Change RtpSender::OnReceiveNACK name and signature
Name changed to follow style.
list replaced with vector to decrease number of included headers.

R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2276833003 .

Cr-Commit-Position: refs/heads/master@{#13938}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 28c6867..9c0d66d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -1263,7 +1263,7 @@
     if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
       LOG(LS_VERBOSE) << "Incoming NACK length: "
                       << rtcpPacketInformation.nackSequenceNumbers.size();
-      _rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers);
+      _rtpRtcp.OnReceivedNack(rtcpPacketInformation.nackSequenceNumbers);
     }
   }
   {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index 5b2eb66..35db67c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -11,7 +11,6 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 
-#include <list>
 #include <map>
 #include <memory>
 #include <vector>
@@ -61,7 +60,7 @@
   uint32_t rtcpPacketTypeFlags;  // RTCPPacketTypeFlags bit field
   uint32_t remoteSSRC;
 
-  std::list<uint16_t> nackSequenceNumbers;
+  std::vector<uint16_t> nackSequenceNumbers;
 
   uint8_t applicationSubType;
   uint32_t applicationName;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index bf151d2..ba7ca7e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -870,8 +870,8 @@
       GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id);
 }
 
-void ModuleRtpRtcpImpl::OnReceivedNACK(
-    const std::list<uint16_t>& nack_sequence_numbers) {
+void ModuleRtpRtcpImpl::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers) {
   for (uint16_t nack_sequence_number : nack_sequence_numbers) {
     send_loss_stats_.AddLostPacket(nack_sequence_number);
   }
@@ -884,7 +884,7 @@
   if (rtt == 0) {
     rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
   }
-  rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt);
+  rtp_sender_.OnReceivedNack(nack_sequence_numbers, rtt);
 }
 
 void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index f8fb211..80e76e3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -11,7 +11,6 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
-#include <list>
 #include <set>
 #include <utility>
 #include <vector>
@@ -312,7 +311,7 @@
   StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
       const override;
 
-  void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
+  void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers);
   void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
 
   void OnRequestSendReport();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 1f68542..5a0dc13 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -721,8 +721,9 @@
   return 0;
 }
 
-void RTPSender::OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
-                               int64_t avg_rtt) {
+void RTPSender::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers,
+    int64_t avg_rtt) {
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                "RTPSender::OnReceivedNACK", "num_seqnum",
                nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index e1dfa9b..7a6677b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -11,7 +11,6 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 
-#include <list>
 #include <map>
 #include <memory>
 #include <utility>
@@ -183,7 +182,7 @@
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
-  void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
+  void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 41b52a2..e11bf1e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <list>
 #include <memory>
 #include <vector>
 
@@ -1637,7 +1636,7 @@
 
   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);
@@ -1650,14 +1649,14 @@
   // 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_);
 }