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platform-external-webrtc/webrtc/test
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michaelt e0729c56d3 Add RtcpRttStats to AudioStream 
BUG=webrtc:6508

Review-Url: https://codereview.webrtc.org/2402333002
Cr-Commit-Position: refs/heads/master@{#14595}
2016-10-11 07:29:34 +00:00
..
fuzzers
Harmonize using-declarations in FEC header formatter fuzzers.
2016-10-05 14:35:01 +00:00
gl
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
ios
Add custom info.plist to modules_unittests
2016-09-22 15:00:57 +00:00
linux
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
mac
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
testsupport
Fix modules_unittests on iOS.
2016-10-03 11:13:33 +00:00
win
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
BUILD.gn
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
call_test.cc
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
call_test.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
configurable_frame_size_encoder.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
configurable_frame_size_encoder.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
constants.cc
Register header extensions in RtpRtcpObserver to avoid log spam.
2015-10-29 14:13:35 +00:00
constants.h
Register header extensions in RtpRtcpObserver to avoid log spam.
2015-10-29 14:13:35 +00:00
DEPS
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
direct_transport.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
direct_transport.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
drifting_clock.cc
Added A/V sync tests with drifting clocks.
2016-02-10 18:54:52 +00:00
drifting_clock.h
Added A/V sync tests with drifting clocks.
2016-02-10 18:54:52 +00:00
encoder_settings.cc
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
encoder_settings.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
fake_audio_device.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
fake_audio_device.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
fake_decoder.cc
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
fake_decoder.h
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
fake_encoder.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
fake_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
fake_network_pipe_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
fake_network_pipe.cc
The current default schedule delay of 30 ms prohibits
2016-09-29 09:02:16 +00:00
fake_network_pipe.h
Remove a few meaningless TODOs
2016-08-03 13:38:20 +00:00
fake_texture_frame.cc
Move fake-handle frame creation into test target.
2016-01-04 21:36:49 +00:00
fake_texture_frame.h
Reland of Update test code to use I420Buffer when writing pixel data. (patchset #1 id:1 of https://codereview.webrtc.org/2343083002/ )
2016-09-19 07:34:52 +00:00
fake_videorenderer.h
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +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
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
frame_generator_capturer.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
frame_generator_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
frame_generator.cc
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
frame_generator.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
frame_utils.cc
New helper function test::ReadI420Buffer, refactor FrameReader to use it.
2016-09-30 11:14:11 +00:00
frame_utils.h
New helper function test::ReadI420Buffer, refactor FrameReader to use it.
2016-09-30 11:14:11 +00:00
gmock.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
gtest.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
layer_filtering_transport.cc
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
layer_filtering_transport.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
mock_transport.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
mock_voe_channel_proxy.h
Add RtcpRttStats to AudioStream
2016-10-11 07:29:34 +00:00
mock_voice_engine.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
null_platform_renderer.cc
…
null_transport.cc
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
null_transport.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
OWNERS
Add stefan@webrtc.org to webrtc/test/OWNERS.
2016-01-21 14:33:03 +00:00
rtcp_packet_parser.cc
Test RtcpParser rewritten to use rtcp packet classes
2016-09-02 16:29:24 +00:00
rtcp_packet_parser.h
Test RtcpParser rewritten to use rtcp packet classes
2016-09-02 16:29:24 +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
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +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
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
run_all_unittests.cc
…
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
…
run_test.h
…
statistics.cc
…
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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
test_suite.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
test_suite.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
test.gyp
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
vcm_capturer.cc
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
vcm_capturer.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
video_capturer.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +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