0

I am developing a colaborative instrument playing game, where multiple users will play an instrument (a synthesizer or sample, using the WebAudio API). On my first prototype I've set up a keyboard that sends note/volume signals via Socket.io to the server, and when the server gets that signal it sends it back to all connected sockets, which will play the corresponding note.

You might have guessed it right: there's a massive amount of lag and inconsistency as to the order of arrival of notes.

What are some efficient ways that I can send the output of WebAudio to the server, and have it broadcast to all connected users, so I have some sort of consistency?

Fernando Gomes
  • 41
  • 3

1 Answers1

0

You could try using a MediaStream by adding an MediaStreamAudioDestinationNode to your audio node graph as a destination and use that stream with either WebRTC or RecordRTC to send to your server.

Here is some info I found you could look at. It does talk about using the getUserMedia method, but both getUserMedia and MediaStreamAudioDestinationNode methods send out a MediaStream constructor. This info has some ideas on how you could send a MediaStream to your sever. However it does say that it needs to be recorded first. Not when it's live and running.

Sending a MediaStream to host Server with WebRTC after it is captured by getUserMedia

I hope this helps :)

Hazmatron
  • 181
  • 2
  • 16