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• Mimic Audacity amplification with Pydub

  16 août 2022, par Unisionzz

  For my music library I have used Audacity for recent years to amplify the music to similar levels of loudness ; technically speaking this is not completely true, but for me it is sufficient. However, as it is tedious to do this all by hand, I decided to write a Python code to automate this process for me. The code after the imported package(s) and defined functions will run in a loop in which the filename changes depending on which song is processed.

  


  The difficult part is that I have not yet been able to find a consistent way to amplify different songs so that when the output files are put through Audacity, it will not want to change the amplitude by more than 0.1 dB(FS).

  


  Below are two attempts which seem to have come closest to the desirable output ; other methods that I have tried were either less succesfull or resulted in clipping.

  


  The first attempt finds the maximum dBFS of the song and then applies a gain in order for the maximum dBFS to equal 0 (I have also tried this method with sound.dBFS and sound.apply_gain, but results seem more mixed than the attempt below) :

  


  from pydub import AudioSegment

def change_amplitude(sound, target_dBFS):
    change_in_dBFS = target_dBFS - sound.max_dBFS
    return sound.apply_gain_stereo(change_in_dBFS)

# Audio is gathered from a hard coded path
s = AudioSegment.from_file(Dir+filename+".mp3", "mp3")
amp_s = change_amplitude(s, 0)
amp_s.export(Dir+filename+".mp3", format = "mp3")


  


  The second attempt finds the amplitude and the maximum allowable amplitude (before clipping), recalculates both to dB and then adds the dB_diff to the sound :

  


  import numpy as np
from pydub import AudioSegment

s = AudioSegment.from_file(Dir+filename+".mp3", "mp3")

# Get dB amplitude of song and maximum allowable value
dB_sound = 20*np.log10(s.max)
dB_max = 20*np.log10(s.max_possible_amplitude)
dB_diff = dB_max - dB_sound

amp_sound = s + dB_diff


  


  Summarizing, I would like to import a music file, amplify it similar to Audacity amplification and then export the file again.

  


• How does MPG determine its default audio track ?

  1er octobre 2022, par codedread

  I have some mpg files that I transcoded from DVDs I bought a long time ago (maybe 20 years ago). ffprobe :

  


  Input #0, mpeg, from 'da-orig.mpg':
  Duration: 00:06:59.44, start: 0.044100, bitrate: 6354 kb/s
  Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, progressive), 720x480 [SAR 8:9 DAR 4:3], Closed Captions, 31 fps, 59.94 tbr, 90k tbn
    Side data:
      cpb: bitrate max/min/avg: 7500000/0/0 buffer size: 1835008 vbv_delay: N/A
  Stream #0:1[0x85]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
  Stream #0:2[0x83]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
  Stream #0:3[0x81]: Audio: ac3, 48000 Hz, mono, fltp, 192 kb/s
  Stream #0:4[0x80]: Audio: ac3, 48000 Hz, mono, fltp, 192 kb/s


  


  This shows there are 4 audio streams. When I play this file in VLC / QuickTime it seems that Audio Track 4 is the default. I'd like to understand how this is chosen. Is it something within the mpg container format or are players choosing the stream that has the lowest id (0x80) ?

  


  More background, when I try to turn this into a mp4 file with the following command :

  


  ffmpeg -i da-orig.mpg -c copy -map 0 da-copy.mp4

  


  I get roughly the same size file, but the default audio track is stream #0:1[0x85].

  


  What I want is an equivalent mp4 file (so the same audio track chosen).

  


• fftools/ffmpeg : drop the -async option

  30 août 2022, par Anton Khirnov

  fftools/ffmpeg : drop the -async option
  It has been deprecated in favor of the aresample filter for almost 10
  
  years.
  Another thing this option can do is drop audio timestamps and have them
  
  generated by the encoding code or the muxer, but
  
   for encoding, this can already be done with the setpts filter
  
   for muxing this should almost never be done as timestamp generation by
  
    the muxer is deprecated, but people who really want to do this can use
  
    the setts bitstream filter
  

  • [DH] doc/ffmpeg.texi
  • [DH] fftools/ffmpeg.c
  • [DH] fftools/ffmpeg.h
  • [DH] fftools/ffmpeg_filter.c
  • [DH] fftools/ffmpeg_mux.c
  • [DH] fftools/ffmpeg_opt.c