rtp_bytestream.cpp

网络MPEG4IP流媒体开发源代码

   * We need to make sure we have some buffering.  We'll buffer
   * about 2 seconds worth, then let the decode task know to go...
   */
  if (m_buffering == 0) {
    uint32_t head_ts, tail_ts;
    if (m_head != NULL) {
      /*
       * Payload type the same.  Make sure we have at least 2 seconds of
       * good data
       */
      if (rtp_ready() == 0) {
	rtp_message(LOG_DEBUG, 
		    "Determined payload type, but rtp bytestream is not ready");
	uint64_t calc;
	do {
	  head_ts = m_head->rtp_pak_ts;
	  tail_ts = m_tail->rtp_pak_ts;
	  calc = (tail_ts - head_ts);
	  calc *= 1000;
	  calc /= m_timescale;
	  if (calc > m_rtp_buffer_time) {
	    rtp_packet *temp = m_head;
	    m_head = m_head->rtp_next;
	    m_tail->rtp_next = m_head;
	    m_head->rtp_prev = m_tail;
	    xfree((void *)temp);
	  }
	} while (calc > m_rtp_buffer_time);
	return 0;
      }
      if (check_rtp_frame_complete_for_payload_type()) {
	head_ts = m_head->rtp_pak_ts;
	tail_ts = m_tail->rtp_pak_ts;
	if (head_ts > tail_ts &&
	    ((head_ts & (1 << 31)) == (tail_ts & (1 << 31)))) {
	  return 0;
	}
	uint64_t calc;
	calc = tail_ts;
	calc -= head_ts;
	calc *= M_LLU;
	calc /= m_timescale;
	if (calc > m_rtp_buffer_time) {
	  if (m_base_ts_set == false) {
	    rtp_message(LOG_NOTICE, "Setting rtp seq and time from 1st pak");
	    set_rtp_base_ts(m_head->rtp_pak_ts);
	    m_rtpinfo_set_from_pak = 1;
	  } else {
	    m_rtpinfo_set_from_pak = 0;
	  }
	  m_buffering = 1;
#if 1
	  rtp_message(LOG_INFO, 
		      "%s buffering complete - seq %d head %u tail %u "LLD, 
		      m_name, m_head->rtp_pak_seq,
		      head_ts, tail_ts, calc);
#endif
	  rtp_done_buffering();
	  
	}
      }
    }

  } else {
    if (decode_thread_waiting != 0) {
      /*
       * We're good with buffering - but the decode thread might have
       * caught up, and will be waiting.  Post a message to kickstart
       * it
       */
      if (check_rtp_frame_complete_for_payload_type()) {
	return (1);
      }
    }
    if (m_recvd_pak == 0) {
      if (m_recvd_pak_timeout == 0) {
	m_recvd_pak_timeout_time = get_time_of_day();
#if 0
	rtp_message(LOG_DEBUG, "%s Starting timeout at %llu", 
		    m_name, m_recvd_pak_timeout_time);
#endif
      } else {
	uint64_t timeout;
	timeout = get_time_of_day() - m_recvd_pak_timeout_time;
	if (m_stream_ondemand && get_max_playtime() != 0.0) {
	  uint64_t range_end = (uint64_t)(get_max_playtime() * 1000.0);
	  if (m_last_realtime + timeout >= range_end) {
	    rtp_message(LOG_DEBUG, 
			"%s Timedout at range end - last "LLU" range end "LLU" "LLU, 
			m_name, m_last_realtime, range_end, timeout);
	    m_eof = 1;
	  }
	} else {
	  // broadcast - perhaps if we time out for a second or 2, we
	  // should re-init rtp ?  We definately need to put some timing
	  // checks here.
	  session_desc_t *sptr = m_fmt->media->parent;
	  if (sptr->time_desc != NULL &&
	      sptr->time_desc->end_time != 0) {
	    time_t this_time;
	    this_time = time(NULL);
	    if (this_time > sptr->time_desc->end_time && 
		timeout >= M_LLU) {
	      m_eof = 1;
	    }
	  }
	  
	}
	
      }
      m_recvd_pak_timeout++;
    } else {
      m_recvd_pak = 0;
      m_recvd_pak_timeout = 0;
    }
  }
  return (m_buffering);
}

int CRtpByteStreamBase::check_rtp_frame_complete_for_payload_type (void)
{
  return (m_head && m_tail->rtp_pak_m == 1);
}

uint64_t CRtpByteStreamBase::rtp_ts_to_msec (uint32_t rtp_ts,
					     uint64_t uts,
					     uint64_t &wrap_offset)
{
  uint64_t timetick;
  uint64_t adjusted_rtp_ts;
  uint64_t adjusted_wc_rtp_ts;
  bool have_wrap = false;

  if (((m_ts & 0x80000000) == 0x80000000) &&
      ((rtp_ts & 0x80000000) == 0)) {
    wrap_offset += (I_LLU << 32);
    have_wrap = true;
  }

  if (m_stream_ondemand) {
    adjusted_rtp_ts = wrap_offset;
    adjusted_rtp_ts += rtp_ts;
    adjusted_wc_rtp_ts = m_base_rtp_ts;

    if (adjusted_wc_rtp_ts > adjusted_rtp_ts) {
      timetick = adjusted_wc_rtp_ts - adjusted_rtp_ts;
      timetick *= M_LLU;
      timetick /= m_timescale;
      if (timetick > m_play_start_time) {
	timetick = 0;
      } else {
	timetick = m_play_start_time - timetick;
      }
    } else {
      timetick = adjusted_rtp_ts - adjusted_wc_rtp_ts;
      timetick *= M_LLU;
      timetick /= m_timescale;
      timetick += m_play_start_time;
    }
  } else {
    // We've got a broadcast scenario here...
    if (m_have_first_pak_ts == false) {
      // We haven't processed the first packet yet - we record
      // the data here.
      m_first_pak_rtp_ts = rtp_ts;
      m_first_pak_ts = uts;
      m_have_first_pak_ts = true;
      rtp_message(LOG_DEBUG, "%s first pak ts %u %llu", 
		  m_name, m_first_pak_rtp_ts, m_first_pak_ts);
      // if we have received RTCP, set the wallclock offset, which
      // triggers the syncronization effort.
      if (m_rtcp_received) {
	// calculate other stuff
	set_wallclock_offset(m_rtcp_ts, m_rtcp_rtp_ts);
      }
    }
    SDL_LockMutex(m_rtp_packet_mutex);
    // fairly simple calculation to calculate the timestamp
    // based on this rtp timestamp, the first pak rtp timestamp and
    // the first packet timestamp.
    int64_t adder;
    if (have_wrap) {
      adder = (int64_t)rtp_ts;
      adder += I_LLU << 32;
      adder -= (int64_t)m_first_pak_rtp_ts;
      // adjust once an hour, to keep errors low
      // we'll adjust the timestamp and rtp timestamp
      adder *= M_LLU;
      adder /= m_timescale;
      m_first_pak_ts += adder;
      m_first_pak_rtp_ts = rtp_ts;
#ifdef DEBUG_RTP_BCAST
      rtp_message(LOG_DEBUG, "%s adjust for wrap - first pak ts is now %llu rtp %u", 
		  m_name, m_first_pak_ts, m_first_pak_rtp_ts);
#endif
    }

    adder = (int64_t)rtp_ts;
    adder -= (int64_t) m_first_pak_rtp_ts;
    adder *= (int64_t)1000;
    adder /= (int64_t)m_timescale;
    timetick = m_first_pak_ts;
    timetick += adder;
    SDL_UnlockMutex(m_rtp_packet_mutex);

#ifdef DEBUG_RTP_BCAST
    rtp_message(LOG_DEBUG, "%s ts %u base %u %llu tp %llu",
		m_name, rtp_ts, m_first_pak_rtp_ts, m_first_pak_ts, 
		timetick);
#endif
  }
  // record time
  m_last_realtime = timetick;
  return (timetick);
}


CRtpByteStream::CRtpByteStream(const char *name,
			       format_list_t *fmt,
			       unsigned int rtp_pt,
			       int ondemand,
			       uint64_t tickpersec,
			       rtp_packet **head, 
			       rtp_packet **tail,
			       int rtp_seq_set,
			       uint16_t rtp_base_seq,
			       int rtp_ts_set,
			       uint32_t rtp_base_ts,
			       int rtcp_received,
			       uint32_t ntp_frac,
			       uint32_t ntp_sec,
			       uint32_t rtp_ts) :
  CRtpByteStreamBase(name, fmt, rtp_pt, ondemand, tickpersec, head, tail,
		     rtp_seq_set, rtp_base_seq, rtp_ts_set, rtp_base_ts, 
		     rtcp_received, ntp_frac, ntp_sec, rtp_ts)
{
  m_buffer = (uint8_t *)malloc(4096);
  m_buffer_len_max = 4096;
  m_bytes_used = m_buffer_len = 0;
}

CRtpByteStream::~CRtpByteStream (void)
{
  free(m_buffer);
  m_buffer = NULL;
}

void CRtpByteStream::reset (void)
{
  m_buffer_len = m_bytes_used = 0;
  CRtpByteStreamBase::reset();
}

uint64_t CRtpByteStream::start_next_frame (uint8_t **buffer, 
					   uint32_t *buflen,
					   void **ud)
{
  uint16_t seq = 0;
  uint32_t rtp_ts = 0;
  uint64_t timetick;
  uint64_t ts = 0;
  int first = 0;
  int finished = 0;
  rtp_packet *rpak;
  int32_t diff;

  diff = m_buffer_len - m_bytes_used;

  if (diff >= 2) {
    // Still bytes in the buffer...
    *buffer = m_buffer + m_bytes_used;
    *buflen = diff;
#ifdef DEBUG_RTP_PAKS
    rtp_message(LOG_DEBUG, "%s Still left - %d bytes", m_name, *buflen);
#endif
#if 0
  rtp_message(LOG_DEBUG, "%s start %02x %02x %02x %02x %02x", m_name,
		  	(*buffer)[0],
		  	(*buffer)[1],
		  	(*buffer)[2],
		  	(*buffer)[3],
		  	(*buffer)[4]);
#endif
    return (m_last_realtime);
  } else {
    m_buffer_len = 0;
    while (finished == 0) {
      rpak = m_head;
      if (rpak == NULL) { 
	// incomplete frame - bail on this
	player_error_message("%s - This should never happen - rtp bytestream"
			     "is incomplete and active", m_name);
	player_error_message("Please report to mpeg4ip");
	player_error_message("first %d seq %u ts %x blen %d",
			     first, seq, rtp_ts, m_buffer_len);
	m_buffer_len = 0;
	m_bytes_used = 0;
	return 0;
      }
	
      remove_packet_rtp_queue(rpak, 0);
      
      if (first == 0) {
	seq = rpak->rtp_pak_seq + 1;
	ts = rpak->pd.rtp_pd_timestamp;
	rtp_ts = rpak->rtp_pak_ts;
	first = 1;
      } else {
	if ((seq != rpak->rtp_pak_seq) ||
	    (rtp_ts != rpak->rtp_pak_ts)) {
	  if (seq != rpak->rtp_pak_seq) {
	    rtp_message(LOG_INFO, "%s missing rtp sequence - should be %u is %u", 
			m_name, seq, rpak->rtp_pak_seq);
	  } else {
	    rtp_message(LOG_INFO, "%s timestamp error - seq %u should be %x is %x", 
			m_name, seq, rtp_ts, rpak->rtp_pak_ts);
	  }
	  m_buffer_len = 0;
	  rtp_ts = rpak->rtp_pak_ts;
	}
	seq = rpak->rtp_pak_seq + 1;
      }
      uint8_t *from;
      uint32_t len;
      from = (uint8_t *)rpak->rtp_data + m_skip_on_advance_bytes;
      len = rpak->rtp_data_len - m_skip_on_advance_bytes;
      if ((m_buffer_len + len) >= m_buffer_len_max) {
	// realloc
	m_buffer_len_max = m_buffer_len + len + 1024;
	m_buffer = (uint8_t *)realloc(m_buffer, m_buffer_len_max);
      }
      memcpy(m_buffer + m_buffer_len, 
	     from,
	     len);
      m_buffer_len += len;
      if (rpak->rtp_pak_m == 1) {
	finished = 1;
      }
      xfree(rpak);
    }
    m_bytes_used = 0;
    *buffer = m_buffer + m_bytes_used;
    *buflen = m_buffer_len - m_bytes_used;
#if 0
  rtp_message(LOG_DEBUG, "%s start %02x %02x %02x %02x %02x", m_name,
		  	(*buffer)[0],
		  	(*buffer)[1],
		  	(*buffer)[2],
		  	(*buffer)[3],
		  	(*buffer)[4]);
#endif
#ifdef DEBUG_RTP_PAKS
    rtp_message(LOG_DEBUG, "%s buffer len %d", m_name, m_buffer_len);
#endif
  }
  timetick = rtp_ts_to_msec(rtp_ts, ts, m_wrap_offset);
  m_ts = rtp_ts;
  
  return (timetick);
}

int CRtpByteStream::skip_next_frame (uint64_t *pts, int *hasSyncFrame,
				     uint8_t **buffer, 
				     uint32_t *buflen,
				     void **ud)
{
  uint64_t ts;
  *hasSyncFrame = -1;  // we don't know if we have a sync frame
  m_buffer_len = m_bytes_used = 0;
  if (m_head == NULL) return 0;
  ts = m_head->rtp_pak_ts;
  do {
    remove_packet_rtp_queue(m_head, 1);
  } while (m_head != NULL && m_head->rtp_pak_ts == ts);

  if (m_head == NULL) return 0;
  init();
  m_buffer_len = m_bytes_used = 0;
  ts = start_next_frame(buffer, buflen, ud);
  *pts = ts;
  return (1);
}

void CRtpByteStream::used_bytes_for_frame (uint32_t bytes)
{
  m_bytes_used += bytes;
#ifdef DEBUG_RTP_PAKS
  rtp_message(LOG_DEBUG, "%s Used %d bytes", m_name, bytes);
#endif
}

int CRtpByteStream::have_no_data (void)
{
  rtp_packet *temp, *first;
  first = temp = m_head;
  if (temp == NULL) return TRUE;
  do {
    if (temp->rtp_pak_m == 1) return FALSE;
    temp = temp->rtp_next;
  } while (temp != NULL && temp != first);
  return TRUE;
}

void CRtpByteStream::flush_rtp_packets (void)
{
  CRtpByteStreamBase::flush_rtp_packets();

  m_bytes_used = m_buffer_len = 0;
}

CAudioRtpByteStream::CAudioRtpByteStream (unsigned int rtp_pt,
					  format_list_t *fmt,
					  int ondemand,
					  uint64_t tps,
					  rtp_packet **head, 
					  rtp_packet **tail,
					  int rtp_seq_set,
					  uint16_t rtp_base_seq,
					  int rtp_ts_set,
					  uint32_t rtp_base_ts,
					  int rtcp_received,
					  uint32_t ntp_frac,
					  uint32_t ntp_sec,
					  uint32_t rtp_ts) :
  CRtpByteStream("audio", 
		 fmt,
		 rtp_pt,
		 ondemand,
		 tps,
		 head, 
		 tail,
		 rtp_seq_set, rtp_base_seq, 
		 rtp_ts_set, rtp_base_ts,
		 rtcp_received,
		 ntp_frac,
		 ntp_sec,
		 rtp_ts)
{
  init();
  m_working_pak = NULL;
}

CAudioRtpByteStream::~CAudioRtpByteStream(void)
{
}

int CAudioRtpByteStream::have_no_data (void)
{
  if (m_head == NULL) return TRUE;
  return FALSE;
}

int CAudioRtpByteStream::check_rtp_frame_complete_for_payload_type (void)
{
  return m_head != NULL;
}

void CAudioRtpByteStream::reset (void)
{
  if (m_working_pak != NULL) {
    xfree(m_working_pak);
    m_working_pak = NULL;
  }
  CRtpByteStream::reset();
}
uint64_t CAudioRtpByteStream::start_next_frame (uint8_t **buffer, 
						uint32_t *buflen,
						void **ud)
{
  uint32_t ts;
  int32_t diff;

  if (m_working_pak != NULL) {
    diff = m_working_pak->rtp_data_len - m_bytes_used;
  } else diff = 0;

  if (diff > 0) {
    // Still bytes in the buffer...
    *buffer = (uint8_t *)m_working_pak->rtp_data + m_bytes_used;
    *buflen = diff;
    ts = m_ts;
#ifdef DEBUG_RTP_PAKS
    rtp_message(LOG_DEBUG, "%s Still left - %d bytes", m_name, *buflen);
#endif
    return (m_last_realtime);
  } else {
    if (m_working_pak) xfree(m_working_pak);
    m_buffer_len = 0;
    m_bytes_used = m_skip_on_advance_bytes;
    m_working_pak = m_head;
    remove_packet_rtp_queue(m_working_pak, 0);
    if (m_have_first_pak_ts && m_seq_recvd != m_working_pak->rtp_pak_seq) {
      rtp_message(LOG_ERR, "%s missing seq should be %d got %d", 
		  m_name, m_seq_recvd, m_working_pak->rtp_pak_seq);
    }
    m_seq_recvd = m_working_pak->rtp_pak_seq + 1;

    *buffer = (uint8_t *)m_working_pak->rtp_data + m_bytes_used;
    *buflen = m_working_pak->rtp_data_len;
    ts = m_working_pak->rtp_pak_ts;
#ifdef DEBUG_RTP_PAKS
    rtp_message(LOG_DEBUG, "%s buffer seq %d ts %x len %d", m_name, 
		m_working_pak->rtp_pak_seq, 
		m_working_pak->rtp_pak_ts, m_buffer_len);
#endif
  }

  // We're going to have to handle wrap better...
  uint64_t retts = rtp_ts_to_msec(ts, 
				  m_working_pak->pd.rtp_pd_timestamp,
				  m_wrap_offset);
  m_ts = ts;
  return retts;

}