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webrtc_m130/webrtc/test
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philipel 19f51434e8 Keep track of the capacity delay error in the FakeNetworkPipe. 
The FakeNetworkPipe use millisecond precision to calculate the delay induced by
the size of the packet being sent. The problem is that it rounds the delay down
to the closest millisecond which can cause a significant error in the actual
throughput. We keep track of that error to compensate the delay cause by
subsequent packets.

BUG=None

Review-Url: https://codereview.webrtc.org/3010653002
Cr-Commit-Position: refs/heads/master@{#19732}
2017-09-07 16:08:50 +00:00
..
android
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
fuzzers
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +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
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
win
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +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
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
fake_decoder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fake_decoder.h
…
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
Keep track of the capacity delay error in the FakeNetworkPipe.
2017-09-07 16:08:50 +00:00
fake_network_pipe.h
Keep track of the capacity delay error in the FakeNetworkPipe.
2017-09-07 16:08:50 +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
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
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +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
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
rtp_file_reader_unittest.cc
…
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
…
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
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
run_loop.cc
…
run_loop.h
…
run_test.cc
…
run_test.h
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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
…
statistics.h
…
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
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
video_codec_settings.h
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video_renderer.cc
…
video_renderer.h
…