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webrtc_m130/webrtc/video
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Stefan Holmer 1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/. 
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 13:29:30 +00:00
..
BUILD.gn
Replace remaining gflags usages with rtc_base/flags
2017-08-31 10:21:39 +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
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +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
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +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 alr tests config
2017-08-31 11:46:45 +00:00
overuse_frame_detector_unittest.cc
Fix places that trigger no-unused-lambda-capture
2017-08-23 11:15:18 +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
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
payload_router.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
payload_router.h
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
picture_id_tests.cc
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +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
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
receive_statistics_proxy.cc
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
receive_statistics_proxy.h
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
replay.cc
Replace remaining gflags usages with rtc_base/flags
2017-08-31 10:21:39 +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
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
rtp_video_stream_receiver.h
Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
2017-08-25 07:49:08 +00:00
screenshare_loopback.cc
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +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
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +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
Add SentToInputFpsRatioPercent UMA metric on send side
2017-08-30 09:30:20 +00:00
send_statistics_proxy.h
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +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
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
video_quality_test.h
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +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
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
video_receive_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_send_stream_tests.cc
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
video_send_stream.cc
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
video_send_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_stream_decoder.cc
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
video_stream_decoder.h
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
video_stream_encoder_unittest.cc
Reland of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:1 of https://codereview.webrtc.org/2995173002/ )
2017-08-18 07:28:40 +00:00
video_stream_encoder.cc
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
video_stream_encoder.h
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00