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Jianjun Zhu 037f3e42f2 Replace absolute path with relative path for GN files. 
Bug: webrtc:7952
Change-Id: I45d889bd976f58386f803d0dc27147ea00a52e56
Reviewed-on: https://chromium-review.googlesource.com/612786
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19357}
2017-08-15 15:57:36 +00:00
..
api
Revert of L16 implementation of the Audio{En,De}coderFactoryTemplate APIs (patchset #5 id:80001 of https://codereview.webrtc.org/2995523002/ )
2017-08-15 14:50:11 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
2017-08-14 15:03:17 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Add a flags field to video timing extension.
2017-08-15 12:33:27 +00:00
examples
AppRTCMobile: Use high resolution app icon on retina screens.
2017-08-15 11:07:12 +00:00
logging
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
media
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
modules
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
pc
Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )
2017-08-15 06:48:03 +00:00
rtc_base
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
rtc_tools
Improved UI for event_log_analyzer tool
2017-08-15 09:04:02 +00:00
sdk
Add a flags field to video timing extension.
2017-08-15 12:33:27 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +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.