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• What's the most desireable way to capture system display and audio in the form of individual encoded audio and video packets in go (language)? [closed]

  11 January 2023, by Tiger Yang

  Question (read the context below first):

  


  For those of you familiar with the capabilities of go, Is there a better way to go about all this? Since ffmpeg is so ubiquitous, I'm sure it's been optomized to perfection, but what's the best way to capture system display and audio in the form of individual encoded audio and video packets in go (language), so that they can be then sent via webtransport-go? I wish for it to prioritize efficiency and low latency, and ideally capture and encode the framebuffer directly like ffmpeg does.

  


  Thanks! I have many other questions about this, but I think it's best to ask as I go.

  


  Context and what I've done so far:

  


  I'm writing a remote desktop software for my personal use because of grievances with current solutions out there. At the moment, it consists of a web app that uses the webtransport API to send input datagrams and receive AV packets on two dedicated unidirectional streams, and the webcodecs API to decode these packets. On the serverside, I originally planned to use python with the aioquic library as a webtransport server. Upon connection and authentication, the server would start ffmpeg as a subprocess with this command:

  


  ffmpeg -init_hw_device d3d11va -filter_complex ddagrab=video_size=1920x1080:framerate=60 -vcodec hevc_nvenc -tune ll -preset p7 -spatial_aq 1 -temporal_aq 1 -forced-idr 1 -rc cbr -b:v 400K -no-scenecut 1 -g 216000 -f hevc -

  


  What I really appreciate about this is that it uses windows' desktop duplication API to copy the framebuffer of my GPU and hand that directly to the on-die hardware encoder with zero round trips to the CPU. I think it's about as efficient and elegant a solution as I can manage. It then outputs the encoded stream to the stdout, which python can read and send to the client.

  


  As for the audio, there is another ffmpeg instance:

  


  ffmpeg -f dshow -channels 2 -sample_rate 48000 -sample_size 16 -audio_buffer_size 15 -i audio="RD Audio (High Definition Audio Device)" -acodec libopus -vbr on -application audio -mapping_family 0 -apply_phase_inv true -b:a 25K -fec false -packet_loss 0 -map 0 -f data -

  


  which listens to a physical loopback interface, which is literally just a short wire bridging the front panel headphone and microphone jacks (I'm aware of the quality loss of converting to analog and back, but the audio is then crushed down to 25kbps so it's fine) ()

  


  Unfortunately, aioquic was not easy to work with IMO, and I found webtransport-go https://github.com/adriancable/webtransport-go, which was a hell of a lot better in both simplicity and documentation. However, now I'm dealing with a whole new language, and I wanna ask: (above)

  


  EDIT: Here's the code for my server so far:

  


  


  


  package main

import (
    "bytes"
    "context"
    "fmt"
    "log"
    "net/http"
    "os/exec"
    "time"

    "github.com/adriancable/webtransport-go"
)

func warn(str string) {
    fmt.Printf("\n===== WARNING ===================================================================================================\n   %s\n=================================================================================================================\n", str)
}

func main() {

    password := []byte("abc")

    videoString := []string{
        "ffmpeg",
        "-init_hw_device", "d3d11va",
        "-filter_complex", "ddagrab=video_size=1920x1080:framerate=60",
        "-vcodec", "hevc_nvenc",
        "-tune", "ll",
        "-preset", "p7",
        "-spatial_aq", "1",
        "-temporal_aq", "1",
        "-forced-idr", "1",
        "-rc", "cbr",
        "-b:v", "500K",
        "-no-scenecut", "1",
        "-g", "216000",
        "-f", "hevc", "-",
    }

    audioString := []string{
        "ffmpeg",
        "-f", "dshow",
        "-channels", "2",
        "-sample_rate", "48000",
        "-sample_size", "16",
        "-audio_buffer_size", "15",
        "-i", "audio=RD Audio (High Definition Audio Device)",
        "-acodec", "libopus",
        "-mapping_family", "0",
        "-b:a", "25K",
        "-map", "0",
        "-f", "data", "-",
    }

    connected := false

    http.HandleFunc("/", func(writer http.ResponseWriter, request *http.Request) {
        session := request.Body.(*webtransport.Session)

        session.AcceptSession()
        fmt.Println("\nAccepted incoming WebTransport connection.")
        fmt.Println("Awaiting authentication...")

        authData, err := session.ReceiveMessage(session.Context()) // Waits here till first datagram
        if err != nil {                                            // if client closes connection before sending anything
            fmt.Println("\nConnection closed:", err)
            return
        }

        if len(authData) >= 2 && bytes.Equal(authData[2:], password) {
            if connected {
                session.CloseSession()
                warn("Client has authenticated, but a session is already taking place! Connection closed.")
                return
            } else {
                connected = true
                fmt.Println("Client has authenticated!\n")
            }
        } else {
            session.CloseSession()
            warn("Client has failed authentication! Connection closed. (" + string(authData[2:]) + ")")
            return
        }

        videoStream, _ := session.OpenUniStreamSync(session.Context())

        videoCmd := exec.Command(videoString[0], videoString[1:]...)
        go func() {
            videoOut, _ := videoCmd.StdoutPipe()
            videoCmd.Start()

            buffer := make([]byte, 15000)
            for {
                len, err := videoOut.Read(buffer)
                if err != nil {
                    break
                }
                if len > 0 {
                    videoStream.Write(buffer[:len])
                }
            }
        }()

        time.Sleep(50 * time.Millisecond)

        audioStream, err := session.OpenUniStreamSync(session.Context())

        audioCmd := exec.Command(audioString[0], audioString[1:]...)
        go func() {
            audioOut, _ := audioCmd.StdoutPipe()
            audioCmd.Start()

            buffer := make([]byte, 15000)
            for {
                len, err := audioOut.Read(buffer)
                if err != nil {
                    break
                }
                if len > 0 {
                    audioStream.Write(buffer[:len])
                }
            }
        }()

        for {
            data, err := session.ReceiveMessage(session.Context())
            if err != nil {
                videoCmd.Process.Kill()
                audioCmd.Process.Kill()

                connected = false

                fmt.Println("\nConnection closed:", err)
                break
            }

            if len(data) == 0 {

            } else if data[0] == byte(0) {
                fmt.Printf("Received mouse datagram: %s\n", data)
            }
        }

    })

    server := &webtransport.Server{
        ListenAddr: ":1024",
        TLSCert:    webtransport.CertFile{Path: "SSL/fullchain.pem"},
        TLSKey:     webtransport.CertFile{Path: "SSL/privkey.pem"},
        QuicConfig: &webtransport.QuicConfig{
            KeepAlive:      false,
            MaxIdleTimeout: 3 * time.Second,
        },
    }

    fmt.Println("Launching WebTransport server at", server.ListenAddr)
    ctx, cancel := context.WithCancel(context.Background())
    if err := server.Run(ctx); err != nil {
        log.Fatal(err)
        cancel()
    }

}

  


  


  



• Cannot play audio from a link using a Discord Bot

  2 March 2020, by Guilhermeffable

  I’m trying to code a bot so me and my friends can hear the local radio on our Discord Server but I’m having this error.

  This is part of my code, it’s the play.js file that handles the playback stuff.

  module.exports = (client,message) => {
  
  
  
  const voiceChannel = message.member.voiceChannel;
  
  const idChannel = voiceChannel.id;
  
  
  
  console.log(idChannel)
  
      //vê se o user está numa sala de voz
  
  if(!voiceChannel) {
  
      return message.channel.send("Precisas de estar num voice channel para usar este comando.")
  
  }
  
  const permissions = voiceChannel.permissionsFor(message.client.user);
  
  
  
   //vê se tem permissões para entrar na sala
  
  if(!permissions.has('CONNECT') || !permissions.has('SPEAK')) {
  
      return message.channel.send("Não tenho permissões para entrar nessa sala.")
  
  }
  
  
  
  voiceChannel.join()
  
      .then(connection => {
  
          console.log("Successfully connected.");
  
          connection.playStream('http://centova.radios.pt:8401/stream.mp3/1')
  
  }).catch(e =>{
  
      console.error(e);
  
  
  
  });

  }

  And this is the error I’m getting:

  TypeError [ERR_INVALID_ARG_TYPE]: The "file" argument must be of type string. Received an instance of         
  
  Object
  
  at validateString (internal/validators.js:117:11)
  
  at normalizeSpawnArguments (child_process.js:406:3)
  
  at Object.spawn (child_process.js:542:16)
  
  at new FfmpegProcess (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\prism-media\src\transcoders\ffmpeg\FfmpegProcess.js:14:33)
  
  at FfmpegTranscoder.transcode (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\prism-media\src\transcoders\ffmpeg\Ffmpeg.js:34:18)
  
  at MediaTranscoder.transcode (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\prism-media\src\transcoders\MediaTranscoder.js:27:31)
  
  at Prism.transcode (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\prism-media\src\Prism.js:13:28)
  
  at AudioPlayer.playUnknownStream (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\discord.js\src\client\voice\player\AudioPlayer.js:97:35)
  
  at VoiceConnection.playStream (C:\Users\guilh\desktop\BOT\orbitalbot\node_modules\discord.js\src\client\voice\VoiceConnection.js:546:24)
  
  at C:\Users\guilh\desktop\BOT\orbitalbot\commands\play.js:24:24 {
  
    code: 'ERR_INVALID_ARG_TYPE'

• Revision 70728: Un chouilla de phpdoc et éviter une notice si ...

  14 March 2013, by kent1@… — Log

  Un chouilla de phpdoc et éviter une notice si $argscontexte?modele? n’est pas défini