From eb61a851d5fd07382d5d99fb9f779691dbdfe761 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Mon, 21 Oct 2013 08:40:56 +0000 Subject: [PATCH] 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@5003 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/tools/tools.gyp | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp index 905c323aa1..b8dc4c142d 100644 --- a/webrtc/tools/tools.gyp +++ b/webrtc/tools/tools.gyp @@ -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)',