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• MKV video playback shows a black picture and audio plays good

  24 mars 2023, par gbriones.gdl

  I have an MKV video file that has something wrong, every time I start my player, the video shows full black picture and the audio starts playing good, if I use the seek bar to position the playback anywhere in the time, then the video shows correct picture but audio stops playing, eventually (it takes several seconds) audio starts playing again in sync with video.

  


  The player I am using is the one in Kodi 20.1.0, I have tried enabling/disabling HW acceleration without success.

  


  I also tried VLC 3.0.16, which has a simmilar behavior with the exception that video starts showing from the very begining when opening the file, just when seeking the audio stops for several seconds.

  


  I also tried using ffmpeg to reencode the video with different codecs without success, I have tried multiple video and audio codec combinations, and also tried changing the pixel format.

  


  Then eventually I managed to fix the video by extracting streams into separate files, then merge them again, without reencoding, I thought that maybe something could be wrong with the metadata of the video, so I copied the same metadata and final output file still works fine. My question is how can I fix my video without having to split and merge streams ?

  


  Here is the info about my input file :

  


  $ ffmpeg -i S01E08.mkv 
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
  configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 70.100 / 56. 70.100
  libavcodec     58.134.100 / 58.134.100
  libavformat    58. 76.100 / 58. 76.100
  libavdevice    58. 13.100 / 58. 13.100
  libavfilter     7.110.100 /  7.110.100
  libswscale      5.  9.100 /  5.  9.100
  libswresample   3.  9.100 /  3.  9.100
  libpostproc    55.  9.100 / 55.  9.100
Input #0, matroska,webm, from 'S01E08.mkv':
  Metadata:
    ENCODER         : Lavf59.27.100
  Duration: 00:25:02.21, start: 0.000000, bitrate: 1382 kb/s
  Stream #0:0: Video: hevc (Main 10), yuv420p10le(tv, bt470bg/unknown/unknown, progressive), 1280x960, 59.94 fps, 59.94 tbr, 1k tbn, 59.94 tbc
    Metadata:
      ENCODER         : Lavc59.37.100 libx265
      DURATION        : 00:25:00.032000000
  Stream #0:1(spa): Audio: aac (HE-AAC), 48000 Hz, stereo, fltp (default)
    Metadata:
      title           : Stereo
      DURATION        : 00:25:02.207000000
  Stream #0:2(jpn): Audio: aac (HE-AAC), 48000 Hz, stereo, fltp
    Metadata:
      title           : Stereo
      DURATION        : 00:25:00.116000000
  Stream #0:3(spa): Subtitle: ass
    Metadata:
      ENCODER         : Lavc59.37.100 ssa
      DURATION        : 00:24:56.585000000
At least one output file must be specified


  


  Here are the steps I followed to fix the video, I had to use mkv format for the outputs because other formats were no good at writing metadata :

  


  ffmpeg -i S01E08.mkv -map 0:v:0 -c:v copy -map_metadata 0 video.mkv
ffmpeg -i S01E08.mkv -map 0:a:0 -c:a copy -map_metadata 0 audio1.mkv
ffmpeg -i S01E08.mkv -map 0:a:1 -c:a copy -map_metadata 0 audio2.mkv
ffmpeg -i S01E08.mkv -map 0:s -c:s copy -map_metadata 0 subs.mkv
ffmpeg -i video.mkv -i audio1.mkv -i audio2.mkv -i subs.mkv -map 0:v -map 1:a -map 2:a -map 3:s -c:v copy -c:a copy -c:s copy -map_metadata 0 -map_metadata 1 -map_metadata 2 -map_metadata 3 -disposition:s:0 0 -default_mode infer_no_subs video_audio.mkv


  


  I had to compare both input and output files :

  


  $ ffmpeg -i video_audio.mkv
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
  configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 70.100 / 56. 70.100
  libavcodec     58.134.100 / 58.134.100
  libavformat    58. 76.100 / 58. 76.100
  libavdevice    58. 13.100 / 58. 13.100
  libavfilter     7.110.100 /  7.110.100
  libswscale      5.  9.100 /  5.  9.100
  libswresample   3.  9.100 /  3.  9.100
  libpostproc    55.  9.100 / 55.  9.100
Input #0, matroska,webm, from 'video_audio.mkv':
  Metadata:
    ENCODER         : Lavf58.76.100
  Duration: 00:25:02.21, start: 0.000000, bitrate: 1382 kb/s
  Stream #0:0: Video: hevc (Main 10), yuv420p10le(tv, bt470bg/unknown/unknown, progressive), 1280x960, 59.94 fps, 59.94 tbr, 1k tbn, 59.94 tbc (default)
    Metadata:
      ENCODER         : Lavc59.37.100 libx265
      DURATION        : 00:25:00.031000000
  Stream #0:1(spa): Audio: aac (HE-AAC), 48000 Hz, stereo, fltp (default)
    Metadata:
      title           : Stereo
      DURATION        : 00:25:02.207000000
  Stream #0:2(jpn): Audio: aac (HE-AAC), 48000 Hz, stereo, fltp
    Metadata:
      title           : Stereo
      DURATION        : 00:25:00.116000000
  Stream #0:3(spa): Subtitle: ass
    Metadata:
      ENCODER         : Lavc59.37.100 ssa
      DURATION        : 00:24:56.585000000
At least one output file must be specified

$ ffprobe -show_streams S01E08.mkv > probe_bad.txt
$ ffprobe -show_streams video_audio.mkv > probe_good.txt
$ diff probe_bad.txt probe_good.txt 
32,33c32,33
< duration_ts=1502207
< duration=1502.207000
---
> duration_ts=N/A
> duration=N/A
40c40
< DISPOSITION:default=0
---
> DISPOSITION:default=1
53c53
< TAG:DURATION=00:25:00.032000000
---
> TAG:DURATION=00:25:00.031000000


  


  Then I thought that maybe the disposition flag in the video stream was causing the error, so I ran ffmpeg to copy all streams from my input video S01E08.mkv and added the disposition flag with the following command :

  


  ffmpeg -i S01E08.mkv -map 0:v -map 0:a:0 -map 0:a:1 -map 0:s -c:v copy -c:a copy -c:s copy -map_metadata 0 -disposition:v:0 default -disposition:s:0 0 -default_mode infer_no_subs test.mkv


  


  Still no luck, test.mkv shows no video when opening the file, I had to double check that disposition flag was changed :

  


  $ ffprobe -show_streams test.mkv > probe_bad2.txt
$ diff probe_bad.txt probe_bad2.txt 
40c40
< DISPOSITION:default=0
---
> DISPOSITION:default=1
53c53
< TAG:DURATION=00:25:00.032000000
---
> TAG:DURATION=00:25:00.031000000


  


  In other words I have to focibly split and merge to make the file work correctly. May be I need to set the duration and duration_ts of the video stream to N/A but I don't know how to do this.

  


• 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()
    }

}

  


  


  



• ffmpeg split audio file into individual segments

  4 novembre 2022, par Martin

  I have a 00:15:24 length .mp3 file that I want to split up into three separate files, ideally using a txt file input like so :

  


  segemnts.txt

  


  00:00:00 00:04:55 seg1
00:04:55 00:08:41 seg2
00:08:41 00:15:24 seg3


  


  How can I do this with a single command ?