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• Play h.264 video stream in html page

  12 juin 2017, par e2l3n

  I am using ffmpeg and vlc on linux to produce MPEG transport stream (mpegts) over HTTP. Basically ffmpeg captures the screen and produces h.264 stream using mpegts and then vlc is used as a server that delivers the stream over http. That’s how I manage to accomplish this.

  ffmpeg -f x11grab -s 1280x800 -r 30 -i :0.0+0,0  -f alsa -ac 2 -i pulse -vcodec libx264 -preset ultrafast -s 1280x800 -threads 0 -f mpegts - | vlc -I dummy - --sout '#std{access=http,mux=ts,dst=:3030}

  And some useful log about the resulting stream :

  Output #0, mpegts, to 'pipe:':
  
    Metadata:
  
      encoder         : Lavf57.72.101
  
      Stream #0:0: Video: h264 (libx264), yuv444p(progressive), 1280x800, q=-1--1, 30 fps, 90k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc57.96.101 libx264
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
  
      Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s
  
      Metadata:
  
        encoder         : Lavc57.96.101 mp2

  All looks good except for that I am unable to play the stream in a HTML page. Could someone point me at some solution to either render the stream in HTML page or change the stream type so that it gets rendered.
  Thank you in advance !

  EDIT :
  I’ve noticed something weird. The <video></video> fails to play mp4 files on Firefox (51.0.1) for Ubuntu (I have also tested on Windows 8 with Chrome).
  I simply use ffmpeg to record screen for 10 seconds and produce .mp4 video (see below).

  ffmpeg -video_size 1280x800 -framerate 25 -f x11grab -i :0.0+0,0 -t 00:00:10 ~/Workspace/videostream/output.mp4

  And then play the output.mp4 video with the following snippet.

  <video width="480" height="320" controls="controls">
  
      <source src="output.mp4" type="video/mp4">
  
  </source></video>

  This simply does not play although I can open the video with any other media player. The media info (using MediaInfo, v0.7.96) of the output.mp4 file is :

  GENERAL: MPEG-4 (Base Media): 1.007 MiB, 10s 0ms; 1 Video stream: AVC; Overall bit rate: 825 Kbps; Writing application: Lavf57.72.101
  
  VIDEO: 2 343 Kbps, 1280*800(16:10), at 25.000 fps, MPEG Video (Version 2) (Main@High 1440)

  However, if I replace the output.mp4 with any other .mp4 file, it does play in Firefox. So, this leads me to conclude that it is not a browser issue but something with the way I exploit ffmpeg. Below is the media info of a proper .mp4 file that successfully runs in the browser.

  GENERAL: MPEG-4 (Base Media/Version 2): 48.4 MiB, 2mn 50s; 1 Video stream: AVC; 1 Audio stream: AAC; Overall bit mode rate: Variable; Overall bit rate: 2 385 Kbps;
  
  VIDEO: 2 256 Kbps, 1280*720(16:9), at 25.000 fps, AVC (Main@L3.1) (CABAC / 3 Ref Frames); ISO media produced by Google Inc.
  
  AUDIO:
  
  126 Kbps, 44.1 KHz, 2 channels, AAC (LC), ISO media produced by Google Inc.

  And here is the media info of the output file that results from redirecting the h.264 mpegts stream using the ffmpeg command with -pix_fmt yuv420p ./ffmpeg -t 00:00:10 -f x11grab -s 1280x800 -r 30 -i :0.0+0,0 -pix_fmt yuv420p -vcodec libx264 -preset ultrafast -s 1280x800 -threads 0 -f mpegts - > ~/Workspace/file.ts :

  GENERAL: MPEG-TS: 2.93 MiB, 9s 960ms; 1 Video stream: MPEG Video; 1 Menu stream: MPEG Video; Overall bit rate mode: Variable; Overall bit rate: 2 465 Kbps; 
  
  VIDEO: 2 343 Kbps, 1280*800(16:10), at 25.000 fps, MPEG Video (Version 2) (Main@High 1440)

• ffmpeg : Setting GOP structure, Picture structure when converting to MPEG-2 [on hold]

  27 juin 2017, par Jude

  I have a yuv file named src9_ref__625.yuv from the following website :
  ftp://vqeg.its.bldrdoc.gov/SDTV/VQEG_PhaseI/TestSequences/Reference/

  It contains 720 pixels (1440 bytes) per horizontal line and has 576 active lines per frame. Each line is in multiplexed 4:2:2 component video format as follows : Cb Y Cr Y ... 720 Y bytes per line, 360 Cb bytes per line, 360 Cr bytes per line.

  How to convert that file to MPEG-2 with following coding parameters :

  • GOP Structure : Automatic
  • Picture Structure : Alw Frame
  • Profile/Level : MP@ML

  ...as suggested in this article :
  http://iie.fing.edu.uy/publicaciones/2009/JL09/JL09.pdf

• 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.