zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Stop using Googletest legacy APIs.

Browse Source 
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}
This commit is contained in:
Mirko Bonadei
2019-01-31 12:20:57 +01:00
committed by Commit Bot
parent 12e5d392cc
commit c84f661b10
77 changed files with 354 additions and 347 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
View File

@ -103,9 +103,9 @@ TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
}
}
INSTANTIATE_TEST_CASE_P(BuiltinAudioEncoderFactoryTest,
AudioEncoderFactoryTest,
::testing::Values(CreateBuiltinAudioEncoderFactory()));
INSTANTIATE_TEST_SUITE_P(BuiltinAudioEncoderFactoryTest,
AudioEncoderFactoryTest,
::testing::Values(CreateBuiltinAudioEncoderFactory()));
TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
using ::testing::ElementsAreArray;

2
modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
View File

@ -103,7 +103,7 @@ TEST_P(SplitIlbcTest, NumFrames) {
// 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.

4
modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -112,6 +112,8 @@ const coding_param param_set[] = {
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

2
modules/audio_coding/codecs/isac/unittest.cc
View File

@ -252,6 +252,6 @@ std::vector<IsacTestParam> TestCases() {
return cases;
}
INSTANTIATE_TEST_CASE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
INSTANTIATE_TEST_SUITE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
} // namespace webrtc

2
modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
View File

@ -160,7 +160,7 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
}
}
INSTANTIATE_TEST_CASE_P(
INSTANTIATE_TEST_SUITE_P(
LegacyEncodedAudioFrame,
SplitBySamplesTest,
::testing::Values(NetEqDecoder::kDecoderPCMu,

2
modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -238,6 +238,6 @@ const coding_param param_set[] = {
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

4
modules/audio_coding/codecs/opus/opus_speed_test.cc
View File

@ -140,6 +140,8 @@ const coding_param param_set[] = {
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

6
modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -810,8 +810,8 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
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

2
modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
View File

@ -82,7 +82,7 @@ TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
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

10
modules/audio_coding/neteq/audio_multi_vector_unittest.cc
View File

@ -309,9 +309,9 @@ TEST_P(AudioMultiVectorTest, CopyChannel) {
}
}
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

30
modules/audio_coding/neteq/neteq_stereo_unittest.cc
View File

@ -364,24 +364,24 @@ void PrintTo(const TestParameters& p, ::std::ostream* os) {
// 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