Recherche avancée

Sur d’autres sites (7709)

• iPhone HTTP Streaming .m3u8 and .ts files - how to create using ffmpeg

  8 septembre 2013, par ChessDev

  I'm trying to get apple-validated http media streams using ffmpeg and am getting errors. Here are some error examples :

  WARNING : Playlist Content-Type is 'application/x-mpegurl', but should
  be one of 'application/vnd.apple.mpegurl', 'audio/x-mpegurl' or
  'audio/mpegurl'.

  WARNING : 258 samples (88.966 %) do not have timestamps in track 256
  (avc1). 4 : us2-1.ts
   

  WARNING : Media segment duration outside of expected duration by 47.733
   % (5.23 vs. 10.00 seconds, limit is 20 %). 40 : us2-19.ts
   

  Average segment duration: 10.16 seconds
  
  Average segment bitrate: 320.12 kbit/s
  
  Average segment structural overhead: 175.89 kbit/s (54.94 %)
  
  
  
  Video codec: avc1
  
  Video resolution: 320x320 pixels
  
  Video frame rate: 29.72, 29.78, 29.82, 30.00, 29.64 fps
  
  Average video bitrate: 100.66 kbit/s
  
  H.264 profile: Baseline
  
  H.264 level: 3.0
  
  
  
  Audio codec: aac 
  
  Audio sample rate: 48000 Hz
  
  Average audio bitrate: 43.57 kbit/s

  ---

  Here is the end file I've been submitting : http://files.chesscomfiles.com/images_users/using/us2.m3u8

  Here is the file I used to create this : http://files.chesscomfiles.com/images_users/using/using-computers-1.mp4

  I've tried these commands, among others :

  ffmpeg -i using-computers-1.mp4 -f mpegts -acodec libfaac -ar 48000
  -ab 64k -s 320x320 -vcodec libx264 -vbsf h264_mp4toannexb -b 96k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 2 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -g 30 -async 2 us2.ts

  ffmpeg -i using-computers-1.mp4 -f mpegts -acodec libfaac -ar 48000
  -ab 64k -s 320x320 -vcodec libx264 -vbsf h264_mp4toannexb -b 96k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -g 30 -async 2 us1.ts

  ffmpeg -i using-computers-1.mp4 -vbsf h264_mp4toannexb -acodec copy -vcodec copy -f mpegts output.ts

  If someone can help me figure out what ffmpeg commands I should be running I'd really appreciate it !

• 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()
    }

}

  


  


  



• Cannot suppress ffmpeg output from ruby

  3 avril 2014, par Danny

  I have ruby on rails app that allows users to upload videos. When a video is added, I have a before_save filter that uses ffmpeg to generate a series of thumbnails. The problem is that ffmpeg is producing tons of console output when I'm saving a video item in the rails console, and when I run my tests.

  My environment :

  • Host Machine : OS X 10.9.2
  • Vagrant Box : Ubuntu 10.04.4
  • ffmpeg version : SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.3
  • ruby version : 1.9.3-p194

  Command I'm running :

  `ffmpeg -v 0 -ss #{timestamp} -i #{video_file.path} -y -f image2 -vcodec mjpeg -vframes 1 -s 640*360 #{thumbnail_path}/thumbnail#{i}.jpg`

  This version of ffmpeg on my VM doesn't seem to care about the "-v 0" option. I've also tried "-loglevel quiet" which causes ffmpeg to error, indicating that the option isn't recognized (both loglevel and v work on my host machine's ffmpeg).

  Tried using both exec() and system(), which both caused execution to hang. Tried to redirecting output to a file by doing :

  `ffmpeg -v 0 -ss #{timestamp} -i #{video_file.path} -y -f image2 -vcodec mjpeg -vframes 1 -s 640*360 #{thumbnail_path}/thumbnail#{i}.jpg > #{thumbnail_path}/output.txt`

  Still see output. Next I tried :

  `ffmpeg -v 0 -ss #{timestamp} -i #{video_file.path} -y -f image2 -vcodec mjpeg -vframes 1 -s 640*360 #{thumbnail_path}/thumbnail#{i}.jpg &> dev/null`

  Still seeing output ! Finally I tried :

  $stdout.reopen("#{thumbnail_path}/output.txt", "w")
  
  $stderr.reopen("#{thumbnail_path}/error.txt", "w")
  
  `ffmpeg -v 0 -ss #{timestamp} -i #{video_file.path} -y -f image2 -vcodec mjpeg -vframes 1 -s 640*360 #{thumbnail_path}/thumbnail#{i}.jpg`
  
  $stdout = STDOUT
  
  $stderr = STDERR

  Holy cow, that worked ! Well, sort of. No more verbose output when running tests, BUT somehow anytime this runs I get kicked out of the rails console.

  Does anyone have a more elegant solution ?