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Merge pull request #485 from thecodingmachine/webrtcOutSound

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FEAT: now play a sound when a user quit the webrtc group
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Kharhamel 2020-12-11 16:28:01 +01:00 committed by GitHub
parent e2c81c4eb1 a1aedaa594
commit 5e7d33e533
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GPG key ID: 4AEE18F83AFDEB23
4 changed files with 11 additions and 3 deletions
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3
front/dist/index.html vendored
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@ -113,6 +113,9 @@
<audio id="audio-webrtc-in"> <audio id="audio-webrtc-in">
<source src="/resources/objects/webrtc-in.mp3" type="audio/mp3"> <source src="/resources/objects/webrtc-in.mp3" type="audio/mp3">
</audio> </audio>
<audio id="audio-webrtc-out">
<source src="/resources/objects/webrtc-out.mp3" type="audio/mp3">
</audio>
<audio id="report-message"> <audio id="report-message">
<source src="/resources/objects/report-message.mp3" type="audio/mp3"> <source src="/resources/objects/report-message.mp3" type="audio/mp3">
</audio> </audio>

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front/dist/resources/objects/webrtc-out.mp3 vendored Normal file
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9
front/src/WebRtc/MediaManager.ts
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@ -38,6 +38,7 @@ export class MediaManager {
microphoneClose: HTMLImageElement; microphoneClose: HTMLImageElement;
microphone: HTMLImageElement; microphone: HTMLImageElement;
webrtcInAudio: HTMLAudioElement; webrtcInAudio: HTMLAudioElement;
private webrtcOutAudio: HTMLAudioElement;
constraintsMedia : MediaStreamConstraints = { constraintsMedia : MediaStreamConstraints = {
audio: true, audio: true,
video: videoConstraint video: videoConstraint
@ -57,8 +58,6 @@ export class MediaManager {
private discussionManager: DiscussionManager; private discussionManager: DiscussionManager;
private userInputManager?: UserInputManager;
private hasCamera = true; private hasCamera = true;
private triggerCloseJistiFrame : Map<String, Function> = new Map<String, Function>(); private triggerCloseJistiFrame : Map<String, Function> = new Map<String, Function>();
@ -67,7 +66,9 @@ export class MediaManager {
this.myCamVideo = HtmlUtils.getElementByIdOrFail<HTMLVideoElement>('myCamVideo'); this.myCamVideo = HtmlUtils.getElementByIdOrFail<HTMLVideoElement>('myCamVideo');
this.webrtcInAudio = HtmlUtils.getElementByIdOrFail<HTMLAudioElement>('audio-webrtc-in'); this.webrtcInAudio = HtmlUtils.getElementByIdOrFail<HTMLAudioElement>('audio-webrtc-in');
this.webrtcOutAudio = HtmlUtils.getElementByIdOrFail<HTMLAudioElement>('audio-webrtc-out');
this.webrtcInAudio.volume = 0.2; this.webrtcInAudio.volume = 0.2;
this.webrtcOutAudio.volume = 0.2;
this.microphoneBtn = HtmlUtils.getElementByIdOrFail<HTMLDivElement>('btn-micro'); this.microphoneBtn = HtmlUtils.getElementByIdOrFail<HTMLDivElement>('btn-micro');
this.microphoneClose = HtmlUtils.getElementByIdOrFail<HTMLImageElement>('microphone-close'); this.microphoneClose = HtmlUtils.getElementByIdOrFail<HTMLImageElement>('microphone-close');
@ -566,6 +567,10 @@ export class MediaManager {
removeActiveScreenSharingVideo(userId: string) { removeActiveScreenSharingVideo(userId: string) {
this.removeActiveVideo(`screen-sharing-${userId}`) this.removeActiveVideo(`screen-sharing-${userId}`)
} }
playWebrtcOutSound(): void {
this.webrtcOutAudio.play();
}
isConnecting(userId: string): void { isConnecting(userId: string): void {
const connectingSpinnerDiv = this.getSpinner(userId); const connectingSpinnerDiv = this.getSpinner(userId);

2
front/src/WebRtc/SimplePeer.ts
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@ -202,8 +202,8 @@ export class SimplePeer {
* This is triggered twice. Once by the server, and once by a remote client disconnecting * This is triggered twice. Once by the server, and once by a remote client disconnecting
*/ */
private closeConnection(userId : number) { private closeConnection(userId : number) {
mediaManager.playWebrtcOutSound();
try { try {
//mediaManager.removeActiveVideo(userId);
const peer = this.PeerConnectionArray.get(userId); const peer = this.PeerConnectionArray.get(userId);
if (peer === undefined) { if (peer === undefined) {
console.warn("closeConnection => Tried to close connection for user "+userId+" but could not find user"); console.warn("closeConnection => Tried to close connection for user "+userId+" but could not find user");