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• WebRTC books – a brief review

  1er janvier 2014, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.

• WebRTC books – a brief review

  30 décembre 2013, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.

• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing