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webrtc_m130/webrtc/video
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philipel 2c53b13a3b Request keyframe if the first received frame is not a keyframe. 
BUG=webrtc:7669

Review-Url: https://codereview.webrtc.org/2874933004
Cr-Commit-Position: refs/heads/master@{#18169}
2017-05-16 15:06:30 +00:00
..
BUILD.gn
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )
2017-04-26 10:38:35 +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
Make EndToEndTest.ContentTypeSwitches more robust by sending more frames before test ends
2017-04-18 16:17:53 +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
Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test.
2017-04-06 17:52:23 +00:00
overuse_frame_detector_unittest.cc
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
overuse_frame_detector.cc
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
overuse_frame_detector.h
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +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
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
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 some unit tests to ReceiveStatsticsProxy.
2017-05-11 12:37:35 +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_stream_receiver_unittest.cc
Request keyframe if the first received frame is not a keyframe.
2017-05-16 15:06:30 +00:00
rtp_stream_receiver.cc
Request keyframe if the first received frame is not a keyframe.
2017-05-16 15:06:30 +00:00
rtp_stream_receiver.h
Request keyframe if the first received frame is not a keyframe.
2017-05-16 15:06:30 +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
Add command-line param to screenshare_loopback to specify a list of slides
2017-04-12 11:52:55 +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
Adding audio DTX to video loopback test.
2017-03-21 11:11:43 +00:00
video_quality_test.cc
Fixing video loopback test with encoder factory.
2017-05-10 11:06:11 +00:00
video_quality_test.h
Add command-line param to screenshare_loopback to specify a list of slides
2017-04-12 11:52:55 +00:00
video_receive_stream_unittest.cc
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
video_receive_stream.cc
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
video_receive_stream.h
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
video_send_stream_tests.cc
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
video_send_stream.cc
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +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
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
vie_encoder.cc
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
vie_encoder.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00