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/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