Remove hardcoded kValueSizeBytes values from variable-length header extensions.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 27 matching lines @@
 constexpr size_t kMaxPaddingLength = 224;
 constexpr size_t kMinAudioPaddingLength = 50;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr size_t kRtpHeaderLength = 12;
 constexpr uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr int kBitrateStatisticsWindowMs = 1000;
 
 constexpr size_t kMinFlexfecPacketsToStoreForPacing = 50;
 
+constexpr RTPExtensionSizeInfo kExtensionSizes[] = {

[danilchap, 2017/05/08 16:25:06]

can you add a comment or change variable name to explain why only these
extensions are selected. (e.g. why PlayoutDelay is included, but AudioLevel and
VideoOrientation are not).

[erikvarga1, 2017/05/09 11:40:03]

Done. I've renamed this to show that it can be used for padding or FEC packets.

+    CreateExtensionSizeInfo<AbsoluteSendTime>(),
+    CreateExtensionSizeInfo<TransmissionOffset>(),
+    CreateExtensionSizeInfo<TransportSequenceNumber>(),
+    CreateExtensionSizeInfo<PlayoutDelayLimits>(),
+};
+
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
     case kEmptyFrame:
       return "empty";
     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
     case kVideoFrameDelta: return "video_delta";
   }
   return "";
@@ skipping 896 matching lines @@
 
   rtc::CritScope lock(&statistics_crit_);
   bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
                             nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
 }
 
 size_t RTPSender::RtpHeaderLength() const {
   rtc::CritScope lock(&send_critsect_);
   size_t rtp_header_length = kRtpHeaderLength;
   rtp_header_length += sizeof(uint32_t) * csrcs_.size();
-  rtp_header_length += rtp_header_extension_map_.GetTotalLengthInBytes();
+  rtp_header_length +=
+      rtp_header_extension_map_.GetTotalLengthInBytes(kExtensionSizes);
   return rtp_header_length;
 }
 
 uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
   rtc::CritScope lock(&send_critsect_);
   uint16_t first_allocated_sequence_number = sequence_number_;
   sequence_number_ += packets_to_send;
   return first_allocated_sequence_number;
 }
 
@@ skipping 282 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc