@webex/plugin-meetings 3.8.0-next.79 → 3.8.0-next.80

package/dist/breakouts/breakout.js

@@ -209,7 +209,7 @@ var Breakout = _webexCore.WebexPlugin.extend({
       sessionId: this.sessionId
     });
   },
-  version: "3.8.0-next.79"
+  version: "3.8.0-next.80"
 });
 var _default = exports.default = Breakout;
 //# sourceMappingURL=breakout.js.map

package/dist/breakouts/index.js

@@ -1110,7 +1110,7 @@ var Breakouts = _webexCore.WebexPlugin.extend({
       this.trigger(_constants.BREAKOUTS.EVENTS.ASK_RETURN_TO_MAIN);
     }
   },
-  version: "3.8.0-next.79"
+  version: "3.8.0-next.80"
 });
 var _default = exports.default = Breakouts;
 //# sourceMappingURL=index.js.map

package/dist/config.js

@@ -108,7 +108,8 @@ var _default = exports.default = {
     reachabilityGetClusterTimeout: 5000,
     logUploadIntervalMultiplicationFactor: 0,
     // if set to 0 or undefined, logs won't be uploaded periodically, if you want periodic logs, recommended value is 1
-    stopIceGatheringAfterFirstRelayCandidate: false
+    stopIceGatheringAfterFirstRelayCandidate: false,
+    enableAudioTwccForMultistream: false
   }
 };
 //# sourceMappingURL=config.js.map

package/dist/config.js.map

@@ -1 +1 @@
-{"version":3,"names":["_common","require","_default","exports","default","meetings","deviceType","WEB","mediaSettings","pstn","reconnection","enabled","autoRejoin","detection","iceReconnectionTimeout","maxRejoinAttempts","stats","enableStatsAnalyzer","autoDowngradeEnabled","interval","analyzerInterval","historyMax","videoPacketLossRatioThreshold","rttThreshold","jitterThreshold","metrics","clientName","clientType","subClientType","mqaMetricsInterval","autoSendMQA","logging","enable","verboseEvents","resolution","maxWidth","maxHeight","idealWidth","idealHeight","screenResolution","bandwidth","audio","video","startBitrate","screenFrameRate","videoShareFrameRate","aspectRatio","autoUploadLogs","enableRtx","receiveTranscription","enableExtmap","enableAutomaticLLM","installedOrgID","undefined","experimental","enableMediaNegotiatedEvent","enableUnifiedMeetings","enableAdhocMeetings","enableTcpReachability","enableTlsReachability","degradationPreferences","maxMacroblocksLimit","iceCandidatesGatheringTimeout","backendIpv6NativeSupport","enableReachabilityChecks","reachabilityGetClusterTimeout","logUploadIntervalMultiplicationFactor","stopIceGatheringAfterFirstRelayCandidate"],"sources":["config.ts"],"sourcesContent":["// @ts-ignore\nimport {deviceType} from '@webex/common';\n\nexport default {\n  // TODO: this needs to be defaulted\n  // to JS_SDK and clients set it to WEB or others\n  // see https://jira-eng-gpk2.cisco.com/jira/browse/SPARK-73167\n  meetings: {\n    deviceType: deviceType.WEB,\n    mediaSettings: {\n      pstn: false,\n    },\n    reconnection: {\n      enabled: false,\n      autoRejoin: true,\n      detection: true,\n      // Timeout duration to wait for ICE to reconnect if a disconnect is received.\n      iceReconnectionTimeout: 10000,\n      // Amount of times attempting to rejoin a meeting during reconnect\n      maxRejoinAttempts: 3,\n    },\n    stats: {\n      // Enable the webrtc stats analyzer that emits quality degradation events\n      enableStatsAnalyzer: true,\n      // Enable the auto downgrade video quality feature\n      autoDowngradeEnabled: false,\n      // 1 second intervals to collect stats data\n      interval: 1000,\n      // we just want to analyze data every 5 sec interval\n      analyzerInterval: 5000,\n      // hold the last 2 minute of a calls data\n      historyMax: 120,\n      // Once packet loss hits this ratio, the video will downgrade\n      videoPacketLossRatioThreshold: 9, // comparison of packets lost / packets received\n      rttThreshold: 500, // 500 ms noticeable quality lag begins based on bandwidth of user\n      jitterThreshold: 500, // 500 ms noticeable quality lag begins based on bandwidth of user\n    },\n    metrics: {\n      // change to your client name else data will be muddled\n      // you do not need a specific format, and you do not need to register it\n      clientName: 'WEBEX_JS_SDK',\n      // TODO: for now this line has to be whitelisted, which is problematic for third party\n      clientType: 'WEBEX_SDK',\n      // Stores the sub client type used when sending metrics\n      subClientType: 'WEB_APP',\n      // send average values MQA in 60 second intervals\n      mqaMetricsInterval: 60000,\n      // send to cisco internal MQA data automatically\n      // we already send CA\n      autoSendMQA: true,\n    },\n    logging: {\n      enable: true,\n      verboseEvents: true,\n    },\n    resolution: {\n      maxWidth: 1280,\n      maxHeight: 720,\n      idealWidth: 1280,\n      idealHeight: 720,\n    },\n    screenResolution: {\n      maxWidth: 1920,\n      maxHeight: 1080,\n      idealWidth: 1920,\n      idealHeight: 1080,\n    },\n    bandwidth: {\n      // please note, these are the maximum bandwidth values\n      // the server supports, minimums have to be tested\n      audio: 64000,\n      video: 4000000,\n      startBitrate: 2000,\n    },\n    screenFrameRate: 10,\n    videoShareFrameRate: 30,\n    aspectRatio: 1.7695852534562213,\n    // When enabled, as calls are ended, it will upload the SDK logs and correlate them\n    autoUploadLogs: true,\n    enableRtx: true,\n    receiveTranscription: false,\n    enableExtmap: false,\n    enableAutomaticLLM: false,\n    installedOrgID: undefined,\n    experimental: {\n      enableMediaNegotiatedEvent: false,\n      enableUnifiedMeetings: true,\n      enableAdhocMeetings: true,\n      enableTcpReachability: false,\n      enableTlsReachability: false,\n    },\n    degradationPreferences: {\n      maxMacroblocksLimit: 8192,\n    },\n    // This only applies to non-multistream meetings\n    iceCandidatesGatheringTimeout: undefined,\n    backendIpv6NativeSupport: false,\n    enableReachabilityChecks: true,\n    reachabilityGetClusterTimeout: 5000,\n    logUploadIntervalMultiplicationFactor: 0, // if set to 0 or undefined, logs won't be uploaded periodically, if you want periodic logs, recommended value is 1\n    stopIceGatheringAfterFirstRelayCandidate: false,\n  },\n};\n"],"mappings":";;;;;;;AACA,IAAAA,OAAA,GAAAC,OAAA;AADA;AAAA,IAAAC,QAAA,GAAAC,OAAA,CAAAC,OAAA,GAGe;EACb;EACA;EACA;EACAC,QAAQ,EAAE;IACRC,UAAU,EAAEA,kBAAU,CAACC,GAAG;IAC1BC,aAAa,EAAE;MACbC,IAAI,EAAE;IACR,CAAC;IACDC,YAAY,EAAE;MACZC,OAAO,EAAE,KAAK;MACdC,UAAU,EAAE,IAAI;MAChBC,SAAS,EAAE,IAAI;MACf;MACAC,sBAAsB,EAAE,KAAK;MAC7B;MACAC,iBAAiB,EAAE;IACrB,CAAC;IACDC,KAAK,EAAE;MACL;MACAC,mBAAmB,EAAE,IAAI;MACzB;MACAC,oBAAoB,EAAE,KAAK;MAC3B;MACAC,QAAQ,EAAE,IAAI;MACd;MACAC,gBAAgB,EAAE,IAAI;MACtB;MACAC,UAAU,EAAE,GAAG;MACf;MACAC,6BAA6B,EAAE,CAAC;MAAE;MAClCC,YAAY,EAAE,GAAG;MAAE;MACnBC,eAAe,EAAE,GAAG,CAAE;IACxB,CAAC;;IACDC,OAAO,EAAE;MACP;MACA;MACAC,UAAU,EAAE,cAAc;MAC1B;MACAC,UAAU,EAAE,WAAW;MACvB;MACAC,aAAa,EAAE,SAAS;MACxB;MACAC,kBAAkB,EAAE,KAAK;MACzB;MACA;MACAC,WAAW,EAAE;IACf,CAAC;IACDC,OAAO,EAAE;MACPC,MAAM,EAAE,IAAI;MACZC,aAAa,EAAE;IACjB,CAAC;IACDC,UAAU,EAAE;MACVC,QAAQ,EAAE,IAAI;MACdC,SAAS,EAAE,GAAG;MACdC,UAAU,EAAE,IAAI;MAChBC,WAAW,EAAE;IACf,CAAC;IACDC,gBAAgB,EAAE;MAChBJ,QAAQ,EAAE,IAAI;MACdC,SAAS,EAAE,IAAI;MACfC,UAAU,EAAE,IAAI;MAChBC,WAAW,EAAE;IACf,CAAC;IACDE,SAAS,EAAE;MACT;MACA;MACAC,KAAK,EAAE,KAAK;MACZC,KAAK,EAAE,OAAO;MACdC,YAAY,EAAE;IAChB,CAAC;IACDC,eAAe,EAAE,EAAE;IACnBC,mBAAmB,EAAE,EAAE;IACvBC,WAAW,EAAE,kBAAkB;IAC/B;IACAC,cAAc,EAAE,IAAI;IACpBC,SAAS,EAAE,IAAI;IACfC,oBAAoB,EAAE,KAAK;IAC3BC,YAAY,EAAE,KAAK;IACnBC,kBAAkB,EAAE,KAAK;IACzBC,cAAc,EAAEC,SAAS;IACzBC,YAAY,EAAE;MACZC,0BAA0B,EAAE,KAAK;MACjCC,qBAAqB,EAAE,IAAI;MAC3BC,mBAAmB,EAAE,IAAI;MACzBC,qBAAqB,EAAE,KAAK;MAC5BC,qBAAqB,EAAE;IACzB,CAAC;IACDC,sBAAsB,EAAE;MACtBC,mBAAmB,EAAE;IACvB,CAAC;IACD;IACAC,6BAA6B,EAAET,SAAS;IACxCU,wBAAwB,EAAE,KAAK;IAC/BC,wBAAwB,EAAE,IAAI;IAC9BC,6BAA6B,EAAE,IAAI;IACnCC,qCAAqC,EAAE,CAAC;IAAE;IAC1CC,wCAAwC,EAAE;EAC5C;AACF,CAAC"}
+{"version":3,"names":["_common","require","_default","exports","default","meetings","deviceType","WEB","mediaSettings","pstn","reconnection","enabled","autoRejoin","detection","iceReconnectionTimeout","maxRejoinAttempts","stats","enableStatsAnalyzer","autoDowngradeEnabled","interval","analyzerInterval","historyMax","videoPacketLossRatioThreshold","rttThreshold","jitterThreshold","metrics","clientName","clientType","subClientType","mqaMetricsInterval","autoSendMQA","logging","enable","verboseEvents","resolution","maxWidth","maxHeight","idealWidth","idealHeight","screenResolution","bandwidth","audio","video","startBitrate","screenFrameRate","videoShareFrameRate","aspectRatio","autoUploadLogs","enableRtx","receiveTranscription","enableExtmap","enableAutomaticLLM","installedOrgID","undefined","experimental","enableMediaNegotiatedEvent","enableUnifiedMeetings","enableAdhocMeetings","enableTcpReachability","enableTlsReachability","degradationPreferences","maxMacroblocksLimit","iceCandidatesGatheringTimeout","backendIpv6NativeSupport","enableReachabilityChecks","reachabilityGetClusterTimeout","logUploadIntervalMultiplicationFactor","stopIceGatheringAfterFirstRelayCandidate","enableAudioTwccForMultistream"],"sources":["config.ts"],"sourcesContent":["// @ts-ignore\nimport {deviceType} from '@webex/common';\n\nexport default {\n  // TODO: this needs to be defaulted\n  // to JS_SDK and clients set it to WEB or others\n  // see https://jira-eng-gpk2.cisco.com/jira/browse/SPARK-73167\n  meetings: {\n    deviceType: deviceType.WEB,\n    mediaSettings: {\n      pstn: false,\n    },\n    reconnection: {\n      enabled: false,\n      autoRejoin: true,\n      detection: true,\n      // Timeout duration to wait for ICE to reconnect if a disconnect is received.\n      iceReconnectionTimeout: 10000,\n      // Amount of times attempting to rejoin a meeting during reconnect\n      maxRejoinAttempts: 3,\n    },\n    stats: {\n      // Enable the webrtc stats analyzer that emits quality degradation events\n      enableStatsAnalyzer: true,\n      // Enable the auto downgrade video quality feature\n      autoDowngradeEnabled: false,\n      // 1 second intervals to collect stats data\n      interval: 1000,\n      // we just want to analyze data every 5 sec interval\n      analyzerInterval: 5000,\n      // hold the last 2 minute of a calls data\n      historyMax: 120,\n      // Once packet loss hits this ratio, the video will downgrade\n      videoPacketLossRatioThreshold: 9, // comparison of packets lost / packets received\n      rttThreshold: 500, // 500 ms noticeable quality lag begins based on bandwidth of user\n      jitterThreshold: 500, // 500 ms noticeable quality lag begins based on bandwidth of user\n    },\n    metrics: {\n      // change to your client name else data will be muddled\n      // you do not need a specific format, and you do not need to register it\n      clientName: 'WEBEX_JS_SDK',\n      // TODO: for now this line has to be whitelisted, which is problematic for third party\n      clientType: 'WEBEX_SDK',\n      // Stores the sub client type used when sending metrics\n      subClientType: 'WEB_APP',\n      // send average values MQA in 60 second intervals\n      mqaMetricsInterval: 60000,\n      // send to cisco internal MQA data automatically\n      // we already send CA\n      autoSendMQA: true,\n    },\n    logging: {\n      enable: true,\n      verboseEvents: true,\n    },\n    resolution: {\n      maxWidth: 1280,\n      maxHeight: 720,\n      idealWidth: 1280,\n      idealHeight: 720,\n    },\n    screenResolution: {\n      maxWidth: 1920,\n      maxHeight: 1080,\n      idealWidth: 1920,\n      idealHeight: 1080,\n    },\n    bandwidth: {\n      // please note, these are the maximum bandwidth values\n      // the server supports, minimums have to be tested\n      audio: 64000,\n      video: 4000000,\n      startBitrate: 2000,\n    },\n    screenFrameRate: 10,\n    videoShareFrameRate: 30,\n    aspectRatio: 1.7695852534562213,\n    // When enabled, as calls are ended, it will upload the SDK logs and correlate them\n    autoUploadLogs: true,\n    enableRtx: true,\n    receiveTranscription: false,\n    enableExtmap: false,\n    enableAutomaticLLM: false,\n    installedOrgID: undefined,\n    experimental: {\n      enableMediaNegotiatedEvent: false,\n      enableUnifiedMeetings: true,\n      enableAdhocMeetings: true,\n      enableTcpReachability: false,\n      enableTlsReachability: false,\n    },\n    degradationPreferences: {\n      maxMacroblocksLimit: 8192,\n    },\n    // This only applies to non-multistream meetings\n    iceCandidatesGatheringTimeout: undefined,\n    backendIpv6NativeSupport: false,\n    enableReachabilityChecks: true,\n    reachabilityGetClusterTimeout: 5000,\n    logUploadIntervalMultiplicationFactor: 0, // if set to 0 or undefined, logs won't be uploaded periodically, if you want periodic logs, recommended value is 1\n    stopIceGatheringAfterFirstRelayCandidate: false,\n    enableAudioTwccForMultistream: false,\n  },\n};\n"],"mappings":";;;;;;;AACA,IAAAA,OAAA,GAAAC,OAAA;AADA;AAAA,IAAAC,QAAA,GAAAC,OAAA,CAAAC,OAAA,GAGe;EACb;EACA;EACA;EACAC,QAAQ,EAAE;IACRC,UAAU,EAAEA,kBAAU,CAACC,GAAG;IAC1BC,aAAa,EAAE;MACbC,IAAI,EAAE;IACR,CAAC;IACDC,YAAY,EAAE;MACZC,OAAO,EAAE,KAAK;MACdC,UAAU,EAAE,IAAI;MAChBC,SAAS,EAAE,IAAI;MACf;MACAC,sBAAsB,EAAE,KAAK;MAC7B;MACAC,iBAAiB,EAAE;IACrB,CAAC;IACDC,KAAK,EAAE;MACL;MACAC,mBAAmB,EAAE,IAAI;MACzB;MACAC,oBAAoB,EAAE,KAAK;MAC3B;MACAC,QAAQ,EAAE,IAAI;MACd;MACAC,gBAAgB,EAAE,IAAI;MACtB;MACAC,UAAU,EAAE,GAAG;MACf;MACAC,6BAA6B,EAAE,CAAC;MAAE;MAClCC,YAAY,EAAE,GAAG;MAAE;MACnBC,eAAe,EAAE,GAAG,CAAE;IACxB,CAAC;;IACDC,OAAO,EAAE;MACP;MACA;MACAC,UAAU,EAAE,cAAc;MAC1B;MACAC,UAAU,EAAE,WAAW;MACvB;MACAC,aAAa,EAAE,SAAS;MACxB;MACAC,kBAAkB,EAAE,KAAK;MACzB;MACA;MACAC,WAAW,EAAE;IACf,CAAC;IACDC,OAAO,EAAE;MACPC,MAAM,EAAE,IAAI;MACZC,aAAa,EAAE;IACjB,CAAC;IACDC,UAAU,EAAE;MACVC,QAAQ,EAAE,IAAI;MACdC,SAAS,EAAE,GAAG;MACdC,UAAU,EAAE,IAAI;MAChBC,WAAW,EAAE;IACf,CAAC;IACDC,gBAAgB,EAAE;MAChBJ,QAAQ,EAAE,IAAI;MACdC,SAAS,EAAE,IAAI;MACfC,UAAU,EAAE,IAAI;MAChBC,WAAW,EAAE;IACf,CAAC;IACDE,SAAS,EAAE;MACT;MACA;MACAC,KAAK,EAAE,KAAK;MACZC,KAAK,EAAE,OAAO;MACdC,YAAY,EAAE;IAChB,CAAC;IACDC,eAAe,EAAE,EAAE;IACnBC,mBAAmB,EAAE,EAAE;IACvBC,WAAW,EAAE,kBAAkB;IAC/B;IACAC,cAAc,EAAE,IAAI;IACpBC,SAAS,EAAE,IAAI;IACfC,oBAAoB,EAAE,KAAK;IAC3BC,YAAY,EAAE,KAAK;IACnBC,kBAAkB,EAAE,KAAK;IACzBC,cAAc,EAAEC,SAAS;IACzBC,YAAY,EAAE;MACZC,0BAA0B,EAAE,KAAK;MACjCC,qBAAqB,EAAE,IAAI;MAC3BC,mBAAmB,EAAE,IAAI;MACzBC,qBAAqB,EAAE,KAAK;MAC5BC,qBAAqB,EAAE;IACzB,CAAC;IACDC,sBAAsB,EAAE;MACtBC,mBAAmB,EAAE;IACvB,CAAC;IACD;IACAC,6BAA6B,EAAET,SAAS;IACxCU,wBAAwB,EAAE,KAAK;IAC/BC,wBAAwB,EAAE,IAAI;IAC9BC,6BAA6B,EAAE,IAAI;IACnCC,qCAAqC,EAAE,CAAC;IAAE;IAC1CC,wCAAwC,EAAE,KAAK;IAC/CC,6BAA6B,EAAE;EACjC;AACF,CAAC"}

package/dist/interpretation/index.js

@@ -373,7 +373,7 @@ var SimultaneousInterpretation = _webexCore.WebexPlugin.extend({
       throw error;
     });
   },
-  version: "3.8.0-next.79"
+  version: "3.8.0-next.80"
 });
 var _default = exports.default = SimultaneousInterpretation;
 //# sourceMappingURL=index.js.map

package/dist/interpretation/siLanguage.js

@@ -18,7 +18,7 @@ var SILanguage = _webexCore.WebexPlugin.extend({
     languageCode: 'number',
     languageName: 'string'
   },
-  version: "3.8.0-next.79"
+  version: "3.8.0-next.80"
 });
 var _default = exports.default = SILanguage;
 //# sourceMappingURL=siLanguage.js.map

package/dist/media/index.js

@@ -120,6 +120,7 @@ Media.createMediaConnection = function (isMultistream, debugId, meetingId, optio
     bundlePolicy = options.bundlePolicy,
     iceCandidatesTimeout = options.iceCandidatesTimeout,
     disableAudioMainDtx = options.disableAudioMainDtx,
+    enableAudioTwcc = options.enableAudioTwcc,
     stopIceGatheringAfterFirstRelayCandidate = options.stopIceGatheringAfterFirstRelayCandidate;
   var iceServers = [];
 
@@ -134,7 +135,8 @@ Media.createMediaConnection = function (isMultistream, debugId, meetingId, optio
   }
   if (isMultistream) {
     var config = {
-      iceServers: iceServers
+      iceServers: iceServers,
+      disableAudioTwcc: !enableAudioTwcc
     };
     if (bundlePolicy) {
       config.bundlePolicy = bundlePolicy;

package/dist/media/index.js.map

@@ -1 +1 @@
-{"version":3,"names":["_internalMediaCore","require","_webCapabilities","_loggerProxy","_interopRequireDefault","_constants","_config","_config2","_browserDetection","ownKeys","e","r","t","_Object$keys","_Object$getOwnPropertySymbols","o","filter","_Object$getOwnPropertyDescriptor","enumerable","push","apply","_objectSpread","arguments","length","Object","forEach","_defineProperty2","default","_Object$getOwnPropertyDescriptors","_Object$defineProperties","_Object$defineProperty","_BrowserDetection","BrowserDetection","isBrowser","Media","getLocalMedia","options","config","sendAudio","sendVideo","sendShare","sharePreferences","isSharing","getMedia","getDisplayMedia","_promise","resolve","undefined","getDirection","forceSendRecv","receive","send","createMediaConnection","isMultistream","debugId","meetingId","_audioStream$outputSt","_videoStream$outputSt","_shareVideoStream$out","_shareAudioStream$out","rtcMetrics","mediaProperties","remoteQualityLevel","enableRtx","enableExtmap","turnServerInfo","bundlePolicy","iceCandidatesTimeout","disableAudioMainDtx","stopIceGatheringAfterFirstRelayCandidate","iceServers","urls","username","credential","password","BrowserInfo","isFirefox","doFullIce","MultistreamRoapMediaConnection","data","addMetrics","closeMetrics","sendMetricsInQueue","Error","mediaDirection","audioStream","videoStream","shareVideoStream","shareAudioStream","RoapMediaConnection","skipInactiveTransceivers","requireH264","sdpMunging","convertPort9to0","addContentSlides","bandwidthLimits","audio","StaticConfig","meetings","bandwidth","video","startBitrate","periodicKeyframes","disableExtmap","disableRtx","localTracks","outputStream","getTracks","screenShareVideo","screenShareAudio","direction","receiveAudio","receiveVideo","receiveShare","customResolution","screenResolution","customShareFrameRate","screenFrameRate","hasSharePreferences","hasCustomConstraints","shareConstraints","hasHighFrameRate","highFrameRate","_Config$meetings","Config","resolution","videoShareFrameRate","aspectRatio","cursor","MEDIA_TRACK_CONSTRAINT","CURSOR","AWLAYS","frameRate","height","idealHeight","width","idealWidth","mediaConfig","navigator","mediaDevices","then","stream","getVideoTracks","applyConstraints","getDisplayMediaParams","defaultWidth","ideal","max","maxWidth","defaultHeight","maxHeight","deviceId","facingMode","fake","process","env","NODE_ENV","getUserMedia","catch","err","logPath","LoggerProxy","logger","error","concat","constraint","toggleStream","stopStream","stop","readyState","mediaSetting","audioVideo","localStream","shareStream","_default","exports"],"sources":["index.ts"],"sourcesContent":["/*!\n * Copyright (c) 2015-2020 Cisco Systems, Inc. See LICENSE file.\n */\n/* globals navigator */\n\nimport {\n  RoapMediaConnection,\n  MultistreamRoapMediaConnection,\n  type MultistreamConnectionConfig,\n} from '@webex/internal-media-core';\nimport {\n  LocalStream,\n  LocalCameraStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  LocalMicrophoneStream,\n} from '@webex/media-helpers';\nimport {RtcMetrics} from '@webex/internal-plugin-metrics';\nimport {BrowserInfo} from '@webex/web-capabilities';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {MEDIA_TRACK_CONSTRAINT} from '../constants';\nimport Config from '../config';\nimport StaticConfig from '../common/config';\nimport BrowserDetection from '../common/browser-detection';\nimport {TurnServerInfo} from '../roap/types';\n\nconst {isBrowser} = BrowserDetection();\n\nexport type BundlePolicy = MultistreamConnectionConfig['bundlePolicy'];\n\n/**\n * MediaDirection\n * @typedef {Object} MediaDirection\n * @property {boolean} sendAudio\n * @property {boolean} receiveAudio\n * @property {boolean} sendVideo\n * @property {boolean} receiveVideo\n * @property {boolean} sendShare\n * @property {boolean} receiveShare\n */\n\n/**\n * SendOptions\n * @typedef {Object} SendOptions\n * @property sendAudio\n * @property sendVideo\n * @property sendShare\n * @property isSharing\n * @property {Object} sharePreferences\n */\n/**\n *\n * @public\n * @export\n * Mimic browser APIs as \"the ultimate browser\".\n * Handles the quirks of each browser.\n * Extends and enhances adapter.js, i.e., the \"media\" file from the web client.\n */\nconst Media: any = {};\n\n/**\n * make a browser call to get the media\n * @param {SendOptions} options\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise}\n */\nMedia.getLocalMedia = (options: any, config: object) => {\n  const {sendAudio, sendVideo, sendShare, sharePreferences, isSharing} = options;\n\n  if (sendAudio || sendVideo) {\n    return Media.getMedia(sendAudio, sendVideo, config);\n  }\n\n  if (sendShare && !isSharing) {\n    return Media.getDisplayMedia(\n      {\n        sendAudio: false,\n        sendShare: true,\n        sharePreferences,\n      },\n      config\n    );\n  }\n\n  return Promise.resolve(undefined);\n};\n\nMedia.getDirection = (forceSendRecv: boolean, receive: boolean, send: boolean) => {\n  if (!receive && !send) {\n    return 'inactive';\n  }\n\n  if (forceSendRecv || (receive && send)) {\n    return 'sendrecv';\n  }\n\n  if (receive) {\n    return 'recvonly';\n  }\n\n  return 'sendonly';\n};\n\n/**\n * creates a webrtc media connection with provided tracks and mediaDirection configuration\n *\n * @param {boolean} isMultistream\n * @param {string} debugId string useful for debugging (will appear in media connection logs)\n * @param {string} meetingId id for the meeting using this connection\n * @param {Object} options\n * @param {Object} [options.mediaProperties] contains mediaDirection and local tracks:\n *                                 audioTrack, videoTrack, shareVideoTrack, and shareAudioTrack\n * @param {string} [options.remoteQualityLevel] LOW|MEDIUM|HIGH applicable only to non-multistream connections\n * @param {boolean} [options.enableRtx] applicable only to non-multistream connections\n * @param {boolean} [options.enableExtmap] applicable only to non-multistream connections\n * @param {Object} [options.turnServerInfo]\n * @param {BundlePolicy} [options.bundlePolicy]\n * @returns {RoapMediaConnection | MultistreamRoapMediaConnection}\n */\nMedia.createMediaConnection = (\n  isMultistream: boolean,\n  debugId: string,\n  meetingId: string,\n  options: {\n    rtcMetrics?: RtcMetrics;\n    mediaProperties: {\n      mediaDirection?: {\n        receiveAudio: boolean;\n        receiveVideo: boolean;\n        receiveShare: boolean;\n        sendAudio: boolean;\n        sendVideo: boolean;\n        sendShare: boolean;\n      };\n      audioStream?: LocalMicrophoneStream;\n      videoStream?: LocalCameraStream;\n      shareVideoStream?: LocalDisplayStream;\n      shareAudioStream?: LocalSystemAudioStream;\n    };\n    remoteQualityLevel?: 'LOW' | 'MEDIUM' | 'HIGH';\n    enableRtx?: boolean;\n    enableExtmap?: boolean;\n    turnServerInfo?: TurnServerInfo;\n    bundlePolicy?: BundlePolicy;\n    iceCandidatesTimeout?: number;\n    disableAudioMainDtx?: boolean;\n    stopIceGatheringAfterFirstRelayCandidate?: boolean;\n  }\n) => {\n  const {\n    rtcMetrics,\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n    iceCandidatesTimeout,\n    disableAudioMainDtx,\n    stopIceGatheringAfterFirstRelayCandidate,\n  } = options;\n\n  const iceServers = [];\n\n  // we might not have any TURN server if TURN discovery failed or wasn't done or we land on a video mesh node\n  if (turnServerInfo?.urls.length > 0) {\n    // TURN-TLS server\n    iceServers.push({\n      urls: turnServerInfo.urls,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    if (disableAudioMainDtx !== undefined) {\n      config.disableAudioMainDtx = disableAudioMainDtx;\n    }\n\n    if (BrowserInfo.isFirefox()) {\n      config.doFullIce = true;\n\n      config.stopIceGatheringAfterFirstRelayCandidate = stopIceGatheringAfterFirstRelayCandidate;\n    }\n\n    return new MultistreamRoapMediaConnection(\n      config,\n      meetingId,\n      /* the rtc metrics objects callbacks */\n      (data) => rtcMetrics?.addMetrics(data),\n      () => rtcMetrics?.closeMetrics(),\n      () => rtcMetrics?.sendMetricsInQueue()\n    );\n  }\n\n  if (!mediaProperties) {\n    throw new Error('mediaProperties have to be provided for non-multistream media connections');\n  }\n\n  const {mediaDirection, audioStream, videoStream, shareVideoStream, shareAudioStream} =\n    mediaProperties;\n\n  return new RoapMediaConnection(\n    {\n      iceServers,\n      iceCandidatesTimeout,\n      skipInactiveTransceivers: false,\n      requireH264: true,\n      sdpMunging: {\n        convertPort9to0: false,\n        addContentSlides: true,\n        bandwidthLimits: {\n          audio: StaticConfig.meetings.bandwidth.audio,\n          video: StaticConfig.meetings.bandwidth.video,\n        },\n        startBitrate: StaticConfig.meetings.bandwidth.startBitrate,\n        periodicKeyframes: 20, // it's always been hardcoded in SDK so for now keeping it that way\n        disableExtmap: !enableExtmap,\n        disableRtx: !enableRtx, // see https://bugs.chromium.org/p/chromium/issues/detail?id=1020642 why we might want to remove RTX from SDP\n      },\n    },\n    {\n      // TODO: RoapMediaConnection is not ready to use stream classes yet, so we pass the raw MediaStreamTrack for now SPARK-460530\n      localTracks: {\n        audio: audioStream?.outputStream?.getTracks()[0],\n        video: videoStream?.outputStream?.getTracks()[0],\n        screenShareVideo: shareVideoStream?.outputStream?.getTracks()[0],\n        screenShareAudio: shareAudioStream?.outputStream?.getTracks()[0], // TODO: add type for screenShareAudio in internal-media-core SPARK-446923\n      } as unknown,\n      direction: {\n        audio: Media.getDirection(true, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(true, mediaDirection.receiveVideo, mediaDirection.sendVideo),\n        screenShareVideo: Media.getDirection(\n          false,\n          mediaDirection.receiveShare,\n          mediaDirection.sendShare\n        ),\n      },\n      remoteQualityLevel,\n    },\n    debugId\n  );\n};\n\n/**\n * generates share streams\n * @param {Object} options parameter\n * @param {Boolean} options.sendAudio send audio from the display share\n * @param {Boolean} options.sendShare send video from the display share\n * @param {Object} options.sharePreferences\n * @param {MediaTrackConstraints} options.sharePreferences.shareConstraints constraints to apply to video\n *   @see {@link https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints}\n * @param {Boolean} options.sharePreferences.highFrameRate if shareConstraints isn't provided, set default values based off of this boolean\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise.<MediaStream>}\n */\nMedia.getDisplayMedia = (\n  options: {\n    sendAudio: boolean;\n    sendShare: boolean;\n    sharePreferences: {\n      shareConstraints: MediaTrackConstraints;\n      highFrameRate: any;\n    };\n  },\n  config: any = {}\n) => {\n  // SDK screen share resolution settings from Webex.init\n  const customResolution = config.screenResolution || {};\n  // user defined screen share frame rate\n  const customShareFrameRate = config.screenFrameRate || null;\n  // user defined share preferences\n  const hasSharePreferences = options.sharePreferences;\n  const hasCustomConstraints = hasSharePreferences && hasSharePreferences.shareConstraints;\n  const hasHighFrameRate = hasSharePreferences && hasSharePreferences.highFrameRate;\n  const {screenResolution, resolution, videoShareFrameRate, screenFrameRate, aspectRatio} =\n    Config.meetings;\n\n  let shareConstraints: any = {\n    cursor: MEDIA_TRACK_CONSTRAINT.CURSOR.AWLAYS,\n    aspectRatio,\n  };\n\n  if (hasCustomConstraints) {\n    shareConstraints = hasSharePreferences.shareConstraints;\n  } else if (hasHighFrameRate) {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: videoShareFrameRate,\n      height: resolution.idealHeight,\n      width: resolution.idealWidth,\n      ...config.resolution,\n    };\n  } else {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: customShareFrameRate || screenFrameRate,\n      height: customResolution.idealHeight || screenResolution.idealHeight,\n      width: customResolution.idealWidth || screenResolution.idealWidth,\n      ...config.screenResolution,\n    };\n  }\n\n  // chrome and webkit based browsers (edge, safari) automatically adjust everything\n  // and we have noticed higher quality with those browser types\n  // firefox specifically has some issues with resolution and frame rate decision making\n  // so we are making it optional and configurable (with defaults) for firefox\n  // to have higher quality, and for developers to control the values\n  // eventually we may have to add the same functionality to chrome, OR conversely, get to with firefox\n\n  if (isBrowser('firefox')) {\n    const mediaConfig: any = {\n      audio: options.sendAudio,\n      video: options.sendShare,\n    };\n\n    return navigator.mediaDevices\n      .getDisplayMedia({audio: options.sendAudio, video: mediaConfig})\n      .then((stream) => {\n        if (options.sendShare && stream.getVideoTracks().length > 0) {\n          // Firefox has a bug with the spec where changing in the height and width only happens\n          // after we get the inital tracks\n          // https://bugzilla.mozilla.org/show_bug.cgi?id=1321221\n          stream.getVideoTracks()[0].applyConstraints(shareConstraints);\n        }\n\n        return stream;\n      });\n  }\n\n  const getDisplayMediaParams: any = {video: options.sendShare ? shareConstraints : false};\n\n  // safari doesn't support sending screen share audio\n  // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getDisplayMedia\n  if (options.sendAudio && isBrowser('safari')) {\n    getDisplayMediaParams.audio = options.sendAudio;\n  }\n\n  return navigator.mediaDevices.getDisplayMedia(getDisplayMediaParams);\n};\n\n/**\n * generates audio and video using constraints (often called after getSupportedDevices)\n * @param {Object|Boolean} audio gum constraints\n * @param {Object|Boolean} video gum constraints\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Object} {streams}\n */\nMedia.getMedia = (audio: any | boolean, video: any | boolean, config: any) => {\n  const defaultWidth = {ideal: config.resolution.idealWidth, max: config.resolution.maxWidth};\n  const defaultHeight = {ideal: config.resolution.idealHeight, max: config.resolution.maxHeight};\n  const mediaConfig = {\n    audio,\n    // TODO: Remove temporary workaround once Firefox fixes low constraint issues\n    // eslint-disable-next-line no-nested-ternary\n    video: video\n      ? isBrowser('firefox') && video.width && video.width.max === 320\n        ? {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: 320,\n            height: 180,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n        : {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: video.width ? video.width : defaultWidth,\n            height: video.height ? video.height : defaultHeight,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n      : false,\n    fake: process.env.NODE_ENV === 'test', // Special case to get fake media for Firefox browser for testing\n  };\n\n  return navigator.mediaDevices.getUserMedia(mediaConfig).catch((err) => {\n    const logPath = 'Media:index#getMedia --> navigator.mediaDevices.getUserMedia';\n\n    LoggerProxy.logger.error(`${logPath} failed - ${err} (${err.constraint})`);\n    throw err;\n  });\n};\n\n/**\n *\n * Toggle a specific stream\n * noop as of now, does nothing\n * @returns {null}\n */\nMedia.toggleStream = () => {};\n\n/**\n * Stop input stream\n * @param {LocalStream} stream A local stream\n * @returns {null}\n */\nMedia.stopStream = (stream: LocalStream) => {\n  if (!stream) {\n    return Promise.resolve();\n  }\n\n  return Promise.resolve().then(() => {\n    try {\n      stream.stop();\n    } catch (e) {\n      LoggerProxy.logger.error(\n        `Media:index#stopStream --> Unable to stop the stream with ready state => ${stream.readyState}, error: ${e}`\n      );\n    }\n  });\n};\n\n/**\n * generates streams for audio video and share\n * @param {object} mediaSetting parameter\n * @param {Object} mediaSetting.sendAudio sendAudio: {Boolean} sendAudio constraints\n * @param {Object} mediaSetting.sendVideo sendVideo: {Boolean} sendVideo constraints\n * @param {Object} mediaSetting.sendShare sendShare: {Boolean} sendShare constraints\n * @param {Object} mediaSetting.isSharing isSharing: {Boolean} isSharing constraints\n * @param {Object} audioVideo parameter\n * @param {Object} audioVideo.audio {deviceId: {String}}\n * @param {Object} audioVideo.video {deviceId: {String}}\n * @param {Object} sharePreferences parameter\n * @param {Object} sharePreferences.shareConstraints parameter\n * @param {Boolean} sharePreferences.highFrameRate parameter\n * @param {Object} config SDK Config\n * @returns {Array} [localStream, shareStream]\n */\nMedia.getUserMedia = (\n  mediaSetting: {\n    sendAudio: object;\n    sendVideo: object;\n    sendShare: object;\n    isSharing: object;\n  },\n  audioVideo: {\n    audio: object;\n    video: object;\n  },\n  sharePreferences: {\n    shareConstraints: object;\n    highFrameRate: boolean;\n  },\n  config: object\n) =>\n  Media.getLocalMedia(\n    {\n      sendAudio: mediaSetting.sendAudio ? audioVideo.audio || mediaSetting.sendAudio : false,\n      sendVideo: mediaSetting.sendVideo ? audioVideo.video || mediaSetting.sendVideo : false,\n    },\n    config\n  ).then((localStream) =>\n    Media.getLocalMedia(\n      {\n        sendShare: mediaSetting.sendShare,\n        isSharing: mediaSetting.isSharing,\n        sharePreferences,\n      },\n      config\n    ).then((shareStream) => [localStream, shareStream])\n  );\n\nexport default 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See LICENSE file.\n */\n/* globals navigator */\n\nimport {\n  RoapMediaConnection,\n  MultistreamRoapMediaConnection,\n  type MultistreamConnectionConfig,\n} from '@webex/internal-media-core';\nimport {\n  LocalStream,\n  LocalCameraStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  LocalMicrophoneStream,\n} from '@webex/media-helpers';\nimport {RtcMetrics} from '@webex/internal-plugin-metrics';\nimport {BrowserInfo} from '@webex/web-capabilities';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {MEDIA_TRACK_CONSTRAINT} from '../constants';\nimport Config from '../config';\nimport StaticConfig from '../common/config';\nimport BrowserDetection from '../common/browser-detection';\nimport {TurnServerInfo} from '../roap/types';\n\nconst {isBrowser} = BrowserDetection();\n\nexport type BundlePolicy = MultistreamConnectionConfig['bundlePolicy'];\n\n/**\n * MediaDirection\n * @typedef {Object} MediaDirection\n * @property {boolean} sendAudio\n * @property {boolean} receiveAudio\n * @property {boolean} sendVideo\n * @property {boolean} receiveVideo\n * @property {boolean} sendShare\n * @property {boolean} receiveShare\n */\n\n/**\n * SendOptions\n * @typedef {Object} SendOptions\n * @property sendAudio\n * @property sendVideo\n * @property sendShare\n * @property isSharing\n * @property {Object} sharePreferences\n */\n/**\n *\n * @public\n * @export\n * Mimic browser APIs as \"the ultimate browser\".\n * Handles the quirks of each browser.\n * Extends and enhances adapter.js, i.e., the \"media\" file from the web client.\n */\nconst Media: any = {};\n\n/**\n * make a browser call to get the media\n * @param {SendOptions} options\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise}\n */\nMedia.getLocalMedia = (options: any, config: object) => {\n  const {sendAudio, sendVideo, sendShare, sharePreferences, isSharing} = options;\n\n  if (sendAudio || sendVideo) {\n    return Media.getMedia(sendAudio, sendVideo, config);\n  }\n\n  if (sendShare && !isSharing) {\n    return Media.getDisplayMedia(\n      {\n        sendAudio: false,\n        sendShare: true,\n        sharePreferences,\n      },\n      config\n    );\n  }\n\n  return Promise.resolve(undefined);\n};\n\nMedia.getDirection = (forceSendRecv: boolean, receive: boolean, send: boolean) => {\n  if (!receive && !send) {\n    return 'inactive';\n  }\n\n  if (forceSendRecv || (receive && send)) {\n    return 'sendrecv';\n  }\n\n  if (receive) {\n    return 'recvonly';\n  }\n\n  return 'sendonly';\n};\n\n/**\n * creates a webrtc media connection with provided tracks and mediaDirection configuration\n *\n * @param {boolean} isMultistream\n * @param {string} debugId string useful for debugging (will appear in media connection logs)\n * @param {string} meetingId id for the meeting using this connection\n * @param {Object} options\n * @param {Object} [options.mediaProperties] contains mediaDirection and local tracks:\n *                                 audioTrack, videoTrack, shareVideoTrack, and shareAudioTrack\n * @param {string} [options.remoteQualityLevel] LOW|MEDIUM|HIGH applicable only to non-multistream connections\n * @param {boolean} [options.enableRtx] applicable only to non-multistream connections\n * @param {boolean} [options.enableExtmap] applicable only to non-multistream connections\n * @param {Object} [options.turnServerInfo]\n * @param {BundlePolicy} [options.bundlePolicy]\n * @returns {RoapMediaConnection | MultistreamRoapMediaConnection}\n */\nMedia.createMediaConnection = (\n  isMultistream: boolean,\n  debugId: string,\n  meetingId: string,\n  options: {\n    rtcMetrics?: RtcMetrics;\n    mediaProperties: {\n      mediaDirection?: {\n        receiveAudio: boolean;\n        receiveVideo: boolean;\n        receiveShare: boolean;\n        sendAudio: boolean;\n        sendVideo: boolean;\n        sendShare: boolean;\n      };\n      audioStream?: LocalMicrophoneStream;\n      videoStream?: LocalCameraStream;\n      shareVideoStream?: LocalDisplayStream;\n      shareAudioStream?: LocalSystemAudioStream;\n    };\n    remoteQualityLevel?: 'LOW' | 'MEDIUM' | 'HIGH';\n    enableRtx?: boolean;\n    enableExtmap?: boolean;\n    turnServerInfo?: TurnServerInfo;\n    bundlePolicy?: BundlePolicy;\n    iceCandidatesTimeout?: number;\n    disableAudioMainDtx?: boolean;\n    enableAudioTwcc?: boolean;\n    stopIceGatheringAfterFirstRelayCandidate?: boolean;\n  }\n) => {\n  const {\n    rtcMetrics,\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n    iceCandidatesTimeout,\n    disableAudioMainDtx,\n    enableAudioTwcc,\n    stopIceGatheringAfterFirstRelayCandidate,\n  } = options;\n\n  const iceServers = [];\n\n  // we might not have any TURN server if TURN discovery failed or wasn't done or we land on a video mesh node\n  if (turnServerInfo?.urls.length > 0) {\n    // TURN-TLS server\n    iceServers.push({\n      urls: turnServerInfo.urls,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n      disableAudioTwcc: !enableAudioTwcc,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    if (disableAudioMainDtx !== undefined) {\n      config.disableAudioMainDtx = disableAudioMainDtx;\n    }\n\n    if (BrowserInfo.isFirefox()) {\n      config.doFullIce = true;\n\n      config.stopIceGatheringAfterFirstRelayCandidate = stopIceGatheringAfterFirstRelayCandidate;\n    }\n\n    return new MultistreamRoapMediaConnection(\n      config,\n      meetingId,\n      /* the rtc metrics objects callbacks */\n      (data) => rtcMetrics?.addMetrics(data),\n      () => rtcMetrics?.closeMetrics(),\n      () => rtcMetrics?.sendMetricsInQueue()\n    );\n  }\n\n  if (!mediaProperties) {\n    throw new Error('mediaProperties have to be provided for non-multistream media connections');\n  }\n\n  const {mediaDirection, audioStream, videoStream, shareVideoStream, shareAudioStream} =\n    mediaProperties;\n\n  return new RoapMediaConnection(\n    {\n      iceServers,\n      iceCandidatesTimeout,\n      skipInactiveTransceivers: false,\n      requireH264: true,\n      sdpMunging: {\n        convertPort9to0: false,\n        addContentSlides: true,\n        bandwidthLimits: {\n          audio: StaticConfig.meetings.bandwidth.audio,\n          video: StaticConfig.meetings.bandwidth.video,\n        },\n        startBitrate: StaticConfig.meetings.bandwidth.startBitrate,\n        periodicKeyframes: 20, // it's always been hardcoded in SDK so for now keeping it that way\n        disableExtmap: !enableExtmap,\n        disableRtx: !enableRtx, // see https://bugs.chromium.org/p/chromium/issues/detail?id=1020642 why we might want to remove RTX from SDP\n      },\n    },\n    {\n      // TODO: RoapMediaConnection is not ready to use stream classes yet, so we pass the raw MediaStreamTrack for now SPARK-460530\n      localTracks: {\n        audio: audioStream?.outputStream?.getTracks()[0],\n        video: videoStream?.outputStream?.getTracks()[0],\n        screenShareVideo: shareVideoStream?.outputStream?.getTracks()[0],\n        screenShareAudio: shareAudioStream?.outputStream?.getTracks()[0], // TODO: add type for screenShareAudio in internal-media-core SPARK-446923\n      } as unknown,\n      direction: {\n        audio: Media.getDirection(true, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(true, mediaDirection.receiveVideo, mediaDirection.sendVideo),\n        screenShareVideo: Media.getDirection(\n          false,\n          mediaDirection.receiveShare,\n          mediaDirection.sendShare\n        ),\n      },\n      remoteQualityLevel,\n    },\n    debugId\n  );\n};\n\n/**\n * generates share streams\n * @param {Object} options parameter\n * @param {Boolean} options.sendAudio send audio from the display share\n * @param {Boolean} options.sendShare send video from the display share\n * @param {Object} options.sharePreferences\n * @param {MediaTrackConstraints} options.sharePreferences.shareConstraints constraints to apply to video\n *   @see {@link https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints}\n * @param {Boolean} options.sharePreferences.highFrameRate if shareConstraints isn't provided, set default values based off of this boolean\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise.<MediaStream>}\n */\nMedia.getDisplayMedia = (\n  options: {\n    sendAudio: boolean;\n    sendShare: boolean;\n    sharePreferences: {\n      shareConstraints: MediaTrackConstraints;\n      highFrameRate: any;\n    };\n  },\n  config: any = {}\n) => {\n  // SDK screen share resolution settings from Webex.init\n  const customResolution = config.screenResolution || {};\n  // user defined screen share frame rate\n  const customShareFrameRate = config.screenFrameRate || null;\n  // user defined share preferences\n  const hasSharePreferences = options.sharePreferences;\n  const hasCustomConstraints = hasSharePreferences && hasSharePreferences.shareConstraints;\n  const hasHighFrameRate = hasSharePreferences && hasSharePreferences.highFrameRate;\n  const {screenResolution, resolution, videoShareFrameRate, screenFrameRate, aspectRatio} =\n    Config.meetings;\n\n  let shareConstraints: any = {\n    cursor: MEDIA_TRACK_CONSTRAINT.CURSOR.AWLAYS,\n    aspectRatio,\n  };\n\n  if (hasCustomConstraints) {\n    shareConstraints = hasSharePreferences.shareConstraints;\n  } else if (hasHighFrameRate) {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: videoShareFrameRate,\n      height: resolution.idealHeight,\n      width: resolution.idealWidth,\n      ...config.resolution,\n    };\n  } else {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: customShareFrameRate || screenFrameRate,\n      height: customResolution.idealHeight || screenResolution.idealHeight,\n      width: customResolution.idealWidth || screenResolution.idealWidth,\n      ...config.screenResolution,\n    };\n  }\n\n  // chrome and webkit based browsers (edge, safari) automatically adjust everything\n  // and we have noticed higher quality with those browser types\n  // firefox specifically has some issues with resolution and frame rate decision making\n  // so we are making it optional and configurable (with defaults) for firefox\n  // to have higher quality, and for developers to control the values\n  // eventually we may have to add the same functionality to chrome, OR conversely, get to with firefox\n\n  if (isBrowser('firefox')) {\n    const mediaConfig: any = {\n      audio: options.sendAudio,\n      video: options.sendShare,\n    };\n\n    return navigator.mediaDevices\n      .getDisplayMedia({audio: options.sendAudio, video: mediaConfig})\n      .then((stream) => {\n        if (options.sendShare && stream.getVideoTracks().length > 0) {\n          // Firefox has a bug with the spec where changing in the height and width only happens\n          // after we get the inital tracks\n          // https://bugzilla.mozilla.org/show_bug.cgi?id=1321221\n          stream.getVideoTracks()[0].applyConstraints(shareConstraints);\n        }\n\n        return stream;\n      });\n  }\n\n  const getDisplayMediaParams: any = {video: options.sendShare ? shareConstraints : false};\n\n  // safari doesn't support sending screen share audio\n  // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getDisplayMedia\n  if (options.sendAudio && isBrowser('safari')) {\n    getDisplayMediaParams.audio = options.sendAudio;\n  }\n\n  return navigator.mediaDevices.getDisplayMedia(getDisplayMediaParams);\n};\n\n/**\n * generates audio and video using constraints (often called after getSupportedDevices)\n * @param {Object|Boolean} audio gum constraints\n * @param {Object|Boolean} video gum constraints\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Object} {streams}\n */\nMedia.getMedia = (audio: any | boolean, video: any | boolean, config: any) => {\n  const defaultWidth = {ideal: config.resolution.idealWidth, max: config.resolution.maxWidth};\n  const defaultHeight = {ideal: config.resolution.idealHeight, max: config.resolution.maxHeight};\n  const mediaConfig = {\n    audio,\n    // TODO: Remove temporary workaround once Firefox fixes low constraint issues\n    // eslint-disable-next-line no-nested-ternary\n    video: video\n      ? isBrowser('firefox') && video.width && video.width.max === 320\n        ? {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: 320,\n            height: 180,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n        : {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: video.width ? video.width : defaultWidth,\n            height: video.height ? video.height : defaultHeight,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n      : false,\n    fake: process.env.NODE_ENV === 'test', // Special case to get fake media for Firefox browser for testing\n  };\n\n  return navigator.mediaDevices.getUserMedia(mediaConfig).catch((err) => {\n    const logPath = 'Media:index#getMedia --> navigator.mediaDevices.getUserMedia';\n\n    LoggerProxy.logger.error(`${logPath} failed - ${err} (${err.constraint})`);\n    throw err;\n  });\n};\n\n/**\n *\n * Toggle a specific stream\n * noop as of now, does nothing\n * @returns {null}\n */\nMedia.toggleStream = () => {};\n\n/**\n * Stop input stream\n * @param {LocalStream} stream A local stream\n * @returns {null}\n */\nMedia.stopStream = (stream: LocalStream) => {\n  if (!stream) {\n    return Promise.resolve();\n  }\n\n  return Promise.resolve().then(() => {\n    try {\n      stream.stop();\n    } catch (e) {\n      LoggerProxy.logger.error(\n        `Media:index#stopStream --> Unable to stop the stream with ready state => ${stream.readyState}, error: ${e}`\n      );\n    }\n  });\n};\n\n/**\n * generates streams for audio video and share\n * @param {object} mediaSetting parameter\n * @param {Object} mediaSetting.sendAudio sendAudio: {Boolean} sendAudio constraints\n * @param {Object} mediaSetting.sendVideo sendVideo: {Boolean} sendVideo constraints\n * @param {Object} mediaSetting.sendShare sendShare: {Boolean} sendShare constraints\n * @param {Object} mediaSetting.isSharing isSharing: {Boolean} isSharing constraints\n * @param {Object} audioVideo parameter\n * @param {Object} audioVideo.audio {deviceId: {String}}\n * @param {Object} audioVideo.video {deviceId: {String}}\n * @param {Object} sharePreferences parameter\n * @param {Object} sharePreferences.shareConstraints parameter\n * @param {Boolean} sharePreferences.highFrameRate parameter\n * @param {Object} config SDK Config\n * @returns {Array} [localStream, shareStream]\n */\nMedia.getUserMedia = (\n  mediaSetting: {\n    sendAudio: object;\n    sendVideo: object;\n    sendShare: object;\n    isSharing: object;\n  },\n  audioVideo: {\n    audio: object;\n    video: object;\n  },\n  sharePreferences: {\n    shareConstraints: object;\n    highFrameRate: boolean;\n  },\n  config: object\n) =>\n  Media.getLocalMedia(\n    {\n      sendAudio: mediaSetting.sendAudio ? audioVideo.audio || mediaSetting.sendAudio : false,\n      sendVideo: mediaSetting.sendVideo ? audioVideo.video || mediaSetting.sendVideo : false,\n    },\n    config\n  ).then((localStream) =>\n    Media.getLocalMedia(\n      {\n        sendShare: mediaSetting.sendShare,\n        isSharing: mediaSetting.isSharing,\n        sharePreferences,\n      },\n      config\n    ).then((shareStream) => [localStream, shareStream])\n  );\n\nexport default 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package/dist/meeting/index.js

@@ -6774,6 +6774,8 @@ var Meeting = exports.default = /*#__PURE__*/function (_StatelessWebexPlugin) {
                 iceCandidatesTimeout: this.config.iceCandidatesGatheringTimeout,
                 // @ts-ignore - config coming from registerPlugin
                 disableAudioMainDtx: this.config.experimental.disableAudioMainDtx,
+                // @ts-ignore - config coming from registerPlugin
+                enableAudioTwcc: this.config.enableAudioTwccForMultistream,
                 stopIceGatheringAfterFirstRelayCandidate:
                 // @ts-ignore - config coming from registerPlugin
                 this.config.stopIceGatheringAfterFirstRelayCandidate