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ilnik 78f2d21e74 Fixed cpu time unittest to be less flaky 
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2725553002
Cr-Commit-Position: refs/heads/master@{#16897}
2017-02-28 10:24:10 +00:00
..
api
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
audio
Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.
2017-02-26 12:18:12 +00:00
base
Fixed cpu time unittest to be less flaky
2017-02-28 10:24:10 +00:00
call
Enable GN check for webrtc/call
2017-02-27 23:57:45 +00:00
common_audio
Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.
2017-02-26 12:18:12 +00:00
common_video
Enable GN check for webrtc/common_video
2017-02-28 00:37:54 +00:00
examples
iOS AppRTCMobile: Fix SocketRocket bug
2017-02-27 13:20:18 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Handle TimeUntilNextProcess in StartupShutdownWithExternalADM
2017-02-28 10:12:57 +00:00
modules
Revert of Move fake_audio_device to its own target. (patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ )
2017-02-28 08:56:28 +00:00
ortc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
p2p
RTCIceCandidatePairStats.nominated collected.
2017-02-27 09:38:08 +00:00
pc
RTCIceCandidatePairStats.nominated collected.
2017-02-27 09:38:08 +00:00
sdk
Revert of Add the OnAddTrack callback for Objective-C wrapper. (patchset #7 id:240001 of https://codereview.webrtc.org/2513063003/ )
2017-02-27 15:04:25 +00:00
stats
Enable GN check in webrtc/stats
2017-02-27 23:53:39 +00:00
system_wrappers
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
test
Make FakeNetworkPipe not busy loop any more.
2017-02-28 10:19:33 +00:00
tools
Add probe logging to RtcEventLog.
2017-02-27 10:18:46 +00:00
video
Create unit test for VideoReceiveStream.
2017-02-28 09:33:09 +00:00
voice_engine
Simplify webrtc::voe::MonitorModule and remove the .cc file.
2017-02-28 09:16:48 +00:00
.gitignore
…
BUILD.gn
Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )
2017-02-26 15:12:50 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.