RecordRTC VS samples

It is all local (and open source) as far as I can tell.

My research; This github looks like it could work https://github.com/muaz-khan/RecordRTC I've found a few APIs and services that I could use but seemed to cost a lot.

I suggest trying Firefox. Also, here's a website where you can test WebRTC screen recording: https://www.webrtc-experiment.com/RecordRTC/ which might be handy for testing it yourself without going through meeting software.

Posting here because I wanted to add it to the Arch Wiki page, but it doesn't work for me. It's possible now to enable the HW accelerated video encoding feature with the command-line flag --enable-features=VaapiVideoEncoder. Testing with the RecordRTC experiment shows that a software codec is still being used (check intel_gpu_top). My HW can only do H264 and VP8, so maybe someone with VP9 HW accelerated encoding will have a better luck with this.

Does recording in Firefox work? test using https://www.webrtc-experiment.com/RecordRTC/

I think this shouldn't be all to hard. I have a repo that only records a microphone and saves it link here. It should get you started to doing it with video as well using the same API even, here is a sample of RecordRTC. If you have any questions feel free to DM me.

For example, Firefox WebRTC can record windows using Portal and dynamically resizes the video to fit whenever you resize the window.

Read this https://github.com/muaz-khan/RecordRTC/issues/97

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.