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platform-external-webrtc/webrtc/test
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brandtr b29e652b10 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )" 
Problem fixed: RTP header extensions were not properly set in tests.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
2016-12-21 14:37:18 +00:00
..
fuzzers
Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
2016-12-21 14:37:18 +00:00
gl
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
ios
Roll chromium_revision 9b5bb47fa0..04e7c673d9 (426837:427632)
2016-10-26 20:38:10 +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
Add WriteIsolatedOutput() functions
2016-12-14 23:03:11 +00:00
win
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
BUILD.gn
Move tools/valgrind-webrtc -> tools-webrtc/valgrind
2016-12-17 20:21:39 +00:00
call_test.cc
Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
2016-12-21 14:37:18 +00:00
call_test.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
configurable_frame_size_encoder.cc
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
configurable_frame_size_encoder.h
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
constants.cc
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
constants.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
direct_transport.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
direct_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
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
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +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
Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
2016-12-19 13:47:28 +00:00
fake_encoder.h
Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
2016-12-19 13:47:28 +00:00
fake_network_pipe_unittest.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
fake_network_pipe.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
frame_generator_capturer.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +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
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
frame_generator.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +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
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
layer_filtering_transport.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
Reland "Update rtt on audio only calls".
2016-11-30 15:51:19 +00:00
mock_voice_engine.h
Added an empty AudioTransportProxy to AudioState.
2016-11-17 14:29:05 +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
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
Replace RTCPUtility RtcpParser with Test RtcpParser
2016-11-02 15:22:04 +00:00
rtcp_packet_parser.h
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +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
Add back video_replay. Disappeared in the gn conversion.
2016-12-20 15:26:58 +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_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
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
vcm_capturer.cc
Replace VideoCaptureDataCallback by VideoSinkInterface.
2016-12-12 08:23:05 +00:00
vcm_capturer.h
Replace VideoCaptureDataCallback by VideoSinkInterface.
2016-12-12 08:23:05 +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