The audio stream is:
o Recorded using live-audio
input.
o Filtered using an HP filter with fc=1500 Hz.
o Encoded using
Opus.
o Transmitted (in loopback) to remote peer using
RTCPeerConnection where it is decoded.
o Finally, the received remote stream is used as source to an <audio>
tag and played out locally.
Press any key to add an effect to the transmitted audio while talking.
Please note that:
o Linux is currently not supported.
o Sample rate and channel configuration must be the same for input and
output sides on Windows.
o Only the Default microphone device can be used for capturing.
For more information, see WebRTC integration with the Web Audio API.