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authorkjellander@webrtc.org <kjellander@webrtc.org>2014-09-02 11:22:06 +0000
committerkjellander@webrtc.org <kjellander@webrtc.org>2014-09-02 11:22:06 +0000
commit732401a298870c3c9e1935bd0565735bc6c61694 (patch)
tree6f1d09667134bdc01dae5f755aac097967229dd7 /build
parent3dbd81358e71cc749fb730b27ccfb73a79aeff88 (diff)
downloadwebrtc-732401a298870c3c9e1935bd0565735bc6c61694.tar.gz
GN: Update webrtc/base to recent GYP changes.
Update the webrtc/base/BUILD.gn file to reflect webrtc/base/base.gyp changes between r6438 and r7011. BUG=3441 TESTED= Trybots + compilation with a standalone WebRTC checkout: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/13359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7022 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'build')
-rw-r--r--build/webrtc.gni8
1 files changed, 8 insertions, 0 deletions
diff --git a/build/webrtc.gni b/build/webrtc.gni
index cae81664..346a0622 100644
--- a/build/webrtc.gni
+++ b/build/webrtc.gni
@@ -17,6 +17,14 @@ declare_args() {
# Disable this to avoid building the Opus audio codec.
include_opus = true
+ # Used to specify an external Jsoncpp include path when not compiling the
+ # library that comes with WebRTC (i.e. build_json == 0).
+ webrtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
+
+ # Used to specify an external OpenSSL include path when not compiling the
+ # library that comes with WebRTC (i.e. build_ssl == 0).
+ webrtc_ssl_root = ""
+
# Adds video support to dependencies shared by voice and video engine.
# This should normally be enabled; the intended use is to disable only
# when building voice engine exclusively.