diff --git a/src/build/common.gypi b/src/build/common.gypi
index b02450f9fb..3d76c1cc69 100644
--- a/src/build/common.gypi
+++ b/src/build/common.gypi
@@ -168,7 +168,7 @@
       ['OS=="ios"', {
         'defines': [
           'WEBRTC_MAC',
-          'MAC_IPHONE', # TODO(sjlee): This should be changed to WEBRTC_IOS.
+          'WEBRTC_IOS',
           'WEBRTC_THREAD_RR',
           'WEBRTC_CLOCK_TYPE_REALTIME',
         ],
diff --git a/src/engine_configurations.h b/src/engine_configurations.h
index a1ed4a904b..691ccf43a3 100644
--- a/src/engine_configurations.h
+++ b/src/engine_configurations.h
@@ -126,7 +126,7 @@
 //  VideoEngine MAC
 // ----------------------------------------------------------------------------
 
-#if defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 // #define CARBON_RENDERING
 #define COCOA_RENDERING
 #endif
@@ -135,7 +135,7 @@
 //  VideoEngine Mobile iPhone
 // ----------------------------------------------------------------------------
 
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
 #define EAGL_RENDERING
 #endif
 
diff --git a/src/modules/audio_device/main/source/audio_device_impl.cc b/src/modules/audio_device/main/source/audio_device_impl.cc
index 496f6174c4..fef1cbc515 100644
--- a/src/modules/audio_device/main/source/audio_device_impl.cc
+++ b/src/modules/audio_device/main/source/audio_device_impl.cc
@@ -38,7 +38,7 @@
  #if defined(LINUX_PULSE)
     #include "audio_device_pulse_linux.h"
  #endif
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
     #include "audio_device_utility_ios.h"
     #include "audio_device_ios.h"
 #elif defined(WEBRTC_MAC)
@@ -160,7 +160,7 @@ WebRtc_Word32 AudioDeviceModuleImpl::CheckPlatform()
 #elif defined(WEBRTC_LINUX)
     platform = kPlatformLinux;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is LINUX");
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
     platform = kPlatformIOS;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is IOS");
 #elif defined(WEBRTC_MAC)
@@ -341,7 +341,7 @@ WebRtc_Word32 AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
 
     // Create the *iPhone* implementation of the Audio Device
     //
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
     if (audioLayer == kPlatformDefaultAudio)
     {
         // Create *iPhone Audio* implementation
@@ -354,7 +354,7 @@ WebRtc_Word32 AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         // Create the Mac implementation of the Device Utility.
         ptrAudioDeviceUtility = new AudioDeviceUtilityIPhone(Id());
     }
-    // END #if defined(MAC_IPHONE)
+    // END #if defined(WEBRTC_IOS)
 
     // Create the *Mac* implementation of the Audio Device
     //
diff --git a/src/modules/audio_device/main/test/audio_device_test_api.cc b/src/modules/audio_device/main/test/audio_device_test_api.cc
index fdd6ad3c8f..bd349fb7c4 100644
--- a/src/modules/audio_device/main/test/audio_device_test_api.cc
+++ b/src/modules/audio_device/main/test/audio_device_test_api.cc
@@ -36,7 +36,7 @@ const char* GetFilename(const char* filename)
   sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", filename);
   return filenameStr[currentStr];
 }
-#elif !defined(MAC_IPHONE)
+#elif !defined(WEBRTC_IOS)
 const char* GetFilename(const char* filename) {
   std::string full_path_filename = webrtc::test::OutputPath() + filename;
   return full_path_filename.c_str();
@@ -1588,7 +1588,7 @@ TEST_F(AudioDeviceAPITest, PlayoutBufferTests) {
 
   CheckInitialPlayoutStates();
   EXPECT_EQ(0, audio_device_->PlayoutBuffer(&bufferType, &sizeMS));
-#if defined(_WIN32) || defined(ANDROID) || defined(MAC_IPHONE)
+#if defined(_WIN32) || defined(ANDROID) || defined(WEBRTC_IOS)
   EXPECT_EQ(AudioDeviceModule::kAdaptiveBufferSize, bufferType);
 #else
   EXPECT_EQ(AudioDeviceModule::kFixedBufferSize, bufferType);
@@ -1625,7 +1625,7 @@ TEST_F(AudioDeviceAPITest, PlayoutBufferTests) {
   EXPECT_EQ(0, audio_device_->PlayoutBuffer(&bufferType, &sizeMS));
   EXPECT_EQ(AudioDeviceModule::kAdaptiveBufferSize, bufferType);
 #endif
-#if defined(ANDROID) || defined(MAC_IPHONE)
+#if defined(ANDROID) || defined(WEBRTC_IOS)
   EXPECT_EQ(-1,
             audio_device_->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
                                           kAdmMinPlayoutBufferSizeMs));
@@ -1770,7 +1770,7 @@ TEST_F(AudioDeviceAPITest, RecordingSampleRate) {
 #elif defined(ANDROID)
   TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
   EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000));
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
   TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
   EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000) ||
               (sampleRate == 8000));
@@ -1789,7 +1789,7 @@ TEST_F(AudioDeviceAPITest, PlayoutSampleRate) {
 #elif defined(ANDROID)
   TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
   EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000));
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
   TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
   EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000) ||
               (sampleRate == 8000));
@@ -1802,7 +1802,7 @@ TEST_F(AudioDeviceAPITest, ResetAudioDevice) {
   EXPECT_EQ(0, audio_device_->SetPlayoutDevice(MACRO_DEFAULT_DEVICE));
   EXPECT_EQ(0, audio_device_->SetRecordingDevice(MACRO_DEFAULT_DEVICE));
 
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
   // Not playing or recording, should just return 0
   EXPECT_EQ(0, audio_device_->ResetAudioDevice());
 
@@ -1838,7 +1838,7 @@ TEST_F(AudioDeviceAPITest, SetPlayoutSpeaker) {
   EXPECT_EQ(0, audio_device_->SetPlayoutDevice(MACRO_DEFAULT_DEVICE));
 
   bool loudspeakerOn(false);
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
   // Not playing or recording, should just return a success
   EXPECT_EQ(0, audio_device_->SetLoudspeakerStatus(true));
   EXPECT_EQ(0, audio_device_->GetLoudspeakerStatus(loudspeakerOn));
diff --git a/src/modules/audio_device/main/test/audio_device_test_func.cc b/src/modules/audio_device/main/test/audio_device_test_func.cc
index c549eea5d4..53cfbe22dc 100644
--- a/src/modules/audio_device/main/test/audio_device_test_func.cc
+++ b/src/modules/audio_device/main/test/audio_device_test_func.cc
@@ -25,7 +25,7 @@ int func_test(int);
 //  main()
 // ----------------------------------------------------------------------------
 
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
 int main(int /*argc*/, char* /*argv*/[])
 {
     func_test(0);
diff --git a/src/modules/audio_device/main/test/func_test_manager.cc b/src/modules/audio_device/main/test/func_test_manager.cc
index 9cf87bf1a8..9b22ede47c 100644
--- a/src/modules/audio_device/main/test/func_test_manager.cc
+++ b/src/modules/audio_device/main/test/func_test_manager.cc
@@ -47,7 +47,7 @@ struct AudioPacket
 };
 
 // Helper functions
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
 char* GetFilename(char* filename)
 {
     return filename;
@@ -2667,7 +2667,7 @@ WebRtc_Word32 FuncTestManager::TestAdvancedMBAPI()
     EXPECT_TRUE(audioDevice->Recording());
     EXPECT_TRUE(audioDevice->Playing());
 
-#if defined(_WIN32_WCE) || defined(MAC_IPHONE)
+#if defined(_WIN32_WCE) || defined(WEBRTC_IOS)
     TEST_LOG("\nResetAudioDevice\n \n");
     if (audioDevice->Recording() && audioDevice->Playing())
     {
@@ -2688,7 +2688,7 @@ WebRtc_Word32 FuncTestManager::TestAdvancedMBAPI()
     }
 #endif
 
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
     bool loudspeakerOn(false);
     TEST_LOG("\nSet playout spaker\n \n");
     if (audioDevice->Recording() && audioDevice->Playing())
diff --git a/src/modules/audio_device/main/test/func_test_manager.h b/src/modules/audio_device/main/test/func_test_manager.h
index 8c01f78585..2fc00087fb 100644
--- a/src/modules/audio_device/main/test/func_test_manager.h
+++ b/src/modules/audio_device/main/test/func_test_manager.h
@@ -22,7 +22,7 @@
 #include "list_wrapper.h"
 #include "resampler.h"
 
-#if defined(MAC_IPHONE) || defined(ANDROID)
+#if defined(WEBRTC_IOS) || defined(ANDROID)
 #define USE_SLEEP_AS_PAUSE
 #else
 //#define USE_SLEEP_AS_PAUSE
diff --git a/src/modules/udp_transport/source/udp_transport_impl.cc b/src/modules/udp_transport/source/udp_transport_impl.cc
index e50db5d196..b4c927947f 100644
--- a/src/modules/udp_transport/source/udp_transport_impl.cc
+++ b/src/modules/udp_transport/source/udp_transport_impl.cc
@@ -30,7 +30,7 @@
 #include <sys/socket.h>
 #include <sys/time.h>
 #include <unistd.h>
-#ifndef MAC_IPHONE
+#ifndef WEBRTC_IOS
 #include <net/if_arp.h>
 #endif
 #endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
diff --git a/src/modules/video_render/main/source/video_render_impl.cc b/src/modules/video_render/main/source/video_render_impl.cc
index 1e0ccb742d..3b3d4cef39 100644
--- a/src/modules/video_render/main/source/video_render_impl.cc
+++ b/src/modules/video_render/main/source/video_render_impl.cc
@@ -24,9 +24,9 @@
 #include "windows/video_render_windows_impl.h"
 #define STANDARD_RENDERING kRenderWindows
 
-// MAC_IPHONE should go before WEBRTC_MAC because WEBRTC_MAC
-// gets defined if MAC_IPHONE is defined
-#elif defined(MAC_IPHONE)
+// WEBRTC_IOS should go before WEBRTC_MAC because WEBRTC_MAC
+// gets defined if WEBRTC_IOS is defined
+#elif defined(WEBRTC_IOS)
 #if defined(IPHONE_GLES_RENDERING)
 #define STANDARD_RENDERING kRenderiPhone
 #include "iPhone/video_render_iphone_impl.h"
@@ -116,7 +116,7 @@ ModuleVideoRenderImpl::ModuleVideoRenderImpl(
         }
         break;
 
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
         case kRenderiPhone:
         {
             VideoRenderIPhoneImpl* ptrRenderer = new VideoRenderIPhoneImpl(_id, videoRenderType, window, _fullScreen);
@@ -276,7 +276,7 @@ ModuleVideoRenderImpl::~ModuleVideoRenderImpl()
             break;
 #endif
 
-#elif defined(MAC_IPHONE)
+#elif defined(WEBRTC_IOS)
             case kRenderiPhone:
             break;
 
@@ -350,7 +350,7 @@ WebRtc_Word32 ModuleVideoRenderImpl::ChangeWindow(void* window)
 
 #ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
 
-#if defined(MAC_IPHONE) // MAC_IPHONE must go before WEBRTC_MAC
+#if defined(WEBRTC_IOS) // WEBRTC_IOS must go before WEBRTC_MAC
     _ptrRenderer = NULL;
     delete _ptrRenderer;
 
diff --git a/src/system_wrappers/interface/tick_util.h b/src/system_wrappers/interface/tick_util.h
index 0ce3d90777..c3432154ea 100644
--- a/src/system_wrappers/interface/tick_util.h
+++ b/src/system_wrappers/interface/tick_util.h
@@ -184,11 +184,11 @@ inline TickTime TickTime::Now()
       if (retval != KERN_SUCCESS) {
         // TODO(wu): Implement CHECK similar to chrome for all the platforms.
         // Then replace this with a CHECK(retval == KERN_SUCCESS);
-#ifndef MAC_IPHONE
+#ifndef WEBRTC_IOS
         asm("int3");
 #else
         __builtin_trap();
-#endif // MAC_IPHONE
+#endif // WEBRTC_IOS
       }
     }
     // Use timebase to convert absolute time tick units into nanoseconds.
diff --git a/src/system_wrappers/source/event.cc b/src/system_wrappers/source/event.cc
index adc33dad33..195ac1f100 100644
--- a/src/system_wrappers/source/event.cc
+++ b/src/system_wrappers/source/event.cc
@@ -13,7 +13,7 @@
 #if defined(_WIN32)
     #include <windows.h>
     #include "event_win.h"
-#elif defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
+#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
     #include <ApplicationServices/ApplicationServices.h>
     #include <pthread.h>
     #include "event_posix.h"
@@ -49,7 +49,7 @@ int EventWrapper::KeyPressed()
     {
         return 0;
     }
-#elif defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
+#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
     bool keyDown = false;
     // loop through all Mac virtual key constant values
     for(int keyIndex = 0; keyIndex <= 0x5C; keyIndex++) 
diff --git a/src/system_wrappers/source/thread_posix.cc b/src/system_wrappers/source/thread_posix.cc
index 173cc1bc60..14b3338224 100644
--- a/src/system_wrappers/source/thread_posix.cc
+++ b/src/system_wrappers/source/thread_posix.cc
@@ -157,8 +157,7 @@ ThreadPosix::~ThreadPosix()
     delete _crit_state;
 }
 
-#define HAS_THREAD_ID !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM) && \
-                      !defined(WEBRTC_MAC)
+#define HAS_THREAD_ID !defined(WEBRTC_IOS) && !defined(WEBRTC_MAC)
 #if HAS_THREAD_ID
 bool ThreadPosix::Start(unsigned int& threadID)
 #else
diff --git a/src/system_wrappers/source/trace_impl.h b/src/system_wrappers/source/trace_impl.h
index 2b85813a88..5a6b59c3f7 100644
--- a/src/system_wrappers/source/trace_impl.h
+++ b/src/system_wrappers/source/trace_impl.h
@@ -24,7 +24,7 @@ namespace webrtc {
 // TODO (hellner) the buffer should be close to how much the system can write to
 //                file. Increasing the buffer will not solve anything. Sooner or
 //                later the buffer is going to fill up anyways.
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
     #define WEBRTC_TRACE_MAX_QUEUE  2000
 #else
     #define WEBRTC_TRACE_MAX_QUEUE  8000
diff --git a/src/voice_engine/test/auto_test/standard/audio_processing_test.cc b/src/voice_engine/test/auto_test/standard/audio_processing_test.cc
index a7a5a07d99..fc759b0b08 100644
--- a/src/voice_engine/test/auto_test/standard/audio_processing_test.cc
+++ b/src/voice_engine/test/auto_test/standard/audio_processing_test.cc
@@ -138,7 +138,7 @@ class AudioProcessingTest : public AfterStreamingFixture {
   }
 };
 
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
 
 TEST_F(AudioProcessingTest, AgcIsOnByDefault) {
   bool agc_enabled = false;
@@ -219,7 +219,7 @@ TEST_F(AudioProcessingTest, DISABLED_TestVoiceActivityDetectionWithObserver) {
   EXPECT_EQ(0, voe_apm_->DeRegisterRxVadObserver(channel_));
 }
 
-#endif   // !MAC_IPHONE && !WEBRTC_ANDROID
+#endif   // !WEBRTC_IOS && !WEBRTC_ANDROID
 
 TEST_F(AudioProcessingTest, EnablingEcAecmShouldEnableEcAecm) {
   // This one apparently applies to Android and iPhone as well.
@@ -377,7 +377,7 @@ TEST_F(AudioProcessingTest, CanStartAndStopDebugRecording) {
   EXPECT_EQ(0, voe_apm_->StopDebugRecording());
 }
 
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
 
 TEST_F(AudioProcessingTest, AgcIsOffByDefaultAndDigital) {
   bool agc_enabled = true;
@@ -412,4 +412,4 @@ TEST_F(AudioProcessingTest, TestVoiceActivityDetection) {
   TryDetectingSpeechAfterSilence();
 }
 
-#endif  // MAC_IPHONE || WEBRTC_ANDROID
+#endif  // WEBRTC_IOS || WEBRTC_ANDROID
diff --git a/src/voice_engine/test/auto_test/standard/dtmf_test.cc b/src/voice_engine/test/auto_test/standard/dtmf_test.cc
index 6aa4625db4..b1b1666528 100644
--- a/src/voice_engine/test/auto_test/standard/dtmf_test.cc
+++ b/src/voice_engine/test/auto_test/standard/dtmf_test.cc
@@ -52,7 +52,7 @@ TEST_F(DtmfTest, TestTwoNonDtmfEvents) {
   EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 110, true));
 }
 
-#ifndef MAC_IPHONE
+#ifndef WEBRTC_IOS
 TEST_F(DtmfTest, ManualCanDisableDtmfPlayoutExceptOnIphone) {
   TEST_LOG("Disabling DTMF playout (no tone should be heard) \n");
   EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, false));
diff --git a/src/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc b/src/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
index 6e56347fd5..edb7f56288 100644
--- a/src/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
+++ b/src/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
@@ -42,14 +42,14 @@ TEST_F(HardwareBeforeStreamingTest,
 
 // Tests that only apply to mobile:
 
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
 TEST_F(HardwareBeforeStreamingTest, ResetsAudioDeviceOnIphone) {
   EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
 }
 #endif
 
 // Tests that only apply to desktop:
-#if !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
+#if !defined(WEBRTC_IOS) & !defined(WEBRTC_ANDROID)
 
 TEST_F(HardwareBeforeStreamingTest, GetSystemCpuLoadSucceeds) {
 #ifdef _WIN32
@@ -163,4 +163,4 @@ TEST_F(HardwareBeforeStreamingTest,
 #endif
 }
 
-#endif // !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
+#endif // !defined(WEBRTC_IOS) & !defined(WEBRTC_ANDROID)
diff --git a/src/voice_engine/test/auto_test/standard/hardware_test.cc b/src/voice_engine/test/auto_test/standard/hardware_test.cc
index 41145e1c88..f9994dac83 100644
--- a/src/voice_engine/test/auto_test/standard/hardware_test.cc
+++ b/src/voice_engine/test/auto_test/standard/hardware_test.cc
@@ -15,7 +15,7 @@
 class HardwareTest : public AfterStreamingFixture {
 };
 
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
 TEST_F(HardwareTest, AbleToQueryForDevices) {
   int num_recording_devices = 0;
   int num_playout_devices = 0;
@@ -84,7 +84,7 @@ TEST_F(HardwareTest, GetSystemCpuLoadWorksExceptOnMacAndAndroid) {
 #endif
 
 TEST_F(HardwareTest, BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer) {
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
   // Ensure the sound device is reset on iPhone.
   EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
   Sleep(2000);
diff --git a/src/voice_engine/test/auto_test/standard/video_sync_test.cc b/src/voice_engine/test/auto_test/standard/video_sync_test.cc
index 65516a1014..b8aedcdb95 100644
--- a/src/voice_engine/test/auto_test/standard/video_sync_test.cc
+++ b/src/voice_engine/test/auto_test/standard/video_sync_test.cc
@@ -14,11 +14,11 @@
 
 #include "voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
 
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
   const int kMinimumReasonableDelayEstimateMs = 30;
 #else
   const int kMinimumReasonableDelayEstimateMs = 45;
-#endif  // !MAC_IPHONE
+#endif  // !WEBRTC_IOS
 
 class VideoSyncTest : public AfterStreamingFixture {
  protected:
diff --git a/src/voice_engine/test/auto_test/standard/volume_test.cc b/src/voice_engine/test/auto_test/standard/volume_test.cc
index d9302816a7..fda867da10 100644
--- a/src/voice_engine/test/auto_test/standard/volume_test.cc
+++ b/src/voice_engine/test/auto_test/standard/volume_test.cc
@@ -87,7 +87,7 @@ TEST_F(VolumeTest, ManualSetVolumeWorks) {
   Sleep(1000);
 }
 
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
 
 TEST_F(VolumeTest, DISABLED_ON_LINUX(DefaultMicrophoneVolumeIsAtMost255)) {
   unsigned int volume = 1000;
@@ -136,9 +136,9 @@ TEST_F(VolumeTest, ManualCanSetChannelScaling) {
   Sleep(2000);
 }
 
-#endif  // !MAC_IPHONE
+#endif  // !WEBRTC_IOS
 
-#if !defined(WEBRTC_ANDROID) && !defined(MAC_IPHONE)
+#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
 
 TEST_F(VolumeTest, InputMutingIsNotEnabledByDefault) {
   bool is_muted = true;
@@ -277,4 +277,4 @@ TEST_F(VolumeTest, ManualTestChannelPanning) {
   EXPECT_FLOAT_EQ(0.8f, right);
 }
 
-#endif  // !WEBRTC_ANDROID && !MAC_IPHONE
+#endif  // !WEBRTC_ANDROID && !WEBRTC_IOS
diff --git a/src/voice_engine/test/auto_test/voe_extended_test.cc b/src/voice_engine/test/auto_test/voe_extended_test.cc
index b90e63c0fc..2fcd6423c5 100644
--- a/src/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/src/voice_engine/test/auto_test/voe_extended_test.cc
@@ -41,7 +41,7 @@ const int RemotePort = 12345; // transmit to this UDP port
 const char* RemoteIP = "192.168.200.1"; // transmit to this IP address
 #endif
 
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
 #define SLEEP_IF_IPHONE(x) SLEEP(x)
 #else
 #define SLEEP_IF_IPHONE(x)
@@ -418,7 +418,7 @@ int VoEExtendedTest::TestBase() {
   TEST_MUSTPASS(voe_base_->Init());
   MARK();
   TEST_MUSTPASS(voe_base_->Terminate());
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   // verify AEC recording
   TEST_MUSTPASS(voe_base_->Init());
   MARK(); // verify output dat-files
@@ -577,7 +577,7 @@ int VoEExtendedTest::TestBase() {
 
   ch = voe_base_->CreateChannel();
 
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
   printf("\nNOTE: Local IP must be set in source code (line %d) \n",
       __LINE__ + 1);
   char* localIp = "127.0.0.1";
@@ -1432,7 +1432,7 @@ int VoEExtendedTest::TestBase() {
   voe_base_->DeleteChannel(ch);
   ch = voe_base_->CreateChannel();
 
-#ifndef MAC_IPHONE
+#ifndef WEBRTC_IOS
   // bind to local IP and try again
   strcpy(localIp, "127.0.0.1");
 #else
@@ -3326,7 +3326,7 @@ int VoEExtendedTest::TestDtmf() {
   SLEEP(1000);
   // Switch codec
   CodecInst ci;
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   ci.channels = 1;
   ci.pacsize = 480;
   ci.plfreq = 16000;
@@ -4596,7 +4596,7 @@ int VoEExtendedTest::TestFile() {
   // Record mixed (speaker + microphone) signal to file
 
 
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
   TEST(StartRecordingSpeakerStereo);
   ANL();
   TEST(StopRecordingSpeakerStereo);
@@ -4633,11 +4633,11 @@ int VoEExtendedTest::TestFile() {
   ANL();
 #else
   TEST_LOG("Skipping stereo record tests -"
-      " MAC_IPHONE or WEBRTC_ANDROID is defined \n");
-#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+      " WEBRTC_IOS or WEBRTC_ANDROID is defined \n");
+#endif // #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
   // Conversion between different file formats
 
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
   TEST_MUSTPASS(voe_base_->StopPlayout(0));
   TEST_MUSTPASS(voe_base_->StopSend(0));
 #endif
@@ -4698,7 +4698,7 @@ int VoEExtendedTest::TestFile() {
           (output_path + "singleUserDemoConv_dummy.pcm").c_str()));MARK();
   AOK();ANL();
 
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
   TEST_MUSTPASS(voe_base_->StartPlayout(0));
   TEST_MUSTPASS(voe_base_->StartSend(0));
 #endif
@@ -4932,7 +4932,7 @@ int VoEExtendedTest::TestHardware() {
   }
   ANL();
 
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
   // GetRecording/PlayoutDeviceStatus
   TEST(Getrecording/PlayoutDeviceStatus);
   ANL();
@@ -5032,8 +5032,8 @@ int VoEExtendedTest::TestHardware() {
   MARK();
 #endif
   ANL();
-#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
-#if defined(MAC_IPHONE)
+#endif // #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS)
   TEST(ResetSoundDevice); ANL();
 
   for (int p=0; p<=60; p+=20)
@@ -5072,8 +5072,8 @@ int VoEExtendedTest::TestHardware() {
   TEST_MUSTPASS(voe_base_->StartSend(0));
   TEST_MUSTPASS(voe_base_->StartPlayout(0));
   TEST_MUSTPASS(voe_base_->StartReceive(0));
-#endif // defined(MAC_IPHONE))
-#ifdef MAC_IPHONE
+#endif // defined(WEBRTC_IOS))
+#ifdef WEBRTC_IOS
   TEST_LOG("\nNOTE: Always run hardware tests also without extended tests "
       "enabled,\nsince the extended tests are pre-streaming tests only.\n");
 #endif
@@ -5111,7 +5111,7 @@ int VoEExtendedTest::TestNetwork() {
 #ifdef WEBRTC_ANDROID
   int sleepTime = 200;
   int sleepTime2 = 250;
-#elif defined(MAC_IPHONE) // MAC_IPHONE needs more delay for getSourceInfo()
+#elif defined(WEBRTC_IOS) // WEBRTC_IOS needs more delay for getSourceInfo()
   int sleepTime = 150;
   int sleepTime2 = 200;
 #else
@@ -5145,7 +5145,7 @@ int VoEExtendedTest::TestNetwork() {
   TEST(GetLocalIP);
   ANL();
 
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
   // Should fail
   TEST_MUSTPASS(!netw->GetLocalIP(NULL, 0)); MARK();
   TEST_ERROR(VE_FUNC_NOT_SUPPORTED);
@@ -5325,7 +5325,7 @@ int VoEExtendedTest::TestNetwork() {
   TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9020));
   TEST_MUSTPASS(voe_base_->StartSend(0));
   SLEEP(sleepTime);
-#ifdef MAC_IPHONE
+#ifdef WEBRTC_IOS
   SLEEP(500); // Need extra pause for some reason
 #endif
 
@@ -5461,7 +5461,7 @@ int VoEExtendedTest::TestNetwork() {
   MARK();
 
   // STATE: external transport is disabled
-#if defined(WEBRTC_ANDROID) || defined(MAC_IPHONE)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
   int testError = VE_FUNC_NOT_SUPPORTED;
 #else
   int testError = VE_EXTERNAL_TRANSPORT_ENABLED;
@@ -7064,7 +7064,7 @@ int VoEExtendedTest::TestRTP_RTCP() {
   TEST_LOG("Turn FEC and VAD on and wait for 4 seconds and ensure that "
     "the jitter is still small...");
   CodecInst cinst;
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   cinst.pltype = 104;
   strcpy(cinst.plname, "isac");
   cinst.plfreq = 32000;
@@ -7238,7 +7238,7 @@ int VoEExtendedTest::TestRTP_RTCP() {
   cinst.channels = 1;
   cinst.rate = 0;
   TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   cinst.pltype = 104;
   strcpy(cinst.plname, "isac");
   cinst.plfreq = 32000;
@@ -7417,7 +7417,7 @@ int VoEExtendedTest::TestVolumeControl()
 
   TEST_MUSTPASS(voe_base_->Init());
   TEST_MUSTPASS(voe_base_->CreateChannel());
-#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE)))
+#if (defined _TEST_HARDWARE_ && (!defined(WEBRTC_IOS)))
 #if defined(_WIN32)
   TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
   TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
@@ -7439,23 +7439,23 @@ int VoEExtendedTest::TestVolumeControl()
   ////////////////////////////
   // Actual test starts here
 
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
   TEST(SetSpeakerVolume);
   ANL();
   TEST_MUSTPASS(-1 != volume->SetSpeakerVolume(256));
   MARK();
   TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
   ANL();
-#endif // #if !defined(MAC_IPHONE)
+#endif // #if !defined(WEBRTC_IOS)
 
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   TEST(SetMicVolume); ANL();
   TEST_MUSTPASS(-1 != volume->SetMicVolume(256)); MARK();
   TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
   ANL();
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#endif // #if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
 
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
   TEST(SetChannelOutputVolumeScaling);
   ANL();
   TEST_MUSTPASS(-1 != volume->SetChannelOutputVolumeScaling(0, (float)-0.1));
@@ -7465,8 +7465,8 @@ int VoEExtendedTest::TestVolumeControl()
   MARK();
   TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
   ANL();
-#endif // #if !defined(MAC_IPHONE)
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#endif // #if !defined(WEBRTC_IOS)
+#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
   TEST(SetOutputVolumePan);
   ANL();
   TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)-0.1,
@@ -7506,7 +7506,7 @@ int VoEExtendedTest::TestVolumeControl()
   MARK();
   TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
   ANL();
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#endif // #if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
 #ifdef _TEST_FILE_
   TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
 #endif
diff --git a/src/voice_engine/test/auto_test/voe_standard_test.cc b/src/voice_engine/test/auto_test/voe_standard_test.cc
index 69eb490e6c..60cc9b2e0a 100644
--- a/src/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/src/voice_engine/test/auto_test/voe_standard_test.cc
@@ -598,7 +598,7 @@ int RunInManualMode(int argc, char** argv) {
 //                                       main
 // ----------------------------------------------------------------------------
 
-#if !defined(MAC_IPHONE)
+#if !defined(WEBRTC_IOS)
 int main(int argc, char** argv) {
   if (argc > 1 && std::string(argv[1]) == "--automated") {
     // This function is defined in automated_mode.cc to avoid macro clashes
@@ -608,4 +608,4 @@ int main(int argc, char** argv) {
 
   return RunInManualMode(argc, argv);
 }
-#endif //#if !defined(MAC_IPHONE)
+#endif //#if !defined(WEBRTC_IOS)
diff --git a/src/voice_engine/voe_audio_processing_impl.cc b/src/voice_engine/voe_audio_processing_impl.cc
index cabecc9dcd..6390970903 100644
--- a/src/voice_engine/voe_audio_processing_impl.cc
+++ b/src/voice_engine/voe_audio_processing_impl.cc
@@ -44,7 +44,7 @@
 
 namespace webrtc {
 
-#if defined(WEBRTC_ANDROID) || defined(MAC_IPHONE) || defined(MAC_IPHONE_SIM)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
 static const EcModes kDefaultEcMode = kEcAecm;
 #else
 static const EcModes kDefaultEcMode = kEcAec;
@@ -183,7 +183,7 @@ int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode) {
     return -1;
   }
 
-#if defined(MAC_IPHONE) || defined(ATA) || defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
   if (mode == kAgcAdaptiveAnalog) {
     _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
         "SetAgcStatus() invalid Agc mode for mobile device");
diff --git a/src/voice_engine/voe_hardware_impl.cc b/src/voice_engine/voe_hardware_impl.cc
index 6bec665b13..b484cf4e8f 100644
--- a/src/voice_engine/voe_hardware_impl.cc
+++ b/src/voice_engine/voe_hardware_impl.cc
@@ -634,7 +634,7 @@ int VoEHardwareImpl::ResetAudioDevice()
         return -1;
     }
 
-#if defined(MAC_IPHONE)
+#if defined(WEBRTC_IOS)
     if (_shared->audio_device()->ResetAudioDevice() < 0)
     {
         _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
diff --git a/src/voice_engine/voice_engine_defines.h b/src/voice_engine/voice_engine_defines.h
index c16e0fe35f..13183dd1ee 100644
--- a/src/voice_engine/voice_engine_defines.h
+++ b/src/voice_engine/voice_engine_defines.h
@@ -457,7 +457,7 @@ namespace webrtc
 #include <sys/time.h>
 #include <time.h>
 #include <AudioUnit/AudioUnit.h>
-#if !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM)
+#if !defined(WEBRTC_IOS)
   #include <CoreServices/CoreServices.h>
   #include <CoreAudio/CoreAudio.h>
   #include <AudioToolbox/DefaultAudioOutput.h>
@@ -517,7 +517,7 @@ namespace
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
 
 // iPhone specific
-#if defined(MAC_IPHONE) || defined(MAC_IPHONE_SIM)
+#if defined(WEBRTC_IOS)
 
 // ----------------------------------------------------------------------------
 //  Enumerators