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Unified Diff: webrtc/api/test/fakeaudiocapturemodule.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: webrtc/api/test/fakeaudiocapturemodule.h
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h
similarity index 98%
rename from talk/app/webrtc/test/fakeaudiocapturemodule.h
rename to webrtc/api/test/fakeaudiocapturemodule.h
index fdac0b9ed252284f9c1b82f07aa2704c27b24135..315c251ab80aa8620f36ae4a8c4a3888d5a04a53 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/webrtc/api/test/fakeaudiocapturemodule.h
@@ -34,8 +34,8 @@
// Note P postfix of a function indicates that it should only be called by the
// processing thread.
-#ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
-#define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
@@ -284,4 +284,4 @@ class FakeAudioCaptureModule
rtc::CriticalSection crit_callback_;
};
-#endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
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