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• Python vlc stream from outside NAT

  7 octobre 2020, par xKedar

  I'm trying to stream, using FFmpeg, my webcam and audio from PC1 to PC2 in another LAN.

  


  PC1 : Public IP address with port forwarding so I can reach it

  


  PC2 : In a different NAT from PC1

  


  I basically run a server on PC1 in order to acquire IP and port from PC2 and reply on the same address

  


      import socket

    localPort   = 1234
    bufferSize  = 1024

    UDPServerSocket = socket.socket(family=socket.AF_INET, type=socket.SOCK_DGRAM)
    UDPServerSocket.bind(("", localPort)) # Bind to address and port

    while(True):
        bytesAddressPair = UDPServerSocket.recvfrom(bufferSize)
        message = bytesAddressPair[0].decode("utf-8")
        address = bytesAddressPair[1]
        # Sending a reply to client
        UDPServerSocket.sendto(str.encode("Hello"), address)
        break

    UDPServerSocket.close()


  


  Then I try to send the stream with the same port number both for the server(localPort) and the client(the one I acquired from address)

  


      import re
    from threading import Thread
    from subprocess import Popen, PIPE

    def detect_devices():
            list_cmd = 'ffmpeg -list_devices true -f dshow -i dummy'.split()
            p = Popen(list_cmd, stderr=PIPE)
            flagcam = flagmic = False
            for line in iter(p.stderr.readline,''):
                if flagcam:
                    cam = re.search('".*"',line.decode(encoding='UTF-8')).group(0)
                    cam = cam if cam else ''
                    flagcam = False
                if flagmic:
                    mic = re.search('".*"',line.decode(encoding='UTF-8')).group(0)
                    mic = mic if mic else ''
                    flagmic = False
                elif 'DirectShow video devices'.encode(encoding='UTF-8') in line:
                    flagcam = True
                elif 'DirectShow audio devices'.encode(encoding='UTF-8') in line:
                    flagmic = True
                elif 'Immediate exit requested'.encode(encoding='UTF-8') in line:
                    break
            return cam, mic   


    class ffmpegThread (Thread):
        def __init__(self, address):
            Thread.__init__(self)
            self.address = address

        def run(self):
            cam, mic = detect_devices()
            command = 'ffmpeg -f dshow -i video='+cam+':audio='+mic+' -profile:v high -pix_fmt yuvj420p -level:v 4.1 -preset ultrafast -tune zerolatency -vcodec libx264 -r 10 -b:v 512k -s 240x160 -acodec aac -ac 2 -ab 32k -ar 44100 -f mpegts -flush_packets 0 -t 40 udp://'+self.address+'?pkt_size=1316?localport='+str(localPort)
            p = Popen(command , stderr=PIPE)
            for line in iter(p.stderr.readline,''):
                if len(line) <5: break
            p.terminate()

    thread1 = ffmpegThread(address[0]+":"+str(address[1]))
    thread1.start()


  


  While on the other side(PC2) I have :

  


      from threading import Thread
    import tkinter as tk
    import vlc

    class myframe(tk.Frame):
        def __init__(self, width=240, height=160):
            self.root = tk.Tk()
            super(myframe, self).__init__(self.root)
            self.root.geometry("%dx%d" % (width, height))
            self.root.wm_attributes("-topmost", 1)
            self.grid()
            self.frame = tk.Frame(self, width=240, height=160)
            self.frame.configure(bg="black")
            self.frame.grid(row=0, column=0, columnspan=2)
            self.play()
            self.root.mainloop()

        def play(self):
            self.player = vlc.Instance().media_player_new()
            self.player.set_mrl('udp://@0.0.0.0:5000')
            self.player.set_hwnd(self.frame.winfo_id())
            self.player.play()

    class guiThread (Thread):
        def __init__(self, nome):
            Thread.__init__(self)
            self.nome = nome

        def run(self):
            app = myframe()


  


  and :

  


      import socket

    msgFromClient       = "Hello UDP Server"
    bytesToSend         = str.encode(msgFromClient)
    serverAddressPort   = ("MYglobal_IPaddress", 1234)
    bufferSize          = 1024
    localPort   = 5000

    # Create a UDP socket at client side
    UDPClientSocket = socket.socket(family=socket.AF_INET, type=socket.SOCK_DGRAM) 
    UDPClientSocket.bind(("", localPort))

    UDPClientSocket.sendto(bytesToSend, serverAddressPort)

    msgFromServer = UDPClientSocket.recvfrom(bufferSize)
    msg = msgFromServer[0].decode("utf-8")
    print(msg)
    UDPClientSocket.close()
    gui = guiThread("ThreadGUI")
    gui.start()


  


  Where I basically try to reach the server both to send my IP:Port and to punch a hole in the NAT in order to be able to get the packages sent from PC1 despite being behind a NAT.

  


  I think it is not working because I can not reach PC2 but I really can not figure out how to fix that because I was expecting that the first part, where I reach PC1 from PC2 was enough in order to establish a connection

  


• ffmpeg rtp-stream with gsm-codec

  15 octobre 2020, par Birgit

  I want to use ffmpeg for encoding and decoding gsm. I built ffmpeg with the --enable-libgsm option.

  


  I can now use the ffmpeg-command-line-tool to read gsm-encoded files, convert files to gsm, and also receive a gsm-encoded rtp stream.
So therefore I think the gsm-encoder and gsm-decoder are working properly.

  


  But for some reason I am not able to send and gsm-encoded rtp-stream.

  


  I tried the following comands :

  


  ffmpeg -re -i test.wav -c:a libgsm -f rtp rtp://127.0.0.1:5000

  


  ffmpeg -re -i test.wav -c:a gsm -f rtp rtp://127.0.0.1:5000

  


  I receive the error : Unsupported codec gsm. Could not write header for output file.

  


  I tried to use gdb to see what's going on. I think the problem is that in the file libavformat/rtpenc.c:49 gsm is not under the supported codecs. Does that mean it is not possible to use ffmpeg to create a gsm-encoded rtp-stream ? Is there a workaround, to overcome this issue ?

  


  I would appreciate any help and hints what I could try. :)

  


• how to send the input data to FFMPEG from a C# program

  18 octobre 2020, par jstuardo

  I need to send a binary stream to FFMPEG so that it sends to an RTMP server.

  


  I did it in a nodejs script using socket.io library and in Linux. It works perfectly.

  


  I need to do the same, but in a Windows Forms application using C#.

  


  This is how I run the ffmpeg.exe application :

  


          _currentProcess = new Process();
        _currentProcess.StartInfo.FileName = _ffmpegExe;
        _currentProcess.StartInfo.Arguments = BuildOptions(framesPerSecond, audioBitRate, audioEncoding, rtmpServer);
        _currentProcess.StartInfo.UseShellExecute = false;
        _currentProcess.StartInfo.CreateNoWindow = true;
        _currentProcess.StartInfo.RedirectStandardInput = true;
        _currentProcess.StartInfo.RedirectStandardError = true;
        _currentProcess.ErrorDataReceived += CurrentProcess_ErrorDataReceived;
        _currentProcess.Start();
        _currentProcess.BeginErrorReadLine();


  


  BuildOptions method is defined this way :

  


      private string BuildOptions(int framesPerSecond, int audioBitRate, string audioEncoding, string rtmpServer)
    {
        string options;
        if (framesPerSecond == 1)
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -r 1 -g 2 -keyint_min 2 -x264opts keyint=2 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate}-f flv {rtmpServer}";
        }
        else if (framesPerSecond == 15)
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency max_muxing_queue_size 1000 -bufsize 5000 -r 15 -g 30 -keyint_min 30 -x264opts keyint=30 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate} -f flv {rtmpServer}";
        }
        else
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -c:a aac -ar {audioBitRate} -b:a {audioEncoding} -bufsize 5000 -f flv {rtmpServer}";
        }

        return options;
    }


  


  I am sending the data to the standard input this way :

  


      public void EncodeAndSend(byte[] data)
    {
        if (_currentProcess != null)
        {
            var streamWriter = _currentProcess.StandardInput;
            streamWriter.Write(Encoding.GetEncoding("ISO-8859-1").GetChars(data));
        }
    }


  


  And finally, this method is for receiving the standard error which receives the result from ffmpeg.exe :

  


      private void CurrentProcess_ErrorDataReceived(object sender, DataReceivedEventArgs e)
    {
        Console.WriteLine(e.Data);
    }


  


  When I run the application, this is shown in the console :

  


  ffmpeg version 4.3.1-2020-10-01-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.2.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
pipe:: Invalid data found when processing input


  


  If I change the EncodeAndSend method to be :

  


      public void EncodeAndSend(byte[] data)
    {
        if (_currentProcess != null)
        {
            var streamWriter = _currentProcess.StandardInput;
            streamWriter.Write(data);
        }
    }


  


  pipe:: Invalid data found when processing input error is not produced, but no more outputs are shown so it seems it is not working.

  


  What is wrong with this ? how can I send the data to the FFMPEG process ?

  


  Finally, I tell you that the binary stream comes from the camera by mean of MediaRecorder in a web page (the same used for my program in nodejs server, so that it is not the issue here)