admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
ehmaldonado ed8c8ede5d Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps 
When set to true, this adds the files necessary to run memcheck as data dependencies, listed in the .gni files.
This will enable us to run memcheck on swarming.

R=kjellander@chromium.org
BUG=chromium:497757
NOTRY=True

Review-Url: https://codereview.webrtc.org/2510033004
Cr-Commit-Position: refs/heads/master@{#15219}
2016-11-23 20:58:45 +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
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
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
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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.