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/call/rtx_receive_stream.h b/call/rtx_receive_stream.h
index c288a27..8ffa440 100644
--- a/call/rtx_receive_stream.h
+++ b/call/rtx_receive_stream.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#ifndef CALL_RTX_RECEIVE_STREAM_H_
+#define CALL_RTX_RECEIVE_STREAM_H_
#include <map>
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
@@ -47,4 +47,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#endif // CALL_RTX_RECEIVE_STREAM_H_