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• How To Extract The Duration From A Webm Using FFprobe

  8 décembre 2018, par Sixtoo

  I have the following code which stores the duration of a video file in a variable called duration :

  for /f %%i in ('ffprobe -select_streams v:0 -show_entries stream^=duration -of default^=noprint_wrappers^=1:nokey^=1 input.avi') do set duration=%%i

  However, when I try to get the duration of a .webm file I get N/A. I used the answer here How to determine webm duration using ffprobe which helped me to be able to see the duration of a webm when using ffprobe. But for some reason I can see the duration in the output of ffprobe but I cannot manage to store it in a variable.

  Please help me with this. Thank you

  ---

  Here is the command and output :

  Command :

  for /f %%i in ('ffprobe -select_streams v:0 -show_entries stream^=duration -of default^=noprint_wrappers^=1:nokey^=1 webm_copy.webm') do set duration=%%i
  
  
  
  echo %duration%

  Output :

  D:\SOFTWARE\ffmpeg\bin\test\go>for /F %i in ('ffprobe -select_streams v:0 -show_entries stream=duration -of default=noprint_wrappers=1:nokey=1 webm_copy.webm') do set duration=%i
  
  ffprobe version N-80066-g566be4f Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 24.100 / 55. 24.100
  
    libavcodec     57. 43.100 / 57. 43.100
  
    libavformat    57. 37.100 / 57. 37.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 46.100 /  6. 46.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, matroska,webm, from 'webm_copy.webm':
  
    Metadata:
  
      encoder         : Lavf57.37.100
  
    Duration: 00:01:31.44, start: 0.000000, bitrate: 278 kb/s
  
      Stream #0:0: Video: vp8, yuv420p, 480x360, SAR 1:1 DAR 4:3, 29.97 fps, 29.97 tbr, 1k tbn (default)
  
      Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
  
  
  
  D:\SOFTWARE\ffmpeg\bin\test\go>set duration=N/A
  
  
  
  D:\SOFTWARE\ffmpeg\bin\test\go>echo N/A
  
  N/A

• libzvbi-teletextdec : split dvb packet to slices

  1er mars 2014, par Marton Balint

  libzvbi-teletextdec : split dvb packet to slices
  Instead of using the demux function of libzvbi to split the packet to slices
  
  (vbi lines), lets do it ourselves.
   eliminates the 1 frame delay between page input and output
  
   handles non-ascending line numbers more gracefully
  
   enables us to return error codes on some invalid packets instead of silently
  
    ignoring them
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] libavcodec/libzvbi-teletextdec.c

• RTP and H.264 (Packetization Mode 1)... Decoding RAW Data... Help understanding the audio and STAP-A packets

  12 février 2014, par Lane

  I am attempting to re-create a video from a Wireshark capture. I have researched extensively and the following links provided me with the most useful information...

  How to convert H.264 UDP packets to playable media stream or file (defragmentation) (and the 2 sub-links)
  H.264 over RTP - Identify SPS and PPS Frames

  ...I understand from these links and RFC (RTP Payload Format for H.264 Video) that...

  • The Wireshark capture shows a client communicating with a server via RTSP/RTP by making the following calls... OPTIONS, DESCRIBE, SETUP, SETUP, then PLAY (both audio and video tracks exist)

  • The RTSP response from PLAY (that contains the Sequence and Picture Parameter Sets) contains the following (some lines excluded)...

  Media Description, name and address (m) : audio 0 RTP/AVP 0
  Media Attribute (a) : rtpmap:0 PCMU/8000/1
  Media Attribute (a) : control:trackID=1
  Media Attribute (a) : x-bufferdelay:0

  Media Description, name and address (m) : video 0 RTP/AVP 98
  Media Attribute (a) : rtpmap:98 H264/90000
  Media Attribute (a) : control:trackID=2
  Media Attribute (a) : fmtp:98 packetization-mode=1 ;profile-level-id=4D0028 ;sprop-parameter-sets=J00AKI2NYCgC3YC1AQEBQAAA+kAAOpg6GAC3IAAzgC7y40MAFuQABnAF3lwWNF3A,KO48gA==

  Media Description, name and address (m) : metadata 0 RTP/AVP 100
  Media Attribute (a) : rtpmap:100 IQ-METADATA/90000
  Media Attribute (a) : control:trackID=3

  ...the packetization-mode=1 means that only NAL Units, STAP-A and FU-A are accepted

  • The streaming RTP packets (video only, DynamicRTP-Type-98) arrive in the following order...

  1x
  [RTP Header]
  0x78 0x00 (Type is 24, meaning STAP-A)
  [Remaining Payload]

   36x
  [RTP Header]
  0x7c (Type is 28, meaning FU-A) then either 0x85 (first) 0x05 (middle) or 0x45 (last)
  [Remaining Payload]

  1x
  [RTP Header]
  0x18 0x00 (Type is 24, meaning STAP-A)
  [Remaining Payload]

  8x
  [RTP Header]
  0x5c (Type is 28, meaning FU-A) then either 0x81 (first) 0x01 (middle) or 0x41 (last)
  [Remaining Payload]

  ...the cycle then repeats... typically there are 29 0x18/0x5c RTP packets for each 0x78/0x7c packet

  • Approximately every 100 packets, there is an audio RTP packet, all have their Marker set to true and their sequence numbers ascend as expected. Sometimes there is an individual RTP audio packet and sometimes there are three, see a sample one here...

  RTP 1042 PT=ITU-T G.711 PCMU, SSRC=0x238E1F29, Seq=31957, Time=1025208762, Mark

  ...also, the type of each audio RTP packet is different (as far as first bytes go... I see 0x4e, 0x55, 0xc5, 0xc1, 0xbc, 0x3c, 0x4d, 0x5f, 0xcc, 0xce, 0xdc, 0x3e, 0xbf, 0x43, 0xc9, and more)

  • From what I gather... to re-create the video, I first need to create a file of the format

  0x000001 [SPS Payload]
  0x000001 [PPS Payload]
  0x000001 [Complete H.264 Frame (NAL Byte, followed by all fragmented RTP payloads without the first 2 bytes)
  0x000001 [Next Frame]
  Etc...

  I made some progress where I can run "ffmpeg -i file" without it saying a bad input format or unable to find codec. But currently it complains something about MP3. My questions are as follows...

  1. Should I be using the SPS and PPS payload returned by the response to the DESCRIBE RTSP call or use the data sent in the first STAP-A RTP packets (0x78 and 0x18) ?

  2. How does the file format change to incorporate the audio track ?

  3. Why is the audio track payload headers all over the place and how can I make sense / utilize them ?

  4. Is my understanding of anything incorrect ?

  Any help is GREATLY appreciated, thanks !