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brettw@chromium.org 9fe11010f7 Fix the RTC+Chromium GN build. 
LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.

Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 19:15:33 +00:00
..
base
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
2014-09-09 15:44:05 +00:00
build
GN: Prefix WebRTC specific variables with "rtc_"
2014-09-07 17:36:10 +00:00
common_audio
Add ctors to ChannelBuffer to enable copying on construction.
2014-09-08 20:27:04 +00:00
common_video
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
examples
Remove former team members from OWNERS and WATCHLISTS
2014-08-26 06:12:08 +00:00
libjingle/xmllite
Fix strange owners files with comments that crashs "git cl presubmit"
2014-09-04 14:25:07 +00:00
modules
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
overrides/webrtc/base
Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
2014-05-21 16:52:14 +00:00
sound
Add talk owners to migrated talk folders
2014-08-28 16:03:58 +00:00
system_wrappers
GN: Prefix WebRTC specific variables with "rtc_"
2014-09-07 17:36:10 +00:00
test
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
tools
Bot Browser files moved to /bot/browser/
2014-09-09 14:50:09 +00:00
video
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_engine
Fix RTT calculations for send-only channels.
2014-09-08 08:45:25 +00:00
voice_engine
Partial revert of r7014 (Android APK refactor)
2014-09-03 14:34:46 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Fix the RTC+Chromium GN build.
2014-09-09 19:15:33 +00:00
call.h
Network up/down signaling in Call.
2014-09-03 16:17:12 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Check before send/receive rtp header extensions.
2014-07-20 15:27:35 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
webrtc_examples.gyp
Fix rm command for class cleanup in r7091
2014-09-06 22:11:28 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Partial revert of r7014 (Android APK refactor)
2014-09-03 14:34:46 +00:00
webrtc.gyp
Create a copy of talk/xmllite under webrtc/xmllite.
2014-09-02 15:41:12 +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.