require('dotenv').config()

const express = require('express');
const app = express();
const Server = require('socket.io');
const path = require('node:path');
const fs = require('node:fs');
let https = require('https');
try {
  https = require('node:https');
} catch (err) {
  console.log('https support is disabled!');
}
const mediasoup = require('mediasoup');

let worker
/**
 * videoCalls
 * |-> Router
 * |-> Producer
 * |-> Consumer 
 * |-> Producer Transport
 * |-> Consumer Transport
 * 
 * '<callId>': {
 *  router: Router,
 *  producer: Producer,
 *  producerTransport: Producer Transport,
 *  consumer: Consumer,
 *  consumerTransport: Consumer Transport
 * }
 * 
 **/
let videoCalls = {}
let socketDetails = {}

app.get('/', (_req, res) => {
  res.send('Hello from mediasoup app!')
})

app.use('/sfu', express.static(path.join(__dirname, 'public')))

// SSL cert for HTTPS access
const options = {
  key: fs.readFileSync('./server/ssl/key.pem', 'utf-8'),
  cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8'),
}

const httpsServer = https.createServer(options, app);

const io = new Server(httpsServer, {
  allowEIO3: true,
  origins: ["*:*"],
  // allowRequest: (req, next) => {
  //   console.log('req', req);
  //   next(null, true)
  // }
});
// const io = new Server(server, { origins: '*:*', allowEIO3: true });

httpsServer.listen(process.env.PORT, () => {
  console.log('Video server listening on port:', process.env.PORT)
})

const peers = io.of('/')

const createWorker = async () => {
  try {
    worker = await mediasoup.createWorker({
      rtcMinPort: 2000,
      rtcMaxPort: 2020,
    })
    console.log(`[createWorker] worker pid ${worker.pid}`);
  
    worker.on('died', error => {
      // This implies something serious happened, so kill the application
      console.error('mediasoup worker has died', error);
      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
    })
    return worker;
  } catch (error) {
    console.log(`ERROR | createWorker | ${error.message}`);
  }
}

// We create a Worker as soon as our application starts
worker = createWorker()

// This is an Array of RtpCapabilities
// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
// list of media codecs supported by mediasoup ...
// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
const mediaCodecs = [
  {
    kind: 'audio',
    mimeType: 'audio/opus',
    clockRate: 48000,
    channels: 2,
  },
  {
    kind: 'video',
    mimeType: 'video/VP8',
    clockRate: 90000,
    parameters: {
      'x-google-start-bitrate': 1000,
    },
  },
]

const closeCall = (callId) => {
  try {
    if (videoCalls[callId]) {
      videoCalls[callId].producer?.close();
      videoCalls[callId].consumer?.close();
      videoCalls[callId]?.consumerTransport?.close();
      videoCalls[callId]?.producerTransport?.close();
      videoCalls[callId]?.router?.close();
      delete videoCalls[callId];
    } else {
      console.log(`The call with id ${callId} has already been deleted`);
    }
  } catch (error) {
    console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
  }
}

const getRtpCapabilities = (callId, callback) => {
  try {
    console.log('[getRtpCapabilities] callId', callId);
    const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
    callback({ rtpCapabilities });
  } catch (error) {
    console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
  }
}

/*
  - Handlers for WS events
  - These are created only when we have a connection with a peer
*/
peers.on('connection', async socket => {
  console.log('[connection] socketId:', socket.id);

  // After making the connection successfully, we send the client a 'connection-success' event
  socket.emit('connection-success', {
    socketId: socket.id
  });

  // It is triggered when the peer is disconnected
  socket.on('disconnect', () => {
    const callId = socketDetails[socket.id];
    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
    delete socketDetails[socket.id];
    closeCall(callId);
  });

  /*
    - This event creates a room with the roomId and the callId sent
    - It will return the rtpCapabilities of that room
    - If the room already exists, it will not create it, but will only return rtpCapabilities
  */
  socket.on('createRoom', async ({ callId }, callback) => {
    try {
      if (callId) {
        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
        if (!videoCalls[callId]) {
          console.log('[createRoom] callId', callId);
          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
          console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
        }
        socketDetails[socket.id] = callId;
        getRtpCapabilities(callId, callback);
      } else {
        console.log(`[createRoom] missing callId ${callId}`);
      }
    } catch (error) {
      console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
    }
  });

  /*
    - Client emits a request to create server side Transport
    - Depending on the sender, producerTransport or consumerTransport is created on that router
    - It will return parameters, these are required for the client to create the RecvTransport 
    from the client.
      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
  */
  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
    try {
      const callId = socketDetails[socket.id];
      console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
      if (sender) {
        if (!videoCalls[callId].producerTransport) {
          videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
        } else {
          console.log(`producerTransport has already been defined | callId ${callId}`);
        }
      } else if (!sender) {
        if (!videoCalls[callId].consumerTransport) {
          videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
        } else {
          console.log(`consumerTransport has already been defined | callId ${callId}`);
        }
      }
    } catch (error) {
      console.log(`ERROR | createWebRtcTransport | callId ${callId} | sender ${sender} | ${error.message}`);
    }
  });

  /*
    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
    - The connection is made to the created transport
  */
  socket.on('transport-connect', async ({ dtlsParameters }) => {
    try {
      const callId = socketDetails[socket.id];
      console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`)
      await videoCalls[callId].producerTransport.connect({ dtlsParameters });
    } catch (error) {
      console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
    }
  });

  /*
    - The event sent by the client (PRODUCER) after successfully connecting to producerTransport
    - For the router with the id callId, we make produce on producerTransport
    - Create the handler on producer at the 'transportclose' event
  */
  socket.on('transport-produce', async ({ kind, rtpParameters, appData }) => {
    try {
      const callId = socketDetails[socket.id];
      console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
        kind,
        rtpParameters,
      });
      console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
  
      videoCalls[callId].producer.on('transportclose', () => {
        const callId = socketDetails[socket.id];
        console.log('transport for this producer closed', callId)
        closeCall(callId);
      });
    } catch (error) {
      console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
    }
  });

  /*
    - The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
    - The connection is made to the created consumerTransport
  */
  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
    try {
      const callId = socketDetails[socket.id];
      console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
      await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
    } catch (error) {
      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
    }
  })

  /*
    - The customer consumes after successfully connecting to consumerTransport
    - The previous step was 'transport-recv-connect', and before that 'createWebRtcTransport'
    - This event is only sent by the consumer
    - The parameters that the consumer consumes are returned
      - The consumer does consumerTransport.consume(params)
  */
  socket.on('consume', async ({ rtpCapabilities }, callback) => {
    try {
      const callId = socketDetails[socket.id];
      console.log('[consume] callId', callId);

      // Check if the router can consume the specified producer
      if (videoCalls[callId].router.canConsume({
        producerId: videoCalls[callId].producer.id,
        rtpCapabilities
      })) {
        console.log('[consume] Can consume', callId);
        // Transport can now consume and return a consumer
        videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
          producerId: videoCalls[callId].producer.id,
          rtpCapabilities,
          paused: true,
        });

        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
        videoCalls[callId].consumer.on('transportclose', () => {
          const callId = socketDetails[socket.id];
          console.log('transport close from consumer', callId);
          closeCall();
        });

        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
        videoCalls[callId].consumer.on('producerclose', () => {
          const callId = socketDetails[socket.id];
          console.log('producer of consumer closed', callId);
          closeCall();
        });

        // From the consumer extract the following params to send back to the Client
        const params = {
          id: videoCalls[callId].consumer.id,
          producerId: videoCalls[callId].producer.id,
          kind: videoCalls[callId].consumer.kind,
          rtpParameters: videoCalls[callId].consumer.rtpParameters,
        };

        // Send the parameters to the client
        callback({ params });
      } else {
        console.log(`[canConsume] Can't consume | callId ${callId}`);
      }
    } catch (error) {
      console.log(`ERROR | consume | callId ${callId} | ${error.message}`)
      callback({ params: { error } });
    }
  });

  /*
    - Event sent by the consumer after consuming to resume the pause
    - When consuming on consumerTransport, it is initially done with paused: true, here we will resume 
  */
  socket.on('consumer-resume', async () => {
    try {
      const callId = socketDetails[socket.id];
      console.log(`[consumer-resume] callId ${callId}`)
      await videoCalls[callId].consumer.resume();
    } catch (error) {
      console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
    }
  });
});

/*
  - Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
  - It will return parameters, these are required for the client to create the RecvTransport 
    from the client.
    - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
    - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
*/
const createWebRtcTransportLayer = async (callId, callback) => {
  try {
    console.log('[createWebRtcTransportLayer] callId', callId);
    // https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
    const webRtcTransport_options = {
      listenIps: [
        {
          ip: process.env.IP, // Listening IPv4 or IPv6.
          announcedIp: process.env.ANNOUNCED_IP, // Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
        }
      ],
      enableUdp: true,
      enableTcp: true,
      preferUdp: true,
    };
    
    // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
    console.log(`callId: ${callId} | transport id: ${transport.id}`)

    // Handler for when DTLS(Datagram Transport Layer Security) changes
    transport.on('dtlsstatechange', dtlsState => {
      console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
      if (dtlsState === 'closed') {
        transport.close();
      }
    });

    // Handler if the transport layer has closed (for various reasons)
    transport.on('close', () => {
      console.log(`transport | closed | calldId ${callId}`);
    });

    const params = {
      id: transport.id,
      iceParameters: transport.iceParameters,
      iceCandidates: transport.iceCandidates,
      dtlsParameters: transport.dtlsParameters,
    };

    // Send back to the client the params
    callback({ params });

    // Set transport to producerTransport or consumerTransport
    return transport;

  } catch (error) {
    console.log(`ERROR | createWebRtcTransportLayer | callId ${callId} | ${error.message}`);
    callback({ params: { error } });
  }
}