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• 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 !

• 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>

• -12909 error decoding h264 stream with intra-refresh

  2 juillet 2024, par ciclopez

  I'm making an iOS app that decodes an h264 stream using video-toolbox. I create the stream with ffmpeg on a PC and send it to an iPhone using RTP. It's working nicely when I use this command to create it :

  



  ffmpeg -y -f:v rawvideo -c:v rawvideo -s 1280x720 -pix_fmt bgra -r 30 -an -i - -pix_fmt yuv420p -c:v libx264 -tune zerolatency -preset fast -b:v 5M -refs 1 -g 30 -profile:v high -level 4.1 -f rtp rtp://192.168.1.100:5678


  



  The iPhone receives and displays all the frames. However, when I enable intra-refresh

  



  -intra-refresh 1


  



  decoding fails with error code -12909 (-8969 on simulator) when VTDecompressionSessionDecodeFrame() is called.

  



  I take care of processing UDP packets to extract NAL Units, so I triple checked this process and discarded a problem with this part of the code.

  



  I didn't find any info about Video-toolbox not supporting intra-refresh, so the question is, does Video-toolbox support intra-refresh ? and if it does, am I missing something in the ffmpeg side that makes the stream not supported by Video-toolbox ?
Do I have to add something to the CMVideoFormatDescriptionRef apart from creating it with SPS and PPS data using CMVideoFormatDescriptionCreateFromH264ParameterSets() ?