Remove RtpPacketToSend::GetHeader as almost unused.

webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
new file mode 100644
index 0000000000000000000000000000000000000000..82a4f68d69150cc7157d4c685a331b77aecbf5a4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 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 "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+namespace webrtc {
+
+void RtpPacketReceived::GetHeader(RTPHeader* header) const {
+  header->markerBit = Marker();
+  header->payloadType = PayloadType();
+  header->sequenceNumber = SequenceNumber();
+  header->timestamp = Timestamp();
+  header->ssrc = Ssrc();
+  std::vector<uint32_t> csrcs = Csrcs();
+  header->numCSRCs = csrcs.size();
+  for (size_t i = 0; i < csrcs.size(); ++i) {
+    header->arrOfCSRCs[i] = csrcs[i];
+  }
+  header->paddingLength = padding_size();
+  header->headerLength = headers_size();
+  header->payload_type_frequency = payload_type_frequency();
+  header->extension.hasTransmissionTimeOffset =
+      GetExtension<TransmissionOffset>(
+          &header->extension.transmissionTimeOffset);
+  header->extension.hasAbsoluteSendTime =
+      GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
+  header->extension.hasTransportSequenceNumber =
+      GetExtension<TransportSequenceNumber>(
+          &header->extension.transportSequenceNumber);
+  header->extension.hasAudioLevel = GetExtension<AudioLevel>(
+      &header->extension.voiceActivity, &header->extension.audioLevel);
+  header->extension.hasVideoRotation =
+      GetExtension<VideoOrientation>(&header->extension.videoRotation);
+  header->extension.hasVideoContentType =
+      GetExtension<VideoContentTypeExtension>(
+          &header->extension.videoContentType);
+  header->extension.has_video_timing =
+      GetExtension<VideoTimingExtension>(&header->extension.video_timing);
+  GetExtension<RtpStreamId>(&header->extension.stream_id);
+  GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
+  GetExtension<RtpMid>(&header->extension.mid);
+  GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
+}
+
+}  // namespace webrtc