Files
frigate/web/src/components/WebRtcPlayer.jsx
Nicolas Mowen d8123d2497 Add go2rtc and add restream role / live source (#4082)
* Pull go2rtc dependency

* Add go2rtc to local services and add to s6

* Add relay controller for go2rtc

* Add restream role

* Add restream role

* Add restream to nginx

* Add camera live source config

* Disable RTMP by default and use restream

* Use go2rtc for camera config

* Fix go2rtc move

* Start restream on frigate start

* Send restream to camera level

* Fix restream

* Make sure jsmpeg works as expected

* Make view rspect live size config

* Tweak player options to fit live view

* Adjust VideoPlayer to accept live option which disables irrelevant controls

* Add multiple options from restream live view

* Add base for webrtc option

* Setup specific restream modules

* Make mp4 the default streaming for now

* Expose 8554 for rtsp relay from go2rtc

* Formatting

* Update docs to suggest new restream method.

* Update docs to reflect restream role

* Update docs to reflect restream role

* Add webrtc player

* Improvements to webRTC

* Support webrtc

* Cleanup

* Adjust rtmp test and add restream test

* Fix tests

* Add restream tests

* Add live view docs and show different options

* Small docs tweak

* Support all stream types

* Update to beta 9 of go2rtc

* Formatting

* Make jsmpeg the default

* Support wss if made from https

* Support wss if made from https

* Use onEffect

* Set url outside onEffect

* Fix passed deps

* Update docs about required host mode

* Try memo instead

* Close websocket on changing camera

* Formatting

* Close pc connection

* Set video source to null on cleanup

* Use full path since go2rtc can't see PATH var

* Adjust audio codec to enable browser audio by default

* Cleanup stream creation

* Add restream tests

* Format tests

* Mock requests

* Adjust paths

* Move stream configs to restream

* Remove live source

* Remove live config

* Use live persistence for which view to use on each camera

* Fix live sizes

* Only use jsmpeg sizes for jsmpeg live

* Set max live size

* Remove access of live config

* Add selector for live view source in web view

* Remove RTMP from default list of roles

* Update docs

* Fix tests

* Fix docs for live view modes

* make default undefined to avoid race condition

* Wait until camera source is loaded to avoid race condition

* Fix tests

* Add config to go2rtc

* Work with config

* Set full path for config

* Set to use stun

* Check for mounted file

* Look for frigate-go2rtc

* Update docs to reflect webRTC configuration.

* Add link to go2rtc config

* Update docs to be more clear

* Update docs to be more clear

* Update format

Co-authored-by: Felipe Santos <felipecassiors@gmail.com>

* Update live docs

* Improve bash startup script

* Add option to force audio compatibility

* Formatting

* Fix mapping

* Fix broken link

* Update go2rtc version

* Get go2rtc webui working

* Add support for mse

* Remove mp4 option

* Undo changes to video player

* Update docs for new live view options

* Make separate path for mse

* Remove unused

* Remove mp4 path

* Try to get go2rtc proxy working

* Try to get go2rtc proxy working

* Remove unused callback

* Allow websocket on restrea dashboard

* Make mse default stream option

* Fix mse sizing

* don't assume roles is defined

* Remove nginx mapping to go2rtc ui

Co-authored-by: Felipe Santos <felipecassiors@gmail.com>
Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
2022-11-02 06:36:09 -05:00

73 lines
2.1 KiB
JavaScript

import { h } from 'preact';
import { baseUrl } from '../api/baseUrl';
import { useEffect } from 'preact/hooks';
export default function WebRtcPlayer({ camera, width, height }) {
const url = `${baseUrl.replace(/^http/, 'ws')}live/webrtc/api/ws?src=${camera}`;
useEffect(() => {
const ws = new WebSocket(url);
ws.onopen = () => {
pc.createOffer().then((offer) => {
pc.setLocalDescription(offer).then(() => {
const msg = { type: 'webrtc/offer', value: pc.localDescription.sdp };
ws.send(JSON.stringify(msg));
});
});
};
ws.onmessage = (ev) => {
const msg = JSON.parse(ev.data);
if (msg.type === 'webrtc/candidate') {
pc.addIceCandidate({ candidate: msg.value, sdpMid: '' });
} else if (msg.type === 'webrtc/answer') {
pc.setRemoteDescription({ type: 'answer', sdp: msg.value });
}
};
const pc = new RTCPeerConnection({
iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
});
pc.onicecandidate = (ev) => {
if (ev.candidate !== null) {
ws.send(
JSON.stringify({
type: 'webrtc/candidate',
value: ev.candidate.toJSON().candidate,
})
);
}
};
pc.ontrack = (ev) => {
const video = document.getElementById('video');
// when audio track not exist in Chrome
if (ev.streams.length === 0) return;
// when audio track not exist in Firefox
if (ev.streams[0].id[0] === '{') return;
// when stream already init
if (video.srcObject !== null) return;
video.srcObject = ev.streams[0];
};
// Safari don't support "offerToReceiveVideo"
// so need to create transeivers manually
pc.addTransceiver('video', { direction: 'recvonly' });
pc.addTransceiver('audio', { direction: 'recvonly' });
return () => {
const video = document.getElementById('video');
video.srcObject = null;
pc.close();
ws.close();
};
}, [url]);
return (
<div>
<video id="video" autoplay playsinline controls muted width={width} height={height} />
</div>
);
}