851a09e71a
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 |
||
---|---|---|
.. | ||
base | ||
build | ||
common_audio | ||
common_video | ||
examples | ||
modules | ||
overrides/webrtc/base | ||
system_wrappers | ||
test | ||
tools | ||
video | ||
video_engine | ||
voice_engine | ||
.gitignore | ||
BUILD.gn | ||
call.h | ||
common_types.h | ||
common.gyp | ||
common.h | ||
config.cc | ||
config.h | ||
engine_configurations.h | ||
experiments.h | ||
frame_callback.h | ||
LICENSE | ||
LICENSE_THIRD_PARTY | ||
OWNERS | ||
PATENTS | ||
PRESUBMIT.py | ||
README.chromium | ||
supplement.gypi | ||
transport.h | ||
typedefs.h | ||
video_engine_tests.isolate | ||
video_receive_stream.h | ||
video_renderer.h | ||
video_send_stream.h | ||
webrtc_examples.gyp | ||
webrtc_perf_tests.isolate | ||
webrtc_tests.gypi | ||
webrtc.gyp |
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.