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platform-external-webrtc/webrtc/video
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asapersson 36e9eb4773 Do not report quality limited resolution stats when quality scaler is disabled. 
WebRTC.Video.QualityLimitedResolutionInPercent is reported as zero for calls when the quality scaler is off (and the degradation preference allows scaling).

Update SetResolutionRestrictionStats to specify if quality scaler is enabled.

BUG=webrtc:7432

Review-Url: https://codereview.webrtc.org/2783213002
Cr-Commit-Position: refs/heads/master@{#17487}
2017-03-31 12:29:12 +00:00
..
BUILD.gn
Low-bandwidth audio testing
2017-03-23 10:40:03 +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
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
encoder_rtcp_feedback_unittest.cc
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +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
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
full_stack_tests_plot.py
Rename full_stack.cc to full_stack_tests.cc.
2016-12-01 08:20:24 +00:00
full_stack_tests.cc
Disable FullStackTest.ScreenshareSlidesVP9_2SL on all platforms
2017-03-24 19:55:13 +00:00
overuse_frame_detector_unittest.cc
Rename adaptation api methods, extended vie_encoder unit test.
2017-02-01 16:38:12 +00:00
overuse_frame_detector.cc
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
overuse_frame_detector.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +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
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +00:00
payload_router.h
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +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
Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets.
2017-03-28 09:44:58 +00:00
receive_statistics_proxy.cc
Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets.
2017-03-28 09:44:58 +00:00
receive_statistics_proxy.h
Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets.
2017-03-28 09:44:58 +00:00
replay.cc
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +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_stream_receiver_unittest.cc
Allow padding packet in video streams.
2017-03-21 12:45:18 +00:00
rtp_stream_receiver.cc
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
rtp_stream_receiver.h
Delete support for sending RTCP RPSI and SLI messages.
2017-03-14 15:54:43 +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
screenshare_loopback.cc
Adding audio DTX to video loopback test.
2017-03-21 11:11:43 +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
Do not report quality limited resolution stats when quality scaler is disabled.
2017-03-31 12:29:12 +00:00
send_statistics_proxy.cc
Do not report quality limited resolution stats when quality scaler is disabled.
2017-03-31 12:29:12 +00:00
send_statistics_proxy.h
Do not report quality limited resolution stats when quality scaler is disabled.
2017-03-31 12:29:12 +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
Adding audio DTX to video loopback test.
2017-03-21 11:11:43 +00:00
video_quality_test.cc
Add back sender_time and receiver_time metrics to full stack tests.
2017-03-30 08:02:15 +00:00
video_quality_test.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
video_receive_stream_unittest.cc
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_receive_stream.cc
Revert of Don't set the priority of the decoder to 'high' on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2745813003/ )
2017-03-17 18:54:25 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream_tests.cc
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
video_send_stream.cc
Define RtpTransportControllerSendInterface.
2017-03-27 12:36:15 +00:00
video_send_stream.h
Define RtpTransportControllerSendInterface.
2017-03-27 12:36:15 +00:00
video_stream_decoder.cc
Delete unsupported method VideoCodingModule::RegisterDecoderTimingCallback.
2017-03-17 12:35:43 +00:00
video_stream_decoder.h
Delete unsupported method VideoCodingModule::RegisterDecoderTimingCallback.
2017-03-17 12:35:43 +00:00
vie_encoder_unittest.cc
Do not report quality limited resolution stats when quality scaler is disabled.
2017-03-31 12:29:12 +00:00
vie_encoder.cc
Do not report quality limited resolution stats when quality scaler is disabled.
2017-03-31 12:29:12 +00:00
vie_encoder.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
vie_remb_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
vie_remb.cc
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
vie_remb.h
Replace scoped_ptr with unique_ptr in webrtc/video/
2016-03-01 19:52:39 +00:00