Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/

gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}

webrtc/api/mediastream_unittest.cc

@@ -12,12 +12,12 @@
 
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/mediastream.h"
-#include "webrtc/api/videotrack.h"
 #include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "dummy_video_cam_1";

webrtc/api/peerconnectioninterface_unittest.cc

@@ -12,7 +12,6 @@
 #include <string>
 #include <utility>
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastream.h"
@@ -22,9 +21,6 @@
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/streamcollection.h"
-#ifdef WEBRTC_ANDROID
-#include "webrtc/api/test/androidtestinitializer.h"
-#endif
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/api/test/fakertccertificategenerator.h"
 #include "webrtc/api/test/fakevideotracksource.h"
@@ -42,6 +38,11 @@
 #include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/mediasession.h"
+#include "webrtc/test/gmock.h"
+
+#ifdef WEBRTC_ANDROID
+#include "webrtc/api/test/androidtestinitializer.h"
+#endif
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kStreamLabel2[] = "local_stream_2";

webrtc/api/proxy_unittest.cc

@@ -13,10 +13,10 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/test/gmock.h"
 
 using ::testing::_;
 using ::testing::DoAll;

webrtc/api/rtpsenderreceiver_unittest.cc

@@ -12,8 +12,6 @@
 #include <string>
 #include <utility>
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/fakemediacontroller.h"
 #include "webrtc/api/localaudiosource.h"
@@ -23,14 +21,16 @@
 #include "webrtc/api/rtpsender.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/videotracksource.h"
 #include "webrtc/api/videotrack.h"
+#include "webrtc/api/videotracksource.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/engine/fakewebrtccall.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channelmanager.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 
 using ::testing::_;
 using ::testing::Exactly;

webrtc/api/statscollector_unittest.cc

@@ -15,8 +15,6 @@
 
 #include "webrtc/api/statscollector.h"
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediastreamtrack.h"
@@ -35,6 +33,8 @@
 #include "webrtc/media/base/test/mock_mediachannel.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channelmanager.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 
 using testing::_;
 using testing::DoAll;

webrtc/audio/audio_receive_stream_unittest.cc

@@ -11,8 +11,6 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
@@ -23,6 +21,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 

webrtc/audio/audio_send_stream_unittest.cc

@@ -11,17 +11,16 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
-#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
+#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 

webrtc/audio/audio_state_unittest.cc

@@ -10,9 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/audio/audio_state.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {

webrtc/base/event_tracer_unittest.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/base/event_tracer.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/system_wrappers/include/static_instance.h"
+#include "webrtc/test/gtest.h"
 
 namespace {
 

webrtc/base/mod_ops_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/mod_ops.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 class TestModOps : public ::testing::Test {

webrtc/base/platform_thread_unittest.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/base/platform_thread.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gtest.h"
 
 namespace rtc {
 namespace {

webrtc/base/random_unittest.cc

@@ -13,9 +13,9 @@
 #include <limits>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/mathutils.h"  // unsigned difference
 #include "webrtc/base/random.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/base/rate_limiter_unittest.cc

@@ -11,13 +11,12 @@
 #include <algorithm>
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/base/rate_statistics_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <algorithm>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/rate_statistics.h"
+#include "webrtc/test/gtest.h"
 
 namespace {
 

webrtc/base/sequenced_task_checker_unittest.cc

@@ -7,12 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/test/gtest.h"
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/sequenced_task_checker.h"
 #include "webrtc/base/task_queue.h"
+#include "webrtc/test/gtest.h"
 
 namespace rtc {
 

webrtc/base/thread_checker_unittest.cc

@@ -12,12 +12,12 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/test/gtest.h"
 
 // Duplicated from base/threading/thread_checker.h so that we can be
 // good citizens there and undef the macro.

webrtc/call/bitrate_allocator_unittest.cc

@@ -12,10 +12,10 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 
 using testing::NiceMock;
 

webrtc/call/bitrate_estimator_tests.cc

@@ -12,8 +12,6 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
@@ -27,8 +25,9 @@
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/mock_voice_engine.h"
 #include "webrtc/test/frame_generator_capturer.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace {

webrtc/call/call_perf_tests.cc

@@ -13,8 +13,6 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
@@ -36,6 +34,7 @@
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"

webrtc/call/call_unittest.cc

@@ -11,11 +11,10 @@
 #include <list>
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {

webrtc/call/mock/mock_rtc_event_log.h

@@ -13,9 +13,8 @@
 
 #include <string>
 
-#include "webrtc/test/gmock.h"
-
 #include "webrtc/call/rtc_event_log.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/call/packet_injection_tests.cc

@@ -10,9 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/test/call_test.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/null_transport.h"
 
 namespace webrtc {

webrtc/call/rampup_tests.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/call/rampup_tests.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 namespace webrtc {

webrtc/call/ringbuffer_unittest.cc

@@ -11,9 +11,9 @@
 
 #include <list>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/random.h"
 #include "webrtc/call/ringbuffer.h"
+#include "webrtc/test/gtest.h"
 
 namespace {
 template <typename T>

webrtc/call/rtc_event_log_unittest.cc

@@ -14,7 +14,6 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
@@ -28,6 +27,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 

webrtc/call/rtc_event_log_unittest_helper.cc

@@ -14,8 +14,8 @@
 
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 

webrtc/common_audio/audio_converter_unittest.cc

@@ -13,12 +13,12 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_audio/audio_ring_buffer_unittest.cc

@@ -12,8 +12,8 @@
 
 #include "webrtc/common_audio/audio_ring_buffer.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_audio/audio_util_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/common_audio/blocker_unittest.cc

@@ -12,8 +12,8 @@
 
 #include "webrtc/common_audio/blocker.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
+#include "webrtc/test/gtest.h"
 
 namespace {
 

webrtc/common_audio/channel_buffer_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_audio/real_fourier_unittest.cc

@@ -12,9 +12,9 @@
 
 #include <stdlib.h>
 
-#include "webrtc/test/gtest.h"
-#include "webrtc/common_audio/real_fourier_openmax.h"
 #include "webrtc/common_audio/real_fourier_ooura.h"
+#include "webrtc/common_audio/real_fourier_openmax.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_audio/resampler/push_resampler_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
-#include "webrtc/base/checks.h"  // force defintion of RTC_DCHECK_IS_ON
+#include "webrtc/base/checks.h"  // RTC_DCHECK_IS_ON
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
+#include "webrtc/test/gtest.h"
 
 // Quality testing of PushResampler is handled through output_mixer_unittest.cc.
 

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

@@ -12,12 +12,12 @@
 #include <cstring>
 #include <memory>
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/common_audio/resampler/resampler_unittest.cc

@@ -8,9 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/test/gtest.h"
 
 // TODO(andrew): this is a work-in-progress. Many more tests are needed.
 

webrtc/common_audio/resampler/sinc_resampler_unittest.cc

@@ -18,13 +18,13 @@
 
 #include <memory>
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/system_wrappers/include/stringize_macros.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/test_suite.h"
 
 using testing::_;

webrtc/common_audio/signal_processing/real_fft_unittest.cc

@@ -10,9 +10,8 @@
 
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/typedefs.h"
-
 #include "webrtc/test/gtest.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace {

webrtc/common_audio/signal_processing/signal_processing_unittest.cc

@@ -11,8 +11,8 @@
 #include <algorithm>
 #include <sstream>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/test/gtest.h"
 
 static const size_t kVector16Size = 9;
 static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963,

webrtc/common_audio/sparse_fir_filter_unittest.cc

@@ -12,9 +12,9 @@
 
 #include "webrtc/common_audio/sparse_fir_filter.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/common_audio/fir_filter.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace {

webrtc/common_audio/vad/mock/mock_vad.h

@@ -12,7 +12,6 @@
 #define WEBRTC_COMMON_AUDIO_VAD_MOCK_MOCK_VAD_H_
 
 #include "webrtc/common_audio/vad/include/vad.h"
-
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {

webrtc/common_audio/vad/vad_core_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <stdlib.h>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 extern "C" {

webrtc/common_audio/vad/vad_filterbank_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <stdlib.h>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 extern "C" {

webrtc/common_audio/vad/vad_gmm_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 extern "C" {

webrtc/common_audio/vad/vad_sp_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <stdlib.h>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 extern "C" {

webrtc/common_audio/vad/vad_unittest.cc

@@ -12,12 +12,11 @@
 
 #include <stdlib.h>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 VadTest::VadTest() {}

webrtc/common_audio/vad/vad_unittest.h

@@ -14,7 +14,6 @@
 #include <stddef.h>  // size_t
 
 #include "webrtc/test/gtest.h"
-
 #include "webrtc/typedefs.h"
 
 namespace {

webrtc/common_audio/wav_file_unittest.cc

@@ -14,9 +14,9 @@
 #include <cmath>
 #include <limits>
 
-#include "webrtc/test/gtest.h"
-#include "webrtc/common_audio/wav_header.h"
 #include "webrtc/common_audio/wav_file.h"
+#include "webrtc/common_audio/wav_header.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {

webrtc/common_audio/wav_header_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <limits>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/wav_header.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/bitrate_adjuster_unittest.cc

@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/common_video/include/bitrate_adjuster.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/h264/pps_parser_unittest.cc

@@ -13,11 +13,10 @@
 #include <limits>
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/common_video/h264/h264_common.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/h264/sps_parser_unittest.cc

@@ -10,12 +10,11 @@
 
 #include "webrtc/common_video/h264/sps_parser.h"
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/common_video/h264/h264_common.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/h264/sps_vui_rewriter_unittest.cc

@@ -10,17 +10,15 @@
 
 #include <vector>
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/stream.h"
-
-#include "webrtc/common_video/h264/sps_vui_rewriter.h"
 #include "webrtc/common_video/h264/h264_common.h"
+#include "webrtc/common_video/h264/sps_vui_rewriter.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/i420_buffer_pool_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_video/include/i420_buffer_pool.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_video/i420_video_frame_unittest.cc

@@ -11,10 +11,10 @@
 #include <math.h>
 #include <string.h>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/test/fake_texture_frame.h"
 #include "webrtc/test/frame_utils.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {

webrtc/common_video/libyuv/libyuv_unittest.cc

@@ -13,9 +13,9 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/test/frame_utils.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/video_frame.h"
 

webrtc/media/engine/nullwebrtcvideoengine_unittest.cc

@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
-
 #include "webrtc/media/engine/nullwebrtcvideoengine.h"
 #include "webrtc/media/engine/webrtcvoiceengine.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
+#include "webrtc/test/gtest.h"
 
 namespace cricket {
 

webrtc/media/engine/payload_type_mapper_unittest.cc

@@ -11,8 +11,8 @@
 #include <set>
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/media/engine/payload_type_mapper.h"
+#include "webrtc/test/gtest.h"
 
 namespace cricket {
 

webrtc/media/engine/simulcast_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/media/engine/simulcast.h"
+#include "webrtc/test/gtest.h"
 
 namespace cricket {
 

webrtc/media/engine/webrtcmediaengine_unittest.cc

@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
-
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/test/gtest.h"
 
 using webrtc::RtpExtension;
 

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

@@ -15,11 +15,11 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace test {

webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

@@ -13,13 +13,13 @@
 #include <algorithm>  // std::min
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 

webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc

@@ -14,12 +14,12 @@
 #include <stdio.h>
 #include <string.h>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace test {

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

@@ -13,7 +13,6 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/platform_thread.h"
@@ -40,6 +39,7 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::testing::AtLeast;

webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace acm2 {

webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
-#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace acm2 {

webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

@@ -11,11 +11,11 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <utility>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc

@@ -11,8 +11,8 @@
 #include <memory>
 #include <utility>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc

@@ -11,11 +11,11 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/vad/mock/mock_vad.h"
 #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/test/gtest.h"
 
 using ::testing::Return;
 using ::testing::_;

webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc

@@ -10,9 +10,9 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 class FilterBanksTest : public testing::Test {

webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc

@@ -7,9 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/test/gtest.h"
+
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 class FiltersTest : public testing::Test {

webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
 class LpcMaskingModelTest : public testing::Test {

webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc

@@ -7,9 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/test/gtest.h"
+
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/test/gtest.h"
 
 static const int kSamples = FRAMESAMPLES/2;
 static const int32_t spec2time_out_expected_1[kSamples] = {

webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <limits>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

@@ -9,8 +9,8 @@
  */
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 struct WebRtcISACStruct;

webrtc/modules/audio_coding/codecs/isac/unittest.cc

@@ -13,11 +13,11 @@
 #include <sstream>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h

@@ -13,9 +13,9 @@
 
 #include <vector>
 
-#include "webrtc/test/gmock.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h

@@ -15,7 +15,6 @@
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc

@@ -10,9 +10,9 @@
 
 #include <memory>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::string;

webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc

@@ -11,11 +11,11 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc

@@ -11,10 +11,10 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "webrtc/test/gtest.h"
 
 using ::testing::Return;
 using ::testing::_;

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::tr1::get;

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 #include <string>
+
 #include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 

webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -17,7 +17,6 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
@@ -34,6 +33,7 @@
 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/system_wrappers/include/data_log.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc

@@ -12,9 +12,9 @@
 
 #include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc

@@ -10,15 +10,15 @@
 
 // Unit tests for DecisionLogic class and derived classes.
 
-#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
+#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc

@@ -15,12 +15,11 @@
 
 #include <string>
 
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-
-#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 
 using testing::_;
 using testing::Invoke;

webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc

@@ -14,9 +14,9 @@
 
 #include <math.h>
 
+#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
 
 namespace webrtc {