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Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1478253002: Add histogram stats for send delay for a sent video stream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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« webrtc/call/call.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/common_types.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('J')
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Index: webrtc/call/rtc_event_log_unittest.cc
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index eb3c14814ca31f174265d234e444078d8a35a059..058f7feabd756d6a7e8c9b958a5a648762e51e2d 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -314,7 +314,8 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
nullptr, // SendTimeObserver*
nullptr, // BitrateStatisticsObserver*
nullptr, // FrameCountObserver*
- nullptr); // SendSideDelayObserver*
+ nullptr, // SendSideDelayObserver*
+ nullptr); // SendPacketObserver*
std::vector<uint32_t> csrcs;
for (unsigned i = 0; i < csrcs_count; i++) {
« webrtc/call/call.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/common_types.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('J')

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