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• Problem with ffmpeg rtp stream to janus webrtc

  13 juin 2020, par XPModder

  I am trying to use ffmpeg and janus-gateway to stream video in the local network. I am piping the h264 video directly in to ffmpeg and from there it gets transferred to janus as an rtp stream. Janus then does the rest.

  



  The problem is, that when I try to open the stream using the streamingtest html page included in janus, I can select the stream, but I never get to see anything. On the console where I started janus, it throws multiple errors starting with : "SDP missing mandatory information"

  



  After starting ffmpeg it shows the SDP in console :

  



  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.20.100
m=video 8004 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; sprop-parameter-sets=J2QAKKwrQDwBE/LAPEiagA==,KO4BNywA; profile-level-id=640028


  



  The command I use to start ffmpeg is the following :

  



  ffmpeg -loglevel quiet -y -i - -c:v copy -preset veryfast -f rtp rtp://127.0.0.1:8004


  



  I assume that I am missing something in the ffmpeg command and therefore there is some kind of problem with the SDP.

  



  Edit : I looked a little further into this and actually looked at the code for janus at the lines that throw the error. Then I traced that to the origin of the error.

  



  Apparently the SDP is supposed to contain some authorization in for of this :

  



  a=... ice-ufrag=?; ice-pwd=?


  



  So how do I find out what this username and password are and how do I get it in to the SDP from ffmpeg ?

  



  Or how do I disable this entirely ?

  


• Decoding H.264 stream using FFmpeg on Java

  25 mai 2020, par maru2213

  I'm trying to decode H.264 stream, which is sent over Socket from an Android application to a computer. And I also want to show the decoded stream using JavaFX. I searched much hours, and decided to use JavaCV / FFmpeg. However I got error from FFmpeg. (I was inspired by this code)

  



  Questions :

  



  

  • Why does FFmpeg make error ?

  


  • Is it a correct way to convert AVFrame to javafx.scene.image.Image ?

  


  



  I'm using :

  



  

  • javacv-platform 1.4.4

  


  • ffmpeg-platform 4.1-1.4.4

  


  



  Code :

  



  This is a part of import and class fields, and method which runs once at the first time. (Actually the content of initialize() is wrapped by try catch.)

  



      import javafx.scene.image.Image;

    private avcodec.AVCodec avCodec;
    private avcodec.AVCodecContext avCodecContext;
    private avutil.AVDictionary avDictionary;
    private avutil.AVFrame avFrame;

    public void initialize() {
        avCodec = avcodec_find_decoder(AV_CODEC_ID_H264);
        if (avCodec == null) {
            throw new RuntimeException("Can't find decoder");
        }
        avCodecContext = avcodec_alloc_context3(avCodec);
        if (avCodecContext == null) {
            throw new RuntimeException("Can't allocate decoder context");
        }
        int result = avcodec_open2(avCodecContext, avCodec, (AVDictionary) null);
        if (result < 0) {
            throw new RuntimeException("Can't open decoder");
        }
        avFrame = av_frame_alloc();
        if (avFrame == null) {
            throw new RuntimeException("Can't allocate frame");
        }
    }


  



  And this is a method which is called every time when I receive a packet from Android. byte[] data is the packet data starting with 0x00, 0x00, 0x00, 0x01.

  



  The place where I get error is number_of_written_bytes. It always gets <0.

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      private void decode(byte[] data) {
        AVPacket avPacket = new AVPacket();
        av_init_packet(avPacket);
        avPacket.pts(AV_NOPTS_VALUE);
        avPacket.dts(AV_NOPTS_VALUE);
        BytePointer bytePointer = new BytePointer(data);
        bytePointer.capacity(data.length);
        avPacket.data(bytePointer);
        avPacket.size(data.length);
        avPacket.pos(-1);

        avcodec_send_packet(avCodecContext, avPacket);
        int result = avcodec_receive_frame(avCodecContext, avFrame);
        if (result >= 0) {
            int bufferOutputSize = av_image_get_buffer_size(avFrame.format(), avFrame.width(), avFrame.height(), 16);
            Pointer pointer = av_malloc(bufferOutputSize);
            BytePointer outputPointer = new BytePointer(pointer);
            int number_of_written_bytes = av_image_copy_to_buffer(outputPointer, bufferOutputSize, avFrame.data(), avFrame.linesize(), avFrame.chroma_location(), avFrame.width(), avFrame.height(), 1);
            if (number_of_written_bytes < 0) {
                //The process always come here.
                throw new RuntimeException("Can't copy image to buffer");
            }

            System.out.println("decode success");
            Image image = new Image(new ByteArrayInputStream(outputPointer.asBuffer().array()));
        } else {
            System.out.println("decode failed");
        }
    }


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• (Ffmpeg) How to play live audio in the browser from received UDP packets using Ffmpeg ?

  26 octobre 2022, par Yousef Alaqra

  I have .NET Core console application which acts as UDP Server and UDP Client

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  • UDP client by receiving audio packet.

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  • UDP server, by sending each received packet.

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  Here's a sample code of the console app :

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  static UdpClient udpListener = new UdpClient();
    static IPEndPoint endPoint = new IPEndPoint(IPAddress.Parse("192.168.1.230"), 6980);
    static IAudioSender audioSender = new UdpAudioSender(new IPEndPoint(IPAddress.Parse("192.168.1.230"), 65535));

    static void Main(string[] args)
    {
        udpListener.Client.SetSocketOption(SocketOptionLevel.Socket, SocketOptionName.ReuseAddress, true);
        udpListener.Client.Bind(endPoint);

        try
        {
            udpListener.BeginReceive(new AsyncCallback(recv), null);
        }
        catch (Exception e)
        {
            throw e;
        }

        Console.WriteLine("Press enter to dispose the running service");
        Console.ReadLine();
    }

    private async static void recv(IAsyncResult res)
    {
        byte[] received = udpListener.EndReceive(res, ref endPoint);
        OnAudioCaptured(received);
        udpListener.BeginReceive(new AsyncCallback(recv), null);
    }


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  On the other side, I have a node js API application, which supposes to execute an FFmpeg command as a child process and to do the following

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  • receive the audio packet as an input from the console app UDP server.

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  • convert the received bytes into WebM

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  • pipe out the result into the response.

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  Finally, in the client-side, I should have an audio element with source value equals to the http://localhost:3000

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  For now, I can only execute this FFmpeg command :

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  ffmpeg -f  s16le  -ar 48000 -ac 2 -i 'udp://192.168.1.230:65535' output.wav


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  Which do the following

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  • Receive UDP packet as an input

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  • Convert the received bytes into the output.wav audio file.

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  How would I execute a child process in the node js server which receives the UDP packets and pipe out the result into the response as Webm ?

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