@epicgames-ps/lib-pixelstreamingfrontend-ue5.5 0.1.4 → 0.3.0

package/dist/esm/PeerConnectionController/PeerConnectionController.js

@@ -0,0 +1,581 @@
+// Copyright Epic Games, Inc. All Rights Reserved.
+var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
+    function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
+    return new (P || (P = Promise))(function (resolve, reject) {
+        function fulfilled(value) { try { step(generator.next(value)); } catch (e) { reject(e); } }
+        function rejected(value) { try { step(generator["throw"](value)); } catch (e) { reject(e); } }
+        function step(result) { result.done ? resolve(result.value) : adopt(result.value).then(fulfilled, rejected); }
+        step((generator = generator.apply(thisArg, _arguments || [])).next());
+    });
+};
+import { Logger } from '@epicgames-ps/lib-pixelstreamingcommon-ue5.5';
+import { OptionParameters, Flags } from '../Config/Config';
+import { AggregatedStats } from './AggregatedStats';
+import { parseRtpParameters, splitSections } from 'sdp';
+import { RTCUtils } from '../Util/RTCUtils';
+/**
+ * Handles the Peer Connection
+ */
+export class PeerConnectionController {
+    /**
+     * Create a new RTC Peer Connection client
+     * @param options - Peer connection Options
+     * @param config - The config for our PS experience.
+     */
+    constructor(options, config, preferredCodec) {
+        this.config = config;
+        this.createPeerConnection(options, preferredCodec);
+    }
+    createPeerConnection(options, preferredCodec) {
+        // Set the ICE transport to relay if TURN enabled
+        if (this.config.isFlagEnabled(Flags.ForceTURN)) {
+            options.iceTransportPolicy = 'relay';
+            Logger.Info('Forcing TURN usage by setting ICE Transport Policy in peer connection config.');
+        }
+        // build a new peer connection with the options
+        this.peerConnection = new RTCPeerConnection(options);
+        this.peerConnection.onsignalingstatechange = (ev) => this.handleSignalStateChange(ev);
+        this.peerConnection.oniceconnectionstatechange = (ev) => this.handleIceConnectionStateChange(ev);
+        this.peerConnection.onicegatheringstatechange = (ev) => this.handleIceGatheringStateChange(ev);
+        this.peerConnection.ontrack = (ev) => this.handleOnTrack(ev);
+        this.peerConnection.onicecandidate = (ev) => this.handleIceCandidate(ev);
+        this.peerConnection.ondatachannel = (ev) => this.handleDataChannel(ev);
+        this.aggregatedStats = new AggregatedStats();
+        this.preferredCodec = preferredCodec;
+        this.updateCodecSelection = true;
+    }
+    /**
+     * Create an offer for the Web RTC handshake and send the offer to the signaling server via websocket
+     * @param offerOptions - RTC Offer Options
+     */
+    createOffer(offerOptions, config) {
+        return __awaiter(this, void 0, void 0, function* () {
+            Logger.Info('Create Offer');
+            const isLocalhostConnection = location.hostname === 'localhost' || location.hostname === '127.0.0.1';
+            const isHttpsConnection = location.protocol === 'https:';
+            let useMic = config.isFlagEnabled(Flags.UseMic);
+            let useCamera = config.isFlagEnabled(Flags.UseCamera);
+            if ((useMic || useCamera) && !(isLocalhostConnection || isHttpsConnection)) {
+                useMic = false;
+                useCamera = false;
+                Logger.Error('Microphone and Webcam access in the browser will not work if you are not on HTTPS or localhost. Disabling mic and webcam access.');
+                Logger.Error("For testing you can enable HTTP microphone access Chrome by visiting chrome://flags/ and enabling 'unsafely-treat-insecure-origin-as-secure'");
+            }
+            this.setupTransceiversAsync(useMic, useCamera).finally(() => {
+                var _a;
+                (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.createOffer(offerOptions).then((offer) => {
+                    var _a;
+                    this.showTextOverlayConnecting();
+                    offer.sdp = this.mungeSDP(offer.sdp, useMic);
+                    (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.setLocalDescription(offer);
+                    this.onSendWebRTCOffer(offer);
+                }).catch(() => {
+                    this.showTextOverlaySetupFailure();
+                });
+            });
+        });
+    }
+    /**
+     *
+     */
+    receiveOffer(offer, config) {
+        var _a;
+        return __awaiter(this, void 0, void 0, function* () {
+            Logger.Info('Receive Offer');
+            (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.setRemoteDescription(offer).then(() => {
+                const isLocalhostConnection = location.hostname === 'localhost' || location.hostname === '127.0.0.1';
+                const isHttpsConnection = location.protocol === 'https:';
+                let useMic = config.isFlagEnabled(Flags.UseMic);
+                let useCamera = config.isFlagEnabled(Flags.UseCamera);
+                if ((useMic || useCamera) && !(isLocalhostConnection || isHttpsConnection)) {
+                    useMic = false;
+                    useCamera = false;
+                    Logger.Error('Microphone and Webcam access in the browser will not work if you are not on HTTPS or localhost. Disabling mic and webcam access.');
+                    Logger.Error("For testing you can enable HTTP microphone access Chrome by visiting chrome://flags/ and enabling 'unsafely-treat-insecure-origin-as-secure'");
+                }
+                // Add our list of preferred codecs, in order of preference
+                this.config.setOptionSettingOptions(OptionParameters.PreferredCodec, this.fuzzyIntersectUEAndBrowserCodecs(offer));
+                this.setupTransceiversAsync(useMic, useCamera).finally(() => {
+                    var _a;
+                    (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.createAnswer().then((Answer) => {
+                        var _a;
+                        Answer.sdp = this.mungeSDP(Answer.sdp, useMic);
+                        return (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.setLocalDescription(Answer);
+                    }).then(() => {
+                        var _a;
+                        this.onSendWebRTCAnswer((_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.currentLocalDescription);
+                    }).catch(() => {
+                        Logger.Error('createAnswer() failed');
+                    });
+                });
+            });
+        });
+    }
+    /**
+     * Set the Remote Descriptor from the signaling server to the RTC Peer Connection
+     * @param answer - RTC Session Descriptor from the Signaling Server
+     */
+    receiveAnswer(answer) {
+        var _a;
+        (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.setRemoteDescription(answer);
+        // Add our list of preferred codecs, in order of preference
+        this.config.setOptionSettingOptions(OptionParameters.PreferredCodec, this.fuzzyIntersectUEAndBrowserCodecs(answer));
+    }
+    /**
+     * Generate Aggregated Stats and then fire a onVideo Stats event
+     */
+    generateStats() {
+        var _a, _b;
+        const audioPromise = this.audioTrack
+            ? (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.getStats(this.audioTrack).then((statsData) => {
+                this.aggregatedStats.processStats(statsData);
+            })
+            : Promise.resolve();
+        const videoPromise = this.videoTrack
+            ? (_b = this.peerConnection) === null || _b === void 0 ? void 0 : _b.getStats(this.videoTrack).then((statsData) => {
+                this.aggregatedStats.processStats(statsData);
+            })
+            : Promise.resolve();
+        Promise.allSettled([audioPromise, videoPromise]).then(() => {
+            this.onVideoStats(this.aggregatedStats);
+            // Update the preferred codec selection based on what was actually negotiated
+            if (this.updateCodecSelection && !!this.aggregatedStats.inboundVideoStats.codecId) {
+                // Construct the qualified codec name from the mimetype and fmtp
+                const codecStats = this.aggregatedStats.codecs.get(this.aggregatedStats.inboundVideoStats.codecId);
+                const codecShortname = codecStats.mimeType.replace('video/', '');
+                let fullCodecName = codecShortname;
+                if (codecStats.sdpFmtpLine && codecStats.sdpFmtpLine.trim() !== '') {
+                    fullCodecName = `${codecShortname} ${codecStats.sdpFmtpLine.trim()}`;
+                }
+                const allBrowserCodecs = this.config.getSettingOption(OptionParameters.PreferredCodec).options;
+                // The list of codecs directly contains the one that was negotiated, select that
+                if (allBrowserCodecs.includes(fullCodecName)) {
+                    this.config.setOptionSettingValue(OptionParameters.PreferredCodec, fullCodecName);
+                    return;
+                }
+                // If we couldn't match on the full name, try to match on just the codec shortname
+                const filteredList = allBrowserCodecs.filter((option) => option.indexOf(codecShortname) !== -1);
+                if (filteredList.length > 0) {
+                    this.config.setOptionSettingValue(OptionParameters.PreferredCodec, filteredList[0]);
+                    return;
+                }
+            }
+        });
+    }
+    /**
+     * Close The Peer Connection
+     */
+    close() {
+        if (this.peerConnection) {
+            this.peerConnection.close();
+            this.peerConnection = null;
+        }
+    }
+    /**
+     * Modify the Session Descriptor
+     * @param sdp - Session Descriptor as a string
+     * @param useMic - Is the microphone in use
+     * @returns A modified Session Descriptor
+     */
+    mungeSDP(sdp, useMic) {
+        let mungedSDP = sdp.replace(/(a=fmtp:\d+ .*level-asymmetry-allowed=.*)\r\n/gm, '$1;x-google-start-bitrate=10000;x-google-max-bitrate=100000\r\n');
+        // set max bitrate to highest bitrate Opus supports
+        let audioSDP = 'maxaveragebitrate=510000;';
+        if (useMic) {
+            // set the max capture rate to 48khz (so we can send high quality audio from mic)
+            audioSDP += 'sprop-maxcapturerate=48000;';
+        }
+        // Force mono or stereo based on whether ?forceMono was passed or not
+        audioSDP += this.config.isFlagEnabled(Flags.ForceMonoAudio) ? 'stereo=0;' : 'stereo=1;';
+        // enable in-band forward error correction for opus audio
+        audioSDP += 'useinbandfec=1';
+        // We use the line 'useinbandfec=1' (which Opus uses) to set our Opus specific audio parameters.
+        mungedSDP = mungedSDP.replace('useinbandfec=1', audioSDP);
+        return mungedSDP;
+    }
+    /**
+     * When a Ice Candidate is received add to the RTC Peer Connection
+     * @param iceCandidate - RTC Ice Candidate from the Signaling Server
+     */
+    handleOnIce(iceCandidate) {
+        var _a;
+        Logger.Info('peerconnection handleOnIce');
+        // // if forcing TURN, reject any candidates not relay
+        if (this.config.isFlagEnabled(Flags.ForceTURN)) {
+            // check if no relay address is found, if so, we are assuming it means no TURN server
+            if (iceCandidate.candidate.indexOf('relay') < 0) {
+                Logger.Info(`Dropping candidate because it was not TURN relay. | Type= ${iceCandidate.type} | Protocol= ${iceCandidate.protocol} | Address=${iceCandidate.address} | Port=${iceCandidate.port} |`);
+                return;
+            }
+        }
+        (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.addIceCandidate(iceCandidate);
+    }
+    /**
+     * When the RTC Peer Connection Signaling server state Changes
+     * @param state - Signaling Server State Change Event
+     */
+    handleSignalStateChange(state) {
+        Logger.Info('signaling state change: ' + state);
+    }
+    /**
+     * Handle when the Ice Connection State Changes
+     * @param state - Ice Connection State
+     */
+    handleIceConnectionStateChange(state) {
+        Logger.Info('ice connection state change: ' + state);
+        this.onIceConnectionStateChange(state);
+    }
+    /**
+     * Handle when the Ice Gathering State Changes
+     * @param state - Ice Gathering State Change
+     */
+    handleIceGatheringStateChange(state) {
+        Logger.Info('ice gathering state change: ' + JSON.stringify(state));
+    }
+    /**
+     * Activates the onTrack method
+     * @param event - The webRtc track event
+     */
+    handleOnTrack(event) {
+        if (event.streams.length < 1 || event.streams[0].id == 'probator') {
+            return;
+        }
+        if (event.track.kind == 'video') {
+            this.videoTrack = event.track;
+        }
+        if (event.track.kind == 'audio') {
+            this.audioTrack = event.track;
+        }
+        this.onTrack(event);
+    }
+    /**
+     * Activates the onPeerIceCandidate
+     * @param event - The peer ice candidate
+     */
+    handleIceCandidate(event) {
+        this.onPeerIceCandidate(event);
+    }
+    /**
+     * Activates the onDataChannel
+     * @param event - The peer's data channel
+     */
+    handleDataChannel(event) {
+        this.onDataChannel(event);
+    }
+    /**
+     * An override method for onTrack for use outside of the PeerConnectionController
+     * @param trackEvent - The webRtc track event
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onTrack(trackEvent) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * An override method for onIceConnectionStateChange for use outside of the PeerConnectionController
+     * @param event - The webRtc iceconnectionstatechange event
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onIceConnectionStateChange(event) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * An override method for onPeerIceCandidate for use outside of the PeerConnectionController
+     * @param peerConnectionIceEvent - The peer ice candidate
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onPeerIceCandidate(peerConnectionIceEvent) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * An override method for onDataChannel for use outside of the PeerConnectionController
+     * @param datachannelEvent - The peer's data channel
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onDataChannel(datachannelEvent) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * Find the intersection between UE and browser codecs, with fuzzy matching if some parameters are mismatched.
+     * @param sdp The remote sdp
+     * @returns The intersection between browser supported codecs and ue supported codecs.
+     */
+    fuzzyIntersectUEAndBrowserCodecs(sdp) {
+        // We want to build an array of all supported codecs on both sides
+        const allSupportedCodecs = new Array();
+        const allUECodecs = this.parseAvailableCodecs(sdp);
+        const allBrowserCodecs = this.config.getSettingOption(OptionParameters.PreferredCodec).options;
+        for (const ueCodec of allUECodecs) {
+            // Check if browser codecs directly matches UE codec (with parameters and everything)
+            if (allBrowserCodecs.includes(ueCodec)) {
+                allSupportedCodecs.push(ueCodec);
+                continue;
+            }
+            // Otherwise check if browser codec at least contains a match for the UE codec name (without parameters).
+            else {
+                const ueCodecNameAndParams = ueCodec.split(' ');
+                const ueCodecName = ueCodecNameAndParams[0];
+                for (const browserCodec of allBrowserCodecs) {
+                    if (browserCodec.includes(ueCodecName)) {
+                        // We pass browser codec here as they option contain extra parameters.
+                        allSupportedCodecs.push(browserCodec);
+                        break;
+                    }
+                }
+            }
+        }
+        return allSupportedCodecs;
+    }
+    /**
+     * Setup tracks on the RTC Peer Connection
+     * @param useMic - is mic in use
+     * @param useCamera - is webcam in use
+     */
+    setupTransceiversAsync(useMic, useCamera) {
+        var _a, _b, _c, _d, _e, _f, _g, _h;
+        return __awaiter(this, void 0, void 0, function* () {
+            let hasVideoReceiver = false;
+            for (const transceiver of (_b = (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.getTransceivers()) !== null && _b !== void 0 ? _b : []) {
+                if (transceiver &&
+                    transceiver.receiver &&
+                    transceiver.receiver.track &&
+                    transceiver.receiver.track.kind === 'video') {
+                    hasVideoReceiver = true;
+                    break;
+                }
+            }
+            // Setup a transceiver for sending webcam video to UE and receiving video from UE
+            if (!useCamera) {
+                if (!hasVideoReceiver) {
+                    (_c = this.peerConnection) === null || _c === void 0 ? void 0 : _c.addTransceiver('video', { direction: 'recvonly' });
+                }
+            }
+            else {
+                yield this.setupVideoSender(hasVideoReceiver);
+            }
+            if (RTCRtpReceiver.getCapabilities && this.preferredCodec != '') {
+                for (const transceiver of (_e = (_d = this.peerConnection) === null || _d === void 0 ? void 0 : _d.getTransceivers()) !== null && _e !== void 0 ? _e : []) {
+                    if (transceiver &&
+                        transceiver.receiver &&
+                        transceiver.receiver.track &&
+                        transceiver.receiver.track.kind === 'video' &&
+                        transceiver.setCodecPreferences) {
+                        // Get our preferred codec from the codecs options drop down
+                        const preferredRTPCodec = this.preferredCodec.split(' ');
+                        const preferredRTCRtpCodecCapability = {
+                            mimeType: 'video/' + preferredRTPCodec[0] /* Name */,
+                            clockRate: 90000 /* All current video formats in browsers have 90khz clock rate */,
+                            sdpFmtpLine: preferredRTPCodec[1] ? preferredRTPCodec[1] : ''
+                        };
+                        // Populate a list of codecs we will support with our preferred one in the first position
+                        const ourSupportedCodecs = [preferredRTCRtpCodecCapability];
+                        // Go through all codecs the browser supports and add them to the list (in any order)
+                        RTCRtpReceiver.getCapabilities('video').codecs.forEach((browserSupportedCodec) => {
+                            // Don't add our preferred codec again, but add everything else
+                            if (browserSupportedCodec.mimeType != preferredRTCRtpCodecCapability.mimeType) {
+                                ourSupportedCodecs.push(browserSupportedCodec);
+                            }
+                            else if ((browserSupportedCodec === null || browserSupportedCodec === void 0 ? void 0 : browserSupportedCodec.sdpFmtpLine) !=
+                                (preferredRTCRtpCodecCapability === null || preferredRTCRtpCodecCapability === void 0 ? void 0 : preferredRTCRtpCodecCapability.sdpFmtpLine)) {
+                                ourSupportedCodecs.push(browserSupportedCodec);
+                            }
+                        });
+                        for (const codec of ourSupportedCodecs) {
+                            if ((codec === null || codec === void 0 ? void 0 : codec.sdpFmtpLine) === undefined || codec.sdpFmtpLine === '') {
+                                // We can't dynamically add members to the codec, so instead remove the field if it's empty
+                                delete codec.sdpFmtpLine;
+                            }
+                        }
+                        transceiver.setCodecPreferences(ourSupportedCodecs);
+                    }
+                }
+            }
+            let hasAudioReceiver = false;
+            for (const transceiver of (_g = (_f = this.peerConnection) === null || _f === void 0 ? void 0 : _f.getTransceivers()) !== null && _g !== void 0 ? _g : []) {
+                if (transceiver &&
+                    transceiver.receiver &&
+                    transceiver.receiver.track &&
+                    transceiver.receiver.track.kind === 'audio') {
+                    hasAudioReceiver = true;
+                    break;
+                }
+            }
+            // Setup a transceiver for sending mic audio to UE and receiving audio from UE
+            if (!useMic) {
+                if (!hasAudioReceiver) {
+                    (_h = this.peerConnection) === null || _h === void 0 ? void 0 : _h.addTransceiver('audio', {
+                        direction: 'recvonly'
+                    });
+                }
+            }
+            else {
+                yield this.setupAudioSender(hasAudioReceiver);
+            }
+        });
+    }
+    setupVideoSender(hasVideoReceiver) {
+        var _a, _b, _c, _d;
+        return __awaiter(this, void 0, void 0, function* () {
+            // set the media send options
+            const mediaSendOptions = {
+                video: true
+            };
+            // Note using webcam on android chrome requires SSL or chrome://flags/ "unsafely-treat-insecure-origin-as-secure"
+            const stream = yield navigator.mediaDevices.getUserMedia(mediaSendOptions);
+            if (stream) {
+                if (hasVideoReceiver) {
+                    for (const transceiver of (_b = (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.getTransceivers()) !== null && _b !== void 0 ? _b : []) {
+                        if (RTCUtils.canTransceiverReceiveVideo(transceiver)) {
+                            for (const track of stream.getTracks()) {
+                                if (track.kind && track.kind == 'video') {
+                                    transceiver.sender.replaceTrack(track);
+                                    transceiver.direction = 'sendrecv';
+                                }
+                            }
+                        }
+                    }
+                }
+                else {
+                    for (const track of stream.getTracks()) {
+                        if (track.kind && track.kind == 'video') {
+                            (_c = this.peerConnection) === null || _c === void 0 ? void 0 : _c.addTransceiver(track, {
+                                direction: 'sendrecv'
+                            });
+                        }
+                    }
+                }
+            }
+            else {
+                if (!hasVideoReceiver) {
+                    (_d = this.peerConnection) === null || _d === void 0 ? void 0 : _d.addTransceiver('video', { direction: 'recvonly' });
+                }
+            }
+        });
+    }
+    setupAudioSender(hasAudioReceiver) {
+        var _a, _b, _c, _d;
+        return __awaiter(this, void 0, void 0, function* () {
+            // set the audio options based on mic usage
+            const audioOptions = {
+                autoGainControl: false,
+                channelCount: 1,
+                echoCancellation: false,
+                latency: 0,
+                noiseSuppression: false,
+                sampleRate: 48000,
+                sampleSize: 16,
+                volume: 1.0
+            };
+            // set the media send options
+            const mediaSendOptions = {
+                video: false,
+                audio: audioOptions
+            };
+            // Note using mic on android chrome requires SSL or chrome://flags/ "unsafely-treat-insecure-origin-as-secure"
+            const stream = yield navigator.mediaDevices.getUserMedia(mediaSendOptions);
+            if (stream) {
+                if (hasAudioReceiver) {
+                    for (const transceiver of (_b = (_a = this.peerConnection) === null || _a === void 0 ? void 0 : _a.getTransceivers()) !== null && _b !== void 0 ? _b : []) {
+                        if (RTCUtils.canTransceiverReceiveAudio(transceiver)) {
+                            for (const track of stream.getTracks()) {
+                                if (track.kind && track.kind == 'audio') {
+                                    transceiver.sender.replaceTrack(track);
+                                    transceiver.direction = 'sendrecv';
+                                }
+                            }
+                        }
+                    }
+                }
+                else {
+                    for (const track of stream.getTracks()) {
+                        if (track.kind && track.kind == 'audio') {
+                            (_c = this.peerConnection) === null || _c === void 0 ? void 0 : _c.addTransceiver(track, {
+                                direction: 'sendrecv'
+                            });
+                        }
+                    }
+                }
+            }
+            else {
+                if (!hasAudioReceiver) {
+                    (_d = this.peerConnection) === null || _d === void 0 ? void 0 : _d.addTransceiver('audio', {
+                        direction: 'recvonly'
+                    });
+                }
+            }
+        });
+    }
+    /**
+     * And override event for when the video stats are fired
+     * @param event - Aggregated Stats
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onVideoStats(event) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * Event to send the RTC offer to the Signaling server
+     * @param offer - RTC Offer
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onSendWebRTCOffer(offer) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * Event to send the RTC Answer to the Signaling server
+     * @param answer - RTC Answer
+     */
+    // eslint-disable-next-line @typescript-eslint/no-unused-vars
+    onSendWebRTCAnswer(answer) {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * An override for showing the Peer connection connecting Overlay
+     */
+    showTextOverlayConnecting() {
+        // Default Functionality: Do Nothing
+    }
+    /**
+     * An override for showing the Peer connection Failed overlay
+     */
+    showTextOverlaySetupFailure() {
+        // Default Functionality: Do Nothing
+    }
+    parseAvailableCodecs(rtcSessionDescription) {
+        // No point in updating the available codecs if on FF
+        if (!RTCRtpReceiver.getCapabilities)
+            return ['Only available on Chrome'];
+        const ueSupportedCodecs = [];
+        const sections = splitSections(rtcSessionDescription.sdp);
+        // discard the session information as we only want media related info
+        sections.shift();
+        sections.forEach((mediaSection) => {
+            const { codecs } = parseRtpParameters(mediaSection);
+            // Filter only for VPX / H26X / AV1
+            const matcher = /(VP\d|H26\d|AV1).*/;
+            codecs.forEach((c) => {
+                const str = c.name +
+                    ' ' +
+                    Object.keys(c.parameters || {})
+                        .map((p) => p + '=' + c.parameters[p])
+                        .join(';');
+                const match = matcher.exec(str);
+                if (match !== null) {
+                    if (c.name == 'VP9') {
+                        // UE answers don't specify profile but we know we want profile 0
+                        c.parameters = {
+                            'profile-id': '0'
+                        };
+                    }
+                    const codecStr = c.name +
+                        ' ' +
+                        Object.keys(c.parameters || {})
+                            .map((p) => p + '=' + c.parameters[p])
+                            .join(';');
+                    ueSupportedCodecs.push(codecStr);
+                }
+            });
+        });
+        return ueSupportedCodecs;
+    }
+}
+//# sourceMappingURL=PeerConnectionController.js.map

package/dist/esm/PeerConnectionController/PeerConnectionController.js.map

@@ -0,0 +1 @@
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package/dist/esm/PeerConnectionController/SessionStats.js

@@ -0,0 +1,7 @@
+// Copyright Epic Games, Inc. All Rights Reserved.
+/**
+ * Session statistics
+ */
+export class SessionStats {
+}
+//# sourceMappingURL=SessionStats.js.map

package/dist/esm/PeerConnectionController/SessionStats.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"SessionStats.js","sourceRoot":"","sources":["../../../src/PeerConnectionController/SessionStats.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD;;GAEG;AACH,MAAM,OAAO,YAAY;CAIxB"}

package/dist/esm/PeerConnectionController/StreamStats.js

@@ -0,0 +1,7 @@
+// Copyright Epic Games, Inc. All Rights Reserved.
+/**
+ * Class to hold the stream stats data coming in from webRtc
+ */
+export class StreamStats {
+}
+//# sourceMappingURL=StreamStats.js.map

package/dist/esm/PeerConnectionController/StreamStats.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"StreamStats.js","sourceRoot":"","sources":["../../../src/PeerConnectionController/StreamStats.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD;;GAEG;AACH,MAAM,OAAO,WAAW;CAKvB"}