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https://github.com/fluencelabs/wasm-bindgen
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* Try to enable all webidls * Separate out unavailable webidl files by reason. * Create record of fully tested WebIDL files * Update notes to reflect new situation with web-idl * Make a blank ident fail, disable the necessary widls. It turns out that all the blank idents came from blank enum variants, which is allowed in webidl apparently.
204 lines
8.1 KiB
Plaintext
Vendored
204 lines
8.1 KiB
Plaintext
Vendored
/* -*- Mode: IDL; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/.
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*
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* The origin of this IDL file is
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* http://w3c.github.io/webrtc-pc/#interface-definition
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*/
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callback RTCSessionDescriptionCallback = void (RTCSessionDescriptionInit description);
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callback RTCPeerConnectionErrorCallback = void (DOMException error);
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callback RTCStatsCallback = void (RTCStatsReport report);
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enum RTCSignalingState {
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"stable",
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"have-local-offer",
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"have-remote-offer",
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"have-local-pranswer",
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"have-remote-pranswer",
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"closed"
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};
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enum RTCIceGatheringState {
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"new",
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"gathering",
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"complete"
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};
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enum RTCIceConnectionState {
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"new",
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"checking",
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"connected",
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"completed",
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"failed",
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"disconnected",
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"closed"
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};
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enum mozPacketDumpType {
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"rtp", // dump unencrypted rtp as the MediaPipeline sees it
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"srtp", // dump encrypted rtp as the MediaPipeline sees it
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"rtcp", // dump unencrypted rtcp as the MediaPipeline sees it
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"srtcp" // dump encrypted rtcp as the MediaPipeline sees it
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};
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callback mozPacketCallback = void (unsigned long level,
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mozPacketDumpType type,
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boolean sending,
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ArrayBuffer packet);
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dictionary RTCDataChannelInit {
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boolean ordered = true;
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unsigned short maxPacketLifeTime;
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unsigned short maxRetransmits;
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DOMString protocol = "";
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boolean negotiated = false;
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unsigned short id;
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// These are deprecated due to renaming in the spec, but still supported for Fx53
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unsigned short maxRetransmitTime;
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};
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dictionary RTCOfferAnswerOptions {
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// boolean voiceActivityDetection = true; // TODO: support this (Bug 1184712)
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};
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dictionary RTCAnswerOptions : RTCOfferAnswerOptions {
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};
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dictionary RTCOfferOptions : RTCOfferAnswerOptions {
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boolean offerToReceiveVideo;
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boolean offerToReceiveAudio;
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boolean iceRestart = false;
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};
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[Pref="media.peerconnection.enabled",
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JSImplementation="@mozilla.org/dom/peerconnection;1",
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Constructor (optional RTCConfiguration configuration,
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optional object? constraints)]
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interface RTCPeerConnection : EventTarget {
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[Throws, StaticClassOverride="mozilla::dom::RTCCertificate"]
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static Promise<RTCCertificate> generateCertificate (AlgorithmIdentifier keygenAlgorithm);
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[Pref="media.peerconnection.identity.enabled"]
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void setIdentityProvider (DOMString provider,
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optional RTCIdentityProviderOptions options);
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[Pref="media.peerconnection.identity.enabled"]
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Promise<DOMString> getIdentityAssertion();
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Promise<RTCSessionDescriptionInit> createOffer (optional RTCOfferOptions options);
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Promise<RTCSessionDescriptionInit> createAnswer (optional RTCAnswerOptions options);
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Promise<void> setLocalDescription (RTCSessionDescriptionInit description);
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Promise<void> setRemoteDescription (RTCSessionDescriptionInit description);
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readonly attribute RTCSessionDescription? localDescription;
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readonly attribute RTCSessionDescription? currentLocalDescription;
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readonly attribute RTCSessionDescription? pendingLocalDescription;
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readonly attribute RTCSessionDescription? remoteDescription;
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readonly attribute RTCSessionDescription? currentRemoteDescription;
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readonly attribute RTCSessionDescription? pendingRemoteDescription;
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readonly attribute RTCSignalingState signalingState;
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Promise<void> addIceCandidate ((RTCIceCandidateInit or RTCIceCandidate)? candidate);
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readonly attribute boolean? canTrickleIceCandidates;
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readonly attribute RTCIceGatheringState iceGatheringState;
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readonly attribute RTCIceConnectionState iceConnectionState;
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[Pref="media.peerconnection.identity.enabled"]
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readonly attribute Promise<RTCIdentityAssertion> peerIdentity;
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[Pref="media.peerconnection.identity.enabled"]
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readonly attribute DOMString? idpLoginUrl;
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[ChromeOnly]
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attribute DOMString id;
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RTCConfiguration getConfiguration ();
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[Deprecated="RTCPeerConnectionGetStreams"]
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sequence<MediaStream> getLocalStreams ();
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[Deprecated="RTCPeerConnectionGetStreams"]
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sequence<MediaStream> getRemoteStreams ();
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void addStream (MediaStream stream);
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// replaces addStream; fails if already added
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// because a track can be part of multiple streams, stream parameters
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// indicate which particular streams should be referenced in signaling
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RTCRtpSender addTrack(MediaStreamTrack track,
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MediaStream stream,
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MediaStream... moreStreams);
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void removeTrack(RTCRtpSender sender);
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RTCRtpTransceiver addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
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optional RTCRtpTransceiverInit init);
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sequence<RTCRtpSender> getSenders();
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sequence<RTCRtpReceiver> getReceivers();
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sequence<RTCRtpTransceiver> getTransceivers();
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// test-only: for testing getContributingSources
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[ChromeOnly]
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DOMHighResTimeStamp mozGetNowInRtpSourceReferenceTime();
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// test-only: for testing getContributingSources
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[ChromeOnly]
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void mozInsertAudioLevelForContributingSource(RTCRtpReceiver receiver,
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unsigned long source,
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DOMHighResTimeStamp timestamp,
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boolean hasLevel,
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byte level);
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[ChromeOnly]
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void mozAddRIDExtension(RTCRtpReceiver receiver, unsigned short extensionId);
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[ChromeOnly]
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void mozAddRIDFilter(RTCRtpReceiver receiver, DOMString rid);
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[ChromeOnly]
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void mozSetPacketCallback(mozPacketCallback callback);
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[ChromeOnly]
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void mozEnablePacketDump(unsigned long level,
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mozPacketDumpType type,
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boolean sending);
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[ChromeOnly]
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void mozDisablePacketDump(unsigned long level,
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mozPacketDumpType type,
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boolean sending);
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void close ();
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attribute EventHandler onnegotiationneeded;
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attribute EventHandler onicecandidate;
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attribute EventHandler onsignalingstatechange;
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attribute EventHandler onaddstream; // obsolete
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attribute EventHandler onaddtrack; // obsolete
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attribute EventHandler ontrack; // replaces onaddtrack and onaddstream.
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attribute EventHandler onremovestream;
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attribute EventHandler oniceconnectionstatechange;
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attribute EventHandler onicegatheringstatechange;
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Promise<RTCStatsReport> getStats (optional MediaStreamTrack? selector);
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// Data channel.
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RTCDataChannel createDataChannel (DOMString label,
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optional RTCDataChannelInit dataChannelDict);
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attribute EventHandler ondatachannel;
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};
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// Legacy callback API
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partial interface RTCPeerConnection {
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// Dummy Promise<void> return values avoid "WebIDL.WebIDLError: error:
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// We have overloads with both Promise and non-Promise return types"
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Promise<void> createOffer (RTCSessionDescriptionCallback successCallback,
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RTCPeerConnectionErrorCallback failureCallback,
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optional RTCOfferOptions options);
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Promise<void> createAnswer (RTCSessionDescriptionCallback successCallback,
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RTCPeerConnectionErrorCallback failureCallback);
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Promise<void> setLocalDescription (RTCSessionDescriptionInit description,
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VoidFunction successCallback,
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RTCPeerConnectionErrorCallback failureCallback);
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Promise<void> setRemoteDescription (RTCSessionDescriptionInit description,
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VoidFunction successCallback,
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RTCPeerConnectionErrorCallback failureCallback);
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Promise<void> addIceCandidate (RTCIceCandidate candidate,
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VoidFunction successCallback,
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RTCPeerConnectionErrorCallback failureCallback);
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Promise<void> getStats (MediaStreamTrack? selector,
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RTCStatsCallback successCallback,
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RTCPeerConnectionErrorCallback failureCallback);
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};
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