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/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 0d58614..7bcb85a 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -12,19 +12,19 @@
#include <string>
#include <vector>
-#include "webrtc/api/test/mock_audio_mixer.h"
-#include "webrtc/audio/audio_receive_stream.h"
-#include "webrtc/audio/conversion.h"
-#include "webrtc/call/rtp_stream_receiver_controller.h"
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
-#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
-#include "webrtc/modules/pacing/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
-#include "webrtc/test/mock_voe_channel_proxy.h"
-#include "webrtc/test/mock_voice_engine.h"
+#include "api/test/mock_audio_mixer.h"
+#include "audio/audio_receive_stream.h"
+#include "audio/conversion.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_voe_channel_proxy.h"
+#include "test/mock_voice_engine.h"
namespace webrtc {
namespace test {