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webrtc_m130/webrtc/video
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Niels Möller 2bf9e73e6b Delete unneeded Start and Stop methods on FlexfecReceiveStream. 
Bug: None
Change-Id: I3013cfc54ed357901f175dd408127eda75e5ba99
Reviewed-on: https://chromium-review.googlesource.com/542735
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19363}
2017-08-16 09:41:27 +00:00
..
BUILD.gn
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +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
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
encoder_rtcp_feedback.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
encoder_rtcp_feedback.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
end_to_end_tests.cc
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
2017-08-14 15:03:17 +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
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +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
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +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
Fix incorrect InterframeDelayMaxInMs histogram metrics
2017-08-15 12:00:33 +00:00
receive_statistics_proxy.cc
Fix incorrect InterframeDelayMaxInMs histogram metrics
2017-08-15 12:00:33 +00:00
receive_statistics_proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
replay.cc
Fix video_replay tool to respect RTX stream and fix default parameters.
2017-07-11 09:38:36 +00:00
report_block_stats_unittest.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
report_block_stats.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +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
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
rtp_video_stream_receiver.cc
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +00:00
rtp_video_stream_receiver.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
screenshare_loopback.cc
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +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
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
send_statistics_proxy.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
send_statistics_proxy.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +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
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
video_quality_test.cc
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
video_quality_test.h
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +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
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
2017-08-11 10:41:44 +00:00
video_receive_stream.h
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
2017-08-11 10:41:44 +00:00
video_send_stream_tests.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
video_send_stream.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
video_send_stream.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +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
video_stream_encoder_unittest.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
video_stream_encoder.cc
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
video_stream_encoder.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00