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  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

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    8 février 2011, par

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    1er décembre 2010, par

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  • 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)

    


  • 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. :)

    


  • 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