Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 6fdac6f63fb08507ec68781c4bff1e59b6ca693e..5bb5474f17835512d01ee2b815d2fe8c19a86414 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -45,6 +45,13 @@ 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 e
erikvarga1
2017/05/09 11:40:03
Done. I've renamed this to show that it can be use
|
+ CreateExtensionSizeInfo<AbsoluteSendTime>(), |
+ CreateExtensionSizeInfo<TransmissionOffset>(), |
+ CreateExtensionSizeInfo<TransportSequenceNumber>(), |
+ CreateExtensionSizeInfo<PlayoutDelayLimits>(), |
+}; |
+ |
const char* FrameTypeToString(FrameType frame_type) { |
switch (frame_type) { |
case kEmptyFrame: |
@@ -961,7 +968,8 @@ 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; |
} |