script: Move code generation and webidl files to new script_bindings crate. (#35157)

Signed-off-by: Josh Matthews <josh@joshmatthews.net>

components/script_bindings/webidls/RTCPeerConnection.webidl

@@ -0,0 +1,153 @@
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+// https://w3c.github.io/webrtc-pc/#interface-definition
+
+[Exposed=Window, Pref="dom_webrtc_enabled"]
+interface RTCPeerConnection : EventTarget {
+    [Throws] constructor(optional RTCConfiguration configuration = {});
+    Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions options = {});
+    Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions options = {});
+    Promise<undefined>                      setLocalDescription(RTCSessionDescriptionInit description);
+    readonly attribute RTCSessionDescription? localDescription;
+    // readonly attribute RTCSessionDescription? currentLocalDescription;
+    // readonly attribute RTCSessionDescription? pendingLocalDescription;
+    Promise<undefined>                      setRemoteDescription(RTCSessionDescriptionInit description);
+    readonly attribute RTCSessionDescription? remoteDescription;
+    // readonly attribute RTCSessionDescription? currentRemoteDescription;
+    // readonly attribute RTCSessionDescription? pendingRemoteDescription;
+    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;
+    // static sequence<RTCIceServer>      getDefaultIceServers();
+    // RTCConfiguration                   getConfiguration();
+    // void                               setConfiguration(RTCConfiguration configuration);
+    undefined                               close();
+             attribute EventHandler           onnegotiationneeded;
+             attribute EventHandler           onicecandidate;
+    //          attribute EventHandler           onicecandidateerror;
+             attribute EventHandler           onsignalingstatechange;
+             attribute EventHandler           oniceconnectionstatechange;
+             attribute EventHandler           onicegatheringstatechange;
+    //          attribute EventHandler           onconnectionstatechange;
+
+    // removed from spec, but still shipped by browsers
+    undefined addStream (MediaStream stream);
+};
+
+dictionary RTCConfiguration {
+    sequence<RTCIceServer>   iceServers;
+    RTCIceTransportPolicy    iceTransportPolicy = "all";
+    RTCBundlePolicy          bundlePolicy = "balanced";
+    RTCRtcpMuxPolicy         rtcpMuxPolicy = "require";
+    DOMString                peerIdentity;
+    // sequence<RTCCertificate> certificates;
+    [EnforceRange]
+    octet                    iceCandidatePoolSize = 0;
+};
+
+enum RTCIceTransportPolicy {
+    "relay",
+    "all"
+};
+
+enum RTCBundlePolicy {
+    "balanced",
+    "max-compat",
+    "max-bundle"
+};
+
+enum RTCRtcpMuxPolicy {
+    // At risk due to lack of implementers' interest.
+    "negotiate",
+    "require"
+};
+
+dictionary RTCIceServer {
+    required (DOMString or sequence<DOMString>) urls;
+             DOMString                          username;
+             DOMString /*(DOMString or RTCOAuthCredential)*/  credential;
+             RTCIceCredentialType               credentialType = "password";
+};
+
+enum RTCIceCredentialType {
+    "password",
+    "oauth"
+};
+
+dictionary RTCOfferAnswerOptions {
+    boolean voiceActivityDetection = true;
+};
+
+dictionary RTCOfferOptions : RTCOfferAnswerOptions {
+    boolean iceRestart = false;
+};
+
+dictionary RTCAnswerOptions : RTCOfferAnswerOptions {
+};
+
+enum RTCIceGatheringState {
+    "new",
+    "gathering",
+    "complete"
+};
+
+enum RTCIceConnectionState {
+    "new",
+    "checking",
+    "connected",
+    "completed",
+    "disconnected",
+    "failed",
+    "closed"
+};
+
+enum RTCSignalingState {
+    "stable",
+    "have-local-offer",
+    "have-remote-offer",
+    "have-local-pranswer",
+    "have-remote-pranswer",
+    "closed"
+};
+
+dictionary RTCRtpCodingParameters {
+  DOMString rid;
+};
+
+dictionary RTCRtpEncodingParameters : RTCRtpCodingParameters {
+  boolean active = true;
+  unsigned long maxBitrate;
+  double scaleResolutionDownBy;
+};
+
+dictionary RTCRtpTransceiverInit {
+  RTCRtpTransceiverDirection direction = "sendrecv";
+  sequence<MediaStream> streams = [];
+  sequence<RTCRtpEncodingParameters> sendEncodings = [];
+};
+
+partial interface RTCPeerConnection {
+    // sequence<RTCRtpSender>      getSenders();
+    // sequence<RTCRtpReceiver>    getReceivers();
+    // sequence<RTCRtpTransceiver> getTransceivers();
+    // RTCRtpSender                addTrack(MediaStreamTrack track,
+    //                                      MediaStream... streams);
+    // void                        removeTrack(RTCRtpSender sender);
+    [Pref="dom_webrtc_transceiver_enabled"]
+    RTCRtpTransceiver           addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
+                                               optional RTCRtpTransceiverInit init = {});
+    attribute EventHandler ontrack;
+};
+
+// https://www.w3.org/TR/webrtc/#rtcpeerconnection-interface-extensions-0
+partial interface RTCPeerConnection {
+  // readonly attribute RTCSctpTransport? sctp;
+  RTCDataChannel createDataChannel(USVString label,
+                                   optional RTCDataChannelInit dataChannelDict = {});
+  attribute EventHandler ondatachannel;
+};