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authorgrunell@chromium.org <grunell@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2012-02-02 14:15:33 +0000
committergrunell@chromium.org <grunell@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2012-02-02 14:15:33 +0000
commit40a96f3ec37c40f4a8161d2a001b7dd924e4a8d3 (patch)
tree69eaf68484eba7eebdf1ea37335d8d63062633ee
parent8f73be933d140c780db67f32ba9990956fad9eff (diff)
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Moved mocks for webrtc unit tests to conditional section.
The purpose is just to clarify that these mocks are only needed when webrtc is enabled. No actual issue is fixed. TEST=Set enable_webrtc to 0, run content_unittests. Review URL: https://chromiumcodereview.appspot.com/9315054 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@120212 0039d316-1c4b-4281-b951-d872f2087c98
-rw-r--r--content/content_tests.gypi8
1 files changed, 4 insertions, 4 deletions
diff --git a/content/content_tests.gypi b/content/content_tests.gypi
index 7f549a4..eda6334 100644
--- a/content/content_tests.gypi
+++ b/content/content_tests.gypi
@@ -58,10 +58,6 @@
'common/test_url_constants.h',
'gpu/gpu_idirect3d9_mock_win.cc',
'gpu/gpu_idirect3d9_mock_win.h',
- 'renderer/media/mock_media_stream_dispatcher.cc',
- 'renderer/media/mock_media_stream_dispatcher.h',
- 'renderer/media/mock_web_peer_connection_handler_client.cc',
- 'renderer/media/mock_web_peer_connection_handler_client.h',
'renderer/mock_content_renderer_client.cc',
'renderer/mock_content_renderer_client.h',
'test/browser_test.h',
@@ -119,10 +115,14 @@
'sources': [
'renderer/media/mock_media_stream_dependency_factory.cc',
'renderer/media/mock_media_stream_dependency_factory.h',
+ 'renderer/media/mock_media_stream_dispatcher.cc',
+ 'renderer/media/mock_media_stream_dispatcher.h',
'renderer/media/mock_media_stream_impl.cc',
'renderer/media/mock_media_stream_impl.h',
'renderer/media/mock_peer_connection_impl.cc',
'renderer/media/mock_peer_connection_impl.h',
+ 'renderer/media/mock_web_peer_connection_handler_client.cc',
+ 'renderer/media/mock_web_peer_connection_handler_client.h',
'test/webrtc_audio_device_test.cc',
'test/webrtc_audio_device_test.h',
],