| /* |
| * Copyright (C) 2012 Google Inc. All rights reserved. |
| * Copyright (C) 2013 Nokia Corporation and/or its subsidiary(-ies). |
| * Copyright (C) 2015, 2016 Ericsson AB. All rights reserved. |
| * Copyright (C) 2017 Apple 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. |
| */ |
| |
| typedef RTCRtpTransceiverDirection RtpTransceiverDirection; |
| |
| [ |
| Conditional=WEB_RTC, |
| EnabledAtRuntime=PeerConnection |
| ] dictionary RTCDataChannelInit { |
| boolean ordered = true; |
| unsigned short maxPacketLifeTime; |
| unsigned short maxRetransmits; |
| USVString protocol = ""; |
| boolean negotiated = false; |
| [EnforceRange] unsigned short id; |
| // FIXME 169644: missing priority |
| }; |
| |
| [ |
| Conditional=WEB_RTC, |
| EnabledAtRuntime=PeerConnection |
| ] dictionary RTCCertificateParameters { |
| DOMString name; |
| DOMString hash; |
| DOMString namedCurve; |
| unsigned long modulusLength; |
| Uint8Array publicExponent; |
| double expires; |
| }; |
| |
| typedef (object or DOMString) AlgorithmIdentifier; |
| |
| [ |
| Conditional=WEB_RTC, |
| EnabledAtRuntime=PeerConnection, |
| ImplementedAs=RTCRtpTransceiverInit |
| ] dictionary RTCRtpTransceiverInit { |
| RtpTransceiverDirection direction = "sendrecv"; |
| sequence<MediaStream> streams = []; |
| // FIXME 169662: missing sendEncodings |
| }; |
| |
| [ |
| ActiveDOMObject, |
| Conditional=WEB_RTC, |
| ConstructorCallWith=Document, |
| EnabledAtRuntime=PeerConnection, |
| ExportMacro=WEBCORE_EXPORT, |
| JSBuiltinConstructor |
| ] interface RTCPeerConnection : EventTarget { |
| // JS built-in constructor handles the optional RTCConfiguration |
| [PrivateIdentifier, CallWith=Document, MayThrowException] void initializeWith(RTCConfiguration configuration); |
| |
| |
| // 4.3.2 Interface Definition |
| // JSBuiltins provide support for legacy signatures |
| [JSBuiltin] Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions offerOptions); |
| [JSBuiltin] Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions answerOptions); |
| |
| [JSBuiltin] Promise<void> setLocalDescription(RTCSessionDescriptionInit description); |
| readonly attribute RTCSessionDescription? localDescription; |
| readonly attribute RTCSessionDescription? currentLocalDescription; |
| readonly attribute RTCSessionDescription? pendingLocalDescription; |
| |
| [JSBuiltin] Promise<void> setRemoteDescription(RTCSessionDescriptionInit description); |
| readonly attribute RTCSessionDescription? remoteDescription; |
| readonly attribute RTCSessionDescription? currentRemoteDescription; |
| readonly attribute RTCSessionDescription? pendingRemoteDescription; |
| |
| [JSBuiltin] Promise<void> addIceCandidate((RTCIceCandidateInit or RTCIceCandidate) candidate); |
| |
| readonly attribute RTCSignalingState signalingState; |
| readonly attribute RTCIceGatheringState iceGatheringState; |
| readonly attribute RTCIceConnectionState iceConnectionState; |
| readonly attribute RTCPeerConnectionState connectionState; |
| // FIXME 169644: missing canTrickleIceCandidates |
| // FIXME 169644: missing defaultIceServers |
| |
| RTCConfiguration getConfiguration(); |
| [MayThrowException] void setConfiguration(RTCConfiguration configuration); |
| void close(); |
| |
| attribute EventHandler onnegotiationneeded; |
| attribute EventHandler onicecandidate; |
| attribute EventHandler onsignalingstatechange; |
| attribute EventHandler oniceconnectionstatechange; |
| attribute EventHandler onicegatheringstatechange; |
| attribute EventHandler onconnectionstatechange; |
| // FIXME 169644: missing onfingerprintfailure and onicecandidateerror |
| |
| // Private API used to implement the overloaded operations above. Queued functions are called by runQueuedOperation(). |
| // See RTCPeerConnectionInternals.js. |
| [PrivateIdentifier] Promise<RTCSessionDescriptionInit> queuedCreateOffer(optional RTCOfferOptions offerOptions); |
| [PrivateIdentifier] Promise<RTCSessionDescriptionInit> queuedCreateAnswer(optional RTCAnswerOptions answerOptions); |
| [PrivateIdentifier] Promise<void> queuedSetLocalDescription(RTCSessionDescription description); |
| [PrivateIdentifier] Promise<void> queuedSetRemoteDescription(RTCSessionDescription description); |
| [PrivateIdentifier] Promise<void> queuedAddIceCandidate(RTCIceCandidate? candidate); |
| |
| |
| // 4.11 Certificate management |
| [CallWith=ExecState] static Promise<RTCCertificate> generateCertificate(AlgorithmIdentifier keygenAlgorithm); |
| |
| // 5.1 RTCPeerConnection extensions |
| // RTP Media API extensions |
| sequence<RTCRtpSender> getSenders(); |
| sequence<RTCRtpReceiver> getReceivers(); |
| sequence<RTCRtpTransceiver> getTransceivers(); |
| |
| [MayThrowException] RTCRtpSender addTrack(MediaStreamTrack track, MediaStream... streams); |
| [MayThrowException] void removeTrack(RTCRtpSender sender); |
| |
| [MayThrowException] RTCRtpTransceiver addTransceiver((MediaStreamTrack or DOMString) track, optional RTCRtpTransceiverInit init); |
| |
| attribute EventHandler ontrack; |
| |
| // 6.1 Peer-to-peer data API |
| // FIXME 169644: missing sctp |
| |
| [MayThrowException] RTCDataChannel createDataChannel([TreatNullAs=EmptyString] USVString label, optional RTCDataChannelInit options); |
| attribute EventHandler ondatachannel; |
| |
| |
| // 8.2 Statistics API |
| Promise<RTCStatsReport> getStats(optional MediaStreamTrack? selector = null); |
| |
| |
| // 9.6 Identity Provider API |
| // FIXME 169644: missing IdP |
| }; |