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webrtc_m130/webrtc/video
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philipel 8054c7ecf8 EndToEndTest.ReceivesAndRetransmitsNack now handles reordering. 
Since packets can be received out of order NACKs could be triggered before any
packet was actually dropped. This cause the test to never drop packets which in
turn caused the block of code which set the |observation_complete_| event to
never execute.

BUG=webrtc:2845

Review-Url: https://codereview.webrtc.org/2613443002
Cr-Commit-Position: refs/heads/master@{#15990}
2017-01-10 13:19:19 +00:00
..
BUILD.gn
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
2017-01-05 15:09:50 +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
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
encoder_rtcp_feedback.cc
Remove OnLocalSsrcChanged and rename EncoderStateFeedback.
2016-10-06 15:35:19 +00:00
encoder_rtcp_feedback.h
Remove OnLocalSsrcChanged and rename EncoderStateFeedback.
2016-10-06 15:35:19 +00:00
end_to_end_tests.cc
EndToEndTest.ReceivesAndRetransmitsNack now handles reordering.
2017-01-10 13:19:19 +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
Add full stack tests:
2016-12-16 14:54:01 +00:00
overuse_frame_detector_unittest.cc
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
overuse_frame_detector.cc
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
overuse_frame_detector.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +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
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
quality_threshold.cc
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
quality_threshold.h
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
receive_statistics_proxy_unittest.cc
ReceiveStatisticsProxy:
2017-01-03 08:44:06 +00:00
receive_statistics_proxy.cc
Lower bad call logging severity
2017-01-09 15:23:33 +00:00
receive_statistics_proxy.h
ReceiveStatisticsProxy:
2017-01-03 08:44:06 +00:00
replay.cc
Add back video_replay. Disappeared in the gn conversion.
2016-12-20 15:26:58 +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.cc
Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
2016-12-20 12:15:59 +00:00
rtp_stream_receiver.h
Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
2016-12-20 12:15:59 +00:00
rtp_streams_synchronizer.cc
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
rtp_streams_synchronizer.h
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
screenshare_loopback.cc
Configure FlexFEC in VideoQualityTest.
2016-11-17 06:47:36 +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
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
send_statistics_proxy.cc
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
send_statistics_proxy.h
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
stats_counter_unittest.cc
Add ability to handle data from multiple streams in RateAccCounter.
2016-11-01 07:21:41 +00:00
stats_counter.cc
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +00:00
stats_counter.h
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +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
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
stream_synchronization.h
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +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
Integrate FlexFEC in video_loopback.
2016-11-17 08:21:19 +00:00
video_quality_test.cc
Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
2016-12-21 14:37:18 +00:00
video_quality_test.h
Configure FlexFEC in VideoQualityTest.
2016-11-17 06:47:36 +00:00
video_receive_stream.cc
Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
2016-12-20 12:15:59 +00:00
video_receive_stream.h
Create VideoReceiver with external VCMTiming object.
2016-12-15 15:11:01 +00:00
video_send_stream_tests.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
video_send_stream.cc
Simplify an always true condition.
2016-12-08 14:24:35 +00:00
video_send_stream.h
Move ownership of PacketRouter from CongestionController to Call.
2016-11-30 11:35:28 +00:00
video_stream_decoder.cc
Delete all of the video_processing module but the denoiser code.
2016-11-30 15:44:59 +00:00
video_stream_decoder.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
vie_encoder_unittest.cc
Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
2016-12-09 11:54:08 +00:00
vie_encoder.cc
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +00:00
vie_encoder.h
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +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