Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 334 insertions, 0 deletions

diff --git a/dom/webidl/RTCStatsReport.webidl b/dom/webidl/RTCStatsReport.webidl
new file mode 100644
index 0000000000..f10e4b56d3
--- /dev/null
+++ b/dom/webidl/RTCStatsReport.webidl
@@ -0,0 +1,334 @@
+/* -*- 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
+ * https://www.w3.org/TR/webrtc-stats/
+ */
+
+enum RTCStatsType {
+  "codec",
+  "inbound-rtp",
+  "outbound-rtp",
+  "remote-inbound-rtp",
+  "remote-outbound-rtp",
+  "media-source",
+  "peer-connection",
+  "csrc",
+  "data-channel",
+  "session",
+  "track",
+  "transport",
+  "candidate-pair",
+  "local-candidate",
+  "remote-candidate"
+};
+
+dictionary RTCStats {
+  DOMHighResTimeStamp timestamp;
+  RTCStatsType type;
+  DOMString id;
+};
+
+dictionary RTCRtpStreamStats : RTCStats {
+  required unsigned long ssrc;
+  required DOMString kind;
+  DOMString mediaType;
+  DOMString transportId;
+  DOMString codecId;
+};
+
+dictionary RTCCodecStats : RTCStats {
+  required unsigned long payloadType;
+  RTCCodecType  codecType;
+  required DOMString     transportId;
+  required DOMString     mimeType;
+  unsigned long clockRate;
+  unsigned long channels;
+  DOMString     sdpFmtpLine;
+};
+
+enum RTCCodecType {
+  "encode",
+  "decode",
+};
+
+dictionary RTCReceivedRtpStreamStats: RTCRtpStreamStats {
+  unsigned long long packetsReceived;
+  long long packetsLost;
+  double jitter;
+  unsigned long discardedPackets; // non-standard alias for packetsDiscarded
+  unsigned long packetsDiscarded;
+};
+
+dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats {
+  required DOMString trackIdentifier;
+  DOMString remoteId;
+  unsigned long framesDecoded;
+  unsigned long framesDropped;
+  unsigned long frameWidth;
+  unsigned long frameHeight;
+  double framesPerSecond;
+  unsigned long long qpSum;
+  double totalDecodeTime;
+  double totalInterFrameDelay;
+  double totalSquaredInterFrameDelay;
+  DOMHighResTimeStamp lastPacketReceivedTimestamp;
+  unsigned long long headerBytesReceived;
+  unsigned long long fecPacketsReceived;
+  unsigned long long fecPacketsDiscarded;
+  unsigned long long bytesReceived;
+  unsigned long nackCount;
+  unsigned long firCount;
+  unsigned long pliCount;
+  double totalProcessingDelay;
+  // Always missing from libwebrtc
+  // DOMHighResTimeStamp  estimatedPlayoutTimestamp;
+  double jitterBufferDelay;
+  unsigned long long jitterBufferEmittedCount;
+  unsigned long long totalSamplesReceived;
+  unsigned long long concealedSamples;
+  unsigned long long silentConcealedSamples;
+  unsigned long long concealmentEvents;
+  unsigned long long insertedSamplesForDeceleration;
+  unsigned long long removedSamplesForAcceleration;
+  double audioLevel;
+  double totalAudioEnergy;
+  double totalSamplesDuration;
+  unsigned long framesReceived;
+};
+
+dictionary RTCRemoteInboundRtpStreamStats : RTCReceivedRtpStreamStats {
+  DOMString localId;
+  double roundTripTime;
+  double totalRoundTripTime;
+  double fractionLost;
+  unsigned long long roundTripTimeMeasurements;
+};
+
+dictionary RTCSentRtpStreamStats : RTCRtpStreamStats {
+  unsigned long packetsSent;
+  unsigned long long bytesSent;
+};
+
+dictionary RTCOutboundRtpStreamStats : RTCSentRtpStreamStats {
+  DOMString remoteId;
+  unsigned long framesEncoded;
+  unsigned long long qpSum;
+  unsigned long nackCount;
+  unsigned long firCount;
+  unsigned long pliCount;
+  unsigned long long headerBytesSent;
+  unsigned long long retransmittedPacketsSent;
+  unsigned long long retransmittedBytesSent;
+  unsigned long long totalEncodedBytesTarget;
+  unsigned long frameWidth;
+  unsigned long frameHeight;
+  unsigned long framesSent;
+  unsigned long hugeFramesSent;
+  double totalEncodeTime;
+};
+
+dictionary RTCRemoteOutboundRtpStreamStats : RTCSentRtpStreamStats {
+  DOMString localId;
+  DOMHighResTimeStamp remoteTimestamp;
+};
+
+dictionary RTCMediaSourceStats : RTCStats {
+  required DOMString trackIdentifier;
+  required DOMString kind;
+};
+
+dictionary RTCPeerConnectionStats : RTCStats {
+  unsigned long dataChannelsOpened;
+  unsigned long dataChannelsClosed;
+};
+
+dictionary RTCRTPContributingSourceStats : RTCStats {
+  unsigned long contributorSsrc;
+  DOMString     inboundRtpStreamId;
+};
+
+dictionary RTCDataChannelStats : RTCStats {
+  DOMString           label;
+  DOMString           protocol;
+  long                dataChannelIdentifier;
+  // RTCTransportId is not yet implemented - Bug 1225723
+  // DOMString transportId;
+  RTCDataChannelState state;
+  unsigned long       messagesSent;
+  unsigned long long  bytesSent;
+  unsigned long       messagesReceived;
+  unsigned long long  bytesReceived;
+};
+
+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 RTCIceCandidateType {
+  "host",
+  "srflx",
+  "prflx",
+  "relay"
+};
+
+dictionary RTCIceCandidateStats : RTCStats {
+  DOMString address;
+  long port;
+  DOMString protocol;
+  RTCIceCandidateType candidateType;
+  long priority;
+  DOMString relayProtocol;
+  // Because we use this internally but don't support RTCIceCandidateStats,
+  // we need to keep the field as ChromeOnly. Bug 1225723
+  [ChromeOnly]
+  DOMString transportId;
+  [ChromeOnly]
+  DOMString proxied;
+};
+
+// This is for tracking the frame rate in about:webrtc
+dictionary RTCVideoFrameHistoryEntryInternal {
+  required unsigned long       width;
+  required unsigned long       height;
+  required unsigned long       rotationAngle;
+  required DOMHighResTimeStamp firstFrameTimestamp;
+  required DOMHighResTimeStamp lastFrameTimestamp;
+  required unsigned long long  consecutiveFrames;
+  required unsigned long       localSsrc;
+  required unsigned long       remoteSsrc;
+};
+
+// Collection over the entries for a single track for about:webrtc
+dictionary RTCVideoFrameHistoryInternal {
+  required DOMString                          trackIdentifier;
+  sequence<RTCVideoFrameHistoryEntryInternal> entries = [];
+};
+
+// Collection over the libwebrtc bandwidth estimation stats
+dictionary RTCBandwidthEstimationInternal {
+  required DOMString  trackIdentifier;
+  long                sendBandwidthBps;    // Estimated available send bandwidth
+  long                maxPaddingBps;       // Cumulative configured max padding
+  long                receiveBandwidthBps; // Estimated available receive bandwidth
+  long                pacerDelayMs;
+  long                rttMs;
+};
+
+// This is used by about:webrtc to report SDP parsing errors
+dictionary RTCSdpParsingErrorInternal {
+  required unsigned long lineNumber;
+  required DOMString     error;
+};
+
+// This is for tracking the flow of SDP for about:webrtc
+dictionary RTCSdpHistoryEntryInternal {
+  required DOMHighResTimeStamp         timestamp;
+  required boolean                     isLocal;
+  required DOMString                   sdp;
+  sequence<RTCSdpParsingErrorInternal> errors = [];
+};
+
+// This is intended to be a list of dictionaries that inherit from RTCStats
+// (with some raw ICE candidates thrown in). Unfortunately, we cannot simply
+// store a sequence<RTCStats> because of slicing. So, we have to have a
+// separate list for each type. Used in c++ gecko code.
+dictionary RTCStatsCollection {
+  sequence<RTCInboundRtpStreamStats>        inboundRtpStreamStats = [];
+  sequence<RTCOutboundRtpStreamStats>       outboundRtpStreamStats = [];
+  sequence<RTCRemoteInboundRtpStreamStats>  remoteInboundRtpStreamStats = [];
+  sequence<RTCRemoteOutboundRtpStreamStats> remoteOutboundRtpStreamStats = [];
+  sequence<RTCMediaSourceStats>             mediaSourceStats = [];
+  sequence<RTCPeerConnectionStats>          peerConnectionStats = [];
+  sequence<RTCRTPContributingSourceStats>   rtpContributingSourceStats = [];
+  sequence<RTCIceCandidatePairStats>        iceCandidatePairStats = [];
+  sequence<RTCIceCandidateStats>            iceCandidateStats = [];
+  sequence<RTCIceCandidateStats>            trickledIceCandidateStats = [];
+  sequence<RTCDataChannelStats>             dataChannelStats = [];
+  sequence<RTCCodecStats>                   codecStats = [];
+
+  // For internal use only
+  sequence<DOMString>                       rawLocalCandidates = [];
+  sequence<DOMString>                       rawRemoteCandidates = [];
+  sequence<RTCVideoFrameHistoryInternal>    videoFrameHistories = [];
+  sequence<RTCBandwidthEstimationInternal>  bandwidthEstimations = [];
+};
+
+// Details that about:webrtc can display about configured ICE servers
+dictionary RTCIceServerInternal {
+  sequence<DOMString> urls = [];
+  required boolean    credentialProvided;
+  required boolean    userNameProvided;
+};
+
+// Details that about:webrtc can display about the RTCConfiguration
+// Chrome only
+dictionary RTCConfigurationInternal {
+  RTCBundlePolicy                bundlePolicy;
+  required boolean               certificatesProvided;
+  sequence<RTCIceServerInternal> iceServers = [];
+  RTCIceTransportPolicy          iceTransportPolicy;
+  required boolean               peerIdentityProvided;
+  DOMString                      sdpSemantics;
+};
+
+dictionary RTCSdpHistoryInternal {
+  required DOMString pcid;
+  sequence<RTCSdpHistoryEntryInternal> sdpHistory = [];
+};
+
+// A collection of RTCStats dictionaries, plus some other info. Used by
+// WebrtcGlobalInformation for about:webrtc, and telemetry.
+dictionary RTCStatsReportInternal : RTCStatsCollection {
+  required DOMString                        pcid;
+  required unsigned long                    browserId;
+  RTCConfigurationInternal                  configuration;
+  DOMString                                 jsepSessionErrors;
+  // TODO demux from RTCStatsReportInternal in bug 1830824
+  sequence<RTCSdpHistoryEntryInternal>      sdpHistory = [];
+  required DOMHighResTimeStamp              timestamp;
+  double                                    callDurationMs;
+  required unsigned long                    iceRestarts;
+  required unsigned long                    iceRollbacks;
+  boolean                                   offerer; // Is the PC the offerer
+  required boolean                          closed; // Is the PC now closed
+};
+
+[Pref="media.peerconnection.enabled",
+ Exposed=Window]
+interface RTCStatsReport {
+
+  // TODO(bug 1586109): Remove this once we no longer need to be able to
+  // construct empty RTCStatsReports from JS.
+  [ChromeOnly]
+  constructor();
+
+  readonly maplike<DOMString, object>;
+};