Sur d’autres sites (8206)

• Having issues making an app with py2app ffmpeg and ffprobe

  23 septembre 2024, par Nuno Mesquita

  I have made an app in OSX Python, that uses FFMEG and ffprobe.

  


  It runs fine in my VS Code.

  


  I want to make a self sustained app, that others can use, without needed to install anything themselves. So, I manually included the static libraries in the bin folder of my main app, and am able to run the app using the libraries inside my bin folder.

  


  I have lost many hours trying to figure this out, and am always getting the same error.

  


  My best shot was being able to open the app inside app/contents/resources and running via Python, but still have the issue saying that ffprobe is not available.

  


  I´m losing my mind...

  


  Should I try a completely different approach ?

  


  For now, I just want my OSX friends to be able to use the app, then I will want it to run on Windows too, but baby steps.

  


  I am not quite sure what to include in setup.py, so I would love to include everything that is in my app folder, to avoid all sorts of errors in missing .pngs and stuff.

  


  I also made sure that libraries are executable (they are about 80MB each (ffmpeg and ffprobe))

  


  Can anyone help =D ???

  


  I described above, tested different ways, checked stuff like redirecting the libraries to the folders of the app :

  


  For the last try my setup.py was :

  


  from setuptools import setup

APP = ['mpc.py']
DATA_FILES = [
    ('bin', ['./bin/ffmpeg', './bin/ffprobe']),
    ('theme/dark', ['./theme/dark/*.png']),
    ('theme/light', ['./theme/light/*.png']),
    ('theme', ['./theme/dark.tcl', './theme/light.tcl']),
    '.',  # Add current directory to ensure everything is included
]

OPTIONS = {
    'argv_emulation': False,
    'packages': ['pydub', 'pygame', 'requests', 'freesound'],
    'includes': ['pydub', 'pygame', 'requests', 'freesound'],
    'resources': ['./bin/ffmpeg', './bin/ffprobe', './theme/dark/*.png', './theme/light/*.png',     './theme/dark.tcl', './theme/light.tcl'],
}

setup(
    app=APP,
    data_files=DATA_FILES,
    options={'py2app': OPTIONS},
    setup_requires=['py2app'],
)


  


  And I also had this in the beginning of my Python script :

  


  # Set ffmpeg and ffprobe paths relative to the app's Resources/bin directory
app_dir = os.path.dirname(os.path.abspath(__file__)) 

# Set ffmpeg and ffprobe paths relative to the app's Resources/bin directory
ffmpeg_path = os.path.join(app_dir, 'bin', 'ffmpeg')
ffprobe_path = os.path.join(app_dir, 'bin', 'ffprobe') 
AudioSegment.converter = ffmpeg_path
AudioSegment.ffprobe = ffprobe_path


  


  Don't know what else to try.

  


• ffmpeg silenceremove - hear what bits are removed

  7 avril 2020, par jimo

  ffmpeg silenceremove is pretty cool. im loving it. i can trim 3 second silences to 2 seconds and reduce a 1.5 hour file of spoken audio down 3 or 4 minutes (depending on the speaker).

  



  once in a while I do hear my choice for stop_threshold (ie-40dB on audio only analog file) does cause the end of a word to be clipped, just here and there when the speaker trails off softly at the end of the word.

  



  is there any way to output what is trimmed to a file ? so I can listen to it and get an idea of just how often this word clipping happens ?

  



  thanks !

  


• waveforms arent as smooth as they should be

  25 mars 2019, par GROVER.

  So I’ve been trying to create a little clone of SoundClouds ’peaks’ waveform. I have noticed that their waveforms are a lot more streamlined
  compared to my sudden shifts in ’loudness’. The dynamic range between peaks never seems to be too steep, and they always represent the perceived loudness of each part of the track pretty well.

  ---

  Here’s an example :

  

  Notice how all the ’drops’ are fairly distinguishable to the ’breakdowns’ in their waveform, but mine is all over the place (apart from the last drop and breakdown, which are kind of similar). There are some minor similarities, but the ’jaggedness’ is still very prominent even in those areas.

  I’m using wav2json as a peaks converter (which is run through the command line and programmed in C++). This is is an example of how I use it :

  /*
  
  *    --channels: mids and min
  
  *    --db-min (minimum level in dB to capture): -35dB
  
  *    --db-max (pretty self explanatory): 6dB
  
  *    -d: use logarithmic instead of linear scale
  
  *    -s (number of peaks to generate): 1800
  
  *    -o (output file): outputfile.json
  
  *    -p (precision of floats): 0
  
  *    -n: no header 
  
  */
  
  exec("wav2json inputfile.wav -s 1800 --channels mid min -d --db-min -35 --db-max 6 -p 0 -o outputfile.json -n");
  
  
  
  $fp     = fopen($tmpOutput, "r");
  
  $json   = fread($fp, filesize($tmpOutput));
  
  // get mids and min from the generated peaks file
  
  $mid    = json_decode($json, true)["mid"];
  
  $min    = json_decode($json, true)["min"];
  
  fclose($fp);
  
  unlink($tmpOutput);
  
  
  
  /* 
  
  *  from here I just combine each mid and min value together and divide by two
  
  *
  
  *  then I normalise all the peaks (instead of each value being between -0.293 to 
  
  *  1.766(just as an example), it is between 0 and 100)
  
  */

  What I’m trying to figure out - and have been trying to for the last few months - is how to get each peak more streamlined and to have the dynamic range of each one look how it actually sounds.

  ---

  What I have tried :

  • ffmpeg eqing
  • actually eqing the highs and lows in a daw and then comparing waveforms
  • using various parameters for wav2json (db min and max, linear etc.)
  • using various compressors and multiband compressors on the track

  All help is appreciated,
  Cheers.