RtpPacketHistory rewritten to use RtpPacket class.

webrtc/modules/rtp_rtcp/source/rtp_sender_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 131 matching lines @@
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(
         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
         &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
         &mock_rtc_event_log_, &send_packet_observer_,
         &retransmission_rate_limiter_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
+    rtp_sender_->SetSendPayloadType(kPayload);
   }
 
   SimulatedClock fake_clock_;
   testing::NiceMock<MockRtcEventLog> mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   testing::StrictMock<MockTransportSequenceNumberAllocator> seq_num_allocator_;
   testing::StrictMock<MockSendPacketObserver> send_packet_observer_;
   RateLimiter retransmission_rate_limiter_;
   std::unique_ptr<RTPSender> rtp_sender_;
   int payload_;
@@ skipping 1459 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc