playcommon.cpp

流媒体传输协议的实现代码,非常有用.可以支持rtsp mms等流媒体传输协议

/**********
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.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
// Copyright (c) 1996-2004, Live Networks, Inc.  All rights reserved
// A common framework, used for the "openRTSP" and "playSIP" applications
// Implementation

#include "playCommon.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#ifdef SUPPORT_REAL_RTSP
#include "../RealRTSP/include/RealRTSP.hh"
#endif

#if defined(__WIN32__) || defined(_WIN32)
#define snprintf _snprintf
#else
#include <signal.h>
#define USE_SIGNALS 1
#endif

// Forward function definitions:
void setupStreams();
void startPlayingStreams();
void tearDownStreams();
void closeMediaSinks();
void subsessionAfterPlaying(void* clientData);
void subsessionByeHandler(void* clientData);
void sessionAfterPlaying(void* clientData = NULL);
void sessionTimerHandler(void* clientData);
void shutdown(int exitCode = 1);
void signalHandlerShutdown(int sig);
void checkForPacketArrival(void* clientData);
void checkInterPacketGaps(void* clientData);
void beginQOSMeasurement();
Boolean setupDestinationRTSPServer();

char const* progName;
UsageEnvironment* env;
Medium* ourClient = NULL;
MediaSession* session = NULL;
TaskToken sessionTimerTask = NULL;
TaskToken arrivalCheckTimerTask = NULL;
TaskToken interPacketGapCheckTimerTask = NULL;
TaskToken qosMeasurementTimerTask = NULL;
Boolean createReceivers = True;
Boolean outputQuickTimeFile = False;
Boolean generateMP4Format = False;
QuickTimeFileSink* qtOut = NULL;
Boolean outputAVIFile = False;
AVIFileSink* aviOut = NULL;
Boolean audioOnly = False;
Boolean videoOnly = False;
char const* singleMedium = NULL;
int verbosityLevel = 0;
double endTime = 0;
double endTimeSlop = -1.0; // extra seconds to play at the end
unsigned interPacketGapMaxTime = 0;
unsigned totNumPacketsReceived = ~0; // used if checking inter-packet gaps
Boolean playContinuously = False;
int simpleRTPoffsetArg = -1;
Boolean sendOptionsRequest = True;
Boolean sendOptionsRequestOnly = False;
Boolean oneFilePerFrame = False;
Boolean notifyOnPacketArrival = False;
Boolean streamUsingTCP = False;
portNumBits tunnelOverHTTPPortNum = 0;
char* username = NULL;
char* password = NULL;
char* proxyServerName = NULL;
unsigned short proxyServerPortNum = 0;
unsigned char desiredAudioRTPPayloadFormat = 0;
char* mimeSubtype = NULL;
unsigned short movieWidth = 240; // default
Boolean movieWidthOptionSet = False;
unsigned short movieHeight = 180; // default
Boolean movieHeightOptionSet = False;
unsigned movieFPS = 15; // default
Boolean movieFPSOptionSet = False;
char* fileNamePrefix = "";
unsigned fileSinkBufferSize = 20000;
unsigned socketInputBufferSize = 0;
Boolean packetLossCompensate = False;
Boolean syncStreams = False;
Boolean generateHintTracks = False;
char* destRTSPURL = NULL;
unsigned qosMeasurementIntervalMS = 0; // 0 means: Don't output QOS data
unsigned statusCode = 0;

struct timeval startTime;

void usage() {
  *env << "Usage: " << progName
       << " [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-e <endTime>] [-E <max-inter-packet-gap-time> [-c] [-s <offset>] [-n] [-O]"
	   << (controlConnectionUsesTCP ? " [-t|-T <http-port>]" : "")
       << " [-u <username> <password>"
	   << (allowProxyServers ? " [<proxy-server> [<proxy-server-port>]]" : "")
       << "]" << (supportCodecSelection ? " [-A <audio-codec-rtp-payload-format-code>|-D <mime-subtype-name>]" : "")
       << " [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F <filename-prefix>] [-b <file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] <url> (or " << progName << " -o [-V] <url>)\n";
  //##### Add "-R <dest-rtsp-url>" #####
  shutdown();
}

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

  progName = argv[0];

  gettimeofday(&startTime, NULL);

#ifdef USE_SIGNALS
  // Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
  signal(SIGHUP, signalHandlerShutdown);
  signal(SIGUSR1, signalHandlerShutdown);
#endif

  unsigned short desiredPortNum = 0;

  // unfortunately we can't use getopt() here, as Windoze doesn't have it
  while (argc > 2) {
    char* const opt = argv[1];
    if (opt[0] != '-') usage();
    switch (opt[1]) {

    case 'p': { // specify start port number
      int portArg;
      if (sscanf(argv[2], "%d", &portArg) != 1) {
	usage();
      }
      if (portArg <= 0 || portArg >= 65536 || portArg&1) {
	*env << "bad port number: " << portArg
		<< " (must be even, and in the range (0,65536))\n";
	usage();
      }
      desiredPortNum = (unsigned short)portArg;
      ++argv; --argc;
      break;
    }

    case 'r': { // do not receive data (instead, just 'play' the stream(s))
      createReceivers = False;
      break;
    }

    case 'q': { // output a QuickTime file (to stdout)
      outputQuickTimeFile = True;
      break;
    }

    case '4': { // output a 'mp4'-format file (to stdout)
      outputQuickTimeFile = True;
      generateMP4Format = True;
      break;
    }

    case 'i': { // output an AVI file (to stdout)
      outputAVIFile = True;
      break;
    }

    case 'I': { // specify input interface... 
      NetAddressList addresses(argv[2]);
      if (addresses.numAddresses() == 0) {
	*env << "Failed to find network address for \"" << argv[2] << "\"";
	break;
      }
      ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
      ++argv; --argc;
      break;
    }

    case 'a': { // receive/record an audio stream only
      audioOnly = True;
      singleMedium = "audio";
      break;
    }

    case 'v': { // receive/record a video stream only
      videoOnly = True;
      singleMedium = "video";
      break;
    }

    case 'V': { // verbose output
      verbosityLevel = 1;
      break;
    }

    case 'e': { // specify end time, or how much to delay after end time
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1) {
	usage();
      }
      if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
	// a 'negative' argument was specified; use this for "endTimeSlop":
	endTime = 0; // use whatever's in the SDP
	endTimeSlop = -arg;
      } else {
	endTime = arg;
	endTimeSlop = 0;
      }
      ++argv; --argc;
      break;
    }

    case 'E': { // specify maximum number of seconds to wait for packets:
      if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'c': { // play continuously
      playContinuously = True;
      break;
    }

    case 's': { // specify an offset to use with "SimpleRTPSource"s
      if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
	usage();
      }
      if (simpleRTPoffsetArg < 0) {
	*env << "offset argument to \"-s\" must be >= 0\n";
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
      sendOptionsRequest = False;
      break;
    }

    case 'o': { // Send only the "OPTIONS" request to the server
      sendOptionsRequestOnly = True;
      break;
    }

    case 'm': { // output multiple files - one for each frame
      oneFilePerFrame = True;
      break;
    }

    case 'n': { // notify the user when the first data packet arrives
      notifyOnPacketArrival = True;
      break;
    }

    case 't': {
      // stream RTP and RTCP over the TCP 'control' connection
      if (controlConnectionUsesTCP) {
	streamUsingTCP = True;
      } else {
	usage();
      }
      break;
    }

    case 'T': {
      // stream RTP and RTCP over a HTTP connection
      if (controlConnectionUsesTCP) {
	if (argc > 3 && argv[2][0] != '-') {
	  // The next argument is the HTTP server port number:
	  if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
	      && tunnelOverHTTPPortNum > 0) {
	    ++argv; --argc;
	    break;
	  }
	}
      }

      // If we get here, the option was specified incorrectly:
      usage();
      break;
    }

    case 'u': { // specify a username and password
      username = argv[2];
      password = argv[3];
      argv+=2; argc-=2;
      if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
	// The next argument is the name of a proxy server:
	proxyServerName = argv[2];
	++argv; --argc;

	if (argc > 3 && argv[2][0] != '-') {
	  // The next argument is the proxy server port number:
	  if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
	    usage();
	  }
	  ++argv; --argc;
	}
      }
      break;
    }

    case 'A': { // specify a desired audio RTP payload format
      unsigned formatArg;
      if (sscanf(argv[2], "%u", &formatArg) != 1
	  || formatArg >= 96) {
	usage();
      }
      desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
      ++argv; --argc;
      break;
    }

    case 'D': { // specify a MIME subtype for a dynamic RTP payload type
      mimeSubtype = argv[2];
      if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat =96;
      ++argv; --argc;
      break;
    }

    case 'w': { // specify a width (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
	usage();
      }
      movieWidthOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'h': { // specify a height (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
	usage();
      }
      movieHeightOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'f': { // specify a frame rate (per second) for an output QT or AVI movie
      if (sscanf(argv[2], "%u", &movieFPS) != 1) {
	usage();
      }
      movieFPSOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'F': { // specify a prefix for the audio and video output files
      fileNamePrefix = argv[2];
      ++argv; --argc;
      break;
    }

    case 'b': { // specify the size of buffers for "FileSink"s
      if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    case 'B': { // specify the size of input socket buffers