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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 janvier 2023, par 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()
    }

}

  


  


  



• how to use ffmplay in android

  8 mai 2016, par Thanh Dat Mai

  I want to show a video with filter. Use this command on pc

  ffplay input.mp4 -vf scale=320:240,curves=psfile=cur.acv

  I was able to show videos. But I do not do this on android, Who can tell me any direction yet or library support for my work. thank you

• Ffmpeg Windows split 1 video to chunks with random length (no re-encode)

  24 janvier 2021, par Anna Lone

  I have 1 big video file and I need to split it to random length chunks without re-encode.

  


  For example chunks from 130 to 240 seconds.

  


  For windows ffmpeg

  


  Tried this and. Nothing in output folder.

  


  $ times=$(ruby -e 's=[]; d=0; while d < 150 do t=rand(15..50); s << (d+t); d=d+t end; puts s.join(",")')
$ echo $times
15,53,96,124,168
$ ffmpeg -i lutherceleb.mp4 -f segment -segment_times $times -c copy -reset_timestamps 1 -map 0 OUTPUT%d.mp4