Unify rtcp packet setters

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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 <algorithm>
@@ skipping 510 matching lines @@
       if (!sequence_number_to_retransmit_) {
         sequence_number_to_retransmit_ =
             rtc::Optional<uint16_t>(header.sequenceNumber);
 
         // Don't ask for retransmission straight away, may be deduped in pacer.
       } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
         observation_complete_.Set();
       } else {
         // Send a NACK as often as necessary until retransmission is received.
         rtcp::Nack nack;
-        nack.From(local_ssrc_);
-        nack.To(remote_ssrc_);
+        nack.SetSenderSsrc(local_ssrc_);
+        nack.SetMediaSsrc(remote_ssrc_);
         uint16_t nack_list[] = {*sequence_number_to_retransmit_};
-        nack.WithList(nack_list, 1);
+        nack.SetPacketIds(nack_list, 1);
         rtc::Buffer buffer = nack.Build();
 
         EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
       }
 
       return SEND_PACKET;
     }
 
     void ModifyAudioConfigs(
         AudioSendStream::Config* send_config,
@@ skipping 3192 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc