commit b62dbbe985c643cf4ee28e4c73c75bb3ef5e4d54
author kjellander <kjellander@webrtc.org> Fri Sep 23 00:38:52 2016 -0700
committer Commit bot <commit-bot@chromium.org> Fri Sep 23 07:38:58 2016 +0000
tree 6e59ec270ff6d4137bf0bf55a7639923d99eb3a9
parent 25f6a39181c9162d26c12ceb8af69e405b2da9e5

GN: Change rtc_source_set targets --> rtc_static_library

This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).

After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()

See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.

NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.

BUG=webrtc:6410, chromium:630755

Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}

webrtc/modules/video_coding/BUILD.gn

diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index 3db9b02..d9b392d 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -8,7 +8,7 @@
 
 import("../../build/webrtc.gni")
 
-rtc_source_set("video_coding") {
+rtc_static_library("video_coding") {
   sources = [
     "codec_database.cc",
     "codec_database.h",
@@ -95,7 +95,7 @@
   ]
 }
 
-rtc_source_set("video_coding_utility") {
+rtc_static_library("video_coding_utility") {
   sources = [
     "utility/frame_dropper.cc",
     "utility/frame_dropper.h",
@@ -127,7 +127,7 @@
   ]
 }
 
-rtc_source_set("webrtc_h264") {
+rtc_static_library("webrtc_h264") {
   sources = [
     "codecs/h264/h264.cc",
     "codecs/h264/include/h264.h",
@@ -178,7 +178,7 @@
     }
   }
 
-  rtc_source_set("webrtc_h264_video_toolbox") {
+  rtc_static_library("webrtc_h264_video_toolbox") {
     sources = [
       "codecs/h264/h264_video_toolbox_decoder.cc",
       "codecs/h264/h264_video_toolbox_decoder.h",
@@ -223,7 +223,7 @@
   }
 }
 
-rtc_source_set("webrtc_i420") {
+rtc_static_library("webrtc_i420") {
   sources = [
     "codecs/i420/i420.cc",
     "codecs/i420/include/i420.h",
@@ -240,7 +240,7 @@
   ]
 }
 
-rtc_source_set("webrtc_vp8") {
+rtc_static_library("webrtc_vp8") {
   sources = [
     "codecs/vp8/default_temporal_layers.cc",
     "codecs/vp8/default_temporal_layers.h",
@@ -278,7 +278,7 @@
   }
 }
 
-rtc_source_set("webrtc_vp9") {
+rtc_static_library("webrtc_vp9") {
   sources = [
     "codecs/vp9/include/vp9.h",
     "codecs/vp9/screenshare_layers.cc",