WebRTC-Audio-Stream-Example VS samples

Here's an example on how to do it: https://github.com/jmcker/WebRTC-Audio-Stream-Example

I record video from a webcam using MediaRecorder 100ms segments in the way described here. When the piece is ready, I send it using socket.io to the server (on a node.js 6.3.0 server). The server writes this piece to the database and gives it to everyone else on the socket. It is known for sure that the correct data is coming to the server, I can save any piece and start the video (then I glue them together and get the recording). The problem is that when the data is returned back to the client, it is invalid.

Thank you so far :) the "complex data processing" is a webpacked DSP hardware simulator, which has to exist in the global scope as it manipulates the DOM and cannot re-instantiate within the AudioWorkletGlobalScope. I have found this, which uses MediaStreamTrackProcessor, MediaStreamTrackGenerator and TransformStream to expose the microphone samples to the global scope and then "cheats" by sending them to an audio HTML component instead of directly to the audiocontext destination, mimicing actual realtime processing. This is what I was looking for, however:

WebRTC: https://webrtc.github.io/samples/

Demos are often snapshots in time, created to showcase an aspect of technology. They are typically small, self-contained, and single-purpose. Demonstrative projects can range from the practical stress-testing of a particular feature to creative, wacky ways to use them. Examples include WebRTC samples and the Dependabot Demo.

This repo includes all the tutorials that are on https://webrtc.github.io/samples/. The aim of the project is to convert all the sample code to a Svelte app.

Maybe you can identify which part of WebRTC is failing using some test webpages like https://webrtc.github.io/samples/

Here's a few samples. The video chat example will be very useful to you.