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• Why does My Discord bot stop playing music

  27 septembre 2021, par Konglnwza

  So My Discord bot will be able to play music for a while then it will stop and i have to restart the bot to fix it.
And if it stopped then i use command skip it will crash the bot with the errors below

  


  C:\Users\User\Desktop\Discord bot\Song\play.js:96&#xA;    server_queue.connection.dispatcher.end();&#xA;                                       ^&#xA;&#xA;TypeError: Cannot read property &#x27;end&#x27; of null&#xA;    at skip_song (C:\Users\User\Desktop\Discord bot\Song\play.js:96:40)&#xA;    at Object.execute (C:\Users\User\Desktop\Discord bot\Song\play.js:65:47)&#xA;    at Client.<anonymous> (C:\Users\User\Desktop\Discord bot\bot.js:78:74)&#xA;    at Client.emit (node:events:394:28)&#xA;    at MessageCreateAction.handle (C:\Users\User\Desktop\Discord bot\node_modules\discord.js\src\client\actions\MessageCreate.js:31:14)&#xA;    at Object.module.exports [as MESSAGE_CREATE] (C:\Users\User\Desktop\Discord bot\node_modules\discord.js\src\client\websocket\handlers\MESSAGE_CREATE.js:4:32)&#xA;    at WebSocketManager.handlePacket (C:\Users\User\Desktop\Discord bot\node_modules\discord.js\src\client\websocket\WebSocketManager.js:384:31)&#xA;    at WebSocketShard.onPacket (C:\Users\User\Desktop\Discord bot\node_modules\discord.js\src\client\websocket\WebSocketShard.js:444:22)&#xA;    at WebSocketShard.onMessage (C:\Users\User\Desktop\Discord bot\node_modules\discord.js\src\client\websocket\WebSocketShard.js:301:10)&#xA;    at WebSocket.onMessage (C:\Users\User\Desktop\Discord bot\node_modules\ws\lib\event-target.js:132:16)&#xA;</anonymous>

  


  I guess the problem is something with ffmpeg or i must have some npms more.

  


  npm i already installed

  


  

  • discord.js

  


  • ytdl-core

  


  • ytdl-search
and also i already installed ffmpeg in my computer and set the path

  


  


  i want to ask that do i have to install any npms more ? or anyone know how to fixed this problem ?

  


  const ytdl = require('ytdl-core');
const ytSearch = require('yt-search');

const queue = new Map();

module.exports = {
    name: 'play',
    aliases: ['skip', 'stop', 'queue', 'leave', 'join'],
    description: 'Joins and play',
    async execute(message, args , cmd, client, Discord){
        const voice_channel = message.member.voice.channel;
        if(!voice_channel) return message.channel.send('เข้าไปอยู่ในดิสก่อนดิวะ :angry: ');//you must in voice channel
        
        
        const server_queue = queue.get(message.guild.id);
        if(cmd==='play' || cmd==='p'){
            if(!args.length) return message.channel.send('จะเปิดอะไรล่ะพี่ :triumph:');//you must have argument
            let song = {};

            if(ytdl.validateURL(args[0])){
                const song_info = await ytdl.getInfo(args[0]);
                song = {title: song_info.videoDetails.title, url: song_info.videoDetails.video_url}
            } else {
                const videoFinder = async (query) => {
                    const videoResult = await ytSearch(query);
                    return (videoResult.videos.length > 1) ? videoResult.videos[0] : null;
                }

                const video = await videoFinder(args.join(' '));
                if(video){
                    song = {title: video.title, url: video.url}
                
                } else{
                    message.channel.send('หาไม่เจอ :cry: ');//cant find song
                }
            }
            //const connection = await voiceChannel.join(); 
            if(!server_queue){
                const queue_constructor = {
                    voice_channel: voice_channel,
                    text_channel: message.channel,
                    connection: null,
                    songs: []
                }

                queue.set(message.guild.id, queue_constructor);
                queue_constructor.songs.push(song);

                try{
                    const connection = await voice_channel.join();
                    queue_constructor.connection = connection;
                    video_player(message.guild, queue_constructor.songs[0]);
                } catch (err) {
                    queue.delete(message.guild.id);
                    message.channel.send('error');
                    throw err;
                }
            } else{
                server_queue.songs.push(song);
                return message.channel.send(`:regional_indicator_k: :regional_indicator_o: :regional_indicator_n: :regional_indicator_g: :star_struck: **${song.title}** ใส่เข้าคิวแล้ว`)//added to queue
            }
            

        }
        else if(cmd === 'skip' || cmd ==='s') skip_song(message, server_queue);
        else if(cmd === 'clear' || cmd==='c') clear_song(message, server_queue);
        else if(cmd === 'join') join_song(message);
        else if(cmd === 'leave') leave_song(message);
        else if(cmd === 'queue' || cmd ==='q') queue_show(message,server_queue,Discord);


    }
}

const video_player = async (guild, song) => {
    const song_queue = queue.get(guild.id);

    if(!song){
        //song_queue.text_channel.send('ไปละบาย :kissing_heart: ');
        song_queue.voice_channel.leave();
        queue.delete(guild.id);
        return;
    }
    const stream = ytdl(song.url,{filter: 'audioonly'},{ highWaterMark: 1<<25 });
    song_queue.connection.play(stream, { seek: 0, volume: 0.5 })
    .on('finish', () => {
        song_queue.songs.shift();
        video_player(guild, song_queue.songs[0]);
    });
    await song_queue.text_channel.send(`:regional_indicator_k: :regional_indicator_o: :regional_indicator_n: :regional_indicator_g: :sunglasses: กำลังเล่นเด็ก ***${song.title}***`);//playing song
}

const skip_song = (message, server_queue) => {
    if(!message.member.voice.channel) return message.channel.send('เข้าดิสก่อนดิ :angry: ');//you must be in voice channel
    if(!server_queue) return message.channel.send('ไม่มีเพลงในคิวแล้ว :relieved: ');//no song in queue
    server_queue.connection.dispatcher.end();
}

const clear_song = (message, server_queue) => {
    if(!message.member.voice.channel) return message.channel.send('เข้าดิสก่อนดิ :angry: ');//you must be in voice channel
    if(!server_queue) return message.channel.send('ไม่มีเพลงในคิวแล้ว :relieved: ');//no song in queue
    server_queue.songs = [];
    server_queue.connection.dispatcher.end();
}

const join_song = (message) => {
    if(!message.member.voice.channel) return message.channel.send('เข้าดิสก่อนดิ :angry: ');//you must be in voice channel
    message.member.voice.channel.join();
}

const leave_song = (message) => {
    if(!message.member.voice.channel) return message.channel.send('เข้าดิสก่อนดิ :angry: ');//you must be in voice channel
    message.member.voice.channel.leave();
}

const queue_show =  (message,server_queue,Discord) => {
    if(!server_queue) return message.channel.send('ไม่มีเพลงในคิวแล้ว :relieved: ');//no song in queue
    const queueList = server_queue.songs.map((song, i) => `[${++i}] - ${song.title}`);
    const queueEmbed = new Discord.MessageEmbed()
    .setDescription(queueList);
    message.channel.send(queueEmbed);
}```


  


• record mediasoup RTP stream using FFmpeg for Firefox

  30 juillet 2024, par Hadi Aghandeh

  I am trying to record WebRTC stream using mediasoup. I could record successfully on chrome and safari 13/14/15. However on Firefox the does not work.

  


  Client side code is a vue js component which gets rtp-compabilities using socket.io and create producers after the server creates the transports. This works good on chrome and safari.

  


  const { connect , createLocalTracks } = require('twilio-video');
const SocketClient = require("socket.io-client");
const SocketPromise = require("socket.io-promise").default;
const MediasoupClient = require("mediasoup-client");

export default {
    data() {
        return {
            errors: [],
            isReady: false,
            isRecording: false,
            loading: false,
            sapio: {
                token: null,
                connectionId: 0
            },
            server: {
                host: 'https://rtc.test',
                ws: '/server',
                socket: null,
            },
            peer: {},
        }
    },
    mounted() {
        this.init();
    },
    methods: {
        async init() {
            await this.startCamera();

            if (this.takeId) {
                await this.recordBySapioServer();
            }
        },
        startCamera() {
            return new Promise( (resolve, reject) => {
                if (window.videoMediaStreamObject) {
                    this.setVideoElementStream(window.videoMediaStreamObject);
                    resolve();
                } else {
                    // Get user media as required
                    try {
                        this.localeStream = navigator.mediaDevices.getUserMedia({
                            audio: true,
                            video: true,
                        }).then((stream) => {
                            this.setVideoElementStream(stream);
                            resolve();
                        })
                    } catch (err) {
                        console.error(err);
                        reject();
                    }
                }
            })
        },
        setVideoElementStream(stream) {
            this.localStream = stream;
            this.$refs.video.srcObject = stream;
            this.$refs.video.muted = true;
            this.$refs.video.play().then((video) => {
                this.isStreaming = true;
                this.height = this.$refs.video.videoHeight;
                this.width = this.$refs.video.videoWidth;
            });
        },
        // first thing we need is connecting to websocket
        connectToSocket() {
            const serverUrl = this.server.host;
            console.log("Connect with sapio rtc server:", serverUrl);

            const socket = SocketClient(serverUrl, {
                path:  this.server.ws,
                transports: ["websocket"],
            });
            this.socket = socket;

            socket.on("connect", () => {
                console.log("WebSocket connected");
                // we ask for rtp-capabilities from server to send to us
                socket.emit('send-rtp-capabilities');
            });

            socket.on("error", (err) => {
                this.loading = true;
                console.error("WebSocket error:", err);
            });

            socket.on("router-rtp-capabilities", async (msg) => {
                const { routerRtpCapabilities, sessionId, externalId } = msg;
                console.log('[rtpCapabilities:%o]', routerRtpCapabilities);
                this.routerRtpCapabilities = routerRtpCapabilities;

                try {
                    const device = new MediasoupClient.Device();
                    // Load the mediasoup device with the router rtp capabilities gotten from the server
                    await device.load({ routerRtpCapabilities });

                    this.peer.sessionId = sessionId;
                    this.peer.externalId = externalId;
                    this.peer.device = device;

                    this.createTransport();
                } catch (error) {
                    console.error('failed to init device [error:%o]', error);
                    socket.disconnect();
                }
            });

            socket.on("create-transport", async (msg) => {
                console.log('handleCreateTransportRequest() [data:%o]', msg);

                try {
                    // Create the local mediasoup send transport
                    this.peer.sendTransport = await this.peer.device.createSendTransport(msg);
                    console.log('send transport created [id:%s]', this.peer.sendTransport.id);

                    // Set the transport listeners and get the users media stream
                    this.handleSendTransportListeners();
                    this.setTracks();
                    this.loading = false;
                } catch (error) {
                    console.error('failed to create transport [error:%o]', error);
                    socket.disconnect();
                }
            });

            socket.on("connect-transport", async (msg) => {
                console.log('handleTransportConnectRequest()');
                try {
                    const action = this.connectTransport;

                    if (!action) {
                        throw new Error('transport-connect action was not found');
                    }

                    await action(msg);
                } catch (error) {
                    console.error('ailed [error:%o]', error);
                }
            });

            socket.on("produce", async (msg) => {
                console.log('handleProduceRequest()');
                try {
                    if (!this.produce) {
                        throw new Error('produce action was not found');
                    }
                    await this.produce(msg);
                } catch (error) {
                    console.error('failed [error:%o]', error);
                }
            });

            socket.on("recording", async (msg) => {
                this.isRecording = true;
            });

            socket.on("recording-error", async (msg) => {
                this.isRecording = false;
                console.error(msg);
            });

            socket.on("recording-closed", async (msg) => {
                this.isRecording = false;
                console.warn(msg)
            });

        },
        createTransport() {
            console.log('createTransport()');

            if (!this.peer || !this.peer.device.loaded) {
                throw new Error('Peer or device is not initialized');
            }

            // First we must create the mediasoup transport on the server side
            this.socket.emit('create-transport',{
                sessionId: this.peer.sessionId
            });
        },
        handleSendTransportListeners() {
            this.peer.sendTransport.on('connect', this.handleTransportConnectEvent);
            this.peer.sendTransport.on('produce', this.handleTransportProduceEvent);
            this.peer.sendTransport.on('connectionstatechange', connectionState => {
                console.log('send transport connection state change [state:%s]', connectionState);
            });
        },
        handleTransportConnectEvent({ dtlsParameters }, callback, errback) {
            console.log('handleTransportConnectEvent()');
            try {
                this.connectTransport = (msg) => {
                    console.log('connect-transport action');
                    callback();
                    this.connectTransport = null;
                };

                this.socket.emit('connect-transport',{
                    sessionId: this.peer.sessionId,
                    transportId: this.peer.sendTransport.id,
                    dtlsParameters
                });

            } catch (error) {
                console.error('handleTransportConnectEvent() failed [error:%o]', error);
                errback(error);
            }
        },
        handleTransportProduceEvent({ kind, rtpParameters }, callback, errback)  {
            console.log('handleTransportProduceEvent()');
            try {
                this.produce = jsonMessage => {
                    console.log('handleTransportProduceEvent callback [data:%o]', jsonMessage);
                    callback({ id: jsonMessage.id });
                    this.produce = null;
                };

                this.socket.emit('produce', {
                    sessionId: this.peer.sessionId,
                    transportId: this.peer.sendTransport.id,
                    kind,
                    rtpParameters
                });
            } catch (error) {
                console.error('handleTransportProduceEvent() failed [error:%o]', error);
                errback(error);
            }
        },
        async recordBySapioServer() {
            this.loading = true;
            this.connectToSocket();
        },
        async setTracks() {
            // Start mediasoup-client's WebRTC producers
            const audioTrack = this.localStream.getAudioTracks()[0];
            this.peer.audioProducer = await this.peer.sendTransport.produce({
                track: audioTrack,
                codecOptions :
                    {
                        opusStereo : 1,
                        opusDtx    : 1
                    }
            });


            let encodings;
            let codec;
            const codecOptions = {videoGoogleStartBitrate : 1000};

            codec = this.peer.device.rtpCapabilities.codecs.find((c) => c.kind.toLowerCase() === 'video');
            if (codec.mimeType.toLowerCase() === 'video/vp9') {
                encodings = { scalabilityMode: 'S3T3_KEY' };
            } else {
                encodings = [
                    { scaleResolutionDownBy: 4, maxBitrate: 500000 },
                    { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
                    { scaleResolutionDownBy: 1, maxBitrate: 5000000 }
                ];
            }
            const videoTrack = this.localStream.getVideoTracks()[0];
            this.peer.videoProducer =await this.peer.sendTransport.produce({
                track: videoTrack,
                encodings,
                codecOptions,
                codec
            });

        },
        startRecording() {
            this.Q.answer.recordingId = this.peer.externalId;
            this.socket.emit("start-record", {
                sessionId: this.peer.sessionId
            });
        },
        stopRecording() {
            this.socket.emit("stop-record" , {
                sessionId: this.peer.sessionId
            });
        },
    },

}





  


  console.log of my ffmpeg process :

  


  // sdp string
[sdpString:v=0
  o=- 0 0 IN IP4 127.0.0.1
  s=FFmpeg
  c=IN IP4 127.0.0.1
  t=0 0
  m=video 25549 RTP/AVP 101 
  a=rtpmap:101 VP8/90000
  a=sendonly
  m=audio 26934 RTP/AVP 100 
  a=rtpmap:100 opus/48000/2
  a=sendonly
  ]

// ffmpeg args
commandArgs:[
  '-loglevel',
  'debug',
  '-protocol_whitelist',
  'pipe,udp,rtp',
  '-fflags',
  '+genpts',
  '-f',
  'sdp',
  '-i',
  'pipe:0',
  '-map',
  '0:v:0',
  '-c:v',
  'copy',
  '-map',
  '0:a:0',
  '-strict',
  '-2',
  '-c:a',
  'copy',
  '-f',
  'webm',
  '-flags',
  '+global_header',
  '-y',
  'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm',
  [length]: 26
]
// ffmpeg log
ffmpeg::process::data [data:'ffmpeg version n4.4']
ffmpeg::process::data [data:' Copyright (c) 2000-2021 the FFmpeg developers']
ffmpeg::process::data [data:'\n']
ffmpeg::process::data [data:'  built with gcc 11.1.0 (GCC)\n']
ffmpeg::process::data [data:'  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-shared --enable-version3\n']
ffmpeg::process::data [data:'  libavutil      56. 70.100 / 56. 70.100\n' +
  '  libavcodec     58.134.100 / 58.134.100\n' +
  '  libavformat    58. 76.100 / 58. 76.100\n' +
  '  libavdevice    58. 13.100 / 58. 13.100\n' +
  '  libavfilter     7.110.100 /  7.110.100\n' +
  '  libswscale      5.  9.100 /  5.  9.100\n' +
  '  libswresample   3.  9.100 /  3.  9.100\n' +
  '  libpostproc    55.  9.100 / 55.  9.100\n' +
  'Splitting the commandline.\n' +
  "Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.\n" +
  "Reading option '-protocol_whitelist' ..."]
ffmpeg::process::data [data:" matched as AVOption 'protocol_whitelist' with argument 'pipe,udp,rtp'.\n" +
  "Reading option '-fflags' ..."]
ffmpeg::process::data [data:" matched as AVOption 'fflags' with argument '+genpts'.\n" +
  "Reading option '-f' ... matched as option 'f' (force format) with argument 'sdp'.\n" +
  "Reading option '-i' ... matched as input url with argument 'pipe:0'.\n" +
  "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:v:0'.\n" +
  "Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
  "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:a:0'.\n" +
  "Reading option '-strict' ...Routing option strict to both codec and muxer layer\n" +
  " matched as AVOption 'strict' with argument '-2'.\n" +
  "Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
  "Reading option '-f' ... matched as option 'f' (force format) with argument 'webm'.\n" +
  "Reading option '-flags' ... matched as AVOption 'flags' with argument '+global_header'.\n" +
  "Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.\n" +
  "Reading option 'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm' ... matched as output url.\n" +
  'Finished splitting the commandline.\n' +
  'Parsing a group of options: global .\n' +
  'Applying option loglevel (set logging level) with argument debug.\n' +
  'Applying option y (overwrite output files) with argument 1.\n' +
  'Successfully parsed a group of options.\n' +
  'Parsing a group of options: input url pipe:0.\n' +
  'Applying option f (force format) with argument sdp.\n' +
  'Successfully parsed a group of options.\n' +
  'Opening an input file: pipe:0.\n' +
  "[sdp @ 0x55604dc58400] Opening 'pipe:0' for reading\n" +
  '[sdp @ 0x55604dc58400] video codec set to: vp8\n' +
  '[sdp @ 0x55604dc58400] audio codec set to: opus\n' +
  '[sdp @ 0x55604dc58400] audio samplerate set to: 48000\n' +
  '[sdp @ 0x55604dc58400] audio channels set to: 2\n' +
  '[udp @ 0x55604dc6c500] end receive buffer size reported is 425984\n' +
  '[udp @ 0x55604dc6c7c0] end receive buffer size reported is 425984\n' +
  '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n' +
  '[udp @ 0x55604dc6d900] end receive buffer size reported is 425984\n' +
  '[udp @ 0x55604dc6d2c0] end receive buffer size reported is 425984\n' +
  '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n']
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Before avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 nb_streams:2\n']
  **mediasoup:Consumer resume() +1s**
  **mediasoup:Channel request() [method:consumer.resume, id:12] +1s**
  **mediasoup:Channel request succeeded [method:consumer.resume, id:12] +0ms**
  **mediasoup:Consumer resume() +1ms**
  **mediasoup:Channel request() [method:consumer.resume, id:13] +0ms**
  **mediasoup:Channel request succeeded [method:consumer.resume, id:13] +0ms**
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Could not find codec parameters for stream 0 (Video: vp8, 1 reference frame, yuv420p): unspecified size\n' +
  "Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options\n"]
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] After avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 frames:0\n' +
  "Input #0, sdp, from 'pipe:0':\n" +
  '  Metadata:\n' +
  '    title           : FFmpeg\n' +
  '  Duration: N/A, bitrate: N/A\n' +
  '  Stream #0:0, 0, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc\n' +
  '  Stream #0:1, 0, 1/48000: Audio: opus, 48000 Hz, stereo, fltp\n' +
  'Successfully opened the file.\n' +
  'Parsing a group of options: output url storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
  'Applying option map (set input stream mapping) with argument 0:v:0.\n' +
  'Applying option c:v (codec name) with argument copy.\n' +
  'Applying option map (set input stream mapping) with argument 0:a:0.\n' +
  'Applying option c:a (codec name) with argument copy.\n' +
  'Applying option f (force format) with argument webm.\n' +
  'Successfully parsed a group of options.\n' +
  'Opening an output file: storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
  "[file @ 0x55604dce5bc0] Setting default whitelist 'file,crypto,data'\n"]
ffmpeg::process::data [data:'Successfully opened the file.\n' +
  '[webm @ 0x55604dce0fc0] dimensions not set\n' +
  'Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument\n' +
  'Error initializing output stream 0:1 -- \n' +
  'Stream mapping:\n' +
  '  Stream #0:0 -> #0:0 (copy)\n' +
  '  Stream #0:1 -> #0:1 (copy)\n' +
  '    Last message repeated 1 times\n' +
  '[AVIOContext @ 0x55604dc6dcc0] Statistics: 0 seeks, 0 writeouts\n' +
  '[AVIOContext @ 0x55604dc69380] Statistics: 210 bytes read, 0 seeks\n']
ffmpeg::process::close



  


  FFmpeg says dimensions not  set and Could not write header for output file when I use Firefox. This might be enough for understanding the problem, but if you need more information you can read how server side is performing.
Server-Side in summary can be something like this :
lets say we initialized worker and router at run time using following functions.

  


      // Start the mediasoup workers
module.exports.initializeWorkers = async () => {
  const { logLevel, logTags, rtcMinPort, rtcMaxPort } = config.worker;

  console.log('initializeWorkers() creating %d mediasoup workers', config.numWorkers);

  for (let i = 0; i < config.numWorkers; ++i) {
    const worker = await mediasoup.createWorker({
      logLevel, logTags, rtcMinPort, rtcMaxPort
    });

    worker.once('died', () => {
      console.error('worker::died worker has died exiting in 2 seconds... [pid:%d]', worker.pid);
      setTimeout(() => process.exit(1), 2000);
    });

    workers.push(worker);
  }
};


  


  module.exports.createRouter = async () => {
  const worker = getNextWorker();

  console.log('createRouter() creating new router [worker.pid:%d]', worker.pid);

  console.log(`config.router.mediaCodecs:${JSON.stringify(config.router.mediaCodecs)}`)

  return await worker.createRouter({ mediaCodecs: config.router.mediaCodecs });
};


  


  We pass router.rtpCompatibilities to the client. clients get the rtpCompatibilities and create a device and loads it. after that a transport must be created at server side.

  


      const handleCreateTransportRequest = async (jsonMessage) => {

  const transport = await createTransport('webRtc', router);

  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}
  
  peer.addTransport(transport);

  peer.socket.emit('create-transport',{
    id: transport.id,
    iceParameters: transport.iceParameters,
    iceCandidates: transport.iceCandidates,
    dtlsParameters: transport.dtlsParameters
  });
};


  


  Then after the client side also created the transport we listen to connect event an at the time of event, we request the server to create connection.

  


  const handleTransportConnectRequest = async (jsonMessage) => {
  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}

  if (!peer) {
    throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
  }

  const transport = peer.getTransport(jsonMessage.transportId);

  if (!transport) {
    throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
  }

  await transport.connect({ dtlsParameters: jsonMessage.dtlsParameters });
  console.log('handleTransportConnectRequest() transport connected');
  peer.socket.emit('connect-transport');
};


  


  Similar thing happen on produce event.

  


  const handleProduceRequest = async (jsonMessage) => {
  console.log('handleProduceRequest [data:%o]', jsonMessage);

  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}

  if (!peer) {
    throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
  }

  const transport = peer.getTransport(jsonMessage.transportId);

  if (!transport) {
    throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
  }

  const producer = await transport.produce({
    kind: jsonMessage.kind,
    rtpParameters: jsonMessage.rtpParameters
  });

  peer.addProducer(producer);

  console.log('handleProducerRequest() new producer added [id:%s, kind:%s]', producer.id, producer.kind);

  peer.socket.emit('produce',{
    id: producer.id,
    kind: producer.kind
  });
};


  


  For Recording, first I create plain transports for audio and video producers.

  


  const rtpTransport = router.createPlainTransport(config.plainRtpTransport);


  


  then rtp transport must be connected to ports :

  


    await rtpTransport.connect({
    ip: '127.0.0.1',
    port: remoteRtpPort,
    rtcpPort: remoteRtcpPort
  });


  


  Then the consumer must also be created.

  


    const rtpConsumer = await rtpTransport.consume({
    producerId: producer.id,
    rtpCapabilities,
    paused: true
  });


  


  After that we can start recording using following code :

  


   this._rtpParameters = args;
    this._process = undefined;
    this._observer = new EventEmitter();
    this._peer = args.peer;

    this._sdpString = createSdpText(this._rtpParameters);
    this._sdpStream = convertStringToStream(this._sdpString);
    // create dir
    const dir = process.env.REOCRDING_PATH ?? 'storage/recordings';
    if (!fs.existsSync(dir)) shelljs.mkdir('-p', dir);
  
    this._extension = 'webm';
    // create file path
    this._path = `${dir}/${args.peer.sessionId}.${this._extension}`
    let loop = 0;
    while(fs.existsSync(this._path)) {
      this._path = `${dir}/${args.peer.sessionId}-${++loop}.${this._extension}`
    }

this._recordingnModel = await Recording.findOne({sessionIds: { $in: [this._peer.sessionId] }})
    this._recordingnModel.files.push(this._path);
    this._recordingnModel.save();

let proc  = ffmpeg(this._sdpStream)
    .inputOptions([
      '-protocol_whitelist','pipe,udp,rtp',
      '-f','sdp',
    ])
    .format(this._extension)
    .output(this._path)
    .size('720x?')
    .on('start', ()=>{
      this._peer.socket.emit('recording');
    })
    .on('end', ()=>{
      let path = this._path.replace('storage/recordings/', '');
      this._peer.socket.emit('recording-closed', {
        url: `${process.env.APP_URL}/recording/file/${path}`
      });
    });

    proc.run();
    this._process =  proc;
  }



  


• audio do not stop recording after pause ffmpeg c++

  15 septembre 2021, par C1ngh10

  I am developing an application that record the screen and the audio from microphone. I implemented the pause function stopping video and audio thread on a condition variable, resuming them with a notify on the same condition variable. This is done in captureAudio(), in the main while. In this way works on macOS and linux, where I use avfoudation and alsa respectively, but on windows, with dshow, keep recording audio during the pause, when the thread is waiting on the condition variable. Does anybody know how can I fix this behaviour ?

  


  #include "ScreenRecorder.h"

using namespace std;

ScreenRecorder::ScreenRecorder() : pauseCapture(false), stopCapture(false), started(false), activeMenu(true) {
    avcodec_register_all();
    avdevice_register_all();

    width = 1920;
    height = 1200;
}

ScreenRecorder::~ScreenRecorder() {

    if (started) {
        value = av_write_trailer(outAVFormatContext);
        if (value < 0) {
            cerr << "Error in writing av trailer" << endl;
            exit(-1);
        }

        avformat_close_input(&inAudioFormatContext);
        if(inAudioFormatContext == nullptr){
            cout << "inAudioFormatContext close successfully" << endl;
        }
        else{
            cerr << "Error: unable to close the inAudioFormatContext" << endl;
            exit(-1);
            //throw "Error: unable to close the file";
        }
        avformat_free_context(inAudioFormatContext);
        if(inAudioFormatContext == nullptr){
            cout << "AudioFormat freed successfully" << endl;
        }
        else{
            cerr << "Error: unable to free AudioFormatContext" << endl;
            exit(-1);
        }
        
        avformat_close_input(&pAVFormatContext);
        if (pAVFormatContext == nullptr) {
            cout << "File close successfully" << endl;
        }
        else {
            cerr << "Error: unable to close the file" << endl;
            exit(-1);
            //throw "Error: unable to close the file";
        }

        avformat_free_context(pAVFormatContext);
        if (pAVFormatContext == nullptr) {
            cout << "VideoFormat freed successfully" << endl;
        }
        else {
            cerr << "Error: unable to free VideoFormatContext" << endl;
            exit(-1);
        }
    }
}

/*==================================== VIDEO ==============================*/

int ScreenRecorder::openVideoDevice() throw() {
    value = 0;
    options = nullptr;
    pAVFormatContext = nullptr;

    pAVFormatContext = avformat_alloc_context();

    string dimension = to_string(width) + "x" + to_string(height);
    av_dict_set(&options, "video_size", dimension.c_str(), 0);   //option to set the dimension of the screen section to record

#ifdef _WIN32
    pAVInputFormat = av_find_input_format("gdigrab");
    if (avformat_open_input(&pAVFormatContext, "desktop", pAVInputFormat, &options) != 0) {
        cerr << "Couldn't open input stream" << endl;
        exit(-1);
    }

#elif defined linux
   
    int offset_x = 0, offset_y = 0;
    string url = ":0.0+" + to_string(offset_x) + "," + to_string(offset_y);  //custom string to set the start point of the screen section
    pAVInputFormat = av_find_input_format("x11grab");
    value = avformat_open_input(&pAVFormatContext, url.c_str(), pAVInputFormat, &options);

    if (value != 0) {
        cerr << "Error in opening input device (video)" << endl;
        exit(-1);
    }
#else

    value = av_dict_set(&options, "pixel_format", "0rgb", 0);
    if (value < 0) {
        cerr << "Error in setting pixel format" << endl;
        exit(-1);
    }

    value = av_dict_set(&options, "video_device_index", "1", 0);

    if (value < 0) {
        cerr << "Error in setting video device index" << endl;
        exit(-1);
    }

    pAVInputFormat = av_find_input_format("avfoundation");

    if (avformat_open_input(&pAVFormatContext, "Capture screen 0:none", pAVInputFormat, &options) != 0) {  //TODO trovare un modo per selezionare sempre lo schermo (forse "Capture screen 0")
        cerr << "Error in opening input device" << endl;
        exit(-1);
    }



#endif
    //set frame per second

    value = av_dict_set(&options, "framerate", "30", 0);
    if (value < 0) {
        cerr << "Error in setting dictionary value (setting framerate)" << endl;
        exit(-1);
    }

    value = av_dict_set(&options, "preset", "medium", 0);
    if (value < 0) {
        cerr << "Error in setting dictionary value (setting preset value)" << endl;
        exit(-1);
    }
    /*
    value = av_dict_set(&options, "vsync", "1", 0);
    if(value < 0){
        cerr << "Error in setting dictionary value (setting vsync value)" << endl;
        exit(-1);
    }
    */

    value = av_dict_set(&options, "probesize", "60M", 0);
    if (value < 0) {
        cerr << "Error in setting probesize value" << endl;
        exit(-1);
    }

    //get video stream infos from context
    value = avformat_find_stream_info(pAVFormatContext, nullptr);
    if (value < 0) {
        cerr << "Error in retrieving the stream info" << endl;
        exit(-1);
    }

    VideoStreamIndx = -1;
    for (int i = 0; i < pAVFormatContext->nb_streams; i++) {
        if (pAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
            VideoStreamIndx = i;
            break;
        }
    }
    if (VideoStreamIndx == -1) {
        cerr << "Error: unable to find video stream index" << endl;
        exit(-2);
    }

    pAVCodecContext = pAVFormatContext->streams[VideoStreamIndx]->codec;
    pAVCodec = avcodec_find_decoder(pAVCodecContext->codec_id/*params->codec_id*/);
    if (pAVCodec == nullptr) {
        cerr << "Error: unable to find decoder video" << endl;
        exit(-1);
    }

    cout << "Insert height and width [h w]: ";   //custom screen dimension to record
    cin >> h >> w;*/


    return 0;
}

/*==========================================  AUDIO  ============================*/

int ScreenRecorder::openAudioDevice() {
    audioOptions = nullptr;
    inAudioFormatContext = nullptr;

    inAudioFormatContext = avformat_alloc_context();
    value = av_dict_set(&audioOptions, "sample_rate", "44100", 0);
    if (value < 0) {
        cerr << "Error: cannot set audio sample rate" << endl;
        exit(-1);
    }
    value = av_dict_set(&audioOptions, "async", "1", 0);
    if (value < 0) {
        cerr << "Error: cannot set audio sample rate" << endl;
        exit(-1);
    }

#if defined linux
    audioInputFormat = av_find_input_format("alsa");
    value = avformat_open_input(&inAudioFormatContext, "hw:0", audioInputFormat, &audioOptions);
    if (value != 0) {
        cerr << "Error in opening input device (audio)" << endl;
        exit(-1);
    }
#endif

#if defined _WIN32
    audioInputFormat = av_find_input_format("dshow");
    value = avformat_open_input(&inAudioFormatContext, "audio=Microfono (Realtek(R) Audio)", audioInputFormat, &audioOptions);
    if (value != 0) {
        cerr << "Error in opening input device (audio)" << endl;
        exit(-1);
    }
#endif

    value = avformat_find_stream_info(inAudioFormatContext, nullptr);
    if (value != 0) {
        cerr << "Error: cannot find the audio stream information" << endl;
        exit(-1);
    }

    audioStreamIndx = -1;
    for (int i = 0; i < inAudioFormatContext->nb_streams; i++) {
        if (inAudioFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            audioStreamIndx = i;
            break;
        }
    }
    if (audioStreamIndx == -1) {
        cerr << "Error: unable to find audio stream index" << endl;
        exit(-2);
    }
}

int ScreenRecorder::initOutputFile() {
    value = 0;

    outAVFormatContext = nullptr;
    outputAVFormat = av_guess_format(nullptr, "output.mp4", nullptr);
    if (outputAVFormat == nullptr) {
        cerr << "Error in guessing the video format, try with correct format" << endl;
        exit(-5);
    }
    avformat_alloc_output_context2(&outAVFormatContext, outputAVFormat, outputAVFormat->name, "..\\media\\output.mp4");
    if (outAVFormatContext == nullptr) {
        cerr << "Error in allocating outAVFormatContext" << endl;
        exit(-4);
    }

    /*===========================================================================*/
    this->generateVideoStream();
    this->generateAudioStream();

    //create an empty video file
    if (!(outAVFormatContext->flags & AVFMT_NOFILE)) {
        if (avio_open2(&outAVFormatContext->pb, "..\\media\\output.mp4", AVIO_FLAG_WRITE, nullptr, nullptr) < 0) {
            cerr << "Error in creating the video file" << endl;
            exit(-10);
        }
    }

    if (outAVFormatContext->nb_streams == 0) {
        cerr << "Output file does not contain any stream" << endl;
        exit(-11);
    }
    value = avformat_write_header(outAVFormatContext, &options);
    if (value < 0) {
        cerr << "Error in writing the header context" << endl;
        exit(-12);
    }
    return 0;
}

/*===================================  VIDEO  ==================================*/

void ScreenRecorder::generateVideoStream() {
    //Generate video stream
    videoSt = avformat_new_stream(outAVFormatContext, nullptr);
    if (videoSt == nullptr) {
        cerr << "Error in creating AVFormatStream" << endl;
        exit(-6);
    }

    outVideoCodec = avcodec_find_encoder(AV_CODEC_ID_MPEG4);  //AV_CODEC_ID_MPEG4
    if (outVideoCodec == nullptr) {
        cerr << "Error in finding the AVCodec, try again with the correct codec" << endl;
        exit(-8);
    }
avcodec_alloc_context3(outAVCodec)
    outVideoCodecContext = avcodec_alloc_context3(outVideoCodec);
    if (outVideoCodecContext == nullptr) {
        cerr << "Error in allocating the codec context" << endl;
        exit(-7);
    }

    //set properties of the video file (stream)
    outVideoCodecContext = videoSt->codec;
    outVideoCodecContext->codec_id = AV_CODEC_ID_MPEG4;
    outVideoCodecContext->codec_type = AVMEDIA_TYPE_VIDEO;
    outVideoCodecContext->pix_fmt = AV_PIX_FMT_YUV420P;
    outVideoCodecContext->bit_rate = 10000000;
    outVideoCodecContext->width = width;
    outVideoCodecContext->height = height;
    outVideoCodecContext->gop_size = 10;
    outVideoCodecContext->global_quality = 500;
    outVideoCodecContext->max_b_frames = 2;
    outVideoCodecContext->time_base.num = 1;
    outVideoCodecContext->time_base.den = 30;
    outVideoCodecContext->bit_rate_tolerance = 400000;

    if (outVideoCodecContext->codec_id == AV_CODEC_ID_H264) {
        av_opt_set(outVideoCodecContext->priv_data, "preset", "slow", 0);
    }

    if (outAVFormatContext->oformat->flags & AVFMT_GLOBALHEADER) {
        outVideoCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
    }

    value = avcodec_open2(outVideoCodecContext, outVideoCodec, nullptr);
    if (value < 0) {
        cerr << "Error in opening the AVCodec" << endl;
        exit(-9);
    }

    outVideoStreamIndex = -1;
    for (int i = 0; i < outAVFormatContext->nb_streams; i++) {
        if (outAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_UNKNOWN) {
            outVideoStreamIndex = i;
        }
    }
    if (outVideoStreamIndex < 0) {
        cerr << "Error: cannot find a free stream index for video output" << endl;
        exit(-1);
    }
    avcodec_parameters_from_context(outAVFormatContext->streams[outVideoStreamIndex]->codecpar, outVideoCodecContext);
}

/*===============================  AUDIO  ==================================*/

void ScreenRecorder::generateAudioStream() {
    AVCodecParameters* params = inAudioFormatContext->streams[audioStreamIndx]->codecpar;
    inAudioCodec = avcodec_find_decoder(params->codec_id);
    if (inAudioCodec == nullptr) {
        cerr << "Error: cannot find the audio decoder" << endl;
        exit(-1);
    }

    inAudioCodecContext = avcodec_alloc_context3(inAudioCodec);
    if (avcodec_parameters_to_context(inAudioCodecContext, params) < 0) {
        cout << "Cannot create codec context for audio input" << endl;
    }

    value = avcodec_open2(inAudioCodecContext, inAudioCodec, nullptr);
    if (value < 0) {
        cerr << "Error: cannot open the input audio codec" << endl;
        exit(-1);
    }

    //Generate audio stream
    outAudioCodecContext = nullptr;
    outAudioCodec = nullptr;
    int i;

    AVStream* audio_st = avformat_new_stream(outAVFormatContext, nullptr);
    if (audio_st == nullptr) {
        cerr << "Error: cannot create audio stream" << endl;
        exit(1);
    }

    outAudioCodec = avcodec_find_encoder(AV_CODEC_ID_AAC);
    if (outAudioCodec == nullptr) {
        cerr << "Error: cannot find requested encoder" << endl;
        exit(1);
    }

    outAudioCodecContext = avcodec_alloc_context3(outAudioCodec);
    if (outAudioCodecContext == nullptr) {
        cerr << "Error: cannot create related VideoCodecContext" << endl;
        exit(1);
    }

    if ((outAudioCodec)->supported_samplerates) {
        outAudioCodecContext->sample_rate = (outAudioCodec)->supported_samplerates[0];
        for (i = 0; (outAudioCodec)->supported_samplerates[i]; i++) {
            if ((outAudioCodec)->supported_samplerates[i] == inAudioCodecContext->sample_rate)
                outAudioCodecContext->sample_rate = inAudioCodecContext->sample_rate;
        }
    }
    outAudioCodecContext->codec_id = AV_CODEC_ID_AAC;
    outAudioCodecContext->sample_fmt = (outAudioCodec)->sample_fmts ? (outAudioCodec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
    outAudioCodecContext->channels = inAudioCodecContext->channels;
    outAudioCodecContext->channel_layout = av_get_default_channel_layout(outAudioCodecContext->channels);
    outAudioCodecContext->bit_rate = 96000;
    outAudioCodecContext->time_base = { 1, inAudioCodecContext->sample_rate };

    outAudioCodecContext->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

    if ((outAVFormatContext)->oformat->flags & AVFMT_GLOBALHEADER) {
        outAudioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
    }

    if (avcodec_open2(outAudioCodecContext, outAudioCodec, nullptr) < 0) {
        cerr << "error in opening the avcodec" << endl;
        exit(1);
    }

    //find a free stream index
    outAudioStreamIndex = -1;
    for (i = 0; i < outAVFormatContext->nb_streams; i++) {
        if (outAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_UNKNOWN) {
            outAudioStreamIndex = i;
        }
    }
    if (outAudioStreamIndex < 0) {
        cerr << "Error: cannot find a free stream for audio on the output" << endl;
        exit(1);
    }

    avcodec_parameters_from_context(outAVFormatContext->streams[outAudioStreamIndex]->codecpar, outAudioCodecContext);
}

int ScreenRecorder::init_fifo()
{
    /* Create the FIFO buffer based on the specified output sample format. */
    if (!(fifo = av_audio_fifo_alloc(outAudioCodecContext->sample_fmt,
        outAudioCodecContext->channels, 1))) {
        fprintf(stderr, "Could not allocate FIFO\n");
        return AVERROR(ENOMEM);
    }
    return 0;
}

int ScreenRecorder::add_samples_to_fifo(uint8_t** converted_input_samples, const int frame_size) {
    int error;
    /* Make the FIFO as large as it needs to be to hold both,
     * the old and the new samples. */
    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
        fprintf(stderr, "Could not reallocate FIFO\n");
        return error;
    }
    /* Store the new samples in the FIFO buffer. */
    if (av_audio_fifo_write(fifo, (void**)converted_input_samples, frame_size) < frame_size) {
        fprintf(stderr, "Could not write data to FIFO\n");
        return AVERROR_EXIT;
    }
    return 0;
}

int ScreenRecorder::initConvertedSamples(uint8_t*** converted_input_samples,
    AVCodecContext* output_codec_context,
    int frame_size) {
    int error;
    /* Allocate as many pointers as there are audio channels.
     * Each pointer will later point to the audio samples of the corresponding
     * channels (although it may be NULL for interleaved formats).
     */
    if (!(*converted_input_samples = (uint8_t**)calloc(output_codec_context->channels,
        sizeof(**converted_input_samples)))) {
        fprintf(stderr, "Could not allocate converted input sample pointers\n");
        return AVERROR(ENOMEM);
    }
    /* Allocate memory for the samples of all channels in one consecutive
     * block for convenience. */
    if (av_samples_alloc(*converted_input_samples, nullptr,
        output_codec_context->channels,
        frame_size,
        output_codec_context->sample_fmt, 0) < 0) {

        exit(1);
    }
    return 0;
}

static int64_t pts = 0;
void ScreenRecorder::captureAudio() {
    int ret;
    AVPacket* inPacket, * outPacket;
    AVFrame* rawFrame, * scaledFrame;
    uint8_t** resampledData;

    init_fifo();

    //allocate space for a packet
    inPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
    if (!inPacket) {
        cerr << "Cannot allocate an AVPacket for encoded video" << endl;
        exit(1);
    }
    av_init_packet(inPacket);

    //allocate space for a packet
    rawFrame = av_frame_alloc();
    if (!rawFrame) {
        cerr << "Cannot allocate an AVPacket for encoded video" << endl;
        exit(1);
    }

    scaledFrame = av_frame_alloc();
    if (!scaledFrame) {
        cerr << "Cannot allocate an AVPacket for encoded video" << endl;
        exit(1);
    }

    outPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
    if (!outPacket) {
        cerr << "Cannot allocate an AVPacket for encoded video" << endl;
        exit(1);
    }

    //init the resampler
    SwrContext* resampleContext = nullptr;
    resampleContext = swr_alloc_set_opts(resampleContext,
        av_get_default_channel_layout(outAudioCodecContext->channels),
        outAudioCodecContext->sample_fmt,
        outAudioCodecContext->sample_rate,
        av_get_default_channel_layout(inAudioCodecContext->channels),
        inAudioCodecContext->sample_fmt,
        inAudioCodecContext->sample_rate,
        0,
        nullptr);
    if (!resampleContext) {
        cerr << "Cannot allocate the resample context" << endl;
        exit(1);
    }
    if ((swr_init(resampleContext)) < 0) {
        fprintf(stderr, "Could not open resample context\n");
        swr_free(&resampleContext);
        exit(1);
    }

    while (true) {
        if (pauseCapture) {
            cout << "Pause audio" << endl;
        }
        cv.wait(ul, [this]() { return !pauseCapture; });

        if (stopCapture) {
            break;
        }

        ul.unlock();

        if (av_read_frame(inAudioFormatContext, inPacket) >= 0 && inPacket->stream_index == audioStreamIndx) {
            //decode audio routing
            av_packet_rescale_ts(outPacket, inAudioFormatContext->streams[audioStreamIndx]->time_base, inAudioCodecContext->time_base);
            if ((ret = avcodec_send_packet(inAudioCodecContext, inPacket)) < 0) {
                cout << "Cannot decode current audio packet " << ret << endl;
                continue;
            }
            
            while (ret >= 0) {
                ret = avcodec_receive_frame(inAudioCodecContext, rawFrame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                    break;
                else if (ret < 0) {
                    cerr << "Error during decoding" << endl;
                    exit(1);
                }
                if (outAVFormatContext->streams[outAudioStreamIndex]->start_time <= 0) {
                    outAVFormatContext->streams[outAudioStreamIndex]->start_time = rawFrame->pts;
                }
                initConvertedSamples(&resampledData, outAudioCodecContext, rawFrame->nb_samples);

                swr_convert(resampleContext,
                    resampledData, rawFrame->nb_samples,
                    (const uint8_t**)rawFrame->extended_data, rawFrame->nb_samp

                add_samples_to_fifo(resampledData, rawFrame->nb_samples);

                //raw frame ready
                av_init_packet(outPacket);
                outPacket->data = nullptr;
                outPacket->size = 0;

                const int frame_size = FFMAX(av_audio_fifo_size(fifo), outAudioCodecContext->frame_size);

                scaledFrame = av_frame_alloc();
                if (!scaledFrame) {
                    cerr << "Cannot allocate an AVPacket for encoded video" << endl;
                    exit(1);
                }

                scaledFrame->nb_samples = outAudioCodecContext->frame_size;
                scaledFrame->channel_layout = outAudioCodecContext->channel_layout;
                scaledFrame->format = outAudioCodecContext->sample_fmt;
                scaledFrame->sample_rate = outAudioCodecContext->sample_rate;
                av_frame_get_buffer(scaledFrame, 0);

                while (av_audio_fifo_size(fifo) >= outAudioCodecContext->frame_size) {

                    ret = av_audio_fifo_read(fifo, (void**)(scaledFrame->data), outAudioCodecContext->frame_size);
                    scaledFrame->pts = pts;
                    pts += scaledFrame->nb_samples;
                    if (avcodec_send_frame(outAudioCodecContext, scaledFrame) < 0) {
                        cout << "Cannot encode current audio packet " << endl;
                        exit(1);
                    }
                    while (ret >= 0) {
                        ret = avcodec_receive_packet(outAudioCodecContext, outPacket);
                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                            break;
                        else if (ret < 0) {
                            cerr << "Error during encoding" << endl;
                            exit(1);
                        }
                        av_packet_rescale_ts(outPacket, outAudioCodecContext->time_base, outAVFormatContext->streams[outAudioStreamIndex]->time_base);

                        outPacket->stream_index = outAudioStreamIndex;

                        write_lock.lock();
                        
                        if (av_write_frame(outAVFormatContext, outPacket) != 0)
                        {
                            cerr << "Error in writing audio frame" << endl;
                        }
                        write_lock.unlock();
                        av_packet_unref(outPacket);
                    }
                    ret = 0;
                }
                av_frame_free(&scaledFrame);
                av_packet_unref(outPacket);
            }
        }
    }
}

int ScreenRecorder::captureVideoFrames() {
    int64_t pts = 0;
    int flag;
    int frameFinished = 0;
    bool endPause = false;
    int numPause = 0;

    ofstream outFile{ "..\\media\\log.txt", ios::out };

    int frameIndex = 0;
    value = 0;

    pAVPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
    if (pAVPacket == nullptr) {
        cerr << "Error in allocating AVPacket" << endl;
        exit(-1);
    }

    pAVFrame = av_frame_alloc();
    if (pAVFrame == nullptr) {
        cerr << "Error: unable to alloc the AVFrame resources" << endl;
        exit(-1);
    }

    outFrame = av_frame_alloc();
    if (outFrame == nullptr) {
        cerr << "Error: unable to alloc the AVFrame resources for out frame" << endl;
        exit(-1);
    }

    int videoOutBuffSize;
    int nBytes = av_image_get_buffer_size(outVideoCodecContext->pix_fmt, outVideoCodecContext->width, outVideoCodecContext->height, 32);
    uint8_t* videoOutBuff = (uint8_t*)av_malloc(nBytes);

    if (videoOutBuff == nullptr) {
        cerr << "Error: unable to allocate memory" << endl;
        exit(-1);
    }

    value = av_image_fill_arrays(outFrame->data, outFrame->linesize, videoOutBuff, AV_PIX_FMT_YUV420P, outVideoCodecContext->width, outVideoCodecContext->height, 1);
    if (value < 0) {
        cerr << "Error in filling image array" << endl;
    }

    SwsContext* swsCtx_;
    if (avcodec_open2(pAVCodecContext, pAVCodec, nullptr) < 0) {
        cerr << "Could not open codec" << endl;
        exit(-1);
    }
    swsCtx_ = sws_getContext(pAVCodecContext->width, pAVCodecContext->height, pAVCodecContext->pix_fmt, outVideoCodecContext->width, outVideoCodecContext->height, outVideoCodecContext->pix_fmt, SWS_BICUBIC,
        nullptr, nullptr, nullptr);

    AVPacket outPacket;
    int gotPicture;

    time_t startTime;
    time(&startTime);

    while (true) {

        if (pauseCapture) {
            cout << "Pause" << endl;
            outFile << "///////////////////   Pause  ///////////////////" << endl;
            cout << "outVideoCodecContext->time_base: " << outVideoCodecContext->time_base.num << ", " << outVideoCodecContext->time_base.den << endl;
        }
        cv.wait(ul, [this]() { return !pauseCapture; });   //pause capture (not busy waiting)
        if (endPause) {
            endPause = false;
        }

        if (stopCapture)  //check if the capture has to stop
            break;
        ul.unlock();

        if (av_read_frame(pAVFormatContext, pAVPacket) >= 0 && pAVPacket->stream_index == VideoStreamIndx) {
            av_packet_rescale_ts(pAVPacket, pAVFormatContext->streams[VideoStreamIndx]->time_base, pAVCodecContext->time_base);
            value = avcodec_decode_video2(pAVCodecContext, pAVFrame, &frameFinished, pAVPacket);
            if (value < 0) {
                cout << "Unable to decode video" << endl;
            }

            if (frameFinished) { //frame successfully decoded
                //sws_scale(swsCtx_, pAVFrame->data, pAVFrame->linesize, 0, pAVCodecContext->height, outFrame->data, outFrame->linesize);
                av_init_packet(&outPacket);
                outPacket.data = nullptr;
                outPacket.size = 0;

                if (outAVFormatContext->streams[outVideoStreamIndex]->start_time <= 0) {
                    outAVFormatContext->streams[outVideoStreamIndex]->start_time = pAVFrame->pts;
                }

                //disable warning on the console
                outFrame->width = outVideoCodecContext->width;
                outFrame->height = outVideoCodecContext->height;
                outFrame->format = outVideoCodecContext->pix_fmt;

                sws_scale(swsCtx_, pAVFrame->data, pAVFrame->linesize, 0, pAVCodecContext->height, outFrame->data, outFrame->linesize);

                avcodec_encode_video2(outVideoCodecContext, &outPacket, outFrame, &gotPicture);

                if (gotPicture) {
                    if (outPacket.pts != AV_NOPTS_VALUE) {
                        outPacket.pts = av_rescale_q(outPacket.pts, videoSt->codec->time_base, videoSt->time_base);
                    }
                    if (outPacket.dts != AV_NOPTS_VALUE) {
                        outPacket.dts = av_rescale_q(outPacket.dts, videoSt->codec->time_base, videoSt->time_base);
                    }

                    //cout << "Write frame " << j++ << " (size = " << outPacket.size / 1000 << ")" << endl;
                    //cout << "(size = " << outPacket.size << ")" << endl;

                    //av_packet_rescale_ts(&outPacket, outVideoCodecContext->time_base, outAVFormatContext->streams[outVideoStreamIndex]->time_base);
                    //outPacket.stream_index = outVideoStreamIndex;

                    outFile << "outPacket->duration: " << outPacket.duration << ", " << "pAVPacket->duration: " << pAVPacket->duration << endl;
                    outFile << "outPacket->pts: " << outPacket.pts << ", " << "pAVPacket->pts: " << pAVPacket->pts << endl;
                    outFile << "outPacket.dts: " << outPacket.dts << ", " << "pAVPacket->dts: " << pAVPacket->dts << endl;

                    time_t timer;
                    double seconds;

                    mu.lock();
                    if (!activeMenu) {
                        time(&timer);
                        seconds = difftime(timer, startTime);
                        int h = (int)(seconds / 3600);
                        int m = (int)(seconds / 60) % 60;
                        int s = (int)(seconds) % 60;

                        std::cout << std::flush << "\r" << std::setw(2) << std::setfill('0') << h << ':'
                            << std::setw(2) << std::setfill('0') << m << ':'
                            << std::setw(2) << std::setfill('0') << s << std::flush;
                    }
                    mu.unlock();

                    write_lock.lock();
                    if (av_write_frame(outAVFormatContext, &outPacket) != 0) {
                        cerr << "Error in writing video frame" << endl;
                    }
                    write_lock.unlock();
                    av_packet_unref(&outPacket);
                }

                av_packet_unref(&outPacket);
                av_free_packet(pAVPacket);  //avoid memory saturation
            }
        }
    }

    outFile.close();

    av_free(videoOutBuff);

    return 0;
}