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Merge branch 'DTSC' of github.com:DDVTECH/DMS into DTSC

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This commit is contained in:
Thulinma 2012-03-22 15:16:17 +01:00
commit 05dd8b41c5
1 changed files with 57 additions and 4 deletions
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61
Connector_RTSP/main.cpp
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@ -55,6 +55,7 @@ int RTSP_Handler( Socket::Connection conn ) {
jrtplib::RTPSessionParams VideoParams;
jrtplib::RTPUDPv6TransmissionParams VideoTransParams;
std::string PreviousRequest = "";
std::string streamname;
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.
@ -102,7 +103,7 @@ int RTSP_Handler( Socket::Connection conn ) {
"t=0 0\r\n" //time the session is active: start-time stop-time; "0 0"=permanent session
"a=recvonly\r\n"//zero or more session attribute lines
"m=video 0 RTP/AVP 98\r\n"//media name and transport address: media port proto fmt ...
"a=control:rtsp://localhost/fifa/video\r\n"//rfc2326 C.1.1, URL for aggregate control on session level
"a=control:" + HTTP_R.url + "\r\n"//rfc2326 C.1.1, URL for aggregate control on session level
"a=rtpmap:98 H264/90000\r\n"//rfc2326 C.1.3, dynamic payload type; see also http://tools.ietf.org/html/rfc1890#section-5
"a=fmtp:98 packetization-mode=0"//codec-specific parameters
"\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");
@ -111,6 +112,7 @@ int RTSP_Handler( Socket::Connection conn ) {
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
}
} else if ( HTTP_R.method == "SETUP" ) {
bool setup_session = false;//whether a session should be setup or not
std::string temp = HTTP_R.GetHeader("Transport");
//Extract the random UTP pair for video data ( RTP/RTCP)
int ClientRTPLoc = temp.find( "client_port=" ) + 12;
@ -121,6 +123,36 @@ int RTSP_Handler( Socket::Connection conn ) {
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 {
do{
if (!ss.connected()){
/// \todo Put stream name-to-file mapping in a separate util file or even class
streamname = std::string(HTTP_R.url.c_str());
unsigned int slash_pos = streamname.rfind('/');
if (slash_pos != std::string::npos) streamname.erase(0, slash_pos);
for (std::string::iterator i=streamname.begin(); i != streamname.end(); ++i){
if (*i == '?'){
streamname.erase(i, streamname.end());
break;
}
if (!isalpha(*i) && !isdigit(*i) && *i != '_'){
streamname.erase(i);
--i;
}else{
*i = tolower(*i);
}
}
streamname = "/tmp/shared_socket_" + streamname;
ss = Socket::Connection(streamname);
if (!ss.connected()){
streamname = "";
HTTP_R.BuildResponse("404", "Not Found");
break; //skip the session below
}
}
setup_session = true;
}while(0);
}
if (setup_session) {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Session", time(NULL) );
/// \todo Add support for audio
@ -216,13 +248,34 @@ int RTSP_Handler( Socket::Connection conn ) {
}
}
if( PlayVideo ) {
/// \todo Select correct source. This should become the DTSC::DTMI or the DTSC::Stream, whatever seems more natural.
std::string VideoBuf = ReadNALU( );
if( VideoBuf == "" ) {
bool no_data_ignore = false;
std::string VideoBuf;
ss.canRead();
switch (ss.ready()) {
case -1:
std::cerr << "Buffer socket is disconnected\n";
break;
case 0://not ready
no_data_ignore = true;
break;
default:
///\todo Make it work!
DTSC::Stream ds;
ss.spool();
if (ds.parsePacket(ss.Received())){
VideoBuf = ds.lastData();
}else{
std::cerr << "Failed to parse packet" << std::endl;
no_data_ignore = true;//perhaps corrupt?
}
break;
}
if(no_data_ignore){}else if( VideoBuf == "" ) {
//videobuffer is empty, no more data.
jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
VideoSession.BYEDestroy(delay,"Out of data",11);
conn.close();
std::cerr << "Buffer empty - closing connection" << std::endl;
} else {
//Send a single NALU (H264 block) here.
VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size(), 98, false, ( 1.0 / 29.917 ) * 90000 );