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}
diff --git a/webrtc/base/autodetectproxy_unittest.cc b/webrtc/base/autodetectproxy_unittest.cc
index bc57304..2ae7a6a 100644
--- a/webrtc/base/autodetectproxy_unittest.cc
+++ b/webrtc/base/autodetectproxy_unittest.cc
@@ -12,7 +12,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/httpcommon.h"
#include "webrtc/base/httpcommon-inl.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/criticalsection_unittest.cc b/webrtc/base/criticalsection_unittest.cc
index 85ef20d..d6990c0 100644
--- a/webrtc/base/criticalsection_unittest.cc
+++ b/webrtc/base/criticalsection_unittest.cc
@@ -17,7 +17,6 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scopedptrcollection.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/logging_unittest.cc b/webrtc/base/logging_unittest.cc
index 3719cde..6047361 100644
--- a/webrtc/base/logging_unittest.cc
+++ b/webrtc/base/logging_unittest.cc
@@ -14,7 +14,6 @@
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/messagequeue_unittest.cc b/webrtc/base/messagequeue_unittest.cc
index 871542d..78024e0 100644
--- a/webrtc/base/messagequeue_unittest.cc
+++ b/webrtc/base/messagequeue_unittest.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/nat_unittest.cc b/webrtc/base/nat_unittest.cc
index f8895eb..8be1be9 100644
--- a/webrtc/base/nat_unittest.cc
+++ b/webrtc/base/nat_unittest.cc
@@ -21,7 +21,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/nullsocketserver_unittest.cc b/webrtc/base/nullsocketserver_unittest.cc
index 2aa38b4..4f22c38 100644
--- a/webrtc/base/nullsocketserver_unittest.cc
+++ b/webrtc/base/nullsocketserver_unittest.cc
@@ -10,7 +10,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/physicalsocketserver_unittest.cc b/webrtc/base/physicalsocketserver_unittest.cc
index 5ff4859..a2fde80 100644
--- a/webrtc/base/physicalsocketserver_unittest.cc
+++ b/webrtc/base/physicalsocketserver_unittest.cc
@@ -18,7 +18,6 @@
#include "webrtc/base/socket_unittest.h"
#include "webrtc/base/testutils.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/proxy_unittest.cc b/webrtc/base/proxy_unittest.cc
index 03dc154..d8a523f 100644
--- a/webrtc/base/proxy_unittest.cc
+++ b/webrtc/base/proxy_unittest.cc
@@ -17,7 +17,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/testechoserver.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using rtc::Socket;
using rtc::Thread;
diff --git a/webrtc/base/sharedexclusivelock_unittest.cc b/webrtc/base/sharedexclusivelock_unittest.cc
index 2857e00..9b64ed7 100644
--- a/webrtc/base/sharedexclusivelock_unittest.cc
+++ b/webrtc/base/sharedexclusivelock_unittest.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/sharedexclusivelock.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/signalthread_unittest.cc b/webrtc/base/signalthread_unittest.cc
index fe6c602..a583aef 100644
--- a/webrtc/base/signalthread_unittest.cc
+++ b/webrtc/base/signalthread_unittest.cc
@@ -11,7 +11,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/signalthread.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc
index 72f02e8..1ed06c3 100644
--- a/webrtc/base/sslstreamadapter_unittest.cc
+++ b/webrtc/base/sslstreamadapter_unittest.cc
@@ -22,7 +22,6 @@
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using ::testing::WithParamInterface;
using ::testing::Values;
diff --git a/webrtc/base/stream_unittest.cc b/webrtc/base/stream_unittest.cc
index 4172a97..8cfd052 100644
--- a/webrtc/base/stream_unittest.cc
+++ b/webrtc/base/stream_unittest.cc
@@ -12,7 +12,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/task_unittest.cc b/webrtc/base/task_unittest.cc
index 3508219..7492436 100644
--- a/webrtc/base/task_unittest.cc
+++ b/webrtc/base/task_unittest.cc
@@ -28,7 +28,6 @@
#include "webrtc/base/taskrunner.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/testclient_unittest.cc b/webrtc/base/testclient_unittest.cc
index 17bf4e6..bdd06b3 100644
--- a/webrtc/base/testclient_unittest.cc
+++ b/webrtc/base/testclient_unittest.cc
@@ -14,7 +14,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/testechoserver.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/thread_checker_unittest.cc b/webrtc/base/thread_checker_unittest.cc
index bcffb52..3381900 100644
--- a/webrtc/base/thread_checker_unittest.cc
+++ b/webrtc/base/thread_checker_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
// Duplicated from base/threading/thread_checker.h so that we can be
// good citizens there and undef the macro.
diff --git a/webrtc/base/thread_unittest.cc b/webrtc/base/thread_unittest.cc
index a826298..7ed4326 100644
--- a/webrtc/base/thread_unittest.cc
+++ b/webrtc/base/thread_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/socketaddress.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#if defined(WEBRTC_WIN)
#include <comdef.h> // NOLINT
diff --git a/webrtc/base/virtualsocket_unittest.cc b/webrtc/base/virtualsocket_unittest.cc
index 68ad23b..2cd2b5e 100644
--- a/webrtc/base/virtualsocket_unittest.cc
+++ b/webrtc/base/virtualsocket_unittest.cc
@@ -22,7 +22,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index dd2a2f8..f590f66 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -27,7 +27,6 @@
#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"
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
diff --git a/webrtc/common_audio/signal_processing/real_fft_unittest.cc b/webrtc/common_audio/signal_processing/real_fft_unittest.cc
index 9bd35cd..fa98836 100644
--- a/webrtc/common_audio/signal_processing/real_fft_unittest.cc
+++ b/webrtc/common_audio/signal_processing/real_fft_unittest.cc
@@ -10,7 +10,6 @@
#include "webrtc/common_audio/signal_processing/include/real_fft.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/webrtc/common_video/libyuv/scaler_unittest.cc b/webrtc/common_video/libyuv/scaler_unittest.cc
index 526e62c..6d02638 100644
--- a/webrtc/common_video/libyuv/scaler_unittest.cc
+++ b/webrtc/common_video/libyuv/scaler_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/common_video/libyuv/include/scaler.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@@ -115,7 +114,12 @@
}
// TODO(mikhal): Converge the test into one function that accepts the method.
-TEST_F(TestScaler, DISABLED_ON_ANDROID(PointScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PointScaleTest DISABLED_PointScaleTest
+#else
+#define MAYBE_PointScaleTest PointScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_PointScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScalePoint;
@@ -182,7 +186,12 @@
ASSERT_EQ(0, fclose(source_file2));
}
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BiLinearScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BilinearScaleTest DISABLED_BiLinearScaleTest
+#else
+#define MAYBE_BilinearScaleTest BiLinearScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BiLinearScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScaleBilinear;
@@ -234,7 +243,12 @@
400, 300);
}
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BoxScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BoxScaleTest DISABLED_BoxScaleTest
+#else
+#define MAYBE_BoxScaleTest BoxScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BoxScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScaleBox;
diff --git a/webrtc/libjingle/xmllite/xmlelement_unittest.cc b/webrtc/libjingle/xmllite/xmlelement_unittest.cc
index 257899a..df8faed 100644
--- a/webrtc/libjingle/xmllite/xmlelement_unittest.cc
+++ b/webrtc/libjingle/xmllite/xmlelement_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using buzz::QName;
using buzz::XmlAttr;
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 a7dd3d4..24ecc69 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 @@
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(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 @@
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 @@
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 @@
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(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 @@
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 @@
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 8de6c91..ef48a48 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 @@
// 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 @@
// 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 @@
}
#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 @@
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 @@
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 @@
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 @@
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 @@
#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 @@
std::vector<ExternalDecoder>());
}
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(16kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
Run(16000, PlatformChecksum("a909560b5ca49fa472b17b7b277195e9",
"f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
"8cffa6abcb3e18e33b9d857666dff66a",
@@ -940,7 +958,7 @@
std::vector<ExternalDecoder>());
}
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(32kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
Run(32000, PlatformChecksum("441aab4b347fb3db4e9244337aca8d8e",
"306e0d990ee6e92de3fbecc0123ece37",
"3e126fe894720c3f85edadcc91964ba5",
@@ -948,7 +966,7 @@
std::vector<ExternalDecoder>());
}
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(48kHzOutput)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
"aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
@@ -956,8 +974,7 @@
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 @@
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 @@
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 @@
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 @@
"9e0a0ab743ad987b55b8e14802769c56"),
16, test::AcmReceiveTestOldApi::kMonoOutput);
}
-
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
-#else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
-TEST_F(AcmSenderBitExactnessOldApi,
- DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms
+#else
+#define MAYBE_IsacSwb30ms IsacSwb30ms
+#endif
+#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 @@
"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::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 @@
"android_arm64_payload"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
-
-#ifdef WEBRTC_CODEC_G722
-#define IF_G722(x) x
-#else
-#define IF_G722(x) DISABLED_##x
#endif
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_20ms DISABLED_G722_20ms
+#else
+#define MAYBE_G722_20ms G722_20ms
+#endif
+#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 @@
"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 @@
"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 @@
// 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 f41a17a..85aaef1 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 e02e92d..1ddc7f2 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 @@
}
};
-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 @@
double drift_factor;
};
-TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
RunTest(100);
}
@@ -314,7 +318,7 @@
}
};
-TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
RunTest(100);
}
@@ -342,7 +346,7 @@
int frame_index_;
};
-TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
RunTest(1000);
}
@@ -361,7 +365,10 @@
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 b3d6d8c..a6b9388 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 @@
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 @@
EXPECT_EQ(1, out_data_[i]);
}
}
+#endif
TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
NetEqOutputType type;
@@ -1171,7 +1171,8 @@
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 @@
--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 3ff3dd8..a27f0bc 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 @@
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 @@
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();
}
-
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
-#else
-#define IF_ISAC(x) DISABLED_##x
#endif
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
+#else
+TEST(AudioCodingModuleTest, TestIsac) {
+#endif
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 @@
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 d7c0411..97471bb 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 @@
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 32f5a3e..567118d 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 @@
}
#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 b2e1198..7f152bf 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 @@
#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 eff791d..d4bb8aa 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 7a035d2..7fb7d2d 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 6a70a3f..b600775 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 @@
// 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 7c99f4f..e4e9048 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 @@
// 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 5462d05..1d1dcc7 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 @@
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 c12ea57..6541a8f 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 @@
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 @@
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 f95067b..50909eb 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 754e124..58471e5 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 @@
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 @@
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 1bd0684..7623131 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 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 d4e1e6e..20715df 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 7650a25..c3d77da 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 @@
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 @@
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 2daa9d7..9a438ff 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 @@
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,
- 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 041a6e0..669bb18 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 a676982..782f9cf 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 5bd8d4e..5410015 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 bf4d376..2fd8fb6 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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;
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 2fbd6df..b4dba1e 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -58,7 +58,6 @@
"testsupport/frame_reader.h",
"testsupport/frame_writer.cc",
"testsupport/frame_writer.h",
- "testsupport/gtest_disable.h",
"testsupport/mock/mock_frame_reader.h",
"testsupport/mock/mock_frame_writer.h",
"testsupport/packet_reader.cc",
diff --git a/webrtc/test/test.gyp b/webrtc/test/test.gyp
index 82c8c25..8f04ce8 100644
--- a/webrtc/test/test.gyp
+++ b/webrtc/test/test.gyp
@@ -142,7 +142,6 @@
'testsupport/frame_reader.h',
'testsupport/frame_writer.cc',
'testsupport/frame_writer.h',
- 'testsupport/gtest_disable.h',
'testsupport/iosfileutils.mm',
'testsupport/mock/mock_frame_reader.h',
'testsupport/mock/mock_frame_writer.h',
diff --git a/webrtc/test/testsupport/fileutils_unittest.cc b/webrtc/test/testsupport/fileutils_unittest.cc
index dff7f22..e205db3 100644
--- a/webrtc/test/testsupport/fileutils_unittest.cc
+++ b/webrtc/test/testsupport/fileutils_unittest.cc
@@ -16,7 +16,6 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#ifdef WIN32
#define chdir _chdir
@@ -66,7 +65,14 @@
}
// Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutputPathFromUnchangedWorkingDir \
+ DISABLED_OutputPathFromUnchangedWorkingDir
+#else
+#define MAYBE_OutputPathFromUnchangedWorkingDir \
+ OutputPathFromUnchangedWorkingDir
+#endif
+TEST_F(FileUtilsTest, MAYBE_OutputPathFromUnchangedWorkingDir) {
std::string path = webrtc::test::OutputPath();
std::string expected_end = "out";
expected_end = kPathDelimiter + expected_end + kPathDelimiter;
@@ -75,7 +81,12 @@
// Tests with current working directory set to a directory higher up in the
// directory tree than the project root dir.
-TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromRootWorkingDir)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutputPathFromRootWorkingDir DISABLED_OutputPathFromRootWorkingDir
+#else
+#define MAYBE_OutputPathFromRootWorkingDir OutputPathFromRootWorkingDir
+#endif
+TEST_F(FileUtilsTest, MAYBE_OutputPathFromRootWorkingDir) {
ASSERT_EQ(0, chdir(kPathDelimiter));
ASSERT_EQ("./", webrtc::test::OutputPath());
}
diff --git a/webrtc/test/testsupport/gtest_disable.h b/webrtc/test/testsupport/gtest_disable.h
deleted file mode 100644
index fdc56ac..0000000
--- a/webrtc/test/testsupport/gtest_disable.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
-#define TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
-
-// Helper macros for platform disables. These can be chained. Example use:
-// TEST_F(ViEStandardIntegrationTest,
-// DISABLED_ON_LINUX(RunsBaseTestWithoutErrors)) { // ...
-//
-// Or, you can disable a whole test class by wrapping all mentions of the test
-// class name inside one of these macros.
-//
-// The platform #defines we are looking at here are set by the build system.
-#ifdef WEBRTC_LINUX
-#define DISABLED_ON_LINUX(test) DISABLED_##test
-#else
-#define DISABLED_ON_LINUX(test) test
-#endif
-
-#ifdef WEBRTC_MAC
-#define DISABLED_ON_MAC(test) DISABLED_##test
-#else
-#define DISABLED_ON_MAC(test) test
-#endif
-
-#ifdef _WIN32
-#define DISABLED_ON_WIN(test) DISABLED_##test
-#else
-#define DISABLED_ON_WIN(test) test
-#endif
-
-// Using some extra magic here to be able to chain Android and iOS macros.
-// http://stackoverflow.com/questions/8231966/why-do-i-need-double-layer-of-indirection-for-macros
-#ifdef WEBRTC_ANDROID
-#define DISABLED_ON_ANDROID_HIDDEN(test) DISABLED_##test
-#define DISABLED_ON_ANDROID(test) DISABLED_ON_ANDROID_HIDDEN(test)
-#else
-#define DISABLED_ON_ANDROID_HIDDEN(test) test
-#define DISABLED_ON_ANDROID(test) DISABLED_ON_ANDROID_HIDDEN(test)
-#endif
-
-#ifdef WEBRTC_IOS
-#define DISABLED_ON_IOS_HIDDEN(test) DISABLED_##test
-#define DISABLED_ON_IOS(test) DISABLED_ON_IOS_HIDDEN(test)
-#else
-#define DISABLED_ON_IOS_HIDDEN(test) test
-#define DISABLED_ON_IOS(test) DISABLED_ON_IOS_HIDDEN(test)
-#endif
-
-#endif // TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 659af2f..f654dbb 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -41,7 +41,6 @@
#include "webrtc/test/rtcp_packet_parser.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video_encoder.h"
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
index 7a3fad8..51db985 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
+++ b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
@@ -16,7 +16,6 @@
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 14dca27..6efa55d 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -101,8 +101,7 @@
EXPECT_STREQ(RTCP_CNAME, char_buffer);
}
-// Flakily hangs on Linux. code.google.com/p/webrtc/issues/detail?id=2178.
-TEST_F(RtpRtcpTest, DISABLED_ON_LINUX(SSRCPropagatesCorrectly)) {
+TEST_F(RtpRtcpTest, SSRCPropagatesCorrectly) {
unsigned int local_ssrc = 1234;
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, local_ssrc));
diff --git a/webrtc/voice_engine/voe_codec_unittest.cc b/webrtc/voice_engine/voe_codec_unittest.cc
index 52aa537..f09e19e 100644
--- a/webrtc/voice_engine/voe_codec_unittest.cc
+++ b/webrtc/voice_engine/voe_codec_unittest.cc
@@ -13,7 +13,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
