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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
fb4d66be9726b849f8af61cafb09ed96bb993604
/
video
/
call_stats_unittest.cc
38c5d93
Reduce locking for CallStats (preparation for TaskQueue).
by Tommi
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/video/call_stats_unittest.cc]
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
01d70a3
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
by asapersson
· 8 years ago
d28db7f
Delete all use of tick_util.h.
by Niels Möller
· 9 years ago
27f982b
Replace scoped_ptr with unique_ptr in webrtc/video/
by kwiberg
· 9 years ago
e2d83d6
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
by sprang
· 9 years ago
7623ce4
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
[Renamed (99%) from webrtc/video_engine/call_stats_unittest.cc]
d3c9447
Nuke TickTime::UseFakeClock.
by Peter Boström
· 9 years ago
8237abf
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
[Renamed (99%) from webrtc/video/call_stats_unittest.cc]
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
[Renamed (99%) from webrtc/video_engine/call_stats_unittest.cc]
ff761fb
modules: more interface -> include renames
by Henrik Kjellander
· 9 years ago
98f5351
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
2328a94
Add average rtt to CallStatsObserver and an average rtt histogram.
by stefan
· 9 years ago
00b8f6b
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
by kwiberg@webrtc.org
· 10 years ago
16825b1
Use int64_t more consistently for times, in particular for RTT values.
by pkasting@chromium.org
· 10 years ago
8084f95
Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
by asapersson@webrtc.org
· 10 years ago
1ae1d0c
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
by asapersson@webrtc.org
· 11 years ago
f5d4cb1
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 11 years ago
8ca8a71
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
by stefan@webrtc.org
· 12 years ago
ccd4b2a
Add a default RTT to CallStats and use different values for buffered/real-time mode.
by stefan@webrtc.org
· 12 years ago
aea96d3
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
by fischman@webrtc.org
· 12 years ago
b2f474e
Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.
by mflodman@webrtc.org
· 12 years ago