Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 191 insertions, 0 deletions

diff --git a/dom/webidl/RTCPeerConnection.webidl b/dom/webidl/RTCPeerConnection.webidl
new file mode 100644
index 0000000000..437e445781
--- /dev/null
+++ b/dom/webidl/RTCPeerConnection.webidl
@@ -0,0 +1,191 @@
+/* -*- Mode: IDL; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/.
+ *
+ * The origin of this IDL file is
+ * http://w3c.github.io/webrtc-pc/#interface-definition
+ */
+
+callback RTCSessionDescriptionCallback = void (RTCSessionDescriptionInit description);
+callback RTCPeerConnectionErrorCallback = void (DOMException error);
+callback RTCStatsCallback = void (RTCStatsReport report);
+
+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"
+};
+
+enum mozPacketDumpType {
+  "rtp", // dump unencrypted rtp as the MediaPipeline sees it
+  "srtp", // dump encrypted rtp as the MediaPipeline sees it
+  "rtcp", // dump unencrypted rtcp as the MediaPipeline sees it
+  "srtcp" // dump encrypted rtcp as the MediaPipeline sees it
+};
+
+callback mozPacketCallback = void (unsigned long level,
+                                   mozPacketDumpType type,
+                                   boolean sending,
+                                   ArrayBuffer packet);
+
+dictionary RTCDataChannelInit {
+  boolean        ordered = true;
+  [EnforceRange]
+  unsigned short maxPacketLifeTime;
+  [EnforceRange]
+  unsigned short maxRetransmits;
+  DOMString      protocol = "";
+  boolean        negotiated = false;
+  [EnforceRange]
+  unsigned short id;
+
+  // These are deprecated due to renaming in the spec, but still supported for Fx53
+  unsigned short maxRetransmitTime;
+};
+
+dictionary RTCOfferAnswerOptions {
+//  boolean voiceActivityDetection = true; // TODO: support this (Bug 1184712)
+};
+
+dictionary RTCAnswerOptions : RTCOfferAnswerOptions {
+};
+
+dictionary RTCOfferOptions : RTCOfferAnswerOptions {
+  boolean offerToReceiveVideo;
+  boolean offerToReceiveAudio;
+  boolean iceRestart = false;
+};
+
+[Pref="media.peerconnection.enabled",
+ JSImplementation="@mozilla.org/dom/peerconnection;1",
+ Exposed=Window]
+interface RTCPeerConnection : EventTarget  {
+  [Throws]
+  constructor(optional RTCConfiguration configuration = {},
+              optional object? constraints);
+
+  [Throws, StaticClassOverride="mozilla::dom::RTCCertificate"]
+  static Promise<RTCCertificate> generateCertificate (AlgorithmIdentifier keygenAlgorithm);
+
+  [Pref="media.peerconnection.identity.enabled"]
+  void setIdentityProvider (DOMString provider,
+                            optional RTCIdentityProviderOptions options = {});
+  [Pref="media.peerconnection.identity.enabled"]
+  Promise<DOMString> getIdentityAssertion();
+  Promise<RTCSessionDescriptionInit> createOffer (optional RTCOfferOptions options = {});
+  Promise<RTCSessionDescriptionInit> createAnswer (optional RTCAnswerOptions options = {});
+  Promise<void> setLocalDescription (optional RTCSessionDescriptionInit description = {});
+  Promise<void> setRemoteDescription (optional RTCSessionDescriptionInit description = {});
+  readonly attribute RTCSessionDescription? localDescription;
+  readonly attribute RTCSessionDescription? currentLocalDescription;
+  readonly attribute RTCSessionDescription? pendingLocalDescription;
+  readonly attribute RTCSessionDescription? remoteDescription;
+  readonly attribute RTCSessionDescription? currentRemoteDescription;
+  readonly attribute RTCSessionDescription? pendingRemoteDescription;
+  readonly attribute RTCSignalingState signalingState;
+  Promise<void> addIceCandidate (optional (RTCIceCandidateInit or RTCIceCandidate) candidate = {});
+  readonly attribute boolean? canTrickleIceCandidates;
+  readonly attribute RTCIceGatheringState iceGatheringState;
+  readonly attribute RTCIceConnectionState iceConnectionState;
+  void restartIce ();
+  [Pref="media.peerconnection.identity.enabled"]
+  readonly attribute Promise<RTCIdentityAssertion> peerIdentity;
+  [Pref="media.peerconnection.identity.enabled"]
+  readonly attribute DOMString? idpLoginUrl;
+
+  [ChromeOnly]
+  attribute DOMString id;
+
+  RTCConfiguration      getConfiguration ();
+  [Deprecated="RTCPeerConnectionGetStreams"]
+  sequence<MediaStream> getLocalStreams ();
+  [Deprecated="RTCPeerConnectionGetStreams"]
+  sequence<MediaStream> getRemoteStreams ();
+  void addStream (MediaStream stream);
+
+  // replaces addStream; fails if already added
+  // because a track can be part of multiple streams, stream parameters
+  // indicate which particular streams should be referenced in signaling
+
+  RTCRtpSender addTrack(MediaStreamTrack track,
+                        MediaStream... streams);
+  void removeTrack(RTCRtpSender sender);
+
+  RTCRtpTransceiver addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
+                                   optional RTCRtpTransceiverInit init = {});
+
+  sequence<RTCRtpSender> getSenders();
+  sequence<RTCRtpReceiver> getReceivers();
+  sequence<RTCRtpTransceiver> getTransceivers();
+
+  [ChromeOnly]
+  void mozSetPacketCallback(mozPacketCallback callback);
+  [ChromeOnly]
+  void mozEnablePacketDump(unsigned long level,
+                           mozPacketDumpType type,
+                           boolean sending);
+  [ChromeOnly]
+  void mozDisablePacketDump(unsigned long level,
+                            mozPacketDumpType type,
+                            boolean sending);
+
+  void close ();
+  attribute EventHandler onnegotiationneeded;
+  attribute EventHandler onicecandidate;
+  attribute EventHandler onsignalingstatechange;
+  attribute EventHandler onaddstream; // obsolete
+  attribute EventHandler onaddtrack;  // obsolete
+  attribute EventHandler ontrack;     // replaces onaddtrack and onaddstream.
+  attribute EventHandler oniceconnectionstatechange;
+  attribute EventHandler onicegatheringstatechange;
+
+  Promise<RTCStatsReport> getStats (optional MediaStreamTrack? selector = null);
+
+  // Data channel.
+  RTCDataChannel createDataChannel (DOMString label,
+                                    optional RTCDataChannelInit dataChannelDict = {});
+  attribute EventHandler ondatachannel;
+};
+
+// Legacy callback API
+
+partial interface RTCPeerConnection {
+
+  // Dummy Promise<void> return values avoid "WebIDL.WebIDLError: error:
+  // We have overloads with both Promise and non-Promise return types"
+
+  Promise<void> createOffer (RTCSessionDescriptionCallback successCallback,
+                             RTCPeerConnectionErrorCallback failureCallback,
+                             optional RTCOfferOptions options = {});
+  Promise<void> createAnswer (RTCSessionDescriptionCallback successCallback,
+                              RTCPeerConnectionErrorCallback failureCallback);
+  Promise<void> setLocalDescription (RTCSessionDescriptionInit description,
+                                     VoidFunction successCallback,
+                                     RTCPeerConnectionErrorCallback failureCallback);
+  Promise<void> setRemoteDescription (RTCSessionDescriptionInit description,
+                                      VoidFunction successCallback,
+                                      RTCPeerConnectionErrorCallback failureCallback);
+  Promise<void> addIceCandidate (RTCIceCandidate candidate,
+                                 VoidFunction successCallback,
+                                 RTCPeerConnectionErrorCallback failureCallback);
+};