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+// Copyright Epic Games, Inc. All Rights Reserved.
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+
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+function webRtcPlayer(parOptions) {
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+ parOptions = typeof parOptions !== 'undefined' ? parOptions : {};
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+
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+ var self = this;
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+ const urlParams = new URLSearchParams(window.location.search);
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+
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+ //**********************
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+ //Config setup
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+ //**********************
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+ this.cfg = typeof parOptions.peerConnectionOptions !== 'undefined' ? parOptions.peerConnectionOptions : {};
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+ this.cfg.sdpSemantics = 'unified-plan';
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+ // this.cfg.rtcAudioJitterBufferMaxPackets = 10;
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+ // this.cfg.rtcAudioJitterBufferFastAccelerate = true;
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+ // this.cfg.rtcAudioJitterBufferMinDelayMs = 0;
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+
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+ // If this is true in Chrome 89+ SDP is sent that is incompatible with UE Pixel Streaming 4.26 and below.
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+ // However 4.27 Pixel Streaming does not need this set to false as it supports `offerExtmapAllowMixed`.
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+ // tdlr; uncomment this line for older versions of Pixel Streaming that need Chrome 89+.
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+ this.cfg.offerExtmapAllowMixed = false;
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+
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+ this.forceTURN = urlParams.has('ForceTURN');
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+ if (this.forceTURN) {
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+ console.log("Forcing TURN usage by setting ICE Transport Policy in peer connection config.");
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+ this.cfg.iceTransportPolicy = "relay";
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+ }
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+
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+ this.cfg.bundlePolicy = "balanced";
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+ this.forceMaxBundle = urlParams.has('ForceMaxBundle');
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+ if (this.forceMaxBundle) {
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+ this.cfg.bundlePolicy = "max-bundle";
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+ }
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+
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+ //**********************
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+ //Variables
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+ //**********************
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+ this.pcClient = null;
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+ this.dcClient = null;
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+ this.tnClient = null;
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+
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+ this.sdpConstraints = {
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+ offerToReceiveAudio: 1, //Note: if you don't need audio you can get improved latency by turning this off.
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+ offerToReceiveVideo: 1,
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+ voiceActivityDetection: false
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+ };
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+
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+ // See https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit for values (this is needed for Firefox to be consistent with Chrome.)
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+ this.dataChannelOptions = { ordered: true };
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+
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+ // This is useful if the video/audio needs to autoplay (without user input) as browsers do not allow autoplay non-muted of sound sources without user interaction.
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+ this.startVideoMuted = typeof parOptions.startVideoMuted !== 'undefined' ? parOptions.startVideoMuted : false;
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+ this.autoPlayAudio = typeof parOptions.autoPlayAudio !== 'undefined' ? parOptions.autoPlayAudio : true;
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+
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+ // To enable mic in browser use SSL/localhost and have ?useMic in the query string.
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+ this.useMic = urlParams.has('useMic');
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+ if (!this.useMic) {
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+ console.log("Microphone access is not enabled. Pass ?useMic in the url to enable it.");
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+ }
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+
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+ // When ?useMic check for SSL or localhost
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+ let isLocalhostConnection = location.hostname === "localhost" || location.hostname === "127.0.0.1";
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+ let isHttpsConnection = location.protocol === 'https:';
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+ if (this.useMic && !isLocalhostConnection && !isHttpsConnection) {
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+ this.useMic = false;
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+ console.error("Microphone access in the browser will not work if you are not on HTTPS or localhost. Disabling mic access.");
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+ console.error("For testing you can enable HTTP microphone access Chrome by visiting chrome://flags/ and enabling 'unsafely-treat-insecure-origin-as-secure'");
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+ }
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+
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+ // Prefer SFU or P2P connection
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+ this.preferSFU = urlParams.has('preferSFU');
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+ console.log(this.preferSFU ?
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+ "The browser will signal it would prefer an SFU connection. Remove ?preferSFU from the url to signal for P2P usage." :
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+ "The browser will signal for a P2P connection. Pass ?preferSFU in the url to signal for SFU usage.");
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+
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+ // Latency tester
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+ this.latencyTestTimings =
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+ {
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+ TestStartTimeMs: null,
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+ UEReceiptTimeMs: null,
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+ UEEncodeMs: null,
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+ UECaptureToSendMs: null,
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+ UETransmissionTimeMs: null,
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+ BrowserReceiptTimeMs: null,
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+ FrameDisplayDeltaTimeMs: null,
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+ Reset: function () {
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+ this.TestStartTimeMs = null;
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+ this.UEReceiptTimeMs = null;
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+ this.UEEncodeMs = null,
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+ this.UECaptureToSendMs = null,
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+ this.UETransmissionTimeMs = null;
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+ this.BrowserReceiptTimeMs = null;
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+ this.FrameDisplayDeltaTimeMs = null;
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+ },
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+ SetUETimings: function (UETimings) {
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+ this.UEReceiptTimeMs = UETimings.ReceiptTimeMs;
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+ this.UEEncodeMs = UETimings.EncodeMs,
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+ this.UECaptureToSendMs = UETimings.CaptureToSendMs,
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+ this.UETransmissionTimeMs = UETimings.TransmissionTimeMs;
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+ this.BrowserReceiptTimeMs = Date.now();
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+ this.OnAllLatencyTimingsReady(this);
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+ },
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+ SetFrameDisplayDeltaTime: function (DeltaTimeMs) {
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+ if (this.FrameDisplayDeltaTimeMs == null) {
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+ this.FrameDisplayDeltaTimeMs = Math.round(DeltaTimeMs);
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+ this.OnAllLatencyTimingsReady(this);
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+ }
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+ },
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+ OnAllLatencyTimingsReady: function (Timings) { }
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+ }
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+
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+ //**********************
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+ //Functions
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+ //**********************
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+
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+ //Create Video element and expose that as a parameter
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+ this.createWebRtcVideo = function () {
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+ var video = document.createElement('video');
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+
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+ video.id = "streamingVideo";
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+ video.playsInline = true;
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+ video.disablepictureinpicture = true;
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+ // video.muted = self.startVideoMuted;
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+ // 音频
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+ video.muted = true;
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+ // 开启全屏
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+ video.style.width = "100%"
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+ video.style.height = "100%"
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+ video.style.objectFit = "fill"
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+ video.style.margin = 0;
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+ video.style.padding = 0;
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+ video.style.top = 0;
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+ video.style.left = 0;
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+ video.style.position = "relative";
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+ video.style.zIndex = 100;
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+ video.style.cursor = "pointer";
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+ // video.style.overflow="hidden";
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+
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+
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+ video.addEventListener('loadedmetadata', function (e) {
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+ if (self.onVideoInitialised) {
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+ self.onVideoInitialised();
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+ }
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+ }, true);
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+
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+ // Check if request video frame callback is supported
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+ if ('requestVideoFrameCallback' in HTMLVideoElement.prototype) {
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+ // The API is supported!
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+
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+ const onVideoFrameReady = (now, metadata) => {
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+
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+ if (metadata.receiveTime && metadata.expectedDisplayTime) {
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+ const receiveToCompositeMs = metadata.presentationTime - metadata.receiveTime;
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+ self.aggregatedStats.receiveToCompositeMs = receiveToCompositeMs;
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+ }
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+
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+
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+ // Re-register the callback to be notified about the next frame.
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+ video.requestVideoFrameCallback(onVideoFrameReady);
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+ };
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+
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+ // Initially register the callback to be notified about the first frame.
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+ video.requestVideoFrameCallback(onVideoFrameReady);
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+ }
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+
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+ return video;
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+ }
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+
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+ this.video = this.createWebRtcVideo();
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+ this.availableVideoStreams = new Map();
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+
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+ function onsignalingstatechange(state) {
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+ console.info('Signaling state change. |', state.srcElement.signalingState, "|")
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+ };
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+
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+ function oniceconnectionstatechange(state) {
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+ console.info('Browser ICE connection |', state.srcElement.iceConnectionState, '|')
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+ };
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+
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+ function onicegatheringstatechange(state) {
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+ console.info('Browser ICE gathering |', state.srcElement.iceGatheringState, '|')
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+ };
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+
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+ function handleOnTrack(e) {
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+ if (e.track) {
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+ console.log('Got track. | Kind=' + e.track.kind + ' | Id=' + e.track.id + ' | readyState=' + e.track.readyState + ' |');
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+ }
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+
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+ if (e.track.kind == "audio") {
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+ handleOnAudioTrack(e.streams[0]);
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+ return;
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+ }
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+ else (e.track.kind == "video")
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+ {
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+ for (const s of e.streams) {
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+ if (!self.availableVideoStreams.has(s.id)) {
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+ self.availableVideoStreams.set(s.id, s);
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+ }
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+ }
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+
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+ self.video.srcObject = e.streams[0];
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+
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+ // All tracks are added "muted" by WebRTC/browser and become unmuted when media is being sent
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+ e.track.onunmute = () => {
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+ self.video.srcObject = e.streams[0];
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+ self.onNewVideoTrack(e.streams);
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+ }
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+ }
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+ };
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+
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+ function handleOnAudioTrack(audioMediaStream) {
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+ // do nothing the video has the same media stream as the audio track we have here (they are linked)
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+ if (self.video.srcObject == audioMediaStream) {
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+ return;
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+ }
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+ // video element has some other media stream that is not associated with this audio track
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+ else if (self.video.srcObject && self.video.srcObject !== audioMediaStream) {
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+ // create a new audio element
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+ let audioElem = document.createElement("Audio");
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+ audioElem.srcObject = audioMediaStream;
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+
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+ // there is no way to autoplay audio (even muted), so we defer audio until first click
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+ if (!self.autoPlayAudio) {
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+
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+ let clickToPlayAudio = function () {
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+ audioElem.play();
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+ self.video.removeEventListener("click", clickToPlayAudio);
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+ };
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+
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+ self.video.addEventListener("click", clickToPlayAudio);
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+ }
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+ // we assume the user has clicked somewhere on the page and autoplaying audio will work
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+ else {
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+ audioElem.play();
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+ }
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+ console.log('Created new audio element to play seperate audio stream.');
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+ }
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+
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+ }
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+
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+ function onDataChannel(dataChannelEvent) {
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+ // This is the primary data channel code path when we are "receiving"
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+ console.log("Data channel created for us by browser as we are a receiving peer.");
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+ self.dcClient = dataChannelEvent.channel;
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+ setupDataChannelCallbacks(self.dcClient);
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+ }
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+
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+ function createDataChannel(pc, label, options) {
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+ // This is the primary data channel code path when we are "offering"
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+ let datachannel = pc.createDataChannel(label, options);
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+ console.log(`Created datachannel (${label})`);
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+ setupDataChannelCallbacks(datachannel);
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+ return datachannel;
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+ }
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+
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+ function setupDataChannelCallbacks(datachannel) {
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+ try {
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+ // Inform browser we would like binary data as an ArrayBuffer (FF chooses Blob by default!)
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+ datachannel.binaryType = "arraybuffer";
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+
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+ datachannel.onopen = function (e) {
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+ console.log("Data channel connected");
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+ if (self.onDataChannelConnected) {
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+ self.onDataChannelConnected();
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+ }
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+ }
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+
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+ datachannel.onclose = function (e) {
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+ console.log("Data channel connected", e);
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+ }
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+
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+ datachannel.onmessage = function (e) {
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+ if (self.onDataChannelMessage) {
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+ self.onDataChannelMessage(e.data);
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+ }
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+ }
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+
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+ datachannel.onerror = function (e) {
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+ console.error("Data channel error", e);
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+ }
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+
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+ return datachannel;
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+ } catch (e) {
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+ console.warn('No data channel', e);
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+ return null;
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+ }
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+ }
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+
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+ function onicecandidate(e) {
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+ let candidate = e.candidate;
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+ if (candidate && candidate.candidate) {
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+ console.log("%c[Browser ICE candidate]", "background: violet; color: black", "| Type=", candidate.type, "| Protocol=", candidate.protocol, "| Address=", candidate.address, "| Port=", candidate.port, "|");
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+ self.onWebRtcCandidate(candidate);
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+ }
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+ };
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+
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+ function handleCreateOffer(pc) {
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+ pc.createOffer(self.sdpConstraints).then(function (offer) {
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+
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+ // Munging is where we modifying the sdp string to set parameters that are not exposed to the browser's WebRTC API
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+ mungeSDPOffer(offer);
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+
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+ // Set our munged SDP on the local peer connection so it is "set" and will be send across
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+ pc.setLocalDescription(offer);
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305
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+ if (self.onWebRtcOffer) {
|
|
|
306
|
+ self.onWebRtcOffer(offer);
|
|
|
307
|
+ }
|
|
|
308
|
+ },
|
|
|
309
|
+ function () { console.warn("Couldn't create offer") });
|
|
|
310
|
+ }
|
|
|
311
|
+
|
|
|
312
|
+ function mungeSDPOffer(offer) {
|
|
|
313
|
+
|
|
|
314
|
+ // turn off video-timing sdp sent from browser
|
|
|
315
|
+ //offer.sdp = offer.sdp.replace("http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "");
|
|
|
316
|
+
|
|
|
317
|
+ // this indicate we support stereo (Chrome needs this)
|
|
|
318
|
+ offer.sdp = offer.sdp.replace('useinbandfec=1', 'useinbandfec=1;stereo=1;sprop-maxcapturerate=48000');
|
|
|
319
|
+
|
|
|
320
|
+ }
|
|
|
321
|
+
|
|
|
322
|
+ function setupPeerConnection(pc) {
|
|
|
323
|
+ //Setup peerConnection events
|
|
|
324
|
+ pc.onsignalingstatechange = onsignalingstatechange;
|
|
|
325
|
+ pc.oniceconnectionstatechange = oniceconnectionstatechange;
|
|
|
326
|
+ pc.onicegatheringstatechange = onicegatheringstatechange;
|
|
|
327
|
+
|
|
|
328
|
+ pc.ontrack = handleOnTrack;
|
|
|
329
|
+ pc.onicecandidate = onicecandidate;
|
|
|
330
|
+ pc.ondatachannel = onDataChannel;
|
|
|
331
|
+ };
|
|
|
332
|
+
|
|
|
333
|
+ function generateAggregatedStatsFunction() {
|
|
|
334
|
+ if (!self.aggregatedStats)
|
|
|
335
|
+ self.aggregatedStats = {};
|
|
|
336
|
+
|
|
|
337
|
+ return function (stats) {
|
|
|
338
|
+ //console.log('Printing Stats');
|
|
|
339
|
+
|
|
|
340
|
+ let newStat = {};
|
|
|
341
|
+
|
|
|
342
|
+ stats.forEach(stat => {
|
|
|
343
|
+ // console.log(JSON.stringify(stat, undefined, 4));
|
|
|
344
|
+ if (stat.type == 'inbound-rtp'
|
|
|
345
|
+ && !stat.isRemote
|
|
|
346
|
+ && (stat.mediaType == 'video' || stat.id.toLowerCase().includes('video'))) {
|
|
|
347
|
+
|
|
|
348
|
+ newStat.timestamp = stat.timestamp;
|
|
|
349
|
+ newStat.bytesReceived = stat.bytesReceived;
|
|
|
350
|
+ newStat.framesDecoded = stat.framesDecoded;
|
|
|
351
|
+ newStat.packetsLost = stat.packetsLost;
|
|
|
352
|
+ newStat.bytesReceivedStart = self.aggregatedStats && self.aggregatedStats.bytesReceivedStart ? self.aggregatedStats.bytesReceivedStart : stat.bytesReceived;
|
|
|
353
|
+ newStat.framesDecodedStart = self.aggregatedStats && self.aggregatedStats.framesDecodedStart ? self.aggregatedStats.framesDecodedStart : stat.framesDecoded;
|
|
|
354
|
+ newStat.timestampStart = self.aggregatedStats && self.aggregatedStats.timestampStart ? self.aggregatedStats.timestampStart : stat.timestamp;
|
|
|
355
|
+
|
|
|
356
|
+ if (self.aggregatedStats && self.aggregatedStats.timestamp) {
|
|
|
357
|
+ if (self.aggregatedStats.bytesReceived) {
|
|
|
358
|
+ // bitrate = bits received since last time / number of ms since last time
|
|
|
359
|
+ //This is automatically in kbits (where k=1000) since time is in ms and stat we want is in seconds (so a '* 1000' then a '/ 1000' would negate each other)
|
|
|
360
|
+ newStat.bitrate = 8 * (newStat.bytesReceived - self.aggregatedStats.bytesReceived) / (newStat.timestamp - self.aggregatedStats.timestamp);
|
|
|
361
|
+ newStat.bitrate = Math.floor(newStat.bitrate);
|
|
|
362
|
+ newStat.lowBitrate = self.aggregatedStats.lowBitrate && self.aggregatedStats.lowBitrate < newStat.bitrate ? self.aggregatedStats.lowBitrate : newStat.bitrate
|
|
|
363
|
+ newStat.highBitrate = self.aggregatedStats.highBitrate && self.aggregatedStats.highBitrate > newStat.bitrate ? self.aggregatedStats.highBitrate : newStat.bitrate
|
|
|
364
|
+ }
|
|
|
365
|
+
|
|
|
366
|
+ if (self.aggregatedStats.bytesReceivedStart) {
|
|
|
367
|
+ newStat.avgBitrate = 8 * (newStat.bytesReceived - self.aggregatedStats.bytesReceivedStart) / (newStat.timestamp - self.aggregatedStats.timestampStart);
|
|
|
368
|
+ newStat.avgBitrate = Math.floor(newStat.avgBitrate);
|
|
|
369
|
+ }
|
|
|
370
|
+
|
|
|
371
|
+ if (self.aggregatedStats.framesDecoded) {
|
|
|
372
|
+ // framerate = frames decoded since last time / number of seconds since last time
|
|
|
373
|
+ newStat.framerate = (newStat.framesDecoded - self.aggregatedStats.framesDecoded) / ((newStat.timestamp - self.aggregatedStats.timestamp) / 1000);
|
|
|
374
|
+ newStat.framerate = Math.floor(newStat.framerate);
|
|
|
375
|
+ newStat.lowFramerate = self.aggregatedStats.lowFramerate && self.aggregatedStats.lowFramerate < newStat.framerate ? self.aggregatedStats.lowFramerate : newStat.framerate
|
|
|
376
|
+ newStat.highFramerate = self.aggregatedStats.highFramerate && self.aggregatedStats.highFramerate > newStat.framerate ? self.aggregatedStats.highFramerate : newStat.framerate
|
|
|
377
|
+ }
|
|
|
378
|
+
|
|
|
379
|
+ if (self.aggregatedStats.framesDecodedStart) {
|
|
|
380
|
+ newStat.avgframerate = (newStat.framesDecoded - self.aggregatedStats.framesDecodedStart) / ((newStat.timestamp - self.aggregatedStats.timestampStart) / 1000);
|
|
|
381
|
+ newStat.avgframerate = Math.floor(newStat.avgframerate);
|
|
|
382
|
+ }
|
|
|
383
|
+ }
|
|
|
384
|
+ }
|
|
|
385
|
+
|
|
|
386
|
+ //Read video track stats
|
|
|
387
|
+ if (stat.type == 'track' && (stat.trackIdentifier == 'video_label' || stat.kind == 'video')) {
|
|
|
388
|
+ newStat.framesDropped = stat.framesDropped;
|
|
|
389
|
+ newStat.framesReceived = stat.framesReceived;
|
|
|
390
|
+ newStat.framesDroppedPercentage = stat.framesDropped / stat.framesReceived * 100;
|
|
|
391
|
+ newStat.frameHeight = stat.frameHeight;
|
|
|
392
|
+ newStat.frameWidth = stat.frameWidth;
|
|
|
393
|
+ newStat.frameHeightStart = self.aggregatedStats && self.aggregatedStats.frameHeightStart ? self.aggregatedStats.frameHeightStart : stat.frameHeight;
|
|
|
394
|
+ newStat.frameWidthStart = self.aggregatedStats && self.aggregatedStats.frameWidthStart ? self.aggregatedStats.frameWidthStart : stat.frameWidth;
|
|
|
395
|
+ }
|
|
|
396
|
+
|
|
|
397
|
+ if (stat.type == 'candidate-pair' && stat.hasOwnProperty('currentRoundTripTime') && stat.currentRoundTripTime != 0) {
|
|
|
398
|
+ newStat.currentRoundTripTime = stat.currentRoundTripTime;
|
|
|
399
|
+ }
|
|
|
400
|
+ });
|
|
|
401
|
+
|
|
|
402
|
+
|
|
|
403
|
+ if (self.aggregatedStats.receiveToCompositeMs) {
|
|
|
404
|
+ newStat.receiveToCompositeMs = self.aggregatedStats.receiveToCompositeMs;
|
|
|
405
|
+ self.latencyTestTimings.SetFrameDisplayDeltaTime(self.aggregatedStats.receiveToCompositeMs);
|
|
|
406
|
+ }
|
|
|
407
|
+
|
|
|
408
|
+ self.aggregatedStats = newStat;
|
|
|
409
|
+
|
|
|
410
|
+ if (self.onAggregatedStats)
|
|
|
411
|
+ self.onAggregatedStats(newStat)
|
|
|
412
|
+ }
|
|
|
413
|
+ };
|
|
|
414
|
+
|
|
|
415
|
+ let setupTransceiversAsync = async function (pc) {
|
|
|
416
|
+
|
|
|
417
|
+ let hasTransceivers = pc.getTransceivers().length > 0;
|
|
|
418
|
+
|
|
|
419
|
+ // Setup a transceiver for getting UE video
|
|
|
420
|
+ pc.addTransceiver("video", { direction: "recvonly" });
|
|
|
421
|
+
|
|
|
422
|
+ // Setup a transceiver for sending mic audio to UE and receiving audio from UE
|
|
|
423
|
+ if (!self.useMic) {
|
|
|
424
|
+ pc.addTransceiver("audio", { direction: "recvonly" });
|
|
|
425
|
+ }
|
|
|
426
|
+ else {
|
|
|
427
|
+ let audioSendOptions = self.useMic ?
|
|
|
428
|
+ {
|
|
|
429
|
+ autoGainControl: false,
|
|
|
430
|
+ channelCount: 1,
|
|
|
431
|
+ echoCancellation: false,
|
|
|
432
|
+ latency: 0,
|
|
|
433
|
+ noiseSuppression: false,
|
|
|
434
|
+ sampleRate: 48000,
|
|
|
435
|
+ volume: 1.0
|
|
|
436
|
+ } : false;
|
|
|
437
|
+
|
|
|
438
|
+ // Note using mic on android chrome requires SSL or chrome://flags/ "unsafely-treat-insecure-origin-as-secure"
|
|
|
439
|
+ const stream = await navigator.mediaDevices.getUserMedia({ video: false, audio: audioSendOptions });
|
|
|
440
|
+ if (stream) {
|
|
|
441
|
+ if (hasTransceivers) {
|
|
|
442
|
+ for (let transceiver of pc.getTransceivers()) {
|
|
|
443
|
+ if (transceiver && transceiver.receiver && transceiver.receiver.track && transceiver.receiver.track.kind === "audio") {
|
|
|
444
|
+ for (const track of stream.getTracks()) {
|
|
|
445
|
+ if (track.kind && track.kind == "audio") {
|
|
|
446
|
+ transceiver.sender.replaceTrack(track);
|
|
|
447
|
+ transceiver.direction = "sendrecv";
|
|
|
448
|
+ }
|
|
|
449
|
+ }
|
|
|
450
|
+ }
|
|
|
451
|
+ }
|
|
|
452
|
+ }
|
|
|
453
|
+ else {
|
|
|
454
|
+ for (const track of stream.getTracks()) {
|
|
|
455
|
+ if (track.kind && track.kind == "audio") {
|
|
|
456
|
+ pc.addTransceiver(track, { direction: "sendrecv" });
|
|
|
457
|
+ }
|
|
|
458
|
+ }
|
|
|
459
|
+ }
|
|
|
460
|
+ }
|
|
|
461
|
+ else {
|
|
|
462
|
+ pc.addTransceiver("audio", { direction: "recvonly" });
|
|
|
463
|
+ }
|
|
|
464
|
+ }
|
|
|
465
|
+ };
|
|
|
466
|
+
|
|
|
467
|
+
|
|
|
468
|
+ //**********************
|
|
|
469
|
+ //Public functions
|
|
|
470
|
+ //**********************
|
|
|
471
|
+
|
|
|
472
|
+ this.setVideoEnabled = function (enabled) {
|
|
|
473
|
+ self.video.srcObject.getTracks().forEach(track => track.enabled = enabled);
|
|
|
474
|
+ }
|
|
|
475
|
+
|
|
|
476
|
+ this.startLatencyTest = function (onTestStarted) {
|
|
|
477
|
+ // Can't start latency test without a video element
|
|
|
478
|
+ if (!self.video) {
|
|
|
479
|
+ return;
|
|
|
480
|
+ }
|
|
|
481
|
+
|
|
|
482
|
+ self.latencyTestTimings.Reset();
|
|
|
483
|
+ self.latencyTestTimings.TestStartTimeMs = Date.now();
|
|
|
484
|
+ onTestStarted(self.latencyTestTimings.TestStartTimeMs);
|
|
|
485
|
+ }
|
|
|
486
|
+
|
|
|
487
|
+ //This is called when revceiving new ice candidates individually instead of part of the offer
|
|
|
488
|
+ this.handleCandidateFromServer = function (iceCandidate) {
|
|
|
489
|
+ let candidate = new RTCIceCandidate(iceCandidate);
|
|
|
490
|
+
|
|
|
491
|
+ console.log("%c[Unreal ICE candidate]", "background: pink; color: black", "| Type=", candidate.type, "| Protocol=", candidate.protocol, "| Address=", candidate.address, "| Port=", candidate.port, "|");
|
|
|
492
|
+
|
|
|
493
|
+ // if forcing TURN, reject any candidates not relay
|
|
|
494
|
+ if (self.forceTURN) {
|
|
|
495
|
+ // check if no relay address is found, if so, we are assuming it means no TURN server
|
|
|
496
|
+ if (candidate.candidate.indexOf("relay") < 0) {
|
|
|
497
|
+ console.warn("Dropping candidate because it was not TURN relay.", "| Type=", candidate.type, "| Protocol=", candidate.protocol, "| Address=", candidate.address, "| Port=", candidate.port, "|")
|
|
|
498
|
+ return;
|
|
|
499
|
+ }
|
|
|
500
|
+ }
|
|
|
501
|
+
|
|
|
502
|
+ self.pcClient.addIceCandidate(candidate).catch(function (e) {
|
|
|
503
|
+ console.error("Failed to add ICE candidate", e);
|
|
|
504
|
+ });
|
|
|
505
|
+ };
|
|
|
506
|
+
|
|
|
507
|
+ //Called externaly to create an offer for the server
|
|
|
508
|
+ this.createOffer = function () {
|
|
|
509
|
+ if (self.pcClient) {
|
|
|
510
|
+ console.log("Closing existing PeerConnection")
|
|
|
511
|
+ self.pcClient.close();
|
|
|
512
|
+ self.pcClient = null;
|
|
|
513
|
+ }
|
|
|
514
|
+ self.pcClient = new RTCPeerConnection(self.cfg);
|
|
|
515
|
+ setupPeerConnection(self.pcClient);
|
|
|
516
|
+
|
|
|
517
|
+ setupTransceiversAsync(self.pcClient).finally(function () {
|
|
|
518
|
+ self.dcClient = createDataChannel(self.pcClient, 'cirrus', self.dataChannelOptions);
|
|
|
519
|
+ handleCreateOffer(self.pcClient);
|
|
|
520
|
+ });
|
|
|
521
|
+
|
|
|
522
|
+ };
|
|
|
523
|
+
|
|
|
524
|
+ //Called externaly when an offer is received from the server
|
|
|
525
|
+ this.receiveOffer = function (offer) {
|
|
|
526
|
+ var offerDesc = new RTCSessionDescription(offer);
|
|
|
527
|
+
|
|
|
528
|
+ if (!self.pcClient) {
|
|
|
529
|
+ console.log("Creating a new PeerConnection in the browser.")
|
|
|
530
|
+ self.pcClient = new RTCPeerConnection(self.cfg);
|
|
|
531
|
+ setupPeerConnection(self.pcClient);
|
|
|
532
|
+
|
|
|
533
|
+ // Put things here that happen post transceiver setup
|
|
|
534
|
+ self.pcClient.setRemoteDescription(offerDesc)
|
|
|
535
|
+ .then(() => {
|
|
|
536
|
+ setupTransceiversAsync(self.pcClient).finally(function () {
|
|
|
537
|
+ self.pcClient.createAnswer()
|
|
|
538
|
+ .then(answer => self.pcClient.setLocalDescription(answer))
|
|
|
539
|
+ .then(() => {
|
|
|
540
|
+ if (self.onWebRtcAnswer) {
|
|
|
541
|
+ self.onWebRtcAnswer(self.pcClient.currentLocalDescription);
|
|
|
542
|
+ }
|
|
|
543
|
+ })
|
|
|
544
|
+ .then(() => {
|
|
|
545
|
+ let receivers = self.pcClient.getReceivers();
|
|
|
546
|
+ for (let receiver of receivers) {
|
|
|
547
|
+ receiver.playoutDelayHint = 0;
|
|
|
548
|
+ }
|
|
|
549
|
+ })
|
|
|
550
|
+ .catch((error) => console.error("createAnswer() failed:", error));
|
|
|
551
|
+ });
|
|
|
552
|
+ });
|
|
|
553
|
+ }
|
|
|
554
|
+ };
|
|
|
555
|
+
|
|
|
556
|
+ //Called externaly when an answer is received from the server
|
|
|
557
|
+ this.receiveAnswer = function (answer) {
|
|
|
558
|
+ var answerDesc = new RTCSessionDescription(answer);
|
|
|
559
|
+ self.pcClient.setRemoteDescription(answerDesc);
|
|
|
560
|
+
|
|
|
561
|
+ let receivers = self.pcClient.getReceivers();
|
|
|
562
|
+ for (let receiver of receivers) {
|
|
|
563
|
+ receiver.playoutDelayHint = 0;
|
|
|
564
|
+ }
|
|
|
565
|
+ };
|
|
|
566
|
+
|
|
|
567
|
+ this.close = function () {
|
|
|
568
|
+ if (self.pcClient) {
|
|
|
569
|
+ console.log("Closing existing peerClient")
|
|
|
570
|
+ self.pcClient.close();
|
|
|
571
|
+ self.pcClient = null;
|
|
|
572
|
+ }
|
|
|
573
|
+ if (self.aggregateStatsIntervalId)
|
|
|
574
|
+ clearInterval(self.aggregateStatsIntervalId);
|
|
|
575
|
+ }
|
|
|
576
|
+
|
|
|
577
|
+ //Sends data across the datachannel
|
|
|
578
|
+ this.send = function (data) {
|
|
|
579
|
+ if (self.dcClient && self.dcClient.readyState == 'open') {
|
|
|
580
|
+ //console.log('Sending data on dataconnection', self.dcClient)
|
|
|
581
|
+ self.dcClient.send(data);
|
|
|
582
|
+ }
|
|
|
583
|
+ };
|
|
|
584
|
+
|
|
|
585
|
+ this.getStats = function (onStats) {
|
|
|
586
|
+ if (self.pcClient && onStats) {
|
|
|
587
|
+ self.pcClient.getStats(null).then((stats) => {
|
|
|
588
|
+ onStats(stats);
|
|
|
589
|
+ });
|
|
|
590
|
+ }
|
|
|
591
|
+ }
|
|
|
592
|
+
|
|
|
593
|
+ this.aggregateStats = function (checkInterval) {
|
|
|
594
|
+ let calcAggregatedStats = generateAggregatedStatsFunction();
|
|
|
595
|
+ let printAggregatedStats = () => { self.getStats(calcAggregatedStats); }
|
|
|
596
|
+ self.aggregateStatsIntervalId = setInterval(printAggregatedStats, checkInterval);
|
|
|
597
|
+ }
|
|
|
598
|
+}
|
|
|
599
|
+
|
|
|
600
|
+export default webRtcPlayer
|
