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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /dom/webidl/RTCStatsReport.webidl
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/webidl/RTCStatsReport.webidl')
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diff --git a/dom/webidl/RTCStatsReport.webidl b/dom/webidl/RTCStatsReport.webidl
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+/* -*- 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>;
+};