diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 5d02b0e..bc4981b 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -14,7 +14,7 @@
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 594ddf5..f83d320 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -25,9 +25,9 @@
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index c37b83b..95fd32e 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -19,7 +19,7 @@
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc
index 1ec361e..5270415 100644
--- a/webrtc/call/congestion_controller.cc
+++ b/webrtc/call/congestion_controller.cc
@@ -20,8 +20,8 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/video_engine/call_stats.h"
diff --git a/webrtc/common_video/video_render_frames.cc b/webrtc/common_video/video_render_frames.cc
index f4ece5e..a7db15c 100644
--- a/webrtc/common_video/video_render_frames.cc
+++ b/webrtc/common_video/video_render_frames.cc
@@ -12,7 +12,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
diff --git a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
index 79d6cbc..7d5b99b 100644
--- a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
+++ b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
@@ -18,7 +18,7 @@
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/examples/android/media_demo/jni/jni_helpers.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 839a143..13f9a5d 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -45,7 +45,7 @@
 config("audio_coding_config") {
   include_dirs = [
     "main/include",
-    "../interface",
+    "../include",
   ]
 }
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 9e812b2..4b080ba 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -24,7 +24,7 @@
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index dbbf8f8..3aee344 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -33,7 +33,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
index 2aece0f..e2df921 100644
--- a/webrtc/modules/audio_coding/main/acm2/call_statistics.h
+++ b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
index c6942ec..6b50dd0 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
index 6fb37d2..088a319 100644
--- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
@@ -81,13 +81,13 @@
       ],
       'include_dirs': [
         'include',
-        '../../interface',
+        '../../include',
         '<(webrtc_root)',
       ],
       'direct_dependent_settings': {
         'include_dirs': [
           'include',
-          '../../interface',
+          '../../include',
           '<(webrtc_root)',
         ],
       },
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module.h b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
index 1792a25..00f8d54 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
@@ -18,7 +18,7 @@
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
index 489df40..1ca6f9d 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
+++ b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
@@ -13,7 +13,7 @@
 
 #include <map>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index 39d4dab..ff6937e 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -14,7 +14,7 @@
 #include <stdio.h>
 
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc
index d0ae783..0466e02 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc
@@ -15,7 +15,7 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/main/test/PCMFile.h
index 8353898..785ed66 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.h
@@ -16,7 +16,7 @@
 
 #include <string>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h
index c79b63e..6bad755 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.h
@@ -15,7 +15,7 @@
 #include <queue>
 
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
index 5b1e07e..857381d 100644
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/test/Channel.h"
 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
index 20b10a3..9d0986a 100644
--- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
@@ -14,7 +14,7 @@
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index d3f6fa6..0252d1c 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -20,7 +20,7 @@
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index 5140c06..806d02b 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -17,7 +17,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/nack.cc b/webrtc/modules/audio_coding/neteq/nack.cc
index fd3d762..011914b 100644
--- a/webrtc/modules/audio_coding/neteq/nack.cc
+++ b/webrtc/modules/audio_coding/neteq/nack.cc
@@ -15,7 +15,7 @@
 #include <algorithm>  // For std::max.
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 9ed0fc8..d9ae15e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -42,7 +42,7 @@
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h
index 723ed8b..64b325e 100644
--- a/webrtc/modules/audio_coding/neteq/packet.h
+++ b/webrtc/modules/audio_coding/neteq/packet.h
@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc
index cf8e028..7ef40bc 100644
--- a/webrtc/modules/audio_coding/neteq/rtcp.cc
+++ b/webrtc/modules/audio_coding/neteq/rtcp.cc
@@ -15,7 +15,7 @@
 #include <algorithm>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 3fbce8b..56ed72f 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -14,7 +14,7 @@
 #include <map>
 #include <stdio.h>
 #include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 enum stereoModes {
     stereoModeMono,
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
index 3bd2df5..489a8b2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 0a41c6e..c9fe11f 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -14,7 +14,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index adce1cf..7a52408 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -34,7 +34,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.cc b/webrtc/modules/audio_coding/neteq/tools/packet.cc
index b8b27af..2b2fcc2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -12,8 +12,8 @@
 
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index 9b17ba8..dad72ea 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 7150bcf..90d5931 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index e3a829b..b7a3109 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -20,7 +20,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index cd7d7e8..2febf68 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -18,7 +18,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
index 6c16192..53371be 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn
index 3b9e276..36391c7 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -9,15 +9,15 @@
 config("audio_conference_mixer_config") {
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
   include_dirs = [
-    "interface",
-    "../interface",
+    "include",
+    "../include",
   ]
 }
 
 source_set("audio_conference_mixer") {
   sources = [
-    "interface/audio_conference_mixer.h",
-    "interface/audio_conference_mixer_defines.h",
+    "include/audio_conference_mixer.h",
+    "include/audio_conference_mixer_defines.h",
     "source/audio_conference_mixer_impl.cc",
     "source/audio_conference_mixer_impl.h",
     "source/audio_frame_manipulator.cc",
diff --git a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
index 5aa3cc4..9d71795 100644
--- a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
+++ b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
@@ -17,8 +17,8 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_conference_mixer.h',
-        'interface/audio_conference_mixer_defines.h',
+        'include/audio_conference_mixer.h',
+        'include/audio_conference_mixer_defines.h',
         'source/audio_frame_manipulator.cc',
         'source/audio_frame_manipulator.h',
         'source/memory_pool.h',
diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
new file mode 100644
index 0000000..7370442
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+class AudioMixerOutputReceiver;
+class MixerParticipant;
+class Trace;
+
+class AudioConferenceMixer : public Module
+{
+public:
+    enum {kMaximumAmountOfMixedParticipants = 3};
+    enum Frequency
+    {
+        kNbInHz           = 8000,
+        kWbInHz           = 16000,
+        kSwbInHz          = 32000,
+        kFbInHz           = 48000,
+        kLowestPossible   = -1,
+        kDefaultFrequency = kWbInHz
+    };
+
+    // Factory method. Constructor disabled.
+    static AudioConferenceMixer* Create(int id);
+    virtual ~AudioConferenceMixer() {}
+
+    // Module functions
+    int64_t TimeUntilNextProcess() override = 0;
+    int32_t Process() override = 0;
+
+    // Register/unregister a callback class for receiving the mixed audio.
+    virtual int32_t RegisterMixedStreamCallback(
+        AudioMixerOutputReceiver* receiver) = 0;
+    virtual int32_t UnRegisterMixedStreamCallback() = 0;
+
+    // Add/remove participants as candidates for mixing.
+    virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
+                                        bool mixable) = 0;
+    // Returns true if a participant is a candidate for mixing.
+    virtual bool MixabilityStatus(
+        const MixerParticipant& participant) const = 0;
+
+    // Inform the mixer that the participant should always be mixed and not
+    // count toward the number of mixed participants. Note that a participant
+    // must have been added to the mixer (by calling SetMixabilityStatus())
+    // before this function can be successfully called.
+    virtual int32_t SetAnonymousMixabilityStatus(
+        MixerParticipant* participant, bool mixable) = 0;
+    // Returns true if the participant is mixed anonymously.
+    virtual bool AnonymousMixabilityStatus(
+        const MixerParticipant& participant) const = 0;
+
+    // Set the minimum sampling frequency at which to mix. The mixing algorithm
+    // may still choose to mix at a higher samling frequency to avoid
+    // downsampling of audio contributing to the mixed audio.
+    virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
+
+protected:
+    AudioConferenceMixer() {}
+};
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
new file mode 100644
index 0000000..5d58f42
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class MixHistory;
+
+// A callback class that all mixer participants must inherit from/implement.
+class MixerParticipant
+{
+public:
+    // The implementation of this function should update audioFrame with new
+    // audio every time it's called.
+    //
+    // If it returns -1, the frame will not be added to the mix.
+    virtual int32_t GetAudioFrame(int32_t id,
+                                  AudioFrame* audioFrame) = 0;
+
+    // Returns true if the participant was mixed this mix iteration.
+    bool IsMixed() const;
+
+    // This function specifies the sampling frequency needed for the AudioFrame
+    // for future GetAudioFrame(..) calls.
+    virtual int32_t NeededFrequency(int32_t id) const = 0;
+
+    MixHistory* _mixHistory;
+protected:
+    MixerParticipant();
+    virtual ~MixerParticipant();
+};
+
+class AudioMixerOutputReceiver
+{
+public:
+    // This callback function provides the mixed audio for this mix iteration.
+    // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
+    // size according to the size parameter.
+    virtual void NewMixedAudio(const int32_t id,
+                               const AudioFrame& generalAudioFrame,
+                               const AudioFrame** uniqueAudioFrames,
+                               const uint32_t size) = 0;
+protected:
+    AudioMixerOutputReceiver() {}
+    virtual ~AudioMixerOutputReceiver() {}
+};
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
index 7ff3957..7536ce9 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
+
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioMixerOutputReceiver;
@@ -74,4 +76,4 @@
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
index d15b7fc..057c37f 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
@@ -8,10 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
+
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -57,4 +59,4 @@
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 2d2cf9d..0ac9eae 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
index bc9a27e..2466112 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -16,10 +16,10 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
index 636698e..9c5d3b9 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace {
diff --git a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
index d4fbd20..293bfa0 100644
--- a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
+++ b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
@@ -10,8 +10,8 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 8875178..b5e503a 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -10,7 +10,7 @@
 
 config("audio_device_config") {
   include_dirs = [
-    "../interface",
+    "../include",
     "include",
     "dummy",  # Contains dummy audio device implementations.
   ]
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 260e793..522010e 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -16,7 +16,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 #define TAG "AudioManager"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h
index 9cceaac..26caf61 100644
--- a/webrtc/modules/audio_device/android/audio_manager.h
+++ b/webrtc/modules/audio_device/android/audio_manager.h
@@ -19,8 +19,8 @@
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index efd5164..a847180 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -17,8 +17,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.h b/webrtc/modules/audio_device/android/audio_track_jni.h
index 43bfcad..067dc6c 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.h
+++ b/webrtc/modules/audio_device/android/audio_track_jni.h
@@ -18,8 +18,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc
index cb5dc29..6289697 100644
--- a/webrtc/modules/audio_device/android/build_info.cc
+++ b/webrtc/modules/audio_device/android/build_info.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/audio_device/android/build_info.h"
 
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/build_info.h b/webrtc/modules/audio_device/android/build_info.h
index d9b2871..1490fa0 100644
--- a/webrtc/modules/audio_device/android/build_info.h
+++ b/webrtc/modules/audio_device/android/build_info.h
@@ -14,7 +14,7 @@
 #include <jni.h>
 #include <string>
 
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/ensure_initialized.cc b/webrtc/modules/audio_device/android/ensure_initialized.cc
index e8197b7..11b2b25 100644
--- a/webrtc/modules/audio_device/android/ensure_initialized.cc
+++ b/webrtc/modules/audio_device/android/ensure_initialized.cc
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 namespace audiodevicemodule {
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index d96388b..4c4d724 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -22,7 +22,7 @@
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi
index 0678d33..c3b6fba 100644
--- a/webrtc/modules/audio_device/audio_device.gypi
+++ b/webrtc/modules/audio_device/audio_device.gypi
@@ -20,13 +20,13 @@
       ],
       'include_dirs': [
         '.',
-        '../interface',
+        '../include',
         'include',
         'dummy',  # Contains dummy audio device implementations.
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          '../interface',
+          '../include',
           'include',
         ],
       },
diff --git a/webrtc/modules/audio_device/include/audio_device.h b/webrtc/modules/audio_device/include/audio_device.h
index c2c2b88..15e0873 100644
--- a/webrtc/modules/audio_device/include/audio_device.h
+++ b/webrtc/modules/audio_device/include/audio_device.h
@@ -12,7 +12,7 @@
 #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
 
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index f26e9f1..5bf6b3b 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -21,7 +21,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/test/audio_device_test_defines.h b/webrtc/modules/audio_device/test/audio_device_test_defines.h
index 5068646b..cc8e3e3 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_defines.h
+++ b/webrtc/modules/audio_device/test/audio_device_test_defines.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef _WIN32
diff --git a/webrtc/modules/audio_processing/agc/agc.cc b/webrtc/modules/audio_processing/agc/agc.cc
index 706b963..fc78f07 100644
--- a/webrtc/modules/audio_processing/agc/agc.cc
+++ b/webrtc/modules/audio_processing/agc/agc.cc
@@ -19,7 +19,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/agc/histogram.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
index 867022d..e4b753b 100644
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -19,7 +19,7 @@
 
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc
index 66a8a2b..25b99d8 100644
--- a/webrtc/modules/audio_processing/agc/agc_unittest.cc
+++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc
@@ -13,7 +13,7 @@
 #include "gmock/gmock.h"
 #include "gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/agc/test_utils.h"
 
diff --git a/webrtc/modules/audio_processing/agc/histogram.cc b/webrtc/modules/audio_processing/agc/histogram.cc
index 1d3035f..5c66727 100644
--- a/webrtc/modules/audio_processing/agc/histogram.cc
+++ b/webrtc/modules/audio_processing/agc/histogram.cc
@@ -13,7 +13,7 @@
 #include <cmath>
 #include <cstring>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h
index 13dbd2e..e362200 100644
--- a/webrtc/modules/audio_processing/agc/mock_agc.h
+++ b/webrtc/modules/audio_processing/agc/mock_agc.h
@@ -14,7 +14,7 @@
 #include "webrtc/modules/audio_processing/agc/agc.h"
 
 #include "gmock/gmock.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 864633f..48c9488 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -15,7 +15,7 @@
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index c657415..668ec11 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -36,7 +36,7 @@
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index f4c36d0..ed20daa 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -14,7 +14,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 using ::testing::Invoke;
 using ::testing::Return;
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index eac165c..eff791d 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -24,7 +24,7 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 555d9d6..1383bbe 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -22,7 +22,7 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index f53d8ac..93a0138 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -22,7 +22,7 @@
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index 506abaf..b7b7595 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -19,7 +19,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_processing/typing_detection.h b/webrtc/modules/audio_processing/typing_detection.h
index 5fa6456..40608f8 100644
--- a/webrtc/modules/audio_processing/typing_detection.h
+++ b/webrtc/modules/audio_processing/typing_detection.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
index 39ec37e..fce144d 100644
--- a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
+++ b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
@@ -18,7 +18,7 @@
 #include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
 #include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad.cc b/webrtc/modules/audio_processing/vad/standalone_vad.cc
index 468b8ff..1209526 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad.cc
@@ -12,8 +12,8 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
index 942008e..5462d05 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -14,7 +14,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
index 8535d1f..1a59559 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -23,7 +23,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
 }
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
index f509af4..a8a4ead 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -21,7 +21,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_processing/vad/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index 8857ee4..d779562 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -14,7 +14,7 @@
 #include <algorithm>
 #include <utility>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 72831c7..2b9e589 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 using webrtc::RtcpBandwidthObserver;
 using webrtc::BitrateObserver;
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index bb53288..5b20287 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -17,8 +17,8 @@
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 40061d3..baf8857 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -15,7 +15,7 @@
 
 #include <deque>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/include/module.h b/webrtc/modules/include/module.h
new file mode 100644
index 0000000..d02aa95
--- /dev/null
+++ b/webrtc/modules/include/module.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class ProcessThread;
+
+class Module {
+ public:
+  // Returns the number of milliseconds until the module wants a worker
+  // thread to call Process.
+  // This method is called on the same worker thread as Process will
+  // be called on.
+  // TODO(tommi): Almost all implementations of this function, need to know
+  // the current tick count.  Consider passing it as an argument.  It could
+  // also improve the accuracy of when the next callback occurs since the
+  // thread that calls Process() will also have it's tick count reference
+  // which might not match with what the implementations use.
+  virtual int64_t TimeUntilNextProcess() = 0;
+
+  // Process any pending tasks such as timeouts.
+  // Called on a worker thread.
+  virtual int32_t Process() = 0;
+
+  // This method is called when the module is attached to a *running* process
+  // thread or detached from one.  In the case of detaching, |process_thread|
+  // will be nullptr.
+  //
+  // This method will be called in the following cases:
+  //
+  // * Non-null process_thread:
+  //   * ProcessThread::RegisterModule() is called while the thread is running.
+  //   * ProcessThread::Start() is called and RegisterModule has previously
+  //     been called.  The thread will be started immediately after notifying
+  //     all modules.
+  //
+  // * Null process_thread:
+  //   * ProcessThread::DeRegisterModule() is called while the thread is
+  //     running.
+  //   * ProcessThread::Stop() was called and the thread has been stopped.
+  //
+  // NOTE: This method is not called from the worker thread itself, but from
+  //       the thread that registers/deregisters the module or calls Start/Stop.
+  virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
+
+ protected:
+  virtual ~Module() {}
+};
+
+// Reference counted version of the Module interface.
+class RefCountedModule : public Module {
+ public:
+  // Increase the reference count by one.
+  // Returns the incremented reference count.
+  virtual int32_t AddRef() const = 0;
+
+  // Decrease the reference count by one.
+  // Returns the decreased reference count.
+  // Returns 0 if the last reference was just released.
+  // When the reference count reaches 0 the object will self-destruct.
+  virtual int32_t Release() const = 0;
+
+ protected:
+  ~RefCountedModule() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h
new file mode 100644
index 0000000..fb63581
--- /dev/null
+++ b/webrtc/modules/include/module_common_types.h
@@ -0,0 +1,810 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#include <assert.h>
+#include <string.h>  // memcpy
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/rotation.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct RTPAudioHeader {
+  uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
+  uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
+  bool isCNG;                         // is this CNG
+  uint8_t channel;                    // number of channels 2 = stereo
+};
+
+const int16_t kNoPictureId = -1;
+const int16_t kMaxOneBytePictureId = 0x7F;    // 7 bits
+const int16_t kMaxTwoBytePictureId = 0x7FFF;  // 15 bits
+const int16_t kNoTl0PicIdx = -1;
+const uint8_t kNoTemporalIdx = 0xFF;
+const uint8_t kNoSpatialIdx = 0xFF;
+const uint8_t kNoGofIdx = 0xFF;
+const size_t kMaxVp9RefPics = 3;
+const size_t kMaxVp9FramesInGof = 0xFF;  // 8 bits
+const size_t kMaxVp9NumberOfSpatialLayers = 8;
+const int kNoKeyIdx = -1;
+
+struct RTPVideoHeaderVP8 {
+  void InitRTPVideoHeaderVP8() {
+    nonReference = false;
+    pictureId = kNoPictureId;
+    tl0PicIdx = kNoTl0PicIdx;
+    temporalIdx = kNoTemporalIdx;
+    layerSync = false;
+    keyIdx = kNoKeyIdx;
+    partitionId = 0;
+    beginningOfPartition = false;
+  }
+
+  bool nonReference;          // Frame is discardable.
+  int16_t pictureId;          // Picture ID index, 15 bits;
+                              // kNoPictureId if PictureID does not exist.
+  int16_t tl0PicIdx;          // TL0PIC_IDX, 8 bits;
+                              // kNoTl0PicIdx means no value provided.
+  uint8_t temporalIdx;        // Temporal layer index, or kNoTemporalIdx.
+  bool layerSync;             // This frame is a layer sync frame.
+                              // Disabled if temporalIdx == kNoTemporalIdx.
+  int keyIdx;                 // 5 bits; kNoKeyIdx means not used.
+  int partitionId;            // VP8 partition ID
+  bool beginningOfPartition;  // True if this packet is the first
+                              // in a VP8 partition. Otherwise false
+};
+
+enum TemporalStructureMode {
+  kTemporalStructureMode1,    // 1 temporal layer structure - i.e., IPPP...
+  kTemporalStructureMode2,    // 2 temporal layers 0-1-0-1...
+  kTemporalStructureMode3     // 3 temporal layers 0-2-1-2-0-2-1-2...
+};
+
+struct GofInfoVP9 {
+  void SetGofInfoVP9(TemporalStructureMode tm) {
+    switch (tm) {
+      case kTemporalStructureMode1:
+        num_frames_in_gof = 1;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = false;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 1;
+        break;
+      case kTemporalStructureMode2:
+        num_frames_in_gof = 2;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = false;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 2;
+
+        temporal_idx[1] = 1;
+        temporal_up_switch[1] = true;
+        num_ref_pics[1] = 1;
+        pid_diff[1][0] = 1;
+        break;
+      case kTemporalStructureMode3:
+        num_frames_in_gof = 4;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = false;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 4;
+
+        temporal_idx[1] = 2;
+        temporal_up_switch[1] = true;
+        num_ref_pics[1] = 1;
+        pid_diff[1][0] = 1;
+
+        temporal_idx[2] = 1;
+        temporal_up_switch[2] = true;
+        num_ref_pics[2] = 1;
+        pid_diff[2][0] = 2;
+
+        temporal_idx[3] = 2;
+        temporal_up_switch[3] = false;
+        num_ref_pics[3] = 2;
+        pid_diff[3][0] = 1;
+        pid_diff[3][1] = 2;
+        break;
+      default:
+        assert(false);
+    }
+  }
+
+  void CopyGofInfoVP9(const GofInfoVP9& src) {
+    num_frames_in_gof = src.num_frames_in_gof;
+    for (size_t i = 0; i < num_frames_in_gof; ++i) {
+      temporal_idx[i] = src.temporal_idx[i];
+      temporal_up_switch[i] = src.temporal_up_switch[i];
+      num_ref_pics[i] = src.num_ref_pics[i];
+      for (size_t r = 0; r < num_ref_pics[i]; ++r) {
+        pid_diff[i][r] = src.pid_diff[i][r];
+      }
+    }
+  }
+
+  size_t num_frames_in_gof;
+  uint8_t temporal_idx[kMaxVp9FramesInGof];
+  bool temporal_up_switch[kMaxVp9FramesInGof];
+  size_t num_ref_pics[kMaxVp9FramesInGof];
+  int16_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
+};
+
+struct RTPVideoHeaderVP9 {
+  void InitRTPVideoHeaderVP9() {
+    inter_pic_predicted = false;
+    flexible_mode = false;
+    beginning_of_frame = false;
+    end_of_frame = false;
+    ss_data_available = false;
+    picture_id = kNoPictureId;
+    max_picture_id = kMaxTwoBytePictureId;
+    tl0_pic_idx = kNoTl0PicIdx;
+    temporal_idx = kNoTemporalIdx;
+    spatial_idx = kNoSpatialIdx;
+    temporal_up_switch = false;
+    inter_layer_predicted = false;
+    gof_idx = kNoGofIdx;
+    num_ref_pics = 0;
+    num_spatial_layers = 1;
+  }
+
+  bool inter_pic_predicted;  // This layer frame is dependent on previously
+                             // coded frame(s).
+  bool flexible_mode;        // This frame is in flexible mode.
+  bool beginning_of_frame;   // True if this packet is the first in a VP9 layer
+                             // frame.
+  bool end_of_frame;  // True if this packet is the last in a VP9 layer frame.
+  bool ss_data_available;  // True if SS data is available in this payload
+                           // descriptor.
+  int16_t picture_id;      // PictureID index, 15 bits;
+                           // kNoPictureId if PictureID does not exist.
+  int16_t max_picture_id;  // Maximum picture ID index; either 0x7F or 0x7FFF;
+  int16_t tl0_pic_idx;     // TL0PIC_IDX, 8 bits;
+                           // kNoTl0PicIdx means no value provided.
+  uint8_t temporal_idx;    // Temporal layer index, or kNoTemporalIdx.
+  uint8_t spatial_idx;     // Spatial layer index, or kNoSpatialIdx.
+  bool temporal_up_switch;  // True if upswitch to higher frame rate is possible
+                            // starting from this frame.
+  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
+                               // layer frame.
+
+  uint8_t gof_idx;  // Index to predefined temporal frame info in SS data.
+
+  size_t num_ref_pics;  // Number of reference pictures used by this layer
+                        // frame.
+  int16_t pid_diff[kMaxVp9RefPics];  // P_DIFF signaled to derive the PictureID
+                                     // of the reference pictures.
+  int16_t ref_picture_id[kMaxVp9RefPics];  // PictureID of reference pictures.
+
+  // SS data.
+  size_t num_spatial_layers;  // Always populated.
+  bool spatial_layer_resolution_present;
+  uint16_t width[kMaxVp9NumberOfSpatialLayers];
+  uint16_t height[kMaxVp9NumberOfSpatialLayers];
+  GofInfoVP9 gof;
+};
+
+// The packetization types that we support: single, aggregated, and fragmented.
+enum H264PacketizationTypes {
+  kH264SingleNalu,  // This packet contains a single NAL unit.
+  kH264StapA,       // This packet contains STAP-A (single time
+                    // aggregation) packets. If this packet has an
+                    // associated NAL unit type, it'll be for the
+                    // first such aggregated packet.
+  kH264FuA,         // This packet contains a FU-A (fragmentation
+                    // unit) packet, meaning it is a part of a frame
+                    // that was too large to fit into a single packet.
+};
+
+struct RTPVideoHeaderH264 {
+  uint8_t nalu_type;  // The NAL unit type. If this is a header for a
+                      // fragmented packet, it's the NAL unit type of
+                      // the original data. If this is the header for an
+                      // aggregated packet, it's the NAL unit type of
+                      // the first NAL unit in the packet.
+  H264PacketizationTypes packetization_type;
+};
+
+union RTPVideoTypeHeader {
+  RTPVideoHeaderVP8 VP8;
+  RTPVideoHeaderVP9 VP9;
+  RTPVideoHeaderH264 H264;
+};
+
+enum RtpVideoCodecTypes {
+  kRtpVideoNone,
+  kRtpVideoGeneric,
+  kRtpVideoVp8,
+  kRtpVideoVp9,
+  kRtpVideoH264
+};
+// Since RTPVideoHeader is used as a member of a union, it can't have a
+// non-trivial default constructor.
+struct RTPVideoHeader {
+  uint16_t width;  // size
+  uint16_t height;
+  VideoRotation rotation;
+
+  bool isFirstPacket;    // first packet in frame
+  uint8_t simulcastIdx;  // Index if the simulcast encoder creating
+                         // this frame, 0 if not using simulcast.
+  RtpVideoCodecTypes codec;
+  RTPVideoTypeHeader codecHeader;
+};
+union RTPTypeHeader {
+  RTPAudioHeader Audio;
+  RTPVideoHeader Video;
+};
+
+struct WebRtcRTPHeader {
+  RTPHeader header;
+  FrameType frameType;
+  RTPTypeHeader type;
+  // NTP time of the capture time in local timebase in milliseconds.
+  int64_t ntp_time_ms;
+};
+
+class RTPFragmentationHeader {
+ public:
+  RTPFragmentationHeader()
+      : fragmentationVectorSize(0),
+        fragmentationOffset(NULL),
+        fragmentationLength(NULL),
+        fragmentationTimeDiff(NULL),
+        fragmentationPlType(NULL) {};
+
+  ~RTPFragmentationHeader() {
+    delete[] fragmentationOffset;
+    delete[] fragmentationLength;
+    delete[] fragmentationTimeDiff;
+    delete[] fragmentationPlType;
+  }
+
+  void CopyFrom(const RTPFragmentationHeader& src) {
+    if (this == &src) {
+      return;
+    }
+
+    if (src.fragmentationVectorSize != fragmentationVectorSize) {
+      // new size of vectors
+
+      // delete old
+      delete[] fragmentationOffset;
+      fragmentationOffset = NULL;
+      delete[] fragmentationLength;
+      fragmentationLength = NULL;
+      delete[] fragmentationTimeDiff;
+      fragmentationTimeDiff = NULL;
+      delete[] fragmentationPlType;
+      fragmentationPlType = NULL;
+
+      if (src.fragmentationVectorSize > 0) {
+        // allocate new
+        if (src.fragmentationOffset) {
+          fragmentationOffset = new size_t[src.fragmentationVectorSize];
+        }
+        if (src.fragmentationLength) {
+          fragmentationLength = new size_t[src.fragmentationVectorSize];
+        }
+        if (src.fragmentationTimeDiff) {
+          fragmentationTimeDiff = new uint16_t[src.fragmentationVectorSize];
+        }
+        if (src.fragmentationPlType) {
+          fragmentationPlType = new uint8_t[src.fragmentationVectorSize];
+        }
+      }
+      // set new size
+      fragmentationVectorSize = src.fragmentationVectorSize;
+    }
+
+    if (src.fragmentationVectorSize > 0) {
+      // copy values
+      if (src.fragmentationOffset) {
+        memcpy(fragmentationOffset, src.fragmentationOffset,
+               src.fragmentationVectorSize * sizeof(size_t));
+      }
+      if (src.fragmentationLength) {
+        memcpy(fragmentationLength, src.fragmentationLength,
+               src.fragmentationVectorSize * sizeof(size_t));
+      }
+      if (src.fragmentationTimeDiff) {
+        memcpy(fragmentationTimeDiff, src.fragmentationTimeDiff,
+               src.fragmentationVectorSize * sizeof(uint16_t));
+      }
+      if (src.fragmentationPlType) {
+        memcpy(fragmentationPlType, src.fragmentationPlType,
+               src.fragmentationVectorSize * sizeof(uint8_t));
+      }
+    }
+  }
+
+  void VerifyAndAllocateFragmentationHeader(const size_t size) {
+    assert(size <= std::numeric_limits<uint16_t>::max());
+    const uint16_t size16 = static_cast<uint16_t>(size);
+    if (fragmentationVectorSize < size16) {
+      uint16_t oldVectorSize = fragmentationVectorSize;
+      {
+        // offset
+        size_t* oldOffsets = fragmentationOffset;
+        fragmentationOffset = new size_t[size16];
+        memset(fragmentationOffset + oldVectorSize, 0,
+               sizeof(size_t) * (size16 - oldVectorSize));
+        // copy old values
+        memcpy(fragmentationOffset, oldOffsets,
+               sizeof(size_t) * oldVectorSize);
+        delete[] oldOffsets;
+      }
+      // length
+      {
+        size_t* oldLengths = fragmentationLength;
+        fragmentationLength = new size_t[size16];
+        memset(fragmentationLength + oldVectorSize, 0,
+               sizeof(size_t) * (size16 - oldVectorSize));
+        memcpy(fragmentationLength, oldLengths,
+               sizeof(size_t) * oldVectorSize);
+        delete[] oldLengths;
+      }
+      // time diff
+      {
+        uint16_t* oldTimeDiffs = fragmentationTimeDiff;
+        fragmentationTimeDiff = new uint16_t[size16];
+        memset(fragmentationTimeDiff + oldVectorSize, 0,
+               sizeof(uint16_t) * (size16 - oldVectorSize));
+        memcpy(fragmentationTimeDiff, oldTimeDiffs,
+               sizeof(uint16_t) * oldVectorSize);
+        delete[] oldTimeDiffs;
+      }
+      // payload type
+      {
+        uint8_t* oldTimePlTypes = fragmentationPlType;
+        fragmentationPlType = new uint8_t[size16];
+        memset(fragmentationPlType + oldVectorSize, 0,
+               sizeof(uint8_t) * (size16 - oldVectorSize));
+        memcpy(fragmentationPlType, oldTimePlTypes,
+               sizeof(uint8_t) * oldVectorSize);
+        delete[] oldTimePlTypes;
+      }
+      fragmentationVectorSize = size16;
+    }
+  }
+
+  uint16_t fragmentationVectorSize;  // Number of fragmentations
+  size_t* fragmentationOffset;       // Offset of pointer to data for each
+                                     // fragmentation
+  size_t* fragmentationLength;       // Data size for each fragmentation
+  uint16_t* fragmentationTimeDiff;   // Timestamp difference relative "now" for
+                                     // each fragmentation
+  uint8_t* fragmentationPlType;      // Payload type of each fragmentation
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
+};
+
+struct RTCPVoIPMetric {
+  // RFC 3611 4.7
+  uint8_t lossRate;
+  uint8_t discardRate;
+  uint8_t burstDensity;
+  uint8_t gapDensity;
+  uint16_t burstDuration;
+  uint16_t gapDuration;
+  uint16_t roundTripDelay;
+  uint16_t endSystemDelay;
+  uint8_t signalLevel;
+  uint8_t noiseLevel;
+  uint8_t RERL;
+  uint8_t Gmin;
+  uint8_t Rfactor;
+  uint8_t extRfactor;
+  uint8_t MOSLQ;
+  uint8_t MOSCQ;
+  uint8_t RXconfig;
+  uint16_t JBnominal;
+  uint16_t JBmax;
+  uint16_t JBabsMax;
+};
+
+// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
+// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
+// loss model. The packet masks are defined in
+// modules/rtp_rtcp/fec_private_tables_random(bursty).h
+enum FecMaskType {
+  kFecMaskRandom,
+  kFecMaskBursty,
+};
+
+// Struct containing forward error correction settings.
+struct FecProtectionParams {
+  int fec_rate;
+  bool use_uep_protection;
+  int max_fec_frames;
+  FecMaskType fec_mask_type;
+};
+
+// Interface used by the CallStats class to distribute call statistics.
+// Callbacks will be triggered as soon as the class has been registered to a
+// CallStats object using RegisterStatsObserver.
+class CallStatsObserver {
+ public:
+  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
+
+  virtual ~CallStatsObserver() {}
+};
+
+struct VideoContentMetrics {
+  VideoContentMetrics()
+      : motion_magnitude(0.0f),
+        spatial_pred_err(0.0f),
+        spatial_pred_err_h(0.0f),
+        spatial_pred_err_v(0.0f) {}
+
+  void Reset() {
+    motion_magnitude = 0.0f;
+    spatial_pred_err = 0.0f;
+    spatial_pred_err_h = 0.0f;
+    spatial_pred_err_v = 0.0f;
+  }
+  float motion_magnitude;
+  float spatial_pred_err;
+  float spatial_pred_err_h;
+  float spatial_pred_err_v;
+};
+
+/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
+ * allows for adding and subtracting frames while keeping track of the resulting
+ * states.
+ *
+ * Notes
+ * - The total number of samples in |data_| is
+ *   samples_per_channel_ * num_channels_
+ *
+ * - Stereo data is interleaved starting with the left channel.
+ *
+ * - The +operator assume that you would never add exactly opposite frames when
+ *   deciding the resulting state. To do this use the -operator.
+ */
+class AudioFrame {
+ public:
+  // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
+  static const size_t kMaxDataSizeSamples = 3840;
+
+  enum VADActivity {
+    kVadActive = 0,
+    kVadPassive = 1,
+    kVadUnknown = 2
+  };
+  enum SpeechType {
+    kNormalSpeech = 0,
+    kPLC = 1,
+    kCNG = 2,
+    kPLCCNG = 3,
+    kUndefined = 4
+  };
+
+  AudioFrame();
+  virtual ~AudioFrame() {}
+
+  // Resets all members to their default state (except does not modify the
+  // contents of |data_|).
+  void Reset();
+
+  // |interleaved_| is not changed by this method.
+  void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
+                   size_t samples_per_channel, int sample_rate_hz,
+                   SpeechType speech_type, VADActivity vad_activity,
+                   int num_channels = 1, uint32_t energy = -1);
+
+  AudioFrame& Append(const AudioFrame& rhs);
+
+  void CopyFrom(const AudioFrame& src);
+
+  void Mute();
+
+  AudioFrame& operator>>=(const int rhs);
+  AudioFrame& operator+=(const AudioFrame& rhs);
+  AudioFrame& operator-=(const AudioFrame& rhs);
+
+  int id_;
+  // RTP timestamp of the first sample in the AudioFrame.
+  uint32_t timestamp_;
+  // Time since the first frame in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t elapsed_time_ms_;
+  // NTP time of the estimated capture time in local timebase in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t ntp_time_ms_;
+  int16_t data_[kMaxDataSizeSamples];
+  size_t samples_per_channel_;
+  int sample_rate_hz_;
+  int num_channels_;
+  SpeechType speech_type_;
+  VADActivity vad_activity_;
+  // Note that there is no guarantee that |energy_| is correct. Any user of this
+  // member must verify that the value is correct.
+  // TODO(henrike) Remove |energy_|.
+  // See https://code.google.com/p/webrtc/issues/detail?id=3315.
+  uint32_t energy_;
+  bool interleaved_;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
+};
+
+inline AudioFrame::AudioFrame()
+    : data_() {
+  Reset();
+}
+
+inline void AudioFrame::Reset() {
+  id_ = -1;
+  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
+  // to an invalid value, or add a new member to indicate invalidity.
+  timestamp_ = 0;
+  elapsed_time_ms_ = -1;
+  ntp_time_ms_ = -1;
+  samples_per_channel_ = 0;
+  sample_rate_hz_ = 0;
+  num_channels_ = 0;
+  speech_type_ = kUndefined;
+  vad_activity_ = kVadUnknown;
+  energy_ = 0xffffffff;
+  interleaved_ = true;
+}
+
+inline void AudioFrame::UpdateFrame(int id,
+                                    uint32_t timestamp,
+                                    const int16_t* data,
+                                    size_t samples_per_channel,
+                                    int sample_rate_hz,
+                                    SpeechType speech_type,
+                                    VADActivity vad_activity,
+                                    int num_channels,
+                                    uint32_t energy) {
+  id_ = id;
+  timestamp_ = timestamp;
+  samples_per_channel_ = samples_per_channel;
+  sample_rate_hz_ = sample_rate_hz;
+  speech_type_ = speech_type;
+  vad_activity_ = vad_activity;
+  num_channels_ = num_channels;
+  energy_ = energy;
+
+  assert(num_channels >= 0);
+  const size_t length = samples_per_channel * num_channels;
+  assert(length <= kMaxDataSizeSamples);
+  if (data != NULL) {
+    memcpy(data_, data, sizeof(int16_t) * length);
+  } else {
+    memset(data_, 0, sizeof(int16_t) * length);
+  }
+}
+
+inline void AudioFrame::CopyFrom(const AudioFrame& src) {
+  if (this == &src) return;
+
+  id_ = src.id_;
+  timestamp_ = src.timestamp_;
+  elapsed_time_ms_ = src.elapsed_time_ms_;
+  ntp_time_ms_ = src.ntp_time_ms_;
+  samples_per_channel_ = src.samples_per_channel_;
+  sample_rate_hz_ = src.sample_rate_hz_;
+  speech_type_ = src.speech_type_;
+  vad_activity_ = src.vad_activity_;
+  num_channels_ = src.num_channels_;
+  energy_ = src.energy_;
+  interleaved_ = src.interleaved_;
+
+  assert(num_channels_ >= 0);
+  const size_t length = samples_per_channel_ * num_channels_;
+  assert(length <= kMaxDataSizeSamples);
+  memcpy(data_, src.data_, sizeof(int16_t) * length);
+}
+
+inline void AudioFrame::Mute() {
+  memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
+}
+
+inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
+  assert((num_channels_ > 0) && (num_channels_ < 3));
+  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+
+  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+    data_[i] = static_cast<int16_t>(data_[i] >> rhs);
+  }
+  return *this;
+}
+
+inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) {
+  // Sanity check
+  assert((num_channels_ > 0) && (num_channels_ < 3));
+  assert(interleaved_ == rhs.interleaved_);
+  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+  if (num_channels_ != rhs.num_channels_) return *this;
+
+  if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
+    vad_activity_ = kVadActive;
+  } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
+    vad_activity_ = kVadUnknown;
+  }
+  if (speech_type_ != rhs.speech_type_) {
+    speech_type_ = kUndefined;
+  }
+
+  size_t offset = samples_per_channel_ * num_channels_;
+  for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
+    data_[offset + i] = rhs.data_[i];
+  }
+  samples_per_channel_ += rhs.samples_per_channel_;
+  return *this;
+}
+
+namespace {
+inline int16_t ClampToInt16(int32_t input) {
+  if (input < -0x00008000) {
+    return -0x8000;
+  } else if (input > 0x00007FFF) {
+    return 0x7FFF;
+  } else {
+    return static_cast<int16_t>(input);
+  }
+}
+}
+
+inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
+  // Sanity check
+  assert((num_channels_ > 0) && (num_channels_ < 3));
+  assert(interleaved_ == rhs.interleaved_);
+  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+  if (num_channels_ != rhs.num_channels_) return *this;
+
+  bool noPrevData = false;
+  if (samples_per_channel_ != rhs.samples_per_channel_) {
+    if (samples_per_channel_ == 0) {
+      // special case we have no data to start with
+      samples_per_channel_ = rhs.samples_per_channel_;
+      noPrevData = true;
+    } else {
+      return *this;
+    }
+  }
+
+  if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
+    vad_activity_ = kVadActive;
+  } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
+    vad_activity_ = kVadUnknown;
+  }
+
+  if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
+
+  if (noPrevData) {
+    memcpy(data_, rhs.data_,
+           sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
+  } else {
+    // IMPROVEMENT this can be done very fast in assembly
+    for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+      int32_t wrap_guard =
+          static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
+      data_[i] = ClampToInt16(wrap_guard);
+    }
+  }
+  energy_ = 0xffffffff;
+  return *this;
+}
+
+inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) {
+  // Sanity check
+  assert((num_channels_ > 0) && (num_channels_ < 3));
+  assert(interleaved_ == rhs.interleaved_);
+  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+
+  if ((samples_per_channel_ != rhs.samples_per_channel_) ||
+      (num_channels_ != rhs.num_channels_)) {
+    return *this;
+  }
+  if ((vad_activity_ != kVadPassive) || rhs.vad_activity_ != kVadPassive) {
+    vad_activity_ = kVadUnknown;
+  }
+  speech_type_ = kUndefined;
+
+  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+    int32_t wrap_guard =
+        static_cast<int32_t>(data_[i]) - static_cast<int32_t>(rhs.data_[i]);
+    data_[i] = ClampToInt16(wrap_guard);
+  }
+  energy_ = 0xffffffff;
+  return *this;
+}
+
+inline bool IsNewerSequenceNumber(uint16_t sequence_number,
+                                  uint16_t prev_sequence_number) {
+  // Distinguish between elements that are exactly 0x8000 apart.
+  // If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false
+  // rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false.
+  if (static_cast<uint16_t>(sequence_number - prev_sequence_number) == 0x8000) {
+    return sequence_number > prev_sequence_number;
+  }
+  return sequence_number != prev_sequence_number &&
+         static_cast<uint16_t>(sequence_number - prev_sequence_number) < 0x8000;
+}
+
+inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
+  // Distinguish between elements that are exactly 0x80000000 apart.
+  // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true,
+  // IsNewer(t2,t1)=false
+  // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
+  if (static_cast<uint32_t>(timestamp - prev_timestamp) == 0x80000000) {
+    return timestamp > prev_timestamp;
+  }
+  return timestamp != prev_timestamp &&
+         static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
+}
+
+inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
+                                     uint16_t sequence_number2) {
+  return IsNewerSequenceNumber(sequence_number1, sequence_number2)
+             ? sequence_number1
+             : sequence_number2;
+}
+
+inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
+  return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
+}
+
+// Utility class to unwrap a sequence number to a larger type, for easier
+// handling large ranges. Note that sequence numbers will never be unwrapped
+// to a negative value.
+class SequenceNumberUnwrapper {
+ public:
+  SequenceNumberUnwrapper() : last_seq_(-1) {}
+
+  // Get the unwrapped sequence, but don't update the internal state.
+  int64_t UnwrapWithoutUpdate(uint16_t sequence_number) {
+    if (last_seq_ == -1)
+      return sequence_number;
+
+    uint16_t cropped_last = static_cast<uint16_t>(last_seq_);
+    int64_t delta = sequence_number - cropped_last;
+    if (IsNewerSequenceNumber(sequence_number, cropped_last)) {
+      if (delta < 0)
+        delta += (1 << 16);  // Wrap forwards.
+    } else if (delta > 0 && (last_seq_ + delta - (1 << 16)) >= 0) {
+      // If sequence_number is older but delta is positive, this is a backwards
+      // wrap-around. However, don't wrap backwards past 0 (unwrapped).
+      delta -= (1 << 16);
+    }
+
+    return last_seq_ + delta;
+  }
+
+  // Only update the internal state to the specified last (unwrapped) sequence.
+  void UpdateLast(int64_t last_sequence) { last_seq_ = last_sequence; }
+
+  // Unwrap the sequence number and update the internal state.
+  int64_t Unwrap(uint16_t sequence_number) {
+    int64_t unwrapped = UnwrapWithoutUpdate(sequence_number);
+    UpdateLast(unwrapped);
+    return unwrapped;
+  }
+
+ private:
+  int64_t last_seq_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/interface/module.h b/webrtc/modules/interface/module.h
index ffd3065..06054c3 100644
--- a/webrtc/modules/interface/module.h
+++ b/webrtc/modules/interface/module.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_INTERFACE_MODULE_H_
-#define MODULES_INTERFACE_MODULE_H_
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
+
+#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
 
 #include "webrtc/typedefs.h"
 
@@ -78,4 +80,4 @@
 
 }  // namespace webrtc
 
-#endif  // MODULES_INTERFACE_MODULE_H_
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 45e93d8..e76a714 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
 
 #include <assert.h>
 #include <string.h>  // memcpy
@@ -807,4 +809,4 @@
 
 }  // namespace webrtc
 
-#endif  // MODULE_COMMON_TYPES_H
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn
index 05cfb4e..70abb51 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -14,8 +14,8 @@
 
 source_set("media_file") {
   sources = [
-    "interface/media_file.h",
-    "interface/media_file_defines.h",
+    "include/media_file.h",
+    "include/media_file_defines.h",
     "source/media_file_impl.cc",
     "source/media_file_impl.h",
     "source/media_file_utility.cc",
diff --git a/webrtc/modules/media_file/include/media_file.h b/webrtc/modules/media_file/include/media_file.h
new file mode 100644
index 0000000..22c727e
--- /dev/null
+++ b/webrtc/modules/media_file/include/media_file.h
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class MediaFile : public Module
+{
+public:
+    // Factory method. Constructor disabled. id is the identifier for the
+    // MediaFile instance.
+    static MediaFile* CreateMediaFile(const int32_t id);
+    static void DestroyMediaFile(MediaFile* module);
+
+    // Put 10-60ms of audio data from file into the audioBuffer depending on
+    // codec frame size. dataLengthInBytes is both an input and output
+    // parameter. As input parameter it indicates the size of audioBuffer.
+    // As output parameter it indicates the number of bytes written to
+    // audioBuffer.
+    // Note: This API only play mono audio but can be used on file containing
+    // audio with more channels (in which case the audio will be converted to
+    // mono).
+    virtual int32_t PlayoutAudioData(
+        int8_t* audioBuffer,
+        size_t& dataLengthInBytes) = 0;
+
+    // Put 10-60ms, depending on codec frame size, of audio data from file into
+    // audioBufferLeft and audioBufferRight. The buffers contain the left and
+    // right channel of played out stereo audio.
+    // dataLengthInBytes is both an input and output parameter. As input
+    // parameter it indicates the size of both audioBufferLeft and
+    // audioBufferRight. As output parameter it indicates the number of bytes
+    // written to both audio buffers.
+    // Note: This API can only be successfully called for WAV files with stereo
+    // audio.
+    virtual int32_t PlayoutStereoData(
+        int8_t* audioBufferLeft,
+        int8_t* audioBufferRight,
+        size_t& dataLengthInBytes) = 0;
+
+    // Open the file specified by fileName (relative path is allowed) for
+    // reading. FileCallback::PlayNotification(..) will be called after
+    // notificationTimeMs of the file has been played if notificationTimeMs is
+    // greater than zero. If loop is true the file will be played until
+    // StopPlaying() is called. When end of file is reached the file is read
+    // from the start. format specifies the type of file fileName refers to.
+    // codecInst specifies the encoding of the audio data. Note that
+    // file formats that contain this information (like WAV files) don't need to
+    // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+    // specify what part of the file should be read. From startPointMs ms to
+    // stopPointMs ms.
+    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+    // mono). Stereo audio is only supported for WAV files.
+    virtual int32_t StartPlayingAudioFile(
+        const char* fileName,
+        const uint32_t notificationTimeMs = 0,
+        const bool loop                         = false,
+        const FileFormats format                = kFileFormatPcm16kHzFile,
+        const CodecInst* codecInst              = NULL,
+        const uint32_t startPointMs       = 0,
+        const uint32_t stopPointMs        = 0) = 0;
+
+    // Prepare for playing audio from stream.
+    // FileCallback::PlayNotification(..) will be called after
+    // notificationTimeMs of the file has been played if notificationTimeMs is
+    // greater than zero. format specifies the type of file fileName refers to.
+    // codecInst specifies the encoding of the audio data. Note that
+    // file formats that contain this information (like WAV files) don't need to
+    // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+    // specify what part of the file should be read. From startPointMs ms to
+    // stopPointMs ms.
+    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+    // mono). Stereo audio is only supported for WAV files.
+    virtual int32_t StartPlayingAudioStream(
+        InStream& stream,
+        const uint32_t notificationTimeMs = 0,
+        const FileFormats    format             = kFileFormatPcm16kHzFile,
+        const CodecInst*     codecInst          = NULL,
+        const uint32_t startPointMs       = 0,
+        const uint32_t stopPointMs        = 0) = 0;
+
+    // Stop playing from file or stream.
+    virtual int32_t StopPlaying() = 0;
+
+    // Return true if playing.
+    virtual bool IsPlaying() = 0;
+
+
+    // Set durationMs to the number of ms that has been played from file.
+    virtual int32_t PlayoutPositionMs(
+        uint32_t& durationMs) const = 0;
+
+    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+    // to file. The audio frame size is determined by the codecInst.pacsize
+    // parameter of the last sucessfull StartRecordingAudioFile(..) call.
+    // Note: bufferLength must be exactly one frame.
+    virtual int32_t IncomingAudioData(
+        const int8_t* audioBuffer,
+        const size_t bufferLength) = 0;
+
+    // Open/creates file specified by fileName for writing (relative path is
+    // allowed). FileCallback::RecordNotification(..) will be called after
+    // notificationTimeMs of audio data has been recorded if
+    // notificationTimeMs is greater than zero.
+    // format specifies the type of file that should be created/opened.
+    // codecInst specifies the encoding of the audio data. maxSizeBytes
+    // specifies the number of bytes allowed to be written to file if it is
+    // greater than zero.
+    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+    // mono). Stereo is only supported for WAV files.
+    virtual int32_t StartRecordingAudioFile(
+        const char*  fileName,
+        const FileFormats    format,
+        const CodecInst&     codecInst,
+        const uint32_t notificationTimeMs = 0,
+        const uint32_t maxSizeBytes       = 0) = 0;
+
+    // Prepare for recording audio to stream.
+    // FileCallback::RecordNotification(..) will be called after
+    // notificationTimeMs of audio data has been recorded if
+    // notificationTimeMs is greater than zero.
+    // format specifies the type of file that stream should correspond to.
+    // codecInst specifies the encoding of the audio data.
+    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+    // mono). Stereo is only supported for WAV files.
+    virtual int32_t StartRecordingAudioStream(
+        OutStream&           stream,
+        const FileFormats    format,
+        const CodecInst&     codecInst,
+        const uint32_t notificationTimeMs = 0) = 0;
+
+    // Stop recording to file or stream.
+    virtual int32_t StopRecording() = 0;
+
+    // Return true if recording.
+    virtual bool IsRecording() = 0;
+
+    // Set durationMs to the number of ms that has been recorded to file.
+    virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0;
+
+    // Return true if recording or playing is stereo.
+    virtual bool IsStereo() = 0;
+
+    // Register callback to receive media file related notifications. Disables
+    // callbacks if callback is NULL.
+    virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
+
+    // Set durationMs to the size of the file (in ms) specified by fileName.
+    // format specifies the type of file fileName refers to. freqInHz specifies
+    // the sampling frequency of the file.
+    virtual int32_t FileDurationMs(
+        const char*  fileName,
+        uint32_t&      durationMs,
+        const FileFormats    format,
+        const uint32_t freqInHz = 16000) = 0;
+
+    // Update codecInst according to the current audio codec being used for
+    // reading or writing.
+    virtual int32_t codec_info(CodecInst& codecInst) const = 0;
+
+protected:
+    MediaFile() {}
+    virtual ~MediaFile() {}
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/include/media_file_defines.h b/webrtc/modules/media_file/include/media_file_defines.h
new file mode 100644
index 0000000..05e7e72
--- /dev/null
+++ b/webrtc/modules/media_file/include/media_file_defines.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+// Callback class for the MediaFile class.
+class FileCallback
+{
+public:
+    virtual ~FileCallback(){}
+
+    // This function is called by MediaFile when a file has been playing for
+    // durationMs ms. id is the identifier for the MediaFile instance calling
+    // the callback.
+    virtual void PlayNotification(const int32_t id,
+                                  const uint32_t durationMs) = 0;
+
+    // This function is called by MediaFile when a file has been recording for
+    // durationMs ms. id is the identifier for the MediaFile instance calling
+    // the callback.
+    virtual void RecordNotification(const int32_t id,
+                                    const uint32_t durationMs) = 0;
+
+    // This function is called by MediaFile when a file has been stopped
+    // playing. id is the identifier for the MediaFile instance calling the
+    // callback.
+    virtual void PlayFileEnded(const int32_t id) = 0;
+
+    // This function is called by MediaFile when a file has been stopped
+    // recording. id is the identifier for the MediaFile instance calling the
+    // callback.
+    virtual void RecordFileEnded(const int32_t id) = 0;
+
+protected:
+    FileCallback() {}
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/interface/media_file.h
index 5b09ad4..3be5f08 100644
--- a/webrtc/modules/media_file/interface/media_file.h
+++ b/webrtc/modules/media_file/interface/media_file.h
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+
+#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -177,4 +179,4 @@
     virtual ~MediaFile() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/interface/media_file_defines.h
index ded71a8..345fe54 100644
--- a/webrtc/modules/media_file/interface/media_file_defines.h
+++ b/webrtc/modules/media_file/interface/media_file_defines.h
@@ -8,11 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+
+#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,4 +50,4 @@
     FileCallback() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/media_file.gypi b/webrtc/modules/media_file/media_file.gypi
index 4ec80c3..b3d8968 100644
--- a/webrtc/modules/media_file/media_file.gypi
+++ b/webrtc/modules/media_file/media_file.gypi
@@ -17,8 +17,8 @@
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
-        'interface/media_file.h',
-        'interface/media_file_defines.h',
+        'include/media_file.h',
+        'include/media_file_defines.h',
         'source/media_file_impl.cc',
         'source/media_file_impl.h',
         'source/media_file_utility.cc',
diff --git a/webrtc/modules/media_file/source/media_file_impl.h b/webrtc/modules/media_file/source/media_file_impl.h
index cdb54d8..6184979 100644
--- a/webrtc/modules/media_file/source/media_file_impl.h
+++ b/webrtc/modules/media_file/source/media_file_impl.h
@@ -12,9 +12,9 @@
 #define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
 #include "webrtc/modules/media_file/source/media_file_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/source/media_file_unittest.cc
index 370d132..4a50c2e 100644
--- a/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/webrtc/modules/media_file/source/media_file_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/webrtc/modules/media_file/source/media_file_utility.cc b/webrtc/modules/media_file/source/media_file_utility.cc
index 61ae442..fad7fe4 100644
--- a/webrtc/modules/media_file/source/media_file_utility.cc
+++ b/webrtc/modules/media_file/source/media_file_utility.cc
@@ -19,7 +19,7 @@
 #include "webrtc/common_audio/wav_header.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/source/media_file_utility.h
index 2823cec..65921bd 100644
--- a/webrtc/modules/media_file/source/media_file_utility.h
+++ b/webrtc/modules/media_file/source/media_file_utility.h
@@ -15,7 +15,7 @@
 #include <stdio.h>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
 
 namespace webrtc {
 class InStream;
diff --git a/webrtc/modules/module_common_types_unittest.cc b/webrtc/modules/module_common_types_unittest.cc
index bc0b7a1..acd5847 100644
--- a/webrtc/modules/module_common_types_unittest.cc
+++ b/webrtc/modules/module_common_types_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/webrtc/modules/pacing/include/paced_sender.h b/webrtc/modules/pacing/include/paced_sender.h
index f142f55..2c8e36e 100644
--- a/webrtc/modules/pacing/include/paced_sender.h
+++ b/webrtc/modules/pacing/include/paced_sender.h
@@ -16,8 +16,8 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h
index 9d461d1..7a65295 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/include/packet_router.h
@@ -19,7 +19,7 @@
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 5d7ae17..4088715 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -16,7 +16,7 @@
 #include <queue>
 #include <set>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/bitrate_prober.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index 563773b..7af74ec 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -12,8 +12,8 @@
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index eecb137..f17f797 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -14,7 +14,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/base/scoped_ptr.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index 4bd9d8c..0734cbf 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -17,9 +17,9 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
index 3dee305..961d6bf 100644
--- a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
+++ b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -14,7 +14,7 @@
 #include <cassert>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
index bb69a8a..56e9c14 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
@@ -13,7 +13,7 @@
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
index b7f9f65..5c45370 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -15,7 +15,7 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
index e867ff7..98a68b3 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
index f837638..f2ff768 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
index 64dfa85..b3df712 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_BASELINEFILE_H_
 
 #include <string>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
index 2881eba..d470324 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
@@ -16,7 +16,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 6b24cf3..6b5f04b 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -24,12 +24,12 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/random.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
index d77447f..171d196 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
@@ -21,7 +21,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
index eee90cf..bf23d09 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -20,7 +20,7 @@
 #include <list>
 #include <map>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/voice_engine/channel.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index b18b9f0..9599b01 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
index a02abc6..b7e4f97 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
@@ -16,7 +16,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet.h b/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 11885a4..4a361c4 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -16,7 +16,7 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
index f70c212..793e064 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
@@ -14,10 +14,10 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index f1faa49..3bcbc0a 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
@@ -15,7 +15,7 @@
 #include <sstream>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index c42647e..0b3741d 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -17,7 +17,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index 9493805..083d1fd 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -17,8 +17,8 @@
 #include "gflags/gflags.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace flags {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 19e4a07..4574faf 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -14,8 +14,8 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 class Observer : public webrtc::RemoteBitrateObserver {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index e277481..bf69872 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -14,8 +14,8 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 int main(int argc, char** argv) {
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
index f2e073a..332590b 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -15,7 +15,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
index 58829b0..93f30e6 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
@@ -15,8 +15,8 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
index b2bc646..64d0e55 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -18,9 +18,9 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index c651424..6239c04 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -10,14 +10,14 @@
 
 source_set("rtp_rtcp") {
   sources = [
-    "interface/fec_receiver.h",
-    "interface/receive_statistics.h",
-    "interface/remote_ntp_time_estimator.h",
-    "interface/rtp_header_parser.h",
-    "interface/rtp_payload_registry.h",
-    "interface/rtp_receiver.h",
-    "interface/rtp_rtcp.h",
-    "interface/rtp_rtcp_defines.h",
+    "include/fec_receiver.h",
+    "include/receive_statistics.h",
+    "include/remote_ntp_time_estimator.h",
+    "include/rtp_header_parser.h",
+    "include/rtp_payload_registry.h",
+    "include/rtp_receiver.h",
+    "include/rtp_rtcp.h",
+    "include/rtp_rtcp_defines.h",
     "mocks/mock_rtp_rtcp.h",
     "source/bitrate.cc",
     "source/bitrate.h",
diff --git a/webrtc/modules/rtp_rtcp/include/fec_receiver.h b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
new file mode 100644
index 0000000..65e85ad
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct FecPacketCounter {
+  FecPacketCounter()
+      : num_packets(0),
+        num_fec_packets(0),
+        num_recovered_packets(0) {}
+
+  size_t num_packets;            // Number of received packets.
+  size_t num_fec_packets;        // Number of received FEC packets.
+  size_t num_recovered_packets;  // Number of recovered media packets using FEC.
+};
+
+class FecReceiver {
+ public:
+  static FecReceiver* Create(RtpData* callback);
+
+  virtual ~FecReceiver() {}
+
+  virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
+                                       const uint8_t* incoming_rtp_packet,
+                                       size_t packet_length,
+                                       uint8_t ulpfec_payload_type) = 0;
+
+  virtual int32_t ProcessReceivedFec() = 0;
+
+  virtual FecPacketCounter GetPacketCounter() const = 0;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
new file mode 100644
index 0000000..b4a7cd0
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#include <map>
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+
+class StreamStatistician {
+ public:
+  virtual ~StreamStatistician();
+
+  virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
+  virtual void GetDataCounters(size_t* bytes_received,
+                               uint32_t* packets_received) const = 0;
+
+  // Gets received stream data counters (includes reset counter values).
+  virtual void GetReceiveStreamDataCounters(
+      StreamDataCounters* data_counters) const = 0;
+
+  virtual uint32_t BitrateReceived() const = 0;
+
+  // Returns true if the packet with RTP header |header| is likely to be a
+  // retransmitted packet, false otherwise.
+  virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
+                                       int64_t min_rtt) const = 0;
+
+  // Returns true if |sequence_number| is received in order, false otherwise.
+  virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
+};
+
+typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
+
+class ReceiveStatistics : public Module {
+ public:
+  virtual ~ReceiveStatistics() {}
+
+  static ReceiveStatistics* Create(Clock* clock);
+
+  // Updates the receive statistics with this packet.
+  virtual void IncomingPacket(const RTPHeader& rtp_header,
+                              size_t packet_length,
+                              bool retransmitted) = 0;
+
+  // Increment counter for number of FEC packets received.
+  virtual void FecPacketReceived(const RTPHeader& header,
+                                 size_t packet_length) = 0;
+
+  // Returns a map of all statisticians which have seen an incoming packet
+  // during the last two seconds.
+  virtual StatisticianMap GetActiveStatisticians() const = 0;
+
+  // Returns a pointer to the statistician of an ssrc.
+  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
+
+  // Sets the max reordering threshold in number of packets.
+  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+
+  // Called on new RTCP stats creation.
+  virtual void RegisterRtcpStatisticsCallback(
+      RtcpStatisticsCallback* callback) = 0;
+
+  // Called on new RTP stats creation.
+  virtual void RegisterRtpStatisticsCallback(
+      StreamDataCountersCallback* callback) = 0;
+};
+
+class NullReceiveStatistics : public ReceiveStatistics {
+ public:
+  void IncomingPacket(const RTPHeader& rtp_header,
+                      size_t packet_length,
+                      bool retransmitted) override;
+  void FecPacketReceived(const RTPHeader& header,
+                         size_t packet_length) override;
+  StatisticianMap GetActiveStatisticians() const override;
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override;
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+  void RegisterRtcpStatisticsCallback(
+      RtcpStatisticsCallback* callback) override;
+  void RegisterRtpStatisticsCallback(
+      StreamDataCountersCallback* callback) override;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
new file mode 100644
index 0000000..56c6e48
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
+
+namespace webrtc {
+
+class Clock;
+class TimestampExtrapolator;
+
+// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
+// time in local timebase.
+// Note that it needs to be trained with at least 2 RTCP SR (by calling
+// |UpdateRtcpTimestamp|) before it can be used.
+class RemoteNtpTimeEstimator {
+ public:
+  explicit RemoteNtpTimeEstimator(Clock* clock);
+
+  ~RemoteNtpTimeEstimator();
+
+  // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
+  // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|.
+  bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
+                           uint32_t rtp_timestamp);
+
+  // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
+  // Returns the NTP timestamp in ms when success. -1 if failed.
+  int64_t Estimate(uint32_t rtp_timestamp);
+
+ private:
+  Clock* clock_;
+  rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
+  RtcpList rtcp_list_;
+  int64_t last_timing_log_ms_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_cvo.h b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
new file mode 100644
index 0000000..2e30d89
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#include "webrtc/common_video/rotation.h"
+
+namespace webrtc {
+
+// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/
+// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is
+// the clockwise angle the frames must be rotated in order to display the frames
+// correctly if the display is rotated in its natural orientation.
+inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) {
+  switch (rotation) {
+    case kVideoRotation_0:
+      return 0;
+    case kVideoRotation_90:
+      return 1;
+    case kVideoRotation_180:
+      return 2;
+    case kVideoRotation_270:
+      return 3;
+  }
+  assert(false);
+  return 0;
+}
+
+inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) {
+  switch (rotation) {
+    case 0:
+      return kVideoRotation_0;
+    case 1:
+      return kVideoRotation_90;
+    case 2:
+      return kVideoRotation_180;
+      break;
+    case 3:
+      return kVideoRotation_270;
+    default:
+      assert(false);
+      return kVideoRotation_0;
+  }
+}
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
new file mode 100644
index 0000000..329de32
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct RTPHeader;
+
+class RtpHeaderParser {
+ public:
+  static RtpHeaderParser* Create();
+  virtual ~RtpHeaderParser() {}
+
+  // Returns true if the packet is an RTCP packet, false otherwise.
+  static bool IsRtcp(const uint8_t* packet, size_t length);
+
+  // Parses the packet and stores the parsed packet in |header|. Returns true on
+  // success, false otherwise.
+  // This method is thread-safe in the sense that it can parse multiple packets
+  // at once.
+  virtual bool Parse(const uint8_t* packet,
+                     size_t length,
+                     RTPHeader* header) const = 0;
+
+  // Registers an RTP header extension and binds it to |id|.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
+                                          uint8_t id) = 0;
+
+  // De-registers an RTP header extension.
+  virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
new file mode 100644
index 0000000..c2f467a
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace webrtc {
+
+// This strategy deals with the audio/video-specific aspects
+// of payload handling.
+class RTPPayloadStrategy {
+ public:
+  virtual ~RTPPayloadStrategy() {}
+
+  virtual bool CodecsMustBeUnique() const = 0;
+
+  virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
+                                   const uint32_t frequency,
+                                   const uint8_t channels,
+                                   const uint32_t rate) const = 0;
+
+  virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
+                                 const uint32_t rate) const = 0;
+
+  virtual RtpUtility::Payload* CreatePayloadType(
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payloadType,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate) const = 0;
+
+  virtual int GetPayloadTypeFrequency(
+      const RtpUtility::Payload& payload) const = 0;
+
+  static RTPPayloadStrategy* CreateStrategy(const bool handling_audio);
+
+ protected:
+  RTPPayloadStrategy() {}
+};
+
+class RTPPayloadRegistry {
+ public:
+  // The registry takes ownership of the strategy.
+  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
+  ~RTPPayloadRegistry();
+
+  int32_t RegisterReceivePayload(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payload_type,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate,
+      bool* created_new_payload_type);
+
+  int32_t DeRegisterReceivePayload(
+      const int8_t payload_type);
+
+  int32_t ReceivePayloadType(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate,
+      int8_t* payload_type) const;
+
+  bool RtxEnabled() const;
+
+  void SetRtxSsrc(uint32_t ssrc);
+
+  bool GetRtxSsrc(uint32_t* ssrc) const;
+
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
+
+  bool IsRtx(const RTPHeader& header) const;
+
+  // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t*
+  // restored_packet, instead of a uint8_t**.
+  // TODO(noahric): Remove this when all callers have been updated.
+  bool RestoreOriginalPacket(uint8_t** restored_packet,
+                             const uint8_t* packet,
+                             size_t* packet_length,
+                             uint32_t original_ssrc,
+                             const RTPHeader& header) const;
+
+  bool RestoreOriginalPacket(uint8_t* restored_packet,
+                             const uint8_t* packet,
+                             size_t* packet_length,
+                             uint32_t original_ssrc,
+                             const RTPHeader& header) const;
+
+  bool IsRed(const RTPHeader& header) const;
+
+  // Returns true if the media of this RTP packet is encapsulated within an
+  // extra header, such as RTX or RED.
+  bool IsEncapsulated(const RTPHeader& header) const;
+
+  bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
+
+  int GetPayloadTypeFrequency(uint8_t payload_type) const;
+
+  bool PayloadTypeToPayload(const uint8_t payload_type,
+                            RtpUtility::Payload*& payload) const;
+
+  void ResetLastReceivedPayloadTypes() {
+    CriticalSectionScoped cs(crit_sect_.get());
+    last_received_payload_type_ = -1;
+    last_received_media_payload_type_ = -1;
+  }
+
+  // This sets the payload type of the packets being received from the network
+  // on the media SSRC. For instance if packets are encapsulated with RED, this
+  // payload type will be the RED payload type.
+  void SetIncomingPayloadType(const RTPHeader& header);
+
+  // Returns true if the new media payload type has not changed.
+  bool ReportMediaPayloadType(uint8_t media_payload_type);
+
+  int8_t red_payload_type() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return red_payload_type_;
+  }
+  int8_t ulpfec_payload_type() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return ulpfec_payload_type_;
+  }
+  int8_t last_received_payload_type() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return last_received_payload_type_;
+  }
+  void set_last_received_payload_type(int8_t last_received_payload_type) {
+    CriticalSectionScoped cs(crit_sect_.get());
+    last_received_payload_type_ = last_received_payload_type;
+  }
+
+  int8_t last_received_media_payload_type() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return last_received_media_payload_type_;
+  };
+
+  bool use_rtx_payload_mapping_on_restore() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return use_rtx_payload_mapping_on_restore_;
+  }
+
+  void set_use_rtx_payload_mapping_on_restore(bool val) {
+    CriticalSectionScoped cs(crit_sect_.get());
+    use_rtx_payload_mapping_on_restore_ = val;
+  }
+
+ private:
+  // Prunes the payload type map of the specific payload type, if it exists.
+  void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const size_t payload_name_length,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate);
+
+  bool IsRtxInternal(const RTPHeader& header) const;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  RtpUtility::PayloadTypeMap payload_type_map_;
+  rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
+  int8_t  red_payload_type_;
+  int8_t ulpfec_payload_type_;
+  int8_t incoming_payload_type_;
+  int8_t  last_received_payload_type_;
+  int8_t  last_received_media_payload_type_;
+  bool rtx_;
+  // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
+  // only understand one RTX PT is no longer needed.
+  int rtx_payload_type_;
+  // Mapping rtx_payload_type_map_[rtx] = associated.
+  std::map<int, int> rtx_payload_type_map_;
+  // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
+  // correct payload type.
+  bool use_rtx_payload_mapping_on_restore_;
+  uint32_t ssrc_rtx_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
new file mode 100644
index 0000000..d257a30
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class RTPPayloadRegistry;
+
+class TelephoneEventHandler {
+ public:
+  virtual ~TelephoneEventHandler() {}
+
+  // The following three methods implement the TelephoneEventHandler interface.
+  // Forward DTMFs to decoder for playout.
+  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
+
+  // Is forwarding of outband telephone events turned on/off?
+  virtual bool TelephoneEventForwardToDecoder() const = 0;
+
+  // Is TelephoneEvent configured with payload type payload_type
+  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
+};
+
+class RtpReceiver {
+ public:
+  // Creates a video-enabled RTP receiver.
+  static RtpReceiver* CreateVideoReceiver(
+      Clock* clock,
+      RtpData* incoming_payload_callback,
+      RtpFeedback* incoming_messages_callback,
+      RTPPayloadRegistry* rtp_payload_registry);
+
+  // Creates an audio-enabled RTP receiver.
+  static RtpReceiver* CreateAudioReceiver(
+      Clock* clock,
+      RtpAudioFeedback* incoming_audio_feedback,
+      RtpData* incoming_payload_callback,
+      RtpFeedback* incoming_messages_callback,
+      RTPPayloadRegistry* rtp_payload_registry);
+
+  virtual ~RtpReceiver() {}
+
+  // Returns a TelephoneEventHandler if available.
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
+  // Registers a receive payload in the payload registry and notifies the media
+  // receiver strategy.
+  virtual int32_t RegisterReceivePayload(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payload_type,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate) = 0;
+
+  // De-registers |payload_type| from the payload registry.
+  virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
+
+  // Parses the media specific parts of an RTP packet and updates the receiver
+  // state. This for instance means that any changes in SSRC and payload type is
+  // detected and acted upon.
+  virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
+                                 const uint8_t* payload,
+                                 size_t payload_length,
+                                 PayloadUnion payload_specific,
+                                 bool in_order) = 0;
+
+  // Returns the currently configured NACK method.
+  virtual NACKMethod NACK() const = 0;
+
+  // Turn negative acknowledgement (NACK) requests on/off.
+  virtual void SetNACKStatus(const NACKMethod method) = 0;
+
+  // Gets the last received timestamp. Returns true if a packet has been
+  // received, false otherwise.
+  virtual bool Timestamp(uint32_t* timestamp) const = 0;
+  // Gets the time in milliseconds when the last timestamp was received.
+  // Returns true if a packet has been received, false otherwise.
+  virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
+
+  // Returns the remote SSRC of the currently received RTP stream.
+  virtual uint32_t SSRC() const = 0;
+
+  // Returns the current remote CSRCs.
+  virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
+
+  // Returns the current energy of the RTP stream received.
+  virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
new file mode 100644
index 0000000..39aeba2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -0,0 +1,641 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#include <set>
+#include <vector>
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+// Forward declarations.
+class ReceiveStatistics;
+class RemoteBitrateEstimator;
+class RtpReceiver;
+class Transport;
+namespace rtcp {
+class TransportFeedback;
+}
+
+class RtpRtcp : public Module {
+ public:
+  struct Configuration {
+    Configuration();
+
+   /*  id                   - Unique identifier of this RTP/RTCP module object
+    *  audio                - True for a audio version of the RTP/RTCP module
+    *                         object false will create a video version
+    *  clock                - The clock to use to read time. If NULL object
+    *                         will be using the system clock.
+    *  incoming_data        - Callback object that will receive the incoming
+    *                         data. May not be NULL; default callback will do
+    *                         nothing.
+    *  incoming_messages    - Callback object that will receive the incoming
+    *                         RTP messages. May not be NULL; default callback
+    *                         will do nothing.
+    *  outgoing_transport   - Transport object that will be called when packets
+    *                         are ready to be sent out on the network
+    *  intra_frame_callback - Called when the receiver request a intra frame.
+    *  bandwidth_callback   - Called when we receive a changed estimate from
+    *                         the receiver of out stream.
+    *  audio_messages       - Telephone events. May not be NULL; default
+    *                         callback will do nothing.
+    *  remote_bitrate_estimator - Estimates the bandwidth available for a set of
+    *                             streams from the same client.
+    *  paced_sender             - Spread any bursts of packets into smaller
+    *                             bursts to minimize packet loss.
+    */
+    bool audio;
+    bool receiver_only;
+    Clock* clock;
+    ReceiveStatistics* receive_statistics;
+    Transport* outgoing_transport;
+    RtcpIntraFrameObserver* intra_frame_callback;
+    RtcpBandwidthObserver* bandwidth_callback;
+    TransportFeedbackObserver* transport_feedback_callback;
+    RtcpRttStats* rtt_stats;
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
+    RtpAudioFeedback* audio_messages;
+    RemoteBitrateEstimator* remote_bitrate_estimator;
+    RtpPacketSender* paced_sender;
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator;
+    BitrateStatisticsObserver* send_bitrate_observer;
+    FrameCountObserver* send_frame_count_observer;
+    SendSideDelayObserver* send_side_delay_observer;
+  };
+
+  /*
+   *   Create a RTP/RTCP module object using the system clock.
+   *
+   *   configuration  - Configuration of the RTP/RTCP module.
+   */
+  static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
+
+  /**************************************************************************
+   *
+   *   Receiver functions
+   *
+   ***************************************************************************/
+
+    virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
+                                       size_t incoming_packet_length) = 0;
+
+    virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
+
+    /**************************************************************************
+    *
+    *   Sender
+    *
+    ***************************************************************************/
+
+    /*
+    *   set MTU
+    *
+    *   size    -  Max transfer unit in bytes, default is 1500
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
+
+    /*
+    *   set transtport overhead
+    *   default is IPv4 and UDP with no encryption
+    *
+    *   TCP                     - true for TCP false UDP
+    *   IPv6                    - true for IP version 6 false for version 4
+    *   authenticationOverhead  - number of bytes to leave for an
+    *                             authentication header
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetTransportOverhead(
+        bool TCP,
+        bool IPV6,
+        uint8_t authenticationOverhead = 0) = 0;
+
+    /*
+    *   Get max payload length
+    *
+    *   A combination of the configuration MaxTransferUnit and
+    *   TransportOverhead.
+    *   Does not account FEC/ULP/RED overhead if FEC is enabled.
+    *   Does not account for RTP headers
+    */
+    virtual uint16_t MaxPayloadLength() const = 0;
+
+    /*
+    *   Get max data payload length
+    *
+    *   A combination of the configuration MaxTransferUnit, headers and
+    *   TransportOverhead.
+    *   Takes into account FEC/ULP/RED overhead if FEC is enabled.
+    *   Takes into account RTP headers
+    */
+    virtual uint16_t MaxDataPayloadLength() const = 0;
+
+    /*
+    *   set codec name and payload type
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RegisterSendPayload(
+        const CodecInst& voiceCodec) = 0;
+
+    /*
+    *   set codec name and payload type
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RegisterSendPayload(
+        const VideoCodec& videoCodec) = 0;
+
+    /*
+    *   Unregister a send payload
+    *
+    *   payloadType - payload type of codec
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
+
+   /*
+    *   (De)register RTP header extension type and id.
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
+                                                   uint8_t id) = 0;
+
+    virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
+
+    /*
+    *   get start timestamp
+    */
+    virtual uint32_t StartTimestamp() const = 0;
+
+    /*
+    *   configure start timestamp, default is a random number
+    *
+    *   timestamp   - start timestamp
+    */
+    virtual void SetStartTimestamp(uint32_t timestamp) = 0;
+
+    /*
+    *   Get SequenceNumber
+    */
+    virtual uint16_t SequenceNumber() const = 0;
+
+    /*
+    *   Set SequenceNumber, default is a random number
+    */
+    virtual void SetSequenceNumber(uint16_t seq) = 0;
+
+    // Returns true if the ssrc matched this module, false otherwise.
+    virtual bool SetRtpStateForSsrc(uint32_t ssrc,
+                                    const RtpState& rtp_state) = 0;
+    virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
+
+    /*
+    *   Get SSRC
+    */
+    virtual uint32_t SSRC() const = 0;
+
+    /*
+    *   configure SSRC, default is a random number
+    */
+    virtual void SetSSRC(uint32_t ssrc) = 0;
+
+    /*
+    *   Set CSRC
+    *
+    *   csrcs   - vector of CSRCs
+    */
+    virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
+
+    /*
+    * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
+    * of values of the enumerator RtxMode.
+    */
+    virtual void SetRtxSendStatus(int modes) = 0;
+
+    /*
+    * Get status of sending RTX (RFC 4588). The returned value can be
+    * a combination of values of the enumerator RtxMode.
+    */
+    virtual int RtxSendStatus() const = 0;
+
+    // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
+    // only the SSRC is set.
+    virtual void SetRtxSsrc(uint32_t ssrc) = 0;
+
+    // Sets the payload type to use when sending RTX packets. Note that this
+    // doesn't enable RTX, only the payload type is set.
+    virtual void SetRtxSendPayloadType(int payload_type,
+                                       int associated_payload_type) = 0;
+
+    // Gets the payload type pair of (RTX, associated) to use when sending RTX
+    // packets.
+    virtual std::pair<int, int> RtxSendPayloadType() const = 0;
+
+    /*
+    *   sends kRtcpByeCode when going from true to false
+    *
+    *   sending - on/off
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetSendingStatus(bool sending) = 0;
+
+    /*
+    *   get send status
+    */
+    virtual bool Sending() const = 0;
+
+    /*
+    *   Starts/Stops media packets, on by default
+    *
+    *   sending - on/off
+    */
+    virtual void SetSendingMediaStatus(bool sending) = 0;
+
+    /*
+    *   get send status
+    */
+    virtual bool SendingMedia() const = 0;
+
+    /*
+    *   get sent bitrate in Kbit/s
+    */
+    virtual void BitrateSent(uint32_t* totalRate,
+                             uint32_t* videoRate,
+                             uint32_t* fecRate,
+                             uint32_t* nackRate) const = 0;
+
+    /*
+    *   Used by the codec module to deliver a video or audio frame for
+    *   packetization.
+    *
+    *   frameType       - type of frame to send
+    *   payloadType     - payload type of frame to send
+    *   timestamp       - timestamp of frame to send
+    *   payloadData     - payload buffer of frame to send
+    *   payloadSize     - size of payload buffer to send
+    *   fragmentation   - fragmentation offset data for fragmented frames such
+    *                     as layers or RED
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendOutgoingData(
+        FrameType frameType,
+        int8_t payloadType,
+        uint32_t timeStamp,
+        int64_t capture_time_ms,
+        const uint8_t* payloadData,
+        size_t payloadSize,
+        const RTPFragmentationHeader* fragmentation = NULL,
+        const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
+
+    virtual bool TimeToSendPacket(uint32_t ssrc,
+                                  uint16_t sequence_number,
+                                  int64_t capture_time_ms,
+                                  bool retransmission) = 0;
+
+    virtual size_t TimeToSendPadding(size_t bytes) = 0;
+
+    // Called on generation of new statistics after an RTP send.
+    virtual void RegisterSendChannelRtpStatisticsCallback(
+        StreamDataCountersCallback* callback) = 0;
+    virtual StreamDataCountersCallback*
+        GetSendChannelRtpStatisticsCallback() const = 0;
+
+    /**************************************************************************
+    *
+    *   RTCP
+    *
+    ***************************************************************************/
+
+    /*
+    *    Get RTCP status
+    */
+    virtual RtcpMode RTCP() const = 0;
+
+    /*
+    *   configure RTCP status i.e on(compound or non- compound)/off
+    *
+    *   method  - RTCP method to use
+    */
+    virtual void SetRTCPStatus(RtcpMode method) = 0;
+
+    /*
+    *   Set RTCP CName (i.e unique identifier)
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetCNAME(const char* c_name) = 0;
+
+    /*
+    *   Get remote CName
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
+                                char cName[RTCP_CNAME_SIZE]) const = 0;
+
+    /*
+    *   Get remote NTP
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RemoteNTP(
+        uint32_t *ReceivedNTPsecs,
+        uint32_t *ReceivedNTPfrac,
+        uint32_t *RTCPArrivalTimeSecs,
+        uint32_t *RTCPArrivalTimeFrac,
+        uint32_t *rtcp_timestamp) const  = 0;
+
+    /*
+    *   AddMixedCNAME
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
+
+    /*
+    *   RemoveMixedCNAME
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
+
+    /*
+    *   Get RoundTripTime
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RTT(uint32_t remoteSSRC,
+                        int64_t* RTT,
+                        int64_t* avgRTT,
+                        int64_t* minRTT,
+                        int64_t* maxRTT) const = 0;
+
+    /*
+    *   Force a send of a RTCP packet
+    *   periodic SR and RR are triggered via the process function
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
+
+    /*
+    *   Force a send of a RTCP packet with more than one packet type.
+    *   periodic SR and RR are triggered via the process function
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendCompoundRTCP(
+        const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
+
+    /*
+    *    Good state of RTP receiver inform sender
+    */
+    virtual int32_t SendRTCPReferencePictureSelection(
+        const uint64_t pictureID) = 0;
+
+    /*
+    *    Send a RTCP Slice Loss Indication (SLI)
+    *    6 least significant bits of pictureID
+    */
+    virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
+
+    /*
+    *   Statistics of the amount of data sent
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t DataCountersRTP(
+        size_t* bytesSent,
+        uint32_t* packetsSent) const = 0;
+
+    /*
+    *   Get send statistics for the RTP and RTX stream.
+    */
+    virtual void GetSendStreamDataCounters(
+        StreamDataCounters* rtp_counters,
+        StreamDataCounters* rtx_counters) const = 0;
+
+    /*
+     *  Get packet loss statistics for the RTP stream.
+     */
+    virtual void GetRtpPacketLossStats(
+        bool outgoing,
+        uint32_t ssrc,
+        struct RtpPacketLossStats* loss_stats) const = 0;
+
+    /*
+    *   Get received RTCP sender info
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
+
+    /*
+    *   Get received RTCP report block
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RemoteRTCPStat(
+        std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
+
+    /*
+    *   (APP) Application specific data
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
+                                                   uint32_t name,
+                                                   const uint8_t* data,
+                                                   uint16_t length) = 0;
+    /*
+    *   (XR) VOIP metric
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetRTCPVoIPMetrics(
+        const RTCPVoIPMetric* VoIPMetric) = 0;
+
+    /*
+    *   (XR) Receiver Reference Time Report
+    */
+    virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
+
+    virtual bool RtcpXrRrtrStatus() const = 0;
+
+    /*
+    *  (REMB) Receiver Estimated Max Bitrate
+    */
+    virtual bool REMB() const = 0;
+
+    virtual void SetREMBStatus(bool enable) = 0;
+
+    virtual void SetREMBData(uint32_t bitrate,
+                             const std::vector<uint32_t>& ssrcs) = 0;
+
+    /*
+    *   (TMMBR) Temporary Max Media Bit Rate
+    */
+    virtual bool TMMBR() const = 0;
+
+    virtual void SetTMMBRStatus(bool enable) = 0;
+
+    /*
+    *   (NACK)
+    */
+
+    /*
+     *  TODO(holmer): Propagate this API to VideoEngine.
+     *  Returns the currently configured selective retransmission settings.
+     */
+    virtual int SelectiveRetransmissions() const = 0;
+
+    /*
+     *  TODO(holmer): Propagate this API to VideoEngine.
+     *  Sets the selective retransmission settings, which will decide which
+     *  packets will be retransmitted if NACKed. Settings are constructed by
+     *  combining the constants in enum RetransmissionMode with bitwise OR.
+     *  All packets are retransmitted if kRetransmitAllPackets is set, while no
+     *  packets are retransmitted if kRetransmitOff is set.
+     *  By default all packets except FEC packets are retransmitted. For VP8
+     *  with temporal scalability only base layer packets are retransmitted.
+     *
+     *  Returns -1 on failure, otherwise 0.
+     */
+    virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
+
+    /*
+    *   Send a Negative acknowledgement packet
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
+
+    /*
+    *   Store the sent packets, needed to answer to a Negative acknowledgement
+    *   requests
+    */
+    virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
+
+    // Returns true if the module is configured to store packets.
+    virtual bool StorePackets() const = 0;
+
+    // Called on receipt of RTCP report block from remote side.
+    virtual void RegisterRtcpStatisticsCallback(
+        RtcpStatisticsCallback* callback) = 0;
+    virtual RtcpStatisticsCallback*
+        GetRtcpStatisticsCallback() = 0;
+    // BWE feedback packets.
+    virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
+
+    /**************************************************************************
+    *
+    *   Audio
+    *
+    ***************************************************************************/
+
+    /*
+    *   set audio packet size, used to determine when it's time to send a DTMF
+    *   packet in silence (CNG)
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
+
+    /*
+    *   Send a TelephoneEvent tone using RFC 2833 (4733)
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendTelephoneEventOutband(uint8_t key,
+                                              uint16_t time_ms,
+                                              uint8_t level) = 0;
+
+    /*
+    *   Set payload type for Redundant Audio Data RFC 2198
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
+
+    /*
+    *   Get payload type for Redundant Audio Data RFC 2198
+    *
+    *   return -1 on failure else 0
+    */
+     virtual int32_t SendREDPayloadType(
+         int8_t& payloadType) const = 0;
+
+     /*
+     * Store the audio level in dBov for header-extension-for-audio-level-
+     * indication.
+     * This API shall be called before transmision of an RTP packet to ensure
+     * that the |level| part of the extended RTP header is updated.
+     *
+     * return -1 on failure else 0.
+     */
+     virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0;
+
+    /**************************************************************************
+    *
+    *   Video
+    *
+    ***************************************************************************/
+
+    /*
+    *   Set the target send bitrate
+    */
+    virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
+
+    /*
+    *   Turn on/off generic FEC
+    */
+    virtual void SetGenericFECStatus(bool enable,
+                                     uint8_t payload_type_red,
+                                     uint8_t payload_type_fec) = 0;
+
+    /*
+    *   Get generic FEC setting
+    */
+    virtual void GenericFECStatus(bool& enable,
+                                     uint8_t& payloadTypeRED,
+                                     uint8_t& payloadTypeFEC) = 0;
+
+
+    virtual int32_t SetFecParameters(
+        const FecProtectionParams* delta_params,
+        const FecProtectionParams* key_params) = 0;
+
+    /*
+    *   Set method for requestion a new key frame
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
+
+    /*
+    *   send a request for a keyframe
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t RequestKeyFrame() = 0;
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
new file mode 100644
index 0000000..6373de2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -0,0 +1,440 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#include <stddef.h>
+#include <list>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/typedefs.h"
+
+#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500    // we assume ethernet
+#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
+#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
+
+namespace webrtc {
+namespace rtcp {
+class TransportFeedback;
+}
+
+const int kVideoPayloadTypeFrequency = 90000;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+struct AudioPayload
+{
+    uint32_t    frequency;
+    uint8_t     channels;
+    uint32_t    rate;
+};
+
+struct VideoPayload
+{
+    RtpVideoCodecTypes   videoCodecType;
+    uint32_t       maxRate;
+};
+
+union PayloadUnion
+{
+    AudioPayload Audio;
+    VideoPayload Video;
+};
+
+enum RTPAliveType
+{
+    kRtpDead   = 0,
+    kRtpNoRtp = 1,
+    kRtpAlive  = 2
+};
+
+enum ProtectionType {
+  kUnprotectedPacket,
+  kProtectedPacket
+};
+
+enum StorageType {
+  kDontRetransmit,
+  kAllowRetransmission
+};
+
+enum RTPExtensionType {
+  kRtpExtensionNone,
+  kRtpExtensionTransmissionTimeOffset,
+  kRtpExtensionAudioLevel,
+  kRtpExtensionAbsoluteSendTime,
+  kRtpExtensionVideoRotation,
+  kRtpExtensionTransportSequenceNumber,
+};
+
+enum RTCPAppSubTypes
+{
+    kAppSubtypeBwe     = 0x00
+};
+
+// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
+enum RTCPPacketType : uint32_t {
+  kRtcpReport = 0x0001,
+  kRtcpSr = 0x0002,
+  kRtcpRr = 0x0004,
+  kRtcpSdes = 0x0008,
+  kRtcpBye = 0x0010,
+  kRtcpPli = 0x0020,
+  kRtcpNack = 0x0040,
+  kRtcpFir = 0x0080,
+  kRtcpTmmbr = 0x0100,
+  kRtcpTmmbn = 0x0200,
+  kRtcpSrReq = 0x0400,
+  kRtcpXrVoipMetric = 0x0800,
+  kRtcpApp = 0x1000,
+  kRtcpSli = 0x4000,
+  kRtcpRpsi = 0x8000,
+  kRtcpRemb = 0x10000,
+  kRtcpTransmissionTimeOffset = 0x20000,
+  kRtcpXrReceiverReferenceTime = 0x40000,
+  kRtcpXrDlrrReportBlock = 0x80000,
+  kRtcpTransportFeedback = 0x100000,
+};
+
+enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
+
+enum RtpRtcpPacketType
+{
+    kPacketRtp        = 0,
+    kPacketKeepAlive = 1
+};
+
+enum NACKMethod
+{
+    kNackOff      = 0,
+    kNackRtcp     = 2
+};
+
+enum RetransmissionMode : uint8_t {
+  kRetransmitOff = 0x0,
+  kRetransmitFECPackets = 0x1,
+  kRetransmitBaseLayer = 0x2,
+  kRetransmitHigherLayers = 0x4,
+  kRetransmitAllPackets = 0xFF
+};
+
+enum RtxMode {
+  kRtxOff                 = 0x0,
+  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
+  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
+                                  // instead of padding.
+};
+
+const size_t kRtxHeaderSize = 2;
+
+struct RTCPSenderInfo
+{
+    uint32_t NTPseconds;
+    uint32_t NTPfraction;
+    uint32_t RTPtimeStamp;
+    uint32_t sendPacketCount;
+    uint32_t sendOctetCount;
+};
+
+struct RTCPReportBlock {
+  RTCPReportBlock()
+      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
+        extendedHighSeqNum(0), jitter(0), lastSR(0),
+        delaySinceLastSR(0) {}
+
+  RTCPReportBlock(uint32_t remote_ssrc,
+                  uint32_t source_ssrc,
+                  uint8_t fraction_lost,
+                  uint32_t cumulative_lost,
+                  uint32_t extended_high_sequence_number,
+                  uint32_t jitter,
+                  uint32_t last_sender_report,
+                  uint32_t delay_since_last_sender_report)
+      : remoteSSRC(remote_ssrc),
+        sourceSSRC(source_ssrc),
+        fractionLost(fraction_lost),
+        cumulativeLost(cumulative_lost),
+        extendedHighSeqNum(extended_high_sequence_number),
+        jitter(jitter),
+        lastSR(last_sender_report),
+        delaySinceLastSR(delay_since_last_sender_report) {}
+
+  // Fields as described by RFC 3550 6.4.2.
+  uint32_t remoteSSRC;  // SSRC of sender of this report.
+  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
+  uint8_t fractionLost;
+  uint32_t cumulativeLost;  // 24 bits valid.
+  uint32_t extendedHighSeqNum;
+  uint32_t jitter;
+  uint32_t lastSR;
+  uint32_t delaySinceLastSR;
+};
+
+struct RtcpReceiveTimeInfo {
+  // Fields as described by RFC 3611 4.5.
+  uint32_t sourceSSRC;
+  uint32_t lastRR;
+  uint32_t delaySinceLastRR;
+};
+
+typedef std::list<RTCPReportBlock> ReportBlockList;
+
+struct RtpState {
+  RtpState()
+      : sequence_number(0),
+        start_timestamp(0),
+        timestamp(0),
+        capture_time_ms(-1),
+        last_timestamp_time_ms(-1),
+        media_has_been_sent(false) {}
+  uint16_t sequence_number;
+  uint32_t start_timestamp;
+  uint32_t timestamp;
+  int64_t capture_time_ms;
+  int64_t last_timestamp_time_ms;
+  bool media_has_been_sent;
+};
+
+class RtpData
+{
+public:
+    virtual ~RtpData() {}
+
+    virtual int32_t OnReceivedPayloadData(
+        const uint8_t* payloadData,
+        const size_t payloadSize,
+        const WebRtcRTPHeader* rtpHeader) = 0;
+
+    virtual bool OnRecoveredPacket(const uint8_t* packet,
+                                   size_t packet_length) = 0;
+};
+
+class RtpFeedback
+{
+public:
+    virtual ~RtpFeedback() {}
+
+    // Receiving payload change or SSRC change. (return success!)
+    /*
+    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
+    */
+    virtual int32_t OnInitializeDecoder(
+        const int8_t payloadType,
+        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+        const int frequency,
+        const uint8_t channels,
+        const uint32_t rate) = 0;
+
+    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
+
+    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
+                                       const bool added) = 0;
+};
+
+class RtpAudioFeedback {
+ public:
+  virtual void OnPlayTelephoneEvent(const uint8_t event,
+                                    const uint16_t lengthMs,
+                                    const uint8_t volume) = 0;
+
+ protected:
+  virtual ~RtpAudioFeedback() {}
+};
+
+class RtcpIntraFrameObserver {
+ public:
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
+
+  virtual void OnReceivedSLI(uint32_t ssrc,
+                             uint8_t picture_id) = 0;
+
+  virtual void OnReceivedRPSI(uint32_t ssrc,
+                              uint64_t picture_id) = 0;
+
+  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
+
+  virtual ~RtcpIntraFrameObserver() {}
+};
+
+class RtcpBandwidthObserver {
+ public:
+  // REMB or TMMBR
+  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
+
+  virtual void OnReceivedRtcpReceiverReport(
+      const ReportBlockList& report_blocks,
+      int64_t rtt,
+      int64_t now_ms) = 0;
+
+  virtual ~RtcpBandwidthObserver() {}
+};
+
+struct PacketInfo {
+  PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
+      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
+
+  PacketInfo(int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : PacketInfo(-1,
+                   arrival_time_ms,
+                   send_time_ms,
+                   sequence_number,
+                   payload_size,
+                   was_paced) {}
+
+  PacketInfo(int64_t creation_time_ms,
+             int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : creation_time_ms(creation_time_ms),
+        arrival_time_ms(arrival_time_ms),
+        send_time_ms(send_time_ms),
+        sequence_number(sequence_number),
+        payload_size(payload_size),
+        was_paced(was_paced) {}
+
+  // Time corresponding to when this object was created.
+  int64_t creation_time_ms;
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock.
+  int64_t arrival_time_ms;
+  // Time corresponding to when the packet was sent, timestamped with the
+  // sender's clock.
+  int64_t send_time_ms;
+  // Packet identifier, incremented with 1 for every packet generated by the
+  // sender.
+  uint16_t sequence_number;
+  // Size of the packet excluding RTP headers.
+  size_t payload_size;
+  // True if the packet was paced out by the pacer.
+  bool was_paced;
+};
+
+class TransportFeedbackObserver {
+ public:
+  TransportFeedbackObserver() {}
+  virtual ~TransportFeedbackObserver() {}
+
+  // Note: Transport-wide sequence number as sequence number. Arrival time
+  // must be set to 0.
+  virtual void AddPacket(uint16_t sequence_number,
+                         size_t length,
+                         bool was_paced) = 0;
+
+  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
+};
+
+class RtcpRttStats {
+ public:
+  virtual void OnRttUpdate(int64_t rtt) = 0;
+
+  virtual int64_t LastProcessedRtt() const = 0;
+
+  virtual ~RtcpRttStats() {};
+};
+
+// Null object version of RtpFeedback.
+class NullRtpFeedback : public RtpFeedback {
+ public:
+  virtual ~NullRtpFeedback() {}
+
+  int32_t OnInitializeDecoder(const int8_t payloadType,
+                              const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                              const int frequency,
+                              const uint8_t channels,
+                              const uint32_t rate) override {
+    return 0;
+  }
+
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
+};
+
+// Null object version of RtpData.
+class NullRtpData : public RtpData {
+ public:
+  virtual ~NullRtpData() {}
+
+  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                const size_t payloadSize,
+                                const WebRtcRTPHeader* rtpHeader) override {
+    return 0;
+  }
+
+  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
+    return true;
+  }
+};
+
+// Null object version of RtpAudioFeedback.
+class NullRtpAudioFeedback : public RtpAudioFeedback {
+ public:
+  virtual ~NullRtpAudioFeedback() {}
+
+  void OnPlayTelephoneEvent(const uint8_t event,
+                            const uint16_t lengthMs,
+                            const uint8_t volume) override {}
+};
+
+// Statistics about packet loss for a single directional connection. All values
+// are totals since the connection initiated.
+struct RtpPacketLossStats {
+  // The number of packets lost in events where no adjacent packets were also
+  // lost.
+  uint64_t single_packet_loss_count;
+  // The number of events in which more than one adjacent packet was lost.
+  uint64_t multiple_packet_loss_event_count;
+  // The number of packets lost in events where more than one adjacent packet
+  // was lost.
+  uint64_t multiple_packet_loss_packet_count;
+};
+
+class RtpPacketSender {
+ public:
+  RtpPacketSender() {}
+  virtual ~RtpPacketSender() {}
+
+  enum Priority {
+    kHighPriority = 0,    // Pass through; will be sent immediately.
+    kNormalPriority = 2,  // Put in back of the line.
+    kLowPriority = 3,     // Put in back of the low priority line.
+  };
+  // Low priority packets are mixed with the normal priority packets
+  // while we are paused.
+
+  // Returns true if we send the packet now, else it will add the packet
+  // information to the queue and call TimeToSendPacket when it's time to send.
+  virtual void InsertPacket(Priority priority,
+                            uint32_t ssrc,
+                            uint16_t sequence_number,
+                            int64_t capture_time_ms,
+                            size_t bytes,
+                            bool retransmission) = 0;
+};
+
+class TransportSequenceNumberAllocator {
+ public:
+  TransportSequenceNumberAllocator() {}
+  virtual ~TransportSequenceNumberAllocator() {}
+
+  virtual uint16_t AllocateSequenceNumber() = 0;
+};
+
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
index 3608165..ec7ec39 100644
--- a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
+++ b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include")
 
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
@@ -43,4 +45,4 @@
   virtual FecPacketCounter GetPacketCounter() const = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
index 6bd5cd8..183eb2e 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -99,4 +101,4 @@
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
index 0ffba21..d271c9b 100644
--- a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
+++ b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
@@ -48,4 +50,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
index c7a0268..6d942cb 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
@@ -7,8 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include "webrtc/common_video/rotation.h"
 
@@ -51,4 +53,4 @@
 }
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
index 2809996..9b2cabc 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
@@ -7,10 +7,12 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -41,4 +43,4 @@
   virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
index 313bef1..4994eb5 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -190,4 +192,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
index 2fb8ac5..db5f46e 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
@@ -8,10 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -100,4 +102,4 @@
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index f907408..d004f67 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -8,14 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include <set>
 #include <vector>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 // Forward declarations.
@@ -638,4 +640,4 @@
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 6936352..ac3954b 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
 
 #include <stddef.h>
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
@@ -437,4 +439,4 @@
 };
 
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index bc4aec8..ca5434e 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -13,9 +13,9 @@
 
 #include "testing/gmock/include/gmock/gmock.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"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index e35a75c..57bea0b 100644
--- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -17,14 +17,14 @@
       ],
       'sources': [
         # Common
-        'interface/fec_receiver.h',
-        'interface/receive_statistics.h',
-        'interface/remote_ntp_time_estimator.h',
-        'interface/rtp_header_parser.h',
-        'interface/rtp_payload_registry.h',
-        'interface/rtp_receiver.h',
-        'interface/rtp_rtcp.h',
-        'interface/rtp_rtcp_defines.h',
+        'include/fec_receiver.h',
+        'include/receive_statistics.h',
+        'include/remote_ntp_time_estimator.h',
+        'include/rtp_header_parser.h',
+        'include/rtp_payload_registry.h',
+        'include/rtp_receiver.h',
+        'include/rtp_rtcp.h',
+        'include/rtp_rtcp_defines.h',
         'source/bitrate.cc',
         'source/bitrate.h',
         'source/byte_io.h',
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 24db39b..b79f6ba 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -14,8 +14,8 @@
 // This header is included to get the nested declaration of Packet structure.
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index f64b537..bb22e1d 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -15,8 +15,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
diff --git a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
index e1791ad..a5de924 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index aad418f..7a9a8be 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -18,7 +18,7 @@
 #include <iterator>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
index f92f014..9ba6ce0 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/ref_count.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
index f577cba..549821f 100644
--- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 07a3693..d95c963 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -16,12 +16,12 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.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/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
 using namespace webrtc;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index fe42990..0cd8012 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 #include <algorithm>
 
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 5b52228..c265c17 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -11,7 +11,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
index 74fc9cd..ccc15ec 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index bc9cf2e..797c788 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -11,7 +11,7 @@
 #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/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index 3c34957..2956bc7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -18,7 +18,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
index 4cc1f38..ad6fd16 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 2723976..fc43de5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -16,7 +16,7 @@
 #include <set>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/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"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index 37b7b88..a697fd2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"  // RTCPReportBlock
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPReportBlock
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 9ec9283..1bdd6d1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -20,8 +20,8 @@
 #include "webrtc/base/thread_annotations.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/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
index f05d512..e1f8737 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -14,7 +14,7 @@
 #include <stddef.h> // size_t, ptrdiff_t
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 18225f9..3519499 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -14,8 +14,8 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index aeef443..c422577 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 1a14b55..d29e3d4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 39b64c6..b47e9b9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -11,7 +11,7 @@
 #include <string>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
index 63db349..d62ecba 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -30,7 +30,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index 2fe9632..6684768 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -19,7 +19,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
index abce7e7..3feca43 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -25,7 +25,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 7be3c2e..263b2d7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -13,7 +13,7 @@
 
 #include <map>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index 520cf7a..ca37750 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -15,7 +15,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index 266bad8..82c813f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index e97d11e..cece640 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -16,8 +16,8 @@
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 00a6ac7..d39991e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -12,7 +12,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video_engine/vie_defines.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 38d1450..a5bfd71 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 0b9bf27..e20c94e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 176852e..b2413f3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -14,8 +14,8 @@
 #include <set>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index e1ebd0c..bd7bd1b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -16,8 +16,8 @@
 #include <string.h>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 46741d5..4ec3531 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 37c3e6e..0e0a34a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
 
 #include "webrtc/base/scoped_ptr.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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 1af2d48..53051dd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -16,8 +16,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 23128df..56f761a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c9b6686..ada6667 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 4c94764..a5e2076 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -12,8 +12,8 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 50f4768..dc544fb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -16,7 +16,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index a134370..8fc0696 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -17,7 +17,7 @@
 
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 1fc9a89..f5df5b3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -14,7 +14,7 @@
 #include <string.h> //memcpy
 
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index fde6d47..332a0f8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -17,9 +17,9 @@
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 6606277..0209510 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index f412542..03ed600 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/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"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index af20f97..1c615e6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -13,8 +13,8 @@
 
 #include <stddef.h> // size_t, ptrdiff_t
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.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"
diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
index 53b678f..4716dfb 100644
--- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
+++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
index 711be4a..2d70289 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
@@ -14,7 +14,7 @@
 #include <stdio.h>
 #include <string>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_transport.h"
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
index bd83962..59742b2 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
@@ -13,7 +13,7 @@
 
 #include <stdlib.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
index 47f2880..c1d91cd 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
@@ -13,7 +13,7 @@
 #include <stdio.h>
 #include <stdlib.h>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
index f394412..02fbe0a 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
 #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/test/channel_transport/udp_transport.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 73334a8..f55b4b5 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -11,12 +11,12 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.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/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 745386d..36456f1 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -15,8 +15,8 @@
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
 using namespace webrtc;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index e9d8112..8faa2dd 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -14,9 +14,9 @@
 #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/receive_statistics.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/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 30a6a1c..16ea540 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -15,9 +15,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.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/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index 163515c..6704cd6 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -10,12 +10,12 @@
 
 source_set("utility") {
   sources = [
-    "interface/audio_frame_operations.h",
-    "interface/file_player.h",
-    "interface/file_recorder.h",
-    "interface/helpers_android.h",
-    "interface/jvm_android.h",
-    "interface/process_thread.h",
+    "include/audio_frame_operations.h",
+    "include/file_player.h",
+    "include/file_recorder.h",
+    "include/helpers_android.h",
+    "include/jvm_android.h",
+    "include/process_thread.h",
     "source/audio_frame_operations.cc",
     "source/coder.cc",
     "source/coder.h",
diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h
new file mode 100644
index 0000000..1551d86
--- /dev/null
+++ b/webrtc/modules/utility/include/audio_frame_operations.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+  // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
+  // operation, meaning src_audio and dst_audio must point to different
+  // buffers. It is the caller's responsibility to ensure that |dst_audio| is
+  // sufficiently large.
+  static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
+                           int16_t* dst_audio);
+  // |frame.num_channels_| will be updated. This version checks for sufficient
+  // buffer size and that |num_channels_| is mono.
+  static int MonoToStereo(AudioFrame* frame);
+
+  // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
+  // operation, meaning |src_audio| and |dst_audio| may point to the same
+  // buffer.
+  static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
+                           int16_t* dst_audio);
+  // |frame.num_channels_| will be updated. This version checks that
+  // |num_channels_| is stereo.
+  static int StereoToMono(AudioFrame* frame);
+
+  // Swap the left and right channels of |frame|. Fails silently if |frame| is
+  // not stereo.
+  static void SwapStereoChannels(AudioFrame* frame);
+
+  // Zeros out the audio and sets |frame.energy| to zero.
+  static void Mute(AudioFrame& frame);
+
+  static int Scale(float left, float right, AudioFrame& frame);
+
+  static int ScaleWithSat(float scale, AudioFrame& frame);
+};
+
+}  // namespace webrtc
+
+#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h
new file mode 100644
index 0000000..4ca134a
--- /dev/null
+++ b/webrtc/modules/utility/include/file_player.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+class FileCallback;
+
+class FilePlayer
+{
+public:
+    // The largest decoded frame size in samples (60ms with 32kHz sample rate).
+    enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
+    enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+
+    // Note: will return NULL for unsupported formats.
+    static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
+                                        const FileFormats fileFormat);
+
+    static void DestroyFilePlayer(FilePlayer* player);
+
+    // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
+    // will be set to the number of samples read (not the number of samples per
+    // channel).
+    virtual int Get10msAudioFromFile(
+        int16_t* outBuffer,
+        size_t& lengthInSamples,
+        int frequencyInHz) = 0;
+
+    // Register callback for receiving file playing notifications.
+    virtual int32_t RegisterModuleFileCallback(
+        FileCallback* callback) = 0;
+
+    // API for playing audio from fileName to channel.
+    // Note: codecInst is used for pre-encoded files.
+    virtual int32_t StartPlayingFile(
+        const char* fileName,
+        bool loop,
+        uint32_t startPosition,
+        float volumeScaling,
+        uint32_t notification,
+        uint32_t stopPosition = 0,
+        const CodecInst* codecInst = NULL) = 0;
+
+    // Note: codecInst is used for pre-encoded files.
+    virtual int32_t StartPlayingFile(
+        InStream& sourceStream,
+        uint32_t startPosition,
+        float volumeScaling,
+        uint32_t notification,
+        uint32_t stopPosition = 0,
+        const CodecInst* codecInst = NULL) = 0;
+
+    virtual int32_t StopPlayingFile() = 0;
+
+    virtual bool IsPlayingFile() const = 0;
+
+    virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+
+    // Set audioCodec to the currently used audio codec.
+    virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+
+    virtual int32_t Frequency() const = 0;
+
+    // Note: scaleFactor is in the range [0.0 - 2.0]
+    virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+
+    // Return the time in ms until next video frame should be pulled (by
+    // calling GetVideoFromFile(..)).
+    // Note: this API reads one video frame from file. This means that it should
+    //       be called exactly once per GetVideoFromFile(..) API call.
+    virtual int32_t TimeUntilNextVideoFrame() { return -1;}
+
+    virtual int32_t StartPlayingVideoFile(
+        const char* /*fileName*/,
+        bool /*loop*/,
+        bool /*videoOnly*/) { return -1;}
+
+    virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
+    {return -1;}
+
+    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
+
+    // Same as GetVideoFromFile(). videoFrame will have the resolution specified
+    // by the width outWidth and height outHeight in pixels.
+    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
+                                     const uint32_t /*outWidth*/,
+                                     const uint32_t /*outHeight*/) {
+      return -1;
+    }
+
+protected:
+    virtual ~FilePlayer() {}
+
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h
new file mode 100644
index 0000000..0c285d4
--- /dev/null
+++ b/webrtc/modules/utility/include/file_recorder.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class FileRecorder
+{
+public:
+
+    // Note: will return NULL for unsupported formats.
+    static FileRecorder* CreateFileRecorder(const uint32_t instanceID,
+                                            const FileFormats fileFormat);
+
+    static void DestroyFileRecorder(FileRecorder* recorder);
+
+    virtual int32_t RegisterModuleFileCallback(
+        FileCallback* callback) = 0;
+
+    virtual FileFormats RecordingFileFormat() const = 0;
+
+    virtual int32_t StartRecordingAudioFile(
+        const char* fileName,
+        const CodecInst& codecInst,
+        uint32_t notification) = 0;
+
+    virtual int32_t StartRecordingAudioFile(
+        OutStream& destStream,
+        const CodecInst& codecInst,
+        uint32_t notification) = 0;
+
+    // Stop recording.
+    // Note: this API is for both audio and video.
+    virtual int32_t StopRecording() = 0;
+
+    // Return true if recording.
+    // Note: this API is for both audio and video.
+    virtual bool IsRecording() const = 0;
+
+    virtual int32_t codec_info(CodecInst& codecInst) const = 0;
+
+    // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
+    virtual int32_t RecordAudioToFile(
+        const AudioFrame& frame,
+        const TickTime* playoutTS = NULL) = 0;
+
+    // Open/create the file specified by fileName for writing audio/video data
+    // (relative path is allowed). audioCodecInst specifies the encoding of the
+    // audio data. videoCodecInst specifies the encoding of the video data.
+    // Only video data will be recorded if videoOnly is true. amrFormat
+    // specifies the amr/amrwb storage format.
+    // Note: the file format is AVI.
+    virtual int32_t StartRecordingVideoFile(
+        const char* fileName,
+        const CodecInst& audioCodecInst,
+        const VideoCodec& videoCodecInst,
+        bool videoOnly = false) = 0;
+
+    // Record the video frame in videoFrame to AVI file.
+    virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) = 0;
+
+protected:
+    virtual ~FileRecorder() {}
+
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/include/helpers_android.h b/webrtc/modules/utility/include/helpers_android.h
new file mode 100644
index 0000000..2840ca9
--- /dev/null
+++ b/webrtc/modules/utility/include/helpers_android.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+
+#include <jni.h>
+#include <string>
+
+// Abort the process if |jni| has a Java exception pending.
+// TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
+#define CHECK_EXCEPTION(jni)        \
+  RTC_CHECK(!jni->ExceptionCheck()) \
+      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
+
+namespace webrtc {
+
+// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
+JNIEnv* GetEnv(JavaVM* jvm);
+
+// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr);
+
+// JNIEnv-helper methods that wraps the API which uses the JNI interface
+// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
+// exception is thrown while calling the method.
+jmethodID GetMethodID(
+    JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jmethodID GetStaticMethodID(
+    JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jclass FindClass(JNIEnv* jni, const char* name);
+
+jobject NewGlobalRef(JNIEnv* jni, jobject o);
+
+void DeleteGlobalRef(JNIEnv* jni, jobject o);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Attach thread to JVM if necessary and detach at scope end if originally
+// attached.
+class AttachThreadScoped {
+ public:
+  explicit AttachThreadScoped(JavaVM* jvm);
+  ~AttachThreadScoped();
+  JNIEnv* env();
+
+ private:
+  bool attached_;
+  JavaVM* jvm_;
+  JNIEnv* env_;
+};
+
+// Scoped holder for global Java refs.
+template<class T>  // T is jclass, jobject, jintArray, etc.
+class ScopedGlobalRef {
+ public:
+  ScopedGlobalRef(JNIEnv* jni, T obj)
+      : jni_(jni), obj_(static_cast<T>(NewGlobalRef(jni, obj))) {}
+  ~ScopedGlobalRef() {
+    DeleteGlobalRef(jni_, obj_);
+  }
+  T operator*() const {
+    return obj_;
+  }
+ private:
+  JNIEnv* jni_;
+  T obj_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/include/helpers_ios.h b/webrtc/modules/utility/include/helpers_ios.h
new file mode 100644
index 0000000..a5a07ac
--- /dev/null
+++ b/webrtc/modules/utility/include/helpers_ios.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+
+#if defined(WEBRTC_IOS)
+
+#include <string>
+
+namespace webrtc {
+namespace ios {
+
+bool CheckAndLogError(BOOL success, NSError* error);
+
+std::string StdStringFromNSString(NSString* nsString);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Returns [NSThread currentThread] description as string.
+// Example: <NSThread: 0x170066d80>{number = 1, name = main}
+std::string GetCurrentThreadDescription();
+
+std::string GetAudioSessionCategory();
+
+// Returns the current name of the operating system.
+std::string GetSystemName();
+
+// Returns the current version of the operating system.
+std::string GetSystemVersion();
+
+// Returns the version of the operating system as a floating point value.
+float GetSystemVersionAsFloat();
+
+// Returns the device type.
+// Examples: ”iPhone” and ”iPod touch”.
+std::string GetDeviceType();
+
+// Returns a more detailed device name.
+// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
+std::string GetDeviceName();
+
+}  // namespace ios
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_IOS)
+
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/include/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h
new file mode 100644
index 0000000..f527dff
--- /dev/null
+++ b/webrtc/modules/utility/include/jvm_android.h
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#include <jni.h>
+#include <string>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+// The JNI interface pointer (JNIEnv) is valid only in the current thread.
+// Should another thread need to access the Java VM, it must first call
+// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface
+// pointer. The native thread remains attached to the VM until it calls
+// DetachCurrentThread() to detach.
+class AttachCurrentThreadIfNeeded {
+ public:
+  AttachCurrentThreadIfNeeded();
+  ~AttachCurrentThreadIfNeeded();
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  bool attached_;
+};
+
+// This class is created by the NativeRegistration class and is used to wrap
+// the actual Java object handle (jobject) on which we can call methods from
+// C++ in to Java. See example in JVM for more details.
+// TODO(henrika): extend support for type of function calls.
+class GlobalRef {
+ public:
+  GlobalRef(JNIEnv* jni, jobject object);
+  ~GlobalRef();
+
+  jboolean CallBooleanMethod(jmethodID methodID, ...);
+  jint CallIntMethod(jmethodID methodID, ...);
+  void CallVoidMethod(jmethodID methodID, ...);
+
+ private:
+  JNIEnv* const jni_;
+  const jobject j_object_;
+};
+
+// Wraps the jclass object on which we can call GetMethodId() functions to
+// query method IDs.
+class JavaClass {
+ public:
+  JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {}
+  ~JavaClass() {}
+
+  jmethodID GetMethodId(const char* name, const char* signature);
+  jmethodID GetStaticMethodId(const char* name, const char* signature);
+  jobject CallStaticObjectMethod(jmethodID methodID, ...);
+
+ protected:
+  JNIEnv* const jni_;
+  jclass const j_class_;
+};
+
+// Adds support of the NewObject factory method to the JavaClass class.
+// See example in JVM for more details on how to use it.
+class NativeRegistration : public JavaClass {
+ public:
+  NativeRegistration(JNIEnv* jni, jclass clazz);
+  ~NativeRegistration();
+
+  rtc::scoped_ptr<GlobalRef> NewObject(
+      const char* name, const char* signature, ...);
+
+ private:
+  JNIEnv* const jni_;
+};
+
+// This class is created by the JVM class and is used to expose methods that
+// needs the JNI interface pointer but its main purpose is to create a
+// NativeRegistration object given name of a Java class and a list of native
+// methods. See example in JVM for more details.
+class JNIEnvironment {
+ public:
+  explicit JNIEnvironment(JNIEnv* jni);
+  ~JNIEnvironment();
+
+  // Registers native methods with the Java class specified by |name|.
+  // Note that the class name must be one of the names in the static
+  // |loaded_classes| array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  rtc::scoped_ptr<NativeRegistration> RegisterNatives(
+      const char* name, const JNINativeMethod *methods, int num_methods);
+
+  // Converts from Java string to std::string.
+  // This method must be called on the construction thread.
+  std::string JavaToStdString(const jstring& j_string);
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  JNIEnv* const jni_;
+};
+
+// Main class for working with Java from C++ using JNI in WebRTC.
+//
+// Example usage:
+//
+//   // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize.
+//   JNIEnv* jni = ::base::android::AttachCurrentThread();
+//   JavaVM* jvm = NULL;
+//   jni->GetJavaVM(&jvm);
+//   jobject context = ::base::android::GetApplicationContext();
+//   webrtc::JVM::Initialize(jvm, context);
+//
+//   // Header (.h) file of example class called User.
+//   rtc::scoped_ptr<JNIEnvironment> env;
+//   rtc::scoped_ptr<NativeRegistration> reg;
+//   rtc::scoped_ptr<GlobalRef> obj;
+//
+//   // Construction (in .cc file) of User class.
+//   User::User() {
+//     // Calling thread must be attached to the JVM.
+//     env = JVM::GetInstance()->environment();
+//     reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,);
+//     obj = reg->NewObject("<init>", ,);
+//   }
+//
+//   // Each User method can now use |reg| and |obj| and call Java functions
+//   // in WebRtcTest.java, e.g. boolean init() {}.
+//   bool User::Foo() {
+//     jmethodID id = reg->GetMethodId("init", "()Z");
+//     return obj->CallBooleanMethod(id);
+//   }
+//
+//   // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize.
+//   JVM::Uninitialize();
+class JVM {
+ public:
+  // Stores global handles to the Java VM interface and the application context.
+  // Should be called once on a thread that is attached to the JVM.
+  static void Initialize(JavaVM* jvm, jobject context);
+  // Clears handles stored in Initialize(). Must be called on same thread as
+  // Initialize().
+  static void Uninitialize();
+  // Gives access to the global Java VM interface pointer, which then can be
+  // used to create a valid JNIEnvironment object or to get a JavaClass object.
+  static JVM* GetInstance();
+
+  // Creates a JNIEnvironment object.
+  // This method returns a NULL pointer if AttachCurrentThread() has not been
+  // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
+  rtc::scoped_ptr<JNIEnvironment> environment();
+
+  // Returns a JavaClass object given class |name|.
+  // Note that the class name must be one of the names in the static
+  // |loaded_classes| array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  JavaClass GetClass(const char* name);
+
+  // TODO(henrika): can we make these private?
+  JavaVM* jvm() const { return jvm_; }
+  jobject context() const { return context_; }
+
+ protected:
+  JVM(JavaVM* jvm, jobject context);
+  ~JVM();
+
+ private:
+  JNIEnv* jni() const { return GetEnv(jvm_); }
+
+  rtc::ThreadChecker thread_checker_;
+  JavaVM* const jvm_;
+  jobject context_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/include/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h
new file mode 100644
index 0000000..56d92f4
--- /dev/null
+++ b/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+
+#include "webrtc/modules/utility/include/process_thread.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+
+namespace webrtc {
+
+class MockProcessThread : public ProcessThread {
+ public:
+  MOCK_METHOD0(Start, void());
+  MOCK_METHOD0(Stop, void());
+  MOCK_METHOD1(WakeUp, void(Module* module));
+  MOCK_METHOD1(PostTask, void(ProcessTask* task));
+  MOCK_METHOD1(RegisterModule, void(Module* module));
+  MOCK_METHOD1(DeRegisterModule, void(Module* module));
+
+  // MOCK_METHOD1 gets confused with mocking this method, so we work around it
+  // by overriding the method from the interface and forwarding the call to a
+  // mocked, simpler method.
+  void PostTask(rtc::scoped_ptr<ProcessTask> task) override {
+    PostTask(task.get());
+  }
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/include/process_thread.h b/webrtc/modules/utility/include/process_thread.h
new file mode 100644
index 0000000..285a5ea
--- /dev/null
+++ b/webrtc/modules/utility/include/process_thread.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+
+#include "webrtc/typedefs.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+class Module;
+
+class ProcessTask {
+ public:
+  ProcessTask() {}
+  virtual ~ProcessTask() {}
+
+  virtual void Run() = 0;
+};
+
+class ProcessThread {
+ public:
+  virtual ~ProcessThread();
+
+  static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name);
+
+  // Starts the worker thread.  Must be called from the construction thread.
+  virtual void Start() = 0;
+
+  // Stops the worker thread.  Must be called from the construction thread.
+  virtual void Stop() = 0;
+
+  // Wakes the thread up to give a module a chance to do processing right
+  // away.  This causes the worker thread to wake up and requery the specified
+  // module for when it should be called back. (Typically the module should
+  // return 0 from TimeUntilNextProcess on the worker thread at that point).
+  // Can be called on any thread.
+  virtual void WakeUp(Module* module) = 0;
+
+  // Queues a task object to run on the worker thread.  Ownership of the
+  // task object is transferred to the ProcessThread and the object will
+  // either be deleted after running on the worker thread, or on the
+  // construction thread of the ProcessThread instance, if the task did not
+  // get a chance to run (e.g. posting the task while shutting down or when
+  // the thread never runs).
+  virtual void PostTask(rtc::scoped_ptr<ProcessTask> task) = 0;
+
+  // Adds a module that will start to receive callbacks on the worker thread.
+  // Can be called from any thread.
+  virtual void RegisterModule(Module* module) = 0;
+
+  // Removes a previously registered module.
+  // Can be called from any thread.
+  virtual void DeRegisterModule(Module* module) = 0;
+};
+
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/audio_frame_operations.h b/webrtc/modules/utility/interface/audio_frame_operations.h
index c2af68a..017352a 100644
--- a/webrtc/modules/utility/interface/audio_frame_operations.h
+++ b/webrtc/modules/utility/interface/audio_frame_operations.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
-#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include "webrtc/typedefs.h"
 
@@ -55,4 +57,4 @@
 
 }  // namespace webrtc
 
-#endif  // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h
index 44f03e4..d6737e1 100644
--- a/webrtc/modules/utility/interface/file_player.h
+++ b/webrtc/modules/utility/interface/file_player.h
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
@@ -108,4 +110,4 @@
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/interface/file_recorder.h
index f2ce785..c7e26f6 100644
--- a/webrtc/modules/utility/interface/file_recorder.h
+++ b/webrtc/modules/utility/interface/file_recorder.h
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
@@ -81,4 +83,4 @@
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/interface/helpers_android.h b/webrtc/modules/utility/interface/helpers_android.h
index 5c73fe4..80dd676 100644
--- a/webrtc/modules/utility/interface/helpers_android.h
+++ b/webrtc/modules/utility/interface/helpers_android.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include <jni.h>
 #include <string>
@@ -84,4 +86,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/helpers_ios.h b/webrtc/modules/utility/interface/helpers_ios.h
index a5edee0..6013ee9 100644
--- a/webrtc/modules/utility/interface/helpers_ios.h
+++ b/webrtc/modules/utility/interface/helpers_ios.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #if defined(WEBRTC_IOS)
 
@@ -56,4 +58,4 @@
 
 #endif  // defined(WEBRTC_IOS)
 
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/interface/jvm_android.h b/webrtc/modules/utility/interface/jvm_android.h
index 0744fdb..a417c1b 100644
--- a/webrtc/modules/utility/interface/jvm_android.h
+++ b/webrtc/modules/utility/interface/jvm_android.h
@@ -8,15 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include <jni.h>
 #include <string>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
@@ -182,4 +184,4 @@
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/mock/mock_process_thread.h b/webrtc/modules/utility/interface/mock/mock_process_thread.h
index fd108a8..c494d4c 100644
--- a/webrtc/modules/utility/interface/mock/mock_process_thread.h
+++ b/webrtc/modules/utility/interface/mock/mock_process_thread.h
@@ -8,10 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
 
-#include "webrtc/modules/utility/interface/process_thread.h"
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
+
+#include "webrtc/modules/utility/include/process_thread.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 
@@ -35,4 +37,4 @@
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/interface/process_thread.h
index 451a5a3..1a7a875 100644
--- a/webrtc/modules/utility/interface/process_thread.h
+++ b/webrtc/modules/utility/interface/process_thread.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
 
 #include "webrtc/typedefs.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -63,4 +65,4 @@
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/source/audio_frame_operations.cc b/webrtc/modules/utility/source/audio_frame_operations.cc
index c07ca1f..fe09d79 100644
--- a/webrtc/modules/utility/source/audio_frame_operations.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index c278cdd..fff8f44 100644
--- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -10,8 +10,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 4ec5f9b..18b690d 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h
index f411db9..2dfe682 100644
--- a/webrtc/modules/utility/source/file_player_impl.h
+++ b/webrtc/modules/utility/source/file_player_impl.h
@@ -14,9 +14,9 @@
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
index 4b65acd..754e124 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -10,7 +10,7 @@
 
 // Unit tests for FilePlayer.
 
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/utility/include/file_player.h"
 
 #include <stdio.h>
 #include <string>
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index 13926de..cbd7c08 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
 #include "webrtc/modules/utility/source/file_recorder_impl.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
index 8ea96bd..01072cd 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/webrtc/modules/utility/source/file_recorder_impl.h
@@ -20,10 +20,10 @@
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
diff --git a/webrtc/modules/utility/source/helpers_android.cc b/webrtc/modules/utility/source/helpers_android.cc
index 25652f2..aea35f8 100644
--- a/webrtc/modules/utility/source/helpers_android.cc
+++ b/webrtc/modules/utility/source/helpers_android.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 #include <android/log.h>
 #include <assert.h>
diff --git a/webrtc/modules/utility/source/helpers_ios.mm b/webrtc/modules/utility/source/helpers_ios.mm
index 90b7c8f..2d0ac09 100644
--- a/webrtc/modules/utility/source/helpers_ios.mm
+++ b/webrtc/modules/utility/source/helpers_ios.mm
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
 namespace ios {
diff --git a/webrtc/modules/utility/source/jvm_android.cc b/webrtc/modules/utility/source/jvm_android.cc
index 648c168..eb37fda 100644
--- a/webrtc/modules/utility/source/jvm_android.cc
+++ b/webrtc/modules/utility/source/jvm_android.cc
@@ -10,7 +10,7 @@
 
 #include <android/log.h>
 
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 #include "webrtc/base/checks.h"
 
diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc
index 04fa887..9c739de 100644
--- a/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/webrtc/modules/utility/source/process_thread_impl.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h
index 4e5861b..0a95665 100644
--- a/webrtc/modules/utility/source/process_thread_impl.h
+++ b/webrtc/modules/utility/source/process_thread_impl.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index e080545..34ed5c5 100644
--- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -10,7 +10,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
diff --git a/webrtc/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi
index 38c9e3e..e5b0a4d 100644
--- a/webrtc/modules/utility/utility.gypi
+++ b/webrtc/modules/utility/utility.gypi
@@ -18,13 +18,13 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_frame_operations.h',
-        'interface/file_player.h',
-        'interface/file_recorder.h',
-        'interface/helpers_android.h',
-        'interface/helpers_ios.h',
-        'interface/jvm_android.h',
-        'interface/process_thread.h',
+        'include/audio_frame_operations.h',
+        'include/file_player.h',
+        'include/file_recorder.h',
+        'include/helpers_android.h',
+        'include/helpers_ios.h',
+        'include/jvm_android.h',
+        'include/process_thread.h',
         'source/audio_frame_operations.cc',
         'source/coder.cc',
         'source/coder.h',
diff --git a/webrtc/modules/video_capture/include/video_capture.h b/webrtc/modules/video_capture/include/video_capture.h
index 09b4502..a032d3c 100644
--- a/webrtc/modules/video_capture/include/video_capture.h
+++ b/webrtc/modules/video_capture/include/video_capture.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
 
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/video_capture/include/video_capture_defines.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_capture/include/video_capture_defines.h b/webrtc/modules/video_capture/include/video_capture_defines.h
index 1dee4fa..f62ddad 100644
--- a/webrtc/modules/video_capture/include/video_capture_defines.h
+++ b/webrtc/modules/video_capture/include/video_capture_defines.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
 #define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 2b8786b..3ec164b 100644
--- a/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_capture/include/video_capture.h"
 #include "webrtc/modules/video_capture/include/video_capture_factory.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_capture/video_capture_impl.cc b/webrtc/modules/video_capture/video_capture_impl.cc
index 4046181..90730cd 100644
--- a/webrtc/modules/video_capture/video_capture_impl.cc
+++ b/webrtc/modules/video_capture/video_capture_impl.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
index 230dea9..bdb079b 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
@@ -19,7 +19,7 @@
 #include <CoreMedia/CoreMedia.h>
 
 #include "webrtc/base/buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
index 6363ab7..5115c4b 100644
--- a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_decoder.h"
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
index da6008b..47f0cf3 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
@@ -13,7 +13,7 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index 029ccd1..4f4e435 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -24,7 +24,7 @@
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index 97f0846..7ebe1a2 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -26,7 +26,7 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
index 67f7b63..94a7b19 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding.h
@@ -21,8 +21,8 @@
 #include <windows.h>
 #endif
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
index fd38d64..c35bbc1 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding_defines.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
 #define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
diff --git a/webrtc/modules/video_coding/main/source/codec_timer.h b/webrtc/modules/video_coding/main/source/codec_timer.h
index 9268e8d..cb7e813 100644
--- a/webrtc/modules/video_coding/main/source/codec_timer.h
+++ b/webrtc/modules/video_coding/main/source/codec_timer.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
index 757ffb0..ae5e633 100644
--- a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
+++ b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
@@ -12,7 +12,7 @@
 
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/source/decoding_state.cc b/webrtc/modules/video_coding/main/source/decoding_state.cc
index cc92f1c..a3da7c6 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state.cc
+++ b/webrtc/modules/video_coding/main/source/decoding_state.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/video_coding/main/source/decoding_state.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/frame_buffer.h"
 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
 #include "webrtc/modules/video_coding/main/source/packet.h"
diff --git a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
index feae701..30b5786 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/decoding_state.h"
 #include "webrtc/modules/video_coding/main/source/frame_buffer.h"
 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.h b/webrtc/modules/video_coding/main/source/encoded_frame.h
index 608578c..dc6bbb8 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.h
+++ b/webrtc/modules/video_coding/main/source/encoded_frame.h
@@ -15,7 +15,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 
diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.h b/webrtc/modules/video_coding/main/source/frame_buffer.h
index ab4ff65..ee38a2e 100644
--- a/webrtc/modules/video_coding/main/source/frame_buffer.h
+++ b/webrtc/modules/video_coding/main/source/frame_buffer.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/source/encoded_frame.h"
 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.h b/webrtc/modules/video_coding/main/source/generic_decoder.h
index 09929e6..c1298bb 100644
--- a/webrtc/modules/video_coding/main/source/generic_decoder.h
+++ b/webrtc/modules/video_coding/main/source/generic_decoder.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/main/source/encoded_frame.h"
 #include "webrtc/modules/video_coding/main/source/timestamp_map.h"
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index bfdd786..bc63411 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -16,7 +16,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/source/frame_buffer.h"
 #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index f4a3638..9bde97c 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -18,7 +18,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/modules/video_coding/main/source/decoding_state.h"
diff --git a/webrtc/modules/video_coding/main/source/media_opt_util.cc b/webrtc/modules/video_coding/main/source/media_opt_util.cc
index 51decbe..002958e 100644
--- a/webrtc/modules/video_coding/main/source/media_opt_util.cc
+++ b/webrtc/modules/video_coding/main/source/media_opt_util.cc
@@ -15,7 +15,7 @@
 #include <limits.h>
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/modules/video_coding/main/source/fec_tables_xor.h"
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/main/source/media_optimization.h
index c4feeff..6ea5c01 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/webrtc/modules/video_coding/main/source/media_optimization.h
@@ -14,7 +14,7 @@
 #include <list>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/source/media_opt_util.h"
 #include "webrtc/modules/video_coding/main/source/qm_select.h"
diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/main/source/packet.cc
index fd5a6ab..3438448 100644
--- a/webrtc/modules/video_coding/main/source/packet.cc
+++ b/webrtc/modules/video_coding/main/source/packet.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/packet.h"
 
 #include <assert.h>
diff --git a/webrtc/modules/video_coding/main/source/packet.h b/webrtc/modules/video_coding/main/source/packet.h
index 80bf532..040cc88 100644
--- a/webrtc/modules/video_coding/main/source/packet.h
+++ b/webrtc/modules/video_coding/main/source/packet.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
 #define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/video_coding/main/source/qm_select.cc b/webrtc/modules/video_coding/main/source/qm_select.cc
index e86d075..be8fcfc 100644
--- a/webrtc/modules/video_coding/main/source/qm_select.cc
+++ b/webrtc/modules/video_coding/main/source/qm_select.cc
@@ -12,7 +12,7 @@
 
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/modules/video_coding/main/source/internal_defines.h"
 #include "webrtc/modules/video_coding/main/source/qm_select_data.h"
diff --git a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
index 6abc0d3..518df34 100644
--- a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
@@ -15,7 +15,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/qm_select.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/source/session_info.h b/webrtc/modules/video_coding/main/source/session_info.h
index 88071e1..37e9768 100644
--- a/webrtc/modules/video_coding/main/source/session_info.h
+++ b/webrtc/modules/video_coding/main/source/session_info.h
@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/source/packet.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
index 58c352d..8d57287 100644
--- a/webrtc/modules/video_coding/main/source/session_info_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/packet.h"
 #include "webrtc/modules/video_coding/main/source/session_info.h"
 
diff --git a/webrtc/modules/video_coding/main/source/timestamp_map.cc b/webrtc/modules/video_coding/main/source/timestamp_map.cc
index c68a5af..d11f949 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_map.cc
+++ b/webrtc/modules/video_coding/main/source/timestamp_map.cc
@@ -11,7 +11,7 @@
 #include <assert.h>
 #include <stdlib.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/source/timestamp_map.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/test/receiver_tests.h b/webrtc/modules/video_coding/main/test/receiver_tests.h
index 6d7b7be..5335c99 100644
--- a/webrtc/modules/video_coding/main/test/receiver_tests.h
+++ b/webrtc/modules/video_coding/main/test/receiver_tests.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/test/test_util.h"
 #include "webrtc/modules/video_coding/main/test/video_source.h"
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index 6717cf2..5fed3b1 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -15,10 +15,10 @@
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/main/source/internal_defines.h"
 #include "webrtc/modules/video_coding/main/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h
index 7459231..a2ecadd 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/webrtc/modules/video_coding/main/test/rtp_player.h
@@ -14,7 +14,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/main/test/test_util.h
index 27f66fe..68b2c23 100644
--- a/webrtc/modules/video_coding/main/test/test_util.h
+++ b/webrtc/modules/video_coding/main/test/test_util.h
@@ -18,7 +18,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 
diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
index 2d874cd..d930805 100644
--- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
+++ b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
@@ -14,7 +14,7 @@
 
 #include <algorithm>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/main/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_processing/main/interface/video_processing.h b/webrtc/modules/video_processing/main/interface/video_processing.h
index 30af99f..377a098 100644
--- a/webrtc/modules/video_processing/main/interface/video_processing.h
+++ b/webrtc/modules/video_processing/main/interface/video_processing.h
@@ -18,8 +18,8 @@
 #ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
 #define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
 #include "webrtc/video_frame.h"
 
diff --git a/webrtc/modules/video_processing/main/source/content_analysis.h b/webrtc/modules/video_processing/main/source/content_analysis.h
index 510c1b4..5b0767a 100644
--- a/webrtc/modules/video_processing/main/source/content_analysis.h
+++ b/webrtc/modules/video_processing/main/source/content_analysis.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
 #define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_processing/main/source/spatial_resampler.h b/webrtc/modules/video_processing/main/source/spatial_resampler.h
index f965a40..9a2d8f5 100644
--- a/webrtc/modules/video_processing/main/source/spatial_resampler.h
+++ b/webrtc/modules/video_processing/main/source/spatial_resampler.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/typedefs.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
 
 #include "webrtc/common_video/libyuv/include/scaler.h"
diff --git a/webrtc/modules/video_processing/main/source/video_decimator.h b/webrtc/modules/video_processing/main/source/video_decimator.h
index 3d4573c..c052c30 100644
--- a/webrtc/modules/video_processing/main/source/video_decimator.h
+++ b/webrtc/modules/video_processing/main/source/video_decimator.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
 #define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_render/external/video_render_external_impl.h b/webrtc/modules/video_render/external/video_render_external_impl.h
index 9230e60..a8b663f 100644
--- a/webrtc/modules/video_render/external/video_render_external_impl.h
+++ b/webrtc/modules/video_render/external/video_render_external_impl.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_render/i_video_render.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
diff --git a/webrtc/modules/video_render/include/video_render.h b/webrtc/modules/video_render/include/video_render.h
index 51fcce1..edd2302 100644
--- a/webrtc/modules/video_render/include/video_render.h
+++ b/webrtc/modules/video_render/include/video_render.h
@@ -20,7 +20,7 @@
  *
  */
 
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_render/include/video_render_defines.h b/webrtc/modules/video_render/include/video_render_defines.h
index f8f4803..1602d66 100644
--- a/webrtc/modules/video_render/include/video_render_defines.h
+++ b/webrtc/modules/video_render/include/video_render_defines.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/interface/incoming_video_stream.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc
 {
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI.cc b/webrtc/modules/video_render/test/testAPI/testAPI.cc
index 256d031..4de554c 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI.cc
+++ b/webrtc/modules/video_render/test/testAPI/testAPI.cc
@@ -32,8 +32,8 @@
 #endif
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_render/include/video_render.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
 #include "webrtc/system_wrappers/include/sleep.h"
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
index dd57397..a8808dd 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
+++ b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
@@ -20,8 +20,8 @@
 
 #import "webrtc/modules/video_render/mac/cocoa_render_view.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_render/include/video_render.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 5533a4c..9cf02ed 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -9,7 +9,8 @@
  */
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/test/layer_filtering_transport.h"
diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h
index 89b6dd0..c7e463e 100644
--- a/webrtc/test/rtp_rtcp_observer.h
+++ b/webrtc/test/rtp_rtcp_observer.h
@@ -16,7 +16,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/test/constants.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc
index 18e7c6d..3f8235e 100644
--- a/webrtc/tools/agc/activity_metric.cc
+++ b/webrtc/tools/agc/activity_metric.cc
@@ -24,7 +24,7 @@
 #include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
 #include "webrtc/modules/audio_processing/vad/standalone_vad.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 static const int kAgcAnalWindowSamples = 100;
 static const double kDefaultActivityThreshold = 0.3;
diff --git a/webrtc/tools/agc/test_utils.cc b/webrtc/tools/agc/test_utils.cc
index 81819c5..a0ed747 100644
--- a/webrtc/tools/agc/test_utils.cc
+++ b/webrtc/tools/agc/test_utils.cc
@@ -14,7 +14,7 @@
 
 #include <algorithm>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 70efe3b..e2dce3f 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -16,10 +16,10 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index f54909e..4849248 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -20,7 +20,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/encoder_settings.h"
diff --git a/webrtc/video/video_capture_input.cc b/webrtc/video/video_capture_input.cc
index 42bc65f..834849c 100644
--- a/webrtc/video/video_capture_input.cc
+++ b/webrtc/video/video_capture_input.cc
@@ -13,8 +13,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_capture/include/video_capture_factory.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
diff --git a/webrtc/video/video_capture_input_unittest.cc b/webrtc/video/video_capture_input_unittest.cc
index e8bc2ad..12ef076 100644
--- a/webrtc/video/video_capture_input_unittest.cc
+++ b/webrtc/video/video_capture_input_unittest.cc
@@ -15,7 +15,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/ref_count.h"
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 333f00d..ebceb56 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -22,6 +22,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/layer_filtering_transport.h"
diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h
index 36a87e3..6b0d302 100644
--- a/webrtc/video/video_send_stream.h
+++ b/webrtc/video/video_send_stream.h
@@ -17,7 +17,7 @@
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/send_statistics_proxy.h"
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 59011a6..e19dc48 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -20,8 +20,8 @@
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/frame_callback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
diff --git a/webrtc/video_engine/call_stats.cc b/webrtc/video_engine/call_stats.cc
index 0b71cc3..4d5338c 100644
--- a/webrtc/video_engine/call_stats.cc
+++ b/webrtc/video_engine/call_stats.cc
@@ -12,7 +12,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
diff --git a/webrtc/video_engine/call_stats.h b/webrtc/video_engine/call_stats.h
index a17330a..d0a0b53 100644
--- a/webrtc/video_engine/call_stats.h
+++ b/webrtc/video_engine/call_stats.h
@@ -15,7 +15,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video_engine/call_stats_unittest.cc b/webrtc/video_engine/call_stats_unittest.cc
index 4fb88df..bfba5cb 100644
--- a/webrtc/video_engine/call_stats_unittest.cc
+++ b/webrtc/video_engine/call_stats_unittest.cc
@@ -12,7 +12,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video_engine/call_stats.h"
 
diff --git a/webrtc/video_engine/encoder_state_feedback.cc b/webrtc/video_engine/encoder_state_feedback.cc
index 1c376b2..16ee8b9 100644
--- a/webrtc/video_engine/encoder_state_feedback.cc
+++ b/webrtc/video_engine/encoder_state_feedback.cc
@@ -13,7 +13,7 @@
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video_engine/vie_encoder.h"
 
diff --git a/webrtc/video_engine/encoder_state_feedback_unittest.cc b/webrtc/video_engine/encoder_state_feedback_unittest.cc
index 9787acc..ea252dc 100644
--- a/webrtc/video_engine/encoder_state_feedback_unittest.cc
+++ b/webrtc/video_engine/encoder_state_feedback_unittest.cc
@@ -20,8 +20,8 @@
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
 #include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_encoder.h"
 
diff --git a/webrtc/video_engine/overuse_frame_detector.h b/webrtc/video_engine/overuse_frame_detector.h
index aff4b43..0deb5ba 100644
--- a/webrtc/video_engine/overuse_frame_detector.h
+++ b/webrtc/video_engine/overuse_frame_detector.h
@@ -17,7 +17,7 @@
 #include "webrtc/base/exp_filter.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video_engine/payload_router.cc b/webrtc/video_engine/payload_router.cc
index 3af3d48..85b294b 100644
--- a/webrtc/video_engine/payload_router.cc
+++ b/webrtc/video_engine/payload_router.cc
@@ -11,8 +11,8 @@
 #include "webrtc/video_engine/payload_router.h"
 
 #include "webrtc/base/checks.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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc
index de39157..acaa400 100644
--- a/webrtc/video_engine/payload_router_unittest.cc
+++ b/webrtc/video_engine/payload_router_unittest.cc
@@ -14,7 +14,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/video_engine/payload_router.h"
 
diff --git a/webrtc/video_engine/report_block_stats.h b/webrtc/video_engine/report_block_stats.h
index dadcc9d..dd43072 100644
--- a/webrtc/video_engine/report_block_stats.h
+++ b/webrtc/video_engine/report_block_stats.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 9140cf1..b4e8332 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -21,9 +21,9 @@
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
 #include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index b1c1bba..20eae7d 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -16,8 +16,8 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 0f4a5a1..96ed2f9 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -22,7 +22,7 @@
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h
index 54aacdb..872fbb6 100644
--- a/webrtc/video_engine/vie_encoder.h
+++ b/webrtc/video_engine/vie_encoder.h
@@ -20,7 +20,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index 2e3b588..c99fa76 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -14,14 +14,14 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index cd069ea..20a9627 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -15,8 +15,8 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/vie_defines.h"
 
diff --git a/webrtc/video_engine/vie_remb.cc b/webrtc/video_engine/vie_remb.cc
index b347f2e..3901d6d 100644
--- a/webrtc/video_engine/vie_remb.cc
+++ b/webrtc/video_engine/vie_remb.cc
@@ -14,8 +14,8 @@
 
 #include <algorithm>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/video_engine/vie_remb.h b/webrtc/video_engine/vie_remb.h
index 9f38259..6a79ffe 100644
--- a/webrtc/video_engine/vie_remb.h
+++ b/webrtc/video_engine/vie_remb.h
@@ -16,9 +16,9 @@
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.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/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video_engine/vie_remb_unittest.cc b/webrtc/video_engine/vie_remb_unittest.cc
index 3289c4b..b6fbf29 100644
--- a/webrtc/video_engine/vie_remb_unittest.cc
+++ b/webrtc/video_engine/vie_remb_unittest.cc
@@ -17,9 +17,9 @@
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video_engine/vie_remb.h"
 
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc
index e7327eb..b2e2713 100644
--- a/webrtc/video_engine/vie_sync_module.cc
+++ b/webrtc/video_engine/vie_sync_module.cc
@@ -12,8 +12,8 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video_engine/stream_synchronization.h"
diff --git a/webrtc/video_engine/vie_sync_module.h b/webrtc/video_engine/vie_sync_module.h
index 2a343b8..dcd8072 100644
--- a/webrtc/video_engine/vie_sync_module.h
+++ b/webrtc/video_engine/vie_sync_module.h
@@ -15,7 +15,7 @@
 #define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video_engine/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index ed47b57..0a3e696 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -19,13 +19,13 @@
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 0f7b543..ba18aaa 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -15,14 +15,14 @@
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
diff --git a/webrtc/voice_engine/level_indicator.cc b/webrtc/voice_engine/level_indicator.cc
index 059b15f..68a837e 100644
--- a/webrtc/voice_engine/level_indicator.cc
+++ b/webrtc/voice_engine/level_indicator.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/voice_engine/level_indicator.h"
 
diff --git a/webrtc/voice_engine/monitor_module.h b/webrtc/voice_engine/monitor_module.h
index 42ea74d..fe915b3 100644
--- a/webrtc/voice_engine/monitor_module.h
+++ b/webrtc/voice_engine/monitor_module.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 #define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index 31b429c..1b4d2e2 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -11,7 +11,7 @@
 #include "webrtc/voice_engine/output_mixer.h"
 
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index 71e55e4..8c0a2de 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -13,9 +13,9 @@
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 311bfa0..9c3d4b1 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -14,7 +14,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 7b45e6d..fb430eb 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -19,7 +19,7 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
diff --git a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
index b4daba5..4f86010 100644
--- a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/base/arraysize.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 780a7f7..1dc15df 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 94592cf..b237414 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -11,7 +11,7 @@
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 714efb4..071d91d 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -15,9 +15,9 @@
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index 7bc7e0e..b041638 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -13,8 +13,8 @@
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index 226e383..921c3e5 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index f0ac959..5cd0acd 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/voice_engine/include/voe_base.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/shared_data.h"
 
 namespace webrtc {
diff --git a/webrtc/voice_engine/voe_file_impl.cc b/webrtc/voice_engine/voe_file_impl.cc
index 7927f9f..c2d5887 100644
--- a/webrtc/voice_engine/voe_file_impl.cc
+++ b/webrtc/voice_engine/voe_file_impl.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/voice_engine/voe_file_impl.h"
 
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index c8761bc..814d619 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -12,7 +12,7 @@
 #include "webrtc/modules/audio_device/android/audio_device_template.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 #endif
 
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"