Web Audio

All posts tagged Web Audio

webrtcH4cKS: ~ Finding the Warts in WebAssembly+WebRTC

Posted by Philipp Hancke on April 11, 2019
Posted in: Reverse-Engineering, Technology.
Tagged: , , , .
4 comments

A while ago we looked at how Zoom was avoiding WebRTC by using WebAssembly to ship their own audio and video codecs instead of using the ones built into the browser’s WebRTC.  I found an interesting branch in Google’s main (and sadly mostly abandoned) WebRTC sample application apprtc this past January. The branch is named wartc… a name which is going to stick as warts!

The repo contains a number of experiments related to compiling the webrtc.org library as WebAssembly and evaluating the performance. From the rapid timeline, this looks to have been a hackathon project. ...  Continue reading

webrtcH4cKS: ~ Autoplay restrictions and WebRTC (Dag-Inge Aas)

Posted by Dag-Inge Aas on May 7, 2018
Posted in: Technology.
Tagged: , , , .
7 comments
Hear No Evil picture

One of the great things about WebRTC is that it is built right into the web platform. The web platform is generally great for WebRTC, but occasionally it can cause huge headaches when specific WebRTC needs do not exactly align with more general browser usage requirements. The latest example of this is has to do with the autoplay of media where sound(s) suddenly went missing for many users. Former webrtcHacks guest author Dag-Inge Aas has been dealing with this first hand. See below for his write-up on browser expectations around the playback of media, the recent Chrome 66+ changes, and some tips and tricks for working around these issues. ...  Continue reading

webrtcH4cKS: ~ Your Browser as a Audio Conference Server with WebRTC & Web Audio (Alexey Aylarov)

Posted by Chad Hart on May 20, 2016
Posted in: Guide.
Tagged: , , , , .
4 comments

Conference calling is a multi-billion dollar industry that is mostly powered by expensive, high-powered conferencing servers. Now you can replicate much of this functionality for free with a modern browser using the combination of WebRTC and WebAudio.

Like with video, multi-party audio can utilize a few architectures:

  1. Full mesh – each client sends their audio to every other client; the individual streams are then combined locally before they come out of your speaker
  2. Mixed with a conferencing server acting as a Multipoint Control Unit (MCU) – the MCU combines each stream and sends a single set to each client
  3. Routed with a conferencing server in a Selective Forwarding Unit (SFU) mode – each client sends a single stream to the server where it is replicated and sent to the others

This architecture represents a fourth type: client-mixed type where one of the clients acts like the server. This provides the server-less benefits of mesh conferencing without the excessive bandwidth usage and stream management challenges. ...  Continue reading

webrtcH4cKS: ~ Shut up! Monitoring audio volume in getUserMedia

Posted by Philipp Hancke on December 10, 2015
Posted in: Guide.
Tagged: , , , , , , , .
1 comment

A few days back my old friend Chris Koehncke, better known as “Kranky” asked me how hard it would be to implement a wild idea he had to monitor what percentage of the time you spent talking instead of listening on a call when using WebRTC. When I said “one day” that made him wonder whether he could offshore it to save money. Well… good luck!

A week later Kranky showed me some code. Wait, he is writing code? It was not bad – it was using the WebAudio API so going in the right direction. It was enough to prod me to finish writing the app for him. ...  Continue reading