Disables unit tests that don't work on Android for Android.
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1747004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/neteq4/decoder_database_unittest.cc b/modules/audio_coding/neteq4/decoder_database_unittest.cc
index 3b2364c..76f5a09 100644
--- a/modules/audio_coding/neteq4/decoder_database_unittest.cc
+++ b/modules/audio_coding/neteq4/decoder_database_unittest.cc
@@ -17,7 +17,9 @@
#include "gmock/gmock.h"
#include "gtest/gtest.h"
+
#include "webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@@ -66,7 +68,7 @@
db.GetRtpPayloadType(kDecoderISAC)); // iSAC is not registered.
}
-TEST(DecoderDatabase, GetDecoder) {
+TEST(DecoderDatabase, DISABLED_ON_ANDROID(GetDecoder)) {
DecoderDatabase db;
const uint8_t kPayloadType = 0;
EXPECT_EQ(DecoderDatabase::kOK,
diff --git a/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
index c0a0fd3..fec25e9 100644
--- a/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
+++ b/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
@@ -21,6 +21,7 @@
#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@@ -201,7 +202,7 @@
scoped_ptr<test::InputAudioFile> input_file_;
};
-TEST_F(NetEqExternalDecoderTest, RunTest) {
+TEST_F(NetEqExternalDecoderTest, DISABLED_ON_ANDROID(RunTest)) {
RunTest(100); // Run 100 laps @ 10 ms each in the test loop.
}
diff --git a/modules/audio_coding/neteq4/neteq_stereo_unittest.cc b/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
index 9c74e03..d6c4150 100644
--- a/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
@@ -20,6 +20,7 @@
#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@@ -270,7 +271,7 @@
}
};
-TEST_P(NetEqStereoTestNoJitter, RunTest) {
+TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
RunTest(8);
}
@@ -295,7 +296,7 @@
double drift_factor;
};
-TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
+TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
RunTest(100);
}
@@ -308,7 +309,7 @@
}
};
-TEST_P(NetEqStereoTestNegativeDrift, RunTest) {
+TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
RunTest(100);
}
@@ -336,7 +337,7 @@
int frame_index_;
};
-TEST_P(NetEqStereoTestDelays, RunTest) {
+TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
RunTest(1000);
}
@@ -355,7 +356,7 @@
int frame_index_;
};
-TEST_P(NetEqStereoTestLosses, RunTest) {
+TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
RunTest(100);
}
diff --git a/modules/audio_coding/neteq4/neteq_unittest.cc b/modules/audio_coding/neteq4/neteq_unittest.cc
index 6072212..1b3af03 100644
--- a/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -23,6 +23,7 @@
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -229,8 +230,10 @@
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
// Load PCMa.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
+#ifndef WEBRTC_ANDROID
// Load iLBC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
+#endif // WEBRTC_ANDROID
// Load iSAC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
// Load iSAC SWB.
@@ -379,7 +382,7 @@
#define MAYBE_TestBitExactness TestBitExactness
#endif
-TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
@@ -394,7 +397,7 @@
DecodeAndCompare(kInputRtpFile, kInputRefFile);
}
-TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
@@ -412,7 +415,7 @@
}
// TODO(hlundin): Re-enable test once the statistics interface is up and again.
-TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
// Use fax mode to avoid time-scaling. This is to simplify the testing of
// packet waiting times in the packet buffer.
neteq_->SetPlayoutMode(kPlayoutFax);
@@ -487,7 +490,8 @@
EXPECT_EQ(100u, waiting_times.size());
}
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
+TEST_F(NetEqDecodingTest,
+ DISABLED_ON_ANDROID(TestAverageInterArrivalTimeNegative)) {
const int kNumFrames = 3000; // Needed for convergence.
int frame_index = 0;
const int kSamples = 10 * 16;
@@ -518,7 +522,8 @@
EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
}
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
+TEST_F(NetEqDecodingTest,
+ DISABLED_ON_ANDROID(TestAverageInterArrivalTimePositive)) {
const int kNumFrames = 5000; // Needed for convergence.
int frame_index = 0;
const int kSamples = 10 * 16;
@@ -549,7 +554,7 @@
EXPECT_EQ(110946, network_stats.clockdrift_ppm);
}
-TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 30;
@@ -642,7 +647,7 @@
EXPECT_GE(delay_after, delay_before - 20 * 16);
}
-TEST_F(NetEqDecodingTest, UnknownPayloadType) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(UnknownPayloadType)) {
const int kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
@@ -653,7 +658,7 @@
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
-TEST_F(NetEqDecodingTest, DecoderError) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
const int kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
@@ -692,7 +697,7 @@
}
}
-TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) {
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
diff --git a/modules/audio_processing/test/audio_processing_unittest.cc b/modules/audio_processing/test/audio_processing_unittest.cc
index ad53117..f93b7b8 100644
--- a/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/modules/audio_processing/test/audio_processing_unittest.cc
@@ -21,6 +21,7 @@
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "gtest/gtest.h"
#include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
@@ -1402,7 +1403,7 @@
// TODO(andrew): Make this test more robust such that it can be run on multiple
// platforms. It currently requires bit-exactness.
#ifdef WEBRTC_AUDIOPROC_BIT_EXACT
-TEST_F(ApmTest, Process) {
+TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) {
GOOGLE_PROTOBUF_VERIFY_VERSION;
webrtc::audioproc::OutputData ref_data;
diff --git a/modules/media_file/source/media_file_unittest.cc b/modules/media_file/source/media_file_unittest.cc
index 8abe7fc..9f3f0cc 100644
--- a/modules/media_file/source/media_file_unittest.cc
+++ b/modules/media_file/source/media_file_unittest.cc
@@ -12,6 +12,7 @@
#include "webrtc/modules/media_file/interface/media_file.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
class MediaFileTest : public testing::Test {
protected:
@@ -27,7 +28,7 @@
webrtc::MediaFile* media_file_;
};
-TEST_F(MediaFileTest, StartPlayingAudioFileWithoutError) {
+TEST_F(MediaFileTest, DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError)) {
// TODO(leozwang): Use hard coded filename here, we want to
// loop through all audio files in future
const std::string audio_file = webrtc::test::ProjectRootPath() +
diff --git a/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
index be89d6d..278a057 100644
--- a/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
+++ b/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -215,4 +215,3 @@
EXPECT_EQ(true, vp8_info.layerSync);
}
} // namespace webrtc
-
diff --git a/modules/video_processing/main/test/unit_test/denoising_test.cc b/modules/video_processing/main/test/unit_test/denoising_test.cc
index 71d133b..8c47917 100644
--- a/modules/video_processing/main/test/unit_test/denoising_test.cc
+++ b/modules/video_processing/main/test/unit_test/denoising_test.cc
@@ -16,10 +16,11 @@
#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
-TEST_F(VideoProcessingModuleTest, Denoising)
+TEST_F(VideoProcessingModuleTest, DISABLED_ON_ANDROID(Denoising))
{
enum { NumRuns = 10 };
uint32_t frameNum = 0;
diff --git a/test/testsupport/gtest_disable.h b/test/testsupport/gtest_disable.h
index b4a661f..257d836 100644
--- a/test/testsupport/gtest_disable.h
+++ b/test/testsupport/gtest_disable.h
@@ -36,4 +36,10 @@
#define DISABLED_ON_WIN(test) test
#endif
+#ifdef WEBRTC_ANDROID
+#define DISABLED_ON_ANDROID(test) DISABLED_##test
+#else
+#define DISABLED_ON_ANDROID(test) test
+#endif
+
#endif // TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_