code: from contextlib import closing
  from PIL import Image
  import subprocess
  from audiotsm import phasevocoder
  from audiotsm.io.wav import WavReader, WavWriter
  from scipy.io import wavfile
  import numpy as np
  import re
  import math
  from shutil import copyfile, rmtree
  import os
  import argparse
  from pytube import YouTube
  
  def downloadFile(url):
      name = YouTube(url).streams.first().download()
      newname = name.replace(' ','_')
      os.rename(name,newname)
      return newname
  
  def getMaxVolume(s):
      maxv = float(np.max(s))
      minv = float(np.min(s))
      return max(maxv,-minv)
  
  def copyFrame(inputFrame,outputFrame):
      src = TEMP_FOLDER+"/frame{:06d}".format(inputFrame+1)+".jpg"
      dst = TEMP_FOLDER+"/newFrame{:06d}".format(outputFrame+1)+".jpg"
      if not os.path.isfile(src):
          return False
      copyfile(src, dst)
      if outputFrame%20 == 19:
          print(str(outputFrame+1)+" time-altered frames saved.")
      return True
  
  def inputToOutputFilename(filename):
      dotIndex = filename.rfind(".")
      return filename[:dotIndex]+"_ALTERED"+filename[dotIndex:]
  
  def createPath(s):
      #assert (not os.path.exists(s)), "The filepath "+s+" already exists. Don't want to overwrite it. Aborting."
  
      try:  
          os.mkdir(s)
      except OSError:  
          assert False, "Creation of the directory %s failed. (The TEMP folder may already exist. Delete or rename it, and try again.)"
  
  def deletePath(s): # Dangerous! Watch out!
      try:  
          rmtree(s,ignore_errors=False)
      except OSError:  
          print ("Deletion of the directory %s failed" % s)
          print(OSError)
  
  def writeELD(start, end, number):
      startFrame = int(start % frameRate)
      startSecond = int((start / frameRate) % 60)
      startMinute = int((start / frameRate / 60) % 60)
      startHour = int((start / frameRate / 60 / 60))
  
      endFrame = int(end % frameRate)
      endSecond = int((end / frameRate) % 60)
      endMinute = int((end / frameRate / 60) % 60)
      endHour = int((end / frameRate / 60 / 60))
  
      eld_file = open(OUTPUT_FILE, "a")
      eld_file.write("{0} 001 V C {4}:{3}:{2}:{1} {8}:{7}:{6}:{5} {4}:{3}:{2}:{1} {8}:{7}:{6}:{5}\r\n".format(
          str(number).zfill(3),
          str(startFrame).zfill(2),
          str(startSecond).zfill(2),
          str(startMinute).zfill(2),
          str(startHour).zfill(2),
          str(endFrame).zfill(2),
          str(endSecond).zfill(2),
          str(endMinute).zfill(2),
          str(endHour).zfill(2)
      ))
      eld_file.close()
  
  
  parser = argparse.ArgumentParser(description='Modifies a video file to play at different speeds when there is sound vs. silence.')
  parser.add_argument('--input_file', type=str,  help='the video file you want modified')
  parser.add_argument('--edl', type=bool,  help='EDL export option. (Supports only cuts off)')
  parser.add_argument('--url', type=str, help='A youtube url to download and process')
  parser.add_argument('--output_file', type=str, default="", help="the output file. (optional. if not included, it'll just modify the input file name)")
  parser.add_argument('--silent_threshold', type=float, default=0.03, help="the volume amount that frames' audio needs to surpass to be consider \"sounded\". It ranges from 0 (silence) to 1 (max volume)")
  parser.add_argument('--sounded_speed', type=float, default=1.00, help="the speed that sounded (spoken) frames should be played at. Typically 1.")
  parser.add_argument('--silent_speed', type=float, default=5.00, help="the speed that silent frames should be played at. 999999 for jumpcutting.")
  parser.add_argument('--frame_margin', type=float, default=1, help="some silent frames adjacent to sounded frames are included to provide context. How many frames on either the side of speech should be included? That's this variable.")
  parser.add_argument('--sample_rate', type=float, default=44100, help="sample rate of the input and output videos")
  parser.add_argument('--frame_rate', type=float, default=30, help="frame rate of the input and output videos. optional... I try to find it out myself, but it doesn't always work.")
  parser.add_argument('--frame_quality', type=int, default=3, help="quality of frames to be extracted from input video. 1 is highest, 31 is lowest, 3 is the default.")
  
  args = parser.parse_args()
  
  
  
  frameRate = args.frame_rate
  SAMPLE_RATE = args.sample_rate
  SILENT_THRESHOLD = args.silent_threshold
  FRAME_SPREADAGE = args.frame_margin
  NEW_SPEED = [args.silent_speed, args.sounded_speed]
  if args.url != None:
      INPUT_FILE = downloadFile(args.url)
  else:
      INPUT_FILE = args.input_file
  URL = args.url
  FRAME_QUALITY = args.frame_quality
  EDL = args.edl
  
  assert INPUT_FILE != None , "why u put no input file, that dum"
      
  OUTPUT_FILE = inputToOutputFilename(INPUT_FILE)
  if len(args.output_file) >= 1:
      OUTPUT_FILE = args.output_file
  else:
      OUTPUT_FILE = inputToOutputFilename(INPUT_FILE)
  
  TEMP_FOLDER = "TEMP"
  AUDIO_FADE_ENVELOPE_SIZE = 400 # smooth out transitiion's audio by quickly fading in/out (arbitrary magic number whatever)
      
  createPath(TEMP_FOLDER)
  
  if not EDL:
      command = "ffmpeg -i "+INPUT_FILE+" -qscale:v "+str(FRAME_QUALITY)+" "+TEMP_FOLDER+"/frame%06d.jpg -hide_banner"
      subprocess.call(command, shell=True)
  
  command = "ffmpeg -i "+INPUT_FILE+" -ab 160k -ac 2 -ar "+str(SAMPLE_RATE)+" -vn "+TEMP_FOLDER+"/audio.wav"
  
  subprocess.call(command, shell=True)
  
  command = "ffmpeg -i "+TEMP_FOLDER+"/input.mp4 2>&1"
  f = open(TEMP_FOLDER+"/params.txt", "w")
  subprocess.call(command, shell=True, stdout=f)
  
  
  
  sampleRate, audioData = wavfile.read(TEMP_FOLDER+"/audio.wav")
  audioSampleCount = audioData.shape[0]
  maxAudioVolume = getMaxVolume(audioData)
  
  f = open(TEMP_FOLDER+"/params.txt", 'r+')
  pre_params = f.read()
  f.close()
  params = pre_params.split('\n')
  for line in params:
      m = re.search('Stream #.*Video.* ([0-9]*) fps',line)
      if m is not None:
          frameRate = float(m.group(1))
  
  samplesPerFrame = sampleRate/frameRate
  
  audioFrameCount = int(math.ceil(audioSampleCount/samplesPerFrame))
  
  hasLoudAudio = np.zeros((audioFrameCount))
  
  
  
  for i in range(audioFrameCount):
      start = int(i*samplesPerFrame)
      end = min(int((i+1)*samplesPerFrame),audioSampleCount)
      audiochunks = audioData[start:end]
      maxchunksVolume = float(getMaxVolume(audiochunks))/maxAudioVolume
      if maxchunksVolume >= SILENT_THRESHOLD:
          hasLoudAudio[i] = 1
  
  chunks = [[0,0,0]]
  shouldIncludeFrame = np.zeros((audioFrameCount))
  for i in range(audioFrameCount):
      start = int(max(0,i-FRAME_SPREADAGE))
      end = int(min(audioFrameCount,i+1+FRAME_SPREADAGE))
      shouldIncludeFrame[i] = np.max(hasLoudAudio[start:end])
      if (i >= 1 and shouldIncludeFrame[i] != shouldIncludeFrame[i-1]): # Did we flip?
          chunks.append([chunks[-1][1],i,shouldIncludeFrame[i-1]])
  
  chunks.append([chunks[-1][1],audioFrameCount,shouldIncludeFrame[i-1]])
  chunks = chunks[1:]
  
  outputAudioData = np.zeros((0,audioData.shape[1]))
  outputPointer = 0
  
  lastExistingFrame = None
  edlFrameNumber = 0
  if EDL and os.path.isfile(OUTPUT_FILE):
      os.remove(OUTPUT_FILE)
  for chunk in chunks:
      if EDL:
          if (chunk[2] == True):
              edlFrameNumber += 1
              writeELD(chunk[0], chunk[1], edlFrameNumber)
          continue
      
      audioChunk = audioData[int(chunk[0]*samplesPerFrame):int(chunk[1]*samplesPerFrame)]
      
      sFile = TEMP_FOLDER+"/tempStart.wav"
      eFile = TEMP_FOLDER+"/tempEnd.wav"
      wavfile.write(sFile,SAMPLE_RATE,audioChunk)
      with WavReader(sFile) as reader:
          with WavWriter(eFile, reader.channels, reader.samplerate) as writer:
              tsm = phasevocoder(reader.channels, speed=NEW_SPEED[int(chunk[2])])
              tsm.run(reader, writer)
      _, alteredAudioData = wavfile.read(eFile)
      leng = alteredAudioData.shape[0]
      endPointer = outputPointer+leng
      outputAudioData = np.concatenate((outputAudioData,alteredAudioData/maxAudioVolume))
  
      #outputAudioData[outputPointer:endPointer] = alteredAudioData/maxAudioVolume
  
      # smooth out transitiion's audio by quickly fading in/out
      
      if leng < AUDIO_FADE_ENVELOPE_SIZE:
          outputAudioData[outputPointer:endPointer] = 0 # audio is less than 0.01 sec, let's just remove it.
      else:
          premask = np.arange(AUDIO_FADE_ENVELOPE_SIZE)/AUDIO_FADE_ENVELOPE_SIZE
          mask = np.repeat(premask[:, np.newaxis],2,axis=1) # make the fade-envelope mask stereo
          outputAudioData[outputPointer:outputPointer+AUDIO_FADE_ENVELOPE_SIZE] *= mask
          outputAudioData[endPointer-AUDIO_FADE_ENVELOPE_SIZE:endPointer] *= 1-mask
  
      startOutputFrame = int(math.ceil(outputPointer/samplesPerFrame))
      endOutputFrame = int(math.ceil(endPointer/samplesPerFrame))
      for outputFrame in range(startOutputFrame, endOutputFrame):
          inputFrame = int(chunk[0]+NEW_SPEED[int(chunk[2])]*(outputFrame-startOutputFrame))
          didItWork = copyFrame(inputFrame,outputFrame)
          if didItWork:
              lastExistingFrame = inputFrame
          else:
              copyFrame(lastExistingFrame,outputFrame)
  
      outputPointer = endPointer
  
  if not EDL:
      wavfile.write(TEMP_FOLDER+"/audioNew.wav",SAMPLE_RATE,outputAudioData)
  
  '''
  outputFrame = math.ceil(outputPointer/samplesPerFrame)
  for endGap in range(outputFrame,audioFrameCount):
      copyFrame(int(audioSampleCount/samplesPerFrame)-1,endGap)
  '''
  
  if not EDL:
      command = "ffmpeg -framerate "+str(frameRate)+" -i "+TEMP_FOLDER+"/newFrame%06d.jpg -i "+TEMP_FOLDER+"/audioNew.wav -strict -2 "+OUTPUT_FILE
      subprocess.call(command, shell=True)
  
  deletePath(TEMP_FOLDER)
  
  
apis: numpy.zeros scipy.io.wavfile.write numpy.repeat numpy.arange numpy.max scipy.io.wavfile.read numpy.min numpy.concatenate