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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 :

    &#xA;&#xA;

    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&#xA;

    &#xA;&#xA;

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

    &#xA;&#xA;

    -intra-refresh 1&#xA;

    &#xA;&#xA;

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

    &#xA;&#xA;

    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.

    &#xA;&#xA;

    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 ?&#xA;Do I have to add something to the CMVideoFormatDescriptionRef apart from creating it with SPS and PPS data using CMVideoFormatDescriptionCreateFromH264ParameterSets() ?

    &#xA;