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webrtc_m130/webrtc/video
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sprang d2702ef110 Fix flaky test VideoSendStreamTest.SendsKeepAlive 
Since the keep-alive payload type is not registered in the payload type
map of FakeNetworkPipe, it will cause a DCHECK to trigger unless we're
able to destroy the call before that.

Just register it in the fake network as media type "any", it will be
discarded early on the receive side anyway.

BUG=webrt:7964

Review-Url: https://codereview.webrtc.org/2979543002
Cr-Commit-Position: refs/heads/master@{#18953}
2017-07-10 15:41:10 +00:00
..
BUILD.gn
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +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
Set overuse detector max frame interval based on target frame rate.
2017-06-15 11:21:07 +00:00
encoder_rtcp_feedback.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
encoder_rtcp_feedback.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
end_to_end_tests.cc
Remove webrtc::VideoEncoderFactory
2017-07-10 10:26:36 +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
Fix issue with zero rtt reports when using FlexFEC and add perf test.
2017-07-05 10:03:02 +00:00
overuse_frame_detector_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
payload_router.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
picture_id_tests.cc
Remove webrtc::VideoEncoderFactory
2017-07-10 10:26:36 +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 interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
receive_statistics_proxy.cc
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
receive_statistics_proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
replay.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_video_stream_receiver.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_video_stream_receiver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
screenshare_loopback.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
send_statistics_proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_quality_test.cc
Remove webrtc::VideoEncoderFactory
2017-07-10 10:26:36 +00:00
video_quality_test.h
Remove webrtc::VideoEncoderFactory
2017-07-10 10:26:36 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_receive_stream.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_send_stream_tests.cc
Fix flaky test VideoSendStreamTest.SendsKeepAlive
2017-07-10 15:41:10 +00:00
video_send_stream.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_send_stream.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
vie_encoder_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
vie_encoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
vie_encoder.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00