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/*
* Copyright (C) 2012 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer
* in the documentation and/or other materials provided with the
* distribution.
* 3. Neither the name of Google Inc. nor the names of its contributors
* may be used to endorse or promote products derived from this
* software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
* OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
// https://w3c.github.io/webrtc-pc/#state-definitions
enum RTCSignalingState {
"stable",
"have-local-offer",
"have-remote-offer",
"have-local-pranswer",
"have-remote-pranswer",
"closed"
};
enum RTCIceConnectionState {
"new",
"checking",
"connected",
"completed",
"failed",
"disconnected",
"closed"
};
enum RTCPeerConnectionState {
"new",
"connecting",
"connected",
"disconnected",
"failed",
"closed"
};
// https://w3c.github.io/webrtc-pc/#interface-definition
[
ActiveScriptWrappable,
Exposed=Window,
LegacyWindowAlias=webkitRTCPeerConnection,
LegacyWindowAlias_Measure
] interface RTCPeerConnection : EventTarget {
[CallWith=ExecutionContext, RaisesException] constructor(optional RTCConfiguration configuration = {});
[CallWith=ScriptState, RaisesException] Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions options = {});
[CallWith=ScriptState, RaisesException] Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions options = {});
[CallWith=ScriptState, RaisesException] Promise<undefined> setLocalDescription(optional RTCSessionDescriptionInit description = {});
readonly attribute RTCSessionDescription? localDescription;
readonly attribute RTCSessionDescription? currentLocalDescription;
readonly attribute RTCSessionDescription? pendingLocalDescription;
[CallWith=ScriptState, RaisesException] Promise<undefined> setRemoteDescription(RTCSessionDescriptionInit description);
readonly attribute RTCSessionDescription? remoteDescription;
readonly attribute RTCSessionDescription? currentRemoteDescription;
readonly attribute RTCSessionDescription? pendingRemoteDescription;
[CallWith=ScriptState, RaisesException, MeasureAs=RTCPeerConnectionAddIceCandidatePromise] Promise<undefined> addIceCandidate(optional RTCIceCandidateInit candidate = {});
readonly attribute RTCSignalingState signalingState;
readonly attribute RTCIceGatheringState iceGatheringState;
readonly attribute RTCIceConnectionState iceConnectionState;
readonly attribute RTCPeerConnectionState connectionState;
readonly attribute boolean? canTrickleIceCandidates;
void restartIce();
[CallWith=ScriptState] RTCConfiguration getConfiguration();
[CallWith=ScriptState, RaisesException] void setConfiguration(optional RTCConfiguration configuration = {});
void close();
attribute EventHandler onnegotiationneeded;
attribute EventHandler onicecandidate;
attribute EventHandler onsignalingstatechange;
attribute EventHandler oniceconnectionstatechange;
attribute EventHandler onconnectionstatechange;
attribute EventHandler onicegatheringstatechange;
attribute EventHandler onicecandidateerror;
// https://w3c.github.io/webrtc-pc/#legacy-interface-extensions
// These methods return Promise<undefined> because having Promise-based versions requires that all overloads return Promises.
[CallWith=ScriptState, RaisesException] Promise<undefined> createOffer(RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional RTCOfferOptions options = {});
[CallWith=ScriptState, RaisesException] Promise<undefined> createAnswer(RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
// TODO(guidou): The failureCallback argument should be non-optional.
// TODO(crbug.com/841185): |failureCallback| is not nullable in the spec.
[CallWith=ScriptState] Promise<undefined> setLocalDescription(RTCSessionDescriptionInit description, VoidFunction successCallback, optional RTCPeerConnectionErrorCallback? failureCallback);
// TODO(guidou): The failureCallback argument should be non-optional.
// TODO(crbug.com/841185): |failureCallback| is not nullable in the spec.
[CallWith=ScriptState] Promise<undefined> setRemoteDescription(RTCSessionDescriptionInit description, VoidFunction successCallback, optional RTCPeerConnectionErrorCallback? failureCallback);
[CallWith=ScriptState, RaisesException, MeasureAs=RTCPeerConnectionAddIceCandidateLegacy] Promise<undefined> addIceCandidate(RTCIceCandidateInit candidate, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
// https://w3c.github.io/webrtc-pc/#sec.stats-model
[CallWith=ScriptState, RaisesException] Promise<RTCStatsReport> getStats(optional MediaStreamTrack? selector = null);
// RTP Media API
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-gettransceivers
[Measure] sequence<RTCRtpTransceiver> getTransceivers();
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getsenders
[Measure] sequence<RTCRtpSender> getSenders();
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getreceivers
[Measure] sequence<RTCRtpReceiver> getReceivers();
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
[Measure, RaisesException] RTCRtpTransceiver addTransceiver(
(MediaStreamTrack or DOMString) trackOrKind,
optional RTCRtpTransceiverInit init = {});
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtrack
[Measure, RaisesException] RTCRtpSender addTrack(MediaStreamTrack track, MediaStream... streams);
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-removetrack
[Measure, RaisesException] void removeTrack(RTCRtpSender sender);
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-ontrack
attribute EventHandler ontrack;
// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api
readonly attribute RTCSctpTransport? sctp;
[CallWith=ScriptState, RaisesException] RTCDataChannel createDataChannel(USVString label, optional RTCDataChannelInit dataChannelDict = {});
attribute EventHandler ondatachannel;
// Certificate management
// https://w3c.github.io/webrtc-pc/#sec.cert-mgmt
[RaisesException, CallWith=ScriptState] static Promise<RTCCertificate> generateCertificate(AlgorithmIdentifier keygenAlgorithm);
[SameObject, RuntimeEnabled=RTCRtpTransport] readonly attribute RTCRtpTransport rtpTransport;
// Non-standard or removed from the spec:
[Measure] sequence<MediaStream> getLocalStreams();
[Measure] sequence<MediaStream> getRemoteStreams();
[Measure, CallWith=ScriptState, RaisesException] void addStream(MediaStream stream);
[Measure, RaisesException] void removeStream(MediaStream stream);
[Measure, RaisesException] RTCDTMFSender createDTMFSender(MediaStreamTrack track);
attribute EventHandler onaddstream;
attribute EventHandler onremovestream;
};
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionerrorcallback
callback RTCPeerConnectionErrorCallback = void (DOMException error);
// https://w3c.github.io/webrtc-pc/#dom-rtcsessiondescriptioncallback
callback RTCSessionDescriptionCallback = void (RTCSessionDescriptionInit description);
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