文件名称:
simplewebrtc.js webrtc网页视频开发帮助工具
开发工具:
文件大小: 15kb
下载次数: 0
上传时间: 2013-03-27
详细说明: 网页视频开发 webrtc ;(function () { var logger = { log: function (){}, warn: function (){}, error: function (){} }; // normalize environment var RTCPeerConnection = null, getUserMedia = null, attachMediaStream = null, reattachMediaStream = null, browser = null, webRTCSupport = true; if (navigator.mozGetUserMedia) { logger.log("This appears to be Firefox"); browser = "firefox"; // The RTCPeerConnection object. RTCPeerConnection = mozRTCPeerConnection; // The RTCSessionDescription object. RTCSessionDescription = mozRTCSessionDescription; // The RTCIceCandidate object. RTCIceCandidate = mozRTCIceCandidate; // Get UserMedia (only difference is the prefix). // Code from Adam Barth. getUserMedia = navigator.mozGetUserMedia.bind(navigator); // Attach a media stream to an element. attachMediaStream = function(element, stream) { element.mozSrcObject = stream; element.play(); }; reattachMediaStream = function(to, from) { to.mozSrcObject = from.mozSrcObject; to.play(); }; // Fake get{Video,Audio}Tracks MediaStream.prototype.getVideoTracks = function() { return []; }; MediaStream.prototype.getAudioTracks = function() { return []; }; } else if (navigator.webkitGetUserMedia) { browser = "chrome"; // The RTCPeerConnection object. RTCPeerConnection = webkitRTCPeerConnection; // Get UserMedia (only difference is the prefix). // Code from Adam Barth. getUserMedia = navigator.webkitGetUserMedia.bind(navigator); // Attach a media stream to an element. attachMediaStream = function(element, stream) { element.autoplay = true; element.src = webkitURL.createObjectURL(stream); }; reattachMediaStream = function(to, from) { to.src = from.src; }; // The representation of tracks in a stream is changed in M26. // Unify them for earlier Chrome versions in the coexisting period. if (!webkitMediaStream.prototype.getVideoTracks) { webkitMediaStream.prototype.getVideoTracks = function() { return this.videoTracks; }; webkitMediaStream.prototype.getAudioTracks = function() { return this.audioTracks; }; } // New syntax of getXXXStreams method in M26. if (!webkitRTCPeerConnection.prototype.getLocalStreams) { webkitRTCPeerConnection.prototype.getLocalStreams = function() { return this.localStreams; }; webkitRTCPeerConnection.prototype.getRemoteStreams = function() { return this.remoteStreams; }; } } else { webRTCSupport = false; throw new Error("Browser does not appear to be WebRTC-capable"); } // emitter that we use as a base function WildEmitter() { this.callbacks = {}; } // Listen on the given `event` with `fn`. Store a group name if present. WildEmitter.prototype.on = function (event, groupName, fn) { var hasGroup = (arguments.length === 3), group = hasGroup ? arguments[1] : undefined, func = hasGroup ? arguments[2] : arguments[1]; func._groupName = group; (this.callbacks[event] = this.callbacks[event] || []).push(func); return this; }; // Adds an `event` listener that will be invoked a single // time then automatically removed. WildEmitter.prototype.once = function (event, fn) { var self = this; function on() { self.off(event, on); fn.apply(this, arguments); } this.on(event, on); return this; }; // Unbinds an entire group WildEmitter.prototype.releaseGroup = function (groupName) { var item, i, len, handlers; for (item in this.callbacks) { handlers = this.callbacks[item]; for (i = 0, len = handlers.length; i < len; i++) { if (handlers[i]._groupName === groupName) { handlers.splice(i, 1); i--; len--; } } } return this; }; // Remove the given callback for `event` or all // registered callbacks. WildEmitter.prototype.off = function (event, fn) { var callbacks = this.callbacks[event], i; if (!callbacks) return this; // remove all handlers if (arguments.length === 1) { delete this.callbacks[event]; return this; } // remove specific handler i = callbacks.indexOf(fn); callbacks.splice(i, 1); return this; }; // Emit `event` with the given args. // also calls any `*` handlers WildEmitter.prototype.emit = function (event) { var args = [].slice.call(arguments, 1), callbacks = this.callbacks[event], specialCallbacks = this.getWildcardCallbacks(event), i, len, item; if (callbacks) { for (i = 0, len = callbacks.length; i < len; ++i) { callbacks[i].apply(this, args); } } if (specialCallbacks) { for (i = 0, len = specialCallbacks.length; i < len; ++i) { specialCallbacks[i].apply(this, [event].concat(args)); } } return this; }; // Helper for for finding special wildcard event handlers that match the event WildEmitter.prototype.getWildcardCallbacks = function (eventName) { var item, split, result = []; for (item in this.callbacks) { split = item.split('*'); if (item === '*' || (split.length === 2 && eventName.slice(0, split[1].length) === split[1])) { result = result.concat(this.callbacks[item]); } } return result; }; function WebRTC(opts) { var self = this, options = opts || {}, config = this.config = { url: 'http://signaling.simplewebrtc.com:8888', log: false, localVideoEl: '', remoteVideosEl: '', autoRequestMedia: false, // makes the entire PC config overridable peerConnectionConfig: { iceServers: browser == 'firefox' ? [{"url":"stun:124.124.124.2"}] : [{"url": "stun:stun.l.google.com:19302"}] }, peerConnectionContraints: { optional: [{"DtlsSrtpKeyAgreement": true}] }, media: { audio:true, video: { mandatory: {}, optional: [] } } }, item, connection; // check for support if (!webRTCSupport) { console.error('Your browser doesn\'t seem to support WebRTC'); } // set options for (item in options) { this.config[item] = options[item]; } // log if configured to if (this.config.log) logger = console; // where we'll store our peer connections this.pcs = {}; // our socket.io connection connection = this.connection = io.connect(this.config.url); connection.on('connect', function () { self.emit('ready', connection.socket.sessionid); self.sessionReady = true; self.testReadiness(); }); connection.on('message', function (message) { var existing = self.pcs[message.from]; if (existing) { existing.handleMessage(message); } else { // create the conversation object self.pcs[message.from] = new Conversation({ id: message.from, parent: self, initiator: false }); self.pcs[message.from].handleMessage(message); } }); connection.on('joined', function (room) { logger.log('got a joined', room); if (!self.pcs[room.id]) { self.startVideoCall(room.id); } }); connection.on('left', function (room) { var conv = self.pcs[room.id]; if (conv) conv.handleStreamRemoved(); }); WildEmitter.call(this); // log events this.on('*', function (event, val1, val2) { logger.log('event:', event, val1, val2); }); // auto request if configured if (this.config.autoRequestMedia) this.startLocalVideo(); } WebRTC.prototype = Object.create(WildEmitter.prototype, { constructor: { value: WebRTC } }); WebRTC.prototype.getEl = function (idOrEl) { if (typeof idOrEl == 'string') { return document.getElementById(idOrEl); } else { return idOrEl; } }; // this accepts either element ID or element // and either the video tag itself or a container // that will be used to put the video tag into. WebRTC.prototype.getLocalVideoContainer = function () { var el = this.getEl(this.config.localVideoEl); if (el && el.tagName === 'VIDEO') { return el; } else { var video = document.createElement('video'); el.appendChild(video); return video; } }; WebRTC.prototype.getRemoteVideoContainer = function () { return this.getEl(this.config.remoteVideosEl); }; WebRTC.prototype.startVideoCall = function (id) { this.pcs[id] = new Conversation({ id: id, parent: this, initiator: true }); this.pcs[id].start(); }; WebRTC.prototype.createRoom = function (name, cb) { if (arguments.length === 2) { this.connection.emit('create', name, cb); } else { this.connection.emit('create', name); } }; WebRTC.prototype.joinRoom = function (name) { this.connection.emit('join', name); this.roomName = name; }; WebRTC.prototype.leaveRoom = function () { if (this.roomName) { this.connection.emit('leave', this.roomName); for (var pc in this.pcs) { this.pcs[pc].end(); } } }; WebRTC.prototype.testReadiness = function () { var self = this; if (this.localStream && this.sessionReady) { // This timeout is a workaround for the strange no-audio bug // as described here: https://code.google.com/p/webrtc/issues/detail?id=1525 // remove timeout when this is fixed. setTimeout(function () { self.emit('readyToCall', self.connection.socket.sessionid); }, 1000); } }; WebRTC.prototype.startLocalVideo = function (element) { var self = this; getUserMedia(this.config.media, function (stream) { attachMediaStream(element || self.getLocalVideoContainer(), stream); self.localStream = stream; self.testReadiness(); }, function () { throw new Error('Failed to get access to local media.'); }); }; WebRTC.prototype.send = function (to, type, payload) { this.connection.emit('message', { to: to, type: type, payload: payload }); }; function Conversation(options) { var self = this; this.id = options.id; this.parent = options.parent; this.initiator = options.initiator; // Create an RTCPeerConnection via the polyfill (adapter.js). this.pc = new RTCPeerConnection(this.parent.config.peerConnectionConfig, this.parent.config.peerConnectionContraints); this.pc.onicecandidate = this.onIceCandidate.bind(this); this.pc.addStream(this.parent.localStream); this.pc.onaddstream = this.handleRemoteStreamAdded.bind(this); this.pc.onremovestream = this.handleStreamRemoved.bind(this); // for re-use this.mediaConstraints = { 'mandatory': { 'OfferToReceiveAudio':true, 'OfferToReceiveVideo':true } }; WildEmitter.call(this); // proxy events to parent this.on('*', function (name, value) { self.parent.emit(name, value, self); }); } Conversation.prototype = Object.create(WildEmitter.prototype, { constructor: { value: Conversation } }); Conversation.prototype.handleMessage = function (message) { if (message.type === 'offer') { logger.log('setting remote description'); this.pc.setRemoteDescription(new RTCSessionDescription(message.payload)); this.answer(); } else if (message.type === 'answer') { this.pc.setRemoteDescription(new RTCSessionDescription(message.payload)); } else if (message.type === 'candidate') { console.log('message.payload', message.payload); var candidate = new RTCIceCandidate({ sdpMLineIndex: message.payload.label, candidate: message.payload.candidate }); this.pc.addIceCandidate(candidate); } }; Conversation.prototype.send = function (type, payload) { this.parent.send(this.id, type, payload); }; Conversation.prototype.onIceCandidate = function (event) { if (this.closed) return; if (event.candidate) { this.send('candidate', { label: event.candidate.sdpMLineIndex, id: event.candidate.sdpMid, candidate: event.candidate.candidate }); } else { logger.log("End of candidates."); } }; Conversation.prototype.start = function () { var self = this; this.pc.createOffer(function (sessionDescription) { logger.log('setting local description'); self.pc.setLocalDescription(sessionDescription); logger.log('sending offer', sessionDescription); self.send('offer', sessionDescription); }, null, this.mediaConstraints); }; Conversation.prototype.end = function () { this.pc.close(); this.handleStreamRemoved(); }; Conversation.prototype.answer = function () { var self = this; logger.log('answer called'); this.pc.createAnswer(function (sessionDescription) { logger.log('setting local description'); self.pc.setLocalDescription(sessionDescription); logger.log('sending answer', sessionDescription); self.send('answer', sessionDescription); }, null, this.mediaConstraints); }; Conversation.prototype.handleRemoteStreamAdded = function (event) { var stream = this.stream = event.stream, el = document.createElement('video'), container = this.parent.getRemoteVideoContainer(); el.id = this.id; attachMediaStream(el, stream); if (container) container.appendChild(el); this.emit('videoAdded', el); }; Conversation.prototype.handleStreamRemoved = function () { var video = document.getElementById(this.id), container = this.parent.getRemoteVideoContainer(); if (video && container) container.removeChild(video); this.emit('videoRemoved', video); delete this.parent.pcs[this.id]; this.closed = true; }; // expose WebRTC window.WebRTC = WebRTC; }()); ...展开收缩
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