Remove RtpPacketToSend::GetHeader as almost unused.

webrtc/modules/rtp_rtcp/source/rtp_packet.cc

 /*
  *  Copyright (c) 2016 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.h"
 
 #include <cstring>
 #include <utility>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/random.h"
 
 namespace webrtc {
 namespace rtp {
 namespace {
 constexpr size_t kFixedHeaderSize = 12;
 constexpr uint8_t kRtpVersion = 2;
 constexpr uint16_t kOneByteExtensionId = 0xBEDE;
@@ skipping 108 matching lines @@
   size_t num_csrc = data()[0] & 0x0F;
   RTC_DCHECK_GE(capacity(), kFixedHeaderSize + num_csrc * 4);
   std::vector<uint32_t> csrcs(num_csrc);
   for (size_t i = 0; i < num_csrc; ++i) {
     csrcs[i] =
         ByteReader<uint32_t>::ReadBigEndian(&data()[kFixedHeaderSize + i * 4]);
   }
   return csrcs;
 }
 
-void Packet::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 = 0;
-  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);
-}
-
 size_t Packet::headers_size() const {
   return payload_offset_;
 }
 
 size_t Packet::payload_size() const {
   return payload_size_;
 }
 
 size_t Packet::padding_size() const {
   return padding_size_;
@@ skipping 383 matching lines @@
 uint8_t* Packet::WriteAt(size_t offset) {
   return buffer_.data() + offset;
 }
 
 void Packet::WriteAt(size_t offset, uint8_t byte) {
   buffer_.data()[offset] = byte;
 }
 
 }  // namespace rtp
 }  // namespace webrtc