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Move audio_e2e_harness into include_tests==1 condition.

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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@5003 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kjellander@webrtc.org 2013-10-21 08:40:56 +00:00
parent 88a310886e
commit eb61a851d5
1 changed files with 13 additions and 13 deletions
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26
webrtc/tools/tools.gyp
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@ -11,19 +11,6 @@
'../build/common.gypi',
],
'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',
@ -113,6 +100,19 @@
'conditions': [
['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)',