commit 92ea95e34af5966555903026f45164afbd7e2088
author Mirko Bonadei <mbonadei@webrtc.org> Fri Sep 15 06:47:31 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Fri Sep 15 05:02:56 2017 +0000
tree 3befb3fcfefff3b35da8a834052543338e15bcc0
parent bb547203bfebcc478b263c4e9ca173c6fd5a0c5d

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}

api/peerconnectioninterface.h

diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index e23b705..13e78e6 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -64,37 +64,37 @@
 // 7. Once a candidate has been gathered, the PeerConnection will call the
 // observer function OnIceCandidate. Send these candidates to the remote peer.
 
-#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
-#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
+#ifndef API_PEERCONNECTIONINTERFACE_H_
+#define API_PEERCONNECTIONINTERFACE_H_
 
 #include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
-#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
-#include "webrtc/api/datachannelinterface.h"
-#include "webrtc/api/dtmfsenderinterface.h"
-#include "webrtc/api/jsep.h"
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/rtcerror.h"
-#include "webrtc/api/rtpreceiverinterface.h"
-#include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/api/stats/rtcstatscollectorcallback.h"
-#include "webrtc/api/statstypes.h"
-#include "webrtc/api/umametrics.h"
-#include "webrtc/call/callfactoryinterface.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
-#include "webrtc/media/base/mediachannel.h"
-#include "webrtc/media/base/videocapturer.h"
-#include "webrtc/p2p/base/portallocator.h"
-#include "webrtc/rtc_base/fileutils.h"
-#include "webrtc/rtc_base/network.h"
-#include "webrtc/rtc_base/rtccertificate.h"
-#include "webrtc/rtc_base/rtccertificategenerator.h"
-#include "webrtc/rtc_base/socketaddress.h"
-#include "webrtc/rtc_base/sslstreamadapter.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/datachannelinterface.h"
+#include "api/dtmfsenderinterface.h"
+#include "api/jsep.h"
+#include "api/mediastreaminterface.h"
+#include "api/rtcerror.h"
+#include "api/rtpreceiverinterface.h"
+#include "api/rtpsenderinterface.h"
+#include "api/stats/rtcstatscollectorcallback.h"
+#include "api/statstypes.h"
+#include "api/umametrics.h"
+#include "call/callfactoryinterface.h"
+#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "media/base/mediachannel.h"
+#include "media/base/videocapturer.h"
+#include "p2p/base/portallocator.h"
+#include "rtc_base/fileutils.h"
+#include "rtc_base/network.h"
+#include "rtc_base/rtccertificate.h"
+#include "rtc_base/rtccertificategenerator.h"
+#include "rtc_base/socketaddress.h"
+#include "rtc_base/sslstreamadapter.h"
 
 namespace rtc {
 class SSLIdentity;
@@ -1224,4 +1224,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
+#endif  // API_PEERCONNECTIONINTERFACE_H_