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• Read, process and save video and audio with FFMPEG

  3 mai 2017, par sysseon

  I want to open a video resource with ffmpeg on Python, get the read frames from the pipe, modify them (e.g. put the timestamp with OpenCV) and write the result to an output video file. I also want to save the audio source with no changes.

  My code (with no audio and two processes) :

  import subprocess as sp
  
  import numpy as np
  
  # import time
  
  # import cv2
  
  
  
  FFMPEG_BIN = "C:/ffmpeg/bin/ffmpeg.exe"
  
  INPUT_VID = 'input.avi'
  
  res = [320, 240]
  
  
  
  command_in = [FFMPEG_BIN,
  
                '-y',  # (optional) overwrite output file if it exists
  
                '-i', INPUT_VID,
  
                '-f', 'image2pipe',  # image2pipe or rawvideo?
  
                '-pix_fmt', 'bgr24',
  
                '-vcodec', 'rawvideo',
  
                '-']
  
  
  
  command_out = [FFMPEG_BIN,
  
                 '-y',  # (optional) overwrite output file if it exists
  
                 '-f', 'rawvideo',
  
                 '-vcodec', 'rawvideo',
  
                 '-s', '320x240',
  
                 '-pix_fmt', 'bgr24',
  
                 '-r', '25',
  
                 '-i', '-',
  
                 # '-i', INPUT_VID,    # Audio
  
                 '-vcodec', 'mpeg4',
  
                 'output.mp4']
  
  
  
  pipe_in = sp.Popen(command_in, stdout=sp.PIPE, stderr=sp.PIPE)
  
  pipe_out = sp.Popen(command_out, stdin=sp.PIPE, stderr=sp.PIPE)
  
  
  
  while True:
  
      # Read 320*240*3 bytes (= 1 frame)
  
      raw_image = pipe_in.stdout.read(res[0] * res[1] * 3)
  
      # Transform the byte read into a numpy array
  
      image = np.fromstring(raw_image, dtype=np.uint8)
  
      image = image.reshape((res[1], res[0], 3))
  
      # Draw some text in the image
  
      # draw_text(image)
  
  
  
      # Show the image with OpenCV (not working, gray image, why?)
  
      # cv2.imshow("VIDEO", image)
  
  
  
      # Write image to output process
  
      pipe_out.stdin.write(image.tostring())
  
  
  
  print 'done'
  
  pipe_in.kill()
  
  pipe_out.kill()

  1. Could it be done with just a process ? (Read the input from a file,
     put it in the input pipe, get the image, process it, and put it in
     the output pipe to be saved into a video file)
  2. How can I save the audio ? In this example, I could use ’-i
     INPUT_VID’ in the second process to get the audio channel, but my
     source will be a RTSP, and I don’t want to create a connection for
     each process. Could I put video+audio in the pipe and rescue and
     separate it with numpy ? How ?
  3. I use a loop to process the frames and wait until I get an error.
     How can I check if all frames are already read ?
  4. Not important, but if I try to show the images with OpenCV
     (cv2.imshow(...)), I only see a gray screen. Why ?

• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

  The post AppRTC : Google’s WebRTC test app and its parameters first appeared on ginger’s thoughts.

• ffmpeg python subprocess error only on ubuntu

  19 avril 2017, par Wonger

  Im working on an application that is splitting videos from youtube into images. I work on a macbook pro for development, but our app servers run on an ubuntu 12.04 server. The current code on our servers running right now is the following

  ffmpeg -i {video_file} -vf fps={fps}

  which we run via a subprocess Popen function call in python. This has very slow performance because it is essentially playing back the whole video file to get the frames. I found another SO post that said you could use -accurate_seek -ss to grab single frames at a specific time, but I am facing some issues with that. When i run the command via command line when SSH into our dev server, it works fine, but when i run it via a subprocess Popen call in python, i get the following error :

  (standard_in) 1: syntax error
  
  ffmpeg version 3.2.4-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.1 (Debian 5.4.1-5) 20170205
  
    configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-5 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray --enable-libass --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  Option accurate_seek (enable/disable accurate seeking with -ss) cannot be applied to output url test.mkv -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to.
  
  Error parsing options for output file test.mkv.
  
  Error opening output files: Invalid argument

  python code :

  import subprocess
  
  command = "for i in {0..3} ; do ffmpeg -accurate_seek -ss `echo $i*60.0 | bc` -i test.mkv -frames:v 1 images/test_img_$i.jpg ; done"
  
  process = subprocess.Popen(command, stdout=subprocess.PIPE, stderr = subprocess.STDOUT, shell=True)
  
  output = process.communicate()
  
  print output[0].replace('\\n' , '\n')

  The thing is, when i run this on my OSX terminal, it works fine, but there is some issue with doing it in ubuntu. Does anyone have experience with this issue ?

  Output when i run the same exact code in osx :

  ffmpeg version 3.0.2 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with Apple LLVM version 7.3.0 (clang-703.0.31)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.0.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libvpx --enable-vda
  
    libavutil      55. 17.103 / 55. 17.103
  
    libavcodec     57. 24.102 / 57. 24.102
  
    libavformat    57. 25.100 / 57. 25.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 31.100 /  6. 31.100
  
    libavresample   3.  0.  0 /  3.  0.  0
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
    Input #0, matroska,webm, from 'test.mkv':
  
    Metadata:
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      MINOR_VERSION   : 0
  
      ENCODER         : Lavf57.25.100
  
    Duration: 00:02:10.20, start: 0.007000, bitrate: 2729 kb/s
  
      Stream #0:0: Video: h264 (High), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:02:10.172000000
  
      Stream #0:1(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        DURATION        : 00:02:10.201000000
  
        [swscaler @ 0x7f9b4b008000] deprecated pixel format used, make sure you did set range correctly
  
  Output #0, image2, to 'images/test_img_0.jpg':
  
    Metadata:
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      MINOR_VERSION   : 0
  
      encoder         : Lavf57.25.100
  
      Stream #0:0: Video: mjpeg, yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 23.98 fps, 23.98 tbn, 23.98 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:02:10.172000000
  
        encoder         : Lavc57.24.102 mjpeg
  
      Side data:
  
        unknown side data type 10 (24 bytes)
  
        Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))
  
    Press [q] to stop, [?] for help
  
  frame=    1 fps=0.0 q=7.2 Lsize=N/A time=00:00:00.04 bitrate=N/A speed=   1x    
  
  video:73kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
  
  ....

  One thing i forgot to mention is that on my mac i used homebrew to install ffmpeg, where as on Ubuntu i used a static build

  other stack overflow reference : Fastest way to extract frames using ffmpeg ?