Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(58)

Unified Diff: webrtc/build/webrtc.gni

Issue 1979933002: Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add externalhmac.{h,cc} files for Chromium build Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/api/BUILD.gn ('k') | webrtc/libjingle/xmllite/BUILD.gn » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/build/webrtc.gni
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 788e60824c592fb38b02886ccfe3235eec09c76c..5c50ab26ee26281d8b5c164789a27b0053d4ab46 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -15,6 +15,9 @@ declare_args() {
# Disable this to avoid building the Opus audio codec.
rtc_include_opus = true
+ # Disable to use absolute header paths for some libraries.
+ rtc_relative_path = true
+
# Used to specify an external Jsoncpp include path when not compiling the
# library that comes with WebRTC (i.e. rtc_build_json == 0).
rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
@@ -36,11 +39,14 @@ declare_args() {
# Disable these to not build components which can be externally provided.
rtc_build_expat = true
rtc_build_json = true
+ rtc_build_libjpeg = true
+ rtc_build_libsrtp = true
rtc_build_libvpx = true
rtc_build_libyuv = true
rtc_build_openmax_dl = true
rtc_build_opus = true
rtc_build_ssl = true
+ rtc_build_usrsctp = true
# Disable by default.
rtc_have_dbus_glib = false
@@ -101,12 +107,19 @@ declare_args() {
# http://www.openh264.org, https://www.ffmpeg.org/
rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
+ # Determines whether QUIC code will be built.
+ rtc_use_quic = false
+
# FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
# by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
# only be initialized once. Projects that initialize FFmpeg externally, such
# as Chromium, must turn this flag off so that WebRTC does not also
# initialize.
rtc_initialize_ffmpeg = !build_with_chromium
+
+ # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
+ # build environments, even if available for Chromium builds.
+ rtc_use_gtk = !build_with_chromium
}
# A second declare_args block, so that declarations within it can
« no previous file with comments | « webrtc/api/BUILD.gn ('k') | webrtc/libjingle/xmllite/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698