webrtc/call/rtc_event_log_unittest.cc - Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call.

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Updated RTP/RTCP module to use setter methods instead of passing the event log pointer in the const… Created 4 years, 10 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

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 b86a6885674d6bdeedcf422d1528c74b08dfffa1..03f6dda8197e4362a225fd61c4485f4ccb6caba3 100644

--- a/webrtc/call/rtc_event_log_unittest.cc

+++ b/webrtc/call/rtc_event_log_unittest.cc

@@ -314,8 +314,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,

nullptr, // SendTimeObserver*

nullptr, // BitrateStatisticsObserver*

nullptr, // FrameCountObserver*

- nullptr, // SendSideDelayObserver*

- nullptr); // RtcEventLog*

+ nullptr); // SendSideDelayObserver*

std::vector<uint32_t> csrcs;

for (unsigned i = 0; i < csrcs_count; i++) {

« webrtc/call/call.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/media/base/mediaengine.h » ('j') | webrtc/modules/rtp_rtcp/include/rtp_rtcp.h » ('J')