mirror of
https://github.com/zhigang1992/DefinitelyTyped.git
synced 2026-03-29 22:38:33 +08:00
Rewrite WebRTC RTCPeerConnection definitions (#12140)
* Rewrite all RTCPeerConnection definitions I used the specification IDL to rewrite all definitions. * RTCPeerConnection: Add legacy interface extensions
This commit is contained in:
Danilo Bargen
committed by
Masahiro Wakame
Masahiro Wakame
parent
ec7779fb1a
commit
513dc337f9
95
webrtc/MediaStream.d.ts
vendored
95
webrtc/MediaStream.d.ts
vendored
@@ -6,6 +6,11 @@
|
||||
// Taken from http://dev.w3.org/2011/webrtc/editor/getusermedia.html
|
||||
// version: W3C Editor's Draft 29 June 2015
|
||||
|
||||
interface ConstrainBooleanParameters {
|
||||
exact?: boolean;
|
||||
ideal?: boolean;
|
||||
}
|
||||
|
||||
interface NumberRange {
|
||||
max?: number;
|
||||
min?: number;
|
||||
@@ -21,6 +26,11 @@ interface ConstrainStringParameters {
|
||||
ideal?: string | string[];
|
||||
}
|
||||
|
||||
interface MediaStreamConstraints {
|
||||
video?: boolean | MediaTrackConstraints;
|
||||
audio?: boolean | MediaTrackConstraints;
|
||||
}
|
||||
|
||||
declare namespace W3C {
|
||||
type LongRange = NumberRange;
|
||||
type DoubleRange = NumberRange;
|
||||
@@ -31,16 +41,49 @@ declare namespace W3C {
|
||||
type ConstrainString = string | string[] | ConstrainStringParameters;
|
||||
}
|
||||
|
||||
interface MediaTrackConstraints extends MediaTrackConstraintSet {
|
||||
advanced?: MediaTrackConstraintSet[];
|
||||
}
|
||||
|
||||
interface MediaTrackConstraintSet {
|
||||
width?: W3C.ConstrainLong;
|
||||
height?: W3C.ConstrainLong;
|
||||
aspectRatio?: W3C.ConstrainDouble;
|
||||
frameRate?: W3C.ConstrainDouble;
|
||||
facingMode?: W3C.ConstrainString;
|
||||
volume?: W3C.ConstrainDouble;
|
||||
sampleRate?: W3C.ConstrainLong;
|
||||
sampleSize?: W3C.ConstrainLong;
|
||||
echoCancellation?: W3C.ConstrainBoolean;
|
||||
latency?: W3C.ConstrainDouble;
|
||||
deviceId?: W3C.ConstrainString;
|
||||
groupId?: W3C.ConstrainString;
|
||||
}
|
||||
|
||||
interface MediaTrackSupportedConstraints {
|
||||
width?: boolean;
|
||||
height?: boolean;
|
||||
aspectRatio?: boolean;
|
||||
frameRate?: boolean;
|
||||
facingMode?: boolean;
|
||||
volume?: boolean;
|
||||
sampleRate?: boolean;
|
||||
sampleSize?: boolean;
|
||||
echoCancellation?: boolean;
|
||||
latency?: boolean;
|
||||
deviceId?: boolean;
|
||||
groupId?: boolean;
|
||||
}
|
||||
|
||||
interface MediaStream extends EventTarget {
|
||||
//id: string;
|
||||
//active: boolean;
|
||||
|
||||
//onactive: EventListener;
|
||||
//oninactive: EventListener;
|
||||
//onaddtrack: (event: MediaStreamTrackEvent) => any;
|
||||
//onremovetrack: (event: MediaStreamTrackEvent) => any;
|
||||
|
||||
clone(): MediaStream;
|
||||
stop(): void;
|
||||
|
||||
@@ -54,12 +97,29 @@ interface MediaStream extends EventTarget {
|
||||
removeTrack(track: MediaStreamTrack): void;
|
||||
}
|
||||
|
||||
interface MediaStreamTrackEvent extends Event {
|
||||
//track: MediaStreamTrack;
|
||||
}
|
||||
|
||||
declare enum MediaStreamTrackState {
|
||||
"live",
|
||||
"ended"
|
||||
}
|
||||
|
||||
interface MediaStreamTrack extends EventTarget {
|
||||
//id: string;
|
||||
//kind: string;
|
||||
//label: string;
|
||||
enabled: boolean;
|
||||
//muted: boolean;
|
||||
//remote: boolean;
|
||||
//readyState: MediaStreamTrackState;
|
||||
|
||||
//onmute: EventListener;
|
||||
//onunmute: EventListener;
|
||||
//onended: EventListener;
|
||||
//onoverconstrained: EventListener;
|
||||
|
||||
clone(): MediaStreamTrack;
|
||||
|
||||
stop(): void;
|
||||
@@ -71,11 +131,39 @@ interface MediaStreamTrack extends EventTarget {
|
||||
}
|
||||
|
||||
interface MediaTrackCapabilities {
|
||||
//width: number | W3C.LongRange;
|
||||
//height: number | W3C.LongRange;
|
||||
//aspectRatio: number | W3C.DoubleRange;
|
||||
//frameRate: number | W3C.DoubleRange;
|
||||
//facingMode: string;
|
||||
//volume: number | W3C.DoubleRange;
|
||||
//sampleRate: number | W3C.LongRange;
|
||||
//sampleSize: number | W3C.LongRange;
|
||||
//echoCancellation: boolean[];
|
||||
latency: number | W3C.DoubleRange;
|
||||
//deviceId: string;
|
||||
//groupId: string;
|
||||
}
|
||||
|
||||
interface MediaTrackSettings {
|
||||
//width: number;
|
||||
//height: number;
|
||||
//aspectRatio: number;
|
||||
//frameRate: number;
|
||||
//facingMode: string;
|
||||
//volume: number;
|
||||
//sampleRate: number;
|
||||
//sampleSize: number;
|
||||
//echoCancellation: boolean;
|
||||
latency: number;
|
||||
//deviceId: string;
|
||||
//groupId: string;
|
||||
}
|
||||
|
||||
interface MediaStreamError {
|
||||
//name: string;
|
||||
//message: string;
|
||||
//constraintName: string;
|
||||
}
|
||||
|
||||
interface NavigatorGetUserMedia {
|
||||
@@ -105,3 +193,10 @@ interface MediaDevices {
|
||||
getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>;
|
||||
enumerateDevices(): Promise<MediaDeviceInfo[]>;
|
||||
}
|
||||
|
||||
interface MediaDeviceInfo {
|
||||
//label: string;
|
||||
//deviceId: string;
|
||||
//kind: string;
|
||||
//groupId: string;
|
||||
}
|
||||
|
||||
@@ -1,114 +1,91 @@
|
||||
/// <reference path="MediaStream.d.ts" />
|
||||
/// <reference path="RTCPeerConnection.d.ts" />
|
||||
|
||||
let defaultIceServers: RTCIceServer[] = RTCPeerConnection.defaultIceServers;
|
||||
if (defaultIceServers.length > 0) {
|
||||
let urls = defaultIceServers[0].urls;
|
||||
}
|
||||
|
||||
|
||||
let voidpromise: Promise<void>;
|
||||
|
||||
var minimalConfig: RTCConfiguration = {};
|
||||
var config: RTCConfiguration = {
|
||||
iceServers: [
|
||||
{
|
||||
// Single url
|
||||
urls: "stun.l.google.com:19302"
|
||||
},
|
||||
{
|
||||
// List of urls and credentials
|
||||
urls: ["another-stun.example.com"],
|
||||
username: "dude",
|
||||
credential: "pass",
|
||||
credentialType: "token"
|
||||
},
|
||||
],
|
||||
iceTransportPolicy: "relay",
|
||||
bundlePolicy: "max-compat",
|
||||
rtcpMuxPolicy: "negotiate",
|
||||
peerIdentity: "dude",
|
||||
certificates: [{ expires: 1337 }],
|
||||
iceCandidatePoolSize: 5
|
||||
// Create a peer connection
|
||||
let ice1: RTCIceServer = {
|
||||
'urls': 'stun:stun.l.google.com:19302',
|
||||
'username': 'john',
|
||||
'credential': '1234',
|
||||
'credentialType': 'password',
|
||||
};
|
||||
var constraints: RTCMediaConstraints =
|
||||
{ mandatory: { OfferToReceiveAudio: true, OfferToReceiveVideo: true } };
|
||||
|
||||
var peerConnection: RTCPeerConnection =
|
||||
new RTCPeerConnection(config, constraints);
|
||||
|
||||
navigator.getUserMedia({ audio: true, video: true },
|
||||
stream => {
|
||||
peerConnection.addStream(stream);
|
||||
},
|
||||
error => {
|
||||
console.log('Error message: ' + error.message);
|
||||
console.log('Error name: ' + error.name);
|
||||
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) => {
|
||||
new RTCPeerConnection({
|
||||
iceServers: [ice1],
|
||||
iceTransportPolicy: 'relay',
|
||||
bundlePolicy: 'max-compat',
|
||||
rtcpMuxPolicy: 'negotiate',
|
||||
peerIdentity: 'dude',
|
||||
certificates: [cert],
|
||||
iceCandidatePoolSize: 5,
|
||||
});
|
||||
|
||||
peerConnection.onaddstream = ev => console.log(ev.type);
|
||||
peerConnection.ondatachannel = ev => console.log(ev.channel);
|
||||
peerConnection.oniceconnectionstatechange = ev => console.log(ev.type);
|
||||
peerConnection.onnegotiationneeded = ev => console.log(ev.type);
|
||||
peerConnection.onopen = ev => console.log(ev.type);
|
||||
peerConnection.onicecandidate = ev => console.log(ev.type);
|
||||
peerConnection.onremovestream = ev => console.log(ev.type);
|
||||
peerConnection.onstatechange = ev => console.log(ev.type);
|
||||
|
||||
peerConnection.createOffer();
|
||||
let offer2: Promise<RTCSessionDescription> = peerConnection.createOffer({
|
||||
voiceActivityDetection: true,
|
||||
iceRestart: false
|
||||
});
|
||||
|
||||
var type: string = RTCSdpType[RTCSdpType.offer];
|
||||
var offer: RTCSessionDescriptionInit = { type: type, sdp: "some sdp" };
|
||||
var sessionDescription = new RTCSessionDescription(offer);
|
||||
// Get/set the configuration
|
||||
let conf: RTCConfiguration = pc2.getConfiguration();
|
||||
pc.setConfiguration(conf);
|
||||
|
||||
peerConnection.setRemoteDescription(sessionDescription).then(
|
||||
() => peerConnection.createAnswer(),
|
||||
error => console.log('Error setting remote description: ' + error + "; offer.sdp=" + offer.sdp)
|
||||
);
|
||||
// Close peer connection
|
||||
pc2.close();
|
||||
|
||||
var webkitSessionDescription = new webkitRTCSessionDescription(offer);
|
||||
// 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);
|
||||
|
||||
// New syntax
|
||||
voidpromise = peerConnection.setLocalDescription(webkitSessionDescription);
|
||||
// Event handlers
|
||||
pc.onnegotiationneeded = ev => console.log(ev.type);
|
||||
pc.onicecandidate = ev => console.log(ev.candidate);
|
||||
pc.onicecandidateerror = ev => console.log(ev.errorText);
|
||||
pc.onsignalingstatechange = ev => console.log(ev.type);
|
||||
pc.oniceconnectionstatechange = ev => console.log(ev.type);
|
||||
pc.onicegatheringstatechange = ev => console.log(ev.type);
|
||||
pc.onconnectionstatechange = ev => console.log(ev.type);
|
||||
pc.ontrack = ev => console.log(ev.receiver);
|
||||
pc.ondatachannel = ev => console.log(ev.channel);
|
||||
|
||||
// Legacy syntax
|
||||
peerConnection.setRemoteDescription(webkitSessionDescription, () => {
|
||||
peerConnection.createAnswer(
|
||||
answer => {
|
||||
peerConnection.setLocalDescription(answer,
|
||||
() => console.log('Set local description'),
|
||||
error => console.log(
|
||||
"Error setting local description from created answer: " + error +
|
||||
"; answer.sdp=" + answer.sdp));
|
||||
},
|
||||
error => console.log("Error creating answer: " + error));
|
||||
},
|
||||
error => console.log('Error setting remote description: ' + error +
|
||||
"; offer.sdp=" + offer.sdp));
|
||||
|
||||
var mozSessionDescription = new mozRTCSessionDescription(offer);
|
||||
|
||||
peerConnection.setRemoteDescription(mozSessionDescription);
|
||||
|
||||
var wkPeerConnection: webkitRTCPeerConnection =
|
||||
new webkitRTCPeerConnection(config, constraints);
|
||||
|
||||
let candidate: RTCIceCandidate = { 'candidate': 'foobar' };
|
||||
voidpromise = peerConnection.addIceCandidate(candidate);
|
||||
|
||||
var mediaTrackConstraintSet: MediaTrackConstraintSet = {};
|
||||
var mediaTrackConstraints: MediaTrackConstraints = mediaTrackConstraintSet;
|
||||
|
||||
wkPeerConnection.getStats(null);
|
||||
|
||||
let mediaStreamTrack: MediaStreamTrack = {
|
||||
enabled: true, id:
|
||||
'id', kind: 'kind', label: 'label', muted: true, onended: () => { }, onmute: () => { },
|
||||
onoverconstrained: () => { }, onunmute: () => { }, readonly: true, readyState: 'string',
|
||||
remote: true, applyConstraints: (): Promise<void> => { return new Promise<void>(() => { }) },
|
||||
clone: ():MediaStreamTrack => { return this;},
|
||||
getCapabilities: ():MediaTrackCapabilities => { return {latency:0};},
|
||||
getConstraints: ():MediaTrackConstraints => { return {}},
|
||||
getSettings: ():MediaTrackSettings => { return { latency: 0}},
|
||||
stop: () => {}, addEventListener: () => {}, dispatchEvent: (evt:Event):boolean => { return false;},
|
||||
removeEventListener: () => {}
|
||||
};
|
||||
wkPeerConnection.getStats(mediaStreamTrack);
|
||||
// Legacy interface extensions
|
||||
pc.createOffer(
|
||||
(sdp: RTCSessionDescription) => console.log(sdp.sdp),
|
||||
(error: DOMException) => console.log(error.message),
|
||||
{iceRestart: true}
|
||||
).then(() => console.log('createOffer complete'));
|
||||
pc.setLocalDescription(
|
||||
{type: 'offer', sdp: 'foobar'},
|
||||
() => console.log('local description set'),
|
||||
(error: DOMException) => console.log(error.message)
|
||||
).then(() => console.log('setLocalDescription complete'));
|
||||
pc.createAnswer(
|
||||
(sdp: RTCSessionDescription) => console.log(sdp.sdp),
|
||||
(error: DOMException) => console.log(error.message)
|
||||
).then(() => console.log('createAnswer complete'));
|
||||
pc.setRemoteDescription(
|
||||
{type: 'answer', sdp: 'foobar'},
|
||||
() => console.log('remote description set'),
|
||||
(error: DOMException) => console.log(error.message)
|
||||
).then(() => console.log('setRemoteDescription complete'));
|
||||
pc.addIceCandidate(
|
||||
{candidate: 'candidate', sdpMid: 'foo', sdpMLineIndex: 1},
|
||||
() => console.log('candidate added'),
|
||||
(error: DOMException) => console.log(error.message)
|
||||
).then(() => console.log('addIceCandidate complete'));
|
||||
pc.getStats(
|
||||
null,
|
||||
(report: RTCStatsReport) => console.log('got report'),
|
||||
(error: DOMException) => console.log(error.message)
|
||||
).then(() => console.log('getStats complete'));
|
||||
|
||||
807
webrtc/RTCPeerConnection.d.ts
vendored
807
webrtc/RTCPeerConnection.d.ts
vendored
@@ -1,405 +1,502 @@
|
||||
// Type definitions for WebRTC
|
||||
// Project: http://dev.w3.org/2011/webrtc/
|
||||
// Definitions by: Ken Smith <https://github.com/smithkl42/>
|
||||
// Type definitions for WebRTC 2016-09-13
|
||||
// Project: https://www.w3.org/TR/webrtc/
|
||||
// Definitions by: Danilo Bargen <https://github.com/dbrgn/>
|
||||
// Definitions: https://github.com/DefinitelyTyped/DefinitelyTyped
|
||||
//
|
||||
// W3 Spec: https://www.w3.org/TR/webrtc/#idl-def-RTCIceServer
|
||||
// W3 Spec: https://www.w3.org/TR/webrtc/
|
||||
//
|
||||
// Note: Commented out definitions clash with definitions in lib.es6.d.ts. I
|
||||
// still kept them in here though, as sometimes they're more specific than the
|
||||
// ES6 library ones.
|
||||
|
||||
/// <reference path="MediaStream.d.ts" />
|
||||
/// <reference path='MediaStream.d.ts' />
|
||||
|
||||
// TODO(1): Get Typescript to have string-enum types as WebRtc is full of string
|
||||
// enums.
|
||||
// https://typescript.codeplex.com/discussions/549207
|
||||
type EventHandler = (event: Event) => void;
|
||||
|
||||
// TODO(2): get Typescript to have union types as WebRtc uses them.
|
||||
// https://typescript.codeplex.com/workitem/1364
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-RTCIceTransportPolicy
|
||||
type RTCIceTransportPolicy = 'public' | '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-RTCCertificate
|
||||
interface RTCCertificate {
|
||||
expires: number;
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcofferansweroptions
|
||||
interface RTCOfferAnswerOptions {
|
||||
voiceActivityDetection?: boolean; // default = true
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-RTCConfiguration
|
||||
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#RTCConfiguration_dictionary
|
||||
interface RTCConfiguration {
|
||||
iceServers ?: RTCIceServer[]; // optional according to mozilla docs
|
||||
iceTransportPolicy ?: RTCIceTransportPolicy; // default = 'all'
|
||||
bundlePolicy ?: RTCBundlePolicy; // default = 'balanced'
|
||||
rtcpMuxPolicy ?: RTCRtcpMuxPolicy; // default = 'require'
|
||||
peerIdentity ?: string; // default = null
|
||||
certificates ?: RTCCertificate[]; // default is auto-generated
|
||||
iceCandidatePoolSize ?: number; // default = 0
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcofferoptions
|
||||
interface RTCOfferOptions extends RTCOfferAnswerOptions {
|
||||
iceRestart?: boolean; // default = false
|
||||
}
|
||||
|
||||
declare var RTCConfiguration: {
|
||||
prototype: RTCConfiguration;
|
||||
new (): RTCConfiguration;
|
||||
};
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcansweroptions
|
||||
interface RTCAnswerOptions extends RTCOfferAnswerOptions {
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-RTCIceCredentialType
|
||||
// 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';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcicecredentialtype
|
||||
type RTCIceCredentialType = 'password' | 'token';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-RTCIceServer
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtciceserver
|
||||
interface RTCIceServer {
|
||||
urls?: any;
|
||||
username?: string;
|
||||
credential?: string;
|
||||
credentialType?: RTCIceCredentialType; // default = 'password'
|
||||
}
|
||||
declare var RTCIceServer: {
|
||||
prototype: RTCIceServer;
|
||||
new (): RTCIceServer;
|
||||
};
|
||||
|
||||
// moz (Firefox) specific prefixes.
|
||||
interface mozRTCPeerConnection extends RTCPeerConnection {
|
||||
}
|
||||
declare var mozRTCPeerConnection: {
|
||||
prototype: mozRTCPeerConnection;
|
||||
new (settings?: RTCConfiguration,
|
||||
constraints?:RTCMediaConstraints): mozRTCPeerConnection;
|
||||
};
|
||||
// webkit (Chrome) specific prefixes.
|
||||
interface webkitRTCPeerConnection extends RTCPeerConnection {
|
||||
}
|
||||
declare var webkitRTCPeerConnection: {
|
||||
prototype: webkitRTCPeerConnection;
|
||||
new (settings?: RTCConfiguration,
|
||||
constraints?:RTCMediaConstraints): webkitRTCPeerConnection;
|
||||
};
|
||||
|
||||
// For Chrome, look at the code here:
|
||||
// https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc&sq=package:chromium&dr=C&l=63
|
||||
interface RTCOptionalMediaConstraint {
|
||||
// When true, will use DTLS/SCTP data channels
|
||||
DtlsSrtpKeyAgreement?: boolean;
|
||||
// When true will use Rtp-based data channels (depreicated)
|
||||
RtpDataChannels?: boolean;
|
||||
//urls: string | string[];
|
||||
username?: string;
|
||||
credential?: string;
|
||||
credentialType?: RTCIceCredentialType; // default = 'password'
|
||||
}
|
||||
|
||||
// ks 12/20/12 - There's more here that doesn't seem to be documented very well yet.
|
||||
// http://www.w3.org/TR/2013/WD-webrtc-20130910/
|
||||
interface RTCMediaConstraints {
|
||||
mandatory?: RTCMediaOfferConstraints;
|
||||
optional?: RTCOptionalMediaConstraint[]
|
||||
// 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;
|
||||
//password: string;
|
||||
}
|
||||
|
||||
interface RTCMediaOfferConstraints {
|
||||
OfferToReceiveAudio: boolean;
|
||||
OfferToReceiveVideo: boolean;
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcicetransport
|
||||
interface RTCIceTransport {
|
||||
//readonly role: RTCIceRole;
|
||||
//readonly component: RTCIceComponent;
|
||||
//readonly state: RTCIceTransportState;
|
||||
readonly gatheringState: RTCIceGatheringState;
|
||||
getLocalCandidates(): RTCIceCandidate[];
|
||||
getRemoteCandidates(): RTCIceCandidate[];
|
||||
getSelectedCandidatePair(): RTCIceCandidatePair | null;
|
||||
getLocalParameters(): RTCIceParameters | null;
|
||||
getRemoteParameters(): RTCIceParameters | null;
|
||||
onstatechange: EventHandler;
|
||||
ongatheringstatechange: EventHandler;
|
||||
onselectedcandidatepairchange: EventHandler;
|
||||
}
|
||||
|
||||
interface RTCSessionDescriptionInit {
|
||||
type: string; // RTCSdpType; See TODO(1)
|
||||
sdp: string;
|
||||
// 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;
|
||||
//readonly state: RTCDtlsTransportState;
|
||||
getRemoteCertificates(): ArrayBuffer[];
|
||||
onstatechange: EventHandler;
|
||||
}
|
||||
|
||||
interface RTCSessionDescription {
|
||||
type?: string; // RTCSdpType; See TODO(1)
|
||||
sdp?: string;
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodeccapability
|
||||
interface RTCRtpCodecCapability {
|
||||
mimeType: string;
|
||||
}
|
||||
declare var RTCSessionDescription: {
|
||||
prototype: RTCSessionDescription;
|
||||
new (descriptionInitDict?: RTCSessionDescriptionInit): RTCSessionDescription;
|
||||
// TODO: Add serializer.
|
||||
// See: http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSdpType)
|
||||
};
|
||||
|
||||
interface webkitRTCSessionDescription extends RTCSessionDescription{
|
||||
type?: string;
|
||||
sdp?: string;
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpheaderextensioncapability
|
||||
interface RTCRtpHeaderExtensionCapability {
|
||||
uri: string;
|
||||
}
|
||||
declare var webkitRTCSessionDescription: {
|
||||
prototype: webkitRTCSessionDescription;
|
||||
new (descriptionInitDict?: RTCSessionDescriptionInit): webkitRTCSessionDescription;
|
||||
};
|
||||
|
||||
interface mozRTCSessionDescription extends RTCSessionDescription{
|
||||
type?: string;
|
||||
sdp?: string;
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcapabilities
|
||||
interface RTCRtpCapabilities {
|
||||
//codecs: RTCRtpCodecCapability[];
|
||||
//headerExtensions: RTCRtpHeaderExtensionCapability[];
|
||||
}
|
||||
declare var mozRTCSessionDescription: {
|
||||
prototype: mozRTCSessionDescription;
|
||||
new (descriptionInitDict?: RTCSessionDescriptionInit): mozRTCSessionDescription;
|
||||
};
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtprtxparameters
|
||||
interface RTCRtpRtxParameters {
|
||||
//ssrc: number;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpfecparameters
|
||||
interface RTCRtpFecParameters {
|
||||
//ssrc: number;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcdtxstatus
|
||||
type RTCDtxStatus = 'disabled' | 'enabled';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcprioritytype
|
||||
type RTCPriorityType = 'very-low' | 'low' | 'medium' | 'high';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpencodingparameters
|
||||
interface RTCRtpEncodingParameters {
|
||||
//ssrc: number;
|
||||
//rtx: RTCRtpRtxParameters;
|
||||
//fec: RTCRtpFecParameters;
|
||||
dtx: RTCDtxStatus;
|
||||
//active: boolean;
|
||||
//priority: RTCPriorityType;
|
||||
//maxBitrate: number;
|
||||
maxFramerate: number;
|
||||
rid: string;
|
||||
scaleResolutionDownBy?: number; // default = 1
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpheaderextensionparameters
|
||||
interface RTCRtpHeaderExtensionParameters {
|
||||
//uri: string;
|
||||
//id: number;
|
||||
encrypted: boolean;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtcpparameters
|
||||
interface RTCRtcpParameters {
|
||||
//cname: string;
|
||||
//reducedSize: boolean;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodecparameters
|
||||
interface RTCRtpCodecParameters {
|
||||
//payloadType: number;
|
||||
mimeType: string;
|
||||
//clockRate: number;
|
||||
channels?: number; // default = 1
|
||||
sdpFmtpLine: string;
|
||||
}
|
||||
|
||||
type RTCDegradationPreference = 'maintain-framerate' | 'maintain-resolution' | 'balanced';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpparameters
|
||||
interface RTCRtpParameters {
|
||||
transactionId: string;
|
||||
//encodings: RTCRtpEncodingParameters[];
|
||||
//headerExtensions: RTCRtpHeaderExtensionParameters[];
|
||||
//rtcp: RTCRtcpParameters;
|
||||
//codecs: RTCRtpCodecParameters[];
|
||||
degradationPreference?: RTCDegradationPreference; // default = 'balanced'
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#dom-rtcrtpcontributingsource
|
||||
interface RTCRtpContributingSource {
|
||||
//readonly timestamp: number;
|
||||
readonly source: number;
|
||||
//readonly audioLevel: number | null;
|
||||
readonly voiceActivityFlag: boolean | null;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcapabilities
|
||||
interface RTCRtcCapabilities {
|
||||
codecs: RTCRtpCodecCapability[];
|
||||
headerExtensions: RTCRtpHeaderExtensionCapability[];
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#dom-rtcrtpsender
|
||||
interface RTCRtpSender {
|
||||
//readonly track?: MediaStreamTrack;
|
||||
//readonly transport?: RTCDtlsTransport;
|
||||
//readonly rtcpTransport?: RTCDtlsTransport;
|
||||
setParameters(parameters?: RTCRtpParameters): Promise<void>;
|
||||
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 {
|
||||
//readonly track?: MediaStreamTrack;
|
||||
//readonly transport?: RTCDtlsTransport;
|
||||
//readonly rtcpTransport?: RTCDtlsTransport;
|
||||
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';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtptransceiver
|
||||
interface RTCRtpTransceiver {
|
||||
readonly mid: string | null;
|
||||
readonly sender: RTCRtpSender;
|
||||
readonly receiver: RTCRtpReceiver;
|
||||
readonly stopped: boolean;
|
||||
readonly direction: RTCRtpTransceiverDirection;
|
||||
setDirection(direction: RTCRtpTransceiverDirection): void;
|
||||
stop(): void;
|
||||
setCodecPreferences(codecs: RTCRtpCodecCapability[]): void;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcrtptransceiverinit
|
||||
interface RTCRtpTransceiverInit {
|
||||
direction?: RTCRtpTransceiverDirection; // default = 'sendrecv'
|
||||
streams: MediaStream[];
|
||||
sendEncodings: RTCRtpEncodingParameters[];
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#dom-rtccertificate
|
||||
interface RTCCertificate {
|
||||
readonly expires: number;
|
||||
getAlgorithm(): string;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcconfiguration
|
||||
interface RTCConfiguration {
|
||||
iceServers?: RTCIceServer[];
|
||||
iceTransportPolicy?: RTCIceTransportPolicy; // default = 'all'
|
||||
bundlePolicy?: RTCBundlePolicy; // default = 'balanced'
|
||||
rtcpMuxPolicy?: RTCRtcpMuxPolicy; // default = 'require'
|
||||
peerIdentity?: string; // default = null
|
||||
certificates?: RTCCertificate[];
|
||||
iceCandidatePoolSize?: number; // default = 0
|
||||
}
|
||||
|
||||
// Compatibility for older definitions on DefinitelyTyped.
|
||||
type RTCPeerConnectionConfig = RTCConfiguration;
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcsctptransport
|
||||
interface RTCSctpTransport {
|
||||
readonly transport: RTCDtlsTransport;
|
||||
readonly maxMessageSize: number;
|
||||
}
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
|
||||
interface RTCDataChannelInit {
|
||||
ordered ?: boolean; // messages must be sent in-order.
|
||||
maxPacketLifeTime ?: number; // unsigned short
|
||||
maxRetransmits ?: number; // unsigned short
|
||||
protocol ?: string; // default = ''
|
||||
negotiated ?: boolean; // default = false;
|
||||
id ?: number; // unsigned short
|
||||
ordered?: boolean; // default = true
|
||||
maxPacketLifeTime?: number;
|
||||
maxRetransmits?: number;
|
||||
protocol?: string; // default = ''
|
||||
negotiated?: boolean; // default = false
|
||||
id?: number;
|
||||
}
|
||||
|
||||
// TODO(1)
|
||||
declare enum RTCSdpType {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcsdptype
|
||||
'offer',
|
||||
'pranswer',
|
||||
'answer'
|
||||
}
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelstate
|
||||
type RTCDataChannelState = 'connecting' | 'open' | 'closing' | 'closed';
|
||||
|
||||
interface RTCMessageEvent {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#event-datachannel-message
|
||||
// At present, this can be an: ArrayBuffer, a string, or a Blob.
|
||||
// See TODO(2)
|
||||
data: any;
|
||||
}
|
||||
|
||||
// TODO(1)
|
||||
declare enum RTCDataChannelState {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannelState
|
||||
'connecting',
|
||||
'open',
|
||||
'closing',
|
||||
'closed'
|
||||
}
|
||||
// https://www.w3.org/TR/websockets/#dom-websocket-binarytype
|
||||
type RTCBinaryType = 'blob' | 'arraybuffer';
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannel
|
||||
interface RTCDataChannel extends EventTarget {
|
||||
label: string;
|
||||
reliable: boolean;
|
||||
readyState: string; // RTCDataChannelState; see TODO(1)
|
||||
bufferedAmount: number;
|
||||
binaryType: string;
|
||||
readonly label: string;
|
||||
readonly ordered: boolean;
|
||||
readonly maxPacketLifeTime: number | null;
|
||||
readonly maxRetransmits: number | null;
|
||||
readonly protocol: string;
|
||||
readonly negotiated: boolean;
|
||||
readonly id: number;
|
||||
readonly readyState: RTCDataChannelState;
|
||||
readonly bufferedAmount: number;
|
||||
bufferedAmountLowThreshold: number;
|
||||
binaryType: RTCBinaryType;
|
||||
|
||||
onopen: (event: Event) => void;
|
||||
onerror: (event: Event) => void;
|
||||
onclose: (event: Event) => void;
|
||||
onmessage: (event: RTCMessageEvent) => void;
|
||||
close(): void;
|
||||
send(data: string | Blob | ArrayBuffer | ArrayBufferView): void;
|
||||
|
||||
close(): void;
|
||||
|
||||
send(data: string): void ;
|
||||
send(data: ArrayBuffer): void;
|
||||
send(data: ArrayBufferView): void;
|
||||
send(data: Blob): void;
|
||||
}
|
||||
declare var RTCDataChannel: {
|
||||
prototype: RTCDataChannel;
|
||||
new (): RTCDataChannel;
|
||||
};
|
||||
|
||||
// https://www.w3.org/TR/webrtc/#rtcdatachannelevent
|
||||
interface RTCDataChannelEvent extends Event {
|
||||
channel: RTCDataChannel;
|
||||
}
|
||||
declare var RTCDataChannelEvent: {
|
||||
prototype: RTCDataChannelEvent;
|
||||
new (eventInitDict: RTCDataChannelEventInit): RTCDataChannelEvent;
|
||||
};
|
||||
|
||||
interface RTCIceCandidateEvent extends Event {
|
||||
candidate: RTCIceCandidate;
|
||||
onopen: EventHandler;
|
||||
onmessage: (event: MessageEvent) => void;
|
||||
onbufferedamountlow: EventHandler;
|
||||
onerror: (event: ErrorEvent) => void;
|
||||
onclose: EventHandler;
|
||||
}
|
||||
|
||||
interface RTCMediaStreamEvent extends Event {
|
||||
stream: MediaStream;
|
||||
// https://www.w3.org/TR/webrtc/#h-rtctrackevent
|
||||
interface RTCTrackEvent extends Event {
|
||||
readonly receiver: RTCRtpReceiver;
|
||||
readonly track: MediaStreamTrack;
|
||||
readonly streams: MediaStream[];
|
||||
readonly transceiver: RTCRtpTransceiver;
|
||||
}
|
||||
|
||||
interface EventInit {
|
||||
// https://www.w3.org/TR/webrtc/#h-rtcpeerconnectioniceevent
|
||||
interface RTCPeerConnectionIceEvent extends Event {
|
||||
readonly candidate: RTCIceCandidate | null;
|
||||
readonly url: string;
|
||||
}
|
||||
|
||||
interface RTCDataChannelEventInit extends EventInit {
|
||||
channel: RTCDataChannel;
|
||||
// https://www.w3.org/TR/webrtc/#h-rtcpeerconnectioniceerrorevent
|
||||
interface RTCPeerConnectionIceErrorEvent extends Event {
|
||||
readonly hostCandidate: string;
|
||||
readonly url: string;
|
||||
readonly errorCode: number;
|
||||
readonly errorText: string;
|
||||
}
|
||||
|
||||
interface RTCVoidCallback {
|
||||
(): void;
|
||||
}
|
||||
interface RTCSessionDescriptionCallback {
|
||||
(sdp: RTCSessionDescription): void;
|
||||
}
|
||||
interface RTCPeerConnectionErrorCallback {
|
||||
(errorInformation: DOMError): void;
|
||||
// https://www.w3.org/TR/webrtc/#h-rtcdatachannelevent
|
||||
interface RTCDataChannelEvent {
|
||||
readonly channel: RTCDataChannel;
|
||||
}
|
||||
|
||||
// TODO(1)
|
||||
declare enum RTCIceGatheringState {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcicegatheringstate-enum
|
||||
'new',
|
||||
'gathering',
|
||||
'complete'
|
||||
// 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>;
|
||||
createAnswer(options?: RTCAnswerOptions): Promise<RTCSessionDescriptionInit>;
|
||||
|
||||
setLocalDescription(description: RTCSessionDescriptionInit): Promise<void>;
|
||||
readonly localDescription: RTCSessionDescription | null;
|
||||
readonly currentLocalDescription: RTCSessionDescription | null;
|
||||
readonly pendingLocalDescription: RTCSessionDescription | null;
|
||||
|
||||
setRemoteDescription(description: RTCSessionDescriptionInit): Promise<void>;
|
||||
readonly remoteDescription: RTCSessionDescription | null;
|
||||
readonly currentRemoteDescription: RTCSessionDescription | null;
|
||||
readonly pendingRemoteDescription: RTCSessionDescription | null;
|
||||
|
||||
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
|
||||
getSenders(): RTCRtpSender[];
|
||||
getReceivers(): RTCRtpReceiver[];
|
||||
getTransceivers(): RTCRtpTransceiver[];
|
||||
addTrack(track: MediaStreamTrack, ...streams: MediaStream[]): RTCRtpSender;
|
||||
removeTrack(sender: RTCRtpSender): void;
|
||||
addTransceiver(trackOrKind: MediaStreamTrack | string, init?: RTCRtpTransceiverInit): RTCRtpTransceiver;
|
||||
ontrack: (event: RTCTrackEvent) => void;
|
||||
|
||||
// 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;
|
||||
|
||||
// Extension: https://www.w3.org/TR/webrtc/#h-rtcpeerconnection-interface-extensions-2
|
||||
getStats(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
|
||||
|
||||
// Extension: https://www.w3.org/TR/webrtc/#legacy-interface-extensions
|
||||
// Deprecated!
|
||||
createOffer(successCallback: RTCSessionDescriptionCallback,
|
||||
failureCallback: RTCPeerConnectionErrorCallback,
|
||||
options?: RTCOfferOptions): Promise<void>;
|
||||
setLocalDescription(description: RTCSessionDescriptionInit,
|
||||
successCallback: () => void,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): Promise<void>;
|
||||
createAnswer(successCallback: RTCSessionDescriptionCallback,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): Promise<void>;
|
||||
setRemoteDescription(description: RTCSessionDescriptionInit,
|
||||
successCallback: () => void,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): Promise<void>;
|
||||
addIceCandidate(candidate: RTCIceCandidateInit | RTCIceCandidate,
|
||||
successCallback: () => void,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): Promise<void>;
|
||||
getStats(selector: MediaStreamTrack | null,
|
||||
successCallback: RTCStatsCallback,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): Promise<void>;
|
||||
}
|
||||
interface RTCPeerConnectionStatic {
|
||||
new(configuration?: RTCConfiguration): RTCPeerConnection;
|
||||
readonly defaultIceServers: RTCIceServer[];
|
||||
|
||||
// Extension: https://www.w3.org/TR/webrtc/#sec.cert-mgmt
|
||||
generateCertificate(keygenAlgorithm: string): Promise<RTCCertificate>;
|
||||
}
|
||||
|
||||
// TODO(1)
|
||||
declare enum RTCIceConnectionState {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCIceConnectionState
|
||||
'new',
|
||||
'checking',
|
||||
'connected',
|
||||
'completed',
|
||||
'failed',
|
||||
'disconnected',
|
||||
'closed'
|
||||
}
|
||||
|
||||
// TODO(1)
|
||||
declare enum RTCSignalingState {
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSignalingState
|
||||
'stable',
|
||||
'have-local-offer',
|
||||
'have-remote-offer',
|
||||
'have-local-pranswer',
|
||||
'have-remote-pranswer',
|
||||
'closed'
|
||||
}
|
||||
|
||||
// This is based on the current implementation of WebRtc in Chrome; the spec is
|
||||
// a little unclear on this.
|
||||
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCStatsReport
|
||||
interface RTCStatsReport {
|
||||
stat(id: string): string;
|
||||
}
|
||||
|
||||
interface RTCStatsCallback {
|
||||
(report: RTCStatsReport): void;
|
||||
}
|
||||
|
||||
interface RTCOfferAnswerOptions {
|
||||
voiceActivityDetection ?: boolean; // default = true
|
||||
}
|
||||
|
||||
interface RTCOfferOptions extends RTCOfferAnswerOptions {
|
||||
iceRestart ?: boolean; // default = false
|
||||
}
|
||||
|
||||
interface RTCAnswerOptions extends RTCOfferAnswerOptions { }
|
||||
|
||||
interface RTCPeerConnection {
|
||||
|
||||
createOffer(options?: RTCOfferOptions): Promise<RTCSessionDescription>;
|
||||
createOffer(successCallback: RTCSessionDescriptionCallback,
|
||||
failureCallback?: RTCPeerConnectionErrorCallback,
|
||||
constraints?: RTCMediaConstraints): void; // Deprecated
|
||||
createAnswer(options?: RTCAnswerOptions): Promise<RTCSessionDescription>;
|
||||
createAnswer(successCallback: RTCSessionDescriptionCallback,
|
||||
failureCallback?: RTCPeerConnectionErrorCallback,
|
||||
constraints?: RTCMediaConstraints): void; // Deprecated
|
||||
setLocalDescription(description: RTCSessionDescription | RTCSessionDescriptionInit): Promise<void>;
|
||||
setLocalDescription(description: RTCSessionDescription,
|
||||
successCallback?: RTCVoidCallback,
|
||||
failureCallback?: RTCPeerConnectionErrorCallback): void; // Deprecated
|
||||
setRemoteDescription(description: RTCSessionDescription | RTCSessionDescriptionInit): Promise<void>;
|
||||
setRemoteDescription(description: RTCSessionDescription,
|
||||
successCallback?: RTCVoidCallback,
|
||||
failureCallback?: RTCPeerConnectionErrorCallback): void;
|
||||
localDescription: RTCSessionDescription;
|
||||
remoteDescription: RTCSessionDescription;
|
||||
signalingState: string; // RTCSignalingState; see TODO(1)
|
||||
updateIce(configuration?: RTCConfiguration,
|
||||
constraints?: RTCMediaConstraints): void;
|
||||
addIceCandidate(candidate: RTCIceCandidate): Promise<void>;
|
||||
addIceCandidate(candidate:RTCIceCandidate,
|
||||
successCallback:() => void,
|
||||
failureCallback:RTCPeerConnectionErrorCallback): void;
|
||||
iceGatheringState: string; // RTCIceGatheringState; see TODO(1)
|
||||
iceConnectionState: string; // RTCIceConnectionState; see TODO(1)
|
||||
getLocalStreams(): MediaStream[];
|
||||
getRemoteStreams(): MediaStream[];
|
||||
createDataChannel(label?: string,
|
||||
dataChannelDict?: RTCDataChannelInit): RTCDataChannel;
|
||||
ondatachannel: (event: RTCDataChannelEvent) => void;
|
||||
addStream(stream: MediaStream, constraints?: RTCMediaConstraints): void;
|
||||
removeStream(stream: MediaStream): void;
|
||||
close(): void;
|
||||
onnegotiationneeded: (event: Event) => void;
|
||||
onconnecting: (event: Event) => void;
|
||||
onopen: (event: Event) => void;
|
||||
onaddstream: (event: RTCMediaStreamEvent) => void;
|
||||
onremovestream: (event: RTCMediaStreamEvent) => void;
|
||||
onstatechange: (event: Event) => void;
|
||||
oniceconnectionstatechange: (event: Event) => void;
|
||||
onicecandidate: (event: RTCIceCandidateEvent) => void;
|
||||
onidentityresult: (event: Event) => void;
|
||||
onsignalingstatechange: (event: Event) => void;
|
||||
getStats(selector: MediaStreamTrack | null): Promise<RTCStatsReport>;
|
||||
getStats(selector: MediaStreamTrack | null,
|
||||
successCallback: RTCStatsCallback,
|
||||
failureCallback: RTCPeerConnectionErrorCallback): void;
|
||||
}
|
||||
declare var RTCPeerConnection: {
|
||||
prototype: RTCPeerConnection;
|
||||
new (configuration: RTCConfiguration,
|
||||
constraints?: RTCMediaConstraints): RTCPeerConnection;
|
||||
};
|
||||
|
||||
interface RTCIceCandidate {
|
||||
candidate: string;
|
||||
sdpMid?: string;
|
||||
sdpMLineIndex?: number;
|
||||
}
|
||||
declare var RTCIceCandidate: {
|
||||
prototype: RTCIceCandidate;
|
||||
new (candidateInitDict?: RTCIceCandidate): RTCIceCandidate;
|
||||
};
|
||||
|
||||
interface webkitRTCIceCandidate extends RTCIceCandidate {
|
||||
candidate: string;
|
||||
sdpMid?: string;
|
||||
sdpMLineIndex?: number;
|
||||
}
|
||||
declare var webkitRTCIceCandidate: {
|
||||
prototype: webkitRTCIceCandidate;
|
||||
new (candidateInitDict?: webkitRTCIceCandidate): webkitRTCIceCandidate;
|
||||
};
|
||||
|
||||
interface mozRTCIceCandidate extends RTCIceCandidate {
|
||||
candidate: string;
|
||||
sdpMid?: string;
|
||||
sdpMLineIndex?: number;
|
||||
}
|
||||
declare var mozRTCIceCandidate: {
|
||||
prototype: mozRTCIceCandidate;
|
||||
new (candidateInitDict?: mozRTCIceCandidate): mozRTCIceCandidate;
|
||||
};
|
||||
|
||||
interface RTCIceCandidateInit {
|
||||
candidate: string;
|
||||
sdpMid?: string;
|
||||
sdpMLineIndex?: number;
|
||||
}
|
||||
declare var RTCIceCandidateInit:{
|
||||
prototype: RTCIceCandidateInit;
|
||||
new (): RTCIceCandidateInit;
|
||||
};
|
||||
|
||||
interface PeerConnectionIceEvent {
|
||||
peer: RTCPeerConnection;
|
||||
candidate: RTCIceCandidate;
|
||||
}
|
||||
declare var PeerConnectionIceEvent: {
|
||||
prototype: PeerConnectionIceEvent;
|
||||
new (): PeerConnectionIceEvent;
|
||||
};
|
||||
|
||||
interface RTCPeerConnectionConfig {
|
||||
iceServers: RTCIceServer[];
|
||||
}
|
||||
declare var RTCPeerConnectionConfig: {
|
||||
prototype: RTCPeerConnectionConfig;
|
||||
new (): RTCPeerConnectionConfig;
|
||||
};
|
||||
|
||||
interface Window{
|
||||
RTCPeerConnection: RTCPeerConnection;
|
||||
webkitRTCPeerConnection: webkitRTCPeerConnection;
|
||||
mozRTCPeerConnection: mozRTCPeerConnection;
|
||||
RTCSessionDescription: RTCSessionDescription;
|
||||
webkitRTCSessionDescription: webkitRTCSessionDescription;
|
||||
mozRTCSessionDescription: mozRTCSessionDescription;
|
||||
RTCIceCandidate: RTCIceCandidate;
|
||||
webkitRTCIceCandidate: webkitRTCIceCandidate;
|
||||
mozRTCIceCandidate: mozRTCIceCandidate;
|
||||
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;
|
||||
}
|
||||
|
||||
@@ -1,14 +1,14 @@
|
||||
# WebRTC Definition Notes
|
||||
|
||||
## The WebRTC specification
|
||||
## The WebRTC specification
|
||||
|
||||
The WebRTC specification is currently a work in progress, but it has been implemented at a basic level in recent versions of Chrome, Opera and (to a lesser extent) Firefox.
|
||||
The latest version of the specification can be found at http://dev.w3.org/2011/webrtc/editor/webrtc.html.
|
||||
The WebRTC specification is currently a work in progress, but it has been
|
||||
implemented at a basic level in recent versions of Chrome, Opera and (to a
|
||||
lesser extent) Firefox. The latest version of the specification can be found
|
||||
at https://www.w3.org/TR/webrtc/.
|
||||
|
||||
This particular set of definitions has been annotated with the vendor-specific prefixes for Chrome (e.g., `webitkit`),
|
||||
but anyone who wants, feel free to add the Mozilla-specific prefixes.
|
||||
This particular set of definitions does not use any vendor-specific prefixes.
|
||||
Instead, you should probably use [adapter.js](https://github.com/webrtc/adapter).
|
||||
|
||||
### Adding the reference to your project
|
||||
|
||||
|
||||
|
||||
The definitions track the currently published working draft. Deprecated
|
||||
features are dropped.
|
||||
|
||||
Reference in New Issue
Block a user