Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 6760b14..57f9e73 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -22,34 +22,34 @@
#include <utility>
#include <vector>
-#include "webrtc/api/fakemetricsobserver.h"
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/media/engine/fakewebrtcvideoengine.h"
-#include "webrtc/p2p/base/p2pconstants.h"
-#include "webrtc/p2p/base/portinterface.h"
-#include "webrtc/p2p/base/sessiondescription.h"
-#include "webrtc/p2p/base/testturnserver.h"
-#include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/pc/dtmfsender.h"
-#include "webrtc/pc/localaudiosource.h"
-#include "webrtc/pc/mediasession.h"
-#include "webrtc/pc/peerconnection.h"
-#include "webrtc/pc/peerconnectionfactory.h"
-#include "webrtc/pc/test/fakeaudiocapturemodule.h"
-#include "webrtc/pc/test/fakeperiodicvideocapturer.h"
-#include "webrtc/pc/test/fakertccertificategenerator.h"
-#include "webrtc/pc/test/fakevideotrackrenderer.h"
-#include "webrtc/pc/test/mockpeerconnectionobservers.h"
-#include "webrtc/rtc_base/asyncinvoker.h"
-#include "webrtc/rtc_base/fakenetwork.h"
-#include "webrtc/rtc_base/gunit.h"
-#include "webrtc/rtc_base/helpers.h"
-#include "webrtc/rtc_base/ssladapter.h"
-#include "webrtc/rtc_base/sslstreamadapter.h"
-#include "webrtc/rtc_base/thread.h"
-#include "webrtc/rtc_base/virtualsocketserver.h"
+#include "api/fakemetricsobserver.h"
+#include "api/mediastreaminterface.h"
+#include "api/peerconnectioninterface.h"
+#include "api/test/fakeconstraints.h"
+#include "media/engine/fakewebrtcvideoengine.h"
+#include "p2p/base/p2pconstants.h"
+#include "p2p/base/portinterface.h"
+#include "p2p/base/sessiondescription.h"
+#include "p2p/base/testturnserver.h"
+#include "p2p/client/basicportallocator.h"
+#include "pc/dtmfsender.h"
+#include "pc/localaudiosource.h"
+#include "pc/mediasession.h"
+#include "pc/peerconnection.h"
+#include "pc/peerconnectionfactory.h"
+#include "pc/test/fakeaudiocapturemodule.h"
+#include "pc/test/fakeperiodicvideocapturer.h"
+#include "pc/test/fakertccertificategenerator.h"
+#include "pc/test/fakevideotrackrenderer.h"
+#include "pc/test/mockpeerconnectionobservers.h"
+#include "rtc_base/asyncinvoker.h"
+#include "rtc_base/fakenetwork.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/helpers.h"
+#include "rtc_base/ssladapter.h"
+#include "rtc_base/sslstreamadapter.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/virtualsocketserver.h"
using cricket::ContentInfo;
using cricket::FakeWebRtcVideoDecoder;