Change RtpSender::OnReceiveNACK name and signature

webrtc/modules/rtp_rtcp/source/rtp_sender.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 703 matching lines @@
   return video_->SelectiveRetransmissions();
 }
 
 int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
   if (!video_)
     return -1;
   video_->SetSelectiveRetransmissions(settings);
   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);
   for (uint16_t seq_no : nack_sequence_numbers) {
     const int32_t bytes_sent = ReSendPacket(seq_no, 5 + avg_rtt);
     if (bytes_sent < 0) {
       // Failed to send one Sequence number. Give up the rest in this nack.
       LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
                       << ", Discard rest of packets";
       break;
@@ skipping 984 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc