webrtc/modules/rtp_rtcp/source/rtp_sender.cc - Issue 2217383002: Use RtpPacketToSend in RtpSenderVideo

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Issue 2217383002: Use RtpPacketToSend in RtpSenderVideo (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Expand comment about non-trivial minor optimization in rtp::Packet::AllocatePayload Created 4 years, 3 months ago

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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 38069688f2ba635e826d245132f172d8e12396c1..54d988ee88258c0139c371432d9aaa12710d212c 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -1093,6 +1093,10 @@ std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {

packet->ReserveExtension<AbsoluteSendTime>();

packet->ReserveExtension<TransmissionOffset>();

packet->ReserveExtension<TransportSequenceNumber>();

+ if (playout_delay_oracle_.send_playout_delay()) {

+ packet->SetExtension<PlayoutDelayLimits>(

+ playout_delay_oracle_.playout_delay());

+ }

return packet;

}