From ff15e6c1a662a5efc710aa714596e1ecbead8f14 Mon Sep 17 00:00:00 2001 From: Erik Zandvliet Date: Tue, 13 Mar 2012 08:42:37 +0100 Subject: [PATCH] "Added documentation to Connector_RTSP/main.cpp for #8" --- Connector_RTSP/main.cpp | 81 ++++++++++++++++++++++++++++++++++------- 1 file changed, 68 insertions(+), 13 deletions(-) diff --git a/Connector_RTSP/main.cpp b/Connector_RTSP/main.cpp index a861aec7..b0b0edd4 100644 --- a/Connector_RTSP/main.cpp +++ b/Connector_RTSP/main.cpp @@ -23,7 +23,9 @@ #include "rtp.h" /// Reads a single NALU from std::cin. Expected is H.264 Bytestream format. +/// Function was used as a way of debugging data. FLV does not contain all the metadata we need, so we had to try different approaches. /// \return The Nalu data. +/// \todo Throw this function away when everything works, it is not needed. std::string ReadNALU( ) { static char Separator[3] = { (char)0x00, (char)0x00, (char)0x01 }; std::string Buffer; @@ -38,62 +40,95 @@ std::string ReadNALU( ) { return Result; } -/// The main function of the connector -/// \param conn A connection with the client +/// The main function of the connector. +/// Used by server_setup.cpp in the bottom of the file, to start up the Connector. +/// This function contains the while loop the accepts connections, and sends them data. +/// \param conn A connection with the client. int RTSP_Handler( Socket::Connection conn ) { - FLV::Tag tag;///< Temporary tag buffer for incoming video data. + /// \todo Convert this to DTSC::DTMI, with an additional DTSC::Stream/ + FLV::Tag tag;// Temporary tag buffer for incoming video data. bool PlayVideo = false; bool PlayAudio = true; + //JRTPlib Objects to handle the RTP connection, which runs "parallel" to RTSP. jrtplib::RTPSession VideoSession; jrtplib::RTPSessionParams VideoParams; jrtplib::RTPUDPv6TransmissionParams VideoTransParams; std::string PreviousRequest = ""; Socket::Connection ss(-1); HTTP::Parser HTTP_R, HTTP_S; + //Some clients appear to expect a single request per connection. Don't know which ones. bool PerRequest = false; + //The main loop of the function while(conn.connected() && !FLV::Parse_Error) { if( HTTP_R.Read(conn ) ) { + //send Debug info to stderr. + //send the appropriate responses on RTSP Commands. fprintf( stderr, "REQUEST:\n%s\n", HTTP_R.BuildRequest().c_str() ); HTTP_S.protocol = "RTSP/1.0"; if( HTTP_R.method == "OPTIONS" ) { + //Always return the requested CSeq value. HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); + //The minimal set of options required for RTSP, add new options here as well if we want to support these. HTTP_S.SetHeader( "Public", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY" ); + //End the HTTP body, IMPORTANT!! Connection hangs otherwise!! 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" ) ); } else if ( HTTP_R.method == "DESCRIBE" ) { + ///\todo Implement DESCRIBE option. + //Don't know if a 501 response is seen as valid. If it is, don't bother changing it. if( HTTP_R.GetHeader( "Accept" ).find( "application/sdp" ) == std::string::npos ) { fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() ); conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) ); } else { HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "Content-Type", "application/sdp" ); - /// \todo Retrieve presence of video and audio data, and process into response - /// \todo Retrieve Packetization mode ( is 0 for now ). Where can I retrieve this? + /// \todo Retrieve presence of video and audio data, and process into response. Can now easily be done through DTSC::DTMI + /// \todo Retrieve Packetization mode ( is 0 for now ). I suppose this is the H264 packetization mode. Can maybe be retrieved from the docs on H64. + /// \todo Send a valid SDP file. + /// \todo Add audio to SDP file. + //This is just a dummy with data that was supposedly right for our teststream. + //SDP Docs: http://tools.ietf.org/html/rfc4566 + //v=0 + //o=- 0 0 IN IP4 ddvtech.com + //s=Fifa Test + //c=IN IP4 127.0.0.1 + //t=0 0 + //a=recvonly + //m=video 0 RTP/AVP 98 + //a=control:rtsp://localhost/fifa/video + //a=rtpmap:98 H264/90000 + //a=fmtp:98 packetization-mode=0 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=0\r\n\r\n");//m=audio 0 RTP/AAP 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" ) ); } } else if ( HTTP_R.method == "SETUP" ) { std::string temp = HTTP_R.GetHeader("Transport"); + //Extract the random UTP pair for video data ( RTP/RTCP) int ClientRTPLoc = temp.find( "client_port=" ) + 12; int PortSpacer = temp.find( "-", ClientRTPLoc ); int RTPClientPort = atoi( temp.substr( ClientRTPLoc, ( PortSpacer - ClientRTPLoc ) ).c_str() ); if( HTTP_S.GetHeader( "Session" ) != "" ) { + //Return an error if a second client tries to connect with an already running stream. 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) ); /// \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 { + /// \todo Add support for audio +// if( HTTP_R.url.find( "audio" ) != std::string::npos ) { +// HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50002-50003" ); +// } else { + //send video data HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50000-50001" ); + //Stub data for testing purposes. This should now be extracted somehow from DTSC::DTMI VideoParams.SetOwnTimestampUnit( ( 1.0 / 29.917 ) * 90000.0 ); VideoParams.SetMaximumPacketSize( 10000 ); - //pick the right port here + //pick the right port here VideoTransParams.SetPortbase( 50000 ); + //create a JRTPlib session int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams, jrtplib::RTPTransmitter::IPv6UDPProto ); if( VideoStatus < 0 ) { std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl; @@ -101,13 +136,15 @@ int RTSP_Handler( Socket::Connection conn ) { } else { std::cerr << "Created video session\n"; } - /// \todo retrieve other client than localhost --> Socket::Connection has no support for this yet? - + + /// \todo Connect with clients other than localhost uint8_t localip[32]; int status = inet_pton( AF_INET6, conn.getHost().c_str(), localip ) ; + //Debug info std::cerr << "Status: " << status << "\n"; jrtplib::RTPIPv6Address addr(localip,RTPClientPort); + //add the destination address to the VideoSession VideoStatus = VideoSession.AddDestination(addr); if (VideoStatus < 0) { std::cerr << jrtplib::RTPGetErrorString(VideoStatus) << std::endl; @@ -115,19 +152,24 @@ int RTSP_Handler( Socket::Connection conn ) { } else { std::cerr << "Destination Set\n"; } + //Stub data for testing purposes. + //Payload type should confirm with the SDP File. 98 == H264 / AVC VideoSession.SetDefaultPayloadType(98); VideoSession.SetDefaultMark(false); + //We have no idea if this timestamp has to correspond with the OwnTimeStampUnit() above. 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" ) ); } } else if( HTTP_R.method == "PLAY" ) { if( HTTP_R.GetHeader( "Range" ).substr(0,4) != "npt=" ) { + //We do not support this, whatever it is. Not needed for minimal compliance. fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() ); conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) ); } else { + //Initializes for actual streaming over the SETUP connection. HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "Session", HTTP_R.GetHeader( "Session" ) ); HTTP_S.SetHeader( "Range", HTTP_R.GetHeader( "Range" ) ); @@ -135,15 +177,20 @@ int RTSP_Handler( Socket::Connection conn ) { 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" ) ); + //Used further down, to start streaming video. + //PlayAudio = true; PlayVideo = true; } } else if( HTTP_R.method == "TEARDOWN" ) { + //If we were sending any stream data at this point, stop it, but keep the setup. HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); 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" ) ); + //PlayAudio = false; PlayVideo = false; } else { + //We do not implement other commands ( yet ) fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() ); conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) ); } @@ -154,14 +201,18 @@ int RTSP_Handler( Socket::Connection conn ) { } } if( PlayVideo ) { - /// \todo Select correct source + /// \todo Select correct source. This should become the DTSC::DTMI or the DTSC::Stream, whatever seems more natural. std::string VideoBuf = ReadNALU( ); if( VideoBuf == "" ) { + //videobuffer is empty, no more data. jrtplib::RTPTime delay = jrtplib::RTPTime(10.0); VideoSession.BYEDestroy(delay,"Out of data",11); conn.close(); } else { + //Send a single NALU (H264 block) here. VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size(), 98, false, ( 1.0 / 29.917 ) * 90000 ); + //we can add delays here as follows: + //don't know if these are nescecary or not, but good for testing nonetheless // jrtplib::RTPTime delay( ( 1.0 / 29.917 ) * 90000 ); // jrtplib::RTPTime::Wait( delay ); } @@ -170,7 +221,11 @@ int RTSP_Handler( Socket::Connection conn ) { return 0; } +//Set Default Port #define DEFAULT_PORT 554 +//Set the function that should be forked for each client #define MAINHANDLER RTSP_Handler +//Set the section in the Config file, though we will not use this yet #define CONFIGSECT RTSP +//Include the main functionality, as well as fork support and everything. #include "../util/server_setup.cpp"