// Copyright Epic Games, Inc. All Rights Reserved.

import { Logger } from '../Logger/Logger';
import { Config, 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 {
    peerConnection: RTCPeerConnection;
    aggregatedStats: AggregatedStats;
    config: Config;
    preferredCodec: string;
    updateCodecSelection: boolean;

    /**
     * Create a new RTC Peer Connection client
     * @param options - Peer connection Options
     * @param config - The config for our PS experience.
     */
    constructor(
        options: RTCConfiguration,
        config: Config,
        preferredCodec: string
    ) {
        this.config = config;
        this.createPeerConnection(options, preferredCodec);
    }

    createPeerConnection(options: RTCConfiguration, preferredCodec: string) {
        // Set the ICE transport to relay if TURN enabled
        if (this.config.isFlagEnabled(Flags.ForceTURN)) {
            options.iceTransportPolicy = 'relay';
            Logger.Log(
                Logger.GetStackTrace(),
                '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: Event) =>
            this.handleSignalStateChange(ev);
        this.peerConnection.oniceconnectionstatechange = (ev: Event) =>
            this.handleIceConnectionStateChange(ev);
        this.peerConnection.onicegatheringstatechange = (ev: Event) =>
            this.handleIceGatheringStateChange(ev);
        this.peerConnection.ontrack = (ev: RTCTrackEvent) =>
            this.handleOnTrack(ev);
        this.peerConnection.onicecandidate = (ev: RTCPeerConnectionIceEvent) =>
            this.handleIceCandidate(ev);
        this.peerConnection.ondatachannel = (ev: RTCDataChannelEvent) =>
            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
     */
    async createOffer(offerOptions: RTCOfferOptions, config: Config) {
        Logger.Log(Logger.GetStackTrace(), 'Create Offer', 6);

        const isLocalhostConnection =
            location.hostname === 'localhost' ||
            location.hostname === '127.0.0.1';
        const isHttpsConnection = location.protocol === 'https:';
        let useMic = config.isFlagEnabled(Flags.UseMic);
        if (useMic && !(isLocalhostConnection || isHttpsConnection)) {
            useMic = false;
            Logger.Error(
                Logger.GetStackTrace(),
                'Microphone access in the browser will not work if you are not on HTTPS or localhost. Disabling mic access.'
            );
            Logger.Error(
                Logger.GetStackTrace(),
                "For testing you can enable HTTP microphone access Chrome by visiting chrome://flags/ and enabling 'unsafely-treat-insecure-origin-as-secure'"
            );
        }

        this.setupTransceiversAsync(useMic).finally(() => {
            this.peerConnection
                ?.createOffer(offerOptions)
                .then((offer: RTCSessionDescriptionInit) => {
                    this.showTextOverlayConnecting();
                    offer.sdp = this.mungeSDP(offer.sdp, useMic);
                    this.peerConnection?.setLocalDescription(offer);
                    this.onSendWebRTCOffer(offer);
                })
                .catch(() => {
                    this.showTextOverlaySetupFailure();
                });
        });
    }

    /**
     *
     */
    async receiveOffer(offer: RTCSessionDescriptionInit, config: Config) {

        Logger.Log(Logger.GetStackTrace(), 'Receive Offer', 6);

        this.peerConnection?.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);
            if (useMic && !(isLocalhostConnection || isHttpsConnection)) {
                useMic = false;
                Logger.Error(
                    Logger.GetStackTrace(),
                    'Microphone access in the browser will not work if you are not on HTTPS or localhost. Disabling mic access.'
                );
                Logger.Error(
                    Logger.GetStackTrace(),
                    "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).finally(() => {
                this.peerConnection
                    ?.createAnswer()
                    .then((Answer: RTCSessionDescriptionInit) => {
                        Answer.sdp = this.mungeSDP(Answer.sdp, useMic);
                        return this.peerConnection?.setLocalDescription(Answer);
                    })
                    .then(() => {
                        this.onSendWebRTCAnswer(
                            this.peerConnection?.currentLocalDescription
                        );
                    })
                    .catch(() => {
                        Logger.Error(
                            Logger.GetStackTrace(),
                            '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: RTCSessionDescriptionInit) {
        this.peerConnection?.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() {
        this.peerConnection?.getStats(null).then((StatsData: RTCStatsReport) => {
            this.aggregatedStats.processStats(StatsData);
            this.onVideoStats(this.aggregatedStats);

            // Update the preferred codec selection based on what was actually negotiated
            if (this.updateCodecSelection && !!this.aggregatedStats.inboundVideoStats.codecId) {
                this.config.setOptionSettingValue(
                    OptionParameters.PreferredCodec,
                    this.aggregatedStats.codecs.get(
                        this.aggregatedStats.inboundVideoStats.codecId
                    )
                );
            }
        });
    }

    /**
     * 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: string, useMic: boolean) {
        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: RTCIceCandidate) {
        Logger.Log(Logger.GetStackTrace(), 'peerconnection handleOnIce', 6);

        // // 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(
                    Logger.GetStackTrace(),
                    `Dropping candidate because it was not TURN relay. | Type= ${iceCandidate.type} | Protocol= ${iceCandidate.protocol} | Address=${iceCandidate.address} | Port=${iceCandidate.port} |`,
                    6
                );
                return;
            }
        }

        this.peerConnection?.addIceCandidate(iceCandidate);
    }

    /**
     * When the RTC Peer Connection Signaling server state Changes
     * @param state - Signaling Server State Change Event
     */
    handleSignalStateChange(state: Event) {
        Logger.Log(
            Logger.GetStackTrace(),
            'signaling state change: ' + state,
            6
        );
    }

    /**
     * Handle when the Ice Connection State Changes
     * @param state - Ice Connection State
     */
    handleIceConnectionStateChange(state: Event) {
        Logger.Log(
            Logger.GetStackTrace(),
            'ice connection state change: ' + state,
            6
        );
        this.onIceConnectionStateChange(state);
    }

    /**
     * Handle when the Ice Gathering State Changes
     * @param state - Ice Gathering State Change
     */
    handleIceGatheringStateChange(state: Event) {
        Logger.Log(
            Logger.GetStackTrace(),
            'ice gathering state change: ' + JSON.stringify(state),
            6
        );
    }

    /**
     * Activates the onTrack method
     * @param event - The webRtc track event
     */
    handleOnTrack(event: RTCTrackEvent) {
        this.onTrack(event);
    }

    /**
     * Activates the onPeerIceCandidate
     * @param event - The peer ice candidate
     */
    handleIceCandidate(event: RTCPeerConnectionIceEvent) {
        this.onPeerIceCandidate(event);
    }

    /**
     * Activates the onDataChannel
     * @param event - The peer's data channel
     */
    handleDataChannel(event: RTCDataChannelEvent) {
        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: RTCTrackEvent) {
        // 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: 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: RTCPeerConnectionIceEvent) {
        // 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: RTCDataChannelEvent) {
        // 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: RTCSessionDescriptionInit) : string[] {
        // We want to build an array of all supported codecs on both sides
        const allSupportedCodecs: Array<string> = new Array<string>();
        const allUECodecs: string[] = this.parseAvailableCodecs(sdp);
        const allBrowserCodecs: string[] = 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: string[] = 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
     */
    async setupTransceiversAsync(useMic: boolean) {

        // Setup a transceiver for receiving video (if we need to)
        let hasVideoReceiver = false;
        for (const transceiver of this.peerConnection?.getTransceivers() ?? []) {
            if (transceiver && transceiver.receiver && transceiver.receiver.track && transceiver.receiver.track.kind === 'video') {
                hasVideoReceiver = true;
                break;
            }
        }
        if(!hasVideoReceiver) {
            this.peerConnection?.addTransceiver('video', { direction: 'recvonly' });
        }

        if (RTCRtpReceiver.getCapabilities && this.preferredCodec != '') {
            for (const transceiver of this.peerConnection?.getTransceivers() ?? []) {
                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: RTCRtpCodecCapability = {
                        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: Array<RTCRtpCodecCapability> = [preferredRTCRtpCodecCapability];

                    // Go through all codecs the browser supports and add them to the list (in any order)
                    RTCRtpReceiver.getCapabilities('video').codecs.forEach((browserSupportedCodec : RTCRtpCodecCapability) => {
                        // Don't add our preferred codec again, but add everything else
                        if(browserSupportedCodec.mimeType != preferredRTCRtpCodecCapability.mimeType) {
                            ourSupportedCodecs.push(browserSupportedCodec);
                        }
                        else if(browserSupportedCodec?.sdpFmtpLine != preferredRTCRtpCodecCapability?.sdpFmtpLine) {
                            ourSupportedCodecs.push(browserSupportedCodec);
                        }
                    });

                    for (const codec of ourSupportedCodecs) {
                        if (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 this.peerConnection?.getTransceivers() ?? []) {
            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) {
                this.peerConnection?.addTransceiver('audio', {
                    direction: 'recvonly'
                });
            }
        } else {
            // 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: MediaStreamConstraints = {
                video: false,
                audio: audioOptions
            };

            // Note using mic on android chrome requires SSL or chrome://flags/ "unsafely-treat-insecure-origin-as-secure"
            const stream = await navigator.mediaDevices.getUserMedia(
                mediaSendOptions
            );
            if (stream) {
                if (hasAudioReceiver) {
                    for (const transceiver of this.peerConnection?.getTransceivers() ?? []) {
                        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') {
                            this.peerConnection?.addTransceiver(track, {
                                direction: 'sendrecv'
                            });
                        }
                    }
                }
            } else {
                if(!hasAudioReceiver) {
                    this.peerConnection?.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: AggregatedStats) {
        // 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: RTCSessionDescriptionInit) {
        // 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: RTCSessionDescriptionInit) {
        // 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: RTCSessionDescriptionInit
    ): Array<string> {
        // No point in updating the available codecs if on FF
        if (!RTCRtpReceiver.getCapabilities)
            return ['Only available on Chrome'];

        const ueSupportedCodecs: Array<string> = [];
        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: string =
                        c.name +
                        ' ' +
                        Object.keys(c.parameters || {})
                            .map((p) => p + '=' + c.parameters[p])
                            .join(';');
                    ueSupportedCodecs.push(codecStr);
                }
            });
        });

        return ueSupportedCodecs;
    }
}