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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 RTCIceGatheringState {
"new",
"gathering",
"complete"
};
enum RTCIceConnectionState {
"new",
"checking",
"connected",
"completed",
"failed",
"disconnected",
"closed"
};
// https://w3c.github.io/webrtc-pc/#interface-definition
// TODO(guidou): Many types are of the wrong type in this interface:
// * Dictionary -> specific dictionary types like RTCConfiguration
// * VoidCallback -> VoidFunction
[
ActiveScriptWrappable,
DependentLifetime,
// TODO(guidou): There should only be one constructor argument.
Constructor(optional RTCConfiguration configuration, optional Dictionary mediaConstraints),
ConstructorCallWith=ExecutionContext,
RaisesException=Constructor
] interface RTCPeerConnection : EventTarget {
[CallWith=ScriptState] Promise<RTCSessionDescription> createOffer(optional RTCOfferOptions options);
[CallWith=ScriptState] Promise<RTCSessionDescription> createAnswer(optional RTCAnswerOptions options);
[CallWith=ScriptState] Promise<void> setLocalDescription(RTCSessionDescriptionInit description);
readonly attribute RTCSessionDescription? localDescription;
// readonly attribute RTCSessionDescription? currentLocalDescription;
// readonly attribute RTCSessionDescription? pendingLocalDescription;
[CallWith=ScriptState] Promise<void> setRemoteDescription(RTCSessionDescriptionInit description);
readonly attribute RTCSessionDescription? remoteDescription;
// readonly attribute RTCSessionDescription? currentRemoteDescription;
// readonly attribute RTCSessionDescription? pendingRemoteDescription;
[CallWith=ScriptState, MeasureAs=RTCPeerConnectionAddIceCandidatePromise] Promise<void> addIceCandidate ((RTCIceCandidateInit or RTCIceCandidate) candidate);
readonly attribute RTCSignalingState signalingState;
readonly attribute RTCIceGatheringState iceGatheringState;
readonly attribute RTCIceConnectionState iceConnectionState;
// readonly attribute boolean? canTrickleIceCandidates;
// RTCConfiguration getConfiguration();
[CallWith=ScriptState, RaisesException] void setConfiguration(RTCConfiguration configuration);
// TODO(guidou): close() should never throw an exception.
[RaisesException] void close();
attribute EventHandler onnegotiationneeded;
attribute EventHandler onicecandidate;
attribute EventHandler onsignalingstatechange;
attribute EventHandler oniceconnectionstatechange;
attribute EventHandler onicegatheringstatechange;
// https://w3c.github.io/webrtc-pc/#legacy-interface-extensions
// These methods return Promise<void> because having Promise-based versions requires that all overloads return Promises.
[CallWith=ScriptState, RaisesException] Promise<void> createOffer(RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional Dictionary rtcOfferOptions);
// TODO(guidou): There should be no mediaConstraints argument.
[CallWith=ScriptState] Promise<void> createAnswer(RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional Dictionary mediaConstraints);
[CallWith=ScriptState] Promise<void> setLocalDescription(RTCSessionDescriptionInit description, VoidCallback successCallback, [Default=Undefined] optional RTCPeerConnectionErrorCallback failureCallback);
// TODO(guidou): The failureCallback argument should be non-optional.
[CallWith=ScriptState] Promise<void> setRemoteDescription(RTCSessionDescriptionInit description, VoidCallback successCallback, [Default=Undefined] optional RTCPeerConnectionErrorCallback failureCallback);
[CallWith=ScriptState, MeasureAs=RTCPeerConnectionAddIceCandidateLegacy] Promise<void> addIceCandidate((RTCIceCandidateInit or RTCIceCandidate) candidate, VoidCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
// TODO(guidou): The selector argument should the first (nullable,
// non-optional) argument, and there should be a third failureCallback
// argument.
[CallWith=ScriptState, LegacyInterfaceTypeChecking] Promise<void> getStats(RTCStatsCallback successCallback, optional MediaStreamTrack? selector);
// TODO(hbos): The spec has an optional |selector| argument that is not
// supported yet. There is a discussion about what to do with it (clarify
// spec, remove it or change it?): https://github.com/w3c/webrtc-stats/issues/116
[CallWith=ScriptState] Promise<RTCStatsReport> getStats();
// RTP Media API
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getsenders
[RuntimeEnabled=RTCRtpSender] sequence<RTCRtpSender> getSenders();
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getreceivers
sequence<RTCRtpReceiver> getReceivers();
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtrack
[RuntimeEnabled=RTCRtpSender, RaisesException] RTCRtpSender addTrack(MediaStreamTrack track, MediaStream... streams);
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-removetrack
[RuntimeEnabled=RTCRtpSender, RaisesException] void removeTrack(RTCRtpSender sender);
// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api
[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);
// Non-standard or removed from the spec:
[Measure] sequence<MediaStream> getLocalStreams();
[Measure] sequence<MediaStream> getRemoteStreams();
[Measure, CallWith=ScriptState, RaisesException] void addStream(MediaStream? stream, optional Dictionary mediaConstraints);
[Measure, RaisesException] void removeStream(MediaStream? stream);
[Measure, RaisesException] RTCDTMFSender createDTMFSender(MediaStreamTrack track);
attribute EventHandler onaddstream;
attribute EventHandler onremovestream;
};