frigate/web/src/components/WebRtcPlayer.jsx

73 lines
2.1 KiB
React
Raw Normal View History

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 14:36:09 +03:00
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>
);
}