webrtc/modules/rtp_rtcp/source/rtp_sender.cc - Issue 2181383002: Add NACK rate throttling for audio channels.

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

Issue 2181383002: Add NACK rate throttling for audio channels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Fixed race in test Created 4 years, 5 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 25016e053f2b43a948f803846fda364423d31dbe..21bc746fe593ed2bb4d1913ccebf39408ddbd6c4 100644

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

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

@@ -127,6 +127,8 @@ RTPSender::RTPSender(

csrcs_(),

rtx_(kRtxOff),

retransmission_rate_limiter_(retransmission_rate_limiter) {

+ RTC_DCHECK(retransmission_rate_limiter_ != nullptr);

// We need to seed the random generator for BuildPaddingPacket() below.

// TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac

// early on in the process.