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f4ded682e913d56bbadf14a391ff3ba37a7419f5
platform-external-webrtc/webrtc/video
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ilnik f4ded682e9 Add SentToInputFpsRatioPercent UMA metric on send side 
New metric is a ratio between SentFramesPerSecond and
InputFramesPerSecond. It's needed to measure how much of input frames
are actually sent.

BUG=webrtc:8178

Review-Url: https://codereview.webrtc.org/3010723002
Cr-Commit-Position: refs/heads/master@{#19596}
2017-08-30 09:30:20 +00:00
..
BUILD.gn
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +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
Reverse |rtx_payload_types| map, and rename.
2017-08-25 11:44:25 +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 a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
overuse_frame_detector_unittest.cc
Fix places that trigger no-unused-lambda-capture
2017-08-23 11:15:18 +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
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
payload_router.h
Delete unneeded includes of atomic32.h.
2017-08-24 10:30:05 +00:00
picture_id_tests.cc
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +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
Report max interframe delay over window insdead of interframe delay sum
2017-08-23 12:24:10 +00:00
receive_statistics_proxy.cc
Report max interframe delay over window insdead of interframe delay sum
2017-08-23 12:24:10 +00:00
receive_statistics_proxy.h
Report max interframe delay over window insdead of interframe delay sum
2017-08-23 12:24:10 +00:00
replay.cc
Update jpeg writer to compile on iOS and document it better
2017-08-28 13:08:33 +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
Reverse |rtx_payload_types| map, and rename.
2017-08-25 11:44:25 +00:00
rtp_video_stream_receiver.h
Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
2017-08-25 07:49:08 +00:00
screenshare_loopback.cc
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +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
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +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
Add SentToInputFpsRatioPercent UMA metric on send side
2017-08-30 09:30:20 +00:00
send_statistics_proxy.h
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +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
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
video_quality_test.h
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +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
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
video_receive_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_send_stream_tests.cc
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
video_send_stream.cc
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
video_send_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +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
Reland of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:1 of https://codereview.webrtc.org/2995173002/ )
2017-08-18 07:28:40 +00:00
video_stream_encoder.cc
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
video_stream_encoder.h
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00