WebRTC samplesWeb Audio input

Add local audio to output

StartStop

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 chrome://flags.
• 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