libdatachannel VS captureSystemAudio

You can use https://github.com/paullouisageneau/libdatachannel for your C/C++ integration needs. It's 10k lines. So the answer is 0. Its required dependencies (I assume this as they are git submodules in deps) are more than 100k lines, though, srtp support making the bulk of it. On my machine it took 11 seconds to compile it.

Irssi is 64k lines (plus its dependencies), so I guess that makes WebRTC complicated.

Can't argue that DCC isn't simple, but perhaps the protocol deviced decades ago is a bit too simple.

Most of the work happened in the libdatachannel! You can check out my PR here[0]

[0] https://github.com/paullouisageneau/libdatachannel/commit/a6...

I am really excited about https://github.com/sepfy/libpeer. It has examples ready for ESP32 etc....

When working on KVS I wasn't familiar with the embedded space at all. I saw 'heavyweight' embedded where you were running on Linux. Then you had RTOS/No OS at all. I wasn't prepared for these devices at all. If we can make WebRTC work in the embedded space I think it will really accelerate what developers are able to build!

Remotely driven cars, security cameras, robots in hospitals that bring iPads to infectious patients etc... Creative people are building amazing things. The WebRTC/video space needs to work harder and support them :)

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I love how diverse the WebRTC space is now. Outside of this implementation you have plenty of other options!

* https://github.com/shinyoshiaki/werift-webrtc (Typescript)

* https://github.com/pion/webrtc (Golang)

* https://github.com/webrtc-rs/webrtc (Rust)

* https://github.com/algesten/str0m (Rust)

* hhttps://github.com/sepfy/libpeer (C/Embedded)

* https://webrtc.googlesource.com/src/ (C++)

* https://github.com/sipsorcery-org/sipsorcery (C#)

* https://github.com/paullouisageneau/libdatachannel (C++)

* https://github.com/elixir-webrtc (Elixir)

* https://github.com/aiortc/aiortc (Python)

* GStreamer’s webrtcbin (C)

See https://github.com/sipsorcery/webrtc-echoes for examples of some running against each other.

This is not true, there are actually multiple WebRTC implementations in different languages besides the reference library: aiortc (python), libdatachannel (C++), sipsorcery (C#),webrtc-rs (rust), werift (Typescript), and Amazon Kinesis (C)

Use any means necessary to transfer your data across devices. Could be IPFS, could be FTP, could be EventSource, WebSocket, WebTransport, Fetch, whatever. See https://github.com/guest271314/secure-file-transfer; offscreen-webrtc, https://github.com/paullouisageneau/libdatachannel.

I've successfully used libdatachannel to Web pages to connect native applications and stream data to the browser.

Alternatively you can use your server as a signaling server for WebRTC (Insertable Streams ("Breakout Box"), or data channels https://github.com/paullouisageneau/libdatachannel), then users (peers) can exchange data themselves and you don't need to store anything, see True End-to-End Encryption with WebRTC Insertable Streams, A complete example for a WebRTC datachannel with manual signaling.

there is also a c++ library that can be used to open a data channel connection, I think a number of SFU “servers?” use this library (I wish I had) https://libdatachannel.org/

For networking and terminating WebRTC connections from cpp libdatachannel is you easiest bet. For transcoding you can pick whatever library you like, if any. The other way would be to use the chrome WebRTC stack and modify it for server use. It can and has been done but is very involved.

Synchronous. Though non-blocking when we read the message from the browser, execute the command, then stream data output from the local application to the browser piped from QuickJS std.popen(), capture_system_audio.js

I've already achieved the requirement multiple ways, already; from using Native Messaging https://github.com/guest271314/native-messaging-espeak-ng, to using GNU Core Utilities tail, to Deno.watchFs() https://github.com/guest271314/fs, et al., see captureSystemAudio. The one approach I have not yet achieved is compiling to Emscripten - with SSML support.

Sure, see https://github.com/guest271314/captureSystemAudio/blob/master/native_messaging/capture_system_audio/background.js, in pertinent part

This is my second time writing C source code capture_system_audio.c. I got a lot of help doing that, too . I basically implemented the same Native Messaging host algorithm in C, C++, Python, WASI/WebAssembly, and JavaScript (Node.js, QuickJS) which is the programming language I write code most using.

I just fetch the nightly node executable, get rid of everything else in the archive, use .mjs extension for import to work without package.json, then I can connect to the host https://github.com/guest271314/native-messaging-nodejs, and start subprocesses, e.g., raw PCM streams, where stdout is streamed to the client (browser), e.g, https://github.com/guest271314/captureSystemAudio/blob/master/native_messaging/capture_system_audio/capture_system_audio_node.js.

Technically I can just serve the raw data as long as the server is capable of opening creating sub-processes, e.g., using EventSource https://plnkr.co/edit/qX5tepJ38BuDOSZj?preview, in which case I would just need to convert this https://github.com/guest271314/captureSystemAudio/blob/master/native_messaging/capture_system_audio/capture_system_audio.c to accept GET request, read the URL, and send the named event stream to the client.

Chromium and Chrome refuses to capture monitor devices on Linux. Read this https://github.com/edisionnano/Screenshare-with-audio-on-Discord-with-Linux README and the README at https://github.com/guest271314/captureSystemAudio very carefully. At best you can capture tab audio, not whatever is output to speakears and headphones, in spite of the systemAudio constraints, which is poorly named, and actually misleading.

How is capturing speechSynthesis.speak() a security vector https://github.com/guest271314/captureSystemAudio?

Capture audio output to speakers or headphones to MP3 or Opus in WebM https://github.com/guest271314/captureSystemAudio/tree/master/native_messaging/capture_system_audio

An example of using Native Messaging to capture "What-U-Hear", sound output to speakers and headphone, launching native GUI's https://github.com/guest271314/captureSystemAudio/tree/master/native_messaging/capture_system_audio; an adapter for espeak-ng so we can capture the output of speech synthesis engine and stream output to the browser https://github.com/guest271314/native-messaging-espeak-ng.