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/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