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webrtc_m130/webrtc/video
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saza 746749237a Add received audio/video call duration metrics based on packets. 
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
2017-07-04 07:19:22 +00:00
..
BUILD.gn
Wire up experiment for improved screenshare bwe.
2017-06-30 20:27:40 +00:00
call_stats_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
call_stats.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
call_stats.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
DEPS
Reland of use allocated encoders in SimulcastEncoderAdapter. (patchset #1 id:1 of https://codereview.webrtc.org/2893003002/ )
2017-05-23 10:32:16 +00:00
encoder_rtcp_feedback_unittest.cc
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
encoder_rtcp_feedback.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
encoder_rtcp_feedback.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
end_to_end_tests.cc
Add received audio/video call duration metrics based on packets.
2017-07-04 07:19:22 +00:00
full_stack_tests_plot.py
Fix lint errors to enable stricter PyLint rules
2017-04-12 19:06:13 +00:00
full_stack_tests.cc
Wire up experiment for improved screenshare bwe.
2017-06-30 20:27:40 +00:00
overuse_frame_detector_unittest.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
overuse_frame_detector.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
overuse_frame_detector.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
OWNERS
Adding Åsa and Erik as video owners.
2016-12-16 10:12:31 +00:00
payload_router_unittest.cc
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
payload_router.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
payload_router.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
picture_id_tests.cc
Test picture sequence id when VideoSendStream is recreated.
2017-06-22 15:57:10 +00:00
quality_threshold_unittest.cc
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +00:00
quality_threshold.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
quality_threshold.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
receive_statistics_proxy_unittest.cc
Add some unit tests to ReceiveStatsticsProxy.
2017-05-11 12:37:35 +00:00
receive_statistics_proxy.cc
Add received interframe delay UMA metrics
2017-07-03 08:15:58 +00:00
receive_statistics_proxy.h
Add received interframe delay UMA metrics
2017-07-03 08:15:58 +00:00
replay.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
report_block_stats_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
report_block_stats.cc
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
report_block_stats.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
rtp_streams_synchronizer.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtp_streams_synchronizer.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtp_video_stream_receiver_unittest.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtp_video_stream_receiver.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtp_video_stream_receiver.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
screenshare_loopback.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
send_delay_stats_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
send_delay_stats.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
send_delay_stats.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
send_statistics_proxy_unittest.cc
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
send_statistics_proxy.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
send_statistics_proxy.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stats_counter_unittest.cc
Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call.
2017-02-06 13:18:35 +00:00
stats_counter.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stats_counter.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stream_synchronization_unittest.cc
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
stream_synchronization.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stream_synchronization.h
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
transport_adapter.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
transport_adapter.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_loopback.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_quality_test.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_quality_test.h
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +00:00
video_receive_stream_unittest.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_receive_stream.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_receive_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream_tests.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_stream_decoder.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_stream_decoder.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
vie_encoder_unittest.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
vie_encoder.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
vie_encoder.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00