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• Discord bot stop playing music in random time of song

  25 janvier 2021, par Jusmejtr

  I have a discord to let me play a random song from the list.

  


  How bot works :
Bot IS connected to firestore Cloud (firebase) where i have economy data from my server. Price for playing random song is 75 coins.

  


  Everything worked as it should, but yesterday I used command, the bot started playing and after a while it stopped playing music and also no other commands worked, bot probably get freezed.

  


  I have no errors in the console until after a minute it showed me this error.

  


  https://pastebin.com/ay9gV75T

  


  The bot is hosted on Heroku and I also added this buildpack to ffmpeg in the settings.

  


  https://github.com/jonathanong/heroku-buildpack-ffmpeg-latest

  


  This is my code :

  


  module.exports = {
    name: "buy-music",
    description: "buy a music",

    async execute(message, config, db){
        const PREFIX = (config.prefix);

        if(message.content === PREFIX + "buy music"){
            const ytdl = require("ytdl-core");
            message.delete();
            let uzivatel = message.author.tag;

            let voiceChannel = message.member.voice.channel;
            if(!voiceChannel) return message.reply("Musíš byť vo voice roomke");

            let cena = 75;
    
            db.collection('economy').doc(uzivatel).get().then(async (q) => {
                if(!q.exists) return message.reply("Nemáš vytvorený účet");
                var hodnota = q.data().money;
                if(hodnota < cena) return message.reply("Nemáš dostatok financií");

                db.collection('statusy').doc('music').get().then(async (asaj) => {
                    let stav = asaj.data().stav;
                    if(stav == "off"){
                        db.collection('statusy').doc('music').update({
                            "stav": "on",
                            "autor": message.author.tag,
                        });
                        hodnota -= cena;
                        db.collection('economy').doc(uzivatel).update({
                            'money': hodnota
                        });
                        function randomhraj(){
                            var pole = [
                     My YT links

                            ]
                            let rnd = Math.floor(Math.random() * pole.length);
                            let output = pole[rnd];
                            return output;
                        }
        
                        try{
                            var pripojenie = await voiceChannel.join();
                            message.reply(`Úspešne si si kúpil chuťovečku`);
                        }catch(error){
                            console.log(`Error pri pripajani do room (music join) ${error}`);
                        }
                        
                        const dispatcher = pripojenie.play(ytdl(randomhraj())).on("finish", async() => {
                            await voiceChannel.leave();
                            await db.collection('statusy').doc('music').update({
                                "stav": "off",
                                "autor": "nikto",
                            });
                        }).on("error", error => {
                            console.log(error)
                        })
                        dispatcher.setVolumeLogarithmic(5 / 5)
                    }else{
                        message.reply("Momentálne si hudbu kúpil niekto iný alebo ak si hudbu kúpil a chceš ju zastaviť použi príkaz *stop");
                    }
                    
                });
            });
    
        }else if(message.content === PREFIX + "stop"){
            message.delete();
            db.collection('statusy').doc('music').get().then((n) => {
                let kto = n.data().autor;
                let meno = message.author.tag;
                if(!message.member.voice.channel) return message.channel.send("Musíš byť vo voice roomke pre stopnutie hudby");
                if(kto == meno){
                    message.member.voice.channel.leave();
                    message.channel.send("Úspešne odpojený");
                    db.collection('statusy').doc('music').update({
                        "stav": "off",
                        "autor": "nikto",
                    });
                }else{
                    message.reply("Zastaviť hudbu môže len ten kto si ju kúpil");
                }
            });
        }
        
    }
}


  


• ffmpeg rtsp stream to YouTube livestream not doing anything

  29 mai 2018, par felixosth

  I’m using C# to initiate ffmpeg. I’ve a onvif bridge server for a CCTV VMS and I’m building a application to enable the user to livestream any CCTV camera to YouTube.

  The rtsp stream to the camera looks like this :

  rtsp ://onvif:bridge@localhost:554/live/xxxxx-xxxguidtocameraxxx-xxxxx

  I’m new to ffmpeg and I’m using this arg :

  -f lavfi -i anullsrc -rtsp_transport udp -i camerastreamurl -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp ://x.rtmp.youtube.com/live2/streamkey -loglevel debug

  It seems like ffmpeg isn’t finding the camera stream, only the fake audio one. It just freezes.

  This is the result of the debug log :

  ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 18.102 / 56. 18.102
  
    libavcodec     58. 19.104 / 58. 19.104
  
    libavformat    58. 17.100 / 58. 17.100
  
    libavdevice    58.  4.100 / 58.  4.100
  
    libavfilter     7. 24.100 /  7. 24.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  Splitting the commandline.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'.
  
  Reading option '-i' ... matched as input url with argument 'anullsrc'.
  
  Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'udp'.
  
  Reading option '-i' ... matched as input url with argument 'rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b'.
  
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  
  Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
  
  Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument '+'.
  
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
  
  Reading option '-strict' ...Routing option strict to both codec and muxer layer
  
   matched as AVOption 'strict' with argument 'experimental'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'rtmp://x.rtmp.youtube.com/live2/xxxxxxxx' ... matched as output url.
  
  Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option loglevel (set logging level) with argument debug.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input url anullsrc.
  
  Applying option f (force format) with argument lavfi.
  
  Successfully parsed a group of options.
  
  Opening an input file: anullsrc.
  
  detected 8 logical cores
  
  [AVFilterGraph @ 0000027a34bad7c0] query_formats: 2 queried, 3 merged, 0 already done, 0 delayed
  
  [Parsed_anullsrc_0 @ 0000027a34badb80] sample_rate:44100 channel_layout:'stereo' nb_samples:1024
  
  [lavfi @ 0000027a34babc80] All info found
  
  Input #0, lavfi, from 'anullsrc':
  
    Duration: N/A, start: 0.000000, bitrate: 705 kb/s
  
      Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: input url rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  Successfully parsed a group of options.
  
  Opening an input file: rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  [tcp @ 0000027a34bb5980] No default whitelist set

  Edit :
  I’m getting this result with minimal args :
  

• FFMPEG av_read_frame() return an I/O error (-5) ?

  7 mars 2019, par vominhtien961476

  I’m using ffmpeg library to capture screen (desktop). Everything works fine when I run my app in debug/release mode. The I/O error comes when this app run in c# window service mode (win 10 OS). I googled but still don’t know why the error come. In the service mode there is no problem when connecting to some rtsp link (Camera) except desktop capture.

  AVInputFormat * pAVInputFormat = av_find_input_format("gdigrab"); 
  
  if (pAVInputFormat != NULL) 
  
  { 
  
     int err = avformat_open_input(&pFormatCtx, "desktop", pAVInputFormat, NULL); 
  
     if(err== 0) 
  
     { 
  
        err = av_read_frame(&pFormatCtx, &packet); 
  
        if (err < 0) 
  
        { 
  
            char errs[AV_ERROR_MAX_STRING_SIZE]; 
  
            av_make_error_string(errs, AV_ERROR_MAX_STRING_SIZE, err ); 
  
            String^ errStr = gcnew String(errs); 
  
            WriteLog("Error in function av_read_frame() return = " + errStr); 
  
            return -1; 
  
       } 
  
     } 
  
  }

  ---