diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi
index 863d63b8ad..26b0de367a 100644
--- a/webrtc/build/common.gypi
+++ b/webrtc/build/common.gypi
@@ -129,7 +129,6 @@
 
         # Exclude internal ADM since Chromium uses its own IO handling.
         'include_internal_audio_device%': 0,
-
       }, {  # Settings for the standalone (not-in-Chromium) build.
         # TODO(andrew): For now, disable the Chrome plugins, which causes a
         # flood of chromium-style warnings. Investigate enabling them:
@@ -370,6 +369,11 @@
            }],
          ],
       }],
+      ['include_internal_audio_device==1', {
+        'defines': [
+          'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
+        ],
+      }],
     ], # conditions
     'direct_dependent_settings': {
       'conditions': [
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index d485af5605..f23905fc27 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -24,8 +24,6 @@ source_set("audio_device") {
     "audio_device_buffer.h",
     "audio_device_generic.cc",
     "audio_device_generic.h",
-    "audio_device_impl.cc",
-    "audio_device_impl.h",
     "audio_device_config.h",
     "dummy/audio_device_dummy.cc",
     "dummy/audio_device_dummy.h",
@@ -50,6 +48,7 @@ source_set("audio_device") {
     include_dirs += [ "android" ]
   }
   if (rtc_include_internal_audio_device) {
+    defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
     sources += [
       "android/audio_device_template.h",
       "android/audio_manager.cc",
@@ -64,6 +63,8 @@ source_set("audio_device") {
       "android/opensles_common.h",
       "android/opensles_player.cc",
       "android/opensles_player.h",
+      "audio_device_impl.cc",
+      "audio_device_impl.h",
       "ios/audio_device_ios.h",
       "ios/audio_device_ios.mm",
       "linux/alsasymboltable_linux.cc",
diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi
index 9add647505..9c9380eb69 100644
--- a/webrtc/modules/audio_device/audio_device.gypi
+++ b/webrtc/modules/audio_device/audio_device.gypi
@@ -38,8 +38,6 @@
         'audio_device_buffer.h',
         'audio_device_generic.cc',
         'audio_device_generic.h',
-        'audio_device_impl.cc',
-        'audio_device_impl.h',
         'audio_device_config.h',
         'dummy/audio_device_dummy.cc',
         'dummy/audio_device_dummy.h',
@@ -101,6 +99,8 @@
             'android/opensles_common.h',
             'android/opensles_player.cc',
             'android/opensles_player.h',
+            'audio_device_impl.cc',
+            'audio_device_impl.h',
             'ios/audio_device_ios.h',
             'ios/audio_device_ios.mm',
             'linux/alsasymboltable_linux.cc',
diff --git a/webrtc/modules/audio_device/audio_device_impl.h b/webrtc/modules/audio_device/audio_device_impl.h
index f2f4996154..efc305ba23 100644
--- a/webrtc/modules/audio_device/audio_device_impl.h
+++ b/webrtc/modules/audio_device/audio_device_impl.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
 
+#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
@@ -194,7 +196,7 @@ public:
 
 public:
     int32_t Id() {return _id;}
-#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(WEBRTC_ANDROID)
     // Only use this acccessor for test purposes on Android.
     AudioManager* GetAndroidAudioManagerForTest() {
       return _audioManagerAndroid.get();
@@ -218,7 +220,7 @@ private:
     AudioDeviceGeneric*          _ptrAudioDevice;
 
     AudioDeviceBuffer            _audioDeviceBuffer;
-#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(WEBRTC_ANDROID)
     rtc::scoped_ptr<AudioManager> _audioManagerAndroid;
 #endif
     int32_t                      _id;
@@ -231,4 +233,6 @@ private:
 
 }  // namespace webrtc
 
+#endif  // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
 #endif  // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 226b858593..e245fca110 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -218,6 +218,9 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
   // Create an internal ADM if the user has not added an external
   // ADM implementation as input to Init().
   if (external_adm == nullptr) {
+#if !defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+    return -1;
+#else
     // Create the internal ADM implementation.
     shared_->set_audio_device(AudioDeviceModuleImpl::Create(
         VoEId(shared_->instance_id(), -1), shared_->audio_device_layer()));
@@ -227,6 +230,7 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
                             "Init() failed to create the ADM");
       return -1;
     }
+#endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
   } else {
     // Use the already existing external ADM implementation.
     shared_->set_audio_device(external_adm);