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