webrtc/api/rtpsenderreceiver_unittest.cc - Issue 1610243002: Move talk/app/webrtc to webrtc/api

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Unified Diff: webrtc/api/rtpsenderreceiver_unittest.cc

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/rtpsenderreceiver_unittest.cc

diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc

similarity index 98%

rename from talk/app/webrtc/rtpsenderreceiver_unittest.cc

rename to webrtc/api/rtpsenderreceiver_unittest.cc

index bcd9ea035022168616bbd43ecf54405f9fc849ac..faca6579f8967a450a8ce338db9b19b0428e4fce 100644

--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc

+++ b/webrtc/api/rtpsenderreceiver_unittest.cc

@@ -28,16 +28,16 @@

#include <string>

#include <utility>

-#include "talk/app/webrtc/audiotrack.h"

-#include "talk/app/webrtc/mediastream.h"

-#include "talk/app/webrtc/remoteaudiosource.h"

-#include "talk/app/webrtc/rtpreceiver.h"

-#include "talk/app/webrtc/rtpsender.h"

-#include "talk/app/webrtc/streamcollection.h"

-#include "talk/app/webrtc/videosource.h"

-#include "talk/app/webrtc/videotrack.h"

#include "testing/gmock/include/gmock/gmock.h"

#include "testing/gtest/include/gtest/gtest.h"

+#include "webrtc/api/audiotrack.h"

+#include "webrtc/api/mediastream.h"

+#include "webrtc/api/remoteaudiosource.h"

+#include "webrtc/api/rtpreceiver.h"

+#include "webrtc/api/rtpsender.h"

+#include "webrtc/api/streamcollection.h"

+#include "webrtc/api/videosource.h"

+#include "webrtc/api/videotrack.h"

#include "webrtc/base/gunit.h"

#include "webrtc/media/base/fakevideocapturer.h"

#include "webrtc/media/base/mediachannel.h"

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