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webrtc_m130/webrtc/test
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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
..
android
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
fuzzers
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +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
…
mac
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
testsupport
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
win
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
call_test.cc
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
call_test.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +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
Add the ability to read/write to WAV files in FakeAudioDevice
2017-03-14 16:01:47 +00:00
direct_transport.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
direct_transport.h
Mark ~DirectTransport with "override."
2017-08-03 14:44:08 +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
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +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
…
field_trial.h
…
frame_generator_capturer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
frame_generator_capturer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
frame_generator_unittest.cc
Update webrtc/test to new VideoFrameBuffer interface
2017-06-11 16:10:16 +00:00
frame_generator.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
frame_generator.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
layer_filtering_transport.h
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +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
…
null_transport.cc
…
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
…
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
…
rtp_rtcp_observer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
run_loop.cc
…
run_loop.h
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run_test.cc
…
run_test.h
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statistics.cc
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statistics.h
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test_main.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
vcm_capturer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
…