diff --git a/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc b/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc
index 3b2364ca6..76f5a099e 100644
--- a/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc
@@ -17,7 +17,9 @@
 
 #include "gmock/gmock.h"
 #include "gtest/gtest.h"
+
 #include "webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -66,7 +68,7 @@ TEST(DecoderDatabase, GetRtpPayloadType) {
             db.GetRtpPayloadType(kDecoderISAC));  // iSAC is not registered.
 }
 
-TEST(DecoderDatabase, GetDecoder) {
+TEST(DecoderDatabase, DISABLED_ON_ANDROID(GetDecoder)) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
   EXPECT_EQ(DecoderDatabase::kOK,
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
index c0a0fd3ba..fec25e985 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
@@ -21,6 +21,7 @@
 #include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -201,7 +202,7 @@ class NetEqExternalDecoderTest : public ::testing::Test {
   scoped_ptr<test::InputAudioFile> input_file_;
 };
 
-TEST_F(NetEqExternalDecoderTest, RunTest) {
+TEST_F(NetEqExternalDecoderTest, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(100);  // Run 100 laps @ 10 ms each in the test loop.
 }
 
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
index 9c74e0391..d6c4150ec 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
@@ -20,6 +20,7 @@
 #include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -270,7 +271,7 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
   }
 };
 
-TEST_P(NetEqStereoTestNoJitter, RunTest) {
+TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(8);
 }
 
@@ -295,7 +296,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
   double drift_factor;
 };
 
-TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
+TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(100);
 }
 
@@ -308,7 +309,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
   }
 };
 
-TEST_P(NetEqStereoTestNegativeDrift, RunTest) {
+TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(100);
 }
 
@@ -336,7 +337,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestDelays, RunTest) {
+TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(1000);
 }
 
@@ -355,7 +356,7 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestLosses, RunTest) {
+TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
   RunTest(100);
 }
 
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index 607221256..1b3af036e 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -23,6 +23,7 @@
 #include "gtest/gtest.h"
 #include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -229,8 +230,10 @@ void NetEqDecodingTest::LoadDecoders() {
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
   // Load PCMa.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
+#ifndef WEBRTC_ANDROID
   // Load iLBC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
+#endif  // WEBRTC_ANDROID
   // Load iSAC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
   // Load iSAC SWB.
@@ -379,7 +382,7 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
 #define MAYBE_TestBitExactness TestBitExactness
 #endif
 
-TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) {
   const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
@@ -394,7 +397,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
   DecodeAndCompare(kInputRtpFile, kInputRefFile);
 }
 
-TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
   const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
@@ -412,7 +415,7 @@ TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
 }
 
 // TODO(hlundin): Re-enable test once the statistics interface is up and again.
-TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
   // Use fax mode to avoid time-scaling. This is to simplify the testing of
   // packet waiting times in the packet buffer.
   neteq_->SetPlayoutMode(kPlayoutFax);
@@ -487,7 +490,8 @@ TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
   EXPECT_EQ(100u, waiting_times.size());
 }
 
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
+TEST_F(NetEqDecodingTest,
+       DISABLED_ON_ANDROID(TestAverageInterArrivalTimeNegative)) {
   const int kNumFrames = 3000;  // Needed for convergence.
   int frame_index = 0;
   const int kSamples = 10 * 16;
@@ -518,7 +522,8 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
   EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
 }
 
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
+TEST_F(NetEqDecodingTest,
+       DISABLED_ON_ANDROID(TestAverageInterArrivalTimePositive)) {
   const int kNumFrames = 5000;  // Needed for convergence.
   int frame_index = 0;
   const int kSamples = 10 * 16;
@@ -549,7 +554,7 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
   EXPECT_EQ(110946, network_stats.clockdrift_ppm);
 }
 
-TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) {
   uint16_t seq_no = 0;
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 30;
@@ -642,7 +647,7 @@ TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
   EXPECT_GE(delay_after, delay_before - 20 * 16);
 }
 
-TEST_F(NetEqDecodingTest, UnknownPayloadType) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(UnknownPayloadType)) {
   const int kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
@@ -653,7 +658,7 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-TEST_F(NetEqDecodingTest, DecoderError) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
   const int kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
@@ -692,7 +697,7 @@ TEST_F(NetEqDecodingTest, DecoderError) {
   }
 }
 
-TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) {
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index ad5311710..f93b7b8bf 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -21,6 +21,7 @@
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/system_wrappers/interface/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"
@@ -1402,7 +1403,7 @@ TEST_F(ApmTest, DebugDump) {
 // TODO(andrew): Make this test more robust such that it can be run on multiple
 // platforms. It currently requires bit-exactness.
 #ifdef WEBRTC_AUDIOPROC_BIT_EXACT
-TEST_F(ApmTest, Process) {
+TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) {
   GOOGLE_PROTOBUF_VERIFY_VERSION;
   webrtc::audioproc::OutputData ref_data;
 
diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/source/media_file_unittest.cc
index 8abe7fc21..9f3f0ccf9 100644
--- a/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/webrtc/modules/media_file/source/media_file_unittest.cc
@@ -12,6 +12,7 @@
 #include "webrtc/modules/media_file/interface/media_file.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 
 class MediaFileTest : public testing::Test {
  protected:
@@ -27,7 +28,7 @@ class MediaFileTest : public testing::Test {
   webrtc::MediaFile* media_file_;
 };
 
-TEST_F(MediaFileTest, StartPlayingAudioFileWithoutError) {
+TEST_F(MediaFileTest, DISABLED_ON_ANDROID(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() +
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
index be89d6da6..278a0577c 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -215,4 +215,3 @@ TEST(TemporalLayersTest, KeyFrame) {
   EXPECT_EQ(true, vp8_info.layerSync);
 }
 }  // namespace webrtc
-
diff --git a/webrtc/modules/video_processing/main/test/unit_test/denoising_test.cc b/webrtc/modules/video_processing/main/test/unit_test/denoising_test.cc
index 71d133bdf..8c4791777 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/denoising_test.cc
+++ b/webrtc/modules/video_processing/main/test/unit_test/denoising_test.cc
@@ -16,10 +16,11 @@
 #include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingModuleTest, Denoising)
+TEST_F(VideoProcessingModuleTest, DISABLED_ON_ANDROID(Denoising))
 {
     enum { NumRuns = 10 };
     uint32_t frameNum = 0;
diff --git a/webrtc/test/testsupport/gtest_disable.h b/webrtc/test/testsupport/gtest_disable.h
index b4a661f4b..257d83612 100644
--- a/webrtc/test/testsupport/gtest_disable.h
+++ b/webrtc/test/testsupport/gtest_disable.h
@@ -36,4 +36,10 @@
 #define DISABLED_ON_WIN(test) test
 #endif
 
+#ifdef WEBRTC_ANDROID
+#define DISABLED_ON_ANDROID(test) DISABLED_##test
+#else
+#define DISABLED_ON_ANDROID(test) test
+#endif
+
 #endif  // TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_