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}

media/base/mediachannel.h

diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 7e7b815..631e695 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -8,35 +8,35 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
-#define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
+#ifndef MEDIA_BASE_MEDIACHANNEL_H_
+#define MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/api/rtpreceiverinterface.h"
-#include "webrtc/api/video/video_timing.h"
-#include "webrtc/call/video_config.h"
-#include "webrtc/media/base/codec.h"
-#include "webrtc/media/base/mediaconstants.h"
-#include "webrtc/media/base/streamparams.h"
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/media/base/videosourceinterface.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/copyonwritebuffer.h"
-#include "webrtc/rtc_base/dscp.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/networkroute.h"
-#include "webrtc/rtc_base/sigslot.h"
-#include "webrtc/rtc_base/socket.h"
-#include "webrtc/rtc_base/window.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/optional.h"
+#include "api/rtpparameters.h"
+#include "api/rtpreceiverinterface.h"
+#include "api/video/video_timing.h"
+#include "call/video_config.h"
+#include "media/base/codec.h"
+#include "media/base/mediaconstants.h"
+#include "media/base/streamparams.h"
+#include "media/base/videosinkinterface.h"
+#include "media/base/videosourceinterface.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/copyonwritebuffer.h"
+#include "rtc_base/dscp.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/networkroute.h"
+#include "rtc_base/sigslot.h"
+#include "rtc_base/socket.h"
+#include "rtc_base/window.h"
 // TODO(juberti): re-evaluate this include
-#include "webrtc/pc/audiomonitor.h"
+#include "pc/audiomonitor.h"
 
 namespace rtc {
 class RateLimiter;
@@ -1254,4 +1254,4 @@
 
 }  // namespace cricket
 
-#endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
+#endif  // MEDIA_BASE_MEDIACHANNEL_H_