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• FFmpeg - Max rtbufsize via dshow ?

  14 septembre 2018, par Nimble

  I recently added an additional 4K capture card to my setup and now I’m dropping frames while initializing and ending recordings. In the past I was encoding a 1080P60 stream and a 4K60 stream simultaneously and had no issues, but now that I am trying to encode 2 4K60 streams at once I’m dropping frames as mentioned before.

  The error displays as :

  [dshow @ 000001499bb17180] real-time buffer [Video (00 Pro Capture HDMI 4K+)] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!

  or

  [dshow @ 00000149944e7080] real-time buffer [AVerMedia HD Capture GC573 1] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!

  10 - 20 times when starting a recording or ending a recording.

  You’d think the solution would be simply increasing my rtbufsize but when I do I just get another error :

  [dshow @ 00000250df6c7080] Value 3000000000.000000 for parameter 'rtbufsize' out of range [0 - 2.14748e+09]
  
  [dshow @ 00000250df6c7080] Error setting option rtbufsize to value 3000M.
  
  video=AVerMedia HD Capture GC573 1:audio=SPDIF/ADAT (1+2) (RME Fireface UC): Result too large

  This same error seems to appear if I try to increase the rtbufsize past 2147.48M on any input so I assume it’s a limitation of FFmpeg and not my hardware ? If it is a baked in limitation of FFmpeg what would be the reasoning ? Any way to bypass or other possible solutions ?

  Full command :

  ffmpeg -y -hide_banner -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M `
  
  -i audio="Analog (1+2) (RME Fireface UC)" `
  
  -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M `
  
  -i audio="ADAT (5+6) (RME Fireface UC)" `
  
  -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M `
  
  -framerate 60 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="ADAT (3+4) (RME Fireface UC)" `
  
  -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M `
  
  -framerate 60 -pixel_format nv12 -i video="AVerMedia HD Capture GC573 1":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" `
  
  -thread_queue_size 9999 -indexmem 9999 -r 25 -f lavfi -rtbufsize 2147.48M -i color=c=black:s=50x50 `
  
  -map 4,0 -map 0 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop `
  
  -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k `
  
  -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:01.768 `
  
  -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
  
  -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Discord\Discord%02d.ts `
  
  -map 4,1 -map 1 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop `
  
  -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k `
  
  -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:01.071 `
  
  -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
  
  -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Soundboard\Soundboard%02d.ts `
  
  -map 2:0,2:1 -map 2:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop `
  
  -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M `
  
  -c:a aac -ar 44100 -b:a 384k -ac 2 -af "atrim=0.086, asetpts=PTS-STARTPTS, aresample=async=250" -vsync 1 -ss 00:00:00.102 `
  
  -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
  
  -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Magewell\Magewell%02d.ts `
  
  -map 3:0,3:1 -map 3:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop `
  
  -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M `
  
  -c:a aac -ar 44100 -b:a 384k -ac 2 -af "pan=mono|c0=c0, aresample=async=250" -vsync 1 `
  
  -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
  
  -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Camera\Camera%02d.ts

  EDIT : Also worth mentioning that I only drop frames when starting and ending recording, everything is fine in the middle. Wonder if I could like "ease" the recording in or something ?

  (09/13/2018) : I was able to stop frames from dropping while starting a recording by re-arranging inputs and outputs, however no matter how I list things I still drop frames ending recordings.

• ffmpeg nvdec not available when adding subtitles to mp4

  17 février 2019, par whatdoineed2do

  I have a mp4 video that I want to BURN-IN subtitles (from a .ass file and via -vf ass=/tmp/conf-notes.ass) with ffmpeg on a machine with a GTX1060 card however ffmpeg reports that it fails to create the NVDEC decoder.

  [h264 @ 0xcb9380] Error creating a NVDEC decoder: 1
  
  [h264 @ 0xcb9380] Failed setup for format cuda: hwaccel initialisation returned error.

  The output file is generated ok but during the creation the CPU is used heavily (I assume for decode) - if I drop the -vf ass=/tmp/conf-notes.ass and perform an mp4->mp4 reencode there are no problems.

  I want to know what I am missing from my command to allow the GPU to perform the decode when an .ass file is added to the chain.

  $ ffmpeg -hwaccel cuvid -i 2019-01-conferance.mp4 -vf ass=/tmp/conf-notes.ass -c:a copy -preset llhq -rc vbr_hq -vb 2M -minrate 500k -maxrate 12M -c:v h264_nvenc export.mp4
  
  ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 8 (GCC)
  
    configuration: --arch=x86_64 --bindir=/usr/bin --datadir=/usr/share/ffmpeg --disable-debug --disable-static --disable-stripping --enable-alsa --enable-avfilter --enable-avresample --enable-bzlib --enable-decklink --enable-doc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gpl --enable-iconv --enable-libass --enable-libbluray --enable-libcdio --enable-libdc1394 --enable-libdrm --enable-libfdk-aac --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libkvazaar --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librsvg --enable-librtmp --enable-librubberband --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtesseract --enable-libtheora --enable-libtwolame --enable-libv4l2 --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxcb-shm --enable-libxcb-xfixes --enable-libxcb-shape --enable-libxml2 --enable-libxvid --enable-libzvbi --enable-lv2 --enable-lzma --enable-libndi_newtek --enable-nonfree --enable-openal --enable-opencl --enable-opengl --enable-postproc --enable-pthreads --enable-sdl2 --enable-shared --enable-version3 --enable-xlib --enable-zlib --extra-cflags=-I/usr/include/cuda --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --optflags='-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --prefix=/usr --shlibdir=/usr/lib64 --enable-cuda --enable-cuvid --enable-libnpp --enable-nvdec --enable-nvenc --enable-runtime-cpudetect
  
    libavutil      56. 14.100 / 56. 14.100
  
    libavcodec     58. 18.100 / 58. 18.100
  
    libavformat    58. 12.100 / 58. 12.100
  
    libavdevice    58.  3.100 / 58.  3.100
  
    libavfilter     7. 16.100 /  7. 16.100
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  1.100 /  5.  1.100
  
    libswresample   3.  1.100 /  3.  1.100
  
    libpostproc    55.  1.100 / 55.  1.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '2019-01-conferance.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: isommp42
  
      creation_time   : 2019-01-12T00:00:06.000000Z
  
    Duration: 00:00:24.07, start: 0.000000, bitrate: 11542 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 11350 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-01-12T00:00:07.000000Z
  
        handler_name    : Mainconcept MP4 Video Media Handler
  
        encoder         : AVC Coding
  
      Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-01-12T00:00:07.000000Z
  
        handler_name    : Mainconcept MP4 Sound Media Handler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> h264 (h264_nvenc))
  
    Stream #0:1 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  [h264 @ 0xcb9380] Error creating a NVDEC decoder: 1
  
  [h264 @ 0xcb9380] Failed setup for format cuda: hwaccel initialisation returned error.
  
  [Parsed_ass_0 @ 0x1241c00] Shaper: FriBidi 1.0.2 (SIMPLE) HarfBuzz-ng 1.7.5 (COMPLEX)
  
  [Parsed_ass_0 @ 0x1241c00] Using font provider fontconfig
  
  [Parsed_ass_0 @ 0x1241c00] Added subtitle file: '/tmp/conf-notes.ass' (2 styles, 101 events)

  The .ass file has been generated from a .srt file via ffmpeg -i foo.srt foo.ass

• WebRTC predictions for 2016

  17 février 2016, par silvia

  I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

  WebRTC Browser support

  I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016 :

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

  Codec Observations

  With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

  However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

  Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

  I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability : I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

  The Enterprise Boundary

  Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary : firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

  The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

  SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

  We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

  Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

  We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

  What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

  I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.

  Summary

  So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress : interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

  —

  It’s worth mentioning Philipp Hancke’s tweet reply to my post :

  https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/ … — we saw some clever people come up with a solution already. Now it needs to be implemented