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webrtc_m130/webrtc/test
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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
..
android
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
fuzzers
Delete old CongestionController class
2017-08-21 10:43:33 +00:00
gl
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ios
Remove MAIN_NIB_FILE from Info.plist because the substitution is broken
2017-07-06 14:09:57 +00:00
linux
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
mac
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
testsupport
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
win
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
call_test.cc
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
call_test.h
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
configurable_frame_size_encoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
configurable_frame_size_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
constants.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
constants.h
Implement timing frames.
2017-06-19 14:18:55 +00:00
DEPS
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
direct_transport.cc
Remove deprecated DirectTransport constructors
2017-08-24 13:56:15 +00:00
direct_transport.h
Make DirectTransport::task_queue_ const
2017-08-29 13:53:21 +00:00
drifting_clock.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
drifting_clock.h
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
2017-03-07 10:51:09 +00:00
encoder_settings.cc
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
2017-02-21 15:28:31 +00:00
encoder_settings.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
fake_audio_device_unittest.cc
Low-bandwidth audio testing
2017-03-23 10:40:03 +00:00
fake_audio_device.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_audio_device.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_decoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_decoder.h
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
fake_encoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_encoder.h
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
fake_network_pipe_unittest.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
fake_network_pipe.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_network_pipe.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_texture_frame.cc
Update webrtc/test to new VideoFrameBuffer interface
2017-06-11 16:10:16 +00:00
fake_texture_frame.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_videorenderer.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
field_trial.cc
Reland "Add APK targets to build libjingle tests for Android."
2016-01-08 13:05:01 +00:00
field_trial.h
Reland "Add APK targets to build libjingle tests for Android."
2016-01-08 13:05:01 +00:00
frame_generator_capturer.cc
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
frame_generator_capturer.h
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
frame_generator_unittest.cc
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
frame_generator.cc
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
frame_generator.h
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
frame_utils.cc
Update webrtc/test to new VideoFrameBuffer interface
2017-06-11 16:10:16 +00:00
frame_utils.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
gmock.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
gtest.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
layer_filtering_transport.cc
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
layer_filtering_transport.h
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
mock_audio_decoder_factory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mock_audio_decoder.cc
Fix Chromium style checker warnings for MockAudioDecoder
2017-06-14 10:31:17 +00:00
mock_audio_decoder.h
Fix Chromium style checker warnings for MockAudioDecoder
2017-06-14 10:31:17 +00:00
mock_audio_encoder_factory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mock_audio_encoder.cc
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
2017-05-02 13:46:30 +00:00
mock_audio_encoder.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mock_transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
mock_voe_channel_proxy.h
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
mock_voice_engine.h
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
null_platform_renderer.cc
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
null_transport.cc
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
null_transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
rtcp_packet_parser.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_packet_parser.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_file_reader_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
rtp_file_reader.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_file_reader.h
Add pcap support to bwe tools. Allow filtering on SSRCs.
2015-07-17 12:27:27 +00:00
rtp_file_writer_unittest.cc
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
rtp_file_writer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_file_writer.h
Adding a new test helper RtpFileWriter and use it in RTPcat
2014-12-01 11:25:04 +00:00
rtp_rtcp_observer.h
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
run_loop.cc
Roll chromium_revision: 28d1981..d3db2ff
2014-11-03 20:10:26 +00:00
run_loop.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
run_test.cc
Adding back platform specific renderer to video loopback test.
2014-06-05 09:32:51 +00:00
run_test.h
Adding back platform specific renderer to video loopback test.
2014-06-05 09:32:51 +00:00
single_threaded_task_queue_unittest.cc
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
single_threaded_task_queue.cc
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
single_threaded_task_queue.h
Use SingleThreadedTaskQueue in DirectTransport
2017-08-22 11:02:52 +00:00
statistics.cc
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
statistics.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
test_main.cc
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
vcm_capturer.cc
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
vcm_capturer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_capturer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_capturer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_codec_settings.h
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video_renderer.cc
Remove a few meaningless TODOs
2016-08-03 13:38:20 +00:00
video_renderer.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00