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WebRTC samples Web Audio input

src/content/peerconnection/webaudio-input/index.html

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WebRTC samplesWeb Audio input

Add local audio to output

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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