Remove RtpPacketToSend::GetHeader as almost unused.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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 482 matching lines @@
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   auto packet = rtp_sender_->AllocatePacket();
   packet->SetPayloadType(kPayload);
   packet->SetMarker(true);
   packet->SetTimestamp(kTimestamp);
   packet->set_capture_time_ms(capture_time_ms);
   const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
   packet->SetExtension<VideoTimingExtension>(kVideoTiming);
   EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
   size_t packet_size = packet->size();
-  webrtc::RTPHeader rtp_header;
-
-  packet->GetHeader(&rtp_header);
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
   EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
                                          kAllowRetransmission,
                                          RtpPacketSender::kNormalPriority));
   EXPECT_EQ(1, transport_.packets_sent());
   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
 
-  transport_.last_sent_packet().GetHeader(&rtp_header);
-  EXPECT_TRUE(rtp_header.extension.has_video_timing);
-  EXPECT_EQ(kStoredTimeInMs,
-            rtp_header.extension.video_timing.pacer_exit_delta_ms);
+  VideoSendTiming video_timing;
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
+      &video_timing));
+  EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
 
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
   rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
                                 PacedPacketInfo());
 
   EXPECT_EQ(2, transport_.packets_sent());
   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
 
-  transport_.last_sent_packet().GetHeader(&rtp_header);
-  EXPECT_TRUE(rtp_header.extension.has_video_timing);
-  EXPECT_EQ(kStoredTimeInMs * 2,
-            rtp_header.extension.video_timing.pacer_exit_delta_ms);
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
+      &video_timing));
+  EXPECT_EQ(kStoredTimeInMs * 2, video_timing.pacer_exit_delta_ms);
 }
 
 TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ skipping 1453 matching lines @@
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTestWithoutPacer,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderVideoTest,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderAudioTest,
                         ::testing::Bool());
 }  // namespace webrtc