webrtc/webrtc
kjellander@webrtc.org 851a09e71a Initial GN work for WebRTC
This CL makes it possible to build the 'webrtc_base'
target using GN.
The majority of our GYP stuff in webrtc/build/common.gypi has been
translated into the configs of webrtc/BUILD.gn.
The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn.

This CL depends on https://codereview.chromium.org/322373002/ for the
jsoncpp BUILD.gn file and the ssl config.
To build inside Chromium, https://codereview.chromium.org/321313006/
needs to be landed first.

BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default
I also ran:
gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true"
but it fails to compile: something is probably wrong with with pkg-config for that.

For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the
WebRTC checkout and applied the following patches:
https://codereview.chromium.org/322373002 (for jsoncpp and ssl config)
https://codereview.chromium.org/321313006 (enable building WebRTC)
Then I built successfully using:
gn gen out/Default && ninja -C out/Default webrtc_base

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:54:03 +00:00
..
base Initial GN work for WebRTC 2014-06-17 08:54:03 +00:00
build Initial GN work for WebRTC 2014-06-17 08:54:03 +00:00
common_audio common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 2014-06-16 10:30:14 +00:00
common_video Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
examples AppRTCDemo(android): support app (UI) & capture rotation. 2014-06-06 18:40:44 +00:00
modules Restore ptypes.txt file 2014-06-17 08:51:01 +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 Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
test Updated W3C getusermedia tests to the latest version of the spec. 2014-06-17 08:46:58 +00:00
tools Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video Implements start bitrate for new video API. 2014-06-16 08:57:39 +00:00
video_engine Add max limit of number for overuses. When limit is reached always apply the rampup delay. 2014-06-16 14:27:19 +00:00
voice_engine Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn Initial GN work for WebRTC 2014-06-17 08:54:03 +00:00
call.h Implements start bitrate for new video API. 2014-06-16 08:57:39 +00:00
common_types.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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 Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-05-12 12:19:19 +00:00
experiments.h Adding API for setting bandwidth estimation configurations. 2014-03-25 10:37:31 +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 Add kjellander@webrtc.org as OWNER for *.isolate 2014-06-10 05:42:53 +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 Roll chromium_revision 260462:266514 2014-04-29 09:36:40 +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 Rename Start/Stop in Video{Send,Receive}Streams. 2014-04-24 11:13:21 +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 Enable pacing by default and remove the option to disable it from the new API. 2014-06-12 15:12:25 +00:00
webrtc_examples.gyp Add webrtc field trials API. 2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Rename neteq4 folder to neteq 2014-06-09 08:10:28 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00

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.