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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: Addressed comments 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.
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc » ('j') | webrtc/video/end_to_end_tests.cc » ('J')

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