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webrtc_m130/webrtc/test
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mbonadei 9452c62f28 Hiding problematic targets from chromium 
Targets that depend on //third_party/gflags are problematic in
chromium because it does not use gflags and if a target depends on
any target in //webrtc/test/BUILD.gn then GN will try to parse the
BUILD.gn file. Gn is eager in this phase of the build files generation
and if a target is not buildable it will complain.

In chromium obviously all the targets that depend on
//third_party/gflags are not buildable.

To be able to enable 'gn check' on the webrtc/base and
webrtc/test directories we have to hide them from chromium.

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2845063002
Cr-Commit-Position: refs/heads/master@{#17913}
2017-04-27 19:29:29 +00:00
..
fuzzers
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ )
2017-04-26 07:38:48 +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
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
testsupport
FileUtilsTest DirExists function added
2017-03-15 13:04:59 +00:00
win
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
BUILD.gn
Hiding problematic targets from chromium
2017-04-27 19:29:29 +00:00
call_test.cc
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
call_test.h
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +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 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
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
constants.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
drifting_clock.cc
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
2017-03-07 10:51:09 +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
Low-bandwidth audio testing
2017-03-23 10:40:03 +00:00
fake_audio_device.h
Low-bandwidth audio testing
2017-03-23 10:40:03 +00:00
fake_decoder.cc
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
fake_decoder.h
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
fake_encoder.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
fake_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
fake_network_pipe_unittest.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
fake_network_pipe.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
fake_network_pipe.h
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
frame_generator_capturer.h
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
frame_generator_unittest.cc
Delete VideoFrame::set_render_time_ms.
2017-01-30 10:43:18 +00:00
frame_generator.cc
Reduce CPU usage in test::FrameGenerator::SquareGenerator.
2017-03-22 11:57:53 +00:00
frame_generator.h
Replace test::FrameGenerator::ChromaGenerator with new FrameGenerator::SquareGenerator The problem with the ChromaGenerator is that the VP8 encoder produce a too low bitrate for each frame. The squaregenerator make the VP8 encoder produce about 600kbit/s at VGA.
2017-02-27 14:52:10 +00:00
frame_utils.cc
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
frame_utils.h
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +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
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
2017-04-10 12:15:48 +00:00
mock_audio_decoder.h
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
2017-04-10 12:15:48 +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
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
mock_voice_engine.h
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
2017-03-27 14:15:49 +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
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
rtcp_packet_parser.h
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +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
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 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
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
vcm_capturer.h
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
video_capturer.cc
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
video_capturer.h
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +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