 isn't on disk (sparse file). [06:21:06 CET] <kepstin> but there's a "collapse" flag that would give you a file like 'AAAA' instead, i.e. move the A bits to start at byte 0 [06:21:53 CET] <kepstin> (it obviously doesn't physically move them on disk, it just edits the extent ranges in the file entry in the filesystem) [06:22:36 CET] <dongs> can I do -ac 6 to just take fisrt 6 channels? [06:23:44 CET] <kepstin> dongs: i'm not sure. [06:24:11 CET] <dongs> im looking @ https://trac.ffmpeg.org/wiki/AudioChannelManipulation but i dont see anything with >6 channels exampels ;d [06:24:20 CET] <dongs> and id rather not extract to 6 .wavs first then re-merge [06:25:01 CET] <kepstin> no need to do that, just specify 6 -map_channel options to pull the channels you want in the right order [06:26:21 CET] <kepstin> might also need -ac6 in combination with that, not sure. [06:26:30 CET] <kepstin> "-ac 6" [06:27:30 CET] <johnnny22-afk> kepstin: So, I could append to a file periodically and when I do that remove part of the head of the file with some fallocate hole-punch trickery while I have a player like mplayer that plays a file and that player would continue playing smoothly forward for as long as it can ? [06:27:51 CET] <johnnny22-afk> as long as it can = as long as there is more data appended to the end ? [06:28:31 CET] <kepstin> johnnny22-afk: that's a good question. If you do the "collapse", I don't know if another program reading the file would make its next read at the same relative position, or suddenly be reading from somewhere else [06:28:41 CET] <kepstin> without the "collapse" flag, that would definitely work fine [06:28:53 CET] <kepstin> (but you wouldn't be able to play from the start with a new player) [06:30:20 CET] <johnnny22-afk> I'm assuming the client would seek from the same absolute position, so it wouldn't be relative to what's been removed in the head.. if that would be the case, would it suddenly jump ? [06:30:42 CET] <dongs> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16 (24 bit), 4608 kb/s [06:30:45 CET] <dongs> looks like it works [06:30:46 CET] <dongs> (ac6 [06:30:52 CET] <kepstin> the problem is that if you do a collapse, the contents of the file at the same absolute position changes [06:31:11 CET] <johnnny22-afk> right, ok [06:31:41 CET] <johnnny22-afk> so, I'd have to make mplayer or some player seek backward by that same offset at the moment this happen. If that's how it behaves. [06:32:09 CET] <kepstin> yeah, i'm not sure how well you could get that to work [06:32:19 CET] <johnnny22-afk> With collapse, an new player instance would need to start playing at a different offset than 0 then ? [06:32:35 CET] <kepstin> this is why doing something like HLS (a playlist file and segments) is common, since it's already handled correctly by players [06:32:37 CET] <johnnny22-afk> i mean, without collapse [06:33:10 CET] <kepstin> yeah, you'd have to remember the most recent cut offset to somehow tell the player to seek that many bytes before reading. [06:33:29 CET] <johnnny22-afk> which is available with -sb on mplayer for example [06:33:49 CET] <kepstin> make sure it's using 64bit offsets :) [06:34:09 CET] <johnnny22-afk> but without collapse would the file size stay the same ? I wouldn't want my hdd to get full. [06:34:45 CET] <kepstin> yeah, it would actually deallocate that section of file from disc so it can be reused [06:34:52 CET] <johnnny22-afk> i'm guessing the filesize increases without collapse :o? [06:35:13 CET] <dongs> now if I can just figure out how to trim 192 samples from start and end [06:35:36 CET] <kepstin> the file length increases, but the size of the data on disk won't [06:35:50 CET] <kepstin> i.e. the number with ls -l will go up, the number with du will stay the same :) [06:36:03 CET] <dongs> or rather, 192 frames @ 24fps, but without having video. [06:36:03 CET] <dongs> hm [06:36:34 CET] <johnnny22-afk> kepstin: would the OS support a file length that is bigger than the available space ? [06:36:53 CET] <kepstin> dongs: the 'atrim' filter can take a number in exact samples [06:37:04 CET] <kepstin> johnnny22-afk: most linux filesystems have supported that for a long time [06:37:26 CET] <kepstin> read up on "sparse files" :) [06:37:27 CET] <dongs> i guess ih ave to calculate it [06:37:30 CET] <johnnny22-afk> kepstin: k k :) im quite ignorante :) [06:38:06 CET] <johnnny22-afk> kepstin: now i gotta figure out if centos 7 supports that :o/ [06:38:51 CET] <kepstin> dongs: 48000Hz / 24Hz * 192 = 384000 samples. [06:39:28 CET] <dongs> mkay. lets see. [06:40:04 CET] <johnnny22-afk> seems like it's using xfs in it's /home [06:40:06 CET] <johnnny22-afk> :) [06:40:54 CET] <johnnny22-afk> now... I assume ffmpeg doesn't support that ? :) [06:41:12 CET] <johnnny22-afk> would be a very cool addition if it doesn't :D [06:41:18 CET] <dongs> kepstin: we have a winner on start_sample [06:41:30 CET] <dongs> i dont suppose end_sample will take a negative offset to mean end of file [06:42:00 CET] <johnnny22-afk> dongs : it just might ! :D [06:42:19 CET] <kepstin> dongs: nothing in the docs :) [06:42:23 CET] <dongs> right [06:42:43 CET] <kepstin> usually you don't have to be as exact on the end, so you could just set a time. [06:43:06 CET] <dongs> no, it must be exact because its part of like 6 reels .mxf [06:43:12 CET] <kepstin> or if you're muxing it with a video that's already cut, you can just use -shortest and have ffmpeg trim it for you [06:43:14 CET] <dongs> but t here's 192 frames padding on both ends [06:43:18 CET] <kepstin> hmm, that's annoying :/ [06:44:16 CET] <dongs> [atrim @ 000001eb9903fa60] Error setting option end_sample to value -384000. [06:44:17 CET] <dongs> :) [06:44:34 CET] <dongs> can I somehow get length in samples after 1st triM? [06:44:54 CET] <kepstin> yeah, i don't think there's a good way to do that other than count the samples in the file and doing some manual math to create the command [06:45:37 CET] <johnnny22-afk> -shortest as kepstin might do it :) [06:45:45 CET] <johnnny22-afk> the trick. maybe [06:45:57 CET] <dongs> yeah, except i have video already processing separately. mxf is separate video +audio packages anyway [06:46:16 CET] <dongs> theres no relation except playlist that combines them [06:47:07 CET] <dongs> thats ok, ill do fist step of dropping 192 samples and from start and then see about trimming the end. [06:49:34 CET] <johnnny22-afk> kepstin : reading https://gist.github.com/CAFxX/571a1558db9a7b393579 [06:53:52 CET] <dongs> the 'sample' window in audacity is a joke [06:53:57 CET] <dongs> they used some weird custom draw control [06:54:03 CET] <dongs> so instead of displaying a proper number its just.. trash [06:54:06 CET] <dongs> because too many samples [06:54:28 CET] <kepstin> there might be some way to get the number out of ffprobe in a scriptable way [06:54:43 CET] <dongs> i only need to do this once really (well, *6) [06:56:05 CET] <dongs> this is so bizzare, if you zoom into audacity it allows the cursor to go betwee n samples [06:56:12 CET] <dongs> ... [06:56:37 CET] <dongs> i guess nobody actually uses it for anything serious [06:56:42 CET] <dongs> time to figure out how to do this all in ffmpeg [06:57:12 CET] <kepstin> the cursor can only ever possibly be between samples, it doesn't make sense to have it "on" a sample since samples are basically points [06:57:54 CET] <dongs> http://i.imgur.com/ZxSNNJB.png [06:57:59 CET] <dongs> each "dot" there is sample [06:58:04 CET] <dongs> but it allows cursor to be between them [06:58:20 CET] <dongs> anyway looks like end trim is only 4k samples, not 384000 [06:58:45 CET] <kepstin> weird that the selection is right on a dot, that doesn't make sense [06:59:00 CET] <dongs> no, i seeked right to the dot [06:59:03 CET] <kepstin> a sample has to either be in a selection or outside, it can't be "on the edge" [06:59:59 CET] <kepstin> but yeah, i wouldn't rely on audacity for counting exact samples of something [07:00:05 CET] <dongs> but beacuse of this i cant use it anyway http://i.imgur.com/wZyzeia.png [07:00:19 CET] <dongs> because they used some weird customdraw counter that only shows stuff if your file is like 30 seconds long [07:00:28 CET] <dongs> not a normal textbox [07:00:32 CET] <kepstin> particularly with its builtin mp3 decoder, since it gets the trimming at the start/end wrong (adds silence to start, cuts a bit off the end) [07:01:10 CET] <dongs> ffprobe is just ffmpeg symlinked right? [07:01:23 CET] <dongs> or does it acutally do something differnt [07:01:41 CET] <kepstin> it's actually a different tool [07:01:58 CET] <kepstin> (obviously most of the work is done in the shared library for both) [07:22:35 CET] <dongs> got it all trimmed and it should match up with video, thanks dudes. [08:17:44 CET] <asm89> my google fu is failing me [08:17:56 CET] <asm89> any good examples of setting up an hls realtime stream with ffserver? [08:25:28 CET] <pzich> I've never done it, but here's what I'm finding for 'ffserver hls' on Google, have you been to all of these? http://stackoverflow.com/a/1098222 http://ffmpeg.gusari.org/viewtopic.php?f=12&t=914 http://www.alkannoide.com/2013/07/04/play-with-ffserver-a-quick-overview/ [08:30:35 CET] <fling> Is it fine to capture dual-angle video using two webcams? [08:30:58 CET] <fling> Is it also fine to use them both for mastering stereo from two mono channels? [08:51:41 CET] <asm89> pzich: nothing that really worked yet, thanks for checking along with me though [09:49:32 CET] <osense> hello [09:49:57 CET] <osense> is it possible to create a "live", so to speak, timelapse video with ffmpeg? [09:50:20 CET] <osense> I.e. capture frames at a low framerate from /dev/video0, but encode them into the video at a higher framerate, e.g. [09:50:25 CET] <osense> ffmpeg -framerate 1 -i /dev/video0 -r 60 out.mp4 [09:50:48 CET] <osense> I get some messages that the driver change the 1 fps to 30 fps [09:51:25 CET] <osense> maybe I better use some other tool to capture stills from the camera and afterwards use ffmpeg to create the video from them? [09:51:59 CET] <solrize> i just built from master and got a lot of compiler warnings about OOB array subscripts.... is that ok? known? [10:20:44 CET] <tumpysans> i need HELP, does anybody have ffmpeg binary with shine-mp3 encoder enabled [10:37:51 CET] <rivarun> hello. i'm trying to convert a 60fps video to 30fps (among other downconversions to save size). after the conversion, ffmpeg reports the output file as being 60fps. http://pastebin.com/KCvVzmDW [10:46:24 CET] <rivarun> hmm, i think -r is what i wanted [10:46:58 CET] <tumpysans> can ffplay playback bluray ? [14:04:57 CET] <tumi> how much ram do you need to build ffmpeg? is 128 MB definitely too much or it could work? I don't care about the speed as long as it doesn't get killed due to lack of memory [14:05:13 CET] <tumi> *too low of course [14:05:22 CET] <tumi> not "too much" [14:05:32 CET] <JEEB> just have enough swap [14:06:03 CET] <JEEB> but generally 128MiB is really limited for *building* pretty much anything [14:06:46 CET] <tumi> enough swap as in how much? 1 GB? 4 GB? [14:07:07 CET] <jkqxz> It will very much depend on your compiler and compile options. [14:07:13 CET] <tumi> what I care about is x264 encoder, and maybe libopus [14:07:25 CET] <JEEB> really depends on how many processes you do at once [14:07:55 CET] <JEEB> and I don't think anyone really has calculated how many MiBs a single GCC/ld/whaetver process takes with the version you have [14:07:55 CET] <tumi> the "-j n" parameter? [14:07:59 CET] <jkqxz> (If you use an old compiler which people were using when normal machines had 128MB of RAM then you'll be just fine.) [14:08:34 CET] <JEEB> I'm just gonna say that if you're doing this to compile on an embedded device, don't [14:09:00 CET] <JEEB> give the build box enough RAM and a few gigs of swap just in case [14:09:10 CET] <JEEB> like I would not go under 1GiB for any linux system now [14:09:11 CET] <tumi> not really embedded, just a really low end vps [14:09:49 CET] <JEEB> anyways, you'd have to measure it yourself with the stuff you're doing and the versions of things you're going to be using [14:10:08 CET] <tumi> that makes sense I guess [14:11:01 CET] <tumi> I also tried to build it on my android phone but the configure script kept complaining about being unable to create a temp file in /tmp [14:11:18 CET] <tumi> even though mktemp seemingly works correctly [15:09:59 CET] <lollisoft> Hi, I have a problem with a screen recording made with 1.16 fps and then edit that video using blender to as my tool of choice. The fps in blender is 10. Can I convert my source easily with ffmpeg? [15:29:07 CET] <J_Darnley> I don't understand your problem. [15:29:14 CET] <J_Darnley> Are the conversion too slow? [15:29:25 CET] <J_Darnley> lollisoft ^ [15:31:34 CET] <lollisoft> No, just starting with a screen capture avi file I have converted to mov. But the framerate is at 1.16 fps and I need 10 fps. I just do build ffmpeg as I have read it would be possible to use this tool to convert 1 to 25 or in my case 1 to 10. (Neclecting the 1.16 fraction)... [15:31:37 CET] <dsl42> -r 10? but the video would be ~10 times faster, or? [15:32:26 CET] <J_Darnley> If you want to duplicate frames to reach 10 fps then just supply -r 10 when encoding [15:32:27 CET] <lollisoft> It is if I do not convert the framerate, because I have a animation at 10 fps and blender is not capable to handle both :-( [15:32:54 CET] <J_Darnley> if you want to speed up the video, then use one of the filters, fps maybe [15:33:02 CET] <lollisoft> Yes, that is what I attempt to do. [15:35:39 CET] <lollisoft> Ahh, I mean make 1 frame to 10 frames and then take the next and so forth. If that is -r 10 doing, then I will have success. Now I try, as macport has build ffmpeg. [15:41:53 CET] <lollisoft> It is telling to be 10 fps, but I have only black screen. All images seem to get lost. My command was ffmpeg -i Product\ Installation\ Windows.avi -r 10 Product\ Installation\ Windows.mov [15:42:48 CET] <J_Darnley> That won't work like you think [15:42:57 CET] <lollisoft> Hmm [15:43:02 CET] <J_Darnley> you can't have the same file as output and input [15:43:20 CET] <J_Darnley> oh its not [15:43:23 CET] <lollisoft> avi to mov. [15:43:24 CET] <dsl42> :) [15:52:34 CET] <lollisoft> It seems to work. Quicktime player can't read the file (eg. black screen), but blender shines here. The mouse movements and speed is as expected. I'll render it now and see if Quicktime then can play the rendered video. [16:18:33 CET] <DHE> other than the main ffmpeg.c, are there code examples of "how to transcode" using libav*? [16:19:42 CET] <furq> http://ffmpeg.org/doxygen/3.0/decoding_encoding_8c-example.html [16:20:48 CET] <DHE> okay.. and libavformat should (hopefully) be rather straight-forward [16:21:33 CET] <furq> http://ffmpeg.org/doxygen/3.0/transcoding_8c-example.html [16:21:35 CET] <furq> there's that as well [16:21:53 CET] <DHE> oh, this might be what I'm looking for [16:21:55 CET] <DHE> thx [16:22:05 CET] <furq> http://ffmpeg.org/doxygen/3.0/examples.html [17:12:50 CET] <Thue> I have created the video https://thuejk.dk/output_10Mbs_quality_good.webm via the command "ffmpeg -r 60 -pattern_type glob -i '*.time.jpg' -quality good -threads 4 -b:v 10M -c:v vp9 output_10Mbs_quality_good.webm". It plays fine in VLC, but Chrome and Firefox won't play it. Firefox says "Video can't be played because the file is corrupt". [17:13:41 CET] Action: J_Darnley guesses thet yuv444p was used [17:13:49 CET] <furq> Thue: -pix_fmt yuv420p [17:14:17 CET] <Thue> Fire Chrome and Firefox does not support the default VP9 format output by ffmpeg? [17:14:28 CET] <Thue> s/fire/so/ [17:14:44 CET] <furq> the default pixel format is the same as the input [17:14:59 CET] <Thue> Ah, thanks :) [17:15:03 CET] <J_Darnley> Or the "best" one supported by the encoder [17:15:08 CET] <furq> yeah [17:15:33 CET] <Thue> Well, that was easier to debug than expected (for you, at least). Thanks :) [17:16:03 CET] <J_Darnley> Only because people come here with the same problem every other day. [17:16:10 CET] <furq> you generally need to specify pix_fmt whenever you convert from images [17:16:41 CET] <J_Darnley> It would be better if browsers supported more or at least told you what the fucking problem is. [17:16:42 CET] <Thue> The error messages (or lack of same) in the browsers are really not helpful, either [17:17:19 CET] <furq> there are open bugs in firefox and chrome to support yuv422/444 but i can't see them going anywhere [17:17:45 CET] <Thue> Are there open bugs in firefox and chrome to make them output better error messages? [17:18:09 CET] <furq> that would make life far too easy [17:18:24 CET] <Thue> Yes, that would be too obvious. What were I thinking? [17:31:11 CET] <Thue> The chrome and firefox bugs for yuv444 support I can find both claim to be fixed: https://bugs.chromium.org/p/chromium/issues/detail?id=104711 https://bugzilla.mozilla.org/show_bug.cgi?id=497023 [19:49:30 CET] <joko> Hello, I am trying to quickly convert the AC3 audio channel of a mkv file to AAC, but w/o any luck... ffmpeg input.mkv -c:v copy -c:a libfdk_aac -vbr 3 output.mkv is what I type, "Output file #0 does not contain any stream" is the message I get [19:50:21 CET] <relaxed> joko: -i input.mkv [19:51:29 CET] <joko> relaxed: doh, my bad [19:52:49 CET] <J_Darnley> I wonder if ffmpeg already overwrote it [19:54:26 CET] <joko> J_Darnley: nope, it seems to be working ;) [20:13:49 CET] <TD-Linux> thebombzen, firefox has fixed 4:4:4 in firefox 46 iirc [20:13:58 CET] <TD-Linux> err that's for Thue [20:14:31 CET] <TD-Linux> this is the bug for vp9 444 https://bugzilla.mozilla.org/show_bug.cgi?id=1190939 [00:00:00 CET] --- Mon Mar 14 2016

1 0

ffmpeg-devel.log.20160312
by burek 13 Mar '16

13 Mar '16

[00:38:10 CET] <Compn> llogan : news entry on opw/gsoc? [00:38:29 CET] <Compn> can you write something up mayyyybe ? :) [00:47:57 CET] <llogan> Compn: unfortunately, i have a work backlog to deal with. [01:58:53 CET] <cone-259> ffmpeg 03Umair Khan 07master:5d64ba9d1829: avcodec/alsdec: fix max bits in ltp prefix code [02:26:53 CET] <cone-259> ffmpeg 03Michael Niedermayer 07master:b4f59beeb4c2: avutil/file: Move av_tempfile() to avutil/file_open ff_tempfile() [02:26:54 CET] <cone-259> ffmpeg 03Michael Niedermayer 07master:9d4ab1380add: Use avpriv_tempfile() [06:44:06 CET] <Timothy_Gu> What's the point of av_uninit? What's too hard about just initializing the variable? [06:44:41 CET] <Timothy_Gu> apparently that macro doesn't work on Clang, and a bunch of bogus warnings comes up [07:24:36 CET] <dinux5> Can anybody tell me the directory in ffmpeg where the struct "FilterParams" is ? [07:47:35 CET] <durandal_1707> dinux5: isnt that in swscale? [07:52:48 CET] <dinux5> Umm. I do not see the exact name there. :/ [07:54:24 CET] <dinux5> It has been used everywhere in mlpenc.c. I need to make modifications. But, do not find the struct itself. :( [08:26:37 CET] <Timothy_Gu> dinux5: have you tried git grep? [08:26:59 CET] <Timothy_Gu> the first result is the one you are looking for [08:27:35 CET] <Timothy_Gu> ~/ffmpeg$ git grep FilterParams Ò libavcodec/mlp.h:typedef struct FilterParams { [08:32:12 CET] <dinux5> Oh. Thank you . :D Unaware of that option. [13:57:59 CET] <kierank> Daemon404: what does [13:58:02 CET] <kierank> 12:57 p.m. <qiubit> "Setting default whitelist 'file', bytes read, could not open source file" [13:58:06 CET] <kierank> This mean? [14:44:56 CET] <mateo`> those avienv/riffenc threads ... [14:47:33 CET] <wm4> well at least he didn't get push access yet [15:54:00 CET] <kierank> 1:47 PM <"wm4> well at least he didn't get push access yet [15:54:02 CET] <kierank> ahahahahha [15:56:50 CET] <cone-663> ffmpeg 03Sasi Inguva 07master:e939dde48d44: avformat/utils: factor update_dts_from_pts() out [15:56:50 CET] <cone-663> ffmpeg 03Sasi Inguva 07master:895dd0967194: lavf/utils: Fix DTS for short H264 streams. [15:56:50 CET] <cone-663> ffmpeg 03Sasi Inguva 07master:6fa5c8235b0c: fate: add test for "Fix DTS for short H264 streams" [15:56:50 CET] <cone-663> ffmpeg 03Michael Niedermayer 07master:4aea2c7d8959: fate: Add ffprobe dependancy to fate-h264-dts_5frames [17:34:50 CET] <ubitux> yay, so i finally have a dump of the text subtitles [17:35:02 CET] <ubitux> 76G... [17:35:11 CET] <ubitux> 3777741 subtitles [17:35:18 CET] <JEEB> not a shabby amount [17:35:20 CET] <Daemon404> RIP ubitux [17:41:21 CET] <wm4> ffprobe * ? [17:44:01 CET] <ubitux> yeah, probably the first thing i'll do, trying to see how much we recognize for a start, and maybe make stats about the distribution of the formats [17:44:48 CET] <rcombs> "the" text subtitles? [17:44:52 CET] <rcombs> like, which ones? [17:45:00 CET] <rcombs> 76GB of _all_ the subtitles? [17:45:27 CET] <ubitux> i got an opensubtitles.org dump [17:45:50 CET] <ubitux> i'm still uncompressing them though [17:45:59 CET] <ubitux> 76G is the compressed size [17:46:04 CET] <Daemon404> lol [17:46:31 CET] <durandal_1707> which compression? [17:46:55 CET] <ubitux> gzip, individual files (not sure why but whatever) [17:51:04 CET] <Daemon404> wut [17:51:10 CET] <Daemon404> tthats the worst possible way to do it [17:51:21 CET] <BBB> I bet theyre not experts in compression [17:51:26 CET] <BBB> theyre just like look it got smaller!" [17:51:31 CET] <BBB> PROFIT!!! SHIP IT!!! [17:52:29 CET] <ubitux> yeah whatever, i'm happy they did it for me :) [17:52:48 CET] <Daemon404> ;p [17:54:06 CET] <philipl> ubitux: isn't all of opensubtitles just basic srt with no formatting? [17:54:52 CET] <ubitux> no, they have many formats, and formatting in it [17:55:13 CET] <ubitux> limited to text subs though, so i won't have vobsub [17:55:20 CET] <philipl> fair enough [17:55:21 CET] <ubitux> not sure if i should be relieved or sad [17:55:32 CET] <philipl> everyone loves vobsub... [17:55:54 CET] <ubitux> well, everyone can go f* himself [18:02:10 CET] <Compn> wow [18:02:17 CET] <Compn> i'd love an opensubtitles.org dump... [18:02:59 CET] <Compn> i remember someone had to get utf16 and other encodings working in mplayer [18:04:11 CET] <Compn> ubitux : any ideas for testing ffmpeg subtitle output vs vsfilter or other renderer? e.g. testing to make sure we render correctly as well ? [18:04:57 CET] <Compn> wonder if we could automate vlc subtitle renderer and then compare the framemd5 against ffmpeg... [18:06:18 CET] <rcombs> does VLC have its own renderer? [18:06:22 CET] <rcombs> I thought it just used libass [18:06:41 CET] <ubitux> Compn: iirc opensubtitles refuses utf16 and the like [18:07:03 CET] <ubitux> Compn: testing rendering is going to be expensive [18:07:30 CET] <Compn> i'd guess that opensubtitles gets users reports like "subtitle doesnt work" and then those nonworking subs get fixed or another version uploaded too. thus shrinking the weird sub samples [18:08:12 CET] <wm4> rcombs: vlc has its own renderer for subtitles other than ass [18:08:17 CET] <rcombs> kek [18:08:32 CET] <Compn> libass wasnt around forever, vlc had something before it existed anyways [18:08:35 CET] <wm4> including its own support for pseudo-html as found in subs [18:08:50 CET] <wm4> mplayer also has its own text renderer [18:09:00 CET] <wm4> it took so long to remove that crap from mpv [18:09:48 CET] <Compn> i dont think testing rendering would be that expensive (well, testing all 70gb will be) , similar to a fate test anyway [18:10:21 CET] <Compn> wm4 : mpv using libass only now ? [18:11:08 CET] <Compn> mplayers' renderer was hacked into a bunch of the video outputs to increase speed. whereas libass was seen as "slow" when it was introduced [18:11:17 CET] <Compn> which i can imagine would be difficult to seperate [18:12:03 CET] <wm4> mplayer also had this awful looking method of rendering to the luma plane only [18:13:06 CET] <Compn> there were lots of tricks for opengl rendering, including in color [18:13:54 CET] <wm4> not really [21:37:28 CET] <Zeranoe> Besides build errors, what would be considered a "critical" bug? [21:41:29 CET] <ubitux> unusable binaries, backdoored code, very important regression (like making unplayable most h264 videos) [21:41:36 CET] <ubitux> dunno what else [21:44:28 CET] <wm4> uh, security issues? [22:03:52 CET] <jamrial> also data loss [22:08:33 CET] <Compn> fails to play common format = critical bug [22:08:41 CET] <Compn> i mean regression in playing common format [22:08:57 CET] <Compn> ubi already said it doh [22:24:06 CET] <cone-869> ffmpeg 03Adam Kent 07master:77e355ffb1d6: avformat/hlsenc: Add support for HLS PLAYLIST types EVENT and VOD [23:09:06 CET] <cone-869> ffmpeg 03James Almer 07master:e71096bb3ab4: fate: add missing h264-dts_5frames ref file [23:17:15 CET] <cone-869> ffmpeg 03James Almer 07master:6224dddaa713: fate: add missing dependencies for fate-h264-dts_5frames [23:52:33 CET] <cone-869> ffmpeg 03Paul B Mahol 07master:a61cd42c8d75: avfilter/smptebars: fix some 75% color values [00:00:00 CET] --- Sun Mar 13 2016

1 0

ffmpeg.log.20160312
by burek 13 Mar '16

13 Mar '16

[00:00:06 CET] <limbo_> c_14: when you say "throws away", I don't know what you mean. [00:00:21 CET] <limbo_> I know data is lost, but not how. [00:00:26 CET] <kepstin> limbo_: multiple things - lowpass filter, perceptual masking, etc. [00:00:39 CET] <kepstin> limbo_: you really have to get deep into theory (more than I know) to understand it all [00:01:18 CET] <c_14> psychoacoustic optimizations as well [00:01:35 CET] <furq> is the max -q for jpeg2000 documented anywhere [00:01:46 CET] <furq> oh. [00:02:02 CET] <furq> Stream #0:0: Video: mjpeg, yuvj444p(pc), 1280x720, q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc [00:02:05 CET] <furq> that was fast [00:02:53 CET] <kepstin> furq: iirc, that shows the acceptible values for '-qp', which aren't the same as the range for '-q' in all codecs [00:03:26 CET] <TD-Linux> limbo_, quantization is the primary way to lose information [00:03:47 CET] Action: kepstin still doesn't know why the crf option for x264 wasn't mapped to -q in ffmpeg, but too late to change that now. [00:03:52 CET] <growery> llogan thanks a lot [00:04:31 CET] <llogan> growery: unfortunately it does not yet have much info on videotoolbox [00:04:49 CET] <limbo_> so, given that. how do I get ffmpeg to encode these files as best I can? I can't listen to all of them 100% after encoding to check for slight lfutrther loss. [00:04:58 CET] <furq> limbo_: what was your actual problem again [00:05:15 CET] <furq> i remember it had something to do with a badly stitched together mp3 file which is unplayable somehow [00:05:52 CET] <kepstin> limbo_: if you need to re-encode the files, and you don't want noticable quality loss, then just set the bitrate higher than you would otherwise need. and avoid re-encoding it multiple times. [00:06:04 CET] <furq> nvm it's still in my buffer [00:06:27 CET] <limbo_> I have 7.1 GB of fucked up (probably) 30kbps mp3 files that I should probably re-encode, since more than on of them may be incorrectly made. [00:06:55 CET] <kepstin> ew, 30kbps. I assume they're all something like mono 16khz voice? [00:06:57 CET] <limbo_> the only file I tried listening to had seeking issues on an ipod. [00:07:07 CET] <furq> have you tried remuxing it? [00:07:19 CET] <furq> failing that, if you need it to work on an ipod then i guess your best bet is reencoding to fdk-aac [00:07:29 CET] <limbo_> kepstin: I don't know about the frequency, but it's all radio recordings. [00:07:49 CET] <furq> i would suggest opus or speex for speech but it's not going to work on an ipod [00:07:59 CET] <limbo_> furq: remuxing how? I tried ffmpeg -codec copy and that was more obviously wrong. [00:08:06 CET] <furq> that's how i would have suggested [00:08:11 CET] <limbo_> since it reported a shorter time. [00:08:11 CET] <kepstin> if they're actually music, than it's almost certain that the bitrate is being shown incorrectly by your tool due to probably missing vbr headers. [00:08:25 CET] <kepstin> particularly if they're stereo and actually sound ok [00:08:41 CET] <furq> can ffmpeg actually decode the whole thing [00:09:21 CET] <limbo_> kepstin: I'm not getting the bitrate from a program, it wasin the description where I got this. [00:09:30 CET] <kepstin> would be a good thing to check first - just decode some samples to a wav and see if they sound ok [00:10:15 CET] <furq> if it is mono and low sample rate then 32kbps isn't so bad [00:11:10 CET] <kepstin> yeah, mono 16kHz would probably be listenable at that. Maybe ok for AM radio recordings? [00:11:15 CET] <limbo_> furq: I'm gett errors like "[mp3 @ 0xd031e0] overread, skip -5 enddists: -1 -1" occasionally [00:11:25 CET] <furq> that sounds fun [00:12:11 CET] <limbo_> How would I quickly check if an encode worked right? since my PC could play them back just fine. [00:12:25 CET] <furq> fwiw you should probably decode the whole thing to wav anyway, unless your ffmpeg was compiled with fdk-aac [00:12:47 CET] <furq> doing that and using the standalone fdk encoder will be much less hassle than recompiling ffmpeg [00:14:09 CET] <limbo_> I'm not touching aac. [00:14:52 CET] <limbo_> I have 7.1 GB of these things, and I'm not looking to increase that much. [00:15:53 CET] <furq> aac is more efficient than mp3, especially at low bitrate [00:17:24 CET] <furq> i wouldn't normally recommend it but it's probably your best bet here [00:22:21 CET] <growery> llogan i am not getting it do i have to compile or do i have to ask for the videotoolbox hardware acceleration on the encoding command? [00:35:35 CET] <limbo_> is there any other way to get he smallest mp3 possible without losing a notible amount of quality? not supplying any options leads to a 47MB file increasing to 57MB. [00:35:57 CET] <TD-Linux> decrease the bitrate or quality setting until it sounds bad [00:37:21 CET] <pzich> yeah, the problem is "notable" is totally relative [00:40:26 CET] <limbo_> TD-Linux: the test file I used is 04:04:58 long. [00:40:44 CET] <TD-Linux> listen to a segment [00:40:51 CET] <limbo_> file says it's 56kbps though. [00:56:40 CET] <growery> llogan i see! i am compiling it [01:23:48 CET] <llogan> growery: once you get it compiled you can encode with "-c:v h264_videotoolbox". i assume you're wanting to encode H.264 and not decode [01:24:07 CET] <llogan> you can look at the commits that contain the word for some info: http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=videotool… [01:34:17 CET] <somethingelse> Is there any way to decode frames in file order [01:34:21 CET] <somethingelse> As opposed to display order? [01:34:41 CET] <somethingelse> (Without doing a second manual buffering and reordering of the frames) [01:35:35 CET] <J_Darnley> I know of no option but look at the help, possibly for instances of dts [01:35:57 CET] <J_Darnley> (not to be confused with the DTS audio format) [01:37:16 CET] <Lucifer333> hi [01:37:32 CET] <Lucifer333> i am looking for the (unofficial) discription of dvd-video file layout [01:37:53 CET] <Lucifer333> does anyone have a link? [03:46:04 CET] <jookiyaya> i heard that aac encoder that comes ffmpeg sucks, is this true? [03:50:33 CET] <TD-Linux> it's better than it used to be [03:50:56 CET] <jookiyaya> which is [03:50:56 CET] <TD-Linux> the correct solution is to not use aac [03:51:02 CET] <jookiyaya> then use what [03:51:04 CET] <kepstin> and it's definitely better than vo-aacenc [03:51:04 CET] <TD-Linux> but it is still behind qaac [03:51:21 CET] <jookiyaya> what's qaac ? [03:51:28 CET] <kepstin> quicktime aac encoder [03:51:32 CET] <jookiyaya> oh apple one [03:51:33 CET] <kepstin> (apple itunes encoder) [03:51:51 CET] <TD-Linux> jookiyaya, opus is currently the best [03:51:58 CET] <jookiyaya> why can't ffmpeg include qaac [03:52:04 CET] <TD-Linux> it's proprietary software [03:52:40 CET] <jookiyaya> opus is better than qaac ? [03:52:43 CET] <TD-Linux> yes [03:52:49 CET] <jookiyaya> says who? [03:52:55 CET] <TD-Linux> listening tests [03:53:00 CET] <jookiyaya> fair enough [03:53:11 CET] <jookiyaya> does opus do 5.1 ? [03:53:19 CET] <TD-Linux> yes [03:53:25 CET] <furq> jookiyaya: http://listening-test.coresv.net/s/scores_by_tracks_en.png [03:53:30 CET] <jookiyaya> and can you use it with mp4 [03:53:41 CET] <furq> no [03:53:54 CET] <TD-Linux> there are now patches for opus in mp4 but I would consider it experimental at best [03:53:59 CET] <TD-Linux> mkv is your best bet right now [03:54:03 CET] <furq> wtf is opus 0.1.9 [03:54:11 CET] <jookiyaya> mp3 is better than faac ? [03:54:11 CET] <TD-Linux> furq, opus-tools [03:54:17 CET] <furq> ah [03:54:37 CET] <TD-Linux> jookiyaya, note that mp3 is run at 128kbps in that test [03:54:39 CET] <furq> i guess that's opus 1.0.3 or one of the 1.1 release candidates then [03:54:48 CET] <furq> so not too old [03:55:07 CET] <jookiyaya> TD-Linux true, that's not a fair comparison then [03:55:43 CET] <TD-Linux> jookiyaya, well it was done because otherwise mp3 would be so far away from the others that it would be meaningless [03:56:14 CET] <TD-Linux> note that faac is not ffmpeg's aac encoder [03:56:32 CET] <furq> that test was done before the ffaac overhaul anyway [03:57:06 CET] <jookiyaya> then where is ffmpeg's aac encoder test? [03:57:12 CET] <furq> there isn't one [03:57:18 CET] <furq> perhaps you'd like to organise one [03:57:52 CET] <TD-Linux> there are some on hydrogenaud.io already [03:58:02 CET] <furq> the only one i found was one guy comparing fdk and ffaac [04:45:57 CET] <jookiyaya> furq i don't see fdk one in that graph [04:50:38 CET] <jookiyaya> <jookiyaya> [18:53] <furq> jookiyaya: http://listening-test.coresv.net/s/scores_by_tracks_en.png [04:50:38 CET] <jookiyaya> [19:49] <+j45> http://lmgtfy.com/?q=the+cake+is+a+lie [04:50:38 CET] <jookiyaya> [19:50] <+j45> the listening tests are rigged [04:50:38 CET] <jookiyaya> [19:50] <jookiyaya> really? [04:55:18 CET] <TD-Linux> that was a portal joke [04:56:03 CET] <jookiyaya> <jookiyaya> i heard opus is better than fdk-aac? why doesn't handbrake support that [04:56:03 CET] <jookiyaya> [19:46] <+j45> because the cake is a lie [04:56:03 CET] <jookiyaya> [19:46] <jookiyaya> huh [04:56:03 CET] <jookiyaya> [19:47] <jookiyaya> cake? [04:56:24 CET] <TD-Linux> you're being trolled [04:56:58 CET] <jookiyaya> i am not [04:59:43 CET] <jookiyaya> these are serious people [05:00:54 CET] <jookiyaya> what aac encoder does google use now? [05:01:18 CET] <jookiyaya> vo-aacenc? [05:16:34 CET] <jookiyaya> [20:07] <bogo_lode> afaict, opus isn't really as good as aac or mp3 in most situations where you'd want to use those. [05:16:34 CET] <jookiyaya> [20:08] <bogo_lode> the big uptake for opus is voip, because it's designed to work well for speech and has low latency options [05:16:34 CET] <jookiyaya> [20:08] <+j45> yes, that's what opus was designed for [05:16:34 CET] <jookiyaya> [20:09] <jookiyaya> why is google starting use opus then [05:16:34 CET] <jookiyaya> [20:09] <bogo_lode> because google uses tons of stupid shit? [05:35:19 CET] <TD-Linux> jookiyaya, yeah that's not true, it is true that it was originally designed for low latency [05:35:26 CET] <TD-Linux> but it ended up outperforming anything else [05:36:35 CET] <TD-Linux> all the data for the picture posted earlier is at http://listening-test.coresv.net/results.htm [05:36:58 CET] <jookiyaya> td-linux can you join #handbrake and tell the people that [05:52:44 CET] <mr_lou> Hello [05:53:13 CET] <durandal_1707> Hello [05:54:01 CET] <mr_lou> I'm creating a home-video-album for my family, on a blu-ray disc. Using ffmpeg to encode video-recordings into a compatible h264 video stream and ac3 audio stream, then muxing them together with tsMuxer. [05:54:28 CET] <mr_lou> At first I used Linear PCM for audio, but since that takes up way too much space compared ot ac3, I switched. [05:55:16 CET] <mr_lou> But after switching to ac3, my blu-ray players won't play the audio. So I've been searching and searching, trying to figure out whether it's an encoding problem by ffmpeg, or something else. [05:57:47 CET] <mr_lou> This is the command line: ffmpeg -i Video.mp4 -acodec ac3 -ab 128k -ar 44100 -vn Audio.ac3 [05:58:35 CET] <mr_lou> Spot anything wrong with that? [06:00:24 CET] <durandal_1707> I dunno, but aren't there special programs for that [06:03:17 CET] <mr_lou> Well there are certainly other programs yes, like Mencoder and Aften to encode ac3. But since I'm already using ffmpeg for the other steps in the process..... [06:50:55 CET] <durandal_1707> mr_lou: perhaps they already have working command combinations [07:02:33 CET] <mr_lou> durandal_1707, They might. I'll try Aften. [08:19:14 CET] <mr_lou> durandal_1707, Well, that didn't exactly improve things. Aften's ac3 audio results in noise.... not sure which I prefer then. ;-) [08:22:45 CET] <durandal_1707> mr_lou: handbrake? [08:28:20 CET] <mr_lou> Trying mencoder now. [08:36:05 CET] <lukeshu> I'm probably just being stupid, but: with ffmpeg 3.0, I'm using ffmpeg to convert rgb24 streams into multipart jpeg/mjpg, but the color is... off. If I encode to something else, it seems fine. [08:37:13 CET] <lukeshu> The command I'm using is: ffmpeg -pixel_format rgb24 -s 640x480 -f rawvideo -i video.rgb24 -q:v 1 -f mpjpg - > video.mjpg [08:37:53 CET] <lukeshu> If I change it to: ffmpeg -pixel_format rgb24 -s 640x480 -f rawvideo -i video.rgb24 -f video.avi [08:37:56 CET] <lukeshu> then it works fine [08:38:40 CET] <lukeshu> Also, with mpjpeg, it prints a warning from swscalar: "deprecated pixel format used, make sure you set range correctly" [08:39:08 CET] <lukeshu> I'm purplexed, any help would be greatly appreciated. [08:41:56 CET] <lukeshu> Nevermind, I've figured it out. [08:42:21 CET] <lukeshu> Apparently ffmpeg's mpjpeg doesn't play nice with stdout instead of a real file. [08:46:39 CET] <mr_lou> durandal_1707, Mencoder seems to work. [08:51:46 CET] <lukeshu> I take it back, it seems to actually be a client issue. [08:54:21 CET] <lukeshu> Any idea why javax.imageio.ImageIO would screw up the colors for an mpjpeg stream comming from ffmpeg, but not other sources. [08:54:22 CET] <lukeshu> ? [11:46:57 CET] <edakiri> Where does the information for preset Ultrafast come from? I find no file with that name. [11:53:25 CET] <jkqxz> It depends on the encoder. (For libx264, see <http://git.videolan.org/?p=x264.git;a=blob;f=common/common.c#l190>.) [13:34:39 CET] <Wader8> hi [13:36:08 CET] <Wader8> Is there any intermediate guide for converting, or just increasing bitrare, I have FRAPS video, want to put it on youtube, but I need to make a good intermediate, in order to avoid further degradation when youtube reconverts it, thus I cannot use the settings youtube uses [13:36:36 CET] <Mavrik> Then upload it as it is? [13:36:38 CET] <Wader8> I always do it like this with video software, but I have to use ffmpeg for Fraps [13:36:49 CET] <Mavrik> No conversion means no quality loss :) [13:37:15 CET] <Wader8> 3 GB will take a long time to upload for 3 minutes of footage [13:37:20 CET] <Mavrik> Mhm. [13:37:32 CET] <Mavrik> Then you want to decrease bitrate I think? (You wrote increase) [13:37:56 CET] <Mavrik> I'd say do losless x264: https://trac.ffmpeg.org/wiki/Encode/H.264 [13:37:59 CET] <Wader8> I don't have that kind of time, I just put it on youtube, embedd into a thread, them im off to other things besides PCs [13:38:21 CET] <Wader8> Fraps is uncompressed - just using MPEG2 would lower size by 100x [13:38:45 CET] <Wader8> well that was a off-the-head number [13:38:56 CET] <Wader8> basically, any codec will decrease size significantly [13:39:34 CET] <Wader8> what about the B frames thing ? [13:40:07 CET] <Wader8> I forgot a bit, which ones are which, I-frames are the ones that are full right ? [13:42:25 CET] <Wader8> Isn't the -preset intended for lowering side ? or does it also do both image quality enhancement ? [13:42:30 CET] <Wader8> size * [13:43:45 CET] <Mavrik> Losless :) [13:43:55 CET] <Mavrik> Do you understand that means that image quality does not change? :) [13:44:06 CET] <Mavrik> preset just controls size vs. encoding time in this cas [13:44:07 CET] <Mavrik> *case [13:44:15 CET] <Mavrik> Also don't forget to encode audio as well [13:50:58 CET] <Wader8> Well the size thing isn't that needed now, it's not going to make that much of a difference, 500MB or 300MB, the time it takes , the intermediate is going to be deleted after upload, so only use slowest for archiving when size is also important to save space [13:51:20 CET] <Wader8> So if I use medium, I would still get as good of a quality as in slow ? [13:51:57 CET] <Wader8> Because I know they do a lot of stuff behind scenes when presets are changed, but I do not know that deep to know if all those things only affect size [13:56:14 CET] <Wader8> Well actually it's going better than I expected, [13:56:48 CET] <Wader8> for 1 minute of footage some 60FPS 1080p shots are 5GB large, but on slow it's going 18 FPS so it shouldn't take long at all [13:57:00 CET] <Wader8> Other videos are in 30 and 15 FPS [14:21:55 CET] <Mavrik> Wader8, why do you keep asking about quality [14:22:01 CET] <Mavrik> When we're talking about LOSLESS. [14:22:09 CET] <Mavrik> No losses = same output as it's the input :P [14:22:23 CET] <Mavrik> Quality doesn't change :) [14:22:58 CET] <Wader8> Not sure, I used CRF=16 as I did with some other project when I was archiving old DVD into X265 [14:23:24 CET] <Wader8> I do not believe X264 and X265 have true lossless [14:23:41 CET] <Wader8> nor I do not need it, never said I wanted lossless, I wanted a good intermediate [14:24:16 CET] <Wader8> I heard some I-Frame only MPEG2 seems to be looking good, but this was some time ago [14:28:02 CET] <Wader8> Here, here's the output it made, from 5GB Fraps, to 137 MB x264 at 14K Mbps variabled http://pastebin.com/H28h92mk [14:28:06 CET] <Mavrik> Wader8, they do have true losless. [14:28:16 CET] <Mavrik> Which is described in the wiki you posted. [14:30:03 CET] <Wader8> That's just increasing bitrate I believe, not sure, I didn't heard from community, X264 wasn't their pick, I always heard I-Frame only MPEG2 to be the choice for actual practical use, ignoring intermediate-only codecs [14:31:08 CET] <Wader8> I think it's something more to it than just increasing bitrate, but heck I'm not even that of an expert on ffmpeg so anyway, I already almost finished with the 4 videos, goign to use this on a monthly basis to come, I just needed some refreshment [14:31:39 CET] <Wader8> 14000 Mbps probably enough, it's not real 60FPS, the FPS ingame keeps fluctuating [14:32:36 CET] <Wader8> Well I used that for like 30FPS, if I was doing some serious HD footage obviously I would double that if it was 60FPS, but this is a Simulator game which has simple graphics so no need [14:33:10 CET] <Wader8> But if I used the super-compression-agressive youtube settings I would get the text all blurred, so that's why I need a good one so the text is clear [14:33:43 CET] <furq> what youtube settings [14:34:30 CET] <Mavrik> Jesus I remember you now. [14:34:41 CET] <Mavrik> You're that moron that already once came here and kept ignoring everything everyone said. [14:34:44 CET] <Mavrik> Like now. [14:34:52 CET] Action: Mavrik moves out of the debate. [14:36:07 CET] <Wader8> furq, the stuff Youtube has officialy on their site, and all the other mainstream people which have videos about "perfect youtube settings for upload", but youtube re-converts that, so you always have to upload a much better version, not give it the crappy version so it makes an even crappier one once it's uploaded, general stupidity out there [14:37:09 CET] <furq> people on youtube, stupid? never [14:37:12 CET] <Wader8> Mavrik, yeah, I don't really follow the system that much, and, I've indicated like 2 times I already figured out what I needed, I posted the pastebin of the results, I'm already uploading, not sure what's your problem [14:37:39 CET] <furq> i assume it's the fact that you inexplicably don't believe that x264 has a lossless mode [14:40:01 CET] <Wader8> furq, I never wanted lossless, I said this 2 times now, if I'm wrong on X264 then I apologize [14:59:48 CET] <jokoon> I want to add text on a small video [14:59:57 CET] <jokoon> is there so free software to do that [15:00:02 CET] <J_Darnley> ffmpeg? [15:00:04 CET] <jokoon> I mean a lightweight thing [15:00:08 CET] <J_Darnley> How small? Where? [15:00:09 CET] <jokoon> I mean with an UI [15:00:21 CET] <J_Darnley> Then no, nothing "light weight" [15:00:45 CET] <J_Darnley> All will probably use ffmpeg behind the scenes anyway [15:00:47 CET] <jokoon> can avidemux do it ? [15:00:56 CET] <J_Darnley> I don't know [15:01:00 CET] <J_Darnley> Read their manual [15:23:50 CET] <great> hi! [15:23:56 CET] <Mavrik> ohai [15:24:06 CET] <great> i have some questions [15:24:11 CET] <great> could you help me [15:24:25 CET] <durandal_1707> nope [15:25:18 CET] <great> we use ffmpeg to convert our videos but this process is so slow [15:25:38 CET] <great> we use ssd servers 12 cores etc. [15:26:21 CET] <great> there is a app called total video converter [15:26:41 CET] <great> and it converts same video with same settings faster than us [15:27:08 CET] <BtbN> probably not the same settings. [15:27:24 CET] <great> yep commands are diffrent [15:27:34 CET] <great> but output is same [15:27:52 CET] <ritsuka> define same [15:28:20 CET] <great> same bitrate and encoder [15:28:28 CET] <J_Darnley> I wonder whether you disabled assembly or threads when compiling ffmpeg (or other libraries) [15:29:16 CET] <BtbN> same bitrate doesn't mean anything [15:29:25 CET] <Mavrik> Yp, same bitrate doesn't mean much. [15:29:33 CET] <great> i complied it with this https://github.com/rdp/ffmpeg-windows-build-helpers [15:29:34 CET] <Mavrik> great, easiest way to tell would be to compare ffmpeg parameters :/ [15:29:48 CET] <Mavrik> great, but if you can at least provide your command line and it's output we can help with optimization [15:29:51 CET] <BtbN> if it uses ultrafast with cbr you get the same bitrate with a notably reduced encoding load, but also much worse quality. [15:30:01 CET] <great> i can send commands wait [15:30:52 CET] <J_Darnley> It takes that script an hour to build libx264, fdkaac and ffmpeg? [15:30:55 CET] <J_Darnley> ONE HOUR? [15:31:10 CET] <J_Darnley> Goog god [15:31:58 CET] <great> http://pastebin.com/UhsWiNpY [15:32:08 CET] <great> it takes more :( [15:32:11 CET] <BtbN> so probably the author never heard of parallel make. [15:32:34 CET] <Mavrik> great, hmm, that's some old ffmpeg commandline there in total converter :) [15:33:04 CET] <Mavrik> great, the biggest hurt I see for you is that you don't pass a "-preset" parameter which controls CPU usage and encoding time [15:33:16 CET] <Mavrik> that's a "general quality vs. CPU used" setting that implies bunch of other [15:33:31 CET] <Mavrik> default is -preset "medium", but it ranges from "ultrafast" to "veryslow" [15:33:48 CET] <great> i guess aisee using ultrafast [15:34:15 CET] <great> is there a quality diffrence [15:34:55 CET] <furq> great: run mediainfo on both files and paste the encoding settings entry [15:35:15 CET] <Mavrik> great, yes, there's a quality difference [15:35:21 CET] <Mavrik> they pass "-vpre ..." and path to some file [15:35:25 CET] <Mavrik> which can be whatever :) [15:35:30 CET] <BtbN> ultrafast degrades h264 to mpeg2 level [15:35:48 CET] <furq> at the same bitrate then faster presets will obviously look worse [15:35:55 CET] <furq> otherwise there wouldn't be much point having slower presets [15:36:05 CET] <Mavrik> Yes. [15:36:12 CET] <Mavrik> Use "fast" or "faster" perhaps [15:36:15 CET] <Mavrik> Wouldn't go beyond that [15:36:45 CET] <Mavrik> But using mediainfo helps too yeah [15:36:48 CET] <great> in -vpre file [15:36:51 CET] <great> preset=faster vprofile=baseline [15:36:57 CET] <furq> well there's your answer [15:37:04 CET] <Mavrik> mhm [15:37:12 CET] <Mavrik> Baseline is fast to encode, preset is set to faster :) [15:37:23 CET] <Mavrik> I strongly advise against encoding to baseline though, looks like arse. [15:37:32 CET] <Mavrik> Just set preset to faster and you should be set [15:38:15 CET] <great> yeah thanks [15:38:32 CET] <great> i will send mediainfo after converts finished [15:38:39 CET] <furq> there's not really any need now [15:39:00 CET] <Mavrik> mhm, we got the reason :) [15:39:48 CET] <great> ok thanks :) [15:40:21 CET] <furq> i should really clean up my ffmpeg mingw build script [15:40:28 CET] <furq> since it takes much less than an hour [16:13:00 CET] <Wader8> why does -i "concat: ||" not work with mp4 anymore [16:13:09 CET] <Wader8> nor with mpegts [16:13:15 CET] <Wader8> worked for vob files [16:13:19 CET] <Wader8> not for AVC/MP4 [16:13:25 CET] <Wader8> i have the same command [16:14:03 CET] <Wader8> maybe i need to add map 0:0, that's the only difference I have in the command, but the source files don't contain any other streams than 1 video and 1 audio [16:15:54 CET] <furq> the concat protocol has never worked for mp4 [16:15:57 CET] <furq> it should work for ts though [16:16:35 CET] <Wader8> the files aren't ts [16:16:44 CET] <Wader8> im not even sure if AVC can be in TS [16:16:59 CET] <furq> sure it can [16:17:16 CET] <Wader8> this just always gets more frustrating by the minute [16:17:19 CET] <furq> https://trac.ffmpeg.org/wiki/Concatenate#demuxer [16:17:29 CET] <furq> that's what you want [16:17:42 CET] <Wader8> i should have finished this 1 and half hours ago to go do other stuff but no im stuck here [16:21:52 CET] <Wader8> could not write header for output file 0# incorrect codec parameters, invalid argument [16:21:59 CET] <Wader8> when i use MYLIST option [16:24:27 CET] <Wader8> anyone please ? [16:25:11 CET] <Wader8> I don't understand how this doesn't work, what kind of nonsense is this that such a basic thing doesn't work [16:28:09 CET] <Wader8_> anyway screw this im going to recovert it later buch of nonsense really, concat doesn't work on MP4, i mean, that's like saying you can't drive up a hill with a car, i better get the heck away from the PC before I blow up [16:29:37 CET] <furq> what a nice young man [16:33:04 CET] <kepstin> "concat doesn't work on MP4" is more like saying "you can't drive through a tree with a car" [16:33:49 CET] <kepstin> (concat protocol, at least) [16:34:08 CET] <furq> what do you mean you can't weld four cars together and go four times as fast [16:34:18 CET] <Mavrik> Well, he apparently read some forums from 1999 [16:34:26 CET] <Mavrik> And doesn't know the heck he's doing. [16:43:13 CET] <JEEB> well to be 100% honest concatenating ISOBMFF is less than simple, which is why L-SMASH doesn't yet do it, and why GPAC's mp4box often does ISOBMFF concat completely wrong [18:02:12 CET] <kubast2> Hey ,how to create .mkv file with empty video track[as long as audio track I'll be adding to the empty file] ,with audio track from .flac file ? [18:04:45 CET] <J_Darnley> "empty" means what exactly? [18:04:54 CET] <J_Darnley> You can create a black video [18:05:05 CET] <J_Darnley> but that is still a video [18:41:01 CET] <dsl42> i guess he meant ffmpeg -i input.flac -c copy out.mkv .. and of course i love dudes who asked questions and leave a minute later ;) [19:02:43 CET] <chungy> heh yeah... and the appropriate extension there would really be .mka :P [19:39:38 CET] <wellen> hello, i have just used ffmpeg on windows cli to get a 720p video to output as just audio m4a file, it took a while eg 1hr as it was 2hrs7mins long... is it not possible to just run a line that quickly outputs only the audio from the video file? [19:40:26 CET] <c_14> ffmpeg -i video -c copy -map 0:a out.m4a [19:41:48 CET] <wellen> hey thanks for quick response - i only had the ffmpeg -h to refer to ... is there a good online manual for ffmpeg i can plow through? [19:42:16 CET] <Mavrik> wellen, wiki tends to describe common operations: https://trac.ffmpeg.org/wiki [19:42:32 CET] <Mavrik> wellen, and online documentation is more useful in my opnion: https://www.ffmpeg.org/documentation.html [19:42:33 CET] <J_Darnley> and http://ffmpeg.org/documentation.html [19:43:09 CET] <wellen> thanks all, sorry for being lazy and just coing here to ask - after watch the cli for over an hour i need to try and make some time back up ! cheers [19:45:43 CET] <wellen> one last question before i go... what exactly did i get ffmpeg to do for an hr with this command... ffmpeg -i e:\files\wipez400\cfiles\videos\prs2016\720p\720pav.mp4 720pav.m4a [19:46:19 CET] <wellen> i mean to ask, what exactly was it doing that too 1hr+ [19:48:06 CET] <J_Darnley> Encoding the video [19:49:08 CET] <wellen> that is what i thought but the end result file 720pav.m4a i would have expected to just be audio ...it is actually the audio and video [19:49:37 CET] <J_Darnley> Of course, m4a is just mp4 with a different extension. [19:50:27 CET] <wellen> ah my stupid assumption was that m4a was just audio file format extension am i very wrong on that ? [19:51:26 CET] <J_Darnley> Sure, apple uses it like that but it is no different to mp4 [19:52:02 CET] <wellen> good to know thanks (and even more reason i should get stuck into the ffmpeg manual no doubt! [19:52:36 CET] <J_Darnley> If you think ffmpeg is doing something unexpected then please come ask us. [19:56:03 CET] <wellen> im normally the person who everyone else comes to for tech answers so i try not to ask others and research first, but this time im breaking my own rules.. sorry (as for ffmpeg doing something unexpected, in my case it was pebcak for sure) [19:59:35 CET] <wellen> found what i need https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio [20:00:00 CET] <wellen> many thanks all for the help bye [20:03:38 CET] <wellen> is there a specific reason that the ffmpeg project does not have an official GUI? [20:04:08 CET] <sfan5> can you imagine the complexity of such a gui? [20:04:21 CET] <sfan5> there are waaay to many options to be able to pack them into a gui [20:05:09 CET] <wellen> i can sfan5..i can also imagine a way through decent ui/ux to actually cover it in a decent way [20:05:39 CET] <sfan5> then go write one [20:05:43 CET] <dsl42> :) [20:05:44 CET] <sfan5> maybe it'll become the ffmpeg official gui [20:06:46 CET] <wellen> has it not been done as it is an intimidating task to everyone? i figured everyone would love a gui but i wondered if it has neveer been done as the original project devs had a reason for it not to be exisitng eg i wondered if they were against ffmpeg project having a gui? [20:08:21 CET] <dsl42> i dont care about a gui [20:08:24 CET] <sfan5> ^ same [20:10:17 CET] <J_Darnley> What toolkit aregoing to to use that will work across all platforms? [20:10:24 CET] <J_Darnley> *are you going [20:11:08 CET] <wellen> i am impressed at the collective effort and talents that have gone (and continue to go into) the ffmpeg project and i have nothing against cli tools either, but i bet ffmpeg would be even more widely adopted by users if the project had a gui as well..i wonder if the collective would be interested in persuing a gui effort at some point [20:11:31 CET] <J_Darnley> Who doesn't use it? [20:11:56 CET] <J_Darnley> It is the backing to every other a/v program out there [20:12:41 CET] <wellen> i have noticed various services using ffmpeg and even making their own basic, limited windows applications to carry out single task functions via ffmpeg [20:16:58 CET] <wellen> so is it accurate to say that an official ffmpeg gui does not exist due to there not being a real need for its own gui? [20:18:48 CET] <J_Darnley> That is certainly a large part [20:20:16 CET] <wellen> also, does it not slightly annoy the ffmpeg developers that others just grab ffmpeg and use it tucked away under their programs as though it is their own work? [20:20:54 CET] <wellen> or are they just happy to provide ffmpeg to anyone and everyones use [20:20:56 CET] <J_Darnley> As long as they follow the license we don't really mind [20:20:57 CET] <furq> only if they're violating the license [20:21:47 CET] <wellen> is that in the legal section? [20:26:13 CET] <wellen> i have read the legal section on the project website... it just seems the usual standard resonable things and of note is the mini section on patents etc and licensing [20:30:38 CET] <wellen> would an official gui be a waste of time? [20:38:59 CET] <wellen> sorry j darnley just noticed your question further up -re toolkit i dont know without giving it serious consideration but im sure there are various options - my first question would be is it worth doing, then the hows [20:43:02 CET] <wellen> gtk+ possibly although this doesnt bode well... "Dirk Hohndel, co-developer of Subsurface and member of Intel's Open-Source Technology Center, criticized the GTK+ developers for being abrasive and ignoring most community requests" [20:44:25 CET] <J_Darnley> If you're seriously considering writing an official gui then definitely email ffmpeg-devel [20:44:34 CET] <J_Darnley> but I don't reckon it would be well recieved [20:44:50 CET] <J_Darnley> It would be a large maintainace burden [20:45:45 CET] <J_Darnley> And I would expect people to complain about portability, libraries, static builds... [20:46:48 CET] <J_Darnley> Since we already link with gdi and other libs on Windows, a win32 specific gui might be an option [20:46:55 CET] <wellen> i understand why it wouldnt be well received, tbh i'd be in that same camp...but i'm not 'every user' and everyone has different needs [20:47:04 CET] <J_Darnley> but I think it would be shot down as "not portable" [20:47:38 CET] <wellen> why would it be not portable? can you elaborate slightly? [20:48:05 CET] <J_Darnley> How can you run the win32 gui on Linux, Mac, Android? [20:49:12 CET] <J_Darnley> I mean using native components and native functions [20:51:12 CET] <wellen> could it not be worked around at idea stage eg wrappers - i would have figured there are various ways/means/methods/workarounds eg if considered at the very early stages [20:51:48 CET] <J_Darnley> Sure: a cross platform toolkit like gtx or qt [20:52:26 CET] <wellen> call me simplistic but also if it was kept lean enough, what is stopping there being 1 download containing version of the gui for each platform? or even cross platform consideration at the code stage of the gui? [20:53:02 CET] <wellen> or is my naiveity showing? [20:55:28 CET] <J_Darnley> Size would be the main reason, especially if you want a statically linked binary [20:56:33 CET] <wellen> what if it was all handcoded in c/c++/c# ? wouldnt that free up most of the concerns as then the only cost would be time and ability of the creator right? [20:59:02 CET] <c_14> I think the biggest problem is that nobody is going to want to maintain it. [21:01:56 CET] <wellen> while i cant blame anyone for feeling that way - it is funny when you think about it... eg the amount of collaborative effort time and talent that has gone into the cli ffmpeg - kinda shows what can result from continued joint effort by many [21:02:43 CET] <J_Darnley> I now think you have no idea how complicated a gui really is. [21:02:45 CET] <ethe> c_14: no one wants to maintain Qt code [21:02:58 CET] <wellen> are you all user of ffmpeg or do any of you also help out on the project as well? [21:03:08 CET] <ethe> or code for *every* os either. [21:03:12 CET] <J_Darnley> Yes, but I'm a shit programmer [21:03:53 CET] <ethe> wellen: I *try* to, but I have the same problem as J_Darnley [21:04:09 CET] <wellen> i call bs i bet you are all just being modest haha [21:04:52 CET] <wellen> as for having no idea how complicated j darnley... most things are... but some like the challenge ;o) [21:08:23 CET] <wellen> also why does it just have to be 'one person'? [21:10:42 CET] <ethe> who else is going to make a gui? It doesn't have to be one person. If you want it to happen, then make it happen [21:11:53 CET] <wellen> for example what if there was a decent effort made to visualise/rough design an official gui whereby everyone was agreed on a few main critieria eg clarity and ease of use [21:12:20 CET] <ethe> There's no point saying "what if"... there isn't. [21:12:32 CET] <wellen> i know often its a case of 'too many cooks' and peoples egos can cause extra grief in group projects but everyone can agree on they just want things to be usable right? [21:15:26 CET] <sfan5> wellen: >clarity and ease of use [21:15:32 CET] <sfan5> that is a wonderful example [21:16:22 CET] <wellen> it is...? [21:16:30 CET] <sfan5> if you want to make it really easy to use you might not want to confuse the user with what a libvpx-vp9 might be and just say VP9 or even just WebM [21:16:37 CET] <sfan5> but that would go again clarity, wouldn't it? [21:16:52 CET] <sfan5> s/again/against/ [21:17:09 CET] <wellen> nothing stopping the gui having 2 tabbed pages eg basic and advanced.... [21:17:23 CET] <furq> i think you're going to need more than two tabs [21:17:24 CET] <sfan5> just deciding on a few main criteria is not going to get a good result [21:17:30 CET] <sfan5> you need to think about every detail [21:17:47 CET] <ethe> wellen: why dont you design a draft for the whole gui? [21:18:01 CET] <furq> `ffmpeg -h full` will give you an idea of the scale of this endeavour [21:18:11 CET] <wellen> most common choices on basic tab (also default) and all the power user settings on the advanced tab..also can have a main options menu to specify by default if gui loads up iin basic or advanced displaying on startup.. right? [21:18:12 CET] Action: dsl42 grabs some popcorn [21:18:23 CET] <sfan5> % ffmpeg -h full 2>/dev/null | wc -l [21:18:23 CET] <sfan5> 8210 [21:18:33 CET] <sfan5> so we need about 821 tabs? [21:18:33 CET] <furq> i got 6400 but this is an old version [21:19:00 CET] <wellen> (butterkist toffee please dsl42) [21:19:10 CET] <furq> i think it'd be charitable to say that 1000 of those are headers and empty lines [21:19:45 CET] <sfan5> adding sort -u gives 3987 [21:20:20 CET] <furq> a lot of those will be mapped to multiple codecs with different value ranges though [21:20:33 CET] <wellen> without being too much of a killjoy - i would be thinking not just the 2 tab approach but also pages eg next and back at bottom right of gui for a wizard type approach [21:20:42 CET] <sfan5> btw [21:20:46 CET] <sfan5> have you looked at handbrake? [21:20:56 CET] <wellen> how come you thought all the options would just be o 2 static pages? [21:20:57 CET] <sfan5> https://handbrake.fr/img/slides/slide2_lin.jpg [21:21:06 CET] <sfan5> it's about as good as it gets [21:21:13 CET] <wellen> yeah i have seen handbrake a while ago [21:21:20 CET] <sfan5> you can't fit every ffmpeg option into a gui [21:21:48 CET] <furq> you can, but it'll be some kind of nightmarish vision of a future gone wrong [21:22:17 CET] <sfan5> like this? https://pbs.twimg.com/media/BylctRxCEAEv-i2.png:orig [21:22:21 CET] <ethe> wellen: handbrake is great, for users. If a user *really* needs 100 obscure flags, then they can just use the cli (the docs really aren't that hard to understand) [21:22:26 CET] <furq> just the codec options needs to cover >400 codecs [21:22:36 CET] <dsl42> :) [21:22:44 CET] <wellen> just seen that handbrake gui and - it could be made simpler (from a user epxerience pov) [21:23:09 CET] <sfan5> huh [21:23:34 CET] <sfan5> if you want to make it that simple you'll need 3 tabs [21:23:35 CET] <dsl42> handbrake only encodes to x264, afaik [21:23:43 CET] <furq> it does x265 and mpeg2 as well [21:23:49 CET] <dsl42> ok [21:23:54 CET] <furq> and maybe xvid [21:24:11 CET] <dsl42> not anymore i guess [21:24:17 CET] <sfan5> vp8 too it seems https://trac.handbrake.fr/wiki/Encoders [21:24:43 CET] <wellen> what would make it simpler is a wizard screens approach (as well as the already mentioned 2 page basic/advance tab) [21:24:45 CET] <dsl42> oh ok, i did not use it for a long time [21:24:47 CET] <furq> mine doesn't do vp8 ;_; [21:25:16 CET] <ethe> does ffmpeg have CoreAudio's encoders yet? (I guess it would if handbrake has them) [21:25:35 CET] <furq> handbrake uses some non-libav* stuff [21:25:43 CET] <sfan5> does ffmpeg have intel quicksync? [21:25:49 CET] <furq> most notably dvd ifo parsing [21:25:59 CET] <furq> which is the only reason i have handbrake installed in the first place [21:26:20 CET] <wellen> btw let me clarify i am thinking usable by users (not power users) [21:26:22 CET] <c_14> sfan5: yes [21:26:27 CET] <c_14> But it's a pain. [21:28:29 CET] <sfan5> fortunately nvenc isn't a pain [21:29:00 CET] <wellen> power users just 'get' the concept that all options are usable and that is what the advanced tab would be for .... the basic gui tab along with wizards would literally guide average users through parameters getting narrower and more defined as each wizard screen is completed.. this would not only allow average users to get things done easier but more importantly would free up the power users and the project devs from being asked for s [21:29:04 CET] <sfan5> wellen: you'll not even be able to mention codec names if you want it that easy [21:29:10 CET] <sfan5> it will just be [21:29:16 CET] <sfan5> input file [....] [21:29:21 CET] <sfan5> output file [....] [21:29:24 CET] <sfan5> format <selectbox> [21:29:57 CET] <wellen> sfan5 - give users some credit..... most know about common codecs, containers etc [21:30:14 CET] <dsl42> most do not, that is my experience [21:30:18 CET] <wellen> in fact sfan i owuld agree on your last thought [21:30:31 CET] <sfan5> i wouldn't expect "users" to know what a container is [21:30:36 CET] <dsl42> right [21:30:52 CET] <sfan5> most people know the "formats" by file extension [21:31:09 CET] <wellen> the user shouldnt need ot get 'stuck in' if all they want to do is alter the resolution etc as the wizard task would not call for a conversion so the codec would remian the same for the usage [21:31:28 CET] <sfan5> i don't know what an user is too you [21:31:40 CET] <sfan5> but i rarely see someone i would call "user" wanting to scale a video [21:32:05 CET] <c_14> What you need is a natural language processing neural network that tries guessing what the user wants and then just does it. [21:32:20 CET] <sfan5> like cdecl.org for ffmpeg [21:32:22 CET] <sfan5> but more advanced :D [21:33:13 CET] <wellen> c_14 ... we need this guy then.... Derek Smart [21:35:10 CET] <furq> derek smart is famous for his intuitive guis [21:38:06 CET] <wellen> intuitive rhymes with guis.... lies? [22:54:12 CET] <skokkk> I want to convert my full video library to .mp4 with minimal quality loss. My library has avi's, flv's, and a few more.. [22:54:23 CET] <skokkk> What is the recommended way to do this? [22:54:36 CET] <skokkk> This is for a few thousand videos [22:55:00 CET] <c_14> You should be able to remux most of them, you'll probably have to reencode the flvs [22:55:14 CET] <skokkk> remux? [22:55:23 CET] <J_Darnley> use -c copy where possible [22:56:15 CET] <skokkk> I'm very new to ffmpeg. What does -c mean? [22:56:48 CET] <furq> codec [22:57:24 CET] <BtbN> why convert it? mp4 isn't exactly a good format, and those formats should play almost everywhere. [22:58:13 CET] <skokkk> not for html5 web on my phone's opera.. [23:55:15 CET] <giusef> Hi all [23:56:14 CET] <giusef> I want ffserver to stream to clients without re-encoding since the feed is already in the right format (to not overload the server). Is that possible [23:56:15 CET] <giusef> ? [23:59:54 CET] <giusef> done :) [00:00:00 CET] --- Sun Mar 13 2016

1 0

ffmpeg-devel.log.20160311
by burek 12 Mar '16

12 Mar '16

[00:13:54 CET] <cone-066> ffmpeg 03Paul B Mahol 07master:681f83361026: avfilter/vf_waveform add support for other scalers for graticule [00:16:41 CET] <cone-066> ffmpeg 03Paul B Mahol 07master:1a5895757db8: avfilter/vf_waveform: fix indentation [00:46:09 CET] <cone-066> ffmpeg 03Michael Niedermayer 07master:b2ab3398f59e: avformat/hlsenc: Fix passing options, regression since bc9a5965c815cf7fd998d8ce14a18b8e861dd9ce [03:23:28 CET] <michel71> Hello all. I have been referred to this IRC to inquire as to how I could contribute to the FFMpeg project. [03:24:25 CET] <michel71> I am currently a graduate student in EE, and my research topic is related to signal processing. Any idea where I could start? [03:33:38 CET] <Shiz> in light of a specific program or in general? [03:34:29 CET] <Compn> michel71 : any experience in C or asm ? [03:36:57 CET] <michel71> In general, especially to those topics that are mathematical in nature. I have experience in C, but almost none in assembly. [03:38:44 CET] <J_Darnley> If you didn't realise, Shiz was asking whether you are looking to apply to GSOC or Outreachy. [03:39:03 CET] <Shiz> yes, that [03:39:09 CET] <Shiz> i worded it rather poorly [03:39:55 CET] <Shiz> gsoc prep is in full force right now, so there's quite a few people whoo come in and ask about in light of it [03:40:25 CET] <rcombs> lovely ticket there [03:40:35 CET] <rcombs> "please use Core Image for& something or other" [03:40:37 CET] <J_Darnley> Yes, I've seen them pop up everywhere [03:41:00 CET] <michel71> Oh, sorry. Nothing related to GSOC or Outreachy. [03:41:17 CET] <J_Darnley> That's fine. I just thought everyone should be clear. [03:41:45 CET] <jamrial> michel71: get familiar with the code. look at existing decoders and encoders, see if you can improve them in some way [03:42:01 CET] <J_Darnley> Start by making sure you can compile it. [03:42:02 CET] <jamrial> maybe look at trac and see if you can find a bug ticket you can fix [03:42:17 CET] <Shiz> yeah the best source for things to work on is probably the trac [03:42:30 CET] <cone-066> ffmpeg 03Ganesh Ajjanagadde 07master:66edd8656b85: lavc/lpc: exploit even symmetry of window function [03:44:14 CET] <michel71> Thanks, I will definately check out the Trac Issue Mailing list. [03:44:32 CET] <Shiz> for new things, the 'enhancement' and 'task' types are probably what you want to filter for, but there's also plenty of bugs out there [03:44:51 CET] <Shiz> http://ffmpeg.org/developer.html this is also generally useful [03:47:42 CET] <michel71> Thanks for the reference, I'll check them out. [03:48:41 CET] <michel71> Any idea on the required mathematical prereqs? Linear Algebra, Numerical Analysis, etc? [03:49:04 CET] <J_Darnley> high school algebra [03:49:19 CET] <J_Darnley> so that you understand variables and math with no numbers [03:49:44 CET] <Shiz> also depends in what area you want to contribute, probably [03:50:20 CET] <Shiz> improving an upscaler has different needs than adding support for a streaming protocol [03:51:29 CET] <michel71> Ah okay, right, sounds good. Thanks for all the info, much appreciated, I'll get to reading. [03:52:12 CET] <Shiz> don't be afraid to ask (not too vague) questions [03:52:24 CET] <Shiz> just be aware of a possible response delay [09:09:44 CET] <ismail> durandal_1707: noone paid but please consider donating me 100k ¬ [09:52:35 CET] <cone-137> ffmpeg 03Paul B Mahol 07master:119c108b31c0: avfilter/vf_waveform: fix ire8 for 25% [10:00:41 CET] <durandal_1707> ismail: 100k, yes Sir [10:02:39 CET] <wm4> ismail: done, subtracted 50k taxes, and 50k no-fun fee [10:03:22 CET] <ismail> wm4: mpv tax? Worth it :P [10:05:27 CET] <wm4> no-fun tax actually [10:06:39 CET] <ismail> you can't tax & fee for the same thing [10:06:50 CET] <wm4> oh yes we can! [10:07:19 CET] <wm4> wow there's a ticket for using coreaudio for decoding [10:10:50 CET] <ubitux> i'm trying to play with .altmacro (gas) to do something like v%(1+1).8H [10:11:17 CET] <ubitux> (to point on a specific simd reg) [10:11:25 CET] <ubitux> but can't make it work [10:11:28 CET] <ubitux> any suggestion? [10:26:14 CET] <cone-137> ffmpeg 03Paul B Mahol 07master:867637caeab5: avfilter/vf_waveform: fix and extend millivolts grat lines [10:26:23 CET] <wbs> ubitux: you may need to wrap it in a separate macro, so that the 1+1 thingie is a macro parameter and gets evaluated there [10:26:45 CET] <ubitux> mmh ok, let's try [10:26:47 CET] <wbs> ubitux: in general, altmacro is problematic, it's one of the few gas features that clang hasn't implemented yet afaik [10:27:10 CET] <wbs> (so when I end up with something where I would like to do arithmetic like that, I go for a more verbose solution than rely on altmacro) [10:27:45 CET] <ubitux> yes, we can't build for android with the clang in the ndk [10:27:58 CET] <ubitux> are we supposed to use gas pp here as well? [10:28:13 CET] <ubitux> anyway, i just discovered noaltmacro because of this, so now i want to use it :$ [10:28:45 CET] <wbs> noaltmacro? [10:29:00 CET] <ubitux> sorry, altmacro [10:29:09 CET] <wbs> ah, k [10:29:16 CET] <wbs> yes, clang builds for 32 bit arm need gas-pp due to this [10:29:37 CET] <ismail> https://llvm.org/bugs/show_bug.cgi?id=18918 looks like stalled. [10:29:40 CET] <wbs> I guess clang says "patch welcome" if you nag them about it; it's a feature that e.g. the linux kernel doesn't use (afaik) so those who improved the gas compat in clang haven't bothered [10:29:53 CET] <wbs> janne filled in a lot of the other details, but this one is still not done [10:30:37 CET] <ubitux> alright, so enabling gaspp is not specific to darwin [10:30:48 CET] <ubitux> but just clang gas [10:31:17 CET] <wbs> no, not specific to clang gas either [10:31:37 CET] <nevcairiel> to any "non-gnu gnu assembler" =p [10:31:37 CET] <wbs> it's for anything that doesn't fulfill everything that our gas asm (either arm or ppc) requires [10:31:54 CET] <wbs> before clang, apple still used binutils gas [10:32:00 CET] <wbs> ... forked from upstream binutils around 1991 [10:32:06 CET] <nevcairiel> haha [10:32:12 CET] <wbs> so it lacked most of the actual gas features [10:32:18 CET] <wbs> this is what gas-pp was written to support initially [10:32:26 CET] <wbs> later clang came along, and didn't support much more gas stuff either [10:32:35 CET] <wbs> then clang was fixed to be able to build the linux kernel, and stuff improved [10:32:52 CET] <wbs> and janne fixed some more stuff, so for aarch64, you don't need gas-pp, but for arm you still need for altmacro [10:32:59 CET] <wbs> and then I fixed stuff for microsoft compilers [10:33:15 CET] <wbs> since they have a completely different syntax, but gas-pp can output that as well [10:33:39 CET] <wbs> so instead of just expanding the macros, it rewrites all the directives (most of the asm itself passes through pretty much unchanged) [10:33:57 CET] <wbs> so today you need it for apple-gas, clang-anything, and ms armasm [10:36:54 CET] <ubitux> http://sprunge.us/hCdW :( [10:37:06 CET] <ubitux> well, whatever :) [10:40:06 CET] <ubitux> anyway, ok for gaspp, thanks [10:40:08 CET] <wbs> move the altmacro/noaltmacro to around the foo %() call [10:40:25 CET] <ubitux> ooh [10:40:30 CET] <ubitux> thanks! [10:40:38 CET] <ubitux> nice :) [10:44:46 CET] <mateo`_> wbs: thanks for the info, i'll give another try at building ffmpeg with clang3.8+gas-pp for arm [10:45:29 CET] <ismail> you can also use -no-integrated-as [10:45:33 CET] <ismail> for clang [10:45:41 CET] <wbs> ah, yes [10:47:06 CET] <mateo`_> ismail: what does -no-integrated-as do ? [10:47:37 CET] <ismail> mateo`_: it disables clang's internal assembler and uses gas [10:47:43 CET] <ismail> which will fix assembler errors [10:55:28 CET] <ubitux> http://sprunge.us/HjER thanks wbs :D [11:05:24 CET] <mateo`_> ismail: thanks a lot, the build succeeded [11:08:59 CET] <ismail> mateo`_: you are welcome. [11:28:24 CET] <nevcairiel> changed all my windows fate boxes to use dash instead of bash, lets see if something breaks next run! [11:29:08 CET] <fritsch> any reason to do so? working towards "sh compatibility"? [11:29:32 CET] <nevcairiel> dash is just much faster on windows for some reason [11:29:56 CET] <ubitux> i'm running my configure with dash locally all the time [11:30:23 CET] <nevcairiel> presumably using busybox would be much faster still [11:30:31 CET] <nevcairiel> but no clue how much that would break [11:31:45 CET] <ubitux> /bin/bash ./configure > /dev/null 11.77s user 0.71s system 78% cpu 15.910 total [11:31:47 CET] <ubitux> /bin/dash ./configure > /dev/null 3.31s user 0.44s system 55% cpu 6.794 total [11:31:55 CET] <cba321> i am also trying to check this channel to comply with a message in the code to "please do not contact google" [11:32:28 CET] <nevcairiel> ubitux: here its something like 150s vs 100s =p [11:32:53 CET] <cba321> is this the right channel to speak to ffmpeg.org [11:34:30 CET] <cba321> why put a message in the code to not let google know anything...if no one from ffmpeg is around [11:35:44 CET] <wm4> cba321: where? [11:36:32 CET] <cba321> wm4...ok i will be happy to report the issue to the right person who put the message in the code...so that google can't find out anything. [11:36:49 CET] <cba321> wm4...are you a ffmpeg developer [11:38:14 CET] <fritsch> cba321: that does not matter - post a link to the file in question ... [11:38:43 CET] <cba321> fritsch...hell yes it does matter....i respect coders wishes...maybe you don't respect anything you damn motherfucker [11:39:03 CET] <BtbN> is this some next level trolling? [11:39:16 CET] <wm4> apparently so [11:39:19 CET] <fritsch> i think it is [11:48:47 CET] <mateo`_> cba321: where is this message located ? [11:49:13 CET] <cba321> mateo...who are you..are you a ffmpeg developer...i am trying to comply with the message. [11:49:30 CET] <ubitux> lol [11:49:37 CET] <ubitux> who knows [11:49:40 CET] <ubitux> better not trust him [11:50:25 CET] <cba321> i don't give a shit who or what you trust....i am trying to comply with a message by a small developer that doesn't want to be runover by big sharks. [11:50:56 CET] <BtbN> Well, the ffmpeg codebase is entirely open, so google already indexed that holy file anyway. [11:50:58 CET] <ubitux> there are a few google spies here [11:51:23 CET] <BtbN> I also highly doubt ffmpeg contains any message like that. [11:51:48 CET] <cba321> BtbN...obviously you don't spend time with the code to try to make anything work [11:52:36 CET] <cba321> BtbN if you don't even know what messages are contained in various projects [11:55:10 CET] <nevcairiel> if you think anyone is going to talk to you after you already insulted the first person, you are surely misguided [11:55:25 CET] <ubitux> nevcairiel: well, you still talk with mats :° [11:55:41 CET] <cba321> nevcairiel...ask me if i give a shit...if the shoe fits...wear it you god damn bastards. [11:56:19 CET] <cba321> nevairel..i respect the lowly..not the giant monsters [11:58:55 CET] <cba321> what made Tifa famous is that she was merely a lowly bartender....who was able to rise to conquer the world as well as some crackpot who thought he was God. [11:59:54 CET] <ritsuka> you know, this channel is called ffmpeg-devel, is not like the people here are totally unrelated with ffmpeg, you can just post your error and be done [12:00:31 CET] <cba321> ritsuka...go to hell...i am trying to make sure information does not fall into the wrong hands...this is at the request of a lowly programmer. [12:00:53 CET] <fritsch> i could not parse your request, sorry ... [12:01:09 CET] <ubitux> cba321: so what are you going to do? [12:01:17 CET] <cba321> fritsch...well maybe you don't know how to configure the right parser for the right formats [12:02:00 CET] <wm4> "checking value of MSG_OOB... 1" dear god why would you do this, glib [12:02:22 CET] <nevcairiel> wm4: configure scripts are full of such crazy things [12:02:45 CET] <wm4> sometimes they seem to be a little bit too crazy [12:03:09 CET] <cba321> nevcairiel...why don't you try to learn grep , egrep, fgrep and so on to save some time looking for things [12:04:18 CET] <fritsch> why don't you learn to use tab completion to save time and write other's names correctly? [12:06:05 CET] <cba321> well if nevcairiel is not the right word...that is too bad...because that is what i see here. [12:14:48 CET] <ubitux> i miss the guy already [12:15:10 CET] <mateo`_> well that was entertaining [12:15:56 CET] <nevcairiel> someone invent the remote punching machine for me [12:18:00 CET] <nevcairiel> the message he saw probably came from libvpx itself or something, which makes it even funnier [12:20:32 CET] <mateo`_> look at #ffmpeg :D [12:21:45 CET] <ubitux> http://sprunge.us/acOK for those not on #ffmpeg [12:21:55 CET] <nevcairiel> should get him K-lined just out of spite =p [12:29:35 CET] <rcombs> oh no, google's going to get our precious& sample files? [13:08:23 CET] <rcombs> "+1 for this enhancement" [13:08:27 CET] <rcombs> how extremely useful [13:09:07 CET] <rcombs> any reason not to just merge the lavc audiotoolbox stuff [13:09:39 CET] <rcombs> might have some issue around end behavior in HE-AAC, but there's no such thing as a regression in a new encoder [13:19:34 CET] <wm4> rcombs: I'm noticing you never seem to set any codeccontext parameters (except sample_fmt) [13:20:01 CET] <wm4> and channels, maybe [13:20:11 CET] <wm4> what about sample rate and channel layout? [13:21:05 CET] <rcombs> audiotoolbox's assumptions around that are kinda weird [13:21:13 CET] <rcombs> seems to expect you to set them early on [13:21:46 CET] <nevcairiel> it expects you to know them before calling the decoder? [13:21:48 CET] <rcombs> but I could probably try reading the values back after sending extradata and each frame [13:21:50 CET] <nevcairiel> that will end up breaking =p [13:21:53 CET] <wm4> even for formats which fully have them in the headers? [13:22:03 CET] <rcombs> nevcairiel: it expects to get _something_, at least [13:22:03 CET] <wm4> like mp3 or ac3 should have [13:24:29 CET] <rcombs> fortunately the bitmap format for channel layouts is the same in lav* as in audiotoolbox [13:25:06 CET] <nevcairiel> its just the wav way! [13:25:06 CET] <rcombs> (iirc the ordering comes from mpeg or the like, so makes sense) [13:25:37 CET] <wm4> well that means nothing, 3 channels still could be either 2.1 or 3.0 etc. [13:25:46 CET] <rcombs> yeah [13:26:18 CET] <rcombs> just means that when moving channel layout bitmaps between lavc and at, I don't have to remap anything [13:26:38 CET] <rcombs> nor when actually moving around audio data [13:30:07 CET] <nevcairiel> using the wave/riff layout is so common that even apple couldnt pretend to invent a new one, i suppose [13:30:33 CET] <andrey_utkin> I am sorry, is Claudio Freire here? I want to ask regarding "AAC encoder 3x performance drop in 3.0 since Oct 2014" topic [13:32:07 CET] <rcombs> andrey_utkin: tl;dr the AAC encoder had some major improvements made focused on output quality/efficiency, and there hasn't been much time spent optimizing that new code for speed yet [13:32:30 CET] <kierank> send patches [13:32:31 CET] <rcombs> or, well, not enough time to get it back near where it used to be on speed [13:32:45 CET] <andrey_utkin> I just want to ask if I can currently speed things up by, like, turning off twopass [13:32:52 CET] <andrey_utkin> because bitrate is not a concern [13:32:59 CET] <rcombs> twopass in AAC? [13:33:13 CET] <nevcairiel> he means twoloop, and no, all the alterntives are crap [13:33:35 CET] <rcombs> you could try `-aac_coder fast`, but you'll pay in quality [13:33:37 CET] <nevcairiel> you could try the fast coder, but i believe its extremely bad [13:33:57 CET] <andrey_utkin> err twoloop [13:34:19 CET] <andrey_utkin> i tried fast coder, no significant change [13:35:03 CET] <andrey_utkin> i am not proficient enough to patch encoder, I can probably fund the development somewhat [13:35:22 CET] <kierank> make some perf reports perhaps [13:37:13 CET] <nevcairiel> you could try vbr, from what I understand a major speed problem is that the psy produces rather bad guesses, making the encoder work hard to reach the target bitrate .. maybe that wouldnt have such a big impact in vbr [13:38:58 CET] <nevcairiel> recent changes to aacenc should have already made it 20-30% faster [13:41:25 CET] <kierank> ak____: hello [14:02:28 CET] <nevcairiel> so i broke my fate boxes earlier today, and for some reason t he fate website has already swallowed their entries [14:02:30 CET] <nevcairiel> how odd [14:02:39 CET] <nevcairiel> usually they take month to disappear if inactive [14:03:57 CET] <ubitux> i haven't updated my --disable-everything fate instance since 11 days and it's still there [14:04:12 CET] <nevcairiel> no clue why those vanished so quickly [14:04:19 CET] <ubitux> did they broke by uploading empty reports or sth? [14:04:21 CET] <nevcairiel> hope they come back when a fixed run goes through now =p [14:04:34 CET] <nevcairiel> dont think the rpeort was empty, just very short [14:04:48 CET] <ubitux> empty or simply broken in a way fate doesn't like them [14:05:04 CET] <ubitux> anyway, better ask Timothy_Gu [14:06:35 CET] <nevcairiel> it failed configure and the report reflected as much [14:06:36 CET] <nevcairiel> oh well [14:07:05 CET] <nevcairiel> i made a copy of the report if Timothy_Gu is interested later [14:30:51 CET] <toot> hi, can someone help me, and say how to add/register a new demuxer? i've searched documentation, but found almost nothing [14:32:08 CET] <Compn> you can check git log for when another demuxer was added [14:32:12 CET] <Compn> and then just copy what was done [14:33:12 CET] <toot> surely, thanks [14:45:32 CET] <wm4> toot: libavformat/allformats.c [14:45:53 CET] <wm4> the configure script parses this (which causes most of the confusion) [14:46:59 CET] <rcombs> yeah, add to allformats.c and libavformat/Makefile [14:50:29 CET] <wm4> rcombs: btw. what's the state of your ordered chapters patch, and does it have edition support? [14:51:06 CET] <rcombs> sitting around largely functional but in need of better error detection and handling (and probably better perf), and yes [14:51:24 CET] <wm4> performance? [14:51:46 CET] <rcombs> opening files to check if their UUIDs are what we're looking for [14:51:51 CET] <wm4> right [14:51:52 CET] <rcombs> it's currently slower than it could be [14:51:58 CET] <rcombs> does more seeks than necessary [14:52:12 CET] <wm4> it should probably just open a raw stream and do some manual ebml parsing [14:52:15 CET] <wm4> does it have edition support? [14:52:41 CET] <rcombs> yes [14:53:24 CET] <wm4> how? [14:53:45 CET] <rcombs> iirc just an AVOption for which edition to use [14:54:17 CET] <wm4> I see [14:54:25 CET] <rcombs> the 3 big things I wanted to add were support for changing extradata (by sending packet side data indicating the change), for providing a list of files to check on the CLI (as opposed to scanning the dir [which would still be the default]), and for exposing the editions through some API [14:55:40 CET] <rcombs> wm4: oh, wait, I just hardcoded it to use the first edition right now [14:55:50 CET] <rcombs> but doing the AVOption thing would be trivial [14:56:06 CET] <rcombs> it's got full support for it and just always picks the first one [14:56:48 CET] <wm4> what I'd like to have is if the demuxer could export the full ordered chapter info, so that API user (mpv) can do it on its own [14:57:22 CET] <rcombs> yeah [14:57:59 CET] <rcombs> I mean, I could stick it in an AV_OPT_TYPE_BIN with some defined struct format if I wanted [14:58:12 CET] <rcombs> and say you're not allowed to modify it [14:58:19 CET] <wm4> maybe glorious side data [14:58:31 CET] <rcombs> is there demuxer-level side data? [14:58:32 CET] <wm4> I also need to add some other features, like subtitle preroll [14:58:35 CET] <rcombs> or metadata I guess [14:58:36 CET] <wm4> I think so [14:58:49 CET] <rcombs> but it seems like we might want a proper API for this sort of thing [14:59:09 CET] <rcombs> remind me how "programs" work right now? [14:59:17 CET] <rcombs> (that's an mpegts thing, right?) [14:59:29 CET] <wm4> ah I was wrong, no demuxer side data yet [14:59:33 CET] <wm4> just AVStream [14:59:39 CET] <wm4> yes [14:59:39 CET] <rcombs> yeah [15:03:20 CET] <nevcairiel> bl [15:03:22 CET] <nevcairiel> woops [15:03:40 CET] <rcombs> nevcairiel's into BL? [15:04:07 CET] <Daemon404> or worse [15:04:08 CET] <Daemon404> arm assembly [15:04:09 CET] <rcombs> or is this ARM or PPC ASM (I forget which uses that syntax for "branch if lower") [15:04:25 CET] <rcombs> Daemon404: hah [15:04:53 CET] <Daemon404> :D [15:05:12 CET] <nevcairiel> just not info shifting focus while typing [15:05:25 CET] <rcombs> ENOTTY [15:15:38 CET] <Daemon404> does mats not understand the concept of a bug tracker [15:31:53 CET] <kierank> J_Darnley: any luck with haar? [15:46:36 CET] <J_Darnley> I haven't been working on it [15:47:29 CET] <J_Darnley> I've been too busy wasting time on the internet trolling and shitposting [15:47:56 CET] <Daemon404> hopefully those shitposts are dank memes [15:49:18 CET] <J_Darnley> No. I am too poor in dank memes [16:04:59 CET] <durandal_170> he got free job at #ffmpeg [16:05:37 CET] <Daemon404> act [16:05:42 CET] <Daemon404> woops [16:05:52 CET] <thardin> did someone say memes? [16:06:11 CET] <Daemon404> not memese [16:06:12 CET] <Daemon404> dank memes [16:06:42 CET] <thardin> ofc. no one wants stank memes [16:10:08 CET] <Daemon404> https://github.com/blog/2119-pull-request-and-issue-reactions [16:10:13 CET] <Daemon404> but we still cant disable PRs [16:11:08 CET] <wm4> imagine a code base where disabling this can't be a simple flag, while for issues it can [16:13:33 CET] <Daemon404> wm4, yes [16:20:59 CET] <J_Darnley> if you disable pull requests people might discover that there are other ways to contribute that don't even have to involve github [16:21:49 CET] <Daemon404> :D [16:57:18 CET] <Gramner> obviously they want you to use their service instead of going somewhere else [17:03:59 CET] <BBB> should by bsf change bump micro or minor? [17:04:16 CET] <durandal_170> mayor [17:04:28 CET] <BBB> that seems mildly aggressive [17:06:39 CET] <BBB> Ill bump micro then? [17:08:29 CET] <BBB> new bsf, maybe & minor? [17:08:30 CET] <Shiz> should cause a library rename [17:08:31 CET] <BBB> I dont know [17:10:42 CET] <iive> it breaks abi and api, of course it should be the mayor [17:11:09 CET] <iive> ;) [17:15:21 CET] <BBB> it only breaks some eggs [17:19:39 CET] <cone-338> ffmpeg 03Ronald S. Bultje 07master:6d8ab358a3c2: lavf: allow BSFs to drop packets. [17:19:40 CET] <cone-338> ffmpeg 03Ronald S. Bultje 07master:2e6636aa8730: vp9: add superframe merging bitstream filter. [17:48:10 CET] <jamrial> BBB: can the new bsf be invoked by the user, for example with the cli, or is it autoinjected only? [17:48:20 CET] <BBB> both [17:48:38 CET] <BBB> it automatically adds itself when muxing vp9 into ivf or mkv [17:48:38 CET] <jamrial> then you should have bumped minor, since it's a new component :p [17:48:44 CET] <jamrial> at least that's how it always goes [17:48:50 CET] <BBB> -ETOOLATE [17:49:03 CET] <nevcairiel> that is rather arbitrary anyway [17:49:08 CET] <nevcairiel> cli users dont care about the versions [17:49:10 CET] <BBB> feel free to change minor, I dont mind [17:50:12 CET] <kierank> where's a good place to upload large files [17:50:42 CET] <wm4> does anyone care about the versions? [17:50:50 CET] <jamrial> mega? [17:53:50 CET] <kierank> upload speed very very slow [17:54:28 CET] <wm4> what's the file size? [17:57:13 CET] <kierank> A bunch of files totalling 20GB or so [17:57:33 CET] <wm4> ok that's huge [17:57:44 CET] <kierank> I guess I have a box.net account [17:57:54 CET] <kierank> I wonder if it'll let me upload directly [17:58:52 CET] <kierank> nope files too big [20:07:19 CET] <jamrial> BBB: i'm getting a lot of "[NULL @ 000001980e815040] Invalid superframe packet size: 15108867 frame size: 249065" error messages decoding a webm file i created using codec copy from another webm as source [20:07:23 CET] <jamrial> decoding doesn't fail, but the md5 output is different than before [20:07:53 CET] <BBB> after my patch just now? [20:07:56 CET] <jamrial> yeah [20:07:59 CET] <BBB> huh... [20:08:05 CET] <BBB> how do I reproduce? [20:08:35 CET] <jamrial> i used the tears of steel vp9 video ubitux or nevcairiel made back when you first benched the decoder [20:08:39 CET] <jamrial> tos3k-vp9-b10000.webm [20:09:20 CET] <jamrial> i did a simple "./ffmpeg -i tos3k-vp9-b10000.webm -c:v copy -t 10 tos.webm" then "./ffmpeg -i tos.webm -f md5 -" [20:09:27 CET] <BBB> let me try... [20:10:00 CET] <jamrial> muxing is also considerably slower now, but i guess it's to be expected since it's running the bsf code [20:12:46 CET] <BBB> it shouldnt be slower [20:12:56 CET] <BBB> I mean its not that much overhead [20:14:32 CET] <BBB> oh well [20:14:38 CET] <BBB> not sure what happened here [20:14:46 CET] <BBB> but it seems like were writing the index in BE but reading it in LE [20:14:50 CET] <BBB> that would be & troubling [20:14:55 CET] <BBB> how did this work originally? [20:19:00 CET] <BBB> (writing a fix) [20:22:54 CET] <BBB> jamrial: fix on ML [20:23:01 CET] <BBB> jamrial: with that, the two MD5s are identical for me [20:24:24 CET] <jamrial> actually, nevermind about the muxing speed thing. the file wasn't cached on memory the first time i ran it so it seemed slower [20:24:34 CET] <BBB> makes sense then [20:24:38 CET] <BBB> ty for testing [20:27:00 CET] <jamrial> i see nothing on the ml [20:27:07 CET] <jamrial> and no prob [20:30:37 CET] <jamrial> ah, there it is [20:33:04 CET] <nevcairiel> how did that ever work =p [20:33:29 CET] <jamrial> BBB: works great now, thanks [20:40:37 CET] <BBB> like my kid sometimes says [20:40:44 CET] <BBB> dont ask me questions like that I dont like it!" [20:42:53 CET] <cone-259> ffmpeg 03Ronald S. Bultje 07master:abedde65d1ce: vp9_superframe: fix endianness of size markers. [20:43:27 CET] <BBB> so I think trac is lacking any further vp9 tickets except some trivial signed integer leftshifts which are silly but Ill fix them anyway [20:47:10 CET] <jamrial> BBB: did you check that argon streams thing someone mentioned the other day? [20:47:26 CET] <BBB> google did [20:47:33 CET] <BBB> I dont have the streams [20:48:07 CET] <jamrial> can't you get your hands on them? would be nice to check your decoder with it [20:48:32 CET] <BBB> I dont think they were run under all combinations of ubsan, asan, helgrind, tsan etc, but just a plain md5 check matches between ffvp9/libvpx for all streams [20:48:35 CET] <jamrial> then again, ffvp9 coverage seems to be above 90% [20:48:42 CET] <BBB> I could probably buy them, but what do I care [21:53:22 CET] <BBB> jamrial: wanna review the other vp9 patch? [22:06:38 CET] <BBB> ty [22:07:41 CET] <cone-259> ffmpeg 03Ronald S. Bultje 07master:b86339a9f8d7: vp9: fix a few signed integer left-shifts. [22:54:36 CET] <llogan> "lavf/riffenc: Improve spec compliance" 28 of 33 messages are one guy talking to himself. [22:55:13 CET] <jamrial> Mats? [22:55:27 CET] <llogan> RE: RE: RE: RE: yes. [22:56:19 CET] <llogan> i guess some of those are replies to real replies, but i still don't get his methodology [22:58:31 CET] <BBB> mats is crazy yes [23:01:52 CET] <BBB> weve told him several times to not do that [23:01:55 CET] <BBB> he just doesnt listen [23:06:03 CET] <llogan> also, why the interest in riff? [23:10:30 CET] <cone-259> ffmpeg 03Paul B Mahol 07master:817d0c6da26b: avfilter/vf_waveform: add parade display mode [23:57:49 CET] <cone-259> ffmpeg 03Shivraj Patil 07master:15ef98afd10b: configure: build fix for P5600 along option --disable-msa [00:00:00 CET] --- Sat Mar 12 2016

1 0

ffmpeg.log.20160311
by burek 12 Mar '16

12 Mar '16

[04:28:42 CET] <jookiyaya> didn't workcan you people tell the difference between x264 video and apple h264 encoded video [05:03:33 CET] <tacosforall> Hi... I'm trying to render an animation but I'm getting inconsistent keyframing in x264. Each frame should appear four times per 30 frames but I'm seeing inconsistent results. I realize this is a well-documented scenario but I've been coding video and audio software for 20 years (really!) and I'm seeing weirdness no matter what I try with keyframes, time bases, and various filter tricks. Can someone give me some top-level guidel [05:05:40 CET] <tacosforall> Specific to what I'm doing, I'm using a concat filter (-f concat -i input.txt) with images and duration lines. The file length is exactly what it should be. However the durations jitter. [05:06:12 CET] <tacosforall> [You really should be able to just specify frames instead of durations, however apparently this is a 10 year open issue that will not get resolved.] So here I am. [05:13:25 CET] <tacosforall> I'm staring at ffprobe output that's clearly a bug. I'll file it. Thank you for the silence, it was soothing. [05:42:04 CET] <fling> Which container to use for opus only stream? [05:47:47 CET] <c_14> ogg? [05:47:56 CET] <c_14> Or any container that supports opus? [07:10:38 CET] <limbo_> what's the best way to re-encode an mp3 without losing any quality when mediainfo doesn't show any duration for it, and itunes shows the incorrect duration? I've already done ffmpeg -i <in> -codec copy <out> [07:19:54 CET] <kepstin> limbo_: you probably want to use the standalone "vbrfix" tool [07:20:08 CET] <kepstin> it'll analyze the mp3 file and add the missing vbr info header [07:25:34 CET] <limbo_> ok. where can I find it and how do I use it? Before I ran it through ffmpeg, mediainfo told my it was constant bitrate. [07:26:11 CET] <kepstin> if the file doesn't have the vbr header, most programs just read the bitrate of the first frame and say the file's cbr with that bitrate [07:26:25 CET] <kepstin> and then they'll often get wrong or missing length estimates [07:26:46 CET] <kepstin> cbr mp3 files don't store the length anywhere - you guess it by dividing file size by bitrate [07:27:08 CET] <kepstin> vbr files need a special header with extra info to determine the length (unless you analyze the entire file) [07:27:44 CET] <kepstin> but yeah, just google "vbrfix". there's a window binary available, and it's packaged in at least some linux distributions. [07:30:45 CET] <limbo_> mediainfo tells me that the outputted files are variable bitrate. [07:37:06 CET] <limbo_> I'm running it, an getting this output (with errors) https://paste.debian.net/414108/ and lines in other runs like "Warning - more than one id3v1 tag found using last found" [07:50:50 CET] <kepstin> fun. your file is actually multiple mpeg audio files (apparently not even the same mpeg version) which have been concatenated without so much as removing the tags [07:50:58 CET] <kepstin> no wonder tools aren't working properly with it [07:51:37 CET] <kepstin> hmm, could be wrong, but either way it's an oddly broken file. [07:57:17 CET] <limbo_> damn. [07:57:24 CET] <limbo_> well, that's 2001 for you. [07:57:49 CET] <limbo_> I have a couple of files, an I guess some of them are fine then. [07:58:44 CET] <limbo_> any suggestions on how best to re-encode these so I can at least play these on an ipod without them failing to seek properly or jumping aroung the place every twenty or so minutes? [08:05:30 CET] <limbo_> They're meant to be about 30kbps, so I don't know what to even re-encode them to. [08:10:16 CET] <kgkg> Hiall! I am trying to forward an IP Camera to youtube but get a "Failed to update header" error. CLI and error: http://pastebin.com/KJX9mZFV [09:22:40 CET] <wmp> hello [09:23:11 CET] <wmp> i have capture dark(night) video from security camera [09:23:22 CET] <wmp> and i want to remove duplicated frames [09:23:31 CET] <wmp> all works on videos with light [09:24:19 CET] <wmp> but when is dark ffmpeg not remove any frames, becouse is differences on single pixels [09:25:22 CET] <wmp> this is my ffmepg options: "-vf "select=gt(scene\,0.00001),setpts=N/(25*TB)" -f segment -segment_time 120 -segment_format mp4 -strftime 1 -reset_timestamps 1" [09:34:03 CET] <jookiyaya> 1.846153846153846] [09:39:40 CET] <aRDa_> hi [09:40:20 CET] <aRDa_> I'm new with ffmpeg I'm recording (from debian jessie) making a video of my screen with the following command: ffmpeg -video_size 1366x768 -f x11grab -r 25 -i :0.0 -qscale 0 video.avi [09:40:36 CET] <aRDa_> but the sound is not recorded, any ideas why ? [11:28:25 CET] <cba321> hello is this the dev channel..i got a message from the compilation of ffmpeg to not report a bug but instead contact ffmpeg.org [11:29:23 CET] <cba321> since the message said "please do not contact Google or someone...i am trying to comply." [11:53:56 CET] <DHE> cba321: there is a distinct dev channel, but you might be better off making a ticket on the trac page... (assuming one doesn't already exist) [12:09:29 CET] <cba321> well it looks like there are no developers around at this time...perhaps a few google spies who need to be incarcerated....especially since some of the code is relevant to vp8/vp9 and so on. [12:10:49 CET] <cba321> nothing is sicker than monstrous 800 pound gorillas trying to steal and capitalize on small people's ideas. [12:14:07 CET] <cba321> and i say a hearty ...fuck you crooked nigger dick suckin Jew bastards... [13:23:50 CET] <andrey_utkin> can I ask anybody how to optimize performance of now-stable ffmpeg's aac encoder? http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/210082 [13:24:09 CET] <andrey_utkin> (link: maillist post by subject "AAC encoder 3x performance drop in 3.0 since Oct 2014") [13:25:04 CET] <andrey_utkin> oh there was a discussion, gonna read [13:46:19 CET] <fenris_mid> saluton. i try to compile ffmpeg git-commit 66edd8656b851a0c85ba25ec293cc66192c363ae and get the following error: libavcodec/h264pred_template.c:1276:19: internal compiler error: in vect_transform_stmt, at tree-vect-stmts.c:8176 [13:46:36 CET] <fenris_mid> does anybody have an idea what the problem could be? [13:46:57 CET] <fenris_mid> is it really just gcc's fault? [13:46:57 CET] <nevcairiel> which compiler? [13:47:17 CET] <fenris_mid> g++ (Debian 6-20160109-1) 6.0.0 20160109 [13:47:38 CET] <nevcairiel> so a pre-release compiler then, likely its fault [13:47:44 CET] <fenris_mid> k [13:48:07 CET] <nevcairiel> best would be to double check with gcc 5, and report the problem to the gcc developers so its fixed before 6 is released [13:49:55 CET] <flux> fenris_mid, I bet the developers would be ecstatic if you were able to infer a minimal reproduciton for it ;) [13:50:23 CET] <flux> probably these vectorization things are quite the bleeding edge stuff anyway in GCC.. [13:51:47 CET] <jkqxz> If you do report it to gcc then you'll probably want to reproduce on something more recent than two months ago first. [13:56:13 CET] <wmp> I have video from security camera, i want to speed up if isn;t motion on video, how can i do this? [14:03:41 CET] <andrey_utkin> wmp, cool idea. Maybe a custom video filter could do that. Analyze frame-to-frame difference and increment the timestamps reverse-proportionally to amount of changes [14:05:56 CET] <BtbN> Or dump it to individual frames, and then compare the difference between the frames, and if it's small enough, discard the later one [14:06:18 CET] <wmp> but i need to do fast ;) [14:07:39 CET] <wmp> now i test this: ffmpeg -y -i short.avi -an -vf "select='gt(scene\,0.02)',setpts=0.1*PTS" output_0.02_0.1.avi [14:11:53 CET] <J_Darnley> That looks like it will create horribly out of order images [14:12:20 CET] <J_Darnley> oh wait, I misread [14:12:24 CET] <J_Darnley> ignore that [14:31:26 CET] <wmp> J_Darnley: unfortunatelly, this dont detect if in room is man, so this dont works... [14:42:35 CET] <J_Darnley> And if a guy stands still in the middel of the frame there won't be any changes either [14:43:05 CET] <J_Darnley> It sounds like you want to go to computer vision levels of analysis [14:43:06 CET] <louiz> You could get a Paranormal Activity effect, automatically. [14:43:19 CET] <J_Darnley> Somethign that is not available in ffmpeg [14:44:02 CET] <wmp> :/ [14:48:07 CET] <wmp> J_Darnley: do you know in what software can i do it? [14:48:33 CET] <J_Darnley> nfi [14:48:42 CET] <J_Darnley> you will probably have to write your own [14:48:55 CET] <wmp> :( [14:53:30 CET] <andrey_utkin> wmp: opencv? [14:55:11 CET] <andrey_utkin> I can connect you with a guy who has written opencv-powered motion detection on frames provided by ffmpeg. The existing code is proprietary, though, so cannot be shared, but you could contact the guy [14:55:21 CET] <BAWSTi> Hey guys, is it normal that by using following command I don't get the file size information? 'ffmpeg -i inputFile.mp2' And if this is the way it is supposed to act, how can I get the file size? [14:56:51 CET] <wmp> andrey_utkin: now i looking for this: http://superuser.com/a/818412 [14:58:33 CET] <andrey_utkin> cool! thanks for sharing wmp. I wonder what "mp" in filter name stands for. Motion ...? [14:59:23 CET] <andrey_utkin> BAWSTi, you can get size by "ls -l file" or "stat file" [15:00:19 CET] <BAWSTi> andrey_utkin: so like ffmpeg -ls -l inputFile.mp2? [15:00:43 CET] <J_Darnley> No! [15:00:58 CET] <J_Darnley> Use the basic unix tool called ls [15:01:37 CET] <J_Darnley> Or are you not on Linux or mac os? [15:01:41 CET] <BAWSTi> oh, yea well that's gonna be a problem since that would require that much more work xD I'm actually using ffmpeg cli on android ;) [15:02:04 CET] <J_Darnley> Well then, how would you get the size of any file? [15:02:28 CET] <andrey_utkin> ffmpeg cli on android - what's that? [15:02:49 CET] <BAWSTi> That's what I'm asking, honestly I just assumed it would be possible with ffmpeg since it shows so much information about a file, why not the size then? : [15:03:38 CET] <J_Darnley> Because ffmpeg is far too big for such a simple use [15:03:56 CET] <J_Darnley> because the filesystem will tell you the size in much less time than ffmpeg [15:04:09 CET] <BAWSTi> There's an example of what it is @andrey_utkin https://github.com/WritingMinds/ffmpeg-android-java [15:04:23 CET] <BAWSTi> alright, thanks for the help anyway J_Darnley :) [15:07:32 CET] <J_Darnley> If you're programming, why the heck would you try and parse the output of some program? [15:07:42 CET] <J_Darnley> Surely there is a function you should be calling with a filename or some file object [15:10:10 CET] <andrey_utkin> J_Darnley, well, in the age of using jQuery for arithmetics... [15:17:10 CET] <wmp> J_Darnley, andrey_utkin, louiz: its possible with ffmpeg :D [15:17:12 CET] <wmp> ffmpeg -i upsy_krotkie.avi -filter_complex "[0:v] mpdecimate=hi=64*120:lo=64*30:frac=0.30 [a] ; [a] setpts='N/(25*TB)' [b] ; [b] fps=25 [c] ; [c] fps=25 [d]" -map "[d]" -c:v libx264 -f mp4 output_upsy_krotkietest.avi [15:17:46 CET] <J_Darnley> I'm glad you managed to get something your satisfied with [15:20:44 CET] <louiz> great [16:51:38 CET] <limbo_> If I have some mp3 files that are probably encodes at30kbps, what's the easiest/best way to re-encode them so I don't lose quality or increase filesize? [16:51:52 CET] <limbo_> It's about 7.1 GB of audio. [16:53:22 CET] <DHE> any re-encoding degrades the audio unless it's a lossless codec (eg: FLAC) [16:53:31 CET] <DHE> at 30 kilobits it probably sounds pretty bad already [16:56:53 CET] <zeryx> hey guys I have some questionable output from a splitting process I'm running [16:56:55 CET] <zeryx> Format mov,mp4,m4a,3gp,3g2,mj2 detected only with low score of 1, misdetection possible![mov,mp4,m4a,3gp,3g2,mj2 moov atom not found/tmp/vd/Hand_on_cereal_field.mp4: Invalid data found when processing input [16:57:10 CET] <zeryx> I'm recieving this error message and I'm unsure why [16:57:32 CET] <zeryx> when I run ffmpeg locally it doesn't complain, but when its run on my remote instance with the static ffmpeg binaries it fails; is this due to a missing library? [16:58:08 CET] <DHE> might be an old version that doesn't detect the file [16:58:18 CET] <DHE> or do you mean running remotely you're pointing at an HTTP url? [16:58:31 CET] <zeryx> running remotely as in on a remote server executing the code & returning an exception [16:58:36 CET] <zeryx> the file is the same [16:58:44 CET] <zeryx> (I'm handling everything myself in that respect) [16:58:54 CET] <zeryx> so you think it might be an older version of the binaries? [16:59:15 CET] <DHE> run it without parameters. there should be a date and version number you can use [17:02:01 CET] <zeryx> 2.8.5 [17:02:23 CET] <zeryx> https://gist.github.com/zeryx/3dbddc3dd1b6da69ee9e [17:04:06 CET] <limbo_> DHE: why does it degrade the quality? Is there something about mp3 compression that I'm missing? also, will ffmpeg pick a sane default based on the input if I don't give it a quality setting. [17:04:10 CET] <limbo_> ? [17:05:30 CET] <kepstin> limbo_: all lossless codecs have an issue with "generation loss" where each time you re-encode, you lose some quality. [17:05:35 CET] <kepstin> all lossy codecs* [17:05:42 CET] <kepstin> lossless ones don't have this issue, obviously [17:06:24 CET] <kepstin> limbo_: if you don't specify any quality setting, ffmpeg will use an arbitrary default that might not match what you want. You should almost always specify something. [17:08:42 CET] <limbo_> kepstin: I don't understand what causes that loss though. [17:09:13 CET] <kepstin> a lossy codec, when encoding, throws away some of the data to make the file smaller [17:09:30 CET] <kepstin> if you decode then reencode, it will throw away more data [17:15:11 CET] <zeryx> DHE I think it was actually just something output to the error stream [17:15:14 CET] <zeryx> that wasn't actually an error [17:15:34 CET] <zeryx> kinda dumb IMO [17:16:42 CET] <zeryx> nah still errors out remotely [17:16:44 CET] <zeryx> odd, works fine locally [17:16:53 CET] <zeryx> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x4f87100] moov atom not found/tmp/vd/Hand_on_cereal_field.mp4: Invalid data found when processing input [17:17:47 CET] <kepstin> zeryx: compare your files to make sure they're actually the same? (e.g. sha1sum them both) [17:18:02 CET] <kepstin> moov atom not found is common if you have a non-streamable mp4 file that's missing the end [17:23:37 CET] <zeryx> gonna check that now, have to do it a bit roundabout since I have no direct access to the file [17:24:46 CET] <zeryx> hah [17:24:48 CET] <zeryx> the sum is different [17:24:50 CET] <zeryx> thats.. odd [17:37:27 CET] <zeryx> hah kepstin I figured it uot [17:37:56 CET] <zeryx> the download link I was being provided wasn't actually a file download, wget somehow is able to differentiate a true file from an http page describing a file [17:38:17 CET] <zeryx> but what I was using in my scala app wasn't able to do that, so it was downloading a borked file with a bunch of garbage in it [17:38:59 CET] <kepstin> zeryx: the link probably had a redirect, and your scala app was maybe using the contents of the redirect page rather than following the redirect? [17:44:00 CET] <zeryx> hmm still seems to be borked [17:44:45 CET] <zeryx> its still different sha1sums [17:45:36 CET] <zeryx> nah its that file specifically [17:45:47 CET] <zeryx> I have another file that I put in there myself in the same directory [17:45:49 CET] <zeryx> works fine [20:41:28 CET] <pomaranc> hi, does anyone know if there is hardware h264 decoding available for linux+nvidia? [20:41:50 CET] <c_14> vdpau [20:43:39 CET] <pomaranc> c_14: yes, but https://trac.ffmpeg.org/wiki/HWAccelIntro says it's deprecated [20:45:34 CET] <pomaranc> and looks like it cannot be used for transcoding [20:47:13 CET] <c_14> In that case, I don't know. [20:47:22 CET] <pomaranc> thanks anyway [20:48:37 CET] <DHE> vdpau is for playback, not decoding for software [20:52:48 CET] <DHE> also vdpau isn't deprecated [20:53:38 CET] <DHE> oh I see what you mean... [21:01:14 CET] <jkqxz> The standalone vdpau decoder is deprecated. The vdpau hwaccel decoder is not. For anything new, you should always use the latter. [21:07:59 CET] <kepstin> my impression was that you couldn't use vdpau outside of X tho, so it's not useful for "headless" transcoding. is that true? [21:09:40 CET] <DHE> that is correct. the decoded frame is only good for rendering on-screen and cannot be drawn back into RAM [21:12:23 CET] <jkqxz> Yes you can - it's not particularly fast to extract it, but it does work. (You can use "ffmpeg -hwaccel vdpau" for transcoding.) [21:13:16 CET] <jkqxz> What you can't do is anything useful with the surface itself other than send it to the screen - there is no hardware transcode path. [21:14:25 CET] <DHE> ah, okay. you have to do excessive pixel copying even to use NVENC [21:20:53 CET] <J_Darnley> Surprise! Using dedicated hardware a vast distance from the cpu and main memory is not very useful for general use. [21:39:03 CET] <lmat> What is the way to upload examples? [21:39:08 CET] <lmat> like audio files to share on irc? [21:41:52 CET] <lmat> found it! http://streams.videolan.org/upload/ [21:41:55 CET] <lmat> it wasn't ffmpeg, but videolan! [21:42:45 CET] <lmat> doh! It just ate my file :-( [22:03:15 CET] <somethingelse> Hi [22:04:20 CET] <somethingelse> I'm trying to build libav* statically via MSVC2015 on windows and link them into a program [22:04:40 CET] <somethingelse> I followed instructions and got the *.a files built (e.g. libavcodec.a) [22:06:03 CET] <somethingelse> http://stackoverflow.com/questions/2096519/from-mingw-static-library-a-to-v… said I might be able to just use the .a's as .lib's [22:06:48 CET] <somethingelse> However, while MSVC didn't complain about the .a files I added to its list of libraries, it doesn't seem to be able to find any libav* symbols, either [22:07:03 CET] <somethingelse> Anyone has experience with this and knows what's up? [22:08:50 CET] <somethingelse> (libav* was built via the MSVC toolchain) [22:14:04 CET] <somethingelse> Oh wait [22:16:57 CET] <somethingelse> Of course [22:17:19 CET] <somethingelse> libav* needs extern "C" [22:18:27 CET] <somethingelse> (It might be unfair, but many c libraries include 'extern "C"' guraded by a C++ #ifdef just to avoid those headaches) [22:18:51 CET] <somethingelse> (Also, many c libraries don't expose structs as public apis) [22:19:16 CET] Action: somethingelse fades out [22:35:13 CET] <Lucifer333> hello [22:35:42 CET] <Lucifer333> maybe its wrong channel, but i have ripped some DVD's as files on disk (yes they are playable and YES I DID BUY THEM) [22:36:12 CET] <Lucifer333> does anyone have a DVD authering tool to edit, coalesce, cut out languages, etc [22:37:24 CET] <llogan> you can use ffmpeg to do some of what I think you may want to do with your DVDs (but not DVD's) [22:37:49 CET] <llogan> as long as the resulting files have been decrypted [22:38:06 CET] <Lucifer333> yes everything is decrypted [22:38:58 CET] <furq> Lucifer333: http://dvdauthor.sourceforge.net/ [22:39:01 CET] <kepstin> the '.vob' files are just mpeg-ps streams. it's as simple as concatenating all the split 1g files from the title you want, and you can use that as input directly to ffmpeg [22:39:16 CET] <furq> assuming you actually want to write it back to a playable dvd [22:39:19 CET] <furq> otherwise you can do everything with ffmpeg [22:39:35 CET] <furq> aside from cutting out individual pgcs ;_; [22:39:42 CET] <Lucifer333> no i dont, i want to convert to mp4 to make it playable on my ipad on the go [22:39:58 CET] <furq> well then you probably shouldn't ask for a dvd authoring tool [22:40:04 CET] <Lucifer333> I payed for the movie so i want to see it on my other devices [22:40:20 CET] <Lucifer333> well the dvd files as is are weird, and split up in weird chucks [22:40:27 CET] <Lucifer333> so they need to be "cleaned up" [22:41:02 CET] <kepstin> they're split up into 1gb chunks because of file size compatibility issues, i think. but you just have to find the ones that belong to the movie and concatenate them. [22:41:24 CET] <furq> if it's a tv show, or poorly-authored, then you might need to use a pgc demuxer such as pgcdemux [22:41:25 CET] <Lucifer333> I read somewhere DVD's are mpeg2 streams [22:41:50 CET] <Lucifer333> oh, the tool needs to be on linux, I dont run anything else [22:41:57 CET] <Lucifer333> pgcdemux, lemmi see [22:42:02 CET] <furq> pgcdemux is windows only afaik [22:42:13 CET] <Lucifer333> maybe i can run it in wine [22:42:18 CET] <llogan> see tools/dvdconcat [22:42:32 CET] <furq> Lucifer333: you might not need to use it [22:42:42 CET] <llogan> *dvd2concat [22:42:51 CET] <furq> it's useful for tv shows where the episodes don't map exactly to the vobs [22:42:54 CET] <kepstin> i wonder if the 'dvdunauthor' tool (included in the dvdauthor package) can do that [22:43:11 CET] <Lucifer333> wow so many tools, thanks for the pointers, [22:43:13 CET] <furq> other than that, the only thing ffmpeg won't do is preserve chapters [22:43:17 CET] <Lucifer333> I need to do some research [22:43:49 CET] <furq> http://dvdauthor.sourceforge.net/doc/dvdunauthor.html [22:43:50 CET] <furq> what a great manpage [22:44:51 CET] <kepstin> well, it has no bugs, that's good to know. [22:48:03 CET] <Lucifer333> http://dvdauthor.sourceforge.net/doc/index.html [22:48:05 CET] <Lucifer333> main page [22:48:21 CET] <furq> those are the dvd author docs [22:48:27 CET] <furq> i'm interested in dvdunauthor [22:49:30 CET] <kepstin> dvdunauthor basically takes a dvd image and turns it into a directory structure that could be used as input to dvdauthor [22:49:48 CET] <Lucifer333> ? [22:49:48 CET] <kepstin> I forget exactly what it preserves. [22:49:59 CET] <furq> http://sprunge.us/QVMN [22:50:04 CET] <furq> someone really doesn't want me to find out what this does [22:52:00 CET] <Lucifer333> do you know the CentOS repo [22:52:05 CET] <Lucifer333> so i can install with yum? [23:07:53 CET] <wirelesslinux> Hi, [23:08:24 CET] <wirelesslinux> I'm trying to use ffmpeg with H264 nvidia support [23:09:27 CET] <wirelesslinux> but when running ffmpeg, I block on error : dl_fn->cu_init(0) - failed with error code 0x3e7 [23:10:04 CET] <llogan> ...not that i have any experience with nvenc [23:10:12 CET] <wirelesslinux> Ok, sorry, first time for me on irc :) [23:13:14 CET] <wirelesslinux> http://pastebin.com/j5Z3T0yP [23:15:38 CET] <kepstin> wirelesslinux: that "modprobe: FATAL: Module nvidia-uvm not found." sounds suspicious. do you have the binary nvidia drivers installed and loaded? [23:16:36 CET] <wirelesslinux> How do I check it ? With lsmod | grep nvidia ? [23:20:44 CET] <wirelesslinux> http://pastebin.com/19cK09hh [23:21:20 CET] <wirelesslinux> And aptitude search uvm return an empty list. [23:28:37 CET] <kepstin> hmm. dvdunauthor does /not/ split up the pgcs by chapter, but it does write timestamps for them into the dvdauthor xml file, at least. [23:34:32 CET] <furq> yeah every tool i can find for this on *nix has been abandoned since 2004 [23:35:13 CET] <kepstin> i mean, you could probably write a shell script that interprets the ifo file and uses dd to extract a particular chapter or whatever :/ [23:35:44 CET] <kepstin> I can't remember if there was any work done on an input to ffmpeg using libdvdread or libdvdnav [23:36:20 CET] <furq> yeah it is odd that ffmpeg seemingly has no concept of ifo files [23:36:23 CET] <Plorkyeran> dvd things in general have been abandoned since 2004 [23:36:23 CET] <jookiyaya> how i tried 10 different players, only one support bluray java menu properly: why is this so difficult [23:36:53 CET] <Plorkyeran> they got to a working state and then have been bitrotting since [23:38:29 CET] <furq> i should see if i can hack something together with libdvdread [23:38:49 CET] <kepstin> jookiyaya: blurays in particular are very difficult [23:39:00 CET] <kepstin> the dvd vm for navigation is a lot simpler :/ [23:39:18 CET] <jookiyaya> if bluray uses java for menu, what does dvd use [23:39:22 CET] <kepstin> (dvdunauthor actually disasembles the nav vm instructions, which is kinda neat) [23:39:29 CET] <kepstin> jookiyaya: custom very simple bytecode [23:39:44 CET] <kepstin> and some still images and mpeg2 videos [23:40:11 CET] <c_14> furq: there's the ifo_palette option [23:40:15 CET] <c_14> or whatever it was called [23:42:14 CET] <kepstin> c_14: that's used for setting the colours when remuxing dvd image subtitles, but it's kind of a sidechannel hack, and obviously doesn't let you read e.g. video chapters and whatnot :) [23:43:00 CET] Action: c_14 wasn't really paying attention and his brain came up with the first result that seemed to match [23:44:31 CET] <limbo_> kepstin: but what about the way mp3 encodes makes it degrade quality? jpg doesn't have this problem. [23:44:40 CET] <furq> yes it does [23:44:44 CET] <kepstin> limbo_: jpg has the same problem, yes... [23:45:00 CET] <limbo_> no, it doesn't. [23:45:04 CET] <furq> the example on the wikipedia page for generation loss is a jpeg [23:45:09 CET] <furq> https://en.wikipedia.org/wiki/Generation_loss [23:45:16 CET] <kepstin> (you can do some modifications to jpeg - rotations, certain crops - without decoding and reencoding, but a full reencode will lose quality) [23:45:58 CET] <kepstin> i mean, if you're only doing it once or twice, you're probably not gonna notice [23:46:11 CET] <furq> you must not have a facebook account [23:46:56 CET] <furq> otherwise you'd see plenty of text screenshots of twitter feeds that have been reencoded by every viral image host on the internet before ending up on your decrepit uncle's wall [23:49:10 CET] <growery> hi [23:49:48 CET] <growery> how can i make ffmpeg hardware accelerated on a mac mini with osx el capitain? [23:51:28 CET] <limbo_> for i in {1..504} ; do ffmpeg -i $i.jpg $((i+1)).jpg; done [23:52:00 CET] <limbo_> I just did that on an imge to re-ecode it 505 times. I'm not seeing any quality loss after number 2. [23:52:30 CET] <limbo_> how many times do I have to do it to get quality loss [23:53:38 CET] <kepstin> limbo_: you might have just gotten lucky; if you re-encode with the exact same settings, it's possible that after a few generations it'll happen to pick the same encoding and give back the same result. [23:53:42 CET] <jkqxz> When you reencode with exactly the same parameters you may well get exactly the same result - in this case you converged to a fixed point very quickly. Try alternating JPEG and something else lossy instead. [23:54:13 CET] <kepstin> alternately, something like cropping a pixel off the top each time will certainly mess it up :) [23:54:32 CET] <limbo_> I know that, because I know how jpeg compression works. [23:55:26 CET] <limbo_> can anyone explain why mp3s might have compession that stops them from being re-encoded without change losslessly? [23:55:26 CET] <kepstin> limbo_: right - so in theory, if you re-encode a file with a the same lossy codec with exactly the same settings as it originally used, and the encoder is deterministic, you *might* get back the identical result. [23:55:46 CET] <kepstin> use a different encoder? all bets are off. [23:56:09 CET] <llogan> growery: FFmpeg has "OS X VideoToolbox support". whatever that is. might be worth a look. [23:56:24 CET] <kepstin> use something like x264 in multithreaded mode? well, then it's not deterministic, so no way :) [23:56:38 CET] <growery> llogan thanks i will search for it now [23:56:47 CET] <furq> limbo_: set -q:v [23:57:13 CET] <c_14> limbo_: because if you take a lossless source and encode it lossy and then decode it again you get something that isn't equivalent to your decoded original lossless source [23:57:27 CET] <c_14> Therefore if you encode the newly decoded content it will encode to something different [23:57:58 CET] <limbo_> but what about the codec used in mp3 makes that happen? [23:58:04 CET] <furq> limbo_: http://imgur.com/a/NVVKn [23:58:05 CET] <llogan> growery: also, this may be of slight interest to you: https://trac.ffmpeg.org/wiki/HWAccelIntro [23:58:10 CET] <furq> that's what i get after 500 passes with -q:v 90 [23:59:03 CET] <llogan> why 90? IIRC, max is 31 [23:59:04 CET] <c_14> limbo_: the mp3 encoder throws away data when it encodes [23:59:40 CET] <furq> because the max isn't documented anywhere i can find and everything else i've ever used for jpeg goes from 1-100 [23:59:47 CET] <c_14> As do all lossy encoders [23:59:50 CET] <llogan> furq: heh. Karnov. [00:00:00 CET] --- Sat Mar 12 2016

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