MediaStreamTrackProcessor/MediaStreamTrackGenerator Latency Demo

This is a demo of using MediaStreamTrackProcessor (MSTP)/MediaStreamTrackGenerator (MSTG) to process an audio MediaStream that is to be sent between two PeerConnections to be rendered at the other end.

• On the input side WebAudio is used to generate a constant audio source which is connected to an AudioWorklet (AW1) which periodically adds a short sine-wave as a signature (this signature is later detected on the render-side so that the latency can be measured).
• The input signal produced by WebAudio is put into a MediaStreamTrack via a MediaStreamDestination. This track is processed using either 1) a MediaStreamTrackProcessor/Transformer/MediaStreamTrackGenerator or 2) a MediaStreamTrackProcessor/Transformer/ShimMediaStreamTrackGenerator pipeline. Processing is passthrough, except that 3, 6 and 9 seconds after the pipeline starts, delays of 100ms, 200ms, and 300ms are added respectively.
• The output signal from the pipeline is sent through a pair of local RTCPeerConnections (PC1 and PC2).
• The track from the receiving RTCPeerConnection (PC2) is connected to WebAudio to render it. The WebAudio AudioContext also has an AudioWorklet (AW2) to detect the signature added by AW1 and measure the end-to-end latency.

Processing occurs on the main thread. If the browser does not support MediaStreamTrackProcessor for audio on the main thread, Jan-Ivar's MediaStreamTrackProcesor polyfill is used instead.

ShimMediaStreamTrackGenerator is Jan-Ivar's MediaStreamTrackGenerator polyfill renamed so that it can be used alongside the native MediaStreamTrackGenerator.

This demo is based on this other demo by Stefan.