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Remove DISABLED_ON_ macros.

Browse Source 
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}
This commit is contained in:
Peter Boström 2016-01-04 22:44:05 +01:00
parent 13f61dfea5
commit e2976c87f7
54 changed files with 524 additions and 271 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/base/autodetectproxy_unittest.cc
View File

@ -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 {

1
webrtc/base/criticalsection_unittest.cc
View File

@ -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 {

1
webrtc/base/logging_unittest.cc
View File

@ -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 {

1
webrtc/base/messagequeue_unittest.cc
View File

@ -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;

1
webrtc/base/nat_unittest.cc
View File

@ -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;

1
webrtc/base/nullsocketserver_unittest.cc
View File

@ -10,7 +10,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/nullsocketserver.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {

1
webrtc/base/physicalsocketserver_unittest.cc
View File

@ -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 {

1
webrtc/base/proxy_unittest.cc
View File

@ -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;

1
webrtc/base/sharedexclusivelock_unittest.cc
View File

@ -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 {

1
webrtc/base/signalthread_unittest.cc
View File

@ -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;

1
webrtc/base/sslstreamadapter_unittest.cc
View File

@ -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;

1
webrtc/base/stream_unittest.cc
View File

@ -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 {

1
webrtc/base/task_unittest.cc
View File

@ -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 {

1
webrtc/base/testclient_unittest.cc
View File

@ -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;

1
webrtc/base/thread_checker_unittest.cc
View File

@ -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.

1
webrtc/base/thread_unittest.cc
View File

@ -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

1
webrtc/base/virtualsocket_unittest.cc
View File

@ -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;

1
webrtc/call/rtc_event_log_unittest.cc
View File

@ -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

1
webrtc/common_audio/signal_processing/real_fft_unittest.cc
View File

@ -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"

22
webrtc/common_video/libyuv/scaler_unittest.cc
View File

@ -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 @@ TEST_F(TestScaler, ScaleSendingBufferTooSmall) {
}
// 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 @@ TEST_F(TestScaler, DISABLED_ON_ANDROID(PointScaleTest)) {
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 @@ TEST_F(TestScaler, DISABLED_ON_ANDROID(BiLinearScaleTest)) {
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;

1
webrtc/libjingle/xmllite/xmlelement_unittest.cc
View File

@ -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;

55
webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -20,7 +20,6 @@
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -164,7 +163,12 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
FrameType last_frame_type_;
};
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
#else
#define MAYBE_AddCodecGetCodec AddCodecGetCodec
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
// Add codec.
for (size_t n = 0; n < codecs_.size(); ++n) {
if (n & 0x1) // Just add codecs with odd index.
@ -188,7 +192,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
}
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
#else
#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
const CodecIdInst codec1(RentACodec::CodecId::kPCMA);
CodecInst codec2 = codec1.inst;
++codec2.pltype;
@ -209,7 +218,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
EXPECT_EQ(true, CodecsEqual(codec2, test_codec));
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
#else
#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
#endif
TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
const CodecIdInst codec1(RentACodec::CodecId::kPCMU);
CodecIdInst codec2(RentACodec::CodecId::kPCMA);
codec2.inst.pltype = codec1.inst.pltype;
@ -229,7 +243,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
EXPECT_EQ(true, CodecsEqual(codec2.inst, test_codec));
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
#else
#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
const CodecIdInst codec(RentACodec::CodecId::kPCMA);
const int payload_type = codec.inst.pltype;
EXPECT_EQ(
@ -247,7 +266,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &ci));
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_SampleRate DISABLED_SampleRate
#else
#define MAYBE_SampleRate SampleRate
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
const RentACodec::CodecId kCodecId[] = {RentACodec::CodecId::kISAC,
RentACodec::CodecId::kISACSWB};
AddSetOfCodecs(kCodecId);
@ -265,7 +289,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
}
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
#else
#define MAYBE_PostdecodingVad PostdecodingVad
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
receiver_->EnableVad();
EXPECT_TRUE(receiver_->vad_enabled());
const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
@ -293,14 +322,13 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
}
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#if defined(WEBRTC_ANDROID)
#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#define MAYBE_LastAudioCodec LastAudioCodec
#endif
TEST_F(AcmReceiverTestOldApi,
DISABLED_ON_ANDROID(IF_ISAC_FLOAT(LastAudioCodec))) {
#if defined(WEBRTC_CODEC_ISAC)
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
const RentACodec::CodecId kCodecId[] = {
RentACodec::CodecId::kISAC, RentACodec::CodecId::kPCMA,
RentACodec::CodecId::kISACSWB, RentACodec::CodecId::kPCM16Bswb32kHz};
@ -363,6 +391,7 @@ TEST_F(AcmReceiverTestOldApi,
EXPECT_TRUE(CodecsEqual(c.inst, codec));
}
}
#endif
} // namespace acm2

127
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -41,7 +41,6 @@
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
using ::testing::AtLeast;
using ::testing::Invoke;
@ -238,7 +237,12 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_InitializedToZero DISABLED_InitializedToZero
#else
#define MAYBE_InitializedToZero InitializedToZero
#endif
TEST_F(AudioCodingModuleTestOldApi, MAYBE_InitializedToZero) {
RegisterCodec();
AudioDecodingCallStats stats;
acm_->GetDecodingCallStatistics(&stats);
@ -253,7 +257,12 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_NetEqCalls DISABLED_NetEqCalls
#else
#define MAYBE_NetEqCalls NetEqCalls
#endif
TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) {
RegisterCodec();
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
@ -320,15 +329,9 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#else
#define IF_ISAC(x) DISABLED_##x
#endif
// Verifies that the RTP timestamp series is not reset when the codec is
// changed.
TEST_F(AudioCodingModuleTestOldApi,
IF_ISAC(TimestampSeriesContinuesWhenCodecChanges)) {
TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
RegisterCodec(); // This registers the default codec.
uint32_t expected_ts = input_frame_.timestamp_;
int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
@ -360,6 +363,7 @@ TEST_F(AudioCodingModuleTestOldApi,
expected_ts += codec_.pacsize;
}
}
#endif
// Introduce this class to set different expectations on the number of encoded
// bytes. This class expects all encoded packets to be 9 bytes (matching one
@ -582,7 +586,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
@ -686,9 +695,16 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
test::AudioLoop audio_loop_;
};
TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
#endif
class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
protected:
@ -838,9 +854,16 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
test::AudioLoop audio_loop_;
};
TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
#endif
// Disabling all of these tests on iOS until file support has been added.
// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
@ -919,12 +942,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
#define IF_ALL_CODECS(x) x
#else
#define IF_ALL_CODECS(x) DISABLED_##x
#endif
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(8kHzOutput)) {
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
Run(8000, PlatformChecksum("908002dc01fc4eb1d2be24eb1d3f354b",
"dcee98c623b147ebe1b40dd30efa896e",
"adc92e173f908f93b96ba5844209815a",
@ -932,7 +950,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(8kHzOutput)) {
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(16kHzOutput)) {
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
Run(16000, PlatformChecksum("a909560b5ca49fa472b17b7b277195e9",
"f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
"8cffa6abcb3e18e33b9d857666dff66a",
@ -940,7 +958,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(16kHzOutput)) {
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(32kHzOutput)) {
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
Run(32000, PlatformChecksum("441aab4b347fb3db4e9244337aca8d8e",
"306e0d990ee6e92de3fbecc0123ece37",
"3e126fe894720c3f85edadcc91964ba5",
@ -948,7 +966,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(32kHzOutput)) {
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(48kHzOutput)) {
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
"aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
@ -956,8 +974,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(48kHzOutput)) {
std::vector<ExternalDecoder>());
}
TEST_F(AcmReceiverBitExactnessOldApi,
IF_ALL_CODECS(48kHzOutputExternalDecoder)) {
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
// Class intended to forward a call from a mock DecodeInternal to Decode on
// the real decoder's Decode. DecodeInternal for the real decoder isn't
// public.
@ -1016,6 +1033,7 @@ TEST_F(AcmReceiverBitExactnessOldApi,
EXPECT_CALL(mock_decoder, Die());
}
#endif
// This test verifies bit exactness for the send-side of ACM. The test setup is
// a chain of three different test classes:
@ -1194,7 +1212,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
rtc::Md5Digest payload_checksum_;
};
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb30ms)) {
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"0b58f9eeee43d5891f5f6c75e77984a3",
@ -1209,7 +1228,7 @@ TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb30ms)) {
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb60ms)) {
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"1ad29139a04782a33daad8c2b9b35875",
@ -1223,15 +1242,15 @@ TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(IsacWb60ms)) {
"9e0a0ab743ad987b55b8e14802769c56"),
16, test::AcmReceiveTestOldApi::kMonoOutput);
}
#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 @@ TEST_F(AcmSenderBitExactnessOldApi,
"android_arm64_payload"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
@ -1324,13 +1344,13 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
test::AcmReceiveTestOldApi::kStereoOutput);
}
#ifdef WEBRTC_CODEC_ILBC
#define IF_ILBC(x) x
#if defined(WEBRTC_ANDROID)
#define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms
#else
#define IF_ILBC(x) DISABLED_##x
#define MAYBE_Ilbc_30ms Ilbc_30ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
#if defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
@ -1342,14 +1362,15 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
"android_arm64_payload"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
#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 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
"android_arm64_payload"),
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
TEST_F(AcmSenderBitExactnessOldApi,
DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
#else
#define MAYBE_G722_stereo_20ms G722_stereo_20ms
#endif
#if defined(WEBRTC_CODEC_G722)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7190ee718ab3d80eca181e5f7140c210",
@ -1375,6 +1402,7 @@ TEST_F(AcmSenderBitExactnessOldApi,
"android_arm64_payload"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
#endif
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
@ -1490,7 +1518,12 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
TEST_F(AcmSetBitRateOldApi, DISABLED_ON_ANDROID(Opus_48khz_20ms_100kbps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
#else
#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
Run(100000, 100888);
}

1
webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
View File

@ -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 {

19
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
View File

@ -21,7 +21,6 @@
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -276,7 +275,12 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
}
};
TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_RunTest DISABLED_RunTest
#else
#define MAYBE_RunTest RunTest
#endif
TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
RunTest(8);
}
@ -301,7 +305,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
double drift_factor;
};
TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
RunTest(100);
}
@ -314,7 +318,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
}
};
TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
RunTest(100);
}
@ -342,7 +346,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
int frame_index_;
};
TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
RunTest(1000);
}
@ -361,7 +365,10 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
int frame_index_;
};
TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
// TODO(pbos): Enable on non-Android, this went failing while being accidentally
// disabled on all platforms and not just Android.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
RunTest(100);
}

16
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -30,7 +30,6 @@
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
@ -930,13 +929,13 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#if defined(WEBRTC_ANDROID)
#define MAYBE_DecoderError DISABLED_DecoderError
#else
#define IF_ISAC(x) DISABLED_##x
#define MAYBE_DecoderError DecoderError
#endif
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
const size_t kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
@ -974,6 +973,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
EXPECT_EQ(1, out_data_[i]);
}
}
#endif
TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
NetEqOutputType type;
@ -1171,7 +1171,8 @@ TEST_F(NetEqBgnTestFade, RunTest) {
CheckBgn(32000);
}
TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(NetEqDecodingTest, SyncPacketInsert) {
WebRtcRTPHeader rtp_info;
uint32_t receive_timestamp = 0;
// For the readability use the following payloads instead of the defaults of
@ -1250,6 +1251,7 @@ TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
--rtp_info.header.ssrc;
EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
}
#endif
// First insert several noise like packets, then sync-packets. Decoding all
// packets should not produce error, statistics should not show any packet loss

50
webrtc/modules/audio_coding/test/Tester.cc
View File

@ -26,7 +26,6 @@
#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
using webrtc::Trace;
@ -42,7 +41,11 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
#else
TEST(AudioCodingModuleTest, TestEncodeDecode) {
#endif
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_encodedecode_trace.txt").c_str());
@ -50,51 +53,54 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
Trace::ReturnTrace();
}
#ifdef WEBRTC_CODEC_RED
#define IF_RED(x) x
#if defined(WEBRTC_CODEC_RED)
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
#else
#define IF_RED(x) DISABLED_##x
TEST(AudioCodingModuleTest, TestRedFec) {
#endif
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_fec_trace.txt").c_str());
webrtc::TestRedFec().Perform();
Trace::ReturnTrace();
}
#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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestWebRtcVadDtx)) {
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
#else
TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
#endif
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_vaddtx_trace.txt").c_str());

38
webrtc/modules/audio_coding/test/target_delay_unittest.cc
View File

@ -17,7 +17,6 @@
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -199,23 +198,50 @@ class TargetDelayTest : public ::testing::Test {
uint8_t payload_[kPayloadLenBytes];
};
TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_OutOfRangeInput DISABLED_OutOfRangeInput
#else
#define MAYBE_OutOfRangeInput OutOfRangeInput
#endif
TEST_F(TargetDelayTest, MAYBE_OutOfRangeInput) {
OutOfRangeInput();
}
TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_NoTargetDelayBufferSizeChanges \
DISABLED_NoTargetDelayBufferSizeChanges
#else
#define MAYBE_NoTargetDelayBufferSizeChanges NoTargetDelayBufferSizeChanges
#endif
TEST_F(TargetDelayTest, MAYBE_NoTargetDelayBufferSizeChanges) {
NoTargetDelayBufferSizeChanges();
}
TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_WithTargetDelayBufferNotChanging \
DISABLED_WithTargetDelayBufferNotChanging
#else
#define MAYBE_WithTargetDelayBufferNotChanging WithTargetDelayBufferNotChanging
#endif
TEST_F(TargetDelayTest, MAYBE_WithTargetDelayBufferNotChanging) {
WithTargetDelayBufferNotChanging();
}
TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_RequiredDelayAtCorrectRange DISABLED_RequiredDelayAtCorrectRange
#else
#define MAYBE_RequiredDelayAtCorrectRange RequiredDelayAtCorrectRange
#endif
TEST_F(TargetDelayTest, MAYBE_RequiredDelayAtCorrectRange) {
RequiredDelayAtCorrectRange();
}
TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
#else
#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
#endif
TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
TargetDelayBufferMinMax();
}

1
webrtc/modules/audio_processing/aec/system_delay_unittest.cc
View File

@ -14,7 +14,6 @@ extern "C" {
}
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
namespace {

1
webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
View File

@ -14,7 +14,6 @@ extern "C" {
#include "webrtc/modules/audio_processing/aec/aec_core.h"
}
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {

1
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
View File

@ -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"

96
webrtc/modules/audio_processing/transient/file_utils_unittest.cc
View File

@ -17,7 +17,6 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -59,7 +58,12 @@ class TransientFileUtilsTest: public ::testing::Test {
const std::string kTestFileNamef;
};
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ConvertByteArrayToFloat DISABLED_ConvertByteArrayToFloat
#else
#define MAYBE_ConvertByteArrayToFloat ConvertByteArrayToFloat
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToFloat) {
float value = 0.0;
EXPECT_EQ(0, ConvertByteArrayToFloat(kPiBytesf, &value));
@ -72,7 +76,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
EXPECT_FLOAT_EQ(kAvogadro, value);
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ConvertByteArrayToDouble DISABLED_ConvertByteArrayToDouble
#else
#define MAYBE_ConvertByteArrayToDouble ConvertByteArrayToDouble
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToDouble) {
double value = 0.0;
EXPECT_EQ(0, ConvertByteArrayToDouble(kPiBytes, &value));
@ -85,7 +94,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
EXPECT_DOUBLE_EQ(kAvogadro, value);
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ConvertFloatToByteArray DISABLED_ConvertFloatToByteArray
#else
#define MAYBE_ConvertFloatToByteArray ConvertFloatToByteArray
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ConvertFloatToByteArray) {
rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[4]);
EXPECT_EQ(0, ConvertFloatToByteArray(kPi, bytes.get()));
@ -98,7 +112,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytesf, 4));
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ConvertDoubleToByteArray DISABLED_ConvertDoubleToByteArray
#else
#define MAYBE_ConvertDoubleToByteArray ConvertDoubleToByteArray
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ConvertDoubleToByteArray) {
rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[8]);
EXPECT_EQ(0, ConvertDoubleToByteArray(kPi, bytes.get()));
@ -111,7 +130,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytes, 8));
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ReadInt16BufferFromFile DISABLED_ReadInt16BufferFromFile
#else
#define MAYBE_ReadInt16BufferFromFile ReadInt16BufferFromFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
std::string test_filename = kTestFileName;
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@ -149,8 +173,13 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
EXPECT_EQ(17631, buffer[kBufferLength - 1]);
}
TEST_F(TransientFileUtilsTest,
DISABLED_ON_IOS(ReadInt16FromFileToFloatBuffer)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ReadInt16FromFileToFloatBuffer \
DISABLED_ReadInt16FromFileToFloatBuffer
#else
#define MAYBE_ReadInt16FromFileToFloatBuffer ReadInt16FromFileToFloatBuffer
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
std::string test_filename = kTestFileName;
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@ -191,8 +220,13 @@ TEST_F(TransientFileUtilsTest,
EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
}
TEST_F(TransientFileUtilsTest,
DISABLED_ON_IOS(ReadInt16FromFileToDoubleBuffer)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ReadInt16FromFileToDoubleBuffer \
DISABLED_ReadInt16FromFileToDoubleBuffer
#else
#define MAYBE_ReadInt16FromFileToDoubleBuffer ReadInt16FromFileToDoubleBuffer
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
std::string test_filename = kTestFileName;
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@ -232,7 +266,12 @@ TEST_F(TransientFileUtilsTest,
EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ReadFloatBufferFromFile DISABLED_ReadFloatBufferFromFile
#else
#define MAYBE_ReadFloatBufferFromFile ReadFloatBufferFromFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
std::string test_filename = kTestFileNamef;
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@ -269,7 +308,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ReadDoubleBufferFromFile DISABLED_ReadDoubleBufferFromFile
#else
#define MAYBE_ReadDoubleBufferFromFile ReadDoubleBufferFromFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
std::string test_filename = kTestFileName;
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@ -306,7 +350,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_WriteInt16BufferToFile DISABLED_WriteInt16BufferToFile
#else
#define MAYBE_WriteInt16BufferToFile WriteInt16BufferToFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
std::string kOutFileName = test::TempFilename(test::OutputPath(),
@ -348,7 +397,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
kBufferLength * sizeof(written_buffer[0])));
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_WriteFloatBufferToFile DISABLED_WriteFloatBufferToFile
#else
#define MAYBE_WriteFloatBufferToFile WriteFloatBufferToFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
std::string kOutFileName = test::TempFilename(test::OutputPath(),
@ -390,7 +444,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
kBufferLength * sizeof(written_buffer[0])));
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_WriteDoubleBufferToFile DISABLED_WriteDoubleBufferToFile
#else
#define MAYBE_WriteDoubleBufferToFile WriteDoubleBufferToFile
#endif
TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
std::string kOutFileName = test::TempFilename(test::OutputPath(),
@ -432,7 +491,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
kBufferLength * sizeof(written_buffer[0])));
}
TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ExpectedErrorReturnValues)) {
#if defined(WEBRTC_IOS)
#define MAYBE_ExpectedErrorReturnValues DISABLED_ExpectedErrorReturnValues
#else
#define MAYBE_ExpectedErrorReturnValues ExpectedErrorReturnValues
#endif
TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
std::string test_filename = kTestFileName;
double value;

9
webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
View File

@ -19,8 +19,7 @@
#include "webrtc/modules/audio_processing/transient/file_utils.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -37,7 +36,11 @@ static const size_t kNumberOfSampleRates =
// The files contain all the results in double precision (Little endian).
// The audio files used with different sample rates are stored in the same
// directory.
TEST(TransientDetectorTest, DISABLED_ON_IOS(CorrectnessBasedOnFiles)) {
#if defined(WEBRTC_IOS)
TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
#else
TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
#endif
for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
int sample_rate_hz = kSampleRatesHz[i];

7
webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
View File

@ -19,7 +19,6 @@
#include "webrtc/modules/audio_processing/transient/file_utils.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -69,7 +68,11 @@ TEST(WPDTreeTest, Construction) {
// It also writes the results in its own set of files in the out directory.
// Matlab and output files contain all the results in double precision (Little
// endian) appended.
TEST(WPDTreeTest, DISABLED_ON_IOS(CorrectnessBasedOnMatlabFiles)) {
#if defined(WEBRTC_IOS)
TEST(WPDTreeTest, DISABLED_CorrectnessBasedOnMatlabFiles) {
#else
TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
#endif
// 10 ms at 16000 Hz.
const size_t kTestBufferSize = 160;
const int kLevels = 3;

7
webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
View File

@ -16,7 +16,6 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -55,7 +54,11 @@ TEST(StandaloneVadTest, Api) {
EXPECT_EQ(kMode, vad->mode());
}
TEST(StandaloneVadTest, DISABLED_ON_IOS(ActivityDetection)) {
#if defined(WEBRTC_IOS)
TEST(StandaloneVadTest, DISABLED_ActivityDetection) {
#else
TEST(StandaloneVadTest, ActivityDetection) {
#endif
rtc::scoped_ptr<StandaloneVad> vad(StandaloneVad::Create());
const size_t kDataLength = kLength10Ms;
int16_t data[kDataLength] = {0};

18
webrtc/modules/media_file/media_file_unittest.cc
View File

@ -12,7 +12,6 @@
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
class MediaFileTest : public testing::Test {
protected:
@ -28,8 +27,14 @@ class MediaFileTest : public testing::Test {
webrtc::MediaFile* media_file_;
};
TEST_F(MediaFileTest, DISABLED_ON_IOS(
DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError))) {
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_StartPlayingAudioFileWithoutError \
DISABLED_StartPlayingAudioFileWithoutError
#else
#define MAYBE_StartPlayingAudioFileWithoutError \
StartPlayingAudioFileWithoutError
#endif
TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
// TODO(leozwang): Use hard coded filename here, we want to
// loop through all audio files in future
const std::string audio_file = webrtc::test::ProjectRootPath() +
@ -47,7 +52,12 @@ TEST_F(MediaFileTest, DISABLED_ON_IOS(
ASSERT_EQ(0, media_file_->StopPlaying());
}
TEST_F(MediaFileTest, DISABLED_ON_IOS(WriteWavFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_WriteWavFile DISABLED_WriteWavFile
#else
#define MAYBE_WriteWavFile WriteWavFile
#endif
TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
// Write file.
static const size_t kHeaderSize = 44;
static const size_t kPayloadSize = 320;

73
webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
View File

@ -24,7 +24,6 @@
#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
namespace testing {
@ -318,8 +317,13 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
EXPECT_EQ(kBwOverusing, overuse_detector_->State());
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance30Kbit3fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance30Kbit3fps \
DISABLED_LowGaussianVariance30Kbit3fps
#else
#define MAYBE_LowGaussianVariance30Kbit3fps LowGaussianVariance30Kbit3fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance30Kbit3fps) {
size_t packet_size = 1200;
int packets_per_frame = 1;
int frame_duration_ms = 333;
@ -375,8 +379,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
EXPECT_EQ(4, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance100Kbit5fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance100Kbit5fps \
DISABLED_LowGaussianVariance100Kbit5fps
#else
#define MAYBE_LowGaussianVariance100Kbit5fps LowGaussianVariance100Kbit5fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit5fps) {
size_t packet_size = 1200;
int packets_per_frame = 2;
int frame_duration_ms = 200;
@ -390,8 +399,13 @@ TEST_F(OveruseDetectorTest,
EXPECT_EQ(13, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(HighGaussianVariance100Kbit5fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_HighGaussianVariance100Kbit5fps \
DISABLED_HighGaussianVariance100Kbit5fps
#else
#define MAYBE_HighGaussianVariance100Kbit5fps HighGaussianVariance100Kbit5fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit5fps) {
size_t packet_size = 1200;
int packets_per_frame = 2;
int frame_duration_ms = 200;
@ -405,8 +419,13 @@ TEST_F(OveruseDetectorTest,
EXPECT_EQ(32, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance100Kbit10fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance100Kbit10fps \
DISABLED_LowGaussianVariance100Kbit10fps
#else
#define MAYBE_LowGaussianVariance100Kbit10fps LowGaussianVariance100Kbit10fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit10fps) {
size_t packet_size = 1200;
int packets_per_frame = 1;
int frame_duration_ms = 100;
@ -420,8 +439,13 @@ TEST_F(OveruseDetectorTest,
EXPECT_EQ(13, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(HighGaussianVariance100Kbit10fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_HighGaussianVariance100Kbit10fps \
DISABLED_HighGaussianVariance100Kbit10fps
#else
#define MAYBE_HighGaussianVariance100Kbit10fps HighGaussianVariance100Kbit10fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit10fps) {
size_t packet_size = 1200;
int packets_per_frame = 1;
int frame_duration_ms = 100;
@ -435,8 +459,13 @@ TEST_F(OveruseDetectorTest,
EXPECT_EQ(32, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance300Kbit30fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance300Kbit30fps \
DISABLED_LowGaussianVariance300Kbit30fps
#else
#define MAYBE_LowGaussianVariance300Kbit30fps LowGaussianVariance300Kbit30fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance300Kbit30fps) {
size_t packet_size = 1200;
int packets_per_frame = 1;
int frame_duration_ms = 33;
@ -492,8 +521,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
EXPECT_EQ(10, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance1000Kbit30fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance1000Kbit30fps \
DISABLED_LowGaussianVariance1000Kbit30fps
#else
#define MAYBE_LowGaussianVariance1000Kbit30fps LowGaussianVariance1000Kbit30fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance1000Kbit30fps) {
size_t packet_size = 1200;
int packets_per_frame = 3;
int frame_duration_ms = 33;
@ -549,8 +583,13 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
EXPECT_EQ(10, frames_until_overuse);
}
TEST_F(OveruseDetectorTest,
DISABLED_ON_ANDROID(LowGaussianVariance2000Kbit30fps)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_LowGaussianVariance2000Kbit30fps \
DISABLED_LowGaussianVariance2000Kbit30fps
#else
#define MAYBE_LowGaussianVariance2000Kbit30fps LowGaussianVariance2000Kbit30fps
#endif
TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance2000Kbit30fps) {
size_t packet_size = 1200;
int packets_per_frame = 6;
int frame_duration_ms = 33;

15
webrtc/modules/utility/source/file_player_unittests.cc
View File

@ -20,7 +20,6 @@
#include "webrtc/base/md5digest.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
DEFINE_bool(file_player_output, false, "Generate reference files.");
@ -82,7 +81,12 @@ class FilePlayerTest : public ::testing::Test {
FILE* output_file_;
};
TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
#if defined(WEBRTC_IOS)
#define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
#else
#define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
#endif
TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
const std::string kFileName =
test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
// The file is longer than this, but keeping the output shorter limits the
@ -93,7 +97,12 @@ TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
}
TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcm16File)) {
#if defined(WEBRTC_IOS)
#define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
#else
#define MAYBE_PlayWavPcm16File PlayWavPcm16File
#endif
TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
const std::string kFileName =
test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
// The file is longer than this, but keeping the output shorter limits the

8
webrtc/modules/video_capture/test/video_capture_unittest.cc
View File

@ -23,7 +23,6 @@
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/video_frame.h"
using rtc::scoped_ptr;
@ -479,7 +478,12 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
// Test frame rate and no picture alarm.
// Flaky on Win32, see webrtc:3270.
TEST_F(VideoCaptureExternalTest, DISABLED_ON_WIN(FrameRate)) {
#if defined(WEBRTC_WIN)
#define MAYBE_FrameRate DISABLED_FrameRate
#else
#define MAYBE_FrameRate FrameRate
#endif
TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
int64_t testTime = 3;
TickTime startTime = TickTime::Now();

37
webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
View File

@ -22,7 +22,6 @@
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/frame_reader.h"
#include "webrtc/test/testsupport/frame_writer.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/test/testsupport/metrics/video_metrics.h"
#include "webrtc/test/testsupport/packet_reader.h"
#include "webrtc/typedefs.h"
@ -814,8 +813,13 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
// low to high to medium. Check that quality and encoder response to the new
// target rate/per-frame bandwidth (for each rate update) is within limits.
// One key frame (first frame only) in sequence.
TEST_F(VideoProcessorIntegrationTest,
DISABLED_ON_ANDROID(ProcessNoLossChangeBitRateVP8)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_ProcessNoLossChangeBitRateVP8 \
DISABLED_ProcessNoLossChangeBitRateVP8
#else
#define MAYBE_ProcessNoLossChangeBitRateVP8 ProcessNoLossChangeBitRateVP8
#endif
TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossChangeBitRateVP8) {
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
@ -846,8 +850,15 @@ TEST_F(VideoProcessorIntegrationTest,
// for the rate control metrics can be lower. One key frame (first frame only).
// Note: quality after update should be higher but we currently compute quality
// metrics averaged over whole sequence run.
#if defined(WEBRTC_ANDROID)
#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
#else
#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
ProcessNoLossChangeFrameRateFrameDropVP8
#endif
TEST_F(VideoProcessorIntegrationTest,
DISABLED_ON_ANDROID(ProcessNoLossChangeFrameRateFrameDropVP8)) {
MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;
@ -874,8 +885,15 @@ TEST_F(VideoProcessorIntegrationTest,
// Run with no packet loss, at low bitrate. During this time we should've
// resized once. Expect 2 key frames generated (first and one for resize).
#if defined(WEBRTC_ANDROID)
#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
DISABLED_ProcessNoLossSpatialResizeFrameDropVP8
#else
#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
ProcessNoLossSpatialResizeFrameDropVP8
#endif
TEST_F(VideoProcessorIntegrationTest,
DISABLED_ON_ANDROID(ProcessNoLossSpatialResizeFrameDropVP8)) {
MAYBE_ProcessNoLossSpatialResizeFrameDropVP8) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;
@ -901,8 +919,13 @@ TEST_F(VideoProcessorIntegrationTest,
// encoding rate mismatch are applied to each layer.
// No dropped frames in this test, and internal spatial resizer is off.
// One key frame (first frame only) in sequence, so no spatial resizing.
TEST_F(VideoProcessorIntegrationTest,
DISABLED_ON_ANDROID(ProcessNoLossTemporalLayersVP8)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_ProcessNoLossTemporalLayersVP8 \
DISABLED_ProcessNoLossTemporalLayersVP8
#else
#define MAYBE_ProcessNoLossTemporalLayersVP8 ProcessNoLossTemporalLayersVP8
#endif
TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossTemporalLayersVP8) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;

15
webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
View File

@ -17,7 +17,6 @@
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -221,7 +220,12 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_inst_, 1));
}
TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_AlignedStrideEncodeDecode DISABLED_AlignedStrideEncodeDecode
#else
#define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
#endif
TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
SetUpEncodeDecode();
encoder_->Encode(input_frame_, NULL, NULL);
EXPECT_GT(WaitForEncodedFrame(), 0u);
@ -237,7 +241,12 @@ TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
EXPECT_EQ(kTestNtpTimeMs, decoded_frame_.ntp_time_ms());
}
TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(DecodeWithACompleteKeyFrame)) {
#if defined(WEBRTC_ANDROID)
#define MAYBE_DecodeWithACompleteKeyFrame DISABLED_DecodeWithACompleteKeyFrame
#else
#define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
#endif
TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
SetUpEncodeDecode();
encoder_->Encode(input_frame_, NULL, NULL);
EXPECT_GT(WaitForEncodedFrame(), 0u);

18
webrtc/modules/video_coding/video_sender_unittest.cc
View File

@ -24,7 +24,6 @@
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
using ::testing::_;
using ::testing::AllOf;
@ -424,8 +423,12 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
int available_bitrate_kbps_;
};
TEST_F(TestVideoSenderWithVp8,
DISABLED_ON_IOS(DISABLED_ON_ANDROID(FixedTemporalLayersStrategy))) {
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_FixedTemporalLayersStrategy DISABLED_FixedTemporalLayersStrategy
#else
#define MAYBE_FixedTemporalLayersStrategy FixedTemporalLayersStrategy
#endif
TEST_F(TestVideoSenderWithVp8, MAYBE_FixedTemporalLayersStrategy) {
const int low_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][0];
const int mid_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][1];
const int high_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][2];
@ -439,8 +442,13 @@ TEST_F(TestVideoSenderWithVp8,
}
}
TEST_F(TestVideoSenderWithVp8,
DISABLED_ON_IOS(DISABLED_ON_ANDROID(RealTimeTemporalLayersStrategy))) {
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_RealTimeTemporalLayersStrategy \
DISABLED_RealTimeTemporalLayersStrategy
#else
#define MAYBE_RealTimeTemporalLayersStrategy RealTimeTemporalLayersStrategy
#endif
TEST_F(TestVideoSenderWithVp8, MAYBE_RealTimeTemporalLayersStrategy) {
Config extra_options;
extra_options.Set<TemporalLayers::Factory>(
new RealTimeTemporalLayersFactory());

8
webrtc/modules/video_processing/test/brightness_detection_test.cc
View File

@ -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;

7
webrtc/modules/video_processing/test/content_metrics_test.cc
View File

@ -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;

7
webrtc/modules/video_processing/test/deflickering_test.cc
View File

@ -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;

37
webrtc/modules/video_processing/test/video_processing_unittest.cc
View File

@ -17,7 +17,6 @@
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@ -92,7 +91,11 @@ void VideoProcessingTest::TearDown() {
vp_ = NULL;
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleNullBuffer)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_HandleNullBuffer) {
#else
TEST_F(VideoProcessingTest, HandleNullBuffer) {
#endif
// TODO(mikhal/stefan): Do we need this one?
VideoProcessing::FrameStats stats;
// Video frame with unallocated buffer.
@ -106,7 +109,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleNullBuffer)) {
EXPECT_EQ(-3, vp_->BrightnessDetection(videoFrame, stats));
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleBadStats)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_HandleBadStats) {
#else
TEST_F(VideoProcessingTest, HandleBadStats) {
#endif
VideoProcessing::FrameStats stats;
vp_->ClearFrameStats(&stats);
rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
@ -120,7 +127,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(HandleBadStats)) {
EXPECT_EQ(-3, vp_->BrightnessDetection(video_frame_, stats));
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(IdenticalResultsAfterReset)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_IdenticalResultsAfterReset) {
#else
TEST_F(VideoProcessingTest, IdenticalResultsAfterReset) {
#endif
VideoFrame video_frame2;
VideoProcessing::FrameStats stats;
// Only testing non-static functions here.
@ -153,7 +164,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(IdenticalResultsAfterReset)) {
EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(FrameStats)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_FrameStats) {
#else
TEST_F(VideoProcessingTest, FrameStats) {
#endif
VideoProcessing::FrameStats stats;
vp_->ClearFrameStats(&stats);
rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
@ -178,7 +193,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(FrameStats)) {
EXPECT_FALSE(vp_->ValidFrameStats(stats));
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(PreprocessorLogic)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_PreprocessorLogic) {
#else
TEST_F(VideoProcessingTest, PreprocessorLogic) {
#endif
// Disable temporal sampling (frame dropping).
vp_->EnableTemporalDecimation(false);
int resolution = 100;
@ -197,7 +216,11 @@ TEST_F(VideoProcessingTest, DISABLED_ON_IOS(PreprocessorLogic)) {
EXPECT_TRUE(vp_->PreprocessFrame(video_frame_) != nullptr);
}
TEST_F(VideoProcessingTest, DISABLED_ON_IOS(Resampler)) {
#if defined(WEBRTC_IOS)
TEST_F(VideoProcessingTest, DISABLED_Resampler) {
#else
TEST_F(VideoProcessingTest, Resampler) {
#endif
enum { NumRuns = 1 };
int64_t min_runtime = 0;

1
webrtc/test/BUILD.gn
View File

@ -58,7 +58,6 @@ source_set("test_support") {
"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",

1
webrtc/test/test.gyp
View File

@ -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',

17
webrtc/test/testsupport/fileutils_unittest.cc
View File

@ -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 @@ TEST_F(FileUtilsTest, ProjectRootPath) {
}
// 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 @@ TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) {
// 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());
}

57
webrtc/test/testsupport/gtest_disable.h
View File

@ -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_

1
webrtc/video/end_to_end_tests.cc
View File

@ -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"

1
webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
View File

@ -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"

3
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
View File

@ -101,8 +101,7 @@ TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
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));

1
webrtc/voice_engine/voe_codec_unittest.cc
View File

@ -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"