Remove DISABLED_ON_ macros.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
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 8de6c9100d3cd243257555aa1ef5fc04a8d87c40..6a2e44102709b5e93fe5f0cc5023893f1f1a89b9 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 +237,11 @@ 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)
+TEST_F(AudioCodingModuleTestOldApi, DISABLED_InitializedToZero) {
+#else
+TEST_F(AudioCodingModuleTestOldApi, InitializedToZero) {
+#endif
   RegisterCodec();
   AudioDecodingCallStats stats;
   acm_->GetDecodingCallStatistics(&stats);
@@ -253,7 +256,11 @@ 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)
+TEST_F(AudioCodingModuleTestOldApi, DISABLED_NetEqCalls) {
+#else
+TEST_F(AudioCodingModuleTestOldApi, NetEqCalls) {
+#endif
   RegisterCodec();
   AudioDecodingCallStats stats;
   const int kNumNormalCalls = 10;
@@ -320,15 +327,9 @@ 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 +361,7 @@ 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 +584,11 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
-TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
+#if defined(WEBRTC_IOS)
+TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_DoTest) {
+#else
+TEST_F(AudioCodingModuleMtTestOldApi, DoTest) {
+#endif
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
@@ -686,9 +692,15 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#if defined(WEBRTC_IOS)
+TEST_F(AcmIsacMtTestOldApi, DISABLED_DoTest) {
+#else
+TEST_F(AcmIsacMtTestOldApi, DoTest) {
+#endif
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
 
 class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
  protected:
@@ -838,9 +850,15 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#if defined(WEBRTC_IOS)
+TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_DoTest) {
+#else
+TEST_F(AcmReRegisterIsacMtTestOldApi, DoTest) {
+#endif
   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.
@@ -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 +1242,14 @@ 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_CODEC_ISAC)
+#if defined(WEBRTC_ANDROID)
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_IsacSwb30ms) {
 #else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
+TEST_F(AcmSenderBitExactnessOldApi, IsacSwb30ms) {
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "5683b58da0fbf2063c7adc2e6bfb3fb8",
@@ -1243,6 +1261,7 @@ 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 +1343,12 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
       test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-#ifdef WEBRTC_CODEC_ILBC
-#define IF_ILBC(x) x
+#if defined(WEBRTC_CODEC_ILBC)
+#if defined(WEBRTC_ANDROID)
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_Ilbc_30ms) {
 #else
-#define IF_ILBC(x) DISABLED_##x
+TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) {
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7b6ec10910debd9af08011d3ed5249f7",
@@ -1342,14 +1360,14 @@ 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_CODEC_G722)
+#if defined(WEBRTC_ANDROID)
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_G722_20ms) {
 #else
-#define IF_G722(x) DISABLED_##x
+TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) {
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7d759436f2533582950d148b5161a36c",
@@ -1362,8 +1380,11 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
       50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
+#if defined(WEBRTC_ANDROID)
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_G722_stereo_20ms) {
+#else
+TEST_F(AcmSenderBitExactnessOldApi, G722_stereo_20ms) {
+#endif
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7190ee718ab3d80eca181e5f7140c210",
@@ -1375,6 +1396,7 @@ 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 +1512,11 @@ 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)
+TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_100kbps) {
+#else
+TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_100kbps) {
+#endif
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
   Run(100000, 100888);
 }