Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

1 files changed, 34 insertions, 0 deletions

diff --git a/dom/webidl/RTCRtpReceiver.webidl b/dom/webidl/RTCRtpReceiver.webidl
new file mode 100644
index 0000000000..59a4f1340f
--- /dev/null
+++ b/dom/webidl/RTCRtpReceiver.webidl
@@ -0,0 +1,34 @@
+/* -*- 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://lists.w3.org/Archives/Public/public-webrtc/2014May/0067.html
+ */
+
+[Pref="media.peerconnection.enabled",
+ Exposed=Window]
+interface RTCRtpReceiver {
+  readonly attribute MediaStreamTrack   track;
+  readonly attribute RTCDtlsTransport?  transport;
+  static RTCRtpCapabilities? getCapabilities(DOMString kind);
+  sequence<RTCRtpContributingSource>    getContributingSources();
+  sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
+  [NewObject]
+  Promise<RTCStatsReport>               getStats();
+
+  // test-only: for testing getContributingSources
+  [ChromeOnly]
+  undefined mozInsertAudioLevelForContributingSource(unsigned long source,
+                                                     DOMHighResTimeStamp timestamp,
+                                                     unsigned long rtpTimestamp,
+                                                     boolean hasLevel,
+                                                     byte level);
+};
+
+//https://w3c.github.io/webrtc-extensions/#rtcrtpreceiver-jitterbuffertarget-rtcrtpreceiver-interface
+partial interface RTCRtpReceiver {
+  [Throws]
+  attribute DOMHighResTimeStamp? jitterBufferTarget;
+};