Merging upstream version 1.71.1+dfsg1.

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

1 files changed, 200 insertions, 0 deletions

diff --git a/vendor/web-sys/webidls/enabled/RTCStatsReport.webidl b/vendor/web-sys/webidls/enabled/RTCStatsReport.webidl
new file mode 100644
index 000000000..a52a2a8f5
--- /dev/null
+++ b/vendor/web-sys/webidls/enabled/RTCStatsReport.webidl
@@ -0,0 +1,200 @@
+/* -*- 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://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcstatsreport-object
+ * http://www.w3.org/2011/04/webrtc/wiki/Stats
+ */
+
+enum RTCStatsType {
+  "inbound-rtp",
+  "outbound-rtp",
+  "csrc",
+  "session",
+  "track",
+  "transport",
+  "candidate-pair",
+  "local-candidate",
+  "remote-candidate"
+};
+
+dictionary RTCStats {
+  DOMHighResTimeStamp timestamp;
+  RTCStatsType type;
+  DOMString id;
+};
+
+dictionary RTCRTPStreamStats : RTCStats {
+  DOMString ssrc;
+  DOMString mediaType;
+  DOMString remoteId;
+  boolean isRemote = false;
+  DOMString mediaTrackId;
+  DOMString transportId;
+  DOMString codecId;
+
+  // Video encoder/decoder measurements, not present in RTCP case
+  double bitrateMean;
+  double bitrateStdDev;
+  double framerateMean;
+  double framerateStdDev;
+
+  // Local only measurements, RTCP related but not communicated via RTCP. Not
+  // present in RTCP case.
+  unsigned long firCount;
+  unsigned long pliCount;
+  unsigned long nackCount;
+};
+
+dictionary RTCInboundRTPStreamStats : RTCRTPStreamStats {
+  unsigned long packetsReceived;
+  unsigned long long bytesReceived;
+  double jitter;
+  unsigned long packetsLost;
+  long roundTripTime;
+
+  // Video decoder measurement, not present in RTCP case
+  unsigned long discardedPackets;
+  unsigned long framesDecoded;
+};
+
+dictionary RTCOutboundRTPStreamStats : RTCRTPStreamStats {
+  unsigned long packetsSent;
+  unsigned long long bytesSent;
+  double targetBitrate;  // config encoder bitrate target of this SSRC in bits/s
+
+  // Video encoder measurements, not present in RTCP case
+  unsigned long droppedFrames;
+  unsigned long framesEncoded;
+};
+
+dictionary RTCMediaStreamTrackStats : RTCStats {
+  DOMString trackIdentifier;      // track.id property
+  boolean remoteSource;
+  sequence<DOMString> ssrcIds;
+  // Stuff that makes sense for video
+  unsigned long frameWidth;
+  unsigned long frameHeight;
+  double framesPerSecond;        // The nominal FPS value
+  unsigned long framesSent;
+  unsigned long framesReceived;  // Only for remoteSource=true
+  unsigned long framesDecoded;
+  unsigned long framesDropped;   // See VideoPlaybackQuality.droppedVideoFrames
+  unsigned long framesCorrupted; // as above.
+  // Stuff that makes sense for audio
+  double audioLevel;               // linear, 1.0 = 0 dBov (from RFC 6464).
+  // AEC stuff on audio tracks sourced from a microphone where AEC is applied
+  double echoReturnLoss;           // in decibels from G.168 (2012) section 3.14
+  double echoReturnLossEnhancement; // as above, section 3.15
+};
+
+dictionary RTCMediaStreamStats : RTCStats {
+  DOMString streamIdentifier;     // stream.id property
+  sequence<DOMString> trackIds;   // Note: stats object ids, not track.id
+};
+
+dictionary RTCRTPContributingSourceStats : RTCStats {
+  unsigned long contributorSsrc;
+  DOMString     inboundRtpStreamId;
+};
+
+dictionary RTCTransportStats: RTCStats {
+  unsigned long bytesSent;
+  unsigned long bytesReceived;
+};
+
+dictionary RTCIceComponentStats : RTCStats {
+  DOMString transportId;
+  long component;
+  unsigned long bytesSent;
+  unsigned long bytesReceived;
+  boolean activeConnection;
+};
+
+enum RTCStatsIceCandidatePairState {
+  "frozen",
+  "waiting",
+  "inprogress",
+  "failed",
+  "succeeded",
+  "cancelled"
+};
+
+dictionary RTCIceCandidatePairStats : RTCStats {
+  DOMString transportId;
+  DOMString localCandidateId;
+  DOMString remoteCandidateId;
+  RTCStatsIceCandidatePairState state;
+  unsigned long long priority;
+  boolean nominated;
+  boolean writable;
+  boolean readable;
+  unsigned long long bytesSent;
+  unsigned long long bytesReceived;
+  DOMHighResTimeStamp lastPacketSentTimestamp;
+  DOMHighResTimeStamp lastPacketReceivedTimestamp;
+  boolean selected;
+  [ChromeOnly]
+  unsigned long componentId; // moz
+};
+
+enum RTCStatsIceCandidateType {
+  "host",
+  "serverreflexive",
+  "peerreflexive",
+  "relayed"
+};
+
+dictionary RTCIceCandidateStats : RTCStats {
+  DOMString componentId;
+  DOMString candidateId;
+  DOMString ipAddress;
+  DOMString transport;
+  long portNumber;
+  RTCStatsIceCandidateType candidateType;
+};
+
+dictionary RTCCodecStats : RTCStats {
+  unsigned long payloadType;       // As used in RTP encoding.
+  DOMString codec;                 // video/vp8 or equivalent
+  unsigned long clockRate;
+  unsigned long channels;          // 2=stereo, missing for most other cases.
+  DOMString parameters;            // From SDP description line
+};
+
+// This is the internal representation of the report in this implementation
+// to be received from c++
+
+dictionary RTCStatsReportInternal {
+  DOMString                               pcid = "";
+  sequence<RTCInboundRTPStreamStats>      inboundRTPStreamStats;
+  sequence<RTCOutboundRTPStreamStats>     outboundRTPStreamStats;
+  sequence<RTCRTPContributingSourceStats> rtpContributingSourceStats;
+  sequence<RTCMediaStreamTrackStats>      mediaStreamTrackStats;
+  sequence<RTCMediaStreamStats>           mediaStreamStats;
+  sequence<RTCTransportStats>             transportStats;
+  sequence<RTCIceComponentStats>          iceComponentStats;
+  sequence<RTCIceCandidatePairStats>      iceCandidatePairStats;
+  sequence<RTCIceCandidateStats>          iceCandidateStats;
+  sequence<RTCCodecStats>                 codecStats;
+  DOMString                               localSdp;
+  DOMString                               remoteSdp;
+  DOMHighResTimeStamp                     timestamp;
+  unsigned long                           iceRestarts;
+  unsigned long                           iceRollbacks;
+  boolean                                 offerer; // Is the PC the offerer
+  boolean                                 closed; // Is the PC now closed
+  sequence<RTCIceCandidateStats>          trickledIceCandidateStats;
+  sequence<DOMString>                     rawLocalCandidates;
+  sequence<DOMString>                     rawRemoteCandidates;
+};
+
+[Pref="media.peerconnection.enabled",
+// TODO: Use MapClass here once it's available (Bug 928114)
+// MapClass(DOMString, object)
+ JSImplementation="@mozilla.org/dom/rtcstatsreport;1"]
+interface RTCStatsReport {
+  readonly maplike<DOMString, object>;
+};