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• 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 ?!

• 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.

• How to get raw bytestring from AudioSegment

  14 juillet 2023, par Yingrjimsch

  Hello I'm trying to get the raw bytestring out of pydub AudioSegment.

  


  I've already tried following :

  


  sound = AudioSegment.from_mp3(file="../source/myfile.mp3")
bytestring = sound.raw_data


  


  and

  


  sound = AudioSegment.from_mp3(file="../source/myfile.mp3")
bytestring = sound._data


  


  but in my case it returns always a bytestring of with len 686866320 Bytes.

  


  if I export it instead to a bytes array like following :

  


  sound = AudioSegment.from_mp3(file="../source/myfile.mp3")
bytes = io.BytesIO()
bytes = sound.export(bytes, format='mp3')


  


  I'm getting a much smaller len of 62301665.

  


  At the end I'd like to have a bytestring and it's real size which (in my case is the real filesize of 62301665) but I cannot figure out how export reduces / compresses the data. The plain datastring of sound.raw_data cannot be played either.

  


  Thank you for any help :)