zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
9b1b4105a48b51020404b3b51d553c0047412b97
platform-external-webrtc/webrtc/test
History
eladalon c0d481a4a6 Protected streams report RTP messages directly to the FlexFec streams 
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
2017-08-02 14:39:07 +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
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
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
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
call_test.cc
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
call_test.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +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
Don't use the Force(Demuxer), Luke.
2017-08-02 13:18:41 +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 webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
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
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
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +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
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
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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
…
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
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_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