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webrtc_m130/webrtc/video
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asapersson 02465b8a11 Add some unit tests to vie_encoder. 
BUG=none

Review-Url: https://codereview.webrtc.org/2801293002
Cr-Commit-Position: refs/heads/master@{#17609}
2017-04-10 08:12:52 +00:00
..
BUILD.gn
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
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
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
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
Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test.
2017-04-06 17:52:23 +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
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
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
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +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
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 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
Allow padding packet in video streams.
2017-03-21 12:45:18 +00:00
rtp_stream_receiver.cc
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +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
Update stats for cpu/quality adaptation changes to excluded time when video is suspended.
2017-04-07 07:57:58 +00:00
send_statistics_proxy.cc
Update stats for cpu/quality adaptation changes to excluded time when video is suspended.
2017-04-07 07:57:58 +00:00
send_statistics_proxy.h
Update stats for cpu/quality adaptation changes to excluded time when video is suspended.
2017-04-07 07:57:58 +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
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
video_receive_stream_unittest.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
video_receive_stream.cc
Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ )
2017-04-04 14:16:21 +00:00
video_receive_stream.h
Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ )
2017-04-04 14:16:21 +00:00
video_send_stream_tests.cc
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
video_send_stream.cc
Fixing some case-sensitive codec name comparisons.
2017-04-03 17:03:35 +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
Add some unit tests to vie_encoder.
2017-04-10 08:12:52 +00:00
vie_encoder.cc
Add some unit tests to vie_encoder.
2017-04-10 08:12:52 +00:00
vie_encoder.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
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