Include files from webrtc/.. paths in rtp_rtcp/

BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
 #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
 
-#include "typedefs.h"
-#include "module_common_types.h"
+#include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/typedefs.h"
 
 #ifndef NULL
     #define NULL    0

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -11,11 +11,11 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 
-#include <gmock/gmock.h>
+#include "testing/gmock/include/gmock/gmock.h"
 
-#include "modules/interface/module.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/H264/bitstream_builder.h

@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class BitstreamBuilder

webrtc/modules/rtp_rtcp/source/H264/bitstream_parser.h

@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class BitstreamParser

webrtc/modules/rtp_rtcp/source/bitrate.h

@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
 
 #include <stdio.h>
+
 #include <list>
 
 #include "webrtc/common_types.h"

webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h

@@ -25,7 +25,7 @@
 // (i.e., more packets/symbols in the code, so larger (k,m), i.e.,  k > 4,
 // m > 3).
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace {
 
@@ -758,4 +758,3 @@ const uint8_t** kPacketMaskBurstyTbl[12] = {
 
 }  // namespace
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
-

webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h

@@ -15,7 +15,7 @@
 // this table are specifically designed to favor recovery to random loss.
 // These packet masks are defined to protect up to maximum of 48 media packets.
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace {
 
@@ -24520,4 +24520,3 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
 
 }  // namespace
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
-

webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h

@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
 
-#include "gmock/gmock.h"
+#include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
 
 namespace webrtc {

webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h

@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
 
-#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/producer_fec.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/rtp_rtcp/source/producer_fec.h"
+#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/producer_fec.h

@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <list>
 
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/fec_test_helper.h"
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-#include "modules/rtp_rtcp/source/producer_fec.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/receiver_fec_unittest.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <list>
 #include <string.h>
 
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
+#include <list>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h"

webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
-#include "typedefs.h"
-#include "common_types.h"
-#include "rtp_utility.h"
-#include "rtcp_sender.h"
-#include "rtcp_receiver.h"
-#include "rtp_rtcp_impl.h"
-#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/typedefs.h"
 
 namespace {
 

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtcp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 
-#include <string.h> //memset
 #include <cassert> //assert
+#include <string.h> //memset
 
-#include "trace.h"
-#include "trace_event.h"
-#include "critical_section_wrapper.h"
-#include "rtcp_utility.h"
-#include "rtp_rtcp_impl.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace
 {

webrtc/modules/rtp_rtcp/source/rtcp_receiver.h

@@ -14,12 +14,12 @@
 #include <map>
 #include <vector>
 
-#include "typedefs.h"
-#include "rtp_utility.h"
-#include "rtcp_utility.h"
-#include "rtp_rtcp_defines.h"
-#include "rtcp_receiver_help.h"
-#include "tmmbr_help.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class ModuleRtpRtcpImpl;

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
 
-#include <string.h>  // memset
 #include <cassert>  // assert
+#include <string.h>  // memset
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 

webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

@@ -12,17 +12,17 @@
 /*
  * This file includes unit tests for the RTCPReceiver.
  */
-#include <gmock/gmock.h>
-#include <gtest/gtest.h>
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 
 // Note: This file has no directory. Lint warning must be ignored.
-#include "common_types.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "modules/rtp_rtcp/source/rtcp_sender.h"
-#include "modules/rtp_rtcp/source/rtcp_receiver.h"
-#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtcp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <algorithm>  // min
 #include <cassert>  // assert
 #include <cstdlib>  // rand
 #include <string.h>  // memcpy
 
-#include "common_types.h"
-#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "system_wrappers/interface/trace_event.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

@@ -15,14 +15,14 @@
 #include <sstream>
 #include <string>
 
-#include "typedefs.h"
-#include "rtcp_utility.h"
-#include "rtp_utility.h"
-#include "rtp_rtcp_defines.h"
-#include "scoped_ptr.h"
-#include "tmmbr_help.h"
-#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc

@@ -13,17 +13,17 @@
  * This file includes unit tests for the RTCPSender.
  */
 
-#include <gmock/gmock.h>
-#include <gtest/gtest.h>
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtcp_utility.cc

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
-#include <cstring> // memcpy
-#include <cmath>   // ceil
 #include <cassert>
+#include <cmath>   // ceil
+#include <cstring> // memcpy
 
 namespace webrtc {
 // RTCPParserV2 : currently read only

webrtc/modules/rtp_rtcp/source/rtcp_utility.h

@@ -13,9 +13,9 @@
 
 #include <cstddef> // size_t, ptrdiff_t
 
-#include "typedefs.h"
-#include "rtp_rtcp_config.h"
-#include "rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace RTCPUtility {

webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <gtest/gtest.h>
 #include <list>
 
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 

webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 
 #include <string.h>  // memcpy
 
 #include <cassert>   // assert
 #include <vector>
 
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+#include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h

@@ -28,9 +28,9 @@
 #include <queue>
 #include <vector>
 
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h"  // NOLINT(build/include)
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc

@@ -9,9 +9,9 @@
  */
 
 
-#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h

@@ -18,10 +18,10 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 
-#include "modules/interface/module_common_types.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc

@@ -13,13 +13,13 @@
  * This file includes unit tests for the VP8 packetizer.
  */
 
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
-#include "compile_assert.h"
+#include "webrtc/system_wrappers/interface/compile_assert.h"
 
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
-#include "typedefs.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc

@@ -10,8 +10,8 @@
 
 #include <cassert>
 
-#include "common_types.h"
-#include "rtp_header_extension.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_header_extension.h

@@ -13,8 +13,8 @@
 
 #include <map>
 
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "typedefs.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc

@@ -13,11 +13,11 @@
  * This file includes unit tests for the RtpHeaderExtensionMap.
  */
 
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
-#include "rtp_header_extension.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtp_packet_history.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 
 #include <assert.h>
 #include <cstring>   // memset
 
-#include "critical_section_wrapper.h"
-#include "rtp_utility.h"
-#include "trace.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_packet_history.h

@@ -15,9 +15,9 @@
 
 #include <vector>
 
-#include "module_common_types.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc

@@ -10,12 +10,12 @@
  * This file includes unit tests for the RTPPacketHistory.
  */
 
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
-#include "clock.h"
-#include "rtp_packet_history.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
 
-#include "gtest/gtest.h"
-#include "gmock/gmock.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 
-#include <string.h>
 #include <cassert>
+#include <string.h>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtp_sender_audio.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
-#include <string.h> //memcpy
 #include <cassert> //assert
+#include <string.h> //memcpy
 
-#include "trace_event.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 RTPSenderAudio::RTPSenderAudio(const int32_t id, Clock* clock,

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -11,14 +11,12 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
-#include "rtp_rtcp_config.h"          // misc. defines (e.g. MAX_PACKET_LENGTH)
-#include "common_types.h"             // Transport
-#include "typedefs.h"
-
-#include "dtmf_queue.h"
-#include "rtp_utility.h"
-
-#include "rtp_sender.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class RTPSenderAudio: public DTMFqueue

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -12,7 +12,7 @@
  * This file includes unit tests for the RTPSender.
  */
 
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/pacing/include/mock/mock_paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -8,21 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtp_sender_video.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "trace_event.h"
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
 
-#include "rtp_utility.h"
-
-#include <string.h> // memcpy
-#include <cassert>  // assert
-#include <cstdlib>  // srand
-
-#include "producer_fec.h"
-#include "rtp_format_vp8.h"
-#include "rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };

webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -13,18 +13,16 @@
 
 #include <list>
 
-#include "typedefs.h"
-#include "common_types.h"               // Transport
-#include "rtp_rtcp_config.h"
-
-#include "rtp_rtcp_defines.h"
-#include "rtp_utility.h"
-
-#include "video_codec_information.h"
-#include "forward_error_correction.h"
-#include "bitrate.h"
-#include "rtp_sender.h"
-#include "producer_fec.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -8,15 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 #include <cassert>
 #include <cmath>  // ceil
 #include <cstring>  // memcpy
 
 #if defined(_WIN32)
+// Order for these headers are important
 #include <Windows.h>  // FILETIME
+
 #include <WinSock.h>  // timeval
+
 #include <MMSystem.h>  // timeGetTime
 #elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
 #include <sys/time.h>  // gettimeofday
@@ -26,8 +29,8 @@
 #include <stdio.h>
 #endif
 
-#include "system_wrappers/interface/tick_util.h"
-#include "system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/interface/trace.h"
 
 #if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
 #define DEBUG_PRINT(...)           \

webrtc/modules/rtp_rtcp/source/rtp_utility.h

@@ -13,10 +13,10 @@
 
 #include <cstddef> // size_t, ptrdiff_t
 
-#include "typedefs.h"
-#include "rtp_header_extension.h"
-#include "rtp_rtcp_config.h"
-#include "rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 enum RtpVideoCodecTypes

webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc

@@ -13,10 +13,10 @@
  * This file conatins unit tests for the ModuleRTPUtility.
  */
 
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "typedefs.h"  // NOLINT(build/include)
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/ssrc_database.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "ssrc_database.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
+#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 
+#include <assert.h>
 #include <stdlib.h>
-#include <cassert>
+
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
 
 #ifdef _WIN32
     #include <windows.h>

webrtc/modules/rtp_rtcp/source/ssrc_database.h

@@ -12,11 +12,11 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
 
 #ifndef WEBRTC_NO_STL
-    #include <map>
+#include <map>
 #endif
 
-#include "system_wrappers/interface/static_instance.h"
-#include "typedefs.h"
+#include "webrtc/system_wrappers/interface/static_instance.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;

webrtc/modules/rtp_rtcp/source/tmmbr_help.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "tmmbr_help.h"
+#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 
 #include <assert.h>
 #include <limits>
 #include <string.h>
-#include "rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 
 namespace webrtc {
 TMMBRSet::TMMBRSet() :

webrtc/modules/rtp_rtcp/source/tmmbr_help.h

@@ -12,13 +12,9 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
 
 #include <vector>
-#include "typedefs.h"
 
-#include "critical_section_wrapper.h"
-
-#ifndef NULL
-    #define NULL    0
-#endif
+#include "webrtc/typedefs.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 class TMMBRSet

webrtc/modules/rtp_rtcp/source/video_codec_information.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
 
-#include "rtp_rtcp_config.h"
-#include "rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 class VideoCodecInformation

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+#include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
 
 #include <assert.h>
 #include <stdlib.h>  // NULL
@@ -265,4 +265,3 @@ int Vp8PartitionAggregator::CalcNumberOfFragments(int large_partition_size,
 }
 
 }  // namespace
-

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h

@@ -13,9 +13,9 @@
 
 #include <vector>
 
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h"  // NOLINT(build/include)
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <stdlib.h>  // NULL
 
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
 
 namespace webrtc {
 
@@ -212,4 +212,3 @@ TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
 }
 
 }  // namespace
-

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "BWEConvergenceTest.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h"
 
 #include <fstream>
 #include <string>
 
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 
 
 BWEConvergenceTestUp::BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps)
@@ -62,5 +62,3 @@ bool BWEConvergenceTestUp::StoppingCriterionMaster()
 {
     return ((_sendrec->BitrateSent() / 1000.0) > (0.9 * _availBWkbps));
 }
-
-

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h

@@ -13,11 +13,9 @@
 
 #include <string>
 
-#include "BWETestBase.h"
-
-#include "typedefs.h"
-
-#include "TestSenderReceiver.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
+#include "webrtc/typedefs.h"
 
 class BWEConvergenceTestUp : public BWEOneWayTest
 {

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc

@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <fstream>
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h"
+
 #include <math.h>
 
-#include "BWEStabilityTest.h"
-#include "TestLoadGenerator.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
+#include <fstream>
+
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 
 
 BWEStabilityTest::BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds)

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h

@@ -13,11 +13,9 @@
 
 #include <string>
 
-#include "BWETestBase.h"
-
-#include "typedefs.h"
-
-#include "TestSenderReceiver.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
+#include "webrtc/typedefs.h"
 
 class BWEStabilityTest : public BWEOneWayTest
 {

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc

@@ -11,18 +11,18 @@
 // BWEStandAlone.cpp : Defines the entry point for the console application.
 //
 
-#include <string>
 #include <stdio.h>
+#include <string>
 
-#include "event_wrapper.h"
-#include "udp_transport.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/channel_transport/udp_transport.h"
 
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
 
-#include "MatlabPlot.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
 
 //#include "vld.h"
 
@@ -197,4 +197,3 @@ int main(int argc, char* argv[])
 
     //return 0;
 }
-

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc

@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "BWETestBase.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
 
 #include <algorithm> // sort
 #include <fstream>
+#include <math.h>
 #include <string>
 #include <vector>
-#include <math.h>
 
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 
 
 double StatVec::Mean()

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h

@@ -14,9 +14,9 @@
 #include <string>
 #include <vector>
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
-#include "TestSenderReceiver.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
 
 
 class StatVec : public std::vector<double>

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc

@@ -11,27 +11,23 @@
 // BWETester.cpp : Defines the entry point for the console application.
 //
 
+#include <time.h>
+
 #include <fstream>
-#include <string>
 #include <iostream>
-#include <ctime>
+#include <string>
 
-#include "event_wrapper.h"
-#include "trace.h"
-
-#include "BWEStabilityTest.h"
-#include "BWEConvergenceTest.h"
-#include "BWETwoWayLimitFinding.h"
-
-#include "MatlabPlot.h"
-
-//#include "vld.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
 
 #ifdef MATLAB
 MatlabEngine eng;
 #endif
 
-
 class testContainer
 {
 public:
@@ -67,7 +63,6 @@ public:
     std::string _instr;
 };
 
-
 // This is were the test cases are created.
 // Syntax:
 // tests->push_back(new testContainer(

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "BWETwoWayLimitFinding.h"
-#include "TestLoadGenerator.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
 
 
 BWETwoWayLimitFinding::BWETwoWayLimitFinding(
@@ -72,4 +72,3 @@ bool BWETwoWayLimitFinding::StoppingCriterionMaster()
 
     return (_forwLimitReached && _revLimitReached);
 }
-

webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h

@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
 #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
 
-#include "BWETestBase.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h"
 
 class BWETwoWayLimitFinding : public BWETest
 {

webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc

@@ -8,19 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "MatlabPlot.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <sstream>
+
 #ifdef MATLAB
 #include "engine.h"
 #endif
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "tick_util.h"
 
-#include <sstream>
-#include <algorithm>
-#include <math.h>
-#include <stdio.h>
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 
 using namespace webrtc;
 

webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h

@@ -15,7 +15,7 @@
 #include <string>
 #include <vector>
 
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;

webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc

@@ -8,15 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm> // for max function
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+
 #include <stdio.h>
 
-#include "TestLoadGenerator.h"
-#include "TestSenderReceiver.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "tick_util.h"
+#include <algorithm>
+
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 
 
 bool SenderThreadFunction(void *obj)

webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h

@@ -13,8 +13,8 @@
 
 #include <stdlib.h>
 
-#include "typedefs.h"
-#include "module_common_types.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/typedefs.h"
 
 class TestSenderReceiver;
 namespace webrtc {

webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc

@@ -8,18 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <stdio.h>
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
 
-#include "rtp_rtcp.h"
-#include "udp_transport.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
+#include <stdio.h>
 #include <stdlib.h>
 
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/test/channel_transport/udp_transport.h"
+
 #define NR_OF_SOCKET_BUFFERS 500
 
 

webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h

@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
 #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
 
-#include "typedefs.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-#include "udp_transport.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/test/channel_transport/udp_transport.h"
+#include "webrtc/typedefs.h"
 
 class TestLoadGenerator;
 namespace webrtc {

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -10,13 +10,13 @@
 
 #include <algorithm>
 #include <vector>
-#include <gtest/gtest.h>
+#include "testing/gtest/include/gtest/gtest.h"
 
-#include "test_api.h"
+#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 using namespace webrtc;
 

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

@@ -11,12 +11,12 @@
 #include <algorithm>
 #include <vector>
 
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
-#include "test_api.h"
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 using namespace webrtc;
 

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc

@@ -13,7 +13,7 @@
 #include <algorithm>
 #include <vector>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"

webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc

@@ -20,12 +20,12 @@
 #include <ctime>
 #include <list>
 
-#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 
-#include "rtp_utility.h"
-#include "testsupport/fileutils.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 //#define VERBOSE_OUTPUT
 

webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc

@@ -45,7 +45,7 @@
 
 #include <math.h>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 #include "webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"

webrtc/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc

@@ -8,15 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <cassert>
 #include <windows.h>
-#include <iostream>
+
+#include <assert.h>
 #include <tchar.h>
 
-#include "rtp_rtcp.h"
-#include "common_types.h"
-#include "rtcp_utility.h"
-#include "tmmbr_help.h"
+#include <iostream>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 
 #define TEST_STR "Test TMMBR."
 #define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
@@ -1031,4 +1033,3 @@ int _tmain(int argc, _TCHAR* argv[])
 
     return 0;
 }
-