webrtc: Require typescript@2.3 and rely on lib.dom.d.ts types (#20242)

types/peerjs/index.d.ts

@@ -2,6 +2,7 @@
 // Project: http://peerjs.com/
 // Definitions by: Toshiya Nakakura <https://github.com/nakakura>
 // Definitions: https://github.com/DefinitelyTyped/DefinitelyTyped
+// TypeScript Version: 2.3
 
 /// <reference types="webrtc" />
 

types/skyway/index.d.ts

@@ -2,6 +2,7 @@
 // Project: http://nttcom.github.io/skyway/
 // Definitions by: Toshiya Nakakura <https://github.com/nakakura>
 // Definitions: https://github.com/DefinitelyTyped/DefinitelyTyped
+// TypeScript Version: 2.3
 
 /// <reference types="webrtc" />
 

types/webrtc/MediaStream.d.ts

@@ -101,11 +101,6 @@ interface MediaStreamTrackEvent extends Event {
     //track: MediaStreamTrack;
 }
 
-declare enum MediaStreamTrackState {
-	"live",
-    "ended"
-}
-
 interface MediaStreamTrack extends EventTarget {
     //id: string;
     //kind: string;

types/webrtc/RTCPeerConnection.d.ts

@@ -27,74 +27,6 @@ interface RTCOfferOptions extends RTCOfferAnswerOptions {
 interface RTCAnswerOptions extends RTCOfferAnswerOptions {
 }
 
-// https://www.w3.org/TR/webrtc/#idl-def-rtcsdptype
-type RTCSdpType = 'offer' | 'pranswer' | 'answer' | 'rollback';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcsessiondescriptioninit
-interface RTCSessionDescriptionInit {
-    type: RTCSdpType;
-    sdp?: string; // If type is 'rollback', this member can be left undefined.
-}
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcsessiondescription
-interface RTCSessionDescription {
-    readonly type: RTCSdpType;
-    readonly sdp: string;
-}
-interface RTCSessionDescriptionStatic {
-    new(descriptionInitDict: RTCSessionDescriptionInit): RTCSessionDescription; // Deprecated
-}
-
-// https://www.w3.org/TR/webrtc/#dom-rtciceprotocol
-type RTCIceProtocol = 'udp' | 'tcp';
-
-// https://www.w3.org/TR/webrtc/#dom-rtcicecandidatetype
-type RTCIceCandidateType = 'host' | 'srflx' | 'prflx' | 'relay';
-
-// https://www.w3.org/TR/webrtc/#dom-rtcicetcpcandidatetype
-type RTCIceTcpCandidateType = 'active' | 'passive' | 'so';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicecandidateinit
-interface RTCIceCandidateInit {
-    candidate: string;
-    sdpMid?: string; // default = null
-    sdpMLineIndex?: number; // default = null
-}
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicecandidate
-interface RTCIceCandidate {
-    readonly candidate: string;
-    readonly sdpMid?: string;
-    readonly sdpMLineIndex?: number;
-    //readonly foundation: string;
-    //readonly priority: number;
-    //readonly ip: string;
-    //readonly protocol: RTCIceProtocol;
-    //readonly port: number;
-    //readonly type: RTCIceCandidateType;
-    //readonly tcpType?: RTCIceTcpCandidateType;
-    //readonly relatedAddress?: string;
-    //readonly relatedPort?: number;
-}
-interface RTCIceCandidateStatic {
-    new(candidateInitDict: RTCIceCandidateInit): RTCIceCandidate;
-}
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicecandidatepair
-interface RTCIceCandidatePair {
-    //local: RTCIceCandidate;
-    //remote: RTCIceCandidate;
-}
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcsignalingstate
-type RTCSignalingState = 'stable' | 'have-local-offer' | 'have-remote-offer' | 'have-local-pranswer' | 'have-remote-pranswer';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicegatheringstate
-type RTCIceGatheringState = 'new' | 'gathering' | 'complete';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtciceconnectionstate
-type RTCIceConnectionState = 'new' | 'checking' | 'connected' | 'completed' | 'failed' | 'disconnected' | 'closed';
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtcpeerconnectionstate
 type RTCPeerConnectionState = 'new' | 'connecting' | 'connected' | 'disconnected' | 'failed' | 'closed';
 
@@ -104,29 +36,12 @@ type RTCIceCredentialType = 'password' | 'token';
 // https://www.w3.org/TR/webrtc/#idl-def-rtciceserver
 interface RTCIceServer {
     //urls: string | string[];
-    username?: string;
-    credential?: string;
     credentialType?: RTCIceCredentialType; // default = 'password'
 }
 
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicetransportpolicy
-type RTCIceTransportPolicy = 'relay' | 'all';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcbundlepolicy
-type RTCBundlePolicy = 'balanced' | 'max-compat' | 'max-bundle';
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtcrtcpmuxpolicy
 type RTCRtcpMuxPolicy = 'negotiate' | 'require';
 
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicerole
-type RTCIceRole = 'controlling' | 'controlled';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicecomponent
-type RTCIceComponent = 'RTP' | 'RTCP';
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcicetransportstate
-type RTCIceTransportState = 'new' | 'checking' | 'connected' | 'completed' | 'failed' | 'disconnected' | 'closed';
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtciceparameters
 interface RTCIceParameters {
     //usernameFragment: string;
@@ -149,9 +64,6 @@ interface RTCIceTransport {
     onselectedcandidatepairchange: EventHandler;
 }
 
-// https://www.w3.org/TR/webrtc/#idl-def-rtcdtlstransportstate
-type RTCDtlsTransportState = 'new' | 'connecting' | 'connected' | 'closed' | 'failed';
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtcdtlstransport
 interface RTCDtlsTransport {
     readonly transport: RTCIceTransport;
@@ -201,7 +113,6 @@ interface RTCRtpEncodingParameters {
     //active: boolean;
     //priority: RTCPriorityType;
     //maxBitrate: number;
-    maxFramerate: number;
     rid: string;
     scaleResolutionDownBy?: number; // default = 1
 }
@@ -228,8 +139,6 @@ interface RTCRtpCodecParameters {
     sdpFmtpLine: string;
 }
 
-type RTCDegradationPreference = 'maintain-framerate' | 'maintain-resolution' | 'balanced';
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtcrtpparameters
 interface RTCRtpParameters {
     transactionId: string;
@@ -263,10 +172,6 @@ interface RTCRtpSender {
     getParameters(): RTCRtpParameters;
     replaceTrack(withTrack: MediaStreamTrack): Promise<void>;
 }
-interface RTCRtpSenderStatic {
-    new(): RTCRtpSender;
-    getCapabilities(kind: string): RTCRtpCapabilities;
-}
 
 // https://www.w3.org/TR/webrtc/#idl-def-rtcrtpreceiver
 interface RTCRtpReceiver {
@@ -276,10 +181,6 @@ interface RTCRtpReceiver {
     getParameters(): RTCRtpParameters;
     getContributingSources(): RTCRtpContributingSource[];
 }
-interface RTCRtpReceiverStatic {
-    new(): RTCRtpReceiver;
-    getCapabilities(kind: string): RTCRtcCapabilities;
-}
 
 // https://www.w3.org/TR/webrtc/#idl-def-rtcrtptransceiverdirection
 type RTCRtpTransceiverDirection = 'sendrecv' | 'sendonly' | 'recvonly' | 'inactive';
@@ -379,7 +280,6 @@ interface RTCTrackEvent extends Event {
 
 // https://www.w3.org/TR/webrtc/#h-rtcpeerconnectioniceevent
 interface RTCPeerConnectionIceEvent extends Event {
-    readonly candidate: RTCIceCandidate | null;
     readonly url: string;
 }
 
@@ -396,18 +296,6 @@ interface RTCDataChannelEvent {
     readonly channel: RTCDataChannel;
 }
 
-// https://www.w3.org/TR/webrtc/#idl-def-rtcsessiondescriptioncallback
-// Deprecated!
-type RTCSessionDescriptionCallback = (sdp: RTCSessionDescription) => void;
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcpeerconnectionerrorcallback
-// Deprecated!
-type RTCPeerConnectionErrorCallback = (error: DOMException) => void;
-
-// https://www.w3.org/TR/webrtc/#idl-def-rtcstatscallback
-// Deprecated!
-type RTCStatsCallback = (report: RTCStatsReport) => void;
-
 // https://www.w3.org/TR/webrtc/#idl-def-rtcpeerconnection
 interface RTCPeerConnection extends EventTarget {
     createOffer(options?: RTCOfferOptions): Promise<RTCSessionDescriptionInit>;
@@ -426,21 +314,12 @@ interface RTCPeerConnection extends EventTarget {
     addIceCandidate(candidate?: RTCIceCandidateInit | RTCIceCandidate): Promise<void>;
 
     readonly signalingState: RTCSignalingState;
-    readonly iceGatheringState: RTCIceGatheringState;
-    readonly iceConnectionState: RTCIceConnectionState;
-    readonly connectionState: RTCPeerConnectionState;
-    readonly canTrickleIceCandidates?: boolean | null;
 
     getConfiguration(): RTCConfiguration;
     setConfiguration(configuration: RTCConfiguration): void;
     close(): void;
 
-    onnegotiationneeded: EventHandler;
-    onicecandidate: (event: RTCPeerConnectionIceEvent) => void;
     onicecandidateerror: (event: RTCPeerConnectionIceErrorEvent) => void;
-    onsignalingstatechange: EventHandler;
-    oniceconnectionstatechange: EventHandler;
-    onicegatheringstatechange: EventHandler;
     onconnectionstatechange: EventHandler;
 
     // Extension: https://www.w3.org/TR/webrtc/#h-rtcpeerconnection-interface-extensions
@@ -455,7 +334,7 @@ interface RTCPeerConnection extends EventTarget {
     // Extension: https://www.w3.org/TR/webrtc/#h-rtcpeerconnection-interface-extensions-1
     readonly sctp: RTCSctpTransport | null;
     createDataChannel(label: string | null, dataChannelDict?: RTCDataChannelInit): RTCDataChannel;
-    ondatachannel: (event: 	RTCDataChannelEvent) => void;
+    ondatachannel: (event: RTCDataChannelEvent) => void;
 
     // Extension: https://www.w3.org/TR/webrtc/#h-rtcpeerconnection-interface-extensions-2
     getStats(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
@@ -488,15 +367,6 @@ interface RTCPeerConnectionStatic {
     generateCertificate(keygenAlgorithm: string): Promise<RTCCertificate>;
 }
 
-declare var RTCPeerConnection: RTCPeerConnectionStatic;
-declare var RTCSessionDescription: RTCSessionDescriptionStatic;
-declare var RTCIceCandidate: RTCIceCandidateStatic;
-//declare var RTCRtpSender: RTCRtpSenderStatic;
-//declare var RTCRtpReceiver: RTCRtpReceiverStatic;
 interface Window {
     RTCPeerConnection: RTCPeerConnectionStatic;
-    RTCSessionDescription: RTCSessionDescriptionStatic;
-    RTCIceCandidate: RTCIceCandidateStatic;
-    RTCRtpSender: RTCRtpSenderStatic;
-    RTCRtpReceiver: RTCRtpReceiverStatic;
 }

types/webrtc/index.d.ts

@@ -2,6 +2,7 @@
 // Project: https://webrtc.org/
 // Definitions by: Toshiya Nakakura <https://github.com/nakakura>
 // Definitions: https://github.com/DefinitelyTyped/DefinitelyTyped
+// TypeScript Version: 2.3
 
 /// <reference path="RTCPeerConnection.d.ts" />
 /// <reference path="MediaStream.d.ts" />

types/webrtc/test/MediaStream.ts

@@ -1,17 +1,17 @@
-var mediaStreamConstraints: MediaStreamConstraints = { audio: true, video: true };
+const mediaStreamConstraints: MediaStreamConstraints = { audio: true, video: true };
 
-var mediaTrackConstraintSet: MediaTrackConstraintSet = {};
-var mediaTrackConstraintArray: MediaTrackConstraintSet[] = [];
-var mediaTrackConstraints: MediaTrackConstraints = mediaTrackConstraintSet;
-var mediaTrackConstraints2: MediaTrackConstraints = { advanced: mediaTrackConstraintArray };
+const mediaTrackConstraintSet: MediaTrackConstraintSet = {};
+const mediaTrackConstraintArray: MediaTrackConstraintSet[] = [];
+const mediaTrackConstraints: MediaTrackConstraints = mediaTrackConstraintSet;
+const mediaTrackConstraints2: MediaTrackConstraints = { advanced: mediaTrackConstraintArray };
 
 navigator.getUserMedia(mediaStreamConstraints,
   stream => {
-    var track: MediaStreamTrack = stream.getTracks()[0];
+    const track: MediaStreamTrack = stream.getTracks()[0];
     console.log('label:' + track.label);
     console.log('ended:' + track.readyState);
-    track.onended = (event:Event) => console.log('Track ended');
-    var objectUrl = URL.createObjectURL(stream);
+    track.onended = (event: Event) => console.log('Track ended');
+    const objectUrl = URL.createObjectURL(stream);
   },
   error => {
     console.log('Error message: ' + error.message);
@@ -20,25 +20,24 @@ navigator.getUserMedia(mediaStreamConstraints,
 
 navigator.webkitGetUserMedia(mediaStreamConstraints,
   stream => {
-    var track: MediaStreamTrack = stream.getTracks()[0];
+    const track: MediaStreamTrack = stream.getTracks()[0];
     console.log('label:' + track.label);
     console.log('ended:' + track.readyState);
-    track.onended = (event:Event) => console.log('Track ended');
-    var objectUrl = URL.createObjectURL(stream);
+    track.onended = (event: Event) => console.log('Track ended');
+    const objectUrl = URL.createObjectURL(stream);
   },
   error => {
     console.log('Error message: ' + error.message);
     console.log('Error name: ' + error.name);
   });
 
-
 navigator.mozGetUserMedia(mediaStreamConstraints,
   stream => {
-    var track: MediaStreamTrack = stream.getTracks()[0];
+    const track: MediaStreamTrack = stream.getTracks()[0];
     console.log('label:' + track.label);
     console.log('ended:' + track.readyState);
-    track.onended = (event:Event) => console.log('Track ended');
-    var objectUrl = URL.createObjectURL(stream);
+    track.onended = (event: Event) => console.log('Track ended');
+    const objectUrl = URL.createObjectURL(stream);
   },
   error => {
     console.log('Error message: ' + error.message);

types/webrtc/test/RTCPeerConnection.ts

@@ -1,21 +1,21 @@
-let defaultIceServers: RTCIceServer[] = RTCPeerConnection.defaultIceServers;
+let defaultIceServers: RTCIceServer[] = window.RTCPeerConnection.defaultIceServers;
 if (defaultIceServers.length > 0) {
-    let urls = defaultIceServers[0].urls;
+    const urls = defaultIceServers[0].urls;
 }
 
 // Create a peer connection
 let ice1: RTCIceServer = {
-    'urls': 'stun:stun.l.google.com:19302',
-    'username': 'john',
-    'credential': '1234',
-    'credentialType': 'password',
+    urls: 'stun:stun.l.google.com:19302',
+    username: 'john',
+    credential: '1234',
+    credentialType: 'password',
 };
-let ice2: RTCIceServer = {'urls': ['stun:stunserver.org', 'stun:stun.example.com']};
-let pc: RTCPeerConnection = new RTCPeerConnection();
+let ice2: RTCIceServer = { urls: ['stun:stunserver.org', 'stun:stun.example.com'] };
+let pc: RTCPeerConnection = new RTCPeerConnection({});
 let pc2: RTCPeerConnection = new RTCPeerConnection({
     iceServers: [ice1, ice2],
 });
-RTCPeerConnection.generateCertificate("sha-256").then((cert: RTCCertificate) => {
+window.RTCPeerConnection.generateCertificate("sha-256").then((cert: RTCCertificate) => {
     new RTCPeerConnection({
         iceServers: [ice1],
         iceTransportPolicy: 'relay',
@@ -35,15 +35,15 @@ pc.setConfiguration(conf);
 pc2.close();
 
 // Offer/answer flow
-let offer: RTCSessionDescriptionInit;
-let answer: RTCSessionDescriptionInit;
 pc.createOffer({iceRestart: true})
-    .then((_offer: RTCSessionDescriptionInit) => offer = _offer);
-pc.setLocalDescription(offer);
-pc2.setRemoteDescription(offer);
-pc2.createAnswer().then((_answer: RTCSessionDescriptionInit) => answer = _answer);
-pc2.setLocalDescription(answer);
-pc.setRemoteDescription(answer);
+    .then((offer: RTCSessionDescriptionInit) => {
+    pc.setLocalDescription(offer);
+    pc2.setRemoteDescription(offer);
+    pc2.createAnswer().then((answer: RTCSessionDescriptionInit) => {
+        pc2.setLocalDescription(answer);
+        pc.setRemoteDescription(answer);
+    });
+});
 
 // Event handlers
 pc.onnegotiationneeded = ev => console.log(ev.type);

types/webrtc/tsconfig.json

@@ -7,7 +7,7 @@
         ],
         "noImplicitAny": true,
         "noImplicitThis": true,
-        "strictNullChecks": false,
+        "strictNullChecks": true,
         "baseUrl": "../",
         "typeRoots": [
             "../"
@@ -18,6 +18,8 @@
     },
     "files": [
         "index.d.ts",
+        "MediaStream.d.ts",
+        "RTCPeerConnection.d.ts",
         "test/MediaStream.ts",
         "test/RTCPeerConnection.ts"
     ]

types/webrtc/tslint.json

@@ -0,0 +1,13 @@
+{
+    "extends": "dtslint/dt.json",
+    "rules": {
+        // TODOs
+        "adjacent-overload-signatures": false,
+        "callable-types": false,
+        "comment-format": false,
+        "dt-header": false,
+        "no-empty-interface": false,
+        "no-useless-files": false,
+        "prefer-method-signature": false
+    }
+}