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// Copyright 2016 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
// https://w3c.github.io/webrtc-pc/#rtcstatsreport-object
[Exposed=Window]
interface RTCStatsReport {
readonly maplike<DOMString, object>;
};
// https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
enum RTCStatsType {
"codec",
"inbound-rtp",
"outbound-rtp",
"remote-inbound-rtp",
"remote-outbound-rtp",
"media-source",
"media-playout",
"peer-connection",
"data-channel",
"transport",
"candidate-pair",
"local-candidate",
"remote-candidate",
"certificate"
};
// https://w3c.github.io/webrtc-pc/#rtciceservertransportprotocol-enum
enum RTCIceServerTransportProtocol {
"udp",
"tcp",
"tls",
};
// https://w3c.github.io/webrtc-pc/#rtcstats-dictionary
dictionary RTCStats {
required DOMHighResTimeStamp timestamp;
required RTCStatsType type;
required DOMString id;
};
// https://www.w3.org/TR/webrtc-stats/#codec-dict*
dictionary RTCCodecStats : RTCStats {
required unsigned long payloadType;
required DOMString transportId;
required DOMString mimeType;
unsigned long clockRate;
unsigned long channels;
DOMString sdpFmtpLine;
};
// https://w3c.github.io/webrtc-stats/#streamstats-dict*
dictionary RTCRtpStreamStats : RTCStats {
required unsigned long ssrc;
required DOMString kind;
DOMString transportId;
DOMString codecId;
// Non-standard and obsolete stats.
DOMString mediaType;
};
// https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
dictionary RTCReceivedRtpStreamStats : RTCRtpStreamStats {
unsigned long long packetsReceived;
long long packetsLost;
double jitter;
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats {
required DOMString trackIdentifier;
DOMString mid;
DOMString remoteId;
unsigned long framesDecoded;
unsigned long keyFramesDecoded;
// Not implemented: unsigned long framesRendered;
unsigned long framesDropped;
unsigned long frameWidth;
unsigned long frameHeight;
double framesPerSecond;
unsigned long long qpSum;
double totalDecodeTime;
double totalInterFrameDelay;
double totalSquaredInterFrameDelay;
unsigned long pauseCount;
double totalPausesDuration;
unsigned long freezeCount;
double totalFreezesDuration;
DOMHighResTimeStamp lastPacketReceivedTimestamp;
unsigned long long headerBytesReceived;
unsigned long long packetsDiscarded;
unsigned long long fecPacketsReceived;
unsigned long long fecPacketsDiscarded;
unsigned long long fecBytesReceived;
unsigned long fecSsrc;
unsigned long long bytesReceived;
unsigned long nackCount;
unsigned long firCount;
unsigned long pliCount;
double totalProcessingDelay;
DOMHighResTimeStamp estimatedPlayoutTimestamp;
double jitterBufferDelay;
double jitterBufferTargetDelay;
unsigned long long jitterBufferEmittedCount;
double jitterBufferMinimumDelay;
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;
DOMString decoderImplementation;
DOMString playoutId;
boolean powerEfficientDecoder;
unsigned long framesAssembledFromMultiplePackets;
double totalAssemblyTime;
// https://w3c.github.io/webrtc-provisional-stats/#RTCInboundRtpStreamStats-dict*
DOMString contentType;
// https://github.com/w3c/webrtc-provisional-stats/issues/40
DOMString googTimingFrameInfo;
unsigned long long retransmittedPacketsReceived;
unsigned long long retransmittedBytesReceived;
unsigned long rtxSsrc;
double totalCorruptionProbability;
double totalSquaredCorruptionProbability;
unsigned long long corruptionMeasurements;
};
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
dictionary RTCRemoteInboundRtpStreamStats : RTCReceivedRtpStreamStats {
DOMString localId;
double roundTripTime;
double totalRoundTripTime;
double fractionLost;
unsigned long long roundTripTimeMeasurements;
};
// https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*
dictionary RTCSentRtpStreamStats : RTCRtpStreamStats {
unsigned long long packetsSent;
unsigned long long bytesSent;
};
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
dictionary RTCOutboundRtpStreamStats : RTCSentRtpStreamStats {
DOMString mid;
DOMString mediaSourceId;
DOMString remoteId;
DOMString rid;
unsigned long encodingIndex;
unsigned long long headerBytesSent;
unsigned long long retransmittedPacketsSent;
unsigned long long retransmittedBytesSent;
unsigned long rtxSsrc;
double targetBitrate;
unsigned long long totalEncodedBytesTarget;
unsigned long frameWidth;
unsigned long frameHeight;
double framesPerSecond;
unsigned long framesSent;
unsigned long hugeFramesSent;
unsigned long framesEncoded;
unsigned long keyFramesEncoded;
unsigned long long qpSum;
double totalEncodeTime;
double totalPacketSendDelay;
RTCQualityLimitationReason qualityLimitationReason;
record<DOMString, double> qualityLimitationDurations;
unsigned long qualityLimitationResolutionChanges;
unsigned long nackCount;
unsigned long firCount;
unsigned long pliCount;
DOMString encoderImplementation;
boolean powerEfficientEncoder;
boolean active;
DOMString scalabilityMode;
// https://w3c.github.io/webrtc-provisional-stats/#RTCOutboundRtpStreamStats-dict*
DOMString contentType;
};
// https://w3c.github.io/webrtc-stats/#rtcqualitylimitationreason-enum
enum RTCQualityLimitationReason {
"none",
"cpu",
"bandwidth",
"other",
};
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
dictionary RTCRemoteOutboundRtpStreamStats : RTCSentRtpStreamStats {
DOMString localId;
DOMHighResTimeStamp remoteTimestamp;
unsigned long long reportsSent;
double roundTripTime;
double totalRoundTripTime;
unsigned long long roundTripTimeMeasurements;
};
// https://w3c.github.io/webrtc-stats/#mediasourcestats-dict*
dictionary RTCMediaSourceStats : RTCStats {
required DOMString trackIdentifier;
required DOMString kind;
};
// https://w3c.github.io/webrtc-stats/#audiosourcestats-dict*
dictionary RTCAudioSourceStats : RTCMediaSourceStats {
double audioLevel;
double totalAudioEnergy;
double totalSamplesDuration;
double echoReturnLoss;
double echoReturnLossEnhancement;
// Not implemented: double droppedSamplesDuration;
// Not implemented: unsigned long droppedSamplesEvents;
// Not implemented: double totalCaptureDelay;
// Not implemented: unsigned long long totalSamplesCaptured;
};
// https://w3c.github.io/webrtc-stats/#videosourcestats-dict*
dictionary RTCVideoSourceStats : RTCMediaSourceStats {
unsigned long width;
unsigned long height;
unsigned long frames;
double framesPerSecond;
};
// https://w3c.github.io/webrtc-stats/#playoutstats-dict*
dictionary RTCAudioPlayoutStats : RTCStats {
required DOMString kind;
double synthesizedSamplesDuration;
unsigned long synthesizedSamplesEvents;
double totalSamplesDuration;
double totalPlayoutDelay;
unsigned long long totalSamplesCount;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
dictionary RTCPeerConnectionStats : RTCStats {
unsigned long dataChannelsOpened;
unsigned long dataChannelsClosed;
};
// https://w3c.github.io/webrtc-stats/#dcstats-dict*
dictionary RTCDataChannelStats : RTCStats {
DOMString label;
DOMString protocol;
unsigned short dataChannelIdentifier;
required RTCDataChannelState state;
unsigned long messagesSent;
unsigned long long bytesSent;
unsigned long messagesReceived;
unsigned long long bytesReceived;
};
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
dictionary RTCTransportStats : RTCStats {
unsigned long long packetsSent;
unsigned long long packetsReceived;
unsigned long long bytesSent;
unsigned long long bytesReceived;
RTCIceRole iceRole;
DOMString iceLocalUsernameFragment;
required RTCDtlsTransportState dtlsState;
RTCIceTransportState iceState;
DOMString selectedCandidatePairId;
DOMString localCertificateId;
DOMString remoteCertificateId;
DOMString tlsVersion;
DOMString dtlsCipher;
RTCDtlsRole dtlsRole;
DOMString srtpCipher;
unsigned long selectedCandidatePairChanges;
// https://w3c.github.io/webrtc-provisional-stats/#RTCTransportStats-dict*
DOMString rtcpTransportStatsId;
};
// https://w3c.github.io/webrtc-stats/#rtcdtlsrole-enum
enum RTCDtlsRole {
"client",
"server",
"unknown",
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
dictionary RTCIceCandidateStats : RTCStats {
required DOMString transportId;
DOMString? address;
long port;
DOMString protocol;
required RTCIceCandidateType candidateType;
long priority;
DOMString url;
RTCIceServerTransportProtocol relayProtocol;
DOMString foundation;
DOMString relatedAddress;
long relatedPort;
DOMString usernameFragment;
RTCIceTcpCandidateType tcpType;
// https://w3c.github.io/webrtc-provisional-stats/#RTCIceCandidateStats-stat*
RTCNetworkType networkType;
// Non-standard and obsolete stats.
// - Removed because `type` reveals same information ("local-candidate" or
// "remote-candidate").
boolean isRemote;
// - Removed because it was renamed `address`.
DOMString? ip;
};
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
dictionary RTCIceCandidatePairStats : RTCStats {
required DOMString transportId;
required DOMString localCandidateId;
required DOMString remoteCandidateId;
required RTCStatsIceCandidatePairState state;
boolean nominated;
unsigned long long packetsSent;
unsigned long long packetsReceived;
unsigned long long bytesSent;
unsigned long long bytesReceived;
DOMHighResTimeStamp lastPacketSentTimestamp;
DOMHighResTimeStamp lastPacketReceivedTimestamp;
double totalRoundTripTime;
double currentRoundTripTime;
double availableOutgoingBitrate;
double availableIncomingBitrate;
unsigned long long requestsReceived;
unsigned long long requestsSent;
unsigned long long responsesReceived;
unsigned long long responsesSent;
unsigned long long consentRequestsSent;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
// Non-standard and obsolete stats.
boolean writable;
unsigned long long priority;
};
// https://w3c.github.io/webrtc-stats/#rtcstatsicecandidatepairstate-enum
enum RTCStatsIceCandidatePairState {
"frozen",
"waiting",
"in-progress",
"failed",
"succeeded"
};
// https://w3c.github.io/webrtc-stats/#certificatestats-dict*
dictionary RTCCertificateStats : RTCStats {
required DOMString fingerprint;
required DOMString fingerprintAlgorithm;
required DOMString base64Certificate;
DOMString issuerCertificateId;
};
// https://w3c.github.io/webrtc-provisional-stats/#rtcnetworktype-enum
enum RTCNetworkType {
"bluetooth",
"cellular",
"ethernet",
"wifi",
"wimax",
"vpn",
"unknown"
};
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