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ehmaldonado 464d50f293 Set rtc_use_memcheck=true for the FYI bot. 
R=kjellander@chromium.org
BUG=chromium:497757
NOTRY=True

Review-Url: https://codereview.webrtc.org/2529753002
Cr-Commit-Position: refs/heads/master@{#15220}
2016-11-23 21:19:30 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
For VPN network, use the underlying network type as its type.
2016-11-23 19:04:33 +00:00
audio
Replace AudioConferenceMixer with AudioMixer.
2016-11-22 14:42:59 +00:00
base
Fix TimeCallback used by BoringSSL.
2016-11-21 22:34:02 +00:00
build
Set rtc_use_memcheck=true for the FYI bot.
2016-11-23 21:19:30 +00:00
call
Start probes only after network is connected.
2016-11-23 00:08:37 +00:00
common_audio
Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )
2016-11-18 09:31:19 +00:00
common_video
Enable QP parsing from CABAC bitstreams
2016-11-22 14:53:39 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Start probes only after network is connected.
2016-11-23 00:08:37 +00:00
modules
Re-enables AudioDeviceTest.StartStopPlayout on Android
2016-11-23 14:49:48 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Let MediaSession generate a FlexFEC SSRC when FlexFEC is active.
2016-11-22 11:38:04 +00:00
sdk
Split avfoundationcapturer classes in separate files.
2016-11-23 08:42:06 +00:00
stats
RTCCodecStats[1] added.
2016-11-23 10:32:14 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps
2016-11-23 20:58:45 +00:00
tools
Remove all references to GYP
2016-11-16 19:11:38 +00:00
video
Try to deflake VideoSendStream tests with FlexFEC.
2016-11-23 12:42:31 +00:00
voice_engine
WebRTC: Replace ProjectRootPath by ResourcePath
2016-11-22 18:43:05 +00:00
.gitignore
…
BUILD.gn
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Unify VideoCodecType to/from string functionality
2016-11-22 18:17:04 +00:00
common_types.h
Unify VideoCodecType to/from string functionality
2016-11-22 18:17:04 +00:00
config.cc
Wire up FlexFEC in VideoEngine2.
2016-11-22 10:16:56 +00:00
config.h
Wire up FlexFEC in VideoEngine2.
2016-11-22 10:16:56 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.