Capture microphone input and stream it to a peer with processing applied to the audio.
The audio stream is:
- Recorded using live web audio input in
- Filtered using an HP filter with fc=1500 Hz.
- Encoded using Opus.
- Transmitted (in loopback) to a remote peer using RTCPeerConnection where it is decoded.
- Finally, the received remote stream is used as source to an <audio> element and played out locally.
Press any key to add an effect to the transmitted audio while talking.
Please note that:
- Sample rate and channel configuration must be the same for input and
output sides on Windows.
- Only the default microphone device can be used for capturing.
For more information, see WebRTC
integration with the Web Audio API.
View source on GitHub