# Audio effects with WebAudio in WebRTC

## Capture microphone input and stream it out to a peer with a processing effect applied to the audio.

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.

Add local audio to output: