9 files changed, 32 insertions, 27 deletions
diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn
index 3b9e2769ac..36391c7abc 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/OWNERS b/webrtc/modules/audio_conference_mixer/OWNERS
index 34bc7389ba..ea2062a9a0 100644
--- a/webrtc/modules/audio_conference_mixer/OWNERS
+++ b/webrtc/modules/audio_conference_mixer/OWNERS
@@ -1,3 +1,8 @@
-andrew@webrtc.org
+minyue@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
index 5aa3cc449b..9d7179504c 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/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
index 7ff39579ee..7370442704 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
@@ -8,12 +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_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"
+#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 +74,4 @@ protected:
 };
 }  // 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/include/audio_conference_mixer_defines.h
index d15b7fca02..5d58f42435 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -8,10 +8,10 @@
  *  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"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -57,4 +57,4 @@ protected:
 };
 }  // 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 2d2cf9dbb8..afb060f46d 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"
 
@@ -50,8 +50,8 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
-int MaxNumChannels(const AudioFrameList* list) {
-  int max_num_channels = 1;
+size_t MaxNumChannels(const AudioFrameList* list) {
+  size_t max_num_channels = 1;
   for (AudioFrameList::const_iterator iter = list->begin();
        iter != list->end();
        ++iter) {
@@ -278,7 +278,7 @@ int32_t AudioConferenceMixerImpl::Process() {
         //                with an API instead of dynamically.
 
         // Find the max channels over all mixing lists.
-        const int num_mixed_channels = std::max(MaxNumChannels(&mixList),
+        const size_t num_mixed_channels = std::max(MaxNumChannels(&mixList),
             std::max(MaxNumChannels(&additionalFramesList),
                      MaxNumChannels(&rampOutList)));
 
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 bc9a27e9f0..2466112769 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 636698e9c1..9c5d3b939d 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 d4fbd205f1..293bfa0db9 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 {