Change RtpSender::OnReceiveNACK name and signature

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  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.
  */
 
 #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>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_annotations.h"
@@ skipping 151 matching lines @@
 
   bool TimeToSendPacket(uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission,
                         int probe_cluster_id);
   size_t TimeToSendPadding(size_t bytes, int probe_cluster_id);
 
   // 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);
 
   bool StorePackets() const;
 
   int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
 
   // Feedback to decide when to stop sending playout delay.
   void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
@@ skipping 223 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_