rtp_transmitter.cpp

来自「MPEG4音频视频压缩编码(含G.711/ACC/H.261等)」· C++ 代码 · 共 802 行 · 第 1/2 页

		// Check and see if we should send the queue first
		if (m_audioQueueCount > 0 && (check_frame & SEND_FIRST)) {

			// send anything that's pending
			SendQueuedAudioFrames();
		}

		// See if we have a jumbo frame
		if (check_frame & IS_JUMBO) {
			SendAudioJumboFrame(pFrame);
			return;
		}

		// Add frame to queue
		m_audioQueue[m_audioQueueCount++] = pFrame;
		m_audioQueueSize += pFrame->GetDataLength();

		// and check to see if we want ot send it now
		if (check_frame & SEND_NOW) {

			// send anything that's pending
			SendQueuedAudioFrames();
		}

	} else {

		// B & D 's original CRtpTransmitter::SendAudioFrame
		OldSendAudioFrame(pFrame);
	}
}

void CAudioRtpTransmitter::SendQueuedAudioFrames(void)
{
	struct iovec iov[m_audioiovMaxCount + 1];
	int ix;
	int i = 0;
	CRtpDestination *rdptr;

	if (m_audioQueueCount == 0) {
		return;
	}

	u_int32_t rtpTimestamp =
		TimestampToRtp(m_audioQueue[0]->GetTimestamp());
	u_int64_t ntpTimestamp =
		TimestampToNtp(m_audioQueue[0]->GetTimestamp());

#ifdef RTP_DEBUG
	debug_message("A ts "U64" rtp %u ntp %u.%u",
		m_audioQueue[0]->GetTimestamp(),
		rtpTimestamp,
		(u_int32_t)(ntpTimestamp >> 32),
		(u_int32_t)ntpTimestamp);
#endif
	rdptr = m_rtpDestination;
	while (rdptr != NULL) {
	  rdptr->send_rtcp(rtpTimestamp, ntpTimestamp);
	  rdptr = rdptr->get_next();
	}

	// moved this from below
	// See if we need to add the header.
	int iov_start = 1;
	int iov_add = 0;
	bool mbit = 1;
	if (m_audio_set_rtp_payload != NULL) {
		iov_start = 0;
		iov_add = m_audio_set_rtp_payload(m_audioQueue,
						  m_audioQueueCount,
						  iov,
						  m_audio_rtp_userdata,
						  &mbit);
	} else {
		// audio payload data
		for (i = 0; i < m_audioQueueCount; i++) {
	  	iov[i + 1].iov_base = m_audioQueue[i]->GetData();
	  	iov[i + 1].iov_len = m_audioQueue[i]->GetDataLength();
		}

		if (m_audio_set_rtp_header != NULL) {
	  		if ((m_audio_set_rtp_header)(iov, 
						     m_audioQueueCount, 
						     m_audio_rtp_userdata, 
						     &mbit)) {
	    			iov_start = 0;
	    			iov_add = 1;
	  		} 
		}
	}
		
	// send packet
	rdptr = m_rtpDestination;
	while (rdptr != NULL) {
	  rdptr->send_iov(&iov[iov_start],
			  iov_add + m_audioQueueCount,
			  rtpTimestamp,
			  mbit ? 1 : 0);
	  rdptr = rdptr->get_next();
	}

	// delete all the pending media frames
	for (ix = 0; ix < m_audioQueueCount; ix++) {
	  if (m_audioQueue[ix]->RemoveReference())
	    delete m_audioQueue[ix];
	  m_audioQueue[ix] = NULL;
	}

	m_audioQueueCount = 0;
	m_audioQueueSize = 0;
}

void CAudioRtpTransmitter::SendAudioJumboFrame(CMediaFrame* pFrame)
{
  u_int32_t rtpTimestamp = 
    TimestampToRtp(pFrame->GetTimestamp());
  u_int64_t ntpTimestamp =
    TimestampToNtp(pFrame->GetTimestamp());
  CRtpDestination *rdptr;

#if RTP_DEBUG
  debug_message("A ts "U64" rtp %u ntp %u.%u",
		pFrame->GetTimestamp(),
		rtpTimestamp,
		(u_int32_t)(ntpTimestamp >> 32),
		(u_int32_t)ntpTimestamp);
#endif
  rdptr = m_rtpDestination;
  while (rdptr != NULL) {
    rdptr->send_rtcp(rtpTimestamp, ntpTimestamp);
    rdptr = rdptr->get_next();
  }

  u_int32_t dataOffset = 0;
  u_int32_t spaceAvailable = m_mtu;


  struct iovec iov[2];
  do {
    iov[1].iov_base = (u_int8_t*)pFrame->GetData() + dataOffset;
    int iov_start = 1, iov_hdr = 0;
    bool mbit;
    if (m_audio_set_rtp_jumbo_frame(iov,
				    dataOffset, 
				    pFrame->GetDataLength(),
				    spaceAvailable,
				    mbit,
				    m_audio_rtp_userdata)) {
      iov_start = 0;
      iov_hdr = 1;
    }
    // send packet
    rdptr = m_rtpDestination;
    while (rdptr != NULL) {
      rdptr->send_iov(&iov[iov_start],
		      1 + iov_hdr,
		      rtpTimestamp,
		      mbit);
      rdptr = rdptr->get_next();
    }
    error_message("data offset %d len %d max %d", dataOffset, 
		  (int)iov[1].iov_len,
		  pFrame->GetDataLength());
    dataOffset += iov[1].iov_len;
  } while (dataOffset < pFrame->GetDataLength());

  if (pFrame->RemoveReference())
    delete pFrame;
}




void CRtpTransmitter::DoStartRtpDestination (const char *destAddr,
					     in_port_t destPort)
{
  CRtpDestination *ptr;

  ptr = m_rtpDestination;
  while (ptr != NULL) {
    if (ptr->Matches(destAddr, destPort)) {
      ptr->start();
      return;
    }
    ptr = ptr->get_next();
  }
}

void CRtpTransmitter::DoStopRtpDestination (const char *destAddr, 
					    in_port_t destPort)
{
  CRtpDestination *ptr, *q;

  q = NULL;
  ptr = m_rtpDestination;
  while (ptr != NULL) {
    if (ptr->Matches(destAddr, destPort)) {
      if (ptr->remove_reference()) {
	if (q == NULL) {
	  m_rtpDestination = ptr->get_next();
	} else {
	  q->set_next(ptr->get_next());
	}
	delete ptr;
      }
      return;
    }
    q = ptr;
    ptr = ptr->get_next();
  }
}

CAudioRtpTransmitter::CAudioRtpTransmitter (CAudioProfile *ap,
					    uint16_t mtu,
					    bool disable_ts_offset) :
  CRtpTransmitter(mtu)
{
  m_audio_profile = ap;
  m_payloadNumber = 97;
  m_audioQueue = NULL;
  m_audio_rtp_userdata = NULL;
  m_audio_set_rtp_payload = NULL;	// plugin function to determind how to build RTP payload
  m_audio_queue_frame = NULL;		// plugin function to determind how to queue frames for RTP
  m_frameno = NULL;			// plugin will allocate counter space
  m_audioiovMaxCount = 0;		// Max number of iov segments for send_iov
	  
  /*  if (m_audioSrcPort == 0) {
    m_audioSrcPort = GetRandomPortBlock();
    }
    if (m_pConfig->GetIntegerValue(CONFIG_RTP_AUDIO_DEST_PORT) == 0) {
      m_pConfig->SetIntegerValue(CONFIG_RTP_AUDIO_DEST_PORT,
				 m_audioSrcPort + 2);
    }
  */
  if (get_audio_rtp_info(ap,
			 &m_frameType,
			 &m_timeScale,
			 &m_payloadNumber, 
			 &m_audioPayloadBytesPerPacket,
			 &m_audioPayloadBytesPerFrame,
			 &m_audioQueueMaxCount,
			 &m_audioiovMaxCount,
			 &m_audio_queue_frame,
			 &m_audio_set_rtp_payload,
			 &m_audio_set_rtp_header,
			 &m_audio_set_rtp_jumbo_frame,
			 &m_audio_rtp_userdata) == false) {
      error_message("rtp transmitter: unknown audio encoding %s",
		    ap->GetStringValue(CFG_AUDIO_ENCODING));
    }

  uint8_t max_queue_from_config = 
    ap->GetIntegerValue(CFG_RTP_MAX_FRAMES_PER_PACKET);
  if (max_queue_from_config != 0 &&
      max_queue_from_config < m_audioQueueMaxCount) {
    m_audioQueueMaxCount = max_queue_from_config;
  }
    if (m_audioiovMaxCount == 0)			// This is purely for backwards compability
      m_audioiovMaxCount = m_audioQueueMaxCount;	// Can go away when lame and faac plugin sets this

#ifdef DEBUG_WRAP_TS
  m_audioRtpTimestampOffset = 0xffff0000;
#else 
    if (disable_ts_offset) {
      m_audioRtpTimestampOffset = 0;
    } else {
      m_audioRtpTimestampOffset = random();
    }
#endif

    m_audioQueueCount = 0;
    m_audioQueueSize = 0;
    
    m_audioQueue = (CMediaFrame**)Malloc(m_audioQueueMaxCount * sizeof(CMediaFrame*));
}

CAudioRtpTransmitter::~CAudioRtpTransmitter (void) 
{
  DoStopTransmit();
  CHECK_AND_FREE(m_audio_rtp_userdata);
  CHECK_AND_FREE(m_frameno);
  CHECK_AND_FREE(m_audioQueue);
}

CVideoRtpTransmitter::CVideoRtpTransmitter (CVideoProfile *vp, 
					    uint16_t mtu,
					    bool disable_ts_offset) :
  CRtpTransmitter(mtu, disable_ts_offset)
{
  m_videoSendFunc = GetVideoRtpTransmitRoutine(vp,
					       &m_frameType, 
					       &m_payloadNumber);
  m_timeScale = 90000;

}

CTextRtpTransmitter::CTextRtpTransmitter (CTextProfile *tp,
					    uint16_t mtu,
					    bool disable_ts_offset) :
  CRtpTransmitter(mtu, disable_ts_offset)
{
  m_textSendFunc = GetTextRtpTransmitRoutine(tp,
					     &m_frameType, 
					     &m_payloadNumber);
  m_timeScale = 90000;
}

static void RtpCallback (struct rtp *session, rtp_event *e) 
{
  // Currently we ignore RTCP packets
  // Just do our required housekeeping
  if (e && (e->type == RX_RTP || e->type == RX_APP)) {
    free(e->data);
  }
}

CRtpDestination::CRtpDestination (const char *destAddr, 
				  in_port_t destPort,
				  in_port_t srcPort,
				  uint8_t payloadNumber,
				  int mcast_ttl,
				  float rtcp_bandwidth)
{

  m_destAddr = strdup(destAddr);
  m_destPort = destPort;
  m_srcPort = srcPort;
  m_payloadNumber = payloadNumber;
  m_mcast_ttl = mcast_ttl;
  m_rtcp_bandwidth = rtcp_bandwidth;
  m_rtpSession = NULL;
  m_next = NULL;
  m_ref_mutex = SDL_CreateMutex();
  m_reference = 1;
}

CRtpDestination::~CRtpDestination (void)
{
  CHECK_AND_FREE(m_destAddr);
  if (m_rtpSession != NULL) {
    rtp_done(m_rtpSession);
    m_rtpSession = NULL;
  }
  SDL_DestroyMutex(m_ref_mutex);
}


void CRtpDestination::start (void) 
{

  while (m_rtpSession == NULL) {
    debug_message("Starting rtp dest %s %d %d", 
		  m_destAddr, m_destPort, m_srcPort);
    if (m_srcPort == 0) m_srcPort = m_destPort;
    m_rtpSession = rtp_init_xmitter(m_destAddr,
				    m_srcPort,
				    m_destPort,
				    m_mcast_ttl, 
				    m_rtcp_bandwidth, 
				    RtpCallback, (uint8_t *)this);
    if (m_rtpSession == NULL) {
      error_message("Couldn't start rtp session");
      SDL_Delay(10);
    }
  }
}
			  
			  
void CRtpDestination::send_rtcp (u_int32_t rtpTimestamp,
				 u_int64_t ntpTimestamp)
{
  if (m_rtpSession != NULL) {
    rtp_send_ctrl_2(m_rtpSession, rtpTimestamp, ntpTimestamp, NULL);
    rtp_update(m_rtpSession);
  }
}

int CRtpDestination::send_iov (struct iovec *piov,
			       int iovCount,
			       u_int32_t rtpTimestamp,
			       int mbit)
{
  if (m_rtpSession != NULL) {
    int ret = rtp_send_data_iov(m_rtpSession,
			     rtpTimestamp,
			     m_payloadNumber,
			     mbit, 
			     0, 
			     NULL, 
			     piov, 
			     iovCount,
			     NULL, 
			     0, 
			     0,
			     0);
    //debug_message("sending iov %d", ret);
    return ret;
    
  }
  return -1;
}