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}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 4c29f6a..b789b29 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -12,27 +12,27 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/audio/audio_send_stream.h"
-#include "webrtc/audio/audio_state.h"
-#include "webrtc/audio/conversion.h"
-#include "webrtc/call/fake_rtp_transport_controller_send.h"
-#include "webrtc/call/rtp_transport_controller_send_interface.h"
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
-#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
-#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
-#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
-#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/rtc_base/task_queue.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_encoder.h"
-#include "webrtc/test/mock_audio_encoder_factory.h"
-#include "webrtc/test/mock_voe_channel_proxy.h"
-#include "webrtc/test/mock_voice_engine.h"
-#include "webrtc/voice_engine/transmit_mixer.h"
+#include "audio/audio_send_stream.h"
+#include "audio/audio_state.h"
+#include "audio/conversion.h"
+#include "call/fake_rtp_transport_controller_send.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/congestion_controller/include/mock/mock_congestion_observer.h"
+#include "modules/congestion_controller/include/send_side_congestion_controller.h"
+#include "modules/pacing/mock/mock_paced_sender.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "rtc_base/ptr_util.h"
+#include "rtc_base/task_queue.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/mock_audio_encoder_factory.h"
+#include "test/mock_voe_channel_proxy.h"
+#include "test/mock_voice_engine.h"
+#include "voice_engine/transmit_mixer.h"
 
 namespace webrtc {
 namespace test {