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webrtc_m130/webrtc/video
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nisse b8f9a32459 Define RtpTransportControllerSendInterface. 
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.

BUG=webrtc:6847, webrtc:7135

Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
2017-03-27 12:36:15 +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 #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +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 #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
overuse_frame_detector.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +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
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
receive_statistics_proxy.cc
Increase kMinRequiredSamples (5 -> 200) for updating histogram stats from OnFrameBufferTimingsUpdated callback.
2017-03-02 16:25:46 +00:00
receive_statistics_proxy.h
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
replay.cc
replay: output rtp header elements for errors
2017-01-26 12:54:04 +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
Allow padding packet in video streams.
2017-03-21 12:45:18 +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
Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call.
2017-02-06 13:18:35 +00:00
send_statistics_proxy.cc
Add traces for some video events.
2017-03-15 17:45:49 +00:00
send_statistics_proxy.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +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
Adding audio DTX to video loopback test.
2017-03-21 11:11:43 +00:00
video_quality_test.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +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
Roll chromium_revision 22b81f6c45..d604de96c9 (459032:459307)
2017-03-24 19:17:33 +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
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
vie_encoder.cc
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
vie_encoder.h
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +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 all use of tick_util.h.
2016-05-10 14:31:58 +00:00
vie_remb.h
Replace scoped_ptr with unique_ptr in webrtc/video/
2016-03-01 19:52:39 +00:00