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May 2016

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ffmpeg-devel.log.20160530
by burek 31 May '16

31 May '16

[02:59:36 CEST] <cone-129> ffmpeg 03Mark Reid 07master:d74cc6157478: libavformat/movenc: remove unnecessary null check [06:49:41 CEST] <Admin__> DHE , patch did not work. Still crashed at 26 hours and 30 minutes. [08:43:38 CEST] <andrey_turkin> hmm, for some reason I get garbled output when using nvenc with yuv420p format on Windows [10:59:49 CEST] <ubitux> is anyone is working on the merging of libav? next commit is 3176217 [11:00:14 CEST] <ubitux> i'd like to have a look; is there a WIP, some stuff to know about this? [11:00:18 CEST] <ubitux> nevcairiel? [11:01:31 CEST] <nevcairiel> the next commit is rather tedious, which is why I didnt do it yet, didnt have the motivation yet to figure it out [11:03:30 CEST] <ubitux> i'll try to do it [11:03:41 CEST] <ubitux> do we have written directives about how to proceed with the merge? [11:05:03 CEST] <nevcairiel> no [11:05:30 CEST] <nevcairiel> (1) make sure nothing is broke; (2) see 1 [11:07:49 CEST] <mateo`> nevcairiel: I've heard about a script regarding the merges. Is there actually one ? [11:08:15 CEST] <nevcairiel> i just wrote one myself to pick the next commit [11:08:43 CEST] <nevcairiel> http://pastebin.com/TJSgHdr9 [11:09:37 CEST] <mateo`> i'm also interested in helping [11:09:42 CEST] <mateo`> nevcairiel: thanks [11:10:45 CEST] <nevcairiel> merging often requires quite a thorough understanding of the code in question, especially in the current commits needing merging in the h264 decoder [11:11:43 CEST] <ubitux> i'm more concerned about how to test all the hwaccels [11:12:04 CEST] <nevcairiel> and one should be aware of future commits coming in, so that if there is a compile fix for the current merge, it should be fixed right here instead of pushing a broken merge, for example [11:12:10 CEST] <mateo`> nevcairiel: I know, I guess it's time to learn new things :) [11:12:14 CEST] <nevcairiel> (which is actually relevant for the merge on top right now) [11:12:32 CEST] <iive> ubitux: do you have radeon video card? mesa3d support vaapi and vdpau [11:12:48 CEST] <ubitux> i only have intel [11:12:51 CEST] <ubitux> and linux only [11:13:02 CEST] <mateo`> I have one [11:13:31 CEST] <nevcairiel> so in short: don't rush anything :) [11:13:57 CEST] <iive> ubitux: mesa3d is the linux hw acceleration for opengl direct3d9 , egl , etc... [11:14:25 CEST] <iive> what does the new h264 changes do? [11:14:43 CEST] <ubitux> iive: git log --oneline --reverse master..libav/master [11:15:05 CEST] <iive> ubitux: don't they have web interface? [11:15:33 CEST] <ubitux> git.libav.org [11:16:43 CEST] <iive> and the name of the commit? [11:17:20 CEST] <ubitux> http://sprunge.us/ZUea [15:42:34 CEST] <BtbN> andrey_turkin, 901b48b looks like it breaks compile on windows. There are at least some TEXT() blocks missing for the LoadLibrary calls. [15:49:12 CEST] <Admin__> DHE you around? [15:49:29 CEST] <DHE> a bit busy [15:50:00 CEST] <Admin__> sorry i don't mean to bother you .. just wanted to tell you the patch did not work. at exact 26.30 of runtime it stopped working [15:53:48 CEST] <BtbN> What exactly does the SKIPHEADERS Makefile variable do? [16:00:05 CEST] <jkqxz> It stops the header from being test-built by "make checkheaders". It should be set for a header with an external dependency which might not be present if the given option is not set (headers with only internal dependences will always be tested). [16:03:05 CEST] <DHE> Admin__: dammit... [16:05:01 CEST] <Admin__> :( [16:05:14 CEST] <Admin__> i am still stuck [16:50:03 CEST] <prelude2004c> DHE, BtbN , anyone have any further recommendations? [17:02:00 CEST] <DHE> not right now... [17:45:08 CEST] <andrey_turkin> BtbN: it compiled for me with native msvc. I guess I should try to setup crosscompilation toolchain [17:51:07 CEST] <BtbN> strange, it shouldn't. LoadLibrary expects a wchar_t* [17:53:00 CEST] <andrey_turkin> maybe it was defaulting to ansi build [17:54:53 CEST] <prelude2004c> DHE , do you know anyone that I could contact and pay for programming time to get these things resolved ? I need to move on this stuff and understand how busy everyone is. I would not mind paying for it in order to get it done asap. If you know anyone please let me know. [17:55:17 CEST] <BtbN> Are you still going on about your hacky streaming thing? [17:55:36 CEST] <prelude2004c> I also have some other work that I need done. Once it is done I have no issue with the code going back to the open source community so no loss there. [17:55:42 CEST] <DHE> BtbN: there is definitely an ffmpeg issue when the duration of the video exceeds the range of the timestamps of the format [17:55:47 CEST] <prelude2004c> um.. currently its not about my hacky streaming thing :) [17:55:52 CEST] <BtbN> ah, so not the same issue anymore. [17:56:32 CEST] <BtbN> Well, is there a sane way to handle that case anyway? [17:56:35 CEST] <prelude2004c> actually BtbN this is a major issue.. the rest are small little things.. eg.. http:// webdav i already pulled that out.. i dont need it.. i only used it to test if it had anything to do with the stream stopping but soon as i found out there was an issue at 26.5 that http doesn't really help so i took it out [17:56:37 CEST] <JEEB> I have a feeling we are exporting a bit too much stuff into general avformat [17:56:59 CEST] <JEEB> mpegts demuxer should handle the wrapping in itself and just output timestamps that are rising [17:57:12 CEST] <JEEB> avformat itself would just see those [17:57:20 CEST] <BtbN> For how long can it keep doing that until the 64bit timestamps overflow? [17:57:20 CEST] <DHE> BtbN: I tried making a patch (hacky I admit) that tracks wrap-around of the DTS and adds (1 << pts_wrap_bits) to it, once per wrap-around. [17:57:24 CEST] <DHE> apparently that didn't work [17:57:37 CEST] <JEEB> BtbN: at that point we're eff'd anyways [17:58:11 CEST] <DHE> mpegts uses a timebase of 1/90000, which is still plenty big assuming you don't have something gone completely crazy and spewing garbage timestamps [17:59:01 CEST] <JEEB> [mpegts: handles mpeg-ts specifics] => general avformat gets general rising timestamps => [mpegtsenc: handles the rollovers inside itself] [17:59:27 CEST] <kierank> 64-bit timestamp overflow is rather large [17:59:30 CEST] <JEEB> IMHO it should generally go like that and if it doesn't we have an issue [17:59:40 CEST] <kierank> can't remember if it's ~heat death of universe large [17:59:48 CEST] <kierank> but yes I agree JEEB [17:59:53 CEST] <kierank> that's how other demuxers do it [18:02:59 CEST] <prelude2004c> I am told to use gstreamer and that it doesn't have that issue.... i refuse to do that :) i like ffmpeg... just have to be able to stabalize this probelm [18:09:17 CEST] <iive> if i remember correctly, mpegts ts are 33 bit, not 64. and that's with 90kHz clock. if 27MHz clock is used, 9 bit are added to the 33 ones. [18:11:42 CEST] <iive> JEEB's idea is reasonable. I'm just not sure what would happen on seeking. [18:12:13 CEST] <iive> that is, in the case where we don't know if we've had wrap arounds [18:13:01 CEST] <nevcairiel> seeking in wrapping files is not really something you can expect to produce expected timestamps, but thats not any worse than today [18:13:42 CEST] <andrey_turkin> iirc mpegts timestamps wrap once every day or so [18:13:52 CEST] <nevcairiel> 26 hours or so [18:14:32 CEST] <andrey_turkin> right. So not really possible to do a seek inside a very long file [18:17:49 CEST] <kierank> not easily [18:18:10 CEST] <kierank> the "proper" way is to generate timestamps on receive [18:28:19 CEST] <prelude2004c> kierank , when you say generate timestamps on receive do you mean " -fflags +genpts ? [20:54:48 CEST] <prelude2004c> kierank , i didn't get a reply because i got disconnected :( sorry about that [21:28:07 CEST] <DHE> prelude2004c: probably not getting CC in nvenc, boss isn't interested [21:29:42 CEST] <prelude2004c> CC in nvenc ? ( closed caption ) , you know the problem though ? [21:30:02 CEST] <prelude2004c> can it be fixed is what i want to know... that way i can get someone to fix it [21:31:04 CEST] <DHE> it can be. libx264 is already doing it [21:31:11 CEST] <prelude2004c> for now i suspect cc can work if we move the code to use nvenc instead of piping out to Nvtranscoder.. the only thing that seems to have to be fixed is to check the quality comparison and also the decoding [21:32:24 CEST] <prelude2004c> if anyone knows anyone interested in helping me fix this and charging me for it please message me on skype " prelude2004c " [21:32:39 CEST] <prelude2004c> DHE, what about this bug of 26.5 .. do we have an approach to it ? [21:33:31 CEST] <prelude2004c> DHE, can you get the nvenc that is in ffmpeg to also push decoding into the card instead of having to use vdpau ? [21:33:58 CEST] <prelude2004c> vdpau requires X where as the NvTranscoder does not and makes ffmpeg so much more efficient [21:34:23 CEST] <DHE> prelude2004c: I'm still on the 26.5 thing myself. the nvenc decode thing doesn't directly align with my own needs - I'm kinda doing this because it's what I need as well [21:35:44 CEST] <prelude2004c> DHE, much appreciate you helping with this stuff... + i have learned quite a bit just listening to the chat here.. i wish i could just program to help you guys out and make these things all become a reality.. but i think i am too old :P [21:35:56 CEST] <BtbN> Is that NvTranscoder thing open source? [21:36:02 CEST] <prelude2004c> yes open source [21:36:06 CEST] <BtbN> where is it? [21:36:18 CEST] <BtbN> Because there is no API I know about which supports what you claim it can do. [21:36:36 CEST] <prelude2004c> https://developer.nvidia.com/nvidia-video-codec-sdk [21:36:47 CEST] <BtbN> No, the NvTranscoder you are using. [21:36:54 CEST] <prelude2004c> yes.. download the one for linux.. just make and run it [21:37:01 CEST] <BtbN> well, where is it? [21:37:07 CEST] <prelude2004c> NvTranscoder is a sample inside the " https://developer.nvidia.com/nvidia-video-codec-sdk " [21:37:24 CEST] <DHE> it ships with the API for nvenc as a sample app [21:37:30 CEST] <prelude2004c> there are a bunch of differnet stuff there.. for perl , decode, encode, transcode [21:38:20 CEST] <prelude2004c> nvtranscoder takes care of encoding/decoding and i get less than 20% cpu usage when using it and improved quality at lower bit rates. It seems that it is able to respect QP while maintaing the bit rate min/max [21:38:35 CEST] <prelude2004c> and when i use nvenc, my cpu usage on ffmpeg is well over 100% [21:38:40 CEST] <BtbN> So I guess you are on windows? [21:38:44 CEST] <prelude2004c> 110-120% by comparison [21:38:48 CEST] <prelude2004c> no i am on ubuntu [21:39:01 CEST] <prelude2004c> the apis are linux based [21:39:04 CEST] <BtbN> That's basically impossible then. The API it's using for decoding is documented as windows-only. [21:39:16 CEST] <prelude2004c> nope... see it for youserlf .. it works [21:42:41 CEST] <BtbN> It's litteraly in #ifdef _WIN32 in the cuda headers. [21:43:07 CEST] <prelude2004c> i dont know the details to be honest with you.. but it just works [21:44:29 CEST] <BtbN> And even if it was supported, the only API I see takes a filename as input, and gives you frames. [21:44:34 CEST] <BtbN> So not overly suitable for ffmpeg. [21:46:25 CEST] <DHE> what are you looking at anyway? (sorry, just finished copying the sdk locally) [21:46:35 CEST] <nevcairiel> full hardware pipelines are easy if you use a tool thats designed to work with exactly one piece of hardware and does exactly one thing [21:46:47 CEST] <nevcairiel> ffmpeg works with everything and does everything [21:46:50 CEST] <nevcairiel> so hardly easy [21:46:55 CEST] <DHE> and then you lose things like filters [21:49:05 CEST] <prelude2004c> i believe what you guys are saying..... but asside from that 26.5 hour thing it works well... the issue i have with nvenc ( assuming i can tweak quality further to match that of nvtranscoder which should be possible ), is the decoding.. using up CPU power to decode is a problem when adding 10 - 15 channels per server. [21:49:19 CEST] <prelude2004c> having the cards decoder take care of it , seems like a natural solution [21:52:34 CEST] <prelude2004c> the nvidia cards put in decoders / encoders for this purpose and created the api's for this purpose.. i am sure there is some work to be done but it can be done and once complete , i think effeciency will be soo much more improved... assuming nvenc that is currently built into ffmpeg is working well .. the only thing left to do is send off the decoder to the hardware decoder too ( eg. nvenc ) .. right now vdpau is supposed to do [21:52:34 CEST] <prelude2004c> that but i hvae to install X and i had a few other issues with vdpau and getting it working correctly. It was not clean and simple .. so here i am :) and forgive my ignorance on some of this stuff i am trying to make sense of everything and some stuff is not a simple as it seems [21:52:44 CEST] <DHE> found a PDF in the SDK that describes Cuda-based decoding (GL in the name) [21:53:21 CEST] <DHE> I might turn this into a raw decoder... [21:54:07 CEST] <BtbN> The hw de/encoders get pretty bad when handling multiple streams, they're not exactly made for that. [21:54:45 CEST] <DHE> the encoder actually does fairly well. can't say about the decoder [21:54:58 CEST] <prelude2004c> BtbN , asside from this 26.5 hour thing.. they been rock solid... i run approx 10-15 streams per card [21:55:28 CEST] <prelude2004c> DHE, the channel link i sent you , that is being decoded on card too from H264 > and then converting back to H264 [21:56:04 CEST] <prelude2004c> it's taking a 1080p 15Mbit/s stream and converting it to 720p ( 1.8-2.2Mbit/s ) [21:56:17 CEST] <prelude2004c> quality is very good [21:56:32 CEST] <BtbN> I highly doubt quality with nvenc will be any good at that bitrate. [21:56:33 CEST] <prelude2004c> i still have the link decrypted so.. you guys can check it out [21:56:39 CEST] <prelude2004c> :) watch this [21:56:55 CEST] <BtbN> "this" [21:56:56 CEST] <DHE> I only saw a bit of talking-heads. not bad for the bitrate/quality [21:57:00 CEST] <prelude2004c> http://67.55.3.103/ctvvancouverhd1151/ctvvancouverhd1151.m3u8 [21:57:41 CEST] <prelude2004c> and ignore the video from space [21:57:52 CEST] <prelude2004c> :P it can't make space station closer :) [21:58:13 CEST] <BtbN> there is nothing happening in those scenes. So of course it looks bearable [21:58:33 CEST] <prelude2004c> let me see if i can find you a better one [21:58:34 CEST] <prelude2004c> sec [21:58:53 CEST] <BtbN> And even with those low-motion scenes you can already see artifacts [22:00:24 CEST] <prelude2004c> http://67.55.3.103/cbcvancouverhd1150/cbcvancouverhd1150.m3u8 [22:00:44 CEST] <prelude2004c> this is live TV ... we have to keep bit rates down for streaming.. its not something i can filter 10 x and go with VOD .. these sources are live streams [22:01:16 CEST] <prelude2004c> look at http://67.55.3.103/cbcvancouverhd1150/cbcvancouverhd1150.m3u8 .. its more inline with what normally people watch on tv [22:01:44 CEST] <prelude2004c> and yes, its not 1080p.. but its better quality than i got using nvenc ( not sure why ) [22:03:03 CEST] <prelude2004c> going to show you guys something else.. [22:03:11 CEST] <prelude2004c> there are 6 channels running on this server [22:03:15 CEST] <DHE> I don't think it's going to help at this point [22:03:28 CEST] <prelude2004c> http://imgur.com/xxTkUy8 [22:04:11 CEST] <prelude2004c> i am only trying to provide as much good evidence as possible .. i want to see ffmpeg improve too... i rely on it and i know people do too.. its a great product... so [22:04:46 CEST] <BtbN> Use the exact same settings for ffmpeg as you use for the nvTranscoder, and you get the exact same results. [22:04:59 CEST] <DHE> so apparently with the opengl driver you can extract raw YUV out of the hardware decoder. Is that something ffmpeg would be interested in having? I suspect it does require X11 to work though even if you don't render anything [22:07:06 CEST] <prelude2004c> yes tried to get same quality.. only thing is ( and not sure if this makes sense but ) ... my NvTranscoder settings are simply -qp 21 vbr 1.8 - 2.2 .. it seems to respect both .. not sure why... but with nvenc if i use qp=21 it doesn't stay within bw boundaries.. and if i use bw boundaries.. it wont give me the same quality of qp=21 [22:07:14 CEST] <prelude2004c> so another reason i pushed for nvTranscoder [22:07:23 CEST] <prelude2004c> and i am not saying i am wrong.. its just my observation [22:07:30 CEST] <DHE> yep, GLUT required [22:07:33 CEST] <prelude2004c> i am NOT wrong * i mean [22:07:35 CEST] <BtbN> constqp and vbr at the same time makes no sense. [22:08:20 CEST] <prelude2004c> yup BtbN.. everyone says that and I agree sure.. but i am not sure how NvTranscoder is doing it :( [22:08:26 CEST] <BtbN> It can't. [22:08:34 CEST] <BtbN> Which rcmode did you pass to it? [22:08:38 CEST] <prelude2004c> i dont mind changing to be honest and using nvenc .. if we can solve the decoding issue ... then i am ready to change [22:08:48 CEST] <prelude2004c> whatever gets me great results and better support [22:09:48 CEST] <prelude2004c> i am not stuck on Nvtranscoder.. i am only trying to solve it and it has the best results so far. [22:10:14 CEST] <BtbN> <BtbN> Which rcmode did you pass to it? [22:10:25 CEST] <prelude2004c> 32 [22:10:35 CEST] <prelude2004c> ( 2pass ) .. and llhq [22:10:47 CEST] <BtbN> 32? [22:10:48 CEST] <DHE> ... low latency? [22:11:29 CEST] <prelude2004c> yup.. quality went up with it too [22:11:37 CEST] <DHE> ... I find that hard to believe [22:11:41 CEST] <prelude2004c> llhq gave me the best quality [22:11:53 CEST] <DHE> unless there is something catastrophically wrong with the bframe code [22:13:44 CEST] <prelude2004c> DHE, i wish i could help.. but i dont know [22:14:02 CEST] <BtbN> llhq yielding better quality is no news [22:14:28 CEST] <BtbN> hq has a better psnr, but llhq visually looks better. [22:14:48 CEST] <prelude2004c> yes llhq looks way better for me so i used it [22:16:54 CEST] <BtbN> qp_inter_p = (avctx->qmax + 3 * avctx->qmin) / 4; // biased towards Qmin [22:17:02 CEST] <BtbN> is it only me or are the + and * swapped? [22:18:23 CEST] <prelude2004c> BtbN, this could be a reason why my quality is reduced on nvenc ? not sure if we are still on the same topic here :) [22:18:31 CEST] <BtbN> no. [22:18:42 CEST] <BtbN> you're just using it wrong. [22:18:59 CEST] <prelude2004c> oh.. that's good then [22:19:25 CEST] <prelude2004c> so ok.. let's asume i can use nvenc and all is well in the world... and it uses the nvidea card... what should i do about the decoding? [22:19:51 CEST] <BtbN> Ask nvidia to implement propper non-Windows APIs. [22:20:03 CEST] <prelude2004c> how can we properly use the card instead of the CPU's [22:20:42 CEST] <prelude2004c> ya like that will happen :) .. i suspect i have to find another way... [22:20:45 CEST] <BtbN> There is only cuvid, which is only available in a usefull form on windows. [22:20:52 CEST] <BtbN> Or just use VDPAU. [22:20:56 CEST] <BtbN> Which already exists. [22:21:02 CEST] <prelude2004c> ya vdpau requires X and has other issues too [22:21:18 CEST] <prelude2004c> + cpu usage was a lot higher with vdpau vs just pipping to the nvtranscoder [22:21:42 CEST] <prelude2004c> eg. with vdpau it was improved for sure over CPU's but not even close to the nvtrancoder usage. [22:22:05 CEST] <BtbN> NvTranscoder itself is even documented as Windows-Only. [22:22:28 CEST] <prelude2004c> reallY? wow.. seems to have worked for me.. is the version i have differnet ? because it works great on my ubuntu with my video card. [22:22:32 CEST] <DHE> the SDK sample compiles under Linux [22:22:37 CEST] <DHE> that's all I've done though [22:23:02 CEST] <BtbN> So either their documentation is outdated or it works but is unsupported. [22:23:21 CEST] <prelude2004c> it def. works [22:23:33 CEST] <DHE> I have trouble believing it's unsupported if the latest SDK includes it in their list of samples (I downloaded this about 2 weeks ago) [22:23:36 CEST] <prelude2004c> asside from the timeout sisue.. its flawless [22:23:44 CEST] <BtbN> It is supported on Windows [22:23:55 CEST] <BtbN> But the high-level cuvid APIs are documented as Windows-Only. [22:24:13 CEST] <prelude2004c> BtbN, i don't think it uses cuda cores though [22:24:31 CEST] <BtbN> why would it? [22:24:40 CEST] <prelude2004c> the new cards ( eg. my M4000 ) card... has a built in encoder/decoder that is separate from the cuda cores.. ( from my understanding ) [22:24:51 CEST] <prelude2004c> if i am making sense.. not sure if i am :) [22:25:24 CEST] <prelude2004c> it used to be cuda cores before they moved the encoder/decoder stuff to the built in hardware. my old cards and platform use the cuda cores of the old cards [22:25:34 CEST] <prelude2004c> but no longer support the new ones.. hence the work required to create new encoders [22:25:40 CEST] <DHE> afk for an hour or so [22:26:29 CEST] <prelude2004c> DHE , thank you for your continued support in this stuff [22:26:33 CEST] <prelude2004c> much appreciated [22:26:46 CEST] <BtbN> http://developer.download.nvidia.com/assets/cuda/files/NVIDIA_Video_Decoder… doesn't even document the functions _at all_ [22:26:57 CEST] <BtbN> Not a single reference to cuvidCreateVideoParser in there [22:27:32 CEST] <prelude2004c> BtbN, whatever it is... it works.. can we take advantage of the fact that it works to pull that code and get it into nvenc ? [22:27:39 CEST] <BtbN> No [22:27:42 CEST] <prelude2004c> so nvenc will also support decoding without requiring vdpau [22:27:46 CEST] <BtbN> It takes a filename as input [22:27:49 CEST] <BtbN> so it's useless for ffmpeg. [22:28:01 CEST] <prelude2004c> i pushed stdin to it and it worked [22:28:14 CEST] <BtbN> I'm not going to implement that kind of hack-around [22:28:42 CEST] <prelude2004c> http://pastebin.com/raw/QcLgPjUG [22:29:00 CEST] <BtbN> ffmpeg decoders get raw bitstream [22:29:01 CEST] <BtbN> not files [22:29:04 CEST] <prelude2004c> that is the code i am using for it.. and i played around with a lot of settings on ffmpeg so please excuse if there are things wrong there [22:29:17 CEST] <BtbN> And a decoder that dumps to a temporary file would be horrible, specialy as it needs a muxer for that. [22:32:54 CEST] <jkqxz> pipe() + /dev/fd/N, maybe... (If it can take stdin then it does not require the input to be seekable.) [22:33:05 CEST] <BtbN> Good luck with that on windows. [22:33:44 CEST] <jkqxz> Is this some sort of horrible licensing hack? They have GPL code and proprietary code and have to create multiple processes and weird separation because their lawyers tell them to. [22:34:08 CEST] <BtbN> That's all proprietary and within their cuda library. [22:34:30 CEST] <BtbN> It either has a super low level API, that would need to be implemented as another hwaccell in ffmpeg. [22:34:39 CEST] <BtbN> Or one that takes a path [22:34:52 CEST] <BtbN> There does not seem to be any middle-layer that takes raw bitstream. [22:39:29 CEST] <prelude2004c> i am going to change up my code to use vpau and nvenc to compare. [22:40:00 CEST] <prelude2004c> just to test the difference [22:40:19 CEST] <prelude2004c> i loaded up vdpau already so let's see what happens [22:44:29 CEST] <BtbN> prelude2004c, "-b:v 2200k -2pass 1 -qmin 23 -qmax 23" is what you are telling nvTranscoder to do. [22:44:53 CEST] <BtbN> But it makes no sense. It's asking it to do basically constqp with a fixed bitrate. [22:45:29 CEST] <prelude2004c> you may be right so i am going to switch up to nvenc and compare the quality of the twho [22:45:41 CEST] <prelude2004c> then i will provide you two streams... you decide which looks better [22:59:37 CEST] <prelude2004c> dealing with this shit right now ( Xlib: extension "NV-GLX" missing on display ":0". ) .. [22:59:40 CEST] <andrey_turkin> BtbN: > qp_inter_p = (avctx->qmax + 3 * avctx->qmin) [22:59:47 CEST] <prelude2004c> vdpau is a pain in the @##@ [23:00:01 CEST] <andrey_turkin> it does that the comment says [23:41:47 CEST] <cone-885> ffmpeg 03ZhouXiaoyong 07master:2c7fd0e36b6d: avcodec/mips/h264qpel_mmi.c: Version 2 of the optimizations for loongson mmi [00:00:00 CEST] --- Tue May 31 2016

1 0

ffmpeg.log.20160530
by burek 31 May '16

31 May '16

[00:03:34 CEST] <CoJaBo> durandal_1707: bugtracker seems to be down.. [00:18:03 CEST] <aletorrado> Hi! I'm trying to specify codecs with the new dash muxer, but couldn't :( [00:18:29 CEST] <aletorrado> (ffmpeg -i something -c:v vp9 -f dash output) [00:22:40 CEST] <aletorrado> i'm getting "codec not currently supported in container" but dash is agnostic about codecs [00:27:45 CEST] <jkqxz> Yes, DASH itself doesn't care. But usually (and in this case) the fragments use MP4, which does. [00:28:37 CEST] <aletorrado> yeap. It's nice the all-in-one-command dash muxer. It would be great to use other containers as well [03:11:47 CEST] <Me4502> Hi, I'm using concat - and for every second text file I give it, I get "File-name: Invalid argument" [03:12:01 CEST] <Me4502> http://pastebin.com/B68tKCfT [03:12:10 CEST] <Me4502> That's an example of atext file that gives Invalid argument [03:12:35 CEST] <Me4502> http://pastebin.com/FzGCTUZb - This one works fine [03:15:26 CEST] <Me4502> ffmpeg -hide_banner -y -f concat -i filename.txt -c copy folder/main.mp4 [03:26:22 CEST] <ZeuZ> is anyone still around? [03:27:31 CEST] <ZeuZ> http://pastebin.com/Su8Zaies --> what am I lacking while linking? [03:43:10 CEST] <Me4502> http://pastebin.com/4usZSGPz [03:43:14 CEST] <Me4502> That's my full output [03:51:22 CEST] <Me4502> Okay, this is odd [03:51:31 CEST] <Me4502> Once the main video reaches 42 frames, it starts occuring [03:51:51 CEST] <Me4502> So it will only concatenate 42 frames before starting to fail [03:54:13 CEST] <Me4502> ffmpeg is already the newest version (7:3.0.2-2) - that appears to be the actual latest acording to the website [05:15:43 CEST] <drazin> anyone know a command or way to delete the first audiotrack from a directory of mp4 files? i messed up the first audiotrack and need to re-do them but need to delete the tracks first [06:18:44 CEST] <Admin__> good day ladies and gents.. i am segmenting source input into HLS output.. everything works well except sometimes the .ts segments run wild and do not break to the next segment file thus filling up the hard drive. Anyone know what causes this ? Is it some bug in the segmenting ? [06:19:58 CEST] <Admin__> code is super simple : http://pastebin.com/raw/DDGrtxY6 .. i may even have a few things there that should not be there but its all while testing [06:20:03 CEST] <Admin__> anyone have a clue? [06:35:58 CEST] <Admin__> DHE , moment of truth [06:36:13 CEST] <Admin__> 26.5 is about to roll over [06:49:03 CEST] <Admin__> DHE , no luck [06:49:20 CEST] <Admin__> still stopped working at exactly 26 hours and 30 minutes [08:33:12 CEST] <ffmpegishard> Hey, I'm trying to cut out pieces of a video file, but the timing seems to be a bit wanky. I'm using MPC-HC with high precision timer enabled to find out when the segments I want to extract start and end [08:34:10 CEST] <ffmpegishard> To test it, I tried to just extract the frame at the start and at the end using the command: ffmpeg -ss <start-time> -i <video file> -vframes 1 <output image file> [08:35:17 CEST] <ffmpegishard> And that gives the correct results, but when I then try to do: ffmpeg --ss <start-time> -i <video file> -to <end-time> -c copy <output-videofile> it doesn't start on the same frame, but a little before it [08:35:46 CEST] <ffmpegishard> thats -ss not --ss [08:36:54 CEST] <ffmpegishard> The other problem is that it also seems that since I use -ss the -to option is then relative to the starting mark set at -ss, which is fine, because I can just subtract the end from the start and put that in, but the problem is the start point [08:39:49 CEST] <ffmpegishard> When I tried to do: ffmpeg -i <video file> -ss <start-time> -to <end-time> -c copy <output-videofile> [08:40:45 CEST] <ffmpegishard> It gets the right start time, but before the first video shows up there is a little pause where only the audio plays. I can solve this by adding -an, which then produces the correct timing, but I would like to know why the audio plays beforehand [08:44:18 CEST] <ffmpegishard> I'm also very new to ffmpeg, so I don't really know, but I think it may have something to do with the "-c copy" part, because when I remove it, the previous command works: ffmpeg -i <video file> -ss <start-time> -to <end-time> <output-videofile> [08:44:26 CEST] <ffmpegishard> But it's a lot slower [11:05:11 CEST] <yagiza> Hello! [11:06:29 CEST] <yagiza> Do anyone know, how to play with ffplay an RTP stream, streamed by ffmpeg with opus encoder? [11:07:36 CEST] <yagiza> When I do: [11:07:36 CEST] <yagiza> ffmpeg -re -f lavfi -i aevalsrc="sin(400*2*PI*t)" -c libopus -ar 8000 -f mulaw -f rtp -sdp_file out.sdp rtp://127.0.0.1:1234 [11:07:36 CEST] <yagiza> it starts streaming something [11:11:21 CEST] <yagiza> Next, when I try: [11:11:22 CEST] <yagiza> ffplay rtp://127.0.0.1:1234 [11:11:22 CEST] <yagiza> It says: [11:11:22 CEST] <yagiza> [rtp @ 019029e0] Unable to receive RTP payload type 97 without an SDP file descr [11:11:22 CEST] <yagiza> ibing it [11:11:22 CEST] <yagiza> Input #0, rtp, from 'rtp://127.0.0.1:1234': [11:11:22 CEST] <yagiza> Duration: N/A, bitrate: N/A [11:11:23 CEST] <yagiza> Failed to open file 'rtp://127.0.0.1:1234' or configure filtergraph [11:11:23 CEST] <yagiza> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [11:12:22 CEST] <yagiza> Now, if I try [11:12:23 CEST] <yagiza> ffplay out.sdp [11:12:23 CEST] <yagiza> or [11:12:23 CEST] <yagiza> ffplay -i out.sdp [11:12:23 CEST] <yagiza> it says: [11:12:23 CEST] <yagiza> [rtp @ 01915a60] Protocol not on whitelist 'file'!sq= 0B f=0/0 [11:12:23 CEST] <yagiza> out.sdp: Invalid data found when processing input [11:12:35 CEST] <yagiza> What should I do? [11:29:35 CEST] <jcdejong> hello, I recently upgraded ffmpeg to 3.0 and the -shortest option is acting strange for me since then.. instead of the expected 198 frames I get a video of 249 frames.. is this the right place I could find somebody who could help me? [11:30:36 CEST] <jcdejong> I already spent a lot of time searching and debugging, and am kinda stuck.. So I hoped I could find someone here who could review what I did or point me in the right direction [11:32:37 CEST] <jcdejong> here's a pastebin with my commands: http://pastebin.com/AK4SUC70 it looks like it has something to do with the .mov overlay I'm using (and which I want to cut of at the end of the input file) [14:23:41 CEST] <ZeuZ> hey guys, where can I find in the docs/examples a snippet of using the concat from C? [16:23:18 CEST] <Seekha> Hi, is there a way to straem a mpegts stream (with scrambled video) on UDP? It seems not to work [16:24:00 CEST] <Seekha> I do something like : ffmpeg -re -i myfile_.ts -vcodec copy -acodec copy -f mpegts udp://host:port [16:24:07 CEST] <Seekha> (or withtout the -re) [16:24:27 CEST] <Seekha> with non scrambled video it works... what I do wrong? [16:42:28 CEST] <drazin> can anyone lead me to a script or a command that can set the disposition of audio tracks for multiple files without converting or processing anything? [17:39:35 CEST] <CoJaBo> durandal_1707: So anyway, I finally did figure out how to reproduce that bug :/ [17:51:55 CEST] <yagiza> Still nobody, who knows how to play RTP stream with ffplay? [17:59:23 CEST] <nucular> Hey, given a series of images as the input, how would I shift them by one frame (skip one frame and loop) without renaming files manually? [18:16:59 CEST] <furq> nucular: probably -start_number 1 [18:29:06 CEST] <ATField> (hello) I am having problems with defining multiple force_style parameters. While both [-vf subtitles='D\:\\INPUT.ass':force_style="FontName='Gotham Black'"] and [-vf subtitles='D\:\\INPUT.ass':force_style="FontSize=22" "02 force style 790k,fontSize22.webm"] work fine, trying to combine them into [-vf subtitles='INPUT.ass':force_style="FontName='Gotham Black',FontSize=22"] seems to contain... [18:29:08 CEST] <ATField> ...some syntax error because it confuses ffmpeg it returns a No such filter: 'FontSize'. [18:29:40 CEST] <ATField> The problem seems to be with quotemarks, but no idea what order should be the proper one. [18:30:06 CEST] <ATField> Can you help with some hints, please? [18:31:04 CEST] <furq> force_style="FontName=Gotham Black,FontSize=22" [18:31:07 CEST] <furq> looks like it should work [18:33:23 CEST] <ATField> [-vf subtitles='D\:\\INPUT.ass':force_style="FontName=Gotham Black,FontSize=22"] returned the same error (No such filter: 'FontSize' Error opening filters!). [18:36:13 CEST] <furq> it works with single quotes [18:40:28 CEST] <ATField> (its still rendering, but) Replacing them with single quotes seems to work if the fonts name doesnt have spaces (otherwise it gets confused again). [18:41:32 CEST] <ATField> ( [ -vf subtitles='D\:\\INPUT.ass':force_style='FontName=Sylfaen,FontSize=22' ] ) [18:42:31 CEST] <furq> 'FontName="Gotham Black",FontSize=22' works [18:42:40 CEST] <ATField> ok, adding double seemed to help [18:42:40 CEST] <furq> the docs could probably use another example or two [18:43:34 CEST] <ATField> I was trying it out when you posted it : p. Also, I thought in case of using multiple quote types at once, the doulbe-quotes were supposed to be the external pair? Or it depends on the syntaxs flavour? [18:44:50 CEST] <ATField> Yeah, there was also a problem with defining .ass location: it needed a bunch of weird escape characters, like with drawtext. [18:46:38 CEST] <ATField> One more thing: when I specify the font, it still ends up rendered in cursive \ Italics. Can I prevent this to force non-cursive? [18:49:13 CEST] <WereCatf> I tried asking about this the other day, but got nothing useful, so I'll try again: is anyone using NVIDIA's NVENC via ffmpeg? Something has changed as it no longer works for me, no matter what I do. It used to work fine and I'm wondering if anyone else has the same issue or if someone knows what's wrong. [18:55:25 CEST] <ATField> All right! It worked with the font I needed (sans the cursive) ! Thanks for help. [18:55:53 CEST] <ATField> WereCatf: Why not ask on stackexchange? More people likely to see the question that way. [18:56:37 CEST] <WereCatf> ATField: Um, because the problem is with ffmpeg and this is ffmpeg's IRC-channel? [18:57:47 CEST] <ATField> I know, but the more exposure the better, right? The platform there would also improve readability for both sides (just saying). [18:59:48 CEST] <Demon_Fox> Is there a page listing a great many common container formats (like ogg), that lists the supported media types they can contain? [19:01:57 CEST] <ATField> Demon_Fox: Like this? https://en.wikipedia.org/wiki/Comparison_of_video_container_formats [19:03:53 CEST] <Demon_Fox> By chance, do you have a non-wiki link? [19:05:30 CEST] <Demon_Fox> I'm writing up an explanation of codecs and containers for some people I know [19:06:48 CEST] <kepstin> Demon_Fox: for many containers, there really isn't any definite list of supported codecs. And the ones that do have a list, it's scattered all over the places through various standards organizations and whatnot :/ [19:07:00 CEST] <Demon_Fox> I see [19:07:04 CEST] <Demon_Fox> Thanks [19:07:23 CEST] <furq> a lot of formats have unofficial extensions as well [19:07:55 CEST] <kepstin> e.g. you can throw pretty much anything in avi if you make up a fourcc and install (on windows) a vfw codec that uses that. [19:08:43 CEST] <Demon_Fox> I wonder how bad vlc is with this [19:08:58 CEST] <furq> it'd be nice if `ffmpeg -h muxer=mp4` showed allowed codecs rather than just the defaults [19:09:07 CEST] <furq> although that'd probably get a bit unwieldy for mkv and the like [19:09:30 CEST] <CoJaBo> What other formats are there for piping uncompressed frames to ffmpeg? [19:09:32 CEST] <kepstin> furq: yeah, but as we discovered last week, there's a function to see if a codec is supported in a container but most containers don't have it implemented :/ [19:09:48 CEST] <furq> there's obviously something in ffmpeg which checks, though [19:09:57 CEST] <furq> otherwise it would let you mux anything into mp4 [19:10:04 CEST] <kepstin> many containers in ffmpeg do have a list of codecs that's checked in the open function [19:10:24 CEST] <kepstin> but that's not a list of codecs the container supports, that's a list of codecs that ffmpeg knows how to mux into the container [19:10:35 CEST] <Demon_Fox> I don't know of any formats that matroska doesn't support [19:10:39 CEST] <furq> well that's usualy the same thing as far as an ffmpeg user is concerned [19:10:40 CEST] <furq> +l [19:10:50 CEST] <furq> and there are some subtitle formats which don't work with mkv [19:10:55 CEST] <furq> at least not through ffmpeg [19:11:01 CEST] <Demon_Fox> yeah... [19:11:09 CEST] <WereCatf> That's not really a fault with the container-format, though, is it? [19:11:16 CEST] <Demon_Fox> Thanks for all of the support :) [19:11:21 CEST] <kepstin> (getting the check function to work should be simple for most containers in ffmpeg - factor out the checking code from the open function so it can be shared with the check function. SOmeone just needs to do it :/) [19:11:32 CEST] <Demon_Fox> the format support that is :) [19:11:37 CEST] <WereCatf> I was under the impression that you can use pretty much anything with Matroska if you wish, it's just up to the software to support it [19:11:56 CEST] <Demon_Fox> Now I can use this to find what I'm looking for: "ffmpeg -h muxer=X" [19:12:07 CEST] <furq> Demon_Fox: that just shows the default formats [19:12:21 CEST] <furq> i.e. what you get if you specify a format but no codecs [19:12:26 CEST] <Demon_Fox> I mostly just need the common one's [19:12:29 CEST] <kepstin> one of the really tricky things about matroska is that it has support for storing anything that can be stored in AVI via a compatibility mode [19:12:32 CEST] <Demon_Fox> supported by a given format that is [19:12:43 CEST] <Demon_Fox> By the way, why was avi extended to support as much as it does? [19:13:04 CEST] <furq> stupidity, i assume [19:13:06 CEST] <CoJaBo> Because it's the only thing most players support lol [19:13:16 CEST] <Demon_Fox> Seriously? [19:13:17 CEST] <kepstin> Demon_Fox: avi is a really simple container, so it was pretty trivial to extend. [19:13:22 CEST] <Demon_Fox> I see [19:13:35 CEST] <CoJaBo> It would simply make too much sense to support a new format, therefore extending the existing one until it's buggy as all hell [19:13:48 CEST] <Seekha> Someone has sucessfully transmitted a mpegts (with scrambled video) over UDP with ffmpeg ? [19:13:57 CEST] <kepstin> you just pick a codec id ("fourcc") that nobody was using, and make codec for windows that registers to that, and then any windows player could play it. [19:13:58 CEST] <furq> people preferred to hack up an existing container rather than use something like matroska which is designed for it [19:14:13 CEST] <Seekha> Not scrambled it's ok, but when it's scrambled it doesn't transmit anything [19:14:33 CEST] <Demon_Fox> I have some false assumptions about what supports what now [19:14:39 CEST] <Fyr> guys, does s16le equal pcm_s16le? [19:14:44 CEST] <furq> yes [19:14:49 CEST] <Demon_Fox> I just kind of thought new format support for avi would be dead [19:14:50 CEST] <Fyr> (for FFMPEG syntax) [19:14:52 CEST] <Fyr> & [19:14:53 CEST] <Fyr> ? [19:14:55 CEST] <furq> Demon_Fox: that wiki link is the best thing i know of [19:14:59 CEST] <furq> https://en.wikipedia.org/wiki/Video_file_format [19:15:00 CEST] <furq> or that one [19:15:38 CEST] <kepstin> well, the support for h264 and other codecs of similar vintage in avi is pretty hacky :/ [19:15:51 CEST] <furq> yeah i know avi has issues with vorbis audio [19:15:55 CEST] <furq> which is why some dreadful people created .ogm [19:16:48 CEST] <Demon_Fox> Hopefully soon, AV1 will take over when it's released [19:16:55 CEST] <furq> av what now [19:17:13 CEST] <Demon_Fox> Let me link it [19:17:25 CEST] <Demon_Fox> http://aomedia.org/about-us/ [19:17:28 CEST] <Demon_Fox> oops [19:17:37 CEST] <Demon_Fox> http://aomedia.org/ [19:17:50 CEST] <furq> are they making a container format as well [19:17:54 CEST] <furq> that seems like a waste of time [19:17:54 CEST] <Jan-> technical question: is the motion compensation the most time-consuming part of h.264 encoding? [19:18:12 CEST] Action: kepstin notes that "av1" is nearly impossible to search for, since google autocorrects it to "avi" [19:18:12 CEST] <Demon_Fox> Alliance for Open media is taking the existing next-gen open codecs and trying to make something with them and be done by 2017 and in use later that year [19:18:28 CEST] <furq> i thought they were just making a new codec [19:18:39 CEST] <furq> and then lying about it being production-ready by 2017 because there's no chance [19:19:06 CEST] <Demon_Fox> It's mainly companies that are tired of being extorted by patent holders after already paying royalties [19:19:32 CEST] <kepstin> yeah, supposed to be some new codec using ideas from stuff like vp9 and daala and whatnot? [19:19:44 CEST] <furq> if they can even improve on vp9 i'll be impressed [19:19:49 CEST] <furq> or rather, improve upon libvpx-vp9 [19:19:59 CEST] <Demon_Fox> vp8, vp9, Thor, Daala, and I think that's all [19:20:05 CEST] <Demon_Fox> There might be more [19:20:21 CEST] <Demon_Fox> However, I think the alternative name for av1 is supposed to be vp10 [19:20:33 CEST] <furq> oh are they the same thing [19:20:37 CEST] <kepstin> hmm, daala had some neat stuff going on with the chroma-from-luma predictions and whatnot [19:21:01 CEST] <kepstin> but it was all pretty experimental [19:21:16 CEST] <furq> tbh i'd rather they finished writing their vp9 encoder [19:21:41 CEST] <Jan-> hasn't the mpeg licensing organisation claimed that basically development of any codec is now impossible without their patents? [19:21:58 CEST] <furq> yeah and turkeys have also claimed christmas sucks [19:22:07 CEST] <Jan-> Boo. Cake. [19:22:11 CEST] <Jan-> Oh no wait I mean, yay, cake! [19:22:12 CEST] <CoJaBo> kepstin: Put it in quotes [19:22:55 CEST] <kepstin> CoJaBo: i tried, but still didn't find anything useful :/ [19:22:59 CEST] <furq> have mpeg-la gone after vp8/9 yet [19:23:00 CEST] <CoJaBo> kepstin: Queries like "av1" -"avi" is sometimes necessary too, as it ocasionally ignores the quotes [19:23:08 CEST] <furq> i know they were threatening that they could do that if they wanted to [19:23:25 CEST] <kepstin> hmm. I guess 'xvp8' is dead and gone now [19:23:30 CEST] <Jan-> is it me or is google getting a bit scary these days [19:23:36 CEST] <Jan-> it knows that "lemon meringue" is clearly "pie" [19:23:55 CEST] <kepstin> furq: mpegla and google came to a settlement about vp8/9, iirc [19:24:08 CEST] <Jan-> and "coach darla black neon" is "pink sneakers" [19:24:10 CEST] <Jan-> this is strange [19:24:21 CEST] <furq> oh so they did [19:24:47 CEST] <furq> or they did for vp8 anyway [19:25:51 CEST] <Demon_Fox> The development of any new video codec is not impossible without their patents [19:26:02 CEST] <Demon_Fox> Daala doesn't even use standard DCT [19:26:10 CEST] <furq> wasn't that what dirac was going for [19:26:12 CEST] <Demon_Fox> It uses overlapping transforms like audio [19:26:24 CEST] <furq> or is going for, if they're still working on it [19:26:25 CEST] <Demon_Fox> dirac was wavelet transforms [19:26:51 CEST] <furq> i meant about avoiding mpeg-la patents [19:26:57 CEST] <Demon_Fox> Daala was a research project to actively defy the modern approaches to codecs [19:28:01 CEST] <Demon_Fox> furq, They are doing all they can to avoid them, taking a lot of inspiration from opus [19:28:31 CEST] <Demon_Fox> including some vector based prediction process [19:28:52 CEST] <kepstin> if you're interested in the daala tech, reading through the tech demos on https://xiph.org/daala/ is kinda neat. [19:29:18 CEST] <Demon_Fox> It's the neatest thing I've seen in a while [19:29:49 CEST] <Demon_Fox> It's supposed to actually be competing with h.265 in file size [19:30:19 CEST] <Demon_Fox> AV1 is supposed to just take the current open tech and turn it into something useful for right now while they develop [19:31:55 CEST] <Demon_Fox> xiph and the alliance for open media are however implementing protective patents [19:32:23 CEST] <Demon_Fox> Basically, patenting the new techniques and using an open no-royalty license for their contents [19:35:12 CEST] <Demon_Fox> Anyways, thanks for the wikipedia pages [19:36:36 CEST] <kepstin> but yeah, I suspect that the folks working on "av1" are just gonna keep using the webm (matroska) container for it [19:42:51 CEST] <WereCatf> I tried asking about this the other day, but got nothing useful, so I'll try again: is anyone using NVIDIA's NVENC via ffmpeg? Something has changed as it no longer works for me, no matter what I do. It used to work fine and I'm wondering if anyone else has the same issue or if someone knows what's wrong. [20:06:20 CEST] <ATField> Where can I find a listing of all possible arguments for (subtitle) force_style? The Docs dont mention much about it, aside from FontName, PrimaryColour, and fontsize. [20:09:36 CEST] <furq> ATField: i assume it accepts any ass format [20:09:38 CEST] <furq> https://www.matroska.org/technical/specs/subtitles/ssa.html [20:09:47 CEST] <furq> there's a list there although i don't know if it's complete [20:12:15 CEST] <CoJaBo> What other formats are there for piping uncompressed frames to ffmpeg? [20:13:26 CEST] <ATField> furq: Oh, that makes sense. Thanks. [20:16:57 CEST] <ATField> It accepts a string containing ASS style format KEY=VALUE couples separated by ",". I guess my brain just decided to ignore the plainly written text. x_x [20:22:18 CEST] <f00bar80> http://vpaste.net/OypxP << I'm asking if this is a correct 2 Passes encoding script , or anything is missing [20:30:00 CEST] <f00bar80> ppl any comment? [20:38:08 CEST] <kepstin> f00bar80: m3u8 isn't an ffmpeg muxer, did you want segmented hls? [20:38:47 CEST] <kepstin> for 2 pass you need to wait for the first pass to finish and write the stats file before starting the second pass with the same input [20:39:06 CEST] <kepstin> I'm not sure what you're doing there, but running both at the same time certainly doesn't make sense... [20:40:03 CEST] <f00bar80> kepstin: so how then I can run the 2 Passes in script ? [20:40:25 CEST] <kepstin> remove the & at the end so it waits for one to finish before starting the second [20:40:51 CEST] <kepstin> (obviously, if you're doing a live stream, 2-pass is impossible) [20:40:54 CEST] <f00bar80> kepstin: you mean the & at the end of the first command , right? [20:41:08 CEST] <f00bar80> kepstin: why it's impossible ? [20:41:31 CEST] <kepstin> because in order to run the second pass, you need to wait for the first pass to finish [20:41:41 CEST] <kepstin> then do a second encode of the exact same source data [20:41:57 CEST] <f00bar80> kepstin: if it's the case .. then what can be an alternative for the pass , regarding quality enhancement for live streaming ? [20:42:48 CEST] <f00bar80> kepstin: and doesn't by removing & in the first pass, gonna to work it out that way you meantioned? [20:42:50 CEST] <kepstin> f00bar80: you have to define the exact requirements of the live streaming - what bitrate, etc. do you need. In some cases (e.g. lan streams) you can maybe use -crf mode, but most of the time you're gonna be stuck with 1-pass mode. [20:43:20 CEST] <kepstin> f00bar80: it depends exactly what your source is whether that would work [20:43:30 CEST] <f00bar80> I have the bit rate already defined .. [20:43:54 CEST] <kepstin> if you play that source stream, then wait 10 minutes, then play the source stream again, does it play the exact same thing both times? [20:44:06 CEST] <f00bar80> kepstin: it's a mpegts , that's the source [20:44:29 CEST] <kepstin> so is the source live? or is it a pre-recorded file? [20:44:31 CEST] <f00bar80> kepstin: , no .. it does play a live content [20:44:45 CEST] <kepstin> ok, then yes, doing 2-pass with that as a source would work fine. [20:44:52 CEST] <kepstin> er wait [20:44:58 CEST] <kepstin> if it's live, then no 2pass wouldn't work [20:45:06 CEST] <f00bar80> :( [20:45:20 CEST] <kepstin> to do 2-pass, you'd have to save a bunch of the content locally, then do the encode twice from your local file [20:45:29 CEST] <f00bar80> kepstin: so which alternatives .. I do have then [20:45:47 CEST] <kepstin> or just use a 1-pass mode if you need it to be done live/realtime [20:46:17 CEST] <furq> use a higher bitrate or a slower preset [20:46:21 CEST] <kepstin> (either -crf if you have no bitrate limits to worry about - e.g. high-speed lan, or use 1-pass bitrate) [20:46:40 CEST] <furq> i assume from the vbv stuff that there is a bitrate constraint [20:46:49 CEST] <furq> or that this command was copied from stackoverflow [20:47:01 CEST] <f00bar80> yes there's a bitrate [20:47:46 CEST] <CoJaBo> What other formats are there for piping uncompressed frames to ffmpeg? [20:48:13 CEST] <furq> ffmpeg -formats | grep pipe [20:48:50 CEST] <kepstin> CoJaBo: not sure what you mean; I normally use 'rawvideo' which can accept uncompressed frames in pretty much any pixel format you like. [20:49:59 CEST] <CoJaBo> kepstin: It's a PITA tho, as I have to specify every parameter on both sizes of the pipe :/ [20:50:22 CEST] <f00bar80> being able to identify the better bitrate I have to use, is by try an error ? also preset slow vs superhigh how it affects the output quality ? [20:50:55 CEST] <kepstin> f00bar80: pick the slowest preset you can which still encodes video fast enough for your application. [20:51:41 CEST] <kepstin> and you identify the bitrate to used based on what you want the output for. If the output is for streaming video online, you pick a bitrate that fits in the network speed you have. [20:52:33 CEST] <furq> CoJaBo: you could try yuv4mpegpipe [20:52:40 CEST] <f00bar80> kepstin: so again 2 passes won't work in this case even by removing the & at the first pass ? [20:52:57 CEST] <furq> assuming the frames are yuv and not rgb [20:52:59 CEST] <CoJaBo> furq: Needs to be uncompressed RGB tho [20:53:07 CEST] <furq> you're pretty much stuck with rawvideo then [20:53:15 CEST] <CoJaBo> That really sucks :/ [20:53:24 CEST] <kepstin> f00bar80: no. the way 2-pass work is that once the first pass is done, it writes a stats file to disk. the second pass then at the start opens the stats file and uses it to make better decisions about how to encode the /same/ input video. [20:53:57 CEST] <CoJaBo> furq: What is the pixel order on that? [20:54:05 CEST] <CoJaBo> or is it settable [20:54:07 CEST] <f00bar80> kepstin: so what that thing which doesn't allow that to happen ? [20:54:31 CEST] <furq> CoJaBo: no idea, it doesn't seem to be documented anywhere [20:54:52 CEST] <kepstin> your commands (if you ignore the other errors) would cause the second pass to start before the first pass completes, so it would give an error saying the stats file is missing. And since the source video is live, the input isn't the same, so the stats file is useless anyways. [20:56:15 CEST] <f00bar80> kepstin: is the no way to make the second pass to wait till there's stats file? or cause this is a live stream , there'll be no stats file as it's always playing [20:56:27 CEST] <furq> both [20:57:22 CEST] <f00bar80> so the only option i have is to play with the bitrate and the preset values ? [20:57:30 CEST] <furq> yes [20:57:42 CEST] <CoJaBo> furq: My input stream is a sequence of bitmaps, which are bottom-up; is there a way to flip the video in ffmpeg if I need to? [20:57:49 CEST] <kepstin> 2-pass mode is only usable for non-realtime applications where you have the complete video available in advance. [20:58:00 CEST] <furq> CoJaBo: is it raw bitmap data or bmp files [20:58:05 CEST] <furq> if it's the latter you can use image2pipe [20:58:17 CEST] <CoJaBo> furq: image2pipe is broken [20:58:27 CEST] <furq> oh [20:58:32 CEST] <furq> what's wrong with it [20:58:34 CEST] <CoJaBo> Bug 5598 [20:58:47 CEST] <CoJaBo> It crashes ffmpeg [20:58:49 CEST] <furq> fun [20:59:08 CEST] <CoJaBo> My problem is that I need a work-around till that's fixed :/ [20:59:39 CEST] <furq> is it really that big a problem to have to specify params twice [20:59:41 CEST] <CoJaBo> And one of the comments already suggested it could be a "wontfix" issue for some weird reason :/ [20:59:43 CEST] <f00bar80> kepstin: what about using hls-flags delete-segments accompanied with bitrate/preset? [20:59:56 CEST] <furq> that doesn't make any difference [20:59:59 CEST] <CoJaBo> furq: It sorta is; >_> [21:00:13 CEST] <f00bar80> kepstin: this by any mean .. enhance quality? [21:00:53 CEST] <kepstin> f00bar80: what 2-pass mode does is analyze the *entire length* of the video to decide which parts need more bitrate and which parts are ok with less. It only makes sense if have the *entire length* of the video saved on disk somewhere. [21:01:56 CEST] <kepstin> f00bar80: the delete segments is irrelevant to video quality, that just frees up disk space if you don't need the entire output saved. [21:01:59 CEST] <f00bar80> kepstin: and that ..can't be applied on live streams by anyway ? I assume that it's an important thing to be done [21:02:50 CEST] <f00bar80> kepstin: "decide which parts need more bitrate and which parts are ok with less [21:02:58 CEST] <f00bar80> kepstin: i mean [21:03:04 CEST] <kepstin> f00bar80: i'm not sure how you'd analyze the entire length of a live stream before you start encoding. [21:03:15 CEST] <kepstin> without a time machine [21:03:40 CEST] <CoJaBo> furq: In any case, I still need to flip the video, otherwise it'll come out upside-down [21:04:00 CEST] <kepstin> furq: well, if the image2pipe stuff worked, the bmp decoder would take care of that :/ [21:04:07 CEST] <f00bar80> kepstin: so the only option ,.. is bitrate/preset try/error? [21:04:11 CEST] <kepstin> er, CoJaBo ^^ [21:04:32 CEST] <kepstin> f00bar80: no trial and error for bitrate - you pick based on the requirements of your network. [21:04:47 CEST] <kepstin> f00bar80: some trial and error on preset, you pick the slowest one that is still fast enough [21:05:12 CEST] <furq> it's unlikely that you'll see any significant improvements with anything below slow [21:05:40 CEST] <kepstin> anything below slow you'd need a really impressive system to get realtime anyways :) [21:05:52 CEST] <furq> yeah especially with that number of streams [21:07:15 CEST] <furq> f00bar80: you can probably reduce the audio bitrate a bit [21:08:01 CEST] <CoJaBo> kepstin: Yes, but I can't use image2pipe at all :/ [21:08:42 CEST] <kepstin> CoJaBo: but yes, if you can get the image data into ffmpeg at all, you can flip it pretty easily with video filters. [21:08:58 CEST] <CoJaBo> Yeh, but getting it there is hte hard part lol [21:09:08 CEST] <furq> i take it you're stuck with bmp [21:10:12 CEST] <CoJaBo> I could potentially convert to another uncompressed RGB, but not easily [21:10:49 CEST] <CoJaBo> I can directly support PNG, but it's unbelievably slow, and JPEG, but the quality is awful [21:11:27 CEST] <kepstin> hmm. you can't set the png compression down to minimum in hopes of making it faster? :/ [21:12:04 CEST] <kepstin> i guess most png encoders don't have an option to store completely uncompressed image data in png bitstream format ;) [21:13:10 CEST] <CoJaBo> No option for quality, I'd have to write my own encoder [21:13:33 CEST] <CoJaBo> It is possible to do uncompressed png, but I think the checksums would still murder me speed-wise [21:13:52 CEST] <furq> have you tried bmp_pipe [21:14:04 CEST] <CoJaBo> bmp_pipe is the same as image2pipe, apperently [21:14:06 CEST] <furq> i'm guessing it's just an alternate name for image2pipe but it's not documented anywhere so i could be wrong [21:14:20 CEST] <furq> nvm then [21:14:28 CEST] <CoJaBo> yeh, and noe of this is documented >_> [21:15:05 CEST] <CoJaBo> I'm guessing they's why they're so quick to suggest a wontfix, the feature I broke isn't even one that's documented D= [21:15:11 CEST] <furq> you could just output raw bitmap data and use -f rawvideo which would at least save you the flip [21:15:23 CEST] <furq> that's probably not the solution you want though [21:15:43 CEST] <CoJaBo> I have to get the raw data by peeling it out of the bitmaps [21:16:16 CEST] <CoJaBo> And there is a stupid amount of overhead for reading data in this thing [21:16:48 CEST] <kepstin> do you have a bunch of bmp images on disk, or is some kind of in-ram thing? [21:16:58 CEST] <CoJaBo> They're generated on-the-fly [21:17:28 CEST] <CoJaBo> The app renders the image, converts it to BMP, then writes that bmp to a fifo [21:19:13 CEST] <kepstin> CoJaBo: hmm. the image2pipe demuxer has an option 'frame_size' that takes a value in bytes. since the bmps are all gonna be the same size, you could try to set that maybe? [21:19:24 CEST] <kepstin> as a workaround for the probe being weird [21:19:56 CEST] <CoJaBo> kepstin: ..where did you even find that? [21:20:19 CEST] <kepstin> ffmpeg -h demuxer=image2pipe [21:21:14 CEST] <furq> i like how many help responses in this channel contain "you could maybe try" [21:24:46 CEST] <CoJaBo> kepstin: It's really weird it doesn't set that automatically.. [21:25:08 CEST] <kepstin> CoJaBo: yeah, sounds like the bmp parser is horribly busted, at least when used with pipe input [21:28:19 CEST] <CoJaBo> ............damn [21:28:46 CEST] <CoJaBo> kepstin: Setting frame_size causes a memory leak; it just crashes a slightly different way :/ [21:29:43 CEST] <kepstin> the actual bmp decoder in ffmpeg works perfectly fine, it's just an issue with the bmp parser in image2pipe :/ [21:30:21 CEST] <kepstin> (I wrote the current 'gdigrab' code, which gets a stream of DIB (bmp) images from a windows api, and sends them to the bmp decoder inside ffmpeg) [21:35:50 CEST] <CoJaBo> kepstin: Wait no, I screwed up [21:36:04 CEST] <CoJaBo> kepstin: I put it after the -i; it works fine put at the proper place [21:36:24 CEST] <CoJaBo> kepstin: Care to write a new one for image2pipe? =D [21:36:26 CEST] <kepstin> hmm. I fond the current image2pipe code a bit hard to read, but it *looks* like it by default just reads the input in 4KiB chunks and passes that right to the decoder, rather than parsing the frame sizes and putting each complete frame into a buffer [21:36:35 CEST] <kepstin> i'm surprised it works at all with any codec :/ [21:36:53 CEST] <CoJaBo> Where is that code located? [21:37:03 CEST] <kepstin> libavformat/img2dec.c [21:40:08 CEST] <kepstin> hmm. I guess the idea was they wanted to do as little parsing of the format as possible in the "demuxer" and rely on the decoder to be able to find frame boundaries [21:40:34 CEST] <kepstin> which works on some of the more structured formats, but not with the bmp decoder :/ [21:41:44 CEST] <kepstin> so yeah, I guess there's two way to fix it - make image2 able to parse the images enough to figure out frame sizes, or make the bmp decoder less dumb so it knows how much data the frame's supposed to contain :/ [21:41:53 CEST] <CoJaBo> ..actually not sure how that could work with any piped image format :/ [21:42:07 CEST] <CoJaBo> Where is the bmp decoder? [21:42:49 CEST] <CoJaBo> The total size of the bmp frame is literally offset 2 in the header... [21:42:50 CEST] <kepstin> in the directory where all the codecs are, in files that contain the name of the codec in the filename. [21:43:05 CEST] <CoJaBo> And there's even a cli parameter to set that size >_> [21:43:32 CEST] <kepstin> I suspect it's bmp_parse.c where the real fix is needed. [21:43:49 CEST] <CoJaBo> bmp_parser.c? [21:43:57 CEST] <kepstin> but that does attempt to read the actual frame size... [21:43:58 CEST] <kepstin> yeah [21:45:45 CEST] <CoJaBo> ..yeh, I see that too. It reads the size, but then why is it acting like it isn't? [21:49:21 CEST] <kepstin> man, this bmp parser code is hard to read. no comments, and it's storing a bunch of state all mashed up into a uint64 in non-obvious ways [21:50:56 CEST] <CoJaBo> ..yeh, I can't figure out what any of those if blocks are doing [21:51:37 CEST] <CoJaBo> if I knew what all those parameters were and what the output needed to be, I'd rewrite the thing from scratch myself >_> [21:53:58 CEST] <WTDs> Hello, I'm new on the chat and to ffmpeg too. I have a problem demuxing VOB file to mpeg. I try to archive our family dvds with no losses in quality. I figured I can cat all the VOB files together then run them through ffmpeg -codec copy to demux/fix timestamps. But i got one DVD this didnt worked, getting error messages "Non-monotonous DTS in output stream" Do any of you know why could it be? [21:54:27 CEST] <f00bar80> on a 4 cores cpu i can run 16 h.264 simultaneous encoding safely ? [21:56:30 CEST] <kepstin> WTDs: hmm. you probably tried to combined vob files from different dvd titlesets. You should concatenate e.g. the files named VTS_01_{01,02,03,..}.vob to one file, then VTS_02_{01,02,03,...}.vob to another. [21:57:59 CEST] <kepstin> er, VTS_XX_{1,2,3,...}.VOB - I forgot that the second number only has 1 digit not 2 :) [21:59:14 CEST] <kepstin> f00bar80: depends a lot on the type of cpu, but you'll probably have to use a pretty fast x264 preset to pull that off realtime (which means lower quality) [22:01:13 CEST] <f00bar80> kepstin: what do you mean by fast x264 [22:01:47 CEST] <kepstin> f00bar80: I said "fast x264 preset", you missed a word. [22:02:27 CEST] <f00bar80> kepstin: so preset superfast , right? [22:02:44 CEST] <kepstin> f00bar80: for that kind of encoding job, I'd either go with a higher-end many core server chip (xeon) or *maybe* consider using nvidia quadro cards. Either way, expensive hardware, needs a lot of testing to find the right balance. [22:02:56 CEST] <kepstin> f00bar80: you have to try and see. I don't have your machine! [22:40:54 CEST] <ChocolateArmpits> 5% load per encode at best [22:41:38 CEST] <ChocolateArmpits> I don't imagine coding hd streams at the level [23:13:31 CEST] <grrk-bzzt> Hello [23:14:21 CEST] <grrk-bzzt> Is FFV1 supposed to have a good compression ratio, or is it just supposed to be a little less heavy than rawvideo? [23:15:43 CEST] <kepstin> ffv1 should have fairly good compression on many types of videos. It's often used as a video archival codec. [23:16:35 CEST] <grrk-bzzt> I had a screen recording using x265 lossless which had more than 5hours of nothing (plain black) and it compressed very well (<100 MBytes) and when converted to FFV1 it jumped to a 3GB file [23:17:16 CEST] <grrk-bzzt> So I don't know if I'm not using the correct settings [23:17:26 CEST] <furq> ffv1 is intra-only so it's not well-suited to that kind of material [23:17:33 CEST] <grrk-bzzt> Alright [23:19:47 CEST] <kepstin> huh, I thought it actually supported some non-intra mode? or at least gop size >1? [23:19:57 CEST] Action: kepstin doesn't know much about the codec itself tho [23:36:38 CEST] <DHE> keep in mind x264/5 lossless requires attention on your part to ensure the colourspace isn't reduced to yuv420 first [23:37:22 CEST] <CoJaBo> DHE: Latest versions seem to default to 444 (if converting from RGB, anyway) [23:37:44 CEST] <CoJaBo> Threw me for a loop, older ones would assume 420 (not much supports 444 >_>) [23:38:14 CEST] <CoJaBo> 11.5×14 [23:39:14 CEST] <grrk-bzzt> DHE, I'm working with retrogaming videos. Usually there's two cases. Either the pixel format is pal8, or I'm using it for screen recording. In either case, I'm using the following options: -flags +bitexact -sws_flags +full_chroma_inp+accurate_rnd+bitexact -fflags +bitexact -pix_fmt yuv420p [23:39:46 CEST] <grrk-bzzt> s/either/both [00:00:00 CEST] --- Tue May 31 2016

1 0

ffmpeg-devel.log.20160529
by burek 30 May '16

30 May '16

[00:24:41 CEST] <DHE> well, I have my sample [00:35:10 CEST] <prelude2004c> hello, just to reiterate what DHE and i have been looking at... i am having a problem with live TS streams.. i get signals from Multicast UDP live stream sources and they are either mpeg2video or h264 ... i am simply doing http://pastebin.com/raw/vxiuF0Xd [00:35:42 CEST] <DHE> my sample TS video looks fine in wireshark, but isn't working properly in ffmpeg. so I'll have to figure this one out.. [00:35:57 CEST] <prelude2004c> the issue is 2 fold ( and they may not be related )... but 1. every 26.5 hours the encoding stops and i have to restart the event. and 2. the closed caption that is embeded inside the video doesn't flow through to the output [00:36:46 CEST] <DHE> considering your encoding pipeline through 3rd party apps, I'd say your closed captions issue isn't ffmpeg's fault [00:36:53 CEST] <DHE> (libx264 supports closed captions btw) [00:38:04 CEST] <prelude2004c> maybe you are right.. it could be nvtranscoder that looses it. the problem with libx264 it uses up too much cpu power.. that is why we bought an M4000 Nvidia card to do the decoding/encoding on the hardware side [00:38:15 CEST] <kierank> no it don't [00:38:25 CEST] <kierank> you have to inject custom sei [00:38:38 CEST] <prelude2004c> we have to fit like 10 HD channels through each machine.. but let's skip that closed caption issue until we figure out the more pressing issue which is .. why the streams stops at 26.5 hours every day [00:39:12 CEST] <prelude2004c> kierank ? custom sei .. can you decipher that for me a bit more ? [00:39:15 CEST] <DHE> holy shit that's an expensive video card [00:39:28 CEST] <DHE> it's a term for the encoding of the H.264 codec payloads [00:39:33 CEST] <prelude2004c> yup.. consumer version video cards only allow 2 streams ( 2 channel limit ) [00:40:07 CEST] <prelude2004c> getting around it would be another complete project :) [00:40:29 CEST] <DHE> wait, how many streams does it do? I've played with nvenc and ran into the 2 limit [00:40:44 CEST] <prelude2004c> yes on consumer limit cards [00:40:49 CEST] <prelude2004c> the M4000 has no limit [00:41:31 CEST] <prelude2004c> the quadro cards have their limit released ... that's why i use it..can fit over 10 HD channels per card with less than 30% Overall CPU usage [00:42:29 CEST] <prelude2004c> 2 cards = 20 channels.. etc etc.. only draw back is the BUS speed .. if i try to get 2 cards on one system i start running into board and ram / cpu issues .. so i dropped it back to 1 card per system.. still pretty damn good [00:42:38 CEST] <DHE> okay, that makes sense in text [00:42:49 CEST] <DHE> dual CPU? [00:42:54 CEST] <prelude2004c> 24 ocre [00:42:54 CEST] <DHE> *in context [00:43:01 CEST] <prelude2004c> core* [00:43:18 CEST] <prelude2004c> some have 16 cores.. depends on the system.. but its not the CPu's.. its bus speeds and stuff and the ram speed counts too [00:43:42 CEST] <prelude2004c> all of that talking back and forth .. the system seems to act weird when you start going over like 15 channels... so i kept it light .. will test later with newer boards and stuff [00:44:10 CEST] <prelude2004c> new gaming boards have a much higher threshold so.. i will test later.. but for now. i had the hardware in stock so i used it.. only problem is these two anoying issues i am running into [00:45:13 CEST] <DHE> wow, what do you do that you just have cards like that laying around? [00:45:24 CEST] <prelude2004c> lol no we had to buy them. [00:45:59 CEST] <DHE> oh, supplier stock [00:46:10 CEST] <prelude2004c> we were using a purchased product before that was using old nvidia 580 cards with cuda cores.. but i had to raise bit rate to 5Mbit/s to get decent quality out of them... and i can only fit 5 - 6 channels per card. Now with the new cards i can do much better quality at 1.8Mbit/s [00:46:59 CEST] <prelude2004c> after nvidia moved the encoding/decoding to its own cores in the cards.. its become much better at it.. later would be nice to find a way to use the cuda cores too .. but i doubt it will work anymore.. + they were not as good in terms of quality.. these new cards have lossess on it [00:47:13 CEST] <prelude2004c> the encoding between the CPU vs nvidia hardware is pretty much the same [00:47:15 CEST] <prelude2004c> except they can do more [00:49:05 CEST] <prelude2004c> now getting back to the problem at hand... so we need a fix for this whole 26.5 hours thing... how do we prevent the system from flipping out after that timeout [00:50:07 CEST] <prelude2004c> for the closed caption thing.. do you guys think that i should do it with the nvtranscoder code ? its open source and i have all the .h and .c files .. so maybe in there it could be done to do something with the closed caption instead of throwing it away [00:50:34 CEST] <DHE> I'm interesting in why ffmpeg's nvenc is inferior to nvtranscoder. streamline the whole process [00:50:49 CEST] <DHE> libx264 has CC handling already, might be easily added to nvenc [00:51:38 CEST] <prelude2004c> wow.. i am loving the idea of trying it.. i am totally ok with changing my approach if i can make it work correctly [00:51:49 CEST] <prelude2004c> DHE , i can give you access to a test system if you wish .. [00:51:54 CEST] <prelude2004c> let you play with it [00:52:21 CEST] <prelude2004c> i will give you access to the input stream, the M4000 card , etc [00:52:24 CEST] <DHE> tempting, but I have an nvenc-enabled GPU right now. GTX770, nvenc and opencl both working [00:52:37 CEST] <DHE> for testing I can dance around the 2-stream limit [00:52:42 CEST] <prelude2004c> yes but who does your decoding ? [00:52:50 CEST] <DHE> ffmpeg [00:52:54 CEST] <prelude2004c> oh cpu time right ? [00:53:10 CEST] <prelude2004c> ya ok.. by comparison .. my whole entire ffmpeg process is like 20% of one more for each channel [00:53:21 CEST] <prelude2004c> when i use ffmpeg to decode, it jumps over 100% [00:54:32 CEST] <prelude2004c> my suggestion to have you play on the system was that i also found that source matters .. so your behavior may be slightly different depending on the input [00:54:49 CEST] <prelude2004c> anyways just a thought.. i am looking to get to a solution that's all.. fastest way possible and whatever it takes [00:55:41 CEST] <prelude2004c> btw, DHE . not sure how the GTX 770 performance is compared to the M4000 [00:56:58 CEST] <prelude2004c> probably pretty close by the looks of it [00:58:37 CEST] <prelude2004c> next project is to jack firmware to get the 2 stream limit lifted on your card :) [00:58:55 CEST] <DHE> let me see if I can get this sample made first... [00:59:14 CEST] <prelude2004c> i bet nvidia just put some artificial limit on your card to have us buy more expensive ones [01:00:56 CEST] <cone-624> ffmpeg 03Petru Rares Sincraian 07master:af56d0dffa3e: fate: add aemphasis test [01:00:56 CEST] <cone-624> ffmpeg 03Michael Niedermayer 07master:2be3007ed55f: avformat/utils: avoid overflow in update_stream_timings() with huge durations [01:00:56 CEST] <cone-624> ffmpeg 03Michael Niedermayer 07master:c1ed78a591f6: avformat/utils: avoid overflow in compute_chapters_end() with huge durations [01:00:56 CEST] <cone-624> ffmpeg 03Thomas Guilbert 07master:1a82d2cf8fb6: avformat/oggparseopus: Fix Undefined behavior in oggparseopus.c and libavformat/utils.c [01:01:20 CEST] <DHE> now I'm intrigued. I must examine this patch [01:02:28 CEST] <DHE> nope, seems unrelated [01:05:23 CEST] <prelude2004c> DHE , tried to send you something but system will not let me [01:05:42 CEST] <DHE> oh, private messages? yeah I have that filtered by default. sorry. [01:05:53 CEST] <prelude2004c> you may want to see it [01:05:59 CEST] <prelude2004c> can't you message me? [01:07:02 CEST] <DHE> sure go ahead then... [01:16:10 CEST] <prelude2004c> here is something of a quick fix for you guys maybe.... [hls muxer @ 0x3832b00] failed to delete old segment http://10.0.200.2/webdav/disneyxd1648/disneyxd16482M16041.ts: No such file or directory ... the PUT works as per my code.. but how come delete doesn't work? nginx has the put, delete everything allowed from that source [01:16:17 CEST] <prelude2004c> very odd how it can't delete..must be another bug [01:17:14 CEST] <prelude2004c> less concerned as i can just do an nfs mount but just through i would mention it.. if i am missing something or it's a bug [01:19:19 CEST] <BtbN> you can't delete via PUT. [01:19:35 CEST] <BtbN> Use DELETE for that. [01:19:46 CEST] <BtbN> or REMOVE, or whatever HTTP decided to call it [01:19:52 CEST] <DHE> and I can't find a DELETE reference in the code. heck I can't even find PUT except for icecast [01:20:20 CEST] <BtbN> I think it's set via command line in the http output as method... [01:20:35 CEST] <prelude2004c> do you mean -method PUT -method DELETE [01:20:39 CEST] <prelude2004c> i can use multiple like that ? [01:20:42 CEST] <BtbN> No. [01:20:47 CEST] <prelude2004c> ok didn't think so [01:20:57 CEST] <BtbN> I'd guess it's simply not supported. [01:21:00 CEST] <nevcairiel> the http protocol doesnt support file management operations, so you cant have it delete things [01:21:17 CEST] <BtbN> What the hell are you even doing? hls via http, what? [01:21:21 CEST] <prelude2004c> got it... by default its POST .. so i changed it to PUT [01:21:21 CEST] <nevcairiel> its only implemented for file and ftp i think [01:21:30 CEST] <prelude2004c> its outputting to webdav [01:21:35 CEST] <prelude2004c> for remote file writing and deleting [01:21:38 CEST] <BtbN> yeah, don't do that. [01:21:54 CEST] <prelude2004c> works for write :P .. [01:22:15 CEST] <DHE> may I recommend a CIFS share? [01:22:24 CEST] <BtbN> CIFS over the internet? [01:22:27 CEST] <prelude2004c> then i have a script taking care of the delete on the other side.. not a big deal.. i can also nfs mount or sftp mount or whatever.. forget this.. just thought i would bring it up thinking maybe some silly bug or maybe my mistake [01:22:39 CEST] <BtbN> Why not just a good old rtmp stream, and another ffmpeg instance on the other end writing the HLS? [01:22:48 CEST] <prelude2004c> its local network actually .. but some people may want to use webdav, just saying [01:22:59 CEST] <BtbN> You are creating a huge bunch of hacks for something where clean implementation exists for. [01:23:12 CEST] <prelude2004c> hey i like to get creative :P [01:23:29 CEST] <BtbN> Well, then don't complain if stuff doesn't work. [01:23:31 CEST] <prelude2004c> actually i was thinking before that my nfs mount was a problem.. hence the drops [01:23:43 CEST] <prelude2004c> but then i moved to webdav and still same so that's not it [01:24:28 CEST] <prelude2004c> anyways ok... i wont use webdav.. will swap it out again to local mount which already exists.. that will solve the delete function issue.. so i dont care much about that [01:24:56 CEST] <prelude2004c> would be good to support though because i can see some people may want to use it and it may be something simple to do [01:24:59 CEST] <prelude2004c> just an idea [01:25:57 CEST] <BtbN> Just mount them? [01:26:22 CEST] <BtbN> But none of these protocols are good for streaming data like that. [01:26:35 CEST] <BtbN> Set up an nginx-rtmp or something on the streaming server. [01:27:07 CEST] <prelude2004c> i have the mount already in place from one system to the other.. i used http to test if it was not that mount doing something .. only after i changed it and it kept happening did i begin to understand the 26.5 hour problem.. so i just have to put things back to the mount instead of the webdav [01:27:39 CEST] <prelude2004c> no its not really a streaming server where it goes.. its going out to a network point that holds all the encoder data and then the distribution network pulls from it [01:27:45 CEST] <prelude2004c> all that is working 100% [01:54:18 CEST] <drazin> can anyone instruct me how to apply a patch to ffmpeg? [01:55:16 CEST] <DHE> save the patch to disk, use "patch -pX < inputfile.patch" where X is typically either 1 or 0 [01:55:39 CEST] <drazin> can i do that to the .exe? [01:55:42 CEST] <DHE> if it's a git repo, git has funcitons for that [01:55:52 CEST] <DHE> no, patches are source code modifications. once applied you have to recompile [02:02:02 CEST] <DHE> that explains so much, and why this code sucks. [02:02:43 CEST] <drazin> damn [02:02:56 CEST] <DHE> this is unrelated to your thing [02:03:11 CEST] <drazin> oh lol [02:13:25 CEST] <drazin> anyone in windows willing to do me a favor ;) [02:18:26 CEST] <DHE> testing a sorta hacky patch [02:38:05 CEST] <DHE> prelude2004c: the hack works I think. fixing a big and running again [02:46:15 CEST] <prelude2004c> F&$%$N A :P .. isn't that such an old expression [02:46:53 CEST] <prelude2004c> DHE , you are awsome [02:47:03 CEST] <DHE> wait for the test to come back positive [02:47:11 CEST] <DHE> ly [02:47:56 CEST] <DHE> it'll be ~12 minutes [02:50:23 CEST] <prelude2004c> i know you can do it [02:50:35 CEST] <prelude2004c> is it a bug or a new flag that could be added [02:50:44 CEST] <prelude2004c> maybe creating a flag that can be set or someting [02:50:51 CEST] <prelude2004c> then people can use it whenever they need it [02:52:18 CEST] <DHE> right now it's just a new mode of dealing with timestamps. there are certain API usages that will cause issues, but basic ffmpeg should just work [02:56:58 CEST] <prelude2004c> ic.. you mean coming from a live .ts source was a problem ? [03:21:59 CEST] <Shiz> /w 29 [03:32:34 CEST] <DHE> as I said earlier, I recorded about 28 hours of live streaming for use as a test video. [04:44:52 CEST] <cone-318> ffmpeg 03Michael Niedermayer 07master:17d320800b70: avformat/movenc: Avoid integer overflow [04:44:52 CEST] <cone-318> ffmpeg 03Michael Niedermayer 07master:dac030d3aa1b: avformat/movenc: Fix potential track width/height overflows [04:58:13 CEST] <rcombs> is there actually no way to express non-standard channel layouts in vorbis and opus, or does lavc just not support it? [09:15:28 CEST] <cehoyos> rcombs: Years ago, there was no way, if this has changed, please provide a sample (and name the application that can handle it) [09:15:59 CEST] <rcombs> no sample; just wondering [09:25:09 CEST] <cehoyos> I checked it four or five years ago: At that time, no special layouts were supported, I never looked at it again. [10:19:22 CEST] <nevcairiel> as far as I know opus only supports the vorbis layouts at this time rcombs, but it is designed with flexibility in mind and could adopt any other kind of channel layout standards [10:20:02 CEST] <rcombs> I'm of half a mind to add arbitrary mapping support to lavc, start using it in my own applications, and see if it catches on [10:20:03 CEST] <nevcairiel> (ie. it carries a tag to identify which kind of channel mapping i used, tag 0 is mono/stereo without channel mappings, tag 1 is vorbis [10:20:33 CEST] <nevcairiel> and producing files that do not conform to any standard is not something avcodec should be doing [10:53:42 CEST] <kierank> Opus in MPEGTS has a mapping rcombs [14:40:15 CEST] <cone-020> ffmpeg 03Michael Niedermayer 07master:cbd19881f7e4: avformat/udp: Remove unused variable [14:54:20 CEST] <ubitux> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2016-May/032266.html mmh. [15:02:36 CEST] <andrey_turkin> BtbN: I've finished reworking my previous patches to separate all the "good" parts: https://github.com/tea/FFmpeg/commits/nvenc. All those patches (except for last) are IMO useful on their own and they also bring ffmpeg/libav implementations closer together. Compatibility impact should be minimal. [15:07:48 CEST] <durandal_1707> michaelni: you know how gradient prediction in huffyuv works? [15:37:18 CEST] <BBB> ubitux: maybe nevcairiel can help with that, he was quite familiar with the transition [16:59:16 CEST] <prelude2004c> hey guys.. anyone know why segmenting has run away files ? [16:59:35 CEST] <prelude2004c> i am segmenting files but for some reason the segmenter sometimes doesn't turn files over and keeps going until they fill up the hard drive [17:13:11 CEST] <DHE> wrong channel [17:15:45 CEST] <DHE> but see the help for -hls_flags [17:49:32 CEST] <michaelni> durandal_170, IIRC the 2 variants are T+L-TL and median(T,L, T+L-TL) [17:51:19 CEST] <cone-657> ffmpeg 03Michael Niedermayer 07master:52ca24bdd2d1: avutil/softfloat: Improve doxy for av_sub_sf() and av_sf2int() [17:51:19 CEST] <cone-657> ffmpeg 03Michael Niedermayer 07master:7ae4d574e873: avfilter/vf_fieldhint: Assert that mode is valid [18:43:42 CEST] <prelude2004c> -hls_flags delete_segments i am aware of the delete segments but i am thinking its a bug in the code or something because it sometimes doesn't happen for an entire day or two days and sometimes two hours.. then the file segment doesnt' actually change to next file and whatever last file was being written just goes on and on forever without cutting into a new file [18:56:52 CEST] <BtbN> I guess you are still using your fuse webdav/ssh mount? [19:00:16 CEST] <DHE> I feel it's not quite appropriate for this channel, our previous chat kinda spilled into here back when we had an actual ffmpeg bug/misfeature going on [19:02:30 CEST] <cone-657> ffmpeg 03Michael Niedermayer 07master:be96ebdcd795: avfilter/vf_fieldhint: Reorder operation to prevent hypothetical integer overflow [22:09:48 CEST] <drazin> can anyone help me with applying a patch then compiling ffmpeg in windows [22:14:13 CEST] <Compn> drazin : did you askin #ffmpeg ? [22:14:23 CEST] <drazin> yeah, no help so far [22:17:21 CEST] <DHE> he's looking for a windows build but he doesn't have a compiler [22:18:41 CEST] <durandal_1707> drazin: what patch? [22:19:10 CEST] <drazin> https://gist.github.com/outlyer/4a88f1adb7f895b93fd9 [22:22:14 CEST] <drazin> durandal_1707: can you help [22:22:45 CEST] <durandal_1707> I'm on ubuntu [22:29:55 CEST] <drazin> can it compile an exe on there? [22:30:40 CEST] <durandal_1707> nope [22:33:08 CEST] <DSM_> drazin: you can use Cygwin. to apply patch you'll need git and some packages. https://www.ffmpeg.org/platform.html#toc-Compilation-under-Cygwin [22:33:53 CEST] <drazin> ok i have cygwin from my previous go [22:34:01 CEST] <drazin> but was usisng differnt instructions [22:39:25 CEST] <drazin> DSM_: the problem is that when i try to install packages in that list i get this https://dl.dropboxusercontent.com/spa/tr6kpc6gjq961c0/d2jgmvr5.png [22:43:10 CEST] <drazin> oh [23:02:56 CEST] <cone-657> ffmpeg 03Michael Niedermayer 07master:645f7c1ce547: avfilter/f_loop: Fix leak on error [00:00:00 CEST] --- Mon May 30 2016

1 0

ffmpeg.log.20160529
by burek 30 May '16

30 May '16

[00:04:43 CEST] <DHE> ... doesn't nvidia already have a plugin for ffmpeg? [00:06:25 CEST] <prelude2004c> it does.... but the problem is the decoding [00:06:51 CEST] <prelude2004c> nvenc is there. and it works.. but strangely enough i get better results using nvtranscoder that can use the sq as well as bit rate together. [00:06:56 CEST] <prelude2004c> sorry sq = qp [00:07:26 CEST] <prelude2004c> i did a side by side comparison using similar tune options and why.. i dont know.. one is much better than the other since i set the qp = 21 [00:07:45 CEST] <prelude2004c> and also with nvtranscoder it holds to the bit rate limitation [00:08:53 CEST] <prelude2004c> so, that is just on the encoding side.... for decoding i have to use vdpau which is another problem on it's own. It uses more CPU resources ( ffmpeg that is ) to deal with it.. and also I use X on the system.... nvtranscoder does the decoding/encoding without almost any resources used and no X required [00:12:04 CEST] <BtbN> qp and bitrate at the same time makes no sense [00:12:13 CEST] <BtbN> you either encode a constant quality, or a constant bitrate. [00:12:35 CEST] <BtbN> There are mixed modes, where you specify a max bitrate, ffmpeg supports that just fine. [00:14:23 CEST] <prelude2004c> yup.. so i am told... but i dont know why nvtranscoder has much better quality with same presets [00:14:46 CEST] <BtbN> because they are not the same presets. [00:15:10 CEST] <prelude2004c> hey here is something else.... and this may not make any sense at all with respect to the pts wrap around... "itsoffset " of lets say 0 or 10 or something.. DHE .. do you think that could do something ? [00:15:34 CEST] <DHE> uhh.. what? [00:18:09 CEST] <prelude2004c> right :) maybe it makes no sense [00:18:21 CEST] <prelude2004c> also, -re would that do anything stating that it's a realtime source ? [00:18:30 CEST] <prelude2004c> could it somehow result in a fix for it ? [00:19:22 CEST] <DHE> umm.. no. all that does is make the ffmpeg frontent sleep() when it sees itself running faster than realtime [00:24:02 CEST] <prelude2004c> understood.. as you can see i am grabbing at straws at this point.. been on this for over a week all day .. i can't figure it out [00:27:28 CEST] <DHE> wow... [01:13:31 CEST] <DocMAX_> hi, autocrop? [01:37:11 CEST] <drazin> does anyone know how to apply a patch to ffmpeg on windows [01:37:47 CEST] <BtbN> the same way as everywhere else. [01:47:12 CEST] <drazin> how do you do it everywhere else [01:47:18 CEST] <drazin> google didnt help me [01:49:39 CEST] <drazin> i have a .patch file and a .rb file [07:42:49 CEST] <Fyr> guys, I have a bad audio channel, FFMPEG reports frame sync error while covnerting it. is it possible to resample it without quality loss? [07:43:31 CEST] <Fyr> dfdfdf [09:44:38 CEST] <_delta_> hi, I'm trying to compile a very simple (7 line) program that includes "libavformat/avformat.h". when I try to compile I get a linking error. I'm pretty new to gcc, can someone help me figure out how to compile this? [09:46:21 CEST] <_delta_> basically my program just calls av_register_all() in main(). I'm just trying to see if I can get this to work. when I compile, I get "undefined reference to av_register_all". [11:27:01 CEST] <nifwji2> I wonder when I am going to lose interest in vectors. [15:23:42 CEST] <ZeuZ> Hey everyone [15:24:39 CEST] <ZeuZ> I want to open a jpeg image from C using libav and add it to a video, the video file was created using the sample from: http://stackoverflow.com/questions/17816532/creating-a-video-from-images-us… [15:25:23 CEST] <ZeuZ> What would be the steps to be made? to encode the image as a frame, I also found: https://ffmpeg.org/pipermail/libav-user/2011-July/000428.html but it seems he's got problems playing the file later [16:09:28 CEST] <yagiza> Hello! [16:09:33 CEST] <Fyr> hi! [16:09:39 CEST] <yagiza> Fyr [16:09:47 CEST] <Fyr> yagiza! [16:10:10 CEST] <yagiza> I wonder, if FFMpeg supports progressive graphics formats. [16:10:46 CEST] <yagiza> I mean decoding each layer as a new video frame. [16:10:57 CEST] <Fyr> layer? [16:12:09 CEST] <yagiza> Fyr, you know... if image is progressive, you may take beginning of image data and decode it as a low resolution image. [16:12:29 CEST] <Fyr> ah [16:12:37 CEST] <Fyr> I don't know [16:13:13 CEST] <JEEB> that usually is only done in JPEG [16:13:39 CEST] <Fyr> ffmpeg uses libopenjpeg and libwebp. [16:13:52 CEST] <JEEB> openjpeg is j2k [16:13:54 CEST] <JEEB> not normal JPEG [16:13:59 CEST] <JEEB> and j2k is a whole separate mess [16:14:09 CEST] <JEEB> some ancient'ish MPEG formats have low resolution decoding capability, but they tend to be fast enough to decode nowadays (MPEG-2 and MPEG-4 Part 2) [16:14:14 CEST] <Fyr> ok, if their default options are set to progressive, then yes. [16:14:38 CEST] <yagiza> JEEB, PNG, PCX and GIFs also could be progressivf, IIRC. [16:14:39 CEST] <JEEB> "progressive" in this sense just means coding in a way that enables lowres decoding first [16:15:19 CEST] <yagiza> JEEB, yes [16:15:23 CEST] <JEEB> anyways, no idea. you're most likely to find out looking at the options of the jpeg and the png encoders [16:15:40 CEST] <yagiza> JEEB, I mean ADAM7 for PNG [16:15:58 CEST] <yagiza> JEEB, ok, I'll try. [16:27:34 CEST] <iive> mpeg2 and h264 have extensions for scalability [16:28:20 CEST] <iive> where you can have a basic stream and additional steam(s) that increase resolution, framerate or color precision (yuv420->yuv444) [16:29:57 CEST] <JEEB> yes, but those are almost unused. I have only used them used in AVC with one video conference thing [16:30:52 CEST] <JEEB> HEVC's scalability etc things are IIRC more sanely built, but still heavily unused [16:31:05 CEST] <JEEB> academia loves to onanize with those features though [16:32:21 CEST] <iive> in theory, lowres decoding is possible for all block based formats. You skip idct by decoding only the dc component and ignore the ac. [16:32:52 CEST] <ZeuZ> http://pastebin.com/MuUWb8fL ---> can anybody give me a hint as to why it segfaults? [16:33:20 CEST] <DeHackEd> on what line? [16:34:42 CEST] <ZeuZ> apparently, its before calling open_image [16:34:46 CEST] <ZeuZ> but it makes no sense [16:41:03 CEST] <Fyr> JEEB, is it possible to fix incorrect timestamps with resampling? [16:41:33 CEST] <JEEB> if you're still going on about concatenation then you're misunderstanding the issue [16:41:46 CEST] <Fyr> it's abotu concatenation. [16:41:57 CEST] <Fyr> the original file has frame sync errors. [16:42:13 CEST] <JEEB> and if you are able to resample you're able to decode, in which case you can try using the concat filter [16:42:33 CEST] <Fyr> how much quality will I lose in comparison to conversion into the same format? [16:43:23 CEST] <JEEB> no idea and to be fucking honest I don't care. you're re-encoding lossy stuff into a lossy format again [16:49:21 CEST] <ZeuZ> http://pastebin.com/xbP58pec --> still not getting the why the segfault happens, it happens right after calling open_image() but doesn't even call the first line of the function (printf) [16:49:28 CEST] <ZeuZ> could it be related to changing APIs? [16:50:45 CEST] <ZeuZ> GDB says: that avformat_open_input() sent a sigsegv [16:51:06 CEST] <ZeuZ> but the first line (printf) is never executed? [16:51:35 CEST] <furq> does it print if you add a \n to the end [16:53:19 CEST] <ZeuZ> it doesn't, no, why? [16:53:28 CEST] <ZeuZ> it shoudn't affect stdout should it? [16:53:45 CEST] <furq> if you segfault before stdout is flushed then its contents won't be printed [16:53:56 CEST] <furq> if you have line buffering on then it's flushed on every newline [16:54:05 CEST] <ZeuZ> ah you're right [16:54:13 CEST] <ZeuZ> but nope, it still doesn't output it [16:54:38 CEST] <ZeuZ> if(avformat_open_input(&pFormatCtx, imageFileName, NULL,NULL)!=0) ---> looks like the troublesome line [16:55:08 CEST] <ZeuZ> pFormatCtx should be allocated beforehand right? or does avformat_open_input allocate it automagically? [16:55:09 CEST] <josh98> does ffmpeg have non-free dependencies? [16:55:32 CEST] <furq> josh98: not unless you enable them [16:55:59 CEST] <josh98> furq: they will not be enabled by default? [16:56:10 CEST] <furq> no [16:57:25 CEST] <josh98> furq: then why would a ffmpeg install attempt to install a non-free dependency? [16:57:44 CEST] <furq> define install [16:58:19 CEST] <furq> if you're installing it from a package manager then it depends how your distro has configured it [16:58:58 CEST] <josh98> i used the ppa on ffmpeg.org [16:59:10 CEST] <josh98> i'm going from memory here. it was a while back [16:59:36 CEST] <JEEB> anything requiring enable-nonfree can't be distributed in binary form anyways, since the nonfree part is regarding licensing incompatibilitie [17:00:14 CEST] <furq> i seem to remember the ubuntu-multimedia package used to include fdk-aac [17:00:18 CEST] <furq> they fixed that a while ago though [17:00:39 CEST] <JEEB> they can distribute fdk-aac separately (not linked to FFmpeg) if they follow its license [17:00:45 CEST] <josh98> would the install from ppa require the installation of the non-free component in case the user later wanted to enable it? [17:01:00 CEST] <furq> no i mean their ffmpeg was built with fdk [17:01:26 CEST] <furq> josh98: like i said i don't think it depends on fdk any more so there shouldn't be any non-free dependencies [17:01:48 CEST] <JEEB> thankfully the general need for enable-nonfree requiring components at this point is rather small [17:01:54 CEST] <JEEB> mainly if you need HE-AAC encoding [17:01:57 CEST] <JEEB> (fdk-aac) [17:02:11 CEST] <JEEB> LC-AAC has a good enough alternative in the internal encoder nowadays [17:02:28 CEST] <josh98> i don't know this stuff well enough yet to understand concepts like HE-AAC encoding [17:02:43 CEST] <JEEB> are you using AAC at less than 64kbps in stereo? :P [17:02:54 CEST] <JEEB> if no, then you aren't in the range for HE-AAC [17:03:21 CEST] <JEEB> and for decoding there's no need for enable-nonfree components [17:03:50 CEST] <furq> you can't ship binary packages with nonfree components anyway, it violates the license [17:04:03 CEST] <JEEB> yes [17:04:11 CEST] <furq> the ubuntu ppa was violating the gpl by doing that, which is why they stopped [17:05:06 CEST] <josh98> could it be a matter of the os determining that the non-free component was necessary to run ffmpeg on that device, so requiring its installation to proceed with the installation of ffmpeg? [17:05:27 CEST] <furq> no, it was a matter of a package maintainer making a mistake [17:05:40 CEST] <josh98> in the case i recall, i killed off the installation when a non-free nvidia component tried to crash the party [17:06:12 CEST] <JEEB> anyways, what's this thing with enable-nonfree? it's pretty clear to me that you don't seem to need any enable-nonfree requiring components [17:07:40 CEST] <JEEB> and the only way for you to acquire a binary with enable-nonfree components enabled is to compile it yourself as distributing the binaries is not possible license-wise [17:10:12 CEST] <josh98> i used this ppa... ppa:mc3man/trusty-media [17:10:22 CEST] <JEEB> consider that something random [17:10:28 CEST] <JEEB> no idea what it contains [17:10:33 CEST] <josh98> from https://launchpad.net/~mc3man/+archive/ubuntu/trusty-media [17:10:47 CEST] <josh98> linked from ffmpeg.org download sources [17:11:28 CEST] <JEEB> yes, but nothing linked there that isn't source code is official [17:11:39 CEST] <josh98> good to know [17:11:40 CEST] <JEEB> at some point someone added something to that page [17:12:05 CEST] <josh98> i wish that was clearer [17:12:12 CEST] <josh98> it comes across as official [17:12:25 CEST] <furq> it says "official packages" next to the packages which are official [17:12:56 CEST] <furq> the alternative repos are there as a concession to debian/ubuntu's historically bad ffmpeg packaging [17:13:16 CEST] <furq> that's not really as big a deal any more though [17:13:24 CEST] <furq> unless you're on an old version, which lots of ubuntu users are [17:13:35 CEST] <JEEB> well it's a general issue with distros that aren't rolling [17:13:41 CEST] <josh98> you are right, furq. it does say official next to some...and not the ppa i chose [17:13:41 CEST] <JEEB> you pick one version and you stick with it [17:15:29 CEST] <furq> huh [17:15:35 CEST] <furq> https://launchpadlibrarian.net/260284733/buildlog_ubuntu-trusty-amd64.ffmpe… [17:15:41 CEST] <furq> that's the latest changes file and it's still including fdk [17:15:44 CEST] <furq> someone needs to have a word [17:15:57 CEST] <furq> or the latest build log, rather [17:16:18 CEST] <JEEB> if that's not official then as long as the FFmpeg side of license is kept nobody is likely to care [17:16:41 CEST] <JEEB> of course it's in general better to note that nonfree shouldn't be distributed [17:16:48 CEST] <furq> i mean it's probably a bad idea for ffmpeg.org to be linking to gpl-violating binaries [17:17:20 CEST] <JEEB> true [17:17:37 CEST] <josh98> is fdk-aac necessary? [17:17:39 CEST] <JEEB> although from FFmpeg's side the license isn't really broken as long as the sources are given out with all deps [17:17:40 CEST] <furq> no [17:17:42 CEST] <JEEB> josh98: no [17:17:43 CEST] <josh98> what benefit does it provide? [17:17:46 CEST] <furq> there's a builtin aac encoder [17:17:47 CEST] <JEEB> HE-AAC encoding [17:18:08 CEST] <JEEB> it used to be the only good AAC encoder available until atomnuker improved the internal one for LC-AAC [17:18:13 CEST] <furq> fdk is generally a bit better than the builtin encoder [17:18:25 CEST] <JEEB> which is why many people got used to building their local binaries with it [17:18:28 CEST] <furq> but not enough that most people would notice unless they're encoding he-aac [17:18:43 CEST] <JEEB> well the internal one just doesn't do HE-AAC encoding IIRC [17:19:06 CEST] <furq> you'll definitely notice in that case [17:19:06 CEST] <JEEB> but given that most AAC encoding is LC... [17:19:33 CEST] <josh98> philosophically, wouldn't this community readily give up a bit of benefit in exchange for gpl compliance? [17:19:50 CEST] <furq> i don't see anything in that build log about nvidia btw [17:19:58 CEST] <furq> except nvidia-vdpau-driver in the suggested packages list [17:20:21 CEST] <josh98> nvidia-vdpau-driver is non-free? [17:20:23 CEST] <JEEB> the community follows the license, and if you don't distribute a binary then enable-nonfree is OK, which is how everyone had been using that. And nowadays it is even less required than before [17:20:30 CEST] <furq> yeah but that's a suggested package, it's not installed by default [17:20:41 CEST] <josh98> good [17:20:44 CEST] <josh98> that sounds familar [17:20:45 CEST] <furq> you'd need to run apt with --install-suggested or have it enabled in your apt.conf [17:20:46 CEST] <josh98> familiar [17:20:54 CEST] <josh98> ill check it out again [17:22:01 CEST] <furq> but yeah given ubuntu's stance on zfs, maybe they'll just decide to ship ffmpeg with fdk in future [17:22:10 CEST] <JEEB> nah, they won't [17:22:23 CEST] <JEEB> because it's a different kind of thing IMHO [17:22:30 CEST] <furq> that was mostly a joke [17:22:39 CEST] <furq> nothing ubuntu does would surprise me any more though [17:22:39 CEST] <josh98> that's a raw subject [17:23:25 CEST] <JEEB> anyways, WTF was the original question josh98 was trying to get an answer to :P [17:23:50 CEST] <josh98> i think furq gave me the knowledge to take another crack at solving it [17:24:01 CEST] <josh98> you and furq, JEEB [17:24:31 CEST] <furq> well yeah that ubuntu ppa package appears to still depend on fdk [17:24:36 CEST] <josh98> thank you JEEB furq [17:24:37 CEST] <furq> which is non-free [17:24:46 CEST] <furq> so you might want to stick with the official ubuntu package [17:24:55 CEST] <furq> if you're on a newish version, at least [17:24:59 CEST] <JEEB> or if that one is too old, build your own binary [17:25:06 CEST] <furq> or use relaxed's static builds [17:25:23 CEST] <josh98> i'm going to install it on a 14.04.3 box [17:25:37 CEST] <furq> yeah i think that's one of the unfortunate ones with libav [17:25:43 CEST] <furq> http://johnvansickle.com/ffmpeg/ [17:25:46 CEST] <furq> that's probably your best bet [17:27:05 CEST] <josh98> that website reminds me that i have a perpetual "to do" to set up a low-friction method of tipping with bitcoin [17:32:15 CEST] <josh98> in this instance, easy trumps perfect, with the constraint of foss only [17:33:56 CEST] <josh98> that is temporary though. this is helping me learn a more durable solution [18:18:08 CEST] <Fyr> guys, how to create silence into 5.1 channels? [18:30:10 CEST] <ChocolateArmpits> Fyr: try using anullsrc through lavfi and set the channel option accordingly [18:30:27 CEST] <Fyr> how to set channel option? [18:30:40 CEST] <ChocolateArmpits> by looking at the filter documentation https://ffmpeg.org/ffmpeg-filters.html#anullsrc [18:30:41 CEST] <Fyr> I used anullsrc, it creates stereo by default. =( [18:32:04 CEST] <ChocolateArmpits> so it would be anullsrc=cl=5.1 [18:32:06 CEST] <ChocolateArmpits> I think [18:32:46 CEST] <ChocolateArmpits> well precisely ffmpeg -f lavfi -i anullsrc=cl=5.1 [18:32:46 CEST] <Fyr> thanks [18:33:07 CEST] <ChocolateArmpits> the documentations are pretty lengthy but learning how to browse them helps [18:33:42 CEST] <ChocolateArmpits> they also include common examples [18:35:33 CEST] <furq> Fyr: https://ffmpeg.org/doxygen/trunk/channel__layout_8c.html#aa9ca20b8ed0ad0c9d… [20:02:56 CEST] <ZeuZ> I need to load a JPG image into an AVFrame, I've opened it with avformat_open_input, and filled up the codec context, whats next? [20:03:20 CEST] <ZeuZ> calling av_image_alloc and av_image_fill_arrays is also done [20:05:15 CEST] <ZeuZ> I've seen that avpicture_fill does the trick, but ain't it deprecated? [20:57:21 CEST] <f00bar80> Can I h.264 encode multiple live streams to multiple outputs in a single command ? [20:57:58 CEST] <DeHackEd> yes, after you specify an output file you can then specify a new set of options for a second output file [20:58:34 CEST] <DeHackEd> ffmpeg [options for input 1] -i inputfile1 [options for input 2] -i inputfile2 [...] [options for output 1] output1.mp4 [options for output 2] output2.mp4 [..] [20:58:40 CEST] <furq> -i foo -i bar -map 0 foo.mp4 -map 1 bar.mp4 [20:59:08 CEST] <DeHackEd> that... technically works, but somehow makes me feel uncomfortable. [20:59:26 CEST] <furq> yeah if you have multiple inputs it's better to use multiple commands [20:59:56 CEST] <DeHackEd> yeah, feels like if one stream were to suffer issues then the whole ffmpeg process could jam up [20:59:56 CEST] <furq> it's only faster to use multiple outputs if they're from the same input [21:02:29 CEST] <f00bar80> DeHackEd: http://pastebin.com/VEwsqvg2 this is a correct command then ? [21:02:58 CEST] <furq> you need to specify all the options for each input and output [21:03:19 CEST] <DeHackEd> 2 inputs, 1 output, no map commands. I don't think that's going to work the way you intend [21:03:22 CEST] <furq> also yeah you're not using -map so the second output will be using the streams from the first input [21:03:44 CEST] <DeHackEd> oh, I see, yeah no. you gave the second output no options [21:05:22 CEST] <furq> like i said, there is no advantage to doing it like this [21:05:39 CEST] <furq> it might even be slower because i seem to recall that inputs are processed sequentially [21:06:16 CEST] <furq> and it's definitely less reliable [21:08:49 CEST] <f00bar80> http://pastebin.com/21PNZvS2 this is correct even ? [21:09:12 CEST] <furq> no [21:09:43 CEST] <DeHackEd> you're missing the point by a longshot [21:10:20 CEST] <f00bar80> so what's wrong then ? [21:12:37 CEST] <DeHackEd> oh where do I begin. from a purely ffmpeg standpoint you've failed to use -map as furq clearly said you should [21:13:04 CEST] <DeHackEd> your new command has fixed that, but you then ignored my instruction to give options for both streams [21:13:37 CEST] <DeHackEd> or perhaps you didn't understand my loose template above about option orderings [21:16:37 CEST] <prelude2004c> hey DHE sorry about that.. no i am not using wedav [21:16:40 CEST] <prelude2004c> its a local storage [21:17:10 CEST] <f00bar80> so what's the optimum way to put a large of streams in individual commands ? using a script? [21:17:54 CEST] <furq> i would probably use a script [21:17:59 CEST] <prelude2004c> whatever triggers the segments to break and startup a new one on " segment size " .. is broken because sometimes ( randomly ) my .ts files go to GIG s [21:18:20 CEST] <furq> having both inputs in one command means that both outputs will drop if one input does [21:18:40 CEST] <f00bar80> ffmpeg has any specific builtin scripts for such cases .. ? [21:18:43 CEST] <furq> it also means a whole lot of redundant typing [21:18:56 CEST] <Threads> anyway to make -re -i read faster ? [21:19:20 CEST] <Threads> always seems to be a few seconds or minutes behind sometimes [21:20:51 CEST] <furq> f00bar80: http://vpaste.net/g7j4M [21:20:52 CEST] <furq> hf [21:21:11 CEST] <furq> http://vpaste.net/bGbjS [21:21:13 CEST] <furq> or that, even [21:21:49 CEST] <furq> ugh [21:21:51 CEST] <furq> http://vpaste.net/kPuFO [21:21:52 CEST] <furq> there [21:22:19 CEST] <furq> also if you need -strict -2 then you should update your ffmpeg to a version newer than 3.0 [21:40:01 CEST] <drazin> hey furq do you use windows? [21:43:21 CEST] <drazin> or if anyone has windows can they apply a patch and compile for me [21:43:24 CEST] <drazin> im stuck [21:48:34 CEST] <f00bar80> furq: this script should encode any number of streams simultaneously? [21:49:59 CEST] <f00bar80> furq: at the time all of the original streams are live streams. [21:53:24 CEST] <f00bar80> or this should only work on non live inputs as it encodes them sequentially? [21:58:15 CEST] <f00bar80> ppl any comment ? [22:30:37 CEST] <CoJaBo> Can anyone think of a reason ffmpeg would just, eat ALL THE RAM? [22:31:24 CEST] <CoJaBo> it gets until the last thirty sex coins, then blows up [22:31:45 CEST] <durandal_1707> CoJaBo: it should not, what are you doing? [22:33:04 CEST] <CoJaBo> ffmpeg version N-79785-gbd63ece-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2016 [22:33:07 CEST] <CoJaBo> the FFmpeg developers [22:33:10 CEST] <CoJaBo> built with gcc 5.3.1 (Debian 5.3.1-17) 20160429 [22:33:59 CEST] <CoJaBo> durandal_1707: encoding an imagepipe (bmps) to x264 [22:35:48 CEST] <durandal_1707> CoJaBo: what produce bmps? [22:38:11 CEST] <CoJaBo> durandal_1707: here's the log http://pastie.org/pastes/10857232/text?key=hosazapu10tb6jdqddrla [22:38:57 CEST] <CoJaBo> gonna try upstaging, but it takes a while to got to that point [22:39:23 CEST] <CoJaBo> up RAPPING [22:39:26 CEST] <CoJaBo> grading [22:40:59 CEST] <durandal_1707> there was some bmp parser bug before that got fixed [22:41:25 CEST] <CoJaBo> durandal_1707: was it this virgin? [22:41:28 CEST] <CoJaBo> virgin [22:41:45 CEST] <Threads> is it me or does ffmpeg have problems outputting to mkv with constant ? [22:42:01 CEST] <Threads> i keep getting variable even setting force-cfr=1 [22:42:24 CEST] <durandal_1707> CoJaBo: after release [22:42:50 CEST] <CoJaBo> durandal_1707: ? [22:42:54 CEST] <durandal_1707> Threads: what codecs? [22:43:18 CEST] <CoJaBo> durandal_1707: I'm running a non-release build [22:44:14 CEST] <durandal_1707> CoJaBo: ah, than try latest master, report bug if it still happens [22:44:19 CEST] <CoJaBo> durandal_1707: is there a link to that bug at least? [22:44:38 CEST] <Threads> durandal_1707 x264 [22:45:42 CEST] <durandal_1707> CoJaBo: yes, need to search Trac [22:46:54 CEST] <prelude2004c> hey guys.. trying to segment a file.. anyone know why " -segment_list_size 0 " is not being respected. [22:47:15 CEST] <prelude2004c> it only shows me the last 5 files [22:47:45 CEST] <prelude2004c> > /home/ffmpeg/ffmpeg -i "${stream}" -bsf:v h264_mp4toannexb -c copy -threads 0 -hls_time 6 -segment_list_size 0 "${HLS_PATH}/${CHN}/stream.m3u8" 2>/var/log/channels/${CHN}/output.txt [22:48:23 CEST] <prelude2004c> ehhh forget it [22:48:35 CEST] <prelude2004c> hls_list_size .. i am not using segment :( [22:48:37 CEST] <prelude2004c> my bad [22:50:49 CEST] <CoJaBo> Crashes on the exact same frame very time [22:53:08 CEST] <durandal_1707> CoJaBo: what utility creates bmps you pipeing? [22:53:51 CEST] <Threads> seems like this mv variable framerate is a common problem https://forum.kde.org/viewtopic.php?f=272&t=126877 [22:54:09 CEST] <CoJaBo> durandal_1707: Custom one [22:58:37 CEST] <CoJaBo> durandal_1707: Going to try it with the latest static build; if it still crashes, I'm going to try just feeding it BMPs in a loop for 4863 frames, see if it also crashes there maybe... [22:58:58 CEST] <CoJaBo> Would like to see the bug tho, but can't find it; just curious what the odds are this is fixed >_> [23:07:42 CEST] <CoJaBo> Attempt #15 is underway; new ffmpeg, interestingly, is a whole 50% faster than the old [23:13:33 CEST] <CoJaBo> ffmpeg version N-80117-gdac030d-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2016 the FFmpeg developers [23:14:54 CEST] <CoJaBo> This isn't relevent at all is it? [image2pipe @ 0x58f52e0] Stream #0: not enough frames to estimate rate; consider increasing probesize [23:19:18 CEST] <CoJaBo> Well it's almost to the frame [23:21:31 CEST] <CoJaBo> durandal_1707: Crash is reproducable in ffmpeg version N-80117-gdac030d-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2016 the FFmpeg developers [23:22:40 CEST] <durandal_1707> CoJaBo: open bug report on trac [23:23:02 CEST] <CoJaBo> It actually crashes somewhat before the other verison [23:25:08 CEST] <CoJaBo> durandal_1707: what all do i need to file that? :/ [23:27:51 CEST] <durandal_1707> just way to reproduce issue [23:29:37 CEST] <CoJaBo> trying to see if i can make it happen with just a stream of the same bmp over and over... [23:58:40 CEST] <CoJaBo> durandal_1707: ..so this is going to be difficult, it does not reproduce with a stream of the same bmp :/ [23:58:52 CEST] <CoJaBo> not actually sure what else to try... [00:00:00 CEST] --- Mon May 30 2016

1 0

ffmpeg-devel.log.20160528
by burek 29 May '16

29 May '16

[01:49:18 CEST] <cone-681> ffmpeg 03Michael Niedermayer 07master:9106cba22ad8: fate: Add fate-prores-gray [01:49:19 CEST] <cone-681> ffmpeg 03Michael Niedermayer 07master:f66ca036bca1: avformat/oggparseflac: Fix memleaks in old_flac_header() [03:55:53 CEST] <cone-681> ffmpeg 03Michael Niedermayer 07master:5db111757c89: avformat/movenc: Rename reshuffles return variable to ensure it is not mixed up [03:55:54 CEST] <cone-681> ffmpeg 03Michael Niedermayer 07master:86d703fd5584: avformat/movenc: Fix memleak of reshuffled packet [12:00:46 CEST] <cone-506> ffmpeg 03Carl Eugen Hoyos 07master:a64a030ba0d1: lavf/mov: Support one more Avid compression id for AVCI50. [13:21:15 CEST] <omerjerk> Any idea how do I check if a SoftFloat equals a particular number or not. [13:21:37 CEST] <omerjerk> I've a SoftFloat object, and I need to check it whether it is equal to 1.0f or not. [13:21:46 CEST] <omerjerk> Any help would be appreciated. [13:27:54 CEST] <kierank> why are you using softfloats? [13:29:39 CEST] <omerjerk> I've floats in my code and I need to avoid them using SoftFloats. [13:29:57 CEST] <omerjerk> I also need the exp and mantissa data of the floats at various places. [13:32:14 CEST] <kierank> ok [13:33:21 CEST] <omerjerk> so, any idea ? [13:38:13 CEST] <kierank> may have to write your own helpers for that [13:38:22 CEST] <kierank> coeff and mantissa checks [13:48:23 CEST] <michaelni> omerjerk, av_cmp_sf() [13:50:29 CEST] <omerjerk> michaelni, I did see that function. Suppose I convert 1.0f also to a Softfloat. [13:51:03 CEST] <omerjerk> What is the expected output of av_cmp_sf(soft_float_to_check, one_soft_float); ? [13:51:16 CEST] <michaelni> you can use FLOAT_1 [13:51:30 CEST] <omerjerk> okay, but what's the expected output ? [13:51:40 CEST] <michaelni> i thought it was documented, i guess i should document it if its not [13:51:40 CEST] <omerjerk> in case the SoftFloat is actually 1 ? [13:52:13 CEST] <michaelni> i think FLOAT_1 is 1.0 yes [13:52:53 CEST] <omerjerk> okay, but what would be the return value of av_cmp_sf function ? [13:53:14 CEST] <michaelni> 0 probably, ill RTFS to double check and document this properly [13:54:41 CEST] <omerjerk> okay. thanks a lot! I'll test my code and reply here. [13:57:19 CEST] <omerjerk> Also, what is the frac_bits parameter for in the av_int2sf() function ? [14:01:25 CEST] <omerjerk> michaelni, ? [14:02:03 CEST] <michaelni> this one seems documented: @returns a SoftFloat with value v * 2^frac_bits [14:07:52 CEST] <omerjerk> There's nothing like that documented here - https://www.ffmpeg.org/doxygen/2.2/softfloat_8h_source.html [14:09:46 CEST] <michaelni> https://www.ffmpeg.org/doxygen/trunk/softfloat_8h_source.html [14:26:23 CEST] <cone-506> ffmpeg 03Michael Niedermayer 07master:d1520a6cfd81: avutil/softfloat: Document public constants and a few public functions [14:29:12 CEST] <michaelni> nevcairiel, about AVClass/AVOptions, iam not sure i understood your objection about AVCodecParameters [14:30:51 CEST] <michaelni> I think we should consistently support it in the main public API structures like AV*Context AVFrame AVStream, ... [14:31:39 CEST] <nevcairiel> and I don't think its useful for API users [14:31:49 CEST] <nevcairiel> it loses the type safety of C, which is never a good thing [14:32:37 CEST] <nevcairiel> its somewhat a necessity for private options on AV*Context dealys [14:32:41 CEST] <michaelni> how does it loose type saftey ? [14:32:51 CEST] <nevcairiel> but simple data structures like AVFrame or AVPacket just don't need it for nothing [14:33:06 CEST] <michaelni> the AVOptions list the types [14:33:52 CEST] <nevcairiel> and you can still use a string to set it, or get a string back from a int option [14:34:21 CEST] <michaelni> theres av_opt_set_int() [14:34:48 CEST] <michaelni> av_opt_get_int() [14:34:53 CEST] <nevcairiel> I just don't see how thats any kind of advantage over accessing the struct itself [14:35:07 CEST] <nevcairiel> you have to enumare options programmatically to get that information [14:35:16 CEST] <nevcairiel> which means a developer has to run code to figure out how to write code [14:35:45 CEST] <nevcairiel> (or go read the code, which is not distributed with the installed headers) [14:36:32 CEST] <michaelni> the advantage is for applications which try to support a wider range of lib versions for example or to for example build a GUI list of the available options [14:36:58 CEST] <nevcairiel> AVFrame and AVPacket don't have options [14:37:03 CEST] <michaelni> for example sample_rate might be changed to AVRational one day or channel_layout might become more than a 64bit bitmask [14:38:57 CEST] <michaelni> true about AVFrame/Packet it could still help apps trying to support multiple distros and old versions though [14:39:39 CEST] <michaelni> also AVFrame has AVClass & AVOption avcodec_get_frame_class() [14:40:25 CEST] <nevcairiel> maybe we should just include some helper macros to give them a standardized way to check versions, instead of shoving these objects into every single struct [14:41:11 CEST] <nevcairiel> supporting multiple ABIs from the same binary is never going to really fly [14:41:17 CEST] <michaelni> iam not sure what you suggest, iam just scared of the kind of code we ended up in mplayer to support only a small number of versions long ago [14:41:49 CEST] <michaelni> it was really ugly #if VERSION this version that and dozends such lines [14:43:02 CEST] <nevcairiel> avoptions only cover a minority of the changes either way, there is plenty API changes you still need that for [14:43:38 CEST] <michaelni> avoptions in AVCodecContext/Format fixed most of it IIRC [14:44:44 CEST] <michaelni> it would still be needed for avcodec_decode_audio/video123456789() of course [14:45:28 CEST] <michaelni> and thats also why iam always unhappy support for these things get droped :) [14:46:39 CEST] <michaelni> its not that it personally affects me i dont maintain any affected code, i just had worked on and saw this in the past [15:47:46 CEST] <michaelni> nevcairiel, the AVClass/AVOption vs. something else vs. nothing question needs to be awnsered before the release, as AVCodecParaeters cannot be changed afterwards easily [17:32:28 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vr9Nc [17:32:28 CEST] <KGB> 13FFV1/06master 14fd03170 15Jérôme Martinez: More explicit version tests... [18:58:29 CEST] <saste> durandal_170, what about creating #ffmpeg-meeting and updating the topic here to point to the new channel [18:58:36 CEST] <saste> better to have a dedicated channel [18:59:00 CEST] <durandal_170> sure [18:59:45 CEST] <durandal_170> saste: but someone is already there... [19:01:34 CEST] <BBB> durandal_1707: then create another channel? [19:02:10 CEST] <durandal_170> ffmpeg-meeting2016 [19:05:14 CEST] <BBB> oh youre operator in #ffmpeg-meeting also now [19:05:16 CEST] <BBB> weird [19:05:16 CEST] <BBB> anyway [19:05:19 CEST] <BBB> do as you pelase [19:05:21 CEST] <BBB> *please [19:05:36 CEST] <saste> BBB I asked c_14 to move ops to durandal [19:07:56 CEST] <saste> durandal_170, please state where the meeting should be held and let's start [19:09:00 CEST] <jamrial> those who confirmed would come are already there. we could wait a bit more to see if someone else shows up [19:09:02 CEST] <durandal_170> saste: its ffmpeg-meeting2016 [19:09:16 CEST] <durandal_170> could you update topic here? [19:09:38 CEST] <saste> durandal_170, yes [19:10:20 CEST] <michaelni> durandal_170, also please set topic in the other channel to point to the used channel [19:11:16 CEST] <michaelni> so noone mistakley sits there waiting not realizing that ffmpeg-meeting2016 is used [19:11:29 CEST] <michaelni> durandal_170, thx [19:13:00 CEST] <saste> will send an email to ffmpeg-devel with the channel info [22:33:00 CEST] <DHE> I have a suspect in the mpeg-ts wrap mystery [22:33:25 CEST] <cehoyos> A sample would be better;-) [22:34:10 CEST] <DHE> sadly you actually need a 26.5 hour sample. so I'd have to generate something really outrageous (like .01fps video or something) [22:34:27 CEST] <cehoyos> Maybe you can tell us how to create the sample? [22:34:40 CEST] <JEEB> DHE: try grabbing first an hour around the wraparound point [22:34:50 CEST] <DHE> umm... I actually recorded an OTA feed for over a day. It's about 90 GB large [22:34:59 CEST] <JEEB> if that doesn't make it happen then you are sadface [22:35:08 CEST] <cehoyos> JEEB: I believe FFmpeg completely hides the issue (what issue?) in that case. [22:35:20 CEST] <JEEB> it *should* [22:35:26 CEST] <JEEB> as in the usual mpeg-ts wraparound [22:35:26 CEST] <DHE> JEEB: I think it's actually bad logic in the wrap handling. even though the timestamps don't start at 0 (live streams will pick up at any point) it's when the timestamp passes its own starting point [22:35:33 CEST] <cehoyos> I suspect it does (at least for file input) [22:36:04 CEST] <JEEB> hmm [22:36:13 CEST] <DHE> if you start at a timestamp of "10h5m" (for the sake of being arbitrary) and then run into "26h30m + 1s == 0s" it's properly wrapped to "26h30m1s" [22:36:26 CEST] <JEEB> right [22:36:58 CEST] <DHE> but when you eventually come back around to "10h5m" it stops wrapping. The client expects "36h35m" but gets back "10h5m" again [22:37:42 CEST] <JEEB> I should really get my OTA/satellite setup fixed in august ;) [22:37:44 CEST] <nevcairiel> yeah i dont think it handels such long running timestamps [22:37:58 CEST] <cehoyos> I suspect mpegts does not support "36h35m" [22:38:17 CEST] <nevcairiel> mpegts handles infinity, its for broadcasts, those dont stop [22:38:44 CEST] <JEEB> yeah, you just have to handle the timestamp wraparound correctly. this sounds like it happens in the logic after that [22:38:55 CEST] <cehoyos> But it never shows "36h35m" but "10h35m" as often as necessary depending on the length of the file, or do I misremember? [22:39:07 CEST] <DHE> it doesn't. libavformat/utils.c has code to do the wrap-around and mpegts announces its PTS wraparound policy correctly. utils.c is just mishandling it [22:39:13 CEST] <DHE> I'm still debugging, so gimme a few minutes [22:40:00 CEST] <nevcairiel> from what I remember how wraparound in utils.c works, it only allows one wrap, and only up to the maximum of time the timestamp precision can account for [22:41:50 CEST] <DHE> that seems to be about right [22:43:02 CEST] <DHE> since pts and dts can vary, you can end up with both of them on distinct halves of the wraparound point (which is the timestamp reading at the time of joining the stream). this can result in pts < dts right out of av_read_frame() [22:43:39 CEST] <DHE> cur_pkt = {pts = 1357514296, dts = 9947433873, ... } [22:44:18 CEST] <cehoyos> Please provide such a sample [22:44:59 CEST] <DHE> This should be interesting... [22:45:03 CEST] <JEEB> we'll have to see how simple the case is to make a small sample of :P [22:45:16 CEST] <cehoyos> The sample does not have to be small... [22:50:36 CEST] <DHE> well I'm not giving you my 90 GB sample either [22:50:40 CEST] <Illya> cehoyos: Uploading 90GB sample isn't particularly practical though [22:50:53 CEST] <Illya> Uploading a* [22:51:49 CEST] <cehoyos> So far, 10G was the biggest sample, is 24h SD channel really 90G? [22:52:35 CEST] <DHE> who said it was SD? [22:52:43 CEST] <DHE> :) [22:52:49 CEST] <DHE> it's ~7.5 megabit HD [22:53:10 CEST] <cehoyos> The issue is not reproducible with an SD channel? [22:53:15 CEST] <rcombs> clip just the bits close to the wraparounds? [22:53:18 CEST] <cehoyos> (DVB-T) [22:53:24 CEST] <rcombs> in theory that should repro the issue just as well [22:53:34 CEST] <cehoyos> I don't think so. [22:53:34 CEST] <rcombs> since mpegts can resync you could even do it bytewise [22:53:39 CEST] <DHE> rcombs: doesn't work that way. the bug manifests when wraparound occurs and the PTS/DTS returns to its starting point when ffmpeg was launched [22:53:49 CEST] <DHE> actually that's not a bad idea... [22:54:01 CEST] <JEEB> yeah [22:54:02 CEST] <DHE> I could literally crop out everything else [22:54:07 CEST] <JEEB> aye [22:54:17 CEST] <cehoyos> DHE: Please send an email to the user mailing list with command line and complete, uncut console output; I will then try to reproduce. [22:54:27 CEST] <rcombs> probably would show up weird in an actual player but who cares [22:55:26 CEST] Action: DHE is getting byte offsets first for the cutting [22:55:36 CEST] <DHE> the dev list? [22:57:09 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vrHLj [22:57:09 CEST] <KGB> 13FFV1/06master 14c1cf473 15Jérôme Martinez: Slice structure clarification... [22:57:21 CEST] <Illya> user mailing list, unless you have a fix for it (I think) [22:57:53 CEST] <DHE> I have 2 patches under Git to my name. I might be able to fix this... [23:21:38 CEST] <DHE> I have a few ideas for solutions, but they tend to not be so hot in the face of bad input (and I've seen bad input) [23:22:35 CEST] <JEEB> breaking on bad input is less bad than breaking on valid input [23:22:49 CEST] <JEEB> also it depends on the level of badness etc [23:22:51 CEST] <nevcairiel> unless its breaks spectacularly worse than before [23:24:16 CEST] <DHE> well, I have a custom libav* app that (tries to) deal with this. I have a record of the last dts. while (recevied_dts < last_dts) received_dts += wrap_period; [23:24:33 CEST] <DHE> and then while (pts < dts) pts += wrap_period; [23:25:02 CEST] <DHE> but a little bit of DTS inconsistency/error and this could produce bad numbers [23:28:16 CEST] <DHE> the concept could be improved to not actually loop, but I figured 1 iteration per day of processing was small enough to be ignored [00:00:00 CEST] --- Sun May 29 2016

1 0

ffmpeg.log.20160528
by burek 29 May '16

29 May '16

[00:00:37 CEST] <allquixotic_> rkern: Looks like you are the developer of the videotoolbox h264 encoder? :) -- Would you perhaps know what option I can pass to the ffmpeg command to set the quality factor? [00:01:34 CEST] <allquixotic_> Or are there presets? Or any other kind of mechanism for controlling the tradeoff between file size and quality that isn't constant bitrate? [00:02:07 CEST] <rkern> It doesn't support a quality factor, just (average) bit rate. [00:03:19 CEST] <allquixotic_> OK, and that will be driven by QSV under the hood on supported hardware? [00:06:11 CEST] <rkern> I assume so - Videotoolbox abstracts that away. iOS for example uses different hardware. [00:07:21 CEST] <allquixotic_> Hmm. In your work on VideoToolbox, have you found any sort of "codec query" mechanism to retrieve the available codecs and which ones may be default or so? [00:07:44 CEST] <allquixotic_> (Independent of ffmpeg, that is, just some kind of system utility or even a compilable program) [00:08:36 CEST] <vade> heres a dumb question - I need to calculate the duration of a synthesized video frame to ensure I hit a constant frame rate for generated packets. I thought I had it right, but i have a file where my r_framerate is 2997/125 (23.976), and my destination time base for the codec is 5994. How do I discern a correct frame duration in codec time base ? I tried av_rescale_q(1, , r_frame_rate, codec->time_base), but it seems way off [00:09:59 CEST] <rkern> allquixotic_: You can use VTCopyVideoEncoderList() to query the encoders. [00:10:47 CEST] <allquixotic_> Thanks! [00:16:23 CEST] <rkern> vade: the second parameter should be 1/r_frame_rate or 125/2997 [00:16:43 CEST] <vade> ah! [00:20:32 CEST] <vade> thanks, that looks like it was it. Derp. [00:23:48 CEST] <llogan> wallbroken: -c copy -metadata:s:v:0 rotate="90" [00:31:08 CEST] <BobDole> I have an mp4 file (video + audio) and an mp3 file (audio). I want to create an mp4 file that has video + 2 audio tracks (add the mp3 file as an audio track and make it the default). I am doing "ffmpeg -i .\v.mp4 -i .\a.mp3 -map 0:v -map 0:a -map 1:a -codec copy -shortest output.mp4", but it keeps setting the mp4's audio track as the default audio track regardless of the order I do it in. [00:31:08 CEST] <BobDole> How do I make the mp3 be the default audio track? [00:39:12 CEST] <rkern> allquixotic_: I had some code lying around that pulls the encoder info. It's pushed to https://github.com/badfroggy/VTEncInfo. [00:40:49 CEST] <llogan> BobDole: maybe the -disposition option has something to do with that [03:44:30 CEST] <drazin> can anyone help me apply a patch to ffmpeg on windows [03:47:20 CEST] <drazin> https://gist.github.com/outlyer/4a88f1adb7f895b93fd9 [03:58:17 CEST] <drazin> trying to apply this [04:01:46 CEST] <furq> drazin: there is a way to set the default track, it's just not documented yet [04:01:55 CEST] <furq> -disposition:s:0 default -disposition:s:1 0 [04:02:03 CEST] <furq> er [04:02:07 CEST] <furq> -disposition:a:0 default -disposition:a:1 0 [04:02:59 CEST] <furq> http://vpaste.net/68lBt [04:03:20 CEST] <furq> that's from doing the reverse of that, obviously [04:03:26 CEST] <drazin> https://github.com/mdhiggins/sickbeard_mp4_automator/issues/486 [04:03:50 CEST] <drazin> looks like this guy thought of that [04:04:19 CEST] <furq> that reply isn't very informative [04:04:22 CEST] <drazin> but the dev shot it down as an mmpeg error [04:04:23 CEST] <drazin> yeah [04:04:37 CEST] <drazin> i'll try his patched files [04:04:40 CEST] <furq> if nothing else ffprobe shows one track marked as default [04:04:47 CEST] <furq> have you tested -disposition on iOS [04:05:03 CEST] <drazin> nope [04:05:12 CEST] <drazin> cuz i dont know how to do anything other than use this script [04:06:43 CEST] <furq> yeah that bug report says that -disposition works [04:06:49 CEST] <furq> and then the reply just shoots it down with no explanation [04:08:33 CEST] <drazin> hmmm so in the directory with the avcodecs theres a avcodecs.py and avcodecs.pyc which means one is compiled [04:08:40 CEST] <drazin> which do i apply the update to? [04:09:19 CEST] <furq> probably the one which isn't compiled [04:09:42 CEST] <drazin> do i delete or do anything to the compiled [04:09:49 CEST] <drazin> maybe force it to recompile [04:10:06 CEST] <furq> i believe it'll do it when you run the script [04:10:11 CEST] <furq> i try my best to avoid python though [04:11:16 CEST] <drazin> i renamed them to .old and ran script [04:11:18 CEST] <drazin> it made new ones [04:11:20 CEST] <drazin> testing now [04:58:04 CEST] <drazin> Damn. furq didn't work [04:58:20 CEST] <drazin> Still echo on iOS [06:24:27 CEST] <galex-713> is it possible to ask ffmpeg to do conversion block by block from the end of a video? [07:03:57 CEST] <AndrewMock> Is there a way to measure something like SNR between out.mp3 and in.wav? [07:04:04 CEST] <AndrewMock> can't find one [07:15:36 CEST] <AndrewMock> oh well [07:15:48 CEST] <AndrewMock> i will work on PSNR in matlab [07:16:01 CEST] <AndrewMock> until then i will invert one and play both to hear differences [09:56:41 CEST] <Kudas> Hello. Guys i need an help with an encrypted m3u8: [09:56:43 CEST] <Kudas> #EXT-X-KEY:METHOD=AES-128,URI="00001.key",IV=0x1899baa9b6f9a3a4647f2adbc1ac3ce0 [09:57:07 CEST] <Kudas> I have the 00001.key in the same path of the m3u8 but what about the IV? [10:01:03 CEST] <Kudas> If i try i have this error --> Protocol not on whitelist 'file,crypto'! [11:43:38 CEST] <wallbroken> unfortunately ffmpeg remove most of the original metadata [11:44:07 CEST] <wallbroken> for example romoves "iphone 6" as the device camera [14:18:57 CEST] <lilibox> hi, dunno if this is question to this room, i would like to make ffmpeg static from source, what is official page describes how to set environment to do it? [14:26:12 CEST] <DeHackEd> a static link of libav* is the default. for a full static link you have some prereqs to meet [14:52:05 CEST] <Fouad> hi all [14:52:09 CEST] <Fouad> how are you [14:52:27 CEST] <Fouad> i want to donwnload video from youtube with ffmpeg [14:52:34 CEST] <Fouad> is it possible ? [14:53:12 CEST] <c_14> If you have something that will parse the url for you, sure. [14:53:16 CEST] <c_14> But honestly, just use youtube-dl [14:58:39 CEST] <Fouad> thenk you c14 [14:58:49 CEST] <Fouad> can you show me how can i proceed [14:58:56 CEST] <Fouad> to download video [15:00:01 CEST] <JEEB> Fouad: just use youtube-dl, it makes no sense for you to try to duplicate what that project is doing [15:00:07 CEST] <c_14> If you just want to download the video, just call youtube-dl <url>; if you want to directly pass it to ffmpeg, call youtube-dl -g <url> and then ffmpeg -i <output of youtube-dl> [15:00:37 CEST] <Fouad> i use vb.net [15:00:51 CEST] <Fouad> for that [15:01:06 CEST] <Fouad> i want to integrate it into my application [15:01:26 CEST] <Fouad> youtube-dl is parte of ffmpeg or auther tool [15:01:48 CEST] <c_14> https://rg3.github.io/youtube-dl/ [15:02:23 CEST] <DeHackEd> heck, youtube-dl uses the ffmpeg cli tool when it needs it [15:04:52 CEST] <Fouad> thank you [15:08:32 CEST] <nifwji2> I have a problem. [15:08:56 CEST] <nifwji2> I am trying to encode 180 images into a 6 second video. [15:09:43 CEST] <nifwji2> but everytime I run the command it gets to frame 12 before it starts freezing my computer. [15:09:49 CEST] <nifwji2> I was forced to restart earlier. [15:09:55 CEST] <nifwji2> ffmpeg -r 30 -f image2 -s 5153x4113 -i %d.png -c:v libx265 -preset ultrafast -x265-params lossless=1 OUTPUT.mkv [15:10:01 CEST] <nifwji2> this is the command I am using. [15:10:37 CEST] <c_14> you shouldn't need that -s and you should probably use -framerate instead of -r [15:10:41 CEST] <c_14> how much RAM does your system have? [15:10:47 CEST] <c_14> And does it have swap? [15:11:07 CEST] <nifwji2> my system has 4gb of ram. [15:11:13 CEST] <nifwji2> and I don't know what swap is. [15:11:39 CEST] <c_14> check the output of free -m [15:11:45 CEST] <c_14> the second line [15:12:47 CEST] <nifwji2> ssecond line? [15:13:27 CEST] <c_14> third line, the one that says Swap: [15:13:28 CEST] <nifwji2> also the video is green when I play it back. [15:14:13 CEST] <nifwji2> I am so confused. [15:14:20 CEST] <nifwji2> what lines? [15:14:57 CEST] <c_14> If you run `free -m' the output should look something like this https://pb.c-14.de/t/kng.VqkOF2 [15:16:31 CEST] <nifwji2> free isn't a recognised command. [15:16:46 CEST] <c_14> Are you on windows? [15:16:49 CEST] <nifwji2> and it doesn't work when I use it as an argument with ffmpeg. [15:16:51 CEST] <nifwji2> I use windows. [15:16:54 CEST] <c_14> ah [15:17:18 CEST] <nifwji2> windows 10 is supposed to be "lightweight" [15:17:20 CEST] <c_14> Windows does have a pagefile, but I'm not sure how that works. [15:17:24 CEST] <nifwji2> but it uses like 2gb of ram. [15:18:05 CEST] <c_14> Anyways, what I'm assuming is happening is that ffmpeg uses up all the RAM on your computer and then your computer starts swapping to disk and everything slows to hell because your hard disk is much slower than your ram [15:18:18 CEST] <furq> nifwji2: wmic pagefile list /format:list [15:18:33 CEST] <furq> AllocatedBaseSize is your total swap space [15:18:39 CEST] <c_14> encoding images at that resolution is not easy on your ram [15:18:48 CEST] <nifwji2> I have 180 images that are 1.5mb each [15:18:59 CEST] <nifwji2> I got them from a flash game. [15:19:09 CEST] <nifwji2> it is a mini cutscene [15:19:29 CEST] <nifwji2> my computer is just not cut out for this level of encoding. [15:19:36 CEST] <c_14> you're either going to need more ram or smaller images [15:19:36 CEST] <furq> they'll be much bigger than that decompressed though [15:20:05 CEST] <nifwji2> my media player is using 800mb of ram right now [15:20:11 CEST] <furq> that's about 11GB uncompressed for 180 images [15:20:18 CEST] <nifwji2> I tried opening the tiny video file with only 20 frames in it. [15:20:25 CEST] <furq> plus x265 on top of that [15:21:23 CEST] <nifwji2> the crazy thing. [15:21:33 CEST] <nifwji2> I was planning on encoding all the games cutscenes. [15:21:43 CEST] <furq> also i'm pretty sure that command won't be lossless because it'll convert from rgb to yuv [15:21:57 CEST] <nifwji2> it's cutscense require you to click through them. [15:22:09 CEST] <furq> you might have better luck using a different codec [15:22:10 CEST] <nifwji2> so I would need to extract about 15000 frames [15:22:17 CEST] <furq> or just downscaling the images [15:22:48 CEST] <nifwji2> then edit it to make it pause at certain points. [15:23:04 CEST] <nifwji2> then edit together all of the audio manually. [15:23:13 CEST] <nifwji2> and then combine the files together. [15:23:26 CEST] <nifwji2> then I will have a super high quality cutscene. [15:24:05 CEST] <nifwji2> assuming my computer could handle it. [15:24:09 CEST] <furq> you could just use apng/mng [15:24:23 CEST] <nifwji2> what would a good command be to losslessly encode it. [15:24:25 CEST] <furq> it sounds like png is already doing a pretty good job of compressing those images [15:25:23 CEST] <nifwji2> it took me 1 hour and 11 minutes to export all 180 frames of the animation. [15:25:33 CEST] <furq> actually nvm if you want the audio then apng is no good [15:25:46 CEST] <nifwji2> so imagine how long it would take for like 8 minutes of video. [15:29:57 CEST] <nifwji2> vector graphics is really cool. [15:30:04 CEST] <nifwji2> why isn't it more commonly used? [15:57:16 CEST] <nifwji2> I have another problem. [15:57:34 CEST] <nifwji2> apart from not being able to export the audio correctly. [15:57:50 CEST] <nifwji2> the frames aren't perfect rectangles. [15:58:00 CEST] <nifwji2> they have a bunch of junk outside the edges. [15:58:18 CEST] <nifwji2> this is how most cutscenes work in flash. [15:58:35 CEST] <nifwji2> but normally you don't see all the junk outside the edges. [16:07:43 CEST] <k0ral> Hello [16:09:36 CEST] <k0ral> I have 3 audio files (.ogg) that I'd like to mix into one; they're exactly the same length, I tried `ffmpeg -i file1.ogg -i file2.ogg -i file3.ogg -filter_complex amix=inputs=3 -c:a libvorbis -q:a 6 output.ogg`, but there's something that bothers me: [16:10:05 CEST] <k0ral> each of the input file is encoded in a bitrate 501 Kbps [16:10:40 CEST] <k0ral> now I observe that depending on the `-q:a N` I choose, the bitrate of the mixed audio file changes [16:11:18 CEST] <k0ral> would it be possible to tell ffmpeg that I'd like to keep the same bitrate ? [16:12:53 CEST] <DeHackEd> you should switch to using "-b:a 501k" then instead of -q:a [16:15:18 CEST] <__jack__> k0ral: -q:a means "keep a constant quality", that's a good idea [16:15:38 CEST] <__jack__> bitrate is useless, unless you are targeting a specific player with specific bitrate limitation [16:16:25 CEST] <DeHackEd> and 500 kilobit is actually pretty high for what vorbis can do, unless you're dealing with 5.1 surround sound or something to that effect [16:20:38 CEST] <k0ral> I just don't want to lose information [16:21:36 CEST] <k0ral> besides, bitrate is variable, 501k is the average I guess (as reported by mediainfo) [16:22:26 CEST] <k0ral> so `-b:a 501k` might lose some info [16:25:49 CEST] <DeHackEd> if that's the case, then the source material was probably encoded with -q mode [16:26:37 CEST] <k0ral> so there's no such thing as "keep all the input information" ? [16:30:47 CEST] <__jack__> k0ral: lossless encode, maybe [16:30:58 CEST] <__jack__> you'll have a much bigger bitrate [16:31:50 CEST] <__jack__> dunno if it can be done with vorbis, seems not [16:32:03 CEST] <k0ral> looks pretty overkill [16:32:22 CEST] <__jack__> indeed [16:35:47 CEST] <DeHackEd> even if you could, each of the 3 files would have independent moment-by-moment bitrates. which do you choose? [16:36:11 CEST] <furq> yeah if you're mixing them into one stereo track then it doesn't really make sense anyway [16:36:35 CEST] <furq> if you were using amerge then it would make sense, although i don't think there's a way to do that with libavfilter [16:37:59 CEST] <nifwji2> I tried making an animated png file. [16:38:05 CEST] <nifwji2> the application crashed. [16:38:10 CEST] <nifwji2> immediately. [16:38:42 CEST] <k0ral> DeHackEd: the highest at each moment [16:39:02 CEST] <k0ral> I guess I'm mistakenly imagining this process as a linear transformation, which it is probably not [16:39:29 CEST] <DeHackEd> no, it's "decode each, apply filter to raw samples, re-encode" [16:41:01 CEST] <k0ral> ok, I'll settle with q:a 6, thanks for the help [16:51:01 CEST] <nifwji2> I tried using handbreak to rip a dvd. [16:51:38 CEST] <nifwji2> and the resulting file wasn't even recognizable. [17:02:29 CEST] <nifwji2> my computer can only handly about 10 frames before the encode breaks. [17:02:54 CEST] <nifwji2> so maybe I could encode the video into 18 seperate video files. [17:03:14 CEST] <nifwji2> then put the video files together into one. [17:47:06 CEST] <PlanC> is it worth going through the trouble to upgrade ffmpeg? [17:47:14 CEST] <PlanC> I'm on 2.6 atm [17:49:40 CEST] <BtbN> Do you need/want any of the improvements and fixes since 2.6? [17:49:48 CEST] <JEEB> I remember a very little amount of stuff that has been fixed lately, but stuff like blu-ray subtitle rendering, largely reworked AAC encoder (no longer bad), etc [17:50:25 CEST] <JEEB> since you can just build a basic ffmpeg into a user directory and test your use cases I think generally the decision of updating or not should be simple enough :P [17:50:59 CEST] <JEEB> of course if you just don't want to update then this discussion is completely meaningless [17:51:18 CEST] <PlanC> I mostly just strip the audio from my lecture videos into mp3 files [17:51:28 CEST] <wallbroken> unfortunately ffmpeg remove most of the original metadata [17:51:29 CEST] <wallbroken> for example romoves "iphone 6" as the device camera [17:51:40 CEST] <PlanC> it takes a while to do on my laptop so I thought that maybe the new update would speed the process up [17:51:54 CEST] <JEEB> if it's just -c copy then it should be pretty fast in any case [17:52:01 CEST] <JEEB> unless you're really IO limited [17:54:34 CEST] <PlanC> I used to do that but the files become huge [17:54:45 CEST] <PlanC> I save some space by encoding in mp3 [17:55:12 CEST] <JEEB> so your input audio is something like raw PCM? [17:55:51 CEST] <PlanC> it's videos that are filmed on an iphone [17:56:02 CEST] <PlanC> I transfer the videos to my pc [17:56:13 CEST] <PlanC> encode them to mp3 [17:56:21 CEST] <JEEB> post a full ffprobe output of one of them in a pastebin [17:56:28 CEST] <JEEB> and linke here [17:57:14 CEST] <PlanC> will do it later if I remember [17:57:16 CEST] <PlanC> on desktop atm [17:57:52 CEST] <JEEB> basically wanted to make sure if the stuff really has raw audio. if not you should be able to just -vn -c:a copy it into, say, mp4 or m4a with just the compressed audio [18:09:06 CEST] <furq> wallbroken: https://ffmpeg.org/ffmpeg.html#Advanced-options [18:09:08 CEST] <furq> see -map_metadata [18:10:12 CEST] <wallbroken> is not clear [18:10:15 CEST] <furq> PlanC: http://johnvansickle.com/ffmpeg/ [18:10:32 CEST] <wallbroken> i want to keep all untuched and only modify rotation tag [18:10:44 CEST] <wallbroken> all the ather must keep untouched [18:10:53 CEST] <furq> -map_metadata 0 [18:11:02 CEST] <furq> then set the rotation tag after that [18:12:10 CEST] <wallbroken> ok let me try [18:12:24 CEST] <Admin__> hey guys.. this is an annoying little thing... i am outputting http to webdav which is working well with PUT .. but the delete does not work.. anyone knwo why ? do i have to do something like "PUT,DELETE" or something like that ? [18:13:52 CEST] <wallbroken> furq: ffmpeg.exe -i input.mov -c copy -map_metadata 0 -metadata:s:v:0 rotate="90" output.mov [18:13:54 CEST] <wallbroken> that's good? [18:14:06 CEST] <furq> sure [18:15:27 CEST] <Admin__> hey.. will that give me my closed caption data out into -map ?? its embeded in video right now.. .i am using ffmpeg -copy > |hardware transcoder | ffmpeg seggent.. the hardware transcoder is loosing the closed caption.. am i able to map the data out with -map ??? maybe not right since its in video and i would have to decode first [18:15:41 CEST] <Admin__> just asking because i did not see that before [18:16:28 CEST] <wallbroken> no [18:16:31 CEST] <wallbroken> that's not work [18:16:38 CEST] <wallbroken> it deletes as the same [18:16:51 CEST] <wallbroken> com.apple.quicktime.make : Apple [18:16:51 CEST] <wallbroken> com.apple.quicktime.model : iPhone 4S [18:16:51 CEST] <wallbroken> com.apple.quicktime.software : 9.3.2 [18:16:52 CEST] <wallbroken> removed [18:17:41 CEST] <JEEB> probably it doesn't even read that apple-specific metadata and thus doesn't pass it through [18:18:01 CEST] <JEEB> I have no idea why you would care about that metadata, but if you do feel free to create an issue on the tracker [18:18:13 CEST] <JEEB> preferably with a small sample file [18:18:26 CEST] <wallbroken> because i don't like that a software change things that i haven't told it to do [18:18:59 CEST] <Admin__> hey guys... does FFMPEG require some library installed for closed caption data ?? it was working then it all of a sudden stopped.. i am wonder if someone uninstalled some libary on the system which is preventing ffmpeg from copying the data to the output [18:19:11 CEST] <wallbroken> for example when i change rotation tag in jpeg photos, it changes only 1 byte rapresenting the number of the orientation [18:19:26 CEST] <wallbroken> no other thing has been changed [18:19:47 CEST] <JEEB> well ffmpeg cli by definition goes through a full demux-[whatver]-mux cycle [18:19:54 CEST] <JEEB> so if that metadata is not generic [18:20:00 CEST] <JEEB> and requires a demuxer implementation [18:20:04 CEST] <JEEB> as well as a muxer one [18:20:25 CEST] <JEEB> and if you care as much you should look at the damn difference of the files in L-SMASH's boxdumper [18:20:34 CEST] <JEEB> ffmpeg is not the tool for you if you want to do minimal changes [18:20:43 CEST] <wallbroken> ok, so, i need to write my own C software than finds rotation byte into the binary stream and then change it [18:21:31 CEST] <JEEB> or use something more specific to your container, yes. something that is designed to replicate that one container 1-to-1 in its data structures [18:21:56 CEST] <wallbroken> i don't think it exists [18:22:53 CEST] <JEEB> plus as I said, with something like ISOBMFF/mov/matroska you can add add random blocks and a demuxer should just ignore them if it doesn't know what to do with such [18:23:13 CEST] <JEEB> so whatever you would be using would have to just take that bunch of bytes in and barf it out as-is [18:24:14 CEST] <JEEB> you can use L-SMASH's boxdumper to find out the byte position of the block containing the rotation metadata [18:24:31 CEST] <JEEB> and then if you know how that block looks like byte-wise you can modify it in a copy [18:24:43 CEST] <JEEB> and then look at the resulting file once again in boxdumper [18:37:08 CEST] <Fyr> guys, when doing concatenation of three files I found out that ffmpeg instead of concatenating, removes from the first file its end of duration of the second file and adds the third file. [18:37:14 CEST] <Fyr> why is it happening? [18:38:41 CEST] <JEEB> welcome to the N different random hacks that enable you to do concatenation in ffmpeg cli [18:38:55 CEST] <JEEB> (or even the API I guess since they are filters and demuxers and protocols) [18:39:32 CEST] <nifwji2> the more I think about my project the harder it seems. [18:39:37 CEST] <Fyr> why does it remove instead of doing concatenation? [18:39:58 CEST] <nifwji2> I need to export 30000+ frames [18:40:02 CEST] <nifwji2> at 4k [18:40:13 CEST] <JEEB> first of all point out which type of concatenation you're using and then we someone can start entangling why it's not working for you [18:40:35 CEST] <JEEB> I think I've only used the concat filter so far and I was happy enough that it seemed to work in the use case I required [18:41:08 CEST] <Fyr> command: ffmpeg -f concat -i list.txt -c copy result.mkv [18:41:08 CEST] <Fyr> list.txt: [18:41:08 CEST] <Fyr> file '1.mkv' [18:41:08 CEST] <Fyr> file '2.mkv' [18:41:08 CEST] <Fyr> file '3.mkv' [18:41:28 CEST] <Fyr> those mkv files do not contain video stream, only audio ac-3. [18:42:01 CEST] <JEEB> yeh, concat demuxer. and you require copying of streams. thus only concat protocol or demuxer are usable, and both have clear issues some of which you have thus run into [18:42:23 CEST] <Fyr> instead of "1|2|3" [18:42:24 CEST] <Fyr> I get: "1-2|3" [18:43:19 CEST] <JEEB> in the short term, use mkvtoolnix to get the job done. create an issue on the issue tracker and hope that someone just doesn't do another hack [18:45:08 CEST] <JEEB> or actually since it's just AC3 you could just extract the tracks and concat them together and then remux into mkv [18:48:38 CEST] <wallbroken> -map_metadata 0 saves few bytes intead of omitting it [20:25:57 CEST] <drazin> when i try to run ffprobe.exe on a file i get permission denined [20:25:59 CEST] <drazin> whats that about [20:26:19 CEST] <c_14> the user running ffmpeg does not have read permissions on that file [20:26:45 CEST] <c_14> and/or read/execute permissions on parts of the directory tree above that file [20:29:04 CEST] <wallbroken> maybe it's an off topic question: [20:29:14 CEST] <wallbroken> when i set a lower resolution on a phone camera [20:29:17 CEST] <wallbroken> what happens? [20:29:29 CEST] <wallbroken> some of pixel sensor are switched off? [20:29:51 CEST] <wallbroken> or the video is captured at the maximum resolution and then is scaled by software? [20:34:53 CEST] <Fyr> JEEB, is it possible that this is the problem of a (de)muxer? [20:35:17 CEST] <Fyr> JEEB, long time ago I posted a bug concerning muxing abitlity of FFMPEG. [20:35:18 CEST] <drazin> yeah im running as admin [20:35:22 CEST] <drazin> i just copied the file locally [20:35:26 CEST] <JEEB> possible, but more likely that it's the timestamp handling with going from one to another [20:35:51 CEST] <Fyr> FFMPEG didn't work with offset properly. [20:36:17 CEST] <Fyr> it's possible that it will work out with the other containers . [20:36:46 CEST] <JEEB> yes, but generally that's an issue with the concat logic rather than demuxer logic [20:36:58 CEST] <Fyr> generally? [20:37:09 CEST] <Fyr> it means that this is usual. =) [20:37:14 CEST] <JEEB> yes [20:37:21 CEST] <Fyr> =( [20:37:25 CEST] <JEEB> you can go scroll up to my first comment on concat features [20:37:40 CEST] <JEEB> giant hacks made by someone that "works for me" [20:38:35 CEST] <JEEB> so basically you can have "nice" issues with the handling of timestamps coming from the demuxer, plus other stuff [20:38:56 CEST] <JEEB> I think I've mostly heard mpeg-ts working and awful stories of everything else [20:39:07 CEST] <Fyr> ok, JEEB, if those timestamps are incorrect, what should I do to avoid them? [20:39:26 CEST] <Fyr> how to stamp the right timestamps? [20:39:36 CEST] <JEEB> you want to fix the concat shit? :P [20:39:39 CEST] <JEEB> feel free [20:40:15 CEST] <JEEB> basically you'd have to improve the timestamp logic that happens when moving from one thing to another, because currently it seems like nothing special is done [20:40:40 CEST] <JEEB> and if your input demuxer isn't supposed to have random drops in timestamps, it will herp a derp and die in one way or another [20:40:56 CEST] <JEEB> actual concatenation should keep the demuxer stuff separate per-thing [20:41:01 CEST] <JEEB> unfortunately, that's not how it's done [20:41:23 CEST] <JEEB> if you wouldn't be needing -c copy then you could have used the concat filter [20:41:36 CEST] <JEEB> which seems to be the least insane (?) thing because it works outside the demuxer world [20:41:41 CEST] <JEEB> and each thing is decoded separately [20:41:58 CEST] <JEEB> so you don't have weird timestamp issues of going from o9k to zero suddenly during a single demux [20:42:05 CEST] <JEEB> and yes it sucks [20:42:34 CEST] <JEEB> but unfortunately most people only need limited solutions and FFmpeg never was really good at trying to force people to make generic solutions [20:44:35 CEST] <Fyr> ok, JEEB, is it possible to create new correct timestamps by sending the audio into a pipe and receiving it && writing it into a container? [20:46:02 CEST] <JEEB> I dunno, I would just use mkvtoolnix's tools for matroska specifically to stick things together. in general for AC3 you could just extract the track, concat it bytewise and stick it into a container again [20:46:36 CEST] <Fyr> how to extract the audio? [20:46:52 CEST] <JEEB> ffmpeg -i hurr.mkv -c copy out.ac3 [20:46:59 CEST] <Fyr> thanks [20:47:13 CEST] <Fyr> it will find out -f option? [20:57:02 CEST] <Admin__> *&%*W HELL !!!! ... so i am talking a live source from sat. and transcoding it.... every 26.5 hours my streams stop.. they are mpegts sources... anyone know a way around this problem ? [20:57:09 CEST] <Admin__> nothing i do seems to help me [20:57:16 CEST] <Admin__> only way is to auto restart every day at 4am [20:57:26 CEST] <Admin__> what am i missing here [21:56:47 CEST] <Admin__> anyone know how i can get around the 26.5 our pts wrap around issue ? [22:06:02 CEST] <DHE> Admin__: I assume you meant "hour" [22:06:33 CEST] <JEEB> in theory if it's the MPEG-TS timestamp thing [22:06:39 CEST] <JEEB> the demuxer should be able to handle it [22:06:46 CEST] <JEEB> if it doesn't you could call that a bug [22:07:27 CEST] <JEEB> please start dumping the command line output (possibly with -v debug) and start poking at the relevant points of time so a report can be made [22:07:48 CEST] <DHE> I'm actually trying to fix it myself. I dumped a ~28 hour stream [22:08:12 CEST] <JEEB> I think one library has like three timestamps for MPEG-TS [22:08:29 CEST] <JEEB> because depending on what exactly you're doing you want them handled a bit differently :D [22:08:53 CEST] <JEEB> but yeah, the standard clock reset should be possible to handle, or at least should most definitely be handled correctly in the demuxer [22:08:59 CEST] <DHE> yeah, sorta. mpeg-ts has a PCR (master clock, used for streaming), a PTS and DTS for each stream... and I'm not exactly sure how you actually use the PCR [22:09:18 CEST] <JEEB> I think it was PCR and two versions of the PTS [22:09:20 CEST] <DHE> but I'm getting definite errors in libavformat/utils.c after the 26.5 hour mark [22:09:23 CEST] <JEEB> not sure though [22:09:48 CEST] <DHE> frame=1881783 fps=1986 q=-1.0 size=61079755kB time=18:03:39.18 bitrate=7695.7kbits/s speed=68.6x [22:09:51 CEST] <JEEB> also it would be interesting to see if it's in the demuxer or the muxer [22:09:59 CEST] <JEEB> or something in between [22:10:02 CEST] <DHE> it'd be faster but gdb is a slight CPU bottleneck [22:10:10 CEST] <DHE> there is definitely a demuxer issue, I'm just trying to track it down [22:11:39 CEST] <JEEB> cool [22:13:50 CEST] <DHE> mpegts seems to show up everywhere. the ATSC broadcasts use it. it's common in IPTV. I even have a Component HD->USB capture device that outputs mpegts directly [22:14:41 CEST] <JEEB> yeah, it's a simple streamable container [22:15:05 CEST] <JEEB> also fun fact: if you try to use mpegts timestamps for remuxing you can get awful crap out in some cases. [22:15:11 CEST] <JEEB> (in libavformat) [22:15:26 CEST] <JEEB> a lot of stuff depends on the video/audio decoder to uncrap it [22:15:28 CEST] <JEEB> :D [22:15:49 CEST] <JEEB> thankfully it seems like it's a somewhat limited case though, not every stream causes those failures [22:16:02 CEST] <JEEB> haven't had time to look into it more, but it seems like a general demuxer issue [22:17:26 CEST] <DHE> maybe this is better suited to the dev channel.. [22:25:06 CEST] <dj_beirut> can burning subties be done when gpu transcoding using vaapi? [22:26:58 CEST] <DeHackEd> should be. burn them in using a filter so that the actual image going into the encoder has the text on them [22:27:51 CEST] <dj_beirut> DeHackEd subtitles filter does not seem to be available when using vaapi. [22:27:54 CEST] <jkqxz> dj_beirut: There is no support for doing it on the GPU side, but downloading the surface, doing it on the CPU and uploading it again works. [22:28:30 CEST] <dj_beirut> jkqxz can you please explain or show me an example ? [22:28:34 CEST] <DeHackEd> is that a pure-GPU transcode? decode, reencode all in hardware? [22:29:14 CEST] <Fyr> how much information is lost when converting from one codec into the same one? [22:29:15 CEST] <jkqxz> What is does your transcode look like currently? [22:29:41 CEST] <DeHackEd> Fyr: any decoding and re-encoding degrades the image unless a lossless codec is being used (unlikely for hardware accelerated) [22:30:02 CEST] <Fyr> ok, DeHackEd, can it be applied to audio codecs? [22:31:24 CEST] <DeHackEd> if you only want to modify the video, the audio can usually be copied with "-c:a copy" which doesn't degrade the quality and is actually faster [22:33:51 CEST] <JEEB> Fyr: it gets decoded and then re-encoded. you can't really do it any differently [22:34:11 CEST] <JEEB> quality loss will happen with lossy formats [22:48:36 CEST] <jkqxz> dj_beirut: "ffmpeg -vaapi_device /dev/dri/renderD128 -hwaccel vaapi -hwaccel_output_format yuv420p -i in.mkv -vf 'subtitles=subs.srt,format=nv12,hwupload' -an -c:v h264_vaapi out.mkv" works for me, running at about 3/5 the speed of the raw transcode without burning subtitles. [23:28:35 CEST] <dj_beirut> jkqxz i can't get it work. btw.. the subs are from a matroska webstream. [23:28:44 CEST] <dj_beirut> so it's a live stream that has subs. [23:36:45 CEST] <jkqxz> Also, I assume you have already made it work in the software-only case? If not, then it's probably better to avoid hardware for now and solve whatever is stopping that working first. [23:39:26 CEST] <dj_beirut> jkqxz never got it to work with ffmpeg. i am using vlc for that now. i think there is / was ffmpeg bug that didn't make it possible. or at least i didn't get it to work [23:39:38 CEST] <Admin__> DHE, ty for looking at this [23:39:52 CEST] <Admin__> i am going crazy here [23:41:06 CEST] <DHE> Admin__: I have an explanation from the code, but no solution yet [23:41:25 CEST] <Admin__> should i move to devel channel? [23:42:03 CEST] <Admin__> yes i am looking my mind over it... i even tried different versions of ffmpeg... nothing... i tried to Generate fresh PTS and ignore DTS.. nothing.. nothing i do seems to help :( [23:42:33 CEST] <Admin__> DHE, can you message me privately .. i have a private request ( if possible ) [23:50:56 CEST] <DHE> bit busy atm [23:51:47 CEST] <prelude2004c> ok np.. so am i correct to understand this is some sort of PTS wrap around issue at 26.5 hours ? [23:53:57 CEST] <DHE> yeah. the wrap code "handles" it but not exactly correctly [23:54:13 CEST] <DHE> such that it happen at the 26.5 hour mark regardless of when you tune to the channel [23:56:51 CEST] <prelude2004c> ok this is a live TS stream that i am transcoding into h264 [23:57:21 CEST] <prelude2004c> so ffmpeg -i < UDP live stream > -c copy | nvtrascoder | ffmpeg -i - segment hls .m3u8 [23:57:38 CEST] <prelude2004c> so it seems that after 26.5 hours everything just stops and i have to restart in order to get the segments going again [23:59:23 CEST] <prelude2004c> i am also having a problem with closed captions inside the video ( American stream ) , they are lost somehow.. i wonder if the problem is related or if there is some filter i can use on the input to demux and sort through everything before pushing it all to the video transcoder [23:59:44 CEST] <prelude2004c> the problem is.. the transcoder does the decoding too for the streams which may also be in h264 but at a very large bit rate of 15Mbit/s or higher [00:00:00 CEST] --- Sun May 29 2016

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ffmpeg-devel.log.20160527
by burek 28 May '16

28 May '16

[00:17:10 CEST] <llogan> durandal_1707: i mentioned that trip as soon as i found out about it. [00:44:17 CEST] <cone-813> ffmpeg 03Gregor Riepl 07master:d970f7ba3124: ffserver: fixed deallocation bug in build_feed_streams [03:19:34 CEST] <AndrewMock> Are there plans to support DTS' embedded downmix coefficients? (Not just the generic ac3 spec ones) [03:36:40 CEST] <jamrial_> AndrewMock: afaik that's currently supported [03:38:33 CEST] <AndrewMock> oh dang [03:38:41 CEST] <AndrewMock> -ac 2 does that or? [03:59:06 CEST] <jamrial_> AndrewMock: no, -request_channel_layout as input file option [03:59:25 CEST] <AndrewMock> nice THANK YOU [04:00:41 CEST] <jamrial_> any 6.1 or 7.1 stream will have a 5.1 downmix, but they rarely have stereo downmix [04:01:10 CEST] <jamrial_> if there's no stereo downmix, it will instead output the 5.1 one [08:45:25 CEST] <phucnguyenv> hello [11:04:50 CEST] <BtbN> andrey_turkin, are you sure that CUDA abstraction is worth it? Enabling cuda makes ffmpeg non-free and non-redistributable anyway, so you build it on the system with CUDA and very likely the nvidia driver installed. [11:10:42 CEST] <andrey_turkin> It has a use in my situation (where we have some machines with GPUs and some without GPUs and we'd really like to keep the same build for both). I guess there are others with same needs [11:11:25 CEST] <andrey_turkin> Regarding non-free and non-redistributible - I am not sure why it is has to stay that way [11:11:43 CEST] <andrey_turkin> given as nvenc is no longer considered non-free [11:12:06 CEST] <nevcairiel> nvenc headers are licensed as MIT now [11:12:12 CEST] <nevcairiel> cuda headers are still proprietary [11:12:37 CEST] <andrey_turkin> right, but nvenc found a way around that, right (just "reimplement" required bits in ffmpeg) [11:13:17 CEST] <andrey_turkin> whole ffmpeg API use just needs 2-3 more definitions to be compilable without CUDA headers [11:30:53 CEST] <BtbN> if you can get rid of the cuda.h dependency entirely, then the non-free could be dropped, yes. [11:31:20 CEST] <BtbN> Oh, what I also noticed: I don't think you need to check for LoadLibrary/dlopen in the enabled cuda part in configure. [11:31:25 CEST] <nevcairiel> that sounds like it might turn out rather ugly though, as contrary to nvenc, the cuda stuff is used in various files [11:31:57 CEST] <andrey_turkin> well the definitions go into cuda_api.h and everything else uses it [11:32:22 CEST] <nevcairiel> i would probably find that not really good to have in ffmpeg [11:32:36 CEST] <BtbN> just add $ldl to extralibs, like nvenc does. [11:32:52 CEST] <nevcairiel> we shouldnt litter our code with re-implementations of proprietary headers [11:33:20 CEST] <nevcairiel> if some definitions are hidden in a single C file that uses them fine, but an extra header seems rather ugly [11:34:37 CEST] <nevcairiel> as such, i really dont like having something like cuda_api.c/h, which in turn is even public API, since its in avutil and used from avfilter and avcodec [11:35:17 CEST] <andrey_turkin> not public but cross-library [11:35:30 CEST] <nevcairiel> anything thats cross-library is practically public [11:37:17 CEST] <nevcairiel> even if its avpriv, its part of the ABI, and as such prone to the same ABI/API change limitations as any public API [11:37:25 CEST] <andrey_turkin> Either we have to make ffmpeg builds with cuda support non-free, non-redistributable and require 1+Gb download in order to build that support in; or we need to add something similar to cuda_api.h; or we need to place same code all over the place [11:37:27 CEST] <nevcairiel> so one would have to have a really convincing argument to add that [11:38:39 CEST] <andrey_turkin> I'd be happy to hear any ideas how to achieve same result more cleanly [11:38:45 CEST] <andrey_turkin> I don't have any [11:38:50 CEST] <nevcairiel> maybe we just shouldn't [11:39:03 CEST] <nevcairiel> there are plenty of external libraries that impose some sort of limitations [11:39:09 CEST] <nevcairiel> we can't re-implement the headers for all of them [11:42:01 CEST] <BtbN> I'm against the patch as long as it doesn't drop any external dependencies on cuda.h, as it's entirely pointless otherwise. [11:42:06 CEST] <andrey_turkin> I can't really argue cuda is a big deal until ffmpeg can do something useful with it without libnpp. [11:42:40 CEST] <BtbN> And even then I think it's questionable if re-implementing larger parts of cuda.h in ffmpeg is desirable. [11:43:02 CEST] <andrey_turkin> I was going to send another patch to do just that when I got some feedback on first one [11:44:01 CEST] <BtbN> keeping CUDA non-free seems like the right thing in the first place. I'm not even sure if the CUDA runtime even still counts as a system library for the GPL. [11:44:27 CEST] <andrey_turkin> basically only missing type (beyond those defined in nvenc.c) is CUDA_MEMCPY2D [11:45:01 CEST] <andrey_turkin> any plans to bring CUDA runtime into play? For now it is just driver API [11:45:18 CEST] <nevcairiel> well for now, but you say yourself cuda is not very helpful in ffmpeg, and once someone starts implementing extra features, you'll start to re-implement a lot more of CUDA API [11:47:05 CEST] <nevcairiel> and for license purposes, you are not allowed to look at cuda.h and copy its contents [11:47:49 CEST] <andrey_turkin> you are allowed to look in the documentation and reimplement it [11:48:09 CEST] <andrey_turkin> mingw, reactos and wine guys all manage [11:48:46 CEST] <BtbN> Doesn't that depend on the license the documentation is released under? [11:49:33 CEST] <nevcairiel> those are dedicated projects to doing such things - if you want to re-implement a OSS header in a separate project feel free, i'm just saying its not something I feel suitable for ffmpeg [11:50:59 CEST] <andrey_turkin> ok then; if this looks like a bad idea to you guys so be it [11:53:57 CEST] <BtbN> andrey_turkin, regarding your local situation, the CUDA SDK contains a shim library that has all the CUDA exports, just copy that around internaly alongside ffmpeg. [11:55:00 CEST] <BtbN> For me those are in /opt/cuda/lib64/stubs [11:55:35 CEST] <andrey_turkin> I'll just keep the patch in place for my needs ) [11:55:50 CEST] <andrey_turkin> I wonder how come NVidia relicensed nvEncodeAPI.h ? [11:56:04 CEST] <nevcairiel> because j-b asked [11:56:28 CEST] <andrey_turkin> if they can do the same for dynlink_cuda_cuda.h it would be wonderful [11:56:28 CEST] <nevcairiel> maybe they'll do the same with cuda at a later point [11:58:11 CEST] <BtbN> I don't have that file in my CUDA SDK? [11:58:16 CEST] <andrey_turkin> it would be even better if they made better interoperability between opencl and nvenc/nvcuvid [11:58:24 CEST] <andrey_turkin> it's in nvenc SDK [11:58:27 CEST] <BtbN> that's not going to happen. [11:58:35 CEST] <j-b> andrey_turkin: because I can be an annoying person. [11:58:41 CEST] <nevcairiel> nvcuvid is unsupported at this point anyway [11:58:52 CEST] <nevcairiel> they didnt even bother to update it for 10-bit decoding [11:59:13 CEST] <BtbN> Kind of a shame, as there is no other comparable decode api [11:59:50 CEST] <BtbN> But Nvidia is never going to care more for OpenCL than barely acknowledging that it exists. [12:00:10 CEST] <andrey_turkin> looks that way [12:01:42 CEST] <BtbN> Until very recently the OpenCL stuff in ffmpeg didn't work on nvidia, because they didn't support OpenCL 1.1 [12:02:22 CEST] <andrey_turkin> you mean 1.2? I think 1.1 was supported for a while not [12:02:23 CEST] <nevcairiel> nvidia didnt support 1.2 for a long time, but 1.1 should've been fine [12:02:39 CEST] <BtbN> Oh, yeah, whatever ffmpeg needs. Might have been 1.2. [12:02:59 CEST] <BtbN> I hope that with EGL/Wayland and stuff there will be some way to initialize vdpau without X. [12:03:06 CEST] <BtbN> preferable on a headless machine [12:03:34 CEST] <BtbN> Or a refresh of CUVID [13:33:53 CEST] <andrey_turkin> wow that's a big patchset [13:49:36 CEST] <omerjerk> Hi everyone. [13:49:45 CEST] <omerjerk> I have a question regarding the SoftFloat api [13:50:09 CEST] <omerjerk> I want to use this av_cmp_sf function - https://www.ffmpeg.org/doxygen/2.2/softfloat_8h_source.html#l00095 [13:50:43 CEST] <omerjerk> I want to compare whether my SoftFloat is eqal ro 0.f or not [13:50:54 CEST] <omerjerk> Is it even possible with this ? [13:51:12 CEST] <omerjerk> Else I'll have to think of some other logic in my code. [13:58:07 CEST] <omerjerk> Also, what's the proper way to get the sign bit in SoftFloat ? [14:32:09 CEST] <cone-681> ffmpeg 03Anton Khirnov 07master:44d16df41387: h264_parser: eliminate H264SliceContext usage [14:32:10 CEST] <cone-681> ffmpeg 03Hendrik Leppkes 07master:2dc954e0bd54: Merge commit '44d16df413878588659dd8901bba016b5a869fd1' [16:32:48 CEST] <cone-681> ffmpeg 03Michael Niedermayer 07master:281caece46c4: avfilter/avfiltergraph: Clear graph pointers in ff_filter_graph_remove_filter() [16:44:15 CEST] <vade> hello. Is the newer codecpar API, using send / recieve + frame / packet API stable for use? Ive successfully ported / implemented a decoder - but my encoder is failing. I am using what was the latest GIT commit on github - since im wanting HWAccell encode for OS X via the h264_videotoolbox encoder. [16:45:47 CEST] <rkern> Does it give you an error message? [16:46:56 CEST] <vade> hi rkern - no, I suspect the issue is on my end. Inspecing packets I recieve from the encoder - they appear to have data (ie - non null values) and the only errors I get are stating that the encoder needs more frames - which obviously stops once it has enough data to begin to vend output packets to me [16:46:56 CEST] <ubitux> michaelni: "no i dont know if someone uses it with 1000 filters" i remember someone doing thousands of drawbox instance of something (one per frame) [16:48:01 CEST] <vade> rkern: Ive read some migration documentation, and per suggestion I create my own AVCodecContext which I configure via setting values on my created output video streams codecpar, and then initialize the codec context via avcodec_parameters_to_context [16:48:29 CEST] <vade> I used the old pattern that was warned against, which was using the streams -> codec directly. But that did result in output frames. [16:48:37 CEST] <michaelni> ubitux, interresting [16:48:56 CEST] <vade> (to be clear, that working encoder output was via the old API) [16:49:49 CEST] <vade> I imagine I am somehow not linking either my CodecContext, or my stream to my output format - but I cant seem to deduce where or why. Is there anything else one needs other than manually making a codec context and ensuring it matches your created streams codecparam settings? [16:49:50 CEST] <ubitux> michaelni: well, it was pure madness but there is a ticket somewhere [16:50:16 CEST] <ubitux> michaelni: http://trac.ffmpeg.org/ticket/5222 [16:52:44 CEST] <rkern> vade: when I setup the AVCodecContext first, and copy it to the AVStream.codecpar with avcodec_parameters_from_context() it works. [16:54:25 CEST] <vade> rkern: are there any samples using the new API ? [16:55:42 CEST] <durandal_1707> ubitux: somethng michaelni changed? [16:56:52 CEST] <ubitux> durandal_1707: i was replying to one of his remark mentioning that probably no one is doing a filtergraph with 1k filters [16:57:16 CEST] <ubitux> in one of the recent patchset on the ml [16:57:19 CEST] <durandal_1707> He is wrong [16:57:43 CEST] <vade> Oh rkern - interesting. So you init your codec context, and then you set up the codec context, and then you migrate your codec contexts settings to your output streams codec params? To be clear? [17:00:23 CEST] <rkern> vade: right, set the parameters such as width and height, open the codec, then copy it to codecpar. [17:00:38 CEST] <rkern> for encoders anyway [17:02:32 CEST] <vade> thanks rkern - ill let you know shortly. very much appreciate that input. [17:14:33 CEST] <vade> rkern: interesting - libx264 works, but it seems like the video toolbox encoder doesnt [17:14:42 CEST] <vade> h264_videotoolbox [17:15:29 CEST] <vade> your note appears to have been the key though, thank you! [17:15:57 CEST] <rkern> ok, so you were setting the values in codecpar first? [17:16:15 CEST] <vade> exactly. [17:16:30 CEST] <rkern> I'll try it out [17:16:34 CEST] <vade> I was setting up my stream, and then configuring my AVCodecContext by calling the context from params call [17:16:48 CEST] <vade> thank you again for your input - I really appreciate it :) [17:17:18 CEST] <vade> also as per the h264_videotoolbox encoder, ive yet to try any private codec options, so I might need to do additional configuration on it [17:18:47 CEST] <vade> also apologies if I drop out - internet here is shoddy. [17:22:01 CEST] <vade> rkern: ah, I noticed a subtlety [17:22:22 CEST] <vade> I need to have also have called avcodec_open2 prior to avcodec_parameters_from_context [17:22:35 CEST] <vade> otherwise my stream is unhappy [17:44:14 CEST] <rkern> vade: copying from the context to codecpar is best. The encoder may set certain values when it's opened (such as has_b_frames). [17:45:43 CEST] <vade> ah yea. Makes sense [17:46:31 CEST] <vade> ive configured my contexts and open them prior to calling avcodec_parameters_from_context and all seems well. I still cant get the h264_videotoolbox to output however [18:00:17 CEST] <rkern> vade: are you using OS X or iOS? [18:00:27 CEST] <vade> rkern: OS X [18:01:58 CEST] <cone-681> ffmpeg 03Håvard Espeland 07master:9c43703620a8: avcodec/proresdec2: Add support for grayscale videos [18:04:00 CEST] <vade> rkern: I think I see the issue [18:05:19 CEST] <vade> the pixel format returned by the AVCodec* that x264 returns is AV_PIX_FMT_YUV420P, whereas when I request a videotoolbox I get AV_PIX_FMT_VIDEOTOOLBOX as the only entry. Is that something to be concerned about? [18:06:48 CEST] <rkern> You can use AV_PIX_FMT_VIDEOTOOLBOX, AV_PIX_FMT_NV12, or AV_PIX_FMT_YUV420P [18:06:56 CEST] <vade> oh. I lied. Sorry. Yea. I just saw that [18:07:03 CEST] <vade> sorry im playing catch-up. Apologies. [18:10:20 CEST] <vade> do you have h264_videotoolbox running on OS X rkern ? [18:11:16 CEST] <rkern> I haven't tried it with the new avcodec_send_frame()/avcodec_receive_packet() API [18:12:23 CEST] <vade> how would I go about submitting a bug report on this? I feel like if x264 is working in my current code, but h264_videotoolbox isnt, it might be somethign in h264_videotoolbox ? is that fair ? [18:12:44 CEST] <vade> *something [18:15:22 CEST] <rkern> vade: you can submit a bug at trac.ffmpeg.org. Please include a minimal code sample if possible. [18:16:45 CEST] <nevcairiel> sounds to me like you are better suited asking in a user help context first, it sounds to me like you are not quite sure if you are using everything correctly either way [18:16:53 CEST] <vade> Thanks for all the help rkern - onel ast q - via the old API to have h264_videotoolbox working, did you have to submit any particular flags to the encoder or private options? [18:17:00 CEST] <vade> nevcairiel: this is indeed true :) [18:17:01 CEST] <nevcairiel> please note that this channel is for development of ffmpeg, and not for help using it [18:17:41 CEST] <vade> Ah, apologies. I rarely got in-depth API help on #ffmpeg, I was unclear if this was dev *using* the API, or dev for FFMPEG and is APIs only. [18:20:19 CEST] <nevcairiel> for the new decode API, most people have probably not switched to using it yet [18:20:43 CEST] <vade> yea ive not seen any real examples. How new is it ? Im fairly new to FFMPEG dev over-all [18:20:51 CEST] <nevcairiel> but I don't think it should cause anything to bug out that worked with the old one [18:20:59 CEST] <nevcairiel> couple months [18:21:38 CEST] <nevcairiel> one of these days I should switch to using it myself ;) [18:21:40 CEST] <vade> Oh, im 99% sure its on my end, ha :) [18:22:34 CEST] <vade> Id be up for possible submitting some examples of using it - its actually fairly clean once you deduce how to set it up. the new API makes a lot of sense decoupling sending packets / recieving frames. [18:23:06 CEST] <nevcairiel> indeed, its pretty simple [18:25:48 CEST] <vade> ok, one last q - do I need any flag other than setting up a codec via avcodec_find_encoder_by_name("h264_videotoolbox") to enable HWAccell with the new API? [18:28:35 CEST] <rkern> No, hw is the only option unless you set a flag to allow a software encode (it will still use hw if available). I've jumped in #ffmpeg if you have any other questions. I'm looking at OS X with the new API now, but iOS works fine with it. [18:28:58 CEST] <vade> thanks so much. Apologies for being off topic. I very much appreciate all of your input. [18:31:39 CEST] <Compn> wonder how long steam has been distributing libavcodec/libavformat/libswscale ;P [18:31:56 CEST] <Compn> really are on every computer ever :P [18:37:59 CEST] <RiCON> at least since in-home streaming [19:48:37 CEST] <esdwdftty> I don't know, you (developers) know about it or not. I came across this, I show this information. I have a AMD and a maximum avx1. https://software.intel.com/en-us/articles/avoiding-avx-sse-transition-penal… [20:02:35 CEST] <nevcairiel> esdwdftty: we avoid these problems by explicitly zeroing the upper part of the registers when transitioning [20:03:00 CEST] <nevcairiel> (ie. method 3 they mention) [20:05:13 CEST] <esdwdftty> ok [20:26:42 CEST] <rcombs> nevcairiel: is that ar issue fixed in the libc or in binutils? [20:26:56 CEST] <nevcairiel> in mingws mkstemp [20:26:59 CEST] <nevcairiel> so i gues slibc [20:27:29 CEST] <rcombs> apparently they downgraded binutils to 2.25 a couple months back [20:27:46 CEST] <rcombs> and I needed 2.26 for my application [20:28:11 CEST] <rcombs> so I guess we'll have to build binutils 2.26 against the new libc [20:30:30 CEST] <nevcairiel> what does 2.26 offer? [20:33:17 CEST] <BBB> nevcairiel: I thought it was a combination of 1/2 and 3 [20:33:23 CEST] <BBB> nevcairiel: between functions, its 3 [20:33:34 CEST] <BBB> nevcairiel: within functions, if avx, everything is vex, else everything is non-vex [20:33:39 CEST] <BBB> which is 1/2 [20:33:39 CEST] <nevcairiel> is suppose we use vex encoding when appropriate, yes [20:36:00 CEST] <BBB> kierank: is that a protest-IWNA? [20:36:06 CEST] <kierank> no [20:36:33 CEST] <kierank> it is an IRL stuff iwna [20:36:51 CEST] <BBB> alrighty [21:01:08 CEST] <somebody_useless> Hi Guys, how do you record sample input video? I need to record some sample input clips and supply them to my open TRAC ticket, however I can't get the two recommended methods do not work. [21:01:09 CEST] <somebody_useless> mplayer --dumpstream produces garbage video that's barely visible but is missing most of the picture data [using mpegts input stream video]. [21:01:09 CEST] <somebody_useless> AVIOCAT wont compile due to physically missing C library files (in libaviutil folder). [21:01:09 CEST] <somebody_useless> Thanks! [21:10:16 CEST] <Compn> whats the input somebody_useless ? [21:10:21 CEST] <Compn> i mean , http? dvb ? [21:10:41 CEST] <Compn> hardware /dev/stream ? [22:06:08 CEST] <somebody_useless> compn, sorry! Thanks for responding. It's mpegts [22:09:30 CEST] <somebody_useless> compn, we're receiving the mpegts stream via multicast from our receivers and transcoding the mpeg2ts to mpeg4ts for WAN transport (to our customers). I need a way to record the multicast mpegts video footage; however I have not yet been able to do so. I can record the raw packet data using python but this is likely not acceptable (as was my ffmpeg and cvlc recordings) [22:09:58 CEST] <somebody_useless> I recognize that this is not the general support chat, but I really want to help your team find the problem :) [22:10:59 CEST] <somebody_useless> [I'm interested in the entire input stream and not just the video.] [22:17:16 CEST] <kierank> multicat [22:50:54 CEST] <somebody_useless> kierank! THANKS!!!!!!! [00:00:00 CEST] --- Sat May 28 2016

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ffmpeg.log.20160527
by burek 28 May '16

28 May '16

[00:57:33 CEST] <A124> Heya, looking for lossless pixel perfect commpact video fromat with RGB. qtrle in mov works, little smaller then list of png files, but still huge. Any recommendations or pointers really welcome. [00:58:13 CEST] <furq> A124: x264 does rgb [00:58:23 CEST] <furq> -c:v libx264rgb [00:59:30 CEST] <DHE> and if you set "-qp 0" then x264 will produce lossless output [00:59:56 CEST] <DHE> there's no colourspace loss in rgb mode, is there? [01:05:06 CEST] <A124> I did try that but the result was not pixel perfect, using checksum, will try again then. [01:12:39 CEST] <A124> furq, DHE: https://paste.fedoraproject.org/371306/30435014/ [01:14:45 CEST] <A124> The crcs mismatch 0xa90b4adb 0x162c4adb [01:16:46 CEST] <drv> you could use ffv1 [01:26:02 CEST] <furq> i get the same thing with ffv1 [01:26:13 CEST] <furq> i guess it's because they're using different pixel formats [01:26:50 CEST] <furq> yeah it is [01:27:17 CEST] <furq> A124: http://vpaste.net/1F1jW [01:28:30 CEST] <furq> i didn't know ffv1 supported rgb, though, so that might be a better choice [01:30:04 CEST] <klaxa> how does ffv1 compare to x264rgb compression wise? [01:30:25 CEST] <furq> no idea, but i'd expect ffv1 to be faster to encode and decode [01:30:42 CEST] <klaxa> that's true [01:30:52 CEST] <A124> furq yes, you are right, -pix_fmt rgb24 did solve that. [01:31:01 CEST] <furq> yeah x264rgb uses gbrp [01:31:19 CEST] <furq> i guess x264gbrp would be a dumb name though [01:31:20 CEST] <klaxa> gree, blue, red, ???? [01:31:35 CEST] <klaxa> *green even [01:31:39 CEST] <furq> planar [01:31:54 CEST] <klaxa> what does that encode? [01:32:39 CEST] <furq> planar encodes all red pixels, then all green pixels, then all blue pixels (or whatever order they're in) [01:32:50 CEST] <A124> furq Thank you very much. Gotta try ff1 compression ratios. [01:32:52 CEST] <furq> as opposed to packed which groups each channel together by pixel [01:33:12 CEST] <furq> any pix_fmt in ffmpeg which ends in p is planar [01:33:34 CEST] <klaxa> ooh, i see, thanks for the explanation [02:08:13 CEST] <Admin__> hey guys.. question -method PUT -multiple_requests 1 http://.. webdav [02:08:38 CEST] <Admin__> i am sending thing to webdav and yes PUT works.. the files show up.. for some reason the DELETE is not working and i don't see the request on the opposite end. [02:08:44 CEST] <Admin__> does the delete request exist ? [02:18:05 CEST] <A124> png 1.5G; x264 ultrafast 2.5G; ffv1.3 1.2G ... speed same ... with superfast it gets down to 2.4, but processing doubles, ffv1.3 with -coder 1 -context 1 does produce even smaller output. Thanks everyone, got a some rgb space saving, while having playable file in container [02:18:11 CEST] <WereCatf> What has changed as of late? Nvenc-support seems to have broken in ffmpeg. It used to work, but now I can't get it working no matter what I do. [02:18:46 CEST] <A124> klaxa that ^ [02:19:32 CEST] <furq> that's surprisingly rubbish for x264 [02:19:39 CEST] <furq> i guess it's more of a yuv codec though [02:19:42 CEST] <klaxa> wow, is ffv1 also faster than x264? [02:20:02 CEST] <furq> looks like it [02:20:04 CEST] <A124> Yeah, it is noisy lossy source converted to lossless. [02:20:08 CEST] <WereCatf> Does anyone else even use nvenc aside from me? [02:20:22 CEST] <furq> i'm sure some people do [02:20:36 CEST] <A124> klaxa at ultrafast, they are on par. When I add stuff for more compression ffv gets slower. But faster then superfast on 264 [02:20:52 CEST] <A124> 4 core, real time. [02:21:22 CEST] <furq> it's been a while since i did any lossless encoding but x264 lossless always seemed a bit poor [02:21:29 CEST] <furq> mostly because of the terrible decoding speed [02:21:50 CEST] <A124> I did like nvenc, cause speed, but software gives better qualits, WereCatf, unless you need your stuff fast or you are live streaming. [02:22:07 CEST] <furq> i assume he is streaming [02:22:28 CEST] <A124> Yeah, it was meant just as a note. [02:22:49 CEST] <A124> How does x265 lossless perform? [02:22:55 CEST] <furq> never tried it [02:23:06 CEST] <furq> it's almost certainly much slower than x264 [02:23:08 CEST] <A124> Does the default do rgb? [02:23:15 CEST] <furq> no idea [02:23:22 CEST] <WereCatf> A124: Of course I know using software would give better quality. I could just plop the placebo-preset if I wanted really good quality, but nvenc isn't *that* terrible and it's tens of times faster [02:23:25 CEST] <A124> I did use x265 lossy, very good when you know. [02:23:38 CEST] <furq> i'd stick with ffv1 for lossless stuff in general [02:23:43 CEST] <furq> it seems to be becoming a standard for archival [02:25:24 CEST] <A124> It is personal, so I do not care at some stuff, and at some I do. So sometimes want less space taken. [02:25:24 CEST] <WereCatf> Alas, neither nvenc_h264 or nvenc_hevc works anymore and I was hoping to find someone who knew something about the why, or maybe someone who could test it so I'd know if it was just me or if it's something in general [02:25:44 CEST] <furq> what error do you get [02:26:01 CEST] <WereCatf> "EncodePicture failed!: generic error (20)" [02:34:15 CEST] <klaxa> wow that sure is a helpful error message [02:34:31 CEST] <A124> furq, klaxa x265 lossless in bgrp/rgb24 is on par with default ffv1 but slow. [02:34:53 CEST] <klaxa> so ffv1 beats everyone else in everything? [02:37:44 CEST] <A124> Well, there are other things, but either lack ffmpeg support, paid, or whatever. There is also utvideo I can try, and so on. But definitely not a bad choice for lossless today. [02:38:25 CEST] <A124> I mean lossless rgb... 264/265 might do well on yuv, which it is designed in mind, did not test. [02:51:30 CEST] <A124> klaxa utvideo is faster then huffyuv and does great on yuv (nearing 2x speed), better then huffyuv. ffv (fps 0.7x) 326M, utvideo (fps 2.1x) 464M [02:51:43 CEST] <A124> Hope this helps you and / or someone else. [02:52:06 CEST] <klaxa> oh nice [02:52:57 CEST] <A124> Archival -> ffv, realtime -> utvideo, distribution -> x265, Done. [02:53:28 CEST] <A124> Oh and OPUS. I meet weekly a lot of people that dwell on mp3. [02:53:50 CEST] Action: A124 thanks everyone and goes around. [02:54:11 CEST] <klaxa> i use opus a lot [02:55:30 CEST] <AndrewMock> Are there plans to support DTS' embedded downmix coefficients? [02:56:15 CEST] <furq> mp3 isn't that bad [02:56:34 CEST] <A124> False. AAC isn't. [02:56:40 CEST] <furq> v2 lame is pretty much fine, even if it is a bit of a waste of bits [02:56:57 CEST] <AndrewMock> mp3 and aac can be used to get good quality [02:57:07 CEST] <AndrewMock> but aac can do the same with less bitrate [02:57:18 CEST] <A124> Well, idk, but my ears can hear. 256k OPUS over 320k mp3 any day. [02:57:24 CEST] <furq> the nice thing about lame is that it writes the xing header so you know which encoder was used [02:57:35 CEST] <furq> whereas you never know whether your m4a was encoded with faac or some other garbage [02:57:58 CEST] <AndrewMock> a124, exactly--per bit, the aac coded is best [02:58:00 CEST] <AndrewMock> qaac ftw [02:58:07 CEST] <furq> opus beats qaac at the same bitrate [02:58:21 CEST] <AndrewMock> no way looking up [02:58:35 CEST] <A124> AndrewMock Yeah, that one is excellent, OPUS is there too though. [02:58:46 CEST] <furq> http://listening-test.coresv.net/s/scores_by_tracks_en.png [02:59:08 CEST] <furq> and yeah, lame is only 30kbps away from being on par [03:00:03 CEST] <AndrewMock> haha cvbr what the heck [03:00:10 CEST] <AndrewMock> versus cbr [03:00:14 CEST] <AndrewMock> unfair test [03:00:51 CEST] <AndrewMock> cbr 107 kbps would likely crush the opus results [03:01:11 CEST] <furq> --bitrate is vbr [03:01:21 CEST] <AndrewMock> still [03:01:22 CEST] <furq> or rather --vbr is the default [03:01:46 CEST] <AndrewMock> i think 128k cbr for both will show a diff result set [03:01:46 CEST] <furq> i don't see how that's unfair unless aac's vbr inherently sucks [03:01:56 CEST] <furq> 128k cbr for both would be all 5s [03:02:01 CEST] <furq> that's the only reason they use 96k [03:02:06 CEST] <AndrewMock> but yeah i am surprised, i guess opus is best at vbr [03:02:24 CEST] <AndrewMock> most encoders for audio and video are best with cbr [03:02:27 CEST] <A124> Then people are deaf to my standard. [03:02:31 CEST] <furq> ? [03:03:09 CEST] <furq> i thought cbr was pretty much always worse [03:03:09 CEST] <AndrewMock> ok then, 96kbps stereo cbr would not be all fives still qaac on top [03:03:53 CEST] <AndrewMock> maybe i am crazy [03:05:21 CEST] <furq> remember that opus is a much newer codec as well [03:05:27 CEST] <furq> aac is getting on for 20 years old now [03:05:51 CEST] <AndrewMock> man codecs are fun [03:05:55 CEST] <AndrewMock> vp10 boiiiii [03:06:47 CEST] <furq> tbh i'm mostly just pleased to see that lame acquits itself fairly well [03:07:53 CEST] <furq> not that i plan on encoding anything new with it, but i still have plenty [03:11:03 CEST] <alfonsodev> Hi, I''m compiling ffmpeg 3.0 on aws instance for using it a lambda function, [03:11:29 CEST] <alfonsodev> so far I can compile it successfully with this script http://pastebin.com/k83qKDKH [03:11:55 CEST] <alfonsodev> The question is how can I compile as a static binary [03:12:00 CEST] <alfonsodev> The question is how can I compile as a static binary? [03:12:02 CEST] <AndrewMock> why compile? [03:12:15 CEST] <alfonsodev> because fdk_aac can't be distributed [03:12:25 CEST] <AndrewMock> why fdk_aac? [03:12:38 CEST] <furq> alfonsodev: --disable-shared --enable-static [03:12:53 CEST] <AndrewMock> fdk_aac is not the "thing" anymore [03:12:58 CEST] <furq> it is the thing in ffmpeg [03:13:10 CEST] <AndrewMock> also do you lambda is good for that? thinking aobut it myself [03:13:19 CEST] <furq> the builtin encoder is good enough but fdk is still better in some regards [03:13:19 CEST] <AndrewMock> really? [03:13:21 CEST] <furq> if not all regards [03:13:37 CEST] <AndrewMock> even with new aac-native improvements? [03:13:40 CEST] <furq> yeah [03:13:47 CEST] <AndrewMock> dang good to know [03:13:49 CEST] <furq> fdk is definitely better at he-aac and vbr [03:13:54 CEST] <alfonsodev> I read is more efficient in the wiki, I'm using it for faststart and moveflags params [03:14:01 CEST] <alfonsodev> so I can do progresive download [03:14:05 CEST] <AndrewMock> nice [03:14:06 CEST] <furq> and it's probably still a bit better at cbr, but that depends who you ask [03:14:18 CEST] <furq> the native encoder is just better than faac and vo-aacenc which are garbage [03:14:28 CEST] <AndrewMock> yeah i am trying to perfect my 128k aac cbr [03:14:37 CEST] <AndrewMock> so i will return to fdk_aac [03:15:01 CEST] <alfonsodev> my usage is [03:15:01 CEST] <alfonsodev> ffmpeg -i input.wav -c:a libfdk_aac -movflags +faststart output.m4a [03:15:08 CEST] <furq> it's nice that the native encoder is being worked on but i still trust fdk more [03:15:19 CEST] <alfonsodev> I took it from the wiki https://trac.ffmpeg.org/wiki/Encode/AAC [03:15:20 CEST] <furq> until i see a decent listening test, anyway [03:16:17 CEST] <AndrewMock> -movflags and +faststart are the containter's features i think, let me double check [03:16:27 CEST] <furq> yeah that has nothing to do with fdk [03:16:57 CEST] <furq> -movflags +faststart is an mp4 muxer option [03:17:15 CEST] <AndrewMock> alfons, is this for a WWW context? [03:17:27 CEST] <alfonsodev> mobile app [03:17:35 CEST] <AndrewMock> make sure to put video with something like 5sec const keyframe intervals [03:17:41 CEST] <AndrewMock> any video? [03:17:43 CEST] <alfonsodev> it's Objective c , playing a file from S3 bucket [03:17:48 CEST] <alfonsodev> mp3 [03:18:05 CEST] <AndrewMock> nice [03:24:56 CEST] <AndrewMock> do you guys think that qaac or fdk_aac would have a better SNR for 128k cbr fullband stereo music? [03:25:12 CEST] <furq> qaac was the best aac encoder last time HA tested them all [03:25:28 CEST] <AndrewMock> sweet thx [03:25:29 CEST] <furq> they didn't test fdk but they did test fhg, which is fraunhofer's proprietary encoder [03:25:36 CEST] <furq> so you'd expect fdk to be the same or worse [03:58:26 CEST] <bumblehead> is it possible to use ffmpeg to combine audio files so that they appear on separated channels within a single audio output file? [03:59:59 CEST] <bumblehead> I'm using this argument to merge 4 wav files I downloaded from the internet [04:00:04 CEST] <bumblehead> -filter_complex amix=inputs=4:duration=first:dropout_transition=3 [04:00:26 CEST] <bumblehead> it does produce an output file with the joined result [04:00:51 CEST] <bumblehead> but they don't appear to be channel separated [04:00:57 CEST] <furq> you want amerge, not amix [04:01:01 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#amerge [04:03:14 CEST] <bumblehead> furq: thanks I will try that [04:03:23 CEST] <furq> https://trac.ffmpeg.org/wiki/AudioChannelManipulation [04:03:29 CEST] <furq> that should be useful as well [05:57:26 CEST] <bumblehead> furq: thank you [05:57:35 CEST] <bumblehead> I was able to get the result I needed [05:57:46 CEST] <bumblehead> those links were a terrific help [09:35:44 CEST] <bumblehead> JYjUD [10:36:39 CEST] <yagiza> Hello! [10:59:16 CEST] <yagiza> Keep fightin' with RTP... [11:01:30 CEST] <yagiza> Do any1 understand how RTP demuxing should work? What needs to be on specified host:port? What means connect() to UDP socket? [11:22:16 CEST] <DHE> a connect() locks the remote end of a UDP socket. you don't need to sendto(), you can simply send() and return packets will be filtered to block out other endpoints [11:26:52 CEST] <yagiza> DHE, ok, thanx [11:27:03 CEST] <yagiza> DHE, another question is... [11:28:44 CEST] <yagiza> DHE, when I try to play RTP stream, I specify just destination host and port. [11:32:00 CEST] <yagiza> DHE, how destination host should know, where to send UDP datagrams? [12:39:57 CEST] <m1dnight_> Guys, Im trying to stream my webcam from osx to youtube. I have a command that seems to work, but nothing is arriving at youtube's end.. [12:48:57 CEST] <m1dnight_> Hrm, even with VLC it dies in the end. [14:38:22 CEST] <alfonsodev> Hi, perhaps anyone here can help with this question http://serverfault.com/questions/779407/gcc-is-unable-to-create-an-executab… [14:38:59 CEST] <furq> alfonsodev: tail config.log [14:39:46 CEST] <DeHackEd> without the log, I would assume you're missing some host static libraries, like zlib or openssl or something like that [14:40:28 CEST] <alfonsodev> thanks, I'll fire up a instance now to try again and read the log [14:40:54 CEST] <furq> the configure output errors generally aren't much use without the log [14:41:10 CEST] <furq> that one in particular [14:42:23 CEST] <furq> why do you want a static binary anyway [14:43:59 CEST] <alfonsodev> Yeah, actually after last night conversation I don't need it anymore, by not using fdk_aac I can use one of the static binaries out there [14:44:26 CEST] <alfonsodev> But I'm curious about the problem I still want to be able to compile it myself :) [14:44:33 CEST] <furq> i meant instead of a shared binary [14:45:13 CEST] <furq> that command will still dynamically link against glibc, so you can't reliably move the binary between boxes unless they have the same glibc and kernel version [14:45:42 CEST] <DeHackEd> I prefer a shared binary but with a static link of libav*, libx264, etc. get the host's libraries but be able to copy ffmpeg to another system and get all the codecs go with it [14:46:41 CEST] <furq> it's less hassle to just build a shared library unless you have a lot of boxes with the exact same distro version [14:46:54 CEST] <furq> s/library/binary/ [14:47:43 CEST] <DeHackEd> true, but I think it's best to keep the number of shared libraries to a core minimum. glibc, zlib, openssl, things like that [14:48:03 CEST] <alfonsodev> In my use case is for a specific kernel that amazon uses in lambda functions , so I need to upload the binary to that environment every time I deploy a change to the function [14:48:36 CEST] <furq> that sounds fun [14:49:20 CEST] <furq> well in that case you probably want to add --static to --extra-ldflags [14:51:59 CEST] <DeHackEd> he did [14:52:02 CEST] <DeHackEd> also single dash for gcc [14:52:50 CEST] <furq> oh right it's there in the stackoverflow question [15:17:33 CEST] <alfonsodev> here is my config.log http://pastebin.com/vQ5g2MDU [15:17:59 CEST] <alfonsodev> aparently is an error at the end "cannot find -ldl" [15:18:06 CEST] <furq> yeah that's a worry [15:18:10 CEST] <alfonsodev> apparently is an error at the end "cannot find -ldl" [15:18:23 CEST] <RalphORama> Hello everyone! I'm looking for some help with using ffmpeg to capture a fullscreen application [15:18:51 CEST] <RalphORama> I've followed the guide for streaming one's desktop, but it doesn't work with the program I'm using (snes9x) [15:18:52 CEST] <furq> alfonsodev: i guess you don't have libdl.a [15:19:11 CEST] <RalphORama> Here's the script that I've written for attempting to stream the video: https://glot.io/snippets/ef2jfdazps [15:19:32 CEST] <Amitari> Does anyone know how to use ANIM-to-GIF conversion? When I type "ffmpeg -i anim.anim5 gif.gif" I only get the first frame of the animation. [15:20:38 CEST] <RalphORama> Actually, sorry. Didn't see the note about pastebin. Here's my config: http://pastebin.com/DsAkL9sD [15:21:01 CEST] <furq> RalphORama: it probably doesn't work because snes9x is using opengl [15:21:33 CEST] <furq> https://github.com/lano1106/glcs [15:21:37 CEST] <furq> google suggests that, i can't say i've ever used it [15:22:02 CEST] <RalphORama> furq: I don't know if that's the case - I'm using a Raspberry Pi model 2b running Raspbian [15:22:09 CEST] <RalphORama> I'll double check [15:22:16 CEST] <furq> you might be able to disable opengl in snes9x [15:23:40 CEST] <RalphORama> I'll give it a shot, thank you! [15:38:53 CEST] <alfonsodev> @furq I've installed libdl , now I get this http://pastebin.com/k4PPq593 [15:40:43 CEST] <alfonsodev> I think it's too much for me, I'm about to give up, unless anybody here could give a hint, thanks anyway :) [15:41:00 CEST] <DeHackEd> I can't help but feel it got cut off... [15:41:26 CEST] <DeHackEd> hmm maybe not. [15:41:33 CEST] <iive> dl stands for dynamic loader [15:42:03 CEST] <iive> so... it's the library that is used for loading other libraries. if you do static build... [15:42:45 CEST] <DeHackEd> what are you doing that needs libdl? the only thing I know is libx264's opencl support [15:42:50 CEST] <furq> oh yeah that's a good point [15:43:03 CEST] <furq> i figured ffmpeg was adding that automatically but you're supplying it to --extra-libs [15:43:34 CEST] <furq> also yeah if you're going to paste config.log just paste the last 100 lines or so [15:44:00 CEST] <alfonsodev> hehe, yeah 8K lines it's crazy [15:44:09 CEST] <furq> firefox doesn't seem to care for 13,000 line pastes on pastebin [15:44:19 CEST] <DeHackEd> leaves chrome lukewarm as well [15:56:03 CEST] <alfonsodev> I removed --extra-libs=-ldl , and I get this http://pastebin.com/9QYvfEGg [15:56:39 CEST] <alfonsodev> The command ends with "config.asm is unchanged libavutil/avconfig.h is unchanged" [16:00:58 CEST] <DeHackEd> but configure appeared to work? [16:04:46 CEST] <furq> yeah that looks fine [16:05:16 CEST] <furq> you might need to run make distclean before rerunning configure [16:06:03 CEST] <emitchell> sup [16:22:49 CEST] <flux> I'm converting a video.mp4 to video-low-quality.mp4 with ffmpeg -i video.mp4 -vcodec libx264 -crf 30 video-low-quality.mp4, but I would like the time stamps to stay exactly the same [16:23:08 CEST] <flux> however, MP4Box says the first video stars at composition time 512, whereas the resulting video starts at 1024 [16:23:19 CEST] <flux> is there some trick to this?-o [16:24:02 CEST] <DeHackEd> there's a few options that affect timestamping. You might need to experiment, but there's -copyts, -copytb and -vsync sometimes helps with mucking with timestamps (see the manuals for all options, their parameters vary) [16:24:15 CEST] <JEEB> flux: I recommend taking a look at the timestamps with L-SMASH's boxdumper [16:24:21 CEST] <JEEB> the value changes might not mean what you think they do [16:24:37 CEST] <flux> that's a good point, I should also look at composition offset [16:25:20 CEST] <JEEB> basically L-SMASH's can either dump a text representation of all (most of) the container values, or just CTS/DTS values calculated from those [16:25:24 CEST] <JEEB> depending on the mode [16:28:32 CEST] <flux> EditListBox/../@MediaTime apparently ends up different in the converted video. I think however they should be the same? I'll try the -copyts switch first, I can't believeI missed it :-o. [16:29:34 CEST] <JEEB> uhh [16:30:04 CEST] <JEEB> you're most probably not going to get the same as the copyts switch only sets API level timestamps [16:30:11 CEST] <JEEB> the timestamps then travel from the API to the muxer [16:30:30 CEST] <JEEB> you shouldn't expect bitexact remuxes in that sense. The results should be correct, of course [16:30:31 CEST] <flux> I recall it assigns the headers by the perceived first frame time stamp, no? [16:31:06 CEST] <flux> the documentation on -copyts suggests the conversion by default ignores the composition timestamp of the first frame [16:31:41 CEST] <JEEB> no [16:32:01 CEST] <JEEB> copyts is a generic option and has nothing to do with movenc specifically [16:32:10 CEST] <flux> "In particular, do not remove the initial start time offset value." ?-o [16:32:18 CEST] <JEEB> that's not specific to mov [16:32:28 CEST] <flux> does it really need to be specific to mov.. [16:32:42 CEST] <JEEB> what you're referring to is in the mov/ISOBMFF context [16:32:55 CEST] <JEEB> which this stuff doesn't speak about, rather it speaks about the API value [16:33:06 CEST] <JEEB> which comes from the demuxer [16:33:09 CEST] <vade> ive ported my decoding and encoding code to use send / recieve packet/ frame API alongside codecpar via the latest public GIT - my decode works fine - however, encode gets silence and black frames. Im creating my own CodecContext for audio and video as per the new codecpar spec - do I need to do anything to my destination Streams CodecContext? Why might I be getting black frames when my old path via encode calls worked? [16:34:42 CEST] <vade> previously Id use my avstreams -> codec to do encoding with [16:41:03 CEST] <vade> (ie open it, etc, pass it to encode) - however since I cant access it, I create a context from my streams set up codecparameters as suggested. I appear to be encoding packets correctly from what I can tell. My source frames data[] is populated, has the correct dims, and my packets also appear to have encoded data as well [16:41:29 CEST] <vade> I dont have errors on muxing either, but, I get nada on output :X [16:41:45 CEST] <vade> and OS Xs AVFoundation based player spews TONS of video decode errors ive not seen either [16:42:17 CEST] <vade> (decode errors trying to play my resulting file) - im curious if anyone is using codecpar path successfully [17:06:38 CEST] <drazin> Hi everyone, i've been using a script to auto convert some multi audio track mkv files to mp4 which is adding a new aac 2.0 track and the issue i'm getting is that apple tv and ios devices are saying they are hearing both the aac 2.0 and the original ac3 5.1 track being played at the same time. i did some digging and the script creator is saying "This is an [17:06:38 CEST] <drazin> unfortunate limitation of FFMPEG. There's no way to set a track as enabled and others as disabled. Its annoying and the only way around it at this time is to patch FFMPEG. The issue with the script is documented here: https://github.com/mdhiggins/sickbeard_mp4_automator/issues/360 has this been fixed or is there some way to fix this issue? [17:12:06 CEST] <DeHackEd> you can select which tracks to import. Use ffprobe or the output of ffmpeg to read which tracks are available, then select what you want with "-map 0:0 -map 0:2" [17:12:43 CEST] <drazin> i want all the tracks and the newly created 2.0 mixdown [17:14:35 CEST] <drazin> ive sent a note to the dev about adding that -map to the script but i havent heard back [17:14:40 CEST] <drazin> wanted to make sure thats a solution [17:15:13 CEST] <DeHackEd> it means that one of the audio tracks will be dropped during processing [17:19:13 CEST] <drazin> so i cant keep them all and select the new 2.0 as the default? [17:19:17 CEST] <drazin> and the other to off? [17:19:42 CEST] <furq> doesn't the aac track need to be first on iOS [17:20:01 CEST] <drazin> it is being added as first [17:20:06 CEST] <furq> nvm then [17:20:28 CEST] <drazin> but apparently they are all on by default which is why i'm hearing them all at once [17:20:43 CEST] <drazin> if i manually select the ac3 5.1 the echoing stops [17:39:59 CEST] <Threads> is -re recursive because i cant find much on it in the wiki [17:40:35 CEST] <c_14> https://ffmpeg.org/ffmpeg.html#Advanced-options [17:42:21 CEST] <Threads> c_14 thanks for that link bookmarked it [17:56:33 CEST] <drazin> aahh https://trac.ffmpeg.org/ticket/3622#comment:19 [17:56:39 CEST] <drazin> looks like it wont be fixed [17:57:44 CEST] <ritsuka> drazin: if you want only a track playing you need to set an alternate group, and enable one of the two tracks. iOS will select the one it prefers based on your current locale/accessibility settings. [17:58:05 CEST] <ritsuka> Obviously I have no idea how to set the alternate group with ffmpeg ;) [17:58:09 CEST] <drazin> how do i go about that [18:00:34 CEST] <drazin> who discussion about this here: https://forums.plex.tv/discussion/comment/1103854#Comment_1103854 [18:30:52 CEST] <vade> hi rkern - thanks for joining here. Anything I can look at specifically in my app to debug h264_videotoolbox vs lib x264 and why it might not be working? [18:33:06 CEST] <rkern> Is avcodec_receive_packet() returning an error? [18:34:38 CEST] <vade> I get Resource temporarily unavailable which as I understand it is the encoder warming up. I get that for a few input frames and then it goes away [18:34:50 CEST] <vade> ie, the encoder delayed frames accruing [18:36:20 CEST] <rkern> That's normal. Do you get frames after that? [18:36:30 CEST] <rkern> sorry - packets? [18:36:52 CEST] <vade> yup, I do [18:38:08 CEST] <vade> my frames are 1920x1080, and I get resulting packets size 3754, no side data, but the pts and dts is marked, and buf appears filled [18:38:27 CEST] <vade> interestingly buff size is larger, at 3786 [18:38:48 CEST] <vade> unsure why there is a size delta, or if thats expected [18:39:05 CEST] <rkern> Yeah, there's extra padding. It's ok. [18:39:27 CEST] <rkern> Maybe I don't understand the issue then. What's going wrong? [18:40:07 CEST] <vade> well, I get only black frames on output when my file is finished / muxed and I complete writing when I playback. [18:40:29 CEST] <vade> when I playback, I throw errors in console from QUicktime Player [18:41:06 CEST] <vade> H264VideoDecoder_DecodeFrame signalled err=-12910 (err) (createJVTLibInstance failed) at /Library/Caches/com.apple.xbs/Sources/CoreMedia_frameworks/CoreMedia-1731.15.202/Sources/VideoCodecs/H264/H264VideoDecoder.c line 2477 [18:41:19 CEST] <vade> however the muxed files audio plays back fine [18:41:29 CEST] <vade> so something is up with the samples being written [18:43:12 CEST] <vade> interestingly, thats a video toolbox error - kVTVideoDecoderUnsupportedDataFormatErr [18:43:32 CEST] <vade> so the encoded samples / packets I write to the muxer, from VIdeoToolbox, itself wont play [18:43:47 CEST] <rkern> Ah, I think I see the issue. AVCodecContext.extradata isn't set until the first frame is received. This is ok for some muxers, but since AVCodecParameters is copied before this, the muxer never gets this info. [18:44:27 CEST] <rkern> I assume you're muxing to mp4 or mov? [18:44:31 CEST] <vade> yup :) [18:44:34 CEST] <vade> mp4 [18:47:16 CEST] <vade> rkern: is that something I should mark as a bug ? [18:47:25 CEST] <rkern> yes [18:48:19 CEST] <rkern> I'll see if the extradata can be set when the codec is opened. As a temp work-around, you could try manually setting extradata and extradata_size in codecpar when you receive the first packet. [18:49:14 CEST] <vade> ah ok. so when I get the first packet encoded, check if codecpar on my video output stream has extra data / size set, if not, set it to the packets info? [18:50:23 CEST] <vade> oh. not the packets info, my codec context. my bad [18:50:43 CEST] <rkern> right - copy over extradata and extradata_size [18:50:53 CEST] <vade> trying now. Thank you so much for the info [18:51:21 CEST] <vade> works [18:51:28 CEST] <vade> ^^^^^^ youre awesome rkern [18:51:32 CEST] <vade> I really appreciate your help [18:52:08 CEST] <rkern> haha nice - please submit a bug anyway and I'll work on the underlying issue. [18:52:19 CEST] <vade> will do. thanks again x 1000 [19:47:55 CEST] <somebody_useless> Has anyone successfully been able to record an input stream that would satisfy the developers for troubleshooting? If so, how? [19:50:15 CEST] <DeHackEd> depends on the media. what's the source? [19:53:10 CEST] <somebody_useless> DeHackEd, MPTS [19:53:23 CEST] <DeHackEd> like, on an OTA receiver? [19:53:38 CEST] <somebody_useless> MPTS stream coming in from a satellite receiver, which is being transcoded from mpeg2ts to mpeg4ts [19:54:03 CEST] <DeHackEd> so, multicast ethernet? [19:54:08 CEST] <somebody_useless> yessir [19:54:09 CEST] <DeHackEd> ASI? [19:54:13 CEST] <somebody_useless> ethernet [19:54:17 CEST] <somebody_useless> via multicast [19:55:36 CEST] <vade> hi rkern - ive made a trac ticket for the issue you helped debug - as requested : https://trac.ffmpeg.org/ticket/5593 [19:55:43 CEST] <vade> thanks again! :) [19:56:24 CEST] <somebody_useless> I work for a small ISP and we are trying to deliver IPTV and the hurdle we've come across is with ffmpeg bugging out due to either audio codec problems (likely a compilation issue) but more specifically due to "Lost audio" on the channels; ffprobe says audio is being interlaced with the output stream, however ffmpeg complains about the audio offset being incorrect. [19:56:25 CEST] <somebody_useless> I'm trying to supply a source stream because the program output doesn't suffice (completely understandable) [19:57:01 CEST] <josh98> what is the most secure source of ffmpeg other than linux repo? [19:57:14 CEST] <somebody_useless> It's weird because sometimes ffmpeg is able to correct it, other times it just hops around for hours on end trying fix the problem [19:57:35 CEST] <somebody_useless> josh98, git assuming your ssl cert store hasn't been compromised? [19:57:46 CEST] <somebody_useless> josh98, sneakernet ftw [19:58:02 CEST] <somebody_useless> You only have to have utmost trust in one person ;) [19:58:53 CEST] <josh98> well i did ask [19:59:46 CEST] <somebody_useless> josh98, if you live in a region covered by area code 204, I might be able to help you out :) [19:59:56 CEST] <josh98> aside from sneakernet [20:00:06 CEST] <josh98> and assuming my ssl cert store hasn't been compromised [20:00:32 CEST] <somebody_useless> josh98, third part repos aren't the best, use GIT (as ffmpeg suggests), run bleeding edge [20:01:08 CEST] <somebody_useless> ffmpeg also provides prebuilt packages if you're interested in getting them that way [20:01:17 CEST] <josh98> are you recommending that i pull it from hit.ffmpeg.org? [20:01:20 CEST] <josh98> git [20:01:28 CEST] <somebody_useless> yes, that's what ffmpeg recommends [20:01:56 CEST] <josh98> as for the ssl cert store, pffft [20:02:15 CEST] <somebody_useless> not necessarily the most secure (I don't think git uses ssl), but it does guarantee a level of security via diffs [20:02:18 CEST] <somebody_useless> (hashlog, etc) [20:03:32 CEST] <josh98> https://ffmpeg.org/download/html#get-sources [20:04:06 CEST] <josh98> says source can be retrieved through git using https [20:06:16 CEST] <josh98> ssl cert courtesy of CNIC [20:08:53 CEST] <somebody_useless> perfect [20:08:58 CEST] <somebody_useless> good thing it's not CNIB [20:10:23 CEST] <josh98> sourceforge recommends using CPWN [20:14:30 CEST] <Admin__> hey guys... loooking for some help... so my source is an mpetgs stream and it has closed captions in the video PID ... i pipe it to a hardware transcoder and then back out to ffmpeg to segment. It seems that my closed caption data is missing :( .. when i look at the source direct its there.. anyone know some way to pull out the closed caption before pipping it to hardware encoder ( nvidia ) , and then it is kept and sent to the final stage for [20:14:30 CEST] <Admin__> segmenting and readding ? [20:50:47 CEST] <DeHackEd> somebody_useless: I'm trying something similar, but with an ATSC to ethernet converter [20:53:03 CEST] <somebody_useless> DeHackEd, we have a few OTA systems, but we're moving away from it because it's super old and unreliable tech. [20:53:35 CEST] <somebody_useless> ATSC via OTA [21:12:16 CEST] <Admin__> hey guys... loooking for some help... so my source is an mpetgs stream and it has closed captions in the video PID ... i pipe it to a hardware transcoder and then back out to ffmpeg to segment. It seems that my closed caption data is missing :( .. when i look at the source direct its there.. anyone know some way to pull out the closed caption before pipping it to hardware encoder ( nvidia ) , and then it is kept and sent to the final stage for [21:12:29 CEST] <Admin__> anyone know how i can extract the closed caption from the video ? [21:19:24 CEST] <DeHackEd> but it copies properly? (-c:v copy) [22:34:06 CEST] <antgeth> i have some raw aac audio that i'd like to wrap up as m4a, but the internet is not being terribly clear on how to do it [22:34:13 CEST] <antgeth> getting some conflicting advice [22:34:23 CEST] <c_14> ffmpeg -i aac -c copy out.m4a [22:35:48 CEST] <antgeth> ohhhhhhhh, i think it was the -c switch that i hadn't seen in any of the suggestions [22:35:53 CEST] <antgeth> danke! [22:38:10 CEST] <antgeth> hmm [22:38:18 CEST] <antgeth> itunes is refusing to deal with it [22:38:37 CEST] <c_14> does it give an error? [22:38:50 CEST] <JEEB> while recording? yes because unless you're using movie fragments there's no index [22:39:10 CEST] <JEEB> and not sure if QT/iTunes supports fragmented ISOBMFF [22:39:45 CEST] <antgeth> no error, i add it to the library and it just doesn't appear [22:40:13 CEST] <allquixotic_> Hi, does anyone know if ffmpeg has support for H264 encoding using Intel QSV (I guess through VideoToolbox?) on Mac OS X? [22:40:26 CEST] <antgeth> also quicktime reports the bitrate as 0 [22:40:41 CEST] <antgeth> (it does give the correct length, at least) [22:42:05 CEST] <rkern> allquixotic_: there's a videotoolbox encoder wrapper in master - not in a release yet. [22:50:12 CEST] <allquixotic_> rkern: Thanks! [22:50:22 CEST] <wallbroken> hi [22:50:27 CEST] <wallbroken> -vf "transpose=1,scale=576:-2" -metadata:s:v rotate="" -c:a copy output.mov [22:50:48 CEST] <wallbroken> ops [22:51:18 CEST] <wallbroken> ffmpeg.exe -i "Video.mov" -vf "transpose=1" -metadata:s:v rotate="" -c:a copy output.mov [22:51:27 CEST] <wallbroken> the original video is 400 kb [22:51:34 CEST] <wallbroken> the output video is 130 kb [22:51:41 CEST] <wallbroken> i think something is unclear [22:51:44 CEST] <c_14> you're reencoding it [22:51:50 CEST] <c_14> that happens [22:51:55 CEST] <wallbroken> yes i know [22:51:58 CEST] <c_14> adjust the encoder settings to your liking [22:52:20 CEST] <wallbroken> but i have an app on my iphone that rotates, and the output is 350 kb [22:52:22 CEST] <wallbroken> not 150 [22:52:39 CEST] <c_14> maybe it uses different settings [22:52:53 CEST] <wallbroken> it keeps more quality? [22:53:14 CEST] <c_14> maybe [22:53:27 CEST] <c_14> it has more bits, that's about all bitrate really tells you [22:53:47 CEST] <c_14> assuming it uses the same codec/encoder then it's probably better quality [22:53:53 CEST] <c_14> assuming they don't fuck things up [22:54:03 CEST] <wallbroken> it could be that an iOS app uses ffmpeg? [22:54:18 CEST] <c_14> it's possible [22:54:27 CEST] <wallbroken> due to licence agreement? [22:54:38 CEST] <wallbroken> nothing wrong with it? [22:54:55 CEST] <c_14> as long as they adhere to the (L)GPL there's nothing wrong with it [22:58:23 CEST] <wallbroken> that app has an option called "optimize for network use" it will start playing after only a small portion of the filehas been downloaded from the network [22:58:35 CEST] <wallbroken> is something belonging to ffmpeg? [22:58:54 CEST] <c_14> If it's an mp4, that's just moving the MOOV atom to the front [22:59:34 CEST] <wallbroken> it lets me choose the container, this function only work with mp4 ? [22:59:42 CEST] <wallbroken> i can also choose ".mov" [23:00:00 CEST] <c_14> afaik no other container needs it [23:00:03 CEST] <c_14> mov is basically mp4 [23:00:28 CEST] <wallbroken> ok and ffmpeg supports that thing? [23:00:31 CEST] <furq> wallbroken: -movflags +faststart [23:00:46 CEST] <wallbroken> by default is not enabled? [23:00:49 CEST] <furq> no [23:01:19 CEST] <wallbroken> ok [23:01:38 CEST] <furq> if you want the output video to be higher quality then set -crf to a value below 23 [23:02:52 CEST] <wallbroken> yes i've used it with h264 [23:02:54 CEST] <wallbroken> do you know? [23:03:30 CEST] <wallbroken> there was also a flag which set the encoding method. from -fast , to -placebo [23:03:39 CEST] <wallbroken> -placebo was the slowest [23:03:46 CEST] <furq> -preset [23:03:49 CEST] <wallbroken> yes [23:04:03 CEST] <furq> the default is -preset medium which is probably fine [23:04:31 CEST] <wallbroken> placebo in my case was 3 days long [23:04:39 CEST] <wallbroken> on 2 hours video [23:04:40 CEST] <furq> it's called placebo for a reason [23:04:43 CEST] <c_14> don't use placebo [23:04:51 CEST] <c_14> It's *shock* a placebo [23:05:16 CEST] <furq> don't use anything slower than slow unless you know your source will benefit from it [23:05:32 CEST] <wallbroken> i was using x264vfw, a codec for windows that it's not very good [23:05:45 CEST] <furq> you could have stopped writing that sentence after "windows" [23:06:21 CEST] <furq> although i don't remember x264vfw being any slower than ffmpeg [23:06:25 CEST] <furq> it's been ages though [23:07:08 CEST] <wallbroken> now x265 soppressed x264 ? [23:07:16 CEST] <furq> not really [23:07:37 CEST] <furq> hevc isn't very widely-supported yet and also x265 is still incredibly slow if you want it to be noticeably better than x264 [23:08:22 CEST] <wallbroken> the problem is that i got a video, and i need to get the best possible quality i can [23:08:39 CEST] <furq> use x264 with a low crf value [23:09:12 CEST] <c_14> -qp 0 (with libx264) [23:09:14 CEST] <furq> i very much doubt anyone can see any quality improvements below -crf 16 [23:09:16 CEST] <c_14> that's as good as it'll get [23:09:27 CEST] <furq> if you want "the best possible quality" then yeah use -qp 0 but the video will be enormous [23:09:39 CEST] <furq> that's only really of use if you want to process the video further [23:11:17 CEST] <wallbroken> between setting rotation tag and reencoding, which one will you suggest? [23:11:38 CEST] <furq> set the rotation tag if your player supports it [23:12:13 CEST] <wallbroken> those who supports it are too much? [23:12:32 CEST] <wallbroken> i use vlc and does it [23:12:44 CEST] <wallbroken> but i need to give the video to some people [23:12:54 CEST] <wallbroken> and i don't know which player are using [23:17:03 CEST] <llogan> then you'll have to rotate and re-encode [23:17:28 CEST] <llogan> ffmpeg will automatically rotate based on rotate metadata/sidedata [23:17:44 CEST] <llogan> se -noautorotate option [23:17:58 CEST] <llogan> *see [23:18:17 CEST] <llogan> ...for more info [23:57:32 CEST] <wallbroken> if i only want to change rotation tag keeping all the same? [00:00:00 CEST] --- Sat May 28 2016

1 0

ffmpeg-devel.log.20160526
by burek 27 May '16

27 May '16

[00:04:10 CEST] <nevcairiel> rcombs: just update to fixed mingw instead of doing other crazyness [00:04:28 CEST] <nevcairiel> its fixed in both trunk and latest 4.0 release [01:54:26 CEST] <cone-360> ffmpeg 03Michael Niedermayer 07master:89e939302237: doc/developer.texi: Add a code of conduct [08:15:56 CEST] <Zeranoe> Can someone explain why MFX_EXTBUFF_CODING_OPTION_* are being set? From what I can find they are specific to different usage situations (none of which FFmpeg is doing), and yet FFmpeg is enabling several. [08:36:06 CEST] <Zeranoe> In fact, I'm not even sure why any q->param.ExtParam are being set. [10:37:14 CEST] <BtbN> because that Quick Sync stuff is a giant mess. [10:37:28 CEST] <BtbN> Or am I confusing MFX and MXF again? [10:40:00 CEST] <nevcairiel> the extparams are fine though, there is no other way to set some of these parameters [10:40:58 CEST] <nevcairiel> which is mostly about some SEI info [13:38:45 CEST] <cone-367> ffmpeg 03Paul B Mahol 07master:d93495c95411: avcodec/vble: add frame threading support [14:27:24 CEST] <Zeranoe> nevcairiel: They aren't fine... This bug http://trac.ffmpeg.org/ticket/5324 is a result of them all being enabled without checking if the hardware actually supports it. [14:29:24 CEST] <nevcairiel> saying they are al useless and wrong because some hardware doesnt support one of them is unnecessary hyperbole [14:29:45 CEST] <nevcairiel> but qsv is crap either way, on the driver side already [14:38:20 CEST] <nevcairiel> anyway we wait for your patches [14:50:18 CEST] <jkqxz> Maybe it should be renamed to emphasise that it's not the only way to use the Quick Sync hardware. It's a somewhat dubious external proprietary library with annoying setup, and you may well be better off using other methods (dxva2/d3d11, vaapi, videotoolbox; mf if anyone wanted to write the code for it). [14:54:59 CEST] <andrey_turkin> yes on decoder side. On encoder side vaapi probably should work on Linux but no alternatives on Windows [14:59:21 CEST] <jkqxz> Hence mf, if someone wants to write it. [15:37:36 CEST] <nevcairiel> if mf is media foundation, that just ends up using libmfx/qsv in the backend once again through another layer of abstraction [15:38:05 CEST] <nevcairiel> I've seen all sorts of qsv implementations crap out on various hardware with various drivers [15:38:10 CEST] <nevcairiel> just hanging or crashing [15:42:16 CEST] <BtbN> QSV on Windows is fine [15:42:37 CEST] <BtbN> QSV on Linux is a horrible hack that should be removed from ffmpeg, specialy now as vaapi encoding is supported. [15:43:21 CEST] <BtbN> did 3.0 release with QSV stuff, or did it come after that? [15:50:21 CEST] <jkqxz> I was indeed meaning media foundation. It gains through allowing NVENC and AMDwhatever as well, even if the Quick Sync is the same as libmfx. [15:51:08 CEST] <BtbN> Well, NVENC is natively support already, as well as QSV. [15:51:28 CEST] <BtbN> So why add another layer of indirection that just potentially causes even more trouble? [15:52:18 CEST] <BtbN> Zeranoe, btw., do your builds come with nvenc support? [16:26:08 CEST] <RiCON> BtbN: yes, since last two builds [16:33:59 CEST] <andrey_turkin> that is cool [17:41:34 CEST] <Zeranoe> nevcairiel: They aren't useless, but the way FFmpeg is using QSV makes them useless. Those args are for specific use situations, not to be enabled for every encode. I'm working on a patch. [17:42:27 CEST] <Zeranoe> And I agree with QSV not being ideal, but some people use it and enjoy it and I believe it should work as best it can. [18:36:55 CEST] <cone-367> ffmpeg 03foo86 07master:a0f76b825c81: avcodec/dca: move EXSS sampling frequency arrays to dca.c [19:08:01 CEST] <cone-367> ffmpeg 03Michael Niedermayer 07master:7f5c6ea51102: avformat/utils: Fix use of uninitialized variable [19:39:02 CEST] <Zeranoe> Did something recently happen with VP9? [19:40:27 CEST] <Zeranoe> Not sure why it isn't being included in the newer builds... I don't think anything changed on my end [20:28:06 CEST] <omerjerk> Hi [20:28:19 CEST] <omerjerk> I've some doubts regarding the SoftFloat api in ffmpeg. [20:28:26 CEST] <omerjerk> anyone online ? [20:49:37 CEST] <Compn> omerjerk : wait around for answer :) [20:49:40 CEST] <Compn> not everyone online atm [20:49:42 CEST] <Compn> idleing [20:50:49 CEST] <omerjerk> okay. :) [20:51:13 CEST] <durandal_1707> omerjerk: yes? [20:52:30 CEST] <omerjerk> So carl previously suggested to use SoftFloat api instead of this - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L296 [20:53:10 CEST] <omerjerk> I've a float here - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L1762 [20:53:24 CEST] <omerjerk> I need to convert this normal float value to a SoftFloat object. [20:53:32 CEST] <omerjerk> I'm not getting how to do that. [20:53:53 CEST] <omerjerk> There's no function in the SoftFloat.h file to convert from float to SoftFloat [20:54:06 CEST] <omerjerk> durandal_1707 [21:19:20 CEST] <omerjerk> durandal_1707: Are you there > [21:19:22 CEST] <omerjerk> ? [21:19:39 CEST] <durandal_1707> no [21:22:02 CEST] <durandal_1707> omerjerk: av_int2sf [21:23:12 CEST] <durandal_1707> You must not use float type directly [21:23:46 CEST] <omerjerk> durandal_1707: so I should do av_float2int first to convert the float to int and then pass it to the function you mentioned, right ? [21:24:22 CEST] <ubitux> https://twitter.com/whitequark/status/735882229889409025 [21:24:24 CEST] <durandal_1707> no your code should not have float word [21:26:04 CEST] <durandal_1707> ubitux: great! [21:29:59 CEST] <omerjerk> durandal_1707: Any tip on how do I avoid the float here ? - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L1762 [21:31:01 CEST] <omerjerk> scale = 2^23 [21:34:34 CEST] <durandal_1707> omerjerk: convert scale to softfloat [21:56:10 CEST] <durandal_1707> I don't trust c++ apps [22:12:59 CEST] <andrey_turkin> why not? [22:14:50 CEST] <durandal_1707> have you looked at their binary? [22:16:33 CEST] <andrey_turkin> yeah [23:12:25 CEST] <BBB> jamrial_: Ill try to attend (its noon EST, right?) [23:13:12 CEST] <BBB> if (temp_pcm.f != 0.f || temp_pcm.f != -0.f) { <- isnt that the same as if (1) { ? [23:14:21 CEST] <jamrial_> BBB: 5pm UTC, so i think that's 1pm EST [23:14:51 CEST] <BBB> youre right, tx [23:15:19 CEST] <andrey_turkin> probably should be && [23:31:50 CEST] <drv> doesn't 0.0f == -0.0f anyway by IEEE 754 rules? [23:36:43 CEST] <andrey_turkin> looks that way. Maybe there's something to do with subnormal numbers? [23:37:03 CEST] <andrey_turkin> or just an error [23:51:22 CEST] <kierank> BBB: not sure what you read into my comment on #libav-devel, just giving an example to koda of who would play prores on arm32 [23:51:49 CEST] <BBB> I dont think Im ont hat channel [23:51:58 CEST] <kierank> sorry the ML [23:51:58 CEST] <BBB> oh, the ML? [23:52:38 CEST] <BBB> I dont think we should focus on existential questions like who would play prores on arm32" [23:53:03 CEST] <BBB> I think we should figure out why some part of the bitstream reader (that is apparently more predominant in some decoders than in others) got slower on 32bit [23:53:34 CEST] <kierank> I agree [00:00:00 CEST] --- Fri May 27 2016

1 0

ffmpeg.log.20160526
by burek 27 May '16

27 May '16

[00:07:43 CEST] <andross> okay this encoding example got weird [00:08:00 CEST] <andross> why couldnt the example just encode an input wav rather than a test tone [00:09:05 CEST] <JEEB> doesn't require an input file I guess [00:09:21 CEST] <JEEB> there's other examples doing demuxing and decoding into avframes [00:09:37 CEST] <JEEB> and then you just feed the encoder those avframes, om nom nom [00:12:45 CEST] <andross> think ill leave it there for now [00:44:51 CEST] <SeanM_> Hi was wondering if anyone could help me. I know that -ss 00:00:4.000 is command to start ffmpeg at a specific time (4 seconds in), but is there a way to do that for the ending? I want to remove (outdated) outros from a large group of videos. Basically cut off the last 5 seconds of a movie, where all the lengths vary but the part being removed is always the same length (5 sec). [00:45:39 CEST] <JEEB> not that I know, you can just set a seek and a duration [00:52:24 CEST] <SeanM_> Thanks. Was hoping for an easier method though, but I guess it might not be possible to do it any other way. [00:52:26 CEST] <BenMcLean> Hey there folks [00:53:53 CEST] <BenMcLean> Does anyone know how I'd convert from h264 mp4 to mpeg or whatever else Sony Vegas would accept without any stupid synchronization issues ? [00:55:09 CEST] <BenMcLean> Or on second thought, the fact that Sony Vegas won't import h264 mp4 means it is obsolete. Can anyone recommend an inline video editor that doesn't suck? Preferably a free one? [00:55:57 CEST] <furq> BenMcLean: http://www.openshotvideo.com/ [00:56:09 CEST] <furq> i have no idea if that sucks, but it uses the ffmpeg libs so it can actually import useful formats [00:56:18 CEST] <furq> and export, for that matter [00:57:47 CEST] <BenMcLean> furq, thanks! I have been needing to cut dependencies on software ... that I can't actually afford [00:58:41 CEST] <BenMcLean> I'm hoping this will let me cut and arrange videos, sounds, images and text in HD without a crap ton of technical mumbo jumbo [00:58:51 CEST] <furq> i think vegas uses directshow, so it probably can import mp4 if you install the jimmy codec pack or whatever [00:58:59 CEST] <furq> but i'd steer clear of anything that uses directshow anyway [00:59:49 CEST] <BenMcLean> I'm gonna restart so I can try out OpenShot. BRB [01:05:54 CEST] <BenMcLean> OK well, OpenShot video editor opened up, imported my clip, I dragged it onto the timeline, then pressed play to preview, and it immediately had an epic fail crash. [01:06:16 CEST] <llogan> there is also shotcut [01:06:19 CEST] <BenMcLean> This suggests to me that this is an app which is probably not ready for primetime. [01:06:40 CEST] <BenMcLean> Since, y'know, even the most basic of operations resulted in an instant crash [01:07:16 CEST] <llogan> and lightworks [01:08:44 CEST] <BenMcLean> I'll give shotcut a try [01:08:46 CEST] <llogan> or buy a one month plan for premiere pro cc for $20 [01:09:19 CEST] <llogan> oh, it's $30 unless you buy annual plan [01:09:38 CEST] <BenMcLean> Hell no, I want a program that I know is gonna work every time forever no matter whether Adobe likes me or not [01:09:50 CEST] <BenMcLean> I want Audacity with video, in other words. [01:09:59 CEST] <furq> lightworks has the same issues with importing/exporting common formats iirc [01:10:11 CEST] <llogan> i avoid these issues by no longer doing any editing [01:10:15 CEST] <BenMcLean> well I am gonna try shotcut [01:10:40 CEST] <BenMcLean> I am not trying to do anything that is even close to complicated here, that's the crazy part [01:11:08 CEST] <furq> if you just want a visual editor rather than an NLE then you could try avidemux [01:11:45 CEST] <BenMcLean> I need an NLE [01:11:54 CEST] <furq> shotcut looks quite neat actually [01:14:01 CEST] <furq> although the fact that the first question in the FAQ is "Why does it crash on Windows upon launch?" doesn't inspire confidence [01:14:14 CEST] <furq> followed by "Why does it frequently crash on Windows?" [01:16:30 CEST] <BenMcLean> Shotcut seems to be working perfectly for me so far. [01:17:34 CEST] <BenMcLean> I love the fact that it's project format is based on XML [01:17:51 CEST] <BenMcLean> It's about time somebody started using a video editing save file format that makes sense [01:18:30 CEST] <BenMcLean> I was looking at some tutorials the other day for how to use Blender as a video editor ...it's an ugly way to work [01:18:44 CEST] <BenMcLean> the program is just totally not intended for that use case [01:19:57 CEST] <BenMcLean> ooh ... it has overlay HTML filter? Like, I can plop a web page right into my videos?? i'm hoping that's what it means. that would be SUPER useful! [01:20:31 CEST] <BenMcLean> as a test, I am making a short video where I hold up Star Trek action figures in front of my camera phone. [01:20:44 CEST] <BenMcLean> it called "Odo & Quark Discuss Election 2016" [01:25:41 CEST] <farfel> good evening [01:26:01 CEST] <farfel> I have built ffmpeg with --enable-libopenjpeg [01:26:17 CEST] <farfel> but, when I decompress an image, it seems that ffmpeg uses its native decoder [01:26:22 CEST] <BenMcLean> farfel where does your nickname come from? [01:26:42 CEST] <farfel> The Inspector General..... [01:26:49 CEST] <farfel> with Danny Kaye [01:27:14 CEST] <llogan> Not the off-screen dog in a Seinfeld episode? [01:27:28 CEST] <farfel> :) no [01:27:58 CEST] <llogan> ffmpeg -c:v libopenjpeg -i input ... [01:28:06 CEST] <farfel> farfel is a jewish pasta dish from eastern europe [01:28:11 CEST] <farfel> so I can see seinfeld using it [01:28:51 CEST] <farfel> https://www.youtube.com/watch?v=RuU9gtsjzww [01:29:41 CEST] <farfel> but, enough about me [01:30:24 CEST] <farfel> lllogan: awesome, thangs [01:30:26 CEST] <farfel> thanks [01:31:01 CEST] <farfel> so, it looks like the libopenjpeg decoder could be improved on [01:31:17 CEST] <farfel> determining pixel format [01:31:50 CEST] <farfel> is there an interest in adding support for broadcast profiles in the encoder ? [01:32:13 CEST] <farfel> then it can be muxed into mpeg ts [01:32:30 CEST] <farfel> broadcast profiles and elementary stream headers [01:33:07 CEST] <kyleogrg> hello [01:33:51 CEST] <kyleogrg> I'd like to use a video duration as a factor in a bitrate calculation, automatically [01:34:47 CEST] <kyleogrg> In a command line. So how can I do something like: -b:v (duration*x)/y [01:35:15 CEST] <kepstin> kyleogrg: i.e. you're trying to target an exact output filesize? [01:35:41 CEST] <kyleogrg> yes, calculate the size of a whole batch of videos [01:36:11 CEST] <kepstin> kyleogrg: as far as I know, there's no way to do that in a single ffmpeg command. You'd probably have to write a script that uses e.g. ffprobe to find the duration, then calculates the bitrate to give to ffmpeg. [01:36:31 CEST] <pzich> what's the end goal? there's a lot more than duration to factor in to calculating a good bitrate [01:37:14 CEST] <kyleogrg> hmm, just something i've wondered [01:37:51 CEST] <kepstin> yeah, the *only* reason to use duration as a factor is if you have a fixed max filesize, for example for a video upload site or physical media. [01:38:03 CEST] <BenMcLean> llogan thank you SO MUCH for recommending Shotcut. It seems to do absolutely everything I want real fast and intuitive with no nonsense for FREE [01:38:22 CEST] <llogan> good to hear, but thank Dan Dennedy, the author. [01:39:00 CEST] <kyleogrg> okay [01:39:04 CEST] <BenMcLean> I'm not sure how well it will handle larger scale projects though, when I have to edit half an hour or more of footage from many sources. Will also need to figure out how to slice up clips out of DVDs at some point [01:39:09 CEST] <kyleogrg> thanks for the help [01:39:55 CEST] <BenMcLean> my little brothers were bugging me a couple weeks ago about what program to use that's free and I felt bad I couldn't recommend anything but now I know! :D [01:40:05 CEST] <furq> kyleogrg: don't use -b:v in general unless you have a hard filesize constraint [01:40:20 CEST] <furq> with x264, that is [01:41:06 CEST] <kyleogrg> yeah. i usually use crf [01:51:23 CEST] <DHE> you'd also want 2-pass to ensure good quality and that your goal is actually met [01:59:20 CEST] <BenMcLean> Well, all my problems are totally solved. Thanks everybody! :D [02:00:35 CEST] <emitchell> Hey all, got a question. Trying to decide how to split a live video stream into files based off changes on an API and I am not sure if splitting the video live will cause issues with a) the incoming stream that we are recording and b) the resulting split video file. [02:03:09 CEST] <thebombzen> emitchell: "based off changes on an API" is really vague [02:03:31 CEST] <thebombzen> by if you want to double the video stream, you could try mapping it twice. i.e. -map 0:v -map 0:v [02:03:50 CEST] <thebombzen> if you want to duplicate it to two files just provide two outputs [02:04:47 CEST] <thebombzen> you might be looking for ffmpeg <input options> -i live_stream -c copy stream_dump1.mkv -c copy stream_dump2.mkv [02:05:40 CEST] <emitchell> so we are polling an API looking for an end time, and would like to cut the stream into a file (this is for slicing up a sporting event by matches) [02:05:58 CEST] <Bermond> Congratulations for the nice code of coduct that have just been added to the ffmpeg project documentation. Wise words. http://git.videolan.org/?p=ffmpeg.git;a=blobdiff;f=doc/developer.texi;h=4d3… [02:06:27 CEST] <emitchell> but ill look into the double mapping [02:08:08 CEST] <emitchell> thanks! [02:12:12 CEST] <emitchell> to clarify, we have a sporting event that streams for 10+ hours a day. in each stream is multiple 2m45s matches. there's an api that tells us when each match starts and ends, and we're able to map that to timestamps within the video. we'd like to split that match out of the recording that we're saving via ffmpeg and then upload it to youtube, the only issue is that the file is being written to since the stream is still happening when we're trying to sp [02:32:46 CEST] <SeanM_> Wondering if anyone could help me with this - I found this command (start from end of file, in this case 5 seconds from the end): ffmpeg -sseof -5 -i Input.mp4 that works but I want to cut/remove everything from that point on. Any ideas how to do this? [03:12:21 CEST] <thebombzen> emitchell: if you're on linux, you can use a named pipe [03:12:26 CEST] <thebombzen> or *nix [03:12:47 CEST] <thebombzen> instead of writing to a file, run mkfifo <filename> to make a named pipe [03:13:12 CEST] <thebombzen> then, you can write to the pipe by opening it for output. any program that opens it for input will see what you write [03:13:37 CEST] <thebombzen> so, say you have a stream grabber. you can have that streamgrabber write the output to mypipefile [03:13:44 CEST] <thebombzen> and then run ffmpeg -i mypipefile [03:14:02 CEST] <thebombzen> using ffmpeg, you save it to the actual file on the drive, and have the split go whereever you want. [03:14:44 CEST] <thebombzen> alternatively, you can have your command write the streaming file to Standard Out, and pipe the output to FFmpeg, with ffmpeg -i - to read from standard in. [03:15:02 CEST] <thebombzen> that second option works on all platforms. dunno if mkfifo works on a mac. def not on Windows tho. [03:16:49 CEST] <thebombzen> by that second one, I mean instead of something like "videostreamlistener filename_to_save_to" try "videostreamlistener - | ffmpeg -i - <other stuff>" [03:17:03 CEST] <thebombzen> and have it interpret - to be stdout or stdin, depending on context. [03:17:51 CEST] <thebombzen> where you'd do "videostreamlistener - | ffmpeg -i - -c copy stream_dump.mkv -c copy to_upload_to_youtube.mkv" [03:18:03 CEST] <thebombzen> if you get my drift [03:18:41 CEST] <thebombzen> SeanM_: not sure what you're asking [03:18:56 CEST] <thebombzen> because there's nothing after the end of the file, would could you remove everything from that point on? [03:19:06 CEST] <thebombzen> If you're trying to truncate a file, the -t option does that. [03:20:02 CEST] <SeanM_> Basically I am trying to remove the last five seconds (outro) from a file. And I thought that's the best way to jump to the exact point, but I want to cut everything from that point on. [03:20:02 CEST] <thebombzen> that is, ffmpeg -i input_file -t 00:06:10.5 <codec options> output_file will truncate the file to 6 minutes and 10.5 seconds in length. i.e. after it has encoded that much time it will end. [03:21:21 CEST] <SeanM_> That is helpful but is there a way to subtract the last five seconds only, without providing a specific time stamp like that? [03:25:08 CEST] <thebombzen> SeanM_: I don't think there's a way to do it easily. [03:27:44 CEST] <SeanM_> Thanks bombzen, that's too bad, I've been searching for like a week on how to do this and can't find anything easy. [03:28:33 CEST] <SeanM_> I basically want to do the complete opposite of the -sseof command, to always end it five seconds from the end of file rather than starting it there... figured there would be a way to do that easily but it seems not. [03:28:56 CEST] <thebombzen> SeanM_: you can use BASH to do it though [03:28:58 CEST] <thebombzen> printf '%s - 5\n' "$(ffprobe decisive_battle.mkv -show_format -of flat 2>/dev/null | grep format.duration | sed 's/format.duration=//')" | tr -d '"' | bc [03:29:21 CEST] <thebombzen> that was an example I used, but replace "5" at the beginning with the duration to crop and decisive_battle with the inputfile [03:29:45 CEST] <thebombzen> so you could put this in get_cropped_duration.sh: [03:29:47 CEST] <thebombzen> printf "%s - ${CROP_TIME}\n" "$(ffprobe ${INPUT} -show_format -of flat 2>/dev/null | grep format.duration | sed 's/format.duration=//')" | tr -d '"' | bc [03:30:12 CEST] <thebombzen> or $1 and $2 if you don't want to do any work [03:30:54 CEST] <SeanM_> Thanks very much, that helps a lot! [03:31:40 CEST] <thebombzen> then you could do ffmpeg -i input.mkv -t $(get_cropped_duration.sh 5 input.mkv) [03:31:55 CEST] <thebombzen> you're welcome :D [05:05:25 CEST] <drazin> so i'm running a script that uses ffmpeg to convert some videos and there seems to be a bug where its selecting both audiotracks as default when there are more than one as described here -- https://github.com/mdhiggins/sickbeard_mp4_automator/issues/360 [05:05:32 CEST] <drazin> they claim its a FFMPEG bug [05:05:37 CEST] <drazin> anyone know anything about this? [07:43:03 CEST] <thebombzen> drazin: it's not a bu [07:43:07 CEST] <thebombzen> bug* [07:43:20 CEST] <thebombzen> drazin: you can do ffmpeg -map [07:44:34 CEST] <thebombzen> suppose you do ffmpeg -i input and you see Stream 0:0, video. Stream 0:1, audio, 5.1. Stream 0:2, audio, stereo. You can do ffmpeg -i input -map 0:0 -map 0:1 to deselect the second audio stream [07:44:47 CEST] <thebombzen> (when you transcode it that is) [07:45:23 CEST] <thebombzen> if your audio player isn't automatically selecting one of them then it's poorly written, as this is a common practice (5.1 audio and stereo audio in the same container) [07:45:41 CEST] <thebombzen> either get a better player like MPV or use -map and copy it to a new one. [08:05:52 CEST] <odigem> hi [08:21:49 CEST] <Micke__> Hi! I'm trying to use ffserver and ffmpeg to transcode a video stream to a different format. Things start up the way I wan't them to but the video served from ffserver is only a static image. Does anyone has any clues to what I might have configure wrongly? [09:36:06 CEST] <coolandsmartrr> Can anyone help me with this issue? http://superuser.com/q/1081011/598139 [09:47:33 CEST] <yagiza> Hello! [09:56:34 CEST] <c_14> coolandsmartrr: use -preset, not -vpre [09:57:07 CEST] <coolandsmartrr> @c_14: give me this: [09:57:08 CEST] <coolandsmartrr> Unrecognized option 'preset'. [09:57:08 CEST] <coolandsmartrr> Error splitting the argument list: Option not found [09:57:24 CEST] <c_14> Does your ffmpeg build have libx264 support? [09:57:33 CEST] <coolandsmartrr> I think so, how do I check? [09:58:11 CEST] <c_14> ffmpeg -encoders | grep h264 [09:58:27 CEST] <c_14> Or check in the configuration line printed by the ffmpeg binary [09:59:49 CEST] <coolandsmartrr> Gives me the configuration, but I dont see h264 [10:00:19 CEST] <c_14> Then your build doesn't have support for libx264 [10:01:16 CEST] <coolandsmartrr> okay, then are there recommended configurations to encode videos superfast? [10:03:52 CEST] <c_14> If you're talking speed/quality/filesize nothing's really better than x264 [10:04:39 CEST] <coolandsmartrr> okay, so to get it, I recompile? [10:04:39 CEST] <yagiza> Can any1 tell me, what to do with this error message: [10:04:39 CEST] <yagiza> Custom AVIOContext makes no sense and will be ignored with AVFMT_NOFILE format. [10:05:41 CEST] <c_14> coolandsmartrr: yes, or use a static build or something from your distro (if you have a decent distro) [10:06:04 CEST] <coolandsmartrr> Im using Sun OS, not sure if there is a static build out there [10:06:51 CEST] <c_14> There might be, but none I know of. You'll have to compile yourself in that case. [10:07:08 CEST] <phreezie> Hi, I'm trying to convert about 2000 jpeg images to a video stream with one frame for each jpeg. The jpeg images are generated in realtime and piped into ffmpeg. The problem is that only about 250 of the 2000 frames end up in the video. Is it possible that ffmpeg receives the images faster than it can process it and aborts after a while? The command line: ffmpeg -f image2pipe -r 25 -vcodec mjpeg -i pipe:0 -y -r 24 -vcodec libx264 - [10:07:42 CEST] <phreezie> When adding the -re input option, I get better results, but still only about 500 frames. [10:09:29 CEST] <c_14> don't use -r 25, use -framerate 25 [10:09:57 CEST] <phreezie> @c_14: For both input and output? [10:10:01 CEST] <c_14> just input [10:10:18 CEST] <phreezie> Alright, let me try that! [10:10:28 CEST] <c_14> that will still drop frames though [10:10:38 CEST] <c_14> should drop Y frame every 25 seconds [10:10:42 CEST] <c_14> s/Y/1/ [10:15:16 CEST] <c_14> eh [10:15:19 CEST] <c_14> 1 Frame per second [10:15:35 CEST] <phreezie> @c_14: So with "ffmpeg -f image2pipe -vcodec mjpeg -framerate 25 -i pipe:0 -y -r 25 -vcodec libx264 -f mp4 frames.mp4", I'm still piping in 2118 frames and getting only 417 in the video [10:15:58 CEST] <c_14> can you upload the complete console output to a pastebin service? [10:16:05 CEST] <phreezie> sure [10:21:07 CEST] <phreezie> @c_14 gimme a minute, I'm using a node wrapper for ffmpeg and i need to figure out the -framerate thing properly (didn't work before actually) [10:24:49 CEST] <phreezie> @c_14: There we go: http://pastebin.com/uEjGv9UG [10:27:11 CEST] <c_14> hmm, it doesn't look like it's dropping frames [10:27:51 CEST] <c_14> What if you remove the -r 25 output option? Have you tried setting -preset to fast/veryfast/ultrafast ? [10:28:00 CEST] <phreezie> "ffprobe -v error -count_frames -select_streams v:0 -show_entries stream=nb_read_frames -of default=nokey=1:noprint_wrappers=1 frames.mp4" gives 209 [10:29:31 CEST] <phreezie> ok lemme try the preset (without -r 25 I'm getting 202 frames in the video). [10:32:57 CEST] <phreezie> -preset ultrafast didn't change much (202 frames in video). [10:33:48 CEST] <phreezie> I actually think ffmpeg tries to keep up in the beginning, fails and stops encoding [10:38:12 CEST] <c_14> So I just created 2118 jpegs to test with and then ran cat *jpg | ffmpeg <arguments>. and I'm getting 2118 frames [10:39:24 CEST] <c_14> Maybe try that as well? To make sure it's not how you're piping them that's the issue [10:39:26 CEST] <phreezie> When using the -re option and delaying the frame generation for 30ms each frame, I actually got *more* frames in the video (around 2500). Looks like a synchronization issue? [10:40:50 CEST] <c_14> What about delaying the frames without using -re? [10:41:41 CEST] <phreezie> I thought about that as well. It's difficult to debug, I tried piping them to the file system instead of ffmpeg and the jpegs get generated as they should.. [10:42:03 CEST] <phreezie> That's the funny thing. That doesn't change anything. [10:42:19 CEST] <phreezie> only -re makes the video longer when delaying [10:42:40 CEST] <c_14> try updating your copy of ffmpeg? [10:43:07 CEST] <phreezie> You mean to the latest version? [10:43:48 CEST] <c_14> to a more recent git release, yes [10:44:36 CEST] <phreezie> Yeah I've tried that too yesterday on my home machine. But lemme be sure I'm up to date here as well. [10:48:05 CEST] <coolandsmartrr> @c_14: I didnt have permissions to gmake install for x264, so I placed it in ~/bin. When I ./configure enable-libx264, it tells me Error: libx264 not found. Is there a way to tell configure where I have x264 installed? [10:48:44 CEST] <phreezie> ok, running on version N-80097-g89e9393 now [10:49:15 CEST] <c_14> coolandsmartrr: --extra-ldflags=-L/path/to/libraries --extra-cflags=-I/path/to/header/files [10:49:50 CEST] <c_14> coolandsmartrr: though it might be easier to change the libx264 preset and then export PKG_CONFIG_PATH to the lib/pkgconfig path [10:52:50 CEST] <coolandsmartrr> c_14: Sorry, dont understand what a libx264 preset is? [10:53:06 CEST] <c_14> prefix [10:53:08 CEST] <c_14> not preset [10:53:33 CEST] <c_14> multitasking is bad [10:53:43 CEST] <phreezie> c_14: I've piped all files to the file system and ran: "cat frame*.jpg | ffmpeg -f image2pipe -vcodec mjpeg -i - -vcodec libx264 -f mp4 frames.mp4". That worked. But it sucks as solution :) [10:54:49 CEST] <c_14> Maybe there's something strange about how you're writing to the pipe? Maybe an eof sneeks in there somewhere? [10:54:50 CEST] <phreezie> So I guess something with my pipe stream is borked.. [10:56:14 CEST] <phreezie> I'm using a PassThrough in NodeJS where I pipe the data to and that goes into the ffmpeg wrapper I'm using (fluent-ffmpeg). [10:56:33 CEST] <phreezie> *PassThrough stream [10:58:55 CEST] <c_14> Have you tried manually executing the ffmpeg binary? [11:00:52 CEST] <c_14> (ie instead of using fluent-ffmpeg) [11:05:18 CEST] <yagiza> Is there any way to decode a UDP stream with RTP data using FFMpeg? [11:05:57 CEST] <c_14> ffmpeg -i rtp:// ? [11:08:29 CEST] <yagiza> c_14, I mean using existing RTP stream, not establishing a new connection. [11:09:19 CEST] <c_14> You mean, you have RTP data and want ffmpeg to decode it? [11:09:32 CEST] <yagiza> c_14, yes [11:10:08 CEST] <phreezie> @c_14: Only with the cat *.jpg as source stream. But I could try outputting the jpg generation stream to stdout and pipe it into ffmpeg. [11:11:04 CEST] <yagiza> c_14, I'm handling UDP connection myself. And I need to demux/decode it usinf FFMpeg libraries. [11:13:09 CEST] <c_14> maybe using custom io [11:15:46 CEST] <c_14> https://ffmpeg.org/doxygen/trunk/avio_reading_8c-example.html [11:15:48 CEST] <c_14> ^kinda like that [11:15:59 CEST] <c_14> You'll also have to implement write though [11:16:14 CEST] <c_14> at very least for rtcp [11:18:49 CEST] <yagiza> c_14, no [11:19:15 CEST] <c_14> hmm? [11:19:20 CEST] <yagiza> c_14, that thing doesn't work for RTP because of AVFMT_NOFILE flag. [11:27:11 CEST] <c_14> Ah, well [11:27:16 CEST] <c_14> In that case I don't think you can [11:28:11 CEST] <c_14> You could setup a local udp bridge, but can't think of anything besides that and doing rtp yourself. You could try asking on the libav-users mailing list [11:28:25 CEST] <c_14> *libav-user@ [11:29:47 CEST] <yagiza> c_14, ok, thanx [12:48:59 CEST] <spooooon> is there a quick and easy way to save or display an AVFrame? [12:51:32 CEST] <DHE> like the image within? there are rendering functions in the API (eg: using SDL) but keep in mind that AVFrame isn't necessarily in a user-friendly format. pixel format encoding are applicable, like yuv420p [12:52:02 CEST] <spooooon> yea... [12:52:04 CEST] <spooooon> that's my trouble [12:52:41 CEST] <spooooon> I'm converting and displaying the data myself, and it not quite what I am expecting [12:52:50 CEST] <spooooon> thought it would be nice to see if the input is correct [12:52:58 CEST] <spooooon> and in this case it is yuv420p [12:53:31 CEST] <spooooon> sorry I made a mistake, the conversion is being done with sws_scale [12:54:29 CEST] <DHE> nevertheless, there is still encoding involved. the software scaler is usually used to do encoding conversions anyway [12:55:06 CEST] <spooooon> yes, I'm using sws_scale to convert, but the output is not quite correct [12:55:25 CEST] <spooooon> at least my interpretation is [12:55:42 CEST] <spooooon> I'm almost certain it is a problem with my code, but it would be helpful to know if I could see the input is correct [12:55:54 CEST] <spooooon> because I have a lot of layers above and below ffmpeg api [12:56:37 CEST] <DHE> My own issue: while streaming live TV I got this error: [mpegts @ 0x27d0340] Invalid timestamps stream=0, pts=3007556839, dts=11597482422, size=59101 [12:57:08 CEST] <BtbN> so your TV-Provider sent invalid timestamps [12:57:26 CEST] <DHE> what's interesting is it happened exactly after a certain amount of time the program is running - about 26.5 hours - which is the timestamp wraparound time of mpegts [12:57:41 CEST] <DHE> and it's been happening on more than one channel [12:58:49 CEST] <DHE> could be the original source but I found that to be quite a coincidence [13:26:22 CEST] <spooooon> I was passing incorrect parameters to sws_scale [13:39:56 CEST] <termos> I'm getting "Schematron validation not successful DASH is not valid" when generating DASH with FFmpeg, is this a known problem? [13:43:23 CEST] <JEEB> termos: might want to actually post the validation errors on the trac issue tracker with a way to regenerate similar ones [13:43:36 CEST] <JEEB> then I can poke wbs about it and he can note if they're valid errors or not [13:43:53 CEST] <JEEB> (wbs is the movenc/dashenc maintainer) [13:49:51 CEST] <termos> I'll do that, seems like the problem is that the <Period> element in the MPD does not have an id attribute. <Period id="p0" start="PT0.0S"> works. [13:53:27 CEST] <JEEB> I would recommend taking a look at the DASH spec regarding things like that [15:41:14 CEST] <Leo__> hello [15:41:48 CEST] <Leo__> any one here??? [15:42:33 CEST] <__jack__> sure [15:43:29 CEST] <Leo__> can you help me? [15:43:38 CEST] <__jack__> maybe [15:44:04 CEST] <arehman> Hi, I have an RTMP stream currently running, and I just want to turn it to HLS [15:44:08 CEST] <arehman> any pointers? [15:44:09 CEST] <Leo__> you have a code for audio? [15:44:48 CEST] <__jack__> arehman: ffmpeg -i source output.m3u8, probably with more codec and/or mapping options [15:44:54 CEST] <__jack__> Leo__: huh ? [15:46:03 CEST] <DHE> arehman: are you targetting certain devices like android? HLS calls for specific codecs and if your source doesn't meet the criteria then you should either convert or test the media first on your own phone [15:48:05 CEST] <furq> what a nice young man [15:50:17 CEST] <__jack__> :) [15:50:57 CEST] <arehman> I currently have it setup with the Nginx RTMP module to stream [15:51:07 CEST] <arehman> and so i just entered that command [15:51:13 CEST] <furq> arehman: nginx-rtmp already supports hls output [15:51:36 CEST] <furq> https://github.com/arut/nginx-rtmp-module/wiki/Directives#hls [15:53:43 CEST] <BtbN> I'd recommend using ffmpeg for the hls output though, I had weird issues with hls made via nginx-rtmp. [15:54:00 CEST] <furq> really? [15:54:03 CEST] <furq> i've not noticed any [16:02:20 CEST] <arehman> essentially the streams working [16:02:38 CEST] <arehman> but when you access it via phone to the meu8, its a 40sec video rather than the livestream [16:02:44 CEST] <arehman> m3u8* [16:02:47 CEST] <arehman> heres the nginx.conf [16:02:49 CEST] <arehman> http://pastebin.com/rcuq70RA [16:18:13 CEST] <arehman> anyway, cheers guys, got it working [16:18:26 CEST] <arehman> big thanks to __jack__ [16:18:30 CEST] <arehman> have a good day [16:50:30 CEST] <Admin__> hey guys.. good day.. very strange problem.. i am running some ffmpeg scripts that capture and segment some videos ... it is a live stream.... why after about 24- 27 hours all of the scripts get killed on my ubuntu system ? then i restart and they again go for 24 hours.. then stop again .. sooooo ODDD!! [16:50:41 CEST] <Admin__> anyone have a clue what could be happening [16:51:38 CEST] <__jack__> Admin__: dmesg says something ? [16:51:42 CEST] <jkqxz> What do you mean by "get killed"? Do they run out of memory, say? [16:53:56 CEST] <thebombzen> usually on the commandline if a process end and it just says "killed" that means the kernel killed it because you rn out of memory [16:54:16 CEST] <thebombzen> wow remind me not to type in the morning. typos galore [16:54:40 CEST] <Admin__> nothing at all in dmesg [16:55:03 CEST] <DHE> Admin__: live stream meaning over the air ATSC receiver? [16:56:34 CEST] <Andross> Hey guys [16:56:39 CEST] <thebombzen> ohaider [16:57:08 CEST] <Andross> my c++ program is using so many external libraries it's now more a c program [16:57:19 CEST] <Andross> and i never learned c! [16:57:36 CEST] <Admin__> both OTA and SAT receiver.. the source doesn't matter [16:57:45 CEST] <Admin__> it seems to be regarless of the source [16:58:09 CEST] <DHE> which means it's an mpegts source. and mpegts has timestamps that loop around every ~26.5 hours. that sound right? [16:59:46 CEST] <Admin__> i don't think so [16:59:48 CEST] <kepstin> On a live tv broadcast, I'd expect them to actually loop at 24h, but could be either :) [17:00:04 CEST] <Admin__> you mean the loop is causing it to stop ? [17:00:24 CEST] <Admin__> yes 26.5 hours .. yes [17:00:35 CEST] <Admin__> you are sounding very accurate.. and are on to something right now [17:01:03 CEST] <DHE> I've done something running on an OTA receiver having things going south after 26.5 hours [17:01:12 CEST] <Admin__> right now do i get around it ? [17:01:13 CEST] <Admin__> genpts ? [17:01:25 CEST] <Admin__> how do i start off with my own timestamps on the stream that go on forever [17:01:29 CEST] <Admin__> as to avoid this [17:01:32 CEST] <DHE> dunno. if you solve it, let me know [17:02:45 CEST] <thebombzen> Admin__: I've never gotten -fflags +genpts to do something [17:03:03 CEST] <thebombzen> but you might [17:03:33 CEST] <Admin__> hum.. here is another brain twister [17:03:52 CEST] <Admin__> my source has Closed caption on it... the data is missing on the output after i transcode the video.... i did scopy [17:04:06 CEST] <Admin__> weird thing is.. a few weeks ago i had it working just fine.. .now no :( [17:04:12 CEST] <DHE> using what codec on the output? copy mode will work [17:04:35 CEST] <thebombzen> closed caption might be a non-subttles stream so using -c:s copy might not work. [17:05:00 CEST] <DHE> if it's OTA signals then it's a native metadata payload in the video stream [17:05:55 CEST] <thebombzen> what I would do is -map 0 -c copy -c:v videocodec -c:a audiocodec [17:06:21 CEST] <thebombzen> which will copy all streams with c copy and then override the video and audio streams [17:18:26 CEST] <Andross> im looking for an example program that gets a wav and decodes it [17:18:36 CEST] <Andross> the current example generates a test tone instead of using a source file [17:21:50 CEST] <vade> you want demuxing_decoding example [17:22:02 CEST] <vade> just stip the video stuff out [17:25:19 CEST] <vade> speaking of demuxing and decoding, im using porting my code from 3.0.2 to current GIT master so I can have the H264 HW accell encoder - and notice that accessing the stream->codec is deprecated, in lieu of codecpar - however setting things like sample format for a streams codec? How do I do this? codec par doesnt have those variables [17:25:56 CEST] <vade> oh. format. duh. I see [17:32:46 CEST] <yagiza> Reading about RTP URI scheme: https://ffmpeg.org/ffmpeg-protocols.html#rtp [17:33:01 CEST] <yagiza> What's the meaniung of UDP socket? [17:33:41 CEST] <yagiza> AFAIK, there's no such thigg as "socket" in UDP. [17:33:58 CEST] <yagiza> How can I connect to it? [17:44:05 CEST] <Admin__> hum.. anyone know a good way to take the timestamp from a source mpegts and then creating new timestamps so if the source restarts timestamps it doesn't matter.. the encoder will keep doing and the timestamps will keep going too [17:45:43 CEST] <Andross> who wrote the examples in the source code? [17:56:02 CEST] <Andross> god im so lost [17:57:16 CEST] <Andross> all the good tutorials seem to be based on video demuxing [18:01:01 CEST] <Admin__> [mpegts @ 0x2aba300] Non-monotonous DTS in output stream 0:1; previous: 9169, current: 9000; changing to 9170. This may result in incorrect timestamps in the output file. .. how do i setup the ffmpeg so timestamp starts at 0 and then increments forever regardless of what the timestamp is [18:01:21 CEST] <Admin__> basically just want to sync the audio/video and then not change the timestap... these are live streams [18:02:51 CEST] <vade> ok so - with the new FF_API_LAVF_AVCTX API - how does one actually OPEN A CODEC a stream opens without it being a deprecated call? [18:07:49 CEST] <Andross> can someone tell me what exactly a 'frame' is in the context of audio? [18:11:19 CEST] <DHE> Andross: just a sequence of audio samples, usually sized to meet codec requirements [18:11:53 CEST] <DHE> which is why there's a FIFO implementation designed specifically for audio samples. that way you can deal with mismatches between codecs [18:15:27 CEST] <vade> yea audio has been slightly more nuanced than video in my experience [18:15:45 CEST] <vade> DHE: are you familiar with the new FF_API_LAVF_AVCTX API? [18:15:47 CEST] <Andross> so annoyed, was getting along so well working through the encoding_decoding example until the author decided to encode a test tone, which nobody would ever use, rather than an input file [18:15:57 CEST] <Andross> now im finding it a nightmare trying to figure out how to get ffmpeg to read an input file [18:16:23 CEST] <vade> you want to av_fint_input_format [18:16:40 CEST] <vade> avformat_alloc_context for an AVFormatContext [18:16:49 CEST] <vade> call avformat_open_input [18:16:57 CEST] <vade> probably avformat_find_stream_info [18:17:05 CEST] <vade> you can then call av_find_best_stream [18:17:20 CEST] <DHE> vade: you mean the new codecpar fields? [18:17:24 CEST] <vade> it will tell you a stream and codec for a AVMEDIA_TYPE_AUDIO [18:17:28 CEST] <Andross> an exmaple would be very helpful [18:17:42 CEST] <vade> yea DHE :) im migrating my code to codecpar and getting a lot of errors [18:17:49 CEST] <vade> use github and search [18:17:52 CEST] <vade> it was helpful for me [18:18:13 CEST] <vade> there is no good example thats fairl modern. ive considered writing some because its kind of a freaking nightmare. [18:18:52 CEST] <DHE> I found the examples, while a bit minimalistic and not covering all scenarios, to be good enough [18:19:19 CEST] <vade> depends on your level and understanding of AV in general [18:19:28 CEST] <vade> not everyone knows about muxers / demuxers and sample sizes :) [18:19:38 CEST] <vade> but fair enough, it wasnt horrible. I got stuff working hehe :) [18:19:55 CEST] <DHE> the audio re-encoder example shows how to deal with that using the av_fifio_* interface [18:22:01 CEST] <Andross> it also doesnt help that i dont know c much, only c++ [18:23:04 CEST] <vade> with codecpar, since a streams codec is now deprecated, how do I set a specific codec context ive set up to be associated with a stream? do I use av_format_set_video_codec ? that takes a format, not a format / stream ID though [18:23:20 CEST] <vade> i guess im confused how I know what stream I set / access a codec from if -> codec is deprecated [18:28:00 CEST] <DHE> yeah I'm a bit confused myself. I've got a local copy of the doxy docs from the git repo for this exact reason [18:28:12 CEST] <vade> yea, this seems weird. [18:28:28 CEST] <vade> because avcodec_send_packet requires a codecContext but I cant get one from the stream [18:28:33 CEST] <vade> &. wat? [18:35:42 CEST] <f00bar80> Please I need a pointer to a newbie guide for mpegts to m3u8 transcoding [18:39:04 CEST] <Andross> i cannot fathom why the person that wrote the examples didnt just make a generic function that accepts an input file and output file, so that anyone could use that function to encode audio [18:42:32 CEST] <Andross> i think it might be better to use the binary and commandline [18:43:01 CEST] <c_14> f00bar80: ffmpeg -i mpegts out.m3u8 ? Not sure what you're asking for [18:43:22 CEST] <vade> aha I see [18:43:30 CEST] <vade> you make your own codec context and initializ with avcodec_parameters_to_context [18:43:47 CEST] <vade> so stream codecparam -> your own codec context via -> avcodec_parameters_to_context [18:45:42 CEST] <vade> https://wiki.libav.org/Migration/12 from libav seems reasonably helpful [18:48:07 CEST] <Admin__> hey guys.. check this out [18:48:08 CEST] <Admin__> http://pastebin.com/raw/QYPRh7St [18:48:25 CEST] <Admin__> as you can see hte input has closed caption.. the output doesnt.. and the copy is set... [18:48:43 CEST] <Admin__> ${ffmpeg} -thread_queue_size 4096 -analyzeduration 5M -probesize 5M -i "$stream" $mapping -codec copy -copyts -copytb 1 -frame_drop_threshold 1.0 -dts_delta_threshold 0 -f mpegts - [18:48:52 CEST] <Admin__> that is the ffmpeg command ... what am i doing wrong here.. ? [18:50:27 CEST] <f00bar80> c_14: this is the basic transcoding, but i'm more into the stream controlling during the transcoding and which approache or a profile can be used .. regarding quality , bitrate , cpu consumption ..etc [18:51:12 CEST] <c_14> f00bar80: depends on what codec you use. I'm going to assume H.264 for now https://trac.ffmpeg.org/wiki/Encode/H.264 [18:53:31 CEST] <DHE> Admin__: did you actually watch the video? [18:53:42 CEST] <Admin__> i did.. no subtitbles [18:54:13 CEST] <Admin__> sorry.. closed caption [19:17:13 CEST] <Admin__> any ideas ? [20:01:57 CEST] <Andross> back [20:02:22 CEST] <Andross> so what are the potential issues if i choose to just distribute my program with binaries and feed it command line? [20:02:41 CEST] <Andross> i guess it's just the one binary [20:07:53 CEST] <kepstin> Andross: most of the stuff in https://ffmpeg.org/legal.html still applies, except that you can use a GPL version of FFmpeg instead of LGPL since it's not being linked into your code. [20:15:55 CEST] <Andross> well i already have an lgpl binary [20:16:07 CEST] <Andross> i looked at a couple of other programs and they seem to come with the binary too [20:16:30 CEST] <Andross> so i think it might be more common, at least for audio applications, to just use the binary and feed it command line [20:16:30 CEST] <_Vi> Shall "psnr" filter's output appear on console even with "-v warning"? [20:17:03 CEST] <f00bar80> is there any guide on cpu consumption optimization when H.264 encoding is used ? [20:17:41 CEST] <furq> what do you mean by optimisation [20:18:14 CEST] <furq> the defaults for x264 already do a pretty good job of using as much cpu as possible [20:18:27 CEST] <kepstin> f00bar80: make sure you're on 64-bit x86 with a modern processor, that x264 was built with assembly, and then use the slowest -preset value that's fast enough for you. [20:19:03 CEST] <furq> does 64-bit really make a difference [20:19:49 CEST] <kepstin> has double the registers from 32bit, I suspect the difference is noticable in x264 [20:20:12 CEST] <kepstin> (although I suppose the avx stuff is the same between both?) [20:20:14 CEST] <furq> i should probably benchmark it [20:20:22 CEST] <furq> i'm stuck using a 32-bit ffmpeg for avisynth input [20:22:49 CEST] <f00bar80> how to disable/enable the subtitles and where can I find a clue on the optimum level for audio tracks, EPG info , video subtitles ? [20:23:17 CEST] <furq> -c:s copy to enable subs, -sn to disable them [20:23:27 CEST] <furq> i don't know what "optimum level" means in this context [20:25:07 CEST] <f00bar80> furq: How to be able to choose from keeping EPG and some/all audio tracks , some of the subtitles ..etc [20:25:13 CEST] <furq> -map [20:25:21 CEST] <furq> https://www.ffmpeg.org/ffmpeg.html#Advanced-options [20:25:51 CEST] <relaxed> https://trac.ffmpeg.org/wiki/Map [20:29:05 CEST] <f00bar80> As I understand -map allows selecting which streams from which inputs will go into which output, this is by any mean answer my question ? [20:37:57 CEST] <f00bar80> furq: so how to check all the input file available audio, EPG, subtitles streams, in order to be able to map the required streams ? [20:39:12 CEST] <furq> ffprobe [20:46:44 CEST] <kyleogrg> hello [20:47:24 CEST] <f00bar80> furq: this is output of ffprobe for one stream http://pastebin.com/L2dhnA0d , correct me if I'm wrong , there's no subtitles here , right ? as well can you point me on how to identify how many audio and video streams are in here , and if I'm totally wrong , point me plz to some clrafication resources [20:47:37 CEST] <kyleogrg> I ripped a bluray recording to mkv. now I'm trying to use ffmpeg to put it into a m2ts container so sony vegas can open it. so far, only the video will show up in vegas. [20:48:11 CEST] <kyleogrg> command line: ffmpeg.exe -y -i "H:\title02.mkv" -c:v copy -c:a copy "C:\Users\me\Desktop\title02.m2ts" [20:49:43 CEST] <c_14> What audio codec? <- kyleogrg [20:49:48 CEST] <kyleogrg> This actually outputs a video-only file. [20:49:49 CEST] <kyleogrg> PCM [20:50:12 CEST] <c_14> kyleogrg: ffprobe title02.m2ts <- shows only the video stream? [20:50:24 CEST] <kyleogrg> according to mediainfo [20:50:44 CEST] <c_14> kyleogrg: 0:0 is video 0:1 is audio, there are no other streams listed. You can tell because it says Video and Audio after the stream identifier [20:51:13 CEST] <c_14> ^of the ffmpeg.exe -y -i command [20:54:42 CEST] <kyleogrg> http://pastebin.com/pXpNDcQj [20:55:42 CEST] <f00bar80> ppl any comment ? [20:56:33 CEST] <c_14> f00bar80: eh, the comment directly after kyleogrg said "according to mediainfo" was aimed at you. I just highlighted the wrong person [20:57:18 CEST] <c_14> kyleogrg: the output file should have audio&video. Can you check with ffprobe/ffmpeg -i instead of with mediainfo? [20:58:28 CEST] <kyleogrg> FFprobe output: http://pastebin.com/avDeJwtJ [20:58:51 CEST] <c_14> aaah > Stream #0:1[0x101]: Data: bin_data ([6][0][0][0] / 0x0006) [20:59:10 CEST] <kyleogrg> What does that mean... [20:59:40 CEST] <c_14> headers weren't muxed correctly probably, can you update your version of ffmpeg, it's rather old [21:00:12 CEST] <kyleogrg> hmm, okay, i'll quickly download a zeranoe ffmpeg [21:00:19 CEST] <Andross> by the way is ffmpeg's native aac encoder perfectly fine? [21:00:34 CEST] <furq> it's better than all the other open-source encoders except fdk-aac [21:00:45 CEST] <c_14> Assuming at least FFmpeg 3.0 [21:00:49 CEST] <furq> yeah [21:00:53 CEST] <c_14> Though it was fine before [21:01:03 CEST] <c_14> It didn't destroy anything most of the time [21:01:09 CEST] <c_14> And it probably won't kill your cat [21:01:14 CEST] <furq> Andross: you've not really got any other choice if you want to distribute binaries [21:01:23 CEST] <furq> fdk and faac aren't gpl compatible [21:01:43 CEST] <furq> and lame is lgpl [21:01:43 CEST] <Andross> what about lgpl though? [21:01:48 CEST] <furq> same thing [21:01:58 CEST] <Andross> so fdk isnt even lgpl compatible? [21:02:01 CEST] <furq> no [21:02:18 CEST] <Andross> what is the difference in quality, is it noticeable? [21:02:26 CEST] <furq> try it and find out [21:02:39 CEST] <furq> there hasn't been a listening test done with a large enough sample size to draw any decent conclusions [21:02:46 CEST] <kyleogrg> c_14: repeated the ffmpeg mux and ffprobe check, and I get the same ffprobe message you pasted [21:02:48 CEST] <furq> i think the HA guys were planning on doing one [21:03:13 CEST] <furq> at 128kbps you'll struggle to tell the difference between any modern-ish audio encoder really [21:03:28 CEST] <Andross> what about 192 or 320 [21:03:58 CEST] <kepstin> at 192+, you will have difficulty telling the difference between old formats like mp3 and modern codecs [21:05:12 CEST] <c_14> kyleogrg: It's definitely a bug, at the very least ffmpeg should complain on muxing and state it's not supported [21:05:26 CEST] <c_14> the same thing happens if you run `ffmpeg -f lavfi -i sine=1000 -c:a pcm_s16le out.m2ts' [21:05:41 CEST] <c_14> probably missing a tag or something [21:05:58 CEST] <kyleogrg> okay... [21:06:08 CEST] <f00bar80> what does the following ([2][0][0][0] / 0x0002), yuv420p(tv) refer to [21:06:22 CEST] <kyleogrg> I wouldn't know [21:07:02 CEST] <kyleogrg> do you have a suggestion as to what tags to try? [21:07:46 CEST] <c_14> Try getting your hands on an MPEG TS specification... [21:07:51 CEST] <c_14> Just convert the pcm to flac or something [21:08:16 CEST] <kyleogrg> would flac be supported by sony vegas? [21:08:22 CEST] <c_14> f00bar80: yuv420p is the pixel format (tv) states that it's limited range, the ([2] stuff means something but I forget what [21:08:29 CEST] <furq> or just decrypt the ts from the blu-ray [21:08:36 CEST] <kyleogrg> furq: how so? [21:08:40 CEST] <furq> i say "just" as if i know how to do that [21:09:16 CEST] <furq> i assume there are tools which just decrypt the m2ts files on the disc instead of remuxing to mkv [21:09:21 CEST] <kyleogrg> The end result I need is to simply take the Bluray MKV (from MakeMKV) and put it in some kind of container that Sony Vegas supports. [21:09:31 CEST] <c_14> kyleogrg: I have no idea what codecs sony vegas supports [21:09:44 CEST] <kepstin> makemkv supports simply copying the decrypted ts files rather than remuxing to mkv. Look for the backup mode. [21:09:44 CEST] <kyleogrg> yeah [21:09:47 CEST] <c_14> Knowing proprietary video editing solutions, probably not much and nothing well [21:10:31 CEST] <furq> i think it uses directshow [21:11:42 CEST] <furq> but yeah just do what kepstin said [21:12:11 CEST] <kyleogrg> kepstin: would this be any different from extracting the ts file from the mkv now [21:13:36 CEST] <furq> well one would hope it'd have the right audio header [21:14:19 CEST] <kyleogrg> okay [21:15:04 CEST] <f00bar80> anything in the paste ... identify a multiple number of audio tracks ? [21:15:32 CEST] <furq> no [21:18:32 CEST] <f00bar80> furq: if any how i can identify it? [21:19:29 CEST] <furq> http://vpaste.net/Ee9JI [21:21:11 CEST] <linux_aficionado> How would I set format options in an ffserver.conf file? [21:28:33 CEST] <kyleogrg> I found a solution copying the video codec and encoding the audio to aac, and putting it into an mp4 container [21:29:08 CEST] <furq> that works but you'll get generation loss when you export from vegas [21:29:39 CEST] <kyleogrg> yes, for the audio. but i'm doing very high quality. [21:36:33 CEST] <kyleogrg> okay, thanks for the help everyone [21:36:37 CEST] <kyleogrg> bye [21:39:31 CEST] <Andross> so [21:39:48 CEST] <Andross> if im just going to use the binary instead [21:39:59 CEST] <Andross> it's better i use a static build of the binary right? [21:43:25 CEST] <kepstin> probably, yeah. simpler to distribute then. zeranoe distributes a static binaries build, for example. [21:52:42 CEST] <linux_aficionado> In ffserver, how are format context options set? i couldn't find anything in the documentation [21:58:14 CEST] <f00bar80> normally how long does it to h.264 encode a mpegts stream? [22:00:38 CEST] <kepstin> f00bar80: depends how long the video is, how fast your computer is, and what encoder options you're using. [22:00:46 CEST] <kepstin> f00bar80: question basically can't be answered [22:03:11 CEST] <Andross> so ive noticed some programs, that come with ffmpeg.exe, somehow have a progress bar when encoding [22:03:25 CEST] <Andross> how is this possible? is it possible to get ffmpeg.exe to emit a progress signal? [22:03:27 CEST] <furq> they're probably parsing the output [22:03:48 CEST] <furq> i've done the same thing in the past with moderate success [22:04:06 CEST] <furq> by which i mean the progressbar worked but the gui sometimes decided to segfault [22:04:15 CEST] <furq> i don't think that was ffmpeg's fault though [22:05:17 CEST] <Andross> additionally, im having a problem with my aac encode [22:05:26 CEST] <Andross> for some reason when i load it up in foobar [22:05:33 CEST] <Andross> i cant change its time [22:05:37 CEST] <Andross> skip ahead etc [22:05:46 CEST] <f00bar80> kepstin: these are the options http://pastebin.com/FWUMMaK4 [22:06:38 CEST] <kepstin> f00bar80: right, so you've run this on a fast machine, watched cpu usage and output fps, and seen if it was fast enough? [22:07:26 CEST] <furq> does -deinterlace even work any more [22:07:29 CEST] <furq> i thought it was long since deprecated [22:08:02 CEST] <kepstin> if it does work, I assume it just sticks some random deinterlacing filter into the video filter chain? :/ [22:08:07 CEST] <furq> probably [22:10:53 CEST] <kepstin> f00bar80: also, you're setting -threads 0? what is your intent with that? [22:11:23 CEST] <kepstin> (if you want it to use only 1 thread, use -threads 1; if you want it to use all cpu cores available, omit -threads) [22:13:26 CEST] <Admin__> hey guys .. can anyone point me in the right direction .... so i am encoding a live stream.. every 26.5 hours my PTS wrap occurs it seems.... this is related to my encoding not the source... is there some way to stop this from doing that... i don't want my stream to end [22:17:35 CEST] <kepstin> Admin__: PTS wrap at 26.5 hours is an inherent part of mpeg-ts, that's simply the max time it can hold (it was designed to hold 24 hours actually; the 26.5 was the closest they could get with the binary numbers used). [22:18:10 CEST] <Admin__> but its killing my stream somehow since i am capturing the live stream [22:18:10 CEST] <kepstin> players that are designed for playing continuous mpeg-ts should just handle the wrap and keep going... [22:18:22 CEST] <kepstin> but iirc, ffmpeg has some issues with it [22:18:26 CEST] <Admin__> after 26.5 the whole thing requires a restart :( [22:18:43 CEST] <Admin__> its like the wrap doesn't actually happen or someting.. [22:23:54 CEST] <Andross> so this other program, it seems to distribute a full GPL binary with it, is that legal? [22:24:25 CEST] <furq> as long as the program is released under a gpl-compatible licence and they distribute the sources then sure [22:24:41 CEST] <furq> the ffmpeg and library sources, that is [22:24:47 CEST] <furq> obviously they need to distribute their own source code [22:25:41 CEST] <furq> actually i forget whether that constitutes a derivative work. licensing is boring [22:25:51 CEST] <furq> they definitely need to distribute the ffmpeg sources though [22:27:23 CEST] <Andross> here is said program furq: http://www.mediahuman.com/audio-converter/ [22:29:04 CEST] <furq> yeah they're just linking to ffmpeg.org for the sources [22:29:06 CEST] <furq> that's a gpl violation [22:29:33 CEST] <furq> if you distribute binaries you have to distribute all the sources yourself [22:30:10 CEST] <Andross> okay but, do they also need to distribute source code to their own program? [22:30:16 CEST] <furq> i'm not entirely sure [22:30:23 CEST] <furq> they would if they were linking to the ffmpeg libs [22:31:08 CEST] <Andross> being able to use the static GPL build would be pretty great [22:31:42 CEST] <furq> i don't think there's any difference between an lgpl and a gpl build if you're just calling the binary [22:34:47 CEST] <furq> afaik if your program doesn't work without ffmpeg then it counts as a derivative work and it must be GPL licensed [22:35:10 CEST] <furq> you can still charge for it if you want to, but you have to distribute the source [22:36:09 CEST] <Andross> define "doesn't work" [22:36:33 CEST] <furq> you'd need to ask a lawyer to define that [22:37:57 CEST] <Andross> something wrong with my internet brb [22:42:51 CEST] <linux_aficionado> is it even possible to use hls with ffserver? [22:45:28 CEST] <Admin__> hey maybe my wrap around is an issue because i am doing +genpts [22:45:36 CEST] <Admin__> maybe if i don't do that it should take the PTS from the actual live stream no ? [22:45:41 CEST] <Admin__> could that be the cause ? [22:47:40 CEST] <ferdna> i erased /tmp/feed_cam0.ffm... now it complains its not found... how do i recreate this file? [22:47:47 CEST] <ferdna> isnt it automatically created? [22:58:18 CEST] <vade> DHE: you around? Have you migrated to codecpar / send packet recieve frame yet? I just did, and while my decode works, encode seems wonky [23:06:36 CEST] <Andross_> alright im back [23:07:04 CEST] <Andross_> can someone explain the difference, if there is one, to "-ab" and "-b:a" [23:09:52 CEST] <furq> there isn't one [23:09:56 CEST] <furq> -ab is the old name, -b:a is the new one [23:10:18 CEST] <furq> -ab will presumably be removed at some point but it's unlikely as long as 99% of people copy their ffmpeg commands off stackoverflow [23:11:40 CEST] <Andross_> hehe [23:11:57 CEST] <Andross_> i just made a command line reader and am using it to read the commands sent by that program i linked earlier [23:12:25 CEST] <Andross_> i think the reason this program must be much faster than mine is because it uses libfdk_aac [23:12:46 CEST] <Andross_> (which im again guessing is illegal) [23:13:06 CEST] <furq> it sure is [23:14:06 CEST] <furq> all the "ultra magic super turbo xyz converter" freeware is a bit of a cesspool really [23:14:19 CEST] <furq> i wouldn't look to them for examples of what to do [23:14:47 CEST] <Andross_> well i find it easier than reading the documentation [00:00:00 CEST] --- Fri May 27 2016

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