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/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 425c213..319e752 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -8,28 +8,28 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include <cmath>
#include <utility>
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/timeutils.h"
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/timeutils.h"
#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
#else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif