Move audio_e2e_harness into include_tests==1 condition.
To avoid compile errors when WebRTC is built as a part of
Chromium.
TEST=ran gclient runhooks locally.
BUG=none
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5003 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/tools/tools.gyp b/tools/tools.gyp
index 905c323..b8dc4c1 100644
--- a/tools/tools.gyp
+++ b/tools/tools.gyp
@@ -12,19 +12,6 @@
],
'targets': [
{
- 'target_name': 'audio_e2e_harness',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/test/test.gyp:channel_transport',
- '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
- ],
- 'sources': [
- 'e2e_quality/audio/audio_e2e_harness.cc',
- ],
- }, # audio_e2e_harness
- {
'target_name': 'video_quality_analysis',
'type': 'static_library',
'dependencies': [
@@ -114,6 +101,19 @@
['include_tests==1', {
'targets' : [
{
+ 'target_name': 'audio_e2e_harness',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/test/test.gyp:channel_transport',
+ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [
+ 'e2e_quality/audio/audio_e2e_harness.cc',
+ ],
+ }, # audio_e2e_harness
+ {
'target_name': 'tools_unittests',
'type': '<(gtest_target_type)',
'dependencies': [