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Membrane WebRTC plugin based on ex_webrtc #1

Merged
merged 17 commits into from
Apr 4, 2024
2 changes: 1 addition & 1 deletion .formatter.exs
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[
inputs: [
"{lib,test,config}/**/*.{ex,exs}",
"{lib,test,config,examples}/**/*.{ex,exs}",
".formatter.exs",
"*.exs"
],
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22 changes: 10 additions & 12 deletions README.md
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# Membrane Template Plugin
# Membrane WebRTC Plugin

[![Hex.pm](https://img.shields.io/hexpm/v/membrane_template_plugin.svg)](https://hex.pm/packages/membrane_template_plugin)
[![API Docs](https://img.shields.io/badge/api-docs-yellow.svg?style=flat)](https://hexdocs.pm/membrane_template_plugin)
[![CircleCI](https://circleci.com/gh/membraneframework/membrane_template_plugin.svg?style=svg)](https://circleci.com/gh/membraneframework/membrane_template_plugin)
[![Hex.pm](https://img.shields.io/hexpm/v/membrane_webrtc_plugin.svg)](https://hex.pm/packages/membrane_webrtc_plugin)
[![API Docs](https://img.shields.io/badge/api-docs-yellow.svg?style=flat)](https://hexdocs.pm/membrane_webrtc_plugin)
[![CircleCI](https://circleci.com/gh/membraneframework/membrane_webrtc_plugin.svg?style=svg)](https://circleci.com/gh/membraneframework/membrane_webrtc_plugin)

This repository contains a template for new plugins.

Check out different branches for other flavors of this template.
Membrane Plugin for sending and receiving streams via WebRTC. It's based on [ex_webrtc](https://github.com/elixir-webrtc/ex_webrtc).

It's a part of the [Membrane Framework](https://membrane.stream).

## Installation

The package can be installed by adding `membrane_template_plugin` to your list of dependencies in `mix.exs`:
The package can be installed by adding `membrane_webrtc_plugin` to your list of dependencies in `mix.exs`:

```elixir
def deps do
[
{:membrane_template_plugin, "~> 0.1.0"}
{:membrane_webrtc_plugin, "~> 0.1.0"}
]
end
```

## Usage

TODO
The `examples` directory shows how to send and receive streams from a web browser.

## Copyright and License

Copyright 2020, [Software Mansion](https://swmansion.com/?utm_source=git&utm_medium=readme&utm_campaign=membrane_template_plugin)
Copyright 2020, [Software Mansion](https://swmansion.com/?utm_source=git&utm_medium=readme&utm_campaign=membrane_webrtc_plugin)

[![Software Mansion](https://logo.swmansion.com/logo?color=white&variant=desktop&width=200&tag=membrane-github)](https://swmansion.com/?utm_source=git&utm_medium=readme&utm_campaign=membrane_template_plugin)
[![Software Mansion](https://logo.swmansion.com/logo?color=white&variant=desktop&width=200&tag=membrane-github)](https://swmansion.com/?utm_source=git&utm_medium=readme&utm_campaign=membrane_webrtc_plugin)

Licensed under the [Apache License, Version 2.0](LICENSE)
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16 changes: 16 additions & 0 deletions examples/assets/browser_to_file/index.html
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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<meta http-equiv="X-UA-Compatible" content="ie=edge">
<title>Membrane WebRTC browser to file example</title>
</head>
<body style="background-color: black; color: white; font-family: Arial, Helvetica, sans-serif; min-height: 100vh; margin: 0px; padding: 5px 0px 5px 0px">
<main>
<h1>Membrane WebRTC browser to file example</h1>
<div id="status">Connecting</div>
</main>
<script src="script.js"></script>
</body>
</html>
51 changes: 51 additions & 0 deletions examples/assets/browser_to_file/script.js
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const pcConfig = { 'iceServers': [{ 'urls': 'stun:stun.l.google.com:19302' },] };
const mediaConstraints = { video: true, audio: true }

const proto = window.location.protocol === "https:" ? "wss:" : "ws:"
const ws = new WebSocket(`${proto}//${window.location.hostname}:8829`);
const conn_status = document.getElementById("status");
ws.onopen = _ => start_connection(ws);
ws.onclose = event => {
conn_status.innerHTML = "Disconnected"
console.log("WebSocket connection was terminated:", event);
}

const start_connection = async (ws) => {
const pc = new RTCPeerConnection(pcConfig);

pc.onicecandidate = event => {
if (event.candidate === null) return;
console.log("Sent ICE candidate:", event.candidate);
ws.send(JSON.stringify({ type: "ice_candidate", data: event.candidate }));
};

const localStream = await navigator.mediaDevices.getUserMedia(mediaConstraints);
for (const track of localStream.getTracks()) {
pc.addTrack(track, localStream);
}

ws.onmessage = async event => {
const { type, data } = JSON.parse(event.data);

switch (type) {
case "sdp_answer":
console.log("Received SDP answer:", data);
await pc.setRemoteDescription(data);
const button = document.createElement('button');
button.innerHTML = "Disconnect";
button.onclick = () => ws.close();
conn_status.innerHTML = "Connected ";
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conn_status.appendChild(button);
break;
case "ice_candidate":
console.log("Recieved ICE candidate:", data);
await pc.addIceCandidate(data);
break;
}
};

const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
console.log("Sent SDP offer:", offer)
ws.send(JSON.stringify({ type: "sdp_offer", data: offer }));
};
16 changes: 16 additions & 0 deletions examples/assets/file_to_browser/index.html
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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<meta http-equiv="X-UA-Compatible" content="ie=edge">
<title>Membrane WebRTC file to browser example</title>
</head>
<body style="background-color: black; color: white; font-family: Arial, Helvetica, sans-serif; min-height: 100vh; margin: 0px; padding: 5px 0px 5px 0px">
<main>
<h1>Membrane WebRTC file to browser example</h1>
<video id="videoPlayer" controls muted autoplay></video>
</main>
<script src="script.js"></script>
</body>
</html>
37 changes: 37 additions & 0 deletions examples/assets/file_to_browser/script.js
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const videoPlayer = document.getElementById("videoPlayer");
const pcConfig = { 'iceServers': [{ 'urls': 'stun:stun.l.google.com:19302' },] };
const proto = window.location.protocol === "https:" ? "wss:" : "ws:"
const ws = new WebSocket(`${proto}//${window.location.hostname}:8829`);
ws.onopen = () => start_connection(ws);
ws.onclose = event => console.log("WebSocket connection was terminated:", event);

const start_connection = async (ws) => {
videoPlayer.srcObject = new MediaStream();

const pc = new RTCPeerConnection(pcConfig);
pc.ontrack = event => videoPlayer.srcObject.addTrack(event.track);
pc.onicecandidate = event => {
if (event.candidate === null) return;

console.log("Sent ICE candidate:", event.candidate);
ws.send(JSON.stringify({ type: "ice_candidate", data: event.candidate }));
};

ws.onmessage = async event => {
const { type, data } = JSON.parse(event.data);

switch (type) {
case "sdp_offer":
console.log("Received SDP offer:", data);
await pc.setRemoteDescription(data);
const answer = await pc.createAnswer();
await pc.setLocalDescription(answer);
ws.send(JSON.stringify({ type: "sdp_answer", data: answer }));
console.log("Sent SDP answer:", answer)
break;
case "ice_candidate":
console.log("Recieved ICE candidate:", data);
await pc.addIceCandidate(data);
}
};
};
72 changes: 72 additions & 0 deletions examples/browser_to_file.exs
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# This example receives audio and video from a browser via WebRTC
# and saves it to a `recording.mkv` file.
# To run it, type `elixir browser_to_file.exs` and open
# http://localhost:8000/index.html in your browser. To finish recording,
# click the `disconnect` button or close the tab.

Logger.configure(level: :info)

Mix.install([
{:membrane_webrtc_plugin, path: "#{__DIR__}/.."},
:membrane_file_plugin,
:membrane_realtimer_plugin,
:membrane_matroska_plugin,
:membrane_opus_plugin,
:membrane_h264_plugin
])

defmodule Example.Pipeline do
use Membrane.Pipeline

alias Membrane.WebRTC

@impl true
def handle_init(_ctx, opts) do
spec =
[
child(:webrtc, %WebRTC.Source{
signaling: {:websocket, port: opts[:port]}
}),
child(:matroska, Membrane.Matroska.Muxer),
get_child(:webrtc)
|> via_out(:output, options: [kind: :audio])
|> child(Membrane.Opus.Parser)
|> get_child(:matroska),
get_child(:webrtc)
|> via_out(:output, options: [kind: :video])
|> get_child(:matroska),
get_child(:matroska)
|> child(:sink, %Membrane.File.Sink{location: "recording.mkv"})
]

{[spec: spec], %{}}
end

@impl true
def handle_element_end_of_stream(:sink, :input, _ctx, state) do
{[terminate: :normal], state}
end

@impl true
def handle_element_end_of_stream(_element, _pad, _ctx, state) do
{[], state}
end
end

{:ok, supervisor, _pipeline} = Membrane.Pipeline.start_link(Example.Pipeline, port: 8829)
Process.monitor(supervisor)

:ok = :inets.start()

{:ok, _server} =
:inets.start(:httpd,
bind_address: ~c"localhost",
port: 8000,
document_root: ~c"#{__DIR__}/assets/browser_to_file",
server_name: ~c"webrtc",
server_root: "/tmp"
)

receive do
{:DOWN, _ref, :process, ^supervisor, _reason} -> :ok
end
94 changes: 94 additions & 0 deletions examples/file_to_browser.exs
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# This example reads a short part of the Big Buck Bunny movie
# from an `.mkv` file and streams it to a browser.
# To run it, type `elixir file_to_browser.exs` and open
# http://localhost:8000/index.html in your browser.
# Note that due to browsers' policy, you need to manually unmute
# audio in the player to hear the sound.

Logger.configure(level: :info)

Mix.install([
{:membrane_webrtc_plugin, path: "#{__DIR__}/.."},
:membrane_file_plugin,
:membrane_realtimer_plugin,
:membrane_matroska_plugin,
:membrane_opus_plugin
])

defmodule Example.Pipeline do
use Membrane.Pipeline

alias Membrane.WebRTC

@impl true
def handle_init(_ctx, opts) do
spec =
child(%Membrane.File.Source{location: "#{__DIR__}/assets/bbb_vp8.mkv"})
|> child(:demuxer, Membrane.Matroska.Demuxer)

{[spec: spec], %{audio_track: nil, video_track: nil, port: opts[:port]}}
end

@impl true
def handle_child_notification({:new_track, {id, info}}, :demuxer, _ctx, state) do
state =
case info.codec do
:opus -> %{state | audio_track: id}
:h264 -> %{state | video_track: id}
:vp8 -> %{state | video_track: id}
end

if state.audio_track && state.video_track do
spec = [
child(:webrtc, %WebRTC.Sink{signaling: {:websocket, port: state.port}}),
get_child(:demuxer)
|> via_out(Pad.ref(:output, state.video_track))
|> child({:realtimer, :video_track}, Membrane.Realtimer)
|> via_in(Pad.ref(:input, :video_track), options: [kind: :video])
|> get_child(:webrtc),
get_child(:demuxer)
|> via_out(Pad.ref(:output, state.audio_track))
|> child({:realtimer, :audio_track}, Membrane.Realtimer)
|> via_in(Pad.ref(:input, :audio_track), options: [kind: :audio])
|> get_child(:webrtc)
]

{[spec: spec], state}
else
{[], state}
end
end

@impl true
def handle_child_notification({:end_of_stream, track}, :webrtc, _ctx, state) do
state = %{state | track => nil}

if !state.audio_track && !state.video_track do
{[terminate: :normal], state}
else
{[], state}
end
end

@impl true
def handle_child_notification(_notification, _child, _ctx, state) do
{[], state}
end
end

{:ok, supervisor, _pipeline} = Membrane.Pipeline.start_link(Example.Pipeline, port: 8829)
Process.monitor(supervisor)
:ok = :inets.start()

{:ok, _server} =
:inets.start(:httpd,
bind_address: ~c"localhost",
port: 8000,
document_root: ~c"#{__DIR__}/assets/file_to_browser",
server_name: ~c"webrtc",
server_root: "/tmp"
)

receive do
{:DOWN, _ref, :process, ^supervisor, _reason} -> :ok
end
2 changes: 0 additions & 2 deletions lib/membrane_template.ex

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