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• Video Conferencing in HTML5 : WebRTC via Web Sockets

  1er janvier 2014, par silvia

  A bit over a week ago I gave a presentation at Web Directions Code 2012 in Melbourne. Maxine and John asked me to speak about something related to HTML5 video, so I went for the new shiny : WebRTC – real-time communication in the browser.

  Presentation slides

  I only had 20 min, so I had to make it tight. I wanted to show off video conferencing without special plugins in Google Chrome in just a few lines of code, as is the promise of WebRTC. To a large extent, I achieved this. But I made some interesting discoveries along the way. Demos are in the slide deck.

  UPDATE : Opera 12 has been released with WebRTC support.

  Housekeeping : if you want to replicate what I have done, you need to install a Google Chrome Web Browser 19+. Then make sure you go to chrome ://flags and activate the MediaStream and PeerConnection experiment(s). Restart your browser and now you can experiment with this feature. Big warning up-front : it’s not production-ready, since there are still changes happening to the spec and there is no compatible implementation by another browser yet.

  Here is a brief summary of the steps involved to set up video conferencing in your browser :

  1. Set up a video element each for the local and the remote video stream.
  2. Grab the local camera and stream it to the first video element.
  3. (*) Establish a connection to another person running the same Web page.
  4. Send the local camera stream on that peer connection.
  5. Accept the remote camera stream into the second video element.

  Now, the most difficult part of all of this – believe it or not – is the signalling part that is required to build the peer connection (marked with (*)). Initially I wanted to run completely without a server and just enter the remote’s IP address to establish the connection. This is, however, not a functionality that the PeerConnection object provides [might this be something to add to the spec ?].

  So, you need a server known to both parties that can provide for the handshake to set up the connection. All the examples that I have seen, such as https://apprtc.appspot.com/, use a channel management server on Google’s appengine. I wanted it all working with HTML5 technology, so I decided to use a Web Socket server instead.

  I implemented my Web Socket server using node.js (code of websocket server). The video conferencing demo is in the slide deck in an iframe – you can also use the stand-alone html page. Works like a treat.

  While it is still using Google’s STUN server to get through NAT, the messaging for setting up the connection is running completely through the Web Socket server. The messages that get exchanged are plain SDP message packets with a session ID. There are OFFER, ANSWER, and OK packets exchanged for each streaming direction. You can see some of it in the below image :

  

  I’m not running a public WebSocket server, so you won’t be able to see this part of the presentation working. But the local loopback video should work.

  At the conference, it all went without a hitch (while the wireless played along). I believe you have to host the WebSocket server on the same machine as the Web page, otherwise it won’t work for security reasons.

  A whole new world of opportunities lies out there when we get the ability to set up video conferencing on every Web page – scary and exciting at the same time !

• Video Conferencing in HTML5 : WebRTC via Web Sockets

  14 juin 2012, par silvia

  A bit over a week ago I gave a presentation at Web Directions Code 2012 in Melbourne. Maxine and John asked me to speak about something related to HTML5 video, so I went for the new shiny : WebRTC – real-time communication in the browser.

  Presentation slides

  I only had 20 min, so I had to make it tight. I wanted to show off video conferencing without special plugins in Google Chrome in just a few lines of code, as is the promise of WebRTC. To a large extent, I achieved this. But I made some interesting discoveries along the way. Demos are in the slide deck.

  UPDATE : Opera 12 has been released with WebRTC support.

  Housekeeping : if you want to replicate what I have done, you need to install a Google Chrome Web Browser 19+. Then make sure you go to chrome ://flags and activate the MediaStream and PeerConnection experiment(s). Restart your browser and now you can experiment with this feature. Big warning up-front : it’s not production-ready, since there are still changes happening to the spec and there is no compatible implementation by another browser yet.

  Here is a brief summary of the steps involved to set up video conferencing in your browser :

  1. Set up a video element each for the local and the remote video stream.
  2. Grab the local camera and stream it to the first video element.
  3. (*) Establish a connection to another person running the same Web page.
  4. Send the local camera stream on that peer connection.
  5. Accept the remote camera stream into the second video element.

  Now, the most difficult part of all of this – believe it or not – is the signalling part that is required to build the peer connection (marked with (*)). Initially I wanted to run completely without a server and just enter the remote’s IP address to establish the connection. This is, however, not a functionality that the PeerConnection object provides [might this be something to add to the spec ?].

  So, you need a server known to both parties that can provide for the handshake to set up the connection. All the examples that I have seen, such as https://apprtc.appspot.com/, use a channel management server on Google’s appengine. I wanted it all working with HTML5 technology, so I decided to use a Web Socket server instead.

  I implemented my Web Socket server using node.js (code of websocket server). The video conferencing demo is in the slide deck in an iframe – you can also use the stand-alone html page. Works like a treat.

  While it is still using Google’s STUN server to get through NAT, the messaging for setting up the connection is running completely through the Web Socket server. The messages that get exchanged are plain SDP message packets with a session ID. There are OFFER, ANSWER, and OK packets exchanged for each streaming direction. You can see some of it in the below image :

  

  I’m not running a public WebSocket server, so you won’t be able to see this part of the presentation working. But the local loopback video should work.

  At the conference, it all went without a hitch (while the wireless played along). I believe you have to host the WebSocket server on the same machine as the Web page, otherwise it won’t work for security reasons.

  A whole new world of opportunities lies out there when we get the ability to set up video conferencing on every Web page – scary and exciting at the same time !

• FFMpeg How to extract individual audio channels from wav/.w64 and insert in .mxf with track tags

  11 septembre 2018, par Vince

  Hi my problem is I have 2 .w64 files (extended wav format) each file has 16 mono channels of audio. I want to be able to extract specific channels of audio from each of those .w64 files and insert those channels into an .mxf file as separate single channel mono audio streams and to be additionally able to set the Tag information on those audio streams. I have tried using -map and so on but it seems to take all the channels from the .w64 files and insert a single audio stream of 16 channels. I apologise in advance as I’m very new to ffmpeg and thanks in advance for any advice any of you can offer. This is all specific to command line usage on windows.
  All the best

  ffmpeg -i "D :\Media\AUDIO_0.W64" -i "D :\media\NO_AUDIO.mxf" -c copy -map 0:0:0 -map 0:0:1 -map 0:0:2 -acodec pcm_s24le -map 1:0 "D :\media\out.mxf"

  ffmpeg version N-67838-g4388e78 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Nov 19 2014 22:02:08 with gcc 4.9.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib
  
  modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw
  
  b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge
  
  r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en
  
  able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
  
   --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-
  
  libx265 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      54. 14.100 / 54. 14.100
  
    libavcodec     56. 12.101 / 56. 12.101
  
    libavformat    56. 14.100 / 56. 14.100
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [w64 @ 0481c580] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, w64, from 'D:\Media\AUDIO_0.W64':
  
    Duration: 00:01:02.56, bitrate: 18432 kb/s
  
      Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 16 channels
  
  , s32 (24 bit), 18432 kb/s
  
  [mxf @ 048890e0] index entry 1564 + TemporalOffset 1 = 1565, which is out of bou
  
  nds
  
  Input #1, mxf, from 'D:\media\NO_AUDIO.mxf':
  
    Metadata:
  
      uid             : adab4424-2f25-4dc7-92ff-29bd000c0000
  
      generation_uid  : adab4424-2f25-4dc7-92ff-29bd000c0001
  
      company_name    : FFmpeg
  
      product_name    : OP1a Muxer
  
      product_version : 57.56.100
  
      product_uid     : adab4424-2f25-4dc7-92ff-29bd000c0002
  
      modification_date: 0000-01-01 00:00:00
  
      timecode        : 00:00:00:00
  
    Duration: 00:01:02.60, start: 0.000000, bitrate: 2404 kb/s
  
      Stream #1:0: Video: mpeg2video (4:2:2), yuv422p(tv), 1920x1080 [SAR 1:1 DAR16:9], max. 50000 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
  
  File 'D:\media\out.mxf' already exists. Overwrite ? [y/N] y
  
  
  
  [mxf @ 04891c60] there must be exactly one video stream and it must be the first
  
   one
  
  Output #0, mxf, to 'D:\media\out.mxf':
  
    Metadata:
  
      encoder         : Lavf56.14.100
  
      Stream #0:0: Audio: pcm_s24le, 48000 Hz, 16 channels, s32 (24 bit), 18432 kb/s
  
      Metadata:
  
        encoder         : Lavc56.12.101 pcm_s24le
  
      Stream #0:1: Audio: pcm_s24le, 48000 Hz, 16 channels, s32 (24 bit), 18432 kb/s
  
      Metadata:
  
        encoder         : Lavc56.12.101 pcm_s24le
  
      Stream #0:2: Audio: pcm_s24le, 48000 Hz, 16 channels, s32 (24 bit), 18432 kb/s
  
      Metadata:
  
        encoder         : Lavc56.12.101 pcm_s24le
  
      Stream #0:3: Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, max. 50000 kb/s, 25 fps, 25 tbn, 25 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s24le (native))
  
    Stream #0:0 -> #0:1 (pcm_s24le (native) -> pcm_s24le (native))
  
    Stream #0:0 -> #0:2 (pcm_s24le (native) -> pcm_s24le (native))
  
    Stream #1:0 -> #0:3 (copy)
  
  
  
  Could not write header for output file #0 (incorrect codec parameters ?): Error
  
  number -1 occurred