Remove DISABLED_ON_ macros.

Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.

This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.

The change also removes gtest_disable.h as an unused include from many
other files.

BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1547343002 .

Cr-Commit-Position: refs/heads/master@{#11150}

25 files changed, 487 insertions, 176 deletions

diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index a7dd3d4484..24ecc694ff 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -20,7 +20,6 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -164,7 +163,12 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   FrameType last_frame_type_;
 };
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
+#else
+#define MAYBE_AddCodecGetCodec AddCodecGetCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
   // Add codec.
   for (size_t n = 0; n < codecs_.size(); ++n) {
     if (n & 0x1)  // Just add codecs with odd index.
@@ -188,7 +192,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
   }
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
+#else
+#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
   const CodecIdInst codec1(RentACodec::CodecId::kPCMA);
   CodecInst codec2 = codec1.inst;
   ++codec2.pltype;
@@ -209,7 +218,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
   EXPECT_EQ(true, CodecsEqual(codec2, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
+#else
+#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
+#endif
+TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
   const CodecIdInst codec1(RentACodec::CodecId::kPCMU);
   CodecIdInst codec2(RentACodec::CodecId::kPCMA);
   codec2.inst.pltype = codec1.inst.pltype;
@@ -229,7 +243,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
   EXPECT_EQ(true, CodecsEqual(codec2.inst, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
+#else
+#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
   const CodecIdInst codec(RentACodec::CodecId::kPCMA);
   const int payload_type = codec.inst.pltype;
   EXPECT_EQ(
@@ -247,7 +266,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
   EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &ci));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SampleRate DISABLED_SampleRate
+#else
+#define MAYBE_SampleRate SampleRate
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
   const RentACodec::CodecId kCodecId[] = {RentACodec::CodecId::kISAC,
                                           RentACodec::CodecId::kISACSWB};
   AddSetOfCodecs(kCodecId);
@@ -265,7 +289,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
   }
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
   receiver_->EnableVad();
   EXPECT_TRUE(receiver_->vad_enabled());
   const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
@@ -293,14 +322,13 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
 
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
 #else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
+#define MAYBE_LastAudioCodec LastAudioCodec
 #endif
-
-TEST_F(AcmReceiverTestOldApi,
-       DISABLED_ON_ANDROID(IF_ISAC_FLOAT(LastAudioCodec))) {
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
   const RentACodec::CodecId kCodecId[] = {
       RentACodec::CodecId::kISAC, RentACodec::CodecId::kPCMA,
       RentACodec::CodecId::kISACSWB, RentACodec::CodecId::kPCM16Bswb32kHz};
@@ -363,6 +391,7 @@ TEST_F(AcmReceiverTestOldApi,
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
+#endif
 
 }  // namespace acm2
 
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 8de6c9100d..ef48a48d94 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -41,7 +41,6 @@
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::AtLeast;
 using ::testing::Invoke;
@@ -238,7 +237,12 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
 
 // Check if the statistics are initialized correctly. Before any call to ACM
 // all fields have to be zero.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_InitializedToZero DISABLED_InitializedToZero
+#else
+#define MAYBE_InitializedToZero InitializedToZero
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_InitializedToZero) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   acm_->GetDecodingCallStatistics(&stats);
@@ -253,7 +257,12 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
 // Insert some packets and pull audio. Check statistics are valid. Then,
 // simulate packet loss and check if PLC and PLC-to-CNG statistics are
 // correctly updated.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NetEqCalls DISABLED_NetEqCalls
+#else
+#define MAYBE_NetEqCalls NetEqCalls
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   const int kNumNormalCalls = 10;
@@ -320,15 +329,9 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
 }
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
-#else
-#define IF_ISAC(x) DISABLED_##x
-#endif
-
 // Verifies that the RTP timestamp series is not reset when the codec is
 // changed.
-TEST_F(AudioCodingModuleTestOldApi,
-       IF_ISAC(TimestampSeriesContinuesWhenCodecChanges)) {
+TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
   RegisterCodec();  // This registers the default codec.
   uint32_t expected_ts = input_frame_.timestamp_;
   int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
@@ -360,6 +363,7 @@ TEST_F(AudioCodingModuleTestOldApi,
     expected_ts += codec_.pacsize;
   }
 }
+#endif
 
 // Introduce this class to set different expectations on the number of encoded
 // bytes. This class expects all encoded packets to be 9 bytes (matching one
@@ -582,7 +586,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
-TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
@@ -686,9 +695,16 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
 
 class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
  protected:
@@ -838,9 +854,16 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
 
 // Disabling all of these tests on iOS until file support has been added.
 // See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
@@ -919,12 +942,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
-#else
-#define IF_ALL_CODECS(x) DISABLED_##x
-#endif
-
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(8kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
   Run(8000, PlatformChecksum("908002dc01fc4eb1d2be24eb1d3f354b",
                              "dcee98c623b147ebe1b40dd30efa896e",
                              "adc92e173f908f93b96ba5844209815a",
@@ -932,7 +950,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(8kHzOutput)) {
       std::vector<ExternalDecoder>());
 }
 
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(16kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
   Run(16000, PlatformChecksum("a909560b5ca49fa472b17b7b277195e9",
                               "f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
                               "8cffa6abcb3e18e33b9d857666dff66a",
@@ -940,7 +958,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(16kHzOutput)) {
       std::vector<ExternalDecoder>());
 }
 
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(32kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
   Run(32000, PlatformChecksum("441aab4b347fb3db4e9244337aca8d8e",
                               "306e0d990ee6e92de3fbecc0123ece37",
                               "3e126fe894720c3f85edadcc91964ba5",
@@ -948,7 +966,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(32kHzOutput)) {
       std::vector<ExternalDecoder>());
 }
 
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(48kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
   Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
                               "aa7c232f63a67b2a72703593bdd172e0",
                               "0155665e93067c4e89256b944dd11999",
@@ -956,8 +974,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(48kHzOutput)) {
       std::vector<ExternalDecoder>());
 }
 
-TEST_F(AcmReceiverBitExactnessOldApi,
-       IF_ALL_CODECS(48kHzOutputExternalDecoder)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
   // Class intended to forward a call from a mock DecodeInternal to Decode on
   // the real decoder's Decode. DecodeInternal for the real decoder isn't
   // public.
@@ -1016,6 +1033,7 @@ TEST_F(AcmReceiverBitExactnessOldApi,
 
   EXPECT_CALL(mock_decoder, Die());
 }
+#endif
 
 // This test verifies bit exactness for the send-side of ACM. The test setup is
 // a chain of three different test classes:
@@ -1194,7 +1212,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   rtc::Md5Digest payload_checksum_;
 };
 
-TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb30ms)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "0b58f9eeee43d5891f5f6c75e77984a3",
@@ -1209,7 +1228,7 @@ TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb30ms)) {
       33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb60ms)) {
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "1ad29139a04782a33daad8c2b9b35875",
@@ -1223,15 +1242,15 @@ TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb60ms)) {
           "9e0a0ab743ad987b55b8e14802769c56"),
       16, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms
 #else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
+#define MAYBE_IsacSwb30ms IsacSwb30ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "5683b58da0fbf2063c7adc2e6bfb3fb8",
@@ -1243,6 +1262,7 @@ TEST_F(AcmSenderBitExactnessOldApi,
           "android_arm64_payload"),
       33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
 TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
@@ -1324,13 +1344,13 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
       test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-#ifdef WEBRTC_CODEC_ILBC
-#define IF_ILBC(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms
 #else
-#define IF_ILBC(x) DISABLED_##x
+#define MAYBE_Ilbc_30ms Ilbc_30ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
+#if defined(WEBRTC_CODEC_ILBC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7b6ec10910debd9af08011d3ed5249f7",
@@ -1342,14 +1362,15 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
           "android_arm64_payload"),
       33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-#ifdef WEBRTC_CODEC_G722
-#define IF_G722(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_20ms DISABLED_G722_20ms
 #else
-#define IF_G722(x) DISABLED_##x
+#define MAYBE_G722_20ms G722_20ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7d759436f2533582950d148b5161a36c",
@@ -1361,9 +1382,15 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
           "android_arm64_payload"),
       50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
+#else
+#define MAYBE_G722_stereo_20ms G722_stereo_20ms
+#endif
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7190ee718ab3d80eca181e5f7140c210",
@@ -1375,6 +1402,7 @@ TEST_F(AcmSenderBitExactnessOldApi,
           "android_arm64_payload"),
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
+#endif
 
 TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
@@ -1490,7 +1518,12 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
 
 // The result on the Android platforms is inconsistent for this test case.
 // On android_rel the result is different from android and android arm64 rel.
-TEST_F(AcmSetBitRateOldApi, DISABLED_ON_ANDROID(Opus_48khz_20ms_100kbps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
+#else
+#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
+#endif
+TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
   Run(100000, 100888);
 }
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
index f41a17ae3a..85aaef1143 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -19,7 +19,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index e02e92dd8f..1ddc7f2edf 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -21,7 +21,6 @@
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -276,7 +275,12 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
   }
 };
 
-TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RunTest DISABLED_RunTest
+#else
+#define MAYBE_RunTest RunTest
+#endif
+TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
   RunTest(8);
 }
 
@@ -301,7 +305,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
   double drift_factor;
 };
 
-TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -314,7 +318,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
   }
 };
 
-TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -342,7 +346,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
   RunTest(1000);
 }
 
@@ -361,7 +365,10 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
+// TODO(pbos): Enable on non-Android, this went failing while being accidentally
+// disabled on all platforms and not just Android.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
+TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
   RunTest(100);
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index b3d6d8c7a7..a6b9388f6c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -30,7 +30,6 @@
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
@@ -930,13 +929,13 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecoderError DISABLED_DecoderError
 #else
-#define IF_ISAC(x) DISABLED_##x
+#define MAYBE_DecoderError DecoderError
 #endif
-
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
@@ -974,6 +973,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
     EXPECT_EQ(1, out_data_[i]);
   }
 }
+#endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   NetEqOutputType type;
@@ -1171,7 +1171,8 @@ TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(32000);
 }
 
-TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of
@@ -1250,6 +1251,7 @@ TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
   --rtp_info.header.ssrc;
   EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 }
+#endif
 
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
diff --git a/webrtc/modules/audio_coding/test/Tester.cc b/webrtc/modules/audio_coding/test/Tester.cc
index 3ff3dd8cd4..a27f0bc58b 100644
--- a/webrtc/modules/audio_coding/test/Tester.cc
+++ b/webrtc/modules/audio_coding/test/Tester.cc
@@ -26,7 +26,6 @@
 #include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using webrtc::Trace;
 
@@ -42,7 +41,11 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
+#else
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_encodedecode_trace.txt").c_str());
@@ -50,51 +53,54 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
   Trace::ReturnTrace();
 }
 
-#ifdef WEBRTC_CODEC_RED
-#define IF_RED(x) x
+#if defined(WEBRTC_CODEC_RED)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
 #else
-#define IF_RED(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TestRedFec) {
 #endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_fec_trace.txt").c_str());
   webrtc::TestRedFec().Perform();
   Trace::ReturnTrace();
 }
+#endif
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
 #else
-#define IF_ISAC(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TestIsac) {
 #endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_isac_trace.txt").c_str());
   webrtc::ISACTest(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
 #else
-#define IF_ALL_CODECS(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TwoWayCommunication) {
 #endif
-
-TEST(AudioCodingModuleTest,
-     DISABLED_ON_ANDROID(IF_ALL_CODECS(TwoWayCommunication))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_twowaycom_trace.txt").c_str());
   webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
+#else
+TEST(AudioCodingModuleTest, TestStereo) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_stereo_trace.txt").c_str());
@@ -102,7 +108,11 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestWebRtcVadDtx)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
+#else
+TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_vaddtx_trace.txt").c_str());
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index d7c0411c92..97471bb566 100644
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -17,7 +17,6 @@
 #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"
 
 namespace webrtc {
 
@@ -199,23 +198,50 @@ class TargetDelayTest : public ::testing::Test {
   uint8_t payload_[kPayloadLenBytes];
 };
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutOfRangeInput DISABLED_OutOfRangeInput
+#else
+#define MAYBE_OutOfRangeInput OutOfRangeInput
+#endif
+TEST_F(TargetDelayTest, MAYBE_OutOfRangeInput) {
   OutOfRangeInput();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NoTargetDelayBufferSizeChanges \
+  DISABLED_NoTargetDelayBufferSizeChanges
+#else
+#define MAYBE_NoTargetDelayBufferSizeChanges NoTargetDelayBufferSizeChanges
+#endif
+TEST_F(TargetDelayTest, MAYBE_NoTargetDelayBufferSizeChanges) {
   NoTargetDelayBufferSizeChanges();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_WithTargetDelayBufferNotChanging \
+  DISABLED_WithTargetDelayBufferNotChanging
+#else
+#define MAYBE_WithTargetDelayBufferNotChanging WithTargetDelayBufferNotChanging
+#endif
+TEST_F(TargetDelayTest, MAYBE_WithTargetDelayBufferNotChanging) {
   WithTargetDelayBufferNotChanging();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RequiredDelayAtCorrectRange DISABLED_RequiredDelayAtCorrectRange
+#else
+#define MAYBE_RequiredDelayAtCorrectRange RequiredDelayAtCorrectRange
+#endif
+TEST_F(TargetDelayTest, MAYBE_RequiredDelayAtCorrectRange) {
   RequiredDelayAtCorrectRange();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
+#else
+#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
+#endif
+TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
   TargetDelayBufferMinMax();
 }
 
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index 32f5a3e6e2..567118d828 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -14,7 +14,6 @@ extern "C" {
 }
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 namespace {
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
index b2e11981fa..7f152bf942 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
@@ -14,7 +14,6 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index eff791d129..d4bb8aa513 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -28,7 +28,6 @@
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "gtest/gtest.h"
 #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
diff --git a/webrtc/modules/audio_processing/transient/file_utils_unittest.cc b/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
index 7a035d2b41..7fb7d2d6a9 100644
--- a/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
@@ -17,7 +17,6 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -59,7 +58,12 @@ class TransientFileUtilsTest: public ::testing::Test {
   const std::string kTestFileNamef;
 };
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToFloat DISABLED_ConvertByteArrayToFloat
+#else
+#define MAYBE_ConvertByteArrayToFloat ConvertByteArrayToFloat
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToFloat) {
   float value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToFloat(kPiBytesf, &value));
@@ -72,7 +76,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
   EXPECT_FLOAT_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToDouble DISABLED_ConvertByteArrayToDouble
+#else
+#define MAYBE_ConvertByteArrayToDouble ConvertByteArrayToDouble
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToDouble) {
   double value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToDouble(kPiBytes, &value));
@@ -85,7 +94,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
   EXPECT_DOUBLE_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertFloatToByteArray DISABLED_ConvertFloatToByteArray
+#else
+#define MAYBE_ConvertFloatToByteArray ConvertFloatToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertFloatToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[4]);
 
   EXPECT_EQ(0, ConvertFloatToByteArray(kPi, bytes.get()));
@@ -98,7 +112,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytesf, 4));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertDoubleToByteArray DISABLED_ConvertDoubleToByteArray
+#else
+#define MAYBE_ConvertDoubleToByteArray ConvertDoubleToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertDoubleToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[8]);
 
   EXPECT_EQ(0, ConvertDoubleToByteArray(kPi, bytes.get()));
@@ -111,7 +130,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytes, 8));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16BufferFromFile DISABLED_ReadInt16BufferFromFile
+#else
+#define MAYBE_ReadInt16BufferFromFile ReadInt16BufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -149,8 +173,13 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
   EXPECT_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest,
-       DISABLED_ON_IOS(ReadInt16FromFileToFloatBuffer)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToFloatBuffer \
+  DISABLED_ReadInt16FromFileToFloatBuffer
+#else
+#define MAYBE_ReadInt16FromFileToFloatBuffer ReadInt16FromFileToFloatBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -191,8 +220,13 @@ TEST_F(TransientFileUtilsTest,
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest,
-       DISABLED_ON_IOS(ReadInt16FromFileToDoubleBuffer)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToDoubleBuffer \
+  DISABLED_ReadInt16FromFileToDoubleBuffer
+#else
+#define MAYBE_ReadInt16FromFileToDoubleBuffer ReadInt16FromFileToDoubleBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -232,7 +266,12 @@ TEST_F(TransientFileUtilsTest,
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadFloatBufferFromFile DISABLED_ReadFloatBufferFromFile
+#else
+#define MAYBE_ReadFloatBufferFromFile ReadFloatBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
   std::string test_filename = kTestFileNamef;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -269,7 +308,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
   EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadDoubleBufferFromFile DISABLED_ReadDoubleBufferFromFile
+#else
+#define MAYBE_ReadDoubleBufferFromFile ReadDoubleBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -306,7 +350,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
   EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteInt16BufferToFile DISABLED_WriteInt16BufferToFile
+#else
+#define MAYBE_WriteInt16BufferToFile WriteInt16BufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -348,7 +397,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteFloatBufferToFile DISABLED_WriteFloatBufferToFile
+#else
+#define MAYBE_WriteFloatBufferToFile WriteFloatBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -390,7 +444,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteDoubleBufferToFile DISABLED_WriteDoubleBufferToFile
+#else
+#define MAYBE_WriteDoubleBufferToFile WriteDoubleBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -432,7 +491,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ExpectedErrorReturnValues)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ExpectedErrorReturnValues DISABLED_ExpectedErrorReturnValues
+#else
+#define MAYBE_ExpectedErrorReturnValues ExpectedErrorReturnValues
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
   std::string test_filename = kTestFileName;
 
   double value;
diff --git a/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc b/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
index 6a70a3f92c..b60077510b 100644
--- a/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
@@ -19,8 +19,7 @@
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
- #include "webrtc/typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 
@@ -37,7 +36,11 @@ static const size_t kNumberOfSampleRates =
 // The files contain all the results in double precision (Little endian).
 // The audio files used with different sample rates are stored in the same
 // directory.
-TEST(TransientDetectorTest, DISABLED_ON_IOS(CorrectnessBasedOnFiles)) {
+#if defined(WEBRTC_IOS)
+TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
+#else
+TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
+#endif
   for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
     int sample_rate_hz = kSampleRatesHz[i];
 
diff --git a/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc b/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
index 7c99f4f161..e4e9048f88 100644
--- a/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
@@ -19,7 +19,6 @@
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -69,7 +68,11 @@ TEST(WPDTreeTest, Construction) {
 // It also writes the results in its own set of files in the out directory.
 // Matlab and output files contain all the results in double precision (Little
 // endian) appended.
-TEST(WPDTreeTest, DISABLED_ON_IOS(CorrectnessBasedOnMatlabFiles)) {
+#if defined(WEBRTC_IOS)
+TEST(WPDTreeTest, DISABLED_CorrectnessBasedOnMatlabFiles) {
+#else
+TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
+#endif
   // 10 ms at 16000 Hz.
   const size_t kTestBufferSize = 160;
   const int kLevels = 3;
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
index 5462d05d37..1d1dcc7066 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -16,7 +16,6 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -55,7 +54,11 @@ TEST(StandaloneVadTest, Api) {
   EXPECT_EQ(kMode, vad->mode());
 }
 
-TEST(StandaloneVadTest, DISABLED_ON_IOS(ActivityDetection)) {
+#if defined(WEBRTC_IOS)
+TEST(StandaloneVadTest, DISABLED_ActivityDetection) {
+#else
+TEST(StandaloneVadTest, ActivityDetection) {
+#endif
   rtc::scoped_ptr<StandaloneVad> vad(StandaloneVad::Create());
   const size_t kDataLength = kLength10Ms;
   int16_t data[kDataLength] = {0};
diff --git a/webrtc/modules/media_file/media_file_unittest.cc b/webrtc/modules/media_file/media_file_unittest.cc
index c12ea57ea2..6541a8fb7c 100644
--- a/webrtc/modules/media_file/media_file_unittest.cc
+++ b/webrtc/modules/media_file/media_file_unittest.cc
@@ -12,7 +12,6 @@
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 class MediaFileTest : public testing::Test {
  protected:
@@ -28,8 +27,14 @@ class MediaFileTest : public testing::Test {
   webrtc::MediaFile* media_file_;
 };
 
-TEST_F(MediaFileTest, DISABLED_ON_IOS(
-       DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  DISABLED_StartPlayingAudioFileWithoutError
+#else
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  StartPlayingAudioFileWithoutError
+#endif
+TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
   // TODO(leozwang): Use hard coded filename here, we want to
   // loop through all audio files in future
   const std::string audio_file = webrtc::test::ProjectRootPath() +
@@ -47,7 +52,12 @@ TEST_F(MediaFileTest, DISABLED_ON_IOS(
   ASSERT_EQ(0, media_file_->StopPlaying());
 }
 
-TEST_F(MediaFileTest, DISABLED_ON_IOS(WriteWavFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteWavFile DISABLED_WriteWavFile
+#else
+#define MAYBE_WriteWavFile WriteWavFile
+#endif
+TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
   // Write file.
   static const size_t kHeaderSize = 44;
   static const size_t kPayloadSize = 320;
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
index f95067b074..50909ebd01 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
@@ -24,7 +24,6 @@
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
 #include "webrtc/test/field_trial.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 namespace testing {
@@ -318,8 +317,13 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
   EXPECT_EQ(kBwOverusing, overuse_detector_->State());
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance30Kbit3fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance30Kbit3fps \
+  DISABLED_LowGaussianVariance30Kbit3fps
+#else
+#define MAYBE_LowGaussianVariance30Kbit3fps LowGaussianVariance30Kbit3fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance30Kbit3fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 333;
@@ -375,8 +379,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
   EXPECT_EQ(4, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit5fps \
+  DISABLED_LowGaussianVariance100Kbit5fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit5fps LowGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -390,8 +399,13 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit5fps \
+  DISABLED_HighGaussianVariance100Kbit5fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit5fps HighGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -405,8 +419,13 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit10fps \
+  DISABLED_LowGaussianVariance100Kbit10fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit10fps LowGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -420,8 +439,13 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit10fps \
+  DISABLED_HighGaussianVariance100Kbit10fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit10fps HighGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -435,8 +459,13 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance300Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance300Kbit30fps \
+  DISABLED_LowGaussianVariance300Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance300Kbit30fps LowGaussianVariance300Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance300Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 33;
@@ -492,8 +521,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
   EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance1000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance1000Kbit30fps \
+  DISABLED_LowGaussianVariance1000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance1000Kbit30fps LowGaussianVariance1000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance1000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 3;
   int frame_duration_ms = 33;
@@ -549,8 +583,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
   EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance2000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance2000Kbit30fps \
+  DISABLED_LowGaussianVariance2000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance2000Kbit30fps LowGaussianVariance2000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance2000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 6;
   int frame_duration_ms = 33;
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
index 754e1242d3..58471e5e8d 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -20,7 +20,6 @@
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 DEFINE_bool(file_player_output, false, "Generate reference files.");
 
@@ -82,7 +81,12 @@ class FilePlayerTest : public ::testing::Test {
   FILE* output_file_;
 };
 
-TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
+#else
+#define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
@@ -93,7 +97,12 @@ TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
   PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
 }
 
-TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcm16File)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
+#else
+#define MAYBE_PlayWavPcm16File PlayWavPcm16File
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 1bd0684d80..7623131226 100644
--- a/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -23,7 +23,6 @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/video_frame.h"
 
 using rtc::scoped_ptr;
@@ -479,7 +478,12 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
 
 // Test frame rate and no picture alarm.
 // Flaky on Win32, see webrtc:3270.
-TEST_F(VideoCaptureExternalTest, DISABLED_ON_WIN(FrameRate)) {
+#if defined(WEBRTC_WIN)
+#define MAYBE_FrameRate DISABLED_FrameRate
+#else
+#define MAYBE_FrameRate FrameRate
+#endif
+TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
   int64_t testTime = 3;
   TickTime startTime = TickTime::Now();
 
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index d4e1e6e3a0..20715df074 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -22,7 +22,6 @@
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 #include "webrtc/typedefs.h"
@@ -814,8 +813,13 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
 // One key frame (first frame only) in sequence.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeBitRateVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeBitRateVP8 \
+  DISABLED_ProcessNoLossChangeBitRateVP8
+#else
+#define MAYBE_ProcessNoLossChangeBitRateVP8 ProcessNoLossChangeBitRateVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossChangeBitRateVP8) {
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
   SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
@@ -846,8 +850,15 @@ TEST_F(VideoProcessorIntegrationTest,
 // for the rate control metrics can be lower. One key frame (first frame only).
 // Note: quality after update should be higher but we currently compute quality
 // metrics averaged over whole sequence run.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  ProcessNoLossChangeFrameRateFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeFrameRateFrameDropVP8)) {
+       MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -874,8 +885,15 @@ TEST_F(VideoProcessorIntegrationTest,
 
 // Run with no packet loss, at low bitrate. During this time we should've
 // resized once. Expect 2 key frames generated (first and one for resize).
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  DISABLED_ProcessNoLossSpatialResizeFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  ProcessNoLossSpatialResizeFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossSpatialResizeFrameDropVP8)) {
+       MAYBE_ProcessNoLossSpatialResizeFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -901,8 +919,13 @@ TEST_F(VideoProcessorIntegrationTest,
 // encoding rate mismatch are applied to each layer.
 // No dropped frames in this test, and internal spatial resizer is off.
 // One key frame (first frame only) in sequence, so no spatial resizing.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossTemporalLayersVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossTemporalLayersVP8 \
+  DISABLED_ProcessNoLossTemporalLayersVP8
+#else
+#define MAYBE_ProcessNoLossTemporalLayersVP8 ProcessNoLossTemporalLayersVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossTemporalLayersVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
diff --git a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 7650a250ce..c3d77da063 100644
--- a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -17,7 +17,6 @@
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -221,7 +220,12 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_inst_, 1));
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AlignedStrideEncodeDecode DISABLED_AlignedStrideEncodeDecode
+#else
+#define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
+#endif
+TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
@@ -237,7 +241,12 @@ TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
   EXPECT_EQ(kTestNtpTimeMs, decoded_frame_.ntp_time_ms());
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(DecodeWithACompleteKeyFrame)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecodeWithACompleteKeyFrame DISABLED_DecodeWithACompleteKeyFrame
+#else
+#define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
+#endif
+TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
diff --git a/webrtc/modules/video_coding/video_sender_unittest.cc b/webrtc/modules/video_coding/video_sender_unittest.cc
index 2daa9d7b2d..9a438ff2b7 100644
--- a/webrtc/modules/video_coding/video_sender_unittest.cc
+++ b/webrtc/modules/video_coding/video_sender_unittest.cc
@@ -24,7 +24,6 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::_;
 using ::testing::AllOf;
@@ -424,8 +423,12 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
   int available_bitrate_kbps_;
 };
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(FixedTemporalLayersStrategy))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_FixedTemporalLayersStrategy DISABLED_FixedTemporalLayersStrategy
+#else
+#define MAYBE_FixedTemporalLayersStrategy FixedTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_FixedTemporalLayersStrategy) {
   const int low_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][0];
   const int mid_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][1];
   const int high_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][2];
@@ -439,8 +442,13 @@ TEST_F(TestVideoSenderWithVp8,
   }
 }
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(RealTimeTemporalLayersStrategy))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_RealTimeTemporalLayersStrategy \
+  DISABLED_RealTimeTemporalLayersStrategy
+#else
+#define MAYBE_RealTimeTemporalLayersStrategy RealTimeTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_RealTimeTemporalLayersStrategy) {
   Config extra_options;
   extra_options.Set<TemporalLayers::Factory>(
       new RealTimeTemporalLayersFactory());
diff --git a/webrtc/modules/video_processing/test/brightness_detection_test.cc b/webrtc/modules/video_processing/test/brightness_detection_test.cc
index 041a6e090f..669bb183e5 100644
--- a/webrtc/modules/video_processing/test/brightness_detection_test.cc
+++ b/webrtc/modules/video_processing/test/brightness_detection_test.cc
@@ -11,11 +11,15 @@
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/modules/video_processing/test/video_processing_unittest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(BrightnessDetection)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_BrightnessDetection DISABLED_BrightnessDetection
+#else
+#define MAYBE_BrightnessDetection BrightnessDetection
+#endif
+TEST_F(VideoProcessingTest, MAYBE_BrightnessDetection) {
   uint32_t frameNum = 0;
   int32_t brightnessWarning = 0;
   uint32_t warningCount = 0;
diff --git a/webrtc/modules/video_processing/test/content_metrics_test.cc b/webrtc/modules/video_processing/test/content_metrics_test.cc
index a676982584..782f9cff59 100644
--- a/webrtc/modules/video_processing/test/content_metrics_test.cc
+++ b/webrtc/modules/video_processing/test/content_metrics_test.cc
@@ -12,11 +12,14 @@
 #include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/modules/video_processing/content_analysis.h"
 #include "webrtc/modules/video_processing/test/video_processing_unittest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(ContentAnalysis)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_ContentAnalysis) {
+#else
+TEST_F(VideoProcessingTest, ContentAnalysis) {
+#endif
   VPMContentAnalysis ca__c(false);
   VPMContentAnalysis ca__sse(true);
   VideoContentMetrics* _cM_c;
diff --git a/webrtc/modules/video_processing/test/deflickering_test.cc b/webrtc/modules/video_processing/test/deflickering_test.cc
index 5bd8d4e003..5410015b06 100644
--- a/webrtc/modules/video_processing/test/deflickering_test.cc
+++ b/webrtc/modules/video_processing/test/deflickering_test.cc
@@ -16,11 +16,14 @@
 #include "webrtc/modules/video_processing/test/video_processing_unittest.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(Deflickering)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Deflickering) {
+#else
+TEST_F(VideoProcessingTest, Deflickering) {
+#endif
   enum { NumRuns = 30 };
   uint32_t frameNum = 0;
   const uint32_t frame_rate = 15;
diff --git a/webrtc/modules/video_processing/test/video_processing_unittest.cc b/webrtc/modules/video_processing/test/video_processing_unittest.cc
index bf4d37619c..2fd8fb6673 100644
--- a/webrtc/modules/video_processing/test/video_processing_unittest.cc
+++ b/webrtc/modules/video_processing/test/video_processing_unittest.cc
@@ -17,7 +17,6 @@
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -92,7 +91,11 @@ void VideoProcessingTest::TearDown() {
   vp_ = NULL;
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleNullBuffer)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleNullBuffer) {
+#else
+TEST_F(VideoProcessingTest, HandleNullBuffer) {
+#endif
   // TODO(mikhal/stefan): Do we need this one?
   VideoProcessing::FrameStats stats;
   // Video frame with unallocated buffer.
@@ -106,7 +109,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleNullBuffer)) {
   EXPECT_EQ(-3, vp_->BrightnessDetection(videoFrame, stats));
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleBadStats)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleBadStats) {
+#else
+TEST_F(VideoProcessingTest, HandleBadStats) {
+#endif
   VideoProcessing::FrameStats stats;
   vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
@@ -120,7 +127,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleBadStats)) {
   EXPECT_EQ(-3, vp_->BrightnessDetection(video_frame_, stats));
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(IdenticalResultsAfterReset)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_IdenticalResultsAfterReset) {
+#else
+TEST_F(VideoProcessingTest, IdenticalResultsAfterReset) {
+#endif
   VideoFrame video_frame2;
   VideoProcessing::FrameStats stats;
   // Only testing non-static functions here.
@@ -153,7 +164,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(IdenticalResultsAfterReset)) {
   EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(FrameStats)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_FrameStats) {
+#else
+TEST_F(VideoProcessingTest, FrameStats) {
+#endif
   VideoProcessing::FrameStats stats;
   vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
@@ -178,7 +193,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(FrameStats)) {
   EXPECT_FALSE(vp_->ValidFrameStats(stats));
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(PreprocessorLogic)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_PreprocessorLogic) {
+#else
+TEST_F(VideoProcessingTest, PreprocessorLogic) {
+#endif
   // Disable temporal sampling (frame dropping).
   vp_->EnableTemporalDecimation(false);
   int resolution = 100;
@@ -197,7 +216,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(PreprocessorLogic)) {
   EXPECT_TRUE(vp_->PreprocessFrame(video_frame_) != nullptr);
 }
 
-TEST_F(VideoProcessingTest, DISABLED_ON_IOS(Resampler)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Resampler) {
+#else
+TEST_F(VideoProcessingTest, Resampler) {
+#endif
   enum { NumRuns = 1 };
 
   int64_t min_runtime = 0;