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• Compiling FFMPEG on CentOS DigitalOcean

  29 juillet 2015, par coder_uk

  I set up a DigitalOcean instance running CentOS 6.5 and successfully followed the guide to compile FFMPEG (https://trac.ffmpeg.org/wiki/CompilationGuide/Centos). Hurrah !

  But of course I realised that by default, DigitalOcean creates a root user and so ffmpeg now lives in /root/bin/ffmpeg. Which isn’t ideal because when I want to exec the ffmpeg bin from nginx, I would have to run nginx as root for it to have permission.

  Questions ...

  1) Long-shot, but presumably if I change the owner of the ffmpeg binary to nginx, it still won’t work, because nginx won’t be able to access the /root folder it is in. Correct ?

  2) I could run nginx as root (’user root’). But this seems like a very bad idea. Correct ?

  3) Which leaves me with the option of creating a new user, and then compiling ffmpeg into its home folder. But : which user ? EC2 creates ’ec2-user’, so should I make my own equivalent for DO ? But then won’t I have to run nginx as that user, else I’ll run into the same problem ?

  Or should I compile ffmpeg into the ’nginx’ home folder, if indeed it has one ? Is that how it is supposed to be done ?

  Since compiling ffmpeg takes ages, I don’t want to keep doing it, and the static files all seem very out of date. Thanks

• 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 ?!