comparison libmpdemux/demux_rtp.cpp @ 10054:1342075f81f0

Added SIP (IP telephony) client support. (This was already supported in the LIVE.COM libraries, so updating the MPlayer code to support it required only relatively minor changes.)

10053:8e84e8deb529

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

extern "C" char* network_username;
extern "C" char* network_password;
int rtspStreamOverTCP = 0; 

extern "C" void demux_open_rtp(demuxer_t* demuxer) {
  do {
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    if (scheduler == NULL) break;
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    if (env == NULL) break;

    RTSPClient* rtspClient = 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 URL:
      char const* url = demuxer->stream->streaming_ctrl->url->url;

      extern int verbose;
      rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
      if (rtspClient == NULL) {
	fprintf(stderr, "Failed to create RTSP client: %s\n",
		env->getResultMsg());
	break;
      }

      // 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;
	sdpDescription
	  = rtspClient->describeWithPassword(url, network_username, password);
      } else {
	sdpDescription = rtspClient->describeURL(url);
      }
      if (sdpDescription == NULL) {
	fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
		url, env->getResultMsg());
	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->mediaSession = mediaSession;
    rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
    rtpState->flags = 0;
    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
    demuxer->priv = rtpState;

    }

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

    // Now that the session is ready to be read, do additional
    // MPlayer codec-specific initialization on each subsession:
    iter.reset();

  dp->pts = 0;
  bufferQueue->savePendingBuffer(dp);
  return True;
}

static void teardownRTSPSession(RTPState* rtpState); // forward

extern "C" void demux_close_rtp(demuxer_t* demuxer) {
  // Reclaim all RTP-related state:

  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  if (rtpState == NULL) return;

  teardownRTSPSession(rtpState);

  UsageEnvironment* env = NULL;
  TaskScheduler* scheduler = NULL;
  if (rtpState->mediaSession != NULL) {
    env = &(rtpState->mediaSession->envir());
    scheduler = &(env->taskScheduler());
  }
  Medium::close(rtpState->mediaSession);
  Medium::close(rtpState->rtspClient);
  delete rtpState->audioBufferQueue;
  delete rtpState->videoBufferQueue;
  delete rtpState->sdpDescription;
  delete rtpState;

  }

  return dp;
}

static void teardownRTSPSession(RTPState* rtpState) {
  RTSPClient* rtspClient = rtpState->rtspClient;
  MediaSession* mediaSession = rtpState->mediaSession;
  if (rtspClient == NULL || mediaSession == NULL) return;

  MediaSubsessionIterator iter(*mediaSession);
  MediaSubsession* subsession;

  while ((subsession = iter.next()) != NULL) {
    rtspClient->teardownMediaSubsession(*subsession);
  }
}

////////// "ReadBuffer" and "ReadBufferQueue" implementation:

10054:1342075f81f0

// 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" void demux_open_rtp(demuxer_t* demuxer) {
  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;
      }

    // 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;

    }

    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();

  dp->pts = 0;
  bufferQueue->savePendingBuffer(dp);
  return True;
}

static void teardownRTSPorSIPSession(RTPState* rtpState); // forward

extern "C" void demux_close_rtp(demuxer_t* demuxer) {
  // Reclaim all RTP-related state:

  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  if (rtpState == NULL) return;

  teardownRTSPorSIPSession(rtpState);

  UsageEnvironment* env = NULL;
  TaskScheduler* scheduler = NULL;
  if (rtpState->mediaSession != NULL) {
    env = &(rtpState->mediaSession->envir());
    scheduler = &(env->taskScheduler());
  }
  Medium::close(rtpState->mediaSession);
  Medium::close(rtpState->rtspClient);
  Medium::close(rtpState->sipClient);
  delete rtpState->audioBufferQueue;
  delete rtpState->videoBufferQueue;
  delete rtpState->sdpDescription;
  delete rtpState;

  }

  return dp;
}

static void teardownRTSPorSIPSession(RTPState* rtpState) {
  MediaSession* mediaSession = rtpState->mediaSession;
  if (mediaSession == NULL) return;
  if (rtpState->rtspClient != NULL) {
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;

    while ((subsession = iter.next()) != NULL) {
      rtpState->rtspClient->teardownMediaSubsession(*subsession);
    }
  } else if (rtpState->sipClient != NULL) {
    rtpState->sipClient->sendBYE();
  }
}

////////// "ReadBuffer" and "ReadBufferQueue" implementation: