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webrtc_m130/webrtc/video
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sprang 1168fd4ed5 What can't loopback test be more like full stack test? 
It can; this CL makes it a lot closer, if not all the way to a merge.
Performance from video_loopback and screenshare_loopback should now
match what we're seeing in FullStackTest, which will make debugging and
assesment of quality differences much easier.

It also adds the ability to view all of the simulcast streams at once,
in separate windows.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2946893003
Cr-Commit-Position: refs/heads/master@{#18703}
2017-06-21 16:00:17 +00:00
..
BUILD.gn
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +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
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
call_stats.h
Replace scoped_ptr with unique_ptr in webrtc/video/
2016-03-01 19:52:39 +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
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
encoder_rtcp_feedback.h
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
end_to_end_tests.cc
Reland of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #1 id:1 of https://codereview.webrtc.org/2922723002/ )
2017-06-08 13:18:53 +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 cropping to VIEEncoder to match simulcast streams resolution
2017-06-16 13:53:48 +00:00
overuse_frame_detector_unittest.cc
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
overuse_frame_detector.cc
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
overuse_frame_detector.h
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
OWNERS
Adding Åsa and Erik as video owners.
2016-12-16 10:12:31 +00:00
payload_router_unittest.cc
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +00:00
payload_router.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
payload_router.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
quality_threshold_unittest.cc
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +00:00
quality_threshold.cc
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +00:00
quality_threshold.h
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +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 traces for some video receive statistics.
2017-05-23 16:06:13 +00:00
receive_statistics_proxy.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
replay.cc
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
rtp_streams_synchronizer.h
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
rtp_video_stream_receiver_unittest.cc
Only append SPS/PPS to bitstream if supplied out of band.
2017-06-21 14:22:40 +00:00
rtp_video_stream_receiver.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
rtp_video_stream_receiver.h
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
screenshare_loopback.cc
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +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
Add ToString method to AggregatedStats and log stats at the end of a call.
2016-11-15 16:20:54 +00:00
send_delay_stats.h
Start integrating StatsCounter class.
2016-06-09 07:09:27 +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
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
send_statistics_proxy.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +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
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.h
Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call.
2017-02-06 13:18:35 +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
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
stream_synchronization.h
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
transport_adapter.cc
Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency
2016-11-03 11:21:47 +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
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +00:00
video_quality_test.cc
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +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
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
video_receive_stream.cc
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
video_receive_stream.h
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
video_send_stream_tests.cc
Fix timing frames and FEC conflict
2017-06-21 15:23:19 +00:00
video_send_stream.cc
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
video_send_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_stream_decoder.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
video_stream_decoder.h
Remove deprecated API
2017-05-08 09:59:38 +00:00
vie_encoder_unittest.cc
Add cropping to VIEEncoder to match simulcast streams resolution
2017-06-16 13:53:48 +00:00
vie_encoder.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
vie_encoder.h
Add cropping to VIEEncoder to match simulcast streams resolution
2017-06-16 13:53:48 +00:00