Stop using Googletest legacy APIs.
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index 6d1cbc6..a548be8 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -103,9 +103,9 @@
}
}
-INSTANTIATE_TEST_CASE_P(BuiltinAudioEncoderFactoryTest,
- AudioEncoderFactoryTest,
- ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+INSTANTIATE_TEST_SUITE_P(BuiltinAudioEncoderFactoryTest,
+ AudioEncoderFactoryTest,
+ ::testing::Values(CreateBuiltinAudioEncoderFactory()));
TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
using ::testing::ElementsAreArray;
diff --git a/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
index 5ec1219..689292f 100644
--- a/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
+++ b/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
@@ -103,7 +103,7 @@
// Also test the maximum number of frames in one packet for 20 and 30 ms.
// The maximum is defined by the largest payload length that can be uniquely
// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
-INSTANTIATE_TEST_CASE_P(
+INSTANTIATE_TEST_SUITE_P(
IlbcTest,
SplitIlbcTest,
::testing::Values(std::pair<int, int>(1, 20), // 1 frame, 20 ms.
diff --git a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index aeca2e8..2075263 100644
--- a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -112,6 +112,8 @@
string("pcm"),
true)};
-INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest, ::testing::ValuesIn(param_set));
+INSTANTIATE_TEST_SUITE_P(AllTest,
+ IsacSpeedTest,
+ ::testing::ValuesIn(param_set));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/isac/unittest.cc b/modules/audio_coding/codecs/isac/unittest.cc
index 6791745..076510b 100644
--- a/modules/audio_coding/codecs/isac/unittest.cc
+++ b/modules/audio_coding/codecs/isac/unittest.cc
@@ -252,6 +252,6 @@
return cases;
}
-INSTANTIATE_TEST_CASE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
+INSTANTIATE_TEST_SUITE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
index 54a362d..2ca1d4c 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -160,7 +160,7 @@
}
}
-INSTANTIATE_TEST_CASE_P(
+INSTANTIATE_TEST_SUITE_P(
LegacyEncodedAudioFrame,
SplitBySamplesTest,
::testing::Values(NetEqDecoder::kDecoderPCMu,
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index cd70821..7552c20 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -238,6 +238,6 @@
string("pcm"))};
// 64 kbps, stereo
-INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
+INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc
index 03b59ed..bf757f6 100644
--- a/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -140,6 +140,8 @@
string("pcm"),
true)};
-INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest, ::testing::ValuesIn(param_set));
+INSTANTIATE_TEST_SUITE_P(AllTest,
+ OpusSpeedTest,
+ ::testing::ValuesIn(param_set));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index de08827..56dfd6a 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -810,8 +810,8 @@
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
-INSTANTIATE_TEST_CASE_P(VariousMode,
- OpusTest,
- Combine(Values(1, 2, 4), Values(0, 1)));
+INSTANTIATE_TEST_SUITE_P(VariousMode,
+ OpusTest,
+ Combine(Values(1, 2, 4), Values(0, 1)));
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
index e26df3a..5c44bc5 100644
--- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
+++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -82,7 +82,7 @@
EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
}
-INSTANTIATE_TEST_CASE_P(
+INSTANTIATE_TEST_SUITE_P(
AudioRingBufferTest,
AudioRingBufferTest,
::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames
diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
index 5b2ec20..d1351d8 100644
--- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -309,9 +309,9 @@
}
}
-INSTANTIATE_TEST_CASE_P(TestNumChannels,
- AudioMultiVectorTest,
- ::testing::Values(static_cast<size_t>(1),
- static_cast<size_t>(2),
- static_cast<size_t>(5)));
+INSTANTIATE_TEST_SUITE_P(TestNumChannels,
+ AudioMultiVectorTest,
+ ::testing::Values(static_cast<size_t>(1),
+ static_cast<size_t>(2),
+ static_cast<size_t>(5)));
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 289a38d..d25e8d6 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -364,24 +364,24 @@
// Instantiate the tests. Each test is instantiated using the function above,
// so that all different parameter combinations are tested.
-INSTANTIATE_TEST_CASE_P(MultiChannel,
- NetEqStereoTestNoJitter,
- ::testing::ValuesIn(GetTestParameters()));
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+ NetEqStereoTestNoJitter,
+ ::testing::ValuesIn(GetTestParameters()));
-INSTANTIATE_TEST_CASE_P(MultiChannel,
- NetEqStereoTestPositiveDrift,
- ::testing::ValuesIn(GetTestParameters()));
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+ NetEqStereoTestPositiveDrift,
+ ::testing::ValuesIn(GetTestParameters()));
-INSTANTIATE_TEST_CASE_P(MultiChannel,
- NetEqStereoTestNegativeDrift,
- ::testing::ValuesIn(GetTestParameters()));
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+ NetEqStereoTestNegativeDrift,
+ ::testing::ValuesIn(GetTestParameters()));
-INSTANTIATE_TEST_CASE_P(MultiChannel,
- NetEqStereoTestDelays,
- ::testing::ValuesIn(GetTestParameters()));
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+ NetEqStereoTestDelays,
+ ::testing::ValuesIn(GetTestParameters()));
-INSTANTIATE_TEST_CASE_P(MultiChannel,
- NetEqStereoTestLosses,
- ::testing::ValuesIn(GetTestParameters()));
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+ NetEqStereoTestLosses,
+ ::testing::ValuesIn(GetTestParameters()));
} // namespace webrtc