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webrtc_m130/webrtc/video
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sprang 892dab52b6 Fix incorrect InterframeDelayMaxInMs histogram metrics 
Two bugs:

1) The max value should only be reported if the average is also
   reported. Otherwise the max might become lower than average.
   (On average).

2) When reporting that max value, actually use the max value.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/3002593002
Cr-Commit-Position: refs/heads/master@{#19352}
2017-08-15 12:00:33 +00:00
..
BUILD.gn
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
call_stats.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
encoder_rtcp_feedback.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
encoder_rtcp_feedback.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
end_to_end_tests.cc
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
2017-08-14 15:03:17 +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
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
overuse_frame_detector_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
overuse_frame_detector.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
overuse_frame_detector.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
payload_router.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
picture_id_tests.cc
Remove webrtc::VideoEncoderFactory
2017-07-10 10:26:36 +00:00
quality_threshold_unittest.cc
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +00:00
quality_threshold.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quality_threshold.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
receive_statistics_proxy_unittest.cc
Fix incorrect InterframeDelayMaxInMs histogram metrics
2017-08-15 12:00:33 +00:00
receive_statistics_proxy.cc
Fix incorrect InterframeDelayMaxInMs histogram metrics
2017-08-15 12:00:33 +00:00
receive_statistics_proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
replay.cc
Fix video_replay tool to respect RTX stream and fix default parameters.
2017-07-11 09:38:36 +00:00
report_block_stats_unittest.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
report_block_stats.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_streams_synchronizer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_video_stream_receiver_unittest.cc
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
rtp_video_stream_receiver.cc
Workaround for PacketBuffer bug.
2017-08-10 16:11:04 +00:00
rtp_video_stream_receiver.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
screenshare_loopback.cc
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
send_delay_stats.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
send_statistics_proxy_unittest.cc
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
send_statistics_proxy.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
send_statistics_proxy.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
stats_counter.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
stream_synchronization.h
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
transport_adapter.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
video_quality_test.cc
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_quality_test.h
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
video_receive_stream_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_receive_stream.cc
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
2017-08-11 10:41:44 +00:00
video_receive_stream.h
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
2017-08-11 10:41:44 +00:00
video_send_stream_tests.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
video_send_stream.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
video_send_stream.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
video_stream_decoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_stream_decoder.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_stream_encoder_unittest.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
video_stream_encoder.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
video_stream_encoder.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00