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June 2018

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ffmpeg-devel.log.20180604
by burek 05 Jun '18

05 Jun '18

[01:06:04 CEST] <Compn> i'll announce mplayer dead when vlc doesnt take 2 minutes to load on my slow computer :P [01:06:17 CEST] <Compn> otherwise mplayer file still is fastest... [01:06:25 CEST] <Compn> i kno he gone :P [03:45:34 CEST] <Chloe> Compn: have you tried mpv? [08:54:26 CEST] <cone-800> ffmpeg 03Vishwanath Dixit 07master:37abfe8c2dd9: avfilter/drawtext: present 'hms' formatted 'pts' in 24h format [10:16:54 CEST] <jya> anyone could point me at some code that would generate a dummy SPS NAL (i only want to check if the resolution is supported by the VideoToolbox H264 decoder) [10:26:05 CEST] <kierank> jya: https://git.videolan.org/?p=x264.git;a=blob;f=encoder/set.c;h=39e5f07d1dc59… [10:26:51 CEST] <jya> kierank: thanks... [11:51:14 CEST] <jdarnley> I shocked! There's no existing pd_2 assembly constant. [11:52:55 CEST] <jdarnley> ah, most of them are words [12:37:21 CEST] <jdarnley> Does anyone know how I can make objdump show the rodata section and disassemble the text section at the same time? [12:37:33 CEST] <jdarnley> Without using -s? [12:41:19 CEST] <jkqxz> "objdump -d -j .rodata -j .text ..."? [12:43:24 CEST] <jdarnley> That disassembles rodata rather than just showing its contents [12:43:51 CEST] <jkqxz> "-d", not "-D". [12:44:18 CEST] <jdarnley> which is what I typed [12:44:30 CEST] <jdarnley> i know -D is --disassemble-all [12:44:51 CEST] <jkqxz> Or is it somehow failing at recognising which segments would contain code for your platform? It works for me on x86-64 Linux. [12:46:05 CEST] <jdarnley> Maybe but this is only Arch Linux so not *that* obscure. [12:48:32 CEST] <jkqxz> <https://0x0.st/s_q7.txt> [12:49:26 CEST] <jdarnley> Maybe I'll blame the objects prodiced by NASM. [12:49:31 CEST] <jdarnley> *produced [12:49:45 CEST] <jdarnley> Thanks anyway [12:53:17 CEST] <jkqxz> It probably depends on the flag in the section header. Can nasm set those? [12:54:11 CEST] <jkqxz> Look at -h output. My .text section from gcc has "CONTENTS, ALLOC, LOAD, READONLY, CODE", while .rodata has "CONTENTS, ALLOC, LOAD, READONLY, DATA". [12:56:25 CEST] <jdarnley> It does set those and .text gets RELOC too [12:57:08 CEST] <jdarnley> .rodata seems to comes before .text in the file [12:57:31 CEST] <jkqxz> My function was trivial enough that it didn't need any relocations! [13:16:14 CEST] <atomnuker> why do you need to look at the disassembly of an assembly file? [13:20:10 CEST] <iive> can't objdump help with that? [14:59:29 CEST] <durandal_1707> kierank: are you sure about av_filter_frame(AVFrame *in, AVFrame **out)? is caller supposed to allocate output frame or callee? [15:00:20 CEST] <kierank> if it's NULL allocate internally, if not let the user provide [15:00:23 CEST] <kierank> but I prefer user [15:00:31 CEST] <kierank> allocating the avframe but not the data planes [15:10:49 CEST] <durandal_1707> kierank: and how to handle cases when filter needs more input frames for 1 output frame, or can give more output frames per 1 input frame? [15:11:15 CEST] <kierank> EAGAIN [15:16:36 CEST] <jkqxz> For non-1:1 cases it might be better to make it avfilter_send_frame() / avfilter_receive_frame(). [15:16:50 CEST] <jkqxz> That would also let you feed input to multiple pads. [15:16:52 CEST] <durandal_1707> but how will it know which of those 2 cases is actually happening [15:18:07 CEST] <jkqxz> You need a bit of state around it anyway, so something there. [15:21:03 CEST] <durandal_1707> jkqxz: how would you signal with avfilter_send_frame() to which input is frame sent? [15:21:21 CEST] <jkqxz> Extra int argument? [15:34:42 CEST] <durandal_1707> what if user calls avfilter_send_frame() too much times? [15:36:04 CEST] <jkqxz> EAGAIN. [15:36:57 CEST] <JEEB> what was the flow with the dec/enc api flow? [15:37:04 CEST] <JEEB> https://www.ffmpeg.org/doxygen/trunk/group__lavc__encdec.html [15:37:17 CEST] <JEEB> also doesn't this one ignore feeding different slots? [15:37:21 CEST] <JEEB> or did I miss something [15:37:31 CEST] <JEEB> such as overlay etc [15:38:50 CEST] <durandal_1707> jkqxz: EAGAIN means give me more.. it already gave too much, and any extra input should be just ignored [15:41:02 CEST] <jkqxz> EAGAIN means "not until something else happens". In this case the something else is you calling receive_frame(). [15:42:49 CEST] <durandal_1707> im running in circles [15:43:45 CEST] <jkqxz> I guess it depends what other people want from this one-filter API. [15:44:13 CEST] <jkqxz> If the 1:1 case is the most interesting then the single call would be simpler and better. [15:56:38 CEST] <nevcairiel> i would use it, but i need 1:n since i want it for deint [15:56:46 CEST] <nevcairiel> so an API like avcodec might be suitable [16:06:42 CEST] <nevcairiel> but i basically only ever use a single filter, so all that boilerplate around it is really a lot of noise [16:09:13 CEST] <jkqxz> Were there any thoughts on how this should be initialised? Do you have some sort of AVFilterSingleContext holding it? [16:11:01 CEST] <jdarnley> I need to stop writing my assembly with do-while loops when the rest of the code assumes a for loop. [16:12:28 CEST] <jdarnley> or just get the condition check right [16:36:17 CEST] <cone-452> ffmpeg 03Jerome Borsboom 07master:f56a0b02cdbf: avcodec/vc1: fix out-of-bounds reference pixel replication [18:14:40 CEST] <jkqxz> rcombs: Do you have any more context for that Apollo Lake issue? I've tried a few variations on it (input vaapi/yuv420p/nv12/p010, scale_vaapi to nv12, encode with H.264/H.265) and not seen any problems. [18:39:28 CEST] <rcombs> jkqxz: oh I totally forgot to mention, input is 10-bit HEVC and output is H264 [18:41:13 CEST] <rcombs> user was able to repro with http://jell.yfish.us/media/jellyfish-3-mbps-hd-hevc-10bit.mkv [18:41:51 CEST] <rcombs> example command that repro'd for the user: [ffmpeg] -hwaccel vaapi -ss 17 -i jellyfish-3-mbps-hd-hevc-10bit.mkv '-vaapi_device' '/dev/dri/renderD128' '-filter_complex' '[0:0]hwupload,scale_vaapi=w=1278:h=538:format=nv12,hwupload[0]' -map '[0]' -t 3 '-codec:0' 'h264_vaapi' '-b:0' '2717k' jellyfish-out2.mp4 [18:42:44 CEST] <rcombs> this ffmpeg is a few months behind at this point, but I don't see any obviously-relevant differences in vaapi code [18:45:16 CEST] <rcombs> and libva+driver are 2.1.0, which is latest [18:45:21 CEST] <rcombs> (but not master) [18:46:23 CEST] <jkqxz> So that's doing a download-upload probably in p010? [18:46:51 CEST] <jkqxz> I don't see anything funny with that file. [19:12:45 CEST] <rcombs> jkqxz: the actual command has `-hwaccel_output_format:0 vaapi`; happens whether with or without that [19:22:57 CEST] <jkqxz> I tried both. [19:23:51 CEST] <rcombs> hmm [19:23:59 CEST] <jkqxz> I guess I should try with the 2.1 release version. [19:24:31 CEST] <rcombs> it's a J3455 [19:24:57 CEST] <rcombs> here's the full thread https://forums.plex.tv/discussion/comment/1665703#Comment_1665703 [19:25:14 CEST] <jkqxz> This is an N4200. [19:26:22 CEST] <rcombs> I dunno how much the graphics vary between them [19:27:04 CEST] <rcombs> "HD Graphics 505" vs "HD Graphics 500" sounds like they _should_ be similar [19:28:08 CEST] <jkqxz> It's 18 vs. 12 execution units and a bit of clock speed. [19:28:15 CEST] <jkqxz> The video stuff should be identical. [19:30:32 CEST] <rcombs> my latest merge is with 0419623cdc [19:33:38 CEST] <jkqxz> Urgh, got it. What the hell. [19:34:35 CEST] <jkqxz> I think I might have had it earlier, but it's totally invisible most of the time. You really do need the jellyfish halfway through with tentacles pointing upwards. [19:46:29 CEST] <jkqxz> <http://ixia.jkqxz.net/~mrt/ffmpeg/apl/in.mp4> -> <http://ixia.jkqxz.net/~mrt/ffmpeg/apl/out.mp4> [19:46:55 CEST] <jkqxz> It does require the H.265 Main 10 input. [19:47:07 CEST] <rcombs> ah [19:48:24 CEST] <jkqxz> Uh, I mean it requires the 10-bit input. [19:48:38 CEST] <jkqxz> Decoding in software and uploading does fail. [19:50:35 CEST] <jkqxz> It's something to do with the scaling used. If you scale to 1280x720 then it doesn't happen. [19:51:00 CEST] <rcombs> huuuuh [19:51:12 CEST] <rcombs> maybe non-mod4? [19:52:22 CEST] <jkqxz> Oh, that might just be passthrough or not. Seems to happen with any non-1280x720 value with that 1280x720 input. [19:54:00 CEST] <jkqxz> Must be a driver issue somehow. I'll see if I can reproduce it on any other system and then raise a bug against the driver. [19:57:31 CEST] <rcombs> cool, thanks [21:25:09 CEST] <gagandeep> kierank: finally, the blurring will be solved, the last inverse transforms that i needed to apply weren't spatial but horizontal-temporal [21:25:18 CEST] <gagandeep> rest of the data is working good [21:25:57 CEST] <kierank> gagandeep: which blurring? [21:27:46 CEST] <gagandeep> i mean, the last inverses were causing problem, so the image quality wasn't perfect [21:28:19 CEST] <durandal_1707> atomnuker: you owe me reviewe [21:28:20 CEST] <gagandeep> the inverse was different and i was not able to see it as the last process was done in threads so was not visible [21:28:47 CEST] <gagandeep> will read in wavelet file of cfhd and perform the final inverse acording to specs [21:29:04 CEST] <gagandeep> other decoding and dequantization is running good [21:30:10 CEST] <gagandeep> though, it is good we were able to spot other nuances in interlacing issue as they could have caused problems with other samples [21:34:33 CEST] <gagandeep> durandal_1707 and kierank: you both were right there is some mixing in 2 frames at the initial transform [21:35:02 CEST] <gagandeep> i assumed mistakingly it was just spatial as they had used temporal afterwards as well [21:41:07 CEST] <atomnuker> durandal_1707: for the nlmeans audio denoiser? [21:43:16 CEST] <durandal_1707> atomnuker: yes [21:44:43 CEST] <atomnuker> "Deprecation of OpenGL and OpenCL" - "Apps built using OpenGL and OpenCL will continue to run in macOS 10.14, but these legacy technologies are deprecated in macOS 10.14." [21:44:56 CEST] <atomnuker> welp, so much for opencl being supported everywhere [21:45:23 CEST] <nevcairiel> do they in the same sentence announce official Vulkan support, or are tehy just screwing everyone into using Metal? [21:45:53 CEST] <atomnuker> lol no, metal only [21:46:26 CEST] <nevcairiel> well if they actually remove OpenGL, I'll just petition work for dropping support entirely [21:49:02 CEST] <atomnuker> where? [21:57:48 CEST] <rcombs> "you know what macOS needed? its very own DirectX" [21:57:59 CEST] <rcombs> ETA on ANGLE Metal backend? [22:09:27 CEST] <iive> i think that there are project that implement d3d9 and d3d11 on vulkan, there is also project that implements vulkan on metal [22:10:14 CEST] <BtbN> ok, so on OSX there will be ANGLE on d3d11 on Metal soonß [22:10:43 CEST] <rcombs> there's a GL-on-Metal implementation, but khronos charges for it [22:14:19 CEST] <Chloe> there's a VK-on-Metal implementation too, which is free and open source [22:15:56 CEST] <rcombs> yeah, which is nice if you're already porting to vulkan [22:40:24 CEST] <cone-139> ffmpeg 03Vishwanath Dixit 07master:146cdf7e4bb0: fftools/ffmpeg: fix for all forced key frames when 'copyts' is enabled [00:00:00 CEST] --- Tue Jun 5 2018

1 0

ffmpeg.log.20180604
by burek 05 Jun '18

05 Jun '18

[00:58:01 CEST] <TheAMM> Actually turns out -async 1 doesn't always work [00:58:18 CEST] <TheAMM> Guess it's that delay in opening up the inputs [01:00:05 CEST] <JEEB> uhh, do you mean the async protocol or the option that now maps into the resampling filter that adds silence? [11:17:24 CEST] <dragmore88> hi, im trying to encode an AAC track with ffmpeg 4.0 from a prores source and i have verified that track 6&7 is the stereo tracks, but the output is silenced.. anyone ? ./ffmpeg -loglevel verbose -i got.mov -filter_complex "[0:a:6][0:a:7]amerge=inputs=2,pan=stereo[aout1]" -map "[aout1]" -acodec aac -b:a 128k result.m4a [11:19:13 CEST] <durandal_1707> dragmore88: pan needs more arguments... [11:19:59 CEST] <durandal_1707> iirc just giving output channel layout will give you silence [11:20:19 CEST] <dragmore88> thx, could u elaborate with an example ? [11:20:42 CEST] <dragmore88> when encoding ac3, i dont need a pan, but the aac encoder complains.. [11:21:53 CEST] <durandal_1707> examples cost extra coins [11:22:00 CEST] <dragmore88> hehe [11:22:08 CEST] <dragmore88> 1 BC pr example? [11:23:37 CEST] <dragmore88> from the track manual adding -ac 2 should be enough, as long as i have specified the input tracks [11:23:38 CEST] <dragmore88> ./ffmpeg -loglevel verbose -i got.mov -filter_complex "[0:a:6][0:a:7]amerge=inputs=2,pan=stereo|[aout1]" -map "[aout1]" -acodec aac -ac 2 -b:a 128k result.m4a [11:23:59 CEST] <dragmore88> still i get a 3Kbps bitrate.. [11:24:16 CEST] <durandal_1707> see pan filter examples here: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [11:24:28 CEST] <dragmore88> yupp.. the one im reffering to [11:24:43 CEST] <dragmore88> "Or use -ac 2 instead of the pan audio filter: [11:24:43 CEST] <dragmore88> ffmpeg -i input1.wav -i input2.wav -filter_complex "[0:a][1:a]amerge=inputs=2[aout]" -map "[aout]" -ac 2 output.mp3 [11:24:44 CEST] <dragmore88> " [11:24:53 CEST] <durandal_1707> pan=stereo|c0=c0+c2|c1=c1+c3 [11:25:21 CEST] <furq> amerge with two inputs should mix to stereo anyway [11:25:49 CEST] <dragmore88> ./ffmpeg -loglevel verbose -i got.mov -filter_complex "[0:a:6][0:a:7]amerge=inputs=2[aout1]" -map "[aout1]" -acodec aac -ac 2 -b:a 128k result.m4a [11:25:53 CEST] <durandal_1707> his tracks are stereo [11:26:00 CEST] <dragmore88> this one seem to work, i removed the pan and added -ac 2 [11:26:20 CEST] <furq> it'll be 8 mono tracks won't it [11:26:35 CEST] <dragmore88> my example is a prores with 2x mono tracks [11:26:41 CEST] <dragmore88> track 6 & 7 [11:26:45 CEST] <dragmore88> -> stereo [11:26:46 CEST] <furq> yeah you don't need to do anything other than amerge then [11:27:18 CEST] <dragmore88> my problem was adding pan to the mix with no |c0=c0 etc.. [11:43:39 CEST] <binarym> hi all. I use the command line here: https://pastebin.com/2A7TiD49 to generate a timestamped HLS stream from a webcam [11:44:19 CEST] <binarym> here is the playlist generated: https://pastebin.com/HbqiX0DY [11:44:39 CEST] <binarym> my problem is that i would like the playlist to wrap too. [11:45:09 CEST] <binarym> and the #EXT-X-MEDIA-SEQUENCE:0 to be incremeted [11:45:13 CEST] <binarym> is it possible with ffmpeg ? [11:46:32 CEST] <binarym> in the HLS stream, in my understanding, if i generate 4 chunks (out00{0,1,2,3}.ts) and the wrap [11:46:40 CEST] <binarym> s/the wrap/then wrap/ [11:47:05 CEST] <binarym> then, the playlist is supposed to "wrap" too: EXT-X-MEDIA-SEQUENCE get incremented [11:47:15 CEST] <binarym> and previous chunk are removed/overwrite ... [11:49:27 CEST] <binarym> ok ... digging into the doc, i discovered the "hls" muxer... let's try it [13:31:24 CEST] <zerodefect> Using the C-API to decode a stream of audio (st 337m) from a memory buffer. I have created a custom AVIOContext to read from the memory buffer - works fine. But... [13:33:20 CEST] <DHE> well don't leave us in suspense! [13:33:55 CEST] <zerodefect> If data from the source is still pending, does it suffice to return 'AVERROR(EAGAIN)'. The documentation for the 'read_packet' function does say, "For stream protocols, must never return 0 but rather a proper AVERROR code". I want to ensure I'm not bending things. [13:34:34 CEST] <zerodefect> Never knew my question were that exciting :) [13:34:40 CEST] <zerodefect> *questions [13:37:10 CEST] <DHE> there is an AVERROR_EOF and I assume the intention is you would use this [13:41:47 CEST] <c_14> Wouldn't you only use EOF if there is no more data? If there will be more data soon", EAGAIN should be the correct return code [13:42:04 CEST] <zerodefect> Thanks both. [13:42:18 CEST] <zerodefect> So if I call 'av_read_frame'... I only want to read one audio frame's worth of data, can I make the assumption that will happen? [13:43:47 CEST] <DHE> I assume you have what basically amounts to a file pre-loaded into RAM. the regular AVFormatContext will parse it as usual. you're just emulating a "file" in RAM [13:45:13 CEST] <zerodefect> @DHE. Fair assumption, but in this case, I'm receiving the data buffer through a network and then loading the payload into the memory buffer. [13:46:56 CEST] <DHE> ah. well in that case EAGAIN makes sense since more data will be available later [13:51:37 CEST] <zerodefect> Yes, those were my thoughts. I understand how EAGAIN works in a codec context - there is an input and output buffer and data is pushed and pulled. In the context of an AVIOContext does it make sense if I think a traditional file workflow? For all the IO reader knows, it is responsbile for pulling the data (there is no concept of push because all the data that will be input already exists - at [13:51:37 CEST] <zerodefect> least in a file workflow). My concern is if this extends to custom IO contexts. [13:51:55 CEST] <zerodefect> Maybe I'm making a mountain out of a mole hill and I just need to suck it up and try it and see what happens :) [14:10:33 CEST] <DHE> EAGAIN loosely means "try again later" [16:53:21 CEST] <hashworks> Hi! I have a video file with hardcoded subtitles. Is there a way to mark that? I'm able to set the language for the video stream, is that what it means? [16:53:35 CEST] <hashworks> hardcoded as in they are in the video stream [16:58:12 CEST] <Mavrik> hashworks: what do you mean mark that? [16:58:34 CEST] <Mavrik> Also the subtitles probably aren't hardcoded if you can disable them (hardcoded usually means they're baked into the video without option of removal) [17:00:51 CEST] <DHE> like, add metadata indicating that the video has a language associated with it? [17:00:59 CEST] <hashworks> Mavrik: Nope, I can't disable them [17:01:09 CEST] <hashworks> They are burned into the video stream [17:01:30 CEST] <hashworks> And I want to indicate that the video has subtitles that can't be adjusted [17:01:36 CEST] <Mavrik> Hmm. [17:02:48 CEST] <Mavrik> There might be a container flag but it's unlikely that's something standard. [17:02:58 CEST] <hashworks> Hm& [17:03:04 CEST] <hashworks> Whats the Video Stream Language Tag for? [17:03:16 CEST] <hashworks> For something like filmed text? [17:03:27 CEST] <Mavrik> Probably yeah [17:03:34 CEST] <Mavrik> I doubt anything uses it tho [17:13:58 CEST] <JEEB> hashworks: basically if you had multiple video tracks with language specific stuff [17:14:16 CEST] <JEEB> otherwise pretty much unused or randomly tagged for content language [17:34:43 CEST] <TAFB_WERK> in the unifi controller, if I want to see how much data went through an AP from friday until sunday, can I do that? Can't seem to see it other than the traffic graph, which doesn't total it up [17:38:02 CEST] <Mavrik> Wrong channel? :) [17:39:38 CEST] <TAFB_WERK> lol, whops :) [17:39:54 CEST] <TAFB_WERK> ./j ##ubnt [19:41:14 CEST] <Daegalus> Question, is there an alternative ffmpeg for windows page? the current linked on is down, and according to Web Archive, has been down for almost a month [19:42:12 CEST] <julesverne> sounds like you might be using a broken link. [19:42:44 CEST] <julesverne> nope. i'm wrong. [19:42:56 CEST] <julesverne> but.. definitely hasn't been down for a month. [19:43:17 CEST] <julesverne> i dloaded it a couple days ago from there. [19:45:26 CEST] <BtbN> well, expired certs do that [19:45:29 CEST] <Daegalus> ah ok, seems like just an Expired cert issue that the maintainer hasnt noticed [19:45:41 CEST] <Daegalus> or gotten around to taking care of [19:45:48 CEST] <julesverne> yep [20:31:19 CEST] <eltoozero> Any solid examples of drawtext text expansion floating around? [20:39:52 CEST] <DHE> -vf drawtext=expansion=normal:text="Frame number %{n}" # Renders a simple frame number in the top-left corner [20:39:55 CEST] <DHE> Something like that? [20:40:08 CEST] <DHE> shell escaping or quoting is up to you [21:14:17 CEST] <eltoozero> @DHE looking for a more complicated example, for instance, can I show text starting at a specific frame number? [21:16:49 CEST] <eltoozero> I found a one-liner exmaple, but it seems like I should be able to specify a textfile= containing a bunch of %{} statements. [00:00:00 CEST] --- Tue Jun 5 2018

1 0

ffmpeg-devel.log.20180603
by burek 04 Jun '18

04 Jun '18

[00:02:51 CEST] <TD-Linux> atomnuker, okay I haven't tried it recently. it seemed to be somewhat hw dependent too, on amd both ways were basically the same [00:03:04 CEST] <TD-Linux> I actually run webrender right now (which also damages the whole screen) [00:03:40 CEST] <atomnuker> I thought the whole point of webrender was to use the compositor's compositor interface and let it handle damage tracking and such [00:04:33 CEST] <atomnuker> so DirectComposition on windows, wl_compositor on linux [00:05:05 CEST] <TD-Linux> no, that's entirely separate (but also being worked on, I think it's actually used on windows) [00:05:32 CEST] <TD-Linux> webrender *could* do damage tracking, it just doesn't [00:10:36 CEST] <atomnuker> welp, everything is awful ¯\_(Ä)_/¯ [12:14:08 CEST] <cone-008> ffmpeg 03Paul B Mahol 07master:d0bf1aa3c5f7: avfilter/avf_showspectrum: improve axes drawing [12:45:25 CEST] <cone-008> ffmpeg 03Paul B Mahol 07master:9add1786ad4c: avfilter/avf_showspectrum: avoid overwritting text [12:45:26 CEST] <cone-008> ffmpeg 03Paul B Mahol 07master:49eda27c6e7a: avfilter/avf_showspectrum: also show sample rate and channel layout [13:00:27 CEST] <atomnuker> jkqxz: can you test vulkan->vaapi mapping if you've got the time? [15:00:01 CEST] <anill> can anyone help me on this https://stackoverflow.com/questions/50664967/convert-rtp-stream-into-h-264-… [15:01:13 CEST] <cone-008> ffmpeg 03Paul B Mahol 07master:983288538669: avfilter/vf_zoompan: do not increase VAR_IN twice, also count from 0 [18:56:03 CEST] <cone-108> ffmpeg 03Paul B Mahol 07master:29e0879b29d1: avfilter/f_drawgraph: fix drawing of first point for non-first metadata key [19:41:01 CEST] <cone-108> ffmpeg 03Mark Thompson 07master:2bd24d4a37e9: v4l2_m2m: Mark V4L2 M2M decoders as unsuitable for probing [21:05:22 CEST] <durandal_1707> Compn: when you will announce that MPlayer is officially dead? [21:48:48 CEST] <durandal_1707> atomnuker: when you gonna post new patches? [22:39:11 CEST] <kierank> durandal_1707: when will you announce libavfilter needs rewriting from scratch [22:43:50 CEST] <durandal_1707> kierank: what would you rewrite exactly? [22:44:00 CEST] <kierank> all of it [22:44:02 CEST] <kierank> it's useless [22:44:04 CEST] <kierank> it leaks memory [22:44:10 CEST] <kierank> it doesn't support any realtime process [22:44:18 CEST] <kierank> api is understandable to nobody [22:44:23 CEST] <kierank> framesync module who knows wtf that does [22:44:23 CEST] <kierank> etc [22:44:32 CEST] <kierank> it's just a big blob of wtf [22:47:56 CEST] <durandal_1707> kierank: where it is useless? where it leaks memory? where it does not support realtime process? [22:48:23 CEST] <kierank> it leaks memory all the time, if you have n streams coming in and one of them is disconnected it will just buffer frames indefinitely [22:48:25 CEST] <kierank> it's full of hacks [22:48:30 CEST] <JEEB> realtime requires the filtering framework thinks of time [22:48:39 CEST] <kierank> the framework just understands frames [22:48:41 CEST] <JEEB> because if you don't get input you can't just keep waiting [22:49:50 CEST] <durandal_1707> you are supposed to handle corner cases manually [22:50:00 CEST] <kierank> with the plethora of documentation sure [22:50:15 CEST] <kierank> the problem is there's no api for just letting your use the filters in your own filtergraph [22:50:19 CEST] <kierank> that's the biggest problem [22:50:26 CEST] <kierank> you have to buy in to all the libavfilter surrounding crap [22:50:31 CEST] <kierank> (i.e framesync) [22:50:55 CEST] <JEEB> yeh [22:51:19 CEST] <JEEB> if I just want to overlay stuff IFF there's stuff there's no way to do that with the filter stuff at the moment [22:51:44 CEST] <kierank> latency of pipeline is a big problem as well [22:51:52 CEST] <kierank> it's completely dynamic [22:52:08 CEST] <kierank> you don't know how many frames are being buffered internally [22:52:12 CEST] <JEEB> yeh [22:52:40 CEST] <JEEB> anyways I need to debug some filter chain stuff during the week because something herps a derp somewhere, which doesn't show up in file coding [22:52:57 CEST] <rcombs> also still no subtitles [22:53:04 CEST] <durandal_1707> filter can know how many frames are buffered if its really needed [22:53:13 CEST] <kierank> across the entire pipeline? [22:53:22 CEST] <kierank> and it won't ever by dynamic because of framesync? [22:53:34 CEST] <rcombs> also you need separate filters for software and hardware frames [22:53:46 CEST] <kierank> my loose understanding of nicolas undocumented masterpiece is that it dynamically adapts to frames arriving [22:54:52 CEST] <rcombs> I also run into some realtime issues with lavf/segment fwiw [22:55:12 CEST] <kierank> durandal_1707: I would like to see filter and filtergraph decoupled [22:55:16 CEST] <kierank> so I can just use filters like codecs [22:55:20 CEST] <kierank> load them, get frames out and that's it [22:55:40 CEST] <durandal_1707> number of frames attached to each fifo can easily be get from filter graphs [22:55:41 CEST] <JEEB> yea, it's a dynamic thing that tries to make sure things are in some sync [22:55:41 CEST] <rcombs> if I have 2 segment muxers running (one for A/V and one for subtitles, which may be sparse), there's no way for the subtitles one to know when the transcode has proceeded far enough to cut a new segment, until a new packet is sent [22:55:53 CEST] <kierank> durandal_1707: yes but's its dynamic [22:56:02 CEST] <kierank> that will never fly in a realtime environment [22:56:17 CEST] <rcombs> so I have to do dumb hacks like assuming the current subtitle segment is finished if the video gets a couple segments ahead of it [22:56:32 CEST] <rcombs> would be good to have heartbeats in lavf for that [22:57:26 CEST] <rcombs> I'd say "write an empty packet with the timestamp set accordingly" but empty packets are kinda overloaded as a concept [22:57:38 CEST] <JEEB> yea [22:57:59 CEST] <JEEB> reminds me of the two ways people already implemented PMT updates [22:58:02 CEST] <JEEB> one was callback-based [22:58:11 CEST] <JEEB> another was AVPackets that came from a specific stream id [22:58:12 CEST] <JEEB> (zero) [23:02:25 CEST] <rcombs> also lavfi has a bit too much boilerplate [23:02:44 CEST] <durandal_1707> rcombs: be more specific [23:06:06 CEST] <durandal_1707> kierank: regarding framesync, one could add option to consume frames asap and not wait for next frames pts [23:06:34 CEST] <rcombs> jkqxz: oh hey do you have an Apollo Lake system [23:07:52 CEST] <rcombs> I've had a user report an issue there where the graph "hwupload,scale_vaapi=w=[X]:h=[Y]:format=nv12,hwupload" with VAAPI for both dec and enc results in the chroma planes getting interleaved wrong in the output [23:08:09 CEST] <rcombs> as if it's either getting nv12 but treating it as yuv420p, or getting yuv420p and treating it as nv12 [23:11:39 CEST] <jkqxz> I don't have an Apollo Lake right now, but I can easily find one to test on. [23:11:42 CEST] <jkqxz> Why hwupload twice there? [23:13:04 CEST] <jkqxz> If you have VAAPI for both encode and decode there then neither of those hwuploads should be doing anything. [23:14:21 CEST] <atomnuker> vaapi chroma planes are messed up - if you map yuv420p (not nv12) vaapi to opencl or vulkan U and V are swapped [23:14:58 CEST] <jkqxz> Oh, is that YV12? [23:15:13 CEST] <BtbN> I420 [23:15:39 CEST] <BtbN> nvenc does the same. As a reaction, the CUDA pix_fmts have it internally swapped [23:16:15 CEST] <durandal_1707> kierank: write how that direct api should roughly look like and i will implement it [23:16:37 CEST] <kierank> av_filter_frame(ctx, in, out) [23:17:13 CEST] <kierank> and your pipeline code is a high level implementor [23:17:14 CEST] <JEEB> yeh, 1) fill as many inputs as you have 2) filter [23:17:14 CEST] <durandal_1707> kierank: that is very trivial, and does not cover multiple inputs/outputs [23:17:27 CEST] <atomnuker> yeah, I'd also like a direct api [23:17:29 CEST] <kierank> durandal_1707: sure, but you handle that in your pipeline code [23:17:37 CEST] <JEEB> for example if you only have the primary inpu tavailable for overlay, you don't fill it [23:17:41 CEST] <JEEB> and then fill the primaryt [23:17:47 CEST] <JEEB> and then press the butan [23:17:48 CEST] <rcombs> jkqxz: yeah, I sprinkle hwuploads all over the place because it makes generating the graph text easier (less special-casing depending on what's using hardware and what's not); since they're no-ops I don't think it matters? [23:18:26 CEST] <durandal_1707> kierank: no comprehendo [23:18:52 CEST] <rcombs> if the user drops the :format=nv12 then it works fine [23:19:15 CEST] <rcombs> but some devices require that iirc [23:19:25 CEST] <durandal_1707> kierank: overlay filtter needs 2 input frames [23:19:33 CEST] <JEEB> durandal_1707: you fill either one or two slots [23:19:42 CEST] <JEEB> primary and if you hav esecondary the secondary [23:19:47 CEST] <JEEB> then you press butan [23:19:52 CEST] <jkqxz> Probably needed on not-i965. That driver has weird magic which copies input frames if it doesn't like the format they start as. [23:20:04 CEST] <durandal_1707> so it would be av_filter_frame(ctx, **in, **out) [23:20:16 CEST] <atomnuker> perfect [23:20:33 CEST] <jkqxz> rcombs: Do you have a picture of the output? [23:20:44 CEST] <rcombs> https://us.v-cdn.net/6025034/uploads/editor/ue/de8zfx6frvn8.png [23:20:53 CEST] <rcombs> vs correct: https://us.v-cdn.net/6025034/uploads/editor/v9/md98p8vih4dy.png [23:22:06 CEST] <durandal_1707> kierank: is that **frame too alien to you? [23:22:34 CEST] <JEEB> durandal_1707: wouldn't each filter instance have slots? [23:22:44 CEST] <JEEB> then you first fill them up as you have them [23:22:57 CEST] <JEEB> then after you have filled, you push [23:23:11 CEST] <JEEB> so the ctx would be the filter context or something [23:27:29 CEST] <durandal_1707> JEEB: how would you know to which input/output frame belongs? [23:27:53 CEST] <JEEB> well you have just gathered them, no? [23:28:14 CEST] <JEEB> it's just like you feed primary/secondary in filter chains right now [23:28:15 CEST] <atomnuker> you expose the filter inputs and outputs array which have descriptions [23:31:19 CEST] <jkqxz> rcombs: That's pretty weird. It's like the chroma samples have been translated horizontally slightly, some to the left and some to the right? The vertical positioning looks right, as do the actual values. [23:31:29 CEST] <jkqxz> I'll try to reproduce it tomorrow. [23:38:40 CEST] <rcombs> thanks [23:52:06 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vhlzC [23:52:06 CEST] <KGB> 13FFV1/06master 1479f5938 15Jérôme Martinez: Version 0 and 1 can have some content after SliceContent... [00:00:00 CEST] --- Mon Jun 4 2018

1 0

ffmpeg.log.20180603
by burek 04 Jun '18

04 Jun '18

[00:00:01 CEST] <Zexaron> for some reason the .exe still wants swresample-3.dll while I removed it completely, greped the files, no mentions [00:00:45 CEST] <Zexaron> Well, there's this FindFFMPEG.cmake but it has other stuff defined which doesn't exist either [00:03:16 CEST] <Zexaron> I cleaned and rebuilt, nothing [00:04:17 CEST] <Zexaron> set(_FFmpeg_DEPS_postproc avutil) [00:04:17 CEST] <Zexaron> set(_FFmpeg_DEPS_swresample avutil) [00:04:40 CEST] <Zexaron> there's device, etc, none of which existed before, and it wasn't needing them [00:07:18 CEST] <Zexaron> also only avutil.h and log.h from avutils has some reference to swresample but doesn't look like it needs them [00:19:24 CEST] <Zexaron> do other ffmpeg dlls need it? [00:24:21 CEST] <Zexaron> Zeranoe's avcodec-58.dll has a string pointing to swresample-3.dll [00:24:40 CEST] <Zexaron> I guess no "load it if I really need it" thing in there as well [00:25:19 CEST] <Zexaron> JEEB: What if that's not intended, bug? [00:25:40 CEST] <JEEB> some audio decoders (opus) need the resampling functions [00:25:42 CEST] <Zexaron> Someone told me before swresample might not be needed and I can remove it [00:25:55 CEST] <JEEB> also libavfilter can need swresample because the audio filtering code uses it [00:26:08 CEST] <JEEB> I mean, you can disable all of that during build time, but otherwise it's all there, cross-library deps :P [00:27:37 CEST] <Zexaron> That's fine, but could it be made to only have to load it when that component is used, such as opus, without having to do a custom build while disabling opus and etc [00:28:06 CEST] <Zexaron> Which is what I'm having to do with Dolphin, so it only loads ffmpeg when needed. [00:40:43 CEST] <Zexaron> I have a thing of being stubborn and persistent [00:43:36 CEST] <Zexaron> I opened avcodec-58.dll in hex, found swresample, it was the last of dependency entries like .da.da.avutil-56.dll...xa.xa.xa.xa.xa.xa.swresample-3.dll........ with a few lines of the end dots representing NULL [00:44:29 CEST] <Zexaron> so I nulled out the .xa ..swresample entry, it gave me an error about a missing ".DLL" so pretty much the same error just without the name [00:45:42 CEST] <Zexaron> JEEB: Then I reverted back to official avcodec-58.dll and created an empty dummy swresample-3.dll - it works, tested and tried the feature inside Dolphing. [00:46:11 CEST] <Zexaron> Now, are you thinking what I'm thinking .... [00:46:47 CEST] <Zexaron> In order to make ffmpeg official ... all I have to do is to ship DUMMY ffmpeg shared libraries with Dolphin .... FINGERS CROSSED !!! [00:49:23 CEST] <Zexaron> In order to make ffmpeg optional* [00:49:55 CEST] <Zexaron> Well I'm still renaming/creating dummies ... drumroll any second [00:52:37 CEST] <Zexaron> Bah, hard requirement, "is either not designed to run on Windows or is corrupt" [01:47:47 CEST] <Zexaron> i'll probably have to do the pointers thing ... but next time, time to shutdown the powah, air conditioner installation, bye! [01:51:34 CEST] <TheAMM> So I have a v4l2 video source and an alsa audio source, and they're not synced up [01:51:56 CEST] <TheAMM> Is there any relevant magic I could do when recording, or do I just itsoffset manually afterwards [04:14:44 CEST] <kepstin> TheAMM: if you the offset is consistent, you could correct for it my using setpts (or asetpts) filters during the capture. [04:15:13 CEST] <kepstin> TheAMM: the problem's probably caused by how ffmpeg initializes inputs one at a time, so it'll always do one before the other :/ [04:15:32 CEST] <kepstin> although it's rare for it to be off far enough to be a problem with only 2 inputs [10:32:55 CEST] <hellozee> I can generate slideshows from images but now I am trying to add transitions between them, specifically the sliding from top one, searched the web but they mostly talk about fades/zoom/pan, what would be the proper approach? [11:19:46 CEST] <lesshaste> I have a video I want to cut out the first 20 seconds of. That I can do but I would like the sound to fade out, rather than just stop abruptly. Is that possible with ffmpeg? [11:38:09 CEST] <Kam_> lesshaste: should be possible, have a look at the 'afade' audio filter. [14:25:49 CEST] <anill> i have an raw stream of RTP packets captured into a file called mycapture.dump, how can i convert it into H.264 via ffmpeg.Do i need to save RTP packets in some specific format or need some development work at ffmpeg side [14:37:59 CEST] <DHE> I don't think the answer has changed since you last asked [14:56:34 CEST] <anill> DHE: iran the command u said, it says invalid data found while processing [15:02:17 CEST] <DHE> when you said rtp I assumed you meant a connection to some internet service, not that you'd recorded a tcpdump/wireshark pcap [15:10:43 CEST] <anill> DHE: i made a custom dump [15:11:37 CEST] <JEEB> the RTP parser probably is in the protocol code [15:11:39 CEST] <JEEB> since you know [15:11:54 CEST] <JEEB> people generally don't rip that info and don't replay it but instead attempt to read it from a file [15:12:12 CEST] <JEEB> so if you really want to do it without replaying the traffic, you will have to rip out the RTP parser from the protocol [15:12:38 CEST] <JEEB> otherwise just open the protocol with a timeout, and start replaying the data to localhost [15:15:52 CEST] <lesshaste> Kam_, thanks! [15:16:01 CEST] <lesshaste> Kam_, can you fade the video too? [15:18:31 CEST] <Kam_> lesshaste: 'fade' and 'tblend' can do this https://ffmpeg.org/ffmpeg-all.html#fade and https://ffmpeg.org/ffmpeg-all.html#blend_002c-tblend [15:18:46 CEST] <lesshaste> Kam_, thanks! [15:32:40 CEST] <anill> JEEB: ist for me right? [15:34:11 CEST] <anill> JEEB: Its for me right? [17:32:13 CEST] <zumba_ad_> morning all. I have few still pictures of an area throughout the day where shadow moves every hour. I'd like to make a nice transition on the timelapse. Can we use ffmpeg to make a nice transition between frames? [17:33:25 CEST] <zumba_ad_> it's not a fade-in/out effect [17:43:01 CEST] <DHE> so, fade between but without black in between? [17:45:36 CEST] <Blacker47> zumba_ad_, maybe something with https://ffmpeg.org/ffmpeg-filters.html#blend_002c-tblend but i can't help futher with it. [17:47:10 CEST] <kepstin> you could also try something like set the input to a really low framerate, then use the "framerate" filter with a higher fps to generate the blended frames between. [18:01:09 CEST] <TheAMM> kepstin: -async 1 was enough [18:01:24 CEST] <TheAMM> (regarding the audio sync on v4l2 and alsa I mentied yesterday) [18:06:38 CEST] <zumba_ad_> ok, taking a look [18:09:04 CEST] <ressected> hello! [18:09:12 CEST] <ressected> long time ffmpeg user here [18:09:37 CEST] <ressected> setup is plex server, chromecast [18:09:39 CEST] <ressected> all linux [18:10:30 CEST] <ressected> got a good copy of The Godfather and TV (for some reason) will not take the AC3 from chromecast and output is a LPCM to my d/a converter [18:11:00 CEST] <ressected> so I need advice on the best way to get AC3 5.1 to two channel AAC for mp4 container [18:11:41 CEST] <ressected> would like to go from AC3 to linear pcm but it doesn't look like the chromecast will take that in an mp4 [18:16:43 CEST] <kepstin> ressected: dunno about "best way", but just throwing "-ac 2" to remix to stereo along with the appropriate codec options should be sufficient. [18:17:04 CEST] <DHE> -ac 2 -c:a aac -b:a 128k # or a higher number for higher bitrate if you prefer [18:18:14 CEST] <ressected> DHE, kepstin, that will do all the summing of the channels in some linear way? [18:19:41 CEST] <kepstin> ressected: yeah, it's got a reasonable preset matrix that it'll use. [18:19:44 CEST] <ressected> DHE, kepstin, e.g. does it just throw away the surround channels or does it try to be "smart" and mix it in like dolby pro-logic type thing [18:19:53 CEST] <DHE> ac3 is an interesting codec. it may specify its own mixing matrix [18:21:00 CEST] <ressected> kepstin, DHE, I will try it [18:22:06 CEST] <ressected> kepstin, DHE, thanks for your help [18:24:55 CEST] <anil> DHE: Hey [18:26:28 CEST] <anil> This info, does JEEB referred to me [18:26:28 CEST] <anil> <JEEB> the RTP parser probably is in the protocol code [18:26:28 CEST] <anil> <JEEB> since you know [18:26:28 CEST] <anil> <JEEB> people generally don't rip that info and don't replay it but instead attempt to read it from a file [18:26:28 CEST] <anil> <JEEB> so if you really want to do it without replaying the traffic, you will have to rip out the RTP parser from the protocol [18:26:28 CEST] <anil> <JEEB> otherwise just open the protocol with a timeout, and start replaying the data to localhost [18:28:30 CEST] <anil> JEEB: are you there? [18:29:04 CEST] <durandal_1707> anil: stop doing that, he will reply if he can [18:29:27 CEST] <DHE> well officially he was nagging me and just quoting jeeeb.... [18:29:47 CEST] <anil> durandal_1707: sorry, will take care [18:37:04 CEST] <ressected> anyway to sleep up the AAC encoding to use more than one core? [18:37:07 CEST] <ressected> speed* [18:38:25 CEST] <klaxa> run more than one encode at once? [18:39:12 CEST] <ressected> klaxa, I'm converting single file from 5.1 ac3 to 2 channel AAC [18:39:45 CEST] <kepstin> ressected: if it's really slow, you're probably accidentally re-encoding the video [18:39:54 CEST] <kepstin> ressected: add a "-c:v copy" :) [18:40:22 CEST] <kepstin> audio encoding is generally fast enough that you just live with it being single-threaded. [18:41:04 CEST] <ressected> ffmpeg -i The.Godfather.1972.1080p.BluRay.x264.AC3.mp4 -ac 2 -c:a aac -b:a 384k -vcodec copy The.Godfather.1972.1080p.BluRay.x264.AAC.mp4 [18:41:07 CEST] <ressected> so thats not right? [18:41:25 CEST] <kepstin> that should be fine, "-vcodec copy" and "-c:v copy" are the same thing [18:41:29 CEST] <klaxa> you could possibly split it up into smaller parts without re-encoding, re-encode the single parts and then concatenate them back together, maybe timestamps get mangled though [18:41:50 CEST] <kepstin> klaxa: doesn't actually work, because of how codecs like aac require preroll packets and padding :/ [18:41:53 CEST] <ressected> its only encoding it at 4x speed [18:41:58 CEST] <klaxa> :V [18:42:03 CEST] <klaxa> thank you bad standards [18:42:05 CEST] <kepstin> (i mean it still can be done, but requires detailed knowledge of how the codec works) [18:42:27 CEST] <ressected> klaxa, I appreciate the hacker mentality [18:42:36 CEST] <kepstin> klaxa: pretty much all audio codecs are like that, because of the overlapped transforms they use [18:42:51 CEST] <ressected> klaxa, by the time I figure that out I might as well just wait the 40 minutes for the encode [18:42:56 CEST] <kepstin> ressected: are you limited by disk speed, maybe? [18:43:35 CEST] <kepstin> ressected: also, that's a kinda ridiculously high bitrate, and the encoder might be running slower just because of that. Try something like 128-192 instead maybe. [18:43:39 CEST] <ressected> kepstin, nah [18:43:57 CEST] <ressected> I thought the higher bit rate the faster it is, due to less compression [18:44:15 CEST] <klaxa> but it has to calculate more bits ;) [18:44:31 CEST] <ressected> crap, okay I'll do 192 [18:44:42 CEST] <ressected> on that system you can hear the smallest artifacts [18:44:44 CEST] <kepstin> depends on the codec, I don't know exactly how the internal aac encoder scales [18:44:52 CEST] <ressected> wont hurt to try [18:45:20 CEST] <kepstin> you might consider using fdk_aac instead, if you can build ffmpeg yourself (although that will probably take longer than the transcode...) [18:45:46 CEST] <ressected> kepstin, I'll consider that in the future [18:46:53 CEST] <kepstin> but yeah, I'd really expect you do be getting a lot faster than 4x unless you have other bottlenecks. What version of ffmpeg are you running? [18:47:42 CEST] <ressected> ffmpeg version 3.1.9 [18:48:17 CEST] <ressected> I am doing it on an NFS share but that should give me 30MB/s bi-directional at minimum [18:49:32 CEST] <BtbN> 3.1. wow. [18:55:11 CEST] <ressected> BtbN, whats wrong with that? [18:55:34 CEST] <ressected> latest avail on fedora 24 [18:55:51 CEST] <BtbN> it's old. [18:56:15 CEST] <JEEB> also fedora 24 is EOL [18:56:16 CEST] <JEEB> rip [18:56:32 CEST] <ressected> thanks but it's what I got [18:56:34 CEST] <JEEB> they only at max support current-2 and even 26 is soon going to end down [18:56:47 CEST] <BtbN> you're not going to get anywhere with that version [18:56:49 CEST] <JEEB> also for the record, Fedora itself doesn't ship FFmpeg [18:56:54 CEST] <ressected> kepstin, from the aforementioned conversion, is the audio likely to be delayed? [18:57:09 CEST] <ressected> kepstin, is there some offset I should be specifying? [18:57:21 CEST] <JEEB> if the input file is OK it should work [18:57:28 CEST] <JEEB> since it just utilizes the timestamps from input [18:57:31 CEST] <kepstin> ressected: shouldn't be, you're not resampling (and ffmpeg sets the timestamps right, so it should be fine in any case) [18:57:53 CEST] <ressected> kep, okay might be my TV then [18:57:55 CEST] <ressected> kepstin, [18:58:07 CEST] <ressected> kepstin, thank you [18:58:28 CEST] <JEEB> well bugs are always possible but relatively unlikely. generally the first recommendation would be to try and build current master and try with it [18:59:45 CEST] <ressected> oh yeah lowering the bitrate sped it up big time [19:01:48 CEST] <EvanR> hello [19:02:26 CEST] <ressected> EvanR, hi [19:02:49 CEST] <EvanR> i think 2 years ago i had a problem with avfoundation capture [19:03:21 CEST] <EvanR> someone in here kernrj i think provided assistance [19:03:43 CEST] <EvanR> i think there was a partial fix, heres the thread https://trac.ffmpeg.org/ticket/5654 [19:03:56 CEST] <EvanR> i got notified about it since someone just commented [21:52:51 CEST] <PaulHere> Hello, in my mp4 file 176 bytes in sequence are equal to FF (inside mdat). Anyone knows why? [21:59:23 CEST] <PaulHere> I think it's the payload of a nal unit, but the nal-header has the forbidden_zero_bit set to 1... any ideas? should I just skip them? [22:11:33 CEST] <PaulHere> ok, so I gues that forbidden_zero == 1 means that there was an error in transmission, and you are supposed to skip it [22:32:31 CEST] <PaulHere> is it possible to rederict libavcoec messages to stdout instead of stderr? [22:33:03 CEST] <BtbN> with ffmpeg.c? no [22:33:27 CEST] <BtbN> as an API user there iirc is a log callback and you can do whatever in there [22:33:37 CEST] <PaulHere> i just wanted to say there is av_log_set_callback [22:33:50 CEST] <PaulHere> yes [22:45:09 CEST] <JEEB> yea as an API client you can do whatever you want [23:04:15 CEST] <DHE> can confirm, I do whatever I want with the API. :) [00:00:00 CEST] --- Mon Jun 4 2018

1 0

ffmpeg-devel.log.20180602
by burek 03 Jun '18

03 Jun '18

[02:10:06 CEST] <cone-026> ffmpeg 03Jacob Trimble 07master:9827bb88e7dc: libavformat/mov: Fix heap buffer overflow. [13:11:21 CEST] <cone-226> ffmpeg 03Paul B Mahol 07master:e28b1fa6e99c: avfilter: add adeclick and adeclip audio filters [13:46:39 CEST] <durandal_1707> atomnuker: i get this one /video/out/wayland_common.c:901:14: error: XDG_TOPLEVEL_STATE_TILED_RIGHT undeclared (first use in this function) --> is some other wayland now needed? [13:49:18 CEST] <JEEB> yes that needs a newer version of wayland-protocols, when merging his changes I bumped that in configure [13:49:28 CEST] <JEEB> and the wayland people told me "just vendor the XML" [13:49:30 CEST] <JEEB> :V [13:50:50 CEST] <JEEB> https://github.com/mpv-player/mpv/commit/d8288877923728dad3c5ce6c735a9b4cde… [14:27:32 CEST] <durandal_1707> JEEB: i have latest protocols installed already... [14:29:32 CEST] <durandal_1707> looks like i need do do full clean [14:29:37 CEST] <JEEB> not sure at which point the headers get generated but most likely you missed some point or something by not running configure (or the wayland-protocols 1.14 is too new, which is why #wayland was OK with just vendoring that crap) [14:47:44 CEST] <gagandeep> kierank: that 2nd frame is working now, i didn't need to do any av_frame stuff as the buffers which held the data of coefficients were not being freed so i could use them directly [14:48:06 CEST] <gagandeep> just working out the offset values and 'prescales' to get bitexact output [14:48:21 CEST] <kierank> you should start doing bitexact testing on all the samples [14:48:42 CEST] <durandal_1707> gagandeep: how many past frames cfhd can use? [14:49:12 CEST] <gagandeep> right now 3d transform is only using 2 frames as group [14:49:24 CEST] <gagandeep> which are passed through temporal transform [14:50:01 CEST] <gagandeep> i don't think it is working like mpeg where past frames are used for construction of new frames [14:50:45 CEST] <gagandeep> it just used lowpass bands of two frames(by doing spatial transform) through a temporal transform [14:52:56 CEST] <gagandeep> a total of 17 subbands are used to reconstruct 2 frames and cycle repeats [14:53:17 CEST] <kierank> gagandeep: sure, but if you are doing threading you need to make sure the first frame has finished decoding (well not decoding per se, but coefficients are done) [14:53:26 CEST] <kierank> gagandeep: did you remove my in place optimisations then? [14:53:49 CEST] <gagandeep> which?, i only removed .capabilities [14:54:31 CEST] <gagandeep> other than that changed how some error handling was done as mountain sample was not having some tags in right place [14:54:33 CEST] <kierank> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/cfhd.c#L305 [14:54:50 CEST] <kierank> i'm surprised you don't need to remove the inplace optimisations [14:55:29 CEST] <gagandeep> i have read the code and i didn't see any problem with any of the transform stuff [14:55:38 CEST] <kierank> interesting [14:56:43 CEST] <gagandeep> though i am assuming the part of code you are asking about is after all subbands are read, right? [14:57:09 CEST] <kierank> during allocation [14:57:11 CEST] <kierank> so [14:57:12 CEST] <kierank> 8:32:33 PM <durandal_1707> kierank: ff_thread_report_progress() ff_thread_await_progress() this are calls for synchronization between different threads, (A)PNG is i think simplest decoder that use them [14:57:23 CEST] <kierank> after bitexact testing I think you need to add threading [14:57:40 CEST] <gagandeep> yeah, will begin to read on threading [14:58:02 CEST] <gagandeep> will look in png [14:58:56 CEST] <kierank> you probably need to copy pointers between threads when frame 1 has finished unpack [15:01:11 CEST] <durandal_1707> gagandeep: you also call ff_thread_release_buffer() after you do not need thread frame any more... [15:01:48 CEST] <gagandeep> ok, guys, will ask when i start looking into it on probably monday [15:01:57 CEST] <gagandeep> then will be asking a lot [15:02:05 CEST] <gagandeep> like a fool [15:02:08 CEST] <gagandeep> :) [15:02:49 CEST] <kierank> that ami_stuff guy needs to come and provide samples [15:03:11 CEST] <gagandeep> also might need to explore how cineformsdk handles threads [15:03:14 CEST] <kierank> gagandeep: you should make new temporal_inverse_filter function [15:03:17 CEST] <kierank> gagandeep: i would not do that [15:03:21 CEST] <gagandeep> k [15:03:22 CEST] <kierank> just look at how ffmpeg does it [15:03:28 CEST] <gagandeep> ok [15:03:35 CEST] <kierank> when you do simd, you can't do if (!temporal_for_highpass) { [15:03:44 CEST] <kierank> well you can but it's inefficient [15:03:56 CEST] <gagandeep> k, will keep in mind [15:04:38 CEST] <kierank> it's good there are simple simd functions [16:27:49 CEST] <rindolf> Hi all! tmm1 : some days ago i sent a patch to http://ffmpeg.org/pipermail/ffmpeg-devel/ but it isn;t there yet [16:29:37 CEST] <JEEB> probably in the moderation queue then [16:29:52 CEST] <JEEB> never 100% understood who was maintaining that [16:47:21 CEST] <cone-226> ffmpeg 03Gyan Doshi 07master:e50a5c9c4eb7: doc/ffmpeg: rewrite Stream Selection chapter [18:20:01 CEST] <JEEB> would be nice if someone could ack https://patchwork.ffmpeg.org/patch/9228/ [18:20:22 CEST] <JEEB> (or tell how this should be "properly" fixed if that's not the way to fix it) [18:21:38 CEST] <JEEB> also the committer is listed as pedro arthur... http://git.videolan.org/?p=ffmpeg.git;a=commit;h=bdf1bbdbb4ebb342c0267d0f77… I wonder if he's on IRC [18:25:01 CEST] <jdarnley> JEEB: from the description you gave it sounds like an okay solution for now [18:25:36 CEST] <jdarnley> If something else is added with depends on the dnn thing then I gues it will need it own "module" in configure [18:25:36 CEST] <JEEB> yea someone on #ffmpeg just had most filters in libavfilter disabled [18:25:41 CEST] <JEEB> yea [18:25:45 CEST] <JEEB> that's the way I felt [18:37:33 CEST] <durandal_1707> atomnuker: actual useful vulkan filter would be lut3d [18:46:30 CEST] <jamrial> JEEB: patch ok [18:48:17 CEST] <JEEB> ok [18:49:29 CEST] <cone-226> ffmpeg 03Jan Ekström 07master:76daf4529bf2: lavfi/Makefile: move dnn_*.o under vf_srcnn [18:49:32 CEST] <JEEB> pushed [19:19:13 CEST] <cone-226> ffmpeg 03Paul B Mahol 07master:af31084399b8: avfilter/vf_lut3d: unbreak haldclut with planar rgb formats [21:48:32 CEST] <cone-226> ffmpeg 03Michael Niedermayer 07master:5ee203076fa1: avcodec/vp3: Fix end of bitstream check in unpack_superblocks() [23:48:01 CEST] <atomnuker> TD-Linux: actually forcing gpu rendering on linux makes things worse [23:48:28 CEST] <atomnuker> firefox with gpu rendering always damages the whole screen, if you leave it disabled it'll damage only areas that need to be damaged [00:00:00 CEST] --- Sun Jun 3 2018

1 0

ffmpeg.log.20180602
by burek 03 Jun '18

03 Jun '18

[00:06:02 CEST] <Ars> DHE, Now i am running as: ffmpeg -i rtmp://Source -codec:v mpeg2video -qscale:v 2 -r 29.97 -g 60 -acodec ac3 -f mpegts udp:IP?pkt_size=1316 [00:06:02 CEST] <Ars> Can you help with correct commanded with CBR muxing ? [00:33:21 CEST] <DHE> qscale isn't good for CBR [00:33:23 CEST] <DHE> so there's that [00:42:53 CEST] <Ars> DHE, so i should remove qscale and add only -muxrate ? [00:44:08 CEST] <DHE> no, you need CBR options [00:45:07 CEST] <Ars> pls help, i am not shure how to add BR options [00:52:16 CEST] <wfbarksdale> its late on friday and i can't figure out how to get this command to not use bframes on re encoding: [00:52:16 CEST] <wfbarksdale> ffmpeg -y -i low-res-audi.mp4 -r 60000/1001 low-res-audi-59-94fps.mp4 [01:06:54 CEST] <c_14> -bf 0 [01:06:57 CEST] <c_14> probably [01:12:13 CEST] <Fusl> hi all. i have a strange issue with, well, not with ffmpeg but with a multicast udp stream transported as rtp container with a h264 and mp2 stream encoded within. our ISP uses multicast to send traffic for their iptv solution, its a small samsung/zenterio (i believe) box and instead of having several of their boxes lying around (which is especially annoying because of the lack of 5ghz/2.4ghz wifi support we [01:12:15 CEST] <Fusl> have to run cables everywhere) we want a central server providing hls streams to be able to use it with our chromecasts that we have deployed everywhere in our house. so far i digged into wireshark and know that they use multicast to transmit their streams to all customers, got ffmpeg to do the magic trickery to join the multicast group and receive the rtp stream from `rtp://@224.1.2.42:3042`. i get the rtp [01:12:16 CEST] <Fusl> stream into ffmpeg but ffmpeg/h264 complains a lot about corrupted data then. the exact command i useed is `./ffmpeg -loglevel error -max_delay 500000 -err_detect ignore_err -y -i rtp://@224.1.2.42:3042 -f null -`. when viewed with ffplay, the picture and audio data *is* from the actual stream, but lots of corrupted data, for example here's a screenshot of what the picture looks like, permanently: [01:12:17 CEST] <Fusl> http://xor.meo.ws/4f3973d8/b5eb/4199/8892/d7f32ed4cdcd.png - the errors that ffmpeg is spewing out are: http://xor.meo.ws/nPPTCZfBSFjcH6ews94yQGotgL2VW2T6.txt [01:12:25 CEST] <Fusl> wow that's a wall of a text there, sorry. [01:15:48 CEST] <Hello71> tl;dr [01:16:23 CEST] <Fusl> tl;dr iptv multicast stream from my ISP plays find on their iptv box but is half corrupted when viewed using ffplay/vlc/mpv/mplayer [01:16:36 CEST] <Hello71> oh well why didn't you start with that [01:16:45 CEST] <Fusl> here's an udp dump if anyone is interested in taking a look: http://xor.meo.ws/tTOwdVD0R_pPAGlZ3WrpO0Wgp8cnbrLL/stream.dump [01:16:53 CEST] <c_14> Fusl: ffmpeg's rtp protocol has numerous deficiencies (especially concerning rtcp) [01:16:54 CEST] <Fusl> Hello71: i didn't realy know where to start, heh [01:17:18 CEST] <c_14> without looking too much into it, it might be better to use an external rtp library and then pass the data to ffmpeg for hls generation [01:17:42 CEST] <Fusl> c_14: would you be able to recommend any? [01:17:47 CEST] <Ars> DHE, how to add CBR options ? [01:18:02 CEST] <Hello71> Ars: http://google.com/search?q=mpeg2+ffmpeg+cbr [01:19:17 CEST] <Ars> Hello71, thank you, in internet many pages, i think best correct source is ffmpeg irc channel :) [01:19:40 CEST] <c_14> Fusl: never really used any that much, you'll probably have to look around yourself [01:26:32 CEST] <ariyasu_> how do i weave with w4fdif? [01:26:39 CEST] <ariyasu_> w3fdif* [01:26:54 CEST] <ariyasu_> i did w3fdif=0:0 but it bobbed [01:30:54 CEST] <furq> ariyasu_: w3fdif doesn't have a bob mode afaik [01:31:07 CEST] <furq> you probably want bwdif=1 [09:13:49 CEST] <_Strix_> hello [09:14:58 CEST] <_Strix_> I need your help to compile and install FFMPEG 4.0 on CentOS 7.1 [09:24:11 CEST] <_Strix_> https://pastebin.com/eK35t6eX [13:30:12 CEST] <Zexaron> Tryint to figure out how to change path where ffmpeg shared DLLs are stored in a program, instead of the root [13:30:34 CEST] <Zexaron> using zeranoe -dev linking, is it hardcoded? [13:31:57 CEST] <furq> that's not a property of the program [13:32:56 CEST] <furq> dlopen/LoadLibrary will look in a predefined list of directories [13:33:26 CEST] <furq> if you want to change it then you need to change an environment variable [13:33:30 CEST] <furq> i think it's just PATH on windows [15:08:47 CEST] <Zexaron> furq not sure why Environmental variables and PATH on windows, this is an obscure directory in my git repos, I built Dolphin-Emu and the ffmpeg DLLs have to be in the root directory, I'd just move them to a subfolder [15:09:13 CEST] <furq> because that's how LoadLibrary works [15:09:23 CEST] <Zexaron> there are some Qt DLLs also in subdirectories, the program when released to public doesn't require any environmental variables [15:09:29 CEST] <Zexaron> changes [15:11:57 CEST] <Zexaron> I was looking if this is possible by editing the .lib or setting some flags in the program, this is an unacceptable solution because nobody will accept bothering 100k users to modify environmental variables [15:12:02 CEST] <furq> if you mean the plugins then those are loaded within the program [15:12:12 CEST] <furq> so you can specify a relative path [15:12:30 CEST] <furq> the libs that the binary is linked to are loaded before the program runs, so they have to be in one of the search path dirs [15:14:03 CEST] <Zexaron> Yeah, qt external may be differently implemented than what I did with ffmpeg now [15:14:52 CEST] <furq> the actual qt runtime libs that come with dolphin are in the root directory on windows [15:17:35 CEST] <Zexaron> yeah, sometimes I bother about the smallest things, the ffmpeg libs have filenames which displace dolphin.exe and with alphabetic arrangement it puts some of them to the top ... ofcourse for users it doesn't matter [15:18:14 CEST] <Zexaron> my hand-eye coordination for some reason doesn't like that [15:49:04 CEST] <kerio> install a rootkit that hides the ffmpeg libraries [16:46:29 CEST] <MrCivvy> Hi! How can I remove everything between x seconds to y seconds of a MP4 video? (using command line) [16:47:08 CEST] <DHE> transcoding or not? [16:47:17 CEST] <MrCivvy> Nope [16:47:33 CEST] <MrCivvy> Want it back in MP4 again. [16:48:01 CEST] <DHE> mp4 isn't related, but I'll assume no [16:48:37 CEST] <MrCivvy> Yeah [16:48:43 CEST] <DHE> if you're not transcoding, you must cut on keyframes. [16:49:08 CEST] <DHE> you can use -ss and -t to select time ranges, get the two pieces you want to keep, then concatenate them. [16:49:35 CEST] <DHE> not sure if you can do it in a single shot without temp files using stream copy. does concat have advanced options that can do that? [16:50:15 CEST] <DHE> hmm... it appears you can [16:53:28 CEST] <MrCivvy> https://pastebin.com/7gY38Lz9 [16:53:35 CEST] <MrCivvy> This is the what I used [16:53:42 CEST] <MrCivvy> Ran into issues though [16:56:46 CEST] <Kam_> Hi, I can not compile FFmpeg anymore since this https://github.com/FFmpeg/FFmpeg/commit/bdf1bbdbb4ebb342c0267d0f77cd06e7171… , it complains about "undefined reference to `avio_open'" and a few other avio_... in libavfilter/dnn_backend_native.o. I'm cross-compiling Unix->Windows with GCC8.1 and MinGW64. [16:58:02 CEST] <JEEB> I guess it complains when libavfilter gets linked? [16:58:21 CEST] <JEEB> I think that author did not add the dependency on libavformat for his filter [16:58:25 CEST] <Kam_> JEEB: yes, exactly. [16:58:29 CEST] <JEEB> anyways that cross-dependency is rather funky... [16:58:37 CEST] <furq> MrCivvy: trim selects the section you want to keep, not the section you want to remove [16:58:46 CEST] <JEEB> also how the hell did he get away with no configure changes [16:59:30 CEST] <JEEB> and since it has no dependencies it gets enabled by default, too [16:59:50 CEST] <furq> you want something like [0:v]trim=0:3622,setpts=PTS-STARTPTS[v0];[0:v]trim=3700[v1];[v0][v1]concat [17:00:13 CEST] <furq> [0:v]trim=0:3622,setpts=PTS-STARTPTS[v0];[0:v]trim=3700,setpts=PTS-STARTPTS[v1];[v0][v1]concat [17:00:42 CEST] <DHE> furq: but he said he didn't want to transcode [17:00:47 CEST] <MrCivvy> Okay, let me try that. [17:00:50 CEST] <furq> yeah but the thing he pasted is transcoding [17:00:56 CEST] <JEEB> Kam_: also for the record the github is just a mirror, git.videolan.org repository is the primary one [17:00:57 CEST] <furq> so i'm guessing he doesn't mind [17:01:55 CEST] <MrCivvy> Okay, I'm pretty new to ffmpeg. I understand transcoding as converting a video from one format to another. :P [17:02:33 CEST] <MrCivvy> Let me if there is more to it. [17:02:37 CEST] <MrCivvy> know* [17:02:40 CEST] <DHE> no, from one codec to another. or changing codec options like image resolution or bitrate [17:02:49 CEST] <MrCivvy> Ahh okay [17:02:51 CEST] <Kam_> JEEB: thanks, I know, I find the github web-interface easy to navigate through [17:02:55 CEST] <MrCivvy> No, I don't want change those. [17:03:20 CEST] <JEEB> Kam_: that's true, the interface can be better there [17:03:43 CEST] <DHE> the "best" way requires writing a concat filter file and letting the concat filter do the heavy lifting [17:03:53 CEST] <DHE> err, demuxer. not filter [17:04:53 CEST] <JEEB> Kam_: ahahaha [17:05:03 CEST] <JEEB> the srcnn filter has a dependency on libavformat [17:05:10 CEST] <JEEB> but he put those dependencies on global list [17:05:31 CEST] <JEEB> as in those additional files [17:06:04 CEST] <JEEB> let me make a patch and let's see if that fixes it for you [17:08:52 CEST] <JEEB> https://github.com/jeeb/ffmpeg/commit/497145ef9e40b00cc747ca8d0b28b69913a59… [17:08:58 CEST] <JEEB> Kam_: ^ see if this helps [17:14:41 CEST] <MrCivvy> DHE and furq: Thanks guys, it works! :) [17:15:05 CEST] <MrCivvy> DHE: I'll read about demuxers. [17:16:00 CEST] <furq> !demuxer concat @MrCivvy [17:16:00 CEST] <nfobot> MrCivvy: http://ffmpeg.org/ffmpeg-formats.html#concat-1 [17:16:19 CEST] <Kam_> JEEB, I'll try and report back in a few minutes [17:16:24 CEST] <JEEB> cheers [17:16:25 CEST] <furq> that's how you would do it without reencoding [17:16:39 CEST] <furq> but obviously it'll only cut on keyframes so it might not be an exact cut [17:16:55 CEST] <MrCivvy> Oh okay. [17:17:04 CEST] <MrCivvy> What if I want an exact cut? [17:17:20 CEST] <furq> then you either have to hope you get lucky and the keyframes coincide with the cut points, or you do the thing i said before [17:17:23 CEST] <furq> which means you have to reencode [17:19:58 CEST] <MrCivvy> Oh okay [17:20:02 CEST] <MrCivvy> Thanks [17:20:13 CEST] <furq> the concat demuxer will be more or less instant (with -c copy) so you might as well try it [17:21:25 CEST] <MrCivvy> Hmm sure. [17:21:55 CEST] <JEEB> Kam_: I'm going out to make a jogging run thing, will check your reply as I get back [17:22:18 CEST] <Kam_> JEEB: ok, thanks [17:23:00 CEST] <furq> preempting the next question [17:23:01 CEST] <furq> MrCivvy: http://vpaste.net/Pq8K0 [17:28:32 CEST] <Kam_> JEEB: it works, thanks! [17:30:01 CEST] <MrCivvy> furq: Sweet! You don't know how thankful I am :D [17:45:20 CEST] <RedSoxFan07> Hi guys! [17:47:22 CEST] <RedSoxFan07> I've copied some Blu-Rays, but the sound doesn't work on my Roku because the sound on those Blu-Rays is in the DTS codec. I'd like to transcode the audio to another codec, perhaps FLAC, but I don't want to have to convert the the video as well because I don't want to lose any video quality. Is it possible to use FFMPEG to just convert the audio in the MKV container? [17:55:29 CEST] <kerio> sure [17:55:31 CEST] <kerio> use -c:v copy [17:56:21 CEST] <JEEB> Kam_: ok, I will then post this on the mailing list [18:04:56 CEST] <JEEB> Kam_: was your use case such that you had libavformat disabled or? [18:07:57 CEST] <JEEB> Kam_: sent to ML, should appear there soon enough [18:08:44 CEST] <Kam_> JEEB: avformat is listed in configure under Libraries here [18:09:28 CEST] <JEEB> ok [18:10:02 CEST] <JEEB> in any case libavformat was not there when linking libavfilter [18:10:14 CEST] <JEEB> which you'd think is the normal case [18:11:43 CEST] <Kam_> I'm expecting to get a avformat-xy.dll, but yes, my FFmpeg configuration is very stripped down and only what I really need is enabled [18:12:01 CEST] <JEEB> ok, that makes sense then [18:12:08 CEST] <JEEB> you most likely had that random filter disabled [18:12:15 CEST] <JEEB> which thus does not add a dependency on libavformat [18:12:36 CEST] <JEEB> and thus since those two dnn_* files were not under that filter [18:12:56 CEST] <JEEB> but instead "global libavfilter files", it would fail linking [18:13:21 CEST] <JEEB> automated testing didn't seem to figure that out because they're genreally built with everything enabled although I'm kind of surprised nobody caught that in review [18:13:35 CEST] <JEEB> "hey you're adding a cross-library dependency there btw" [18:15:04 CEST] <Kam_> JEEB: output from configure https://pastebin.com/FqHVEKi7 [18:15:35 CEST] <JEEB> yeh [18:15:58 CEST] <JEEB> that might have not happened if you had disabled all of libavfilter [18:16:08 CEST] <JEEB> but you had libavfilter enabled, but the filters disabled [18:16:58 CEST] <JEEB> it appeared on the archives https://patchwork.ffmpeg.org/patch/9228/ [18:17:11 CEST] <Kam_> I'm not sure, I'd need to re-investigate, I think I had to have libavfilter enabled because I'm linking against it with another application that expects it [18:17:44 CEST] <JEEB> but in any case, your config showed a bug [18:17:50 CEST] <JEEB> which I've now posted a patch to fix [18:19:43 CEST] <Kam_> thanks for that! [18:36:35 CEST] <lesshaste> I have a video I want to cut out the first 20 seconds of. That I can do but I would like the sound to fade out, rather than just stop abruptly. Is that possible with ffmpeg? [18:50:59 CEST] <JEEB> Kam_: and now it's in master [18:56:21 CEST] <Kam_> JEEB: wow great, that was fast! [18:57:51 CEST] <JEEB> usually stuff that's clearly borked gets fixed rather quickly [18:58:09 CEST] <JEEB> unless it's a component that nobody really understands [18:58:15 CEST] <JEEB> or if the breaking doesn't get found [18:58:55 CEST] <Kam_> JEEB: I've seen your zlib-pkg_config improvement, is it already in? I've not contacted the zlib developers regarding moving the DLLs in the CMake script, yet. [18:59:16 CEST] <JEEB> it should be [18:59:41 CEST] <MrCivvy> Why does transcoding at -q 0.0 causes the output file to be larger than the original? [18:59:57 CEST] <JEEB> since I built zlib and noticed it's there and I was rather annoyed by having to manually add flags [19:00:23 CEST] <JEEB> Kam_: yea, it's in since early May http://git.videolan.org/?p=ffmpeg.git;a=commit;h=8d1263fa199ff6dcfc8f0cc94f… [19:04:12 CEST] <Kam_> MrCivvy: I guess one reason is because it's trying to retain/compress the compression artifacts from your source as well (if the source is not lossless). [19:17:22 CEST] <MrCivvy> Oh okay. That's interesting. [19:22:46 CEST] <Kam_> MrCivvy: my suggestion would be, if you are cutting on non-keyframes to re-encode only until the next keyframe. copy all untouched GOPs to avoid unnecessary transcoding and concat everything together in the end. [19:52:37 CEST] <Zexaron> Hello [19:53:15 CEST] <Zexaron> Shipping ffmpeg Shared DLL's is okay, I did read up on license, but im not that suere [19:54:35 CEST] <Zexaron> Currently a custom .lib is shipped with only some codecs, an idea was to switch it over to dynamic linking, and then the user downloads optionally, because the size is bigger, people wouldn't agree shipping the whole thing as standard [19:54:57 CEST] <Zexaron> But they might, so I was thinking if it's legal to do that? [19:55:16 CEST] <Zexaron> There's no changes to the ffmpeg code [19:56:28 CEST] <Zexaron> That said, I found a big problem, the Program doesn't want to start because of missing ffmpeg dlls, I need to make this so it only needs the DLLs when a certain feture is enabled, a warning dialog, I hope this doesn't require having custom .libs [19:56:47 CEST] <Zexaron> Because I can just use Zeranoe's, much easier [20:05:23 CEST] <furq> the zeranoe libs are gplv3 because they're linked with x264 [20:05:47 CEST] <furq> that's the only real issue other than the size [20:06:56 CEST] <furq> dolphin is gplv2 so that's not an issue [20:07:26 CEST] <furq> honestly though you should compile your own libs if you're going to ship them [20:07:51 CEST] <furq> you need like four components out of thousands so you'd be saving a ton of space [20:12:38 CEST] <Zexaron> furq: 4 components? Actually there's an attempt to update framedumping, including trying to use VCE x264 and other HWAccell stuff, the custom libs create dependency on a maintainer having a ffmpeg build environment working, plus people want to use other codecs like lagarith [20:13:00 CEST] <furq> even so you could strip out a ton of stuff [20:13:19 CEST] <furq> you don't need any decoders or filters, for starters [20:13:38 CEST] <Zexaron> Sure, but that's for later, right now DLLs aren't going to be shipped, but an autodownloader will be provided, optionally. [20:14:07 CEST] <Zexaron> I don't know yet how to make it really optional tho, as the program doesn't start when they're missing. [20:14:23 CEST] <furq> you dlopen them at runtime [20:14:40 CEST] <furq> that all sounds like a lot of hassle though [20:14:53 CEST] <furq> it's really not difficult to build your own customised libs [20:15:06 CEST] <furq> you could even static link it into the binary since you're already gpl-compatible [20:16:56 CEST] <Zexaron> I know it's difficulty, I myself am not reliable and may not keep such environment in working shape for years [20:17:16 CEST] <Zexaron> I mean not difficult, I just can't be relied upon to always be there [20:17:18 CEST] <DHE> and you could remove anything you don't need. like decoders for codecs you won't be needing. for an emulator, that sounds like almost all of them. and encoder filters. you can cut the ffmpeg libs size in half or even lower. [20:18:40 CEST] <furq> it'd probably be more than half [20:19:03 CEST] <furq> you can drop avfilter, avdevice, swscale and swresample completely [20:19:23 CEST] <furq> and then you'd only need one muxer from avformat and maybe a few encoders from avcodec [20:19:41 CEST] <Zexaron> Look, decision has already been made, thanks, but it really won't affect at this time, this can be done later when it's known which codecs people use the most, avfilters, avpostprocess, avdevices aren't included already, sure demuxers and decoders could be removed, but I simply am not patient enough to deal with ffmpeg for weeks before, it can come later after people can use the full feature set, now they can't use anything [20:20:31 CEST] <furq> isn't dolphin already bundling lavc and lavf [20:21:14 CEST] <Zexaron> There is audio dumping, I will see if I can drop swresample, but scaling might come handing, I think some kind of image scaling to correct aspect is used somewhere, not sure if ffmpeg tho [20:22:07 CEST] <Zexaron> the shipped ffmpeg libs are for windows only I think, I'm not sure tho [20:22:16 CEST] <furq> that seems like the kind of thing an emulator would have fast internal routines for [20:22:23 CEST] <Zexaron> Other OS use the OS installed ffmpeg, linux and Mac [20:23:02 CEST] <furq> you probably don't want to get too fancy with it then or you're going to end up dealing with api breakage [20:23:30 CEST] <furq> granted if you're still on ubuntu 14 then you don't really deserve working software [20:23:33 CEST] <Zexaron> DHE: Sure, but again, development and upgrade is coming, nobody knows yet which codecs will be used, I'd be wasting time figuring that out right now [20:23:56 CEST] <furq> i take it someone's already got their heart set on writing a plugin downloader [20:24:17 CEST] <furq> because that seems like a really excessive thing to do considering you'd hardly be adding any code size (as far as i can tell) [20:24:30 CEST] <Zexaron> All I know is, that the existing stuff MPEG4 visual and AVI is probably going to be cut, but I guess if user overrides they could use it, won't be default anymore [20:24:42 CEST] <DHE> Zexaron: I'm working on the assumption that you're encoding video from an emulator out to a video file. so you don't need a decoder at all, maybe some simple "raw" codecs at worst (and even those you don't need) [20:24:45 CEST] <furq> you should drop xvid completely really [20:25:36 CEST] <furq> i would just have x264, libvpx, and a few lossless codecs [20:25:56 CEST] <furq> probably rawvideo, huffyuv, ffvhuff, ut video, ffv1, and lagarith if someone already asked for it [20:26:02 CEST] <Zexaron> DHE: yes you're right, however it does make sense, just at a later time when the feature works and things are glued down, then we can downsize ffmpeg, it makes more sense to do that later. [20:27:03 CEST] <Zexaron> The primary thing right now is, I need to handle ffmpeg's DLL presence, so that they aren't loaded at program start, but rather when the feature that needs them is enabled [20:27:39 CEST] <furq> like i said, that's a fair amount of extra effort [20:28:23 CEST] <Zexaron> Well, I guess the only way to figure out how much extra work it is, is to try it, anyway, I think it'll be worth it anyway. [20:28:33 CEST] <furq> there's no portable way of doing it [20:28:43 CEST] <furq> unless you pull in some windows dlopen wrapper [20:28:53 CEST] <Zexaron> No need to, these ffmpeg ways are only for Win [20:29:02 CEST] <Zexaron> DLLs* [20:30:10 CEST] <Zexaron> Or wait, you mean the code necessary to open it later [21:17:30 CEST] <Zexaron> Heh looks like on non-win os they don't load swresample https://github.com/dolphin-emu/dolphin/blob/master/CMakeLists.txt#L453 [21:20:48 CEST] <furq> oh yeah i guess they use swscale to convert to yuv [21:21:04 CEST] <furq> that's probably important [23:29:27 CEST] <Zexaron> Hmm, why aren't muxers, demuxers, encoders, decoders split into their own DLL [23:29:50 CEST] <Zexaron> In this case I wouldn't need any decoders and demuxers [23:30:10 CEST] <JEEB> nobody came up with a proper interface for that [23:30:26 CEST] <JEEB> and everyone seemed to be content building their own limited versions for their own use case on windows [23:30:28 CEST] <Zexaron> or encoders+muxers in their DLL and for demuxers+decoders, [23:30:30 CEST] <JEEB> (and mac) [23:30:31 CEST] <Zexaron> more sense [23:30:52 CEST] <JEEB> right now you have decoders and encoders in avcodec, demuxers and muxers in avformat [23:30:57 CEST] <DHE> libavinput and libavoutput ? [23:31:03 CEST] <Zexaron> oh sorry lol : [23:31:49 CEST] <Zexaron> so it's already separate, just, not the right mix for this case [23:32:17 CEST] <JEEB> yes [23:32:20 CEST] <JEEB> there is no single libffmpeg [23:32:34 CEST] <Zexaron> DHE - yeah! ... but now if it comes to this, then should try to look if it won't break the existing usecases [23:32:35 CEST] <JEEB> and it sounded like you wanted more separation [23:32:54 CEST] <JEEB> lik each module being a separate library within those libraries [23:33:09 CEST] <JEEB> (each decoder/encoder/demuxer/muxer being its own thing) [23:33:26 CEST] <Zexaron> I don't know enough to be able to gauge if something needs only avformat and not avcodec, or vice versa. [23:33:32 CEST] <Zexaron> Out there [23:33:50 CEST] <JEEB> anyways, start worrying about optimizing after you've got a PoC running [23:34:11 CEST] <Zexaron> So I got rid of swresample in this case just now, seems like it wasn't used, but defined as a dependency in vcxproj [23:36:26 CEST] <Zexaron> It's a contribution I'm making to an open source project, Dolphin-Emu, it's not my program. Yes I'm including the full avformat/avcodec and not building ffmpeg with less features because not sure yet, but the problem is, program won't work without DLLs present, while the feature is very optional, so I have to do extra work to make the DLLs load later [00:00:00 CEST] --- Sun Jun 3 2018

1 0

ffmpeg-devel.log.20180601
by burek 02 Jun '18

02 Jun '18

[00:00:22 CEST] <jamrial> jkqxz: atomnuker already calls that in find_device() and prints other fields. do the same for driverVersion [00:02:15 CEST] <jkqxz> Mesa seems to be version 0x04801063, which isn't very inspiring. [00:05:45 CEST] <jkqxz> Windows: Intel = 0x00010003, AMD = 0x00800014. Hmm. [00:17:22 CEST] <jamrial> jkqxz: the latest windows AMD vulkan driver is 2.0.20 according to radeon settings [00:17:35 CEST] <jamrial> that 0x14 could be the 20 [00:28:04 CEST] <jamrial> jkqxz: that mesa driver version is 18.1.99 probably [00:32:48 CEST] <BtbN> philipl, according to most review sites it has both [00:33:12 CEST] <BtbN> I'm also still unsure if it is an Optimus Laptop, or if the Intel card comes disabled. It doesn't say Optimus or Intel anywhere [01:30:25 CEST] <philipl> BtbN: certainly if it's the only GPU, I'd expect both. [01:30:47 CEST] <philipl> I've been writing a silly little capability checker tool for nvdec/nvenc. I should stick it on github soon. [08:26:14 CEST] <leinne> why does ff_matroska_audio_muxer.audio_codec defaults to AV_CODEC_ID_VORBIS? where does that makes sense? [08:28:36 CEST] <atomnuker> got a better suggestion? [08:29:08 CEST] <atomnuker> aac would be better imo, patch welcome [08:30:17 CEST] <leinne> AV_CODEC_ID_AAC for matroska and AV_CODEC_ID_OPUS for ogg [11:01:08 CEST] <cone-404> ffmpeg 03Paul B Mahol 07master:0d35413e5dbe: avcodec/dnxhdenc: do not free nonexistent slice threads data when frame threading is used [11:04:12 CEST] <BtbN> philipl, I still expect the thing to be Optimus. Dedicated nvidia laptop GPUs don't really exist anymore. [11:06:53 CEST] <KGB> [13FFV1] 15michaelni closed pull request #114: Change ffv1 0,1,3 to informative type (06master...06change-ffv1-0,1,3-to-informative-type) 02https://git.io/vhnfn [11:52:47 CEST] <cone-404> ffmpeg 03Jérôme Martinez 07master:5205b3289e46: avcodec/dpx: Support for RGBA 12-bit packed decoding [16:22:50 CEST] <anill> Hi FFMPEG people, i have a raw dump of RTP packets and want to convert it to .h264, can it be done via ffmpeg or needs some development ? [16:33:58 CEST] <KGB> [13FFV1] 15dericed opened pull request #116: Review ietf cellar ffv1 02 secdir early xia 2018 06 01 (06master...06review-ietf-cellar-ffv1-02-secdir-early-xia-2018-06-01) 02https://git.io/vhCn0 [17:31:47 CEST] <jdarnley> WTF. This SIMD code I just wrote doesn't produce the same file but it does decode to the same *.out.rawvideo file. [17:32:37 CEST] <jdarnley> thanks fat [17:32:42 CEST] <jdarnley> *fate [19:03:18 CEST] <durandal_1707> development is pretty much dead [19:04:45 CEST] <kierank> durandal_1707: ? [19:06:20 CEST] <durandal_1707> kierank: ml activity, irc activity too [19:06:31 CEST] <atomnuker> its friday [19:07:48 CEST] <kierank> durandal_1707: there is all important http server in ffmpeg [19:07:51 CEST] <kierank> what are you talking about [19:07:54 CEST] <kierank> important work [19:12:49 CEST] <jamrial> durandal_1707: there's that dnn avfilter stuff, vulkan hwcontext, opencl filters, assorted mov stuff, and someone trying to rewrite ffserver [19:13:13 CEST] <jamrial> one thing is "there's no development", another is "current development doesn't seem interesting to me" [19:15:00 CEST] <jamrial> also this guy fixing all the issues in the vc1 decoder [19:16:11 CEST] <JEEB> jkqxz: the guy linked https://gitee.com/hwren/avs2_samples [19:16:48 CEST] <JEEB> seems like he uploaded them 6 hours ago, lol [19:23:59 CEST] <jkqxz> I saw those. They're all labelled as made with xavs2, so I imagine they were just created in response to your question. [19:24:17 CEST] <jkqxz> Not sure that's really very useful for testing anything, since I could do the same thing. [19:25:02 CEST] <JEEB> true that [19:25:16 CEST] <JEEB> I didn't check that much but I kind of wondered if that was exactly what he did :P [20:11:30 CEST] <Compn> anyone want to work on a broadcast transcription workflow for a US company? they sent mail directly to me heh [21:05:59 CEST] <durandal_1707> jamrial: that can be hardly called an serious developement [21:06:31 CEST] <klaxa> thanks, i guess [21:06:54 CEST] <klaxa> i'll be blunt, you're passive agressiveness doesn't feel right with me [21:06:57 CEST] <JEEB> don't take anything he says too heavily, he seems to try to be hfunny [21:07:00 CEST] <JEEB> yes [21:07:21 CEST] <JEEB> I like what durandal codes (generally), but the whole "trying to be funny on the internet" thing isn't working [21:07:30 CEST] <JEEB> it's just rude and generally not funny [21:07:31 CEST] <nevcairiel> i dont consider all his filters useful either, but i dont go around insulting him for it [21:07:36 CEST] <jamrial> klaxa: ignore him. he tries to be funny but it just rubs people the wrong way [21:07:51 CEST] <JEEB> nevcairiel: exactly [21:08:07 CEST] <jamrial> durandal_1707: really, no need to be a dick. joking or not [21:08:17 CEST] <JEEB> ^ [21:08:27 CEST] <klaxa> well seeing the rather long thread about CoC shit, this is kind of what lead to this development morale, everyone says "just ignore it" [21:08:32 CEST] <klaxa> i mean i don't disagree [21:09:22 CEST] <klaxa> but it still doesn't sit right with me, i thought we all want to create great code, no need to be unfriendly [21:10:37 CEST] <durandal_1707> where is atrac9 ? improved fft & mdct ? where is Dolby Vision, ProRes RAW, etc [21:11:41 CEST] <atomnuker> its coming [21:12:25 CEST] <kierank> durandal_1707: https://filmplusgear.com/prores-raw-testfile3 [21:12:50 CEST] <durandal_1707> and binary decoder? [21:13:11 CEST] <kierank> needs final cut pro x [21:13:13 CEST] <kierank> I guess, or a mac [21:13:16 CEST] <kierank> I don't have a mac [21:13:33 CEST] <cone-194> ffmpeg 03Gyan Doshi 07master:841c1efc78ac: lavc/libx265: allow users to set closed GOP via generic lavc flag [21:13:40 CEST] <durandal_1707> i should stfu or will get to jail [21:16:56 CEST] <kierank> durandal_1707: ? [21:18:20 CEST] <JEEB> durandal_1707: dolby vision has some white papers linked but no spec AFAIK [21:18:34 CEST] <JEEB> also which profile of dolby vision? [21:18:43 CEST] <durandal_1707> kierank: it is bad idea to mess with rotten fruitTM [21:19:46 CEST] <JEEB> oh, prores raw now has test files? lol [21:20:16 CEST] <kierank> durandal_1707: can you help my student with frame threading [21:20:20 CEST] <durandal_1707> JEEB: dunno, i more like to work on obsolete stuff, so nobody gonna sue me, shit I did not obsolete stuff recently... [21:20:21 CEST] <kierank> how he implement it for IP files [21:22:54 CEST] <durandal_1707> kierank: i'm not expert of that field - threads synchronization, i even failed/abandoned to write multithreaded flac encoder - which should be trivial - just write trail separately [21:25:18 CEST] <durandal_1707> guess, if h264 can have frame threading cfhd can too (for P frames?) [21:27:00 CEST] <durandal_1707> kierank: cfhd have P frames or? [21:27:04 CEST] <kierank> yes [21:32:30 CEST] <durandal_1707> kierank: ff_thread_report_progress() ff_thread_await_progress() this are calls for synchronization between different threads, (A)PNG is i think simplest decoder that use them [21:56:13 CEST] <Chloe> mmh, I've been pretty unwell and unable to check mailing list. But I agree with wm4. [21:57:35 CEST] <Chloe> I think the CoC (or what's there already) should be entirely removed (at least at first), as it is unenforced and just creates confusion. [22:01:04 CEST] <durandal_1707> first get well [22:02:51 CEST] <Chloe> durandal_1707: unlikely to happen for several months I think (at least two). [22:04:56 CEST] <durandal_1707> :( [00:00:00 CEST] --- Sat Jun 2 2018

1 0

ffmpeg.log.20180601
by burek 02 Jun '18

02 Jun '18

[00:00:17 CEST] <gcl_5> how do i disable warnings and let to pass progress info? [00:01:25 CEST] <furq> gcl_5: -v error -stats [00:02:29 CEST] <gcl_5> thxs furq [01:51:21 CEST] <NasKar> can you automate options like lame when installing from ports with a command like "make -C /usr/ports/multimedia/ffmpeg clean install BATCH=yes" ? [01:54:11 CEST] <furq> automate how [01:54:27 CEST] <NasKar> with a batch script [01:54:58 CEST] <furq> what bit do you want to automate though [01:55:34 CEST] <NasKar> lame, smb, ass, opus and x265 [01:56:21 CEST] <NasKar> the command I referenced will install ffmpeg from ports but won't choose the options I want. [03:20:39 CEST] <bashprogfortysix> sup guys pour a little cherry wine how come these ffmpeg recordings with ffmpeg -video_size 1024x768 -framerate 25 -f x11grab -i :0.0+100,200 -f pulse -ac 2 -i default output.mkv https://trac.ffmpeg.org/wiki/Capture/Desktop are no good. just huge files, though i gues my opion is to find a better output file condition and or just convert the output file with bitrate plus.... window size, maybe find coordinates [03:20:41 CEST] <bashprogfortysix> to square out..... [03:22:38 CEST] <bashprogfortysix> oh i just noticed the slower preset in that link so maybe ill look that way first [03:25:52 CEST] <bashprogfortysix> looks like that smaller preset nails it since lossless is still lossless lol! [05:04:27 CEST] <sirjaymz> As TrueHD is in experimental, CLI came back and indicated '-strict -2' are necessary to use.. after specifying, the CLI window indicates something along the lines of 'mono and stereo are currently only usable ' or something along those lines..is there a schedule I can follow on expected feature complete timeframe? [05:10:06 CEST] <zumba_addict> Is it possible to move each frame 5px to the right and rotate each frame maybe 0.1 degree? [08:18:02 CEST] <bashprogfortysix> ffmpeg presets are pretty interesting [08:18:12 CEST] <bashprogfortysix> thought the super slow does take awhile [09:09:29 CEST] <bashprogfortysix> hey does any one have much experience with recording av to usb vhs from the vcr , i got a sample but find that in particular to watching the vhs in realtime on my desktop [09:10:14 CEST] <bashprogfortysix> https://pastebin.com/g8AzuKwy [09:10:34 CEST] <bashprogfortysix> this isnt working to where maybe its how actually these vhs' are playing though and [09:11:06 CEST] <bashprogfortysix> i was watching and the audio was way lagging not bad but recording this way mplayer plays the output files much worse than ffplay [09:11:45 CEST] <bashprogfortysix> i guess haha [09:12:28 CEST] <bashprogfortysix> though if i get on it i expect to get the recording exactly as the vhsss are played importantly [09:14:44 CEST] <bashprogfortysix> thought to let you know how to play the vhs youd be familiar :) just from /dev/video and arecord - | aplay - [09:15:28 CEST] <bashprogfortysix> i also discovered my system had gst-launch-1.0 and started using that with it also so thats how my setup is doin it... [11:05:55 CEST] <Zexaron> Hello [11:06:56 CEST] <Zexaron> how do I check for the shared DLL version that gets loaded, I thought checking with defines but then I realized those are defined inside the source [11:07:27 CEST] <Zexaron> or, the source doesn't get used in the program, only the DLLs, so it would work ? [11:10:35 CEST] <BtbN> avcodec_version, avfilter_version, ... [11:38:23 CEST] <Zexaron> BtbN: well I did it with minor and micro, major is already tied to the filename, higher major isn't problem but want to strictly avoid anything lower [12:29:31 CEST] <BtbN> that's exactly what those functions are for. [12:42:21 CEST] <Zexaron> ok [14:33:28 CEST] <HarryFox> Hi guys i'm new here, so i begin you be patient a little :P [14:35:29 CEST] <HarryFox> I have a task. There is a movie splited by 3 or more parts, and i need to stream it in real time as a single movie with single timeline, and only what i want to know, is it possible at all with ffmpeg? [14:37:09 CEST] <DHE> concatenate? yes. there's a few ways to do it. search the wiki for concatenate [14:43:09 CEST] <HarryFox> Not exactly concat. I need to stream it in real time while it downloading. But headers(or metadata, sorry for illiteracy, and for my English too) must be as it a full single movie. [14:44:20 CEST] <Mavrik> You're describing concatenation :)~ [14:45:06 CEST] <DHE> concatenating from network sources can work, though I'm concerned that the second, third and any others may timeout while the first is playing [14:45:24 CEST] <DHE> unless "network sources" is something simple like a samba share [14:50:37 CEST] <HarryFox> And I could send headers(metadata) for the stream as if the stream a full movie with single timeline? It matters because of subtitles [14:50:54 CEST] <HarryFox> If it so, it's great news for me! [14:52:11 CEST] <Mavrik> Well, just use a right container. [14:52:18 CEST] <Mavrik> E.g. MPEG-TS doesn't have any headers [14:52:27 CEST] <Mavrik> Because it's meant for broadcast TV [14:52:34 CEST] <Mavrik> So you just start sending it and concatenating it [14:53:41 CEST] <DHE> timestamps are a container thing and you'll be generating the container as you go... if not outright transcoding the video/audio/subtitle streams [14:54:06 CEST] <HarryFox> What I've done for this time it's trying to split every part of a movie to smaller parts as .ts files, made for it approximate m3u8, but it works ugly because every splitted part of video not exactly 10 second duration [14:56:36 CEST] <HarryFox> So it's great news. It importants to me to know that it possible to do. Don't want to waste time for nothing :P [14:56:46 CEST] <HarryFox> Thank you so much guys! [14:57:36 CEST] <HarryFox> Will try to find solution in google (but something tells me that i will come back :c) [15:07:33 CEST] <friki_> Hi! I can see at changelog of ffmpeg4.0 that ffserver is removed. Is there a replacement? I'm trying to serve a http live streaming (not HLS to avoid some delays). Is still possible? [15:09:42 CEST] <DHE> use an external app that's better suited. the nginx-rtmp module (on top of nginx obviously) is popular [15:10:08 CEST] <DHE> alternatively ffmpeg has an HLS output muxer which you can just write to disk in a directory served by a static content HTTP server (apache, nginx, whatever) [15:10:31 CEST] <DHE> ... but it's http live streaming literally what HLS stands for? [15:10:53 CEST] <friki_> any recomendation to reduce the delay as possible? [15:12:06 CEST] <friki_> DHE: by "http live streaming" I mean an mp4 direct download. I don't know what's the common name for this type of streamings [15:12:07 CEST] <DHE> HTTP is just not friendly in that regard. you may be best suited with a streaming protocol directly. like rtmp [15:12:12 CEST] <furq> there's no good way of streaming to browsers without delay [15:12:39 CEST] <friki_> furq: yap, it seems to be right [15:12:46 CEST] <furq> if you don't care about browsers then nginx-rtmp is fine [15:12:47 CEST] <DHE> when you say "delay" what do you mean? time from click to video starts playing? or it's a live stream and you want the delay between the camera and the user screen to be as short as possible? [15:13:46 CEST] <friki_> delay between original stream is generated and displayed at the browser [15:14:21 CEST] <friki_> "delay between the camera and the user screen to be as short as possible?" that's it [15:14:45 CEST] <DHE> yeah. ditch the browser. use RTMP... [15:15:45 CEST] <DHE> of course that means you'll need a player app that can do that. VLC maybe? [15:16:20 CEST] <furq> vlc and mpv both work fine [15:16:25 CEST] <furq> preferably mpv for obvious reasons [15:17:20 CEST] <friki_> HLS with short segments... i think it's the best applicable option [15:17:52 CEST] <furq> if you need browser support then you've no other choice really [15:17:54 CEST] <friki_> i want to watch the video inside a web browser and avoid flash [15:18:04 CEST] <furq> you're stuck with hls and dash [15:18:05 CEST] <furq> and hls is simpler [15:18:13 CEST] <DHE> even if a segment is 1 second (and I don't recommend that) it's still a good 3-4 seconds of latency on users. [15:18:27 CEST] <furq> yeah you're looking at at least 10 seconds latency for hls [15:18:34 CEST] <DHE> and that's not getting into encoder latency [15:18:39 CEST] <friki_> furq: some years ago there are more options based over http [15:18:46 CEST] <furq> encoder latency is pretty trivial compared to hls latency [15:18:50 CEST] <furq> and other buffers and whatnot [15:19:05 CEST] <DHE> yes. they still exist, but they are not realtime. they're not suitable for videoconferencing or whatever you're doing [15:19:11 CEST] <furq> don't trip my "someone is enabling -tune zerolatency to send a stream to twitch" alarm [15:19:23 CEST] <furq> my delicate heart can't take it [15:19:27 CEST] <DHE> hahaha [15:19:37 CEST] <friki_> DHE: how can i search more info about this options? :-) [15:19:48 CEST] <DHE> which options? [15:19:58 CEST] <furq> the only other option i know of that works in all browsers is webrtc [15:20:11 CEST] <furq> and ffmpeg has nothing that will help you with that [15:20:13 CEST] <friki_> i want to explore all http options [15:20:24 CEST] <furq> and also that restricts you to baseline h264 [15:20:28 CEST] <friki_> I already know about HLS and DASH (based on segment files) [15:20:40 CEST] <furq> there isn't really anything else [15:20:49 CEST] <JEEB> as soon as you enter browsers, you're fscked regarding HTTP [15:20:55 CEST] <JEEB> because XHR will buffer the whole thing [15:21:02 CEST] <furq> browsers will only play mp4, you generally can't livestream mp4, and i don't think fmp4 works outside of hls/dash [15:21:14 CEST] <furq> browsers meaning all browsers [15:21:22 CEST] <friki_> can't do a "direct download" live streaming? [15:21:27 CEST] <furq> not to browsers [15:21:39 CEST] <JEEB> that's why the idiots keep working around it with trying to stream with websockets instead since that's implemented with streaming apis [15:21:44 CEST] <furq> not for live streams anyway [15:21:47 CEST] <furq> obviously you can do that for vod [15:21:51 CEST] <JEEB> s/with trying/by trying/ [15:22:01 CEST] <friki_> i mean, http://myserver.com/live.mp4 [15:22:01 CEST] <JEEB> and lower latency stuff, yes, webrtc [15:22:16 CEST] <furq> friki_: that doesn't work for live streaming for a variety of reasons [15:22:25 CEST] <JEEB> friki_: if some browser supports fragmented isobmff it might work with a video tag [15:22:26 CEST] <DHE> .mp4 works fine for pre-rendered videos. after the fact streaming, youtube style [15:22:28 CEST] <JEEB> most likely not though [15:22:40 CEST] <JEEB> but if you have the time and the way to serve that, feel free [15:22:58 CEST] <friki_> .mp4 .mov .mkv (i don't mind the file format) [15:23:05 CEST] <furq> well browsers will only play mp4 and webm [15:23:17 CEST] <furq> and webm restricts you to vp9 [15:23:23 CEST] <furq> and you really don't want to use that for livestreaming [15:27:07 CEST] <friki_> take a look at this live audio stream: http://rac1.radiocat.net/ [15:27:20 CEST] <furq> yeah audio is simpler [15:27:30 CEST] <friki_> can't do it with video? [15:27:41 CEST] <furq> icecast will serve webm [15:27:46 CEST] <furq> i can't vouch for how well it works [15:27:58 CEST] <furq> the problem with webm is it means you have to use libvpx which is incredibly slow [15:36:52 CEST] <jphelps> Hello. I have a commercial DVD which has an A/V sync problem with one of its audio tracks. When played with mplayer using the -mc option (mplayer -mc 0 dvd://1), the a/v problem dissappears. I am trying to generate a valid movie file without transcoding that does nto have this problem. [15:36:58 CEST] <jphelps> Any suggestions? [15:41:19 CEST] <relaxed> does it happen at the start? [15:44:09 CEST] <jphelps> No, it looks like a progressive shift [15:44:18 CEST] <jphelps> It is not very bad in any case [16:05:24 CEST] <chrysalis> is there a practical way to identify and cut/merge videos at keyframes, because from what i understand using -ss breaks seeking? [16:08:06 CEST] <relaxed> chrysalis: have you seen https://trac.ffmpeg.org/wiki/Seeking ? [16:10:56 CEST] <relaxed> you mean without transcoding? [16:11:23 CEST] <chrysalis> yes, copy [16:13:42 CEST] <relaxed> how does it break seeking? [16:15:51 CEST] <chrysalis> i dont know, im confused, i just need to know the proper way to chop up a video with ffmpeg and replicate what i accomplish with bandicut using keyframes [16:17:50 CEST] <sbraz> chrysalis: if you -c:v copy, it won't cut at -ss but at the nearest keyframe i think [16:18:20 CEST] <sbraz> maybe not the "nearest", but the next keyframe or something, anyway it will properly cut on keyframe [16:19:21 CEST] <anill> Hi FFMPEG people, i have a raw dump of RTP packets and want to convert it to .h264, can it be done via ffmpeg? [16:19:39 CEST] <chrysalis> yes, but as i understand it it matters if you put -ss before or after -i file [16:19:52 CEST] <chrysalis> this is whats confusing me [16:20:18 CEST] <anill> chrysalis: is it for me? [16:21:22 CEST] <sbraz> chrysalis: if you copy it doesn't matter i think [16:21:39 CEST] <sbraz> it matters if you transcode, then if you put -ss after, it will do an exact seek and start transcoding there [16:21:55 CEST] <chrysalis> it might cut at the keyframe, but itll play from the -ss position or sometghing like that [16:24:18 CEST] <chrysalis> im not transcoding im copying, so you are saying it doesnt matter in this case? [16:34:50 CEST] <furq> iirc it depends on the demuxer whether or not it'll cut properly at the nearest keyframe if you also have audio [16:35:04 CEST] <furq> obviously the video stream will always cut at the nearest keyframe, but it may or may not seek the audio to that point before cutting [16:57:17 CEST] <Satao> Hi all! I'm using the dashenc output to file, but it's writing to a temp file. Is it possible to change this behaviour? [16:57:53 CEST] <Satao> or better... writing to a temp file and then moves/renames the file [17:04:42 CEST] <TAFB_WERK> ffplay plays my ip camera perfect, ffmpeg does not. Can I stream data from ffplay to ffmpeg? :) [17:08:07 CEST] <DHE> that's a surprise, unless the ffmpeg output has some kind of limitation that causes it to sputter. the camera does expect realtime streaming. [17:08:26 CEST] <DHE> or a version mismatch? [17:12:23 CEST] <kepstin> TAFB_WERK: ffmpeg isn't a player, I'm not sure what you mean by "plays my ip camera perfect"? [17:13:19 CEST] <kepstin> i mean, you can sort of play video with ffmpeg by using an output format that shows stuff on the screen, but there's no attempt to time frames correctly there, so... [17:17:36 CEST] <TAFB_WERK> kepstin: how can you say ffmpeg isn't a play? -c copy to "display" = player :) [17:17:58 CEST] <TAFB_WERK> DHE: ffmpeg recording sputters, ffplay is smooth [17:18:08 CEST] <Hello71> ffmpeg buffers [17:18:20 CEST] <DHE> he just said ffmpeg isn't a player because, among other things, it doesn't bother with timing and stuff. it wants to transfer data in and out as quick as possible [17:18:57 CEST] <kepstin> if you want to play something, use a tool designed for that, which will among other things correctly handle displaying frames at the correct time. [17:19:18 CEST] <DHE> and sync the audio properly [17:19:19 CEST] <TAFB_WERK> i'm trying to live stream my crappy ip camera to youtube, skips, studders, etc. looks fine in ffplay [17:19:31 CEST] <TAFB_WERK> the stream has no audio [17:19:39 CEST] <TAFB_WERK> I have to inject blank audio to get youtube to stream it [17:19:50 CEST] <kepstin> oh, so you're *not* using ffmpeg as a player, you're using it as transcoder and network streaming tool. [17:20:02 CEST] <TAFB_WERK> ffmpeg -rtsp_transport tcp -allowed_media_types video -i "rtsp://admin:admin@192.168.0.65:554/0" -c copy -f h264 - | ffmpeg -f h264 -r 30 -i - -f lavfi -i anullsrc -b:a 2k -c:a aac -c:v copy -f flv "rtmp://a.rtmp.youtube.com/live2 [17:20:04 CEST] <kepstin> now we're starting to know enough to be able to help you :) [17:20:21 CEST] <DHE> ... wat? why are you piping ffmpeg into ffmpeg? [17:20:50 CEST] <TAFB_WERK> someone here gave me the command, worked fine with my old camera, not so much with my new one. [17:20:52 CEST] <furq> changing the framerate i guess [17:20:54 CEST] <kepstin> TAFB_WERK: that command will discard frame timing and cause all sort of problems (you're saying the stream is 30fps, but if the camera isn't actually doing 30fps it'll cause issues) [17:21:12 CEST] <furq> but yeah that's probably going to work badly with a live input [17:21:36 CEST] <furq> MAYBE that's why it's stuttering [17:22:23 CEST] <TAFB_WERK> kepstin: the camera has a variable framerate :( i know :( [17:23:09 CEST] <furq> doesn't flv support vfr [17:23:20 CEST] <furq> actually nvm i bet youtube doesn't [17:23:34 CEST] <TAFB_WERK> youtube for sure doesn't [17:23:39 CEST] <kepstin> furq: yeah, it does, but the -r30 in that command makes it cfr, so... [17:23:46 CEST] <furq> well yeah i'm saying he could skip that step [17:24:01 CEST] <DHE> but only in the second half. the first half isn't forcing a framerate. [17:24:06 CEST] <kepstin> a single ffmpeg command (no pipe), optionally with an fps filter to make it cfr, should work fine [17:24:08 CEST] <furq> although if you want to make it cfr then might i suggest -vf fps=30 [17:24:26 CEST] <DHE> besides, variable FPS coercion and stream copy to raw h264 bitstream are not compatible [17:24:36 CEST] <TAFB_WERK> i'll give that try, thx. [17:24:37 CEST] <furq> yeah [17:25:54 CEST] <kepstin> i'd expect yt live stuff should handle vfr stuff ok? they allow streaming from phones and such, iirc. [17:26:18 CEST] <HarryFox> Few I found the answer on previous question, but result is unsatisfactory for me. Sad but concat of [17:26:32 CEST] <DHE> sure, but he's doing full h264 stream copy the whole way through and discarding timing information in the pipe [17:26:37 CEST] <HarryFox> mp4 streams it's too expensive by comp resources [17:31:50 CEST] <manuelschneid3r> hi [17:32:03 CEST] <manuelschneid3r> how can I avoid metadata to be removed by ffmpeg [17:32:03 CEST] <TAFB_WERK> https://www.youtube.com/watch?v=dCBM91HT0Ck [17:32:20 CEST] <TAFB_WERK> seems to be streaming pretty smooth with -vf fps=30 [17:33:59 CEST] <Hello71> what's the modern uncompressed or minimally (lossless) compressed video format? [17:34:47 CEST] <Hello71> yuv4mpegpipe? [17:35:26 CEST] <DHE> a number of options depending on CPU and size preferences. h264 with -qp 0 (colourspace permitting), ffv1, huffyuv are all popular choices [17:37:29 CEST] <furq> Hello71: y4m is a container [17:37:59 CEST] <HarryFox> Guys I hope you can help me with another question: I split a video to 15 sec .ts files, but duration of the files not direcly 15 sec and it brings troubles for m3u generation. So question is how can i get every file exact duration? [17:38:10 CEST] <furq> rawvideo is uncompressed, ffvhuff is the fastest lossless codec (but nothing other than libavcodec supports it) [17:38:16 CEST] <Hello71> doesn't it only hold uncompressed yuv [17:38:20 CEST] <furq> yes [17:38:25 CEST] <Hello71> y4m that is [17:38:28 CEST] <furq> yes [17:38:55 CEST] <furq> ffvhuff is fast enough that there's no point not using it [17:39:07 CEST] <furq> but it obviously doesn't compress very well [17:39:11 CEST] <Hello71> what do I put it in? [17:39:23 CEST] <furq> mkv or nut are fine [17:39:55 CEST] <Hello71> I think by default -c:v libx264 -f mp4 - | ffmpeg won't work [17:40:44 CEST] <furq> it sure won't [17:41:13 CEST] <furq> Hello71: ffmpeg -codecs | grep EV..S [17:41:17 CEST] <furq> will list every lossless video encoder [17:41:34 CEST] <furq> HarryFox: if you're splitting without reencoding then you're constrained by keyframes [17:42:05 CEST] <furq> there's no solution other than reencoding or just not caring that your segments aren't all the same duration [17:42:12 CEST] <furq> i would go with the latter because it doesn't matter that much in my experience [17:43:34 CEST] <Hello71> huh, are LS new? I could swear it used to be fewer characters [17:45:03 CEST] <Hello71> also it's weird that h264 is marked S but hevc isn't [17:50:21 CEST] <Hello71> oh cool, ffserver might be getting unkilled [18:06:53 CEST] <anill> Hi FFMPEG people, i have a raw dump of RTP packets and want to convert it to .h264, can it be done via ffmpeg? [18:11:03 CEST] <BtbN> write a server that replay them? [18:27:43 CEST] <HarryFox> furq: in my case it's very important to know exact duration, becaouse the creation of playlist file dependce on it. [18:28:39 CEST] <HarryFox> Anyway now I'm trying to use ffprobe and catch em stdout with python [18:31:09 CEST] <ChocolateArmpits> HarryFox, you can ask ffprobe for specific properties rather than parsing all of the output [18:33:42 CEST] <HarryFox> yeah i know but somewhat it doesn't work at 100% with python subprocess, and stdout is None [18:34:10 CEST] <HarryFox> but I think I found solution. Working on it [18:36:02 CEST] <Hello71> unless you're sure what you're doing you should use check_call [18:47:14 CEST] <HarryFox> Hello71: yeah it's much more convenient than Popen, thanks [18:50:11 CEST] <Hello71> well no shit, https://docs.python.org/3/library/subprocess.html: "The recommended approach to invoking subprocesses is to use the run() function for all use cases it can handle. For more advanced use cases, the underlying Popen interface can be used directly." [19:01:46 CEST] <HarryFox> Hello71: run() has been added with python 3.5 but in my case need python2.7 code compatible. It's not option for me [19:02:21 CEST] <Hello71> if you want to capture the output then I guess it doesn't work [19:04:52 CEST] <HarryFox> check_call it doesn't, but check_output is perfect [20:14:01 CEST] <galassi_> Hiya guys, I'm trying to figure out why ffmpeg seems to possibly hang upon creating/opening a file upon attempting to stream from screen. [20:15:16 CEST] <Durandal> Maybe it's not getting any data from the source? [20:15:29 CEST] <galassi_> Possible, should I post the log? [20:15:30 CEST] <Durandal> What is your commend line [20:16:07 CEST] <galassi_> ffmpeg -loglevel 'debug' -report -nostdin -video_size 1024x768 -framerate 25 -t 00:00:10 -f x11grab -i :0.0+100,200 output.mp4 [20:16:42 CEST] <galassi_> it seems to stop recording the log at Successfully parsed a group of options. [20:16:42 CEST] <galassi_> Opening an input file: :0.0+100,200. [20:16:54 CEST] <galassi_> then it just hangs [20:17:03 CEST] <galassi_> no crashes, and no output file is created [20:17:29 CEST] <Durandal> Don't -f and -I need to come first? [20:17:35 CEST] <galassi_> I have to ctrl c to stop it running [20:17:41 CEST] <galassi_> will try that [20:19:07 CEST] <Durandal> If you intend to output only 10 seconds of capture then -t needs to go after -i [20:21:43 CEST] <galassi_> what would that command look like? I'm getting no such file or directory atm [20:23:08 CEST] <galassi_> I just want to test and see if ffmpeg is able to run on this computer. I was given the task "Investigate why FFMPEG Isn't Working on Rushmore", and that is the extent of my knowledge at this point. [20:29:28 CEST] <galassi_> it seems to hang at the same point when performing a grab with -i at this position: [20:29:29 CEST] <galassi_> ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg -loglevel 'debug' [20:31:33 CEST] <Durandal> this is a similar command line that I use on windows [20:32:12 CEST] <Durandal> ffmpeg -f gdigrab -i desktop -framerate 30 -t 10 "out.mp4" [20:32:40 CEST] <DHE> order of args does matter. options before "-i" are configuration options for the input. otherwise they are configuration options for the output [20:32:50 CEST] <Durandal> The general structure should be ffmpeg [GLOBAL PARAMS] [INPUT PARAMETERS] [INPUTS] [OUTPUT PARAMETERS] [OUTPUT] [20:32:53 CEST] <Durandal> yeah [20:33:40 CEST] <Durandal> ffmpeg -loglevel 'debug' -f x11grab -i :0.0 -video_size cif -framerate 25 /tmp/out.mpg [20:34:23 CEST] <DHE> I would use -s instead of video_size for the input, and make sure you set it on the left side of -i [20:34:42 CEST] <DHE> oh wait, I'm looking at the wrong options. ignore that [20:36:17 CEST] <galassi_> yeah still hanging on opening an input file with: ffmpeg -loglevel 'debug' -f x11grab -i :0.0 -video_size cif -framerate 25 /tmp/out.mpg [20:36:33 CEST] <galassi_> should I be looking at x11grab? [20:36:35 CEST] <Durandal> ffmpeg -loglevel 'debug' -f x11grab -i :0.0 -framerate 5 -preset ultrafast out.mp4 [20:36:44 CEST] <Durandal> works for me on ubuntu [20:36:56 CEST] <Durandal> do you need to demonstrate the x11grab works, or just ffmpeg in general? [20:37:00 CEST] <DHE> framerate is an x11grab option, put it on the left side [20:37:02 CEST] <kepstin> '-framerate' is also an input option, but yeah - sounds like the issue is in the connection to X to do the screen grabbing [20:38:15 CEST] <galassi_> oh I am being told now, it's a setting in x2go that is the issue. [20:38:45 CEST] <galassi_> there is a bug here related to it being disabled, but it has been addressed. [20:39:55 CEST] <galassi_> just didn't get relayed to me until now. [20:39:55 CEST] <DHE> so it's not really an X11 server? [20:40:06 CEST] <galassi_> I guess not. [20:41:11 CEST] <galassi_> or at least something in x2go is disabled so that x11grab will not work. [20:42:13 CEST] <galassi_> Upon my boss seeing this chat screen, he immediately apologized about assigning this task so I guess it's all "sorted" atm. [20:42:35 CEST] <galassi_> The mystery remains for now. [20:42:41 CEST] <galassi_> ty for your help though guys [21:32:10 CEST] <Xogium> hi :) I've downloaded a bunch of sounds which claimed to be stereo and 5.0 channels both, but I only got the 5.0 version. Is there any easy way to get these to stereo without too much tweaking and such ? [21:32:34 CEST] <Xogium> note that I can't even play anything else than stereo here [21:32:57 CEST] <c_14> ffmpeg -i file -ac 2 stereofile [21:33:06 CEST] <c_14> might need codec/bitrate settings [21:33:19 CEST] <Xogium> won't that cause an unbalanced sound ? [21:33:25 CEST] <c_14> how so? [21:33:29 CEST] <c_14> should do proper downmixing [21:34:10 CEST] <Xogium> I'm not sure, it felt like I was missing 1 or 2 channels from the original [21:34:22 CEST] <Xogium> a weird kind of flanger effect too [21:34:29 CEST] <c_14> is the 5.0 version correctly tagged? [21:34:34 CEST] <c_14> channels all in the right places? [21:34:47 CEST] <Xogium> it should be [21:34:55 CEST] <Xogium> how can I verify this to make sure ? [21:37:29 CEST] <c_14> first check that the layout listed by ffprobe/ffmpeg is the one you're expecting. should list 5.0 [21:37:42 CEST] <c_14> but if they're incorrectly labeled you'd have to listen to each channel on it's own [21:37:50 CEST] <c_14> or remove channels to find out which ones they actually are [21:39:55 CEST] <Xogium> heh [21:40:16 CEST] <Xogium> the rest of the files appears to be labled right, just this one that wasn't [21:40:50 CEST] <Xogium> I definitely need to report this to the guys [21:42:04 CEST] <Xogium> thank you c_14 :) [21:42:17 CEST] <DHE> depending on the file format, you might be able to probe the file to see if it has multiple audio streams, and select which you want [21:42:28 CEST] <Xogium> yeah that's wave [21:42:32 CEST] <DHE> oh... [21:45:14 CEST] <Xogium> I was going crazy with that file and tried answer from stackoverflow& They started to downmix in a weird way and using a filter and all, and I was like, surely downmixing properly can't be so complicated [21:49:45 CEST] <DHE> "properly" can be is relative. how to mix 5 speakers down to 2 without clipping while still maintaining the right volumes for each speaker? [21:58:39 CEST] <Xogium> DHE: yeah [21:59:13 CEST] <Xogium> but I mean, for me setting to 2 channels was enough [23:20:00 CEST] <Ars> Hi guys [23:20:00 CEST] <Ars> I am transcoding live stream to mpeg2 [23:20:00 CEST] <Ars> On VLC its perfect, but hardware doesn't accept stream an shows: PCR repetition errors [23:20:31 CEST] <Ars> how fox PCR errors [23:20:35 CEST] <Ars> fix [23:21:40 CEST] <JEEB> Ars: the mpegts muxer should in theory generate proper PCR, how are your timestamps and what is your FFmpeg version? [23:22:08 CEST] <JEEB> I've had issues with UDP multicast and really cringily strict hardware receivers, but not PCR so far [23:23:26 CEST] <Ars> JEEB, ffmpeg version 4.0-static [23:24:37 CEST] <Ars> i am using multicast [23:24:45 CEST] <Ars> UDP [23:30:37 CEST] <DHE> the mpegts muxer works best and is most compliant when run in CBR muxing mode. set -muxrate to a number that you will definitely never exceed (including the overhead of mpegts itself) [23:30:57 CEST] <DHE> typically add your peak bitrates from all streams, add 10%, and you have a good estimate for the muxrate [23:39:46 CEST] <Ars> DHE, thank you i will try [00:00:00 CEST] --- Sat Jun 2 2018

1 0

ffmpeg-devel.log.20180531
by burek 01 Jun '18

01 Jun '18

[02:31:53 CEST] <atomnuker> jkqxz: could you test again with my branch? [12:40:09 CEST] <jdarnley> Dammit! You were working before. [12:41:21 CEST] <jdarnley> Why do you fail when I changes another part of the code? [12:41:43 CEST] <jdarnley> And how do you in avio_close? [12:41:51 CEST] <jdarnley> *fail in [12:48:25 CEST] <kierank> gagandeep: can you send me screenshot of mountain frame [12:48:28 CEST] <gagandeep> kierank: how do i check whether the 2nd frame is working, you have any ideas [12:48:40 CEST] <kierank> does it look bad? [12:48:45 CEST] <gagandeep> yeah, sure, didnt see your message [12:48:58 CEST] <atomnuker> jkqxz: replied to your email with a new patch to fix stuff [12:49:14 CEST] <atomnuker> and also add vulkan->vaapi mapping support (tested, it works!) [12:50:12 CEST] <gagandeep> kierank:https://imgur.com/a/zk5iRU1 [12:50:13 CEST] <atomnuker> yep, shader based gpu filtering without any copying is here [12:50:32 CEST] <kierank> gagandeep: cool, second frame bad I guess? [12:50:38 CEST] <gagandeep> i can't make it out, as testcfhd is problematic with these samples [12:50:47 CEST] <gagandeep> i don't know if i am seeing second frame [12:51:08 CEST] <gagandeep> like some metric built in ffmpeg to check [12:51:36 CEST] <gagandeep> i allocated an av_frame and referenced it's data pointers to decoded data [12:52:06 CEST] <gagandeep> then when p frame is required as sample type = 1, i av_frame_ref(pic, my saved frame) [12:52:31 CEST] <gagandeep> pic is the frame you had defined to store final output [12:54:06 CEST] <gagandeep> kierank: you see the screens, if you want i can upload video file for mountain [12:54:18 CEST] <gagandeep> transcoded ofcourse [12:57:18 CEST] <BtbN> I wonder if the MX150 has nvenc capabilities [12:57:23 CEST] <BtbN> and nvdec [13:02:46 CEST] <gagandeep> kierank: ? [13:03:00 CEST] <kierank> gagandeep: sorry on phone [13:03:06 CEST] <gagandeep> ok [13:04:30 CEST] <durandal_1707> kierank is on vacations all the time [13:06:40 CEST] <kierank> durandal_1707: lies [13:06:43 CEST] <kierank> gagandeep: ok back [13:07:00 CEST] <kierank> gagandeep: looks ok, no? [13:07:09 CEST] <kierank> there is cineform plugin for vlc [13:07:14 CEST] <kierank> which I used before and it worked [13:07:24 CEST] <gagandeep> ah, ok, will see with that [13:08:03 CEST] <gagandeep> i will send the patch and you can then comment on how better the things can be done, right [13:08:35 CEST] <kierank> ok [13:09:01 CEST] <gagandeep> k, working on cleaning up the code now [13:13:35 CEST] <kierank> gagandeep: btw you can get matrix [13:14:01 CEST] <kierank> seems to be what a lot of people use [13:14:02 CEST] <durandal_1707> ubitux: why you wrote ebur128 and not itur one? what is different? [13:14:04 CEST] <kierank> for irc bridging [13:14:09 CEST] <kierank> durandal_1707: international politics [13:14:35 CEST] Action: jdarnley is dumb and makes his own obvious bugs. [13:18:21 CEST] <gagandeep> ok [13:37:39 CEST] <ubitux> durandal_1707: ebur128 was kinda trending when i wrote it [13:38:12 CEST] <JEEB> also it seems to work? [13:39:43 CEST] <nevcairiel> arent both based on the same calculation anyway [14:08:28 CEST] <durandal_1707> nevcairiel: they are very similar [15:01:01 CEST] <ubitux> ah you're actually refering to the ITU paper that EBU-R128 is based on? [15:01:17 CEST] <nevcairiel> personally i dont even know any other [15:01:34 CEST] <ubitux> yeah well, i used that paper then [15:01:36 CEST] <nevcairiel> R128 was based on BS.1770, BS.1770-2 backported some changes R128 made [15:01:40 CEST] <ubitux> yep, that [15:01:43 CEST] <nevcairiel> so its really quite similar [16:00:00 CEST] <nevcairiel> wbs: in case you're interested i reported the last miscompilation i've been seeing https://developercommunity.visualstudio.com/content/problem/264519/miscompi… [16:08:20 CEST] <nevcairiel> all the other issues went away with 15.7 already [16:33:05 CEST] <cone-115> ffmpeg 03John Cox 07master:8d4fc2ea06a7: avfilter: use av_clip_uintp2_c where clip is variable [21:54:51 CEST] <cone-115> ffmpeg 03Gyan Doshi 07master:04b570817b2b: doc/ffmpeg: remove non-existent vdt option [22:00:44 CEST] <philipl> BtbN: wouldn't be surprised if one or the other was missing [22:16:15 CEST] <kierank> durandal_1707: do you know about bayer? [22:17:41 CEST] <durandal_1707> kierank: yes, he is nice guy, he is my first neighbour [22:18:04 CEST] <kierank> durandal_1707: on a serious point, would you be able to help gagandeep with bayer [22:18:10 CEST] <kierank> because I don't know anything about it [22:19:08 CEST] <durandal_1707> bayer is solo red/blue/green "pixel"? [22:19:26 CEST] <JEEB> some overlap thing I think? [22:19:47 CEST] <JEEB> https://en.wikipedia.org/wiki/Bayer_filter [22:20:41 CEST] <durandal_1707> i know even less, i never played with it, i only have i think some raw image files, i downloaded, which are very big and i think are bayer... [22:20:54 CEST] <JEEB> but yea, basically mosaic/demosaic [22:21:09 CEST] <JEEB> (not that I have poked it either, I only read about it on camera firmware forums) [22:21:35 CEST] <durandal_1707> sws supports what kind of bayer? [22:21:46 CEST] <kierank> who is bayer expert in ffmpeg? [22:21:47 CEST] <kierank> carl? [22:23:53 CEST] <durandal_1707> lol, sws supports lot of bayer, you can look at its code [22:24:16 CEST] <atomnuker> https://github.com/mpv-player/mpv/blob/master/video/img_format.c#L434 [22:24:21 CEST] <durandal_1707> do you know what kind of bayer is stored in cfhd? [22:25:13 CEST] <durandal_1707> atomnuker: lol [22:25:47 CEST] <jkqxz> atomnuker: Now radv fails a bit later: <https://0x0.st/s20-.txt>. [22:26:39 CEST] <JEEB> atomnuker: thank you for those short laughs [22:30:06 CEST] <jkqxz> (Currently waiting patiently for configure on Windows so I can investigate YUV planar formats there further.) [22:30:09 CEST] <kierank> durandal_1707: dunno [22:30:23 CEST] <kierank> durandal_1707: but i guess prores raw will have same problem [22:45:34 CEST] <atomnuker> jkqxz: hmm, seems like its using the host memory extension, hence why alloc mem has an extension [22:45:48 CEST] <atomnuker> I think if you disable it it should run [22:46:46 CEST] <atomnuker> to do that, change VK_EXT_EXTERNAL_MEMORY_HOST_EXTENSION_NAME to some string like "blah" [22:52:54 CEST] <atomnuker> I'll get an AMD card as soon as they release a new one, which will be after I move in a few months [22:53:17 CEST] <durandal_1707> atomnuker: where you gonna move? [22:53:38 CEST] <jkqxz> Right, that seems to work (output looks ok). [22:53:54 CEST] <jkqxz> Segfaults on close, though: <https://0x0.st/s20A.txt>. [23:04:11 CEST] <atomnuker> fixed, missing NULL check in the lavfi common deinit code [23:04:42 CEST] <atomnuker> (pushed to my github) [23:05:50 CEST] <atomnuker> could you try to run avgblur_vulkan on AMD? [23:08:44 CEST] <jkqxz> Looks pretty blurry to me. (And painfully slow because of the YUV->RGB, upload, blur, download, RGB->YUV.) [23:09:43 CEST] <jkqxz> Oh, the bottom edge of a 1080p video is broken. [23:18:40 CEST] <atomnuker> jkqxz: try https://pars.ee/temp/fix_exporting.diff, it should fix the host mapped extensioin [23:18:46 CEST] <atomnuker> *extension [23:24:02 CEST] <jkqxz> Same stacktrace as above. [23:32:15 CEST] <jkqxz> Intel/Windows on Skylake really doesn't have the planar YUV formats. [23:33:33 CEST] <atomnuker> https://vulkan.gpuinfo.org/ lists all of them as supporting it [23:33:52 CEST] <jkqxz> There are so many version numbers in this driver. I'm currently using the most recent released one from the Intel website, which is probably called either 15.45.23.4860 or maybe 21.20.16.4860. [23:35:42 CEST] <jkqxz> The other Windows machine I can easily check is a Haswell, so not useful because Intel decided that Windows users can't have Vulkan there. [23:37:16 CEST] <jkqxz> The version numbers on that website do not match either of the ones I gave above. Let me see if I can find a 0.xxx.yyyy version number somewhere... [23:44:24 CEST] <jkqxz> Can't find anything. Is there a driver version string available in the API that I could print somewhere? [23:52:54 CEST] <jamrial> jkqxz: vkGetPhysicalDeviceProperties() maybe? for VkPhysicalDeviceProperties->driverVersion [00:00:00 CEST] --- Fri Jun 1 2018

1 0

ffmpeg.log.20180531
by burek 01 Jun '18

01 Jun '18

[00:58:16 CEST] <slavanap> Hi! I have a list of segments in video I have to mute, could you suggest the best effective way to do it with ffmpeg. No video reencoding needed. [00:58:24 CEST] <zap0> when requesting a greyscale output, how are coloured inputs converted? [00:59:13 CEST] <slavanap> zap0, I guess it's converted to YUV, then U:=0, V:=0 [00:59:53 CEST] <zap0> slavanap, output an audio track, as WAV or some dumb fast format. edit it. make a new video by streamin in vision from original file and audio frmo edited file [01:00:31 CEST] <zap0> slavanap, ok.. makes sense if orogina is YUV.. what if it's RGB.. just a YUV convertin first internally? [01:01:02 CEST] <zap0> original/ [01:01:49 CEST] <slavanap> zap0, about "muting" I expected ffmpeg has already some internal filters to do it. [01:02:27 CEST] <slavanap> zap0, RBG -> YUV, then using just Y seems the most correct from scientific point of view. [01:02:33 CEST] <zap0> like there is a pix_fmt grey14 or grey16 etc.. is the internal represntation of the Y floating point? or bigger than 8? [01:02:59 CEST] <zap0> slavanap, muting: yeah i imagine there is a filter, but it would only do 1 span. [01:03:43 CEST] <zap0> on command line ffmpeg -filters [01:04:56 CEST] <zap0> ffmpeg -filters | grep volume [01:13:51 CEST] <zap0> slavanap https://stackoverflow.com/questions/29215197/mute-specified-sections-of-an-… [01:14:05 CEST] <slavanap> zap0, grep for RGB_TO_Y_CCIR -- that's for getting Luma (Y) from RGB. ffmpeg supports GRAY8 & GRAY16 formats -- that's greppable too. [01:15:02 CEST] <slavanap> I can't find out the exact placement of colorspace conversions now. [01:15:19 CEST] <slavanap> zap0, thanks for SO question & answer! [01:15:28 CEST] <zap0> i hope it works! [01:17:37 CEST] <slavanap> zap0, I just believe ffmpeg does something like RGB -> YUV -> Y -> RGB for grayscale output in RGB and input in RGB, preserving the color bitness. Moveover, may formats supports grayscale natively, so some of conversions might not be necessary [01:17:54 CEST] <slavanap> s/may/some/ [01:23:58 CEST] <furq> slavanap: -af "volume=0:enable=between(t\, 12\, 34)+between(t\, 56\, 78)" [01:24:21 CEST] <slavanap> Thanks [01:24:53 CEST] <slavanap> zap0, and I believe y -> RGB is just r=g=b=y, but notice there're different YUV formats. [01:24:56 CEST] <slavanap> good night [01:25:50 CEST] <zap0> :) [01:26:50 CEST] <voip_> hello guys [01:26:51 CEST] <voip_> I need take rtmp live stream and without transcoding video/audio push multicast UDP. [01:26:51 CEST] <voip_> Please help with right command [03:49:37 CEST] <Hello71> how does this command work: ffmpeg -ss 3 -i input.mp4 -vf "select=gt(scene\,0.4)" -frames:v 5 -vsync vfr -vf fps=fps=1/600 out%02d.jpg [03:50:30 CEST] <Hello71> find all frames matching scene > 0.4, take the first one every 10 minutes, and stop at 5 frames? [03:58:12 CEST] <furq> wrong order [03:58:44 CEST] <furq> specifying -vf twice will just override the first one [03:59:29 CEST] <furq> if it's supposed to be -vf select,fps then that'll pretty much do what you said [03:59:34 CEST] <furq> plus it'll skip the first 3 seconds [03:59:53 CEST] <furq> i suspect you actually want -vf fps,select though [04:01:01 CEST] <Hello71> I don't think that does anything [04:01:08 CEST] <Hello71> for most videos then select would do nothing [04:01:17 CEST] <Hello71> anyways I decided I actually want thumbnail filter [04:01:54 CEST] <furq> thumbnail buffers the entire selection [04:02:08 CEST] <furq> so you're probably going to run out of memory if you ask it to buffer 10 minutes of uncompressed frames [04:03:36 CEST] <Hello71> well I'm planning on using fps=5,thumbnail=100 or something [04:04:12 CEST] <furq> that'd help [04:04:18 CEST] <furq> -skip_frame nokey might help as well [04:04:29 CEST] <furq> since scenecuts are more likely to be on keyframes [04:07:54 CEST] <Hello71> ty [04:21:23 CEST] <Hello71> also, TIL the ffmpeg jpeg encoder is terrible [04:22:04 CEST] <Hello71> why is it so much worse than imagemagick [04:23:25 CEST] <atomnuker> turn on huffman optimizations [04:24:11 CEST] <furq> isn't it on by default [04:24:19 CEST] <Hello71> oh, nvm that last part, turns out a few K is a big difference to imagemagick [04:24:28 CEST] <Hello71> thought 74 and 80 should be about the same, apparently not [06:01:36 CEST] <nai> hi, i have an issue trying to record my screen. here is my command: [06:01:38 CEST] <nai> ffmpeg -framerate 30 -f x11grab -i :0.0 -f mp4 -c:v libx264 -crf 0 -preset ultrafast out.mp4 [06:02:12 CEST] <nai> the resulting file plays nice in mpv and ffplay, but firefox cannot play it. instead, it gives the following error: "Video can't be played because the file is corrupt" [06:02:25 CEST] <nai> here is the output of ffprobe on that file: https://up.nai.im/ffprobe.txt [06:05:34 CEST] <furq> nai: add -pix_fmt yuv420p [06:06:12 CEST] <nai> furq: doesn't solve it [06:06:44 CEST] <kepstin> lossless always uses the 444 profile in h264, which firefox doesn't support [06:06:59 CEST] <kepstin> you'll have to use a non-lossless mode *and* the yuv420p pix_fmt option [06:07:32 CEST] <kepstin> if you want working lossless video and 4:4:4 sampling in firefox & chrome, you should use vp9 instead of h264. [06:07:50 CEST] <nai> perfect answer, thanks! [06:08:15 CEST] <nai> i was precisely wondering the benefits of webm/vp9 over traditional formats [06:08:29 CEST] <kepstin> (note that the libvpx-vp9 encoder is a lot slower than x264 - you might consider capturing with x264 then transcoding to vp9 if it's too slow to do vp9 live) [06:08:30 CEST] <atomnuker> don't use fucking mp4 [06:08:44 CEST] <atomnuker> if ffmpeg dies hard the file will be corrupt [06:08:46 CEST] <nai> kepstin: thanks [06:08:56 CEST] <kepstin> and, what atomnuker said ;) [06:09:51 CEST] <nai> am i to infer that mp4 stores data at the end of files? [06:10:57 CEST] <kepstin> mp4 stores some codec initialization data together with a variable-size index field, which means the init data can't be stored at the start of the file when initially encoding the video [06:11:10 CEST] <kepstin> and so it's written at the end... and the video is unplayable without the init data [06:11:42 CEST] <kepstin> mkv is a reasonable alternative. you can always remux the file later into a different container. [06:11:51 CEST] <nai> got it, thanks [06:12:30 CEST] <nai> i was initially using mkv, but ran into another problem, maybe you can enlighten me on this one: the mimetype of the resulting files was application/octet-stream instead of video/x-matroska [06:12:47 CEST] <kepstin> that sounds like a problem with your web server configuration [06:12:59 CEST] <nai> no, i'm talking about the mimetype returned by xdg-mime [06:13:02 CEST] <kepstin> (but most browsers can't play mkv anyways, so it doesn't really matter) [06:13:26 CEST] <kepstin> hmm, well, that's probably a bug there. on linux there's no mime type stored in the file, so tools just try to read file headers and guess [06:13:29 CEST] <nai> this isn't a very important issue, it just makes xdg-open confused when trying to open the file, and ends up opening it with my browser instead of my video player [06:14:41 CEST] <nai> yeah, my file manager (pcmanfm) uses another method to detect file types, possibly based on extensions, and therefore is able to open the correct application [06:14:57 CEST] Action: kepstin notes that xdg-mime reports 'application/x-matroska' correctly on his system - maybe yours is just out of date or missing file type definitions? [06:15:04 CEST] <nai> but apparently something i was doing wrong prevents xdg-mime from detecting a matroska file [06:15:17 CEST] <nai> wait, i'll give you a test command [06:15:48 CEST] <nai> ffmpeg -framerate 30 -f x11grab -i :0.0 -f matroska -c:v libx264 -pix_fmt yuv420p -crf 0 -preset ultrafast out.mkv [06:16:02 CEST] <nai> xdg-mime query filetype out.mkv => application/octet-stream [06:16:31 CEST] <kepstin> I get video/x-matroska there. You probably have out of date software or a distro packaging issue :/ [06:16:47 CEST] <nai> hm, interesting [06:16:53 CEST] <nai> i'm on arch's bleeding edge [06:17:11 CEST] <kepstin> so, distro packaging issue then ;) [06:17:22 CEST] <nai> yup. alright, thanks for everything! :) [06:17:46 CEST] <kepstin> (i'm joking, but i have had friends run into issues with arch packages a fair bit, so..?) [06:18:14 CEST] <nai> yeah, might be. i honestly don't have the insight to know what the cause might be [06:18:52 CEST] <kepstin> also check what 'file --mime-type out.mkv' reports [06:19:20 CEST] <kepstin> i dunno exactly how xdg-mime works, but maybe it uses the same data as file? [06:19:24 CEST] <nai> application/octet-stream too [06:19:47 CEST] <nai> my "file" command comes from https://www.darwinsys.com/file/ [06:20:03 CEST] <nai> apparently it's standard among Linux distros [06:20:11 CEST] <nai> i have version 5.33-3 [06:20:56 CEST] <kepstin> yeah, so there's something wrong with the 'magic' database on your system that 'file' uses to detect file types. [06:24:10 CEST] <nai> alright, i want to go to the end of this. could you upload your /usr/share/file/misc/magic.mgc file? here's mine https://up.nai.im/magic.mgc [06:24:45 CEST] <nai> my server serves that as plain text, sorry about that, wrong mime type configuration :') [06:26:30 CEST] <nai> ah, i've found a bug report https://bugs.launchpad.net/ubuntu/+source/file/+bug/420963 [06:27:13 CEST] <nai> ...from 2012 [06:27:17 CEST] <nai> 2010* [09:33:03 CEST] <momomo> Don't block me. One line. Looking for a Linux Sysadmin in Europe, for a great job oppurtonity in Stockholm city. Well payed, permanent / temporary (based on your preference). Immediate Accommodation. One crux, has to also know Elastic Search. Need to be filled immediately. [09:33:24 CEST] <lohroc> I want to use ffmpeg to downscale from 1080p .mkv to 720p .mp4 but I'm getting the error element type mismatch 3 != 0 [09:34:57 CEST] <lohroc> also can I resize the video and change the extension at once? [10:10:11 CEST] <momomo> Don't block me. One line. Looking for a Linux Sysadmin in Europe, for a great job oppurtonity in Stockholm city. Well payed, permanent / temporary (based on your preference). Immediate Accommodation. One crux, has to also know Elastic Search. Need to be filled immediately. [13:00:50 CEST] <subi> why zeranoe's win build doesn't include ffserver.exe? also why the 4.0 version couldn't initialise SDL Audio through WASAPI? [13:02:06 CEST] <durandal_1707> subi: ffserver is removed in 4.0 [13:02:43 CEST] <subi> then where's the feature now? in ffmpeg.exe? [13:03:41 CEST] <klaxa> it's in development [13:03:58 CEST] <klaxa> i'm writing a replacement [13:04:28 CEST] <klaxa> maybe it will even be useful [13:07:28 CEST] <subi> is it zeranoe specific or the original ffmpeg project that makes those changes? [13:08:48 CEST] <furq> it was removed upstream because it never worked properly and nobody maintained it [13:11:05 CEST] <subi> then what's left is obs studio? [13:11:44 CEST] <furq> OBS doesn't have a server does it [13:11:45 CEST] <subi> something for live broadcast [13:11:57 CEST] <furq> you can still do screen capture with ffmpeg, you just need an external streaming server [13:12:10 CEST] <furq> nginx-rtmp is a popular one [13:57:51 CEST] <TarquinWJ> JEEB: think it was you who helped me yesterday... I am running this command: [13:57:51 CEST] <TarquinWJ> ffmpeg.exe -i source.mp4 -map 0:v -c:v libvpx-vp9 -pix_fmt yuv420p -b:v 2M -vf scale=1024:576 -r 30 -f dash -format_options dash_segment_type=webm -seg_duration 3 -media_seg_name "\$Number\$.chk" -init_seg_name 0.chk output.mpd [13:57:51 CEST] <TarquinWJ> it works, I get a dashed video. but MediaInfo says that the first fragment is "Format : MPEG-4" rather than "Format : WebM" which I see coming from YouTube [13:57:51 CEST] <TarquinWJ> Am I doing something wrong? [14:01:24 CEST] <TarquinWJ> it almost feels like I have got an old version rather than the latest, which is possible, but I downloaded the latest snapshot here: [14:01:25 CEST] <TarquinWJ> ttps://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 [14:07:54 CEST] <JEEB> TarquinWJ: not sure ffmpeg.c take options like that? [14:08:04 CEST] <JEEB> also isn't dash_segment_type its own option? [14:08:10 CEST] <JEEB> -dash_segment_type "webm" [14:08:21 CEST] <JEEB> not an option for "format_options" [14:12:14 CEST] <TarquinWJ> hah, well at least that was a simple fix :D [14:12:38 CEST] Action: TarquinWJ going to hang head in shame now [14:13:33 CEST] <JEEB> hint: ffmpeg -h muxer=dash [14:13:41 CEST] <JEEB> should list all AVOptions specific for that muxer [14:13:51 CEST] <TarquinWJ> thanks :) [14:14:57 CEST] <TarquinWJ> always nice to realise you just misunderstood the documentation, rather than something being impossible [14:15:15 CEST] <JEEB> also if you're building, just use the git.videolan.org git repo, much simpler to update :P [14:15:20 CEST] <JEEB> and you know exactly the hash you're on [14:21:39 CEST] <TarquinWJ> I see why I was confused; documentation does not show "-dash_segment_type", it shows "dash_segment_type" without the "-", so it looks like a value for the previous option [14:22:04 CEST] <TarquinWJ> see here: https://www.ffmpeg.org/ffmpeg-all.html#dash-2 [15:09:01 CEST] <JEEB> TarquinWJ: and that is because the dash in front is specific to ffmpeg.c :) [15:09:10 CEST] <JEEB> while those things just list the options for each module [16:32:12 CEST] <aeikum> hi all, quick question about av_frame ref counting with the new decoder API. in short, is it OK to have more unrefs than refs? specifically, avcodec_receive_frame() says it calls av_frame_unref(), but this is unbalanced on the very first call. do i need to call av_frame_ref() first? [16:34:03 CEST] <aeikum> i guess the intention is that the client can do av_frame_alloc(), repeat receive() with refs managed internally, then av_frame_free(), without having to bother with refcounting at all [16:34:05 CEST] <aeikum> just wanted to confirm [16:40:59 CEST] <anill> Hi users, can anyone explain me how can i can use ffmpeg for packetization. [16:44:11 CEST] <anill> Can i use FFMPEG to extrach the H.264 content from RTP stream.? [16:52:39 CEST] <kepstin> aeikum: calling av_frame_unref on a frame that doesn't have any allocated buffers isn't an error, it'll just reset some of the fields in the frame structure. [16:52:59 CEST] <aeikum> awesome, makes sense. thank you [17:03:40 CEST] <anill> Can i use FFMPEG to extract the H.264 content from RTP stream.? [17:04:13 CEST] <DHE> anill: yes, just a question of what you want to write it to. mp4 file? [17:04:24 CEST] <DHE> or maybe clarify what you mean by packetization [17:07:39 CEST] <anill> DHE: i mean if i have decrypted RTP stream, i want to get the raw H.264 contents from it or maily look at this https://tools.ietf.org/html/rfc6184#page-16 [17:09:19 CEST] <anill> DHE:i have two requirements for writing one is writing into .h264 file and another is directly in MP4 file [17:09:54 CEST] <DHE> okay. that's easy. [17:10:02 CEST] <anill> great [17:10:14 CEST] <anill> can you please explain me a little bit [17:10:22 CEST] <DHE> Well I'm guessing it would be enough to do: ffmpeg -i rtp://..... -c copy -map 0:v output.mp4 -c copy -map 0:v output.h264 [17:10:39 CEST] <DHE> will write both an .mp4 file and a raw h264 variant [17:11:05 CEST] <DHE> actually hold on. I don't think ffmpeg recognizes that as a default format... [17:11:31 CEST] <DHE> okay, I'm wrong. it does. [17:11:50 CEST] <anill> can i give it a file rather than rtp://.... [17:11:58 CEST] <DHE> sure [17:12:59 CEST] <anill> another thing there are 3 packetization modes Single NAL unit mode, Non Interleaved Mode and Interleaved Mode [17:13:11 CEST] <anill> as shown here https://tools.ietf.org/html/rfc6184#page-16 [17:13:52 CEST] <anill> if i run the above command what would be the packteziation mode ? [17:13:54 CEST] <anill> any idea [17:14:27 CEST] <DHE> yes I see that. I don't know the specifics here but I expect ffmpeg would output in non-interleaved mode. internally it runs in DTS order which seems to match the description of non-interleaved [17:15:50 CEST] <zevarito> Does anyone know if h264/hls encoder uses already written .ts segments to build new ones, update m3u8 etc.? or forget about it once it has been written on disk? [17:15:51 CEST] <anill> DHE: you seems to me very knowlegeable, can i ask you a silly question, do you have any link for the document stating ffmpeg runs in DTS order. [17:20:46 CEST] <DHE> anill: it's part of the doxygen stuff for the raw APIs. the muxer needs monotonically increasing DTS order for its inputs [17:26:29 CEST] <anill> DHE: does ffmpeg take pcap file of decrypted RTP stream and get the raw H.264 file and MP4 file? [17:26:47 CEST] <anill> i mean i wanted to know the type of file input that can be given to ffmpeg [17:32:33 CEST] <anill> DHE: u there? [17:35:47 CEST] <DHE> it won't read a pcap file. you might be able to dump the stream in the pcap to a file and read that maybe? [17:36:08 CEST] <DHE> maybe. I really don't know if that will work. my gut says no though [17:36:18 CEST] <kepstin> anill: ffmpeg won't take pcap input, no. the ffmpeg rtp stuff is designed to receive network packets directly, so you'd have to have something to parse it and either send over net or use libavformat directly. [17:37:17 CEST] <kepstin> there's probably some specialty network analysis tools that can save audio/video from rtp streams in pcap (wireshark might have something?) but it's not something the ffmpeg cli can do. [17:38:42 CEST] <anill> kepstin: what i do is i capture the RTP stream into file, its a raw file just a dump of all RTP packets [17:39:32 CEST] <anill> so as you said i need to use libavformat to dump all the RTP packets into file is it [17:39:34 CEST] <anill> ? [17:39:40 CEST] <kepstin> a dump of the rtp packets isn't sufficient, iirc rtp relies on external framing instead of storing packet lengths internally so you need to at least store the length with each packet. [17:40:05 CEST] <kepstin> (i could be wrong about that, been a while since i've worked with this tuff) [17:40:44 CEST] <anill> whats iirc? [17:40:56 CEST] <kepstin> "if i recall correctly" [17:41:15 CEST] <kepstin> (or "remember") [17:42:22 CEST] <anill> do u have an idea of packetization mode in ffmpeg? [17:42:56 CEST] <voip_> hi guys [17:43:02 CEST] <voip_> I need take rtmp live stream and without transcoding video/audio push multicast UDP. [17:43:02 CEST] <voip_> Please help with right command [17:43:29 CEST] <anill> kepstin: do u have an idea of packetization mode in ffmpeg? [17:43:48 CEST] <kepstin> anill: packetization mode of input? output? what format? [17:44:30 CEST] <DHE> voip_: ffmpeg -i rtmp://...... -c copy -f mpegts -pkt_size 1316 -muxrate 8M -bitrate 8M udp://239.0.0.1:12345 [17:45:03 CEST] <DHE> where the source video is CBR and never exceeds ~8 megabits. substitute in the options you want like multicast IP/port and of course the rtmp source [17:45:14 CEST] <voip_> DHE, thank you! [17:45:15 CEST] <kepstin> voip_: although given your name, maybe you want rtp instead? [17:45:54 CEST] <voip_> its:)))) udp is perfect :)) [17:46:29 CEST] <kepstin> there's lots of ways to put video into udp, and you didn't say which way you wanted. [17:47:03 CEST] <anill> kepstin: although DHE, helped me much in this, what i want is i have decryped RTP stream and i want H.264 data from it, there are packetization mode defined here https://tools.ietf.org/html/rfc6184#page-16, i want to get the output file .h264 and .MP4 [17:47:11 CEST] <kepstin> mpeg-ts is used in iptv broadcast stuff with multicast udp, so that's fairly common at least. [17:47:37 CEST] <voip_> DHE one more question, for monitoring and restart ffmpeg what program you will recommend ? I was google for "monit", but not found goof guide [17:48:12 CEST] <kepstin> anill: that packetization mode is being done by whatever is sending the video, not by ffmpeg... ? [17:52:24 CEST] <anill> kepstin: you asking me or telling, confused [17:55:13 CEST] <anill> kepstin: u there ? [17:56:15 CEST] <kepstin> anill: I don't know what you're trying to ask - the information about packetization format is for the rtp stream, which is not what you're writing with ffmpeg. [17:57:41 CEST] <anill> kepstin: You mean to say ffmpeg has nothing to do with packteziation mode, right? [17:58:35 CEST] <kepstin> well, it has to be able to parse whatever mode you're sending, but I expect that it can handle common standard modes fine. [18:02:24 CEST] <anill> so this command ffmpeg -i rtp://..... -c copy -map 0:v output.mp4 -c copy -map 0:v output.h264 will output the rtp://... into the .h264 and MP4 format [18:04:48 CEST] <anill> kepstin: am i correct? [18:05:57 CEST] <kepstin> anill: it should yeah. [18:06:32 CEST] <kepstin> anill: that should work to capture a live (network) rtp stream that is being sent to the computer ffmpeg is running on. [18:07:51 CEST] <anill> kepstin: ya but in my case i have stream being captured into a file, and also no clue how to give file as input to ffmpeg [18:14:36 CEST] <voip_> Guys for monitoring ffmpeg and restarting if it stops i am going to use Monit utility. [18:14:36 CEST] <voip_> But didn't found any "best practice" guide how to configure for ffmpeg [18:14:36 CEST] <voip_> Or you can recommend other software ? [19:20:41 CEST] <wilsoniya> Does ffmpeg maintain a corpus of videos of various a/v codecs and container formats (both valid and potentially invalid/corrupted) which are used for testing? [19:28:06 CEST] <leif_> Does anyone know what might cause ffmpeg to output the warning: "Could not update timestamps for skipped samples"? [19:28:28 CEST] <leif_> (And the resulting file is very short. Like, less than a seecond of the two minute file I was expecting.) [19:29:01 CEST] <klaxa> wilsoniya: there is https://www.ffmpeg.org/fate.html [19:33:05 CEST] <leif_> kepstin: Err...I didn't use the command line tool, but the C API. [19:33:29 CEST] <kepstin> well, that's helpful to know too :) [19:33:55 CEST] <leif_> When I use the command line, it works fine with the exact same filtergraph. Which makes me think its a problem with seeking. [19:34:16 CEST] <leif_> (But the thing is I get it even when I don't do any seeking.) [19:35:50 CEST] <kepstin> kinda hard to tell what's going on without any sample code or information about the intput, but a guess from the ffmpeg code is that you might have an issue with how the stream timebase is configured. [19:38:21 CEST] <leif_> kepstin: Mmm...that might make sense. Would that be a thing I have to set manually, or a thing ffmpeg guesses at automatically? [19:38:51 CEST] <leif_> (My codebase is very asynchronous, making it hard to make a small sample. But I'll see if I can put one together.) [19:39:12 CEST] <kepstin> there's some steps you're required to do to make sure information is passed between different parts of the code, maybe look at some of the examples for reference? [19:42:57 CEST] <leif_> kepstin: Hmm...I've done that. But its possible I've missed something. [19:43:03 CEST] <leif_> I'll try to get a small sample together. [19:46:54 CEST] <leif_> https://gist.github.com/LeifAndersen/4fe6316aa7a35a243a161c3d9c1877c7 [19:47:14 CEST] <leif_> kepstin: Okay, that is (more ore less), what I do to open a file. [19:47:43 CEST] <kepstin> oh, fun, you're going through some kind of bindings rather than using the api directly :/ [19:48:33 CEST] <leif_> kepstin: Yes I am. Although I am the poor chump responsible for making those bindings... [19:48:53 CEST] <kepstin> I was gonna suggest going to the person who developed the bindings for help, but, uh... :) [19:48:54 CEST] <leif_> I can easily translate it into C proper if you'd like. [19:49:02 CEST] <leif_> lol... [19:51:20 CEST] <leif_> If I do translate it into C, is there any easy way to test if its right? [19:53:42 CEST] <kepstin> not sure what you mean. compile and run it, see if it works? [19:54:53 CEST] <leif_> kepstin:Err...its the 'see if it works' part that's hard. Because if I do translate it to C, then all I have is a stream of data. So I'm not sure how I can make sure that stream is 'the right stream'(tm). [19:57:17 CEST] <kepstin> tbh, I'd suggest attempting to write code that does what you want in C, then working on your bindings afterwards so they replicate what you've got working in C [19:58:03 CEST] <kepstin> note that ffmpeg's api is a very synchronous design, so building something async on top of it is probably gonna be difficult - and you have to be careful with threads [20:00:54 CEST] <leif_> kepstin: Oh ya. Its all single threaded. It just uses callbacks to sort of jump back and forth between encoding and decoding. [20:01:34 CEST] <leif_> or at least, it only uses one OS level thread. I do use a few 'green' threads so to speak. [20:01:40 CEST] <leif_> Anyway, thanks for the suggestion. :) [20:03:50 CEST] <wilsoniya> klaxa: exactly what I needed! Thank you! [22:24:52 CEST] <Zexaron> Hello [22:26:08 CEST] <Zexaron> I'm not sure why is the program that uses a specific ffmpeg version having to check what version it's using, or I'm understanding that wrong, it has stuff like this #if (LIBAVCODEC_VERSION_MICRO >= 100 && LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 33, 100)) || \ [22:26:58 CEST] <durandal_1707> mainly to check if it is Libav of FFmpeg... [22:29:27 CEST] <Zexaron> durandal_1707_: but FFmpeg is custom built and integrated into the program without the need for shared DLLs, it's as specific as you can get, and I only the developer that includes ffmpeg support knows the details of the custom feature set he used when building those ffmpeg libs (unless that can be analyzed in the .lib) [22:30:41 CEST] <Zexaron> This whole thing is being rewritten from top to bottom, there may be old code from 2009 still there, it's at this line: https://github.com/dolphin-emu/dolphin/blob/master/Source/Core/VideoCommon/… [22:31:07 CEST] <Zexaron> Sorry, not 2009, ffmpeg got introduced quite a bit later [22:35:32 CEST] <Zexaron> This should explain things https://github.com/dolphin-emu/dolphin/commit/04158dfe158785b9fcb3c11278d74… [22:35:42 CEST] <Zexaron> Still relevant for 4.0 ? [22:36:42 CEST] <Zexaron> All the old stuff is being removed, 4.0 will be supported by default and prefferred, there will be an override to do different versions, but not supported [22:37:01 CEST] <Zexaron> So I don't care about any other version except ffmpeg 4.0 [22:37:55 CEST] <JEEB> then just keep the stuff that builds with 4.0, that's just bw compat for non-codecpar versions, which by now are quite old [22:41:36 CEST] <Zexaron> The goal is to go straight to hw accel codecs, so this non-codecpar might not be needed, if that's not working right, even if cpu codecs are allowed the newer stuff will be used, but i don't know what codecpar even is yet [22:42:16 CEST] <Zexaron> JEEB: the ffmpeg currently used is 3.4.2 I think, I already updated it to 4.0 and had no trouble building with existing code [22:42:30 CEST] <Zexaron> in my dev branch [22:42:37 CEST] <JEEB> 3.4 already had codecpar anyways. it's by now old, so all that's to support some really older stuff [22:43:30 CEST] <Zexaron> Okay, just making sure, thanks. [23:59:13 CEST] <gcl_5> i get some repetitive warnings so i disable with -loglevel error, but i still want "the progress bar" -> time=00:35:24.87 bitrate= [00:00:00 CEST] --- Fri Jun 1 2018

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