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// TestRtsp.cpp : 定义控制台应用程序的入口点。
//

#include "stdafx.h"

//#include "liveMedia.hh"
//#include "BasicUsageEnvironment.hh"

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
**********/
// Copyright (c) 1996-2013, Live Networks, Inc.  All rights reserved
// A demo application, showing how to create and run a RTSP client (that can potentially receive multiple streams concurrently).
//
// NOTE: This code - although it builds a running application - is intended only to illustrate how to develop your own RTSP
// client application.  For a full-featured RTSP client application - with much more functionality, and many options - see
// "openRTSP": http://www.live555.com/openRTSP/

//TestRtspClient.cpp

#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"

// Forward function definitions:

// RTSP 'response handlers':
void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString);
void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString);
void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString);

// Other event handler functions:
void subsessionAfterPlaying(void* clientData); // called when a stream's subsession (e.g., audio or video substream) ends
void subsessionByeHandler(void* clientData); // called when a RTCP "BYE" is received for a subsession
void streamTimerHandler(void* clientData);
// called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")

// The main streaming routine (for each "rtsp://" URL):
void openURL(UsageEnvironment& env, char const* progName, char const* rtspURL);

// Used to iterate through each stream's 'subsessions', setting up each one:
void setupNextSubsession(RTSPClient* rtspClient);

// Used to shut down and close a stream (including its "RTSPClient" object):
void shutdownStream(RTSPClient* rtspClient, int exitCode = 1);

// A function that outputs a string that identifies each stream (for debugging output).  Modify this if you wish:
UsageEnvironment& operator<<(UsageEnvironment& env, const RTSPClient& rtspClient) {
 return env << "[URL:\"" << rtspClient.url() << "\"]: ";
}

// A function that outputs a string that identifies each subsession (for debugging output).  Modify this if you wish:
UsageEnvironment& operator<<(UsageEnvironment& env, const MediaSubsession& subsession) {
 return env << subsession.mediumName() << "/" << subsession.codecName();
}

void usage(UsageEnvironment& env, char const* progName) {
 env << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>\n";
 env << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)\n";
}

char eventLoopWatchVariable = 0;

int main(int argc, char** argv) {
 // Begin by setting up our usage environment:
 TaskScheduler* scheduler = BasicTaskScheduler::createNew();
 UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

 // We need at least one "rtsp://" URL argument:
 if (argc < 2) {
  usage(*env, argv[0]);
  return 1;
 }

 // There are argc-1 URLs: argv[1] through argv[argc-1].  Open and start streaming each one:
 for (int i = 1; i <= argc-1; ++i) {
  openURL(*env, argv[0], argv[i]);
 }

 // All subsequent activity takes place within the event loop:
 env->taskScheduler().doEventLoop(&eventLoopWatchVariable);
 // This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.

 return 0;

 // If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
 // and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
 // then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
 /*
 env->reclaim(); env = NULL;
 delete scheduler; scheduler = NULL;
 */
}

// Define a class to hold per-stream state that we maintain throughout each stream's lifetime:

class StreamClientState {
public:
 StreamClientState();
 virtual ~StreamClientState();

public:
 MediaSubsessionIterator* iter;
 MediaSession* session;
 MediaSubsession* subsession;
 TaskToken streamTimerTask;
 double duration;
};

// If you're streaming just a single stream (i.e., just from a single URL, once), then you can define and use just a single
// "StreamClientState" structure, as a global variable in your application.  However, because - in this demo application - we're
// showing how to play multiple streams, concurrently, we can't do that.  Instead, we have to have a separate "StreamClientState"
// structure for each "RTSPClient".  To do this, we subclass "RTSPClient", and add a "StreamClientState" field to the subclass:

class ourRTSPClient: public RTSPClient {
public:
 static ourRTSPClient* createNew(UsageEnvironment& env, char const* rtspURL,
  int verbosityLevel = 0,
  char const* applicationName = NULL,
  portNumBits tunnelOverHTTPPortNum = 0);

protected:
 ourRTSPClient(UsageEnvironment& env, char const* rtspURL,
  int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum);
 // called only by createNew();
 virtual ~ourRTSPClient();

public:
 StreamClientState scs;
};

// Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
// In practice, this might be a class (or a chain of classes) that decodes and then renders the incoming audio or video.
// Or it might be a "FileSink", for outputting the received data into a file (as is done by the "openRTSP" application).
// In this example code, however, we define a simple 'dummy' sink that receives incoming data, but does nothing with it.

class DummySink: public MediaSink {
public:
 static DummySink* createNew(UsageEnvironment& env,
  MediaSubsession& subsession, // identifies the kind of data that's being received
  char const* streamId = NULL); // identifies the stream itself (optional)

private:
 DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId);
 // called only by "createNew()"
 virtual ~DummySink();

 static void afterGettingFrame(void* clientData, unsigned frameSize,
  unsigned numTruncatedBytes,
 struct timeval presentationTime,
  unsigned durationInMicroseconds);

 void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
 struct timeval presentationTime, unsigned durationInMicroseconds);

private:
 // redefined virtual functions:
 virtual Boolean continuePlaying();

private:
 u_int8_t* fReceiveBuffer;
 MediaSubsession& fSubsession;
 char* fStreamId;
};

#define RTSP_CLIENT_VERBOSITY_LEVEL 1 // by default, print verbose output from each "RTSPClient"

static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTSPClient"s) are currently in use.

void openURL(UsageEnvironment& env, char const* progName, char const* rtspURL) {
 // Begin by creating a "RTSPClient" object.  Note that there is a separate "RTSPClient" object for each stream that we wish
 // to receive (even if more than stream uses the same "rtsp://" URL).
 RTSPClient* rtspClient = ourRTSPClient::createNew(env, rtspURL, RTSP_CLIENT_VERBOSITY_LEVEL, progName);
 if (rtspClient == NULL) {
  env << "Failed to create a RTSP client for URL \"" << rtspURL << "\": " << env.getResultMsg() << "\n";
  return;
 }

 ++rtspClientCount;

 // Next, send a RTSP "DESCRIBE" command, to get a SDP description for the stream.
 // Note that this command - like all RTSP commands - is sent asynchronously; we do not block, waiting for a response.
 // Instead, the following function call returns immediately, and we handle the RTSP response later, from within the event loop:
 rtspClient->sendDescribeCommand(continueAfterDESCRIBE);
}


// Implementation of the RTSP 'response handlers':

void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString) {
 do {
  UsageEnvironment& env = rtspClient->envir(); // alias
  StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

  if (resultCode != 0) {
   env << *rtspClient << "Failed to get a SDP description: " << resultString << "\n";
   delete[] resultString;
   break;
  }

  char* const sdpDescription = resultString;
  env << *rtspClient << "Got a SDP description:\n" << sdpDescription << "\n";

  // Create a media session object from this SDP description:
  scs.session = MediaSession::createNew(env, sdpDescription);
  delete[] sdpDescription; // because we don't need it anymore
  if (scs.session == NULL) {
   env << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << "\n";
   break;
  } else if (!scs.session->hasSubsessions()) {
   env << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)\n";
   break;
  }

  // Then, create and set up our data source objects for the session.  We do this by iterating over the session's 'subsessions',
  // calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
  // (Each 'subsession' will have its own data source.)
  scs.iter = new MediaSubsessionIterator(*scs.session);
  setupNextSubsession(rtspClient);
  return;
 } while (0);

 // An unrecoverable error occurred with this stream.
 shutdownStream(rtspClient);
}

// By default, we request that the server stream its data using RTP/UDP.
// If, instead, you want to request that the server stream via RTP-over-TCP, change the following to True:
//RTP over udp还是 over tcp
//REQUEST_STREAMING_OVER_TCP=false  over udp;
//REQUEST_STREAMING_OVER_TCP=true  over tcp

#define REQUEST_STREAMING_OVER_TCP False

void setupNextSubsession(RTSPClient* rtspClient) {
 UsageEnvironment& env = rtspClient->envir(); // alias
 StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

 scs.subsession = scs.iter->next();
 if (scs.subsession != NULL) {
  if (!scs.subsession->initiate()) {
   env << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n";
   setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
  } else {
   env << *rtspClient << "Initiated the \"" << *scs.subsession
    << "\" subsession (client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << ")\n";

   // Continue setting up this subsession, by sending a RTSP "SETUP" command:
   rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
  }
  return;
 }

 // We've finished setting up all of the subsessions.  Now, send a RTSP "PLAY" command to start the streaming:
 if (scs.session->absStartTime() != NULL) {
  // Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command:
  rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime());
 } else {
  scs.duration = scs.session->playEndTime() - scs.session->playStartTime();
  rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY);
 }
}

void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString) {
 do {
  UsageEnvironment& env = rtspClient->envir(); // alias
  StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

  if (resultCode != 0) {
   env << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << "\n";
   break;
  }

  env << *rtspClient << "Set up the \"" << *scs.subsession
   << "\" subsession (client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << ")\n";

  // Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
  // (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
  // after we've sent a RTSP "PLAY" command.)

  scs.subsession->sink = DummySink::createNew(env, *scs.subsession, rtspClient->url());
  // perhaps use your own custom "MediaSink" subclass instead
  if (scs.subsession->sink == NULL) {
   env << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
    << "\" subsession: " << env.getResultMsg() << "\n";
   break;
  }

  env << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n";
  scs.subsession->miscPtr = rtspClient; // a hack to let subsession handle functions get the "RTSPClient" from the subsession
  scs.subsession->sink->startPlaying(*(scs.subsession->readSource()),
   subsessionAfterPlaying, scs.subsession);
  // Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
  if (scs.subsession->rtcpInstance() != NULL) {
   scs.subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, scs.subsession);
  }
 } while (0);
 delete[] resultString;

 // Set up the next subsession, if any:
 setupNextSubsession(rtspClient);
}

void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString) {
 Boolean success = False;

 do {
  UsageEnvironment& env = rtspClient->envir(); // alias
  StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

  if (resultCode != 0) {
   env << *rtspClient << "Failed to start playing session: " << resultString << "\n";
   break;
  }

  // Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
  // using a RTCP "BYE").  This is optional.  If, instead, you want to keep the stream active - e.g., so you can later
  // 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
  // (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)

  //延迟定时器
  if (scs.duration > 0) {
   unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration.  (This is optional.)
   scs.duration += delaySlop;
   unsigned uSecsToDelay = (unsigned)(scs.duration*1000000);
   scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
  }

  env << *rtspClient << "Started playing session";
  if (scs.duration > 0) {
   env << " (for up to " << scs.duration << " seconds)";
  }
  env << "...\n";

  success = True;
 } while (0);
 delete[] resultString;

 if (!success) {
  // An unrecoverable error occurred with this stream.
  shutdownStream(rtspClient);
 }
}


// Implementation of the other event handlers:

void subsessionAfterPlaying(void* clientData) {
 MediaSubsession* subsession = (MediaSubsession*)clientData;
 RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr);

 // Begin by closing this subsession's stream:
 Medium::close(subsession->sink);
 subsession->sink = NULL;

 // Next, check whether *all* subsessions' streams have now been closed:
 MediaSession& session = subsession->parentSession();
 MediaSubsessionIterator iter(session);
 while ((subsession = iter.next()) != NULL) {
  if (subsession->sink != NULL) return; // this subsession is still active
 }

 // All subsessions' streams have now been closed, so shutdown the client:
 shutdownStream(rtspClient);
}

void subsessionByeHandler(void* clientData) {
 MediaSubsession* subsession = (MediaSubsession*)clientData;
 RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
 UsageEnvironment& env = rtspClient->envir(); // alias

 env << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession\n";

 // Now act as if the subsession had closed:
 subsessionAfterPlaying(subsession);
}

void streamTimerHandler(void* clientData) {
 ourRTSPClient* rtspClient = (ourRTSPClient*)clientData;
 StreamClientState& scs = rtspClient->scs; // alias

 scs.streamTimerTask = NULL;

 // Shut down the stream:
 shutdownStream(rtspClient);
}

void shutdownStream(RTSPClient* rtspClient, int exitCode) {
 UsageEnvironment& env = rtspClient->envir(); // alias
 StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias

 // First, check whether any subsessions have still to be closed:
 if (scs.session != NULL) {
  Boolean someSubsessionsWereActive = False;
  MediaSubsessionIterator iter(*scs.session);
  MediaSubsession* subsession;

  while ((subsession = iter.next()) != NULL) {
   if (subsession->sink != NULL) {
    Medium::close(subsession->sink);
    subsession->sink = NULL;

    if (subsession->rtcpInstance() != NULL) {
     subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
    }

    someSubsessionsWereActive = True;
   }
  }

  if (someSubsessionsWereActive) {
   // Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
   // Don't bother handling the response to the "TEARDOWN".
   rtspClient->sendTeardownCommand(*scs.session, NULL);
  }
 }

 env << *rtspClient << "Closing the stream.\n";
 Medium::close(rtspClient);
 // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.

 if (--rtspClientCount == 0) {
  // The final stream has ended, so exit the application now.
  // (Of course, if you're embedding this code into your own application, you might want to comment this out,
  // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
  exit(exitCode);
 }
}


// Implementation of "ourRTSPClient":

ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, char const* rtspURL,
          int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum) {
           return new ourRTSPClient(env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum);
}

ourRTSPClient::ourRTSPClient(UsageEnvironment& env, char const* rtspURL,
        int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum)
        : RTSPClient(env,rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum, -1) {
}

ourRTSPClient::~ourRTSPClient() {
}


// Implementation of "StreamClientState":

StreamClientState::StreamClientState()
: iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0) {
}

StreamClientState::~StreamClientState() {
 delete iter;
 if (session != NULL) {
  // We also need to delete "session", and unschedule "streamTimerTask" (if set)
  UsageEnvironment& env = session->envir(); // alias

  env.taskScheduler().unscheduleDelayedTask(streamTimerTask);
  Medium::close(session);
 }
}


// Implementation of "DummySink":

// Even though we're not going to be doing anything with the incoming data, we still need to receive it.
// Define the size of the buffer that we'll use:
#define DUMMY_SINK_RECEIVE_BUFFER_SIZE 100000

DummySink* DummySink::createNew(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId) {
 return new DummySink(env, subsession, streamId);
}

DummySink::DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId)
: MediaSink(env),
fSubsession(subsession) {
 fStreamId = strDup(streamId);
 fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];
}

DummySink::~DummySink() {
 delete[] fReceiveBuffer;
 delete[] fStreamId;
}

void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned durationInMicroseconds) {
 DummySink* sink = (DummySink*)clientData;
 sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
}

// If you don't want to see debugging output for each received frame, then comment out the following line:
#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1

void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
 // We've just received a frame of data.  (Optionally) print out information about it:
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
 if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
 envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
 if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
 char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
 sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
 envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
 if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
  envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
 }
#ifdef DEBUG_PRINT_NPT
 envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
#endif
 envir() << "\n";
#endif

 // Then continue, to request the next frame of data:
 continuePlaying();
}

Boolean DummySink::continuePlaying() {
 if (fSource == NULL) return False; // sanity check (should not happen)

 // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:
 fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,
  afterGettingFrame, this,
  onSourceClosure, this);
 return True;
}

 

//int _tmain(int argc, _TCHAR* argv[])
//{
// 
// TaskScheduler *scheduler= BasicTaskScheduler::createNew();
//
// UsageEnvironment *env=BasicUsageEnvironment::createNew(*scheduler);
// RTSPServer *rtspServer=RTSPServer::createNew(*env,8554);
// *env<<"play streams from this server using the URL\n\t"
//  <<rtspServer->rtspURLPrefix()<<"<filename>.\n";
//
// ServerMediaSession *sms=ServerMediaSession::createNew(*env,"test.264");
//
// sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env,"test.264",true));
//
// rtspServer->addServerMediaSession(sms);
//
// env->taskScheduler().doEventLoop();    // does not return
// 
//
// return 0;
//}
//

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