Code refactoring for video-rtc.js

This commit is contained in:
Alex X
2025-09-30 12:12:29 +03:00
parent dd7130d2d4
commit 7d9862202a
+14 -14
View File
@@ -185,7 +185,7 @@ export class VideoRTC extends HTMLElement {
/** @param {Function} isSupported */
codecs(isSupported) {
return this.CODECS
.filter(codec => this.media.indexOf(codec.indexOf('vc1') > 0 ? 'video' : 'audio') >= 0)
.filter(codec => this.media.includes(codec.includes('vc1') ? 'video' : 'audio'))
.filter(codec => isSupported(`video/mp4; codecs="${codec}"`)).join();
}
@@ -350,23 +350,23 @@ export class VideoRTC extends HTMLElement {
const modes = [];
if (this.mode.indexOf('mse') >= 0 && ('MediaSource' in window || 'ManagedMediaSource' in window)) {
if (this.mode.includes('mse') && ('MediaSource' in window || 'ManagedMediaSource' in window)) {
modes.push('mse');
this.onmse();
} else if (this.mode.indexOf('hls') >= 0 && this.video.canPlayType('application/vnd.apple.mpegurl')) {
} else if (this.mode.includes('hls') && this.video.canPlayType('application/vnd.apple.mpegurl')) {
modes.push('hls');
this.onhls();
} else if (this.mode.indexOf('mp4') >= 0) {
} else if (this.mode.includes('mp4')) {
modes.push('mp4');
this.onmp4();
}
if (this.mode.indexOf('webrtc') >= 0 && 'RTCPeerConnection' in window) {
if (this.mode.includes('webrtc') && 'RTCPeerConnection' in window) {
modes.push('webrtc');
this.onwebrtc();
}
if (this.mode.indexOf('mjpeg') >= 0) {
if (this.mode.includes('mjpeg')) {
if (modes.length) {
this.onmessage['mjpeg'] = msg => {
if (msg.type !== 'error' || msg.value.indexOf(modes[0]) !== 0) return;
@@ -490,7 +490,7 @@ export class VideoRTC extends HTMLElement {
const pc = new RTCPeerConnection(this.pcConfig);
pc.addEventListener('icecandidate', ev => {
if (ev.candidate && this.mode.indexOf('webrtc/tcp') >= 0 && ev.candidate.protocol === 'udp') return;
if (ev.candidate && this.mode.includes('webrtc/tcp') && ev.candidate.protocol === 'udp') return;
const candidate = ev.candidate ? ev.candidate.toJSON().candidate : '';
this.send({type: 'webrtc/candidate', value: candidate});
@@ -518,7 +518,7 @@ export class VideoRTC extends HTMLElement {
this.onmessage['webrtc'] = msg => {
switch (msg.type) {
case 'webrtc/candidate':
if (this.mode.indexOf('webrtc/tcp') >= 0 && msg.value.indexOf(' udp ') > 0) return;
if (this.mode.includes('webrtc/tcp') && msg.value.includes(' udp ')) return;
pc.addIceCandidate({candidate: msg.value, sdpMid: '0'}).catch(er => {
console.warn(er);
@@ -530,7 +530,7 @@ export class VideoRTC extends HTMLElement {
});
break;
case 'error':
if (msg.value.indexOf('webrtc/offer') < 0) return;
if (!msg.value.includes('webrtc/offer')) return;
pc.close();
}
};
@@ -549,7 +549,7 @@ export class VideoRTC extends HTMLElement {
*/
async createOffer(pc) {
try {
if (this.media.indexOf('microphone') >= 0) {
if (this.media.includes('microphone')) {
const media = await navigator.mediaDevices.getUserMedia({audio: true});
media.getTracks().forEach(track => {
pc.addTransceiver(track, {direction: 'sendonly'});
@@ -560,7 +560,7 @@ export class VideoRTC extends HTMLElement {
}
for (const kind of ['video', 'audio']) {
if (this.media.indexOf(kind) >= 0) {
if (this.media.includes(kind)) {
pc.addTransceiver(kind, {direction: 'recvonly'});
}
}
@@ -587,9 +587,9 @@ export class VideoRTC extends HTMLElement {
}
if (stream.getAudioTracks().length > 0) rtcPriority += 0x102;
if (this.mseCodecs.indexOf('hvc1.') >= 0) msePriority += 0x230;
if (this.mseCodecs.indexOf('avc1.') >= 0) msePriority += 0x210;
if (this.mseCodecs.indexOf('mp4a.') >= 0) msePriority += 0x101;
if (this.mseCodecs.includes('hvc1.')) msePriority += 0x230;
if (this.mseCodecs.includes('avc1.')) msePriority += 0x210;
if (this.mseCodecs.includes('mp4a.')) msePriority += 0x101;
if (rtcPriority >= msePriority) {
this.video.srcObject = stream;