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• 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

• Powerful Video Analytics and Audio Analytics for Piwik

  10 novembre 2016, par InnoCraft — Plugins, Press Releases

  Over the years, one of the most frequently requested feature by users was to be able to measure how videos and audios are watched and engaged with on your website. We are finally able to announce that it is here ! We are very excited to launch Media Analytics, which will help you understand and grow your audience.

  This article is a showcase of the new powerful video and audio analytics product built for Piwik.

  Why media analytics ?

  We all love media content such as videos as they can make our experiences on websites and apps so much more interesting. A growing number of websites now utilize media files in one form or another : a video presentation of a product or service, a video tutorial teaching you how to do something or interviews with key speakers. Also many creators and distributors are publishing audio files such as podcasts or music songs, and even broadcasting live video events such as music concerts or an entire conference online.

  Whenever you publish videos or audio media on your websites or applications, Media Analytics provides you with clear insights on how your audience interacts with your content. It helps you see what content works and why – so you can better understand and further grow your business !

  

  Valuable insights in Real time

  

  See where your audience comes from.

  How will Media Analytics help me grow ?

  • Better understand your audience : who are the users playing videos and for how long, how often, and where have they dropped off.
  • Gain quick insights into how interaction with your media changes over time with easy to use graphs and report overviews.
  • Get closer to your users by seeing every action of your visitors before and after they utilized your media.
  • View valuable insights in Real time : ‘most popular content right now’, your real time audience map, and more.
  • See where your audience comes from. Drill down right from continents to specifics such as cities.
  • Share and export media analytics reports with your colleagues by creating custom email reports.
  • Video and audio players are supported either automatically (for Youtube, Vimeo, HTML5…) or via a simple custom player integration.
  • No data limit and 100% privacy and data ownership.

  Best of all, it is easy to use and understand, and integrates perfectly with Piwik. Media Analytics complements other reports to give you a 360 degree view of how your users engage with your content.

  Learn more on the official website : www.media-analytics.net

  How do I get Media Analytics ?

  All premium plugins come with our 14 day money back guarantee and 1-click installation & updates. Customers get all product updates for free.

  Learn more about Media Analytics Get it now on the Marketplace

  Media Analytics is available for purchase and download on the Marketplace.

  If you are not using Piwik yet, you can also signup for a free trial of Piwik Cloud (including Media Analytics !).

  Have a question about this product ? Get in touch.

• Stream audio from STDIN through VLC for SDR

  13 décembre 2018, par doghousedean

  I’m trying to get an audio stream from a Raspberry pi 2 to my windows desktop.

  The RTL SDR on my pi is great when using spyserver but there are some certain applications that cannot connect to it, echoes the meteor detector for example

  Streaming the audio from the pi to my desktop could work with the stereo mixer input device

  My idea is to use VLC but cannot figure it out

  $ rtl_fm -f 143.050 -M wbfm -s 180k -E deemp | cvlc - -v --sout '#standard{access=http,mux=ogg,dst=localhost:8080}'
  
  VLC media player 3.0.3 Vetinari (revision 3.0.3-1-0-gc2bb759264)
  
  Found 1 device(s):
  
    0:  Realtek, RTL2838UHIDIR, SN: 00000001
  
  
  
  Using device 0: Generic RTL2832U OEM
  
  [01d919d8] dbus interface error: Failed to connect to the D-Bus session daemon: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11
  
  [01d919d8] main interface error: no suitable interface module
  
  [01d14938] main libvlc error: interface "dbus,none" initialization failed
  
  [01d92be8] main interface error: no suitable interface module
  
  [01d14938] main libvlc error: interface "globalhotkeys,none" initialization failed
  
  [01d92c50] dummy interface: using the dummy interface module...
  
  [73a01670] http access out warning: "localhost" HTTP host might be ignored in multiple-host configurations, use at your own risks.
  
  [73a01670] http access out: Consider passing --http-host=IP on the command line instead.
  
  [73a02dd0] mux_ogg mux: Open
  
  Found Rafael Micro R820T tuner
  
  Tuner gain set to automatic.
  
  [R82XX] PLL not locked!
  
  Tuned to 286143 Hz.
  
  Oversampling input by: 6x.
  
  Oversampling output by: 1x.
  
  Buffer size: 7.59ms
  
  Exact sample rate is: 1080000.025749 Hz
  
  [R82XX] PLL not locked!
  
  Sampling at 1080000 S/s.
  
  Output at 180000 Hz.
  
  [73a070b8] ps demux warning: this does not look like an MPEG PS stream, continuing anyway
  
  [73a070b8] ps demux warning: garbage at input from 509, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 89140, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 195382, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 279916, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 326867, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 434452, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 438851, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 454391, trying to resync...
  
  [73a070b8] ps demux warning: found sync code
  
  [73a070b8] ps demux warning: garbage at input from 547348, trying to resync...

  I figured I could send the rtl_fm output to ffmpeg and then to vlc but I just can’t figure out the options I need.

  $ rtl_fm -f 143.050 -M wbfm -s 180k -E deemp | ffmpeg -acodec pcm_u8 -f oga - | vlc - -I dummy --sout '#standard{access=http,mux=ogg,dst=localhost:8080}'

  This gives similar output to above with the differences below

  Output #0, oga, to 'pipe:':
  
  Output file #0 does not contain any stream
  
  Signal caught, exiting!
  
  
  
  User cancel, exiting...
  
  [739070b8] mjpeg demux error: cannot peek
  
  [73c00520] main input error: Your input can't be opened
  
  [73c00520] main input error: VLC is unable to open the MRL 'fd://0'. Check the log for details.
  
  [73902dd0] mux_ogg mux: Close