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• avcodec : Remove libaacplus

  24 janvier 2016, par Timothy Gu

  avcodec : Remove libaacplus
  TODO : bump minor
  It’s inferior in quality to fdk-aac and has an arguably more problematic
  
  license.
  As early as 2012, a HydrogenAudio user reported :
  > It has however one huge advantage : much better quality at low bitrates than
  
  > faac and libaacplus.
  (https://hydrogenaud.io/index.php?PHPSESSID=ckiq394pdglka0kj2fin6ij8t7&topic=95989.msg804633#msg804633)
  I myself have made a few spectrograms for a comparison of the two
  
  encoders as well. The FDK output is consistently better than the
  
  libaacplus one, in all bitrates I tested.
  libaacplus license is 3GPP + LGPLv2. 3GPP copyright notice is completely
  
  proprietory, as follows :
  > No part may be reproduced except as authorized by written permission.
  
  >
  
  > The copyright and the foregoing restriction extend to reproduction in
  
  > all media.
  
  >
  
  > © 2008, 3GPP Organizational Partners (ARIB, ATIS, CCSA, ETSI, TTA, TTC).
  
  >
  
  > All rights reserved.
  (The latest 26410-d00 zip from 3GPP has the same notice, but the copyright
  
  year is changed to 2015)
  The copyright part of the FDK AAC license (section 2) is a copyleft
  
  license that permits redistribution under certain conditions (and
  
  therefore the LGPL + libfdk-aac combination is not prohibited by
  
  configure) :
  > Redistribution and use in source and binary forms, with or without
  
  > modification, are permitted without payment of copyright license fees
  
  > provided that you satisfy the following conditions :
  
  >
  
  > You must retain the complete text of this software license in
  
  > redistributions of the FDK AAC Codec or your modifications thereto in
  
  > source code form.
  
  >
  
  > You must retain the complete text of this software license in the
  
  > documentation and/or other materials provided with redistributions of
  
  > the FDK AAC Codec or your modifications thereto in binary form.
  
  >
  
  > You must make available free of charge copies of the complete source
  
  > code of the FDK AAC Codec and your modifications thereto to recipients
  
  > of copies in binary form.
  
  >
  
  > The name of Fraunhofer may not be used to endorse or promote products
  
  > derived from this library without prior written permission.
  
  >
  
  > You may not charge copyright license fees for anyone to use, copy or
  
  > distribute the FDK AAC Codec software or your modifications thereto.
  
  >
  
  > Your modified versions of the FDK AAC Codec must carry prominent
  
  > notices stating that you changed the software and the date of any
  
  > change. For modified versions of the FDK AAC Codec, the term
  
  > "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the
  
  > term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec
  
  > Library for Android."
  

  • [DH] Changelog
  • [DH] LICENSE.md
  • [DH] configure
  • [DH] doc/general.texi
  • [DH] doc/platform.texi
  • [DH] libavcodec/Makefile
  • [DH] libavcodec/allcodecs.c
  • [DH] libavcodec/libaacplus.c

• What H.264 encoding profile is considered "safe" in 2017 ?

  12 avril 2017, par degenerate

  Have streaming services adopted a certain H.264 profile as "standard" or "safe" for use in 2017 ? For example are all modern Youtube videos at least "High" profile for H.264 ?

  Or does Youtube continue to serve all encoding profiles, down to "Baseline" depending on hardware ?

  I would like to start encoding my videos with "High" profile or higher, but cannot find any recent documentation on which profile is standard practice or safe to use.

  Searching google is not fruitful :

  Old blog post from 2008 :
  http://blog.mediacoderhq.com/h264-profiles-and-levels/

  Old blog post from 2014 :
  http://leightronix.com/blog/when-to-choose-high-main-and-baseline-while-encoding-h-264/

  I cannot find any updated 2017 information on this.

• Trying to convert an mp3 file to a Numpy Array, and ffmpeg just hangs

  5 juillet 2016, par Rich

  I’m working on a music classification methodology with Scikit-learn, and the first step in that process is converting a music file to a numpy array.

  After unsuccessfully trying to call ffmpeg from a python script, I decided to simply pipe the file in directly :

  FFMPEG_BIN = "ffmpeg"
  
  cwd = (os.getcwd())
  
  dcwd = (cwd + "/temp")
  
  if not os.path.exists(dcwd): os.makedirs(dcwd)
  
  
  
  folder_path = sys.argv[1]
  
  f = open("test.txt","a")
  
  
  
  for f in glob.glob(os.path.join(folder_path, "*.mp3")):
  
      ff = f.replace("./", "/")
  
      print("Name: " + ff)
  
      aa = (cwd + ff)
  
  
  
      command = [ FFMPEG_BIN,
  
          '-i',  aa,
  
          '-f', 's16le',
  
          '-acodec', 'pcm_s16le',
  
          '-ar', '22000', # ouput will have 44100 Hz
  
          '-ac', '1', # stereo (set to '1' for mono)
  
          '-']
  
  
  
      pipe = sp.Popen(command, stdout=sp.PIPE, bufsize=10**8)
  
      raw_audio = pipe.proc.stdout.read(88200*4)
  
      audio_array = numpy.fromstring(raw_audio, dtype="int16")
  
      print (str(audio_array))
  
      f.write(audio_array + "\n")

  The problem is, when I run the file, it starts ffmpeg and then does nothing :

  [mp3 @ 0x1446540] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from '/home/don/Code/Projects/MC/Music/Spaz.mp3':
  
    Metadata:
  
      title           : Spaz
  
      album           : Seeing souns
  
      artist          : N*E*R*D
  
      genre           : Hip-Hop
  
      encoder         : Audiograbber 1.83.01, LAME dll 3.96, 320 Kbit/s, Joint Stereo, Normal quality
  
      track           : 5/12
  
      date            : 2008
  
    Duration: 00:03:50.58, start: 0.000000, bitrate: 320 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
  Output #0, s16le, to 'pipe:':
  
    Metadata:
  
      title           : Spaz
  
      album           : Seeing souns
  
      artist          : N*E*R*D
  
      genre           : Hip-Hop
  
      date            : 2008
  
      track           : 5/12
  
      encoder         : Lavf56.4.101
  
      Stream #0:0: Audio: pcm_s16le, 22000 Hz, mono, s16, 352 kb/s
  
      Metadata:
  
        encoder         : Lavc56.1.100 pcm_s16le
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help

  It just sits there, hanging, for far longer than the song is. What am I doing wrong here ?,