@webex/plugin-meetings 3.0.0-beta.2 → 3.0.0-beta.20

package/src/statsAnalyzer/mqaUtil.ts

@@ -0,0 +1,290 @@
+/* eslint-disable prefer-destructuring */
+
+import {mean, max} from 'lodash';
+
+import {STATS} from '../constants';
+
+export const getAudioReceiverMqa = ({audioReceiver, statsResults, lastMqaDataSent}) => {
+  const mediaType = STATS.AUDIO_CORRELATE;
+  const sendrecvType = STATS.RECEIVE_DIRECTION;
+
+  audioReceiver.common.common.direction = statsResults[mediaType].direction;
+  audioReceiver.common.transportType = statsResults.connectionType.remote.transport[0];
+  // add rtpPacket info inside common as also for call analyzer
+  audioReceiver.common.rtpPackets =
+    statsResults[mediaType][sendrecvType].totalPacketsReceived -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsReceived || 0;
+  // Hop by hop are numbers and not percentage so we compare on what we sent the last min
+  // collect the packets received for the last min
+  audioReceiver.streams[0].common.rtpPackets = audioReceiver.common.rtpPackets;
+  audioReceiver.common.mediaHopByHopLost =
+    statsResults[mediaType][sendrecvType].totalPacketsLost -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsLost || 0;
+  audioReceiver.common.rtpHopByHopLost =
+    statsResults[mediaType][sendrecvType].totalPacketsLost -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsLost || 0;
+
+  // @ts-ignore
+  audioReceiver.streams[0].common.maxRtpJitter =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;
+  audioReceiver.streams[0].common.meanRtpJitter =
+    mean(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;
+  audioReceiver.streams[0].common.rtpJitter = audioReceiver.streams[0].common.maxRtpJitter;
+
+  // Fec packets do come in as part of the FEC only for audio
+  const fecRecovered =
+    statsResults[mediaType][sendrecvType].fecPacketsReceived -
+    lastMqaDataSent[mediaType][sendrecvType].fecPacketsReceived -
+    (statsResults[mediaType][sendrecvType].fecPacketsDiscarded -
+      lastMqaDataSent[mediaType][sendrecvType].fecPacketsDiscarded);
+
+  audioReceiver.streams[0].common.rtpEndToEndLost =
+    statsResults[mediaType][sendrecvType].totalPacketsLost -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsLost -
+      fecRecovered || 0;
+
+  audioReceiver.streams[0].common.framesDropped =
+    statsResults[mediaType][sendrecvType].totalSamplesDecoded -
+      lastMqaDataSent[mediaType][sendrecvType].totalSamplesDecoded || 0;
+  audioReceiver.streams[0].common.renderedFrameRate =
+    (audioReceiver.streams[0].common.framesDropped * 100) / 60 || 0;
+  audioReceiver.streams[0].common.framesReceived =
+    statsResults[mediaType][sendrecvType].totalSamplesReceived -
+      lastMqaDataSent[mediaType][sendrecvType].totalSamplesReceived || 0;
+  audioReceiver.streams[0].common.concealedFrames =
+    statsResults[mediaType][sendrecvType].concealedSamples -
+      lastMqaDataSent[mediaType][sendrecvType].concealedSamples || 0;
+  audioReceiver.streams[0].common.receivedBitrate =
+    ((statsResults[mediaType][sendrecvType].totalBytesReceived -
+      lastMqaDataSent[mediaType][sendrecvType].totalBytesReceived) *
+      8) /
+      60 || 0;
+
+  audioReceiver.common.rtpBitrate = audioReceiver.streams[0].common.receivedBitrate;
+};
+
+export const getAudioSenderMqa = ({audioSender, statsResults, lastMqaDataSent}) => {
+  const mediaType = STATS.AUDIO_CORRELATE;
+  const sendrecvType = STATS.SEND_DIRECTION;
+
+  audioSender.common.common.direction = statsResults[mediaType].direction;
+  audioSender.common.transportType = statsResults.connectionType.local.transport[0];
+
+  // @ts-ignore
+  audioSender.common.maxRemoteJitter =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+  audioSender.common.meanRemoteJitter =
+    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+
+  audioSender.common.rtpPackets =
+    statsResults[mediaType][sendrecvType].totalPacketsSent -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsSent || 0;
+  audioSender.streams[0].common.rtpPackets = audioSender.common.rtpPackets;
+  // From candidate-pair
+  audioSender.common.availableBitrate =
+    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;
+  // Calculate based on how much packets lost of received compated to how to the client sent
+
+  const totalpacketsLostForaMin =
+    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver -
+    lastMqaDataSent[mediaType][sendrecvType].totalPacketsLostOnReceiver;
+
+  audioSender.common.remoteLossRate =
+    totalpacketsLostForaMin > 0
+      ? (totalpacketsLostForaMin * 100) / audioSender.common.rtpPackets
+      : 0; // This is the packets sent with in last min || 0;
+
+  // @ts-ignore
+  audioSender.common.maxRoundTripTime =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;
+  audioSender.common.meanRoundTripTime =
+    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;
+  audioSender.common.roundTripTime = audioSender.common.maxRoundTripTime;
+
+  // Calculate the outgoing bitrate
+  const totalBytesSentInaMin =
+    statsResults[mediaType][sendrecvType].totalBytesSent -
+    lastMqaDataSent[mediaType][sendrecvType].totalBytesSent;
+
+  audioSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin
+    ? (totalBytesSentInaMin * 8) / 60
+    : 0;
+  audioSender.common.rtpBitrate = audioSender.streams[0].common.transmittedBitrate;
+
+  audioSender.streams[0].transmittedKeyFrames =
+    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded -
+      lastMqaDataSent[mediaType][sendrecvType].totalKeyFramesEncoded || 0;
+  audioSender.streams[0].requestedKeyFrames =
+    statsResults[mediaType][sendrecvType].totalFirCount -
+      lastMqaDataSent[mediaType][sendrecvType].totalFirCount || 0;
+};
+
+export const getVideoReceiverMqa = ({
+  videoReceiver,
+  statsResults,
+  lastMqaDataSent,
+  isShareStream = false,
+}) => {
+  const mediaType = isShareStream ? STATS.SHARE_CORRELATE : STATS.VIDEO_CORRELATE;
+  const sendrecvType = STATS.RECEIVE_DIRECTION;
+
+  videoReceiver.common.common.direction = statsResults[mediaType].direction;
+  videoReceiver.common.transportType = statsResults.connectionType.remote.transport[0];
+  // collect the packets received for the last min
+  videoReceiver.common.rtpPackets =
+    statsResults[mediaType][sendrecvType].totalPacketsReceived -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsReceived || 0;
+  videoReceiver.streams[0].common.rtpPackets = videoReceiver.common.rtpPackets;
+
+  const totalPacketLoss =
+    statsResults[mediaType][sendrecvType].totalPacketsLost -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsLost || 0;
+
+  // Hope by hop are numbers and not percentage so we compare on what we sent the last min
+  // this is including packet lost
+  videoReceiver.common.mediaHopByHopLost = totalPacketLoss;
+  videoReceiver.common.rtpHopByHopLost = totalPacketLoss;
+
+  // End to end packetloss is after recovery
+  videoReceiver.streams[0].common.rtpEndToEndLost = totalPacketLoss;
+
+  // calculate this values
+
+  // @ts-ignore
+  videoReceiver.common.maxRemoteJitter =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+  videoReceiver.common.meanRemoteJitter =
+    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+
+  videoReceiver.streams[0].common.rtpJitter = videoReceiver.common.maxRemoteJitter;
+  // videoReceiver.streams[0].common.rtpJitter = (statsResults.resolutions[mediaType][sendrecvType].jitterBufferDelay - lastMqaDataSent.resolutions[mediaType][sendrecvType].jitterBufferDelay) / (statsResults.resolutions[mediaType][sendrecvType].jitterBufferEmittedCount - lastMqaDataSent.resolutions[mediaType][sendrecvType].jitterBufferEmittedCount) * 1000 || 0;
+
+  // Calculate the outgoing bitrate
+  const totalBytesReceivedInaMin =
+    statsResults[mediaType][sendrecvType].totalBytesReceived -
+    lastMqaDataSent[mediaType][sendrecvType].totalBytesReceived;
+
+  videoReceiver.streams[0].common.receivedBitrate = totalBytesReceivedInaMin
+    ? (totalBytesReceivedInaMin * 8) / 60
+    : 0;
+  videoReceiver.common.rtpBitrate = videoReceiver.streams[0].common.receivedBitrate;
+
+  // From tracks //TODO: calculate a proper one
+  const totalFrameReceivedInaMin =
+    statsResults.resolutions[mediaType][sendrecvType].framesReceived -
+    lastMqaDataSent.resolutions[mediaType][sendrecvType].framesReceived;
+  const totalFrameDecodedInaMin =
+    statsResults.resolutions[mediaType][sendrecvType].framesDecoded -
+    lastMqaDataSent.resolutions[mediaType][sendrecvType].framesDecoded;
+
+  videoReceiver.streams[0].common.receivedFrameRate = totalFrameReceivedInaMin
+    ? (totalFrameReceivedInaMin * 100) / 60
+    : 0;
+  videoReceiver.streams[0].common.renderedFrameRate = totalFrameDecodedInaMin
+    ? (totalFrameDecodedInaMin * 100) / 60
+    : 0;
+
+  videoReceiver.streams[0].common.framesDropped =
+    statsResults.resolutions[mediaType][sendrecvType].framesDropped -
+    lastMqaDataSent.resolutions[mediaType][sendrecvType].framesDropped;
+  videoReceiver.streams[0].receivedHeight =
+    statsResults.resolutions[mediaType][sendrecvType].height;
+  videoReceiver.streams[0].receivedWidth = statsResults.resolutions[mediaType][sendrecvType].width;
+  videoReceiver.streams[0].receivedFrameSize =
+    (statsResults.resolutions[mediaType][sendrecvType].height *
+      statsResults.resolutions[mediaType][sendrecvType].height) /
+    256;
+
+  videoReceiver.streams[0].receivedKeyFrames =
+    statsResults[mediaType][sendrecvType].keyFramesDecoded -
+      lastMqaDataSent[mediaType][sendrecvType].keyFramesDecoded || 0;
+  videoReceiver.streams[0].requestedKeyFrames =
+    statsResults[mediaType][sendrecvType].totalPliCount -
+      lastMqaDataSent[mediaType][sendrecvType].totalPliCount || 0;
+};
+
+export const getVideoSenderMqa = ({
+  videoSender,
+  statsResults,
+  lastMqaDataSent,
+  isShareStream = false,
+}) => {
+  const mediaType = isShareStream ? STATS.SHARE_CORRELATE : STATS.VIDEO_CORRELATE;
+  const sendrecvType = STATS.SEND_DIRECTION;
+
+  videoSender.common.common.direction = statsResults[mediaType].direction;
+  videoSender.common.transportType = statsResults.connectionType.local.transport[0];
+
+  // @ts-ignore
+  videoSender.common.maxRemoteJitter =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+  videoSender.common.meanRemoteJitter =
+    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;
+
+  videoSender.common.rtpPackets =
+    statsResults[mediaType][sendrecvType].totalPacketsSent -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsSent || 0;
+  videoSender.common.availableBitrate =
+    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;
+  // Calculate based on how much packets lost of received compated to how to the client sent
+
+  const totalpacketsLostForaMin =
+    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver -
+    lastMqaDataSent[mediaType][sendrecvType].totalPacketsLostOnReceiver;
+
+  videoSender.common.remoteLossRate =
+    totalpacketsLostForaMin > 0
+      ? (totalpacketsLostForaMin * 100) / (videoSender.common.rtpPackets + totalpacketsLostForaMin)
+      : 0; // This is the packets sent with in last min || 0;
+
+  // @ts-ignore
+  videoSender.common.maxRoundTripTime =
+    // @ts-ignore
+    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;
+  videoSender.common.meanRoundTripTime =
+    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;
+  videoSender.common.roundTripTime = videoSender.common.maxRoundTripTime;
+
+  videoSender.streams[0].common.rtpPackets =
+    statsResults[mediaType][sendrecvType].totalPacketsSent -
+      lastMqaDataSent[mediaType][sendrecvType].totalPacketsSent || 0;
+
+  // Calculate the outgoing bitrate
+  const totalBytesSentInaMin =
+    statsResults[mediaType][sendrecvType].totalBytesSent -
+    lastMqaDataSent[mediaType][sendrecvType].totalBytesSent;
+
+  videoSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin
+    ? (totalBytesSentInaMin * 8) / 60
+    : 0;
+
+  videoSender.common.rtpBitrate = videoSender.streams[0].common.transmittedBitrate;
+
+  videoSender.streams[0].transmittedKeyFrames =
+    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded -
+      lastMqaDataSent[mediaType][sendrecvType].totalKeyFramesEncoded || 0;
+  videoSender.streams[0].requestedKeyFrames =
+    statsResults[mediaType][sendrecvType].totalFirCount -
+      lastMqaDataSent[mediaType][sendrecvType].totalFirCount || 0;
+
+  // From tracks //TODO: calculate a proper one
+  const totalFrameSentInaMin =
+    statsResults.resolutions[mediaType][sendrecvType].framesSent -
+    (lastMqaDataSent.resolutions[mediaType][sendrecvType].framesSent || 0);
+
+  videoSender.streams[0].common.transmittedFrameRate = totalFrameSentInaMin
+    ? (totalFrameSentInaMin * 100) / 60
+    : 0;
+  videoSender.streams[0].transmittedHeight =
+    statsResults.resolutions[mediaType][sendrecvType].height;
+  videoSender.streams[0].transmittedWidth = statsResults.resolutions[mediaType][sendrecvType].width;
+  videoSender.streams[0].transmittedFrameSize =
+    (statsResults.resolutions[mediaType][sendrecvType].height *
+      statsResults.resolutions[mediaType][sendrecvType].width) /
+    254;
+};

package/src/transcription/{index.js → index.ts}

@@ -6,6 +6,12 @@ import {v4 as uuidv4} from 'uuid';
  * @class Transcription
  */
 export default class Transcription {
+  webSocketUrl: any;
+  sessionID: any;
+  members: any;
+  memberCSIs: any;
+  webSocket: any;
+
   /**
    * @param {string} webSocketUrl
    * @param {sessionID} sessionId
@@ -13,7 +19,7 @@ export default class Transcription {
    * @constructor
    * @memberof Transcription
    */
-  constructor(webSocketUrl, sessionId, members) {
+  constructor(webSocketUrl: string, sessionId: any, members: object) {
     this.webSocketUrl = webSocketUrl;
     this.sessionID = sessionId;
     this.members = members;
@@ -32,24 +38,26 @@ export default class Transcription {
    * @param {string} sessionID
    * @returns {void}
    */
-  connect(token) {
+  connect(token: string) {
     this.webSocket.onopen = () => {
-      this.webSocket.send(JSON.stringify({
-        id: uuidv4(),
-        type: 'authorization',
-        data: {token: `Bearer ${token}`},
-        trackingId: `webex-js-sdk_${this.sessionID}${Date.now.toString()}`
-      }));
+      this.webSocket.send(
+        JSON.stringify({
+          id: uuidv4(),
+          type: 'authorization',
+          data: {token: `Bearer ${token}`},
+          trackingId: `webex-js-sdk_${this.sessionID}${Date.now.toString()}`,
+        })
+      );
     };
   }
 
   /**
    * Sets callback to invoke when the web socket connection is closed.
    *
-   * @param {function} callback
+   * @param {Function} callback
    * @returns {void}
    */
-  onCloseSocket(callback) {
+  onCloseSocket(callback: any) {
     if (callback) {
       this.webSocket.onclose = (event) => {
         callback(event);
@@ -61,10 +69,10 @@ export default class Transcription {
    * Sets callback to invoke when a web socket connection error occurs.
    * An event parameter will be send to the callback.
    *
-   * @param {function} callback
+   * @param {Function} callback
    * @returns {void}
    */
-  onErrorSocket(callback) {
+  onErrorSocket(callback: any) {
     if (callback) {
       this.webSocket.onerror = (event) => {
         callback(event);
@@ -79,7 +87,7 @@ export default class Transcription {
    * @param {object} csis
    * @returns {object}
    */
-  getSpeaker(csis) {
+  private getSpeaker(csis: any) {
     for (const csi of csis) {
       const member = this.memberCSIs[csi];
 
@@ -88,31 +96,33 @@ export default class Transcription {
       }
     }
 
-    return Object.values(this.members.membersCollection.members)
-      .find((member) => {
-        const memberCSIs = member.participant.status.csis;
-        let selfIsSpeaking = false;
-
-        for (const csi of csis) {
-          if (memberCSIs.includes(csi)) {
-            this.memberCSIs[csi] = member;
-            selfIsSpeaking = true;
-            break;
-          }
+    return Object.values(this.members.membersCollection.members).find((member: any) => {
+      const memberCSIs = member.participant.status.csis;
+      let selfIsSpeaking = false;
+
+      for (const csi of csis) {
+        if (memberCSIs.includes(csi)) {
+          this.memberCSIs[csi] = member;
+          selfIsSpeaking = true;
+          break;
         }
+      }
 
-        return selfIsSpeaking;
-      });
+      return selfIsSpeaking;
+    });
   }
 
   /**
    * Sends transcription data to given callback as it arrives.
    *
-   * @param {function} callback
+   * @param {Function} callback
    * @returns {void}
    */
-  subscribe(callback) {
-    let data, csis, speaker, transcription;
+  subscribe(callback: any) {
+    let data;
+    let csis;
+    let speaker;
+    let transcription;
 
     this.webSocket.onmessage = (event) => {
       data = JSON.parse(event.data);
@@ -123,15 +133,13 @@ export default class Transcription {
       this.webSocket.send(JSON.stringify({messageID: data.id, type: 'ack'}));
 
       if (transcription) {
-        callback(
-          {
-            id: data.data?.voiceaPayload?.transcript_id,
-            personID: speaker?.id,
-            transcription,
-            timestamp: data.timestamp,
-            type: data?.data?.voiceaPayload?.type
-          }
-        );
+        callback({
+          id: data.data?.voiceaPayload?.transcript_id,
+          personID: speaker?.id,
+          transcription,
+          timestamp: data.timestamp,
+          type: data?.data?.voiceaPayload?.type,
+        });
       }
     };
   }
@@ -144,4 +152,3 @@ export default class Transcription {
     this.webSocket.close();
   }
 }
-