house_win / WindowsNoEditor /Samples /PixelStreaming /WebServers /SignallingWebServer /scripts /webRtcPlayer.js
| // Copyright Epic Games, Inc. All Rights Reserved. | |
| // universal module definition - read https://www.davidbcalhoun.com/2014/what-is-amd-commonjs-and-umd/ | |
| (function (root, factory) { | |
| if (typeof define === 'function' && define.amd) { | |
| // AMD. Register as an anonymous module. | |
| define(["./adapter"], factory); | |
| } else if (typeof exports === 'object') { | |
| // Node. Does not work with strict CommonJS, but | |
| // only CommonJS-like environments that support module.exports, | |
| // like Node. | |
| module.exports = factory(require("./adapter")); | |
| } else { | |
| // Browser globals (root is window) | |
| root.webRtcPlayer = factory(root.adapter); | |
| } | |
| }(this, function (adapter) { | |
| function webRtcPlayer(parOptions) { | |
| parOptions = typeof parOptions !== 'undefined' ? parOptions : {}; | |
| var self = this; | |
| //********************** | |
| //Config setup | |
| //********************** | |
| this.cfg = typeof parOptions.peerConnectionOptions !== 'undefined' ? parOptions.peerConnectionOptions : {}; | |
| this.cfg.sdpSemantics = 'unified-plan'; | |
| // this.cfg.rtcAudioJitterBufferMaxPackets = 10; | |
| // this.cfg.rtcAudioJitterBufferFastAccelerate = true; | |
| // this.cfg.rtcAudioJitterBufferMinDelayMs = 0; | |
| // If this is true in Chrome 89+ SDP is sent that is incompatible with UE Pixel Streaming 4.26 and below. | |
| // However 4.27 Pixel Streaming does not need this set to false as it supports `offerExtmapAllowMixed`. | |
| // tdlr; uncomment this line for older versions of Pixel Streaming that need Chrome 89+. | |
| this.cfg.offerExtmapAllowMixed = false; | |
| //********************** | |
| //Variables | |
| //********************** | |
| this.pcClient = null; | |
| this.dcClient = null; | |
| this.tnClient = null; | |
| this.sdpConstraints = { | |
| offerToReceiveAudio: 1, //Note: if you don't need audio you can get improved latency by turning this off. | |
| offerToReceiveVideo: 1, | |
| voiceActivityDetection: false | |
| }; | |
| // See https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit for values (this is needed for Firefox to be consistent with Chrome.) | |
| this.dataChannelOptions = {ordered: true}; | |
| // 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. | |
| this.startVideoMuted = typeof parOptions.startVideoMuted !== 'undefined' ? parOptions.startVideoMuted : false; | |
| this.autoPlayAudio = typeof parOptions.autoPlayAudio !== 'undefined' ? parOptions.autoPlayAudio : true; | |
| // To enable mic in browser use SSL/localhost and have ?useMic in the query string. | |
| const urlParams = new URLSearchParams(window.location.search); | |
| this.useMic = urlParams.has('useMic'); | |
| if(!this.useMic) | |
| { | |
| console.log("Microphone access is not enabled. Pass ?useMic in the url to enable it."); | |
| } | |
| // When ?useMic check for SSL or localhost | |
| let isLocalhostConnection = location.hostname === "localhost" || location.hostname === "127.0.0.1"; | |
| let isHttpsConnection = location.protocol === 'https:'; | |
| if(this.useMic && !isLocalhostConnection && !isHttpsConnection) | |
| { | |
| this.useMic = false; | |
| console.error("Microphone access in the browser will not work if you are not on HTTPS or localhost. Disabling mic access."); | |
| console.error("For testing you can enable HTTP microphone access Chrome by visiting chrome://flags/ and enabling 'unsafely-treat-insecure-origin-as-secure'"); | |
| } | |
| // Latency tester | |
| this.latencyTestTimings = | |
| { | |
| TestStartTimeMs: null, | |
| UEReceiptTimeMs: null, | |
| UEPreCaptureTimeMs: null, | |
| UEPostCaptureTimeMs: null, | |
| UEPreEncodeTimeMs: null, | |
| UEPostEncodeTimeMs: null, | |
| UETransmissionTimeMs: null, | |
| BrowserReceiptTimeMs: null, | |
| FrameDisplayDeltaTimeMs: null, | |
| Reset: function() | |
| { | |
| this.TestStartTimeMs = null; | |
| this.UEReceiptTimeMs = null; | |
| this.UEPreCaptureTimeMs = null; | |
| this.UEPostCaptureTimeMs = null; | |
| this.UEPreEncodeTimeMs = null; | |
| this.UEPostEncodeTimeMs = null; | |
| this.UETransmissionTimeMs = null; | |
| this.BrowserReceiptTimeMs = null; | |
| this.FrameDisplayDeltaTimeMs = null; | |
| }, | |
| SetUETimings: function(UETimings) | |
| { | |
| this.UEReceiptTimeMs = UETimings.ReceiptTimeMs; | |
| this.UEPreCaptureTimeMs = UETimings.PreCaptureTimeMs; | |
| this.UEPostCaptureTimeMs = UETimings.PostCaptureTimeMs; | |
| this.UEPreEncodeTimeMs = UETimings.PreEncodeTimeMs; | |
| this.UEPostEncodeTimeMs = UETimings.PostEncodeTimeMs; | |
| this.UETransmissionTimeMs = UETimings.TransmissionTimeMs; | |
| this.BrowserReceiptTimeMs = Date.now(); | |
| this.OnAllLatencyTimingsReady(this); | |
| }, | |
| SetFrameDisplayDeltaTime: function(DeltaTimeMs) | |
| { | |
| if(this.FrameDisplayDeltaTimeMs == null) | |
| { | |
| this.FrameDisplayDeltaTimeMs = Math.round(DeltaTimeMs); | |
| this.OnAllLatencyTimingsReady(this); | |
| } | |
| }, | |
| OnAllLatencyTimingsReady: function(Timings){} | |
| } | |
| //********************** | |
| //Functions | |
| //********************** | |
| //Create Video element and expose that as a parameter | |
| this.createWebRtcVideo = function() { | |
| var video = document.createElement('video'); | |
| video.id = "streamingVideo"; | |
| video.playsInline = true; | |
| video.disablepictureinpicture = true; | |
| video.muted = self.startVideoMuted;; | |
| video.addEventListener('loadedmetadata', function(e){ | |
| if(self.onVideoInitialised){ | |
| self.onVideoInitialised(); | |
| } | |
| }, true); | |
| // Check if request video frame callback is supported | |
| if ('requestVideoFrameCallback' in HTMLVideoElement.prototype) { | |
| // The API is supported! | |
| const onVideoFrameReady = (now, metadata) => { | |
| if(metadata.receiveTime && metadata.expectedDisplayTime) | |
| { | |
| const receiveToCompositeMs = metadata.presentationTime - metadata.receiveTime; | |
| self.aggregatedStats.receiveToCompositeMs = receiveToCompositeMs; | |
| } | |
| // Re-register the callback to be notified about the next frame. | |
| video.requestVideoFrameCallback(onVideoFrameReady); | |
| }; | |
| // Initially register the callback to be notified about the first frame. | |
| video.requestVideoFrameCallback(onVideoFrameReady); | |
| } | |
| return video; | |
| } | |
| this.video = this.createWebRtcVideo(); | |
| onsignalingstatechange = function(state) { | |
| console.info('signaling state change:', state) | |
| }; | |
| oniceconnectionstatechange = function(state) { | |
| console.info('ice connection state change:', state) | |
| }; | |
| onicegatheringstatechange = function(state) { | |
| console.info('ice gathering state change:', state) | |
| }; | |
| handleOnTrack = function(e) { | |
| console.log('handleOnTrack', e.streams); | |
| if (e.track) | |
| { | |
| console.log('Got track - ' + e.track.kind + ' id=' + e.track.id + ' readyState=' + e.track.readyState); | |
| } | |
| if(e.track.kind == "audio") | |
| { | |
| handleOnAudioTrack(e.streams[0]); | |
| return; | |
| } | |
| else(e.track.kind == "video" && self.video.srcObject !== e.streams[0]) | |
| { | |
| self.video.srcObject = e.streams[0]; | |
| console.log('Set video source from video track ontrack.'); | |
| return; | |
| } | |
| }; | |
| handleOnAudioTrack = function(audioMediaStream) | |
| { | |
| // do nothing the video has the same media stream as the audio track we have here (they are linked) | |
| if(self.video.srcObject == audioMediaStream) | |
| { | |
| return; | |
| } | |
| // video element has some other media stream that is not associated with this audio track | |
| else if(self.video.srcObject && self.video.srcObject !== audioMediaStream) | |
| { | |
| // create a new audio element | |
| let audioElem = document.createElement("Audio"); | |
| audioElem.srcObject = audioMediaStream; | |
| // there is no way to autoplay audio (even muted), so we defer audio until first click | |
| if(!self.autoPlayAudio) { | |
| let clickToPlayAudio = function() { | |
| audioElem.play(); | |
| self.video.removeEventListener("click", clickToPlayAudio); | |
| }; | |
| self.video.addEventListener("click", clickToPlayAudio); | |
| } | |
| // we assume the user has clicked somewhere on the page and autoplaying audio will work | |
| else { | |
| audioElem.play(); | |
| } | |
| console.log('Created new audio element to play seperate audio stream.'); | |
| } | |
| } | |
| setupDataChannel = function(pc, label, options) { | |
| try { | |
| let datachannel = pc.createDataChannel(label, options); | |
| console.log(`Created datachannel (${label})`) | |
| // Inform browser we would like binary data as an ArrayBuffer (FF chooses Blob by default!) | |
| datachannel.binaryType = "arraybuffer"; | |
| datachannel.onopen = function (e) { | |
| console.log(`data channel (${label}) connect`) | |
| if(self.onDataChannelConnected){ | |
| self.onDataChannelConnected(); | |
| } | |
| } | |
| datachannel.onclose = function (e) { | |
| console.log(`data channel (${label}) closed`) | |
| } | |
| datachannel.onmessage = function (e) { | |
| //console.log(`Got message (${label})`, e.data) | |
| if (self.onDataChannelMessage) | |
| self.onDataChannelMessage(e.data); | |
| } | |
| return datachannel; | |
| } catch (e) { | |
| console.warn('No data channel', e); | |
| return null; | |
| } | |
| } | |
| onicecandidate = function (e) { | |
| console.log('ICE candidate', e) | |
| if (e.candidate && e.candidate.candidate) { | |
| self.onWebRtcCandidate(e.candidate); | |
| } | |
| }; | |
| handleCreateOffer = function (pc) { | |
| pc.createOffer(self.sdpConstraints).then(function (offer) { | |
| // Munging is where we modifying the sdp string to set parameters that are not exposed to the browser's WebRTC API | |
| mungeSDPOffer(offer); | |
| // Set our munged SDP on the local peer connection so it is "set" and will be send across | |
| pc.setLocalDescription(offer); | |
| if (self.onWebRtcOffer) { | |
| self.onWebRtcOffer(offer); | |
| } | |
| }, | |
| function () { console.warn("Couldn't create offer") }); | |
| } | |
| mungeSDPOffer = function (offer) { | |
| // turn off video-timing sdp sent from browser | |
| //offer.sdp = offer.sdp.replace("http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", ""); | |
| // this indicate we support stereo (Chrome needs this) | |
| offer.sdp = offer.sdp.replace('useinbandfec=1', 'useinbandfec=1;stereo=1;sprop-maxcapturerate=48000'); | |
| } | |
| setupPeerConnection = function (pc) { | |
| if (pc.SetBitrate) | |
| console.log("Hurray! there's RTCPeerConnection.SetBitrate function"); | |
| //Setup peerConnection events | |
| pc.onsignalingstatechange = onsignalingstatechange; | |
| pc.oniceconnectionstatechange = oniceconnectionstatechange; | |
| pc.onicegatheringstatechange = onicegatheringstatechange; | |
| pc.ontrack = handleOnTrack; | |
| pc.onicecandidate = onicecandidate; | |
| }; | |
| generateAggregatedStatsFunction = function(){ | |
| if(!self.aggregatedStats) | |
| self.aggregatedStats = {}; | |
| return function(stats){ | |
| //console.log('Printing Stats'); | |
| let newStat = {}; | |
| stats.forEach(stat => { | |
| // console.log(JSON.stringify(stat, undefined, 4)); | |
| if (stat.type == 'inbound-rtp' | |
| && !stat.isRemote | |
| && (stat.mediaType == 'video' || stat.id.toLowerCase().includes('video'))) { | |
| newStat.timestamp = stat.timestamp; | |
| newStat.bytesReceived = stat.bytesReceived; | |
| newStat.framesDecoded = stat.framesDecoded; | |
| newStat.packetsLost = stat.packetsLost; | |
| newStat.bytesReceivedStart = self.aggregatedStats && self.aggregatedStats.bytesReceivedStart ? self.aggregatedStats.bytesReceivedStart : stat.bytesReceived; | |
| newStat.framesDecodedStart = self.aggregatedStats && self.aggregatedStats.framesDecodedStart ? self.aggregatedStats.framesDecodedStart : stat.framesDecoded; | |
| newStat.timestampStart = self.aggregatedStats && self.aggregatedStats.timestampStart ? self.aggregatedStats.timestampStart : stat.timestamp; | |
| if(self.aggregatedStats && self.aggregatedStats.timestamp){ | |
| if(self.aggregatedStats.bytesReceived){ | |
| // bitrate = bits received since last time / number of ms since last time | |
| //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) | |
| newStat.bitrate = 8 * (newStat.bytesReceived - self.aggregatedStats.bytesReceived) / (newStat.timestamp - self.aggregatedStats.timestamp); | |
| newStat.bitrate = Math.floor(newStat.bitrate); | |
| newStat.lowBitrate = self.aggregatedStats.lowBitrate && self.aggregatedStats.lowBitrate < newStat.bitrate ? self.aggregatedStats.lowBitrate : newStat.bitrate | |
| newStat.highBitrate = self.aggregatedStats.highBitrate && self.aggregatedStats.highBitrate > newStat.bitrate ? self.aggregatedStats.highBitrate : newStat.bitrate | |
| } | |
| if(self.aggregatedStats.bytesReceivedStart){ | |
| newStat.avgBitrate = 8 * (newStat.bytesReceived - self.aggregatedStats.bytesReceivedStart) / (newStat.timestamp - self.aggregatedStats.timestampStart); | |
| newStat.avgBitrate = Math.floor(newStat.avgBitrate); | |
| } | |
| if(self.aggregatedStats.framesDecoded){ | |
| // framerate = frames decoded since last time / number of seconds since last time | |
| newStat.framerate = (newStat.framesDecoded - self.aggregatedStats.framesDecoded) / ((newStat.timestamp - self.aggregatedStats.timestamp) / 1000); | |
| newStat.framerate = Math.floor(newStat.framerate); | |
| newStat.lowFramerate = self.aggregatedStats.lowFramerate && self.aggregatedStats.lowFramerate < newStat.framerate ? self.aggregatedStats.lowFramerate : newStat.framerate | |
| newStat.highFramerate = self.aggregatedStats.highFramerate && self.aggregatedStats.highFramerate > newStat.framerate ? self.aggregatedStats.highFramerate : newStat.framerate | |
| } | |
| if(self.aggregatedStats.framesDecodedStart){ | |
| newStat.avgframerate = (newStat.framesDecoded - self.aggregatedStats.framesDecodedStart) / ((newStat.timestamp - self.aggregatedStats.timestampStart) / 1000); | |
| newStat.avgframerate = Math.floor(newStat.avgframerate); | |
| } | |
| } | |
| } | |
| //Read video track stats | |
| if(stat.type == 'track' && (stat.trackIdentifier == 'video_label' || stat.kind == 'video')) { | |
| newStat.framesDropped = stat.framesDropped; | |
| newStat.framesReceived = stat.framesReceived; | |
| newStat.framesDroppedPercentage = stat.framesDropped / stat.framesReceived * 100; | |
| newStat.frameHeight = stat.frameHeight; | |
| newStat.frameWidth = stat.frameWidth; | |
| newStat.frameHeightStart = self.aggregatedStats && self.aggregatedStats.frameHeightStart ? self.aggregatedStats.frameHeightStart : stat.frameHeight; | |
| newStat.frameWidthStart = self.aggregatedStats && self.aggregatedStats.frameWidthStart ? self.aggregatedStats.frameWidthStart : stat.frameWidth; | |
| } | |
| if(stat.type =='candidate-pair' && stat.hasOwnProperty('currentRoundTripTime') && stat.currentRoundTripTime != 0){ | |
| newStat.currentRoundTripTime = stat.currentRoundTripTime; | |
| } | |
| }); | |
| if(self.aggregatedStats.receiveToCompositeMs) | |
| { | |
| newStat.receiveToCompositeMs = self.aggregatedStats.receiveToCompositeMs; | |
| self.latencyTestTimings.SetFrameDisplayDeltaTime(self.aggregatedStats.receiveToCompositeMs); | |
| } | |
| self.aggregatedStats = newStat; | |
| if(self.onAggregatedStats) | |
| self.onAggregatedStats(newStat) | |
| } | |
| }; | |
| setupTracksToSendAsync = async function(pc){ | |
| // Setup a transceiver for getting UE video | |
| pc.addTransceiver("video", { direction: "recvonly" }); | |
| // Setup a transceiver for sending mic audio to UE and receiving audio from UE | |
| if(!self.useMic) | |
| { | |
| pc.addTransceiver("audio", { direction: "recvonly" }); | |
| } | |
| else | |
| { | |
| let audioSendOptions = self.useMic ? | |
| { | |
| autoGainControl: false, | |
| channelCount: 1, | |
| echoCancellation: false, | |
| latency: 0, | |
| noiseSuppression: false, | |
| sampleRate: 16000, | |
| volume: 1.0 | |
| } : false; | |
| // Note using mic on android chrome requires SSL or chrome://flags/ "unsafely-treat-insecure-origin-as-secure" | |
| const stream = await navigator.mediaDevices.getUserMedia({video: false, audio: audioSendOptions}); | |
| if(stream) | |
| { | |
| for (const track of stream.getTracks()) { | |
| if(track.kind && track.kind == "audio") | |
| { | |
| pc.addTransceiver(track, { direction: "sendrecv" }); | |
| } | |
| } | |
| } | |
| else | |
| { | |
| pc.addTransceiver("audio", { direction: "recvonly" }); | |
| } | |
| } | |
| }; | |
| //********************** | |
| //Public functions | |
| //********************** | |
| this.setVideoEnabled = function(enabled) { | |
| self.video.srcObject.getTracks().forEach(track => track.enabled = enabled); | |
| } | |
| this.startLatencyTest = function(onTestStarted) { | |
| // Can't start latency test without a video element | |
| if(!self.video) | |
| { | |
| return; | |
| } | |
| self.latencyTestTimings.Reset(); | |
| self.latencyTestTimings.TestStartTimeMs = Date.now(); | |
| onTestStarted(self.latencyTestTimings.TestStartTimeMs); | |
| } | |
| //This is called when revceiving new ice candidates individually instead of part of the offer | |
| //This is currently not used but would be called externally from this class | |
| this.handleCandidateFromServer = function(iceCandidate) { | |
| console.log("ICE candidate: ", iceCandidate); | |
| let candidate = new RTCIceCandidate(iceCandidate); | |
| self.pcClient.addIceCandidate(candidate).then(_=>{ | |
| console.log('ICE candidate successfully added'); | |
| }); | |
| }; | |
| //Called externaly to create an offer for the server | |
| this.createOffer = function() { | |
| if(self.pcClient){ | |
| console.log("Closing existing PeerConnection") | |
| self.pcClient.close(); | |
| self.pcClient = null; | |
| } | |
| self.pcClient = new RTCPeerConnection(self.cfg); | |
| setupTracksToSendAsync(self.pcClient).finally(function() | |
| { | |
| setupPeerConnection(self.pcClient); | |
| self.dcClient = setupDataChannel(self.pcClient, 'cirrus', self.dataChannelOptions); | |
| handleCreateOffer(self.pcClient); | |
| }); | |
| }; | |
| //Called externaly when an answer is received from the server | |
| this.receiveAnswer = function(answer) { | |
| console.log('Received answer:'); | |
| console.log(answer); | |
| var answerDesc = new RTCSessionDescription(answer); | |
| self.pcClient.setRemoteDescription(answerDesc); | |
| let receivers = self.pcClient.getReceivers(); | |
| for(let receiver of receivers) | |
| { | |
| receiver.playoutDelayHint = 0; | |
| } | |
| }; | |
| this.close = function(){ | |
| if(self.pcClient){ | |
| console.log("Closing existing peerClient") | |
| self.pcClient.close(); | |
| self.pcClient = null; | |
| } | |
| if(self.aggregateStatsIntervalId) | |
| clearInterval(self.aggregateStatsIntervalId); | |
| } | |
| //Sends data across the datachannel | |
| this.send = function(data){ | |
| if(self.dcClient && self.dcClient.readyState == 'open'){ | |
| //console.log('Sending data on dataconnection', self.dcClient) | |
| self.dcClient.send(data); | |
| } | |
| }; | |
| this.getStats = function(onStats){ | |
| if(self.pcClient && onStats){ | |
| self.pcClient.getStats(null).then((stats) => { | |
| onStats(stats); | |
| }); | |
| } | |
| } | |
| this.aggregateStats = function(checkInterval){ | |
| let calcAggregatedStats = generateAggregatedStatsFunction(); | |
| let printAggregatedStats = () => { self.getStats(calcAggregatedStats); } | |
| self.aggregateStatsIntervalId = setInterval(printAggregatedStats, checkInterval); | |
| } | |
| }; | |
| return webRtcPlayer; | |
| })); | |