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1.1.20
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Yiming Wang committed Jun 29, 2023
1 parent a2f95c3 commit bd8764d
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2 changes: 1 addition & 1 deletion package.json
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Original file line number Diff line number Diff line change
@@ -1,6 +1,6 @@
{
"name": "connect-rtc-js",
"version": "1.1.19",
"version": "1.1.20",
"description": "Amazon Connect softphone library",
"license": "Apache-2.0",
"main": "./src/js/connect-rtc.js",
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160 changes: 41 additions & 119 deletions release/connect-rtc-debug.js
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158 changes: 40 additions & 118 deletions release/connect-rtc.js
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Original file line number Diff line number Diff line change
Expand Up @@ -10035,50 +10035,26 @@ var RtcSession = function () {

case 11:
_context2.next = 13;
return Promise.all(tracks.map(function () {
var _ref3 = (0, _asyncToGenerator3.default)( /*#__PURE__*/_regenerator2.default.mark(function _callee(track) {
var self;
return _regenerator2.default.wrap(function _callee$(_context) {
while (1) {
switch (_context.prev = _context.next) {
case 0:
if (!_this11._legacyStatsReportSupport) {
_context.next = 5;
break;
return Promise.all(tracks.map((0, _asyncToGenerator3.default)( /*#__PURE__*/_regenerator2.default.mark(function _callee() {
return _regenerator2.default.wrap(function _callee$(_context) {
while (1) {
switch (_context.prev = _context.next) {
case 0:
return _context.abrupt('return', _this11._pc.getStats().then(function (rawStats) {
var digestedStats = (0, _rtpStats.extractMediaStatsFromStats)(timestamp, rawStats, streamType);
if (!digestedStats) {
throw new Error('Failed to extract MediaRtpStats from RTCStatsReport for stream type ' + streamType);
}
return digestedStats;
}));

self = _this11;
return _context.abrupt('return', new Promise(function (resolve, reject) {
self._pc.getStats(function (rawStats) {
var digestedStats = (0, _rtpStats.extractMediaStatsFromStats)(timestamp, rawStats.result(), streamType);
if (!digestedStats) {
reject(new Error('Failed to extract MediaRtpStats from RTCStatsReport for stream type ' + streamType));
}
resolve(digestedStats);
}, track);
}));

case 5:
return _context.abrupt('return', _this11._pc.getStats().then(function (rawStats) {
var digestedStats = (0, _rtpStats.extractMediaStatsFromStats)(timestamp, rawStats, streamType);
if (!digestedStats) {
throw new Error('Failed to extract MediaRtpStats from RTCStatsReport for stream type ' + streamType);
}
return digestedStats;
}));

case 6:
case 'end':
return _context.stop();
}
case 1:
case 'end':
return _context.stop();
}
}, _callee, _this11);
}));

return function (_x3) {
return _ref3.apply(this, arguments);
};
}()));
}
}, _callee, _this11);
}))));

case 13:
return _context2.abrupt('return', _context2.sent);
Expand Down Expand Up @@ -10733,92 +10709,38 @@ function _interopRequireDefault(obj) { return obj && obj.__esModule ? obj : { de
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')) {
if (statsReport.type === 'inbound-rtp' && streamType === 'audio_input') {
// inbound-rtp: Stats for stream from Server to CCP, as seen on the browser
reportType = statsReport.type;
extractedStats = {
timestamp: timestamp,
packetsLost: statsReport.packetsLost,
// packetsCount: number of packet received by CCP, as seen on the browser
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
jbMilliseconds: Math.floor((0, _utils.when_defined)(statsReport.jitter, 0) * 1000),
// Multiplying audioLevel by 32768 aligns its value with the legacy getStats API.
audioLevel: (0, _utils.is_defined)(statsReport.audioLevel) ? Math.floor(statsReport.audioLevel * 32768) : null
};
// 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')) {
} else if (statsReport.type === 'outbound-rtp' && streamType === 'audio_output') {
// outbound-rtp: Stats for stream from CCP to Server, as seen on the browser
extractedStats = extractedStats || {};
// packetsCount: number of packet sent by CCP, as seen on the browser
extractedStats.packetsCount = statsReport.packetsSent;
} else if (statsReport.type === 'media-source' && streamType === 'audio_output') {
extractedStats = extractedStats || {};
// Multiplying audioLevel by 32768 aligns its value with the legacy getStats API.
extractedStats.audioLevel = (0, _utils.is_defined)(statsReport.audioLevel) ? Math.floor(statsReport.audioLevel * 32768) : null;
} else if (statsReport.type === 'remote-inbound-rtp' && streamType === 'audio_output') {
// remote-inbound-rtp: Stats for stream from CCP to Server, as seen on Server side
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
};
extractedStats = extractedStats || {};
extractedStats.timestamp = timestamp;
extractedStats.packetsLost = statsReport.packetsLost;
extractedStats.rttMilliseconds = (0, _utils.is_defined)(statsReport.roundTripTime) ? Math.floor(statsReport.roundTripTime * 1000) : null;
extractedStats.jbMilliseconds = Math.floor((0, _utils.when_defined)(statsReport.jitter, 0) * 1000);
}
}
});
Expand Down Expand Up @@ -11034,7 +10956,7 @@ var SessionReport = exports.SessionReport = function () {
this._noRemoteIceCandidateFailure = null;
this._setRemoteDescriptionFailure = null;
this._streamStats = [];
this._rtcJsVersion = "1.1.19";
this._rtcJsVersion = "1.1.20";
}
/**
*Timestamp when RTCSession started.
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5 changes: 2 additions & 3 deletions release/connect-rtc.min.js
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