@abrar71/lib-jitsi-meet 0.0.0

package/dist/esm/modules/qualitycontrol/QualityController.js

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+import { getLogger } from '@jitsi/logger';
+import { JitsiConferenceEvents } from '../../JitsiConferenceEvents';
+import { RTCEvents } from '../../service/RTC/RTCEvents';
+import { DEFAULT_LAST_N, LAST_N_UNLIMITED, VIDEO_CODECS_BY_COMPLEXITY, VIDEO_QUALITY_LEVELS } from '../../service/RTC/StandardVideoQualitySettings';
+import RTCStats from '../RTCStats/RTCStats';
+import { RTCStatsEvents } from '../RTCStats/RTCStatsEvents';
+import { isValidNumber } from '../util/MathUtil';
+import { CodecSelection } from './CodecSelection';
+import { ReceiverAudioController } from './ReceiveAudioController';
+import ReceiveVideoController from './ReceiveVideoController';
+import SendVideoController from './SendVideoController';
+const logger = getLogger('qc:QualityController');
+// Period for which the client will wait for the cpu limitation flag to be reset in the peerconnection stats before it
+// attempts to rectify the situation by attempting a codec switch.
+const LIMITED_BY_CPU_TIMEOUT = 60000;
+// The min. value that lastN will be set to while trying to fix video qaulity issues.
+const MIN_LAST_N = 3;
+var QualityLimitationReason;
+(function (QualityLimitationReason) {
+    QualityLimitationReason["BANDWIDTH"] = "bandwidth";
+    QualityLimitationReason["CPU"] = "cpu";
+    QualityLimitationReason["NONE"] = "none";
+})(QualityLimitationReason || (QualityLimitationReason = {}));
+/* eslint-disable require-jsdoc */
+export class FixedSizeArray {
+    constructor(size) {
+        this._maxSize = size;
+        this._data = [];
+    }
+    add(item) {
+        if (this._data.length >= this._maxSize) {
+            this._data.shift();
+        }
+        this._data.push(item);
+    }
+    get(index) {
+        if (index < 0 || index >= this._data.length) {
+            throw new Error('Index out of bounds');
+        }
+        return this._data[index];
+    }
+    size() {
+        return this._data.length;
+    }
+}
+/* eslint-enable require-jsdoc */
+/**
+ * QualityController class that is responsible for maintaining optimal video quality experience on the local endpoint
+ * by controlling the codec, encode resolution and receive resolution of the remote video streams. It also makes
+ * adjustments based on the outbound and inbound rtp stream stats reported by the underlying peer connection.
+ */
+export class QualityController {
+    /**
+     *
+     * @param {JitsiConference} conference - The JitsiConference instance.
+     * @param {Object} options - video quality settings passed through config.js.
+     */
+    constructor(conference, options) {
+        this._audioController = new ReceiverAudioController(conference);
+        this._conference = conference;
+        const { jvb, p2p } = options;
+        this._codecController = new CodecSelection(conference, { jvb,
+            p2p });
+        this._enableAdaptiveMode = options.enableAdaptiveMode ?? true;
+        this._encodeTimeStats = new Map();
+        this._isLastNRampupBlocked = false;
+        this._lastNRampupTime = options.lastNRampupTime;
+        this._receiveVideoController = new ReceiveVideoController(conference);
+        this._sendVideoController = new SendVideoController(conference);
+        this._conference.on(JitsiConferenceEvents._MEDIA_SESSION_STARTED, (session) => {
+            this._codecController.selectPreferredCodec(session);
+            this._receiveVideoController.onMediaSessionStarted(session);
+            this._sendVideoController.onMediaSessionStarted(session);
+        });
+        this._conference.on(JitsiConferenceEvents._MEDIA_SESSION_ACTIVE_CHANGED, () => this._sendVideoController.configureConstraintsForLocalSources());
+        this._conference.on(JitsiConferenceEvents.CONFERENCE_VISITOR_CODECS_CHANGED, (codecList) => this._codecController.updateVisitorCodecs(codecList));
+        // Debounce the calls to codec selection when there is a burst of joins and leaves.
+        const debouncedSelectCodec = this._debounce(() => this._codecController.selectPreferredCodec(this._conference.jvbJingleSession), 1000);
+        this._conference.on(JitsiConferenceEvents.USER_JOINED, debouncedSelectCodec.bind(this));
+        this._conference.on(JitsiConferenceEvents.USER_LEFT, debouncedSelectCodec.bind(this));
+        this._conference.rtc.on(RTCEvents.SENDER_VIDEO_CONSTRAINTS_CHANGED, (videoConstraints) => this._sendVideoController.onSenderConstraintsReceived(videoConstraints));
+        this._conference.on(JitsiConferenceEvents.ENCODE_TIME_STATS_RECEIVED, (tpc, stats) => this._processOutboundRtpStats(tpc, stats));
+    }
+    /**
+     * Creates a debounced function that delays the execution of the provided function until after the specified delay
+     * has elapsed. Unlike typical debounce implementations, the timer does not reset when the function is called again
+     * within the delay period.
+     *
+     * @param {Function} func - The function to be debounced.
+     * @param {number} delay - The delay in milliseconds.
+     * @returns {Function} - The debounced function.
+     */
+    _debounce(func, delay) {
+        return function (...args) {
+            /* eslint-disable @typescript-eslint/no-invalid-this */
+            if (!this._timer) {
+                this._timer = setTimeout(() => {
+                    this._timer = null;
+                    func.apply(this, args);
+                }, delay);
+            }
+            /* eslint-enable @typescript-eslint/no-invalid-this */
+        };
+    }
+    /**
+     * Adjusts the lastN value so that fewer remote video sources are received from the bridge in an attempt to improve
+     * encode resolution of the outbound video streams based on cpuLimited parameter passed. If cpuLimited is false,
+     * the lastN value will slowly be ramped back up to the channelLastN value set in config.js.
+     *
+     * @param {boolean} cpuLimited - whether the endpoint is cpu limited or not.
+     * @returns boolean - Returns true if an action was taken, false otherwise.
+     */
+    _lowerOrRaiseLastN(cpuLimited) {
+        const lastN = this.receiveVideoController.getLastN();
+        let newLastN = lastN;
+        if (cpuLimited && (lastN !== LAST_N_UNLIMITED && lastN <= MIN_LAST_N)) {
+            return false;
+        }
+        // If channelLastN is not set or set to -1 in config.js, the client will ramp up lastN to only up to 25.
+        let { channelLastN = DEFAULT_LAST_N } = this._conference.options.config;
+        channelLastN = channelLastN === LAST_N_UNLIMITED ? DEFAULT_LAST_N : channelLastN;
+        if (cpuLimited) {
+            const videoStreamsReceived = this._conference.getForwardedSources().length;
+            newLastN = Math.floor(videoStreamsReceived / 2);
+            if (newLastN < MIN_LAST_N) {
+                newLastN = MIN_LAST_N;
+            }
+            // Increment lastN by 1 every LAST_N_RAMPUP_TIME (60) secs.
+        }
+        else if (lastN < channelLastN) {
+            newLastN++;
+        }
+        if (newLastN === lastN) {
+            return false;
+        }
+        const isStillLimitedByCpu = newLastN < channelLastN;
+        this.receiveVideoController.setLastNLimitedByCpu(isStillLimitedByCpu);
+        logger.info(`QualityController - setting lastN=${newLastN}, limitedByCpu=${isStillLimitedByCpu}`);
+        this.receiveVideoController.setLastN(newLastN);
+        return true;
+    }
+    /**
+     * Adjusts the requested resolution for remote video sources by updating the receiver constraints in an attempt to
+     * improve the encode resolution of the outbound video streams.
+     * @return {void}
+     */
+    _maybeLowerReceiveResolution() {
+        const currentConstraints = this.receiveVideoController.getCurrentReceiverConstraints();
+        const individualConstraints = currentConstraints.constraints;
+        let maxHeight = 0;
+        if (individualConstraints && Object.keys(individualConstraints).length) {
+            for (const value of Object.values(individualConstraints)) {
+                const v = value;
+                maxHeight = Math.max(maxHeight, v.maxHeight);
+            }
+        }
+        const currentLevel = VIDEO_QUALITY_LEVELS.findIndex(lvl => lvl.height <= maxHeight);
+        // Do not lower the resolution to less than 180p.
+        if (VIDEO_QUALITY_LEVELS[currentLevel].height === 180) {
+            return;
+        }
+        this.receiveVideoController.setPreferredReceiveMaxFrameHeight(VIDEO_QUALITY_LEVELS[currentLevel + 1].height);
+    }
+    /**
+     * Updates the codec preference order for the local endpoint on the active media session and switches the video
+     * codec if needed.
+     *
+     * @param {number} trackId - The track ID of the local video track for which stats have been captured.
+     * @returns {boolean} - Returns true if video codec was changed.
+     */
+    _maybeSwitchVideoCodec(trackId) {
+        const stats = this._encodeTimeStats.get(trackId);
+        const { codec, encodeResolution, localTrack } = stats.get(stats.size() - 1);
+        const codecsByVideoType = VIDEO_CODECS_BY_COMPLEXITY[localTrack.getVideoType()];
+        const codecIndex = codecsByVideoType.findIndex(val => val === codec.toLowerCase());
+        // Do nothing if the encoder is using the lowest complexity codec already.
+        if (codecIndex === codecsByVideoType.length - 1) {
+            return false;
+        }
+        if (!this._limitedByCpuTimeout) {
+            this._limitedByCpuTimeout = window.setTimeout(() => {
+                this._limitedByCpuTimeout = undefined;
+                const updatedStats = this._encodeTimeStats.get(trackId);
+                const latestSourceStats = updatedStats.get(updatedStats.size() - 1);
+                const expectedSendResolution = Math.min(localTrack.maxEnabledResolution, localTrack.getCaptureResolution());
+                // If the encoder is still limited by CPU, switch to a lower complexity codec.
+                if (latestSourceStats.qualityLimitationReason === QualityLimitationReason.CPU
+                    || encodeResolution < expectedSendResolution) {
+                    return this.codecController.changeCodecPreferenceOrder(localTrack, codec);
+                }
+            }, LIMITED_BY_CPU_TIMEOUT);
+        }
+        return false;
+    }
+    /**
+     * Adjusts codec, lastN or receive resolution based on the send resolution (of the outbound streams) and limitation
+     * reported by the browser in the WebRTC stats. Recovery is also attempted if the limitation goes away. No action
+     * is taken if the adaptive mode has been disabled through config.js.
+     *
+     * @param {ISourceStats} sourceStats - The outbound-rtp stats for a local video track.
+     * @returns {void}
+     */
+    _performQualityOptimizations(sourceStats) {
+        // Do not attempt run time adjustments if the adaptive mode is disabled.
+        if (!this._enableAdaptiveMode) {
+            return;
+        }
+        const { encodeResolution, localTrack, qualityLimitationReason, tpc } = sourceStats;
+        // Older browser versions might not report the resolution in the stats.
+        if (!isValidNumber(encodeResolution)) {
+            return;
+        }
+        const trackId = localTrack.rtcId;
+        if (encodeResolution === tpc.calculateExpectedSendResolution(localTrack)) {
+            if (this._limitedByCpuTimeout) {
+                window.clearTimeout(this._limitedByCpuTimeout);
+                this._limitedByCpuTimeout = undefined;
+            }
+            if (qualityLimitationReason === QualityLimitationReason.NONE
+                && this.receiveVideoController.isLastNLimitedByCpu()) {
+                if (!this._lastNRampupTimeout && !this._isLastNRampupBlocked) {
+                    RTCStats.sendStatsEntry(RTCStatsEvents.ENCODER_CPU_RESTRICTED_EVENT, null, false);
+                    // Ramp up the number of received videos if CPU limitation no longer exists. If the cpu
+                    // limitation returns as a consequence, do not attempt to ramp up again, continue to
+                    // increment the lastN value otherwise until it is equal to the channelLastN value.
+                    this._lastNRampupTimeout = window.setTimeout(() => {
+                        this._lastNRampupTimeout = undefined;
+                        const updatedStats = this._encodeTimeStats.get(trackId);
+                        const latestSourceStats = updatedStats.get(updatedStats.size() - 1);
+                        if (latestSourceStats.qualityLimitationReason === QualityLimitationReason.CPU) {
+                            this._isLastNRampupBlocked = true;
+                        }
+                        else {
+                            this._lowerOrRaiseLastN(false /* raise */);
+                        }
+                    }, this._lastNRampupTime);
+                }
+            }
+            return;
+        }
+        // Do nothing if the limitation reason is bandwidth since the browser will dynamically adapt the outbound
+        // resolution based on available uplink bandwith. Otherwise,
+        // 1. Switch the codec to the lowest complexity one incrementally.
+        // 2. Switch to a lower lastN value, cutting the receive videos by half in every iteration until
+        // MIN_LAST_N value is reached.
+        // 3. Lower the receive resolution of individual streams up to 180p.
+        if (qualityLimitationReason === QualityLimitationReason.CPU) {
+            if (this._lastNRampupTimeout) {
+                window.clearTimeout(this._lastNRampupTimeout);
+                this._lastNRampupTimeout = undefined;
+                this._isLastNRampupBlocked = true;
+            }
+            RTCStats.sendStatsEntry(RTCStatsEvents.ENCODER_CPU_RESTRICTED_EVENT, null, true);
+            const codecSwitched = this._maybeSwitchVideoCodec(trackId);
+            if (!codecSwitched && !this._limitedByCpuTimeout) {
+                const lastNChanged = this._lowerOrRaiseLastN(true /* lower */);
+                if (!lastNChanged) {
+                    this.receiveVideoController.setReceiveResolutionLimitedByCpu(true);
+                    this._maybeLowerReceiveResolution();
+                }
+            }
+        }
+    }
+    /**
+     * Processes the outbound RTP stream stats as reported by the WebRTC peerconnection and makes runtime adjustments
+     * to the client for better quality experience if the adaptive mode is enabled.
+     *
+     * @param {TraceablePeerConnection} tpc - The underlying WebRTC peerconnection where stats have been captured.
+     * @param {Map<number, IOutboundRtpStats>} stats - Outbound-rtp stream stats per SSRC.
+     * @returns void
+     */
+    _processOutboundRtpStats(tpc, stats) {
+        const activeSession = this._conference.getActiveMediaSession();
+        // Process stats only for the active media session.
+        if (activeSession.peerconnection !== tpc) {
+            return;
+        }
+        const statsPerTrack = new Map();
+        for (const ssrc of stats.keys()) {
+            const { codec, encodeTime, qualityLimitationReason, resolution, timestamp } = stats.get(ssrc);
+            const track = tpc.getTrackBySSRC(ssrc);
+            const trackId = track.rtcId;
+            let existingStats = statsPerTrack.get(trackId);
+            const encodeResolution = Math.min(resolution?.height, resolution?.width);
+            const ssrcStats = {
+                encodeResolution,
+                encodeTime,
+                qualityLimitationReason
+            };
+            if (existingStats) {
+                existingStats.codec = codec;
+                existingStats.timestamp = timestamp;
+                existingStats.trackStats.push(ssrcStats);
+            }
+            else {
+                existingStats = {
+                    codec,
+                    timestamp,
+                    trackStats: [ssrcStats]
+                };
+                statsPerTrack.set(trackId, existingStats);
+            }
+        }
+        // Aggregate the stats for multiple simulcast streams with different SSRCs but for the same video stream.
+        for (const trackId of statsPerTrack.keys()) {
+            const { codec, timestamp, trackStats } = statsPerTrack.get(trackId);
+            const totalEncodeTime = trackStats
+                .map((stat) => stat.encodeTime)
+                .reduce((totalValue, currentValue) => totalValue + currentValue, 0);
+            const avgEncodeTime = totalEncodeTime / trackStats.length;
+            const { qualityLimitationReason = QualityLimitationReason.NONE } = trackStats
+                .find((stat) => stat.qualityLimitationReason !== QualityLimitationReason.NONE) ?? {};
+            const encodeResolution = trackStats
+                .map((stat) => stat.encodeResolution)
+                .reduce((resolution, currentValue) => Math.max(resolution, currentValue), 0);
+            const localTrack = this._conference.getLocalVideoTracks().find(t => t.rtcId === trackId);
+            const exisitingStats = this._encodeTimeStats.get(trackId);
+            const sourceStats = {
+                avgEncodeTime,
+                codec,
+                encodeResolution,
+                localTrack,
+                qualityLimitationReason,
+                timestamp,
+                tpc
+            };
+            if (exisitingStats) {
+                exisitingStats.add(sourceStats);
+            }
+            else {
+                // Save stats for only the last 5 mins.
+                const data = new FixedSizeArray(300);
+                data.add(sourceStats);
+                this._encodeTimeStats.set(trackId, data);
+            }
+            logger.debug(`Encode stats for ${localTrack}: codec=${codec}, time=${avgEncodeTime},`
+                + `resolution=${encodeResolution}, qualityLimitationReason=${qualityLimitationReason}`);
+            this._performQualityOptimizations(sourceStats);
+        }
+    }
+    /**
+     * Gets the audio controller instance.
+     */
+    get audioController() {
+        return this._audioController;
+    }
+    /**
+     * Gets the codec controller instance.
+     */
+    get codecController() {
+        return this._codecController;
+    }
+    /**
+     * Gets the receive video controller instance.
+     */
+    get receiveVideoController() {
+        return this._receiveVideoController;
+    }
+    /**
+     * Gets the send video controller instance.
+     */
+    get sendVideoController() {
+        return this._sendVideoController;
+    }
+}
+//# sourceMappingURL=QualityController.js.map

package/dist/esm/modules/qualitycontrol/QualityController.js.map

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package/dist/esm/modules/qualitycontrol/ReceiveAudioController.js

@@ -0,0 +1,73 @@
+import { getLogger } from '@jitsi/logger';
+import { isEqual } from 'lodash-es';
+import { ReceiverAudioSubscription } from '../../service/RTC/ReceiverAudioSubscription';
+const logger = getLogger('qc:ReceiveAudioController');
+/**
+ * Controller for managing audio subscriptions in a Jitsi conference. It allows subscribing to remote audio streams
+ * based on different modes such as ALL, EXCLUDE, INCLUDE, and NONE.
+ */
+export class ReceiverAudioController {
+    /**
+     * Creates a new instance of the ReceiverAudioController.
+     */
+    constructor(conference) {
+        this._rtc = conference.rtc;
+        this._sourceList = [];
+        this._subscriptionMode = ReceiverAudioSubscription.ALL;
+    }
+    /**
+     * Gets the current audio subscription option.
+     *
+     * @returns {ReceiverAudioSubscription} The current audio subscription option.
+     */
+    get audioSubscription() {
+        return this._subscriptionMode;
+    }
+    /**
+     * Mutes or unmutes the remote audio streams based on the provided parameter.
+     *
+     * @param {boolean} muted - Indicates whether the remote audio should be muted or not.
+     * @returns {void}
+     */
+    muteRemoteAudio(muted) {
+        this.setAudioSubscriptionMode({
+            mode: muted ? ReceiverAudioSubscription.NONE : ReceiverAudioSubscription.ALL
+        });
+    }
+    /**
+     * Sets the audio subscription options.
+     *
+     * @param message The audio subscription message containing the mode and optional source list.
+     * @returns {void}
+     */
+    setAudioSubscriptionMode(message) {
+        if ((message.mode == ReceiverAudioSubscription.NONE || message.mode == ReceiverAudioSubscription.ALL)
+            && this._subscriptionMode == message.mode) {
+            logger.debug(`Ignoring ReceiverAudioSubscription with mode: ${message.mode}, no change needed.`);
+            return;
+        }
+        this._subscriptionMode = message.mode;
+        if (message.mode == ReceiverAudioSubscription.INCLUDE
+            || message.mode == ReceiverAudioSubscription.EXCLUDE) {
+            if (!message.list?.length) {
+                this._subscriptionMode = message.mode == ReceiverAudioSubscription.INCLUDE
+                    ? ReceiverAudioSubscription.NONE : ReceiverAudioSubscription.ALL;
+            }
+            else if (this._subscriptionMode == message.mode && isEqual(this._sourceList, message.list)) {
+                logger.debug(`Ignoring ReceiverAudioSubscription with mode: ${message.mode},`
+                    + ` sourceList: ${message.list.join(', ')}, no change needed.`);
+                return;
+            }
+            this._sourceList = message.list || [];
+        }
+        else {
+            // Clear the source list for ALL or NONE modes.
+            this._sourceList = [];
+        }
+        this._rtc.sendReceiverAudioSubscriptionMessage({
+            list: this._sourceList,
+            mode: this._subscriptionMode
+        });
+    }
+}
+//# sourceMappingURL=ReceiveAudioController.js.map

package/dist/esm/modules/qualitycontrol/ReceiveAudioController.js.map

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