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mangle.js
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mangle.js
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// using shiftwidth=2 since reformating all the adapter code...
function mangle(stats) {
// Mangle chrome stats to spec stats. Just chrome stats.
var needsMangling = false;
Object.keys(stats).forEach(function(id) {
if (stats[id].type === 'googComponent') {
needsMangling = true;
}
});
if (!needsMangling) {
return stats;
}
// taken from https://github.com/fippo/adapter/tree/getstats-mangling
var standardReport = {};
Object.keys(stats).forEach(function(id) {
if (!stats[id].id) {
stats[id].id = id; // backfill, might have been removed by client.
}
var standardStats = stats[id];
// Step 1: translate to standard types and attribute names.
switch (standardStats.type) {
case 'ssrc':
standardStats.trackIdentifier = standardStats.googTrackId;
// FIXME: not defined in spec, probably whether the track is
// remote?
standardStats.remoteSource =
standardStats.id.indexOf('recv') !== -1;
standardStats.ssrc = parseInt(standardStats.ssrc, 10);
/* done in client since it needs access to PC
if (!standardStats.mediaType && standardStats.googTrackId) {
// look up track kind in local or remote streams.
var streams = standardStats.remoteSource ?
pc.getRemoteStreams() : pc.getLocalStreams();
for (var i = 0; i < streams.length && !standardStats.mediaType;
i++) {
var tracks = streams[i].getTracks();
for (var j = 0; j < tracks.length; j++) {
if (tracks[j].id === standardStats.googTrackId) {
standardStats.mediaType = tracks[j].kind;
}
}
}
}
*/
// FIXME: 'only makes sense' <=> not set?
if (standardStats.googFrameWidthReceived ||
standardStats.googFrameWidthSent) {
standardStats.frameWidth = parseInt(
standardStats.googFrameWidthReceived ||
standardStats.googFrameWidthSent, 10);
}
if (standardStats.googFrameHeightReceived ||
standardStats.googFrameHeightSent) {
standardStats.frameHeight = parseInt(
standardStats.googFrameHeightReceived ||
standardStats.googFrameHeightSent, 10);
}
if (standardStats.googFrameRateInput ||
standardStats.googFrameRateReceived) {
// FIXME: might be something else not available currently
standardStats.framesPerSecond = parseInt(
standardStats.googFrameRateInput ||
standardStats.googFrameRateReceived, 10);
}
/* FIXME unfortunately the current stats (googFrameRateSent,
* googFrameRateReceived, googFrameRateDecoded) so we can not
* calculate the cumulative amount.
* FIXME (spec) Firefox has frameRateMean why is this
* not part of the spec?
*/
if (standardStats.googFrameRateSent) {
standardStats.framesSent = 0;
}
if (standardStats.googFrameRateReceived) {
standardStats.framesReceived = 0;
}
if (standardStats.googFrameRateDecoded) {
standardStats.framesDecoded = 0;
}
// FIXME: both on sender and receiver?
if (standardStats.mediaType === 'video') {
standardStats.framesDropped = 0;
}
if (standardStats.audioInputLevel ||
standardStats.audioOutputLevel) {
standardStats.audioLevel = parseInt(
standardStats.audioInputLevel ||
standardStats.audioOutputLevel, 10) / 32767.0;
}
if (standardStats.googJitterReceived) {
standardStats.jitter = 1.0 * parseInt(
standardStats.googJitterReceived, 10);
}
// FIXME: fractionLost
if (standardStats.googFirsReceived || standardStats.googFirsSent) {
standardStats.firCount = parseInt(
standardStats.googFirsReceived ||
standardStats.googFirsSent, 10);
}
if (standardStats.googPlisReceived || standardStats.googPlisSent) {
standardStats.pliCount = parseInt(
standardStats.googPlisReceived ||
standardStats.googPlisSent, 10);
}
if (standardStats.googNacksReceived ||
standardStats.googNacksSent) {
standardStats.nackCount = parseInt(
standardStats.googNacksReceived ||
standardStats.googNacksSent, 10);
}
// FIXME: no SLI stats yet?
if (standardStats.bytesSent) {
standardStats.bytesSent = parseInt(standardStats.bytesSent, 10);
}
if (standardStats.bytesReceived) {
standardStats.bytesReceived = parseInt(
standardStats.bytesReceived, 10);
}
if (standardStats.packetsSent) {
standardStats.packetsSent = parseInt(
standardStats.packetsSent, 10);
}
if (standardStats.packetsReceived) {
standardStats.packetsReceived = parseInt(
standardStats.packetsReceived, 10);
}
if (standardStats.packetsLost) {
standardStats.packetsLost = parseInt(
standardStats.packetsLost, 10);
}
if (standardStats.googEchoCancellationReturnLoss) {
standardStats.echoReturnLoss = 1.0 * parseInt(
standardStats.googEchoCancellationReturnLoss, 10);
standardStats.echoReturnLossEnhancement = 1.0 * parseInt(
standardStats.googEchoCancellationReturnLossEnhancement, 10);
}
if (standardStats.googRtt) {
// This is the RTCP RTT.
standardStats.roundTripTime = parseInt(standardStats.googRtt, 10);
}
if (standardStats.googEncodeUsagePercent) {
standardStats.googEncodeUsagePercent = parseInt(standardStats.googEncodeUsagePercent, 10);
}
break;
case 'localcandidate':
case 'remotecandidate':
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
standardStats.portNumber = parseInt(standardStats.portNumber, 10);
standardStats.priority = parseInt(standardStats.priority, 10);
// FIXME: addressSourceUrl?
// FIXME: https://github.com/w3c/webrtc-stats/issues/12
break;
case 'googCandidatePair':
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
standardStats.transportId = standardStats.googChannelId;
// FIXME: maybe set depending on iceconnectionstate and read/write?
//standardStats.state = 'FIXME'; // enum
// FIXME: could be calculated from candidate priorities and role.
//standardStats.priority = 'FIXME'; // unsigned long long
standardStats.writable = standardStats.googWritable === 'true';
standardStats.readable = standardStats.googReadable === 'true';
// assumption: nominated is readable and writeable.
standardStats.nominated = standardStats.readable &&
standardStats.writable;
// FIXME: missing from spec
standardStats.selected =
standardStats.googActiveConnection === 'true';
standardStats.bytesSent = parseInt(standardStats.bytesSent, 10);
standardStats.bytesReceived = parseInt(
standardStats.bytesReceived, 10);
// FIXME: packetsSent is not in spec?
// FIXME: no packetsReceived?
standardStats.packetsSent = parseInt(
standardStats.packetsSent, 10);
standardStats.packetsDiscardedOnSend = parseInt(
standardStats.packetsDiscardedOnSend, 10);
// This is the STUN RTT.
standardStats.roundTripTime = parseInt(standardStats.googRtt, 10);
// backfilled later from videoBWE.
standardStats.availableOutgoingBitrate = 0.0;
standardStats.availableIncomingBitrate = 0.0;
standardStats.requestsSent >>>= 0;
standardStats.responsesReceived >>>= 0;
standardStats.requestsReceived >>>= 0;
standardStats.responsesSent >>>= 0;
standardStats.consentRequestsSent >>>= 0;
break;
case 'googComponent':
// additional RTCTransportStats created later since we
// want the normalized fields and complete snowball.
break;
case 'googCertificate':
standardStats.type = 'certificate'; // FIXME spec: undefined in spec.
standardStats.fingerprint = standardStats.googFingerprint;
standardStats.fingerprintAlgorithm =
standardStats.googFingerprintAlgorithm;
standardStats.base64Certificate = standardStats.googDerBase64;
standardStats.issuerCertificateId = null; // FIXME spec: undefined what 'no issuer' is.
break;
case 'VideoBwe':
standardStats.availableOutgoingBitrate = 1.0 *
parseInt(standardStats.googAvailableSendBandwidth, 10);
standardStats.availableIncomingBitrate = 1.0 *
parseInt(standardStats.googAvailableReceiveBandwidth, 10);
// not really standard...
standardStats.googBucketDelay = parseInt(standardStats.googBucketDelay, 10);
standardStats.googTransmitBitrate = parseInt(standardStats.googTransmitBitrate, 10);
standardStats.googRetransmitBitrate = parseInt(standardStats.googRetransmitBitrate, 10);
standardStats.googTargetEncBitrate = parseInt(standardStats.googTargetEncBitrate, 10);
break;
case 'rtcstats-device-report':
standardStats.batteryLevel = parseInt(standardStats.batteryLevel, 10);
break;
default:
break;
}
standardReport[standardStats.id] = standardStats;
});
// Step 2: fix things spanning multiple reports.
Object.keys(standardReport).forEach(function(id) {
var report = standardReport[id];
var other, newId, sdp;
switch (report.type) {
case 'googCandidatePair':
report.type = 'candidatepair';
if (standardReport.bweforvideo) {
report.availableOutgoingBitrate =
standardReport.bweforvideo.availableOutgoingBitrate;
report.availableIncomingBitrate =
standardReport.bweforvideo.availableIncomingBitrate;
standardReport[report.id] = report;
}
break;
case 'googComponent':
// create a new report since we don't carry over all fields.
other = standardReport[report.selectedCandidatePairId];
newId = 'transport_' + report.id;
standardReport[newId] = {
type: 'transport',
timestamp: report.timestamp,
id: newId,
bytesSent: other && other.bytesSent || 0,
bytesReceived: other && other.bytesReceived || 0,
// FIXME (spec): rtpcpTransportStatsId: rtcp-mux is required so...
activeConnection: other && other.selected,
selectedCandidatePairId: report.selectedCandidatePairId,
localCertificateId: report.localCertificateId,
remoteCertificateId: report.remoteCertificateId
};
break;
case 'ssrc':
newId = 'rtpstream_' + report.id;
// Workaround for https://code.google.com/p/webrtc/issues/detail?id=4808 (fixed in M46)
/* it is not apparently. This can be set to the empty string.
if (!report.googCodecName) {
report.googCodecName = 'VP8';
}
*/
standardReport[newId] = {
//type: 'notastandalonething',
timestamp: report.timestamp,
id: newId,
ssrc: report.ssrc,
mediaType: report.mediaType,
associateStatsId: 'rtcpstream_' + report.id,
isRemote: false,
mediaTrackId: 'mediatrack_' + report.id,
transportId: report.transportId,
};
if (report.googCodecName && report.googCodecName.length) {
standardReport.codecId = 'codec_' + report.googCodecName
}
if (report.mediaType === 'video') {
standardReport[newId].firCount = report.firCount;
standardReport[newId].pliCount = report.pliCount;
standardReport[newId].nackCount = report.nackCount;
standardReport[newId].sliCount = report.sliCount; // undefined yet
}
if (report.remoteSource) {
standardReport[newId].type = 'inboundrtp';
standardReport[newId].packetsReceived = report.packetsReceived;
standardReport[newId].bytesReceived = report.bytesReceived;
standardReport[newId].packetsLost = report.packetsLost;
} else {
standardReport[newId].type = 'outboundrtp';
standardReport[newId].packetsSent = report.packetsSent;
standardReport[newId].bytesSent = report.bytesSent;
standardReport[newId].roundTripTime = report.roundTripTime;
// TODO: targetBitrate
}
// FIXME: this is slightly more complicated. inboundrtp can have packetlost
// but so can outboundrtp via rtcp (isRemote = true)
// need to unmux with opposite type and put loss into remote report.
newId = 'rtcpstream_' + report.id;
standardReport[newId] = {
//type: 'notastandalonething',
timestamp: report.timestamp,
id: newId,
ssrc: report.ssrc,
associateStatsId: 'rtpstream_' + report.id,
isRemote: true,
mediaTrackId: 'mediatrack_' + report.id,
transportId: report.transportId,
codecId: 'codec_' + report.googCodecName
};
if (report.remoteSource) {
standardReport[newId].type = 'outboundrtp';
standardReport[newId].packetsSent = report.packetsSent;
standardReport[newId].bytesSent = report.bytesSent;
standardReport[newId].roundTripTime = report.roundTripTime;
standardReport[newId].packetsLost = report.packetsLost;
} else {
standardReport[newId].type = 'inboundrtp';
standardReport[newId].packetsReceived = report.packetsReceived;
standardReport[newId].bytesReceived = report.bytesReceived;
standardReport[newId].packetsLost = report.packetsLost;
}
// FIXME: one of these is not set?
if (report.jitter) {
standardReport[newId].jitter = report.jitter;
}
newId = 'mediatrack_' + report.id;
standardReport[newId] = {
type: 'track',
timestamp: report.timestamp,
id: newId,
trackIdentifier: report.trackIdentifier,
remoteSource: report.remoteSource,
ssrcIds: ['rtpstream_' + report.id, 'rtcpstream_' + report.id]
};
if (report.mediaType === 'audio') {
standardReport[newId].audioLevel = report.audioLevel;
if (report.id.indexOf('send') !== -1) {
standardReport[newId].echoReturnLoss = report.echoReturnLoss;
standardReport[newId].echoReturnLossEnhancement =
report.echoReturnLossEnhancement;
}
} else if (report.mediaType === 'video') {
standardReport[newId].frameWidth = report.frameWidth;
standardReport[newId].frameHeight = report.frameHeight;
standardReport[newId].framesPerSecond = report.framesPerSecond;
if (report.remoteSource) {
standardReport[newId].framesReceived = report.framesReceived;
standardReport[newId].framesDecoded = report.framesDecoded;
standardReport[newId].framesDropped = report.framesDropped;
standardReport[newId].framesCorrupted = report.framesCorrupted;
} else {
standardReport[newId].framesSent = report.framesSent;
}
}
// We have one codec item per codec name.
// This might be wrong (in theory) since with unified plan
// we can have multiple m-lines and codecs and different
// payload types/parameters but unified is not supported yet.
/*
if (!standardReport['codec_' + report.googCodecName]) {
// determine payload type (from offer) and negotiated (?spec)
// parameters (from answer). (parameters not negotiated yet)
if (pc.localDescription &&
pc.localDescription.type === 'offer') {
sdp = pc.localDescription.sdp;
} else if (pc.remoteDescription &&
pc.remoteDescription.type === 'offer') {
sdp = pc.remoteDescription.sdp;
}
if (sdp) {
// TODO: use a SDP library instead of this regexp-stringsoup approach.
var match = sdp.match(new RegExp('a=rtpmap:(\\d+) ' +
report.googCodecName + '\\/(\\d+)(?:\\/(\\d+))?'));
if (match) {
newId = 'codec_' + report.id;
standardReport[newId] = {
type: 'codec', // FIXME (spec)
timestamp: report.timestamp,
id: newId,
codec: report.googCodecName,
payloadType: parseInt(match[1], 10),
clockRate: parseInt(match[2], 10),
channels: parseInt(match[3] || '1', 10),
parameters: ''
};
}
}
}
*/
break;
default:
break;
}
});
// Step 3: fiddle the transport in between transport and rtp stream
Object.keys(standardReport).forEach(function(id) {
var report = standardReport[id];
if (report.type === 'transprort') {
// RTCTransport has a pointer to the selectedCandidatePair...
var other = standardReport[report.selectedCandidatePairId];
if (other) {
other.transportId = report.id;
}
// but no pointers to the rtpstreams running over it?!
// instead, we rely on having added 'transport_'
Object.keys(standardReport).forEach(function(otherid) {
other = standardReport[otherid];
if ((other.type === 'inboundrtp' ||
other.type === 'outboundrtp') &&
report.id === 'transport_' + other.transportId) {
other.transportId = report.id;
}
});
}
});
return standardReport;
}