RTP on Android MediaPlayer

2019-02-06 16:22发布

站内文章 / Android
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问题:

I've implemented RTSP on Android MediaPlayer using VLC as rtsp server with this code:

# vlc -vvv /home/marco/Videos/pippo.mp4 --sout 
#rtp{dst=192.168.100.246,port=6024-6025,sdp=rtsp://192.168.100.243:8080/test.sdp}

and on the Android project: 

Uri videoUri = Uri.parse("rtsp://192.168.100.242:8080/test.sdp"); 
videoView.setVideoURI(videoUri); 
videoView.start();

This works fine but if I'd like also to play live stream RTP so I copied the sdp file into the sdcard (/mnt/sdcard/test.sdp) and setting vlc: 

# vlc -vvv /home/marco/Videos/pippo.mp4 --sout 
#rtp{dst=192.168.100.249,port=6024-6025}

I tried to play the stream RTP setting the path of the sdp file locally: 

Uri videoUri = Uri.parse("/mnt/sdcard/test.sdp");
videoView.setVideoURI(videoUri); 
videoView.start();

But I got an error: 

D/MediaPlayer( 9616): Couldn't open file on client side, trying server side 
W/MediaPlayer( 9616): info/warning (1, 26) 
I/MediaPlayer( 9616): Info (1,26) 
E/PlayerDriver(   76): Command PLAYER_INIT completed with an error or info PVMFFailure 
E/MediaPlayer( 9616): error (1, -1)
E/MediaPlayer( 9616): Error (1,-1) 
D/VideoView( 9616): Error: 1,-1

Does anyone know where's the problem? I'm I wrong or it's not possible to play RTP on MediaPlayer? Cheers Giorgio

回答1:

I have a partial solution for you.

I'm currently working on a Ra&D project involving RTP streaming of medias from a server to Android clients.

By doing this work, I contribute to my own library called smpte2022lib you may find here : http://sourceforge.net/projects/smpte-2022lib/.

Helped with such library (the Java implementation is currently the best one) you may be able to parse RTP multicast streams coming from professional streaming equipements, VLC RTP sessions, ...

I already tested it successfully with streams coming from captured profesionnal RTP streams with SMPTE-2022 2D-FEC or with simple streams generated with VLC.

Unfortunately I cannot put a code-snippet here as the project using it is actually under copyright, but I ensure you you can use it simply by parsing UDP streams helped with RtpPacket constructor.

If the packets are valid RTP packets (the bytes) they will be decoded as such.

At this moment of time, I wrap the call to RtpPacket's constructor to a thread that actually stores the decoded payload as a media file. Then I will call the VideoView with this file as parameter.

Crossing fingers ;-)

Kind Regards,

David Fischer



回答2:

Possible in android using ( not mediaPlayer but other stuff further down the stack) but do you really want do pursue RTSP/RTP when the rest of the media ecosystem does not??

IMO - there are far better media/stream approaches under the umbrella of HTML5/WebRTC. Like look at what 'Ondello' is doing with streams.

That said, here is some old-project code for android/RTSP/SDP/RTP using 'netty' and 'efflux'. It will negotiate some portions of 'Sessions' on SDP file providers. Cant remember whether it would actually play the audio portion of Youtube/RTSP stuff, but that is what my goal was at the time. ( i think that it worked using AMR-NB codec but , there were tons of issues and i dropped RTSP on android like a bad habit!)

on Git.... 

        @Override
        public void mediaDescriptor(Client client, String descriptor)
        {
            // searches for control: session and media arguments.
            final String target = "control:";
            Log.d(TAG, "Session Descriptor\n" + descriptor);
            int position = -1;
            while((position = descriptor.indexOf(target)) > -1)
            {
                descriptor = descriptor.substring(position + target.length());
                resourceList.add(descriptor.substring(0, descriptor.indexOf('\r')));
            }
        }
        private int nextPort()
        {
            return (port += 2) - 2;
        }       


        private void getRTPStream(TransportHeader transport){

            String[] words;
            // only want 2000 part of 'client_port=2000-2001' in the Transport header in the response

            words = transport.getParameter("client_port").substring(transport.getParameter("client_port").indexOf("=") +1).split("-");
            port_lc = Integer.parseInt(words[0]);

            words = transport.getParameter("server_port").substring(transport.getParameter("server_port").indexOf("=") +1).split("-");
            port_rm = Integer.parseInt(words[0]);

            source = transport.getParameter("source").substring(transport.getParameter("source").indexOf("=") +1);          
            ssrc = transport.getParameter("ssrc").substring(transport.getParameter("ssrc").indexOf("=") +1);
            // assume dynamic Packet type = RTP , 99
            getRTPStream(session, source, port_lc, port_rm, 99);
            //getRTPStream("sessiona", source, port_lc, port_rm, 99);
            Log.d(TAG, "raw parms " +port_lc +" " +port_rm +" " +source );
//          String[] words = session.split(";");
        Log.d(TAG, "session: " +session);   
        Log.d(TAG, "transport: " +transport.getParameter("client_port") 
                +" "  +transport.getParameter("server_port") +" "  +transport.getParameter("source") 
                +" "  +transport.getParameter("ssrc"));

        }

        private void getRTPStream(String session, String source, int portl, int portr, int payloadFormat ){
            // what do u do with ssrc?
            InetAddress addr;
            try {
                addr = InetAddress.getLocalHost();
                // Get IP Address
 //             LAN_IP_ADDR = addr.getHostAddress();
                LAN_IP_ADDR = "192.168.1.125";
                Log.d(TAG, "using client IP addr " +LAN_IP_ADDR);

            } catch (UnknownHostException e1) {
                // TODO Auto-generated catch block
                e1.printStackTrace();
            }


            final CountDownLatch latch = new CountDownLatch(2);

            RtpParticipant local1 = RtpParticipant.createReceiver(new RtpParticipantInfo(1), LAN_IP_ADDR, portl, portl+=1);
     //       RtpParticipant local1 = RtpParticipant.createReceiver(new RtpParticipantInfo(1), "127.0.0.1", portl, portl+=1);
            RtpParticipant remote1 = RtpParticipant.createReceiver(new RtpParticipantInfo(2), source, portr, portr+=1);


            remote1.getInfo().setSsrc( Long.parseLong(ssrc, 16));
            session1 = new SingleParticipantSession(session, payloadFormat, local1, remote1);

           Log.d(TAG, "remote ssrc " +session1.getRemoteParticipant().getInfo().getSsrc());

            session1.init();

            session1.addDataListener(new RtpSessionDataListener() {
                @Override
                public void dataPacketReceived(RtpSession session, RtpParticipantInfo participant, DataPacket packet) {
     //               System.err.println("Session 1 received packet: " + packet + "(session: " + session.getId() + ")");
                    //TODO close the file, flush the buffer
//                  if (_sink != null) _sink.getPackByte(packet);
                    getPackByte(packet);

     //             System.err.println("Ssn 1  packet seqn: typ: datasz "  +packet.getSequenceNumber()  + " " +packet.getPayloadType() +" " +packet.getDataSize());
     //             System.err.println("Ssn 1  packet sessn: typ: datasz "  + session.getId() + " " +packet.getPayloadType() +" " +packet.getDataSize());
 //                   latch.countDown();
                }

            });
     //       DataPacket packet = new DataPacket();
      //      packet.setData(new byte[]{0x45, 0x45, 0x45, 0x45});
     //       packet.setSequenceNumber(1);
     //       session1.sendDataPacket(packet);


//        try {
       //       latch.await(2000, TimeUnit.MILLISECONDS);
     //     } catch (Exception e) {
   //         fail("Exception caught: " + e.getClass().getSimpleName() + " - " + e.getMessage());

 //      }
        }
 //TODO  below should collaborate with the audioTrack object and should write to the AT buffr
        // audioTrack write was blocking forever 

    public void getPackByte(DataPacket packet) {
            //TODO this is getting called but not sure why only one time 
            // or whether it is stalling in mid-exec??

            //TODO on firstPacket write bytes and start audioTrack
            // AMR-nb frames at 12.2 KB or format type 7 frames are handled . 
            // after the normal header, the getDataArray contains extra 10 bytes of dynamic header that are bypassed by 'limit'


            // real value for the frame separator comes in the input stream at position 1 in the data array
            // returned by 

//          int newFrameSep = 0x3c;
            // bytes avail = packet.getDataSize() - limit;

//          byte[] lbuf = new byte[packet.getDataSize()];
//          if ( packet.getDataSize() > 0)
//              lbuf = packet.getDataAsArray();
            //first frame includes the 1 byte frame header whose value should be used 
            // to write subsequent frame separators 
            Log.d(TAG, "getPackByt start and play");

            if(!started){
                Log.d(TAG, " PLAY  audioTrak");
                track.play();
                started = true;
            }

//          track.write(packet.getDataAsArray(), limit, (packet.getDataSize() - limit));
            track.write(packet.getDataAsArray(), 0, packet.getDataSize() );
            Log.d(TAG, "getPackByt aft write");

//          if(!started && nBytesRead > minBufferSize){
    //          Log.d(TAG, " PLAY  audioTrak");
        //      track.play();
        //  started = true;}
            nBytesRead += packet.getDataSize(); 
            if (nBytesRead % 500 < 375) Log.d(TAG, " getPackByte plus 5K received");
        }       
    }


回答3:

Unfortunately it is not possible to play an RTP Stream with the Android MediaPlayer.

Solutions to this problems include the decoding of the RTP Stream with ffmpeg. Tutorials on how to compile ffmpeg for Android can be found on the Web.



标签： android media-player rtp
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