demux_rtp.cpp

自己移植的linux下的流媒体播放器原代码,支持mms协议,支持ftp和http协议.

////////// Routines (with C-linkage) that interface between "MPlayer"
////////// and the "LIVE.COM Streaming Media" libraries:

extern "C" {
// on MinGW, we must include windows.h before the things it conflicts
#ifdef __MINGW32__    // with.  they are each protected from
#include <windows.h>  // windows.h, but not the other way around.
#endif
#include "demux_rtp.h"
#include "stheader.h"
}
#include "demux_rtp_internal.h"

#include "BasicUsageEnvironment.hh"
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include <unistd.h>

extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize,
				     int* file_format) {
  *file_format = DEMUXER_TYPE_RTP;
  stream_t* newStream = NULL;
  do {
    char* sdpDescription = (char*)malloc(fileSize+1);
    if (sdpDescription == NULL) break;

    ssize_t numBytesRead = read(fd, sdpDescription, fileSize);
    if (numBytesRead != fileSize) break;
    sdpDescription[fileSize] = '\0'; // to be safe

    newStream = (stream_t*)calloc(sizeof (stream_t), 1);
    if (newStream == NULL) break;

    // Store the SDP description in the 'priv' field, for later use:
    newStream->priv = sdpDescription; 
  } while (0);
  return newStream;
}

extern "C" int _rtsp_streaming_seek(int /*fd*/, off_t /*pos*/,
				    streaming_ctrl_t* /*streaming_ctrl*/) {
  return -1; // For now, we don't handle RTSP stream seeking
}

extern "C" int rtsp_streaming_start(stream_t* stream) {
  stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek;

  return 0;
}

// A data structure representing input data for each stream:
class ReadBufferQueue {
public:
  ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
		  char const* tag);
  virtual ~ReadBufferQueue();

  FramedSource* readSource() const { return fReadSource; }
  RTPSource* rtpSource() const { return fRTPSource; }
  demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  char const* tag() const { return fTag; }

  char blockingFlag; // used to implement synchronous reads

  // For A/V synchronization:
  Boolean prevPacketWasSynchronized;
  float prevPacketPTS;
  ReadBufferQueue** otherQueue;

  // The 'queue' actually consists of just a single "demux_packet_t"
  // (because the underlying OS does the actual queueing/buffering):
  demux_packet_t* dp;

  // However, we sometimes inspect buffers before delivering them.
  // For this, we maintain a queue of pending buffers:
  void savePendingBuffer(demux_packet_t* dp);
  demux_packet_t* getPendingBuffer();

private:
  demux_packet_t* pendingDPHead;
  demux_packet_t* pendingDPTail;

  FramedSource* fReadSource;
  RTPSource* fRTPSource;
  demuxer_t* fOurDemuxer;
  char const* fTag; // used for debugging
};

// A structure of RTP-specific state, kept so that we can cleanly
// reclaim it:
typedef struct RTPState {
  char const* sdpDescription;
  RTSPClient* rtspClient;
  SIPClient* sipClient;
  MediaSession* mediaSession;
  ReadBufferQueue* audioBufferQueue;
  ReadBufferQueue* videoBufferQueue;
  unsigned flags;
  struct timeval firstSyncTime;
};

extern "C" char* network_username;
extern "C" char* network_password;
static char* openURL_rtsp(RTSPClient* client, char const* url) {
  // If we were given a user name (and optional password), then use them: 
  if (network_username != NULL) {
    char const* password = network_password == NULL ? "" : network_password;
    return client->describeWithPassword(url, network_username, password);
  } else {
    return client->describeURL(url);
  }
}

static char* openURL_sip(SIPClient* client, char const* url) {
  // If we were given a user name (and optional password), then use them: 
  if (network_username != NULL) {
    char const* password = network_password == NULL ? "" : network_password;
    return client->inviteWithPassword(url, network_username, password);
  } else {
    return client->invite(url);
  }
}

int rtspStreamOverTCP = 0; 

extern "C" int audio_id, video_id, dvdsub_id;
extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  Boolean success = False;
  do {
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    if (scheduler == NULL) break;
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    if (env == NULL) break;

    RTSPClient* rtspClient = NULL;
    SIPClient* sipClient = NULL;

    if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
    demuxer->stream->eof = 0; // just in case 

    // Look at the stream's 'priv' field to see if we were initiated
    // via a SDP description:
    char* sdpDescription = (char*)(demuxer->stream->priv);
    if (sdpDescription == NULL) {
      // We weren't given a SDP description directly, so assume that
      // we were given a RTSP or SIP URL:
      char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
      char const* url = demuxer->stream->streaming_ctrl->url->url;
      extern int verbose;
      if (strcmp(protocol, "rtsp") == 0) {
	rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
	if (rtspClient == NULL) {
	  fprintf(stderr, "Failed to create RTSP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sdpDescription = openURL_rtsp(rtspClient, url);
      } else { // SIP
	unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
	sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
					 verbose, "MPlayer");
	if (sipClient == NULL) {
	  fprintf(stderr, "Failed to create SIP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sipClient->setClientStartPortNum(8000);
	sdpDescription = openURL_sip(sipClient, url);
      }

      if (sdpDescription == NULL) {
	fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
		url, env->getResultMsg());
	break;
      }
    }

    // Now that we have a SDP description, create a MediaSession from it:
    MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
    if (mediaSession == NULL) break;


    // Create a 'RTPState' structure containing the state that we just created,
    // and store it in the demuxer's 'priv' field, for future reference:
    RTPState* rtpState = new RTPState;
    rtpState->sdpDescription = sdpDescription;
    rtpState->rtspClient = rtspClient;
    rtpState->sipClient = sipClient;
    rtpState->mediaSession = mediaSession;
    rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
    rtpState->flags = 0;
    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
    demuxer->priv = rtpState;

    // Create RTP receivers (sources) for each subsession:
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;
    unsigned desiredReceiveBufferSize;
    while ((subsession = iter.next()) != NULL) {
      // Ignore any subsession that's not audio or video:
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	desiredReceiveBufferSize = 100000;
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	desiredReceiveBufferSize = 2000000;
      } else {
	continue;
      }

      if (!subsession->initiate()) {
	fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
      } else {
	fprintf(stderr, "Initiated \"%s/%s\" RTP subsession\n", subsession->mediumName(), subsession->codecName());

	// Set the OS's socket receive buffer sufficiently large to avoid
	// incoming packets getting dropped between successive reads from this
	// subsession's demuxer.  Depending on the bitrate(s) that you expect,
	// you may wish to tweak the "desiredReceiveBufferSize" values above.
	int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
	int receiveBufferSize
	  = increaseReceiveBufferTo(*env, rtpSocketNum,
				    desiredReceiveBufferSize);
	if (verbose > 0) {
	  fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
		  subsession->mediumName(), receiveBufferSize);
	}

	if (rtspClient != NULL) {
	  // Issue a RTSP "SETUP" command on the chosen subsession:
	  if (!rtspClient->setupMediaSubsession(*subsession, False,
						rtspStreamOverTCP)) break;
	}
      }
    }

    if (rtspClient != NULL) {
      // Issue a RTSP aggregate "PLAY" command on the whole session:
      if (!rtspClient->playMediaSession(*mediaSession)) break;
    } else if (sipClient != NULL) {
      sipClient->sendACK(); // to start the stream flowing
    }

    // Now that the session is ready to be read, do additional
    // MPlayer codec-specific initialization on each subsession:
    iter.reset();
    while ((subsession = iter.next()) != NULL) {
      if (subsession->readSource() == NULL) continue; // not reading this

      unsigned flags = 0;
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	rtpState->audioBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "audio");
	rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
	rtpCodecInitialize_audio(demuxer, subsession, flags);
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	rtpState->videoBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "video");
	rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
	rtpCodecInitialize_video(demuxer, subsession, flags);
      }
      rtpState->flags |= flags;
    }
    success = True;
  } while (0);
  if (!success) return NULL; // an error occurred

  // Hack: If audio and video are demuxed together on a single RTP stream,
  // then create a new "demuxer_t" structure to allow the higher-level
  // code to recognize this:
  if (demux_is_multiplexed_rtp_stream(demuxer)) {
    stream_t* s = new_ds_stream(demuxer->video);
    demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
			       audio_id, video_id, dvdsub_id, NULL);
    demuxer = new_demuxers_demuxer(od, od, od);
  }

  return demuxer;
}

extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
}

extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
}

static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
				 Boolean mustGetNewData,
				 float& ptsBehind); // forward

extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
  // Get a filled-in "demux_packet" from the RTP source, and deliver it.
  // Note that this is called as a synchronous read operation, so it needs
  // to block in the (hopefully infrequent) case where no packet is
  // immediately available.

  while (1) {
    float ptsBehind;
    demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
    if (dp == NULL) return 0;