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author | kjellander@webrtc.org <kjellander@webrtc.org> | 2014-09-02 11:22:06 +0000 |
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committer | kjellander@webrtc.org <kjellander@webrtc.org> | 2014-09-02 11:22:06 +0000 |
commit | 732401a298870c3c9e1935bd0565735bc6c61694 (patch) | |
tree | 6f1d09667134bdc01dae5f755aac097967229dd7 /build | |
parent | 3dbd81358e71cc749fb730b27ccfb73a79aeff88 (diff) | |
download | webrtc-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.gni | 8 |
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. |