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• What is “interoperable TTML” ?

  19 septembre 2012, par silvia

  I’ve just tried to come to terms with the latest state of TTML, the Timed Text Markup Language.

  TTML has been specified by the W3C Timed Text Working Group and released as a RECommendation v1.0 in November 2010. Since then, several organisations have tried to adopt it as their caption file format. This includes the SMPTE, the EBU (European Broadcasting Union), and Microsoft.

  Both, Microsoft and the EBU actually looked at TTML in detail and decided that in order to make it usable for their use cases, a restriction of its functionalities is needed.

  EBU-TT

  The EBU released EBU-TT, which restricts the set of valid attributes and feature. “The EBU-TT format is intended to constrain the features provided by TTML, especially to make EBU-TT more suitable for the use with broadcast video and web video applications.” (see EBU-TT).

  In addition, EBU-specific namespaces were introduce to extend TTML with EBU-specific data types, e.g. ebuttdt:frameRateMultiplierType or ebuttdt:smpteTimingType. Similarly, a bunch of metadata elements were introduced, e.g. ebuttm:documentMetadata, ebuttm:documentEbuttVersion, or ebuttm:documentIdentifier.

  The use of namespaces as an extensibility mechanism will ascertain that EBU-TT files continue to be valid TTML files. However, any vanilla TTML parser will not know what to do with these custom extensions and will drop them on the floor.

  Simple Delivery Profile

  With the intention to make TTML ready for “internet delivery of Captions originated in the United States”, Microsoft proposed a “Simple Delivery Profile for Closed Captions (US)” (see Simple Profile). The Simple Profile is also a restriction of TTML.

  Unfortunately, the Microsoft profile is not the same as the EBU-TT profile : for example, it contains the “set” element, which is not conformant in EBU-TT. Similarly, the supported style features are different, e.g. Simple Profile supports “display-region”, while EBU-TT does not. On the other hand, EBU-TT supports monospace, sans-serif and serif fonts, while the Simple profile does not.

  Thus files created for the Simple Delivery Profile will not work on players that expect EBU-TT and the reverse.

  Fortunately, the Simple Delivery Profile does not introduce any new namespaces and new features, so at least it is an explicit subpart of TTML and not both a restriction and extension like EBU-TT.

  SMPTE-TT

  SMPTE also created a version of the TTML standard called SMPTE-TT. SMPTE did not decide on a subset of TTML for their purposes – it was simply adopted as a complete set. “This Standard provides a framework for timed text to be supported for content delivered via broadband means,…” (see SMPTE-TT).

  However, SMPTE extended TTML in SMPTE-TT with an ability to store a binary blob with captions in another format. This allows using SMPTE-TT as a transport format for any caption format and is deemed to help with “backwards compatibility”.

  Now, instead of specifying a profile, SMPTE decided to define how to convert CEA-608 captions to SMPTE-TT. Even if it’s not called a “profile”, that’s actually what it is. It even has its own namespace : “m608 :”.

  Conclusion

  With all these different versions of TTML, I ask myself what a video player that claims support for TTML will do to get something working. The only chance it has is to implement all the extensions defined in all the different profiles. I pity the player that has to deal with a SMPTE-TT file that has a binary blob in it and is expected to be able to decode this.

  Now, what is a caption author supposed to do when creating TTML ? They obviously cannot expect all players to be able to play back all TTML versions. Should they create different files depending on what platform they are targeting, i.e. a EBU-TT version, a SMPTE-TT version, a vanilla TTML version, and a Simple Delivery Profile version ? Should they by throwing all the features of all the versions into one TTML file and hope that the players will pick out the right things that they require and drop the rest on the floor ?

  Maybe the best way to progress would be to make a list of the “safe” features : those features that every TTML profile supports. That may be the best way to get an “interoperable TTML” file. Here’s me hoping that this minimal set of features doesn’t just end up being the usual (starttime, endtime, text) triple.

  UPDATE :

  I just found out that UltraViolet have their own profile of SMPTE-TT called CFF-TT (see UltraViolet FAQ and spec). They are making some SMPTE-TT fields optional, but introduce a new @forcedDisplayMode attribute under their own namespace “cff :”.

• ffmpeg rtsp stream to YouTube livestream not doing anything

  29 mai 2018, par felixosth

  I’m using C# to initiate ffmpeg. I’ve a onvif bridge server for a CCTV VMS and I’m building a application to enable the user to livestream any CCTV camera to YouTube.

  The rtsp stream to the camera looks like this :

  rtsp ://onvif:bridge@localhost:554/live/xxxxx-xxxguidtocameraxxx-xxxxx

  I’m new to ffmpeg and I’m using this arg :

  -f lavfi -i anullsrc -rtsp_transport udp -i camerastreamurl -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp ://x.rtmp.youtube.com/live2/streamkey -loglevel debug

  It seems like ffmpeg isn’t finding the camera stream, only the fake audio one. It just freezes.

  This is the result of the debug log :

  ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 18.102 / 56. 18.102
  
    libavcodec     58. 19.104 / 58. 19.104
  
    libavformat    58. 17.100 / 58. 17.100
  
    libavdevice    58.  4.100 / 58.  4.100
  
    libavfilter     7. 24.100 /  7. 24.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  Splitting the commandline.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'.
  
  Reading option '-i' ... matched as input url with argument 'anullsrc'.
  
  Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'udp'.
  
  Reading option '-i' ... matched as input url with argument 'rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b'.
  
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  
  Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
  
  Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument '+'.
  
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
  
  Reading option '-strict' ...Routing option strict to both codec and muxer layer
  
   matched as AVOption 'strict' with argument 'experimental'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'rtmp://x.rtmp.youtube.com/live2/xxxxxxxx' ... matched as output url.
  
  Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option loglevel (set logging level) with argument debug.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input url anullsrc.
  
  Applying option f (force format) with argument lavfi.
  
  Successfully parsed a group of options.
  
  Opening an input file: anullsrc.
  
  detected 8 logical cores
  
  [AVFilterGraph @ 0000027a34bad7c0] query_formats: 2 queried, 3 merged, 0 already done, 0 delayed
  
  [Parsed_anullsrc_0 @ 0000027a34badb80] sample_rate:44100 channel_layout:'stereo' nb_samples:1024
  
  [lavfi @ 0000027a34babc80] All info found
  
  Input #0, lavfi, from 'anullsrc':
  
    Duration: N/A, start: 0.000000, bitrate: 705 kb/s
  
      Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: input url rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  Successfully parsed a group of options.
  
  Opening an input file: rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  [tcp @ 0000027a34bb5980] No default whitelist set

  Edit :
  I’m getting this result with minimal args :
  

• ffmpeg problems with streaming mp4 over udp in local network

  28 novembre 2019, par AJ Cole

  I’m streaming mp4 video files (some of them are avi converted to mp4 with ffmpeg earlier) over udp://232.255.23.23:1234 from linux (embedded) with ffmpeg v3.4.2 to multiple linux (antix) machines that play the stream with MPV, all of this happens in local network so I expected it to work flawlessly, but unfortunately it doesn’t.

  Here are the original commands I tried to use :

  ffmpeg

  ffmpeg -re -i PATH_TO_FILE.mp4 -c copy -f mpegts udp://232.255.23.23:1234

  mpv

  mpv --no-config --geometry=[geo settings] --no-border udp://232.255.23.23:1234

  This seemed to woork good, however a problem appeared that on the displaying end, the stream is actually much longer than the streamed content itself. The mp4 files in total have 5 minutes 36 seconds, and mpv plays the entire stream loop in >= 6 minutes. I think it’s happening because of dropped frames, that mpv waits to recover or something and therefore extends the length of the actual content. This cannot work in my case, as I have a precise time gap for displaying the stream and it cannot be longer than the streamed content.
  All the content is made in 1680x800 resolution and is displayed on a screen with 1680x1050 resoltion (positioned with mpv geometry)

  It appears that using this command for mpv :

  mpv --no-config --framedrop=no --geometry=[geo settings] --no-border udp://232.255.23.23:1234

  made the duration correct, however this introduces huge artifacts in the videos sometimes.

  I read that using -re for streaming can cause these frame drops, so I tried putting a static number of fps for both file input and output stream, for example :

  ffmpeg -re -i PATH_TO_FILE.mp4 -c copy -r 25 -f mpegts udp://232.255.23.23:1234

  This reads the file at native framerate and outputs the stream at 25fps, and it appears to have the timing duration correct, but it also causes occasional articats and I think has worse qualit overall. Output from mpv when one of the artifacts happened :

  [ffmpeg/video] h264: cabac decode of qscale diff failed at 85 19
  
  [ffmpeg/video] h264: error while decoding MB 85 19, bytestream 85515

  I also tried using --untimed or --no-cache in mpv, but this causes stutters in the video

  I’m also getting requent Invalid video timestamp warnings in MPV, for example : Invalid video timestamp: 1.208333 -> -8.711667

  Playing in mpv without --no-config and with --untimed added also causes frequent artifacts :

  V: -00:00:00 / 00:00:00 Cache:  0s+266KB
  
  [ffmpeg/video] h264: Invalid NAL unit 8, skipping.
  
  V: -00:00:00 / 00:00:00 Cache:  0s+274KB
  
  [ffmpeg/video] h264: Reference 4 >= 4
  
  [ffmpeg/video] h264: error while decoding MB 6 0, bytestream 31474
  
  [ffmpeg/video] h264: error while decoding MB 78 49, bytestream -12
  
  V: 00:00:06 / 00:00:00 Cache:  5s+11KB
  
  Invalid video timestamp: 6.288333 -> -8.724933
  
  V: -00:00:05 / 00:00:00 Cache:  3s+0KB
  
  [ffmpeg/video] h264: Invalid NAL unit 8, skipping.
  
  [ffmpeg/video] h264: error while decoding MB 59 24, bytestream -27
  
  V: -00:00:04 / 00:00:00 Cache:  3s+0KB
  
  [ffmpeg/video] h264: Reference 4 >= 3
  
  [ffmpeg/video] h264: error while decoding MB 5 2, bytestream 13402
  
  V: -00:00:03 / 00:00:00 Cache:  2s+0KB
  
  [ffmpeg/video] h264: Reference 5 >= 4
  
  [ffmpeg/video] h264: error while decoding MB 51 21, bytestream 9415

  I tried playing the stream with ffplay and it also caused the videos to be "played" 20 seconds longer.
  Is there any way to keep the streaming duration intact and prevent those huge artifacts ? These aren’t any huge video files, they are few MB each, everything happens in local network so the latencies are minimal.

  Output from ffmpeg when streaming one of the files :

   libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SDM.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.48.100
  
    Duration: 00:00:20.00, start: 0.000000, bitrate: 1883 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1680x800 [SAR 1:1 DAR 21:10], 1880 kb/s, 24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
  Output #0, mpegts, to 'udp://232.255.23.23:1234':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.83.100
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1680x800 [SAR 1:1 DAR 21:10], q=2-31, 1880 kb/s, 24 fps, 24 tbr, 90k tbn, 25 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=  480 fps= 24 q=-1.0 Lsize=    5009kB time=00:00:19.87 bitrate=2064.7kbits/s speed=   1x
  
  video:4592kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 9.082929%

  Edit : all the files don’t contain any audio, so it should be even less traffic on the network