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webrtc/webrtc
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kjellander@webrtc.org 9bef551ba1 GN: Fix include paths for WebRTC in Chromium build. 
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.

This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.

However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.

BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-13 09:02:54 +00:00
..
base Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 2014-07-11 19:09:59 +00:00
build Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +00:00
common_audio common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 2014-07-03 13:38:53 +00:00
common_video GN: Implement BUILD.gn for common_video. 2014-07-03 17:04:12 +00:00
examples WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
modules GN: Fix include paths for WebRTC in Chromium build. 2014-07-13 09:02:54 +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
system_wrappers Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 2014-07-11 19:09:59 +00:00
test Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 2014-07-10 10:35:12 +00:00
tools Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +00:00
video_engine Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 2014-07-11 19:09:59 +00:00
voice_engine Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 2014-07-11 19:09:59 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Fix include paths for WebRTC in Chromium build. 2014-07-13 09:02:54 +00:00
call.h Remove GetDefaultConfigs() from Call. 2014-07-07 04:45:15 +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 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +00:00
engine_configurations.h Add boilerplate code for H.264. 2014-07-04 12:42:07 +00:00
experiments.h Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 2014-07-08 13:59:46 +00:00
frame_callback.h Wire up statistics in video receive stream of new API 2014-02-07 12:06:29 +00:00
LICENSE Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
supplement.gypi TSan: Move suppressions to source file. 2014-06-27 09:18:51 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_receive_stream.h Remove GetDefaultConfigs() from Call. 2014-07-07 04:45:15 +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 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 2014-07-08 13:59:46 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.