Minor fixes in FEC and RtpSender{,Video}. (pt. 2)

webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 858 matching lines @@
   RTC_DCHECK_GT(num_media_packets, 0);
   int sequence_number = start_seq_num;
   int time_stamp = random_.Rand<int>();
 
   for (int i = 0; i < num_media_packets; ++i) {
     std::unique_ptr<ForwardErrorCorrection::Packet> media_packet(
         new ForwardErrorCorrection::Packet());
     constexpr uint32_t kMinPacketSize = kRtpHeaderSize;
     const uint32_t kMaxPacketSize = IP_PACKET_SIZE - kRtpHeaderSize -
                                     kTransportOverhead -
-                                    ForwardErrorCorrection::PacketOverhead();
+                                    fec_.MaxPacketOverhead();
     media_packet->length = random_.Rand(kMinPacketSize, kMaxPacketSize);
 
     // Generate random values for the first 2 bytes
     media_packet->data[0] = random_.Rand<uint8_t>();
     media_packet->data[1] = random_.Rand<uint8_t>();
 
     // The first two bits are assumed to be 10 by the FEC encoder.
     // In fact the FEC decoder will set the two first bits to 10 regardless of
     // what they actually were. Set the first two bits to 10 so that a memcmp
     // can be performed for the whole restored packet.
@@ skipping 35 matching lines @@
                                         PacketList* dst) {
   RTC_DCHECK_GT(n, 0);
   int i = 0;
   for (const auto& packet : src) {
     if (i % n == 0) {
       dst->emplace_back(new ForwardErrorCorrection::Packet(*packet));
     }
     ++i;
   }
 }