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Erik Språng 491bd534ef Disable some PortAllocatorTest on tsan due to flakiness 
TBR=kjellander@webrtc.org
BUG=4743

Review URL: https://codereview.webrtc.org/1160033005

Cr-Commit-Position: refs/heads/master@{#9369}
2015-06-04 12:57:59 +00:00
..
base
Remove win32toolhelp.h.
2015-06-04 08:25:12 +00:00
build
Roll chromium_revision ccef3cb..7779e7d (331232:332119)
2015-06-01 09:49:28 +00:00
common_audio
Add LappedTransform accessors.
2015-05-29 01:01:43 +00:00
common_video
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
2015-05-29 13:05:52 +00:00
modules
Change JitterBuffer::GetNackList to return a std::vector<uint16_t>.
2015-06-03 22:03:46 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Disable some PortAllocatorTest on tsan due to flakiness
2015-06-04 12:57:59 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Document the time unit in EventWrapper.
2015-06-02 00:33:51 +00:00
test
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
tools
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
video
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
video_engine
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
voice_engine
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
…
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_encoder.h
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Revert "Convert native handles to buffers before encoding."
2015-06-02 13:04:31 +00:00
video_receive_stream.h
Configure default render delay as 10 ms.
2015-05-27 15:59:21 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.