NACK WebRTC pseudocode .py

# code from nack_module.cc
on_packet(rtp):
  if rtp.seq_num < newest_seq_num:  # out of order or retransmnission
     return
  add_missing(newest_seq_num + 1, rtp.seq_num)
  if rtp.is_first_packet_of_keyframe:  
     keyframes.insert(rtp.seq_num)

  newest_seq_num = packet.seq_num
  if nack = generate_nack(SEQ):
     send_nack(nack)

add_missing(from_seq_num, to_seq_num):
  packets_lost.remove_older_than(10000) # MAX_PACKET_AGE (sequence numbers)
  if packets_lost.size() > 1000: # MAX_SIZE
     packets_lost.remove_until_keyframe()
  if packets_lost.size() > 1000: # Still too big
     packets_lost.clear()
     sender.request_keyframe()
     return
  for seq_num in range(from_seq_num, to_seq_num):
    packets_lost.insert(seq_num)

# called when a new decodable frame is ready
cleanup_to(to_seq_num):
  packets_lost.remove(seq_num < to_seq_num);

generate_nack(mode):
  seq_nums = []
  for packet in packets_lost:
    if packet.retries > 10: # MAX_RETRIES
        packets_lost.remove(packet)
    if (mode == SEQ and packet.seq_num < newest_seq_num) or
       (mode == TIME and packet.sent_at + rtt > now()):
        seq_nums.insert(packet.seq_num)
        packet.sent_at = now()
  return seq_nums
  
every_20_msec():
  if nack = generate_nack(TIME):
     send_nack(nack)

# code from rtp_rtcp_impl.cc
send_nack(seq_nums):
   if now - time_last_full_nack_sent < 1.5 * RTT:  
       seq_nums.filter(seq_num > seq_num_last_nack_sent)
   else:
       time_last_full_nack_sent = now
   if seq_nums.length > 253:
       seq_nums = seq_nums[:-253]
   send(new RtcpNack(seq_nums))