From dd2222d748042d82dd8f07aefab2edd516c1e3c7 Mon Sep 17 00:00:00 2001 From: Erik Zandvliet Date: Thu, 1 Sep 2011 01:47:13 +0200 Subject: [PATCH] RTSP: Blokkerig beeld, hapert veel, maar iig hebben we bewegend beeld --- Connector_RTSP/main.cpp | 130 ++++++++++++++-------------------------- 1 file changed, 45 insertions(+), 85 deletions(-) diff --git a/Connector_RTSP/main.cpp b/Connector_RTSP/main.cpp index 8feea13d..9a1b584f 100644 --- a/Connector_RTSP/main.cpp +++ b/Connector_RTSP/main.cpp @@ -22,6 +22,8 @@ //JRTPLIB #include "rtp.h" +/// Reads a single NALU from std::cin. Expected is H.264 Bytestream format. +/// \return The Nalu data. std::string ReadNALU( ) { static char Separator[3] = { (char)0x00, (char)0x00, (char)0x01 }; std::string Buffer; @@ -39,16 +41,9 @@ std::string ReadNALU( ) { /// The main function of the connector /// \param conn A connection with the client int RTSP_Handler( Socket::Connection conn ) { - bool ready4data = false; FLV::Tag tag;///< Temporary tag buffer for incoming video data. bool PlayVideo = false; bool PlayAudio = true; - bool InitVideo = false; - bool InitAudio = true; - bool VideoMeta = false; - int RTPClientPort; - int RTCPClientPort; - bool inited; jrtplib::RTPSession VideoSession; jrtplib::RTPSessionParams VideoParams; jrtplib::RTPUDPv4TransmissionParams VideoTransParams; @@ -59,13 +54,13 @@ int RTSP_Handler( Socket::Connection conn ) { while(conn.connected() && !FLV::Parse_Error) { if( HTTP_R.Read(conn ) ) { fprintf( stderr, "REQUEST:\n%s\n", HTTP_R.BuildRequest().c_str() ); - if( HTTP_R.GetHeader( "User-Agent" ).find( "RealMedia Player Version" ) != std::string::npos) { - PerRequest = true; - } +// if( HTTP_R.GetHeader( "User-Agent" ).find( "RealMedia Player Version" ) != std::string::npos) { +// PerRequest = true; +// } HTTP_S.protocol = "RTSP/1.0"; if( HTTP_R.method == "OPTIONS" ) { HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); - HTTP_S.SetHeader( "Public", "DESCRIBE, SETUP, TEARDOWN, PLAY" ); + HTTP_S.SetHeader( "Public", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY" ); HTTP_S.SetBody( "\r\n\r\n" ); fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() ); conn.write( HTTP_S.BuildResponse( "200", "OK" ) ); @@ -76,7 +71,9 @@ int RTSP_Handler( Socket::Connection conn ) { } else { HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "Content-Type", "application/sdp" ); - HTTP_S.SetBody( "v=0\r\no=- 0 0 IN IP4 ddvtech.com\r\ns=Fifa Test\r\nc=IN IP4 127.0.0.1\r\nt=0 0\r\na=recvonly\r\nm=video 0 RTP/AVP 98\r\na=rtpmap:98 H264/700000\r\n\r\n" );//a=fmtp:98 packetization-mode=1\r\n\r\n" ); + /// \todo Retrieve presence of video and audio data, and process into response + /// \todo Retrieve Packetization mode ( is 1 for now ). Where can I retrieve this? + HTTP_S.SetBody( "v=0\r\no=- 0 0 IN IP4 ddvtech.com\r\ns=Fifa Test\r\nc=IN IP4 127.0.0.1\r\nt=0 0\r\na=recvonly\r\nm=video 0 RTP/AVP 98\r\na=control:rtsp://localhost/fifa/video\r\na=rtpmap:98 H264/90000\r\na=fmtp:98 packetization-mode=1\r\nm=audio 0 RTP/AVP 96\r\na=control:rtsp://localhost/fifa/audio\r\na=rtpmap:96 mpeg4-generic/16000/2\r\n\r\n"); fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() ); conn.write( HTTP_S.BuildResponse( "200", "OK" ) ); } @@ -84,16 +81,44 @@ int RTSP_Handler( Socket::Connection conn ) { std::string temp = HTTP_R.GetHeader("Transport"); int ClientRTPLoc = temp.find( "client_port=" ) + 12; int PortSpacer = temp.find( "-", ClientRTPLoc ); - int PortEnd = ( temp.find( ";", PortSpacer ) ); - RTPClientPort = atoi( temp.substr( ClientRTPLoc, ( PortSpacer - ClientRTPLoc ) ).c_str() ); - RTCPClientPort = atoi( temp.substr( PortSpacer + 1 , ( PortEnd - ( PortSpacer + 1 ) ) ).c_str() ); + int RTPClientPort = atoi( temp.substr( ClientRTPLoc, ( PortSpacer - ClientRTPLoc ) ).c_str() ); if( HTTP_S.GetHeader( "Session" ) != "" ) { fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "459", "Aggregate Operation Not Allowed" ).c_str() ); conn.write( HTTP_S.BuildResponse( "459", "Aggregate Operation Not Allowed" ) ); } else { HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "Session", time(NULL) ); - HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50000-50001" ); + /// \todo "Random" generation of server_ports + if( HTTP_R.url.find( "audio" ) != std::string::npos ) { + HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50002-50003" ); + } else { + HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50000-50001" ); + VideoParams.SetOwnTimestampUnit( ( 1.0 / 29.917 ) * 90000.0 ); + VideoParams.SetMaximumPacketSize( 10000 ); + //pick the right port here + VideoTransParams.SetPortbase( 50000 ); + int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams ); + if( VideoStatus < 0 ) { + std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl; + exit( -1 ); + } else { + std::cerr << "Created video session\n"; + } + /// \todo retrieve other client than localhost --> Socket::Connection has no support for this yet? + uint8_t localip[]={127,0,0,1}; + jrtplib::RTPIPv4Address addr(localip,RTPClientPort); + + VideoStatus = VideoSession.AddDestination(addr); + if (VideoStatus < 0) { + std::cerr << jrtplib::RTPGetErrorString(VideoStatus) << std::endl; + exit(-1); + } else { + std::cerr << "Destination Set\n"; + } + VideoSession.SetDefaultPayloadType(98); + VideoSession.SetDefaultMark(false); + VideoSession.SetDefaultTimestampIncrement( ( 1.0 / 29.917 ) * 90000 ); + } HTTP_S.SetBody( "\r\n\r\n" ); fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() ); conn.write( HTTP_S.BuildResponse( "200", "OK" ) ); @@ -127,85 +152,20 @@ int RTSP_Handler( Socket::Connection conn ) { if( PerRequest ) { conn.close(); } -/* if( ( PlayVideo ) && !inited ) { - ss = Socket::Connection("/tmp/shared_socket_fifa"); - if (!ss.connected()){ - #if DEBUG >= 1 - fprintf(stderr, "Could not connect to server!\n"); - #endif - conn.close(); - break; - } - #if DEBUG >= 3 - fprintf(stderr, "Everything connected, starting to send video data...\n"); - #endif - inited = true; - }*/ } if( PlayVideo ) { - if( !InitVideo ) { - VideoParams.SetOwnTimestampUnit( 1.0/90000.0 ); - VideoParams.SetMaximumPacketSize( 10000 ); - VideoTransParams.SetPortbase( 50000 ); - int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams ); - if( VideoStatus < 0 ) { - std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl; - exit( -1 ); - } else { - std::cerr << "Created video session\n"; - } - uint8_t localip[]={127,0,0,1}; - jrtplib::RTPIPv4Address addr(localip,RTPClientPort); - - VideoStatus = VideoSession.AddDestination(addr); - if (VideoStatus < 0) { - std::cerr << jrtplib::RTPGetErrorString(VideoStatus) << std::endl; - exit(-1); - } else { - std::cerr << "Destination Set\n"; - } - VideoSession.SetDefaultPayloadType(98); - VideoSession.SetDefaultMark(false); - VideoSession.SetDefaultTimestampIncrement(0); - InitVideo = true; - } -// std::cerr << "Retrieving NALU from stdin\n"; + /// \todo Select correct source std::string VideoBuf = ReadNALU( ); if( VideoBuf == "" ) { jrtplib::RTPTime delay = jrtplib::RTPTime(10.0); VideoSession.BYEDestroy(delay,"Out of data",11); conn.close(); } else { -// std::cerr << "NALU Retrieved:\n"; -// std::cerr << "\t" << (int)VideoBuf[0] << " " << (int)VideoBuf[2] -// << " " << (int)VideoBuf[3] << " " << (int)VideoBuf[4] << "\n"; - VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size(), 98, false, ( 700000.0 / VideoBuf.size() ) ); + VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size() );//, 98, true, ( 1.0 / 29.917 ) ); +// jrtplib::RTPTime delay( 1.0 / 29.917 ); +// jrtplib::RTPTime::Wait( delay ); } } -/* - switch (ss.ready()){ - case -1: - conn.close(); - #if DEBUG >= 1 - fprintf(stderr, "Source socket is disconnected.\n"); - #endif - break; - case 0: break;//not ready yet - default: - if (tag.SockLoader(ss)){//able to read a full packet? - if( tag.data[ 0 ] == 0x09 ) { - if( ( ( tag.data[ 11 ] & 0x0F ) == 7 ) ) { //&& ( tag.data[ 12 ] == 1 ) ) { - fprintf(stderr, "Video contains NALU\n" ); - } - } - if( tag.data[ 0 ] == 0x08 ) { - if( ( tag.data[ 11 ] == 0xAF ) && ( tag.data[ 12 ] == 0x01 ) ) { - fprintf(stderr, "Audio Contains Raw AAC\n"); - } - } - } - break; - }*/ } return 0; }