Change RtpSender::OnReceiveNACK name and signature

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.
  */
 
-#include <list>
 #include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ skipping 1608 matching lines @@
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
   const int32_t kPacketSize = 1400;
   const int32_t kNumPackets = 30;
 
   retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
 
   rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
   const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
-  std::list<uint16_t> sequence_numbers;
+  std::vector<uint16_t> sequence_numbers;
   for (int32_t i = 0; i < kNumPackets; ++i) {
     sequence_numbers.push_back(kStartSequenceNumber + i);
     fake_clock_.AdvanceTimeMilliseconds(1);
     SendPacket(fake_clock_.TimeInMilliseconds(), kPacketSize);
   }
   EXPECT_EQ(kNumPackets, transport_.packets_sent_);
 
   fake_clock_.AdvanceTimeMilliseconds(1000 - kNumPackets);
 
   // Resending should work - brings the bandwidth up to the limit.
   // NACK bitrate is capped to the same bitrate as the encoder, since the max
   // protection overhead is 50% (see MediaOptimization::SetTargetRates).
-  rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
+  rtp_sender_->OnReceivedNack(sequence_numbers, 0);
   EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
 
   // Must be at least 5ms in between retransmission attempts.
   fake_clock_.AdvanceTimeMilliseconds(5);
 
   // Resending should not work, bandwidth exceeded.
-  rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
+  rtp_sender_->OnReceivedNack(sequence_numbers, 0);
   EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
 }
 
 // Verify that all packets of a frame have CVO byte set.
 TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
   RTPVideoHeader hdr = {0};
   hdr.rotation = kVideoRotation_90;
 
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
@@ skipping 15 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