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• network : Add RFC 8305 style "Happy Eyeballs"/"Fast Fallback" helper function

  10 août 2018, par Martin Storsjö

  network : Add RFC 8305 style "Happy Eyeballs"/"Fast Fallback" helper function
  For cases with dual stack (IPv4 + IPv6) connectivity, but where one
  
  stack potentially is less reliable, strive to trying to connect over
  
  both protocols in parallel, using whichever address connected first.
  In cases with a hostname resolving to multiple IPv4 and IPv6
  
  addresses, the current connection mechanism would try all addresses
  
  in the order returned by getaddrinfo (with all IPv6 addresses ordered
  
  before the IPv4 addresses normally). If connection attempts to the
  
  IPv6 addresses return quickly with an error, this was no problem, but
  
  if they were unsuccessful leading up to timeouts, the connection process
  
  would have to wait for timeouts on all IPv6 target addresses before
  
  attempting any IPv4 address.
  Similar to what RFC 8305 suggests, reorder the list of addresses to
  
  try connecting to, interleaving address families. After starting one
  
  connection attempt, start another one in parallel after a small delay
  
  (200 ms as suggested by the RFC).
  For cases with unreliable IPv6 but reliable IPv4, this should make
  
  connection attempts work as reliably as with plain IPv4, with only an
  
  extra 200 ms of connection delay.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavformat/network.c
  • [DBH] libavformat/network.h

• Convert mp3 to AAC with mpeg-2 container (FFMPEG)

  18 mars 2016, par jsurf

  I’m trying to convert an mp3 audio file to an AAC file with FFMPEG, and I need the audio to be wrapped in an MPEG-2 container.
  The resulting AAC file needs to be AAC-LC (Low Complexity), 1-channel, CBR mode, 44100 sample rate, and 48kb/s bitrate, so I use this command :

  ffmpeg -y -i input.mp3 -ar 44100 -ab 48k -acodec libfdk_aac -ac 1 output.aac

  But when I examine the ADTS headers, the audio file is always being wrapped in an MPEG-4 container. I have tried all the codecs listed here but I still end up with an mpeg-4 container wrapped around the audio : http://trac.ffmpeg.org/wiki/AACEncodingGuide.

  Here are the headers I get when examining the AAC output file :

  mpeg_type : ’MPEG4’,
  profile : 2,
  profile_name : ’AAC LC’,
  sample_freq : 44100,
  channel_config : 1,
  channels : 1,
  frame_length : 139,
  buffer_fullness : 157,
  number_of_frames : 1,
  frames_per_sec : 43.06640625

  Any ideas as to why ffmpeg wraps an mp4 container around the audio ? Can I get around this somehow ? Are there any other encoders I can try aside from FFMPEG ? I was giving FAAC encoder a shot and it gives me the proper encoding and ADTS headers, but alas it does not support mp3, only WAV.

• Setting qscale programmatically when using MPEG4 encoder ( for constant quality / VBR)

  14 février 2019, par Dennis

  i implemented the possibility to encode various self-rendered video-frames with MPEG4 codec and create an .mp4 video file. This works fine. Now i want to add the possibility to define a quality slider (0-100%) to parameterize a factor for constant quality (VBR). I don’t know how to do that.

  I found out that -qscale seems to do what i want, so i looked in ffmpeg_opt.c what happens there and tried the same :

  config.codecContext->flags |= AV_CODEC_FLAG_QSCALE;
  
  config.codecContext->global_quality = FF_QP2LAMBDA * QualityLvl;

  with :

  • "config.codecContext" being the code context
  • "FF_QP2LAMBDA" being 118
  • "QualityLvl" is the "factor for constant quality" (has to be an int between 1 and 31 according to this :
    https://trac.ffmpeg.org/wiki/Encode/MPEG-4)

  The problem is, that it actually doesn’t matter if "QualityLvl" is 1,2 or 30 it always results in the same file size and a visually same(?) video file. I would have expected file size and quality differences ?!