Unify rtcp packet setters

webrtc/call/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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 121 matching lines @@
   size_t payload_size = packet_size - rtp_packet.headers_size();
   uint8_t* payload = rtp_packet.AllocatePayload(payload_size);
   for (size_t i = 0; i < payload_size; i++) {
     payload[i] = prng->Rand<uint8_t>();
   }
   return rtp_packet;
 }
 
 rtc::Buffer GenerateRtcpPacket(Random* prng) {
   rtcp::ReportBlock report_block;
-  report_block.To(prng->Rand<uint32_t>());  // Remote SSRC.
-  report_block.WithFractionLost(prng->Rand(50));
+  report_block.SetMediaSsrc(prng->Rand<uint32_t>());  // Remote SSRC.
+  report_block.SetFractionLost(prng->Rand(50));
 
   rtcp::SenderReport sender_report;
-  sender_report.From(prng->Rand<uint32_t>());  // Sender SSRC.
-  sender_report.WithNtp(
-      NtpTime(prng->Rand<uint32_t>(), prng->Rand<uint32_t>()));
-  sender_report.WithPacketCount(prng->Rand<uint32_t>());
-  sender_report.WithReportBlock(report_block);
+  sender_report.SetSenderSsrc(prng->Rand<uint32_t>());
+  sender_report.SetNtp(NtpTime(prng->Rand<uint32_t>(), prng->Rand<uint32_t>()));
+  sender_report.SetPacketCount(prng->Rand<uint32_t>());
+  sender_report.AddReportBlock(report_block);
 
   return sender_report.Build();
 }
 
 void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
                                 VideoReceiveStream::Config* config,
                                 Random* prng) {
   // Create a map from a payload type to an encoder name.
   VideoReceiveStream::Decoder decoder;
   decoder.payload_type = prng->Rand(0, 127);
@@ skipping 291 matching lines @@
                                          MediaType::VIDEO, rtcp_packet.data(),
                                          rtcp_packet.size());
 
   RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 
 }  // namespace webrtc