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• FFMPEG command from Python 3.5 does not actually create audio file

  20 décembre 2017, par Nathan Blaine

  I have a Django web application that accepts user uploaded videos/audio and saves them into a folder ’../WebAppDirectory/media/recordings’.

  I am then using a speech to text API to get a rough transcription of the audio. This is working fine for .wav and .mp4 files, but the web app also accepts videos (.MOV) that I would like to first convert to .wav, then pass off to the API.

  Using ffmpeg from my command line like this

  ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav

  Correctly creates the .wav file from the original .MOV.

  However, when I run this from python with

  subprocess.check_call(command, shell=True)

  ffmpeg responds with

  File ’upload_sample.wav’ already exists. Overwrite ? [y/N]

  While Python tells me

  FileNotFoundError : [Errno 2] No such file or directory : ’C :\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav’

  It is also worth noting that I do not see a ’upload_sample.wav’ file in the media/recordings/ directory.

  This leads me to believe that maybe Python and ffmpeg are looking in different folders, but I am not sure where I am going wrong. When I print the command from the subprocess.check_call and copy/paste it into cmd, the file is created as expected.

  Hoping someone with some experience with ffmpeg/Python subprocess can help shed some light ! Here are the files I am working with :

  Folder Structure

  DjangoWebApp
  
  |---media
  
  |---|---imgs
  
  |---|---recordings
  
  |---|---|---upload_sample.MOV
  
  |---uploaded_audio_to_text.py

  uploaded_audio_to_text.py

  import speech_recognition as sr
  
  from os import path
  
  import os
  
  import subprocess
  
  
  
  
  
  def speech_to_text(file_name):
  
      AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', file_name)
  
      print("Looking at path: ",AUDIO_FILE)
  
      # get extension
  
      AUDIO_FILE_EXT = os.path.splitext(AUDIO_FILE)[1]
  
  
  
      if(AUDIO_FILE_EXT == '.MOV'):
  
          print("File is not .wav: ", AUDIO_FILE_EXT, "found. Converting...")
  
          # We will use subprocess and ffmpeg to convert this .MOV file to .wav, so we can send to API
  
          temp_wav = os.path.splitext(file_name)[0] + '.wav'
  
          print("New audio file will be: ", temp_wav)
  
          # build CMD ffmpeg command
  
          command = "ffmpeg -i "
  
          command += AUDIO_FILE
  
          command += " -ab 160k -ac 2 -ar 44100 -vn "
  
          command += temp_wav
  
  
  
          print("Attempting to run this command: \n",command)
  
          print(subprocess.check_call(command, shell=True))
  
          print("Past Subprocess.call")
  
          AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', temp_wav)
  
          print("AUDIO_FILE now set to: ", AUDIO_FILE)
  
  
  
      else:
  
          # continue with what we are doing
  
          pass
  
  
  
  
  
      r = sr.Recognizer()
  
      with sr.AudioFile(AUDIO_FILE) as source:
  
          audio = r.record(source)  # read the entire audio file
  
          text_transcription = "Sentinel"
  
          # recognize speech using Microsoft Bing Voice Recognition
  
          BING_KEY = "MY_KEY_:)"
  
          try:
  
              text_transcription = r.recognize_bing(audio, key=BING_KEY)
  
          except sr.UnknownValueError:
  
              print("Microsoft Bing Voice Recognition could not understand audio")
  
          except sr.RequestError as e:
  
              print("Could not request results from Microsoft Bing Voice Recognition service; {0}".format(e))
  
  
  
      return text_transcription
  
  
  
  
  
  #my tests
  
  my_relative_file_path = "upload_sample.MOV"
  
  print(speech_to_text(my_relative_file_path))

  Console output (traceback and my print()’s)

  Looking at path:  C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV 
  
  File is not .wav:  .MOV found. Converting... 
  
  New audio file will be:  upload_sample.wav Attempting to run this command:
  
   ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav 
  
  ffmpeg version git-2017-12-18-74f408c Copyright (c) 2000-2017 the FFmpeg developers   built with gcc 7.2.0 (GCC)   
  
  ----REMOVED SOME FFMPEG OUTPUT FOR BREVITY----
  
  File 'upload_sample.wav' already exists. Overwrite ? [y/N] y 
  
  Stream mapping:   Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'upload_sample.wav':   Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      com.apple.quicktime.creationdate: 2017-12-19T16:06:10-0500
  
      com.apple.quicktime.make: Apple
  
      com.apple.quicktime.model: iPhone 6
  
      com.apple.quicktime.software: 10.3.3
  
      ISFT            : Lavf58.3.100
  
      Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-12-19T21:06:11.000000Z
  
        handler_name    : Core Media Data Handler
  
        encoder         : Lavc58.8.100 pcm_s16le size=    1036kB time=00:00:06.01 bitrate=1411.3kbits/s speed=N/A     video:0kB audio:1036kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.007352% 
  
  0 
  
  Traceback (most recent call last): Past Subprocess.call   
  
  File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 53, in <module> 
  
  AUDIO_FILE now set to:  C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav
  
      print(speech_to_text(my_relative_file_path))   
  
  File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 36, in speech_to_text
  
      with sr.AudioFile(AUDIO_FILE) as source:   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\site-packages\speech_recognition\__init__.py", line 203, in __enter__
  
      self.audio_reader = wave.open(self.filename_or_fileobject, "rb")   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 499, in open
  
      return Wave_read(f)   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 159, in __init__
  
      f = builtins.open(f, 'rb') 
  
  FileNotFoundError: [Errno 2] No such file or directory: 'C:\\Users\\Nathan\\Desktop\\MeetingRecorderWebAPP\\media\\recordings\\upload_sample.wav'
  
  
  
  Process finished with exit code 1
  
  </module>

• FFmpeg - selecting appropriate bitrate for VP9 encoding

  11 janvier 2018, par fastily

  I am looking to encode a 4k video shot with iPhone 6s in VP9 in the best quality possible.

  For reference, stream data of the video I would like to encode, via ffprobe :

  Duration: 00:00:10.48, start: 0.000000, bitrate: 46047 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 3840x2160, 45959 kb/s, 29.98 fps, 29.97 tbr, 600 tbn, 1200 tbc (default)
  
      Metadata:
  
        creation_time   : 2017-03-13T21:12:56.000000Z
  
        handler_name    : Core Media Data Handler
  
        encoder         : H.264
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 79 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-03-13T21:12:56.000000Z
  
        handler_name    : Core Media Data Handler

  I am using the following FFmpeg commands, based on these instructions (see Best Quality (Slowest) Recommended Settings section).

  1. ffmpeg -i INPUT.mov -c:v libvpx-vp9 -pass 1 -b:v 46000K -threads 4 -speed 4 -g 9999 -an -f webm -y /dev/null
  2. ffmpeg -I INPUT.mov -c:v libvpx-vp9 -pass 2 -b:v 46000K -threads 4 -speed 0 -g 9999 -an -f webm OUTPUT.webm

  Is there a best practice to select an optimal -b:v value such that the resulting video is visually indistinguishable from the original ? I have tried values ranging from 36000K-46000K, but these result in massive files with an overall bitrate exceeding the target bitrate.

  Thanks in advance !

• Using ffmpeg to build a streaming server to stream static media files (broadcast behaviour)

  15 février 2018, par MiDaa

  I’ve read some online articles and SO questions, most of them are about streaming MY video to SERVER like youtube or switch.

  This is about a project of interest, here are what it should do.

  • Work on a Linux server
  • Serve media(preferably multiple format like mp4 mkv) files to client through rtp protocol maybe ?
  • Server could set a specific time to start the streaming or end it
  • Server could pause and resume the streaming(?)
  • Multiple clients connect and play the stream at same time(sounds like a basic feature)

  After some research, I found that ffmpeg is a great open-source candidate for such a project but as a newbie in this area, I’m having a tough time understanding how this whole thing work.

  As this(ffmpeg doc) states, it looks like just a one liner command. But I don’t find anything fit my feature listed above.

  Can ffmpeg be used to achieve those ? If not appriciate any suggesstion on where I should be looking at.

  EDIT :

  • Target devices : iPad,iPhone, Android phones should be able to watch the stream using a web browser(assume a modern browser)