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February 2018

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ffmpeg-devel.log.20180207
by burek 08 Feb '18

08 Feb '18

[00:14:04 CET] <cone-205> ffmpeg 03James Almer 07master:8885a29e5dc8: avformat/Makefile: fix fifo and fifo_test muxer build objects [01:20:13 CET] <Compn> wow [01:20:20 CET] <Compn> http://blog.chiariglione.org/wp-content/uploads/2018/01/HEVC.jpg [01:20:23 CET] <Compn> nightmare fuel [01:32:37 CET] <philipl> Compn: The blog post is amusing. The end of innovation! [01:34:08 CET] <Compn> philipl : its very amusing. [01:34:25 CET] <Compn> too bad no ffmpeg mentions :( [01:36:08 CET] <Compn> his graph at the end seems BS too [01:36:21 CET] <Compn> vcds were 4x smaller than mpeg2 :P [01:36:24 CET] <Compn> dvds that is [01:36:26 CET] Action: Compn runs [01:36:45 CET] <philipl> heh [01:37:59 CET] <philipl> jya: Well, I got it working after much tediousness. The problem was that the NVImage code assumes the data it's given is contiguous, which in this case it's not. The two planes are separate allocations. After fixing that assumption, it worked. But of course, it ends up being higher CPU consumption than software for vp9. Hah. [01:38:44 CET] <philipl> I suspect because it's doing the NV12->I420 conversion/copy [01:39:07 CET] <philipl> So there's a one copy penalty and then it's a non-trivial copy on top of that. [01:39:33 CET] <philipl> Might be doing two extra copies for all I know. There's a copy when creating an NVImage and then another on the way out [01:45:08 CET] <philipl> jya: could move it to doing the conversion immediately in SetData to elide a copy. Of course, at that point NVImage isn't really an NVImage, but just a regular image that knows how to convert NV12 when being initialised. [02:13:33 CET] <BBB> Compn: the blog post is about standardization and monetization thereof through the patent process [02:13:45 CET] <BBB> Compn: it doesnt really intersect with ffmpeg in any way [02:14:19 CET] <BBB> Compn: (well, and how aom broke it) [02:14:50 CET] <Compn> it dances around the issue of open source and open video coding [02:17:21 CET] <BBB> open video is aom [02:17:32 CET] <BBB> open source is orthogonal to mpegla/aom [02:18:00 CET] <BBB> the iso process is open source, and mpegla doesnt care as long as you pay them somehow [02:18:13 CET] <Compn> fair enough [02:18:30 CET] <BBB> to say that ffmpeg has to be mentioned is kind of like insisting that shell is mentioned when we talk about oil [02:18:35 CET] <BBB> :-p [02:20:47 CET] <Compn> i like it when ffmpeg is mentioned :P [02:22:07 CET] <BBB> me too ;) [02:22:36 CET] <BBB> Ill write a blog post sometime soon and be sure to mention ffmpeg [02:22:40 CET] <c3r1c3-Win> Well it doesn't hurt to mention Shell... or Carl's Jr. [02:23:15 CET] <Compn> cia using ffmpeg in secret spy tool heh [02:23:47 CET] Action: Compn reading old news [02:26:31 CET] <BBB> I guess everyone uses ffmpeg somehow? [02:26:35 CET] <BBB> its quite widespread nowadays [02:31:39 CET] <Chloe> Pretty sure everyone uses ffmpeg [02:32:20 CET] <Chloe> If they use chrome, Firefox, Kodi or VLC (which I would assume covers the majority) [02:40:07 CET] <klaxa> even if not directly then indirectly [02:54:19 CET] <philipl> jya: whelp. I got rid of the extra copy and still clearly using more CPU than software :-) [02:54:33 CET] <philipl> Of course, the pixel format conversion is still happening [02:59:30 CET] <SortaCore> that yuvj stuff is tripping me up [02:59:48 CET] <SortaCore> is it equivlaent to setting color_range to jpeg and switching to yuv equivalent? [03:00:26 CET] <SortaCore> out of all the codecs only libx264 is accepting yuvj420p [03:00:42 CET] <SortaCore> (h264_qsv, h264_nvenc, libopenh264 all reject) [03:12:18 CET] <cone-205> ffmpeg 03James Almer 07master:f7aacf4ab765: avformat/mov: add VP8 codec support [03:44:01 CET] <cone-205> ffmpeg 03James Almer 07master:36c85d6e773d: api: add missing version bumps and APIChanges entries [03:49:31 CET] <cone-205> ffmpeg 03James Almer 07master:e8323c42c19f: doc/APIChanges: fix lavu version numbers in a few recent entries [04:31:26 CET] <cone-205> ffmpeg 03James Almer 07master:8318bf175135: avformat/fifo_test: remove network.h include [07:32:12 CET] <rcombs> tmm1: *poke* [08:57:25 CET] <jya> philipl: our shaders still need rgb, so there's a conversion happening.. [08:57:51 CET] <jya> but I had similar result in the past (though with vaapi), I only spent a couple of hours on it and didn't investigate much [08:58:14 CET] <jya> without opengl being the default rendering engine on linux, there wasn't much point anyway... [08:58:25 CET] <jya> linux uses the basic rgb compositor, it's awful [09:45:30 CET] <slasktrat> Hi. I've updated an one year old, unclosed ticket (http://trac.ffmpeg.org/ticket/6165) do I need to send a notification somewhere? [09:46:12 CET] <slasktrat> Uploaded sample and should make it very easy to reproduce [09:57:19 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #104: [WIP] Metadata (06master...06Metadata) 02https://git.io/vAfOw [10:25:50 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #105: JPEG2000-RCT exception clarification (06master...06Background) 02https://git.io/vAfsG [11:00:28 CET] <Chloe> wm4: any idea how I should conditionally include fifo_test? I think marton suggested a configure flag but wouldnt this then require a custom flag just to test fate [11:18:05 CET] <Chloe> Also I guess I should be deprecating .next within the components after the lavfi patch [11:38:58 CET] <wm4> Chloe: hm no idea, that's nasty [11:40:16 CET] <jkqxz> The next links aren't user-visible, just delete them immediately. There's no reason for them to continue to exist. [11:47:21 CET] <atomnuker> jkqxz: can you check out http://ffmpeg.org/pipermail/ffmpeg-devel/2018-February/224942.html [11:51:28 CET] <jkqxz> I replied to that five minutes ago. [11:51:34 CET] <jkqxz> How did you even generate the files there? [11:53:01 CET] <nevcairiel> maybe he did an in-tree build [11:58:16 CET] <jkqxz> Oh, huh. That is possible, isn't it. [12:03:55 CET] <Chloe> jkqxz: ok, well I cant remove them until FF_API_NEXT is removed, cause the compat code needs it still [12:05:05 CET] <jkqxz> Isn't it easily rewritten to not use them? [12:05:26 CET] <jkqxz> Still, even if they stay you don't need to do anything now because they aren't user-visible. [12:26:21 CET] <cone-875> ffmpeg 03Rostislav Pehlivanov 07master:33d632d40e14: lavfi: add a gitignore file for OpenCL compiled kernels [12:33:07 CET] <wm4> libswresample/soxr_resample.c:46: undefined reference to `soxr_io_spec' [12:33:15 CET] <wm4> why does linking so suxsr fail now [12:35:41 CET] <wm4> and only when running fate [12:35:57 CET] <wm4> when building tests/checkasm/checkasm [12:36:01 CET] Action: wm4 just disable suxsr [12:42:53 CET] <cone-875> ffmpeg 03Richard Shaffer 07master:651d5f963921: avformat/hls: Support metadata updates from subdemuxers [13:43:19 CET] <Chloe> is anyone able to look at/push '[PATCH] cmdutils: fix finding next codec for id'? Currently `-codecs` is broken in master. [13:47:20 CET] <JEEB> I will take a look at if if no-one else does when I finish work [17:12:57 CET] <philipl> jya: ah well [17:13:06 CET] <philipl> BtbN: so yeah, it really doesn't help today. [17:15:07 CET] <jya> philipl: have you looked at a profile to see where the time is spent? [17:15:27 CET] <jya> firefox has a built-in profiler now.. that will help [17:16:29 CET] <jya> https://developer.mozilla.org/en-US/docs/Mozilla/Performance/Profiling_with… [17:16:52 CET] <jya> it's quite easy, there's an extension you need to install, and from there it's pretty much self-explanatory [17:16:54 CET] <philipl> jya: I haven't, but given how isolated the change is, there's only things going on that weren't there before - the copy back to system memory and then the NV12->I420 conversion. [17:17:03 CET] <jya> if you start with e10s disabled, that helps too [17:17:12 CET] <jya> cause profiling across multiprocess is a pain [17:17:53 CET] <jya> nv12->i420 shouldn't take much time... there's a libyuv function to do that that is SSE accelerated [17:18:01 CET] <jya> oh, did you enable GL rendering? [17:18:06 CET] <philipl> I didn't. [17:18:27 CET] <JEEB> also there's libp2p which I think had some SIMD as well for various packing types [17:18:30 CET] <jya> you need to, otherwise there's so much time spent in the compositor it's hard for anything else to be of relevance [17:18:35 CET] <philipl> ah [17:18:42 CET] <JEEB> https://github.com/sekrit-twc/libp2p [17:18:50 CET] <nevcairiel> If you wanted to truely optimize things, you could do the conversion during the readback, that comes practically for free then, but of course you need custom readback then =p [17:18:59 CET] <jya> JEEB: libyuv is in firefox tree, trying to keep the changes small [17:18:59 CET] <JEEB> :) [17:19:03 CET] <JEEB> jya: right [17:19:05 CET] <philipl> JEEB: the thing firefox uses has avx2. Seems up to date. [17:19:10 CET] <JEEB> alright [17:19:40 CET] <philipl> nevcairiel: Yeah. I elided one copy that way but that was on the system side already [17:19:54 CET] <philipl> The original code path was read-back -> copy -> copy -> convert -> copy [17:19:59 CET] <jya> philipl: in about:config , set layers.acceleration.force-enabled to true [17:20:00 CET] <jya> restart [17:20:28 CET] <jya> philipl: indeed, we already copy the content coming out from ffmpeg [17:20:48 CET] <nevcairiel> that seems something worthy of getting rid of [17:20:53 CET] <jya> with out of process compositing, it becomes almost impossible to refcount those memory blob [17:21:20 CET] <atomnuker> JEEB: stop trying to flog that thing, its useless [17:21:34 CET] <JEEB> atomnuker: ok, if it's useless then it's useless [17:21:55 CET] <jya> on windows and mac we do reduce the amount of copy, what we do is directly upload the yuv buffer into either an IOSurface (mac) or a IMF image (windows) [17:21:58 CET] <JEEB> it seemed to have some SIMD and since zimg wasn't completely useless I thought that would be similar :P [17:22:27 CET] <atomnuker> well it turns out we have some simd too [17:22:39 CET] <JEEB> yea but then why does nevcairiel write his own SIMD? [17:22:46 CET] <JEEB> (Ž4@) [17:22:54 CET] <nevcairiel> because sws is the spawn of the devil [17:22:59 CET] <atomnuker> and I'll believe something is faster than c:v v210 when I see it [17:23:15 CET] <JEEB> yea but we're dealing with NV12 here [17:23:17 CET] <JEEB> not V210 [17:23:20 CET] <nevcairiel> noone wants v210 anyway [17:23:30 CET] <JEEB> other than broadcasters [17:23:45 CET] <JEEB> hw packing formats ahoy [17:40:30 CET] <philipl> jya: even with GL compositing turned on, it's still easy to see CPU is lower with software. I also can't get the profiler to show any of the ffmpeg/nvimage calls [17:42:35 CET] <atomnuker> philipl: did firefox really do that many copies per frame? [17:43:00 CET] <philipl> atomnuker: It does a lot. [17:44:52 CET] <philipl> AVFrame -> copied to Image -> converted to I420 -> converted to RGB and then who knows how many times after that [17:46:40 CET] <philipl> I combined the first copy and the I420 conversion but CPU is stil noticeably higher than software [18:42:26 CET] <jya> philipl: shouldn't be more than 2 copies. [18:42:46 CET] <jya> one copy from FFmpeg AVFrame to create an Image [18:42:58 CET] <jya> and the copy from Image to the OpenGL surface [19:40:57 CET] <feliwir> i just got a warning: "avcodec_copy_context is obsolet" [19:41:08 CET] <feliwir> do i not need to copy the codec_ctx anymore? [19:42:03 CET] <nevcairiel> there is no good reason to ever copy a context [19:44:00 CET] <feliwir> nevcairiel, http://dranger.com/ffmpeg/tutorial01.html [19:44:08 CET] <feliwir> "Note that we must not use the AVCodecContext from the video stream directly! So we have to use avcodec_copy_context() to copy the context to a new location (after allocating memory for it, of course). " [19:44:21 CET] <nevcairiel> those guides are probably a decade old by now, if not more [19:44:29 CET] <feliwir> updated in 2015 [19:44:33 CET] <feliwir> it says that on the first page [19:44:53 CET] <feliwir> so i can use the original context without worries for decoding? [19:44:59 CET] <nevcairiel> no. [19:45:07 CET] <feliwir> uhm? [19:45:25 CET] <nevcairiel> there is no original context anymore (well there is, but its deprecated) [19:45:30 CET] <feliwir> didn't you just say that there isn't a good reason to copy a context [19:45:40 CET] <nevcairiel> there is more options then copying a context [19:45:44 CET] <nevcairiel> you could create a new one =p [19:45:56 CET] <nevcairiel> you should read the official examples [19:46:23 CET] <wm4> avcodec_copy_context() should not be used [19:46:26 CET] <wm4> it's pretty broken [19:46:57 CET] <feliwir> nevcairiel, where can i find the official "small examples". And please don't link me to ffplay :D [19:47:13 CET] <wm4> ah I see it's properly deprecated [19:47:20 CET] <wm4> so at least that warns the user [19:47:31 CET] <wm4> feliwir: there are none, all examples are either non-official, or broken, or both [19:47:42 CET] <feliwir> https://github.com/feliwir/arda/blob/master/src/video/video.cpp#L145 i guess those parts are pretty wrong then? It always used to work, but i want to use the correct way [19:47:51 CET] <feliwir> wm4, guessed so tbh [19:48:01 CET] <nevcairiel> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/demuxing_decodi… should be mostly ok [19:48:35 CET] <nevcairiel> well it still uses the old decode api, but at least not the old context =p [19:49:11 CET] <feliwir> i already use the new decode api in the link above [19:49:21 CET] <feliwir> i guess i must just use avcodec_parameters_to_context instead of context copy [19:51:34 CET] <feliwir> lol, the parameter order was changed for avcodec_parameters_to_context [19:51:48 CET] <feliwir> the vlc example has it the wrong way round [19:52:29 CET] <wm4> I doubt that ever happened [19:55:19 CET] <feliwir> that's my signature: "int avcodec_parameters_from_context(AVCodecParameters* par, AVCodecContext* codec)" [19:55:38 CET] <feliwir> this is the VLC usage: "avcodec_parameters_to_context(*dec_ctx, st->codecpar))" [19:55:41 CET] <nevcairiel> thats from, not to [19:55:52 CET] <nevcairiel> it always goes (destination, source) [19:55:56 CET] <feliwir> what a bummer -.- [19:56:18 CET] <feliwir> my bad :D [20:14:42 CET] <uartie> av_register_all just got deprecated... is there an example on how to change user code to new api? [20:15:21 CET] <uartie> documentation seems to still use av_register_all [20:17:24 CET] <jkqxz> Just don't call it. [20:17:44 CET] <uartie> thanks [20:42:27 CET] <Chloe> Guess I should update the docs too huh. [20:43:07 CET] <cone-875> ffmpeg 03James Almer 07master:cf666651b499: avformat: fix stream_segment muxer build dependencies [20:45:03 CET] <jkqxz> I just sent a patch to remove the (non-lavfi) register functions from the examples. [20:45:18 CET] <Chloe> jkqxz: can you take a look at `cmdutils: fix finding next codec for id`? [20:45:29 CET] <jkqxz> Sure. [20:46:32 CET] <Chloe> `-codecs` in ffmpeg is currently broken, this is a fix for that (most of the cmdutils code should probably be rewritten, but I'd rather put out a quick fix for something which is actually broken than leave it too long) [20:46:48 CET] <jkqxz> Right, it currently ends up in a loop. [20:46:52 CET] <Chloe> yes [20:47:31 CET] <Chloe> if you could push it too (if you think it's ok), that'd be nice. [20:50:06 CET] <jkqxz> It doesn't work. [20:50:16 CET] <jkqxz> That strcmp() isn't doing the right thing. [20:50:47 CET] <Chloe> it's not? [20:50:48 CET] <jkqxz> I think you need to check for the thing after prev in iteration order? That is, find prev, then find the next one. [20:51:50 CET] <jkqxz> " DEV.LS h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (decoders: h264 h264_v4l2m2m ) (encoders: libx264 libx264rgb nvenc nvenc_h264 )" [20:51:55 CET] <jkqxz> I have more H.264 encoders than that. [20:52:34 CET] <jkqxz> The list is not in strcmp() order, so once you've found one you only look for ones after it in the alphabet. [20:53:15 CET] <jkqxz> Hence you find libx264, then skip over all the h264_foo (amf, vaapi, v4l2m2m) because they're < "l", then find the deprecated nvenc names because they are > "l". [20:58:48 CET] <cone-875> ffmpeg 03James Almer 07master:9d8fb095e6f8: doc/APIchanges: fix a recent depreacted function name [21:06:42 CET] <cone-875> ffmpeg 03James Almer 07master:3f0a41367eb9: doc/APIchanges: mention a few more recently introduced and deprecated functions [21:31:00 CET] <rcombs> tmm1: ping [21:59:22 CET] <Chloe> jkqxz: https://0x0.st/scjy.patch how does this look [22:08:45 CET] <Chloe> mmh, i dont know how reply-to works in my client, I set the header but it didnt seem to work [00:00:00 CET] --- Thu Feb 8 2018

1 0

ffmpeg.log.20180207
by burek 08 Feb '18

08 Feb '18

[02:16:35 CET] <Uuperich> hi, I got a quick question [02:17:08 CET] <lightbulb6> hello, how do you pad with transparency using ffmpeg? i tried `ffmpeg -i foo.png -filter pad=iw+4:ih+4:2:2:black@0 test.png` but the resulting padding is black instead of fully transparent [02:17:47 CET] <Uuperich> I'm converting a bunch of webm to .mp3 files on a debian server i have lying around here. For some reason the output has varying speedups? [02:18:09 CET] <Uuperich> ranging from 14.x times to 19.x times [02:18:26 CET] <lightbulb6> Uuperich: it's the encoding speed [02:18:32 CET] <lightbulb6> it's normal for it to fluctuate [02:18:40 CET] <lightbulb6> depending on your server's load ad the moment for example [02:19:03 CET] <Uuperich> but the output mp3 also are garbled [02:19:12 CET] <DHE> Uuperich: well what's your command? [02:19:16 CET] <lightbulb6> post the command line you use [02:19:23 CET] <Uuperich> for FILE in raws/*.webm; do echo -e "Processing video '\e[32m$FILE\e[0m'"; ffmpeg -i "${FILE}" -y "${FILE%.webm}.mp3"; done; [02:19:47 CET] <Uuperich> is what i'm using right now, before it also specified bitrate and khz lemme look for it [02:19:57 CET] <lightbulb6> hmm, are you using the latest git build of ffmpeg? [02:20:02 CET] <Uuperich> funny thing is that next one worked fine yesterday when I tried it via ssh from work [02:20:26 CET] <Uuperich> for FILE in raws/*.webm; do echo -e "Processing video '\e[32m$FILE\e[0m'"; ffmpeg -i "${FILE}" -vn -ab 128k -ar 44100 -y "${/audio/FILE%.webm}.mp3"; done; [02:20:34 CET] <Uuperich> that worked yesterday from my work machine [02:20:38 CET] <Uuperich> the build might be a bit older [02:20:52 CET] <lightbulb6> if it's from debian repos it's likely very old [02:20:59 CET] <Uuperich> the server is only a hobby so it idles around for months at a time [02:21:14 CET] <Uuperich> ok lemme try updating it [02:21:43 CET] <lightbulb6> try these builds for hassle-free testing https://johnvansickle.com/ffmpeg/ [02:22:21 CET] <lightbulb6> sadly these don't work anymore on my debian vps because the kernel is too old, you may have better luck [02:25:35 CET] <lightbulb6> also if possible, you could upload somewhere the files which ffmpeg failed to encode (decode?) properly, so others can have a look and maybe spot something useful [02:25:41 CET] <Uuperich> ok thanks for the quick advice, i'll try from work once more tomorrow since it really confuses me how that worked fine yesterday [02:25:56 CET] <Uuperich> actually they're already available at audio.animu.date [02:26:30 CET] <Uuperich> the script fetches webm from a webserver and converts it to mp3 locally (since I cant execute scripts on my webhoster), then uploads them again [02:26:42 CET] <DHE> lightbulb6: in theory you can specify an alpha channel in your colour selection, but you'll need to use a pix_fmt that has alpha support as well depending on what you're up to [02:26:46 CET] <Uuperich> and a still very scarce website at least shows all the mp3s [02:27:23 CET] <Uuperich> you're supposed to hear some samples of japanese [02:27:50 CET] <Uuperich> what actually comes out sounds like small samples brutally sped up [02:27:53 CET] <lightbulb6> if it's for the web, you could use opus instead of mp3, as it's royalty free and higher quality than mp3, allows you to save on bandidth for example [02:28:15 CET] <DHE> obvious question. if you output to .wav instead (just try one or two sample videos) does it sound okay? [02:28:36 CET] <Uuperich> will try that once my server is done updating, gimme a sec [02:30:16 CET] <lightbulb6> DHE: great, `format=rgba,pad=iw+4:ih+4:2:2:black@0` did the trick, thanks! [02:31:02 CET] <lightbulb6> i've already tried the filter before, but silly me put it after the pad filter rather than before it :P [02:31:17 CET] <lightbulb6> and all i got was black padding around the image [02:32:25 CET] <DHE> I thought you might have to do 0x00000000 (RRGGBBAA) or such [02:33:12 CET] <lightbulb6> the "@0" in "black@0" means fully transparent black, it's equivalent of 0x00000000 [02:33:36 CET] <lightbulb6> see https://ffmpeg.org/ffmpeg-utils.html#Color for details [02:34:37 CET] <Uuperich> outputting in .wav fixes the speed issue but introduces a shitload of noice [02:34:41 CET] <Uuperich> *noise [02:35:54 CET] <Swervz> Hey would anyone know how to join thousands of fragments downloaded from a streaming site into a video? I found the wiki on Concatenate but that looks like it only works for videos inside a container already. Also found some video but it was not in English so I couldn't understand it [02:37:11 CET] <klaxa> hmm what's a "fragment" then? [02:37:30 CET] <klaxa> if ffmpeg can read it, it can concatenate it [02:37:35 CET] <klaxa> probably [02:38:11 CET] <lightbulb6> if it's mpeg dash or something like it, you can just concatenate the files directly [02:38:22 CET] <Swervz> No idea what format its in, I tried using media info on it but I just get the "floating point division by zero" error [02:38:36 CET] <Swervz> It's a HTTP stream it seems, I guess dash? [02:38:37 CET] <lightbulb6> you may need an init chunk [02:38:44 CET] <klaxa> what does ffprobe or ffmpeg -i say? [02:40:08 CET] <Uuperich> short update: neither .wav nor .opus give any improvement [02:40:09 CET] <Swervz> let me pastebin the output [02:41:12 CET] <lightbulb6> try searching for the smallest file in the directory, that would probably be the init/header file, and to get a playable movie out of it you could do `cat init.m4s chk1.m4s chk2.m4s chkn.m4s > movie.m4s` [02:41:50 CET] <lightbulb6> it works with webm dash streams, so it should work with mpeg dash as well [02:42:30 CET] <lightbulb6> Uuperich: so does the issue occur with all the webms or some select ones? [02:42:59 CET] <Uuperich> every single one of them [02:43:19 CET] <Uuperich> it's 33 little clips, recorded via mpv [02:43:35 CET] <Uuperich> the weird thing is it worked fine yesterday when I did it manually [02:44:45 CET] <Uuperich> let me try it manually one more time just to be sure it works in theory [02:45:46 CET] <lightbulb6> Uuperich: none of the mp3s on that site you linked to work for me, as in they won't even play [02:46:36 CET] <Swervz> Wtf I was just playing the video in my brower and it kept pausing every second or so and the audio was glitching then my screen went black a couple times and blue with like static [02:47:23 CET] <Uuperich> lightbulb6, exactly, and the files do the same locally [02:47:58 CET] <Uuperich> lightbulb6, try it again now [02:48:11 CET] <Uuperich> when I do it manually it apparently works fine [02:48:40 CET] <lightbulb6> yeah, works now [02:49:12 CET] <Uuperich> any bash experts? :> [02:49:39 CET] <lightbulb6> and what if you make it a simplistic python script rather than bash? [02:49:42 CET] <lightbulb6> may just work [02:50:06 CET] <klaxa> well pastebin the script [02:50:38 CET] <Uuperich> https://pastebin.com/anNBhV2X [02:50:53 CET] <Uuperich> I might just opt for python just to try it [02:51:18 CET] <Uuperich> I'm not that versed with linux yet but this is puzzling the heck out of me [02:52:30 CET] <lightbulb6> have you confirmed the wget part correctly downloads the webms? [02:52:31 CET] <klaxa> huh that doesn't look wrong at all [02:52:44 CET] <klaxa> if it works when he runs it manually i would think the files are ok [02:52:56 CET] <klaxa> can you try removing the echo with the ansi color codes? [02:53:17 CET] <klaxa> even though that should not do anything at all [02:54:47 CET] <Uuperich> well let's try [02:55:52 CET] <Uuperich> upload done, some filesizes changed [02:56:12 CET] <Uuperich> ...and chrome already barfs at the files [02:58:12 CET] <Uuperich> i really need some sleep so I'll try around with another computer tomorrow, that used "ideapad" which is my current scriptserver does some weird things [02:58:40 CET] <Uuperich> if I remember I'll report back, thanks so far for confirming I'm not a total dunce or insance [02:58:43 CET] <Uuperich> *insane [03:21:11 CET] <Swervz> Sorry I had to reboot [03:26:52 CET] <Swervz> klaxa, Here is the output of ffprobe on one of the files https://pastebin.com/yj5YXxTp [03:27:13 CET] <Swervz> Oh it didnt preserve the line endings :| [03:27:45 CET] <Swervz> should be fixed now [03:28:12 CET] <klaxa> have you tried this: <lightbulb6> try searching for the smallest file in the directory, that would probably be the init/header file, and to get a playable movie out of it you could do `cat init.m4s chk1.m4s chk2.m4s chkn.m4s > movie.m4s` ? [03:29:24 CET] <Swervz> I'm not sure there is one [03:30:13 CET] <klaxa> hmm... no idea then sorry :/ [03:30:16 CET] <Swervz> Could it be a f4m file maybe [03:30:27 CET] <Swervz> I think that got filtered out originally when I was downloading [03:31:05 CET] <Swervz> Oh yeah that's it. In XML format it seems [03:34:19 CET] <Swervz> How would I concat them on windows? The header is .f4m and the fragments have no file extension [03:37:09 CET] <lightbulb6> the .f4m is an xml file? [03:37:31 CET] <lightbulb6> the header should be binary as far as i know... [03:38:05 CET] <lightbulb6> hls/dash uses an xml manifest, usually with the extension ".mpd" [03:38:52 CET] <lightbulb6> this file usually describes the available fragment so the player knows which fragments it can download at which time [03:41:04 CET] <Swervz> Apprently its called Abode Dynamic Streaming for Flash [03:41:35 CET] <Swervz> metadata is in base64 [03:43:03 CET] <lightbulb6> sorry, never heard of it [03:43:16 CET] <lightbulb6> can you upload that f4m file somewhere? [03:44:41 CET] <Swervz> Lets hope it doesnt have any personal info https://pastebin.com/KZ1p6Prk [03:45:18 CET] <Swervz> I think I may have found a php script to do this but I think it still uses ffmpeg [03:47:24 CET] <Swervz> Oh it can have embedded auth info so I deleted the paste because it was public and not unlisted [03:48:01 CET] <Swervz> I forgot they shown up on the front page of pastebin for anyone to see [03:48:40 CET] <lightbulb6> anyways i don't think it's going to be easy [03:48:51 CET] <lightbulb6> have fun with their drm and encryption stuff [03:49:08 CET] <lightbulb6> what tool did you use to download the files in the first place? [03:51:39 CET] <Swervz> I got the URL from fiddler then just wrote a script to download each one [03:52:18 CET] <Swervz> they're numbered like _Seg1-Frag1 so I just had to find the last fragment and then say download 1-last [03:54:09 CET] <lightbulb6> hmm, there's that bootstrapInfo section [03:54:28 CET] <lightbulb6> it may contain the header neccesary to decode the following chunks [03:54:56 CET] <lightbulb6> also the drmAdditionalHeader [03:55:06 CET] <Swervz> I hate adobe drm [03:55:43 CET] <lightbulb6> i'm just guessing at this point, but you could try to base64 decode these two strings, then concatenate the files (binary) [03:56:02 CET] <lightbulb6> and then further concatenate this header with the fragments [03:56:17 CET] <Swervz> I tried decoding them already but I just got a malformed input [03:56:30 CET] <Swervz> I think I need a drm key [03:56:57 CET] <Swervz> er decryption key [03:57:05 CET] <Swervz> probably more trouble that its worth at this point [03:58:32 CET] <Swervz> "This stream is encrypted with FlashAccess DRM. Decryption of such streams isn't currently possible with this script." [03:58:40 CET] <Swervz> Guess I'm SOL then [03:58:54 CET] <lightbulb6> it's from that php script? [03:59:18 CET] <Swervz> yeah [04:00:19 CET] <lightbulb6> well then [04:00:48 CET] <lightbulb6> so you were right about that SOL thing [04:01:19 CET] <Swervz> hmm well thankyou anyway I guess I'll have to find some other way to get the content [04:03:12 CET] <Swervz> I just wanted to watch a TV show on a phone with datacap when out but it didn't support their player so I was trying to download it from their site, guess they forced people into getting the content by other means even when they would willingly pay [04:04:01 CET] <lightbulb6> yeah, drm is anti-consumer [04:04:11 CET] <lightbulb6> the bad guys usually find ways around anyways [04:04:38 CET] <prelude2004c> hey everyone.. looking for some help [04:04:54 CET] <prelude2004c> ${ffmpeg} -hwaccel cuvid -c:v mpeg2_cuvid -deint 2 -resize 1280x720 -hwaccel_device $gpuenc -max_alloc 100000000 -threads 1 -vsync 0 -fflags +discardcorrupt -i "$stream" .. it wont deinterlace for some reason [11:59:37 CET] <OnceMe> is it possible to convert mkv to wmv without losing quality or minimal quality loss? [11:59:51 CET] <OnceMe> I need wmv for my powerpoint presentation, and pp does not accept any other format then wmv [12:01:42 CET] <BtbN> no [12:02:07 CET] <OnceMe> why not? :( [12:02:07 CET] <JEEB> well for minimal quality loss you try setting the quantizer quite low - you will make the file quite a bit larger most likely, but it should maybe play? [12:02:11 CET] <BtbN> wmv is terrible and does not accept any codecs that are usually in mkv [12:02:22 CET] <BtbN> And to my knowledge all the wmv-video-encoders are terrible [12:02:32 CET] <OnceMe> how cloudconverter converts it from mkv to wmv? [12:02:38 CET] <OnceMe> should I convert it from mkv to avi and then to wmv? [12:02:39 CET] <BtbN> they re-encode it. [12:02:45 CET] <JEEB> stream copying will most likely not work, so without loss of quality isn't going to happen [12:02:51 CET] <JEEB> which is what BtbN is noting [12:03:13 CET] <OnceMe> can I re-encode it with ffmpeg? [12:03:18 CET] <OnceMe> I tried setting -q 8 [12:03:18 CET] <JEEB> yes [12:03:22 CET] <BtbN> unless you use an insanely high bitrate you will get a very noticeable quality hit [12:03:24 CET] <OnceMe> but quallity loss is still shitty [12:03:25 CET] <furq> ffmpeg doesn't have a vc-1 encoder iirc [12:03:32 CET] <furq> which is your best bet for a codec that isn't garbage [12:03:34 CET] <JEEB> it does have wmv3 tho I think? [12:03:42 CET] <BtbN> it has some wmv encoder, but it's bad [12:03:48 CET] <furq> it has wmv1 and wmv2 [12:03:51 CET] <JEEB> oh, nope [12:03:52 CET] <JEEB> wmv2 [12:03:53 CET] <JEEB> yea [12:04:16 CET] <JEEB> OnceMe: you just need to know the quantizer range for a given format :P [12:04:20 CET] <OnceMe> so ffmpeg -i msc.mkv -q 8 -s 640x480 -vcodec wmv2 myvideo.wmv ? [12:04:20 CET] <JEEB> s/format/encoder/ [12:04:37 CET] <JEEB> something like that, yes, if you want to scale too :P [12:04:45 CET] <BtbN> I'm pretty sure recent versions of office support .mp4 with h264 and aac just fine. [12:05:01 CET] <OnceMe> I used SimpleScreenRecorder to record my screen and used H.264 codec [12:05:07 CET] <BtbN> which you should be able to stream-copy from your mkv [12:05:18 CET] <furq> https://support.office.com/en-ie/article/Video-and-audio-file-formats-suppo… [12:05:21 CET] <furq> yeah [12:05:24 CET] <furq> PowerPoint 2013, PowerPoint 2016, PowerPoint 2016 for Mac: .mp4 files encoded with H.264 video and AAC audio [12:05:48 CET] <OnceMe> what about windows? [12:05:53 CET] <furq> that is for windows [12:06:02 CET] <OnceMe> so PP 2013 should work with mkv? [12:06:07 CET] <furq> no [12:06:07 CET] <BtbN> mp4... [12:06:17 CET] <furq> it should work with the streams from the mkv remuxed into mp4 [12:06:19 CET] <OnceMe> ok so converting to mp4 from mkv is easier? [12:06:21 CET] <OnceMe> :D [12:06:32 CET] <furq> -i foo.mkv -c copy bar.mp4 [12:06:57 CET] <furq> or -i foo.mkv -c:v copy -c:a aac bar.mp4 [12:07:01 CET] <furq> if your audio isn't aac already [12:07:46 CET] <OnceMe> ok [12:07:51 CET] <OnceMe> but what about mkv to wmv? [12:07:55 CET] <OnceMe> so ffmpeg -i msc.mkv -q 8 -s 640x480 -vcodec wmv2 myvideo.wmv ? [12:07:57 CET] <OnceMe> didnt worked well [12:08:01 CET] <OnceMe> still have shitty quallity [12:08:01 CET] <furq> shrug [12:08:08 CET] <furq> the wmv encoder in ffmpeg is probably garbage [12:08:11 CET] <furq> and there's no vc-1 encoder [12:08:23 CET] <furq> i would probably try to find something that'll transcode it to vc- [12:08:23 CET] <furq> 1 [12:08:31 CET] <OnceMe> how? [12:09:24 CET] <BtbN> https://www.google.com/search?q=vc-1+encoder like that [12:12:29 CET] <JEEB> OnceMe: as I noted check the darn range of values of quantizer :P [12:12:55 CET] <OnceMe> ? [12:12:58 CET] <JEEB> wmv3/vc-1 is better of course but we don't have it because by the time it came around nobody cared [12:13:51 CET] <furq> i would just forget about supporting office 2010 [12:14:06 CET] <JEEB> OnceMe: as in, try going up/down with the value :P [12:19:27 CET] <OnceMe> -q 30? [12:21:41 CET] <OnceMe> I mean -q 1 [13:04:05 CET] <OnceMe> powerpoint 2013 does not play video in full screen [13:04:14 CET] <OnceMe> I mean its "full screen" but grayed out box around [13:04:34 CET] <iranen> https://fosdem.org/2018/schedule/event/rust_av/ [15:58:30 CET] <TAFB_WORK> I use this to stream to youtube live but it doesn't work anymore, youtube live dashboard says "Your encoder is sending data faster than realtime (multipleseconds of video each second). You must rate limit your livevideo upload to approximately 1 second of video each second." please help: https://pastebin.com/7X2tAApC [15:59:49 CET] <sfan5> is there a reason you're not using a single ffmpeg command? [16:00:03 CET] <TAFB_WORK> the 4k ip camera stream sudders a lot if I do [16:02:55 CET] <sfan5> does adding -re to the second ffmpeg command work? [16:03:08 CET] <TAFB_WORK> let me try :) [16:05:20 CET] <TAFB_WORK> nope, same error on youtube [16:05:44 CET] <TAFB_WORK> stupid ip camera has a veriable framerate, 27fps to 30fps or so, hard to get it nice. [16:09:24 CET] <kepstin> youtube's happier with a constant framerate, so you should use an fps filter (or -r output option) to correct that [16:09:29 CET] <kepstin> and you should drop the -r input option [16:13:55 CET] <TAFB_WORK> -r is already on the output [16:14:01 CET] <TAFB_WORK> oh [16:14:03 CET] <TAFB_WORK> i see what you mean [17:18:55 CET] <bledo> Hi, anyone knows who to set pixel format, size and framerate of rawvideo inside a mpegts format over udp? [17:19:48 CET] <bledo> i'm trying to open a multicast stream whit encoded audio and raw video [17:26:42 CET] <DHE> mpegts doesn't have a codec code for rawvideo. you can encode it, but pulling it back out will be tricky [17:52:27 CET] <dkc> hello folks, I'm writting RTP demuxers for video and audio streams. For these demuxers, PTS are supposed to be inferred from RTP timestamps. First question: is that correct that PTS for each stream is expressed in AVStream->time_base units. Second question: these PTS start at epoch, what do you I need to set to let FFmpeg know that and sync audio/video properly? [18:11:19 CET] <JEEB> for A/V sync all you need to do is to make sure that the timestamps are relevant to each other (as in, if you apply both streams' relevant time base the AVPackets of things that happen during the same time should be during the same time) . and yes, AVPackets come out of a demuxer with the AVStream's time_base [18:11:25 CET] <JEEB> which is something that a demuxer will set, yes [18:12:16 CET] <JEEB> timestamps do not have to start from zero so the demuxer doesn't have to do anything for that [18:12:42 CET] <JEEB> you just have to make sure that the timestamps on the streams' relevant time_bases make sensew [18:15:48 CET] <dkc> okay, but if one have a timebase of 1/90000 and the other of 1/48000, to make them relevant to one another, you need to specify when is time zero, where they both started to count, right? or am I missing something [18:19:46 CET] <JEEB> it just means that a packet that's supposed to be at point 1s in your source has 48000 in audio, and 90000 in video [18:20:07 CET] <JEEB> demuxer's job is to make sure the timestamps are read correctly and are in sync between streams [18:20:28 CET] <JEEB> (of course if the input timestamps are bonkers you can't do too much about that :P) [18:25:21 CET] <dkc> To give you a bit of context, on both streams RTP timestamp are derived from their own media clock, so RTP timestamps for two audio packets 1s apart will have a difference a 48000. From that I can easily compute the absolute PTS since epoch but as you said, that makes video and audio timestamps look totaly unrelated [18:37:14 CET] <dkc> so I've been using the RTP/SDP AVInputFormat, but should I derived my own input format to tweak PTS according to what I want to do? [19:04:21 CET] <saml> good morning [19:04:38 CET] <saml> ffmpeg -i [input] -r 24 -filter:v "setpts=0.999*PTS" -y [output] vs ffmpeg -i [input] -r 24 -y [output] [19:04:41 CET] <saml> what's better? [19:23:14 CET] <furq> are you trying to do a 23.976 to 24fps conversion [19:38:52 CET] <saml> furq, yes [19:39:11 CET] <saml> https://toolstud.io/video/framerate.php?inputfps=23.976&compare=network&dur… I found those from here [19:39:11 CET] <furq> well yeah don't do either of those unless you actually want to reencode [19:39:19 CET] <saml> i'm doing the interpolation. simple -r [19:39:30 CET] <saml> oh is there another way? [19:39:56 CET] <furq> just use mkvmerge/mkvtoolnix and change the fps [19:39:57 CET] <saml> i'm doing this during (re)encoding. but I want to know other way as well [19:40:00 CET] <furq> oh [19:40:15 CET] Action: saml googles mkvmerge .. [19:40:26 CET] <furq> if you want to reencode then definitely use setpts or else you'll end up with dup frames [19:40:29 CET] <saml> i'm mostly .mp4 [19:40:37 CET] <furq> l-smash can probably do it [19:40:48 CET] <furq> ffmpeg will do it but you need to demux and then remux [19:40:48 CET] <saml> wow so many things. thanks [19:41:15 CET] <saml> are you a programmer? [19:41:22 CET] <saml> or professional video person? [19:41:25 CET] <furq> the former [19:41:29 CET] <furq> not really on ffmpeg though [19:41:30 CET] <saml> where do I find people like you for hire? [19:42:06 CET] <furq> shrug [19:42:17 CET] <saml> i guess steal an engineer from youtube or instagram [19:42:42 CET] <furq> anyway yeah if you're reencoding anyway then just use setpts and atempo [19:43:07 CET] <saml> http://l-smash.github.io/l-smash/ wow it's asian [19:43:09 CET] <furq> if you're not then those won't work [19:44:03 CET] <furq> if l-smash doesn't do it then something like ffmpeg -i foo.mp4 -an -c:v copy -f h264 - | ffmpeg -r 24 -i - -i foo.mp4 -c:v copy -c:a aac -af atempo=1.001 bar.mp4 [19:44:12 CET] <furq> but again, only if you're not reencoding [19:44:59 CET] <Chloe> saml: check the ffmpeg site for consulting [19:47:02 CET] <saml> oh nice [20:39:52 CET] <colekas> hello, I have a question regarding HLS streaming using FFmpeg from John Van Sickle - ffmpeg version N-89882-g4dbae00bac-static [20:40:36 CET] <colekas> looking at the code, in the .m3u8 the BANDWIDTH parameter in the EXT-X-STREAM-INF is a 10th of my bitrate value [20:41:04 CET] <colekas> I see in the code that there's a divide by 10 but I am curious why. Nothing in any of the specs I've looked at suggest that's correct [20:49:23 CET] <JEEB> at least that's a version from the git repo http://git.videolan.org/?p=ffmpeg.git;a=commit;h=4dbae00bac [20:51:52 CET] <JEEB> and bit rates are in the master playlist [20:52:57 CET] <JEEB> so `ff_hls_write_stream_info` [20:53:09 CET] <JEEB> that writes the #EXT-X-STREAM-INF:BANDWIDTH [20:54:02 CET] <JEEB> colekas: that is `bandwidth += bandwidth / 10;` [20:54:22 CET] <JEEB> so bandwidth is set to bandwidth + bandwidth / 10 [20:54:26 CET] <JEEB> I guess 10% buffer or so? [20:54:30 CET] <JEEB> it shouldn't be 1/10 [20:55:01 CET] <JEEB> although it only seems to take into mention the bit_rate field, I wonder if this works at all with CRF+VBV/HRD [20:55:51 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/hlsenc.c;h=cc13c… [20:55:54 CET] <JEEB> this is the current code [20:56:51 CET] <JEEB> it should be video bit rate + audio bit rate + 10% [20:56:53 CET] <JEEB> more or less [20:58:50 CET] <colekas> oh [20:58:53 CET] <colekas> i see [20:59:50 CET] <colekas> I'm using libx264 and trying the VBV constrained RC model setting maxrate, bufsize and CFR [21:00:09 CET] <JEEB> yea, CRF doesn't set the bit_rate field [21:00:17 CET] <JEEB> that thing should be patched to also take maxrate into mention [21:00:20 CET] <BtbN> the bitrate is written there before a single frame is ever encoded. So it's nothing more than a wild guess. [21:00:32 CET] <JEEB> it's the avcodec value I guess? [21:00:53 CET] Action: JEEB closed the tab already but stream->xxx->bit_rate sounded like the bit rate value [21:01:06 CET] <JEEB> so most likely it was just your audio or something? [21:01:15 CET] <BtbN> for stuff like crf encoding it's pretty much useless [21:01:15 CET] <JEEB> since it begins with zero, then adds the video, then adds the audio [21:01:36 CET] <JEEB> BtbN: yea - I usually utilize the maxrate value if set in that case [21:01:40 CET] <JEEB> since you need some value there [21:02:04 CET] <JEEB> how close you end up to that maxrate depends on your CRF and complexity of content [21:03:00 CET] <colekas> right, and following this guide http://slhck.info/video/2017/03/01/rate-control.html under Constrained Encoding (VBV) I *do not* want to set the bitrate parameter when doing maxrate/bufsize/crf right? [21:03:29 CET] <JEEB> yea [21:03:45 CET] <JEEB> it's a missing thing in the HLS main playlist writing logic [21:04:03 CET] <JEEB> (I've done it in ISML manifest writing, and wbs did it in DASH IIRC) [21:05:11 CET] <colekas> ah, I see [21:06:25 CET] <colekas> yes, you're right, it's only taking the set audio bitrate and adding it + 10% to the BANDWIDTH flag [21:06:39 CET] <JEEB> yup, since the default value for bit_rate is zero [21:06:44 CET] <JEEB> in the AVCodecContext structure [21:06:57 CET] <JEEB> and it being zero with CRF is 100% correct [21:08:53 CET] <furq> i guess maxrate is what you actually want there if this is for adaptive streaming [21:09:23 CET] <JEEB> I think generally things that write such info out first check bit_rate, then maxrate, and finally zarro if nothing [21:09:29 CET] <JEEB> if I recall things correctly [21:09:42 CET] <furq> maxrate then bitrate seems more sensible [21:10:09 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/movenc.c;h=f4334… [21:10:24 CET] <JEEB> yea, I just followed what wbs (?) did with the DASH muxer [21:10:38 CET] <furq> fair enough [21:10:39 CET] <JEEB> since people tend to expect the bit rate they set if they set a bit rate [21:10:49 CET] <furq> yeah you can probably argue it either way [21:10:53 CET] <furq> either's better than nothing anyway [21:10:57 CET] <JEEB> yeh [21:11:29 CET] <JEEB> it makes sense either way :P maxrate would mean the capped rate over whatever buffer, and bit_rate would mean the overall average bit rate without the spikes taken into mention [21:12:15 CET] <furq> i guess it depends on how players actually use that value [21:12:30 CET] <furq> i always figured their detection was just "if this starts buffering then drop down" [21:12:35 CET] <furq> but maybe it's more sophisticated [21:32:05 CET] Action: rk[ghost] waves [21:32:08 CET] <rk[ghost]> :) [21:32:44 CET] <rk[ghost]> i have some flacs, i want to convert to .ogg for streaming. is default setting for highest quality ogg or do i need an extra flag? [21:36:47 CET] <furq> opus defaults to 128kbps which is what you want, idk about vorbis but probably add -q:a 8 [21:36:57 CET] <furq> you should really be using opus nowadays though [21:37:34 CET] <furq> actually -q 8 is 256kbps so probably drop down to 4-6 [21:38:52 CET] <rk[ghost]> can you give me a quick explanation of opus vs. vorbis.. all the codec jazz twists my neural network in to spaghetti [21:38:58 CET] <sfan5> hydrogenaudio says to use -q 5 [21:39:14 CET] <furq> opus is newer and better [21:39:18 CET] <furq> and maybe more widely supported [21:39:21 CET] <furq> certainly by browsers [21:39:37 CET] Action: rk[ghost] uses mplayer to stream XD [21:40:00 CET] <rk[ghost]> are both apart of the .ogg container? (i still get confused on the structure of audio jazzes :P) [21:40:12 CET] <sfan5> .ogg can hold both [21:40:14 CET] <furq> they're both usually muxed into ogg, yes [21:40:21 CET] <rk[ghost]> ah [21:40:28 CET] <rk[ghost]> it being abstracted away, i didn't notice.. [21:40:37 CET] <furq> just -i foo.flac foo.opus will get you more or less what you want [21:40:48 CET] <furq> (also .opus is still ogg) [21:40:53 CET] <rk[ghost]> ah, hmm. [21:41:02 CET] <furq> you can mess around with the bitrate if you want but 128kbps vbr is recommended [21:41:12 CET] <furq> for music, at least [21:41:13 CET] <rk[ghost]> i see. so ogg is a container and opus/vorbis are the codecs.. or am i simplifying too much? [21:41:19 CET] <sfan5> no that's correct [21:41:20 CET] <furq> no that's exactly right [21:41:33 CET] <rk[ghost]> ha- imagine that... [21:41:48 CET] <rk[ghost]> thanks ya'll:) [21:41:51 CET] <furq> there are other codecs you can mux into ogg but none that are of any significance nowadays [21:41:55 CET] <rk[ghost]> =P [21:41:57 CET] <rk[ghost]> aye. [21:42:10 CET] <sfan5> if you want a better overview, https://en.wikipedia.org/wiki/Vorbis and related pages should be a good reference [21:42:20 CET] <sfan5> furq: flac in .ogg :) [21:42:29 CET] <rk[ghost]> so the -q flag is for quality and it is integer based which maps to some bitrate in the end? [21:42:42 CET] <sfan5> basically [21:42:44 CET] <rk[ghost]> sfan5: thanks but going that route i usually get lost in the details [21:42:45 CET] <furq> nobody uses ogg flac though and it's poorly supported [21:42:53 CET] <rk[ghost]> ah i see. [21:42:56 CET] <furq> rk[ghost]: yes but -q is only for certain codecs [21:42:59 CET] <rk[ghost]> well, my case is slightly special [21:43:05 CET] <sfan5> it doesn't map exactly but just ends up roughly at some bitrate [21:43:10 CET] <rk[ghost]> i am mostly interested in streaming to myself the highest quality i can muster.. [21:43:12 CET] <furq> vorbis uses -q, opus doesn't [21:43:21 CET] <furq> you'd use e.g. -b:a 192kbps if you wanted higher quality opus [21:43:26 CET] <furq> or er- [21:43:27 CET] <rk[ghost]> currently playing around with icecast2/ices [21:43:29 CET] <furq> -b:a 192k [21:43:44 CET] <furq> but yeah the opus guys recommend 128k for transparency [21:43:49 CET] <rk[ghost]> i have some random recordings i took on a .wav recording device that i already converted everything to .flac [21:44:12 CET] <rk[ghost]> ah, coolio. thanks. bitrate makes a little more sense to me as random # require me knowing the mapping.. [21:44:20 CET] <furq> i'm pretty sure icecast supports flac btw [21:44:29 CET] <rk[ghost]> i appreciate all ya'lls quick and straightforward help!! [21:44:31 CET] <sfan5> well bitrate is also arbitrary to some degree [21:44:39 CET] <sfan5> 128k mp3 is not comparable to 128k opus [21:44:41 CET] <sfan5> like at all [21:44:51 CET] <rk[ghost]> furq: you know i thought about testing just to find out.. then somehow that simple route escape my plans. hahaha. [21:45:22 CET] <rk[ghost]> sfan5: noted, thanks. [21:45:42 CET] <rk[ghost]> ps) thanks anyone who is a developer for ffmpeg.. you have treated me so well. [21:47:49 CET] <furq> just tested it, flac works fine over icecast [21:47:59 CET] <furq> idk how well source clients support it but ffmpeg does [21:58:55 CET] <rk[ghost]> furq: when i try to do it.. it crashes, maybe i am doing something wrong.. [21:59:06 CET] <furq> what's the error [21:59:13 CET] <rk[ghost]> the mount point is available, then i try to connect with mplayer [21:59:18 CET] <rk[ghost]> then the mount point disappears [21:59:23 CET] <rk[ghost]> i looked at ices log and i get [21:59:58 CET] <furq> oh i have no idea about ices [22:00:01 CET] <furq> ffmpeg -re -f lavfi -i sine -c:a flac -f ogg icecast:// [22:00:02 CET] <furq> that works [22:00:39 CET] <rk[ghost]> ah, okay.. hmm. i suppose if ffmpeg works in place of ices.. i should investigate further.. [22:00:55 CET] <rk[ghost]> ... my system is ignoring my copy/paste.. ha;P [22:01:41 CET] <rk[ghost]> very very strange [22:02:43 CET] <rk[ghost]> i hit shift+insert on one screen, and it pastes the error, .. i come back to IRC and it pastes a youtube link -.- [22:02:57 CET] Action: rk[ghost] cuddles with inconsistent computertek [22:03:21 CET] <rk[ghost]> anyhoot, i get "corrupt or missing data in file $file" [22:03:45 CET] <rk[ghost]> in the ices log.. i will investigate ices capabilities futher. maybe it is unable to support flac or maybe i need to tell it specifically what it is. [22:03:52 CET] <rk[ghost]> thanks again for your assistance. [22:04:43 CET] <furq> ffmpeg will stream fine to icecast but the support for playlists isn't exactly great [22:06:04 CET] <rk[ghost]> ah, i noticed that based on how you invoked it [22:06:10 CET] <rk[ghost]> but i can bashfu... =P!!! [22:23:31 CET] <jrun> https://paste.pound-python.org/show/4AbQB15Gq4c4wfKgsObg/ [22:23:47 CET] <jrun> that's using cuda on gentoo with mpv [22:24:07 CET] <jrun> let me know how to give more detail if you need any, please. [22:29:07 CET] <jrun> very verbose output, it might help: [22:29:11 CET] <jrun> https://paste.pound-python.org/show/kVXQXTs45m8fDDDNDbbA/ [22:47:05 CET] <FLXX> hi [23:15:33 CET] <saml> wow muxer (l-smash) bro [23:16:12 CET] <saml> muxer -i 'a.mp4?fps=24' -o muxer.24.mp4; ffmpeg -i muxer24.mp4 -i a.mp4 -filter_complex psnr -f null - gives infinity [23:16:20 CET] <saml> it really didn't touch frames at all [23:18:14 CET] <saml> ffmpeg -i a.mp4 -r 24 -filter_complex 'setpts=0.999*PTS' r.24.mp4; and psnr is 42 against muxer24.mp4 [23:25:52 CET] <kepstin> saml: that second ffmpeg command re-encodes the video, so that would be expected yeah. [23:26:35 CET] <kepstin> if you want to set to constant fps without re-encoding, something like "ffmpeg -r 24 -i a.mp4 -c copy r.24.mp4" would do that. [23:26:37 CET] <saml> ah i see [23:26:56 CET] <saml> yeah i tried -r as input and gave me lower psnr [23:27:14 CET] <kepstin> probably just some frames misaligned (not the same exact timestamp) [23:27:52 CET] <kepstin> obviously you can't use psnr to compare files which had their framerates converted using different methods [23:28:11 CET] <saml> yup [23:28:55 CET] <saml> ffmpeg -r 24 -i ski.mp4 -c copy out.mp4; out.mp4 is still 23.98 fps [23:29:43 CET] <saml> i got 120fps video. when I did muxer -i 'a.mp4?fps=24' -o out.mp4; out.mp4 is now slow mo [23:30:07 CET] <saml> because muxer doesn't drop frames [23:30:14 CET] <furq> well yeah that's the whole point [23:30:15 CET] <saml> that makes sense [23:30:37 CET] <furq> you can't drop frames without reencoding [23:30:47 CET] <furq> at least not in most codecs [23:30:49 CET] <saml> ah i see [23:31:23 CET] <saml> is there one script to convert frame rates? [23:31:39 CET] Action: saml scrolls up the log [23:33:11 CET] <saml> so, input -r plus atempo filter. argument to the filter is target-fps/source-fps [23:33:16 CET] Action: saml tries [23:33:25 CET] <furq> i don't think -r does anything as an input option to mp4 [23:37:53 CET] <saml> 120000/1001 fps to 24 fps is too much [23:38:09 CET] <saml> for audio tempo to be slowed down 0.2 [23:47:11 CET] <mont3z> Hi, I'm having some issues to decode a RTP opus stream using ffmpeg. The decoder issues no error but the raw frame size is the same as the encoded frame. Everything works fine using libOpus without ffmpeg. Has anyone experience this kind of issue with ffmpeg? [23:47:37 CET] <mont3z> My problem is the same as the one described here: https://stackoverflow.com/questions/41589934/decoding-opus-using-libavcodec… [00:00:00 CET] --- Thu Feb 8 2018

1 0

ffmpeg-devel.log.20180206
by burek 07 Feb '18

07 Feb '18

[00:27:48 CET] <rcombs> anyone wanna review https://patchwork.ffmpeg.org/patch/7498/ [00:28:13 CET] <rcombs> maybe tmm1 [00:33:32 CET] <nevcairiel> didnt someone else post a patch that tried to do that [00:34:33 CET] <nevcairiel> apple sure could use some pain for their weird design [00:57:09 CET] <Chloe> https://github.com/eintw1ck/FFmpeg/blob/array/libavformat/allformats.c#L566 this might actually be an issue [00:57:27 CET] <Chloe> (using lavd stuff in the old api wrapper for lavf) [00:58:15 CET] <Chloe> oh actually dont worry. I just need to use indev_list instead [02:15:18 CET] <atomnuker> jkqxz: I've sent an opencl related patch on the ml, could you take a look? [02:25:22 CET] <philipl> At least it's not me timing out all the time for a change. [03:20:49 CET] <iive> so quiet in here. [03:22:02 CET] <Chloe> iive: should there be discussions about anything in particular? [03:22:56 CET] <iive> maybe about spaceX launch, of Falcon Heavy, that is supposed to be tomorrow? [03:23:11 CET] <iive> i remember people around here following stuff like that. [03:27:09 CET] <rcombs> tmm1: got a sample that hits that videotoolbox failure case? I'm not exactly sure which code path it'd go down right now [03:38:14 CET] <tmm1> used to happen on live tuner streams for me [03:38:36 CET] <tmm1> i can try that patch tomorrow and see [03:41:38 CET] <FishPencil> Is anyone working on adding support for 8 and 10-bit support in the same FFmpeg build? x264 and x265 both support mulibitdepth now [03:41:38 CET] <rcombs> thanks [03:42:44 CET] <tmm1> does avcodec_send_packet() invoke AVCodec.receive_frame ? [03:44:32 CET] <rcombs> tmm1: yes [03:44:59 CET] <rcombs> if there's not already a buffered frame [03:45:39 CET] <tmm1> huh i didn't realize it buffered frames [03:47:17 CET] <rcombs> didn't used to [03:47:26 CET] <rcombs> it was tweaked to simplify the internal implementation [03:54:50 CET] <rcombs> >If we wanted to we could actually add two more side boosters and make it Falcon Super Heavy [04:37:44 CET] <tmm1> is there a way to query the size of the packet buffer? [04:38:20 CET] <tmm1> i.e. after you call avcodec_send_packet() multiple times [04:48:08 CET] <rcombs> tmm1: lavc itself will only buffer one frame (no packets) [04:48:43 CET] <rcombs> individual codecs can buffer whatever they want internally, and there's no generic way to query that [04:49:27 CET] <rcombs> you have to call receive until you get EINVAL whenever you call send, though [05:47:57 CET] <tmm1> sure one frame on the output, but wont incoming packets queue up? [11:07:54 CET] <BtbN> So, Firefox is now using ffmpeg completely it seems. With all the new hwaccel stuff, using vaapi/vdpau/cuvid hwaccel should be super easy for them, but it's still missing? [11:10:51 CET] <atomnuker> yes, I've been wondering exactly that as well [11:11:36 CET] <atomnuker> they just need to add a few calls during init to run whatever hwaccel is found, and then some code to map what they get to opengl [11:12:18 CET] <atomnuker> jya: any info why no one has bothered to do so? [11:13:21 CET] <atomnuker> I've noticed things at firefox move at glacial pace, the wayland patches were around for literal years before they got merged just recently [11:18:38 CET] <atomnuker> oh wait, I think I know why [11:19:12 CET] <atomnuker> they don't use opengl compositing on anything but windows afaik, not even on linux and not even if you enable all flags and force it [11:20:33 CET] <atomnuker> so even if they did implement it they'd still need to download them to system ram, compose and then back to the gpu [11:20:59 CET] <atomnuker> would still take less cpu though [11:27:52 CET] <BtbN> Even hwaccel with plain stupid hwdownload/sw output would be a huge improvement and would need zero changes outside of ffmpeg code [11:28:15 CET] <BtbN> Does Chrome even support vaapi? [11:28:21 CET] <BtbN> Or vdpau, or any video hwaccel on Linux? [11:39:39 CET] <atomnuker> maybe TD-Linux knows [11:39:51 CET] <TD-Linux> they added vaapi [11:41:43 CET] <TD-Linux> firefox still doesn't use gl on linux [11:41:46 CET] <TD-Linux> so it's less useful [11:49:03 CET] <jya> atomnuker: what's that about? [11:49:27 CET] <nevcairiel> BtbN: browsers are obsessed with decoding actually working, which is why the simple method of just turning on hwaccel is often not s omething they want to do, but add complex checks based on hardware and video properties that ffmpeg doesnt do [11:50:05 CET] <jya> BtbN: if you saw the amount of crashes we get due to bad drivers on windows... how could we ever hope for drivers to be in any better shapes on linux [11:50:22 CET] <BtbN> I understand that, and I totally agree with not enabling it by default. [11:50:31 CET] <BtbN> But at least an option for the experimental testers? [11:50:54 CET] <jya> TD-Linux: well, we can use opengl rendering, it's just disabled by default. It causes kernel crash with many versions of X [11:50:57 CET] <nevcairiel> you could see how the amount of work required is not really warranted if its not going to be used by anyoen that matters :D [11:50:58 CET] <BtbN> I'm not aware of the Firefox Codebase, but with current ffmpeg hwaccel, it can't be a lot of code [11:51:15 CET] <jya> BtbN: the issue isn't in the decoder, that's trivial [11:51:57 CET] <jya> it's the compositor. adding vdpau decoding is great, but then how do you paint it and do whatever is required to do with some crazy css? [11:51:59 CET] <TD-Linux> jya, yeah. I really would like to see a whitelist though, especially because we happily enable webgl 2.0 on most hardware yet can't manage drawing quads. but that's offtopic :) [11:52:33 CET] <BtbN> jya, you slap a hwdownload filter in the chain and get software frames like you do with the software decoder. Not nearly as efficient, but an improvement over no hwaccel at all. [11:52:41 CET] <jya> TD-Linux: it's only in ubuntu 17.10 that I finally managed to not get complete freezing or the kernel stopping to respond . and that's not in our code [11:53:20 CET] <TD-Linux> jya, that's not my experience, but regardless is solved by a whitelist. [11:53:30 CET] <jya> BtbN: actually, our tests show the opposite. We had a prototype with vaapi running, doing the gpu read killed all the performance, there was no worthy benefit [11:53:59 CET] <nevcairiel> didnt vaapi have like three ways to do gpu reads, with only one being fast [11:54:04 CET] <TD-Linux> yeah also remember that firefox uses ffvp9 whereas chromium uses libvpx so the improvement is going to be way smaller just due to that [11:54:07 CET] <BtbN> Yeah, GPU reads from VAAPI are special [11:54:08 CET] <jya> and it was even worse with nvidia [11:54:12 CET] <BtbN> But I think that was recently solved. [11:54:35 CET] <BtbN> nvidia is super fast with read-backs from my experience. Barely any difference when doing transcoding via cuvid->nvenc [11:54:50 CET] <BtbN> Difference between on-device transcode and download + re-uploading frames [11:54:52 CET] <jya> BtbN: the "recently" is what worries me [11:55:08 CET] <BtbN> It's not super recently, 3.4 should have it [11:55:11 CET] <jya> heck, we still have a fair amount of crash report of people using Libav 8 [11:55:16 CET] <TD-Linux> BtbN, is that also true with vaapi? firefox would only implement vaapi or maybe vdpau, not cuvid. [11:55:37 CET] <BtbN> vaapi and vdpau would work for pretty much everything [11:55:52 CET] <jya> i think vaapi is the way to go [11:55:52 CET] <BtbN> well, except for VP9 for vdpau [11:55:55 CET] <BtbN> that's only in cuvid [11:56:00 CET] <jya> especially with the future of vdpau in taters [11:56:03 CET] <BtbN> vaapi is pretty much Intel-Only [11:56:09 CET] <BtbN> Nvidia is all abouv nvdec now [11:56:20 CET] <nevcairiel> all the apis are vendor exclusive, which is why linux is so terrible to support [11:56:22 CET] <BtbN> vdpau still works, but does not get new features, those go to nvdec [11:56:28 CET] <jya> yeah, until next year when it will all about "*****" [11:58:09 CET] <BtbN> ffmpeg currently does a decent job abstracting vdpau/vaapi/nvdec hwaccels. [11:58:24 CET] <jya> yes, from the decoding side of things [11:58:31 CET] <jya> that still requires custom compositing [11:58:52 CET] <BtbN> Are you sure for nvdec a copy-back is a problem? From my experience it's barely noticeable if at all. [11:59:12 CET] <jya> and ffmpeg hwaccel isn't that great when it comes to uniformity. I mean to retrieve the pointer, where you get it from varies depending on the API [11:59:23 CET] <jya> that may have changed though, I haven't looked in over a year [11:59:37 CET] <BtbN> There are abstracted hwdownload functions [11:59:46 CET] <BtbN> and a filter that just gives you software frames no matter what hwformat you throw at it [12:00:00 CET] <BtbN> For at least the 3 or 4 major hwaccel APIs [12:00:09 CET] <BtbN> (Not sure about the OSX stuff) [12:00:15 CET] <jya> last time i looked, you had to take the 4th element of the array to get the handle for vdpau [12:00:19 CET] <jya> while nvidia was at 0 [12:00:23 CET] <nevcairiel> re: nvidia copyback, on low-end systems it can be the difference between 4k 60fps working or not, but thats mostly because you also need to upload again eventually, which just eats into memory bandwidth, but those will also struggle with software decode (if not more so), so its nothing really lost [12:00:45 CET] <BtbN> That's how the HWACCEL pix_fmts work [12:00:50 CET] <BtbN> but you don't need to touch that manually anymore [12:00:56 CET] <jya> good [12:00:57 CET] <BtbN> Downloading them to software is abstracted now [12:01:25 CET] <jya> may be worth looking again, but we have more pressured needs [12:01:28 CET] <BtbN> You just run them through the hwdownload avfilter, and you get whatever software format is inside. Mostly NV12 [12:01:45 CET] <nevcairiel> every hwaccel pixfmt is a special type anyway, if you need to access them directly for on-gpu processing, you need to know how to handle each one invidually anyway, so which data member they are in is rather irrelevant since you need special code anyway. if all you need is download, there are generic functions now that do that for you [12:02:03 CET] <nevcairiel> also if all you want is download, dont use avfilter for that, thats just silly [12:02:08 CET] <nevcairiel> hwcontext has a download function [12:02:13 CET] <jya> ok [12:02:21 CET] <jya> there's doc for that somewhere? [12:02:23 CET] <BtbN> unless you already have a filter chain anyway [12:02:29 CET] <BtbN> for pix_fmt conversion or whatever [12:02:38 CET] <jya> we do not, we don't have have libavfilter in our tree [12:02:41 CET] <nevcairiel> avutil/hwcontext.h contains most of that [12:02:52 CET] <BtbN> Yeah, downloading directly is better then [12:03:33 CET] <atomnuker> BtbN: amd went with vaapi as well [12:03:39 CET] <nevcairiel> also with amf [12:03:48 CET] <BtbN> Isn't AMF encode-only and windows-only? [12:04:30 CET] <atomnuker> jya: I think you'd be surprised how difficult it is to crash mesa on intel with opengl/vulkan, I've never done so despite the crazy things I've done [12:04:55 CET] <nevcairiel> its decode too, and the sdk works on linux, but i dont think their linux drivers support it yet, but they say they are supposed to in the future [12:05:14 CET] <BtbN> great [12:05:17 CET] <jya> atomnuker: as always with you, it's so easy... I suggest you take on the task.. it's open source after all... let me find the link to the bug you can take [12:05:24 CET] <BtbN> I can see why all the vendors create their own encode API [12:05:35 CET] <atomnuker> not sure about how stable amd is though, and I do know nvidia binaries are prone to crashing on valid vulkan uses [12:05:38 CET] <BtbN> But for decode it would be really nice if they could all just properly support something common [12:05:41 CET] <nevcairiel> probably because the vaapi encode api is so terrible :D [12:05:56 CET] <BtbN> like, vdpau, if nvidia would not have let it die [12:06:01 CET] <nevcairiel> its so super low level [12:06:12 CET] <BtbN> vaapi is just tailored around Intel GPUs [12:06:24 CET] <BtbN> any other vendor trying to use it will run into trouble [12:06:33 CET] <jya> https://bugzilla.mozilla.org/show_bug.cgi?id=1210727 [12:06:40 CET] <jya> atomnuker: feel free to take it ^ [12:06:56 CET] <jya> and that's the easy part of just doing the decoding. [12:08:49 CET] <BtbN> Talking about weird (driver?) bugs, I have this very weird issue on my Laptop right now that the menu in Firefox is not visible(but interactible) if I logout of my X session and then re-login without reboot. [12:09:07 CET] <BtbN> On first X login everything works just fine. [12:09:14 CET] <jya> is that with the default layout? [12:09:17 CET] <BtbN> Yes [12:09:26 CET] <BtbN> The menu is there and works, I can click things. Just can't see it. [12:09:33 CET] <jya> when is the last time you've done a config refresh? it usually helps, espeically on linux.. [12:09:56 CET] <BtbN> This happenes on a new user as well. Some other non-firefox menus are also weird, but not invisible. So I doubt it's Firefoxs fault [12:10:38 CET] <BtbN> Like, the Plasma Task-Bar is half-transparent-greyish instead of its normal color. [12:10:41 CET] <jya> i had plenty of weird things when I changed to gnome [12:10:55 CET] <BtbN> I blame mesa and/or Plasma so far [12:11:15 CET] <BtbN> Luckily I don't usually restart my X session [12:11:21 CET] <jya> not sure which distrib you're using, but there's been great progress in the past 3-4 months [12:11:27 CET] <BtbN> Gentoo [12:11:34 CET] <BtbN> I'm not on the very latest Plasma though [12:11:35 CET] <jya> oh I misread [12:11:41 CET] <jya> it's if you logout [12:11:50 CET] <jya> I've stopped login out and back in [12:11:51 CET] <BtbN> It's if I logout and re-login without reboot [12:11:59 CET] <BtbN> Which I don't usually do anyway [12:12:00 CET] <jya> sometimes I get a complete hang on my dell xps 15 [12:12:12 CET] <jya> that if I had nvidia enabled [12:12:28 CET] <BtbN> I just happened to do it when I first setup this Laptop, so stuff was broken. I spent hours debugging, gave up, went to bed, and when I came back the next day (and rebooted in turn) everything was fine. [12:12:32 CET] <jya> I have a gigabyte aero 14 [12:12:39 CET] <jya> when I do that the screen turns black [12:12:56 CET] <BtbN> This is a thinkpad T550 [12:13:05 CET] <BtbN> With a super shitty Broadwell CPU/GPU [12:13:16 CET] <jya> and the keyboard is no longer functional. so I must do a hard reset. but if doing so then the trackpad will no longer work until i reboot under windows. [12:13:17 CET] <BtbN> (And a horrible nvidia GPU, which I just started ignoring) [12:13:25 CET] <jya> then the trackpad works again, and i go back into linux [12:13:49 CET] <BtbN> sounds like something needs firmware that only the Windows driver can upload correctly [12:14:19 CET] <jya> yes I believe so... it's a crappy trackpad , there's plenty of fixes for asus and othrer device. doesn't work with mine [12:14:44 CET] <jya> (I love my aero otherwise, stays super cool and very silent.. unlike the dell xps 15) [12:14:54 CET] <BtbN> My laptop has incredible thermal issues as well. Watching YouTube puts the CPU into thermal throttle to ~1GHZ [12:15:17 CET] <BtbN> So to watch Videos I'm forced to download them and play them with mpv or VLC, which is why I'm so interested in full hwaccel [12:15:56 CET] <jya> on the xps 15 I solved this doing so: https://www.ultrabookreview.com/14875-fix-throttling-xps-15/ [12:17:38 CET] <BtbN> I'm pretty sure it's a firmware issue with my CPU [12:18:01 CET] <BtbN> It starts throtling at 65°C, but the fans do not turn on at all until it reaches 60°C [12:18:22 CET] <BtbN> software control of the fan seems to be impossible [12:27:02 CET] <atomnuker> jya: thanks, I'll look into it, have to start compiling because of wayland anyway [15:28:55 CET] <kierank> jya: weird, i have not had that issue [15:30:30 CET] <jya> kierank: which issue is that? [15:30:56 CET] <kierank> with the xps 15 [16:20:30 CET] <jdarnley> Does anyone know how there are no vsynth3 tests for the vc2 encoder? [16:21:42 CET] <jdarnley> does adding them to RESIZE_OFF doit? [16:22:47 CET] <jdarnley> ah [16:22:49 CET] <jdarnley> "VSYNTH3_OFF = $(RESIZE_OFF) $(INC_PAR_OFF)" [16:51:55 CET] <philipl> BtbN: Is the avcodec built by mozilla just upstream code? You could build your own with a patch to rename the cuvid decoder so that it provides 'vp8' and 'vp9' and that would be a drop in replacement. Don't even need download logic. [16:52:16 CET] <BtbN> It does not bundle ffmpeg [16:52:30 CET] <BtbN> it uses system ffmpeg via some elaborate dynamic loading and version detection logic [16:52:37 CET] <philipl> libmozavcodec.so does not agree with you [16:53:01 CET] <BtbN> Are you sure it's not just a wrapper? [16:53:21 CET] <BtbN> At least jya just complained about users using libav 8 [16:53:27 CET] <philipl> I ran strings on it. [16:53:35 CET] <philipl> Yes, he did. Which confused me. Maybe he can clarify [16:53:41 CET] <philipl> but this library is a libavcodec. [16:53:44 CET] <philipl> I ran strings on it [16:53:53 CET] <jya> BtbN: it is not a wrapper [16:54:11 CET] <jya> it's libavcodec with ffvp8, ffvp9 and flac [16:54:25 CET] <jya> this is for use on *all* platform [16:54:44 CET] <jya> for other codecs, and where libavcodec system is present, then it will be dynamically loaded [16:55:25 CET] <philipl> jya: so, vp8/9 and flac come from the bundled one, and h264/aac/etc come from the system one? [16:55:40 CET] <jya> I can only officially confirm the former :) [16:55:49 CET] <philipl> heh. [16:56:00 CET] <jya> we certainly do not tell people to install anything in particular [16:56:20 CET] <philipl> BtbN: So yeah, if you just do a patched build that renames the codecs, cuvid should just work transparently. [16:56:25 CET] <philipl> I'm going to try it :-) [16:56:26 CET] <jya> but yes, if a system ffmpeg is present, and it supports h264, aac or mp3 it will be used [16:56:36 CET] <jya> what's cuvid? [16:56:41 CET] <BtbN> old name for nvdec [16:56:48 CET] <jya> ah ok [16:56:53 CET] <philipl> jya: but not just a name. It's written as a full decoder rather than an hwaccel [16:56:53 CET] <BtbN> but still present as a stand-alone decoder [16:57:08 CET] <philipl> So it does download-to-system-memory automatically if you ask for a SW format [16:57:15 CET] <jya> cool [16:57:26 CET] <BtbN> It behaves like a normal decoder unless you explicitly ask it not to [16:57:36 CET] <jya> philipl: you may find it easier to patch the existing code [16:57:46 CET] <jya> happy to give you pointers [16:57:59 CET] <philipl> jya: I can rebuild ffmpeg quicker than Firefox to prove the concept. [16:57:59 CET] <BtbN> From how I understand it, that mozavcodec is stripped down to what the lawyers okayed [16:58:10 CET] <BtbN> so cuvid won't be even in there [16:58:12 CET] <philipl> BtbN: command line is available from 'strings' [16:58:15 CET] <philipl> Rebuild away [16:58:53 CET] <philipl> jya: I'll ping you if I fail to get it building. Last time I built firefox was about 15 years ago. [16:59:02 CET] <jya> philipl: https://searchfox.org/mozilla-central/source/dom/media/platforms/ffmpeg/FFm… here is where from the codec mimetype, the ffmpeg codec_id is returned [16:59:17 CET] <jya> so my guess here is that say for h264 you return the cuvid one [16:59:31 CET] <philipl> jya: yes. that would be it, and similarly for vp8/9 [16:59:40 CET] <jya> and here is where you ask what format you want: https://searchfox.org/mozilla-central/source/dom/media/platforms/ffmpeg/FFm… [17:00:01 CET] <jya> the issue now is that we have no ffmpeg fallback mechanism [17:00:09 CET] <jya> so if say ffvp9 fails [17:00:26 CET] <jya> then it will attempt ffvp9 from the system libavocdec, and if that fails that will use libvpx [17:00:55 CET] <jya> we would need something that attempt to use the cuvid one and if that fails fall back to AV_CODEC_ID_H264 [17:01:05 CET] <philipl> jya: So, if I was to move libmozavcodec out of the way, it would fall back to system for all codecs? [17:01:18 CET] <jya> philipl: there's a much easier way :) [17:01:33 CET] <BtbN> It's probably this that needs to be changed to use cuvid: https://searchfox.org/mozilla-central/source/dom/media/platforms/ffmpeg/FFm… [17:02:03 CET] <philipl> jya: some config key? [17:02:04 CET] <jya> philipl: go to the about:config page, search for the media.ffvpx.enabled preference and set it to false [17:02:17 CET] <BBB> ;( fallback to libvpx ;( [17:02:26 CET] <philipl> jya: ok. [17:02:33 CET] <BBB> jya: on a more practical note, do you have statistics on how often that happens? [17:02:49 CET] <jya> BBB: for vp9? [17:02:51 CET] <jya> never [17:02:54 CET] <BBB> for example [17:03:03 CET] <BBB> oh, cool [17:03:09 CET] <jya> unless played with the pref [17:03:11 CET] <BBB> *silent voice* and vp8? [17:03:16 CET] <jya> same thing [17:03:23 CET] <BBB> *pheew* [17:03:37 CET] <jya> and we favour *our* ffvp8/ffvp9 over the system vp9 [17:03:48 CET] <jya> just because we control that code and it's a known entity [17:03:53 CET] <BBB> yup [17:04:03 CET] Action: BBB goes slack in a corner again [17:04:11 CET] <jya> libvpx was kept for webrtc (which still doesn't use it) [17:04:19 CET] <BBB> I was scared something was wrong and this fallback mechanism was added to cover up my failures [17:04:31 CET] <jya> because that libvpx decoder isn't used for webrtc, but the one deep in the webrtc.org code [17:04:59 CET] <jya> BBB: no no, that fallback was added so I could test, and if close to release we found something was screwed up we could disable the code path [17:05:36 CET] <jya> ohhh... actually we can use libvpx [17:05:44 CET] <jya> vp8/vp9 with transparency layer [17:05:55 CET] <jya> only the libvpx code was patched up [17:06:10 CET] <jya> forgotten about that one.. it's so messy [17:06:13 CET] <BBB> ah yes I remember that [17:06:15 CET] <BBB> yes its messy [17:06:17 CET] <BBB> sorry about that [17:06:25 CET] <jya> i've seen such horrible videos with transparency [17:06:27 CET] <BBB> its not hard to add to native vp8/9 decoders but someone needs to care [17:06:39 CET] <jya> like the keyframes don't match between the two streams [17:06:47 CET] <BBB> yikes [17:07:21 CET] <jya> so now when you seek, if the transparency isn't a keyframe, we skip backward until we find one [17:07:35 CET] <jya> and we skip backward because sometimes the keyframe is just at the start [17:07:49 CET] <jya> that kills our cache handling and we have a bug opened that seeking is now slow [17:08:04 CET] <jya> from always the same guy, who always manage to create the worse encoding of all kind [17:11:21 CET] <nevcairiel> I like ignoring such people and their artifical samples =p [17:12:50 CET] <nevcairiel> if you try to fix every broken sample out there, you go insane [17:14:21 CET] <wm4> seems pretty sane to require that transparency data has the same keyframe flags at least [17:14:25 CET] <wm4> but really this is such a shit hack [17:16:49 CET] <jamrial> <@nevcairiel> if you try to fix every broken sample out there, you go insane [17:16:51 CET] <jamrial> that explains so much about ffmpeg :p [17:59:29 CET] <BBB> wm4: the sad part is that the samples created in the wild with the tools given to us dont actually do that [17:59:39 CET] <BBB> so requiring something that is not actually true by default is & not going to work [17:59:40 CET] <BBB> :( [17:59:47 CET] <BBB> I dont disagree it *should* be that way [17:59:49 CET] <BBB> but it is not :/ [18:00:27 CET] <nevcairiel> just have to make a stand and fail decoding any samples that are just crap like that [18:00:36 CET] <nevcairiel> or hire assassins to purge the world of bad developers that enabled this [18:01:23 CET] <wm4> what tools do create it= [18:01:26 CET] <wm4> ? [18:01:37 CET] <philipl> jya: where's the part that builds libmozavcodec? [18:14:53 CET] <philipl> jya: found it [18:19:43 CET] <philipl> jya: Yeah, so even the cuvid decoder won't actually drop in and work as it doesn't return any of the SW formats that firefox expects. It does NV12 and P010 [18:36:22 CET] <Chloe> why do all commits have to build in sequence [18:37:04 CET] <Chloe> What's the issue with not merging two commits which are very different (i.e. shouldnt be merged), but they are dependent [18:38:02 CET] <wm4> because that makes bisecting easier [18:42:24 CET] <nevcairiel> yeah its super annoying when you bisect and stuff doesnt build [18:42:26 CET] <DHE> `git bisect` is a great thing. [18:46:14 CET] <Chloe> how are you using `git bisect`? [18:47:25 CET] <DHE> any time I run into a regression, I pick 2 versions of $SOFTWARE, one working and one broken, and run "git bisect start; git bisect good $WORKINGVERSION ; git bisect bad $BROKENVERSION" [18:47:45 CET] <DHE> and git picks a revision in between for you to test. you report good or bad and it repeats the process until it nails the commit that broke it [18:48:05 CET] <DHE> but for this to work, each version needs to build and work properly. and preferably be small-ish, easy to digest updates [18:49:17 CET] <Chloe> ah right that's interesting, sounds useful [18:50:06 CET] <hanna> git bisect is also a good reason not to use merge commits [18:50:15 CET] <hanna> git needs to test each merge base every time [18:50:33 CET] <nevcairiel> that works surprisingly well however [18:50:40 CET] <hanna> it works but it takes longer [18:51:07 CET] <hanna> although I guess git could make an effort to pick a bisection that avoids merge commits if possible [18:51:11 CET] <hanna> which should be possible for small merge branches [18:51:19 CET] <hanna> not sure if it does this, though? [18:51:41 CET] <nevcairiel> in our case you can just skip the steps where you end up on the libav branch, so it doesnt really add any extra steps [18:51:54 CET] <nevcairiel> we even have a helper script that just does this for you by checking if ffmpeg.c exists [18:51:57 CET] <hanna> ah [18:52:02 CET] <hanna> well that sounds useful [18:52:08 CET] <hanna> but it also sounds like a hack [18:52:32 CET] <nevcairiel> well "git bisect skip" is a thing, so [18:52:36 CET] <hanna> anyway git needs to be replaced by a patch-based system :^) [18:52:43 CET] <hanna> everybody uses git in a patch-based workflow anyway [18:53:07 CET] <hanna> state based 3-way merge is broken [18:54:14 CET] <hanna> https://pijul.org/ is the latest meme [18:55:07 CET] <Chloe> tools/bisect-create? [19:21:08 CET] <iive> If there are still people who are interested in SpaceX stuff, falcon heavy is scheduled for launch in 2 hours. get popcorn for the fireworks :D [19:22:38 CET] <jdarnley> I'll watch the gif highlights for any fireworks [19:49:53 CET] <durandal_1707> iive: im watching p0rn instead [19:57:59 CET] <cone-205> ffmpeg 03Josh de Kock 07master:7e8eba2d8755: lavc: add new API for iterating codecs and codec parsers [19:58:00 CET] <cone-205> ffmpeg 03Josh de Kock 07master:61974537610d: lavf/rtp: replace linked list with array [19:58:01 CET] <cone-205> ffmpeg 03Josh de Kock 07master:598d5f8579b3: lavf: move fifo test muxer into separate file [19:58:02 CET] <cone-205> ffmpeg 03Josh de Kock 07master:0694d8702421: lavf: add new API for iterating muxers and demuxers [19:58:03 CET] <cone-205> ffmpeg 03Josh de Kock 07master:0fd475704e87: lavd: add new API for iterating input and output devices [19:58:04 CET] <cone-205> ffmpeg 03Josh de Kock 07master:cdc78058c78d: cmdutils: make use of new iteration APIs [19:58:05 CET] <cone-205> ffmpeg 03Josh de Kock 07master:26d879c1cea1: lavc/bsf: make BSF iteration the same as other iterators [20:00:05 CET] <jamrial> Chloe: that set wasn't fully reviewed and oked [20:00:32 CET] <jamrial> why did you push it? [20:00:40 CET] <durandal_1707> oh nooo, what have you done! [20:01:52 CET] <Chloe> you serious? I've been asking for comments for ages, and again, people wait until after things are pushed. I fixed all the issues which were described [20:02:20 CET] <Chloe> but if that's not good enough then fuck it, revert and fix it yourself. [20:03:10 CET] <atomnuker> well that was a bit rude of him, michaelni just emailed him -f lavfi didn't work [20:03:30 CET] <atomnuker> I'll try to fix it [20:03:54 CET] <JEEB> huh, so michael now posts under an alias of Muhammad? [20:04:01 CET] <JEEB> I thought that was a separate person :P [20:04:01 CET] <durandal_1707> yes [20:04:20 CET] <atomnuker> misremembered, oh well [20:04:25 CET] <JEEB> it happens [20:04:56 CET] <JEEB> anyways, that wasn't nice. since while `-f LIBAVDEVICE` probably is a hack, it still used to work [20:07:23 CET] <jamrial> seriously, no patch in the set has gotten an ok. there were unaddressed design suggestions [20:07:25 CET] <jamrial> there was even a "competing" set doing the same thing in a different way [20:07:45 CET] <jamrial> this needed at the very least a ping before being pushed. it's a massive API change [20:09:13 CET] <jamrial> i don't understand the rush, and much less the reaction he had at being told hadn't even gotten one ok yet... [20:10:46 CET] <hanna> so michaelni unironically sockpuppets? [20:10:48 CET] <durandal_1707> people cant wait for comments that happen like every 96h [20:10:54 CET] <hanna> does he use it to agree with himself? :^) [20:11:36 CET] <durandal_1707> michaelni is not problem here [20:11:45 CET] <JEEB> hanna: he misremembered :P [20:12:14 CET] <wm4> jamrial: maybe it's because he's been trying to get in for weeks and it was all bikeshedding and people going "hey this patch set was there for weeks but suddenly I have an important opinion about it" [20:12:58 CET] <jamrial> wouldn't you have an opinion about a full API replacement? [20:13:28 CET] <hanna> oh now I get it [20:13:49 CET] <hanna> fucking pronouns man [20:13:52 CET] <hanna> just say atomnuker [20:14:09 CET] <jya> philipl: that's easy fix, we have backend for both nv12 and 10 bits image. [20:14:23 CET] <JEEB> I was trying to think if the "allocate the whole thing for each request of a list" thing made any sense [20:14:41 CET] <JEEB> but then didn't have that much time to give it (Ž4@) [20:14:46 CET] <jya> 10 bits image only works on OGL machines though, so Linux and max [20:15:26 CET] <atomnuker> hmm, I can't replicate "./ffmpeg -f lavfi -i aevalsrc=0 -f null -" or any lavfi source [20:16:12 CET] <JEEB> yea [20:16:22 CET] <JEEB> he said to me he had tested that and full FATE worked for him [20:16:30 CET] <JEEB> he could have replied to that mail at least :P [20:18:14 CET] <jamrial> he mentioned lavfi worked for him in a reply to the thread, just not to Faiz's email [20:18:53 CET] <JEEB> right [20:19:01 CET] <jamrial> i don't know if because he fixed it before pushing, or because it was fixed in a subsequent patch in the set, meaning a bisect would be able to reproduce that failure [20:21:17 CET] <JEEB> at least there's just a single cmdutils commit [20:21:32 CET] <JEEB> and before that it utilizes the old API? [20:23:32 CET] <philipl> jya: so how do I express that NV12 has only two planes in the YCbCrBuffer? It's enough for the plane[3] data to be null? [20:23:35 CET] <philipl> I don't see any type flag [20:25:43 CET] <JEEB> set pix_fmt to NV12 and use the two first planes [20:25:49 CET] <JEEB> set third to NULL I think [20:26:06 CET] <philipl> jya: this is the firefox specific buffer metadata [20:26:09 CET] <philipl> JEEB: sorry [20:26:28 CET] <JEEB> oh [20:41:36 CET] <atomnuker> is there any way to prevent ffmpeg.c doing a full decode to figure out parameters before doing a second decode to actually decode? [20:42:09 CET] <tmm1> isn't there some way to skip find_stream_info [20:44:25 CET] <JEEB> probesize/analyzeduration was it? [20:49:35 CET] <wm4> tmm1: by not calling the function, and if ffmpeg.c always calls it, an option to prevent it should be added [20:49:47 CET] <wm4> unfortunately random things tend to break when skipping it [20:55:37 CET] <tmm1> i thought i saw some commit where you could set probesize to 0 or -1 to skip [20:55:41 CET] <tmm1> but don't remember if it was committed [21:33:59 CET] <iive> T-12minutes [21:37:04 CET] <durandal_1707> iive: climax approaching [22:06:26 CET] <philipl> jya: Looks like YCbCrBuffer is not usable today to pass NV12. I see assertions in MediaData.cpp that plane[1] and plane[2] have the same dimensions and that all three planes are present. [22:06:54 CET] <jya> not nv12 no [22:07:13 CET] <jya> but there's a nv12 buffer , let me think [22:08:09 CET] <jya> https://searchfox.org/mozilla-central/source/gfx/layers/ImageContainer.h#916 [22:08:38 CET] <jya> philipl: that code may not have been well tested I'm afraid [22:08:51 CET] <jya> i saw some bad things a while ago when i went over that code [22:10:23 CET] <jya> also, you'll need to add libavcodec.58.so support there https://searchfox.org/mozilla-central/source/dom/media/platforms/ffmpeg/FFm… [22:10:59 CET] <jya> and if doing so, you can take bug 1435212 and ask me for review [22:11:03 CET] <philipl> jya: SO MediaData returns a PlanarYCbCrData, which can store NV12. So enabling that is relatively simple, I think. Then we need to see whether the consumer knows or whether more adapting is required. [22:12:09 CET] <jya> not quite [22:13:21 CET] <jya> MediaData *takes* a YCbCrBuffer, but it can also take an image [22:13:33 CET] <jya> so for that you create a NV12Image instead [22:15:23 CET] <jya> https://searchfox.org/mozilla-central/source/dom/media/MediaData.cpp#428 [22:15:34 CET] <jya> so you can create a VideoData from an image [22:16:23 CET] <philipl> jya: Yeah, of course, this is way ahead of knowing what you're dealing with - you don't know if it will come back as YUV420P or NV12 [22:16:32 CET] <jya> so if nv12 buffer, rather than recreate the YCbCrBuffer and make VideoData::CreateAndCopy create the YCbCrImage for you, you need to create the nv12 image yourself [22:16:58 CET] <jya> well, it doesn't matter what it is, it takes an Image [22:17:05 CET] <jya> an Image can be of many types [22:17:54 CET] <jya> so you need to add that special case there: https://searchfox.org/mozilla-central/source/dom/media/platforms/ffmpeg/FFm… [22:18:15 CET] <jya> if the pixel type is nv12, then handle the case and return early [22:18:22 CET] <philipl> jya: Yeah, I get it [22:18:45 CET] <jya> that type of type is a bit easier to handle in C++ than in C! [22:19:37 CET] Action: jya grrmrmrmbl i receive my neighbour wifi signal better than my own [22:22:03 CET] <iive> durandal_1707, https://youtu.be/wbSwFU6tY1c?t=37m47s double culmination ;) [22:23:02 CET] <philipl> jya: Ok, so I create an NVImage, and to do that I need to create a PlanarYCbCrData and fill that in for it. Then I call CreateFromImage to get the VideoData [22:23:06 CET] <durandal_1707> iive: its all cgi [22:23:29 CET] <jya> philipl: looks like it [22:23:31 CET] <iive> of course, the earth is flat. [22:24:06 CET] <jya> \o/ ... the i9-7980EX has finally shipped [22:24:29 CET] <durandal_1707> iive: im serious, look at that frames with car in space [22:25:43 CET] <jya> philipl: if you have any questions, lodge them to the bug I pointed to earlier today, or put them on mozilla's irc #media and I'll get back to it tomorrow [22:26:56 CET] <philipl> jya: Ok. Will do [22:28:49 CET] <feliwir> hey, i have a video file that should be playable with ffmpeg but isn't picked up [22:28:59 CET] <feliwir> could anyone maybe take a look? [22:30:55 CET] <feliwir> https://ufile.io/qrhak [22:31:07 CET] <feliwir> this should be an EA container with vp6 content [22:34:35 CET] <feliwir> ffmpeg-probe just says "Invalid data found when processing input" [22:35:16 CET] <feliwir> nvm it's contained in another ea format [22:35:57 CET] <JEEB> :) [22:42:55 CET] <feliwir> well, now i really good a file that doesn't work [22:44:09 CET] <feliwir> https://youtu.be/E-j66dZo3Ms?t=2m50s [22:44:23 CET] <feliwir> those videos in the topright don't have alpha in ffmpeg [22:46:41 CET] <feliwir> it only finds a single vp6 stream, but doesn't recognize the alpha channel [22:47:31 CET] <feliwir> https://ufile.io/7hwh5 [22:47:55 CET] <feliwir> I fixed vp6 alpha in ea files already a while ago [22:48:02 CET] <feliwir> but those are other files from newer games [22:49:44 CET] <feliwir> there is some "ALhd" tag [22:50:13 CET] <feliwir> https://github.com/FFmpeg/FFmpeg/blob/9ec8790ac4787d3d514c5fa27b66d581614fd… [22:50:19 CET] <feliwir> which is not supported [22:57:31 CET] <philipl> jya: Well, I managed to get some recognisable frames on the screen with messed up chroma and then the process crashed. Progress. [22:57:39 CET] <philipl> Had to disable seccomp sandbox for cuda [22:58:09 CET] <jya> Not surprising that the sandbox would kick in. [23:00:13 CET] <philipl> jya: It's not showing me a backtrace from the content process [23:00:53 CET] <jya> Disable multi-process, much easier to debug [23:01:38 CET] <jya> If you start Firefox using Mach, you do ./mach run --disable-e10s [23:02:34 CET] <philipl> jya: thanks [23:08:50 CET] <cone-205> ffmpeg 03Menno 07master:204c7caf0d77: avcodec/libopus: support disabling phase inversion. [23:08:51 CET] <cone-205> ffmpeg 03Jérôme Martinez 07master:698d5eb5bf83: avcodec/ffv1: Support for RGBA64 and GBRAP16 [23:08:52 CET] <cone-205> ffmpeg 03Michael Niedermayer 07master:8f6a71f7f13d: avformat/bintext: Implement bin_probe() [23:10:09 CET] <SortaCore> well, good thing I read the backlog [23:10:20 CET] <SortaCore> otherwise I'd never have got so intimate with steven's connection [23:11:19 CET] <philipl> jya: Ok, so my UV plane representation is wrong. Need to work out what it's expecting. [23:12:14 CET] <philipl> jya: hard to tell if I'm supposed to just fill in one plane or I'm supposed to set it up to look at the interleaved plane as two planes (using skip and offsets) [23:12:20 CET] <jya> As I said, it's very likely the nv12 image code is broken [23:12:22 CET] <philipl> The only example I can see is a test that does this [23:12:25 CET] <philipl> :-/ [23:13:07 CET] <jya> When I added the 10 bits support, I had to go over all images, and the calculations and handling of strides was all wrong [23:14:18 CET] <jya> (For nv12) [23:14:45 CET] <jya> I think I created a bug thst it was broken, but as it's currently unused no one cared. [23:32:36 CET] <philipl> jya: getting closer. Got something that looks more correct before it crashes [00:00:00 CET] --- Wed Feb 7 2018

1 0

ffmpeg.log.20180206
by burek 07 Feb '18

07 Feb '18

[00:00:36 CET] <pzich> [00:00:38 CET] <pzich> ~ [00:01:00 CET] Last message repeated 2 time(s). [01:06:41 CET] <intrac> I'm trying to check for corruption or continuity problems in a set of .ts files. if I play the files with ffplay, at various points I get errors like: [01:06:43 CET] <intrac> [h264 @ 0x7f6e781f9a00] error while decoding MB 0 54, bytestream 45085 [01:06:47 CET] <intrac> [h264 @ 0x7f6e781f9a00] concealing 1729 DC, 1729 AC, 1729 MV errors in I frame [01:07:22 CET] <intrac> is there a way to get ffmpeg to scan a file and only report back if it encounters these errors? [01:07:35 CET] <intrac> I've tried this suggestion without success: https://superuser.com/questions/100288/how-can-i-check-the-integrity-of-a-v… [01:07:38 CET] <furq> maybe ffmpeg -xerror [01:07:43 CET] <furq> i forget if that bails on warnings though [01:08:33 CET] <furq> if you just want it to log the warnings then you'll need -v warn, not -v error [01:09:48 CET] <intrac> furq: I've tried 'error' and 'warning', but they don't show the same errors that ffplay does. [01:10:36 CET] <intrac> the 'concealing X errors in I frame' [01:10:55 CET] <intrac> also, it'd be helpful if the timecode was output as well [01:12:18 CET] <intrac> it doesn't look like it's actually scanning through the whole file as it completes within a second or two [01:12:35 CET] <intrac> and it's a 1 hour HD recording [01:14:12 CET] <intrac> fairly sure that superuser posting is incorrect. it's not reading the frames, just the header/meta/basic file formats [01:14:39 CET] <furq> that command will decode the streams [01:15:04 CET] <furq> maybe add -enable_er 0 before -i [01:15:43 CET] <intrac> furq: but at what level? it might extract the elementary stream, but I can't see that it's actually decoding every frame from the h264 stream in a couple of seconds [01:16:01 CET] <intrac> and it might need to do a full decode to spot these errors [01:38:03 CET] <intrac> hm, maybe ffprobe can extract the packet data [02:58:34 CET] <FishPencil> Does FFmpeg support x264 and x265's multi bitdepth? [03:28:10 CET] <kepstin> FishPencil: not yet (you need to use a pre-multi-bit-depth x264 or it'll fail to compile) [03:28:28 CET] <kepstin> at least in the 3.4 release and last time i checked git master [03:43:34 CET] <kepstin> FishPencil: x265 multi bit depth already works with ffmpeg [03:45:54 CET] <DHE> x265 has had it for years now. x264 is, what a month or two? [03:46:41 CET] <kepstin> committed in december 2017, yeah [03:46:54 CET] <kepstin> (iirc) [03:47:24 CET] <DHE> yep, committed december 24th [03:47:39 CET] <DHE> oddly, authored january 6th 2017... that sat for a while [03:47:54 CET] <kepstin> it's been a wip on their mailing list for a long time [03:49:20 CET] <FishPencil> How is it done? [03:50:50 CET] <kepstin> I haven't looked closely, but I expect it basically works by compiling the library multiple times (with symbol prefixes) and then linking them together with some wrapper code that calls the correct functions based on runtime config [03:51:02 CET] <DHE> that's what the patch implies [03:51:16 CET] <FishPencil> oh sorry, how do I use it in FFmpeg [03:51:21 CET] <FishPencil> is it done with -pix_fmt? [03:51:41 CET] <DHE> yeah, give it a 10bit format and it'll use the 10bit x264 code [03:51:44 CET] <kepstin> FishPencil: the x265 wrapper in ffmpeg just selects based on the pix_fmt used, yeah [03:52:16 CET] <kepstin> FishPencil: you can see this with "ffmpeg -h encoder=libx265", the list of pixel formats at the top of the output shows what the library can do [03:52:32 CET] <kepstin> I expect the x264 wrapper will work similarly when it's added [03:53:01 CET] <DHE> it's supported now in the Git version. support was added pretty quick. like 2 days later [03:53:29 CET] <kepstin> oh, it is supported in ffmpeg? I took a look through the commits but didn't see anything obvious [03:53:35 CET] <kepstin> maybe I just didn't look back far enough [03:54:44 CET] <kepstin> https://git.videolan.org/?p=ffmpeg.git;a=commit;h=c6558e8840fbb2386bf8742e4… well there it is [04:00:34 CET] <FishPencil> What is a common 10-bit pix_fmt FFmpeg supports? [04:03:43 CET] <kepstin> FishPencil: you'll almost always be using yuv420p10le [05:21:14 CET] <prelude2004c> hey guys.. can i get some help.. ${ffmpeg} -hwaccel cuvid -c:v mpeg2_cuvid -deint 2 -resize 1280x720 -hwaccel_device $gpuenc -max_alloc 100000000 -threads 1 -vsync 0 -fflags +discardcorrupt -i "$stream" .. it wont deinterlace for some reason [05:21:18 CET] <prelude2004c> i still get only 29.97fps [05:21:30 CET] <prelude2004c> when i do it the other way with -vf yadif it works but then uses up all my CPU [05:21:36 CET] <prelude2004c> i'm trying to deinterlace on decoder [05:28:20 CET] <prelude2004c> tried deint bob which is same but no lucky.. still at 29.97fps [05:48:36 CET] <prelude2004c> anyone around? [06:30:40 CET] <Diag> prelude2004c: yadif is frame interpolation eh? [06:36:09 CET] <prelude2004c> i was trying yadif but that is usually on the encoder side right? [06:36:15 CET] <prelude2004c> i'm trying to deinterlace on the decoder side [06:36:34 CET] <Diag> uhhh [06:36:41 CET] <prelude2004c> reading things i'm told like [06:36:53 CET] <prelude2004c> -c:v mpeg2_cuvid -deint bob [06:37:01 CET] <Diag> the video should be deinterlaced when its decoded really [06:37:11 CET] <Diag> im not familiar with how ffmpeg does it [06:37:11 CET] <prelude2004c> and output shows : Stream #0:0 -> #0:0 (mpeg2video (mpeg2_cuvid) -> hevc (hevc_nvenc)) [06:37:20 CET] <Diag> >nvenc [06:37:22 CET] <Diag> theres your problem [06:37:31 CET] <Diag> dont ever expect anything nvidia makes to work [06:37:35 CET] <Diag> however [06:37:40 CET] <prelude2004c> well transcoding normally works [06:37:41 CET] <Diag> How fast is the video being encoded? [06:37:46 CET] <Diag> in terms of fps [06:37:47 CET] <prelude2004c> realtime [06:37:55 CET] <Diag> so its locked to 29.97? [06:37:57 CET] <prelude2004c> comes in at 29.95 interlaced [06:38:05 CET] <prelude2004c> yes sorry 97 [06:38:08 CET] <Diag> 29.95 interlaced? grody [06:38:11 CET] <Diag> oh [06:38:12 CET] <Diag> lol [06:38:13 CET] <prelude2004c> and i want to deinterlace to 59.94 [06:38:27 CET] <Diag> you sure you dont got that backwards? [06:38:40 CET] <Diag> or do you want the interpolation [06:38:41 CET] <prelude2004c> i tried to use a -vf yadif but cpu was through the roof [06:38:52 CET] <Diag> yeah its probably pretty intensive [06:38:55 CET] <Diag> what res video are you using [06:38:58 CET] <prelude2004c> i dont care how it works :) i just want to convert to progressive [06:39:14 CET] <prelude2004c> its coming in at 1080i , i am converting to 720p [06:39:25 CET] <Diag> whats it coming from? [06:39:31 CET] <prelude2004c> fiber link [06:39:48 CET] <Diag> what sort of cpu is this running o [06:39:49 CET] <Diag> on* [06:39:57 CET] <prelude2004c> xeon's [06:40:05 CET] <prelude2004c> but.. i run like 15 channels on the system [06:40:06 CET] <Diag> and what do you consider "through the roof" [06:40:07 CET] <Diag> for usage [06:40:15 CET] <prelude2004c> so.. i have to use decoder / encoder on the GPU [06:40:26 CET] <Diag> gpu cant yadif [06:40:38 CET] <prelude2004c> ya but its supposed to -deint 2 [06:40:46 CET] <prelude2004c> on cuvid isn't it? [06:41:21 CET] <Diag> yeah [06:41:21 CET] <prelude2004c> the idea is.. source is mpeg2video .. i use cuvid to decode and deinterlace, then pass it to nvenc to transcode back to 720p nvenc [06:41:26 CET] <prelude2004c> hevc [06:41:51 CET] <prelude2004c> even on the decoder i can -resize so.. i dont' even have to do anything there. [06:42:01 CET] <Diag> What do you consider through the roof though [06:42:03 CET] <Diag> for cpu usage [06:42:09 CET] <prelude2004c> like 200% per channel [06:42:15 CET] <Diag> 200% being...? [06:42:16 CET] <Diag> 2 cores? [06:42:20 CET] <prelude2004c> ya pretty much [06:42:25 CET] <Diag> what xeons? [06:42:37 CET] <prelude2004c> considering with GPU work only 30% right now [06:42:47 CET] <prelude2004c> 5660 i think or something like that [06:43:02 CET] <prelude2004c> X5650 [06:43:21 CET] <prelude2004c> they are multhreaded 6's [06:43:32 CET] <Diag> so not a pos [06:44:24 CET] <Diag> dunno though [06:44:29 CET] <prelude2004c> it's weird.. i just want to deinterlace source.. and then do whatever. [06:44:32 CET] <Diag> interpolation is pretty cpu heavy [06:44:39 CET] <Diag> why yadif, why not just blend or whatever [06:44:50 CET] <prelude2004c> blend ? how [06:44:56 CET] <Diag> idk is that not a thing [06:44:59 CET] <prelude2004c> not familiar with that [06:45:55 CET] <Diag> tblend? [06:45:59 CET] <Diag> it appears? I dk [06:46:10 CET] <Diag> >The tblend (time blend) filter takes two consecutive frames from one single stream, and outputs the result obtained by blending the new frame on top of the old frame. [06:47:06 CET] <Diag> idunno, kinda talkin outta my ass here as im not familiar with ffmpeg [06:47:19 CET] <Diag> I just know blend is a popular shit because its low overhead [06:48:01 CET] <prelude2004c> sounds like fancy filters [06:49:45 CET] <Diag> dunnno [06:49:57 CET] <Diag> youre probably doin some crazy ass shit [06:50:04 CET] <Diag> Since i know someones gonna ask [06:50:11 CET] <Diag> you should post the full command youre running [06:51:04 CET] <prelude2004c> minterpolate < i tried that too [06:51:17 CET] <prelude2004c> crazy?? not too crazy no [06:51:58 CET] <Diag> No i mean like [06:52:23 CET] <Diag> you probably got 29.97 video thats 15 full frames a second [06:52:35 CET] <Diag> and youre tryna boost it up to 60full frames a second [06:52:44 CET] <Diag> thatd drive your cpu usage through the roof if it was interpolating it all [07:03:03 CET] <prelude2004c> well its 29.95 interlaced [07:03:09 CET] <prelude2004c> and i want 59.94 deinterlaced [07:03:21 CET] <prelude2004c> just not working.. trying different things and trying to be as efficient as possible. [07:03:34 CET] <prelude2004c> nobody seems to know :P [07:03:36 CET] <Diag> you can only be so efficient... [08:10:57 CET] <kazuma_> you just bob it prelude2004c [08:46:52 CET] <prelude2004c> kazuma_, I did bob it [08:46:56 CET] <prelude2004c> i did -deint bob [08:47:00 CET] <prelude2004c> still nothing .. doesn't do it [09:15:42 CET] <kazuma_> prelude2004c [09:15:51 CET] <kazuma_> -vf yadif=1:0 [10:00:45 CET] <slasktrat> Hi. I've updated an one year old, unclosed ticket (http://trac.ffmpeg.org/ticket/6165) do I need to send a notification somewhere? [10:14:22 CET] <blop> what ./configure option enables --vo=opengl:backend=x11egles ? [10:24:44 CET] <blop> ah i already have egl-drm egl-helpers egl-x11 [10:25:03 CET] <blop> in mpv [12:14:17 CET] <Kapricornus> Hello [12:29:19 CET] <Projec[T]MayheM> Hello guys [12:29:28 CET] <Projec[T]MayheM> anyone can please help me with a ffmpeg segment error? [12:31:41 CET] <Projec[T]MayheM> I'm recording an rtsp stream in 1 minute chunks but some segments show a 1 hour plus 1 minute duration [12:31:46 CET] <Projec[T]MayheM> with ffprobe I get this: [mpegts @ 0x2e86960] DTS discontinuity in stream 0: packet 31 with DTS 575999, packet 32 with DTS 337043505 [12:32:11 CET] <Projec[T]MayheM> the DTS jumps from 500k to 337M [12:32:15 CET] <Projec[T]MayheM> anyone? [12:35:00 CET] <JEEB> mpegts can have fun timestamp stuff [12:35:43 CET] <Projec[T]MayheM> any way to reset this? [12:35:53 CET] <JEEB> unless that mpeg-ts is written by you, in which case you should check the input timestamps from RTSP [12:36:39 CET] <Projec[T]MayheM> I do this "ffmpeg -y -i rtsp://10.28.2.195/proxyStream-1 -c copy -f segment -segment_time 60 -reset_timestamps 1 -segment_atclocktime 1 -strftime 1 test%Y-%m-%d_%H-%M-%Stest.ts [12:36:43 CET] <Projec[T]MayheM> ops [12:36:44 CET] <Projec[T]MayheM> ffmpeg -y -i rtsp://10.28.2.195/proxyStream-1 -c copy -f segment -segment_time 60 -reset_timestamps 1 -segment_atclocktime 1 -strftime 1 test%Y-%m-%d_%H-%M-%Stest.ts [12:36:55 CET] <Projec[T]MayheM> is there a flag I can use to reset the timestamps? [12:38:31 CET] <Projec[T]MayheM> maybe a setpts filter? [13:00:11 CET] <zerodefect> Using the C-API to change the properties of the overlay filter dynamically. I'm using the 'avfilter_graph_send_command' commands to update the overlay properties of the filter before pushing in a new frame. [13:00:18 CET] <zerodefect> I'm getting an error: 'ENOSYS' or "Function not implemented" [13:01:04 CET] <zerodefect> Now I'm not to sure on the difference between the 'cmd' and 'arg' params to 'avfilter_graph_send_command' [13:01:49 CET] <zerodefect> So far, I'm specifying "x=10:y=10:W=720:H=576:w=253:h=253" for cmd and NULL for arg :D [13:16:26 CET] <furq> pretty sure that should be arg and cmd should be "overlay" [13:24:34 CET] <styler2go> Is it possible to use ffmepg as a rstp stream relay? (I receive a rtsp stream and provide it from there to other devices) It's to lower the data bandwith from my security camera to a lan-connected device [13:24:35 CET] <zerodefect> @furq Does the "overlay" not part of the 'target' ? [13:25:22 CET] <zerodefect> @furq Does the "overlay" string not belong to the 'target' parameter? [13:25:46 CET] <furq> i could only find one example of someone using it and it had the filter name in target and cmd [13:27:40 CET] <zerodefect> Are you able to share that example with me ? :) [13:27:59 CET] <furq> https://github.com/PromyLOPh/pianobar/blob/master/src/player.c#L99 [13:28:05 CET] <furq> it's not that enlightening really [13:29:08 CET] <jesopo> Hello guys [13:33:23 CET] <zerodefect> @furq Interesting that the filter's name is 'volume' and it looks like the property being adjusted is 'volume'. I think that is why they match [13:33:39 CET] <zerodefect> ...but its interesting that the property value goes in arg. [13:35:59 CET] <Projec[T]MayheM> I'm recording an rtsp stream in 1 minute chunks but some segments show a 1 hour plus 1 minute duration. with ffprobe I get this: [mpegts @ 0x2e86960] DTS discontinuity in stream 0: packet 31 with DTS 575999, packet 32 with DTS 337043505, the DTS jumps from 500k to 337M resulting in 1 hour jump. Anyone? [13:55:32 CET] <JEEB> Projec[T]MayheM: as I noted check the timestamps from the input, in the worst case use tcdump to dump the traffic to be able to replay it for testing purposes [13:55:53 CET] <JEEB> and yes, I have no idea how well the segmenting works [13:57:17 CET] <Projec[T]MayheM> thanks dude [14:17:47 CET] <styler2go> I am trying to receive a rtsp stream buzt ffmpeg keep sspamming: Past duration 6.499992 too large [14:17:50 CET] <styler2go> what does that mean? [14:34:12 CET] <TheoTheo> Is anybody familiar with the message: "Input pad "default" with type audio of the filter instance "in" of afade not connected to any source"?. Trying to get a fade filter working in C code using the libavfilter library [14:35:22 CET] <TheoTheo> Am using version 2.8.13 of FFmpeg btw [15:04:19 CET] <tommy``> guys how can i change framerate from 23.976 to 25 and stretch audio 0.95904 ? [15:07:47 CET] <Nacht> Is 'master_pl_name' something new of HLS ? [15:08:16 CET] <JEEB> the hls muxer has recently gotten a lot of new stuff by akamai and the chinaffmpeg folk [15:08:33 CET] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html#hls-2 [15:08:39 CET] <JEEB> see the options here [15:09:09 CET] <JEEB> TheoTheo: you haven't connected the filters yet? [15:09:10 CET] <Nacht> Yeah I was looking at it, and suddenly saw the master function. Couldn't recall seeing it before. I'm sure I was looking for it in the past [15:09:26 CET] <JEEB> yea, the hls muxer until very recently only did the per-profile playlists [15:11:38 CET] <TheoTheo> JEEB: Still can't figure out the order of things to make a filter graph that works for me [15:12:21 CET] <TheoTheo> Does a fade need to be preceded by a diffreent filter? [15:55:05 CET] <waldner> I'm using the following command to segment a stream coming from a camera: ffmpeg -i 'rtsp://.....' -c copy -f segment -segment_time 10 /tmp/capture-%03d.mp4 [15:55:15 CET] <tommy``> is t possible capture stream like amazon prime? [15:55:19 CET] <tommy``> with ffmpeg [15:55:33 CET] <waldner> however after writing 3 files, ffmpeg exits by itself without apparent error messages [15:55:38 CET] <waldner> am I doing something wrong? [15:57:04 CET] <JEEB> -v verbose or -v debug [16:00:28 CET] <waldner> this is how it ends with -v debug: https://pastebin.com/raw/ThySknxZ [16:00:41 CET] <waldner> the camera is a cisco IPC 6400 [16:02:28 CET] <waldner> in fact it stops after 30 seconds even without segmentation [16:02:32 CET] <waldner> just dumping to a file [16:02:40 CET] <waldner> (just tried it now) [16:07:40 CET] <waldner> ffmpeg version 3.4.1 Copyright (c) 2000-2017 the FFmpeg developers [16:10:18 CET] <waldner> if I select another camera channel it doesn't stop...I guess this can be a workaround for now [16:17:04 CET] <waldner> another thing, for old segments to be automatically deleted from disk is hls the only muxer? I can't find a similar option for -f segment [16:33:31 CET] <waldner> (segment_wrap is useless with -strftime 1) [00:00:00 CET] --- Wed Feb 7 2018

1 0

ffmpeg-devel.log.20180205
by burek 06 Feb '18

06 Feb '18

[00:17:18 CET] <Compn> Chloe : hmm wsl confuses me. it sounds like a very specialized elf binary would work, but it would have to be compiled anyways so this is basically just running a linux kernel in userspace ? [00:17:43 CET] <Compn> but "without any linux kernel code" lol [00:17:52 CET] <Compn> so its like a linux emulator ? [00:18:01 CET] <Chloe> Compn: it maps syscalls [00:18:04 CET] <Chloe> afaik [00:18:38 CET] <Chloe> wsl does just run windows binaries too (or it should) [00:18:40 CET] <DHE> "running a linux kernel in userspace" is UML [00:19:03 CET] <Chloe> so the idea was to be able to test FATE 'natively' [00:19:54 CET] <Compn> reminds me of colinux [00:21:28 CET] <kierank> michaelni: is there a reason why i couldn't make the argument to idct_pub void? [00:22:18 CET] <kierank> is it public api? [00:23:23 CET] <kierank> idct_put* [00:30:47 CET] <michaelni> kierank, i dont think its public, so it could be changed to void* ... i think [00:35:15 CET] <kierank> ok I might do that [01:11:27 CET] <kierank> seems too much effort actually have to chaange a ton of stuff [02:54:44 CET] <cone-415> ffmpeg 03Michael Niedermayer 07master:118e1b0b3370: avcodec/utvideodec: Fix bytes left check in decode_frame() [09:24:33 CET] <volkan> hi everyone [09:24:58 CET] <volkan> i have an question about continuously stream2file [09:25:22 CET] <volkan> i have an audio device who is play. mp2 files on disk [09:25:44 CET] <volkan> i wanna create mp2 files and put 1-2sec buffer and write microphone2file [09:26:03 CET] <volkan> is it possible [14:57:44 CET] <Chloe> Could some other people take a look at my add-new-iteration APIs set? Itd be nice to get some input from people other than someone who made a conflicting patch and Nicolas [15:00:42 CET] <durandal_1707> Chloe: others are very busy folks [15:01:55 CET] <Chloe> durandal_1707: so I can just push 7 days after patch submit date I guess? [15:09:31 CET] <atomnuker> just email the ml you'll push after a day if no one objects [15:10:42 CET] <Chloe> atomnuker: ok [15:12:06 CET] <wm4> I did that too but CE lashed out anyway [15:17:19 CET] <durandal_1707> Chloe: if there are no requests for changes, yes [16:51:55 CET] <hanna> re: http://ffmpeg.org/pipermail/ffmpeg-devel/2018-February/224911.html I'm sure that this patch is still bugged FWIW [16:52:07 CET] <hanna> I can reproduce A/V desync relatively often with loudnorm active (removing and reinserting fixes it) [16:52:20 CET] <hanna> but I don't know how to fix it any better [16:52:45 CET] <hanna> I guess maybe we should compute the delta and always add it to the s->pts? [16:53:00 CET] <JEEB> is this timestamp discontinuity stuff from the input? [16:53:10 CET] <JEEB> so you were at PTS 1024 at some point, and suddenly you get PTS 12? [16:53:11 CET] <hanna> I'm not sure [16:53:16 CET] <JEEB> (seek f.ex.) [16:53:23 CET] <hanna> I only observed it when watching e.g. long twitch videos [16:53:32 CET] <hanna> it's difficult to reproduce for sane files [16:53:49 CET] <hanna> but near the end of a 2-hour long twitch video it might be several seconds out of sync [16:53:50 CET] <JEEB> twitch IIRC is HLS, which has MPEG-TS, which has discontinuities [16:53:54 CET] <JEEB> hmm [16:53:58 CET] <hanna> then yeah I guess it's that [16:54:05 CET] <JEEB> but yea, it shouldn't be as quick as 2h [16:54:17 CET] <JEEB> MPEG-TS discontinuities happen at ~26h [16:54:31 CET] <JEEB> since the time base is 90000 and the timestamp field is 33bit [16:54:34 CET] <hanna> ah [16:54:45 CET] <sfan5> 26:29:40.39 to be exact [16:54:49 CET] <JEEB> :) [16:55:07 CET] <JEEB> of course it depends on how high you start muxing with to begin with [16:55:17 CET] <JEEB> but generally muxers start at zero [16:56:39 CET] <wm4> a good way is to use the latest PTS you got for input, and add/subtract all buffering [16:57:07 CET] <wm4> that's still incorrect for timestamp resets, but good enough and will recover quickly [17:00:46 CET] <hanna> I guess on a timestamp discontinuity you'd correctly need to flush the buffers and reset the filter? [17:01:05 CET] <hanna> although that also nukes internal state which might make sense to persist after a discontinuity [17:01:08 CET] <hanna> especially for small ones [17:01:33 CET] <nevcairiel> if anything it should really just copy input tiemstamps, a simple filter like that really has no business messing with timestamps [17:01:55 CET] <wm4> if it's something like dynaudnorm that buffers 15 seconds, it could matter, otherwise probably not [17:02:05 CET] <hanna> it might buffer variable amounts of input [17:02:19 CET] <wm4> nevcairiel: yeah but those copy all data to an internal ringbuffer, making timestamps hard to track [17:16:43 CET] <cone-765> ffmpeg 03Muhammad Faiz 07master:caa4bd7a9fe9: avcodec/me_cmp: remove ff_me_cmp_init_static() [18:31:43 CET] <SortaCore> sup young bois [00:00:00 CET] --- Tue Feb 6 2018

1 0

ffmpeg.log.20180205
by burek 06 Feb '18

06 Feb '18

[08:41:19 CET] <volkan> Hello guys [08:41:20 CET] <volkan> ffmpeg -f avfoundation -i ":0" -acodec mp2 -ab 32k -ac 1 ./AD/channel10-title2.adp [08:41:30 CET] <volkan> when i try to output this put error [08:41:36 CET] <volkan> Unable to find a suitable output format for './AD/channel10-title2.adp' [08:41:37 CET] <volkan> ./AD/channel10-title2.adp: Invalid argument [08:43:18 CET] <volkan> i find the answer [08:43:27 CET] <volkan> -f mp2 works now on output side [09:26:23 CET] <volkan> hi everyone [09:26:23 CET] <volkan> robUx4 has left IRC (Ping timeout: 256 seconds) [09:26:23 CET] <volkan> volkan [09:26:25 CET] <volkan> i have an question about continuously stream2file [09:26:27 CET] <volkan> volkan [09:26:29 CET] <volkan> i have an audio device who is play. mp2 files on disk [09:26:31 CET] <volkan> volkan [09:26:33 CET] <volkan> i wanna create mp2 files and put 1-2sec buffer and write microphone2file [09:26:35 CET] <volkan> volkan [09:26:37 CET] <volkan> is it possible [14:16:14 CET] <volkan> Hi , can any one try to write stream to file when i try to write file is locked, but uTorrent makes different things i don't know it's downloading and i'm watching same time [14:16:33 CET] <volkan> is it possible [14:17:35 CET] <BtbN> what? [14:17:49 CET] <Icedream> uTorrent downloads blocks of the file, you can't watch an incomplete file while it's being downloaded via a torrent [14:18:09 CET] <Icedream> you may be able to do that with HTTP-downloaded files because they are downloaded from beginning to end and not in random blocks [14:18:11 CET] <BtbN> you can, if the right blocks are downloaded in the right order. [14:18:25 CET] <BtbN> Some Torrent-Applications have a mode to download them in order [14:18:26 CET] <Icedream> yes but only rarely can clients be configured to do just that [14:18:49 CET] <BtbN> uTorrent is basically malware at this point in time though [14:20:02 CET] <volkan> is it possible on FFmpeg-cli [14:20:21 CET] <volkan> sudo ffmpeg -f concat -f avfoundation -i ":0" -acodec mp2 -ab 32k -ac 1 -f mp2 -b:v 1024k -bufsize 1024k ./AD/channel10-title2.adp [14:20:28 CET] <volkan> sorry not thi [14:20:31 CET] <volkan> s [14:20:55 CET] <volkan> on OSX "ffmpeg -f avfoundation -i ":0" -acodec mp2 -ab 32k -ac 1 -f mp2 ./AD/channel10-title2.adp" [16:10:31 CET] <microchip_> stevenli_: fix your shit [16:11:50 CET] <durandal_1707> stevenliu: fix your connection [16:44:06 CET] <Nacht> Hmm. This is interesting. I have a TS file created with FFMPEG , which has B-frames in it. So I check the file using MPEG-2 TS package analyser, and I see that the DTS flags are enabled, and that the DTS differs from the PTS. Which is to be expected. Yet when I use FFprobe, it says the DTS times are equal to that of the PTS. [16:46:22 CET] <DHE> depends how far you inspect. only B-frames will have this "issue" and ffprobe may not scan the whole file, or you might not have checked far enough [16:47:35 CET] <Nacht> It doesn't have to scan the whole file. The first packet would already be enough. [16:47:47 CET] <Nacht> ffprobe says; frame,133200,133200 [16:47:50 CET] <kepstin> but the first packet might not have any B frames. [16:48:04 CET] <Nacht> MPEG-TS analyser says PTS: 133200 DTS: 126000 [16:48:18 CET] <Nacht> And I checked the bytes, the analyser is correct [16:49:20 CET] <JEEB> Nacht: yea that seems like a thing in the mpegts demuxer; if you can figure it out I recommend posting a patch, otherwise a sample in a ticket [16:49:27 CET] <JEEB> I've seen similar things with just demuxing [16:49:37 CET] <JEEB> a lot of timestamp fixing seems to be done during decoding [16:50:10 CET] <JEEB> as in, if your workflow demuxes and then decodes your timestamps are OK, but if you just demux MPEG-TS you can possibly have fun [16:50:46 CET] <Nacht> I can't say I know much C, but I'll have a look at the code and see if I can find something [16:51:11 CET] <JEEB> libavformat/mpegts.cd [16:51:12 CET] <JEEB> libavformat/mpegts.c [16:51:22 CET] <Nacht> JEEB: Cheers mate. Ill have a look [16:51:23 CET] <JEEB> I also have some TODOs regarding it [16:51:36 CET] <JEEB> but I just have too many things "I should get to at some point" [16:51:38 CET] <JEEB> :V [16:51:50 CET] <Nacht> JEEB: Life would be boring if you hadn't :) [17:13:34 CET] <slasktrat> Hi. I have a strange issue with remuxing a multicast transport stream. All is good as long as there is no dvb_subtitles involved, but as soon as I include a dvbsub track the output is unwatchable (both picture and audio is stuttering and I get a lot of artifacts). Command: ffmpeg -i udp://@239.1.1.1:1234 -map 0 -c copy -f mpegts udp://@239.2.1.1:1234 [17:15:02 CET] <slasktrat> I have also tried to re-encode the dvbsubs, but same result. [17:15:52 CET] <slasktrat> If/when there is present subs for a longer period (more than a couple of seconds) the stream is stable while the subs are present [17:16:12 CET] <slasktrat> Seems to be same issue as reported here: http://trac.ffmpeg.org/ticket/6165 btw [17:34:58 CET] <kazuma_> slasktrat ffmpeg cannot handle dvbsubs [17:35:03 CET] <kazuma_> or any image based subs [17:35:34 CET] <kazuma_> you would need to ocr them to subrib with another application, then mux them back in [17:35:48 CET] <kazuma_> subrip* [17:37:43 CET] <volkan> Hi, i have an question about audio files while i'm listening audio can ffmpeg write over on the file input is stream output is an file [17:38:34 CET] <kazuma_> im not sure i understand the question [17:38:44 CET] <kazuma_> you want to re stream audio? [17:40:44 CET] <volkan> i don't know what is re stream but i wanna do basically stream2file i can do it but file is locked [17:41:13 CET] <volkan> can FFmpeg write block an block on the audio file [17:47:49 CET] <kepstin> volkan: the file locking when open thing is a windows peculiarity; playing files as they're written by ffmpeg works fine on other operating systems (assuming the container supports it) [17:48:20 CET] <slasktrat> kazuma_: I don't want ffmpeg to do anything with the dvbsubs, except pass them through. and it works in the term of that I can see the subtitles in the remuxed stream, but the stream is corrupt somehow so playback is full of artifacts [17:48:41 CET] <slasktrat> If I dump to .ts file instead of multicasting everything works also [17:48:54 CET] <volkan> i use Macintosh [17:49:50 CET] Action: kepstin doesn't know enough about mac to know what could cause that to work/not work. [17:50:11 CET] <volkan> my command is [17:50:50 CET] <volkan> ffmpeg -f avfoundation -i ":0" -acodec mp2 -ab 32k -ac 1 -f mp2 out.mp2 [17:50:53 CET] <kepstin> volkan: basically, ffmpeg isn't doing anything to lock the file, it just opens it and writes to it. [17:51:18 CET] <volkan> can ffmpeg write an block by block so i can open file [17:52:34 CET] <volkan> some torrent client's write over file and i can watch [18:09:22 CET] <goraj_> Hi, can you recommend me a software to use as a subtitle editor or to label video sequences? [18:10:02 CET] <goraj_> I guess someone here has probably done this in the past [18:10:56 CET] <slasktrat> goraj_: http://www.nikse.dk/subtitleedit [18:11:29 CET] <JummerBimmer> hey hey i found this channel on ffmpeg.org [18:11:45 CET] <JummerBimmer> i got some problems with moving watermark i dont understand how do i move it across the video i want to be [18:11:48 CET] <JummerBimmer> can anyone help? [18:12:52 CET] <goraj_> thank you slasktrat [18:13:00 CET] <slasktrat> np [18:13:06 CET] <JummerBimmer> ain_w-overlay_w-10:30 [out] ,i change numbers and it barely moves up i dont get it,what if i want to watermark in left side of the video [18:13:24 CET] <JummerBimmer> or lets say move it right further,none of these 2 numbers change seem to move it to left or right,just up.. [18:14:26 CET] <JummerBimmer> been sitting 3 hours changing those numbers lol [18:14:54 CET] <JummerBimmer> or numbers go from 0 to lets say 999? [18:14:59 CET] <JummerBimmer> i change like 20 30 max [18:32:58 CET] <SortaCore> C++ use nvenc how [18:45:50 CET] <SortaCore> C++ use nvenc how?!! [18:45:58 CET] <SortaCore> also steven what's up with your connection [18:48:01 CET] <DHE> not really any different from any other codec iirc [18:49:31 CET] <SortaCore> it's hwaccel'd though [18:49:37 CET] <SortaCore> so when I'm transcoding, say from RTSP to file [18:49:58 CET] <SortaCore> I have cuda, cuvid, dxva2 hwaccels [20:10:03 CET] <prelude2004c> hey everyone, looking for a bit of help.. i'm using cuvid to decode however the source seems interlaced and I'm having a hard time getting it to deinterlace on the nvidia card ( decoder ) .. it seeems c:v h264_cuvid -deint 2 only works for h264 source. [20:10:31 CET] <prelude2004c> any way to deinterlace mpeg2 source ? I tried on the transcoding with yadif but kills the CPU quite a bit. [20:12:43 CET] <c_14> mpeg2_cuvid also has that option it seams [20:25:50 CET] <prelude2004c> do i have to enable that i guess when i compile ? [20:25:52 CET] <prelude2004c> probably i guess right [20:33:26 CET] <prelude2004c> c_14, i don't have any mpeg2_cuvid option [20:33:43 CET] <prelude2004c> when i do ffmpeg -decoders |grep cuvid [21:06:33 CET] <c_14> maybe update your ffmpeg? [21:06:38 CET] <c_14> it's there for me [21:16:44 CET] <prelude2004c> ya but then i have other issues.. i will try [22:08:55 CET] <ddubya> Is there a QFile-like object that gives a detatchable file handle? I would like to use QFile::open() but keep the file around when QFile is destructed [22:09:09 CET] <ddubya> I mean keep it open (the file handle) [22:10:32 CET] <volkan> hey everyone, i wanna ask something [22:10:41 CET] <volkan> can ffmpeg write on file without locking [22:11:02 CET] <volkan> i wanna do something like this ffmpeg -f avfoundation -i ":0" -acodec mp2 -ab 32k -ac 1 -f mp2 out.adp [22:11:15 CET] <JEEB> it depends on what the file access APIs on a specific system [22:11:35 CET] <JEEB> I know that the standard windows fopen etc APIs will lock a file [22:11:46 CET] <JEEB> linux doesn't [22:11:53 CET] <JEEB> macos I have no idea about :P [22:12:19 CET] <volkan> Some torrent client's have good feature it downloading and i can watch [22:12:50 CET] <volkan> when i try to read file incomplete file header error [22:13:42 CET] <JEEB> anyways, no idea if macos locks your file with a normal fopen [22:13:49 CET] <JEEB> which avfoundation sounds like :P [22:15:03 CET] <volkan> i use unix for read operation [22:15:15 CET] <volkan> mac write shared folder which is mounted SMB [22:20:16 CET] <volkan> i think ffmpeg write buffer and then writes all buffer to disk [22:20:25 CET] <volkan> first,Buffer and last disk [22:20:36 CET] <JEEB> no [22:21:00 CET] <JEEB> with some containers the index comes last, but pretty sure when the muxer actually writes buffers, they get written [22:22:07 CET] <volkan> i dont use muxer , can muxer help me ? [22:22:30 CET] <JEEB> whatever you think -f mp2 does mate [22:22:57 CET] <volkan> 1 min i recorded my sound and when i try to ls -al command to show how much kb , out put is zero byte [22:24:05 CET] <JEEB> that can be due to a lot of stuff - including your OS just not having the size field updated [22:26:13 CET] <JEEB> also just checked on windows; I can play welp.mp2 just fine [22:26:35 CET] <JEEB> so it seems like the windows-specific APIs got used at some point [22:26:59 CET] <JEEB> but the file size is 0KiB until I refresh things specifically [22:27:49 CET] <volkan> can you post the test field commands [22:28:08 CET] <volkan> if windows help me i choice windows :) [22:28:27 CET] <JEEB> maybe you should rather see if mac locks your files or not actually [22:28:32 CET] <JEEB> http://up-cat.net/p/7cc2b5f1 [22:28:49 CET] <JEEB> very basic test command which I can't believe I even went through the effort to poke [22:31:49 CET] <volkan> i found the solution [22:32:44 CET] <volkan> i add at the end of command "-report" [22:36:01 CET] <JEEB> I seriously doubt that is what fixed anything you had as a problem, but if that makes you happy I am fine with that. [22:38:25 CET] <volkan> i know i missed somethings [22:39:01 CET] <volkan> ffmpeg writes every 256b data i can listen what i recorded [00:00:00 CET] --- Tue Feb 6 2018

1 0

ffmpeg-devel.log.20180204
by burek 05 Feb '18

05 Feb '18

[00:34:20 CET] <cone-481> ffmpeg 03Muhammad Faiz 07master:3caecf7ce8e9: avcodec: do not use init_static_data on some codecs [11:57:33 CET] <volkan> hello guys [11:57:33 CET] <volkan> i have and question about pcm_f32le 2 wav convert [11:57:41 CET] <volkan> can someone help me [12:08:27 CET] <volkan> when i try to convert i failed it [12:08:27 CET] <volkan> Assertion failed: (el >= 0), function calc_energy, file psymodel.c, line 576. [13:57:53 CET] <cone-804> ffmpeg 03Jun Zhao 07master:0519e66a90f0: examples: Add a VA-API transcode example. [14:49:44 CET] <Joobla> hello,does anyone know how to put watermark at 90% degree? [14:50:39 CET] <Joobla> or 60% or whatever,how to lean watermark to go from top to bottom is it possible? [15:02:56 CET] <DHE> not appropriate for this channel [15:05:25 CET] <Compn> Joobla : try #ffmpeg [15:05:33 CET] <Compn> this channel is for development only [15:23:10 CET] <cone-804> ffmpeg 03wm4 07master:ff46124b0df1: rtsp: rename certain options after a deprecation period [15:23:12 CET] <cone-804> ffmpeg 03wm4 07master:48bc9fffd11c: id3v2: fix unsynchronization [18:32:54 CET] <KGB> [13FFV1] 15michaelni closed pull request #103: len_count related descriptions (06master...06len_count) 02https://git.io/vNpU8 [19:42:36 CET] <Chloe> I've been playing with getting FATE to run mingw builds under WSL, but https://github.com/ffmpeg/ffmpeg/blob/master/libavformat/file.c#L135 can't seem to resolve /mnt/d etc (windows drive mounts). I haven't been able to get a debugger to work under WSL (for debugging .exes) [19:43:28 CET] <nevcairiel> you are basically creating a windows executable, it wont see WSL paths [19:43:33 CET] <Chloe> ffmpeg.exe can, however, resolve windows paths directly under linux [19:43:58 CET] <Chloe> nevcairiel: yeah [19:44:44 CET] <Chloe> So I was wondering, could I do something which replaced '/mnt/d/' etc with 'D:/' if a fate-wsl flag was set in configure or something [19:45:34 CET] <nevcairiel> cant you symlink the fate samples so that its in the same relative path or something [19:45:56 CET] <Chloe> fate seems to want to use absolute paths [19:46:02 CET] <Chloe> though relative paths would work [19:46:54 CET] <nevcairiel> my samples dir is relative in my fate config at least [19:50:49 CET] <Chloe> nevcairiel: ill try an out-of tree build [21:56:52 CET] <Chloe> now WSL doesnt even want to run the compiled ffmpeg.exe (but it runs outside of WSL), maybe this is more pain than it's worth for now [21:58:01 CET] <Chloe> current diff if anyone cares about WSL https://0x0.st/scz5.diff [00:00:00 CET] --- Mon Feb 5 2018

1 0

ffmpeg.log.20180204
by burek 05 Feb '18

05 Feb '18

[02:47:25 CET] <FishPencil> Do bluray disc's use YUV420p? [02:51:29 CET] <kepstin> FishPencil: yes (at least the h264 stuff does, I think it's restricted to that by the permitted profiles) [02:52:39 CET] <FishPencil> Is YUV420p the most widely used color encoding method? [03:02:13 CET] <kepstin> FishPencil: for computer video, yes (or formats with the same sampling, but different memory arrangements, like NV12 or YUYV) [03:39:16 CET] <DHE> FishPencil: yes, unfortunately. some players (Quicktime, TVs with media support) just don't support the higher stuff [06:20:24 CET] <FishPencil> Is there a better way to pack in that information without losing quality? 420p seems pretty good [08:57:50 CET] <Bear10> By any chance does anyone know the correct job title for someone experienced in multimedia / rtmp / ott? [08:58:16 CET] <Bear10> Basically someone who knows the ins and outs to make a streaming platform [09:58:50 CET] <TaZeR> ffmpeg is pretty good, are there competing codecs though? [10:03:15 CET] <durandal_1707> TaZeR: ffmpeg is not codec, its library [10:03:55 CET] <TaZeR> oh, is there another competing one that i would be familiar with maybe? [10:04:22 CET] <TaZeR> they usually have such catchy names [10:04:31 CET] <TaZeR> well ffmpeg is catchy to me lol [10:04:44 CET] <durandal_1707> TaZeR: opensource? no [10:05:06 CET] <TaZeR> oh i see [10:25:33 CET] <furq> TaZeR: it depends what you want to do [10:25:40 CET] <furq> there's nothing else that does as much as the ffmpeg libs can [10:26:18 CET] <furq> but if you e.g. just want to encode rawvideo frames to vp9 then you might just want to use libvpx [10:26:51 CET] <furq> or vpxenc if you're talking about the cli [11:43:42 CET] <volkan> hello guys [11:44:09 CET] <volkan> i have and question about pcm_f32le 2 wav convert [12:08:39 CET] <volkan> guys please look at this message [12:08:39 CET] <volkan> Assertion failed: (el >= 0), function calc_energy, file psymodel.c, line 576. [12:08:44 CET] <volkan> what is thiss ? [12:36:16 CET] <yusa> Dont know, what are compiling and how and on which maschine? [12:38:13 CET] <yusa> probably you forwarded a parameter greater or equal than zero to a function which expects negative values for this specific parameter :) [12:58:26 CET] <volkan> i only use ffmpeg-cli tools [13:08:17 CET] <atomnuker> upload the file somewhere [13:08:40 CET] <atomnuker> also you're converting to aac for some reason [13:18:03 CET] <volkan> hi @atomnuker [13:24:03 CET] <volkan> i uploaded to dropbox link is https://www.dropbox.com/sh/70dwd368tp9yx4z/AAC05LwVXSSpleNuIQx6qnG-a?dl=0 [14:24:57 CET] <Joobla> hello,could anyone please help me with ffmpeg? [14:25:18 CET] <volkan> what do u want to [14:25:31 CET] <Joobla> i want to position watermark and it should lean not be in straith line i have no idea how to do it :( [14:25:59 CET] <volkan> what type of watermark like soundriver ? [14:27:11 CET] <Joobla> watermark image [14:35:29 CET] <volkan> this can be possible but you need to watermark image which is transparent background [14:54:01 CET] <Joobla> soo anyone knows how to put watermark a bit rotated? [14:54:12 CET] <Joobla> i have transparent background mage [14:54:32 CET] <Joobla> problem in settings is overlay=main_w-overlay_w-5:main_h-overlay_h-2 [out] [14:54:39 CET] <Joobla> i can move and down left and right np with this [14:54:52 CET] <Joobla> but how i rotate watermark a little angle why some %.. [14:55:08 CET] <volkan> i think rotate png file or use imageMagick [14:55:30 CET] <volkan> pr write an script to for it [15:00:32 CET] <furq> !filter rotate @Joobla [15:00:32 CET] <nfobot> Joobla: http://ffmpeg.org/ffmpeg-filters.html#rotate [15:18:03 CET] <Joobla> @furq l [15:18:14 CET] <Joobla> where do you put the code? [15:18:40 CET] <Joobla> i try imagemagic too [15:22:17 CET] <Joobla> that feeling when you need to rotate image in ffmpeg and it feels like rocket science - https://www.youtube.com/watch?v=rQSmfR1_SVE [18:05:46 CET] <Joobla> can anyone help me with some basic ffmpeg? [18:10:54 CET] <zort> can anybody tell me what's wrong with my command? https://pastebin.com/CXGz7bgw [18:14:18 CET] <zort> this works fine: https://pastebin.com/Bix7wpJP [18:15:27 CET] <zort> oh mp3 can't contain aac... [19:26:41 CET] <rapha> hi! [19:30:33 CET] <rapha> i've the exact same problem as the guy at https://superuser.com/questions/538829/repairing-corrupt-mp4 (even same scenario, same error message -"moov atom not found"-), alas the file is 6.2GB in size, so the apparent solution using `untrunc` does not work. is it still possible to attach the final metadata block somehow? [19:32:13 CET] <BtbN> probably not [19:33:07 CET] <rapha> hmm [19:36:10 CET] <rapha> that sounds unlikely unless Trump has now found a way to discriminate against rallies of bytes too large to his likings [19:36:22 CET] <SortaCore> cut the file in half byte size and add mov atom to both [19:36:29 CET] <SortaCore> could be >4GB size is killing a seek op? [19:36:56 CET] <BtbN> what? [19:37:02 CET] <BtbN> mp4 does not have a 4GB limit [19:37:09 CET] <SortaCore> yea [19:37:14 CET] <BtbN> it's just a damaged mp4 file [19:37:22 CET] <BtbN> mp4 is 100% useless without a proper moov atom [19:37:28 CET] <SortaCore> yea [19:39:05 CET] <kazuma_> what is moov atom? [19:39:24 CET] <SortaCore> basically metadata about how to read the video, normally appended at the end of the mp4 [19:39:36 CET] <SortaCore> if a mp4 recording is cut off, then the atom isn't added, so the file becomes unreadable [19:39:50 CET] <kazuma_> i see [19:43:05 CET] <SortaCore> my suspicion is whatever is trying to read or add the mov atom is using a 32-bit seek func and messing up [19:43:13 CET] <SortaCore> but either way the atom's missing [19:43:55 CET] <SortaCore> I see in the comment untrunc doesn't work for large files [22:46:31 CET] <BaroqueLarouche> Hi! Ive been trying to get libav and friends to work in my C++ application and I cant get it to work. I am trying to generate a H.264 video with audio generated from a WAV file + frame grab from OpenGL using openFrameworks. All I get is grabled audio and black images. Can anyone help me figuring out whats wrong ? [22:46:33 CET] <BaroqueLarouche> https://github.com/mlarouche/PlaylistVideoGenerator/blob/master/src/VideoWr… [22:46:53 CET] <BaroqueLarouche> ^ Heres the libav usage part in my code [00:00:00 CET] --- Mon Feb 5 2018

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ffmpeg-devel.log.20180203
by burek 04 Feb '18

04 Feb '18

[00:17:45 CET] <atomnuker> huh, a patch from someone with a @google.com address [00:18:19 CET] <BtbN> there are a lot of them [00:19:02 CET] <atomnuker> last one I remember was the lavf mp4 editing thing [00:26:19 CET] <Compn> atomnuker : google has several places they use ffmpeg in. at least google video, google docs, google drive... also whatever security fuzz oss testing they are doing in secret [00:26:24 CET] <Compn> plus youtube team [00:28:43 CET] <jamrial_> And Chromium [00:29:54 CET] <Compn> oh yeah them :D [00:30:09 CET] <Compn> and android [00:30:37 CET] <JEEB> chromium seems like the bigges thing, which also seems to have prompted the fuzzing [00:55:37 CET] <cone-117> ffmpeg 03Jérôme Martinez 07master:e3d946b3f48f: avcodec/ffv1enc: mark RGB48 support as non-experimental [01:41:29 CET] <Chloe> why is this guy trying to optimise iteration of components [01:41:42 CET] <Chloe> is O(k) lookup really that bad [02:04:16 CET] <SortaCore> I guess it's not O(k) with him [02:04:26 CET] <DHE> *groan* [12:56:36 CET] <wm4> michaelni: I've caught swresample corruption memory/crashing with very large compensation distance in mpv (not sure if my fault or libswresample) [13:02:53 CET] <michaelni> will look later, but how can i reproduce it ? [13:27:10 CET] <wm4> michaelni: mpv 196950e905, mpv something.mkv --no-config --audio-pitch-correction=no, hit [ very quickly during playback (or just keep it pressed) [13:30:58 CET] <wm4> also resampling with an extreme ration (100x smaller output rate than input) and then feeding a small frame leaves you with no returned sample data, even if the intention is to drain the resampler instead of feeding more data into it [14:10:21 CET] <JEEB> was there a patch posted for libaom for FFmpeg yet? [14:10:28 CET] <JEEB> I can find one for the IVF muxer but that's it [14:10:55 CET] <RiCON> JEEB: not for ffmpeg, only for libav, iirc [14:11:47 CET] <JEEB> right [14:16:43 CET] <jamrial> JEEB: BBB ported/updated it here https://github.com/rbultje/ffmpeg/commits/libaom_av1_decoder [14:23:06 CET] <JEEB> cool [14:40:46 CET] <JEEB> jamrial: nice, I just forward-ported it to current HEAD, let's see if this works with the latest libaom [14:47:57 CET] <JEEB> seems to work, nice [14:54:46 CET] <JEEB> oh huh [14:55:00 CET] <JEEB> it just silently took my enable-libaom [14:55:08 CET] <JEEB> but then failed in the check :/ [14:55:55 CET] <RiCON> configure seems to do that a lot [14:56:11 CET] <wm4> configure turns out to be a fragile mess! [14:56:36 CET] <RiCON> or silently disable explicitely enabled libs missing some deps [15:02:30 CET] <JEEB> ok, now I know why it fails [15:02:40 CET] <JEEB> it doesn't add the include flags for some reason from pkg-config [15:03:15 CET] <JEEB> right, the pkg-config based check is disabled in the patch [15:06:55 CET] <jamrial> it's using check_lib() instead of require() or require_pkg_config() [15:07:01 CET] <jamrial> which doesn't abort on failure [15:07:17 CET] <JEEB> yup [15:10:15 CET] <wm4> rcombs: FOSDEM just had a stream with a talk about HEIF, and there was someone who asked the presenter "are you happy with the format?" and mentioned your patch [15:11:05 CET] <kierank> wm4: you should come to FOSDEM [15:11:14 CET] <wm4> too late [15:11:43 CET] <nevcairiel> FOSDEM somehow always happens to be on a vacation weekend for me, I just back back an hour ago from denmark, last year I was in australia =p [15:12:01 CET] <nevcairiel> got back* [15:13:00 CET] <JEEB> grr, need to remember the parameter order for require_pkg_config [15:13:31 CET] <nevcairiel> just do what everyone does and copy some other library [15:13:33 CET] <JEEB> ok, the internal name, the pkg-config name [15:13:45 CET] <JEEB> and header and func [15:13:47 CET] <JEEB> ooh-kay [15:16:07 CET] <wm4> man HLS live streams always either fall behind or underrun (due to being too far ahead) for me [15:16:49 CET] <JEEB> yayifications, that worked [15:16:55 CET] <JEEB> let's see if I can make it now [15:17:08 CET] <JEEB> muh battery [15:20:35 CET] <JEEB> ok, now I finally got to libaom API having changed :) [15:20:41 CET] <JEEB> and my battery is dying [15:20:47 CET] <JEEB> so I guess this is most of it for today [15:29:25 CET] <jamrial> so the wrapper is already outdated? :p [15:29:43 CET] <JEEB> yea [15:29:51 CET] <JEEB> probably just arg count thing [15:30:08 CET] <jamrial> no point bothering then until they tag a stable release [15:31:13 CET] <JEEB> yea [15:31:56 CET] <BBB> JEEB: I disabled the pkg-config because Im using uninstalled libs [15:31:59 CET] <BBB> sorry for not documenting that [15:32:08 CET] <BBB> my use case is kind of weird that way [15:32:29 CET] <JEEB> BBB: it was quick so np [15:35:15 CET] <atomnuker> Tempted to see how many frames per day I get with using libaom on my first gen rpi but don't want to contribute industrial amounts to global warming [15:35:49 CET] <RiCON> wouldn't the rpi be more "frame per month"? [15:39:40 CET] <TD-Linux> realtime with rav1e :^) [16:05:40 CET] <JEEB> ok, this seems to work with current libaom HEAD https://github.com/jeeb/ffmpeg/commits/libaom_av1_decoder [16:22:41 CET] <funman> atomnuker: you did not upload your slides on fosdem? [17:29:38 CET] <durandal_1707> ffmpeg description should also contain word filter, not just stream and convert [17:37:17 CET] <BBB> JEEB: ty [17:37:32 CET] <BBB> TD-Linux: x264 preset=ultrafast is also realtime, doesnt mean its good :-p [17:39:01 CET] <RiCON> even medium is more or less realtime, iirc [17:41:20 CET] <DHE> that's highly CPU dependent though. my core i7, sure for not-too-busy action [18:03:59 CET] <atomnuker> funman: I did, check the talk's page [18:42:00 CET] <rcombs> wm4: lol link? [18:43:23 CET] <wm4> rcombs: I think recordings come out later [18:43:28 CET] <rcombs> ah [18:43:56 CET] <rcombs> I'm internet famous \o/ [18:44:07 CET] <wm4> he didn't say your name [18:44:16 CET] <wm4> just that your patch didn't work with most of their samples [18:44:29 CET] <rcombs> well yeah, 'cause tiling [18:44:39 CET] <rcombs> always knew bitching about shitty format design would get me places [18:45:02 CET] <durandal_1707> nobody mentioned me? :( [18:45:23 CET] <rcombs> have you bitched about shitty format design in an HEIF patch's commit message [18:55:22 CET] <durandal_1707> atomnuker: dont get drunk! [19:14:24 CET] <KGB> [13FFV1] 15dericed opened pull request #102: support method to output lines only to version0,1,3 or version4 (06master...06split-ffv1-version-outputs) 02https://git.io/vNxce [22:53:52 CET] <cone-481> ffmpeg 03Nikolas Bowe 07master:c86490c49a3d: avfilter/af_join: Fix crash in join filter [22:53:52 CET] <cone-481> ffmpeg 03Michael Niedermayer 07master:8e50bd61e4ff: avcodec/wavpack: Fix integer overflow in FFABS [00:00:00 CET] --- Sun Feb 4 2018

1 0

ffmpeg.log.20180203
by burek 04 Feb '18

04 Feb '18

[00:03:58 CET] <jfmcarreira> heyy guys [00:04:13 CET] <jfmcarreira> which is the function of avutils to convert AVFrame pixel format? [00:05:00 CET] <ddubya> swscale or avfilter functions [00:05:30 CET] <jfmcarreira> ddubya: which one? cant find it by googling it [00:05:47 CET] <ddubya> more than one... [00:09:54 CET] <jfmcarreira> seems like a very simple thing to have several function for it :D [00:18:49 CET] <ddubya> haha melt hacked to use cuvid [00:20:01 CET] <ddubya> jfmcarreira, jfmcarreira ffmpeg has to supports a gazillion codecs, pixel format etc etc. Given that, it doesn't seem too crazy [00:22:53 CET] <jfmcarreira> ddubya: fair enoug. I was looking at one of the examples and if i understand correctly i have to create a sws context and then call the the sws function [00:23:47 CET] <JEEB> with swscale directly, yes [00:24:08 CET] <jfmcarreira> ddubya: actually i am developing a video player and I took the pixel description idea from ffmpeg. seems quiet nice the way ffmpeg is able to use the same function for a series of different formats. however as I cant implement all i need to convert some of them to know fmts like YUV420p [00:24:11 CET] <JEEB> I prefer avfilter myself nowadays since it handles the whole avframe side of things [00:24:31 CET] <JEEB> because what you get from the decoder is an AVFrame [00:24:46 CET] <JEEB> and usually if you need to pass things around into encoders etc those take AVFrames in [00:25:10 CET] <ddubya> avfilter is cleaner, no need for avpicture junk [00:25:15 CET] <JEEB> https://github.com/jeeb/matroska_thumbnails/blob/master/src/matroska_thumbn… [00:25:21 CET] <JEEB> this is my old swscale usage [00:25:28 CET] <JEEB> nowadays I use avfilter [00:25:43 CET] <JEEB> (avfilter also lets you do deinterlacing etc) [00:27:15 CET] <jfmcarreira> JEEB: ddubya so i will go for avfilter. since I am already able to convert AVFrame to my own class where I save the pixel values. I just need to convert AVFrame to a known format [00:28:23 CET] <jfmcarreira> JEEB: the example you shown is wuth avfilter? [00:28:33 CET] <ddubya> yep just match format to display format [00:29:20 CET] <JEEB> jfmcarreira: there are avfilter examples in doc/examples [00:29:22 CET] <JEEB> of FFmpeg [00:29:31 CET] <JEEB> not perfect, but some sort of filtering is there [00:29:46 CET] <JEEB> also I recommend looking at the avfilter doxygen @ ffmpeg.org [00:30:45 CET] <jfmcarreira> ddubya: normally i do not want to do that because the app is suppose to be used for video processing so I want to shown things as in source (but I am not being able to do so for every piece of fmt :( ) [00:31:53 CET] <ddubya> jfmcarreira, well you can convert to a wider gamut format for display. RGB is OK I guess [00:32:02 CET] <hojuruku> Cu5tosLimen: i got timecode problems with what the vaapi ffmpeg driver / mesa / libva is creating vs libx264 [00:32:22 CET] <hojuruku> JEEB: knows the story, what software do i have to use to analyse the corrupted mp4 / mkv files image streams? [00:32:45 CET] <JEEB> for mkv mkvinfo [00:32:49 CET] <JEEB> it can output the timeline [00:33:01 CET] <JEEB> for mp4 there are also various tools [00:45:44 CET] <SortaCore> so err, it was decode with hw_device_ctx dxva2, then encode with derived device context nvenc right? [00:47:03 CET] <jfmcarreira> where can I get a list of avfilters? [00:48:17 CET] <kepstin> jfmcarreira: https://www.ffmpeg.org/ffmpeg-filters.html is a good place to start. [00:48:37 CET] <kepstin> if you're asking how to get a list of filters available in *your particular ffmpeg build* there's an api for that. [00:49:22 CET] <kepstin> jfmcarreira: but for general format conversion and whatnot, you're probably just gonna need the scale filter (and maybe format, depending how you're using the api) [00:55:58 CET] <jfmcarreira> kepstin: i think that list helps [00:56:23 CET] <jfmcarreira> i guess i found what I need [00:56:30 CET] <jfmcarreira> is the "format" filtet [00:57:30 CET] <kepstin> note that the 'format' filter doesn't do much on its own - it just requests a particular format at a spot in the filter chain, and then a (possibly auto-inserted) scale filter before it will do the actual conversion. [00:57:45 CET] <jfmcarreira> :( [01:02:07 CET] <jfmcarreira> yeah I tested in command line and I converted from PAL to BGR and I asked for YUV420 [01:05:53 CET] <jfmcarreira> cant I request the video decoder to output AVFrame in a specific pixel format? [01:24:13 CET] <prelude2004c> hey guys..quick question.. i have some filter like -vf fps=xx.xx,scale_npp,etc... i want to had motion Interpolation [01:24:19 CET] <prelude2004c> not sure where to add it or what it should look like [01:25:19 CET] <sfan5> replace the fps filter with minterpolate [01:30:56 CET] <prelude2004c> i tried just -vf minterpolate=fps=60 and nothing else and it says : Impossible to convert between the formats supported by the filter 'graph 0 input from stream 0:0' and the filter 'auto_scaler_0' [01:31:59 CET] <sfan5> don't you need a "hwupload" before scale_npp? [01:32:48 CET] <DHE> which tells me frame is probably a hardware frame so really it needs downloading first [01:32:57 CET] <prelude2004c> right nwo i am using : -vf 'a53_split,fps=29.97,a53_combine=ratio=5,scale_npp=w=1280:h=720' [01:33:23 CET] <prelude2004c> and i want to do some motion Interpolation to fix the 29.97 to smooth video closer to 60fps [01:34:19 CET] <prelude2004c> at least i'm reading on that is how it's done.. just not sure best way to make it wokr [01:34:20 CET] <prelude2004c> at least i'm reading on that is how it's done.. just not sure best way to make it work [01:47:39 CET] <prelude2004c> anyone? [01:47:44 CET] <prelude2004c> i'm stuck [01:49:24 CET] <SortaCore> put hwupload before scale_npp and hwdownload after, you might not need the hwupload one [01:55:54 CET] <prelude2004c> oh.. ok trying [01:56:54 CET] <prelude2004c> like this ? ..._combine=ratio=5,scale_npp=w=1280:h=720,hwdownload,minterpolate=fps=60 [01:57:30 CET] <prelude2004c> Impossible to convert between the formats supported by the filter 'Parsed_minterpolate_5' and the filter 'auto_scaler_0' [02:00:44 CET] <prelude2004c> i tried both with hwupload and hwdownloa and just hwdownload and i had the same issue occur [02:03:49 CET] <SortaCore> hwdownload after minterpolate [02:04:27 CET] <prelude2004c> same error [02:04:27 CET] <prelude2004c> Impossible to convert between the formats supported by the filter 'Parsed_scale_npp_4' and the filter 'auto_scaler_0' [02:04:42 CET] <prelude2004c> .... ,a53_combine=ratio=5,hwupload,scale_npp=w=1280:h=720,minterpolate=fps=60,hwdownload' [02:05:02 CET] <prelude2004c> -vf 'a53_split,fps=29.97,a53_combine=ratio=5,hwupload,scale_npp=w=1280:h=720,minterpolate=fps=60,hwdownload' [02:18:16 CET] <prelude2004c> :( still nothing.. [02:18:46 CET] <prelude2004c> tried just hwupload,minterpolate=fps=60,hwdownload [02:19:03 CET] <prelude2004c> it gave me some error [hwupload @ 0x3d203c0] A hardware device reference is required to upload frames to. [02:19:14 CET] <prelude2004c> tried without upload.. still errors out.. nothing i do seems to work [02:44:17 CET] <prelude2004c> no other ideas ? i'm using cuvid for decoding [02:44:53 CET] <DHE> we know. we suspected that the hardware decoding is actually making the software filtering act up [02:46:01 CET] <prelude2004c> any other suggestion? for some reason source for video is coming in at 29.97 but in fast motion or text scrolling its stuttering like crazy [02:46:12 CET] <prelude2004c> i want to smooth the video somehow so its more fluid [02:46:22 CET] <prelude2004c> is interpolation the only way to do it? [02:46:29 CET] <DHE> interlaced? [02:47:18 CET] <prelude2004c> no [02:48:41 CET] <prelude2004c> mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc [02:58:04 CET] <DHE> prelude2004c: ffprobe (source) -show_frames | grep interlaced [02:58:42 CET] <furq> and/or ffmpeg -i foo -vf idet -f null - [02:59:56 CET] <DHE> nice [03:00:12 CET] <furq> obviously if it's flagged as interlaced you don't need to bother with that [03:00:59 CET] <furq> maybe a better question would be "why do you want this to be 60fps" [03:01:46 CET] <furq> if it is actually 30p then fps is just going to dup every frame and minterpolate is going to be incredibly slow and probably give artifacts [03:02:15 CET] <DHE> mpeg2 1920x1080 makes me suspect OTA broadcasts which are actually 1080i [03:02:42 CET] <DHE> (god I hate that but that's the standard) [03:02:55 CET] <furq> yeah [03:03:40 CET] <furq> i'm not sure where else you'd get 1080 mpeg2, and i don't know of anyone broadcasting 1080p30 in mpeg2 [03:03:54 CET] <furq> but i'm sure someone out there is more than happy to disappoint me [03:04:27 CET] <furq> anyway [03:04:30 CET] <DHE> HD standards for broadcast are 720p60 and 1080i30. but it's all I got... [03:04:36 CET] <furq> prelude2004c: if it is interlaced then just -vf yadif=1 [03:04:45 CET] <furq> so you'd better hope it is if you really need this to be 60p [03:05:09 CET] <furq> in fact doesn't nvenc have a decent double-rate deinterlacer [03:29:33 CET] <prelude2004c> furq, not sure about double-rate de-interlacer [03:29:35 CET] <prelude2004c> checking that now [03:30:46 CET] <prelude2004c> interlaced_frame=1 [03:30:47 CET] <prelude2004c> interlaced_frame=0 looks like some frames are interlaced [03:30:56 CET] <prelude2004c> so just yadif=1 ? [03:31:41 CET] <furq> prelude2004c: -deint bob -c:v h264_cuvid -i foo [03:32:06 CET] <furq> there's also -deint adaptive, idk which is better [03:32:11 CET] <furq> bob will definitely double the framerate [03:33:23 CET] <prelude2004c> so i put that after -i right? [03:33:29 CET] <prelude2004c> sorry [03:33:33 CET] <prelude2004c> silly me.. you said it.. ok testing [03:35:15 CET] <prelude2004c> ok so this is what i get when i use the -deint bob [03:35:16 CET] <prelude2004c> Codec AVOption deint (Set deinterlacing mode) specified for input file #0 (udp://235.1.2.246:758?ttl=100&reuse=1&include=1&sources=10.187.124.10&fifo_size=64000&buffer_size=100000&overrun_nonfatal=1) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some decoder which was not actually used [03:35:17 CET] <prelude2004c> for any stream. [03:35:48 CET] <furq> oh wait it's mpeg2 isn't it [03:36:51 CET] <furq> -c:v mpeg2_cuvid if you weren't already doing that [03:36:58 CET] <furq> if you were then shrug [03:37:01 CET] <furq> i've never really used nvenc [03:38:43 CET] <prelude2004c> nah.. doesn't like -c:v mpeg2_cuvid [03:41:09 CET] <prelude2004c> -drop_second_field 1 should also drop it no ? but still doesn't help me much [03:42:18 CET] <prelude2004c> ehh doesn't like that too :( Shittzel [03:51:57 CET] <prelude2004c> ok so -deint doubles up the frames.. [03:52:07 CET] <prelude2004c> so that might do what i want but .. i'm using cuvid decoding [03:52:29 CET] <prelude2004c> not 100% sure what i'm doing wrong here.. doesn't like it it seems . i can't -deint bob [03:52:36 CET] <prelude2004c> probably because mpeg2video source [03:52:39 CET] <prelude2004c> not h264 [04:30:14 CET] <prelude2004c> wow.. ok yadif=1 seems to double up frames [04:30:39 CET] <prelude2004c> but.. kills my CPU on the decoding side.. i'm using hw_accel cuvid.. should it not do it on the GPU ? [04:30:50 CET] <prelude2004c> any other way to get similar result without killing system ? [09:39:02 CET] <xxx_> hello, does ffmpeg support closed caption? [09:52:00 CET] Last message repeated 1 time(s). [11:05:23 CET] <Fyr> guys, mkvalidator gives: [11:05:23 CET] <Fyr> >>WRN311: The Cues entry for timecode 1634424 ms is listed after entry 1642951 ms [11:05:35 CET] <Fyr> the file was made by FFMPEG. [11:05:48 CET] <Fyr> what is the problem with the file? [11:06:15 CET] <Fyr> I've a few thousands of these warnings. [11:08:28 CET] <JEEB> could be the input to the muxer having weird timestamps, but without any details on the input etc nothing can really be said :P [11:33:49 CET] <kazuma_> how can i demux/extract 1 audio channel? [11:34:06 CET] <kazuma_> using -ac 1 , downmixes the both channels into 1 [11:43:35 CET] <Fyr> JEEB, is there a way to find out what index_space is necessary? I want to optimize the output MKV, however, there is no guide to use -reserve_index_space option. [11:44:29 CET] <Fyr> I muxed the file, but I don't know what value to set to the -reserve_index_space option. [11:44:44 CET] <Fyr> mkvinfo doesn't show something useful. [14:18:02 CET] <volkan> hello guys [14:18:20 CET] <volkan> any one can help me i have an serious problem [14:18:54 CET] <iive> nobody can help you until you say what the problem is... [14:23:25 CET] <Fyr> the problem seems to be solved. =) [14:24:31 CET] <ovi> hello [14:24:39 CET] <ovi> i have a little problem [14:24:52 CET] <ovi> i am trying to set a list of jpg files as input to ffmpeg [14:25:02 CET] <ovi> but it is interepted as output [14:25:27 CET] <furq> pastebin the command line [14:25:50 CET] <ovi> ./ffmpeg -v debug -r 16 -f image2 -s 640x480 -i ./frames/0/*-video.jpg -c:v libx264 -crf 25 -pix_fmt yuv420p ./videos/tl_camera-0.flv [14:26:26 CET] <furq> add -pattern_type glob and quote the input filename [14:26:28 CET] <ovi> i have tried with cat and pipe but is not working because cat will complain too many arguments [14:26:31 CET] <furq> otherwise your shell will interpret the glob [14:26:56 CET] <ovi> in -pattern_type glob in front of -i ? [14:26:59 CET] <furq> yeah [14:28:04 CET] <Fyr> ovi, ffmpeg -f image2 -framerate 16 -pattern_type glob -i "frames/0/*-video.jpg" camera.flv [14:28:09 CET] <ovi> cool. thanks a lot [14:28:17 CET] <ovi> it is already running [14:28:32 CET] <ovi> thank you very much! [14:34:29 CET] <wiak> ffmpeg.exe -i "test.mkv" -vcodec libvpx-vp9 "AVOptions: -row-mt=1 --tile-columns=6 --auto-alt-ref=1 --lag-in-frames=25 --aq-mode=0 --threads=16 --target-bitrate=3000 --cpu-used=0" -acodec libopus test_out.mkv [14:34:32 CET] <wiak> hmm probobly not right [14:46:19 CET] <volkan> my problem is i have an pcm_f32le file [14:46:31 CET] <volkan> when i try to convert to wav it will be full noise [14:46:41 CET] <volkan> my cli looks like ffmpeg -acodec pcm_f32le -i channel10-title1.adp -ar 48000 out.wav [15:05:47 CET] <kepstin> volkan: how do you know it's "pcm_f32le"? if you're getting noise from that, it's probably not actually... [15:06:05 CET] <volkan> how can i know it [15:06:21 CET] <volkan> this sound file created honeywell annoucement device [15:06:41 CET] <volkan> actually honeywell has an program called adpcm2wav converter [15:07:33 CET] <furq> volkan: pcm_f32le is lpcm, not adpcm [15:07:56 CET] <furq> what does ffmpeg autodetect it as if you remove -acodec pcm_f32le [15:09:04 CET] <kepstin> ah, adpcm. ffmpeg has a lot of different adpcm decoders, be tricky to guess the right one. but given that this is telephony stuff it's probably g722 or g726? [15:10:03 CET] <volkan> i sniffed network but [15:10:45 CET] <volkan> rtp packet start 80 4E [15:11:13 CET] <kepstin> huh? network sniffing? then how did you make this file? [15:11:20 CET] <kepstin> is this a network capture or something? [15:12:26 CET] <kepstin> anyways, like I said, the codec is probably either g722 or g726 (the former is more likely, i think) [15:12:41 CET] <volkan> no this file created by device [15:13:03 CET] <furq> 722 is wideband isn't it [15:13:12 CET] <volkan> http://www.networksorcery.com/enp/protocol/rtp.htm [15:13:22 CET] <furq> it'll probably be 721 or 726, both of which are covered by adpcm_g726 [15:13:42 CET] <furq> so if ffmpeg can't autodetect it then try that [15:13:47 CET] <kepstin> hmm, well, either way it'll be one of those two [15:14:12 CET] <volkan> when i tried the g726 my mac is return this error Error while opening decoder for input stream #0:0 : Not yet implemented in FFmpeg, patches welcome [15:14:21 CET] <volkan> to * [15:14:28 CET] <furq> uh [15:14:33 CET] <furq> paste the command line and full output [15:14:38 CET] <volkan> ffmpeg version 3.4.1 Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 9.0.0 (clang-900.0.39.2) configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbo [15:14:46 CET] <furq> to a pastebin [15:14:57 CET] <volkan> ok 1sec please [15:15:52 CET] <volkan> https://paste.ee/p/GdKnF [15:16:02 CET] <volkan> pastebin is blocked our country sorry for that [15:18:41 CET] <kepstin> hmm, I think that filename might be confusing ffmpeg; there's an 'adp' demuxer for iirc some sort of game audio or something. [15:19:38 CET] <furq> you could try adding -ac 1 before -i, but yeah i'm guessing it's misdetecting the container [15:19:45 CET] <furq> and i have no idea what the right container would be [15:19:50 CET] <kepstin> try "-f g726" [15:20:03 CET] <furq> oh [15:20:08 CET] <furq> nice grepping [15:20:09 CET] <kepstin> or g726le, g722, etc [15:20:37 CET] <furq> do those exist [15:20:39 CET] <furq> i don't have any of those [15:21:26 CET] <kepstin> hmm, they're in my build. [15:22:59 CET] <volkan> ffmpeg -f g722 -ac 1 -i channel10-title2.adp out.wav [15:24:38 CET] <volkan> out file length is 29 min original file length is 4min [15:24:58 CET] <furq> try -f g726 [15:25:22 CET] <furq> kepstin: nvm they're new in 3.4 [15:25:25 CET] <volkan> in g726 53 min [15:25:34 CET] <furq> i'm still on 3.3 here [15:25:54 CET] <volkan> g726le same as g726 53 min [15:26:20 CET] <kepstin> volkan: if you play the resulting wav files are they just noise, or do they sound almost correct but distorted? [15:26:22 CET] <furq> volkan: pastebin the output of ffprobe foo.adp [15:27:02 CET] <volkan> honeywell provide this command line tool https://ibb.co/mEKYb6 [15:27:50 CET] <volkan> https://paste.ee/p/rbcsj [15:28:37 CET] <kepstin> volkan: so, does just "ffmpeg -i channel10-title1.adp out.wav" work? [15:28:49 CET] <furq> adpcm_dtk is a gamecube format, so probably not [15:29:23 CET] <kepstin> i guess at this point, it's just back to going through the list of adpcm codecs ffmpeg supports and trying them to see [15:29:34 CET] <furq> also i notice that image says it's converting to adpcm [15:29:55 CET] <furq> what file did you give that adpcm converter [15:30:22 CET] <kepstin> is that the tool that made the file, maybe? [15:30:30 CET] <furq> yeah i assume so [15:30:30 CET] <volkan> out file length is same as original wav file [15:30:33 CET] <volkan> but noisy [15:30:55 CET] <furq> lol [15:30:56 CET] <volkan> is honeywell encode this sound file like an aes or something else [15:31:05 CET] <furq> if you have the original file then what are you doing this for [15:31:16 CET] <furq> are you just testing this honeywell tool or something [15:31:55 CET] <volkan> some announcement recording and i don't have it original file i need to convert and stream [15:32:06 CET] <volkan> but honeywell tool only work with windows [15:33:54 CET] <kepstin> hmm. the only other ffmpeg-supported adpcm codec that seems possible is adpcm_ms [15:34:03 CET] <kepstin> since most of the rest are game formats [15:34:29 CET] <volkan> i tried adpcm_ms but failed says no packet found [15:35:03 CET] <volkan> i have an sniffed network data rtp stream can we recognize on stream [15:35:48 CET] <volkan> https://ibb.co/nxpfpR [15:36:11 CET] <volkan> blue background is raw rtp data [17:17:46 CET] <Purple_Minion> How do you set the aspect ratio in ffserver? putting -aspect 4:3 in ffmpeg creates a proper file that tells the player to stretch it when playing, but ffserver can't do this it seems. [17:20:57 CET] <BtbN> Keep in mind that you really should not be using ffserver. And nobody really has any idea how to use it, so help with it is unlikely. [17:21:15 CET] <Purple_Minion> oh I didn't know that. Is there a better program? [17:21:37 CET] <Purple_Minion> crashed twice already [17:22:02 CET] <klaxa> people usually recommend nginx-rtmp or hls or mpeg-dash [17:22:43 CET] <Purple_Minion> Which one do you think is the easiest to configure? [17:23:27 CET] <klaxa> use hls and let ffmpeg write segments to disk which are served by a webserver [17:23:37 CET] <klaxa> i think that shouldn't be too hard to set up really [17:23:47 CET] <klaxa> requires your devices to support hls though [17:23:48 CET] <Purple_Minion> How much delay is there? [17:23:57 CET] <klaxa> quite some [17:24:00 CET] <Purple_Minion> I'll look up hls [17:24:40 CET] <klaxa> hls works by splitting up your video into small chunks and maintaining a playlist file of these chunks [17:24:52 CET] <klaxa> clients dowload the playlist file and then download the segments [17:25:01 CET] <Purple_Minion> Could VLC be used as well? [17:25:03 CET] <klaxa> the player stiches them together again [17:25:04 CET] <klaxa> yes [17:25:12 CET] <Purple_Minion> VLC to stream I mean [17:25:16 CET] <klaxa> ah [17:25:18 CET] <klaxa> maybe? [17:25:26 CET] <klaxa> not for hls though i think [17:25:32 CET] <klaxa> but i never checked [17:25:43 CET] <kazuma_> hls/nginx is the way to go [17:26:20 CET] <kazuma_> you let ffmpeg write the m3u8/.ts segments to a dir nginx monitors [17:26:36 CET] <kazuma_> the load up the .m3u8 on your device/mediaplayer/browser whatever [17:27:25 CET] <Purple_Minion> That's too much delay for video conferencing [17:27:39 CET] <klaxa> yes [17:27:43 CET] <Purple_Minion> RTSP seems like a great system [17:31:22 CET] <Purple_Minion> ffplay rtsp://nomad.ignorelist.com:5554/tv.mpeg2 [17:34:22 CET] <Purple_Minion> ffserver works somewhat :) [17:52:21 CET] <IchGucksLive> hi i woudt like to downbsize a HD to a SD so projektx can grab it as it wont do HD cuts [17:52:39 CET] <IchGucksLive> in showes Stream #0:0[0x25f]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt470bg), 720x576 [SAR 16:11 DAR 20:11], 25 fps, 50 tbr, 90k tbn, 50 tbc [17:53:02 CET] <IchGucksLive> this side showes a script https://avpres.net/FFmpeg/HD_SD.html [17:53:41 CET] <furq> 720x576 is sd [17:54:16 CET] <IchGucksLive> ffmpeg -i MillNut.ts -filter:v "colormatrix=bt709:bt601,scale=768:576:flags=lanczos,pad=768:432:0:72"-c:a copy SDmillNut.ts [17:54:53 CET] <IchGucksLive> OH thanks furq [17:55:01 CET] <IchGucksLive> but it cand be read [17:56:30 CET] <IchGucksLive> it showes pictuer decoing not posible for mpeg-4/H.264 [17:56:54 CET] <furq> you'll probably need to transcode to mpeg2video though [17:56:55 CET] <furq> then [17:57:14 CET] <furq> what are you trying to do with projectx anyway [17:57:43 CET] <IchGucksLive> cutting down its a 4episode file to the episodes [17:57:59 CET] <furq> you could just do that with ffmpeg [17:58:00 CET] <IchGucksLive> im on a old 14.04 ubuntu [17:58:04 CET] <furq> or maybe something like avidemux if you want a gui for it [17:58:48 CET] <IchGucksLive> avidemux read the file only at 2.6 version i only can run 2.5.4 [18:00:09 CET] <furq> there are static builds for avidemux on their site [18:00:12 CET] <furq> http://avidemux.sourceforge.net/download.html [18:03:11 CET] <IchGucksLive> 2.7 read the file [18:06:46 CET] <IchGucksLive> Workes gread Thanks [18:09:11 CET] <Purple_Minion> ffmpeg -i yourfile -vcodec mpeg2video -acodec copy out.avi [18:09:25 CET] <BtbN> -c:v and -c:a [18:10:34 CET] <furq> also yeah don't actually do that [18:10:51 CET] <Purple_Minion> audio not correct? [18:10:57 CET] <furq> he doesn't need to reencode it [18:11:06 CET] <Purple_Minion> I thought it needs to become mpeg2 [18:11:09 CET] <volkan> hello guys [18:11:12 CET] <furq> only to import it into projectx [18:11:20 CET] <furq> which is no longer required [18:11:36 CET] <furq> although if you were going to reencode something to m2v you'd need a whole lot more parameters than that to make it look good [18:11:37 CET] <IchGucksLive> for only cutting it is not needed to transform [18:11:40 CET] <furq> none of which i remember, sadly [18:12:01 CET] <IchGucksLive> need to check if the TV can run this from the NAS after AVIDMUX 2.7 [18:12:47 CET] <furq> IchGucksLive: make sure you set avidemux to copy the streams [18:13:05 CET] <furq> you shouldn't need to reencode it as long as you're ok with the cut points possibly being slightly out [18:13:48 CET] <IchGucksLive> i only did a Mux TS seting [18:14:02 CET] <furq> you can also do it with ffmpeg, it's just a bit annoying to get the cuts with no gui [18:14:03 CET] <IchGucksLive> auti and viodio cop [18:14:07 CET] <furq> ok good [18:14:32 CET] <IchGucksLive> -ss -t is for ffmpeg [18:14:38 CET] <furq> right [18:14:42 CET] <furq> and -c copy [18:15:41 CET] <IchGucksLive> but the gui is perfect as procktX has now been history if that workes for me [18:15:57 CET] <IchGucksLive> Modern Days from know [18:16:28 CET] <IchGucksLive> Thanks im off [19:13:17 CET] <Purple_Minion> Does anyone have an opinion on gst-rtsp-server ? [19:13:51 CET] <furq> that's only a library isn't it [19:15:10 CET] <Purple_Minion> I think you're right. I wonder what uses it? [19:15:15 CET] <furq> fwiw i wouldn't expect rtsp to have much (any?) less latency than rtmp [19:15:29 CET] <furq> the high latency thing is if you use hls or dash so that browsers can view the stream [19:15:48 CET] <furq> which nginx-rtmp will do for you, but you can also just use the rtmp feed [19:17:11 CET] <Purple_Minion> https://stackoverflow.com/questions/36759875/streaming-h264-file-using-gst-… [19:18:47 CET] <furq> http://vpaste.net/umpW3 [19:18:51 CET] <furq> doesn't look like anything uses it [19:21:31 CET] <Purple_Minion> nginex [19:21:35 CET] <Purple_Minion> oops [19:45:25 CET] <roggy545> isnt it possble to run a http stream anymore with just ffmpeg now that ffserver is dropped? [19:48:38 CET] <DHE> use a format that's HTTP friendly (typically DASH or HLS) and run a boring old HTTP server like Apache [19:51:12 CET] <klaxa> you can also feel adventurous and try this: https://github.com/klaxa/mkvserver_mk2 [19:51:20 CET] <klaxa> it uses ffmpeg's http server api [20:06:37 CET] <furq> someone in here mentioned that you can push mpegts over icecast [20:06:45 CET] <furq> which is basically http [20:06:50 CET] <furq> i tried that and didn't have the best results [20:07:23 CET] <furq> it seemed to work quite well on the one in five occasions that mpv wouldn't just bail out before starting playback [20:09:54 CET] <roggy545> Apache already installed so I think I'll try that first [21:00:45 CET] <utack> is there a reason input jpg and converting to yuv420p significantly alters the color? [21:02:14 CET] <furq> yes [21:02:40 CET] <utack> is there a way around or, or do i have to deal with it? [21:02:42 CET] <furq> it's converting from full to limited range and it sounds like either swscale is doing it wrong or whatever you're viewing it with isn't handling that well [21:03:11 CET] <utack> i did not intentionally insert any flters, maybe zscale is better? [21:03:26 CET] <furq> -pix_fmt yuv420p is the same as -vf format=yuv420p [21:03:41 CET] <furq> also i take it your output is a video [21:04:14 CET] <utack> yeah, i am trying to compress dashcam footage into a video stream to store it at someting like 15% the size [21:04:33 CET] <utack> so the quality is not super critical or the color, but i was wondering about it [21:04:37 CET] <furq> x264 accepts yuvj420p as input [21:04:48 CET] <kepstin> utack: your dashcam footage is a bunch of individual jpeg files? or an mjpeg stream? [21:04:59 CET] <furq> that should fix the colours but i don't know how it affects anything else in the world it has to interact with [21:05:08 CET] <utack> individual jpegs, it is actually a gopro in interval mode, not some cheap dashcam [21:05:44 CET] <kepstin> utack: does the resulting video appear to have lower contrast (washed out dark colour, grey light colors)? [21:05:46 CET] <furq> i assume passing it yuvj420p does the same thing as x264 --fullrange but i don't actually know [21:06:05 CET] <furq> i vaguely remember there being more than one way of getting full range [21:06:35 CET] <utack> it is a bit brighter and a bit washed out kepstin [21:06:45 CET] <furq> yeah that sounds about right [21:06:45 CET] <utack> i'll upload a sample, one second [21:07:01 CET] <utack> ok so that is yuv420 limitation? guess i can use yuv444 then? [21:07:12 CET] <kepstin> my guess is that the camera is writing limited range to jpeg, ffmpeg is reading it as full range and then doing an extra conversion to limited range [21:07:30 CET] <furq> utack: full/limited range has nothing to do with the subsampling [21:07:34 CET] <furq> yuv444 will have the same issue [21:07:35 CET] <utack> ah [21:07:42 CET] <furq> like i said, try with yuvj420p [21:07:53 CET] <utack> did not solve it unfortunately [21:07:56 CET] <furq> oh [21:08:03 CET] <furq> well yeah kepstin is probably right then [21:09:08 CET] <furq> so uh [21:10:01 CET] <kepstin> i dunno, adding "-pix_fmt yuv420p" as an *input* option might work around the issue [21:10:10 CET] <furq> there is a proper way to do this but i forget what it is now [21:10:15 CET] <furq> something like -vf colorspace [21:10:41 CET] <furq> maybe -vf zscale=rin=limited,format=yuv420p [21:10:53 CET] <furq> instead of pix_fmt obv [21:11:45 CET] <kepstin> -vf scale=in_range=tv:out_range=tv should also work, i think [21:12:08 CET] <kepstin> (it's a no-op, just retagging the input video as tv range so the player will handle it correctly) [21:13:28 CET] <utack> hmh limited input range messes with it a LOT more, that does not seem to be the case [21:13:48 CET] <furq> what are you playing the video in [21:13:58 CET] <utack> mpv, for both the result and the jpg [21:14:03 CET] <furq> opengl vo? [21:14:05 CET] <utack> yep [21:14:13 CET] <furq> yeah that's probably not it then [21:15:46 CET] <utack> probably not. thank you for thinking about it, but given that i just want to "archive" it at lower quality, the little color shift is quite acceptable either way [21:16:43 CET] <furq> maybe -vf colorspace=irange=tv:range=tv:format=yuv420p [21:16:50 CET] <furq> failing that i'd suspect maybe ICC profiles or something like that [21:17:12 CET] <furq> if mpv even supports those [21:34:35 CET] <wiak> any idea why this iant working ffmpeg.exe -i "test.mkv" -vcodec libvpx-vp9 -row-mt 1 -tile-columns 6 -auto-alt-ref 1 -lag-in-frames 25 -aq-mode 0 -threads 16 -target-bitrate 3000 -cpu-used 0 -acodec libopus test_out.mkv [21:34:36 CET] <wiak> ? [21:38:06 CET] <utack> well what does it say goes wrong? [21:39:04 CET] <utack> also vp9 really benefits from 2pass, i have been told. and is there a reason you are using row mt and tile columns both? [21:39:07 CET] <sfan5> your target bitrate is quite low [21:39:12 CET] <sfan5> you probably wanted 3000k [21:39:43 CET] <sfan5> unless that option actually accepts kbit/s, i'd prefer using -b:v anyway [21:40:57 CET] <wiak> libvpx-vp9 defaults to 2pass anyway [21:41:09 CET] <utack> ah [21:42:32 CET] <wiak> i know the quality of the settings,l they look awesome when i encode with vpxenc [21:42:55 CET] <wiak> you can see some of the samples here https://awesome.nwgat.ninja/party/ [21:43:32 CET] <wiak> but the issues is that i get invalid argument on row-mt for some werid reason [21:44:46 CET] <sfan5> looking at ffmpeg -h encoder=libvpx-vp9 that option does not seem to exist [21:45:02 CET] <sfan5> not here at least, my libvpx is probably too old [21:45:37 CET] <furq> row-mt is only in git builds [21:45:45 CET] <furq> or 1.7 if that came out yet [21:46:29 CET] <wiak> https://patchwork.ffmpeg.org/patch/2745/ [21:46:42 CET] <wiak> is ffmpeg dev so slow? [21:47:29 CET] <furq> that says it was accepted the same day [21:47:30 CET] <wiak> the patch was added march of last year.. [21:47:41 CET] <utack> wiak libvpx 1.7 was only released a few days ago, not even arch has it yet [21:48:05 CET] <wiak> row-mt was added in 1.6.1* [21:48:17 CET] <furq> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/libvpxenc.c#L111 [21:48:20 CET] <wiak> ffmpeg row-mt support was added march of last year [21:48:50 CET] <furq> what are you asking then [21:49:02 CET] <wiak> how to pass arguments to vpxenc? ;P [21:49:15 CET] <utack> only worked for me when rebuilding ffmpeg 3.4 with libvpx 1.7 in place [21:49:25 CET] <furq> yeah i'm pretty sure row-mt is new in 1.7 [21:49:35 CET] <furq> it's been around for a while but never cut into a release [21:49:57 CET] <furq> until last week obv [21:50:15 CET] <wiak> can you look at my options and see if am doing it wrong? [21:50:17 CET] <furq> https://github.com/webmproject/libvpx/releases/tag/v1.7.0 [21:50:19 CET] <furq> yeah there you go [21:50:32 CET] <furq> wiak: are you sure you have an ffmpeg build with libvpx 1.7 [21:50:42 CET] <wiak> lol probobly not stupid me [21:50:45 CET] <wiak> totally forgot [21:50:56 CET] <wiak> zaraone ffmpeg 3.4 i think [21:51:17 CET] <wiak> https://groups.google.com/a/webmproject.org/forum/#!topic/codec-devel/oiHjg… [21:51:25 CET] <furq> the latest zeranoe git build has 1.7 [21:51:43 CET] <utack> you can always pipe to a libvpx git that is straightfoward to build. [21:51:54 CET] <utack> ffmpeg -i source.mkv -pix_fmt yuv420p -f rawvideo - | ~/libvpx/vpxenc ... --width=1920 -height=1080 - -o out.vp9 [21:51:57 CET] <utack> something like this [21:52:21 CET] <wiak> aah thanks [21:52:38 CET] <utack> that is how i used row-mt before libvpx 1.7 was released [21:52:54 CET] <wiak> ive used row-mt almost a year lol [21:53:05 CET] <utack> with ffmpeg? [21:53:06 CET] <furq> i should grab a build and check it out [21:53:23 CET] <furq> i don't care enough about vp9 to have bothered building with git libvpx [21:53:24 CET] <wiak> utack no i just use ffmpeg to export to y4m hehe [21:53:28 CET] <furq> but that's mostly because it's so slow [21:53:33 CET] <wiak> well not anymore [21:53:44 CET] <wiak> still slower than h264 but alot faster than last [21:53:49 CET] <furq> that's good to know [21:53:51 CET] <wiak> last= before [21:54:12 CET] <furq> In tests[1] of encoding HD videos with 4 column tiles, the improved VP9 MT encoder achieved speedups over the original of 11% with 2 threads, 27% with 4 threads, 101% with 8 threads, and 135% with 16 threads. [21:54:14 CET] <wiak> my ryen 1700x used to do 25% now it uses 80% [21:54:17 CET] <furq> that sounds about right lol [21:54:23 CET] <wiak> ryzen* [21:54:30 CET] <furq> yeah i got about 33% with 8 threads [21:54:42 CET] <utack> any reason you use tile columns and row-mt both? [21:55:23 CET] <furq> hopefully you get an even bigger improvement on stuff that can only use two tiles [21:55:28 CET] <furq> that was my big problem with it before [21:55:41 CET] <furq> sd stuff would hardly be any faster than hd because it just wouldn't multithread [21:56:08 CET] <utack> i used row mt to encode 960x540 yesterday and it was using something like 500% cpu [21:56:37 CET] <furq> that's all right [21:57:12 CET] <furq> one of the av1 developers pinky promised me that libaom would have frame threading [21:57:20 CET] <utack> :D [21:57:28 CET] <furq> it better or else he's a dead man [21:58:42 CET] <utack> i mean all the "big" content providers probably don't care, they slice stuff anyways and piece it together on keyframes? [21:59:02 CET] <utack> you could probably also implement that in a bash script for your consumer machine quite easily? [22:35:32 CET] <seb_> hello [22:35:38 CET] <utack> hi [22:36:54 CET] <seb_> i have a little question about PPM format [22:37:27 CET] <seb_> i see that format could be in Binary or ASCII [22:38:25 CET] <seb_> No luck for me, i think that by default the output format is Binary and not ascii [22:38:50 CET] <seb_> is it a way to change de output format ? [22:39:22 CET] <iranen> when do you think you can use ffmpeg to compress av1? [22:40:53 CET] <seb_> I use ffmeg to make thumbnail [22:45:42 CET] <kepstin> iranen: I'd expect it to be added fairly soon once the aom av1 encoder has a stable api. [22:46:42 CET] <furq> yeah there hasn't been a release of libaom yet [22:46:46 CET] <furq> presumably once there is then ffmpeg will add it [22:47:02 CET] <furq> it's not like there aren't any aom developers in here who could add it [22:47:44 CET] <utack> on the plus side if they add it in the current state, by christmas you might be done encoding your first 60 minute video :D [22:48:31 CET] <iranen> ty for the info guys :) [22:50:25 CET] <iranen> I'm really excited about av1 [22:50:28 CET] <DHE> even x264's API isn't quite stable... close, but not exactly [22:51:13 CET] <DHE> iranen: I am as well, but what I really need is support in hardware, like TVs, Android, etc [22:51:44 CET] <iranen> well sure [22:52:09 CET] <kepstin> hmm, there's that exciting x264 api break just recently when they added multi-bit-depth support in one lib :) [22:53:55 CET] <seb_> How to have with strftime() milliseconds? [22:57:45 CET] <seb_> i found https://github.com/InSoundz/ffmpeg-strftime-milliseconds [22:57:49 CET] <seb_> yessss [22:58:24 CET] <seb_> it coud be add to ffmpeg [23:09:46 CET] <SortaCore> *hunts for C++ nvenc hwaccel example* [00:00:00 CET] --- Sun Feb 4 2018

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