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• Getting accurate time from FFMPeg with Objective C (Audio Queue Services)

  2 avril 2012, par Winston

  My iPhone app plays an audio file using FFMPeg.

  I'm getting the elapsed time (to show to user) from the playing audio (in minutes and seconds after converting from microseconds, given by FFMPeg) like so :

  AudioTimeStamp currentTimeStamp;
  
  AudioQueueGetCurrentTime (audioQueue, NULL, &currentTimeStamp, NULL);
  
  
  
  getFFMPEGtime = currentTimeStamp.mSampleTime/self.basicAudioDescription.mSampleRate;
  
  
  
  self.currentAudioTime = [NSString stringWithFormat: @"%02d:%02d",
  
                              (int) getFFMPEGtime / (int)60000000,
  
                              (int) ((getFFMPEGtime % 60000000)/1000000)];

  Everything works fine, but when I scrub back or forward to play another portion of the song, the elapsed time will go back to zero, no matter the current position. The timer will always zero out.

  I know I'm suposed to do some math to keep track of the old time and the new time, maybe constructing another clock or so, perhaps implementing another callback function, etc... I'm not sure what way I should go.

  My questions are :

  1) What's the best approach to keep track of the elapsed time when going back/forward in a song, avoiding the clock to always going back to zero ?

  2) Should I look deeply into FFMPeg functions or should I stick with Objective-C and Cocoa Touch for solving this problem ?

  Please, I need some advices/ideas from experienced programmers. I'm stuck. Thanks beforehand !

• 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.

  


• ffmpeg convert wav to mp3 using php variable

  17 janvier 2016, par Jeff

  I am working on a php function to upload a .wav and also create an .mp3 version of it in the same folder. So far everything is working how it should except for when I try using :

   shell_exec('ffmpeg -i ' . $file_path . '-f mp2 ' . $mp3name); 

  within the function it isn’t creating the .mp3 in the beats folder.

  I tried

  shell_exec('ffmpeg -i test.wav -f mp2 test.mp3'); 

  on its own and it worked great.

  php Function :

  function upload_a_sound($user_id, $file_temp, $file_extn, $name, $uploader, $keywords) {
  
      $timecode = substr(md5(time()), 0, 10);
  
      $mp3name = 'beats/' . $timecode . '.mp3';
  
      $file_path = 'beats/' . $timecode . '.' . $file_extn;
  
      //$date = date('m-d-Y');
  
      move_uploaded_file($file_temp, $file_path);
  
      shell_exec('ffmpeg -i ' . $file_path . '-f mp2 ' . $mp3name); 
  
      require ('classAudioFile.php');
  
      $AF = new AudioFile;
  
      $AF->loadFile($file_path);
  
      $AF->visual_width=200;
  
      $AF->visual_height=200;
  
      $AF->visual_graph_color="#c491db";
  
      $AF->visual_background_color="#000000";
  
      $AF->visual_grid=false;
  
      $AF->visual_border=false;
  
      $AF->visual_graph_mode=0;
  
      $AF->getVisualization ('images/song/' . $timecode . '.png');
  
      $imageloc = 'images/song/' . $timecode . '.png';
  
      mysql_query("INSERT INTO `content` VALUES ('', '', '$name', '$uploader', '$keywords', '$file_path', '$imageloc', '$mp3name')");
  
  
  
  }