RecordRTC | WebRTC JavaScript library for audio/video as well as screen | Video Utils library

I am using PrimeNG in my Angular project. I am trying to make the table elements show in "stack" mode when responsive, which should be a simple thing according to the documentation.

Yet it does not work in my code, the rows don't get stacked for small screen sizes:

Here is my component.html:

I'm trying to record videos on browsers of mobile devices and send that videos to my PHP server. But when I inspect/debug my code in PHP the array $_FILES is empty. I'm sure that something is wrong in my code of JavaScript because of my lack of knowledge.

Here is my HTML / Javascript code :

I am working on a project where I need the user to be able to record screen, audio and microphone. At the moment I could only make it recognize screen and audio.

First I am capturing the screen and the audio from it and saving it to a variable. And then I am capturing that variable to show the a video component.

I'm working on integrating screen capturing in a framework I'm expanding. I'm requesting the MediaStream of the screen through the getDisplayMedia method and recording it using the RecordRTC library. Both Firefox and Chrome allow the user specify what exactly will be shared, like the entire screen, a specific window or a single tab (only Chrome). I noticed the choice here significantly affects the resulting video file size. The results below are from 30 second recordings, where the browser window filled the entire screen.

Chrome:

Entire Screen: 3.44 MB

Window: 0.81 MB

Tab: 0.15 MB

Firefox:

Entire Screen: 5.23 MB

Window: 3.56 MB

Ofcourse when recording a window opposed to the entire screen the resolution becomes slightly smaller. Like for the firefox recording: entire screen = 2560x1440 and window = 2488x1376, but I don't think that should make that much of a difference.

I've tried looking at the Chromium source code (as that's open-source and Chrome based) to figure out what the difference is between the different options, but can't seem to figure out what is going on. None of my Google searches were successful either.

Does anyone know what the reason for these large differences is?

I'm on Ubuntu 20.04 if that makes a difference.

If i use the following code to record a canvas animation:

Is there some formula like frames per second X resolution to determine the bitsPerSecond? I can't understand what values I'm supposed to be using. I want to specify the bitsPerSecond for 720p, 1080p and 4k video. I'm not sure if file type matters but this will most likely befor webm or mp4. I'm afraid some of my files are unnecessarily large while others I'm not using enough bits causing video glitches.

I did find values listed here.... https://restream.io/blog/what-is-a-good-upload-speed-for-streaming/ But even that im not sure how to convert over.

I am using RecordRTC https://github.com/muaz-khan/RecordRTC which is a wrapper for the MediaRecorder.

I've followed a bunch of tutorials on Google's speech to text, and have it all working fine locally. My setup is using websockets (socket.io) to communicate between a client Angular app and a node/express backend that does the server-side call to the speech API. I am using streaming-recognize ( https://cloud.google.com/speech-to-text/docs/streaming-recognize ) to read mic stream and return results.

This works fully locally, but I have an issue when running it on the gcloud app deploy instance in that I haven't actually installed the SoX dependency (done locally via brew install sox . This is a requirement for their example of setting up the mic stream.

I think I need to set up a virtual machine instance which I can provision with SoX, but also feel this seems a bit overkill - is there an alternative? I did try to manually parse and send the mic data stream as Uint8Array/ArrayBuffer chunkns with some success, but not much. I also read some hypothesis about non-SoX approaches to processing user mic stream, but to no avail. Eg with recordrtc.

The question is - what do I need to do to get this working in gcloud? Set up a vm instance, install sox, and use that? Or is there a SoX-free way to get this running? Guidance welcomed!

Here's the server error I get on gcloud - it seems to me because it has not got SoX on its path:

Bringing this over from softwareengineering. Was told this question may be better for stackoverflow.

I am sending a video stream of data to another peer and want to reassemble that data and make that stream the source of a video element. I record the data using the npm package RecordRTC and am getting a Blob of data every 1s.

I send it over an WebRTC Data Channel and initially tried to reassemble the data using the MediaSource API but turns out that MediaSource doesn't support data with a mimetype of video/webm;codecs=vp8,pcm. Are there any thoughts on how to reassemble this stream? Is it possible to modify the MediaSource API?

My only requirement of this stream of data is that the audio be encoded with pcm but if you have any thoughts or questions please let me know!

P.S. I thought opinion based questioned weren't for stackoverflow so that's why I posted there first.

I have a small test application to record the camera and sent the file to a directory on my server. The main file is as follow: