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• KLV data in RTP stream

  18 septembre 2013, par Ardoramor

  I have implemented RFC6597 to stream KLV is RTP SMPTE336M packets. Currently, my SDP looks like this :

  v=2
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=Unnamed
  
  i=N/A
  
  c=IN IP4 192.168.1.6
  
  t=0 0
  
  a=recvonly
  
  m=video 8202 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  a=fmtp:96 packetization-mode=1;profile-level-id=428028;sprop-parameter-sets=Z0KAKJWgKA9E,aM48gA==;
  
  a=control:trackID=0
  
  m=application 8206 RTP/AVP 97
  
  a=rtpmap:97 smpte336m/1000
  
  a=control:trackID=1

  I try to remux the RTP stream with FFmpeg like so :

  ffmpeg.exe -i test.sdp -map 0:0 -map 0:1 -c:v copy -c:d copy test.m2ts

  I get the following output with FFmpeg :

  ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      52. 18.100 / 52. 18.100
  
    libavcodec     54. 92.100 / 54. 92.100
  
    libavformat    54. 63.104 / 54. 63.104
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 42.103 /  3. 42.103
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  [aac @ 0000000002137900] Sample rate index in program config element does not match the sample rate index configured by the container.
  
      Last message repeated 1 times
  
  [aac @ 0000000002137900] decode_pce: Input buffer exhausted before END element found
  
  [h264 @ 00000000002ce540] Missing reference picture, default is 0
  
  [h264 @ 00000000002ce540] decode_slice_header error
  
  [sdp @ 00000000002cfa80] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, sdp, from 'C:\Users\dragan\Documents\Workspace\Android\uvlens\tests\test.sdp':
  
    Metadata:
  
      title           : Unnamed
  
      comment         : N/A
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Audio: aac, 32000 Hz, 58 channels, fltp
  
      Stream #0:1: Video: h264 (Baseline), yuv420p, 640x480, 14.83 tbr, 90k tbn, 180k tbc
  
      Stream #0:2: Data: none
  
  File 'C:\Users\dragan\Documents\Workspace\Android\uvlens\tests\test.m2ts' already exists. Overwrite ? [y/N] y
  
  Output #0, mpegts, to 'C:\Users\dragan\Documents\Workspace\Android\uvlens\tests\test.m2ts':
  
    Metadata:
  
      title           : Unnamed
  
      comment         : N/A
  
      encoder         : Lavf54.63.104
  
      Stream #0:0: Video: h264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
  
      Stream #0:1: Data: none
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
    Stream #0:2 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  [mpegts @ 0000000002159940] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 8583659665 >= 8583656110
  
  av_interleaved_write_frame(): Invalid argument

  The problem is that KLV stream packets do not contain have a DTS field. According to the RFC6597 STMPE336M, RTP packet structure is the same as a standard structure :

  4.1.  RTP Header Usage
  
  
  
  This payload format uses the RTP packet header fields as described in
  
  the table below:
  
  
  
  +-----------+-------------------------------------------------------+
  
  | Field     | Usage                                                 |
  
  +-----------+-------------------------------------------------------+
  
  | Timestamp | The RTP Timestamp encodes the instant along a         |
  
  |           | presentation timeline that the entire KLVunit encoded |
  
  |           | in the packet payload is to be presented.  When one   |
  
  |           | KLVunit is placed in multiple RTP packets, the RTP    |
  
  |           | timestamp of all packets comprising that KLVunit MUST |
  
  |           | be the same.  The timestamp clock frequency is        |
  
  |           | defined as a parameter to the payload format          |
  
  |           | (Section 6).                                          |
  
  |           |                                                       |
  
  | M-bit     | The RTP header marker bit (M) is used to demarcate    |
  
  |           | KLVunits.  Senders MUST set the marker bit to '1' for |
  
  |           | any RTP packet that contains the final byte of a      |
  
  |           | KLVunit.  For all other packets, senders MUST set the |
  
  |           | RTP header marker bit to '0'.  This allows receivers  |
  
  |           | to pass a KLVunit for parsing/decoding immediately    |
  
  |           | upon receipt of the last RTP packet comprising the    |
  
  |           | KLVunit.  Without this, a receiver would need to wait |
  
  |           | for the next RTP packet with a different timestamp to |
  
  |           | arrive, thus signaling the end of one KLVunit and the |
  
  |           | start of another.                                     |
  
  +-----------+-------------------------------------------------------+
  
  
  
  The remaining RTP header fields are used as specified in [RFC3550].

  Header from RFC3550 :

  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  
  |V=2|P|X|  CC   |M|     PT      |       sequence number         |
  
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  
  |                           timestamp                           |
  
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  
  |           synchronization source (SSRC) identifier            |
  
  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
  
  |            contributing source (CSRC) identifiers             |
  
  |                             ....                              |
  
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

  RFC's note about placement of KLV data into RTP packet :

  KLVunits small enough to fit into a single RTP
  packet (RTP packet size is up to the implementation but should
  consider underlying transport/network factors such as MTU
  limitations) are placed directly into the payload of the RTP packet,
  with the first byte of the KLVunit (which is the first byte of a KLV
  Universal Label Key) being the first byte of the RTP packet payload.

  My question is where does FFmpeg keep looking for the DTS ?

  Does it interpret the Timestamp field of the RTP packet header as DTS ? If so, I've verified that the timestamps increase (although at different rates) but are not equal to what FFmpeg prints out :

  8583659665 >= 8583656110

• FFMPEG Detect volume of streaming (PHP)

  22 septembre 2013, par Mohamed Mostafa

  I spent last 4 days trying to acheive that but with no luck,

  I am trying to detect volume of streaming link or save audio file, using the FFmpeg I tried every single command line.

  ffmpeg -f lavfi -i amovie=sample1.aac,volumedetect -f null -y test.txt

  Output

  There was a problem! Array (
  
      [0] => FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  
      [1] => built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
  
      [2] => configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  
      [3] => libavutil 50.15. 1 / 50.15. 1
  
      [4] => libavcodec 52.72. 2 / 52.72. 2
  
      [5] => libavformat 52.64. 2 / 52.64. 2
  
      [6] => libavdevice 52. 2. 0 / 52. 2. 0
  
      [7] => libavfilter 1.19. 0 / 1.19. 0
  
      [8] => libswscale 0.11. 0 / 0.11. 0
  
      [9] => libpostproc 51. 2. 0 / 51. 2. 0
  
      [10] => Unknown input format: 'lavf'
  
  )

  Basically my problem now is :

   Unknown input format: 'lavf'

  Any help please

  My FFMpeg Version is

  [root@bea ~]# ffmpeg -formats | grep lavfi

  FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
  configuration : —prefix=/usr —libdir=/usr/lib64 —shlibdir=/usr/lib64 —mandir=/usr/share/man —incdir=/usr/include —disable-avisynth —extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector —param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' —enable-avfilter —enable-avfilter-lavf —enable-libdc1394 —enable-libdirac —enable-libfaac —enable-libfaad —enable-libfaadbin —enable-libgsm —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-librtmp —enable-libschroedinger —enable-libspeex —enable-libtheora —enable-libx264 —enable-gpl —enable-nonfree —enable-postproc —enable-pthreads —enable-shared —enable-swscale —enable-vdpau —enable-version3 —enable-x11grab
  libavutil 50.15. 1 / 50.15. 1
  libavcodec 52.72. 2 / 52.72. 2
  libavformat 52.64. 2 / 52.64. 2
  libavdevice 52. 2. 0 / 52. 2. 0
  libavfilter 1.19. 0 / 1.19. 0
  libswscale 0.11. 0 / 0.11. 0
  libpostproc 51. 2. 0 / 51. 2. 0

  From PHP info

  ffmpeg

  ffmpeg-php version 0.6.0-svn
  ffmpeg-php built on Sep 21 2013 15:38:20
  ffmpeg-php gd support enabled
  ffmpeg libavcodec version Lavc52.72.2
  ffmpeg libavformat version Lavf52.64.2
  ffmpeg swscaler version SwS0.11.0

  Directive Local Value Master Value
  ffmpeg.allow_persistent 0 0
  ffmpeg.show_warnings 0 0

• ffmpeg not finding audio streams ?

  24 septembre 2013, par Jim Miller

  I'm doing some video conversion with ffmpeg v. N-54271-g7f866c1 (a fresh pull from the git sources in late June 2013) on Fedora 19. One of the things I want to do is to concatenate two videos and then convert the result to an mp4. The following code is working well for me :

  ffmpeg -i video_a.mov -i video_b.mov -acodec libfaac -vcodec libx264 
  
       -preset fast -crf 22  -s 940x528 -pix_fmt yuv420p 
  
       -filter_complex '[0:1] [0:0] [1:1] [1:0]  concat=n=2:v=1:a=1 [v] [a]' 
  
       -map '[v]' -map '[a]' output.mp4

  except for a couple of older hunks of test video I've been using. On those, ffmpeg isn't finding the audio stream, and so the above call dies with ffmpeg complaining that Stream specifier ':1' in filtergraph description [0:1][0:0][1:1][1:0] concat=n=2:v=1:a=1 [v] [a] matches no streams. The catch, of course, is that there is an audio stream in the video ; it's just not getting found.

  When ffmpeg starts, I get a description of the input like so :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/var/www/priv/videorising7/raw_take_video/v2261-MTQxMzgwMDM4NzAx.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      creation_time   : 2011-10-13 16:08:18
  
    Duration: 00:00:25.52, start: 0.000000, bitrate: 49 kb/s
  
      Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 608x342, 47 kb/s, 10.03 fps, 10 tbr, 1k tbn, 2k tbc
  
      Metadata:
  
        creation_time   : 2011-10-13 16:08:18
  
        handler_name    : Apple Alias Data Handler

  I suppose the video might just be so old (2006 ?) that I should be lucky that it plays at all. However, I'm able to run these videos through some other ffmpeg jobs (converting from .mov to .mp4, for instance), but those don't require explicitly referencing the audio and video tracks. Any insights out there ?