Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
index 8de6c9100d3cd243257555aa1ef5fc04a8d87c40..ef48a48d94d275a9fad0fdeec7adc2e21280e912 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 @@ 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); |
} |
+#endif |
-#ifdef WEBRTC_CODEC_ISAC |
-#define IF_ISAC_FLOAT(x) x |
+#if defined(WEBRTC_ANDROID) |
+#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms |
#else |
-#define IF_ISAC_FLOAT(x) DISABLED_##x |
+#define MAYBE_IsacSwb30ms IsacSwb30ms |
#endif |
- |
-TEST_F(AcmSenderBitExactnessOldApi, |
- DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) { |
+#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); |
} |
+#endif |
-#ifdef WEBRTC_CODEC_G722 |
-#define IF_G722(x) x |
+#if defined(WEBRTC_ANDROID) |
+#define MAYBE_G722_20ms DISABLED_G722_20ms |
#else |
-#define IF_G722(x) DISABLED_##x |
+#define MAYBE_G722_20ms G722_20ms |
#endif |
- |
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) { |
+#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); |
} |