in monitoring-stack-cdk/resources/frontend/connect-rtc.js [10445:10539]
function extractMediaStatsFromStats(timestamp, stats, streamType) {
var extractedStats = null;
var reportType = null;
var packetsSent = null;
stats.forEach(function (statsReport) {
if (statsReport) {
if (statsReport.type === 'ssrc') {
reportType = statsReport.type;
// Legacy report. Legacy report names stats with google specific names.
if (parseInt(statsReport.stat('packetsSent')) && statsReport.stat('mediaType') == 'audio' && streamType === 'audio_output') {
extractedStats = {
timestamp: timestamp,
packetsCount: parseInt(statsReport.stat('packetsSent')),
bytesSent: parseInt(statsReport.stat('bytesSent')),
audioLevel: (0, _utils.when_defined)(parseInt(statsReport.stat('audioInputLevel'))),
packetsLost: (0, _utils.is_defined)(statsReport.stat('packetsLost')) ? Math.max(0, statsReport.stat('packetsLost')) : 0,
procMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googCurrentDelayMs'))),
rttMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googRtt'))),
jbMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googJitterReceived')))
};
} else if (parseInt(statsReport.stat('packetsReceived')) && statsReport.stat('mediaType') == 'audio' && streamType === 'audio_input') {
extractedStats = {
timestamp: timestamp,
packetsCount: parseInt(statsReport.stat('packetsReceived')),
bytesReceived: parseInt(statsReport.stat('bytesReceived')),
audioLevel: (0, _utils.when_defined)(parseInt(statsReport.stat('audioOutputLevel'))),
packetsLost: (0, _utils.is_defined)(parseInt(statsReport.stat('packetsLost'))) ? Math.max(0, statsReport.stat('packetsLost')) : 0,
procMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googCurrentDelayMs'))),
jbMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googJitterReceived')))
};
} else if ((0, _utils.is_defined)(statsReport.packetsSent) && statsReport.mediaType == 'video' && streamType === 'video_input') {
extractedStats = {
timestamp: timestamp,
packetsCount: parseInt(statsReport.stat('packetsSent')),
bytesSent: parseInt(statsReport.stat('bytesSent')),
packetsLost: (0, _utils.is_defined)(statsReport.stat('packetsLost')) ? Math.max(0, statsReport.stat('packetsLost')) : 0,
rttMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googRtt'))),
procMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googCurrentDelayMs'))),
frameRateSent: (0, _utils.when_defined)(parseFloat(statsReport.stat('googFrameRateSent')))
};
} else if (typeof statsReport.packetsReceived !== 'undefined' && statsReport.mediaType == 'video' && streamType === 'video_output') {
extractedStats = {
timestamp: timestamp,
packetsCount: parseInt(statsReport.stat('packetsSent')),
bytesReceived: parseInt(statsReport.stat('bytesReceived')),
packetsLost: (0, _utils.is_defined)(parseInt(statsReport.stat('packetsLost'))) ? Math.max(0, statsReport.stat('packetsLost')) : 0,
frameRateReceived: (0, _utils.when_defined)(parseFloat(statsReport.stat('statsReport.googFrameRateReceived'))),
procMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googCurrentDelayMs'))),
jbMilliseconds: (0, _utils.when_defined)(parseInt(statsReport.stat('googJitterReceived')))
};
}
// Standardized report for input stream stats
} else if (statsReport.type === 'inbound-rtp' && !statsReport.isRemote && (streamType === 'audio_input' || streamType === 'video_input')) {
reportType = statsReport.type;
extractedStats = {
timestamp: timestamp,
packetsLost: statsReport.packetsLost,
packetsCount: statsReport.packetsReceived,
jbMilliseconds: (0, _utils.when_defined)(statsReport.jitter, 0) * 1000
};
// Standardized report for packets sent
} else if (statsReport.type === 'outbound-rtp' && !statsReport.isRemote && (streamType === 'audio_output' || streamType === 'video_output')) {
// outbound-rtp report can appear either before or after extractedStats object is created
if (extractedStats && !extractedStats.packetsCount) {
extractedStats.packetsCount = statsReport.packetsSent;
} else {
packetsSent = statsReport.packetsSent;
}
// Standardized report for remaining output stream stats
} else if (statsReport.type === 'remote-inbound-rtp' && (streamType === 'audio_output' || streamType === 'video_output')) {
reportType = statsReport.type;
extractedStats = {
timestamp: timestamp,
packetsLost: statsReport.packetsLost,
packetsCount: packetsSent,
rttMilliseconds: Number.isInteger(statsReport.roundTripTime) ? statsReport.roundTripTime : (0, _utils.is_defined)(statsReport.roundTripTime) ? statsReport.roundTripTime * 1000 : null,
jbMilliseconds: (0, _utils.when_defined)(statsReport.jitter, 0) * 1000
};
// Case for Firefox 65 and below for getting remaining output stream stats
} else if (statsReport.type === 'inbound-rtp' && statsReport.isRemote && (streamType === 'audio_output' || streamType === 'video_output')) {
reportType = statsReport.type;
extractedStats = {
timestamp: timestamp,
packetsLost: statsReport.packetsLost,
packetsCount: packetsSent,
rttMilliseconds: Number.isInteger(statsReport.roundTripTime) ? statsReport.roundTripTime : (0, _utils.is_defined)(statsReport.roundTripTime) ? statsReport.roundTripTime * 1000 : null,
jbMilliseconds: (0, _utils.when_defined)(statsReport.jitter, 0) * 1000
};
}
}
});
return extractedStats ? new MediaRtpStats(extractedStats, reportType, streamType) : null;
}